Threads.cpp revision 0f11b51a57bc9062c4fe8af73747319cedabc5d6
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title 189#ifndef DEBUG_CPU_USAGE 190 __unused 191#endif 192 ) { 193#ifdef DEBUG_CPU_USAGE 194 // get current thread's delta CPU time in wall clock ns 195 double wcNs; 196 bool valid = mCpuUsage.sampleAndEnable(wcNs); 197 198 // record sample for wall clock statistics 199 if (valid) { 200 mWcStats.sample(wcNs); 201 } 202 203 // get the current CPU number 204 int cpuNum = sched_getcpu(); 205 206 // get the current CPU frequency in kHz 207 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 208 209 // check if either CPU number or frequency changed 210 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 211 mCpuNum = cpuNum; 212 mCpukHz = cpukHz; 213 // ignore sample for purposes of cycles 214 valid = false; 215 } 216 217 // if no change in CPU number or frequency, then record sample for cycle statistics 218 if (valid && mCpukHz > 0) { 219 double cycles = wcNs * cpukHz * 0.000001; 220 mHzStats.sample(cycles); 221 } 222 223 unsigned n = mWcStats.n(); 224 // mCpuUsage.elapsed() is expensive, so don't call it every loop 225 if ((n & 127) == 1) { 226 long long elapsed = mCpuUsage.elapsed(); 227 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 228 double perLoop = elapsed / (double) n; 229 double perLoop100 = perLoop * 0.01; 230 double perLoop1k = perLoop * 0.001; 231 double mean = mWcStats.mean(); 232 double stddev = mWcStats.stddev(); 233 double minimum = mWcStats.minimum(); 234 double maximum = mWcStats.maximum(); 235 double meanCycles = mHzStats.mean(); 236 double stddevCycles = mHzStats.stddev(); 237 double minCycles = mHzStats.minimum(); 238 double maxCycles = mHzStats.maximum(); 239 mCpuUsage.resetElapsed(); 240 mWcStats.reset(); 241 mHzStats.reset(); 242 ALOGD("CPU usage for %s over past %.1f secs\n" 243 " (%u mixer loops at %.1f mean ms per loop):\n" 244 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 245 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 246 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 247 title.string(), 248 elapsed * .000000001, n, perLoop * .000001, 249 mean * .001, 250 stddev * .001, 251 minimum * .001, 252 maximum * .001, 253 mean / perLoop100, 254 stddev / perLoop100, 255 minimum / perLoop100, 256 maximum / perLoop100, 257 meanCycles / perLoop1k, 258 stddevCycles / perLoop1k, 259 minCycles / perLoop1k, 260 maxCycles / perLoop1k); 261 262 } 263 } 264#endif 265}; 266 267// ---------------------------------------------------------------------------- 268// ThreadBase 269// ---------------------------------------------------------------------------- 270 271AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 272 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 273 : Thread(false /*canCallJava*/), 274 mType(type), 275 mAudioFlinger(audioFlinger), 276 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 277 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 278 mParamStatus(NO_ERROR), 279 //FIXME: mStandby should be true here. Is this some kind of hack? 280 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 281 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 282 // mName will be set by concrete (non-virtual) subclass 283 mDeathRecipient(new PMDeathRecipient(this)) 284{ 285} 286 287AudioFlinger::ThreadBase::~ThreadBase() 288{ 289 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 290 for (size_t i = 0; i < mConfigEvents.size(); i++) { 291 delete mConfigEvents[i]; 292 } 293 mConfigEvents.clear(); 294 295 mParamCond.broadcast(); 296 // do not lock the mutex in destructor 297 releaseWakeLock_l(); 298 if (mPowerManager != 0) { 299 sp<IBinder> binder = mPowerManager->asBinder(); 300 binder->unlinkToDeath(mDeathRecipient); 301 } 302} 303 304status_t AudioFlinger::ThreadBase::readyToRun() 305{ 306 status_t status = initCheck(); 307 if (status == NO_ERROR) { 308 ALOGI("AudioFlinger's thread %p ready to run", this); 309 } else { 310 ALOGE("No working audio driver found."); 311 } 312 return status; 313} 314 315void AudioFlinger::ThreadBase::exit() 316{ 317 ALOGV("ThreadBase::exit"); 318 // do any cleanup required for exit to succeed 319 preExit(); 320 { 321 // This lock prevents the following race in thread (uniprocessor for illustration): 322 // if (!exitPending()) { 323 // // context switch from here to exit() 324 // // exit() calls requestExit(), what exitPending() observes 325 // // exit() calls signal(), which is dropped since no waiters 326 // // context switch back from exit() to here 327 // mWaitWorkCV.wait(...); 328 // // now thread is hung 329 // } 330 AutoMutex lock(mLock); 331 requestExit(); 332 mWaitWorkCV.broadcast(); 333 } 334 // When Thread::requestExitAndWait is made virtual and this method is renamed to 335 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 336 requestExitAndWait(); 337} 338 339status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 340{ 341 status_t status; 342 343 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 344 Mutex::Autolock _l(mLock); 345 346 mNewParameters.add(keyValuePairs); 347 mWaitWorkCV.signal(); 348 // wait condition with timeout in case the thread loop has exited 349 // before the request could be processed 350 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 351 status = mParamStatus; 352 mWaitWorkCV.signal(); 353 } else { 354 status = TIMED_OUT; 355 } 356 return status; 357} 358 359void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 360{ 361 Mutex::Autolock _l(mLock); 362 sendIoConfigEvent_l(event, param); 363} 364 365// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 366void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 367{ 368 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 369 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 370 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 371 param); 372 mWaitWorkCV.signal(); 373} 374 375// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 376void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 377{ 378 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 379 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 380 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 381 mConfigEvents.size(), pid, tid, prio); 382 mWaitWorkCV.signal(); 383} 384 385void AudioFlinger::ThreadBase::processConfigEvents() 386{ 387 Mutex::Autolock _l(mLock); 388 processConfigEvents_l(); 389} 390 391// post condition: mConfigEvents.isEmpty() 392void AudioFlinger::ThreadBase::processConfigEvents_l() 393{ 394 while (!mConfigEvents.isEmpty()) { 395 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 396 ConfigEvent *event = mConfigEvents[0]; 397 mConfigEvents.removeAt(0); 398 // release mLock before locking AudioFlinger mLock: lock order is always 399 // AudioFlinger then ThreadBase to avoid cross deadlock 400 mLock.unlock(); 401 switch (event->type()) { 402 case CFG_EVENT_PRIO: { 403 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 404 // FIXME Need to understand why this has be done asynchronously 405 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 406 true /*asynchronous*/); 407 if (err != 0) { 408 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 409 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 410 } 411 } break; 412 case CFG_EVENT_IO: { 413 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 414 { 415 Mutex::Autolock _l(mAudioFlinger->mLock); 416 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 417 } 418 } break; 419 default: 420 ALOGE("processConfigEvents() unknown event type %d", event->type()); 421 break; 422 } 423 delete event; 424 mLock.lock(); 425 } 426} 427 428void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 429{ 430 const size_t SIZE = 256; 431 char buffer[SIZE]; 432 String8 result; 433 434 bool locked = AudioFlinger::dumpTryLock(mLock); 435 if (!locked) { 436 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 437 write(fd, buffer, strlen(buffer)); 438 } 439 440 snprintf(buffer, SIZE, "io handle: %d\n", mId); 441 result.append(buffer); 442 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 443 result.append(buffer); 444 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 445 result.append(buffer); 446 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 447 result.append(buffer); 448 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 449 result.append(buffer); 450 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 451 result.append(buffer); 452 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 453 result.append(buffer); 454 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 455 result.append(buffer); 456 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 457 result.append(buffer); 458 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 459 result.append(buffer); 460 461 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 462 result.append(buffer); 463 result.append(" Index Command"); 464 for (size_t i = 0; i < mNewParameters.size(); ++i) { 465 snprintf(buffer, SIZE, "\n %02d ", i); 466 result.append(buffer); 467 result.append(mNewParameters[i]); 468 } 469 470 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 471 result.append(buffer); 472 for (size_t i = 0; i < mConfigEvents.size(); i++) { 473 mConfigEvents[i]->dump(buffer, SIZE); 474 result.append(buffer); 475 } 476 result.append("\n"); 477 478 write(fd, result.string(), result.size()); 479 480 if (locked) { 481 mLock.unlock(); 482 } 483} 484 485void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 486{ 487 const size_t SIZE = 256; 488 char buffer[SIZE]; 489 String8 result; 490 491 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 492 write(fd, buffer, strlen(buffer)); 493 494 for (size_t i = 0; i < mEffectChains.size(); ++i) { 495 sp<EffectChain> chain = mEffectChains[i]; 496 if (chain != 0) { 497 chain->dump(fd, args); 498 } 499 } 500} 501 502void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 503{ 504 Mutex::Autolock _l(mLock); 505 acquireWakeLock_l(uid); 506} 507 508String16 AudioFlinger::ThreadBase::getWakeLockTag() 509{ 510 switch (mType) { 511 case MIXER: 512 return String16("AudioMix"); 513 case DIRECT: 514 return String16("AudioDirectOut"); 515 case DUPLICATING: 516 return String16("AudioDup"); 517 case RECORD: 518 return String16("AudioIn"); 519 case OFFLOAD: 520 return String16("AudioOffload"); 521 default: 522 ALOG_ASSERT(false); 523 return String16("AudioUnknown"); 524 } 525} 526 527void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 528{ 529 getPowerManager_l(); 530 if (mPowerManager != 0) { 531 sp<IBinder> binder = new BBinder(); 532 status_t status; 533 if (uid >= 0) { 534 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 535 binder, 536 getWakeLockTag(), 537 String16("media"), 538 uid); 539 } else { 540 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 541 binder, 542 getWakeLockTag(), 543 String16("media")); 544 } 545 if (status == NO_ERROR) { 546 mWakeLockToken = binder; 547 } 548 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 549 } 550} 551 552void AudioFlinger::ThreadBase::releaseWakeLock() 553{ 554 Mutex::Autolock _l(mLock); 555 releaseWakeLock_l(); 556} 557 558void AudioFlinger::ThreadBase::releaseWakeLock_l() 559{ 560 if (mWakeLockToken != 0) { 561 ALOGV("releaseWakeLock_l() %s", mName); 562 if (mPowerManager != 0) { 563 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 564 } 565 mWakeLockToken.clear(); 566 } 567} 568 569void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 570 Mutex::Autolock _l(mLock); 571 updateWakeLockUids_l(uids); 572} 573 574void AudioFlinger::ThreadBase::getPowerManager_l() { 575 576 if (mPowerManager == 0) { 577 // use checkService() to avoid blocking if power service is not up yet 578 sp<IBinder> binder = 579 defaultServiceManager()->checkService(String16("power")); 580 if (binder == 0) { 581 ALOGW("Thread %s cannot connect to the power manager service", mName); 582 } else { 583 mPowerManager = interface_cast<IPowerManager>(binder); 584 binder->linkToDeath(mDeathRecipient); 585 } 586 } 587} 588 589void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 590 591 getPowerManager_l(); 592 if (mWakeLockToken == NULL) { 593 ALOGE("no wake lock to update!"); 594 return; 595 } 596 if (mPowerManager != 0) { 597 sp<IBinder> binder = new BBinder(); 598 status_t status; 599 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 600 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 601 } 602} 603 604void AudioFlinger::ThreadBase::clearPowerManager() 605{ 606 Mutex::Autolock _l(mLock); 607 releaseWakeLock_l(); 608 mPowerManager.clear(); 609} 610 611void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 612{ 613 sp<ThreadBase> thread = mThread.promote(); 614 if (thread != 0) { 615 thread->clearPowerManager(); 616 } 617 ALOGW("power manager service died !!!"); 618} 619 620void AudioFlinger::ThreadBase::setEffectSuspended( 621 const effect_uuid_t *type, bool suspend, int sessionId) 622{ 623 Mutex::Autolock _l(mLock); 624 setEffectSuspended_l(type, suspend, sessionId); 625} 626 627void AudioFlinger::ThreadBase::setEffectSuspended_l( 628 const effect_uuid_t *type, bool suspend, int sessionId) 629{ 630 sp<EffectChain> chain = getEffectChain_l(sessionId); 631 if (chain != 0) { 632 if (type != NULL) { 633 chain->setEffectSuspended_l(type, suspend); 634 } else { 635 chain->setEffectSuspendedAll_l(suspend); 636 } 637 } 638 639 updateSuspendedSessions_l(type, suspend, sessionId); 640} 641 642void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 643{ 644 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 645 if (index < 0) { 646 return; 647 } 648 649 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 650 mSuspendedSessions.valueAt(index); 651 652 for (size_t i = 0; i < sessionEffects.size(); i++) { 653 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 654 for (int j = 0; j < desc->mRefCount; j++) { 655 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 656 chain->setEffectSuspendedAll_l(true); 657 } else { 658 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 659 desc->mType.timeLow); 660 chain->setEffectSuspended_l(&desc->mType, true); 661 } 662 } 663 } 664} 665 666void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 667 bool suspend, 668 int sessionId) 669{ 670 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 671 672 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 673 674 if (suspend) { 675 if (index >= 0) { 676 sessionEffects = mSuspendedSessions.valueAt(index); 677 } else { 678 mSuspendedSessions.add(sessionId, sessionEffects); 679 } 680 } else { 681 if (index < 0) { 682 return; 683 } 684 sessionEffects = mSuspendedSessions.valueAt(index); 685 } 686 687 688 int key = EffectChain::kKeyForSuspendAll; 689 if (type != NULL) { 690 key = type->timeLow; 691 } 692 index = sessionEffects.indexOfKey(key); 693 694 sp<SuspendedSessionDesc> desc; 695 if (suspend) { 696 if (index >= 0) { 697 desc = sessionEffects.valueAt(index); 698 } else { 699 desc = new SuspendedSessionDesc(); 700 if (type != NULL) { 701 desc->mType = *type; 702 } 703 sessionEffects.add(key, desc); 704 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 705 } 706 desc->mRefCount++; 707 } else { 708 if (index < 0) { 709 return; 710 } 711 desc = sessionEffects.valueAt(index); 712 if (--desc->mRefCount == 0) { 713 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 714 sessionEffects.removeItemsAt(index); 715 if (sessionEffects.isEmpty()) { 716 ALOGV("updateSuspendedSessions_l() restore removing session %d", 717 sessionId); 718 mSuspendedSessions.removeItem(sessionId); 719 } 720 } 721 } 722 if (!sessionEffects.isEmpty()) { 723 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 724 } 725} 726 727void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 728 bool enabled, 729 int sessionId) 730{ 731 Mutex::Autolock _l(mLock); 732 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 733} 734 735void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 736 bool enabled, 737 int sessionId) 738{ 739 if (mType != RECORD) { 740 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 741 // another session. This gives the priority to well behaved effect control panels 742 // and applications not using global effects. 743 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 744 // global effects 745 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 746 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 747 } 748 } 749 750 sp<EffectChain> chain = getEffectChain_l(sessionId); 751 if (chain != 0) { 752 chain->checkSuspendOnEffectEnabled(effect, enabled); 753 } 754} 755 756// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 757sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 758 const sp<AudioFlinger::Client>& client, 759 const sp<IEffectClient>& effectClient, 760 int32_t priority, 761 int sessionId, 762 effect_descriptor_t *desc, 763 int *enabled, 764 status_t *status) 765{ 766 sp<EffectModule> effect; 767 sp<EffectHandle> handle; 768 status_t lStatus; 769 sp<EffectChain> chain; 770 bool chainCreated = false; 771 bool effectCreated = false; 772 bool effectRegistered = false; 773 774 lStatus = initCheck(); 775 if (lStatus != NO_ERROR) { 776 ALOGW("createEffect_l() Audio driver not initialized."); 777 goto Exit; 778 } 779 780 // Allow global effects only on offloaded and mixer threads 781 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 782 switch (mType) { 783 case MIXER: 784 case OFFLOAD: 785 break; 786 case DIRECT: 787 case DUPLICATING: 788 case RECORD: 789 default: 790 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 791 lStatus = BAD_VALUE; 792 goto Exit; 793 } 794 } 795 796 // Only Pre processor effects are allowed on input threads and only on input threads 797 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 798 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 799 desc->name, desc->flags, mType); 800 lStatus = BAD_VALUE; 801 goto Exit; 802 } 803 804 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 805 806 { // scope for mLock 807 Mutex::Autolock _l(mLock); 808 809 // check for existing effect chain with the requested audio session 810 chain = getEffectChain_l(sessionId); 811 if (chain == 0) { 812 // create a new chain for this session 813 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 814 chain = new EffectChain(this, sessionId); 815 addEffectChain_l(chain); 816 chain->setStrategy(getStrategyForSession_l(sessionId)); 817 chainCreated = true; 818 } else { 819 effect = chain->getEffectFromDesc_l(desc); 820 } 821 822 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 823 824 if (effect == 0) { 825 int id = mAudioFlinger->nextUniqueId(); 826 // Check CPU and memory usage 827 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 828 if (lStatus != NO_ERROR) { 829 goto Exit; 830 } 831 effectRegistered = true; 832 // create a new effect module if none present in the chain 833 effect = new EffectModule(this, chain, desc, id, sessionId); 834 lStatus = effect->status(); 835 if (lStatus != NO_ERROR) { 836 goto Exit; 837 } 838 effect->setOffloaded(mType == OFFLOAD, mId); 839 840 lStatus = chain->addEffect_l(effect); 841 if (lStatus != NO_ERROR) { 842 goto Exit; 843 } 844 effectCreated = true; 845 846 effect->setDevice(mOutDevice); 847 effect->setDevice(mInDevice); 848 effect->setMode(mAudioFlinger->getMode()); 849 effect->setAudioSource(mAudioSource); 850 } 851 // create effect handle and connect it to effect module 852 handle = new EffectHandle(effect, client, effectClient, priority); 853 lStatus = handle->initCheck(); 854 if (lStatus == OK) { 855 lStatus = effect->addHandle(handle.get()); 856 } 857 if (enabled != NULL) { 858 *enabled = (int)effect->isEnabled(); 859 } 860 } 861 862Exit: 863 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 864 Mutex::Autolock _l(mLock); 865 if (effectCreated) { 866 chain->removeEffect_l(effect); 867 } 868 if (effectRegistered) { 869 AudioSystem::unregisterEffect(effect->id()); 870 } 871 if (chainCreated) { 872 removeEffectChain_l(chain); 873 } 874 handle.clear(); 875 } 876 877 *status = lStatus; 878 return handle; 879} 880 881sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 882{ 883 Mutex::Autolock _l(mLock); 884 return getEffect_l(sessionId, effectId); 885} 886 887sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 888{ 889 sp<EffectChain> chain = getEffectChain_l(sessionId); 890 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 891} 892 893// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 894// PlaybackThread::mLock held 895status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 896{ 897 // check for existing effect chain with the requested audio session 898 int sessionId = effect->sessionId(); 899 sp<EffectChain> chain = getEffectChain_l(sessionId); 900 bool chainCreated = false; 901 902 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 903 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 904 