Threads.cpp revision 462fd2fa9eef642b0574aa7409de0bde3fec8d43
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 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 285 for (size_t i = 0; i < mConfigEvents.size(); i++) { 286 delete mConfigEvents[i]; 287 } 288 mConfigEvents.clear(); 289 290 mParamCond.broadcast(); 291 // do not lock the mutex in destructor 292 releaseWakeLock_l(); 293 if (mPowerManager != 0) { 294 sp<IBinder> binder = mPowerManager->asBinder(); 295 binder->unlinkToDeath(mDeathRecipient); 296 } 297} 298 299status_t AudioFlinger::ThreadBase::readyToRun() 300{ 301 status_t status = initCheck(); 302 if (status == NO_ERROR) { 303 ALOGI("AudioFlinger's thread %p ready to run", this); 304 } else { 305 ALOGE("No working audio driver found."); 306 } 307 return status; 308} 309 310void AudioFlinger::ThreadBase::exit() 311{ 312 ALOGV("ThreadBase::exit"); 313 // do any cleanup required for exit to succeed 314 preExit(); 315 { 316 // This lock prevents the following race in thread (uniprocessor for illustration): 317 // if (!exitPending()) { 318 // // context switch from here to exit() 319 // // exit() calls requestExit(), what exitPending() observes 320 // // exit() calls signal(), which is dropped since no waiters 321 // // context switch back from exit() to here 322 // mWaitWorkCV.wait(...); 323 // // now thread is hung 324 // } 325 AutoMutex lock(mLock); 326 requestExit(); 327 mWaitWorkCV.broadcast(); 328 } 329 // When Thread::requestExitAndWait is made virtual and this method is renamed to 330 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 331 requestExitAndWait(); 332} 333 334status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 335{ 336 status_t status; 337 338 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 339 Mutex::Autolock _l(mLock); 340 341 mNewParameters.add(keyValuePairs); 342 mWaitWorkCV.signal(); 343 // wait condition with timeout in case the thread loop has exited 344 // before the request could be processed 345 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 346 status = mParamStatus; 347 mWaitWorkCV.signal(); 348 } else { 349 status = TIMED_OUT; 350 } 351 return status; 352} 353 354void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 355{ 356 Mutex::Autolock _l(mLock); 357 sendIoConfigEvent_l(event, param); 358} 359 360// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 361void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 362{ 363 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 364 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 365 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 366 param); 367 mWaitWorkCV.signal(); 368} 369 370// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 371void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 372{ 373 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 374 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 375 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 376 mConfigEvents.size(), pid, tid, prio); 377 mWaitWorkCV.signal(); 378} 379 380void AudioFlinger::ThreadBase::processConfigEvents() 381{ 382 Mutex::Autolock _l(mLock); 383 processConfigEvents_l(); 384} 385 386// post condition: mConfigEvents.isEmpty() 387void AudioFlinger::ThreadBase::processConfigEvents_l() 388{ 389 while (!mConfigEvents.isEmpty()) { 390 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 391 ConfigEvent *event = mConfigEvents[0]; 392 mConfigEvents.removeAt(0); 393 // release mLock before locking AudioFlinger mLock: lock order is always 394 // AudioFlinger then ThreadBase to avoid cross deadlock 395 mLock.unlock(); 396 switch (event->type()) { 397 case CFG_EVENT_PRIO: { 398 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 399 // FIXME Need to understand why this has be done asynchronously 400 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 401 true /*asynchronous*/); 402 if (err != 0) { 403 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 404 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 405 } 406 } break; 407 case CFG_EVENT_IO: { 408 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 409 { 410 Mutex::Autolock _l(mAudioFlinger->mLock); 411 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 412 } 413 } break; 414 default: 415 ALOGE("processConfigEvents() unknown event type %d", event->type()); 416 break; 417 } 418 delete event; 419 mLock.lock(); 420 } 421} 422 423void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 424{ 425 const size_t SIZE = 256; 426 char buffer[SIZE]; 427 String8 result; 428 429 bool locked = AudioFlinger::dumpTryLock(mLock); 430 if (!locked) { 431 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 432 write(fd, buffer, strlen(buffer)); 433 } 434 435 snprintf(buffer, SIZE, "io handle: %d\n", mId); 436 result.append(buffer); 437 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 438 result.append(buffer); 439 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 440 result.append(buffer); 441 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 442 result.append(buffer); 443 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 444 result.append(buffer); 445 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 446 result.append(buffer); 447 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 448 result.append(buffer); 449 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 450 result.append(buffer); 451 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 452 result.append(buffer); 453 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 454 result.append(buffer); 455 456 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 457 result.append(buffer); 458 result.append(" Index Command"); 459 for (size_t i = 0; i < mNewParameters.size(); ++i) { 460 snprintf(buffer, SIZE, "\n %02d ", i); 461 result.append(buffer); 462 result.append(mNewParameters[i]); 463 } 464 465 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 466 result.append(buffer); 467 for (size_t i = 0; i < mConfigEvents.size(); i++) { 468 mConfigEvents[i]->dump(buffer, SIZE); 469 result.append(buffer); 470 } 471 result.append("\n"); 472 473 write(fd, result.string(), result.size()); 474 475 if (locked) { 476 mLock.unlock(); 477 } 478} 479 480void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 481{ 482 const size_t SIZE = 256; 483 char buffer[SIZE]; 484 String8 result; 485 486 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 487 write(fd, buffer, strlen(buffer)); 488 489 for (size_t i = 0; i < mEffectChains.size(); ++i) { 490 sp<EffectChain> chain = mEffectChains[i]; 491 if (chain != 0) { 492 chain->dump(fd, args); 493 } 494 } 495} 496 497void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 498{ 499 Mutex::Autolock _l(mLock); 500 acquireWakeLock_l(uid); 501} 502 503String16 AudioFlinger::ThreadBase::getWakeLockTag() 504{ 505 switch (mType) { 506 case MIXER: 507 return String16("AudioMix"); 508 case DIRECT: 509 return String16("AudioDirectOut"); 510 case DUPLICATING: 511 return String16("AudioDup"); 512 case RECORD: 513 return String16("AudioIn"); 514 case OFFLOAD: 515 return String16("AudioOffload"); 516 default: 517 ALOG_ASSERT(false); 518 return String16("AudioUnknown"); 519 } 520} 521 522void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 523{ 524 getPowerManager_l(); 525 if (mPowerManager != 0) { 526 sp<IBinder> binder = new BBinder(); 527 status_t status; 528 if (uid >= 0) { 529 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 530 binder, 531 getWakeLockTag(), 532 String16("media"), 533 uid); 534 } else { 535 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 536 binder, 537 getWakeLockTag(), 538 String16("media")); 539 } 540 if (status == NO_ERROR) { 541 mWakeLockToken = binder; 542 } 543 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 544 } 545} 546 547void AudioFlinger::ThreadBase::releaseWakeLock() 548{ 549 Mutex::Autolock _l(mLock); 550 releaseWakeLock_l(); 551} 552 553void AudioFlinger::ThreadBase::releaseWakeLock_l() 554{ 555 if (mWakeLockToken != 0) { 556 ALOGV("releaseWakeLock_l() %s", mName); 557 if (mPowerManager != 0) { 558 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 559 } 560 mWakeLockToken.clear(); 561 } 562} 563 564void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 565 Mutex::Autolock _l(mLock); 566 updateWakeLockUids_l(uids); 567} 568 569void AudioFlinger::ThreadBase::getPowerManager_l() { 570 571 if (mPowerManager == 0) { 572 // use checkService() to avoid blocking if power service is not up yet 573 sp<IBinder> binder = 574 defaultServiceManager()->checkService(String16("power")); 575 if (binder == 0) { 576 ALOGW("Thread %s cannot connect to the power manager service", mName); 577 } else { 578 mPowerManager = interface_cast<IPowerManager>(binder); 579 binder->linkToDeath(mDeathRecipient); 580 } 581 } 582} 583 584void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 585 586 getPowerManager_l(); 587 if (mWakeLockToken == NULL) { 588 ALOGE("no wake lock to update!"); 589 return; 590 } 591 if (mPowerManager != 0) { 592 sp<IBinder> binder = new BBinder(); 593 status_t status; 594 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 595 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 596 } 597} 598 599void AudioFlinger::ThreadBase::clearPowerManager() 600{ 601 Mutex::Autolock _l(mLock); 602 releaseWakeLock_l(); 603 mPowerManager.clear(); 604} 605 606void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 607{ 608 sp<ThreadBase> thread = mThread.promote(); 609 if (thread != 0) { 610 thread->clearPowerManager(); 611 } 612 ALOGW("power manager service died !!!"); 613} 614 615void AudioFlinger::ThreadBase::setEffectSuspended( 616 const effect_uuid_t *type, bool suspend, int sessionId) 617{ 618 Mutex::Autolock _l(mLock); 619 setEffectSuspended_l(type, suspend, sessionId); 620} 621 622void AudioFlinger::ThreadBase::setEffectSuspended_l( 623 const effect_uuid_t *type, bool suspend, int sessionId) 624{ 625 sp<EffectChain> chain = getEffectChain_l(sessionId); 626 if (chain != 0) { 627 if (type != NULL) { 628 chain->setEffectSuspended_l(type, suspend); 629 } else { 630 chain->setEffectSuspendedAll_l(suspend); 631 } 632 } 633 634 updateSuspendedSessions_l(type, suspend, sessionId); 635} 636 637void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 638{ 639 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 640 if (index < 0) { 641 return; 642 } 643 644 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 645 mSuspendedSessions.valueAt(index); 646 647 for (size_t i = 0; i < sessionEffects.size(); i++) { 648 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 649 for (int j = 0; j < desc->mRefCount; j++) { 650 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 651 chain->setEffectSuspendedAll_l(true); 652 } else { 653 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 654 desc->mType.timeLow); 655 chain->setEffectSuspended_l(&desc->mType, true); 656 } 657 } 658 } 659} 660 661void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 662 bool suspend, 663 int sessionId) 664{ 665 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 666 667 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 668 669 if (suspend) { 670 if (index >= 0) { 671 sessionEffects = mSuspendedSessions.valueAt(index); 672 } else { 673 mSuspendedSessions.add(sessionId, sessionEffects); 674 } 675 } else { 676 if (index < 0) { 677 return; 678 } 679 sessionEffects = mSuspendedSessions.valueAt(index); 680 } 681 682 683 int key = EffectChain::kKeyForSuspendAll; 684 if (type != NULL) { 685 key = type->timeLow; 686 } 687 index = sessionEffects.indexOfKey(key); 688 689 sp<SuspendedSessionDesc> desc; 690 if (suspend) { 691 if (index >= 0) { 692 desc = sessionEffects.valueAt(index); 693 } else { 694 desc = new SuspendedSessionDesc(); 695 if (type != NULL) { 696 desc->mType = *type; 697 } 698 sessionEffects.add(key, desc); 699 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 700 } 701 desc->mRefCount++; 702 } else { 703 if (index < 0) { 704 return; 705 } 706 desc = sessionEffects.valueAt(index); 707 if (--desc->mRefCount == 0) { 708 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 709 sessionEffects.removeItemsAt(index); 710 if (sessionEffects.isEmpty()) { 711 ALOGV("updateSuspendedSessions_l() restore removing session %d", 712 sessionId); 713 mSuspendedSessions.removeItem(sessionId); 714 } 715 } 716 } 717 if (!sessionEffects.isEmpty()) { 718 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 719 } 720} 721 722void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 723 bool enabled, 724 int sessionId) 725{ 726 Mutex::Autolock _l(mLock); 727 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 728} 729 730void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 731 bool enabled, 732 int sessionId) 733{ 734 if (mType != RECORD) { 735 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 736 // another session. This gives the priority to well behaved effect control panels 737 // and applications not using global effects. 738 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 739 // global effects 740 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 741 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 742 } 743 } 744 745 sp<EffectChain> chain = getEffectChain_l(sessionId); 746 if (chain != 0) { 747 chain->checkSuspendOnEffectEnabled(effect, enabled); 748 } 749} 750 751// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 752sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 753 const sp<AudioFlinger::Client>& client, 754 const sp<IEffectClient>& effectClient, 755 int32_t priority, 756 int sessionId, 757 effect_descriptor_t *desc, 758 int *enabled, 759 status_t *status) 760{ 761 sp<EffectModule> effect; 762 sp<EffectHandle> handle; 763 status_t lStatus; 764 sp<EffectChain> chain; 765 bool chainCreated = false; 766 bool effectCreated = false; 767 bool effectRegistered = false; 768 769 lStatus = initCheck(); 770 if (lStatus != NO_ERROR) { 771 ALOGW("createEffect_l() Audio driver not initialized."); 772 goto Exit; 773 } 774 775 // Allow global effects only on offloaded and mixer threads 776 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 777 switch (mType) { 778 case MIXER: 779 case OFFLOAD: 780 break; 781 case DIRECT: 782 case DUPLICATING: 783 case RECORD: 784 default: 785 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 786 lStatus = BAD_VALUE; 787 goto Exit; 788 } 789 } 790 791 // Only Pre processor effects are allowed on input threads and only on input threads 792 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 793 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 794 desc->name, desc->flags, mType); 795 lStatus = BAD_VALUE; 796 goto Exit; 797 } 798 799 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 800 801 { // scope for mLock 802 Mutex::Autolock _l(mLock); 803 804 // check for existing effect chain with the requested audio session 805 chain = getEffectChain_l(sessionId); 806 if (chain == 0) { 807 // create a new chain for this session 808 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 809 chain = new EffectChain(this, sessionId); 810 addEffectChain_l(chain); 811 chain->setStrategy(getStrategyForSession_l(sessionId)); 812 chainCreated = true; 813 } else { 814 effect = chain->getEffectFromDesc_l(desc); 815 } 816 817 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 818 819 if (effect == 0) { 820 int id = mAudioFlinger->nextUniqueId(); 821 // Check CPU and memory usage 822 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 823 if (lStatus != NO_ERROR) { 824 goto Exit; 825 } 826 effectRegistered = true; 827 // create a new effect module if none present in the chain 828 effect = new EffectModule(this, chain, desc, id, sessionId); 829 lStatus = effect->status(); 830 if (lStatus != NO_ERROR) { 831 goto Exit; 832 } 833 effect->setOffloaded(mType == OFFLOAD, mId); 834 835 lStatus = chain->addEffect_l(effect); 836 if (lStatus != NO_ERROR) { 837 goto Exit; 838 } 839 effectCreated = true; 840 841 effect->setDevice(mOutDevice); 842 effect->setDevice(mInDevice); 843 effect->setMode(mAudioFlinger->getMode()); 844 effect->setAudioSource(mAudioSource); 845 } 846 // create effect handle and connect it to effect module 847 handle = new EffectHandle(effect, client, effectClient, priority); 848 lStatus = effect->addHandle(handle.get()); 849 if (enabled != NULL) { 850 *enabled = (int)effect->isEnabled(); 851 } 852 } 853 854Exit: 855 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 856 Mutex::Autolock _l(mLock); 857 if (effectCreated) { 858 chain->removeEffect_l(effect); 859 } 860 if (effectRegistered) { 861 AudioSystem::unregisterEffect(effect->id()); 862 } 863 if (chainCreated) { 864 removeEffectChain_l(chain); 865 } 866 handle.clear(); 867 } 868 869 *status = lStatus; 870 return handle; 871} 872 873sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 874{ 875 Mutex::Autolock _l(mLock); 876 return getEffect_l(sessionId, effectId); 877} 878 879sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 880{ 881 sp<EffectChain> chain = getEffectChain_l(sessionId); 882 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 883} 884 885// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 886// PlaybackThread::mLock held 887status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 888{ 889 // check for existing effect chain with the requested audio session 890 int sessionId = effect->sessionId(); 891 sp<EffectChain> chain = getEffectChain_l(sessionId); 892 bool chainCreated = false; 893 894 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 895 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 896 this, effect->desc().name, effect->desc().flags); 897 898 if (chain == 0) { 899 // create a new chain for this session 900 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 901 chain = new EffectChain(this, sessionId); 902 addEffectChain_l(chain); 903 chain->setStrategy(getStrategyForSession_l(sessionId)); 904 chainCreated = true; 905 } 906 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 907 908 if (chain->getEffectFromId_l(effect->id()) != 0) { 909 ALOGW("addEffect_l() %p effect %s already present in chain %p", 910 this, effect->desc().name, chain.get()); 911 return BAD_VALUE; 912 } 913 914 effect->setOffloaded(mType == OFFLOAD, mId); 915 916 status_t status = chain->addEffect_l(effect); 917 if (status != NO_ERROR) { 918 if (chainCreated) { 919 removeEffectChain_l(chain); 920 } 921 return status; 922 } 923 924 effect->setDevice(mOutDevice); 925 effect->setDevice(mInDevice); 926 effect->setMode(mAudioFlinger->getMode()); 927 effect->setAudioSource(mAudioSource); 928 return NO_ERROR; 929} 930 931void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 932 933 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 934 effect_descriptor_t desc = effect->desc(); 935 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 936 detachAuxEffect_l(effect->id()); 937 } 938 939 sp<EffectChain> chain = effect->chain().promote(); 940 if (chain != 0) { 941 // remove effect chain if removing last effect 942 if (chain->removeEffect_l(effect) == 0) { 943 removeEffectChain_l(chain); 944 } 945 } else { 946 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 947 } 948} 949 950void AudioFlinger::ThreadBase::lockEffectChains_l( 951 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 952{ 953 effectChains = mEffectChains; 954 for (size_t i = 0; i < mEffectChains.size(); i++) { 955 mEffectChains[i]->lock(); 956 } 957} 958 959void AudioFlinger::ThreadBase::unlockEffectChains( 960 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 961{ 962 for (size_t i = 0; i < effectChains.size(); i++) { 963 effectChains[i]->unlock(); 964 } 965} 966 967sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 968{ 969 Mutex::Autolock _l(mLock); 970 return getEffectChain_l(sessionId); 971} 972 973sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 974{ 975 size_t size = mEffectChains.size(); 976 for (size_t i = 0; i < size; i++) { 977 if (mEffectChains[i]->sessionId() == sessionId) { 978 return mEffectChains[i]; 979 } 980 } 981 return 0; 982} 983 984void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 985{ 986 Mutex::Autolock _l(mLock); 987 size_t size = mEffectChains.size(); 988 for (size_t i = 0; i < size; i++) { 989 mEffectChains[i]->setMode_l(mode); 990 } 991} 992 993void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 994 EffectHandle *handle, 995 bool unpinIfLast) { 996 997 Mutex::Autolock _l(mLock); 998 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 999 // delete the effect module if removing last handle on it 1000 if (effect->removeHandle(handle) == 0) { 1001 if (!effect->isPinned() || unpinIfLast) { 1002 removeEffect_l(effect); 1003 AudioSystem::unregisterEffect(effect->id()); 1004 } 1005 } 1006} 1007 1008// ---------------------------------------------------------------------------- 1009// Playback 1010// ---------------------------------------------------------------------------- 1011 1012AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1013 AudioStreamOut* output, 1014 audio_io_handle_t id, 1015 audio_devices_t device, 1016 type_t type) 1017 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1018 mNormalFrameCount(0), mMixBuffer(NULL), 1019 mSuspended(0), mBytesWritten(0), 1020 mActiveTracksGeneration(0), 1021 // mStreamTypes[] initialized in constructor body 1022 mOutput(output), 1023 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1024 mMixerStatus(MIXER_IDLE), 1025 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1026 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1027 mBytesRemaining(0), 1028 mCurrentWriteLength(0), 1029 mUseAsyncWrite(false), 1030 mWriteAckSequence(0), 1031 mDrainSequence(0), 1032 mSignalPending(false), 1033 mScreenState(AudioFlinger::mScreenState), 1034 // index 0 is reserved for normal mixer's submix 1035 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1036 // mLatchD, mLatchQ, 1037 mLatchDValid(false), mLatchQValid(false) 1038{ 1039 snprintf(mName, kNameLength, "AudioOut_%X", id); 1040 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1041 1042 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1043 // it would be safer to explicitly pass initial masterVolume/masterMute as 1044 // parameter. 