this, effect->desc().name, effect->desc().flags); 905 906 if (chain == 0) { 907 // create a new chain for this session 908 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 909 chain = new EffectChain(this, sessionId); 910 addEffectChain_l(chain); 911 chain->setStrategy(getStrategyForSession_l(sessionId)); 912 chainCreated = true; 913 } 914 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 915 916 if (chain->getEffectFromId_l(effect->id()) != 0) { 917 ALOGW("addEffect_l() %p effect %s already present in chain %p", 918 this, effect->desc().name, chain.get()); 919 return BAD_VALUE; 920 } 921 922 effect->setOffloaded(mType == OFFLOAD, mId); 923 924 status_t status = chain->addEffect_l(effect); 925 if (status != NO_ERROR) { 926 if (chainCreated) { 927 removeEffectChain_l(chain); 928 } 929 return status; 930 } 931 932 effect->setDevice(mOutDevice); 933 effect->setDevice(mInDevice); 934 effect->setMode(mAudioFlinger->getMode()); 935 effect->setAudioSource(mAudioSource); 936 return NO_ERROR; 937} 938 939void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 940 941 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 942 effect_descriptor_t desc = effect->desc(); 943 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 944 detachAuxEffect_l(effect->id()); 945 } 946 947 sp<EffectChain> chain = effect->chain().promote(); 948 if (chain != 0) { 949 // remove effect chain if removing last effect 950 if (chain->removeEffect_l(effect) == 0) { 951 removeEffectChain_l(chain); 952 } 953 } else { 954 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 955 } 956} 957 958void AudioFlinger::ThreadBase::lockEffectChains_l( 959 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 960{ 961 effectChains = mEffectChains; 962 for (size_t i = 0; i < mEffectChains.size(); i++) { 963 mEffectChains[i]->lock(); 964 } 965} 966 967void AudioFlinger::ThreadBase::unlockEffectChains( 968 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 969{ 970 for (size_t i = 0; i < effectChains.size(); i++) { 971 effectChains[i]->unlock(); 972 } 973} 974 975sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 976{ 977 Mutex::Autolock _l(mLock); 978 return getEffectChain_l(sessionId); 979} 980 981sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 982{ 983 size_t size = mEffectChains.size(); 984 for (size_t i = 0; i < size; i++) { 985 if (mEffectChains[i]->sessionId() == sessionId) { 986 return mEffectChains[i]; 987 } 988 } 989 return 0; 990} 991 992void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 993{ 994 Mutex::Autolock _l(mLock); 995 size_t size = mEffectChains.size(); 996 for (size_t i = 0; i < size; i++) { 997 mEffectChains[i]->setMode_l(mode); 998 } 999} 1000 1001void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1002 EffectHandle *handle, 1003 bool unpinIfLast) { 1004 1005 Mutex::Autolock _l(mLock); 1006 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1007 // delete the effect module if removing last handle on it 1008 if (effect->removeHandle(handle) == 0) { 1009 if (!effect->isPinned() || unpinIfLast) { 1010 removeEffect_l(effect); 1011 AudioSystem::unregisterEffect(effect->id()); 1012 } 1013 } 1014} 1015 1016// ---------------------------------------------------------------------------- 1017// Playback 1018// ---------------------------------------------------------------------------- 1019 1020AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1021 AudioStreamOut* output, 1022 audio_io_handle_t id, 1023 audio_devices_t device, 1024 type_t type) 1025 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1026 mNormalFrameCount(0), mMixBuffer(NULL), 1027 mSuspended(0), mBytesWritten(0), 1028 mActiveTracksGeneration(0), 1029 // mStreamTypes[] initialized in constructor body 1030 mOutput(output), 1031 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1032 mMixerStatus(MIXER_IDLE), 1033 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1034 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1035 mBytesRemaining(0), 1036 mCurrentWriteLength(0), 1037 mUseAsyncWrite(false), 1038 mWriteAckSequence(0), 1039 mDrainSequence(0), 1040 mSignalPending(false), 1041 mScreenState(AudioFlinger::mScreenState), 1042 // index 0 is reserved for normal mixer's submix 1043 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1044 // mLatchD, mLatchQ, 1045 mLatchDValid(false), mLatchQValid(false) 1046{ 1047 snprintf(mName, kNameLength, "AudioOut_%X", id); 1048 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1049 1050 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1051 // it would be safer to explicitly pass initial masterVolume/masterMute as 1052 // parameter. 1053 // 1054 // If the HAL we are using has support for master volume or master mute, 1055 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1056 // and the mute set to false). 1057 mMasterVolume = audioFlinger->masterVolume_l(); 1058 mMasterMute = audioFlinger->masterMute_l(); 1059 if (mOutput && mOutput->audioHwDev) { 1060 if (mOutput->audioHwDev->canSetMasterVolume()) { 1061 mMasterVolume = 1.0; 1062 } 1063 1064 if (mOutput->audioHwDev->canSetMasterMute()) { 1065 mMasterMute = false; 1066 } 1067 } 1068 1069 readOutputParameters(); 1070 1071 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1072 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1073 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1074 stream = (audio_stream_type_t) (stream + 1)) { 1075 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1076 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1077 } 1078 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1079 // because mAudioFlinger doesn't have one to copy from 1080} 1081 1082AudioFlinger::PlaybackThread::~PlaybackThread() 1083{ 1084 mAudioFlinger->unregisterWriter(mNBLogWriter); 1085 delete[] mMixBuffer; 1086} 1087 1088void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1089{ 1090 dumpInternals(fd, args); 1091 dumpTracks(fd, args); 1092 dumpEffectChains(fd, args); 1093} 1094 1095void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1096{ 1097 const size_t SIZE = 256; 1098 char buffer[SIZE]; 1099 String8 result; 1100 1101 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1102 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1103 const stream_type_t *st = &mStreamTypes[i]; 1104 if (i > 0) { 1105 result.appendFormat(", "); 1106 } 1107 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1108 if (st->mute) { 1109 result.append("M"); 1110 } 1111 } 1112 result.append("\n"); 1113 write(fd, result.string(), result.length()); 1114 result.clear(); 1115 1116 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1117 result.append(buffer); 1118 Track::appendDumpHeader(result); 1119 for (size_t i = 0; i < mTracks.size(); ++i) { 1120 sp<Track> track = mTracks[i]; 1121 if (track != 0) { 1122 track->dump(buffer, SIZE); 1123 result.append(buffer); 1124 } 1125 } 1126 1127 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1128 result.append(buffer); 1129 Track::appendDumpHeader(result); 1130 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1131 sp<Track> track = mActiveTracks[i].promote(); 1132 if (track != 0) { 1133 track->dump(buffer, SIZE); 1134 result.append(buffer); 1135 } 1136 } 1137 write(fd, result.string(), result.size()); 1138 1139 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1140 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1141 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1142 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1143} 1144 1145void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1146{ 1147 const size_t SIZE = 256; 1148 char buffer[SIZE]; 1149 String8 result; 1150 1151 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1152 result.append(buffer); 1153 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1154 result.append(buffer); 1155 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1156 ns2ms(systemTime() - mLastWriteTime)); 1157 result.append(buffer); 1158 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1159 result.append(buffer); 1160 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1161 result.append(buffer); 1162 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1163 result.append(buffer); 1164 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1165 result.append(buffer); 1166 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1167 result.append(buffer); 1168 write(fd, result.string(), result.size()); 1169 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1170 1171 dumpBase(fd, args); 1172} 1173 1174// Thread virtuals 1175 1176void AudioFlinger::PlaybackThread::onFirstRef() 1177{ 1178 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1179} 1180 1181// ThreadBase virtuals 1182void AudioFlinger::PlaybackThread::preExit() 1183{ 1184 ALOGV(" preExit()"); 1185 // FIXME this is using hard-coded strings but in the future, this functionality will be 1186 // converted to use audio HAL extensions required to support tunneling 1187 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1188} 1189 1190// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1191sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1192 const sp<AudioFlinger::Client>& client, 1193 audio_stream_type_t streamType, 1194 uint32_t sampleRate, 1195 audio_format_t format, 1196 audio_channel_mask_t channelMask, 1197 size_t *pFrameCount, 1198 const sp<IMemory>& sharedBuffer, 1199 int sessionId, 1200 IAudioFlinger::track_flags_t *flags, 1201 pid_t tid, 1202 int uid, 1203 status_t *status) 1204{ 1205 size_t frameCount = *pFrameCount; 1206 sp<Track> track; 1207 status_t lStatus; 1208 1209 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1210 1211 // client expresses a preference for FAST, but we get the final say 1212 if (*flags & IAudioFlinger::TRACK_FAST) { 1213 if ( 1214 // not timed 1215 (!isTimed) && 1216 // either of these use cases: 1217 ( 1218 // use case 1: shared buffer with any frame count 1219 ( 1220 (sharedBuffer != 0) 1221 ) || 1222 // use case 2: callback handler and frame count is default or at least as large as HAL 1223 ( 1224 (tid != -1) && 1225 ((frameCount == 0) || 1226 (frameCount >= mFrameCount)) 1227 ) 1228 ) && 1229 // PCM data 1230 audio_is_linear_pcm(format) && 1231 // mono or stereo 1232 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1233 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1234#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1235 // hardware sample rate 1236 (sampleRate == mSampleRate) && 1237#endif 1238 // normal mixer has an associated fast mixer 1239 hasFastMixer() && 1240 // there are sufficient fast track slots available 1241 (mFastTrackAvailMask != 0) 1242 // FIXME test that MixerThread for this fast track has a capable output HAL 1243 // FIXME add a permission test also? 1244 ) { 1245 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1246 if (frameCount == 0) { 1247 frameCount = mFrameCount * kFastTrackMultiplier; 1248 } 1249 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1250 frameCount, mFrameCount); 1251 } else { 1252 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1253 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1254 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1255 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1256 audio_is_linear_pcm(format), 1257 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1258 *flags &= ~IAudioFlinger::TRACK_FAST; 1259 // For compatibility with AudioTrack calculation, buffer depth is forced 1260 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1261 // This is probably too conservative, but legacy application code may depend on it. 1262 // If you change this calculation, also review the start threshold which is related. 1263 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1264 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1265 if (minBufCount < 2) { 1266 minBufCount = 2; 1267 } 1268 size_t minFrameCount = mNormalFrameCount * minBufCount; 1269 if (frameCount < minFrameCount) { 1270 frameCount = minFrameCount; 1271 } 1272 } 1273 } 1274 *pFrameCount = frameCount; 1275 1276 if (mType == DIRECT) { 1277 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1278 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1279 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1280 "for output %p with format %d", 1281 sampleRate, format, channelMask, mOutput, mFormat); 1282 lStatus = BAD_VALUE; 1283 goto Exit; 1284 } 1285 } 1286 } else if (mType == OFFLOAD) { 1287 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1288 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1289 "for output %p with format %d", 1290 sampleRate, format, channelMask, mOutput, mFormat); 1291 lStatus = BAD_VALUE; 1292 goto Exit; 1293 } 1294 } else { 1295 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1296 ALOGE("createTrack_l() Bad parameter: format %d \"" 1297 "for output %p with format %d", 1298 format, mOutput, mFormat); 1299 lStatus = BAD_VALUE; 1300 goto Exit; 1301 } 1302 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1303 if (sampleRate > mSampleRate*2) { 1304 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1305 lStatus = BAD_VALUE; 1306 goto Exit; 1307 } 1308 } 1309 1310 lStatus = initCheck(); 1311 if (lStatus != NO_ERROR) { 1312 ALOGE("Audio driver not initialized."); 1313 goto Exit; 1314 } 1315 1316 { // scope for mLock 1317 Mutex::Autolock _l(mLock); 1318 1319 // all tracks in same audio session must share the same routing strategy otherwise 1320 // conflicts will happen when tracks are moved from one output to another by audio policy 1321 // manager 1322 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1323 for (size_t i = 0; i < mTracks.size(); ++i) { 1324 sp<Track> t = mTracks[i]; 1325 if (t != 0 && !t->isOutputTrack()) { 1326 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1327 if (sessionId == t->sessionId() && strategy != actual) { 1328 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1329 strategy, actual); 1330 lStatus = BAD_VALUE; 1331 goto Exit; 1332 } 1333 } 1334 } 1335 1336 if (!isTimed) { 1337 track = new Track(this, client, streamType, sampleRate, format, 1338 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1339 } else { 1340 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1341 channelMask, frameCount, sharedBuffer, sessionId, uid); 1342 } 1343 1344 // new Track always returns non-NULL, 1345 // but TimedTrack::create() is a factory that could fail by returning NULL 1346 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1347 if (lStatus != NO_ERROR) { 1348 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1349 track.clear(); 1350 goto Exit; 1351 } 1352 1353 mTracks.add(track); 1354 1355 sp<EffectChain> chain = getEffectChain_l(sessionId); 1356 if (chain != 0) { 1357 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1358 track->setMainBuffer(chain->inBuffer()); 1359 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1360 chain->incTrackCnt(); 1361 } 1362 1363 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1364 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1365 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1366 // so ask activity manager to do this on our behalf 1367 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1368 } 1369 } 1370 1371 lStatus = NO_ERROR; 1372 1373Exit: 1374 *status = lStatus; 1375 return track; 1376} 1377 1378uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1379{ 1380 return latency; 1381} 1382 1383uint32_t AudioFlinger::PlaybackThread::latency() const 1384{ 1385 Mutex::Autolock _l(mLock); 1386 return latency_l(); 1387} 1388uint32_t AudioFlinger::PlaybackThread::latency_l() const 1389{ 1390 if (initCheck() == NO_ERROR) { 1391 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1392 } else { 1393 return 0; 1394 } 1395} 1396 1397void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1398{ 1399 Mutex::Autolock _l(mLock); 1400 // Don't apply master volume in SW if our HAL can do it for us. 1401 if (mOutput && mOutput->audioHwDev && 1402 mOutput->audioHwDev->canSetMasterVolume()) { 1403 mMasterVolume = 1.0; 1404 } else { 1405 mMasterVolume = value; 1406 } 1407} 1408 1409void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1410{ 1411 Mutex::Autolock _l(mLock); 1412 // Don't apply master mute in SW if our HAL can do it for us. 1413 if (mOutput && mOutput->audioHwDev && 1414 mOutput->audioHwDev->canSetMasterMute()) { 1415 mMasterMute = false; 1416 } else { 1417 mMasterMute = muted; 1418 } 1419} 1420 1421void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1422{ 1423 Mutex::Autolock _l(mLock); 1424 mStreamTypes[stream].volume = value; 1425 broadcast_l(); 1426} 1427 1428void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1429{ 1430 Mutex::Autolock _l(mLock); 1431 mStreamTypes[stream].mute = muted; 1432 broadcast_l(); 1433} 1434 1435float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1436{ 1437 Mutex::Autolock _l(mLock); 1438 return mStreamTypes[stream].volume; 1439} 1440 1441// addTrack_l() must be called with ThreadBase::mLock held 1442status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1443{ 1444 status_t status = ALREADY_EXISTS; 1445 1446 // set retry count for buffer fill 1447 track->mRetryCount = kMaxTrackStartupRetries; 1448 if (mActiveTracks.indexOf(track) < 0) { 1449 // the track is newly added, make sure it fills up all its 1450 // buffers before playing. This is to ensure the client will 1451 // effectively get the latency it requested. 1452 if (!track->isOutputTrack()) { 1453 TrackBase::track_state state = track->mState; 1454 mLock.unlock(); 1455 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1456 mLock.lock(); 1457 // abort track was stopped/paused while we released the lock 1458 if (state != track->mState) { 1459 if (status == NO_ERROR) { 1460 mLock.unlock(); 1461 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1462 mLock.lock(); 1463 } 1464 return INVALID_OPERATION; 1465 } 1466 // abort if start is rejected by audio policy manager 1467 if (status != NO_ERROR) { 1468 return PERMISSION_DENIED; 1469 } 1470#ifdef ADD_BATTERY_DATA 1471 // to track the speaker usage 1472 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1473#endif 1474 } 1475 1476 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1477 track->mResetDone = false; 1478 track->mPresentationCompleteFrames = 0; 1479 mActiveTracks.add(track); 1480 mWakeLockUids.add(track->uid()); 1481 mActiveTracksGeneration++; 1482 mLatestActiveTrack = track; 1483 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1484 if (chain != 0) { 1485 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1486 track->sessionId()); 1487 chain->incActiveTrackCnt(); 1488 } 1489 1490 status = NO_ERROR; 1491 } 1492 1493 ALOGV("signal playback thread"); 1494 broadcast_l(); 1495 1496 return status; 1497} 1498 1499bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1500{ 1501 track->terminate(); 1502 // active tracks are removed by threadLoop() 1503 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1504 track->mState = TrackBase::STOPPED; 1505 if (!trackActive) { 1506 removeTrack_l(track); 1507 } else if (track->isFastTrack() || track->isOffloaded()) { 1508 track->mState = TrackBase::STOPPING_1; 1509 } 1510 1511 return trackActive; 1512} 1513 1514void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1515{ 1516 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1517 mTracks.remove(track); 1518 deleteTrackName_l(track->name()); 1519 // redundant as track is about to be destroyed, for dumpsys only 1520 track->mName = -1; 1521 if (track->isFastTrack()) { 1522 int index = track->mFastIndex; 1523 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1524 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1525 mFastTrackAvailMask |= 1 << index; 1526 // redundant as track is about to be destroyed, for dumpsys only 1527 track->mFastIndex = -1; 1528 } 1529 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1530 if (chain != 0) { 1531 chain->decTrackCnt(); 1532 } 1533} 1534 1535void AudioFlinger::PlaybackThread::broadcast_l() 1536{ 1537 // Thread could be blocked waiting for async 1538 // so signal it to handle state changes immediately 1539 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1540 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1541 mSignalPending = true; 1542 mWaitWorkCV.broadcast(); 1543} 1544 1545String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1546{ 1547 Mutex::Autolock _l(mLock); 1548 if (initCheck() != NO_ERROR) { 1549 return String8(); 1550 } 1551 1552 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1553 const String8 out_s8(s); 1554 free(s); 1555 return out_s8; 1556} 1557 1558// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1559void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1560 AudioSystem::OutputDescriptor desc; 1561 void *param2 = NULL; 1562 1563 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1564 param); 1565 1566 