1045 // 1046 // If the HAL we are using has support for master volume or master mute, 1047 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1048 // and the mute set to false). 1049 mMasterVolume = audioFlinger->masterVolume_l(); 1050 mMasterMute = audioFlinger->masterMute_l(); 1051 if (mOutput && mOutput->audioHwDev) { 1052 if (mOutput->audioHwDev->canSetMasterVolume()) { 1053 mMasterVolume = 1.0; 1054 } 1055 1056 if (mOutput->audioHwDev->canSetMasterMute()) { 1057 mMasterMute = false; 1058 } 1059 } 1060 1061 readOutputParameters(); 1062 1063 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1064 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1065 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1066 stream = (audio_stream_type_t) (stream + 1)) { 1067 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1068 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1069 } 1070 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1071 // because mAudioFlinger doesn't have one to copy from 1072} 1073 1074AudioFlinger::PlaybackThread::~PlaybackThread() 1075{ 1076 mAudioFlinger->unregisterWriter(mNBLogWriter); 1077 delete[] mMixBuffer; 1078} 1079 1080void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1081{ 1082 dumpInternals(fd, args); 1083 dumpTracks(fd, args); 1084 dumpEffectChains(fd, args); 1085} 1086 1087void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1088{ 1089 const size_t SIZE = 256; 1090 char buffer[SIZE]; 1091 String8 result; 1092 1093 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1094 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1095 const stream_type_t *st = &mStreamTypes[i]; 1096 if (i > 0) { 1097 result.appendFormat(", "); 1098 } 1099 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1100 if (st->mute) { 1101 result.append("M"); 1102 } 1103 } 1104 result.append("\n"); 1105 write(fd, result.string(), result.length()); 1106 result.clear(); 1107 1108 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1109 result.append(buffer); 1110 Track::appendDumpHeader(result); 1111 for (size_t i = 0; i < mTracks.size(); ++i) { 1112 sp<Track> track = mTracks[i]; 1113 if (track != 0) { 1114 track->dump(buffer, SIZE); 1115 result.append(buffer); 1116 } 1117 } 1118 1119 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1120 result.append(buffer); 1121 Track::appendDumpHeader(result); 1122 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1123 sp<Track> track = mActiveTracks[i].promote(); 1124 if (track != 0) { 1125 track->dump(buffer, SIZE); 1126 result.append(buffer); 1127 } 1128 } 1129 write(fd, result.string(), result.size()); 1130 1131 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1132 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1133 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1134 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1135} 1136 1137void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1138{ 1139 const size_t SIZE = 256; 1140 char buffer[SIZE]; 1141 String8 result; 1142 1143 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1144 result.append(buffer); 1145 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1146 result.append(buffer); 1147 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1148 ns2ms(systemTime() - mLastWriteTime)); 1149 result.append(buffer); 1150 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1151 result.append(buffer); 1152 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1153 result.append(buffer); 1154 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1155 result.append(buffer); 1156 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1157 result.append(buffer); 1158 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1159 result.append(buffer); 1160 write(fd, result.string(), result.size()); 1161 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1162 1163 dumpBase(fd, args); 1164} 1165 1166// Thread virtuals 1167 1168void AudioFlinger::PlaybackThread::onFirstRef() 1169{ 1170 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1171} 1172 1173// ThreadBase virtuals 1174void AudioFlinger::PlaybackThread::preExit() 1175{ 1176 ALOGV(" preExit()"); 1177 // FIXME this is using hard-coded strings but in the future, this functionality will be 1178 // converted to use audio HAL extensions required to support tunneling 1179 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1180} 1181 1182// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1183sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1184 const sp<AudioFlinger::Client>& client, 1185 audio_stream_type_t streamType, 1186 uint32_t sampleRate, 1187 audio_format_t format, 1188 audio_channel_mask_t channelMask, 1189 size_t frameCount, 1190 const sp<IMemory>& sharedBuffer, 1191 int sessionId, 1192 IAudioFlinger::track_flags_t *flags, 1193 pid_t tid, 1194 int uid, 1195 status_t *status) 1196{ 1197 sp<Track> track; 1198 status_t lStatus; 1199 1200 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1201 1202 // client expresses a preference for FAST, but we get the final say 1203 if (*flags & IAudioFlinger::TRACK_FAST) { 1204 if ( 1205 // not timed 1206 (!isTimed) && 1207 // either of these use cases: 1208 ( 1209 // use case 1: shared buffer with any frame count 1210 ( 1211 (sharedBuffer != 0) 1212 ) || 1213 // use case 2: callback handler and frame count is default or at least as large as HAL 1214 ( 1215 (tid != -1) && 1216 ((frameCount == 0) || 1217 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1218 ) 1219 ) && 1220 // PCM data 1221 audio_is_linear_pcm(format) && 1222 // mono or stereo 1223 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1224 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1225#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1226 // hardware sample rate 1227 (sampleRate == mSampleRate) && 1228#endif 1229 // normal mixer has an associated fast mixer 1230 hasFastMixer() && 1231 // there are sufficient fast track slots available 1232 (mFastTrackAvailMask != 0) 1233 // FIXME test that MixerThread for this fast track has a capable output HAL 1234 // FIXME add a permission test also? 1235 ) { 1236 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1237 if (frameCount == 0) { 1238 frameCount = mFrameCount * kFastTrackMultiplier; 1239 } 1240 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1241 frameCount, mFrameCount); 1242 } else { 1243 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1244 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1245 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1246 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1247 audio_is_linear_pcm(format), 1248 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1249 *flags &= ~IAudioFlinger::TRACK_FAST; 1250 // For compatibility with AudioTrack calculation, buffer depth is forced 1251 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1252 // This is probably too conservative, but legacy application code may depend on it. 1253 // If you change this calculation, also review the start threshold which is related. 1254 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1255 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1256 if (minBufCount < 2) { 1257 minBufCount = 2; 1258 } 1259 size_t minFrameCount = mNormalFrameCount * minBufCount; 1260 if (frameCount < minFrameCount) { 1261 frameCount = minFrameCount; 1262 } 1263 } 1264 } 1265 1266 if (mType == DIRECT) { 1267 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1268 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1269 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1270 "for output %p with format %d", 1271 sampleRate, format, channelMask, mOutput, mFormat); 1272 lStatus = BAD_VALUE; 1273 goto Exit; 1274 } 1275 } 1276 } else if (mType == OFFLOAD) { 1277 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1278 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1279 "for output %p with format %d", 1280 sampleRate, format, channelMask, mOutput, mFormat); 1281 lStatus = BAD_VALUE; 1282 goto Exit; 1283 } 1284 } else { 1285 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1286 ALOGE("createTrack_l() Bad parameter: format %d \"" 1287 "for output %p with format %d", 1288 format, mOutput, mFormat); 1289 lStatus = BAD_VALUE; 1290 goto Exit; 1291 } 1292 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1293 if (sampleRate > mSampleRate*2) { 1294 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1295 lStatus = BAD_VALUE; 1296 goto Exit; 1297 } 1298 } 1299 1300 lStatus = initCheck(); 1301 if (lStatus != NO_ERROR) { 1302 ALOGE("Audio driver not initialized."); 1303 goto Exit; 1304 } 1305 1306 { // scope for mLock 1307 Mutex::Autolock _l(mLock); 1308 1309 // all tracks in same audio session must share the same routing strategy otherwise 1310 // conflicts will happen when tracks are moved from one output to another by audio policy 1311 // manager 1312 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1313 for (size_t i = 0; i < mTracks.size(); ++i) { 1314 sp<Track> t = mTracks[i]; 1315 if (t != 0 && !t->isOutputTrack()) { 1316 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1317 if (sessionId == t->sessionId() && strategy != actual) { 1318 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1319 strategy, actual); 1320 lStatus = BAD_VALUE; 1321 goto Exit; 1322 } 1323 } 1324 } 1325 1326 if (!isTimed) { 1327 track = new Track(this, client, streamType, sampleRate, format, 1328 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1329 } else { 1330 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1331 channelMask, frameCount, sharedBuffer, sessionId, uid); 1332 } 1333 1334 // new Track always returns non-NULL, 1335 // but TimedTrack::create() is a factory that could fail by returning NULL 1336 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1337 if (lStatus != NO_ERROR) { 1338 track.clear(); 1339 goto Exit; 1340 } 1341 1342 mTracks.add(track); 1343 1344 sp<EffectChain> chain = getEffectChain_l(sessionId); 1345 if (chain != 0) { 1346 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1347 track->setMainBuffer(chain->inBuffer()); 1348 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1349 chain->incTrackCnt(); 1350 } 1351 1352 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1353 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1354 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1355 // so ask activity manager to do this on our behalf 1356 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1357 } 1358 } 1359 1360 lStatus = NO_ERROR; 1361 1362Exit: 1363 *status = lStatus; 1364 return track; 1365} 1366 1367uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1368{ 1369 return latency; 1370} 1371 1372uint32_t AudioFlinger::PlaybackThread::latency() const 1373{ 1374 Mutex::Autolock _l(mLock); 1375 return latency_l(); 1376} 1377uint32_t AudioFlinger::PlaybackThread::latency_l() const 1378{ 1379 if (initCheck() == NO_ERROR) { 1380 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1381 } else { 1382 return 0; 1383 } 1384} 1385 1386void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1387{ 1388 Mutex::Autolock _l(mLock); 1389 // Don't apply master volume in SW if our HAL can do it for us. 1390 if (mOutput && mOutput->audioHwDev && 1391 mOutput->audioHwDev->canSetMasterVolume()) { 1392 mMasterVolume = 1.0; 1393 } else { 1394 mMasterVolume = value; 1395 } 1396} 1397 1398void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1399{ 1400 Mutex::Autolock _l(mLock); 1401 // Don't apply master mute in SW if our HAL can do it for us. 1402 if (mOutput && mOutput->audioHwDev && 1403 mOutput->audioHwDev->canSetMasterMute()) { 1404 mMasterMute = false; 1405 } else { 1406 mMasterMute = muted; 1407 } 1408} 1409 1410void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1411{ 1412 Mutex::Autolock _l(mLock); 1413 mStreamTypes[stream].volume = value; 1414 broadcast_l(); 1415} 1416 1417void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1418{ 1419 Mutex::Autolock _l(mLock); 1420 mStreamTypes[stream].mute = muted; 1421 broadcast_l(); 1422} 1423 1424float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1425{ 1426 Mutex::Autolock _l(mLock); 1427 return mStreamTypes[stream].volume; 1428} 1429 1430// addTrack_l() must be called with ThreadBase::mLock held 1431status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1432{ 1433 status_t status = ALREADY_EXISTS; 1434 1435 // set retry count for buffer fill 1436 track->mRetryCount = kMaxTrackStartupRetries; 1437 if (mActiveTracks.indexOf(track) < 0) { 1438 // the track is newly added, make sure it fills up all its 1439 // buffers before playing. This is to ensure the client will 1440 // effectively get the latency it requested. 1441 if (!track->isOutputTrack()) { 1442 TrackBase::track_state state = track->mState; 1443 mLock.unlock(); 1444 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1445 mLock.lock(); 1446 // abort track was stopped/paused while we released the lock 1447 if (state != track->mState) { 1448 if (status == NO_ERROR) { 1449 mLock.unlock(); 1450 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1451 mLock.lock(); 1452 } 1453 return INVALID_OPERATION; 1454 } 1455 // abort if start is rejected by audio policy manager 1456 if (status != NO_ERROR) { 1457 return PERMISSION_DENIED; 1458 } 1459#ifdef ADD_BATTERY_DATA 1460 // to track the speaker usage 1461 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1462#endif 1463 } 1464 1465 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1466 track->mResetDone = false; 1467 track->mPresentationCompleteFrames = 0; 1468 mActiveTracks.add(track); 1469 mWakeLockUids.add(track->uid()); 1470 mActiveTracksGeneration++; 1471 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1472 if (chain != 0) { 1473 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1474 track->sessionId()); 1475 chain->incActiveTrackCnt(); 1476 } 1477 1478 status = NO_ERROR; 1479 } 1480 1481 ALOGV("signal playback thread"); 1482 broadcast_l(); 1483 1484 return status; 1485} 1486 1487bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1488{ 1489 track->terminate(); 1490 // active tracks are removed by threadLoop() 1491 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1492 track->mState = TrackBase::STOPPED; 1493 if (!trackActive) { 1494 removeTrack_l(track); 1495 } else if (track->isFastTrack() || track->isOffloaded()) { 1496 track->mState = TrackBase::STOPPING_1; 1497 } 1498 1499 return trackActive; 1500} 1501 1502void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1503{ 1504 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1505 mTracks.remove(track); 1506 deleteTrackName_l(track->name()); 1507 // redundant as track is about to be destroyed, for dumpsys only 1508 track->mName = -1; 1509 if (track->isFastTrack()) { 1510 int index = track->mFastIndex; 1511 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1512 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1513 mFastTrackAvailMask |= 1 << index; 1514 // redundant as track is about to be destroyed, for dumpsys only 1515 track->mFastIndex = -1; 1516 } 1517 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1518 if (chain != 0) { 1519 chain->decTrackCnt(); 1520 } 1521} 1522 1523void AudioFlinger::PlaybackThread::broadcast_l() 1524{ 1525 // Thread could be blocked waiting for async 1526 // so signal it to handle state changes immediately 1527 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1528 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1529 mSignalPending = true; 1530 mWaitWorkCV.broadcast(); 1531} 1532 1533String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1534{ 1535 Mutex::Autolock _l(mLock); 1536 if (initCheck() != NO_ERROR) { 1537 return String8(); 1538 } 1539 1540 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1541 const String8 out_s8(s); 1542 free(s); 1543 return out_s8; 1544} 1545 1546// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1547void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1548 AudioSystem::OutputDescriptor desc; 1549 void *param2 = NULL; 1550 1551 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1552 param); 1553 1554 switch (event) { 1555 case AudioSystem::OUTPUT_OPENED: 1556 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1557 desc.channelMask = mChannelMask; 1558 desc.samplingRate = mSampleRate; 1559 desc.format = mFormat; 1560 desc.frameCount = mNormalFrameCount; // FIXME see 1561 // AudioFlinger::frameCount(audio_io_handle_t) 1562 desc.latency = latency(); 1563 param2 = &desc; 1564 break; 1565 1566 case AudioSystem::STREAM_CONFIG_CHANGED: 1567 param2 = ¶m; 1568 case AudioSystem::OUTPUT_CLOSED: 1569 default: 1570 break; 1571 } 1572 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1573} 1574 1575void AudioFlinger::PlaybackThread::writeCallback() 1576{ 1577 ALOG_ASSERT(mCallbackThread != 0); 1578 mCallbackThread->resetWriteBlocked(); 1579} 1580 1581void AudioFlinger::PlaybackThread::drainCallback() 1582{ 1583 ALOG_ASSERT(mCallbackThread != 0); 1584 mCallbackThread->resetDraining(); 1585} 1586 1587void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1588{ 1589 Mutex::Autolock _l(mLock); 1590 // reject out of sequence requests 1591 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1592 mWriteAckSequence &= ~1; 1593 mWaitWorkCV.signal(); 1594 } 1595} 1596 1597void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1598{ 1599 Mutex::Autolock _l(mLock); 1600 // reject out of sequence requests 1601 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1602 mDrainSequence &= ~1; 1603 mWaitWorkCV.signal(); 1604 } 1605} 1606 1607// static 