switch (event) { 1567 case AudioSystem::OUTPUT_OPENED: 1568 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1569 desc.channelMask = mChannelMask; 1570 desc.samplingRate = mSampleRate; 1571 desc.format = mFormat; 1572 desc.frameCount = mNormalFrameCount; // FIXME see 1573 // AudioFlinger::frameCount(audio_io_handle_t) 1574 desc.latency = latency(); 1575 param2 = &desc; 1576 break; 1577 1578 case AudioSystem::STREAM_CONFIG_CHANGED: 1579 param2 = ¶m; 1580 case AudioSystem::OUTPUT_CLOSED: 1581 default: 1582 break; 1583 } 1584 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1585} 1586 1587void AudioFlinger::PlaybackThread::writeCallback() 1588{ 1589 ALOG_ASSERT(mCallbackThread != 0); 1590 mCallbackThread->resetWriteBlocked(); 1591} 1592 1593void AudioFlinger::PlaybackThread::drainCallback() 1594{ 1595 ALOG_ASSERT(mCallbackThread != 0); 1596 mCallbackThread->resetDraining(); 1597} 1598 1599void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1600{ 1601 Mutex::Autolock _l(mLock); 1602 // reject out of sequence requests 1603 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1604 mWriteAckSequence &= ~1; 1605 mWaitWorkCV.signal(); 1606 } 1607} 1608 1609void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1610{ 1611 Mutex::Autolock _l(mLock); 1612 // reject out of sequence requests 1613 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1614 mDrainSequence &= ~1; 1615 mWaitWorkCV.signal(); 1616 } 1617} 1618 1619// static 1620int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1621 void *param __unused, 1622 void *cookie) 1623{ 1624 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1625 ALOGV("asyncCallback() event %d", event); 1626 switch (event) { 1627 case STREAM_CBK_EVENT_WRITE_READY: 1628 me->writeCallback(); 1629 break; 1630 case STREAM_CBK_EVENT_DRAIN_READY: 1631 me->drainCallback(); 1632 break; 1633 default: 1634 ALOGW("asyncCallback() unknown event %d", event); 1635 break; 1636 } 1637 return 0; 1638} 1639 1640void AudioFlinger::PlaybackThread::readOutputParameters() 1641{ 1642 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1643 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1644 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1645 if (!audio_is_output_channel(mChannelMask)) { 1646 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1647 } 1648 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1649 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1650 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1651 } 1652 mChannelCount = popcount(mChannelMask); 1653 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1654 if (!audio_is_valid_format(mFormat)) { 1655 LOG_FATAL("HAL format %d not valid for output", mFormat); 1656 } 1657 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1658 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1659 mFormat); 1660 } 1661 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1662 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1663 mFrameCount = mBufferSize / mFrameSize; 1664 if (mFrameCount & 15) { 1665 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1666 mFrameCount); 1667 } 1668 1669 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1670 (mOutput->stream->set_callback != NULL)) { 1671 if (mOutput->stream->set_callback(mOutput->stream, 1672 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1673 mUseAsyncWrite = true; 1674 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1675 } 1676 } 1677 1678 // Calculate size of normal mix buffer relative to the HAL output buffer size 1679 double multiplier = 1.0; 1680 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1681 kUseFastMixer == FastMixer_Dynamic)) { 1682 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1683 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1684 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1685 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1686 maxNormalFrameCount = maxNormalFrameCount & ~15; 1687 if (maxNormalFrameCount < minNormalFrameCount) { 1688 maxNormalFrameCount = minNormalFrameCount; 1689 } 1690 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1691 if (multiplier <= 1.0) { 1692 multiplier = 1.0; 1693 } else if (multiplier <= 2.0) { 1694 if (2 * mFrameCount <= maxNormalFrameCount) { 1695 multiplier = 2.0; 1696 } else { 1697 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1698 } 1699 } else { 1700 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1701 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1702 // track, but we sometimes have to do this to satisfy the maximum frame count 1703 // constraint) 1704 // FIXME this rounding up should not be done if no HAL SRC 1705 uint32_t truncMult = (uint32_t) multiplier; 1706 if ((truncMult & 1)) { 1707 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1708 ++truncMult; 1709 } 1710 } 1711 multiplier = (double) truncMult; 1712 } 1713 } 1714 mNormalFrameCount = multiplier * mFrameCount; 1715 // round up to nearest 16 frames to satisfy AudioMixer 1716 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1717 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1718 mNormalFrameCount); 1719 1720 delete[] mMixBuffer; 1721 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1722 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1723 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1724 memset(mMixBuffer, 0, normalBufferSize); 1725 1726 // force reconfiguration of effect chains and engines to take new buffer size and audio 1727 // parameters into account 1728 // Note that mLock is not held when readOutputParameters() is called from the constructor 1729 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1730 // matter. 1731 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1732 Vector< sp<EffectChain> > effectChains = mEffectChains; 1733 for (size_t i = 0; i < effectChains.size(); i ++) { 1734 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1735 } 1736} 1737 1738 1739status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1740{ 1741 if (halFrames == NULL || dspFrames == NULL) { 1742 return BAD_VALUE; 1743 } 1744 Mutex::Autolock _l(mLock); 1745 if (initCheck() != NO_ERROR) { 1746 return INVALID_OPERATION; 1747 } 1748 size_t framesWritten = mBytesWritten / mFrameSize; 1749 *halFrames = framesWritten; 1750 1751 if (isSuspended()) { 1752 // return an estimation of rendered frames when the output is suspended 1753 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1754 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1755 return NO_ERROR; 1756 } else { 1757 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1758 } 1759} 1760 1761uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1762{ 1763 Mutex::Autolock _l(mLock); 1764 uint32_t result = 0; 1765 if (getEffectChain_l(sessionId) != 0) { 1766 result = EFFECT_SESSION; 1767 } 1768 1769 for (size_t i = 0; i < mTracks.size(); ++i) { 1770 sp<Track> track = mTracks[i]; 1771 if (sessionId == track->sessionId() && !track->isInvalid()) { 1772 result |= TRACK_SESSION; 1773 break; 1774 } 1775 } 1776 1777 return result; 1778} 1779 1780uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1781{ 1782 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1783 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1784 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1785 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1786 } 1787 for (size_t i = 0; i < mTracks.size(); i++) { 1788 sp<Track> track = mTracks[i]; 1789 if (sessionId == track->sessionId() && !track->isInvalid()) { 1790 return AudioSystem::getStrategyForStream(track->streamType()); 1791 } 1792 } 1793 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1794} 1795 1796 1797AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1798{ 1799 Mutex::Autolock _l(mLock); 1800 return mOutput; 1801} 1802 1803AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1804{ 1805 Mutex::Autolock _l(mLock); 1806 AudioStreamOut *output = mOutput; 1807 mOutput = NULL; 1808 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1809 // must push a NULL and wait for ack 1810 mOutputSink.clear(); 1811 mPipeSink.clear(); 1812 mNormalSink.clear(); 1813 return output; 1814} 1815 1816// this method must always be called either with ThreadBase mLock held or inside the thread loop 1817audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1818{ 1819 if (mOutput == NULL) { 1820 return NULL; 1821 } 1822 return &mOutput->stream->common; 1823} 1824 1825uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1826{ 1827 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1828} 1829 1830status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1831{ 1832 if (!isValidSyncEvent(event)) { 1833 return BAD_VALUE; 1834 } 1835 1836 Mutex::Autolock _l(mLock); 1837 1838 for (size_t i = 0; i < mTracks.size(); ++i) { 1839 sp<Track> track = mTracks[i]; 1840 if (event->triggerSession() == track->sessionId()) { 1841 (void) track->setSyncEvent(event); 1842 return NO_ERROR; 1843 } 1844 } 1845 1846 return NAME_NOT_FOUND; 1847} 1848 1849bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1850{ 1851 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1852} 1853 1854void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1855 const Vector< sp<Track> >& tracksToRemove) 1856{ 1857 size_t count = tracksToRemove.size(); 1858 if (count > 0) { 1859 for (size_t i = 0 ; i < count ; i++) { 1860 const sp<Track>& track = tracksToRemove.itemAt(i); 1861 if (!track->isOutputTrack()) { 1862 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1863#ifdef ADD_BATTERY_DATA 1864 // to track the speaker usage 1865 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1866#endif 1867 if (track->isTerminated()) { 1868 AudioSystem::releaseOutput(mId); 1869 } 1870 } 1871 } 1872 } 1873} 1874 1875void AudioFlinger::PlaybackThread::checkSilentMode_l() 1876{ 1877 if (!mMasterMute) { 1878 char value[PROPERTY_VALUE_MAX]; 1879 if (property_get("ro.audio.silent", value, "0") > 0) { 1880 char *endptr; 1881 unsigned long ul = strtoul(value, &endptr, 0); 1882 if (*endptr == '\0' && ul != 0) { 1883 ALOGD("Silence is golden"); 1884 // The setprop command will not allow a property to be changed after 1885 // the first time it is set, so we don't have to worry about un-muting. 1886 setMasterMute_l(true); 1887 } 1888 } 1889 } 1890} 1891 1892// shared by MIXER and DIRECT, overridden by DUPLICATING 1893ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1894{ 1895 // FIXME rewrite to reduce number of system calls 1896 mLastWriteTime = systemTime(); 1897 mInWrite = true; 1898 ssize_t bytesWritten; 1899 1900 // If an NBAIO sink is present, use it to write the normal mixer's submix 1901 if (mNormalSink != 0) { 1902#define mBitShift 2 // FIXME 1903 size_t count = mBytesRemaining >> mBitShift; 1904 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1905 ATRACE_BEGIN("write"); 1906 // update the setpoint when AudioFlinger::mScreenState changes 1907 uint32_t screenState = AudioFlinger::mScreenState; 1908 if (screenState != mScreenState) { 1909 mScreenState = screenState; 1910 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1911 if (pipe != NULL) { 1912 pipe->setAvgFrames((mScreenState & 1) ? 1913 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1914 } 1915 } 1916 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1917 ATRACE_END(); 1918 if (framesWritten > 0) { 1919 bytesWritten = framesWritten << mBitShift; 1920 } else { 1921 bytesWritten = framesWritten; 1922 } 1923 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1924 if (status == NO_ERROR) { 1925 size_t totalFramesWritten = mNormalSink->framesWritten(); 1926 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1927 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1928 mLatchDValid = true; 1929 } 1930 } 1931 // otherwise use the HAL / AudioStreamOut directly 1932 } else { 1933 // Direct output and offload threads 1934 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1935 if (mUseAsyncWrite) { 1936 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1937 mWriteAckSequence += 2; 1938 mWriteAckSequence |= 1; 1939 ALOG_ASSERT(mCallbackThread != 0); 1940 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1941 } 1942 // FIXME We should have an implementation of timestamps for direct output threads. 1943 // They are used e.g for multichannel PCM playback over HDMI. 1944 bytesWritten = mOutput->stream->write(mOutput->stream, 1945 (char *)mMixBuffer + offset, mBytesRemaining); 1946 if (mUseAsyncWrite && 1947 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1948 // do not wait for async callback in case of error of full write 1949 mWriteAckSequence &= ~1; 1950 ALOG_ASSERT(mCallbackThread != 0); 1951 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1952 } 1953 } 1954 1955 mNumWrites++; 1956 mInWrite = false; 1957 mStandby = false; 1958 return bytesWritten; 1959} 1960 1961void AudioFlinger::PlaybackThread::threadLoop_drain() 1962{ 1963 if (mOutput->stream->drain) { 1964 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1965 if (mUseAsyncWrite) { 1966 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1967 mDrainSequence |= 1; 1968 ALOG_ASSERT(mCallbackThread != 0); 1969 mCallbackThread->setDraining(mDrainSequence); 1970 } 1971 mOutput->stream->drain(mOutput->stream, 1972 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1973 : AUDIO_DRAIN_ALL); 1974 } 1975} 1976 1977void AudioFlinger::PlaybackThread::threadLoop_exit() 1978{ 1979 // Default implementation has nothing to do 1980} 1981 1982/* 1983The derived values that are cached: 1984 - mixBufferSize from frame count * frame size 1985 - activeSleepTime from activeSleepTimeUs() 1986 - idleSleepTime from idleSleepTimeUs() 1987 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1988 - maxPeriod from frame count and sample rate (MIXER only) 1989 1990The parameters that affect these derived values are: 1991 - frame count 1992 - frame size 1993 - sample rate 1994 - device type: A2DP or not 1995 - device latency 1996 - format: PCM or not 1997 - active sleep time 1998 - idle sleep time 1999*/ 2000 2001void AudioFlinger::PlaybackThread::cacheParameters_l() 2002{ 2003 mixBufferSize = mNormalFrameCount * mFrameSize; 2004 activeSleepTime = activeSleepTimeUs(); 2005 idleSleepTime = idleSleepTimeUs(); 2006} 2007 2008void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2009{ 2010 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2011 this, streamType, mTracks.size()); 2012 Mutex::Autolock _l(mLock); 2013 2014 size_t size = mTracks.size(); 2015 for (size_t i = 0; i < size; i++) { 2016 sp<Track> t = mTracks[i]; 2017 if (t->streamType() == streamType) { 2018 t->invalidate(); 2019 } 2020 } 2021} 2022 2023status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2024{ 2025 int session = chain->sessionId(); 2026 int16_t *buffer = mMixBuffer; 2027 bool ownsBuffer = false; 2028 2029 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2030 if (session > 0) { 2031 // Only one effect chain can be present in direct output thread and it uses 2032 // the mix buffer as input 2033 if (mType != DIRECT) { 2034 size_t numSamples = mNormalFrameCount * mChannelCount; 2035 buffer = new int16_t[numSamples]; 2036 memset(buffer, 0, numSamples * sizeof(int16_t)); 2037 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2038 ownsBuffer = true; 2039 } 2040 2041 // Attach all tracks with same session ID to this chain. 2042 for (size_t i = 0; i < mTracks.size(); ++i) { 2043 sp<Track> track = mTracks[i]; 2044 if (session == track->sessionId()) { 2045 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2046 buffer); 2047 track->setMainBuffer(buffer); 2048 chain->incTrackCnt(); 2049 } 2050 } 2051 2052 // indicate all active tracks in the chain 2053 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2054 sp<Track> track = mActiveTracks[i].promote(); 2055 if (track == 0) { 2056 continue; 2057 } 2058 if (session == track->sessionId()) { 2059 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2060 chain->incActiveTrackCnt(); 2061 } 2062 } 2063 } 2064 2065 chain->setInBuffer(buffer, ownsBuffer); 2066 chain->setOutBuffer(mMixBuffer); 2067 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2068 // chains list in order to be processed last as it contains output stage effects 2069 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2070 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2071 // after track specific effects and before output stage 2072 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2073 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2074 // Effect chain for other sessions are inserted at beginning of effect 2075 // chains list to be processed before output mix effects. Relative order between other 2076 // sessions is not important 2077 size_t size = mEffectChains.size(); 2078 size_t i = 0; 2079 for (i = 0; i < size; i++) { 2080 if (mEffectChains[i]->sessionId() < session) { 2081 break; 2082 } 2083 } 2084 mEffectChains.insertAt(chain, i); 2085 checkSuspendOnAddEffectChain_l(chain); 2086 2087 return NO_ERROR; 2088} 2089 2090size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2091{ 2092 int session = chain->sessionId(); 2093 2094 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2095 2096 for (size_t i = 0; i < mEffectChains.size(); i++) { 2097 if (chain == mEffectChains[i]) { 2098 mEffectChains.removeAt(i); 2099 // detach all active tracks from the chain 2100 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2101 sp<Track> track = mActiveTracks[i].promote(); 2102 if (track == 0) { 2103 continue; 2104 } 2105 if (session == track->sessionId()) { 2106 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2107 chain.get(), session); 2108 chain->decActiveTrackCnt(); 2109 } 2110 } 2111 2112 // detach all tracks with same session ID from this chain 2113 for (size_t i = 0; i < mTracks.size(); ++i) { 2114 sp<Track> track = mTracks[i]; 2115 if (session == track->sessionId()) { 2116 track->setMainBuffer(mMixBuffer); 2117 chain->decTrackCnt(); 2118 } 2119 } 2120 break; 2121 } 2122 } 2123 return mEffectChains.size(); 2124} 2125 2126status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2127 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2128{ 2129 Mutex::Autolock _l(mLock); 2130 return attachAuxEffect_l(track, EffectId); 2131} 2132 2133status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2134 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2135{ 2136 status_t status = NO_ERROR; 2137 2138 if (EffectId == 0) { 2139 track->setAuxBuffer(0, NULL); 2140 } else { 2141 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2142 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2143 if (effect != 0) { 2144 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2145 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2146 } else { 2147 status = INVALID_OPERATION; 2148 } 2149 } else { 2150 status = BAD_VALUE; 2151 } 2152 } 2153 return status; 2154} 2155 2156void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2157{ 2158 for (size_t i = 0; i < mTracks.size(); ++i) { 2159 sp<Track> track = mTracks[i]; 2160 if (track->auxEffectId() == effectId) { 2161 attachAuxEffect_l(track, 0); 2162 } 2163 } 2164} 2165 2166bool AudioFlinger::PlaybackThread::threadLoop() 2167{ 2168 Vector< sp<Track> > tracksToRemove; 2169 2170 standbyTime = systemTime(); 2171 2172 // MIXER 2173 nsecs_t lastWarning = 0; 2174 2175 // DUPLICATING 2176 // FIXME could this be made local to while loop? 2177 writeFrames = 0; 2178 2179 int lastGeneration = 0; 2180 2181 cacheParameters_l(); 2182 sleepTime = idleSleepTime; 2183 2184 if (mType == MIXER) { 2185 sleepTimeShift = 0; 2186 } 2187 2188 CpuStats cpuStats; 2189 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2190 2191 acquireWakeLock(); 2192 2193 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2194 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2195 // and then that string will be logged at the next convenient opportunity. 