1608int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1609 void *param, 1610 void *cookie) 1611{ 1612 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1613 ALOGV("asyncCallback() event %d", event); 1614 switch (event) { 1615 case STREAM_CBK_EVENT_WRITE_READY: 1616 me->writeCallback(); 1617 break; 1618 case STREAM_CBK_EVENT_DRAIN_READY: 1619 me->drainCallback(); 1620 break; 1621 default: 1622 ALOGW("asyncCallback() unknown event %d", event); 1623 break; 1624 } 1625 return 0; 1626} 1627 1628void AudioFlinger::PlaybackThread::readOutputParameters() 1629{ 1630 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1631 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1632 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1633 if (!audio_is_output_channel(mChannelMask)) { 1634 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1635 } 1636 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1637 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1638 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1639 } 1640 mChannelCount = popcount(mChannelMask); 1641 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1642 if (!audio_is_valid_format(mFormat)) { 1643 LOG_FATAL("HAL format %d not valid for output", mFormat); 1644 } 1645 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1646 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1647 mFormat); 1648 } 1649 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1650 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1651 mFrameCount = mBufferSize / mFrameSize; 1652 if (mFrameCount & 15) { 1653 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1654 mFrameCount); 1655 } 1656 1657 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1658 (mOutput->stream->set_callback != NULL)) { 1659 if (mOutput->stream->set_callback(mOutput->stream, 1660 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1661 mUseAsyncWrite = true; 1662 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1663 } 1664 } 1665 1666 // Calculate size of normal mix buffer relative to the HAL output buffer size 1667 double multiplier = 1.0; 1668 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1669 kUseFastMixer == FastMixer_Dynamic)) { 1670 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1671 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1672 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1673 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1674 maxNormalFrameCount = maxNormalFrameCount & ~15; 1675 if (maxNormalFrameCount < minNormalFrameCount) { 1676 maxNormalFrameCount = minNormalFrameCount; 1677 } 1678 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1679 if (multiplier <= 1.0) { 1680 multiplier = 1.0; 1681 } else if (multiplier <= 2.0) { 1682 if (2 * mFrameCount <= maxNormalFrameCount) { 1683 multiplier = 2.0; 1684 } else { 1685 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1686 } 1687 } else { 1688 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1689 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1690 // track, but we sometimes have to do this to satisfy the maximum frame count 1691 // constraint) 1692 // FIXME this rounding up should not be done if no HAL SRC 1693 uint32_t truncMult = (uint32_t) multiplier; 1694 if ((truncMult & 1)) { 1695 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1696 ++truncMult; 1697 } 1698 } 1699 multiplier = (double) truncMult; 1700 } 1701 } 1702 mNormalFrameCount = multiplier * mFrameCount; 1703 // round up to nearest 16 frames to satisfy AudioMixer 1704 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1705 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1706 mNormalFrameCount); 1707 1708 delete[] mMixBuffer; 1709 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1710 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1711 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1712 memset(mMixBuffer, 0, normalBufferSize); 1713 1714 // force reconfiguration of effect chains and engines to take new buffer size and audio 1715 // parameters into account 1716 // Note that mLock is not held when readOutputParameters() is called from the constructor 1717 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1718 // matter. 1719 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1720 Vector< sp<EffectChain> > effectChains = mEffectChains; 1721 for (size_t i = 0; i < effectChains.size(); i ++) { 1722 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1723 } 1724} 1725 1726 1727status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1728{ 1729 if (halFrames == NULL || dspFrames == NULL) { 1730 return BAD_VALUE; 1731 } 1732 Mutex::Autolock _l(mLock); 1733 if (initCheck() != NO_ERROR) { 1734 return INVALID_OPERATION; 1735 } 1736 size_t framesWritten = mBytesWritten / mFrameSize; 1737 *halFrames = framesWritten; 1738 1739 if (isSuspended()) { 1740 // return an estimation of rendered frames when the output is suspended 1741 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1742 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1743 return NO_ERROR; 1744 } else { 1745 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1746 } 1747} 1748 1749uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1750{ 1751 Mutex::Autolock _l(mLock); 1752 uint32_t result = 0; 1753 if (getEffectChain_l(sessionId) != 0) { 1754 result = EFFECT_SESSION; 1755 } 1756 1757 for (size_t i = 0; i < mTracks.size(); ++i) { 1758 sp<Track> track = mTracks[i]; 1759 if (sessionId == track->sessionId() && !track->isInvalid()) { 1760 result |= TRACK_SESSION; 1761 break; 1762 } 1763 } 1764 1765 return result; 1766} 1767 1768uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1769{ 1770 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1771 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1772 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1773 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1774 } 1775 for (size_t i = 0; i < mTracks.size(); i++) { 1776 sp<Track> track = mTracks[i]; 1777 if (sessionId == track->sessionId() && !track->isInvalid()) { 1778 return AudioSystem::getStrategyForStream(track->streamType()); 1779 } 1780 } 1781 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1782} 1783 1784 1785AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1786{ 1787 Mutex::Autolock _l(mLock); 1788 return mOutput; 1789} 1790 1791AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1792{ 1793 Mutex::Autolock _l(mLock); 1794 AudioStreamOut *output = mOutput; 1795 mOutput = NULL; 1796 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1797 // must push a NULL and wait for ack 1798 mOutputSink.clear(); 1799 mPipeSink.clear(); 1800 mNormalSink.clear(); 1801 return output; 1802} 1803 1804// this method must always be called either with ThreadBase mLock held or inside the thread loop 1805audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1806{ 1807 if (mOutput == NULL) { 1808 return NULL; 1809 } 1810 return &mOutput->stream->common; 1811} 1812 1813uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1814{ 1815 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1816} 1817 1818status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1819{ 1820 if (!isValidSyncEvent(event)) { 1821 return BAD_VALUE; 1822 } 1823 1824 Mutex::Autolock _l(mLock); 1825 1826 for (size_t i = 0; i < mTracks.size(); ++i) { 1827 sp<Track> track = mTracks[i]; 1828 if (event->triggerSession() == track->sessionId()) { 1829 (void) track->setSyncEvent(event); 1830 return NO_ERROR; 1831 } 1832 } 1833 1834 return NAME_NOT_FOUND; 1835} 1836 1837bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1838{ 1839 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1840} 1841 1842void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1843 const Vector< sp<Track> >& tracksToRemove) 1844{ 1845 size_t count = tracksToRemove.size(); 1846 if (count > 0) { 1847 for (size_t i = 0 ; i < count ; i++) { 1848 const sp<Track>& track = tracksToRemove.itemAt(i); 1849 if (!track->isOutputTrack()) { 1850 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1851#ifdef ADD_BATTERY_DATA 1852 // to track the speaker usage 1853 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1854#endif 1855 if (track->isTerminated()) { 1856 AudioSystem::releaseOutput(mId); 1857 } 1858 } 1859 } 1860 } 1861} 1862 1863void AudioFlinger::PlaybackThread::checkSilentMode_l() 1864{ 1865 if (!mMasterMute) { 1866 char value[PROPERTY_VALUE_MAX]; 1867 if (property_get("ro.audio.silent", value, "0") > 0) { 1868 char *endptr; 1869 unsigned long ul = strtoul(value, &endptr, 0); 1870 if (*endptr == '\0' && ul != 0) { 1871 ALOGD("Silence is golden"); 1872 // The setprop command will not allow a property to be changed after 1873 // the first time it is set, so we don't have to worry about un-muting. 1874 setMasterMute_l(true); 1875 } 1876 } 1877 } 1878} 1879 1880// shared by MIXER and DIRECT, overridden by DUPLICATING 1881ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1882{ 1883 // FIXME rewrite to reduce number of system calls 1884 mLastWriteTime = systemTime(); 1885 mInWrite = true; 1886 ssize_t bytesWritten; 1887 1888 // If an NBAIO sink is present, use it to write the normal mixer's submix 1889 if (mNormalSink != 0) { 1890#define mBitShift 2 // FIXME 1891 size_t count = mBytesRemaining >> mBitShift; 1892 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1893 ATRACE_BEGIN("write"); 1894 // update the setpoint when AudioFlinger::mScreenState changes 1895 uint32_t screenState = AudioFlinger::mScreenState; 1896 if (screenState != mScreenState) { 1897 mScreenState = screenState; 1898 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1899 if (pipe != NULL) { 1900 pipe->setAvgFrames((mScreenState & 1) ? 1901 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1902 } 1903 } 1904 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1905 ATRACE_END(); 1906 if (framesWritten > 0) { 1907 bytesWritten = framesWritten << mBitShift; 1908 } else { 1909 bytesWritten = framesWritten; 1910 } 1911 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1912 if (status == NO_ERROR) { 1913 size_t totalFramesWritten = mNormalSink->framesWritten(); 1914 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1915 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1916 mLatchDValid = true; 1917 } 1918 } 1919 // otherwise use the HAL / AudioStreamOut directly 1920 } else { 1921 // Direct output and offload threads 1922 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1923 if (mUseAsyncWrite) { 1924 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1925 mWriteAckSequence += 2; 1926 mWriteAckSequence |= 1; 1927 ALOG_ASSERT(mCallbackThread != 0); 1928 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1929 } 1930 // FIXME We should have an implementation of timestamps for direct output threads. 1931 // They are used e.g for multichannel PCM playback over HDMI. 1932 bytesWritten = mOutput->stream->write(mOutput->stream, 1933 mMixBuffer + offset, mBytesRemaining); 1934 if (mUseAsyncWrite && 1935 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1936 // do not wait for async callback in case of error of full write 1937 mWriteAckSequence &= ~1; 1938 ALOG_ASSERT(mCallbackThread != 0); 1939 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1940 } 1941 } 1942 1943 mNumWrites++; 1944 mInWrite = false; 1945 1946 return bytesWritten; 1947} 1948 1949void AudioFlinger::PlaybackThread::threadLoop_drain() 1950{ 1951 if (mOutput->stream->drain) { 1952 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1953 if (mUseAsyncWrite) { 1954 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1955 mDrainSequence |= 1; 1956 ALOG_ASSERT(mCallbackThread != 0); 1957 mCallbackThread->setDraining(mDrainSequence); 1958 } 1959 mOutput->stream->drain(mOutput->stream, 1960 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1961 : AUDIO_DRAIN_ALL); 1962 } 1963} 1964 1965void AudioFlinger::PlaybackThread::threadLoop_exit() 1966{ 1967 // Default implementation has nothing to do 1968} 1969 1970/* 1971The derived values that are cached: 1972 - mixBufferSize from frame count * frame size 1973 - activeSleepTime from activeSleepTimeUs() 1974 - idleSleepTime from idleSleepTimeUs() 1975 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1976 - maxPeriod from frame count and sample rate (MIXER only) 1977 1978The parameters that affect these derived values are: 1979 - frame count 1980 - frame size 1981 - sample rate 1982 - device type: A2DP or not 1983 - device latency 1984 - format: PCM or not 1985 - active sleep time 1986 - idle sleep time 1987*/ 1988 1989void AudioFlinger::PlaybackThread::cacheParameters_l() 1990{ 1991 mixBufferSize = mNormalFrameCount * mFrameSize; 1992 activeSleepTime = activeSleepTimeUs(); 1993 idleSleepTime = idleSleepTimeUs(); 1994} 1995 1996void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1997{ 1998 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1999 this, streamType, mTracks.size()); 2000 Mutex::Autolock _l(mLock); 2001 2002 size_t size = mTracks.size(); 2003 for (size_t i = 0; i < size; i++) { 2004 sp<Track> t = mTracks[i]; 2005 if (t->streamType() == streamType) { 2006 t->invalidate(); 2007 } 2008 } 2009} 2010 2011status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2012{ 2013 int session = chain->sessionId(); 2014 int16_t *buffer = mMixBuffer; 2015 bool ownsBuffer = false; 2016 2017 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2018 if (session > 0) { 2019 // Only one effect chain can be present in direct output thread and it uses 2020 // the mix buffer as input 2021 if (mType != DIRECT) { 2022 size_t numSamples = mNormalFrameCount * mChannelCount; 2023 buffer = new int16_t[numSamples]; 2024 memset(buffer, 0, numSamples * sizeof(int16_t)); 2025 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2026 ownsBuffer = true; 2027 } 2028 2029 // Attach all tracks with same session ID to this chain. 2030 for (size_t i = 0; i < mTracks.size(); ++i) { 2031 sp<Track> track = mTracks[i]; 2032 if (session == track->sessionId()) { 2033 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2034 buffer); 2035 track->setMainBuffer(buffer); 2036 chain->incTrackCnt(); 2037 } 2038 } 2039 2040 // indicate all active tracks in the chain 2041 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2042 sp<Track> track = mActiveTracks[i].promote(); 2043 if (track == 0) { 2044 continue; 2045 } 2046 if (session == track->sessionId()) { 2047 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2048 chain->incActiveTrackCnt(); 2049 } 2050 } 2051 } 2052 2053 chain->setInBuffer(buffer, ownsBuffer); 2054 chain->setOutBuffer(mMixBuffer); 2055 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2056 // chains list in order to be processed last as it contains output stage effects 2057 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2058 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2059 // after track specific effects and before output stage 2060 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2061 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2062 // Effect chain for other sessions are inserted at beginning of effect 2063 // chains list to be processed before output mix effects. Relative order between other 2064 // sessions is not important 2065 size_t size = mEffectChains.size(); 2066 size_t i = 0; 2067 for (i = 0; i < size; i++) { 2068 if (mEffectChains[i]->sessionId() < session) { 2069 break; 2070 } 2071 } 2072 mEffectChains.insertAt(chain, i); 2073 checkSuspendOnAddEffectChain_l(chain); 2074 2075 return NO_ERROR; 2076} 2077 2078size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2079{ 2080 int session = chain->sessionId(); 2081 2082 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2083 2084 for (size_t i = 0; i < mEffectChains.size(); i++) { 2085 if (chain == mEffectChains[i]) { 2086 mEffectChains.removeAt(i); 2087 // detach all active tracks from the chain 2088 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2089 sp<Track> track = mActiveTracks[i].promote(); 2090 if (track == 0) { 2091 continue; 2092 } 2093 if (session == track->sessionId()) { 2094 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2095 chain.get(), session); 2096 chain->decActiveTrackCnt(); 2097 } 2098 } 2099 2100 // detach all tracks with same session ID from this chain 2101 for (size_t i = 0; i < mTracks.size(); ++i) { 2102 sp<Track> track = mTracks[i]; 2103 if (session == track->sessionId()) { 2104 track->setMainBuffer(mMixBuffer); 2105 chain->decTrackCnt(); 2106 } 2107 } 2108 break; 2109 } 2110 } 2111 return mEffectChains.size(); 2112} 2113 2114status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2115 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2116{ 2117 Mutex::Autolock _l(mLock); 2118 return attachAuxEffect_l(track, EffectId); 2119} 2120 2121status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2122 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2123{ 2124 status_t status = NO_ERROR; 2125 2126 if (EffectId == 0) { 2127 track->setAuxBuffer(0, NULL); 2128 } else { 2129 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2130 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2131 if (effect != 0) { 2132 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2133 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2134 } else { 2135 status = INVALID_OPERATION; 2136 } 2137 } else { 2138 status = BAD_VALUE; 2139 } 2140 } 2141 return status; 2142} 2143 2144void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2145{ 2146 for (size_t i = 0; i < mTracks.size(); ++i) { 2147 sp<Track> track = mTracks[i]; 2148 if (track->auxEffectId() == effectId) { 2149 attachAuxEffect_l(track, 0); 2150 } 2151 } 2152} 2153 2154bool AudioFlinger::PlaybackThread::threadLoop() 2155{ 2156 Vector< sp<Track> > tracksToRemove; 2157 2158 standbyTime = systemTime(); 2159 2160 // MIXER 2161 nsecs_t lastWarning = 0; 2162 2163 // DUPLICATING 2164 // FIXME could this be made local to while loop? 2165 writeFrames = 0; 2166 2167 int lastGeneration = 0; 2168 2169 cacheParameters_l(); 2170 sleepTime = idleSleepTime; 2171 2172 if (mType == MIXER) { 2173 sleepTimeShift = 0; 2174 } 2175 2176 CpuStats cpuStats; 2177 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2178 2179 acquireWakeLock(); 2180 2181 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2182 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2183 // and then that string will be logged at the next convenient opportunity. 