2196 const char *logString = NULL; 2197 2198 checkSilentMode_l(); 2199 2200 while (!exitPending()) 2201 { 2202 cpuStats.sample(myName); 2203 2204 Vector< sp<EffectChain> > effectChains; 2205 2206 processConfigEvents(); 2207 2208 { // scope for mLock 2209 2210 Mutex::Autolock _l(mLock); 2211 2212 if (logString != NULL) { 2213 mNBLogWriter->logTimestamp(); 2214 mNBLogWriter->log(logString); 2215 logString = NULL; 2216 } 2217 2218 if (mLatchDValid) { 2219 mLatchQ = mLatchD; 2220 mLatchDValid = false; 2221 mLatchQValid = true; 2222 } 2223 2224 if (checkForNewParameters_l()) { 2225 cacheParameters_l(); 2226 } 2227 2228 saveOutputTracks(); 2229 if (mSignalPending) { 2230 // A signal was raised while we were unlocked 2231 mSignalPending = false; 2232 } else if (waitingAsyncCallback_l()) { 2233 if (exitPending()) { 2234 break; 2235 } 2236 releaseWakeLock_l(); 2237 mWakeLockUids.clear(); 2238 mActiveTracksGeneration++; 2239 ALOGV("wait async completion"); 2240 mWaitWorkCV.wait(mLock); 2241 ALOGV("async completion/wake"); 2242 acquireWakeLock_l(); 2243 standbyTime = systemTime() + standbyDelay; 2244 sleepTime = 0; 2245 2246 continue; 2247 } 2248 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2249 isSuspended()) { 2250 // put audio hardware into standby after short delay 2251 if (shouldStandby_l()) { 2252 2253 threadLoop_standby(); 2254 2255 mStandby = true; 2256 } 2257 2258 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2259 // we're about to wait, flush the binder command buffer 2260 IPCThreadState::self()->flushCommands(); 2261 2262 clearOutputTracks(); 2263 2264 if (exitPending()) { 2265 break; 2266 } 2267 2268 releaseWakeLock_l(); 2269 mWakeLockUids.clear(); 2270 mActiveTracksGeneration++; 2271 // wait until we have something to do... 2272 ALOGV("%s going to sleep", myName.string()); 2273 mWaitWorkCV.wait(mLock); 2274 ALOGV("%s waking up", myName.string()); 2275 acquireWakeLock_l(); 2276 2277 mMixerStatus = MIXER_IDLE; 2278 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2279 mBytesWritten = 0; 2280 mBytesRemaining = 0; 2281 checkSilentMode_l(); 2282 2283 standbyTime = systemTime() + standbyDelay; 2284 sleepTime = idleSleepTime; 2285 if (mType == MIXER) { 2286 sleepTimeShift = 0; 2287 } 2288 2289 continue; 2290 } 2291 } 2292 // mMixerStatusIgnoringFastTracks is also updated internally 2293 mMixerStatus = prepareTracks_l(&tracksToRemove); 2294 2295 // compare with previously applied list 2296 if (lastGeneration != mActiveTracksGeneration) { 2297 // update wakelock 2298 updateWakeLockUids_l(mWakeLockUids); 2299 lastGeneration = mActiveTracksGeneration; 2300 } 2301 2302 // prevent any changes in effect chain list and in each effect chain 2303 // during mixing and effect process as the audio buffers could be deleted 2304 // or modified if an effect is created or deleted 2305 lockEffectChains_l(effectChains); 2306 } // mLock scope ends 2307 2308 if (mBytesRemaining == 0) { 2309 mCurrentWriteLength = 0; 2310 if (mMixerStatus == MIXER_TRACKS_READY) { 2311 // threadLoop_mix() sets mCurrentWriteLength 2312 threadLoop_mix(); 2313 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2314 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2315 // threadLoop_sleepTime sets sleepTime to 0 if data 2316 // must be written to HAL 2317 threadLoop_sleepTime(); 2318 if (sleepTime == 0) { 2319 mCurrentWriteLength = mixBufferSize; 2320 } 2321 } 2322 mBytesRemaining = mCurrentWriteLength; 2323 if (isSuspended()) { 2324 sleepTime = suspendSleepTimeUs(); 2325 // simulate write to HAL when suspended 2326 mBytesWritten += mixBufferSize; 2327 mBytesRemaining = 0; 2328 } 2329 2330 // only process effects if we're going to write 2331 if (sleepTime == 0 && mType != OFFLOAD) { 2332 for (size_t i = 0; i < effectChains.size(); i ++) { 2333 effectChains[i]->process_l(); 2334 } 2335 } 2336 } 2337 // Process effect chains for offloaded thread even if no audio 2338 // was read from audio track: process only updates effect state 2339 // and thus does have to be synchronized with audio writes but may have 2340 // to be called while waiting for async write callback 2341 if (mType == OFFLOAD) { 2342 for (size_t i = 0; i < effectChains.size(); i ++) { 2343 effectChains[i]->process_l(); 2344 } 2345 } 2346 2347 // enable changes in effect chain 2348 unlockEffectChains(effectChains); 2349 2350 if (!waitingAsyncCallback()) { 2351 // sleepTime == 0 means we must write to audio hardware 2352 if (sleepTime == 0) { 2353 if (mBytesRemaining) { 2354 ssize_t ret = threadLoop_write(); 2355 if (ret < 0) { 2356 mBytesRemaining = 0; 2357 } else { 2358 mBytesWritten += ret; 2359 mBytesRemaining -= ret; 2360 } 2361 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2362 (mMixerStatus == MIXER_DRAIN_ALL)) { 2363 threadLoop_drain(); 2364 } 2365if (mType == MIXER) { 2366 // write blocked detection 2367 nsecs_t now = systemTime(); 2368 nsecs_t delta = now - mLastWriteTime; 2369 if (!mStandby && delta > maxPeriod) { 2370 mNumDelayedWrites++; 2371 if ((now - lastWarning) > kWarningThrottleNs) { 2372 ATRACE_NAME("underrun"); 2373 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2374 ns2ms(delta), mNumDelayedWrites, this); 2375 lastWarning = now; 2376 } 2377 } 2378} 2379 2380 } else { 2381 usleep(sleepTime); 2382 } 2383 } 2384 2385 // Finally let go of removed track(s), without the lock held 2386 // since we can't guarantee the destructors won't acquire that 2387 // same lock. This will also mutate and push a new fast mixer state. 2388 threadLoop_removeTracks(tracksToRemove); 2389 tracksToRemove.clear(); 2390 2391 // FIXME I don't understand the need for this here; 2392 // it was in the original code but maybe the 2393 // assignment in saveOutputTracks() makes this unnecessary? 2394 clearOutputTracks(); 2395 2396 // Effect chains will be actually deleted here if they were removed from 2397 // mEffectChains list during mixing or effects processing 2398 effectChains.clear(); 2399 2400 // FIXME Note that the above .clear() is no longer necessary since effectChains 2401 // is now local to this block, but will keep it for now (at least until merge done). 2402 } 2403 2404 threadLoop_exit(); 2405 2406 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2407 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2408 // put output stream into standby mode 2409 if (!mStandby) { 2410 mOutput->stream->common.standby(&mOutput->stream->common); 2411 } 2412 } 2413 2414 releaseWakeLock(); 2415 mWakeLockUids.clear(); 2416 mActiveTracksGeneration++; 2417 2418 ALOGV("Thread %p type %d exiting", this, mType); 2419 return false; 2420} 2421 2422// removeTracks_l() must be called with ThreadBase::mLock held 2423void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2424{ 2425 size_t count = tracksToRemove.size(); 2426 if (count > 0) { 2427 for (size_t i=0 ; i<count ; i++) { 2428 const sp<Track>& track = tracksToRemove.itemAt(i); 2429 mActiveTracks.remove(track); 2430 mWakeLockUids.remove(track->uid()); 2431 mActiveTracksGeneration++; 2432 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2433 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2434 if (chain != 0) { 2435 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2436 track->sessionId()); 2437 chain->decActiveTrackCnt(); 2438 } 2439 if (track->isTerminated()) { 2440 removeTrack_l(track); 2441 } 2442 } 2443 } 2444 2445} 2446 2447status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2448{ 2449 if (mNormalSink != 0) { 2450 return mNormalSink->getTimestamp(timestamp); 2451 } 2452 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2453 uint64_t position64; 2454 int ret = mOutput->stream->get_presentation_position( 2455 mOutput->stream, &position64, ×tamp.mTime); 2456 if (ret == 0) { 2457 timestamp.mPosition = (uint32_t)position64; 2458 return NO_ERROR; 2459 } 2460 } 2461 return INVALID_OPERATION; 2462} 2463// ---------------------------------------------------------------------------- 2464 2465AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2466 audio_io_handle_t id, audio_devices_t device, type_t type) 2467 : PlaybackThread(audioFlinger, output, id, device, type), 2468 // mAudioMixer below 2469 // mFastMixer below 2470 mFastMixerFutex(0) 2471 // mOutputSink below 2472 // mPipeSink below 2473 // mNormalSink below 2474{ 2475 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2476 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2477 "mFrameCount=%d, mNormalFrameCount=%d", 2478 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2479 mNormalFrameCount); 2480 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2481 2482 // FIXME - Current mixer implementation only supports stereo output 2483 if (mChannelCount != FCC_2) { 2484 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2485 } 2486 2487 // create an NBAIO sink for the HAL output stream, and negotiate 2488 mOutputSink = new AudioStreamOutSink(output->stream); 2489 size_t numCounterOffers = 0; 2490 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2491 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2492 ALOG_ASSERT(index == 0); 2493 2494 // initialize fast mixer depending on configuration 2495 bool initFastMixer; 2496 switch (kUseFastMixer) { 2497 case FastMixer_Never: 2498 initFastMixer = false; 2499 break; 2500 case FastMixer_Always: 2501 initFastMixer = true; 2502 break; 2503 case FastMixer_Static: 2504 case FastMixer_Dynamic: 2505 initFastMixer = mFrameCount < mNormalFrameCount; 2506 break; 2507 } 2508 if (initFastMixer) { 2509 2510 // create a MonoPipe to connect our submix to FastMixer 2511 NBAIO_Format format = mOutputSink->format(); 2512 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2513 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2514 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2515 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2516 const NBAIO_Format offers[1] = {format}; 2517 size_t numCounterOffers = 0; 2518 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2519 ALOG_ASSERT(index == 0); 2520 monoPipe->setAvgFrames((mScreenState & 1) ? 2521 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2522 mPipeSink = monoPipe; 2523 2524#ifdef TEE_SINK 2525 if (mTeeSinkOutputEnabled) { 2526 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2527 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2528 numCounterOffers = 0; 2529 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2530 ALOG_ASSERT(index == 0); 2531 mTeeSink = teeSink; 2532 PipeReader *teeSource = new PipeReader(*teeSink); 2533 numCounterOffers = 0; 2534 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2535 ALOG_ASSERT(index == 0); 2536 mTeeSource = teeSource; 2537 } 2538#endif 2539 2540 // create fast mixer and configure it initially with just one fast track for our submix 2541 mFastMixer = new FastMixer(); 2542 FastMixerStateQueue *sq = mFastMixer->sq(); 2543#ifdef STATE_QUEUE_DUMP 2544 sq->setObserverDump(&mStateQueueObserverDump); 2545 sq->setMutatorDump(&mStateQueueMutatorDump); 2546#endif 2547 FastMixerState *state = sq->begin(); 2548 FastTrack *fastTrack = &state->mFastTracks[0]; 2549 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2550 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2551 fastTrack->mVolumeProvider = NULL; 2552 fastTrack->mGeneration++; 2553 state->mFastTracksGen++; 2554 state->mTrackMask = 1; 2555 // fast mixer will use the HAL output sink 2556 state->mOutputSink = mOutputSink.get(); 2557 state->mOutputSinkGen++; 2558 state->mFrameCount = mFrameCount; 2559 state->mCommand = FastMixerState::COLD_IDLE; 2560 // already done in constructor initialization list 2561 //mFastMixerFutex = 0; 2562 state->mColdFutexAddr = &mFastMixerFutex; 2563 state->mColdGen++; 2564 state->mDumpState = &mFastMixerDumpState; 2565#ifdef TEE_SINK 2566 state->mTeeSink = mTeeSink.get(); 2567#endif 2568 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2569 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2570 sq->end(); 2571 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2572 2573 // start the fast mixer 2574 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2575 pid_t tid = mFastMixer->getTid(); 2576 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2577 if (err != 0) { 2578 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2579 kPriorityFastMixer, getpid_cached, tid, err); 2580 } 2581 2582#ifdef AUDIO_WATCHDOG 2583 // create and start the watchdog 2584 mAudioWatchdog = new AudioWatchdog(); 2585 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2586 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2587 tid = mAudioWatchdog->getTid(); 2588 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2589 if (err != 0) { 2590 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2591 kPriorityFastMixer, getpid_cached, tid, err); 2592 } 2593#endif 2594 2595 } else { 2596 mFastMixer = NULL; 2597 } 2598 2599 switch (kUseFastMixer) { 2600 case FastMixer_Never: 2601 case FastMixer_Dynamic: 2602 mNormalSink = mOutputSink; 2603 break; 2604 case FastMixer_Always: 2605 mNormalSink = mPipeSink; 2606 break; 2607 case FastMixer_Static: 2608 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2609 break; 2610 } 2611} 2612 2613AudioFlinger::MixerThread::~MixerThread() 2614{ 2615 if (mFastMixer != NULL) { 2616 FastMixerStateQueue *sq = mFastMixer->sq(); 2617 FastMixerState *state = sq->begin(); 2618 if (state->mCommand == FastMixerState::COLD_IDLE) { 2619 int32_t old = android_atomic_inc(&mFastMixerFutex); 2620 if (old == -1) { 2621 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2622 } 2623 } 2624 state->mCommand = FastMixerState::EXIT; 2625 sq->end(); 2626 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2627 mFastMixer->join(); 2628 // Though the fast mixer thread has exited, it's state queue is still valid. 2629 // We'll use that extract the final state which contains one remaining fast track 2630 // corresponding to our sub-mix. 2631 state = sq->begin(); 2632 ALOG_ASSERT(state->mTrackMask == 1); 2633 FastTrack *fastTrack = &state->mFastTracks[0]; 2634 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2635 delete fastTrack->mBufferProvider; 2636 sq->end(false /*didModify*/); 2637 delete mFastMixer; 2638#ifdef AUDIO_WATCHDOG 2639 if (mAudioWatchdog != 0) { 2640 mAudioWatchdog->requestExit(); 2641 mAudioWatchdog->requestExitAndWait(); 2642 mAudioWatchdog.clear(); 2643 } 2644#endif 2645 } 2646 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2647 delete mAudioMixer; 2648} 2649 2650 2651uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2652{ 2653 if (mFastMixer != NULL) { 2654 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2655 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2656 } 2657 return latency; 2658} 2659 2660 2661void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2662{ 2663 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2664} 2665 2666ssize_t AudioFlinger::MixerThread::threadLoop_write() 2667{ 2668 // FIXME we should only do one push per cycle; confirm this is true 2669 // Start the fast mixer if it's not already running 2670 if (mFastMixer != NULL) { 2671 FastMixerStateQueue *sq = mFastMixer->sq(); 2672 FastMixerState *state = sq->begin(); 2673 if (state->mCommand != FastMixerState::MIX_WRITE && 2674 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2675 if (state->mCommand == FastMixerState::COLD_IDLE) { 2676 int32_t old = android_atomic_inc(&mFastMixerFutex); 2677 if (old == -1) { 2678 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2679 } 2680#ifdef AUDIO_WATCHDOG 2681 if (mAudioWatchdog != 0) { 2682 mAudioWatchdog->resume(); 2683 } 2684#endif 2685 } 2686 state->mCommand = FastMixerState::MIX_WRITE; 2687 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2688 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2689 sq->end(); 2690 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2691 if (kUseFastMixer == FastMixer_Dynamic) { 2692 mNormalSink = mPipeSink; 2693 } 2694 } else { 2695 sq->end(false /*didModify*/); 2696 } 2697 } 2698 return PlaybackThread::threadLoop_write(); 2699} 2700 2701void AudioFlinger::MixerThread::threadLoop_standby() 2702{ 2703 // Idle the fast mixer if it's currently running 2704 if (mFastMixer != NULL) { 2705 FastMixerStateQueue *sq = mFastMixer->sq(); 2706 FastMixerState *state = sq->begin(); 2707 if (!(state->mCommand & FastMixerState::IDLE)) { 2708 state->mCommand = FastMixerState::COLD_IDLE; 2709 state->mColdFutexAddr = &mFastMixerFutex; 2710 state->mColdGen++; 2711 mFastMixerFutex = 0; 2712 sq->end(); 2713 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2714 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2715 if (kUseFastMixer == FastMixer_Dynamic) { 2716 mNormalSink = mOutputSink; 2717 } 2718#ifdef AUDIO_WATCHDOG 2719 if (mAudioWatchdog != 0) { 2720 mAudioWatchdog->pause(); 2721 } 2722#endif 2723 } else { 2724 sq->end(false /*didModify*/); 2725 } 2726 } 2727 PlaybackThread::threadLoop_standby(); 2728} 2729 2730// Empty implementation for standard mixer 2731// Overridden for offloaded playback 2732void AudioFlinger::PlaybackThread::flushOutput_l() 2733{ 2734} 2735 2736bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2737{ 2738 return false; 2739} 2740 2741bool AudioFlinger::PlaybackThread::shouldStandby_l() 2742{ 2743 return !mStandby; 2744} 2745 2746bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2747{ 2748 Mutex::Autolock _l(mLock); 2749 return waitingAsyncCallback_l(); 2750} 2751 2752// shared by MIXER and DIRECT, overridden by DUPLICATING 2753void AudioFlinger::PlaybackThread::threadLoop_standby() 2754{ 2755 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2756 mOutput->stream->common.standby(&mOutput->stream->common); 2757 if (mUseAsyncWrite != 0) { 2758 // discard any pending drain or write ack by incrementing sequence 2759 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2760 mDrainSequence = (mDrainSequence + 2) & ~1; 2761 ALOG_ASSERT(mCallbackThread != 0); 2762 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2763 mCallbackThread->setDraining(mDrainSequence); 2764 } 2765} 2766 2767void AudioFlinger::MixerThread::threadLoop_mix() 2768{ 2769 // obtain the presentation timestamp of the next output buffer 2770 int64_t pts; 2771 status_t status = INVALID_OPERATION; 2772 2773 if (mNormalSink != 0) { 2774 status = mNormalSink->getNextWriteTimestamp(&pts); 2775 } else { 2776 status = mOutputSink->getNextWriteTimestamp(&pts); 2777 } 2778 2779 if (status != NO_ERROR) { 2780 pts = AudioBufferProvider::kInvalidPTS; 2781 } 2782 2783 // mix buffers... 2784 mAudioMixer->process(pts); 2785 mCurrentWriteLength = mixBufferSize; 2786 // increase sleep time progressively when application underrun condition clears. 2787 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2788 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2789 // such that we would underrun the audio HAL. 2790 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2791 sleepTimeShift--; 2792 } 2793 sleepTime = 0; 2794 standbyTime = systemTime() + standbyDelay; 2795 //TODO: delay standby when effects have a tail 2796} 2797 2798void AudioFlinger::MixerThread::threadLoop_sleepTime() 2799{ 2800 // If no tracks are ready, sleep once for the duration of an output 2801 // buffer size, then write 0s to the output 2802 if (sleepTime == 0) { 2803 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2804 sleepTime = activeSleepTime >> sleepTimeShift; 2805 if (sleepTime < kMinThreadSleepTimeUs) { 2806 sleepTime = kMinThreadSleepTimeUs; 2807 } 2808 // reduce sleep time in case of consecutive application underruns to avoid 2809 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2810 // duration we would end up writing less data than needed by the audio HAL if 2811 // the condition persists. 2812 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2813 sleepTimeShift++; 2814 } 2815 } else { 2816 sleepTime = idleSleepTime; 2817 } 2818 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2819 memset(mMixBuffer, 0, mixBufferSize); 2820 sleepTime = 0; 2821 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2822 "anticipated start"); 2823 } 2824 // TODO add standby time extension fct of effect tail 2825} 2826 2827// prepareTracks_l() must be called with ThreadBase::mLock held 2828AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2829 Vector< sp<Track> > *tracksToRemove) 2830{ 2831 2832 mixer_state mixerStatus = MIXER_IDLE; 2833 // find out which tracks need to be processed 2834 size_t count = mActiveTracks.size(); 2835 size_t mixedTracks = 0; 2836 size_t tracksWithEffect = 0; 2837 // counts only _active_ fast tracks 2838 size_t fastTracks = 0; 2839 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2840 2841 float masterVolume = mMasterVolume; 2842 bool masterMute = mMasterMute; 2843 2844 if (masterMute) { 2845 masterVolume = 0; 2846 } 2847 // Delegate master volume control to effect in output mix effect chain if needed 2848 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2849 if (chain != 0) { 2850 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2851 chain->setVolume_l(&v, &v); 2852 masterVolume = (float)((v + (1 << 23)) >> 24); 2853 chain.clear(); 2854 } 2855 2856 // prepare a new state to push 2857 FastMixerStateQueue *sq = NULL; 2858 FastMixerState *state = NULL; 2859 bool didModify = false; 2860 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2861 if (mFastMixer != NULL) { 2862 sq = mFastMixer->sq(); 2863 state = sq->begin(); 2864 } 2865 2866 for (size_t i=0 ; i<count ; i++) { 2867 const sp<Track> t = mActiveTracks[i].promote(); 2868 if (t == 0) { 2869 continue; 2870 } 2871 2872 // this const just means the local variable doesn't change 2873 Track* const track = t.get(); 2874 2875 // process fast tracks 2876 if (track->isFastTrack()) { 2877 2878 // It's theoretically possible (though unlikely) for a fast track to be created 2879 // and then removed within the same normal mix cycle. This is not a problem, as 2880 // the track never becomes active so it's fast mixer slot is never touched. 2881 // The converse, of removing an (active) track and then creating a new track 2882 // at the identical fast mixer slot within the same normal mix cycle, 2883 // is impossible because the slot isn't marked available until the end of each cycle. 2884 int j = track->mFastIndex; 2885 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2886 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2887 FastTrack *fastTrack = &state->mFastTracks[j]; 2888 2889 // Determine whether the track is currently in underrun condition, 2890 // and whether it had a recent underrun. 2891 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2892 FastTrackUnderruns underruns = ftDump->mUnderruns; 2893 uint32_t recentFull = (underruns.mBitFields.mFull - 2894 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2895 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2896 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2897 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2898 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2899 uint32_t recentUnderruns = recentPartial + recentEmpty; 2900 track->mObservedUnderruns = underruns; 2901 // don't count underruns that occur while stopping or pausing 2902 // or stopped which can occur when flush() is called while active 2903 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2904 recentUnderruns > 0) { 2905 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2906 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2907 } 2908 2909 // This is similar to the state machine for normal tracks, 2910 // with a few modifications for fast tracks. 2911 bool isActive = true; 2912 switch (track->mState) { 2913 case TrackBase::STOPPING_1: 2914 // track stays active in STOPPING_1 state until first underrun 2915 if (recentUnderruns > 0 || track->isTerminated()) { 2916 track->mState = TrackBase::STOPPING_2; 2917 } 2918 break; 2919 case TrackBase::PAUSING: 2920 // ramp down is not yet implemented 2921 track->setPaused(); 2922 break; 2923 case TrackBase::RESUMING: 2924 // ramp up is not yet implemented 2925 track->mState = TrackBase::ACTIVE; 2926 break; 2927 case TrackBase::ACTIVE: 2928 if (recentFull > 0 || recentPartial > 0) { 2929 // track has provided at least some frames recently: reset retry count 2930 track->mRetryCount = kMaxTrackRetries; 2931 } 2932 if (recentUnderruns == 0) { 2933 // no recent underruns: stay active 2934 break; 2935 } 2936 // there has recently been an underrun of some kind 2937 if (track->sharedBuffer() == 0) { 2938 // were any of the recent underruns "empty" (no frames available)? 2939 if (recentEmpty == 0) { 2940 // no, then ignore the partial underruns as they are allowed indefinitely 2941 break; 2942 } 2943 // there has recently been an "empty" underrun: decrement the retry counter 2944 if (--(track->mRetryCount) > 0) { 2945 break; 2946 } 2947 // indicate to client process that the track was disabled because of underrun; 2948 // it will then automatically call start() when data is available 2949 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2950 // remove from active list, but state remains ACTIVE [confusing but true] 2951 isActive = false; 2952 break; 2953 } 2954 // fall through 2955 case TrackBase::STOPPING_2: 2956 case TrackBase::PAUSED: 2957 case TrackBase::STOPPED: 2958 case TrackBase::FLUSHED: // flush() while active 2959 // Check for presentation complete if track is inactive 2960 // We have consumed all the buffers of this track. 2961 // This would be incomplete if we auto-paused on underrun 2962 { 2963 size_t audioHALFrames = 2964 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2965 size_t framesWritten = mBytesWritten / mFrameSize; 2966 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2967 // track stays in active list until presentation is complete 2968 break; 2969 } 2970 } 2971 if (track->isStopping_2()) { 2972 track->mState = TrackBase::STOPPED; 2973 } 2974 if (track->isStopped()) { 2975 // Can't reset directly, as fast mixer is still polling this track 2976 // track->reset(); 2977 // So instead mark this track as needing to be reset after push with ack 2978 resetMask |= 1 << i; 2979 } 2980 isActive = false; 2981 break; 2982 case TrackBase::IDLE: 2983 default: 2984 LOG_FATAL("unexpected track state %d", track->mState); 2985 } 2986 2987 if (isActive) { 2988 // was it previously inactive? 2989 if (!(state->mTrackMask & (1 << j))) { 2990 ExtendedAudioBufferProvider *eabp = track; 2991 VolumeProvider *vp = track; 2992 fastTrack->mBufferProvider = eabp; 2993 fastTrack->mVolumeProvider = vp; 2994 fastTrack->mSampleRate = track->mSampleRate; 2995 fastTrack->mChannelMask = track->mChannelMask; 2996 fastTrack->mGeneration++; 2997 state->mTrackMask |= 1 << j; 2998 didModify = true; 2999 // no acknowledgement required for newly active tracks 3000 } 3001 // cache the combined master volume and stream type volume for fast mixer; this 3002 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3003 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3004 ++fastTracks; 3005 } else { 3006 // was it previously active? 3007 if (state->mTrackMask & (1 << j)) { 3008 fastTrack->mBufferProvider = NULL; 3009 fastTrack->mGeneration++; 3010 state->mTrackMask &= ~(1 << j); 3011 didModify = true; 3012 // If any fast tracks were removed, we must wait for acknowledgement 3013 // because we're about to decrement the last sp<> on those tracks. 3014 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3015 } else { 3016 LOG_FATAL("fast track %d should have been active", j); 3017 } 3018 tracksToRemove->add(track); 3019 // Avoids a misleading display in dumpsys 3020 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3021 } 3022 continue; 3023 } 3024 3025 { // local variable scope to avoid goto warning 3026 3027 audio_track_cblk_t* cblk = track->cblk(); 3028 3029 // The first time a track is added we wait 3030 // for all its buffers to be filled before processing it 3031 int name = track->name(); 3032 // make sure that we have enough frames to mix one full buffer. 3033 // enforce this condition only once to enable draining the buffer in case the client 3034 // app does not call stop() and relies on underrun to stop: 3035 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3036 // during last round 3037 size_t desiredFrames; 3038 uint32_t sr = track->sampleRate(); 3039 if (sr == mSampleRate) { 3040 desiredFrames = mNormalFrameCount; 3041 } else { 3042 // +1 for rounding and +1 for additional sample needed for interpolation 3043 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3044 // add frames already consumed but not yet released by the resampler 3045 // because mAudioTrackServerProxy->framesReady() will include these frames 3046 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3047#if 0 3048 // the minimum track buffer size is normally twice the number of frames necessary 3049 // to fill one buffer and the resampler should not leave more than one buffer worth 3050 // of unreleased frames after each pass, but just in case... 3051 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3052#endif 3053 } 3054 uint32_t minFrames = 1; 3055 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3056 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3057 minFrames = desiredFrames; 3058 } 3059 3060 size_t framesReady = track->framesReady(); 3061 if ((framesReady >= minFrames) && track->isReady() && 3062 !track->isPaused() && !track->isTerminated()) 3063 { 3064 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3065 3066 mixedTracks++; 3067 3068 // track->mainBuffer() != mMixBuffer means there is an effect chain 3069 // connected to the track 3070 chain.clear(); 3071 if (track->mainBuffer() != mMixBuffer) { 3072 chain = getEffectChain_l(track->sessionId()); 3073 // Delegate volume control to effect in track effect chain if needed 3074 if (chain != 0) { 3075 tracksWithEffect++; 3076 } else { 3077 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3078 "session %d", 3079 name, track->sessionId()); 3080 } 3081 } 3082 3083 3084 int param = AudioMixer::VOLUME; 3085 if (track->mFillingUpStatus == Track::FS_FILLED) { 3086 // no ramp for the first volume setting 3087 track->mFillingUpStatus = Track::FS_ACTIVE; 3088 if (track->mState == TrackBase::RESUMING) { 3089 track->mState = TrackBase::ACTIVE; 3090 param = AudioMixer::RAMP_VOLUME; 3091 } 3092 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3093 // FIXME should not make a decision based on mServer 3094 } else if (cblk->mServer != 0) { 3095 // If the track is stopped before the first frame was mixed, 3096 // do not apply ramp 3097 param = AudioMixer::RAMP_VOLUME; 3098 } 3099 3100 // compute volume for this track 3101 uint32_t vl, vr, va; 3102 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3103 vl = vr = va = 0; 3104 if (track->isPausing()) { 3105 track->setPaused(); 3106 } 3107 } else { 3108 3109 // read original volumes with volume control 3110 float typeVolume = mStreamTypes[track->streamType()].volume; 3111 float v = masterVolume * typeVolume; 3112 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3113 uint32_t vlr = proxy->getVolumeLR(); 3114 vl = vlr & 0xFFFF; 3115 vr = vlr >> 16; 3116 // track volumes come from shared memory, so can't be trusted and must be clamped 3117 if (vl > MAX_GAIN_INT) { 3118 ALOGV("Track left volume out of range: %04X", vl); 3119 vl = MAX_GAIN_INT; 3120 } 3121 if (vr > MAX_GAIN_INT) { 3122 ALOGV("Track right volume out of range: %04X", vr); 3123 vr = MAX_GAIN_INT; 3124 } 3125 // now apply the master volume and stream type volume 3126 vl = (uint32_t)(v * vl) << 12; 3127 vr = (uint32_t)(v * vr) << 12; 3128 // assuming master volume and stream type volume each go up to 1.0, 3129 // vl and vr are now in 8.24 format 3130 3131 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3132 // send level comes from shared memory and so may be corrupt 3133 if (sendLevel > MAX_GAIN_INT) { 3134 ALOGV("Track send level out of range: %04X", sendLevel); 3135 sendLevel = MAX_GAIN_INT; 3136 } 3137 va = (uint32_t)(v * sendLevel); 3138 } 3139 3140 // Delegate volume control to effect in track effect chain if needed 3141 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3142 // Do not ramp volume if volume is controlled by effect 3143 param = AudioMixer::VOLUME; 3144 track->mHasVolumeController = true; 3145 } else { 3146 // force no volume ramp when volume controller was just disabled or removed 3147 // from effect chain to avoid volume spike 3148 if (track->mHasVolumeController) { 3149 param = AudioMixer::VOLUME; 3150 } 3151 track->mHasVolumeController = false; 3152 } 3153 3154 // Convert volumes from 8.24 to 4.12 format 3155 // This additional clamping is needed in case chain->setVolume_l() overshot 3156 vl = (vl + (1 << 11)) >> 12; 3157 if (vl > MAX_GAIN_INT) { 3158 vl = MAX_GAIN_INT; 3159 } 3160 vr = (vr + (1 << 11)) >> 12; 3161 if (vr > MAX_GAIN_INT) { 3162 vr = MAX_GAIN_INT; 3163 } 3164 3165 if (va > MAX_GAIN_INT) { 3166 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3167 } 3168 3169 // XXX: these things DON'T need to be done each time 3170 mAudioMixer->setBufferProvider(name, track); 3171 mAudioMixer->enable(name); 3172 3173 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3174 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3175 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3176 mAudioMixer->setParameter( 3177 name, 3178 AudioMixer::TRACK, 3179 AudioMixer::FORMAT, (void *)track->format()); 3180 mAudioMixer->setParameter( 3181 name, 3182 AudioMixer::TRACK, 3183 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3184 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3185 uint32_t maxSampleRate = mSampleRate * 2; 3186 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3187 if (reqSampleRate == 0) { 3188 reqSampleRate = mSampleRate; 3189 } else if (reqSampleRate > maxSampleRate) { 3190 reqSampleRate = maxSampleRate; 3191 } 3192 mAudioMixer->setParameter( 3193 name, 3194 AudioMixer::RESAMPLE, 3195 AudioMixer::SAMPLE_RATE, 3196 (void *)reqSampleRate); 3197 mAudioMixer->setParameter( 3198 name, 3199 AudioMixer::TRACK, 3200 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3201 mAudioMixer->setParameter( 3202 name, 3203 AudioMixer::TRACK, 3204 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3205 3206 // reset retry count 3207 track->mRetryCount = kMaxTrackRetries; 3208 3209 // If one track is ready, set the mixer ready if: 3210 // - the mixer was not ready during previous round OR 3211 // - no other track is not ready 3212 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3213 mixerStatus != MIXER_TRACKS_ENABLED) { 3214 mixerStatus = MIXER_TRACKS_READY; 3215 } 3216 } else { 3217 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3218 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3219 } 3220 // clear effect chain input buffer if an active track underruns to avoid sending 3221 // previous audio buffer again to effects 3222 chain = getEffectChain_l(track->sessionId()); 3223 if (chain != 0) { 3224 chain->clearInputBuffer(); 3225 } 3226 3227 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3228 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3229 track->isStopped() || track->isPaused()) { 3230 // We have consumed all the buffers of this track. 3231 // Remove it from the list of active tracks. 3232 // TODO: use actual buffer filling status instead of latency when available from 3233 // audio HAL 3234 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3235 size_t framesWritten = mBytesWritten / mFrameSize; 3236 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3237 if (track->isStopped()) { 3238 track->reset(); 3239 } 3240 tracksToRemove->add(track); 3241 } 3242 } else { 3243 // No buffers for this track. Give it a few chances to 3244 // fill a buffer, then remove it from active list. 3245 if (--(track->mRetryCount) <= 0) { 3246 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3247 tracksToRemove->add(track); 3248 // indicate to client process that the track was disabled because of underrun; 3249 // it will then automatically call start() when data is available 3250 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3251 // If one track is not ready, mark the mixer also not ready if: 3252 // - the mixer was ready during previous round OR 3253 // - no other track is ready 3254 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3255 mixerStatus != MIXER_TRACKS_READY) { 3256 mixerStatus = MIXER_TRACKS_ENABLED; 3257 } 3258 } 3259 mAudioMixer->disable(name); 3260 } 3261 3262 } // local variable scope to avoid goto warning 3263track_is_ready: ; 3264 3265 } 3266 3267 // Push the new FastMixer state if necessary 3268 bool pauseAudioWatchdog = false; 3269 if (didModify) { 3270 state->mFastTracksGen++; 3271 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3272 if (kUseFastMixer == FastMixer_Dynamic && 3273 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3274 state->mCommand = FastMixerState::COLD_IDLE; 3275 state->mColdFutexAddr = &mFastMixerFutex; 3276 state->mColdGen++; 3277 mFastMixerFutex = 0; 3278 if (kUseFastMixer == FastMixer_Dynamic) { 3279 mNormalSink = mOutputSink; 3280 } 3281 // If we go into cold idle, need to wait for acknowledgement 3282 // so that fast mixer stops doing I/O. 3283 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3284 pauseAudioWatchdog = true; 3285 } 3286 } 3287 if (sq != NULL) { 3288 sq->end(didModify); 3289 sq->push(block); 3290 } 3291#ifdef AUDIO_WATCHDOG 3292 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3293 mAudioWatchdog->pause(); 3294 } 3295#endif 3296 3297 // Now perform the deferred reset on fast tracks that have stopped 3298 while (resetMask != 0) { 3299 size_t i = __builtin_ctz(resetMask); 3300 ALOG_ASSERT(i < count); 3301 resetMask &= ~(1 << i); 3302 sp<Track> t = mActiveTracks[i].promote(); 3303 if (t == 0) { 3304 continue; 3305 } 3306 Track* track = t.get(); 3307 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3308 track->reset(); 3309 } 3310 3311 // remove all the tracks that need to be... 3312 removeTracks_l(*tracksToRemove); 3313 3314 // mix buffer must be cleared if all tracks are connected to an 3315 // effect chain as in this case the mixer will not write to 3316 // mix buffer and track effects will accumulate into it 3317 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3318 (mixedTracks == 0 && fastTracks > 0))) { 3319 // FIXME as a performance optimization, should remember previous zero status 3320 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3321 } 3322 3323 // if any fast tracks, then status is ready 3324 mMixerStatusIgnoringFastTracks = mixerStatus; 3325 if (fastTracks > 0) { 3326 mixerStatus = MIXER_TRACKS_READY; 3327 } 3328 return mixerStatus; 3329} 3330 3331// getTrackName_l() must be called with ThreadBase::mLock held 3332int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3333{ 3334 return mAudioMixer->getTrackName(channelMask, sessionId); 3335} 3336 3337// deleteTrackName_l() must be called with ThreadBase::mLock held 3338void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3339{ 3340 ALOGV("remove track (%d) and delete from mixer", name); 3341 mAudioMixer->deleteTrackName(name); 3342} 3343 3344// checkForNewParameters_l() must be called with ThreadBase::mLock held 3345bool AudioFlinger::MixerThread::checkForNewParameters_l() 3346{ 3347 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3348 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3349 bool reconfig = false; 3350 3351 while (!mNewParameters.isEmpty()) { 3352 3353 if (mFastMixer != NULL) { 3354 FastMixerStateQueue *sq = mFastMixer->sq(); 3355 FastMixerState *state = sq->begin(); 3356 if (!(state->mCommand & FastMixerState::IDLE)) { 3357 previousCommand = state->mCommand; 3358 state->mCommand = FastMixerState::HOT_IDLE; 3359 sq->end(); 3360 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3361 } else { 3362 sq->end(false /*didModify*/); 3363 } 3364 } 3365 3366 status_t status = NO_ERROR; 3367 String8 keyValuePair = mNewParameters[0]; 3368 AudioParameter param = AudioParameter(keyValuePair); 3369 int value; 3370 3371 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3372 reconfig = true; 3373 } 3374 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3375 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3376 status = BAD_VALUE; 3377 } else { 3378 // no need to save value, since it's constant 3379 reconfig = true; 3380 } 3381 } 3382 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3383 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3384 status = BAD_VALUE; 3385 } else { 3386 // no need to save value, since it's constant 3387 reconfig = true; 3388 } 3389 } 3390 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3391 // do not accept frame count changes if tracks are open as the track buffer 3392 // size depends on frame count and correct behavior would not be guaranteed 3393 // if frame count is changed after track creation 3394 if (!mTracks.isEmpty()) { 3395 status = INVALID_OPERATION; 3396 } else { 3397 reconfig = true; 3398 } 3399 } 3400 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3401#ifdef ADD_BATTERY_DATA 3402 // when changing the audio output device, call addBatteryData to notify 3403 // the change 3404 if (mOutDevice != value) { 3405 uint32_t params = 0; 3406 // check whether speaker is on 3407 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3408 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3409 } 3410 3411 audio_devices_t deviceWithoutSpeaker 3412 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3413 // check if any other device (except speaker) is on 3414 if (value & deviceWithoutSpeaker ) { 3415 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3416 } 3417 3418 if (params != 0) { 3419 addBatteryData(params); 3420 } 3421 } 3422#endif 3423 3424 // forward device change to effects that have requested to be 3425 // aware of attached audio device. 3426 if (value != AUDIO_DEVICE_NONE) { 3427 mOutDevice = value; 3428 for (size_t i = 0; i < mEffectChains.size(); i++) { 3429 mEffectChains[i]->setDevice_l(mOutDevice); 3430 } 3431 } 3432 } 3433 3434 if (status == NO_ERROR) { 3435 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3436 keyValuePair.string()); 3437 if (!mStandby && status == INVALID_OPERATION) { 3438 mOutput->stream->common.standby(&mOutput->stream->common); 3439 mStandby = true; 3440 mBytesWritten = 0; 3441 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3442 keyValuePair.string()); 3443 } 3444 if (status == NO_ERROR && reconfig) { 3445 readOutputParameters(); 3446 delete mAudioMixer; 3447 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3448 for (size_t i = 0; i < mTracks.size() ; i++) { 3449 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3450 if (name < 0) { 3451 break; 3452 } 3453 mTracks[i]->mName = name; 3454 } 3455 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3456 } 3457 } 3458 3459 mNewParameters.removeAt(0); 3460 3461 mParamStatus = status; 3462 mParamCond.signal(); 3463 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3464 // already timed out waiting for the status and will never signal the condition. 