2184 const char *logString = NULL; 2185 2186 checkSilentMode_l(); 2187 2188 while (!exitPending()) 2189 { 2190 cpuStats.sample(myName); 2191 2192 Vector< sp<EffectChain> > effectChains; 2193 2194 processConfigEvents(); 2195 2196 { // scope for mLock 2197 2198 Mutex::Autolock _l(mLock); 2199 2200 if (logString != NULL) { 2201 mNBLogWriter->logTimestamp(); 2202 mNBLogWriter->log(logString); 2203 logString = NULL; 2204 } 2205 2206 if (mLatchDValid) { 2207 mLatchQ = mLatchD; 2208 mLatchDValid = false; 2209 mLatchQValid = true; 2210 } 2211 2212 if (checkForNewParameters_l()) { 2213 cacheParameters_l(); 2214 } 2215 2216 saveOutputTracks(); 2217 if (mSignalPending) { 2218 // A signal was raised while we were unlocked 2219 mSignalPending = false; 2220 } else if (waitingAsyncCallback_l()) { 2221 if (exitPending()) { 2222 break; 2223 } 2224 releaseWakeLock_l(); 2225 mWakeLockUids.clear(); 2226 mActiveTracksGeneration++; 2227 ALOGV("wait async completion"); 2228 mWaitWorkCV.wait(mLock); 2229 ALOGV("async completion/wake"); 2230 acquireWakeLock_l(); 2231 standbyTime = systemTime() + standbyDelay; 2232 sleepTime = 0; 2233 2234 continue; 2235 } 2236 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2237 isSuspended()) { 2238 // put audio hardware into standby after short delay 2239 if (shouldStandby_l()) { 2240 2241 threadLoop_standby(); 2242 2243 mStandby = true; 2244 } 2245 2246 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2247 // we're about to wait, flush the binder command buffer 2248 IPCThreadState::self()->flushCommands(); 2249 2250 clearOutputTracks(); 2251 2252 if (exitPending()) { 2253 break; 2254 } 2255 2256 releaseWakeLock_l(); 2257 mWakeLockUids.clear(); 2258 mActiveTracksGeneration++; 2259 // wait until we have something to do... 2260 ALOGV("%s going to sleep", myName.string()); 2261 mWaitWorkCV.wait(mLock); 2262 ALOGV("%s waking up", myName.string()); 2263 acquireWakeLock_l(); 2264 2265 mMixerStatus = MIXER_IDLE; 2266 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2267 mBytesWritten = 0; 2268 mBytesRemaining = 0; 2269 checkSilentMode_l(); 2270 2271 standbyTime = systemTime() + standbyDelay; 2272 sleepTime = idleSleepTime; 2273 if (mType == MIXER) { 2274 sleepTimeShift = 0; 2275 } 2276 2277 continue; 2278 } 2279 } 2280 // mMixerStatusIgnoringFastTracks is also updated internally 2281 mMixerStatus = prepareTracks_l(&tracksToRemove); 2282 2283 // compare with previously applied list 2284 if (lastGeneration != mActiveTracksGeneration) { 2285 // update wakelock 2286 updateWakeLockUids_l(mWakeLockUids); 2287 lastGeneration = mActiveTracksGeneration; 2288 } 2289 2290 // prevent any changes in effect chain list and in each effect chain 2291 // during mixing and effect process as the audio buffers could be deleted 2292 // or modified if an effect is created or deleted 2293 lockEffectChains_l(effectChains); 2294 } // mLock scope ends 2295 2296 if (mBytesRemaining == 0) { 2297 mCurrentWriteLength = 0; 2298 if (mMixerStatus == MIXER_TRACKS_READY) { 2299 // threadLoop_mix() sets mCurrentWriteLength 2300 threadLoop_mix(); 2301 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2302 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2303 // threadLoop_sleepTime sets sleepTime to 0 if data 2304 // must be written to HAL 2305 threadLoop_sleepTime(); 2306 if (sleepTime == 0) { 2307 mCurrentWriteLength = mixBufferSize; 2308 } 2309 } 2310 mBytesRemaining = mCurrentWriteLength; 2311 if (isSuspended()) { 2312 sleepTime = suspendSleepTimeUs(); 2313 // simulate write to HAL when suspended 2314 mBytesWritten += mixBufferSize; 2315 mBytesRemaining = 0; 2316 } 2317 2318 // only process effects if we're going to write 2319 if (sleepTime == 0 && mType != OFFLOAD) { 2320 for (size_t i = 0; i < effectChains.size(); i ++) { 2321 effectChains[i]->process_l(); 2322 } 2323 } 2324 } 2325 // Process effect chains for offloaded thread even if no audio 2326 // was read from audio track: process only updates effect state 2327 // and thus does have to be synchronized with audio writes but may have 2328 // to be called while waiting for async write callback 2329 if (mType == OFFLOAD) { 2330 for (size_t i = 0; i < effectChains.size(); i ++) { 2331 effectChains[i]->process_l(); 2332 } 2333 } 2334 2335 // enable changes in effect chain 2336 unlockEffectChains(effectChains); 2337 2338 if (!waitingAsyncCallback()) { 2339 // sleepTime == 0 means we must write to audio hardware 2340 if (sleepTime == 0) { 2341 if (mBytesRemaining) { 2342 ssize_t ret = threadLoop_write(); 2343 if (ret < 0) { 2344 mBytesRemaining = 0; 2345 } else { 2346 mBytesWritten += ret; 2347 mBytesRemaining -= ret; 2348 } 2349 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2350 (mMixerStatus == MIXER_DRAIN_ALL)) { 2351 threadLoop_drain(); 2352 } 2353if (mType == MIXER) { 2354 // write blocked detection 2355 nsecs_t now = systemTime(); 2356 nsecs_t delta = now - mLastWriteTime; 2357 if (!mStandby && delta > maxPeriod) { 2358 mNumDelayedWrites++; 2359 if ((now - lastWarning) > kWarningThrottleNs) { 2360 ATRACE_NAME("underrun"); 2361 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2362 ns2ms(delta), mNumDelayedWrites, this); 2363 lastWarning = now; 2364 } 2365 } 2366} 2367 2368 mStandby = false; 2369 } else { 2370 usleep(sleepTime); 2371 } 2372 } 2373 2374 // Finally let go of removed track(s), without the lock held 2375 // since we can't guarantee the destructors won't acquire that 2376 // same lock. This will also mutate and push a new fast mixer state. 2377 threadLoop_removeTracks(tracksToRemove); 2378 tracksToRemove.clear(); 2379 2380 // FIXME I don't understand the need for this here; 2381 // it was in the original code but maybe the 2382 // assignment in saveOutputTracks() makes this unnecessary? 2383 clearOutputTracks(); 2384 2385 // Effect chains will be actually deleted here if they were removed from 2386 // mEffectChains list during mixing or effects processing 2387 effectChains.clear(); 2388 2389 // FIXME Note that the above .clear() is no longer necessary since effectChains 2390 // is now local to this block, but will keep it for now (at least until merge done). 2391 } 2392 2393 threadLoop_exit(); 2394 2395 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2396 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2397 // put output stream into standby mode 2398 if (!mStandby) { 2399 mOutput->stream->common.standby(&mOutput->stream->common); 2400 } 2401 } 2402 2403 releaseWakeLock(); 2404 mWakeLockUids.clear(); 2405 mActiveTracksGeneration++; 2406 2407 ALOGV("Thread %p type %d exiting", this, mType); 2408 return false; 2409} 2410 2411// removeTracks_l() must be called with ThreadBase::mLock held 2412void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2413{ 2414 size_t count = tracksToRemove.size(); 2415 if (count > 0) { 2416 for (size_t i=0 ; i<count ; i++) { 2417 const sp<Track>& track = tracksToRemove.itemAt(i); 2418 mActiveTracks.remove(track); 2419 mWakeLockUids.remove(track->uid()); 2420 mActiveTracksGeneration++; 2421 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2422 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2423 if (chain != 0) { 2424 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2425 track->sessionId()); 2426 chain->decActiveTrackCnt(); 2427 } 2428 if (track->isTerminated()) { 2429 removeTrack_l(track); 2430 } 2431 } 2432 } 2433 2434} 2435 2436status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2437{ 2438 if (mNormalSink != 0) { 2439 return mNormalSink->getTimestamp(timestamp); 2440 } 2441 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2442 uint64_t position64; 2443 int ret = mOutput->stream->get_presentation_position( 2444 mOutput->stream, &position64, ×tamp.mTime); 2445 if (ret == 0) { 2446 timestamp.mPosition = (uint32_t)position64; 2447 return NO_ERROR; 2448 } 2449 } 2450 return INVALID_OPERATION; 2451} 2452// ---------------------------------------------------------------------------- 2453 2454AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2455 audio_io_handle_t id, audio_devices_t device, type_t type) 2456 : PlaybackThread(audioFlinger, output, id, device, type), 2457 // mAudioMixer below 2458 // mFastMixer below 2459 mFastMixerFutex(0) 2460 // mOutputSink below 2461 // mPipeSink below 2462 // mNormalSink below 2463{ 2464 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2465 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2466 "mFrameCount=%d, mNormalFrameCount=%d", 2467 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2468 mNormalFrameCount); 2469 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2470 2471 // FIXME - Current mixer implementation only supports stereo output 2472 if (mChannelCount != FCC_2) { 2473 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2474 } 2475 2476 // create an NBAIO sink for the HAL output stream, and negotiate 2477 mOutputSink = new AudioStreamOutSink(output->stream); 2478 size_t numCounterOffers = 0; 2479 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2480 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2481 ALOG_ASSERT(index == 0); 2482 2483 // initialize fast mixer depending on configuration 2484 bool initFastMixer; 2485 switch (kUseFastMixer) { 2486 case FastMixer_Never: 2487 initFastMixer = false; 2488 break; 2489 case FastMixer_Always: 2490 initFastMixer = true; 2491 break; 2492 case FastMixer_Static: 2493 case FastMixer_Dynamic: 2494 initFastMixer = mFrameCount < mNormalFrameCount; 2495 break; 2496 } 2497 if (initFastMixer) { 2498 2499 // create a MonoPipe to connect our submix to FastMixer 2500 NBAIO_Format format = mOutputSink->format(); 2501 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2502 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2503 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2504 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2505 const NBAIO_Format offers[1] = {format}; 2506 size_t numCounterOffers = 0; 2507 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2508 ALOG_ASSERT(index == 0); 2509 monoPipe->setAvgFrames((mScreenState & 1) ? 2510 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2511 mPipeSink = monoPipe; 2512 2513#ifdef TEE_SINK 2514 if (mTeeSinkOutputEnabled) { 2515 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2516 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2517 numCounterOffers = 0; 2518 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2519 ALOG_ASSERT(index == 0); 2520 mTeeSink = teeSink; 2521 PipeReader *teeSource = new PipeReader(*teeSink); 2522 numCounterOffers = 0; 2523 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2524 ALOG_ASSERT(index == 0); 2525 mTeeSource = teeSource; 2526 } 2527#endif 2528 2529 // create fast mixer and configure it initially with just one fast track for our submix 2530 mFastMixer = new FastMixer(); 2531 FastMixerStateQueue *sq = mFastMixer->sq(); 2532#ifdef STATE_QUEUE_DUMP 2533 sq->setObserverDump(&mStateQueueObserverDump); 2534 sq->setMutatorDump(&mStateQueueMutatorDump); 2535#endif 2536 FastMixerState *state = sq->begin(); 2537 FastTrack *fastTrack = &state->mFastTracks[0]; 2538 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2539 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2540 fastTrack->mVolumeProvider = NULL; 2541 fastTrack->mGeneration++; 2542 state->mFastTracksGen++; 2543 state->mTrackMask = 1; 2544 // fast mixer will use the HAL output sink 2545 state->mOutputSink = mOutputSink.get(); 2546 state->mOutputSinkGen++; 2547 state->mFrameCount = mFrameCount; 2548 state->mCommand = FastMixerState::COLD_IDLE; 2549 // already done in constructor initialization list 2550 //mFastMixerFutex = 0; 2551 state->mColdFutexAddr = &mFastMixerFutex; 2552 state->mColdGen++; 2553 state->mDumpState = &mFastMixerDumpState; 2554#ifdef TEE_SINK 2555 state->mTeeSink = mTeeSink.get(); 2556#endif 2557 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2558 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2559 sq->end(); 2560 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2561 2562 // start the fast mixer 2563 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2564 pid_t tid = mFastMixer->getTid(); 2565 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2566 if (err != 0) { 2567 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2568 kPriorityFastMixer, getpid_cached, tid, err); 2569 } 2570 2571#ifdef AUDIO_WATCHDOG 2572 // create and start the watchdog 2573 mAudioWatchdog = new AudioWatchdog(); 2574 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2575 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2576 tid = mAudioWatchdog->getTid(); 2577 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2578 if (err != 0) { 2579 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2580 kPriorityFastMixer, getpid_cached, tid, err); 2581 } 2582#endif 2583 2584 } else { 2585 mFastMixer = NULL; 2586 } 2587 2588 switch (kUseFastMixer) { 2589 case FastMixer_Never: 2590 case FastMixer_Dynamic: 2591 mNormalSink = mOutputSink; 2592 break; 2593 case FastMixer_Always: 2594 mNormalSink = mPipeSink; 2595 break; 2596 case FastMixer_Static: 2597 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2598 break; 2599 } 2600} 2601 2602AudioFlinger::MixerThread::~MixerThread() 2603{ 2604 if (mFastMixer != NULL) { 2605 FastMixerStateQueue *sq = mFastMixer->sq(); 2606 FastMixerState *state = sq->begin(); 2607 if (state->mCommand == FastMixerState::COLD_IDLE) { 2608 int32_t old = android_atomic_inc(&mFastMixerFutex); 2609 if (old == -1) { 2610 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2611 } 2612 } 2613 state->mCommand = FastMixerState::EXIT; 2614 sq->end(); 2615 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2616 mFastMixer->join(); 2617 // Though the fast mixer thread has exited, it's state queue is still valid. 2618 // We'll use that extract the final state which contains one remaining fast track 2619 // corresponding to our sub-mix. 2620 state = sq->begin(); 2621 ALOG_ASSERT(state->mTrackMask == 1); 2622 FastTrack *fastTrack = &state->mFastTracks[0]; 2623 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2624 delete fastTrack->mBufferProvider; 2625 sq->end(false /*didModify*/); 2626 delete mFastMixer; 2627#ifdef AUDIO_WATCHDOG 2628 if (mAudioWatchdog != 0) { 2629 mAudioWatchdog->requestExit(); 2630 mAudioWatchdog->requestExitAndWait(); 2631 mAudioWatchdog.clear(); 2632 } 2633#endif 2634 } 2635 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2636 delete mAudioMixer; 2637} 2638 2639 2640uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2641{ 2642 if (mFastMixer != NULL) { 2643 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2644 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2645 } 2646 return latency; 2647} 2648 2649 2650void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2651{ 2652 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2653} 2654 2655ssize_t AudioFlinger::MixerThread::threadLoop_write() 2656{ 2657 // FIXME we should only do one push per cycle; confirm this is true 2658 // Start the fast mixer if it's not already running 2659 if (mFastMixer != NULL) { 2660 FastMixerStateQueue *sq = mFastMixer->sq(); 2661 FastMixerState *state = sq->begin(); 2662 if (state->mCommand != FastMixerState::MIX_WRITE && 2663 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2664 if (state->mCommand == FastMixerState::COLD_IDLE) { 2665 int32_t old = android_atomic_inc(&mFastMixerFutex); 2666 if (old == -1) { 2667 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2668 } 2669#ifdef AUDIO_WATCHDOG 2670 if (mAudioWatchdog != 0) { 2671 mAudioWatchdog->resume(); 2672 } 2673#endif 2674 } 2675 state->mCommand = FastMixerState::MIX_WRITE; 2676 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2677 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2678 sq->end(); 2679 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2680 if (kUseFastMixer == FastMixer_Dynamic) { 2681 mNormalSink = mPipeSink; 2682 } 2683 } else { 2684 sq->end(false /*didModify*/); 2685 } 2686 } 2687 return PlaybackThread::threadLoop_write(); 2688} 2689 2690void AudioFlinger::MixerThread::threadLoop_standby() 2691{ 2692 // Idle the fast mixer if it's currently running 2693 if (mFastMixer != NULL) { 2694 FastMixerStateQueue *sq = mFastMixer->sq(); 2695 FastMixerState *state = sq->begin(); 2696 if (!(state->mCommand & FastMixerState::IDLE)) { 2697 state->mCommand = FastMixerState::COLD_IDLE; 2698 state->mColdFutexAddr = &mFastMixerFutex; 2699 state->mColdGen++; 2700 mFastMixerFutex = 0; 2701 sq->end(); 2702 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2703 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2704 if (kUseFastMixer == FastMixer_Dynamic) { 2705 mNormalSink = mOutputSink; 2706 } 2707#ifdef AUDIO_WATCHDOG 2708 if (mAudioWatchdog != 0) { 2709 mAudioWatchdog->pause(); 2710 } 2711#endif 2712 } else { 2713 sq->end(false /*didModify*/); 2714 } 2715 } 2716 PlaybackThread::threadLoop_standby(); 2717} 2718 2719// Empty implementation for standard mixer 2720// Overridden for offloaded playback 2721void AudioFlinger::PlaybackThread::flushOutput_l() 2722{ 2723} 2724 2725bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2726{ 2727 return false; 2728} 2729 2730bool AudioFlinger::PlaybackThread::shouldStandby_l() 2731{ 2732 return !mStandby; 2733} 2734 2735bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2736{ 2737 Mutex::Autolock _l(mLock); 2738 return waitingAsyncCallback_l(); 2739} 2740 2741// shared by MIXER and DIRECT, overridden by DUPLICATING 2742void AudioFlinger::PlaybackThread::threadLoop_standby() 2743{ 2744 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2745 mOutput->stream->common.standby(&mOutput->stream->common); 2746 if (mUseAsyncWrite != 0) { 2747 // discard any pending drain or write ack by incrementing sequence 2748 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2749 mDrainSequence = (mDrainSequence + 2) & ~1; 2750 ALOG_ASSERT(mCallbackThread != 0); 2751 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2752 mCallbackThread->setDraining(mDrainSequence); 2753 } 2754} 2755 2756void AudioFlinger::MixerThread::threadLoop_mix() 2757{ 2758 // obtain the presentation timestamp of the next output buffer 2759 int64_t pts; 2760 status_t status = INVALID_OPERATION; 2761 2762 if (mNormalSink != 0) { 2763 status = mNormalSink->getNextWriteTimestamp(&pts); 2764 } else { 2765 status = mOutputSink->getNextWriteTimestamp(&pts); 2766 } 2767 2768 if (status != NO_ERROR) { 2769 pts = AudioBufferProvider::kInvalidPTS; 2770 } 2771 2772 // mix buffers... 2773 mAudioMixer->process(pts); 2774 mCurrentWriteLength = mixBufferSize; 2775 // increase sleep time progressively when application underrun condition clears. 2776 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2777 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2778 // such that we would underrun the audio HAL. 2779 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2780 sleepTimeShift--; 2781 } 2782 sleepTime = 0; 2783 standbyTime = systemTime() + standbyDelay; 2784 //TODO: delay standby when effects have a tail 2785} 2786 2787void AudioFlinger::MixerThread::threadLoop_sleepTime() 2788{ 2789 // If no tracks are ready, sleep once for the duration of an output 2790 // buffer size, then write 0s to the output 2791 if (sleepTime == 0) { 2792 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2793 sleepTime = activeSleepTime >> sleepTimeShift; 2794 if (sleepTime < kMinThreadSleepTimeUs) { 2795 sleepTime = kMinThreadSleepTimeUs; 2796 } 2797 // reduce sleep time in case of consecutive application underruns to avoid 2798 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2799 // duration we would end up writing less data than needed by the audio HAL if 2800 // the condition persists. 2801 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2802 sleepTimeShift++; 2803 } 2804 } else { 2805 sleepTime = idleSleepTime; 2806 } 2807 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2808 memset(mMixBuffer, 0, mixBufferSize); 2809 sleepTime = 0; 2810 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2811 "anticipated start"); 2812 } 2813 // TODO add standby time extension fct of effect tail 2814} 2815 2816// prepareTracks_l() must be called with ThreadBase::mLock held 2817AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2818 Vector< sp<Track> > *tracksToRemove) 2819{ 2820 2821 mixer_state mixerStatus = MIXER_IDLE; 2822 // find out which tracks need to be processed 2823 size_t count = mActiveTracks.size(); 2824 size_t mixedTracks = 0; 2825 size_t tracksWithEffect = 0; 2826 // counts only _active_ fast tracks 2827 size_t fastTracks = 0; 2828 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2829 2830 float masterVolume = mMasterVolume; 2831 bool masterMute = mMasterMute; 2832 2833 if (masterMute) { 2834 masterVolume = 0; 2835 } 2836 // Delegate master volume control to effect in output mix effect chain if needed 2837 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2838 if (chain != 0) { 2839 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2840 chain->setVolume_l(&v, &v); 2841 masterVolume = (float)((v + (1 << 23)) >> 24); 2842 chain.clear(); 2843 } 2844 2845 // prepare a new state to push 2846 FastMixerStateQueue *sq = NULL; 2847 FastMixerState *state = NULL; 2848 bool didModify = false; 2849 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2850 if (mFastMixer != NULL) { 2851 sq = mFastMixer->sq(); 2852 state = sq->begin(); 2853 } 2854 2855 for (size_t i=0 ; i<count ; i++) { 2856 const sp<Track> t = mActiveTracks[i].promote(); 2857 if (t == 0) { 2858 continue; 2859 } 2860 2861 // this const just means the local variable doesn't change 2862 Track* const track = t.get(); 2863 2864 // process fast tracks 2865 if (track->isFastTrack()) { 2866 2867 // It's theoretically possible (though unlikely) for a fast track to be created 2868 // and then removed within the same normal mix cycle. This is not a problem, as 2869 // the track never becomes active so it's fast mixer slot is never touched. 2870 // The converse, of removing an (active) track and then creating a new track 2871 // at the identical fast mixer slot within the same normal mix cycle, 2872 // is impossible because the slot isn't marked available until the end of each cycle. 2873 int j = track->mFastIndex; 2874 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2875 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2876 FastTrack *fastTrack = &state->mFastTracks[j]; 2877 2878 // Determine whether the track is currently in underrun condition, 2879 // and whether it had a recent underrun. 2880 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2881 FastTrackUnderruns underruns = ftDump->mUnderruns; 2882 uint32_t recentFull = (underruns.mBitFields.mFull - 2883 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2884 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2885 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2886 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2887 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2888 uint32_t recentUnderruns = recentPartial + recentEmpty; 2889 track->mObservedUnderruns = underruns; 2890 // don't count underruns that occur while stopping or pausing 2891 // or stopped which can occur when flush() is called while active 2892 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2893 recentUnderruns > 0) { 2894 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2895 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2896 } 2897 2898 // This is similar to the state machine for normal tracks, 2899 // with a few modifications for fast tracks. 2900 bool isActive = true; 2901 switch (track->mState) { 2902 case TrackBase::STOPPING_1: 2903 // track stays active in STOPPING_1 state until first underrun 2904 if (recentUnderruns > 0 || track->isTerminated()) { 2905 track->mState = TrackBase::STOPPING_2; 2906 } 2907 break; 2908 case TrackBase::PAUSING: 2909 // ramp down is not yet implemented 2910 track->setPaused(); 2911 break; 2912 case TrackBase::RESUMING: 2913 // ramp up is not yet implemented 2914 track->mState = TrackBase::ACTIVE; 2915 break; 2916 case TrackBase::ACTIVE: 2917 if (recentFull > 0 || recentPartial > 0) { 2918 // track has provided at least some frames recently: reset retry count 2919 track->mRetryCount = kMaxTrackRetries; 2920 } 2921 if (recentUnderruns == 0) { 2922 // no recent underruns: stay active 2923 break; 2924 } 2925 // there has recently been an underrun of some kind 2926 if (track->sharedBuffer() == 0) { 2927 // were any of the recent underruns "empty" (no frames available)? 2928 if (recentEmpty == 0) { 2929 // no, then ignore the partial underruns as they are allowed indefinitely 2930 break; 2931 } 2932 // there has recently been an "empty" underrun: decrement the retry counter 2933 if (--(track->mRetryCount) > 0) { 2934 break; 2935 } 2936 // indicate to client process that the track was disabled because of underrun; 2937 // it will then automatically call start() when data is available 2938 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2939 // remove from active list, but state remains ACTIVE [confusing but true] 2940 isActive = false; 2941 break; 2942 } 2943 // fall through 2944 case TrackBase::STOPPING_2: 2945 case TrackBase::PAUSED: 2946 case TrackBase::STOPPED: 2947 case TrackBase::FLUSHED: // flush() while active 2948 // Check for presentation complete if track is inactive 2949 // We have consumed all the buffers of this track. 2950 // This would be incomplete if we auto-paused on underrun 2951 { 2952 size_t audioHALFrames = 2953 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2954 size_t framesWritten = mBytesWritten / mFrameSize; 2955 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2956 // track stays in active list until presentation is complete 2957 break; 2958 } 2959 } 2960 if (track->isStopping_2()) { 2961 track->mState = TrackBase::STOPPED; 2962 } 2963 if (track->isStopped()) { 2964 // Can't reset directly, as fast mixer is still polling this track 2965 // track->reset(); 2966 // So instead mark this track as needing to be reset after push with ack 2967 resetMask |= 1 << i; 2968 } 2969 isActive = false; 2970 break; 2971 case TrackBase::IDLE: 2972 default: 2973 LOG_FATAL("unexpected track state %d", track->mState); 2974 } 2975 2976 if (isActive) { 2977 // was it previously inactive? 2978 if (!(state->mTrackMask & (1 << j))) { 2979 ExtendedAudioBufferProvider *eabp = track; 2980 VolumeProvider *vp = track; 2981 fastTrack->mBufferProvider = eabp; 2982 fastTrack->mVolumeProvider = vp; 2983 fastTrack->mSampleRate = track->mSampleRate; 2984 fastTrack->mChannelMask = track->mChannelMask; 2985 fastTrack->mGeneration++; 2986 state->mTrackMask |= 1 << j; 2987 didModify = true; 2988 // no acknowledgement required for newly active tracks 2989 } 2990 // cache the combined master volume and stream type volume for fast mixer; this 2991 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2992 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2993 ++fastTracks; 2994 } else { 2995 // was it previously active? 2996 if (state->mTrackMask & (1 << j)) { 2997 fastTrack->mBufferProvider = NULL; 2998 fastTrack->mGeneration++; 2999 state->mTrackMask &= ~(1 << j); 3000 didModify = true; 3001 // If any fast tracks were removed, we must wait for acknowledgement 3002 // because we're about to decrement the last sp<> on those tracks. 3003 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3004 } else { 3005 LOG_FATAL("fast track %d should have been active", j); 3006 } 3007 tracksToRemove->add(track); 3008 // Avoids a misleading display in dumpsys 3009 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3010 } 3011 continue; 3012 } 3013 3014 { // local variable scope to avoid goto warning 3015 3016 audio_track_cblk_t* cblk = track->cblk(); 3017 3018 // The first time a track is added we wait 3019 // for all its buffers to be filled before processing it 3020 int name = track->name(); 3021 // make sure that we have enough frames to mix one full buffer. 3022 // enforce this condition only once to enable draining the buffer in case the client 3023 // app does not call stop() and relies on underrun to stop: 3024 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3025 // during last round 3026 size_t desiredFrames; 3027 uint32_t sr = track->sampleRate(); 3028 if (sr == mSampleRate) { 3029 desiredFrames = mNormalFrameCount; 3030 } else { 3031 // +1 for rounding and +1 for additional sample needed for interpolation 3032 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3033 // add frames already consumed but not yet released by the resampler 3034 // because mAudioTrackServerProxy->framesReady() will include these frames 3035 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3036 // the minimum track buffer size is normally twice the number of frames necessary 3037 // to fill one buffer and the resampler should not leave more than one buffer worth 3038 // of unreleased frames after each pass, but just in case... 3039 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3040 } 3041 uint32_t minFrames = 1; 3042 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3043 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3044 minFrames = desiredFrames; 3045 } 3046 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 3047 size_t framesReady; 3048 if (track->sharedBuffer() == 0) { 3049 framesReady = track->framesReady(); 3050 } else if (track->isStopped()) { 3051 framesReady = 0; 3052 } else { 3053 framesReady = 1; 3054 } 3055 if ((framesReady >= minFrames) && track->isReady() && 3056 !track->isPaused() && !track->isTerminated()) 3057 { 3058 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3059 3060 mixedTracks++; 3061 3062 // track->mainBuffer() != mMixBuffer means there is an effect chain 3063 // connected to the track 3064 chain.clear(); 3065 if (track->mainBuffer() != mMixBuffer) { 3066 chain = getEffectChain_l(track->sessionId()); 3067 // Delegate volume control to effect in track effect chain if needed 3068 if (chain != 0) { 3069 tracksWithEffect++; 3070 } else { 3071 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3072 "session %d", 3073 name, track->sessionId()); 3074 } 3075 } 3076 3077 3078 int param = AudioMixer::VOLUME; 3079 if (track->mFillingUpStatus == Track::FS_FILLED) { 3080 // no ramp for the first volume setting 3081 track->mFillingUpStatus = Track::FS_ACTIVE; 3082 if (track->mState == TrackBase::RESUMING) { 3083 track->mState = TrackBase::ACTIVE; 3084 param = AudioMixer::RAMP_VOLUME; 3085 } 3086 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3087 // FIXME should not make a decision based on mServer 3088 } else if (cblk->mServer != 0) { 3089 // If the track is stopped before the first frame was mixed, 3090 // do not apply ramp 3091 param = AudioMixer::RAMP_VOLUME; 3092 } 3093 3094 // compute volume for this track 3095 uint32_t vl, vr, va; 3096 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3097 vl = vr = va = 0; 3098 if (track->isPausing()) { 3099 track->setPaused(); 3100 } 3101 } else { 3102 3103 // read original volumes with volume control 3104 float typeVolume = mStreamTypes[track->streamType()].volume; 3105 float v = masterVolume * typeVolume; 3106 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3107 uint32_t vlr = proxy->getVolumeLR(); 3108 vl = vlr & 0xFFFF; 3109 vr = vlr >> 16; 3110 // track volumes come from shared memory, so can't be trusted and must be clamped 3111 if (vl > MAX_GAIN_INT) { 3112 ALOGV("Track left volume out of range: %04X", vl); 3113 vl = MAX_GAIN_INT; 3114 } 3115 if (vr > MAX_GAIN_INT) { 3116 ALOGV("Track right volume out of range: %04X", vr); 3117 vr = MAX_GAIN_INT; 3118 } 3119 // now apply the master volume and stream type volume 3120 vl = (uint32_t)(v * vl) << 12; 3121 vr = (uint32_t)(v * vr) << 12; 3122 // assuming master volume and stream type volume each go up to 1.0, 3123 // vl and vr are now in 8.24 format 3124 3125 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3126 // send level comes from shared memory and so may be corrupt 3127 if (sendLevel > MAX_GAIN_INT) { 3128 ALOGV("Track send level out of range: %04X", sendLevel); 3129 sendLevel = MAX_GAIN_INT; 3130 } 3131 va = (uint32_t)(v * sendLevel); 3132 } 3133 3134 // Delegate volume control to effect in track effect chain if needed 3135 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3136 // Do not ramp volume if volume is controlled by effect 3137 param = AudioMixer::VOLUME; 3138 track->mHasVolumeController = true; 3139 } else { 3140 // force no volume ramp when volume controller was just disabled or removed 3141 // from effect chain to avoid volume spike 3142 if (track->mHasVolumeController) { 3143 param = AudioMixer::VOLUME; 3144 } 3145 track->mHasVolumeController = false; 3146 } 3147 3148 // Convert volumes from 8.24 to 4.12 format 3149 // This additional clamping is needed in case chain->setVolume_l() overshot 3150 vl = (vl + (1 << 11)) >> 12; 3151 if (vl > MAX_GAIN_INT) { 3152 vl = MAX_GAIN_INT; 3153 } 3154 vr = (vr + (1 << 11)) >> 12; 3155 if (vr > MAX_GAIN_INT) { 3156 vr = MAX_GAIN_INT; 3157 } 3158 3159 if (va > MAX_GAIN_INT) { 3160 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3161 } 3162 3163 // XXX: these things DON'T need to be done each time 3164 mAudioMixer->setBufferProvider(name, track); 3165 mAudioMixer->enable(name); 3166 3167 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3168 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3169 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3170 mAudioMixer->setParameter( 3171 name, 3172 AudioMixer::TRACK, 3173 AudioMixer::FORMAT, (void *)track->format()); 3174 mAudioMixer->setParameter( 3175 name, 3176 AudioMixer::TRACK, 3177 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3178 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3179 uint32_t maxSampleRate = mSampleRate * 2; 3180 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3181 if (reqSampleRate == 0) { 3182 reqSampleRate = mSampleRate; 3183 } else if (reqSampleRate > maxSampleRate) { 3184 reqSampleRate = maxSampleRate; 3185 } 3186 mAudioMixer->setParameter( 3187 name, 3188 AudioMixer::RESAMPLE, 3189 AudioMixer::SAMPLE_RATE, 3190 (void *)reqSampleRate); 3191 mAudioMixer->setParameter( 3192 name, 3193 AudioMixer::TRACK, 3194 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3195 mAudioMixer->setParameter( 3196 name, 3197 AudioMixer::TRACK, 3198 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3199 3200 // reset retry count 3201 track->mRetryCount = kMaxTrackRetries; 3202 3203 // If one track is ready, set the mixer ready if: 3204 // - the mixer was not ready during previous round OR 3205 // - no other track is not ready 3206 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3207 mixerStatus != MIXER_TRACKS_ENABLED) { 3208 mixerStatus = MIXER_TRACKS_READY; 3209 } 3210 } else { 3211 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3212 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3213 } 3214 // clear effect chain input buffer if an active track underruns to avoid sending 3215 // previous audio buffer again to effects 3216 chain = getEffectChain_l(track->sessionId()); 3217 if (chain != 0) { 3218 chain->clearInputBuffer(); 3219 } 3220 3221 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3222 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3223 track->isStopped() || track->isPaused()) { 3224 // We have consumed all the buffers of this track. 3225 // Remove it from the list of active tracks. 3226 // TODO: use actual buffer filling status instead of latency when available from 3227 // audio HAL 3228 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3229 size_t framesWritten = mBytesWritten / mFrameSize; 3230 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3231 if (track->isStopped()) { 3232 track->reset(); 3233 } 3234 tracksToRemove->add(track); 3235 } 3236 } else { 3237 // No buffers for this track. Give it a few chances to 3238 // fill a buffer, then remove it from active list. 3239 if (--(track->mRetryCount) <= 0) { 3240 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3241 tracksToRemove->add(track); 3242 // indicate to client process that the track was disabled because of underrun; 3243 // it will then automatically call start() when data is available 3244 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3245 // If one track is not ready, mark the mixer also not ready if: 3246 // - the mixer was ready during previous round OR 3247 // - no other track is ready 3248 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3249 mixerStatus != MIXER_TRACKS_READY) { 3250 mixerStatus = MIXER_TRACKS_ENABLED; 3251 } 3252 } 3253 mAudioMixer->disable(name); 3254 } 3255 3256 } // local variable scope to avoid goto warning 3257track_is_ready: ; 3258 3259 } 3260 3261 // Push the new FastMixer state if necessary 3262 bool pauseAudioWatchdog = false; 3263 if (didModify) { 3264 state->mFastTracksGen++; 3265 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3266 if (kUseFastMixer == FastMixer_Dynamic && 3267 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3268 state->mCommand = FastMixerState::COLD_IDLE; 3269 state->mColdFutexAddr = &mFastMixerFutex; 3270 state->mColdGen++; 3271 mFastMixerFutex = 0; 3272 if (kUseFastMixer == FastMixer_Dynamic) { 3273 mNormalSink = mOutputSink; 3274 } 3275 // If we go into cold idle, need to wait for acknowledgement 3276 // so that fast mixer stops doing I/O. 3277 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3278 pauseAudioWatchdog = true; 3279 } 3280 } 3281 if (sq != NULL) { 3282 sq->end(didModify); 3283 sq->push(block); 3284 } 3285#ifdef AUDIO_WATCHDOG 3286 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3287 mAudioWatchdog->pause(); 3288 } 3289#endif 3290 3291 // Now perform the deferred reset on fast tracks that have stopped 3292 while (resetMask != 0) { 3293 size_t i = __builtin_ctz(resetMask); 3294 ALOG_ASSERT(i < count); 3295 resetMask &= ~(1 << i); 3296 sp<Track> t = mActiveTracks[i].promote(); 3297 if (t == 0) { 3298 continue; 3299 } 3300 Track* track = t.get(); 3301 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3302 track->reset(); 3303 } 3304 3305 // remove all the tracks that need to be... 3306 removeTracks_l(*tracksToRemove); 3307 3308 // mix buffer must be cleared if all tracks are connected to an 3309 // effect chain as in this case the mixer will not write to 3310 // mix buffer and track effects will accumulate into it 3311 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3312 (mixedTracks == 0 && fastTracks > 0))) { 3313 // FIXME as a performance optimization, should remember previous zero status 3314 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3315 } 3316 3317 // if any fast tracks, then status is ready 3318 mMixerStatusIgnoringFastTracks = mixerStatus; 3319 if (fastTracks > 0) { 3320 mixerStatus = MIXER_TRACKS_READY; 3321 } 3322 return mixerStatus; 3323} 3324 3325// getTrackName_l() must be called with ThreadBase::mLock held 3326int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3327{ 3328 return mAudioMixer->getTrackName(channelMask, sessionId); 3329} 3330 3331// deleteTrackName_l() must be called with ThreadBase::mLock held 3332void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3333{ 3334 ALOGV("remove track (%d) and delete from mixer", name); 3335 mAudioMixer->deleteTrackName(name); 3336} 3337 3338// checkForNewParameters_l() must be called with ThreadBase::mLock held 3339bool AudioFlinger::MixerThread::checkForNewParameters_l() 3340{ 3341 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3342 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3343 bool reconfig = false; 3344 3345 while (!mNewParameters.isEmpty()) { 3346 3347 if (mFastMixer != NULL) { 3348 FastMixerStateQueue *sq = mFastMixer->sq(); 3349 FastMixerState *state = sq->begin(); 3350 if (!(state->mCommand & FastMixerState::IDLE)) { 3351 previousCommand = state->mCommand; 3352 state->mCommand = FastMixerState::HOT_IDLE; 3353 sq->end(); 3354 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3355 } else { 3356 sq->end(false /*didModify*/); 3357 } 3358 } 3359 3360 status_t status = NO_ERROR; 3361 String8 keyValuePair = mNewParameters[0]; 3362 AudioParameter param = AudioParameter(keyValuePair); 3363 int value; 3364 3365 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3366 reconfig = true; 3367 } 3368 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3369 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3370 status = BAD_VALUE; 3371 } else { 3372 // no need to save value, since it's constant 3373 reconfig = true; 3374 } 3375 } 3376 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3377 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3378 status = BAD_VALUE; 3379 } else { 3380 // no need to save value, since it's constant 3381 reconfig = true; 3382 } 3383 } 3384 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3385 // do not accept frame count changes if tracks are open as the track buffer 3386 // size depends on frame count and correct behavior would not be guaranteed 3387 // if frame count is changed after track creation 3388 if (!mTracks.isEmpty()) { 3389 status = INVALID_OPERATION; 3390 } else { 3391 reconfig = true; 3392 } 3393 } 3394 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3395#ifdef ADD_BATTERY_DATA 3396 // when changing the audio output device, call addBatteryData to notify 3397 // the change 3398 if (mOutDevice != value) { 3399 uint32_t params = 0; 3400 // check whether speaker is on 3401 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3402 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3403 } 3404 3405 audio_devices_t deviceWithoutSpeaker 3406 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3407 // check if any other device (except speaker) is on 3408 if (value & deviceWithoutSpeaker ) { 3409 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3410 } 3411 3412 if (params != 0) { 3413 addBatteryData(params); 3414 } 3415 } 3416#endif 3417 3418 // forward device change to effects that have requested to be 3419 // aware of attached audio device. 