3465 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3466 } 3467 3468 if (!(previousCommand & FastMixerState::IDLE)) { 3469 ALOG_ASSERT(mFastMixer != NULL); 3470 FastMixerStateQueue *sq = mFastMixer->sq(); 3471 FastMixerState *state = sq->begin(); 3472 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3473 state->mCommand = previousCommand; 3474 sq->end(); 3475 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3476 } 3477 3478 return reconfig; 3479} 3480 3481 3482void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3483{ 3484 const size_t SIZE = 256; 3485 char buffer[SIZE]; 3486 String8 result; 3487 3488 PlaybackThread::dumpInternals(fd, args); 3489 3490 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3491 result.append(buffer); 3492 write(fd, result.string(), result.size()); 3493 3494 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3495 const FastMixerDumpState copy(mFastMixerDumpState); 3496 copy.dump(fd); 3497 3498#ifdef STATE_QUEUE_DUMP 3499 // Similar for state queue 3500 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3501 observerCopy.dump(fd); 3502 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3503 mutatorCopy.dump(fd); 3504#endif 3505 3506#ifdef TEE_SINK 3507 // Write the tee output to a .wav file 3508 dumpTee(fd, mTeeSource, mId); 3509#endif 3510 3511#ifdef AUDIO_WATCHDOG 3512 if (mAudioWatchdog != 0) { 3513 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3514 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3515 wdCopy.dump(fd); 3516 } 3517#endif 3518} 3519 3520uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3521{ 3522 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3523} 3524 3525uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3526{ 3527 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3528} 3529 3530void AudioFlinger::MixerThread::cacheParameters_l() 3531{ 3532 PlaybackThread::cacheParameters_l(); 3533 3534 // FIXME: Relaxed timing because of a certain device that can't meet latency 3535 // Should be reduced to 2x after the vendor fixes the driver issue 3536 // increase threshold again due to low power audio mode. The way this warning 3537 // threshold is calculated and its usefulness should be reconsidered anyway. 3538 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3539} 3540 3541// ---------------------------------------------------------------------------- 3542 3543AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3544 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3545 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3546 // mLeftVolFloat, mRightVolFloat 3547{ 3548} 3549 3550AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3551 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3552 ThreadBase::type_t type) 3553 : PlaybackThread(audioFlinger, output, id, device, type) 3554 // mLeftVolFloat, mRightVolFloat 3555{ 3556} 3557 3558AudioFlinger::DirectOutputThread::~DirectOutputThread() 3559{ 3560} 3561 3562void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3563{ 3564 audio_track_cblk_t* cblk = track->cblk(); 3565 float left, right; 3566 3567 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3568 left = right = 0; 3569 } else { 3570 float typeVolume = mStreamTypes[track->streamType()].volume; 3571 float v = mMasterVolume * typeVolume; 3572 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3573 uint32_t vlr = proxy->getVolumeLR(); 3574 float v_clamped = v * (vlr & 0xFFFF); 3575 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3576 left = v_clamped/MAX_GAIN; 3577 v_clamped = v * (vlr >> 16); 3578 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3579 right = v_clamped/MAX_GAIN; 3580 } 3581 3582 if (lastTrack) { 3583 if (left != mLeftVolFloat || right != mRightVolFloat) { 3584 mLeftVolFloat = left; 3585 mRightVolFloat = right; 3586 3587 // Convert volumes from float to 8.24 3588 uint32_t vl = (uint32_t)(left * (1 << 24)); 3589 uint32_t vr = (uint32_t)(right * (1 << 24)); 3590 3591 // Delegate volume control to effect in track effect chain if needed 3592 // only one effect chain can be present on DirectOutputThread, so if 3593 // there is one, the track is connected to it 3594 if (!mEffectChains.isEmpty()) { 3595 mEffectChains[0]->setVolume_l(&vl, &vr); 3596 left = (float)vl / (1 << 24); 3597 right = (float)vr / (1 << 24); 3598 } 3599 if (mOutput->stream->set_volume) { 3600 mOutput->stream->set_volume(mOutput->stream, left, right); 3601 } 3602 } 3603 } 3604} 3605 3606 3607AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3608 Vector< sp<Track> > *tracksToRemove 3609) 3610{ 3611 size_t count = mActiveTracks.size(); 3612 mixer_state mixerStatus = MIXER_IDLE; 3613 3614 // find out which tracks need to be processed 3615 for (size_t i = 0; i < count; i++) { 3616 sp<Track> t = mActiveTracks[i].promote(); 3617 // The track died recently 3618 if (t == 0) { 3619 continue; 3620 } 3621 3622 Track* const track = t.get(); 3623 audio_track_cblk_t* cblk = track->cblk(); 3624 // Only consider last track started for volume and mixer state control. 3625 // In theory an older track could underrun and restart after the new one starts 3626 // but as we only care about the transition phase between two tracks on a 3627 // direct output, it is not a problem to ignore the underrun case. 3628 sp<Track> l = mLatestActiveTrack.promote(); 3629 bool last = l.get() == track; 3630 3631 // The first time a track is added we wait 3632 // for all its buffers to be filled before processing it 3633 uint32_t minFrames; 3634 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3635 minFrames = mNormalFrameCount; 3636 } else { 3637 minFrames = 1; 3638 } 3639 3640 if ((track->framesReady() >= minFrames) && track->isReady() && 3641 !track->isPaused() && !track->isTerminated()) 3642 { 3643 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3644 3645 if (track->mFillingUpStatus == Track::FS_FILLED) { 3646 track->mFillingUpStatus = Track::FS_ACTIVE; 3647 // make sure processVolume_l() will apply new volume even if 0 3648 mLeftVolFloat = mRightVolFloat = -1.0; 3649 if (track->mState == TrackBase::RESUMING) { 3650 track->mState = TrackBase::ACTIVE; 3651 } 3652 } 3653 3654 // compute volume for this track 3655 processVolume_l(track, last); 3656 if (last) { 3657 // reset retry count 3658 track->mRetryCount = kMaxTrackRetriesDirect; 3659 mActiveTrack = t; 3660 mixerStatus = MIXER_TRACKS_READY; 3661 } 3662 } else { 3663 // clear effect chain input buffer if the last active track started underruns 3664 // to avoid sending previous audio buffer again to effects 3665 if (!mEffectChains.isEmpty() && last) { 3666 mEffectChains[0]->clearInputBuffer(); 3667 } 3668 3669 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3670 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3671 track->isStopped() || track->isPaused()) { 3672 // We have consumed all the buffers of this track. 3673 // Remove it from the list of active tracks. 3674 // TODO: implement behavior for compressed audio 3675 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3676 size_t framesWritten = mBytesWritten / mFrameSize; 3677 if (mStandby || !last || 3678 track->presentationComplete(framesWritten, audioHALFrames)) { 3679 if (track->isStopped()) { 3680 track->reset(); 3681 } 3682 tracksToRemove->add(track); 3683 } 3684 } else { 3685 // No buffers for this track. Give it a few chances to 3686 // fill a buffer, then remove it from active list. 3687 // Only consider last track started for mixer state control 3688 if (--(track->mRetryCount) <= 0) { 3689 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3690 tracksToRemove->add(track); 3691 // indicate to client process that the track was disabled because of underrun; 3692 // it will then automatically call start() when data is available 3693 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3694 } else if (last) { 3695 mixerStatus = MIXER_TRACKS_ENABLED; 3696 } 3697 } 3698 } 3699 } 3700 3701 // remove all the tracks that need to be... 3702 removeTracks_l(*tracksToRemove); 3703 3704 return mixerStatus; 3705} 3706 3707void AudioFlinger::DirectOutputThread::threadLoop_mix() 3708{ 3709 size_t frameCount = mFrameCount; 3710 int8_t *curBuf = (int8_t *)mMixBuffer; 3711 // output audio to hardware 3712 while (frameCount) { 3713 AudioBufferProvider::Buffer buffer; 3714 buffer.frameCount = frameCount; 3715 mActiveTrack->getNextBuffer(&buffer); 3716 if (buffer.raw == NULL) { 3717 memset(curBuf, 0, frameCount * mFrameSize); 3718 break; 3719 } 3720 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3721 frameCount -= buffer.frameCount; 3722 curBuf += buffer.frameCount * mFrameSize; 3723 mActiveTrack->releaseBuffer(&buffer); 3724 } 3725 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3726 sleepTime = 0; 3727 standbyTime = systemTime() + standbyDelay; 3728 mActiveTrack.clear(); 3729} 3730 3731void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3732{ 3733 if (sleepTime == 0) { 3734 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3735 sleepTime = activeSleepTime; 3736 } else { 3737 sleepTime = idleSleepTime; 3738 } 3739 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3740 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3741 sleepTime = 0; 3742 } 3743} 3744 3745// getTrackName_l() must be called with ThreadBase::mLock held 3746int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3747 int sessionId __unused) 3748{ 3749 return 0; 3750} 3751 3752// deleteTrackName_l() must be called with ThreadBase::mLock held 3753void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3754{ 3755} 3756 3757// checkForNewParameters_l() must be called with ThreadBase::mLock held 3758bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3759{ 3760 bool reconfig = false; 3761 3762 while (!mNewParameters.isEmpty()) { 3763 status_t status = NO_ERROR; 3764 String8 keyValuePair = mNewParameters[0]; 3765 AudioParameter param = AudioParameter(keyValuePair); 3766 int value; 3767 3768 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3769 // do not accept frame count changes if tracks are open as the track buffer 3770 // size depends on frame count and correct behavior would not be garantied 3771 // if frame count is changed after track creation 3772 if (!mTracks.isEmpty()) { 3773 status = INVALID_OPERATION; 3774 } else { 3775 reconfig = true; 3776 } 3777 } 3778 if (status == NO_ERROR) { 3779 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3780 keyValuePair.string()); 3781 if (!mStandby && status == INVALID_OPERATION) { 3782 mOutput->stream->common.standby(&mOutput->stream->common); 3783 mStandby = true; 3784 mBytesWritten = 0; 3785 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3786 keyValuePair.string()); 3787 } 3788 if (status == NO_ERROR && reconfig) { 3789 readOutputParameters(); 3790 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3791 } 3792 } 3793 3794 mNewParameters.removeAt(0); 3795 3796 mParamStatus = status; 3797 mParamCond.signal(); 3798 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3799 // already timed out waiting for the status and will never signal the condition. 3800 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3801 } 3802 return reconfig; 3803} 3804 3805uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3806{ 3807 uint32_t time; 3808 if (audio_is_linear_pcm(mFormat)) { 3809 time = PlaybackThread::activeSleepTimeUs(); 3810 } else { 3811 time = 10000; 3812 } 3813 return time; 3814} 3815 3816uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3817{ 3818 uint32_t time; 3819 if (audio_is_linear_pcm(mFormat)) { 3820 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3821 } else { 3822 time = 10000; 3823 } 3824 return time; 3825} 3826 3827uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3828{ 3829 uint32_t time; 3830 if (audio_is_linear_pcm(mFormat)) { 3831 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3832 } else { 3833 time = 10000; 3834 } 3835 return time; 3836} 3837 3838void AudioFlinger::DirectOutputThread::cacheParameters_l() 3839{ 3840 PlaybackThread::cacheParameters_l(); 3841 3842 // use shorter standby delay as on normal output to release 3843 // hardware resources as soon as possible 3844 if (audio_is_linear_pcm(mFormat)) { 3845 standbyDelay = microseconds(activeSleepTime*2); 3846 } else { 3847 standbyDelay = kOffloadStandbyDelayNs; 3848 } 3849} 3850 3851// ---------------------------------------------------------------------------- 3852 3853AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3854 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3855 : Thread(false /*canCallJava*/), 3856 mPlaybackThread(playbackThread), 3857 mWriteAckSequence(0), 3858 mDrainSequence(0) 3859{ 3860} 3861 3862AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3863{ 3864} 3865 3866void AudioFlinger::AsyncCallbackThread::onFirstRef() 3867{ 3868 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3869} 3870 3871bool AudioFlinger::AsyncCallbackThread::threadLoop() 3872{ 3873 while (!exitPending()) { 3874 uint32_t writeAckSequence; 3875 uint32_t drainSequence; 3876 3877 { 3878 Mutex::Autolock _l(mLock); 3879 while (!((mWriteAckSequence & 1) || 3880 (mDrainSequence & 1) || 3881 exitPending())) { 3882 mWaitWorkCV.wait(mLock); 3883 } 3884 3885 if (exitPending()) { 3886 break; 3887 } 3888 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3889 mWriteAckSequence, mDrainSequence); 3890 writeAckSequence = mWriteAckSequence; 3891 mWriteAckSequence &= ~1; 3892 drainSequence = mDrainSequence; 3893 mDrainSequence &= ~1; 3894 } 3895 { 3896 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3897 if (playbackThread != 0) { 3898 if (writeAckSequence & 1) { 3899 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3900 } 3901 if (drainSequence & 1) { 3902 playbackThread->resetDraining(drainSequence >> 1); 3903 } 3904 } 3905 } 3906 } 3907 return false; 3908} 3909 3910void AudioFlinger::AsyncCallbackThread::exit() 3911{ 3912 ALOGV("AsyncCallbackThread::exit"); 3913 Mutex::Autolock _l(mLock); 3914 requestExit(); 3915 mWaitWorkCV.broadcast(); 3916} 3917 3918void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3919{ 3920 Mutex::Autolock _l(mLock); 3921 // bit 0 is cleared 3922 mWriteAckSequence = sequence << 1; 3923} 3924 3925void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3926{ 3927 Mutex::Autolock _l(mLock); 3928 // ignore unexpected callbacks 3929 if (mWriteAckSequence & 2) { 3930 mWriteAckSequence |= 1; 3931 mWaitWorkCV.signal(); 3932 } 3933} 3934 3935void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3936{ 3937 Mutex::Autolock _l(mLock); 3938 // bit 0 is cleared 3939 mDrainSequence = sequence << 1; 3940} 3941 3942void AudioFlinger::AsyncCallbackThread::resetDraining() 3943{ 3944 Mutex::Autolock _l(mLock); 3945 // ignore unexpected callbacks 3946 if (mDrainSequence & 2) { 3947 mDrainSequence |= 1; 3948 mWaitWorkCV.signal(); 3949 } 3950} 3951 3952 3953// ---------------------------------------------------------------------------- 3954AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3955 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3956 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3957 mHwPaused(false), 3958 mFlushPending(false), 3959 mPausedBytesRemaining(0) 3960{ 3961 //FIXME: mStandby should be set to true by ThreadBase constructor 3962 mStandby = true; 3963} 3964 3965void AudioFlinger::OffloadThread::threadLoop_exit() 3966{ 3967 if (mFlushPending || mHwPaused) { 3968 // If a flush is pending or track was paused, just discard buffered data 3969 flushHw_l(); 3970 } else { 3971 mMixerStatus = MIXER_DRAIN_ALL; 3972 threadLoop_drain(); 3973 } 3974 mCallbackThread->exit(); 3975 PlaybackThread::threadLoop_exit(); 3976} 3977 3978AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3979 Vector< sp<Track> > *tracksToRemove 3980) 3981{ 3982 size_t count = mActiveTracks.size(); 3983 3984 mixer_state mixerStatus = MIXER_IDLE; 3985 bool doHwPause = false; 3986 bool doHwResume = false; 3987 3988 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3989 3990 // find out which tracks need to be processed 3991 for (size_t i = 0; i < count; i++) { 3992 sp<Track> t = mActiveTracks[i].promote(); 3993 // The track died recently 3994 if (t == 0) { 3995 continue; 3996 } 3997 Track* const track = t.get(); 3998 audio_track_cblk_t* cblk = track->cblk(); 3999 // Only consider last track started for volume and mixer state control. 4000 // In theory an older track could underrun and restart after the new one starts 4001 // but as we only care about the transition phase between two tracks on a 4002 // direct output, it is not a problem to ignore the underrun case. 4003 sp<Track> l = mLatestActiveTrack.promote(); 4004 bool last = l.get() == track; 4005 4006 if (track->isPausing()) { 4007 track->setPaused(); 4008 if (last) { 4009 if (!mHwPaused) { 4010 doHwPause = true; 4011 mHwPaused = true; 4012 } 4013 // If we were part way through writing the mixbuffer to 4014 // the HAL we must save this until we resume 4015 // BUG - this will be wrong if a different track is made active, 4016 // in that case we want to discard the pending data in the 4017 // mixbuffer and tell the client to present it again when the 4018 // track is resumed 4019 mPausedWriteLength = mCurrentWriteLength; 4020 mPausedBytesRemaining = mBytesRemaining; 4021 mBytesRemaining = 0; // stop writing 4022 } 4023 tracksToRemove->add(track); 4024 } else if (track->framesReady() && track->isReady() && 4025 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4026 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4027 if (track->mFillingUpStatus == Track::FS_FILLED) { 4028 track->mFillingUpStatus = Track::FS_ACTIVE; 4029 // make sure processVolume_l() will apply new volume even if 0 4030 mLeftVolFloat = mRightVolFloat = -1.0; 4031 if (track->mState == TrackBase::RESUMING) { 4032 track->mState = TrackBase::ACTIVE; 4033 if (last) { 4034 if (mPausedBytesRemaining) { 4035 // Need to continue write that was interrupted 4036 mCurrentWriteLength = mPausedWriteLength; 4037 mBytesRemaining = mPausedBytesRemaining; 4038 mPausedBytesRemaining = 0; 4039 } 4040 if (mHwPaused) { 4041 doHwResume = true; 4042 mHwPaused = false; 4043 // threadLoop_mix() will handle the case that we need to 4044 // resume an interrupted write 4045 } 4046 // enable write to audio HAL 4047 sleepTime = 0; 4048 } 4049 } 4050 } 4051 4052 if (last) { 4053 sp<Track> previousTrack = mPreviousTrack.promote(); 4054 if (previousTrack != 0) { 4055 if (track != previousTrack.get()) { 4056 // Flush any data still being written from last track 4057 mBytesRemaining = 0; 4058 if (mPausedBytesRemaining) { 4059 // Last track was paused so we also need to flush saved 4060 // mixbuffer state and invalidate track so that it will 4061 // re-submit that unwritten data when it is next resumed 4062 mPausedBytesRemaining = 0; 4063 // Invalidate is a bit drastic - would be more efficient 4064 // to have a flag to tell client that some of the 4065 // previously written data was lost 4066 previousTrack->invalidate(); 4067 } 4068 // flush data already sent to the DSP if changing audio session as audio 4069 // comes from a different source. Also invalidate previous track to force a 4070 // seek when resuming. 4071 if (previousTrack->sessionId() != track->sessionId()) { 4072 previousTrack->invalidate(); 4073 mFlushPending = true; 4074 } 4075 } 4076 } 4077 mPreviousTrack = track; 4078 // reset retry count 4079 track->mRetryCount = kMaxTrackRetriesOffload; 4080 mActiveTrack = t; 4081 mixerStatus = MIXER_TRACKS_READY; 4082 } 4083 } else { 4084 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4085 if (track->isStopping_1()) { 4086 // Hardware buffer can hold a large amount of audio so we must 4087 // wait for all current track's data to drain before we say 4088 // that the track is stopped. 4089 if (mBytesRemaining == 0) { 4090 // Only start draining when all data in mixbuffer 4091 // has been written 4092 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4093 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4094 // do not drain if no data was ever sent to HAL (mStandby == true) 4095 if (last && !mStandby) { 4096 // do not modify drain sequence if we are already draining. This happens 4097 // when resuming from pause after drain. 4098 if ((mDrainSequence & 1) == 0) { 4099 sleepTime = 0; 4100 standbyTime = systemTime() + standbyDelay; 4101 mixerStatus = MIXER_DRAIN_TRACK; 4102 mDrainSequence += 2; 4103 } 4104 if (mHwPaused) { 4105 // It is possible to move from PAUSED to STOPPING_1 without 4106 // a resume so we must ensure hardware is running 4107 doHwResume = true; 4108 mHwPaused = false; 4109 } 4110 } 4111 } 4112 } else if (track->isStopping_2()) { 4113 // Drain has completed or we are in standby, signal presentation complete 4114 if (!