3420 if (value != AUDIO_DEVICE_NONE) { 3421 mOutDevice = value; 3422 for (size_t i = 0; i < mEffectChains.size(); i++) { 3423 mEffectChains[i]->setDevice_l(mOutDevice); 3424 } 3425 } 3426 } 3427 3428 if (status == NO_ERROR) { 3429 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3430 keyValuePair.string()); 3431 if (!mStandby && status == INVALID_OPERATION) { 3432 mOutput->stream->common.standby(&mOutput->stream->common); 3433 mStandby = true; 3434 mBytesWritten = 0; 3435 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3436 keyValuePair.string()); 3437 } 3438 if (status == NO_ERROR && reconfig) { 3439 readOutputParameters(); 3440 delete mAudioMixer; 3441 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3442 for (size_t i = 0; i < mTracks.size() ; i++) { 3443 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3444 if (name < 0) { 3445 break; 3446 } 3447 mTracks[i]->mName = name; 3448 } 3449 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3450 } 3451 } 3452 3453 mNewParameters.removeAt(0); 3454 3455 mParamStatus = status; 3456 mParamCond.signal(); 3457 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3458 // already timed out waiting for the status and will never signal the condition. 3459 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3460 } 3461 3462 if (!(previousCommand & FastMixerState::IDLE)) { 3463 ALOG_ASSERT(mFastMixer != NULL); 3464 FastMixerStateQueue *sq = mFastMixer->sq(); 3465 FastMixerState *state = sq->begin(); 3466 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3467 state->mCommand = previousCommand; 3468 sq->end(); 3469 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3470 } 3471 3472 return reconfig; 3473} 3474 3475 3476void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3477{ 3478 const size_t SIZE = 256; 3479 char buffer[SIZE]; 3480 String8 result; 3481 3482 PlaybackThread::dumpInternals(fd, args); 3483 3484 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3485 result.append(buffer); 3486 write(fd, result.string(), result.size()); 3487 3488 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3489 const FastMixerDumpState copy(mFastMixerDumpState); 3490 copy.dump(fd); 3491 3492#ifdef STATE_QUEUE_DUMP 3493 // Similar for state queue 3494 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3495 observerCopy.dump(fd); 3496 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3497 mutatorCopy.dump(fd); 3498#endif 3499 3500#ifdef TEE_SINK 3501 // Write the tee output to a .wav file 3502 dumpTee(fd, mTeeSource, mId); 3503#endif 3504 3505#ifdef AUDIO_WATCHDOG 3506 if (mAudioWatchdog != 0) { 3507 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3508 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3509 wdCopy.dump(fd); 3510 } 3511#endif 3512} 3513 3514uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3515{ 3516 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3517} 3518 3519uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3520{ 3521 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3522} 3523 3524void AudioFlinger::MixerThread::cacheParameters_l() 3525{ 3526 PlaybackThread::cacheParameters_l(); 3527 3528 // FIXME: Relaxed timing because of a certain device that can't meet latency 3529 // Should be reduced to 2x after the vendor fixes the driver issue 3530 // increase threshold again due to low power audio mode. The way this warning 3531 // threshold is calculated and its usefulness should be reconsidered anyway. 3532 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3533} 3534 3535// ---------------------------------------------------------------------------- 3536 3537AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3538 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3539 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3540 // mLeftVolFloat, mRightVolFloat 3541{ 3542} 3543 3544AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3545 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3546 ThreadBase::type_t type) 3547 : PlaybackThread(audioFlinger, output, id, device, type) 3548 // mLeftVolFloat, mRightVolFloat 3549{ 3550} 3551 3552AudioFlinger::DirectOutputThread::~DirectOutputThread() 3553{ 3554} 3555 3556void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3557{ 3558 audio_track_cblk_t* cblk = track->cblk(); 3559 float left, right; 3560 3561 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3562 left = right = 0; 3563 } else { 3564 float typeVolume = mStreamTypes[track->streamType()].volume; 3565 float v = mMasterVolume * typeVolume; 3566 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3567 uint32_t vlr = proxy->getVolumeLR(); 3568 float v_clamped = v * (vlr & 0xFFFF); 3569 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3570 left = v_clamped/MAX_GAIN; 3571 v_clamped = v * (vlr >> 16); 3572 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3573 right = v_clamped/MAX_GAIN; 3574 } 3575 3576 if (lastTrack) { 3577 if (left != mLeftVolFloat || right != mRightVolFloat) { 3578 mLeftVolFloat = left; 3579 mRightVolFloat = right; 3580 3581 // Convert volumes from float to 8.24 3582 uint32_t vl = (uint32_t)(left * (1 << 24)); 3583 uint32_t vr = (uint32_t)(right * (1 << 24)); 3584 3585 // Delegate volume control to effect in track effect chain if needed 3586 // only one effect chain can be present on DirectOutputThread, so if 3587 // there is one, the track is connected to it 3588 if (!mEffectChains.isEmpty()) { 3589 mEffectChains[0]->setVolume_l(&vl, &vr); 3590 left = (float)vl / (1 << 24); 3591 right = (float)vr / (1 << 24); 3592 } 3593 if (mOutput->stream->set_volume) { 3594 mOutput->stream->set_volume(mOutput->stream, left, right); 3595 } 3596 } 3597 } 3598} 3599 3600 3601AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3602 Vector< sp<Track> > *tracksToRemove 3603) 3604{ 3605 size_t count = mActiveTracks.size(); 3606 mixer_state mixerStatus = MIXER_IDLE; 3607 3608 // find out which tracks need to be processed 3609 for (size_t i = 0; i < count; i++) { 3610 sp<Track> t = mActiveTracks[i].promote(); 3611 // The track died recently 3612 if (t == 0) { 3613 continue; 3614 } 3615 3616 Track* const track = t.get(); 3617 audio_track_cblk_t* cblk = track->cblk(); 3618 3619 // The first time a track is added we wait 3620 // for all its buffers to be filled before processing it 3621 uint32_t minFrames; 3622 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3623 minFrames = mNormalFrameCount; 3624 } else { 3625 minFrames = 1; 3626 } 3627 // Only consider last track started for volume and mixer state control. 3628 // This is the last entry in mActiveTracks unless a track underruns. 3629 // As we only care about the transition phase between two tracks on a 3630 // direct output, it is not a problem to ignore the underrun case. 3631 bool last = (i == (count - 1)); 3632 3633 if ((track->framesReady() >= minFrames) && track->isReady() && 3634 !track->isPaused() && !track->isTerminated()) 3635 { 3636 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3637 3638 if (track->mFillingUpStatus == Track::FS_FILLED) { 3639 track->mFillingUpStatus = Track::FS_ACTIVE; 3640 // make sure processVolume_l() will apply new volume even if 0 3641 mLeftVolFloat = mRightVolFloat = -1.0; 3642 if (track->mState == TrackBase::RESUMING) { 3643 track->mState = TrackBase::ACTIVE; 3644 } 3645 } 3646 3647 // compute volume for this track 3648 processVolume_l(track, last); 3649 if (last) { 3650 // reset retry count 3651 track->mRetryCount = kMaxTrackRetriesDirect; 3652 mActiveTrack = t; 3653 mixerStatus = MIXER_TRACKS_READY; 3654 } 3655 } else { 3656 // clear effect chain input buffer if the last active track started underruns 3657 // to avoid sending previous audio buffer again to effects 3658 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3659 mEffectChains[0]->clearInputBuffer(); 3660 } 3661 3662 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3663 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3664 track->isStopped() || track->isPaused()) { 3665 // We have consumed all the buffers of this track. 3666 // Remove it from the list of active tracks. 3667 // TODO: implement behavior for compressed audio 3668 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3669 size_t framesWritten = mBytesWritten / mFrameSize; 3670 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3671 if (track->isStopped()) { 3672 track->reset(); 3673 } 3674 tracksToRemove->add(track); 3675 } 3676 } else { 3677 // No buffers for this track. Give it a few chances to 3678 // fill a buffer, then remove it from active list. 3679 // Only consider last track started for mixer state control 3680 if (--(track->mRetryCount) <= 0) { 3681 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3682 tracksToRemove->add(track); 3683 } else if (last) { 3684 mixerStatus = MIXER_TRACKS_ENABLED; 3685 } 3686 } 3687 } 3688 } 3689 3690 // remove all the tracks that need to be... 3691 removeTracks_l(*tracksToRemove); 3692 3693 return mixerStatus; 3694} 3695 3696void AudioFlinger::DirectOutputThread::threadLoop_mix() 3697{ 3698 size_t frameCount = mFrameCount; 3699 int8_t *curBuf = (int8_t *)mMixBuffer; 3700 // output audio to hardware 3701 while (frameCount) { 3702 AudioBufferProvider::Buffer buffer; 3703 buffer.frameCount = frameCount; 3704 mActiveTrack->getNextBuffer(&buffer); 3705 if (buffer.raw == NULL) { 3706 memset(curBuf, 0, frameCount * mFrameSize); 3707 break; 3708 } 3709 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3710 frameCount -= buffer.frameCount; 3711 curBuf += buffer.frameCount * mFrameSize; 3712 mActiveTrack->releaseBuffer(&buffer); 3713 } 3714 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3715 sleepTime = 0; 3716 standbyTime = systemTime() + standbyDelay; 3717 mActiveTrack.clear(); 3718} 3719 3720void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3721{ 3722 if (sleepTime == 0) { 3723 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3724 sleepTime = activeSleepTime; 3725 } else { 3726 sleepTime = idleSleepTime; 3727 } 3728 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3729 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3730 sleepTime = 0; 3731 } 3732} 3733 3734// getTrackName_l() must be called with ThreadBase::mLock held 3735int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3736 int sessionId) 3737{ 3738 return 0; 3739} 3740 3741// deleteTrackName_l() must be called with ThreadBase::mLock held 3742void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3743{ 3744} 3745 3746// checkForNewParameters_l() must be called with ThreadBase::mLock held 3747bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3748{ 3749 bool reconfig = false; 3750 3751 while (!mNewParameters.isEmpty()) { 3752 status_t status = NO_ERROR; 3753 String8 keyValuePair = mNewParameters[0]; 3754 AudioParameter param = AudioParameter(keyValuePair); 3755 int value; 3756 3757 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3758 // do not accept frame count changes if tracks are open as the track buffer 3759 // size depends on frame count and correct behavior would not be garantied 3760 // if frame count is changed after track creation 3761 if (!mTracks.isEmpty()) { 3762 status = INVALID_OPERATION; 3763 } else { 3764 reconfig = true; 3765 } 3766 } 3767 if (status == NO_ERROR) { 3768 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3769 keyValuePair.string()); 3770 if (!mStandby && status == INVALID_OPERATION) { 3771 mOutput->stream->common.standby(&mOutput->stream->common); 3772 mStandby = true; 3773 mBytesWritten = 0; 3774 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3775 keyValuePair.string()); 3776 } 3777 if (status == NO_ERROR && reconfig) { 3778 readOutputParameters(); 3779 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3780 } 3781 } 3782 3783 mNewParameters.removeAt(0); 3784 3785 mParamStatus = status; 3786 mParamCond.signal(); 3787 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3788 // already timed out waiting for the status and will never signal the condition. 3789 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3790 } 3791 return reconfig; 3792} 3793 3794uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3795{ 3796 uint32_t time; 3797 if (audio_is_linear_pcm(mFormat)) { 3798 time = PlaybackThread::activeSleepTimeUs(); 3799 } else { 3800 time = 10000; 3801 } 3802 return time; 3803} 3804 3805uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3806{ 3807 uint32_t time; 3808 if (audio_is_linear_pcm(mFormat)) { 3809 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3810 } else { 3811 time = 10000; 3812 } 3813 return time; 3814} 3815 3816uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3817{ 3818 uint32_t time; 3819 if (audio_is_linear_pcm(mFormat)) { 3820 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3821 } else { 3822 time = 10000; 3823 } 3824 return time; 3825} 3826 3827void AudioFlinger::DirectOutputThread::cacheParameters_l() 3828{ 3829 PlaybackThread::cacheParameters_l(); 3830 3831 // use shorter standby delay as on normal output to release 3832 // hardware resources as soon as possible 3833 if (audio_is_linear_pcm(mFormat)) { 3834 standbyDelay = microseconds(activeSleepTime*2); 3835 } else { 3836 standbyDelay = kOffloadStandbyDelayNs; 3837 } 3838} 3839 3840// ---------------------------------------------------------------------------- 3841 3842AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3843 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3844 : Thread(false /*canCallJava*/), 3845 mPlaybackThread(playbackThread), 3846 mWriteAckSequence(0), 3847 mDrainSequence(0) 3848{ 3849} 3850 3851AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3852{ 3853} 3854 3855void AudioFlinger::AsyncCallbackThread::onFirstRef() 3856{ 3857 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3858} 3859 3860bool AudioFlinger::AsyncCallbackThread::threadLoop() 3861{ 3862 while (!exitPending()) { 3863 uint32_t writeAckSequence; 3864 uint32_t drainSequence; 3865 3866 { 3867 Mutex::Autolock _l(mLock); 3868 mWaitWorkCV.wait(mLock); 3869 if (exitPending()) { 3870 break; 3871 } 3872 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3873 mWriteAckSequence, mDrainSequence); 3874 writeAckSequence = mWriteAckSequence; 3875 mWriteAckSequence &= ~1; 3876 drainSequence = mDrainSequence; 3877 mDrainSequence &= ~1; 3878 } 3879 { 3880 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3881 if (playbackThread != 0) { 3882 if (writeAckSequence & 1) { 3883 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3884 } 3885 if (drainSequence & 1) { 3886 playbackThread->resetDraining(drainSequence >> 1); 3887 } 3888 } 3889 } 3890 } 3891 return false; 3892} 3893 3894void AudioFlinger::AsyncCallbackThread::exit() 3895{ 3896 ALOGV("AsyncCallbackThread::exit"); 3897 Mutex::Autolock _l(mLock); 3898 requestExit(); 3899 mWaitWorkCV.broadcast(); 3900} 3901 3902void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3903{ 3904 Mutex::Autolock _l(mLock); 3905 // bit 0 is cleared 3906 mWriteAckSequence = sequence << 1; 3907} 3908 3909void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3910{ 3911 Mutex::Autolock _l(mLock); 3912 // ignore unexpected callbacks 3913 if (mWriteAckSequence & 2) { 3914 mWriteAckSequence |= 1; 3915 mWaitWorkCV.signal(); 3916 } 3917} 3918 3919void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3920{ 3921 Mutex::Autolock _l(mLock); 3922 // bit 0 is cleared 3923 mDrainSequence = sequence << 1; 3924} 3925 3926void AudioFlinger::AsyncCallbackThread::resetDraining() 3927{ 3928 Mutex::Autolock _l(mLock); 3929 // ignore unexpected callbacks 3930 if (mDrainSequence & 2) { 3931 mDrainSequence |= 1; 3932 mWaitWorkCV.signal(); 3933 } 3934} 3935 3936 3937// ---------------------------------------------------------------------------- 3938AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3939 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3940 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3941 mHwPaused(false), 3942 mFlushPending(false), 3943 mPausedBytesRemaining(0), 3944 mPreviousTrack(NULL) 3945{ 3946} 3947 3948void AudioFlinger::OffloadThread::threadLoop_exit() 3949{ 3950 if (mFlushPending || mHwPaused) { 3951 // If a flush is pending or track was paused, just discard buffered data 3952 flushHw_l(); 3953 } else { 3954 mMixerStatus = MIXER_DRAIN_ALL; 3955 threadLoop_drain(); 3956 } 3957 mCallbackThread->exit(); 3958 PlaybackThread::threadLoop_exit(); 3959} 3960 3961AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3962 Vector< sp<Track> > *tracksToRemove 3963) 3964{ 3965 size_t count = mActiveTracks.size(); 3966 3967 mixer_state mixerStatus = MIXER_IDLE; 3968 bool doHwPause = false; 3969 bool doHwResume = false; 3970 3971 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3972 3973 // find out which tracks need to be processed 3974 for (size_t i = 0; i < count; i++) { 3975 sp<Track> t = mActiveTracks[i].promote(); 3976 // The track died recently 3977 if (t == 0) { 3978 continue; 3979 } 3980 Track* const track = t.get(); 3981 audio_track_cblk_t* cblk = track->cblk(); 3982 if (mPreviousTrack != NULL) { 3983 if (t.get() != mPreviousTrack) { 3984 // Flush any data still being written from last track 3985 mBytesRemaining = 0; 3986 if (mPausedBytesRemaining) { 3987 // Last track was paused so we also need to flush saved 3988 // mixbuffer state and invalidate track so that it will 3989 // re-submit that unwritten data when it is next resumed 3990 mPausedBytesRemaining = 0; 3991 // Invalidate is a bit drastic - would be more efficient 3992 // to have a flag to tell client that some of the 3993 // previously written data was lost 3994 mPreviousTrack->invalidate(); 3995 } 3996 } 3997 } 3998 mPreviousTrack = t.get(); 3999 bool last = (i == (count - 1)); 4000 if (track->isPausing()) { 4001 track->setPaused(); 4002 if (last) { 4003 if (!mHwPaused) { 4004 doHwPause = true; 4005 mHwPaused = true; 4006 } 4007 // If we were part way through writing the mixbuffer to 4008 // the HAL we must save this until we resume 4009 // BUG - this will be wrong if a different track is made active, 4010 // in that case we want to discard the pending data in the 4011 // mixbuffer and tell the client to present it again when the 4012 // track is resumed 4013 mPausedWriteLength = mCurrentWriteLength; 4014 mPausedBytesRemaining = mBytesRemaining; 4015 mBytesRemaining = 0; // stop writing 4016 } 4017 tracksToRemove->add(track); 4018 } else if (track->framesReady() && track->isReady() && 4019 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4020 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4021 if (track->mFillingUpStatus == Track::FS_FILLED) { 4022 track->mFillingUpStatus = Track::FS_ACTIVE; 4023 // make sure processVolume_l() will apply new volume even if 0 4024 mLeftVolFloat = mRightVolFloat = -1.0; 4025 if (track->mState == TrackBase::RESUMING) { 4026 track->mState = TrackBase::ACTIVE; 4027 if (last) { 4028 if (mPausedBytesRemaining) { 4029 // Need to continue write that was interrupted 4030 mCurrentWriteLength = mPausedWriteLength; 4031 mBytesRemaining = mPausedBytesRemaining; 4032 mPausedBytesRemaining = 0; 4033 } 4034 if (mHwPaused) { 4035 doHwResume = true; 4036 mHwPaused = false; 4037 // threadLoop_mix() will handle the case that we need to 4038 // resume an interrupted write 4039 } 4040 // enable write to audio HAL 4041 sleepTime = 0; 4042 } 4043 } 4044 } 4045 4046 if (last) { 4047 // reset retry count 4048 track->mRetryCount = kMaxTrackRetriesOffload; 4049 mActiveTrack = t; 4050 mixerStatus = MIXER_TRACKS_READY; 4051 } 4052 } else { 4053 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4054 if (track->isStopping_1()) { 4055 // Hardware buffer can hold a large amount of audio so we must 4056 // wait for all current track's data to drain before we say 4057 // that the track is stopped. 