(mDrainSequence & 1) || !last || mStandby) { 4115 track->mState = TrackBase::STOPPED; 4116 size_t audioHALFrames = 4117 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4118 size_t framesWritten = 4119 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4120 track->presentationComplete(framesWritten, audioHALFrames); 4121 track->reset(); 4122 tracksToRemove->add(track); 4123 } 4124 } else { 4125 // No buffers for this track. Give it a few chances to 4126 // fill a buffer, then remove it from active list. 4127 if (--(track->mRetryCount) <= 0) { 4128 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4129 track->name()); 4130 tracksToRemove->add(track); 4131 // indicate to client process that the track was disabled because of underrun; 4132 // it will then automatically call start() when data is available 4133 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4134 } else if (last){ 4135 mixerStatus = MIXER_TRACKS_ENABLED; 4136 } 4137 } 4138 } 4139 // compute volume for this track 4140 processVolume_l(track, last); 4141 } 4142 4143 // make sure the pause/flush/resume sequence is executed in the right order. 4144 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4145 // before flush and then resume HW. This can happen in case of pause/flush/resume 4146 // if resume is received before pause is executed. 4147 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4148 mOutput->stream->pause(mOutput->stream); 4149 if (!doHwPause) { 4150 doHwResume = true; 4151 } 4152 } 4153 if (mFlushPending) { 4154 flushHw_l(); 4155 mFlushPending = false; 4156 } 4157 if (!mStandby && doHwResume) { 4158 mOutput->stream->resume(mOutput->stream); 4159 } 4160 4161 // remove all the tracks that need to be... 4162 removeTracks_l(*tracksToRemove); 4163 4164 return mixerStatus; 4165} 4166 4167void AudioFlinger::OffloadThread::flushOutput_l() 4168{ 4169 mFlushPending = true; 4170} 4171 4172// must be called with thread mutex locked 4173bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4174{ 4175 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4176 mWriteAckSequence, mDrainSequence); 4177 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4178 return true; 4179 } 4180 return false; 4181} 4182 4183// must be called with thread mutex locked 4184bool AudioFlinger::OffloadThread::shouldStandby_l() 4185{ 4186 bool trackPaused = false; 4187 4188 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4189 // after a timeout and we will enter standby then. 4190 if (mTracks.size() > 0) { 4191 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4192 } 4193 4194 return !mStandby && !trackPaused; 4195} 4196 4197 4198bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4199{ 4200 Mutex::Autolock _l(mLock); 4201 return waitingAsyncCallback_l(); 4202} 4203 4204void AudioFlinger::OffloadThread::flushHw_l() 4205{ 4206 mOutput->stream->flush(mOutput->stream); 4207 // Flush anything still waiting in the mixbuffer 4208 mCurrentWriteLength = 0; 4209 mBytesRemaining = 0; 4210 mPausedWriteLength = 0; 4211 mPausedBytesRemaining = 0; 4212 if (mUseAsyncWrite) { 4213 // discard any pending drain or write ack by incrementing sequence 4214 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4215 mDrainSequence = (mDrainSequence + 2) & ~1; 4216 ALOG_ASSERT(mCallbackThread != 0); 4217 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4218 mCallbackThread->setDraining(mDrainSequence); 4219 } 4220} 4221 4222// ---------------------------------------------------------------------------- 4223 4224AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4225 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4226 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4227 DUPLICATING), 4228 mWaitTimeMs(UINT_MAX) 4229{ 4230 addOutputTrack(mainThread); 4231} 4232 4233AudioFlinger::DuplicatingThread::~DuplicatingThread() 4234{ 4235 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4236 mOutputTracks[i]->destroy(); 4237 } 4238} 4239 4240void AudioFlinger::DuplicatingThread::threadLoop_mix() 4241{ 4242 // mix buffers... 4243 if (outputsReady(outputTracks)) { 4244 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4245 } else { 4246 memset(mMixBuffer, 0, mixBufferSize); 4247 } 4248 sleepTime = 0; 4249 writeFrames = mNormalFrameCount; 4250 mCurrentWriteLength = mixBufferSize; 4251 standbyTime = systemTime() + standbyDelay; 4252} 4253 4254void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4255{ 4256 if (sleepTime == 0) { 4257 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4258 sleepTime = activeSleepTime; 4259 } else { 4260 sleepTime = idleSleepTime; 4261 } 4262 } else if (mBytesWritten != 0) { 4263 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4264 writeFrames = mNormalFrameCount; 4265 memset(mMixBuffer, 0, mixBufferSize); 4266 } else { 4267 // flush remaining overflow buffers in output tracks 4268 writeFrames = 0; 4269 } 4270 sleepTime = 0; 4271 } 4272} 4273 4274ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4275{ 4276 for (size_t i = 0; i < outputTracks.size(); i++) { 4277 outputTracks[i]->write(mMixBuffer, writeFrames); 4278 } 4279 mStandby = false; 4280 return (ssize_t)mixBufferSize; 4281} 4282 4283void AudioFlinger::DuplicatingThread::threadLoop_standby() 4284{ 4285 // DuplicatingThread implements standby by stopping all tracks 4286 for (size_t i = 0; i < outputTracks.size(); i++) { 4287 outputTracks[i]->stop(); 4288 } 4289} 4290 4291void AudioFlinger::DuplicatingThread::saveOutputTracks() 4292{ 4293 outputTracks = mOutputTracks; 4294} 4295 4296void AudioFlinger::DuplicatingThread::clearOutputTracks() 4297{ 4298 outputTracks.clear(); 4299} 4300 4301void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4302{ 4303 Mutex::Autolock _l(mLock); 4304 // FIXME explain this formula 4305 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4306 OutputTrack *outputTrack = new OutputTrack(thread, 4307 this, 4308 mSampleRate, 4309 mFormat, 4310 mChannelMask, 4311 frameCount, 4312 IPCThreadState::self()->getCallingUid()); 4313 if (outputTrack->cblk() != NULL) { 4314 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4315 mOutputTracks.add(outputTrack); 4316 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4317 updateWaitTime_l(); 4318 } 4319} 4320 4321void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4322{ 4323 Mutex::Autolock _l(mLock); 4324 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4325 if (mOutputTracks[i]->thread() == thread) { 4326 mOutputTracks[i]->destroy(); 4327 mOutputTracks.removeAt(i); 4328 updateWaitTime_l(); 4329 return; 4330 } 4331 } 4332 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4333} 4334 4335// caller must hold mLock 4336void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4337{ 4338 mWaitTimeMs = UINT_MAX; 4339 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4340 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4341 if (strong != 0) { 4342 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4343 if (waitTimeMs < mWaitTimeMs) { 4344 mWaitTimeMs = waitTimeMs; 4345 } 4346 } 4347 } 4348} 4349 4350 4351bool AudioFlinger::DuplicatingThread::outputsReady( 4352 const SortedVector< sp<OutputTrack> > &outputTracks) 4353{ 4354 for (size_t i = 0; i < outputTracks.size(); i++) { 4355 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4356 if (thread == 0) { 4357 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4358 outputTracks[i].get()); 4359 return false; 4360 } 4361 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4362 // see note at standby() declaration 4363 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4364 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4365 thread.get()); 4366 return false; 4367 } 4368 } 4369 return true; 4370} 4371 4372uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4373{ 4374 return (mWaitTimeMs * 1000) / 2; 4375} 4376 4377void AudioFlinger::DuplicatingThread::cacheParameters_l() 4378{ 4379 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4380 updateWaitTime_l(); 4381 4382 MixerThread::cacheParameters_l(); 4383} 4384 4385// ---------------------------------------------------------------------------- 4386// Record 4387// ---------------------------------------------------------------------------- 4388 4389AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4390 AudioStreamIn *input, 4391 uint32_t sampleRate, 4392 audio_channel_mask_t channelMask, 4393 audio_io_handle_t id, 4394 audio_devices_t outDevice, 4395 audio_devices_t inDevice 4396#ifdef TEE_SINK 4397 , const sp<NBAIO_Sink>& teeSink 4398#endif 4399 ) : 4400 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4401 mInput(input), mActiveTracksGen(0), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4402 // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear 4403 // are set by readInputParameters() 4404 // mRsmpInIndex LEGACY 4405 mReqChannelCount(popcount(channelMask)), 4406 mReqSampleRate(sampleRate) 4407 // mBytesRead is only meaningful while active, and so is cleared in start() 4408 // (but might be better to also clear here for dump?) 4409#ifdef TEE_SINK 4410 , mTeeSink(teeSink) 4411#endif 4412{ 4413 snprintf(mName, kNameLength, "AudioIn_%X", id); 4414 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4415 4416 readInputParameters(); 4417} 4418 4419 4420AudioFlinger::RecordThread::~RecordThread() 4421{ 4422 mAudioFlinger->unregisterWriter(mNBLogWriter); 4423 delete[] mRsmpInBuffer; 4424 delete mResampler; 4425 delete[] mRsmpOutBuffer; 4426} 4427 4428void AudioFlinger::RecordThread::onFirstRef() 4429{ 4430 run(mName, PRIORITY_URGENT_AUDIO); 4431} 4432 4433bool AudioFlinger::RecordThread::threadLoop() 4434{ 4435 nsecs_t lastWarning = 0; 4436 4437 inputStandBy(); 4438 4439 // used to verify we've read at least once before evaluating how many bytes were read 4440 bool readOnce = false; 4441 4442 // used to request a deferred sleep, to be executed later while mutex is unlocked 4443 bool doSleep = false; 4444 4445reacquire_wakelock: 4446 sp<RecordTrack> activeTrack; 4447 int activeTracksGen; 4448 { 4449 Mutex::Autolock _l(mLock); 4450 size_t size = mActiveTracks.size(); 4451 activeTracksGen = mActiveTracksGen; 4452 if (size > 0) { 4453 // FIXME an arbitrary choice 4454 activeTrack = mActiveTracks[0]; 4455 acquireWakeLock_l(activeTrack->uid()); 4456 if (size > 1) { 4457 SortedVector<int> tmp; 4458 for (size_t i = 0; i < size; i++) { 4459 tmp.add(mActiveTracks[i]->uid()); 4460 } 4461 updateWakeLockUids_l(tmp); 4462 } 4463 } else { 4464 acquireWakeLock_l(-1); 4465 } 4466 } 4467 4468 // start recording 4469 for (;;) { 4470 TrackBase::track_state activeTrackState; 4471 Vector< sp<EffectChain> > effectChains; 4472 4473 // sleep with mutex unlocked 4474 if (doSleep) { 4475 doSleep = false; 4476 usleep(kRecordThreadSleepUs); 4477 } 4478 4479 { // scope for mLock 4480 Mutex::Autolock _l(mLock); 4481 4482 processConfigEvents_l(); 4483 // return value 'reconfig' is currently unused 4484 bool reconfig = checkForNewParameters_l(); 4485 4486 // check exitPending here because checkForNewParameters_l() and 4487 // checkForNewParameters_l() can temporarily release mLock 4488 if (exitPending()) { 4489 break; 4490 } 4491 4492 // if no active track(s), then standby and release wakelock 4493 size_t size = mActiveTracks.size(); 4494 if (size == 0) { 4495 standbyIfNotAlreadyInStandby(); 4496 // exitPending() can't become true here 4497 releaseWakeLock_l(); 4498 ALOGV("RecordThread: loop stopping"); 4499 // go to sleep 4500 mWaitWorkCV.wait(mLock); 4501 ALOGV("RecordThread: loop starting"); 4502 goto reacquire_wakelock; 4503 } 4504 4505 if (mActiveTracksGen != activeTracksGen) { 4506 activeTracksGen = mActiveTracksGen; 4507 SortedVector<int> tmp; 4508 for (size_t i = 0; i < size; i++) { 4509 tmp.add(mActiveTracks[i]->uid()); 4510 } 4511 updateWakeLockUids_l(tmp); 4512 // FIXME an arbitrary choice 4513 activeTrack = mActiveTracks[0]; 4514 } 4515 4516 if (activeTrack->isTerminated()) { 4517 removeTrack_l(activeTrack); 4518 mActiveTracks.remove(activeTrack); 4519 mActiveTracksGen++; 4520 continue; 4521 } 4522 4523 activeTrackState = activeTrack->mState; 4524 switch (activeTrackState) { 4525 case TrackBase::PAUSING: 4526 standbyIfNotAlreadyInStandby(); 4527 mActiveTracks.remove(activeTrack); 4528 mActiveTracksGen++; 4529 mStartStopCond.broadcast(); 4530 doSleep = true; 4531 continue; 4532 4533 case TrackBase::RESUMING: 4534 mStandby = false; 4535 if (mReqChannelCount != activeTrack->channelCount()) { 4536 mActiveTracks.remove(activeTrack); 4537 mActiveTracksGen++; 4538 mStartStopCond.broadcast(); 4539 continue; 4540 } 4541 if (readOnce) { 4542 mStartStopCond.broadcast(); 4543 // record start succeeds only if first read from audio input succeeds 4544 if (mBytesRead < 0) { 4545 mActiveTracks.remove(activeTrack); 4546 mActiveTracksGen++; 4547 continue; 4548 } 4549 activeTrack->mState = TrackBase::ACTIVE; 4550 } 4551 break; 4552 4553 case TrackBase::ACTIVE: 4554 break; 4555 4556 case TrackBase::IDLE: 4557 doSleep = true; 4558 continue; 4559 4560 default: 4561 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4562 } 4563 4564 lockEffectChains_l(effectChains); 4565 } 4566 4567 // thread mutex is now unlocked, mActiveTracks unknown, activeTrack != 0, kept, immutable 4568 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4569 4570 for (size_t i = 0; i < effectChains.size(); i ++) { 4571 // thread mutex is not locked, but effect chain is locked 4572 effectChains[i]->process_l(); 4573 } 4574 4575 AudioBufferProvider::Buffer buffer; 4576 buffer.frameCount = mFrameCount; 4577 status_t status = activeTrack->getNextBuffer(&buffer); 4578 if (status == NO_ERROR) { 4579 readOnce = true; 4580 size_t framesOut = buffer.frameCount; 4581 if (mResampler == NULL) { 4582 // no resampling 4583 while (framesOut) { 4584 size_t framesIn = mFrameCount - mRsmpInIndex; 4585 if (framesIn > 0) { 4586 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4587 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4588 activeTrack->mFrameSize; 4589 if (framesIn > framesOut) { 4590 framesIn = framesOut; 4591 } 4592 mRsmpInIndex += framesIn; 4593 framesOut -= framesIn; 4594 if (mChannelCount == mReqChannelCount) { 4595 memcpy(dst, src, framesIn * mFrameSize); 4596 } else { 4597 if (mChannelCount == 1) { 4598 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4599 (int16_t *)src, framesIn); 4600 } else { 4601 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4602 (int16_t *)src, framesIn); 4603 } 4604 } 4605 } 4606 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4607 void *readInto; 4608 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4609 readInto = buffer.raw; 4610 framesOut = 0; 4611 } else { 4612 readInto = mRsmpInBuffer; 4613 mRsmpInIndex = 0; 4614 } 4615 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4616 mBufferSize); 4617 if (mBytesRead <= 0) { 4618 // TODO: verify that it's benign to use a stale track state 4619 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4620 { 4621 ALOGE("Error reading audio input"); 4622 // Force input into standby so that it tries to 4623 // recover at next read attempt 4624 inputStandBy(); 4625 doSleep = true; 4626 } 4627 mRsmpInIndex = mFrameCount; 4628 framesOut = 0; 4629 buffer.frameCount = 0; 4630 } 4631#ifdef TEE_SINK 4632 else if (mTeeSink != 0) { 4633 (void) mTeeSink->write(readInto, 4634 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4635 } 4636#endif 4637 } 4638 } 4639 } else { 4640 // resampling 4641 4642 // avoid busy-waiting if client doesn't keep up 4643 bool madeProgress = false; 4644 4645 // keep mRsmpInBuffer full so resampler always has sufficient input 4646 for (;;) { 4647 int32_t rear = mRsmpInRear; 4648 ssize_t filled = rear - mRsmpInFront; 4649 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 4650 // exit once there is enough data in buffer for resampler 4651 if ((size_t) filled >= mRsmpInFrames) { 4652 break; 4653 } 4654 size_t avail = mRsmpInFramesP2 - filled; 4655 // Only try to read full HAL buffers. 4656 // But if the HAL read returns a partial buffer, use it. 4657 if (avail < mFrameCount) { 4658 ALOGE("insufficient space to read: avail %d < mFrameCount %d", 4659 avail, mFrameCount); 4660 break; 4661 } 4662 // If 'avail' is non-contiguous, first read past the nominal end of buffer, then 4663 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4664 rear &= mRsmpInFramesP2 - 1; 4665 mBytesRead = mInput->stream->read(mInput->stream, 4666 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4667 if (mBytesRead <= 0) { 4668 ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); 4669 break; 4670 } 4671 ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); 4672 size_t framesRead = mBytesRead / mFrameSize; 4673 ALOG_ASSERT(framesRead > 0); 4674 madeProgress = true; 4675 // If 'avail' was non-contiguous, we now correct for reading past end of buffer. 4676 size_t part1 = mRsmpInFramesP2 - rear; 4677 if (framesRead > part1) { 4678 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4679 (framesRead - part1) * mFrameSize); 4680 } 4681 mRsmpInRear += framesRead; 4682 } 4683 4684 if (!madeProgress) { 4685 ALOGV("Did not make progress"); 4686 usleep(((mFrameCount * 1000) / mSampleRate) * 1000); 4687 } 4688 4689 // resampler accumulates, but we only have one source track 4690 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4691 mResampler->resample(mRsmpOutBuffer, framesOut, 4692 this /* AudioBufferProvider* */); 4693 // ditherAndClamp() works as long as all buffers returned by 4694 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4695 if (mReqChannelCount == 1) { 4696 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4697 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4698 // the resampler always outputs stereo samples: 4699 // do post stereo to mono conversion 4700 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4701 framesOut); 4702 } else { 4703 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4704 } 4705 // now done with mRsmpOutBuffer 4706 4707 } 4708 if (mFramestoDrop == 0) { 4709 activeTrack->releaseBuffer(&buffer); 4710 } else { 4711 if (mFramestoDrop > 0) { 4712 mFramestoDrop -= buffer.frameCount; 4713 if (mFramestoDrop <= 0) { 4714 clearSyncStartEvent(); 4715 } 4716 } else { 4717 mFramestoDrop += buffer.frameCount; 4718 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4719 mSyncStartEvent->isCancelled()) { 4720 ALOGW("Synced record %s, session %d, trigger session %d", 4721 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4722 activeTrack->sessionId(), 4723 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4724 clearSyncStartEvent(); 4725 } 4726 } 4727 } 4728 activeTrack->clearOverflow(); 4729 } 4730 // client isn't retrieving buffers fast enough 4731 else { 4732 if (!activeTrack->setOverflow()) { 4733 nsecs_t now = systemTime(); 4734 if ((now - lastWarning) > kWarningThrottleNs) { 4735 ALOGW("RecordThread: buffer overflow"); 4736 lastWarning = now; 4737 } 4738 } 4739 // Release the processor for a while before asking for a new buffer. 4740 // This will give the application more chance to read from the buffer and 4741 // clear the overflow. 4742 doSleep = true; 4743 } 4744 4745 // enable changes in effect chain 4746 unlockEffectChains(effectChains); 4747 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4748 } 4749 4750 standbyIfNotAlreadyInStandby(); 4751 4752 { 4753 Mutex::Autolock _l(mLock); 4754 for (size_t i = 0; i < mTracks.size(); i++) { 4755 sp<RecordTrack> track = mTracks[i]; 4756 track->invalidate(); 4757 } 4758 mActiveTracks.clear(); 4759 mActiveTracksGen++; 4760 mStartStopCond.broadcast(); 4761 } 4762 4763 releaseWakeLock(); 4764 4765 ALOGV("RecordThread %p exiting", this); 4766 return false; 4767} 4768 4769void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4770{ 4771 if (!mStandby) { 4772 inputStandBy(); 4773 mStandby = true; 4774 } 4775} 4776 4777void AudioFlinger::RecordThread::inputStandBy() 4778{ 4779 mInput->stream->common.standby(&mInput->stream->common); 4780} 4781 4782sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4783 const sp<AudioFlinger::Client>& client, 4784 uint32_t sampleRate, 4785 audio_format_t format, 4786 audio_channel_mask_t channelMask, 4787 size_t *pFrameCount, 4788 int sessionId, 4789 int uid, 4790 IAudioFlinger::track_flags_t *flags, 4791 pid_t tid, 4792 status_t *status) 4793{ 4794 size_t frameCount = *pFrameCount; 4795 sp<RecordTrack> track; 4796 status_t lStatus; 4797 4798 lStatus = initCheck(); 4799 if (lStatus != NO_ERROR) { 4800 ALOGE("createRecordTrack_l() audio driver not initialized"); 4801 goto Exit; 4802 } 4803 // client expresses a preference for FAST, but we get the final say 4804 if (*flags & IAudioFlinger::TRACK_FAST) { 4805 if ( 4806 // use case: callback handler and frame count is default or at least as large as HAL 4807 ( 4808 (tid != -1) && 4809 ((frameCount == 0) || 4810 (frameCount >= mFrameCount)) 4811 ) && 4812 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4813 // mono or stereo 4814 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4815 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4816 // hardware sample rate 4817 (sampleRate == mSampleRate) && 4818 // record thread has an associated fast recorder 4819 hasFastRecorder() 4820 // FIXME test that RecordThread for this fast track has a capable output HAL 4821 // FIXME add a permission test also? 4822 ) { 4823 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4824 if (frameCount == 0) { 4825 frameCount = mFrameCount * kFastTrackMultiplier; 4826 } 4827 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4828 frameCount, mFrameCount); 4829 } else { 4830 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4831 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4832 "hasFastRecorder=%d tid=%d", 4833 frameCount, mFrameCount, format, 4834 audio_is_linear_pcm(format), 4835 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4836 *flags &= ~IAudioFlinger::TRACK_FAST; 4837 // For compatibility with AudioRecord calculation, buffer depth is forced 4838 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4839 // This is probably too conservative, but legacy application code may depend on it. 4840 // If you change this calculation, also review the start threshold which is related. 4841 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4842 size_t mNormalFrameCount = 2048; // FIXME 4843 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4844 if (minBufCount < 2) { 4845 minBufCount = 2; 4846 } 4847 size_t minFrameCount = mNormalFrameCount * minBufCount; 4848 if (frameCount < minFrameCount) { 4849 frameCount = minFrameCount; 4850 } 4851 } 4852 } 4853 *pFrameCount = frameCount; 4854 4855 // FIXME use flags and tid similar to createTrack_l() 4856 4857 { // scope for mLock 4858 Mutex::Autolock _l(mLock); 4859 4860 track = new RecordTrack(this, client, sampleRate, 4861 format, channelMask, frameCount, sessionId, uid); 4862 4863 lStatus = track->initCheck(); 4864 if (lStatus != NO_ERROR) { 4865 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4866 track.clear(); 4867 goto Exit; 4868 } 4869 mTracks.add(track); 4870 4871 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4872 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4873 mAudioFlinger->btNrecIsOff(); 4874 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4875 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4876 4877 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4878 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4879 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4880 // so ask activity manager to do this on our behalf 4881 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4882 } 4883 } 4884 lStatus = NO_ERROR; 4885 4886Exit: 4887 *status = lStatus; 4888 return track; 4889} 4890 4891status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4892 AudioSystem::sync_event_t event, 4893 int triggerSession) 4894{ 4895 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4896 sp<ThreadBase> strongMe = this; 4897 status_t status = NO_ERROR; 4898 4899 if (event == AudioSystem::SYNC_EVENT_NONE) { 4900 clearSyncStartEvent(); 4901 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4902 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4903 triggerSession, 4904 recordTrack->sessionId(), 4905 syncStartEventCallback, 4906 this); 4907 // Sync event can be cancelled by the trigger session if the track is not in a 4908 // compatible state in which case we start record immediately 4909 if (mSyncStartEvent->isCancelled()) { 4910 clearSyncStartEvent(); 4911 } else { 4912 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4913 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4914 } 4915 } 4916 4917 { 4918 // This section is a rendezvous between binder thread executing start() and RecordThread 4919 AutoMutex lock(mLock); 4920 if (mActiveTracks.size() > 0) { 4921 // FIXME does not work for multiple active tracks 4922 if (mActiveTracks.indexOf(recordTrack) != 0) { 4923 status = -EBUSY; 4924 } else if (recordTrack->mState == TrackBase::PAUSING) { 4925 recordTrack->mState = TrackBase::ACTIVE; 4926 } 4927 return status; 4928 } 4929 4930 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4931 recordTrack->mState = TrackBase::IDLE; 4932 mActiveTracks.add(recordTrack); 4933 mActiveTracksGen++; 4934 mLock.unlock(); 4935 status_t status = AudioSystem::startInput(mId); 4936 mLock.lock(); 4937 // FIXME should verify that mActiveTrack is still == recordTrack 4938 if (status != NO_ERROR) { 4939 mActiveTracks.remove(recordTrack); 4940 mActiveTracksGen++; 4941 clearSyncStartEvent(); 4942 return status; 4943 } 4944 // FIXME LEGACY 4945 mRsmpInIndex = mFrameCount; 4946 mRsmpInFront = 0; 4947 mRsmpInRear = 0; 4948 mRsmpInUnrel = 0; 4949 mBytesRead = 0; 4950 if (mResampler != NULL) { 4951 mResampler->reset(); 4952 } 4953 // FIXME hijacking a playback track state name which was intended for start after pause; 4954 // here 'STARTING_2' would be more accurate 4955 recordTrack->mState = TrackBase::RESUMING; 4956 // signal thread to start 4957 ALOGV("Signal record thread"); 4958 mWaitWorkCV.broadcast(); 4959 // do not wait for mStartStopCond if exiting 4960 if (exitPending()) { 4961 mActiveTracks.remove(recordTrack); 4962 mActiveTracksGen++; 4963 status = INVALID_OPERATION; 4964 goto startError; 4965 } 4966 // FIXME incorrect usage of wait: no explicit predicate or loop 4967 mStartStopCond.wait(mLock); 4968 if (mActiveTracks.indexOf(recordTrack) < 0) { 4969 ALOGV("Record failed to start"); 4970 status = BAD_VALUE; 4971 goto startError; 4972 } 4973 ALOGV("Record started OK"); 4974 return status; 4975 } 4976 4977startError: 4978 AudioSystem::stopInput(mId); 4979 clearSyncStartEvent(); 4980 return status; 4981} 4982 4983void AudioFlinger::RecordThread::clearSyncStartEvent() 4984{ 4985 if (mSyncStartEvent != 0) { 4986 mSyncStartEvent->cancel(); 4987 } 4988 mSyncStartEvent.clear(); 4989 mFramestoDrop = 0; 4990} 4991 4992void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4993{ 4994 sp<SyncEvent> strongEvent = event.promote(); 4995 4996 if (strongEvent != 0) { 4997 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4998 me->handleSyncStartEvent(strongEvent); 4999 } 5000} 5001 5002void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5003{ 5004 if (event == mSyncStartEvent) { 5005 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5006 // from audio HAL 5007 mFramestoDrop = mFrameCount * 2; 5008 } 5009} 5010 5011bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5012 ALOGV("RecordThread::stop"); 5013 AutoMutex _l(mLock); 5014 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5015 return false; 5016 } 5017 // note that threadLoop may still be processing the track at this point [without lock] 5018 recordTrack->mState = TrackBase::PAUSING; 5019 // do not wait for mStartStopCond if exiting 5020 if (exitPending()) { 5021 return true; 5022 } 5023 // FIXME incorrect usage of wait: no explicit predicate or loop 5024 mStartStopCond.wait(mLock); 5025 // if we have been restarted, recordTrack is in mActiveTracks here 5026 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5027 ALOGV("Record stopped OK"); 5028 return true; 5029 } 5030 return false; 5031} 5032 5033bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5034{ 5035 return false; 5036} 5037 5038status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5039{ 5040#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5041 if (!isValidSyncEvent(event)) { 5042 return BAD_VALUE; 5043 } 5044 5045 int eventSession = event->triggerSession(); 5046 status_t ret = NAME_NOT_FOUND; 5047 5048 Mutex::Autolock _l(mLock); 5049 5050 for (size_t i = 0; i < mTracks.size(); i++) { 5051 sp<RecordTrack> track = mTracks[i]; 5052 if (eventSession == track->sessionId()) { 5053 (void) track->setSyncEvent(event); 5054 ret = NO_ERROR; 5055 } 5056 } 5057 return ret; 5058#else 5059 return BAD_VALUE; 5060#endif 5061} 5062 5063// destroyTrack_l() must be called with ThreadBase::mLock held 5064void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5065{ 5066 track->terminate(); 5067 track->mState = TrackBase::STOPPED; 5068 // active tracks are removed by threadLoop() 5069 if (mActiveTracks.indexOf(track) < 0) { 5070 removeTrack_l(track); 5071 } 5072} 5073 5074void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5075{ 5076 mTracks.remove(track); 5077 // need anything related to effects here? 5078} 5079 5080void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5081{ 5082 dumpInternals(fd, args); 5083 dumpTracks(fd, args); 5084 dumpEffectChains(fd, args); 5085} 5086 5087void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5088{ 5089 const size_t SIZE = 256; 5090 char buffer[SIZE]; 5091 String8 result; 5092 5093 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5094 result.append(buffer); 5095 5096 if (mActiveTracks.size() > 0) { 5097 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5098 result.append(buffer); 5099 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 5100 result.append(buffer); 5101 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5102 result.append(buffer); 5103 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 5104 result.append(buffer); 5105 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 5106 result.append(buffer); 5107 } else { 5108 result.append("No active record client\n"); 5109 } 5110 5111 write(fd, result.string(), result.size()); 5112 5113 dumpBase(fd, args); 5114} 5115 5116void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5117{ 5118 const size_t SIZE = 256; 5119 char buffer[SIZE]; 5120 String8 result; 5121 5122 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 5123 result.append(buffer); 5124 RecordTrack::appendDumpHeader(result); 5125 for (size_t i = 0; i < mTracks.size(); ++i) { 5126 sp<RecordTrack> track = mTracks[i]; 5127 if (track != 0) { 5128 track->dump(buffer, SIZE); 5129 result.append(buffer); 5130 } 5131 } 5132 5133 size_t size = mActiveTracks.size(); 5134 if (size > 0) { 5135 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5136 result.append(buffer); 5137 RecordTrack::appendDumpHeader(result); 5138 for (size_t i = 0; i < size; ++i) { 5139 sp<RecordTrack> track = mActiveTracks[i]; 5140 track->dump(buffer, SIZE); 5141 result.append(buffer); 5142 } 5143 5144 } 5145 write(fd, result.string(), result.size()); 5146} 5147 5148// AudioBufferProvider interface 5149status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5150{ 5151 int32_t rear = mRsmpInRear; 5152 int32_t front = mRsmpInFront; 5153 ssize_t filled = rear - front; 5154 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 5155 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5156 front &= mRsmpInFramesP2 - 1; 5157 size_t part1 = mRsmpInFramesP2 - front; 5158 if (part1 > (size_t) filled) { 5159 part1 = filled; 5160 } 5161 size_t ask = buffer->frameCount; 5162 ALOG_ASSERT(ask > 0); 5163 if (part1 > ask) { 5164 part1 = ask; 5165 } 5166 if (part1 == 0) { 5167 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5168 ALOGE("RecordThread::getNextBuffer() starved"); 5169 buffer->raw = NULL; 5170 buffer->frameCount = 0; 5171 mRsmpInUnrel = 0; 5172 return NOT_ENOUGH_DATA; 5173 } 5174 5175 buffer->raw = mRsmpInBuffer + front * mChannelCount; 5176 buffer->frameCount = part1; 5177 mRsmpInUnrel = part1; 5178 return NO_ERROR; 5179} 5180 5181// AudioBufferProvider interface 5182void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5183{ 5184 size_t stepCount = buffer->frameCount; 5185 if (stepCount == 0) { 5186 return; 5187 } 5188 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 5189 mRsmpInUnrel -= stepCount; 5190 mRsmpInFront += stepCount; 5191 buffer->raw = NULL; 5192 buffer->frameCount = 0; 5193} 5194 5195bool AudioFlinger::RecordThread::checkForNewParameters_l() 5196{ 5197 bool reconfig = false; 5198 5199 while (!mNewParameters.isEmpty()) { 5200 status_t status = NO_ERROR; 5201 String8 keyValuePair = mNewParameters[0]; 5202 AudioParameter param = AudioParameter(keyValuePair); 5203 int value; 5204 audio_format_t reqFormat = mFormat; 5205 uint32_t reqSamplingRate = mReqSampleRate; 5206 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5207 5208 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5209 reqSamplingRate = value; 5210 reconfig = true; 5211 } 5212 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5213 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5214 status = BAD_VALUE; 5215 } else { 5216 reqFormat = (audio_format_t) value; 5217 reconfig = true; 5218 } 5219 } 5220 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5221 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5222 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5223 status = BAD_VALUE; 5224 } else { 5225 reqChannelMask = mask; 5226 reconfig = true; 5227 } 5228 } 5229 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5230 // do not accept frame count changes if tracks are open as the track buffer 5231 // size depends on frame count and correct behavior would not be guaranteed 5232 // if frame count is changed after track creation 5233 if (mActiveTracks.size() > 0) { 5234 status = INVALID_OPERATION; 5235 } else { 5236 reconfig = true; 5237 } 5238 } 5239 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5240 // forward device change to effects that have requested to be 5241 // aware of attached audio device. 5242 for (size_t i = 0; i < mEffectChains.size(); i++) { 5243 mEffectChains[i]->setDevice_l(value); 5244 } 5245 5246 // store input device and output device but do not forward output device to audio HAL. 5247 // Note that status is ignored by the caller for output device 5248 // (see AudioFlinger::setParameters() 5249 if (audio_is_output_devices(value)) { 5250 mOutDevice = value; 5251 status = BAD_VALUE; 5252 } else { 5253 mInDevice = value; 5254 // disable AEC and NS if the device is a BT SCO headset supporting those 5255 // pre processings 5256 if (mTracks.size() > 0) { 5257 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5258 mAudioFlinger->btNrecIsOff(); 5259 for (size_t i = 0; i < mTracks.size(); i++) { 5260 sp<RecordTrack> track = mTracks[i]; 5261 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5262 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5263 } 5264 } 5265 } 5266 } 5267 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5268 mAudioSource != (audio_source_t)value) { 5269 // forward device change to effects that have requested to be 5270 // aware of attached audio device. 5271 for (size_t i = 0; i < mEffectChains.size(); i++) { 5272 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5273 } 5274 mAudioSource = (audio_source_t)value; 5275 } 5276 5277 if (status == NO_ERROR) { 5278 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5279 keyValuePair.string()); 5280 if (status == INVALID_OPERATION) { 5281 inputStandBy(); 5282 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5283 keyValuePair.string()); 5284 } 5285 if (reconfig) { 5286 if (status == BAD_VALUE && 5287 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5288 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5289 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5290 <= (2 * reqSamplingRate)) && 5291 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5292 <= FCC_2 && 5293 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5294 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5295 status = NO_ERROR; 5296 } 5297 if (status == NO_ERROR) { 5298 readInputParameters(); 5299 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5300 } 5301 } 5302 } 5303 5304 mNewParameters.removeAt(0); 5305 5306 mParamStatus = status; 5307 mParamCond.signal(); 5308 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5309 // already timed out waiting for the status and will never signal the condition. 5310 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5311 } 5312 return reconfig; 5313} 5314 5315String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5316{ 5317 Mutex::Autolock _l(mLock); 5318 if (initCheck() != NO_ERROR) { 5319 return String8(); 5320 } 5321 5322 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5323 const String8 out_s8(s); 5324 free(s); 5325 return out_s8; 5326} 5327 5328void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5329 AudioSystem::OutputDescriptor desc; 5330 const void *param2 = NULL; 5331 5332 switch (event) { 5333 case AudioSystem::INPUT_OPENED: 5334 case AudioSystem::INPUT_CONFIG_CHANGED: 5335 desc.channelMask = mChannelMask; 5336 desc.samplingRate = mSampleRate; 5337 desc.format = mFormat; 5338 desc.frameCount = mFrameCount; 5339 desc.latency = 0; 5340 param2 = &desc; 5341 break; 5342 5343 case AudioSystem::INPUT_CLOSED: 5344 default: 5345 break; 5346 } 5347 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5348} 5349 5350void AudioFlinger::RecordThread::readInputParameters() 5351{ 5352 delete[] mRsmpInBuffer; 5353 // mRsmpInBuffer is always assigned a new[] below 5354 delete[] mRsmpOutBuffer; 5355 mRsmpOutBuffer = NULL; 5356 delete mResampler; 5357 mResampler = NULL; 5358 5359 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5360 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5361 mChannelCount = popcount(mChannelMask); 5362 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5363 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5364 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5365 } 5366 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5367 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5368 mFrameCount = mBufferSize / mFrameSize; 5369 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5370 // 1 full output buffer, regardless of the alignment of the available input. 5371 mRsmpInFrames = mFrameCount * 3; 5372 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5373 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5374 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5375 mRsmpInFront = 0; 5376 mRsmpInRear = 0; 5377 mRsmpInUnrel = 0; 5378 5379 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5380 mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); 5381 mResampler->setSampleRate(mSampleRate); 5382 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5383 // resampler always outputs stereo 5384 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5385 } 5386 mRsmpInIndex = mFrameCount; 5387} 5388 5389uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5390{ 5391 Mutex::Autolock _l(mLock); 5392 if (initCheck() != NO_ERROR) { 5393 return 0; 5394 } 5395 5396 return mInput->stream->get_input_frames_lost(mInput->stream); 5397} 5398 5399uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5400{ 5401 Mutex::Autolock _l(mLock); 5402 uint32_t result = 0; 5403 if (getEffectChain_l(sessionId) != 0) { 5404 result = EFFECT_SESSION; 5405 } 5406 5407 for (size_t i = 0; i < mTracks.size(); ++i) { 5408 if (sessionId == mTracks[i]->sessionId()) { 5409 result |= TRACK_SESSION; 5410 break; 5411 } 5412 } 5413 5414 return result; 5415} 5416 5417KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5418{ 5419 KeyedVector<int, bool> ids; 5420 Mutex::Autolock _l(mLock); 5421 for (size_t j = 0; j < mTracks.size(); ++j) { 5422 sp<RecordThread::RecordTrack> track = mTracks[j]; 5423 int sessionId = track->sessionId(); 5424 if (ids.indexOfKey(sessionId) < 0) { 5425 ids.add(sessionId, true); 5426 } 5427 } 5428 return ids; 5429} 5430 5431AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5432{ 5433 Mutex::Autolock _l(mLock); 5434 AudioStreamIn *input = mInput; 5435 mInput = NULL; 5436 return input; 5437} 5438 5439// this method must always be called either with ThreadBase mLock held or inside the thread loop 5440audio_stream_t* AudioFlinger::RecordThread::stream() const 5441{ 5442 if (mInput == NULL) { 5443 return NULL; 5444 } 5445 return &mInput->stream->common; 5446} 5447 5448status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5449{ 5450 // only one chain per input thread 5451 if (mEffectChains.size() != 0) { 5452 return INVALID_OPERATION; 5453 } 5454 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5455 5456 chain->setInBuffer(NULL); 5457 chain->setOutBuffer(NULL); 5458 5459 checkSuspendOnAddEffectChain_l(chain); 5460 5461 mEffectChains.add(chain); 5462 5463 return NO_ERROR; 5464} 5465 5466size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5467{ 5468 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5469 ALOGW_IF(mEffectChains.size() != 1, 5470 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5471 chain.get(), mEffectChains.size(), this); 5472 if (mEffectChains.size() == 1) { 5473 mEffectChains.removeAt(0); 5474 } 5475 return 0; 5476} 5477 5478}; // namespace android 5479