4058 if (mBytesRemaining == 0) { 4059 // Only start draining when all data in mixbuffer 4060 // has been written 4061 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4062 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4063 // do not drain if no data was ever sent to HAL (mStandby == true) 4064 if (last && !mStandby) { 4065 sleepTime = 0; 4066 standbyTime = systemTime() + standbyDelay; 4067 mixerStatus = MIXER_DRAIN_TRACK; 4068 mDrainSequence += 2; 4069 if (mHwPaused) { 4070 // It is possible to move from PAUSED to STOPPING_1 without 4071 // a resume so we must ensure hardware is running 4072 mOutput->stream->resume(mOutput->stream); 4073 mHwPaused = false; 4074 } 4075 } 4076 } 4077 } else if (track->isStopping_2()) { 4078 // Drain has completed or we are in standby, signal presentation complete 4079 if (!(mDrainSequence & 1) || !last || mStandby) { 4080 track->mState = TrackBase::STOPPED; 4081 size_t audioHALFrames = 4082 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4083 size_t framesWritten = 4084 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4085 track->presentationComplete(framesWritten, audioHALFrames); 4086 track->reset(); 4087 tracksToRemove->add(track); 4088 } 4089 } else { 4090 // No buffers for this track. Give it a few chances to 4091 // fill a buffer, then remove it from active list. 4092 if (--(track->mRetryCount) <= 0) { 4093 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4094 track->name()); 4095 tracksToRemove->add(track); 4096 } else if (last){ 4097 mixerStatus = MIXER_TRACKS_ENABLED; 4098 } 4099 } 4100 } 4101 // compute volume for this track 4102 processVolume_l(track, last); 4103 } 4104 4105 // make sure the pause/flush/resume sequence is executed in the right order. 4106 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4107 // before flush and then resume HW. This can happen in case of pause/flush/resume 4108 // if resume is received before pause is executed. 4109 if (doHwPause || (mFlushPending && !mHwPaused && (count != 0))) { 4110 mOutput->stream->pause(mOutput->stream); 4111 if (!doHwPause) { 4112 doHwResume = true; 4113 } 4114 } 4115 if (mFlushPending) { 4116 flushHw_l(); 4117 mFlushPending = false; 4118 } 4119 if (doHwResume) { 4120 mOutput->stream->resume(mOutput->stream); 4121 } 4122 4123 // remove all the tracks that need to be... 4124 removeTracks_l(*tracksToRemove); 4125 4126 return mixerStatus; 4127} 4128 4129void AudioFlinger::OffloadThread::flushOutput_l() 4130{ 4131 mFlushPending = true; 4132} 4133 4134// must be called with thread mutex locked 4135bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4136{ 4137 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4138 mWriteAckSequence, mDrainSequence); 4139 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4140 return true; 4141 } 4142 return false; 4143} 4144 4145// must be called with thread mutex locked 4146bool AudioFlinger::OffloadThread::shouldStandby_l() 4147{ 4148 bool TrackPaused = false; 4149 4150 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4151 // after a timeout and we will enter standby then. 4152 if (mTracks.size() > 0) { 4153 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4154 } 4155 4156 return !mStandby && !TrackPaused; 4157} 4158 4159 4160bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4161{ 4162 Mutex::Autolock _l(mLock); 4163 return waitingAsyncCallback_l(); 4164} 4165 4166void AudioFlinger::OffloadThread::flushHw_l() 4167{ 4168 mOutput->stream->flush(mOutput->stream); 4169 // Flush anything still waiting in the mixbuffer 4170 mCurrentWriteLength = 0; 4171 mBytesRemaining = 0; 4172 mPausedWriteLength = 0; 4173 mPausedBytesRemaining = 0; 4174 if (mUseAsyncWrite) { 4175 // discard any pending drain or write ack by incrementing sequence 4176 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4177 mDrainSequence = (mDrainSequence + 2) & ~1; 4178 ALOG_ASSERT(mCallbackThread != 0); 4179 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4180 mCallbackThread->setDraining(mDrainSequence); 4181 } 4182} 4183 4184// ---------------------------------------------------------------------------- 4185 4186AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4187 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4188 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4189 DUPLICATING), 4190 mWaitTimeMs(UINT_MAX) 4191{ 4192 addOutputTrack(mainThread); 4193} 4194 4195AudioFlinger::DuplicatingThread::~DuplicatingThread() 4196{ 4197 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4198 mOutputTracks[i]->destroy(); 4199 } 4200} 4201 4202void AudioFlinger::DuplicatingThread::threadLoop_mix() 4203{ 4204 // mix buffers... 4205 if (outputsReady(outputTracks)) { 4206 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4207 } else { 4208 memset(mMixBuffer, 0, mixBufferSize); 4209 } 4210 sleepTime = 0; 4211 writeFrames = mNormalFrameCount; 4212 mCurrentWriteLength = mixBufferSize; 4213 standbyTime = systemTime() + standbyDelay; 4214} 4215 4216void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4217{ 4218 if (sleepTime == 0) { 4219 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4220 sleepTime = activeSleepTime; 4221 } else { 4222 sleepTime = idleSleepTime; 4223 } 4224 } else if (mBytesWritten != 0) { 4225 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4226 writeFrames = mNormalFrameCount; 4227 memset(mMixBuffer, 0, mixBufferSize); 4228 } else { 4229 // flush remaining overflow buffers in output tracks 4230 writeFrames = 0; 4231 } 4232 sleepTime = 0; 4233 } 4234} 4235 4236ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4237{ 4238 for (size_t i = 0; i < outputTracks.size(); i++) { 4239 outputTracks[i]->write(mMixBuffer, writeFrames); 4240 } 4241 return (ssize_t)mixBufferSize; 4242} 4243 4244void AudioFlinger::DuplicatingThread::threadLoop_standby() 4245{ 4246 // DuplicatingThread implements standby by stopping all tracks 4247 for (size_t i = 0; i < outputTracks.size(); i++) { 4248 outputTracks[i]->stop(); 4249 } 4250} 4251 4252void AudioFlinger::DuplicatingThread::saveOutputTracks() 4253{ 4254 outputTracks = mOutputTracks; 4255} 4256 4257void AudioFlinger::DuplicatingThread::clearOutputTracks() 4258{ 4259 outputTracks.clear(); 4260} 4261 4262void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4263{ 4264 Mutex::Autolock _l(mLock); 4265 // FIXME explain this formula 4266 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4267 OutputTrack *outputTrack = new OutputTrack(thread, 4268 this, 4269 mSampleRate, 4270 mFormat, 4271 mChannelMask, 4272 frameCount, 4273 IPCThreadState::self()->getCallingUid()); 4274 if (outputTrack->cblk() != NULL) { 4275 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4276 mOutputTracks.add(outputTrack); 4277 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4278 updateWaitTime_l(); 4279 } 4280} 4281 4282void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4283{ 4284 Mutex::Autolock _l(mLock); 4285 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4286 if (mOutputTracks[i]->thread() == thread) { 4287 mOutputTracks[i]->destroy(); 4288 mOutputTracks.removeAt(i); 4289 updateWaitTime_l(); 4290 return; 4291 } 4292 } 4293 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4294} 4295 4296// caller must hold mLock 4297void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4298{ 4299 mWaitTimeMs = UINT_MAX; 4300 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4301 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4302 if (strong != 0) { 4303 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4304 if (waitTimeMs < mWaitTimeMs) { 4305 mWaitTimeMs = waitTimeMs; 4306 } 4307 } 4308 } 4309} 4310 4311 4312bool AudioFlinger::DuplicatingThread::outputsReady( 4313 const SortedVector< sp<OutputTrack> > &outputTracks) 4314{ 4315 for (size_t i = 0; i < outputTracks.size(); i++) { 4316 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4317 if (thread == 0) { 4318 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4319 outputTracks[i].get()); 4320 return false; 4321 } 4322 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4323 // see note at standby() declaration 4324 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4325 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4326 thread.get()); 4327 return false; 4328 } 4329 } 4330 return true; 4331} 4332 4333uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4334{ 4335 return (mWaitTimeMs * 1000) / 2; 4336} 4337 4338void AudioFlinger::DuplicatingThread::cacheParameters_l() 4339{ 4340 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4341 updateWaitTime_l(); 4342 4343 MixerThread::cacheParameters_l(); 4344} 4345 4346// ---------------------------------------------------------------------------- 4347// Record 4348// ---------------------------------------------------------------------------- 4349 4350AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4351 AudioStreamIn *input, 4352 uint32_t sampleRate, 4353 audio_channel_mask_t channelMask, 4354 audio_io_handle_t id, 4355 audio_devices_t outDevice, 4356 audio_devices_t inDevice 4357#ifdef TEE_SINK 4358 , const sp<NBAIO_Sink>& teeSink 4359#endif 4360 ) : 4361 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4362 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4363 // mRsmpInIndex set by readInputParameters() 4364 mReqChannelCount(popcount(channelMask)), 4365 mReqSampleRate(sampleRate) 4366 // mBytesRead is only meaningful while active, and so is cleared in start() 4367 // (but might be better to also clear here for dump?) 4368#ifdef TEE_SINK 4369 , mTeeSink(teeSink) 4370#endif 4371{ 4372 snprintf(mName, kNameLength, "AudioIn_%X", id); 4373 4374 readInputParameters(); 4375} 4376 4377 4378AudioFlinger::RecordThread::~RecordThread() 4379{ 4380 delete[] mRsmpInBuffer; 4381 delete mResampler; 4382 delete[] mRsmpOutBuffer; 4383} 4384 4385void AudioFlinger::RecordThread::onFirstRef() 4386{ 4387 run(mName, PRIORITY_URGENT_AUDIO); 4388} 4389 4390bool AudioFlinger::RecordThread::threadLoop() 4391{ 4392 AudioBufferProvider::Buffer buffer; 4393 4394 nsecs_t lastWarning = 0; 4395 4396 inputStandBy(); 4397 sp<RecordTrack> activeTrack; 4398 { 4399 Mutex::Autolock _l(mLock); 4400 activeTrack = mActiveTrack; 4401 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4402 } 4403 4404 // used to verify we've read at least once before evaluating how many bytes were read 4405 bool readOnce = false; 4406 4407 // used to request a deferred sleep, to be executed later while mutex is unlocked 4408 bool doSleep = false; 4409 4410 // start recording 4411 for (;;) { 4412 TrackBase::track_state activeTrackState; 4413 Vector< sp<EffectChain> > effectChains; 4414 4415 // sleep with mutex unlocked 4416 if (doSleep) { 4417 doSleep = false; 4418 usleep(kRecordThreadSleepUs); 4419 } 4420 4421 { // scope for mLock 4422 Mutex::Autolock _l(mLock); 4423 if (exitPending()) { 4424 break; 4425 } 4426 processConfigEvents_l(); 4427 // return value 'reconfig' is currently unused 4428 bool reconfig = checkForNewParameters_l(); 4429 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4430 SortedVector<int> tmp; 4431 tmp.add(mActiveTrack->uid()); 4432 updateWakeLockUids_l(tmp); 4433 } 4434 // make a stable copy of mActiveTrack 4435 activeTrack = mActiveTrack; 4436 if (activeTrack == 0) { 4437 standby(); 4438 // exitPending() can't become true here 4439 releaseWakeLock_l(); 4440 ALOGV("RecordThread: loop stopping"); 4441 // go to sleep 4442 mWaitWorkCV.wait(mLock); 4443 ALOGV("RecordThread: loop starting"); 4444 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4445 continue; 4446 } 4447 4448 if (activeTrack->isTerminated()) { 4449 removeTrack_l(activeTrack); 4450 mActiveTrack.clear(); 4451 continue; 4452 } 4453 4454 activeTrackState = activeTrack->mState; 4455 switch (activeTrackState) { 4456 case TrackBase::PAUSING: 4457 standby(); 4458 mActiveTrack.clear(); 4459 mStartStopCond.broadcast(); 4460 doSleep = true; 4461 continue; 4462 4463 case TrackBase::RESUMING: 4464 mStandby = false; 4465 if (mReqChannelCount != activeTrack->channelCount()) { 4466 mActiveTrack.clear(); 4467 mStartStopCond.broadcast(); 4468 continue; 4469 } 4470 if (readOnce) { 4471 mStartStopCond.broadcast(); 4472 // record start succeeds only if first read from audio input succeeds 4473 if (mBytesRead < 0) { 4474 mActiveTrack.clear(); 4475 continue; 4476 } 4477 activeTrack->mState = TrackBase::ACTIVE; 4478 } 4479 break; 4480 4481 case TrackBase::ACTIVE: 4482 break; 4483 4484 case TrackBase::IDLE: 4485 doSleep = true; 4486 continue; 4487 4488 default: 4489 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4490 } 4491 4492 lockEffectChains_l(effectChains); 4493 } 4494 4495 // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable 4496 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4497 4498 for (size_t i = 0; i < effectChains.size(); i ++) { 4499 // thread mutex is not locked, but effect chain is locked 4500 effectChains[i]->process_l(); 4501 } 4502 4503 buffer.frameCount = mFrameCount; 4504 status_t status = activeTrack->getNextBuffer(&buffer); 4505 if (status == NO_ERROR) { 4506 readOnce = true; 4507 size_t framesOut = buffer.frameCount; 4508 if (mResampler == NULL) { 4509 // no resampling 4510 while (framesOut) { 4511 size_t framesIn = mFrameCount - mRsmpInIndex; 4512 if (framesIn > 0) { 4513 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4514 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4515 activeTrack->mFrameSize; 4516 if (framesIn > framesOut) { 4517 framesIn = framesOut; 4518 } 4519 mRsmpInIndex += framesIn; 4520 framesOut -= framesIn; 4521 if (mChannelCount == mReqChannelCount) { 4522 memcpy(dst, src, framesIn * mFrameSize); 4523 } else { 4524 if (mChannelCount == 1) { 4525 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4526 (int16_t *)src, framesIn); 4527 } else { 4528 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4529 (int16_t *)src, framesIn); 4530 } 4531 } 4532 } 4533 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4534 void *readInto; 4535 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4536 readInto = buffer.raw; 4537 framesOut = 0; 4538 } else { 4539 readInto = mRsmpInBuffer; 4540 mRsmpInIndex = 0; 4541 } 4542 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4543 mBufferSize); 4544 if (mBytesRead <= 0) { 4545 // TODO: verify that it's benign to use a stale track state 4546 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4547 { 4548 ALOGE("Error reading audio input"); 4549 // Force input into standby so that it tries to 4550 // recover at next read attempt 4551 inputStandBy(); 4552 doSleep = true; 4553 } 4554 mRsmpInIndex = mFrameCount; 4555 framesOut = 0; 4556 buffer.frameCount = 0; 4557 } 4558#ifdef TEE_SINK 4559 else if (mTeeSink != 0) { 4560 (void) mTeeSink->write(readInto, 4561 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4562 } 4563#endif 4564 } 4565 } 4566 } else { 4567 // resampling 4568 4569 // resampler accumulates, but we only have one source track 4570 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4571 // alter output frame count as if we were expecting stereo samples 4572 if (mChannelCount == 1 && mReqChannelCount == 1) { 4573 framesOut >>= 1; 4574 } 4575 mResampler->resample(mRsmpOutBuffer, framesOut, 4576 this /* AudioBufferProvider* */); 4577 // ditherAndClamp() works as long as all buffers returned by 4578 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4579 if (mChannelCount == 2 && mReqChannelCount == 1) { 4580 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4581 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4582 // the resampler always outputs stereo samples: 4583 // do post stereo to mono conversion 4584 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4585 framesOut); 4586 } else { 4587 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4588 } 4589 // now done with mRsmpOutBuffer 4590 4591 } 4592 if (mFramestoDrop == 0) { 4593 activeTrack->releaseBuffer(&buffer); 4594 } else { 4595 if (mFramestoDrop > 0) { 4596 mFramestoDrop -= buffer.frameCount; 4597 if (mFramestoDrop <= 0) { 4598 clearSyncStartEvent(); 4599 } 4600 } else { 4601 mFramestoDrop += buffer.frameCount; 4602 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4603 mSyncStartEvent->isCancelled()) { 4604 ALOGW("Synced record %s, session %d, trigger session %d", 4605 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4606 activeTrack->sessionId(), 4607 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4608 clearSyncStartEvent(); 4609 } 4610 } 4611 } 4612 activeTrack->clearOverflow(); 4613 } 4614 // client isn't retrieving buffers fast enough 4615 else { 4616 if (!activeTrack->setOverflow()) { 4617 nsecs_t now = systemTime(); 4618 if ((now - lastWarning) > kWarningThrottleNs) { 4619 ALOGW("RecordThread: buffer overflow"); 4620 lastWarning = now; 4621 } 4622 } 4623 // Release the processor for a while before asking for a new buffer. 4624 // This will give the application more chance to read from the buffer and 4625 // clear the overflow. 4626 doSleep = true; 4627 } 4628 4629 // enable changes in effect chain 4630 unlockEffectChains(effectChains); 4631 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4632 } 4633 4634 standby(); 4635 4636 { 4637 Mutex::Autolock _l(mLock); 4638 for (size_t i = 0; i < mTracks.size(); i++) { 4639 sp<RecordTrack> track = mTracks[i]; 4640 track->invalidate(); 4641 } 4642 mActiveTrack.clear(); 4643 mStartStopCond.broadcast(); 4644 } 4645 4646 releaseWakeLock(); 4647 4648 ALOGV("RecordThread %p exiting", this); 4649 return false; 4650} 4651 4652void AudioFlinger::RecordThread::standby() 4653{ 4654 if (!mStandby) { 4655 inputStandBy(); 4656 mStandby = true; 4657 } 4658} 4659 4660void AudioFlinger::RecordThread::inputStandBy() 4661{ 4662 mInput->stream->common.standby(&mInput->stream->common); 4663} 4664 4665sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4666 const sp<AudioFlinger::Client>& client, 4667 uint32_t sampleRate, 4668 audio_format_t format, 4669 audio_channel_mask_t channelMask, 4670 size_t frameCount, 4671 int sessionId, 4672 int uid, 4673 IAudioFlinger::track_flags_t *flags, 4674 pid_t tid, 4675 status_t *status) 4676{ 4677 sp<RecordTrack> track; 4678 status_t lStatus; 4679 4680 lStatus = initCheck(); 4681 if (lStatus != NO_ERROR) { 4682 ALOGE("createRecordTrack_l() audio driver not initialized"); 4683 goto Exit; 4684 } 4685 // client expresses a preference for FAST, but we get the final say 4686 if (*flags & IAudioFlinger::TRACK_FAST) { 4687 if ( 4688 // use case: callback handler and frame count is default or at least as large as HAL 4689 ( 4690 (tid != -1) && 4691 ((frameCount == 0) || 4692 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4693 ) && 4694 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4695 // mono or stereo 4696 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4697 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4698 // hardware sample rate 4699 (sampleRate == mSampleRate) && 4700 // record thread has an associated fast recorder 4701 hasFastRecorder() 4702 // FIXME test that RecordThread for this fast track has a capable output HAL 4703 // FIXME add a permission test also? 4704 ) { 4705 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4706 if (frameCount == 0) { 4707 frameCount = mFrameCount * kFastTrackMultiplier; 4708 } 4709 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4710 frameCount, mFrameCount); 4711 } else { 4712 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4713 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4714 "hasFastRecorder=%d tid=%d", 4715 frameCount, mFrameCount, format, 4716 audio_is_linear_pcm(format), 4717 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4718 *flags &= ~IAudioFlinger::TRACK_FAST; 4719 // For compatibility with AudioRecord calculation, buffer depth is forced 4720 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4721 // This is probably too conservative, but legacy application code may depend on it. 4722 // If you change this calculation, also review the start threshold which is related. 4723 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4724 size_t mNormalFrameCount = 2048; // FIXME 4725 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4726 if (minBufCount < 2) { 4727 minBufCount = 2; 4728 } 4729 size_t minFrameCount = mNormalFrameCount * minBufCount; 4730 if (frameCount < minFrameCount) { 4731 frameCount = minFrameCount; 4732 } 4733 } 4734 } 4735 4736 // FIXME use flags and tid similar to createTrack_l() 4737 4738 { // scope for mLock 4739 Mutex::Autolock _l(mLock); 4740 4741 track = new RecordTrack(this, client, sampleRate, 4742 format, channelMask, frameCount, sessionId, uid); 4743 4744 lStatus = track->initCheck(); 4745 if (lStatus != NO_ERROR) { 4746 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4747 track.clear(); 4748 goto Exit; 4749 } 4750 mTracks.add(track); 4751 4752 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4753 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4754 mAudioFlinger->btNrecIsOff(); 4755 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4756 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4757 4758 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4759 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4760 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4761 // so ask activity manager to do this on our behalf 4762 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4763 } 4764 } 4765 lStatus = NO_ERROR; 4766 4767Exit: 4768 *status = lStatus; 4769 return track; 4770} 4771 4772status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4773 AudioSystem::sync_event_t event, 4774 int triggerSession) 4775{ 4776 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4777 sp<ThreadBase> strongMe = this; 4778 status_t status = NO_ERROR; 4779 4780 if (event == AudioSystem::SYNC_EVENT_NONE) { 4781 clearSyncStartEvent(); 4782 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4783 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4784 triggerSession, 4785 recordTrack->sessionId(), 4786 syncStartEventCallback, 4787 this); 4788 // Sync event can be cancelled by the trigger session if the track is not in a 4789 // compatible state in which case we start record immediately 4790 if (mSyncStartEvent->isCancelled()) { 4791 clearSyncStartEvent(); 4792 } else { 4793 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4794 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4795 } 4796 } 4797 4798 { 4799 // This section is a rendezvous between binder thread executing start() and RecordThread 4800 AutoMutex lock(mLock); 4801 if (mActiveTrack != 0) { 4802 if (recordTrack != mActiveTrack.get()) { 4803 status = -EBUSY; 4804 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4805 mActiveTrack->mState = TrackBase::ACTIVE; 4806 } 4807 return status; 4808 } 4809 4810 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4811 recordTrack->mState = TrackBase::IDLE; 4812 mActiveTrack = recordTrack; 4813 mLock.unlock(); 4814 status_t status = AudioSystem::startInput(mId); 4815 mLock.lock(); 4816 // FIXME should verify that mActiveTrack is still == recordTrack 4817 if (status != NO_ERROR) { 4818 mActiveTrack.clear(); 4819 clearSyncStartEvent(); 4820 return status; 4821 } 4822 mRsmpInIndex = mFrameCount; 4823 mBytesRead = 0; 4824 if (mResampler != NULL) { 4825 mResampler->reset(); 4826 } 4827 // FIXME hijacking a playback track state name which was intended for start after pause; 4828 // here 'STARTING_2' would be more accurate 4829 mActiveTrack->mState = TrackBase::RESUMING; 4830 // signal thread to start 4831 ALOGV("Signal record thread"); 4832 mWaitWorkCV.broadcast(); 4833 // do not wait for mStartStopCond if exiting 4834 if (exitPending()) { 4835 mActiveTrack.clear(); 4836 status = INVALID_OPERATION; 4837 goto startError; 4838 } 4839 // FIXME incorrect usage of wait: no explicit predicate or loop 4840 mStartStopCond.wait(mLock); 4841 if (mActiveTrack == 0) { 4842 ALOGV("Record failed to start"); 4843 status = BAD_VALUE; 4844 goto startError; 4845 } 4846 ALOGV("Record started OK"); 4847 return status; 4848 } 4849 4850startError: 4851 AudioSystem::stopInput(mId); 4852 clearSyncStartEvent(); 4853 return status; 4854} 4855 4856void AudioFlinger::RecordThread::clearSyncStartEvent() 4857{ 4858 if (mSyncStartEvent != 0) { 4859 mSyncStartEvent->cancel(); 4860 } 4861 mSyncStartEvent.clear(); 4862 mFramestoDrop = 0; 4863} 4864 4865void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4866{ 4867 sp<SyncEvent> strongEvent = event.promote(); 4868 4869 if (strongEvent != 0) { 4870 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4871 me->handleSyncStartEvent(strongEvent); 4872 } 4873} 4874 4875void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4876{ 4877 if (event == mSyncStartEvent) { 4878 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4879 // from audio HAL 4880 mFramestoDrop = mFrameCount * 2; 4881 } 4882} 4883 4884bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4885 ALOGV("RecordThread::stop"); 4886 AutoMutex _l(mLock); 4887 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4888 return false; 4889 } 4890 // note that threadLoop may still be processing the track at this point [without lock] 4891 recordTrack->mState = TrackBase::PAUSING; 4892 // do not wait for mStartStopCond if exiting 4893 if (exitPending()) { 4894 return true; 4895 } 4896 // FIXME incorrect usage of wait: no explicit predicate or loop 4897 mStartStopCond.wait(mLock); 4898 // if we have been restarted, recordTrack == mActiveTrack.get() here 4899 if (exitPending() || recordTrack != mActiveTrack.get()) { 4900 ALOGV("Record stopped OK"); 4901 return true; 4902 } 4903 return false; 4904} 4905 4906bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4907{ 4908 return false; 4909} 4910 4911status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4912{ 4913#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4914 if (!isValidSyncEvent(event)) { 4915 return BAD_VALUE; 4916 } 4917 4918 int eventSession = event->triggerSession(); 4919 status_t ret = NAME_NOT_FOUND; 4920 4921 Mutex::Autolock _l(mLock); 4922 4923 for (size_t i = 0; i < mTracks.size(); i++) { 4924 sp<RecordTrack> track = mTracks[i]; 4925 if (eventSession == track->sessionId()) { 4926 (void) track->setSyncEvent(event); 4927 ret = NO_ERROR; 4928 } 4929 } 4930 return ret; 4931#else 4932 return BAD_VALUE; 4933#endif 4934} 4935 4936// destroyTrack_l() must be called with ThreadBase::mLock held 4937void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4938{ 4939 track->terminate(); 4940 track->mState = TrackBase::STOPPED; 4941 // active tracks are removed by threadLoop() 4942 if (mActiveTrack != track) { 4943 removeTrack_l(track); 4944 } 4945} 4946 4947void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4948{ 4949 mTracks.remove(track); 4950 // need anything related to effects here? 4951} 4952 4953void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4954{ 4955 dumpInternals(fd, args); 4956 dumpTracks(fd, args); 4957 dumpEffectChains(fd, args); 4958} 4959 4960void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4961{ 4962 const size_t SIZE = 256; 4963 char buffer[SIZE]; 4964 String8 result; 4965 4966 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4967 result.append(buffer); 4968 4969 if (mActiveTrack != 0) { 4970 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4971 result.append(buffer); 4972 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4973 result.append(buffer); 4974 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4975 result.append(buffer); 4976 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4977 result.append(buffer); 4978 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4979 result.append(buffer); 4980 } else { 4981 result.append("No active record client\n"); 4982 } 4983 4984 write(fd, result.string(), result.size()); 4985 4986 dumpBase(fd, args); 4987} 4988 4989void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4990{ 4991 const size_t SIZE = 256; 4992 char buffer[SIZE]; 4993 String8 result; 4994 4995 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4996 result.append(buffer); 4997 RecordTrack::appendDumpHeader(result); 4998 for (size_t i = 0; i < mTracks.size(); ++i) { 4999 sp<RecordTrack> track = mTracks[i]; 5000 if (track != 0) { 5001 track->dump(buffer, SIZE); 5002 result.append(buffer); 5003 } 5004 } 5005 5006 if (mActiveTrack != 0) { 5007 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5008 result.append(buffer); 5009 RecordTrack::appendDumpHeader(result); 5010 mActiveTrack->dump(buffer, SIZE); 5011 result.append(buffer); 5012 5013 } 5014 write(fd, result.string(), result.size()); 5015} 5016 5017// AudioBufferProvider interface 5018status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5019{ 5020 size_t framesReq = buffer->frameCount; 5021 size_t framesReady = mFrameCount - mRsmpInIndex; 5022 int channelCount; 5023 5024 if (framesReady == 0) { 5025 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 5026 if (mBytesRead <= 0) { 5027 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 5028 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5029 // Force input into standby so that it tries to 5030 // recover at next read attempt 5031 inputStandBy(); 5032 // FIXME an awkward place to sleep, consider using doSleep when this is pulled up 5033 usleep(kRecordThreadSleepUs); 5034 } 5035 buffer->raw = NULL; 5036 buffer->frameCount = 0; 5037 return NOT_ENOUGH_DATA; 5038 } 5039 mRsmpInIndex = 0; 5040 framesReady = mFrameCount; 5041 } 5042 5043 if (framesReq > framesReady) { 5044 framesReq = framesReady; 5045 } 5046 5047 if (mChannelCount == 1 && mReqChannelCount == 2) { 5048 channelCount = 1; 5049 } else { 5050 channelCount = 2; 5051 } 5052 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5053 buffer->frameCount = framesReq; 5054 return NO_ERROR; 5055} 5056 5057// AudioBufferProvider interface 5058void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5059{ 5060 mRsmpInIndex += buffer->frameCount; 5061 buffer->frameCount = 0; 5062} 5063 5064bool AudioFlinger::RecordThread::checkForNewParameters_l() 5065{ 5066 bool reconfig = false; 5067 5068 while (!mNewParameters.isEmpty()) { 5069 status_t status = NO_ERROR; 5070 String8 keyValuePair = mNewParameters[0]; 5071 AudioParameter param = AudioParameter(keyValuePair); 5072 int value; 5073 audio_format_t reqFormat = mFormat; 5074 uint32_t reqSamplingRate = mReqSampleRate; 5075 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5076 5077 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5078 reqSamplingRate = value; 5079 reconfig = true; 5080 } 5081 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5082 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5083 status = BAD_VALUE; 5084 } else { 5085 reqFormat = (audio_format_t) value; 5086 reconfig = true; 5087 } 5088 } 5089 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5090 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5091 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5092 status = BAD_VALUE; 5093 } else { 5094 reqChannelMask = mask; 5095 reconfig = true; 5096 } 5097 } 5098 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5099 // do not accept frame count changes if tracks are open as the track buffer 5100 // size depends on frame count and correct behavior would not be guaranteed 5101 // if frame count is changed after track creation 5102 if (mActiveTrack != 0) { 5103 status = INVALID_OPERATION; 5104 } else { 5105 reconfig = true; 5106 } 5107 } 5108 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5109 // forward device change to effects that have requested to be 5110 // aware of attached audio device. 5111 for (size_t i = 0; i < mEffectChains.size(); i++) { 5112 mEffectChains[i]->setDevice_l(value); 5113 } 5114 5115 // store input device and output device but do not forward output device to audio HAL. 5116 // Note that status is ignored by the caller for output device 5117 // (see AudioFlinger::setParameters() 5118 if (audio_is_output_devices(value)) { 5119 mOutDevice = value; 5120 status = BAD_VALUE; 5121 } else { 5122 mInDevice = value; 5123 // disable AEC and NS if the device is a BT SCO headset supporting those 5124 // pre processings 5125 if (mTracks.size() > 0) { 5126 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5127 mAudioFlinger->btNrecIsOff(); 5128 for (size_t i = 0; i < mTracks.size(); i++) { 5129 sp<RecordTrack> track = mTracks[i]; 5130 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5131 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5132 } 5133 } 5134 } 5135 } 5136 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5137 mAudioSource != (audio_source_t)value) { 5138 // forward device change to effects that have requested to be 5139 // aware of attached audio device. 5140 for (size_t i = 0; i < mEffectChains.size(); i++) { 5141 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5142 } 5143 mAudioSource = (audio_source_t)value; 5144 } 5145 5146 if (status == NO_ERROR) { 5147 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5148 keyValuePair.string()); 5149 if (status == INVALID_OPERATION) { 5150 inputStandBy(); 5151 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5152 keyValuePair.string()); 5153 } 5154 if (reconfig) { 5155 if (status == BAD_VALUE && 5156 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5157 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5158 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5159 <= (2 * reqSamplingRate)) && 5160 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5161 <= FCC_2 && 5162 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5163 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5164 status = NO_ERROR; 5165 } 5166 if (status == NO_ERROR) { 5167 readInputParameters(); 5168 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5169 } 5170 } 5171 } 5172 5173 mNewParameters.removeAt(0); 5174 5175 mParamStatus = status; 5176 mParamCond.signal(); 5177 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5178 // already timed out waiting for the status and will never signal the condition. 5179 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5180 } 5181 return reconfig; 5182} 5183 5184String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5185{ 5186 Mutex::Autolock _l(mLock); 5187 if (initCheck() != NO_ERROR) { 5188 return String8(); 5189 } 5190 5191 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5192 const String8 out_s8(s); 5193 free(s); 5194 return out_s8; 5195} 5196 5197void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5198 AudioSystem::OutputDescriptor desc; 5199 void *param2 = NULL; 5200 5201 switch (event) { 5202 case AudioSystem::INPUT_OPENED: 5203 case AudioSystem::INPUT_CONFIG_CHANGED: 5204 desc.channelMask = mChannelMask; 5205 desc.samplingRate = mSampleRate; 5206 desc.format = mFormat; 5207 desc.frameCount = mFrameCount; 5208 desc.latency = 0; 5209 param2 = &desc; 5210 break; 5211 5212 case AudioSystem::INPUT_CLOSED: 5213 default: 5214 break; 5215 } 5216 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5217} 5218 5219void AudioFlinger::RecordThread::readInputParameters() 5220{ 5221 delete[] mRsmpInBuffer; 5222 // mRsmpInBuffer is always assigned a new[] below 5223 delete[] mRsmpOutBuffer; 5224 mRsmpOutBuffer = NULL; 5225 delete mResampler; 5226 mResampler = NULL; 5227 5228 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5229 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5230 mChannelCount = popcount(mChannelMask); 5231 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5232 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5233 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5234 } 5235 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5236 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5237 mFrameCount = mBufferSize / mFrameSize; 5238 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5239 5240 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5241 int channelCount; 5242 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5243 // stereo to mono post process as the resampler always outputs stereo. 5244 if (mChannelCount == 1 && mReqChannelCount == 2) { 5245 channelCount = 1; 5246 } else { 5247 channelCount = 2; 5248 } 5249 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5250 mResampler->setSampleRate(mSampleRate); 5251 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5252 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5253 5254 // optmization: if mono to mono, alter input frame count as if we were inputing 5255 // stereo samples 5256 if (mChannelCount == 1 && mReqChannelCount == 1) { 5257 mFrameCount >>= 1; 5258 } 5259 5260 } 5261 mRsmpInIndex = mFrameCount; 5262} 5263 5264unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5265{ 5266 Mutex::Autolock _l(mLock); 5267 if (initCheck() != NO_ERROR) { 5268 return 0; 5269 } 5270 5271 return mInput->stream->get_input_frames_lost(mInput->stream); 5272} 5273 5274uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5275{ 5276 Mutex::Autolock _l(mLock); 5277 uint32_t result = 0; 5278 if (getEffectChain_l(sessionId) != 0) { 5279 result = EFFECT_SESSION; 5280 } 5281 5282 for (size_t i = 0; i < mTracks.size(); ++i) { 5283 if (sessionId == mTracks[i]->sessionId()) { 5284 result |= TRACK_SESSION; 5285 break; 5286 } 5287 } 5288 5289 return result; 5290} 5291 5292KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5293{ 5294 KeyedVector<int, bool> ids; 5295 Mutex::Autolock _l(mLock); 5296 for (size_t j = 0; j < mTracks.size(); ++j) { 5297 sp<RecordThread::RecordTrack> track = mTracks[j]; 5298 int sessionId = track->sessionId(); 5299 if (ids.indexOfKey(sessionId) < 0) { 5300 ids.add(sessionId, true); 5301 } 5302 } 5303 return ids; 5304} 5305 5306AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5307{ 5308 Mutex::Autolock _l(mLock); 5309 AudioStreamIn *input = mInput; 5310 mInput = NULL; 5311 return input; 5312} 5313 5314// this method must always be called either with ThreadBase mLock held or inside the thread loop 5315audio_stream_t* AudioFlinger::RecordThread::stream() const 5316{ 5317 if (mInput == NULL) { 5318 return NULL; 5319 } 5320 return &mInput->stream->common; 5321} 5322 5323status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5324{ 5325 // only one chain per input thread 5326 if (mEffectChains.size() != 0) { 5327 return INVALID_OPERATION; 5328 } 5329 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5330 5331 chain->setInBuffer(NULL); 5332 chain->setOutBuffer(NULL); 5333 5334 checkSuspendOnAddEffectChain_l(chain); 5335 5336 mEffectChains.add(chain); 5337 5338 return NO_ERROR; 5339} 5340 5341size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5342{ 5343 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5344 ALOGW_IF(mEffectChains.size() != 1, 5345 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5346 chain.get(), mEffectChains.size(), this); 5347 if (mEffectChains.size() == 1) { 5348 mEffectChains.removeAt(0); 5349 } 5350 return 0; 5351} 5352 5353}; // namespace android 5354