Threads.cpp revision 9cae217050aa1347d4ac5053c305754879e3f97f
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 285 for (size_t i = 0; i < mConfigEvents.size(); i++) { 286 delete mConfigEvents[i]; 287 } 288 mConfigEvents.clear(); 289 290 mParamCond.broadcast(); 291 // do not lock the mutex in destructor 292 releaseWakeLock_l(); 293 if (mPowerManager != 0) { 294 sp<IBinder> binder = mPowerManager->asBinder(); 295 binder->unlinkToDeath(mDeathRecipient); 296 } 297} 298 299void AudioFlinger::ThreadBase::exit() 300{ 301 ALOGV("ThreadBase::exit"); 302 // do any cleanup required for exit to succeed 303 preExit(); 304 { 305 // This lock prevents the following race in thread (uniprocessor for illustration): 306 // if (!exitPending()) { 307 // // context switch from here to exit() 308 // // exit() calls requestExit(), what exitPending() observes 309 // // exit() calls signal(), which is dropped since no waiters 310 // // context switch back from exit() to here 311 // mWaitWorkCV.wait(...); 312 // // now thread is hung 313 // } 314 AutoMutex lock(mLock); 315 requestExit(); 316 mWaitWorkCV.broadcast(); 317 } 318 // When Thread::requestExitAndWait is made virtual and this method is renamed to 319 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 320 requestExitAndWait(); 321} 322 323status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 324{ 325 status_t status; 326 327 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 328 Mutex::Autolock _l(mLock); 329 330 mNewParameters.add(keyValuePairs); 331 mWaitWorkCV.signal(); 332 // wait condition with timeout in case the thread loop has exited 333 // before the request could be processed 334 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 335 status = mParamStatus; 336 mWaitWorkCV.signal(); 337 } else { 338 status = TIMED_OUT; 339 } 340 return status; 341} 342 343void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 344{ 345 Mutex::Autolock _l(mLock); 346 sendIoConfigEvent_l(event, param); 347} 348 349// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 350void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 351{ 352 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 353 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 354 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 355 param); 356 mWaitWorkCV.signal(); 357} 358 359// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 360void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 361{ 362 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 363 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 364 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 365 mConfigEvents.size(), pid, tid, prio); 366 mWaitWorkCV.signal(); 367} 368 369void AudioFlinger::ThreadBase::processConfigEvents() 370{ 371 mLock.lock(); 372 while (!mConfigEvents.isEmpty()) { 373 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 374 ConfigEvent *event = mConfigEvents[0]; 375 mConfigEvents.removeAt(0); 376 // release mLock before locking AudioFlinger mLock: lock order is always 377 // AudioFlinger then ThreadBase to avoid cross deadlock 378 mLock.unlock(); 379 switch(event->type()) { 380 case CFG_EVENT_PRIO: { 381 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 382 // FIXME Need to understand why this has be done asynchronously 383 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 384 true /*asynchronous*/); 385 if (err != 0) { 386 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 387 "error %d", 388 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 389 } 390 } break; 391 case CFG_EVENT_IO: { 392 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 393 mAudioFlinger->mLock.lock(); 394 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 395 mAudioFlinger->mLock.unlock(); 396 } break; 397 default: 398 ALOGE("processConfigEvents() unknown event type %d", event->type()); 399 break; 400 } 401 delete event; 402 mLock.lock(); 403 } 404 mLock.unlock(); 405} 406 407void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 408{ 409 const size_t SIZE = 256; 410 char buffer[SIZE]; 411 String8 result; 412 413 bool locked = AudioFlinger::dumpTryLock(mLock); 414 if (!locked) { 415 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 416 write(fd, buffer, strlen(buffer)); 417 } 418 419 snprintf(buffer, SIZE, "io handle: %d\n", mId); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 432 result.append(buffer); 433 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 434 result.append(buffer); 435 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 436 result.append(buffer); 437 438 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 439 result.append(buffer); 440 result.append(" Index Command"); 441 for (size_t i = 0; i < mNewParameters.size(); ++i) { 442 snprintf(buffer, SIZE, "\n %02d ", i); 443 result.append(buffer); 444 result.append(mNewParameters[i]); 445 } 446 447 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 448 result.append(buffer); 449 for (size_t i = 0; i < mConfigEvents.size(); i++) { 450 mConfigEvents[i]->dump(buffer, SIZE); 451 result.append(buffer); 452 } 453 result.append("\n"); 454 455 write(fd, result.string(), result.size()); 456 457 if (locked) { 458 mLock.unlock(); 459 } 460} 461 462void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 463{ 464 const size_t SIZE = 256; 465 char buffer[SIZE]; 466 String8 result; 467 468 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 469 write(fd, buffer, strlen(buffer)); 470 471 for (size_t i = 0; i < mEffectChains.size(); ++i) { 472 sp<EffectChain> chain = mEffectChains[i]; 473 if (chain != 0) { 474 chain->dump(fd, args); 475 } 476 } 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 480{ 481 Mutex::Autolock _l(mLock); 482 acquireWakeLock_l(uid); 483} 484 485String16 AudioFlinger::ThreadBase::getWakeLockTag() 486{ 487 switch (mType) { 488 case MIXER: 489 return String16("AudioMix"); 490 case DIRECT: 491 return String16("AudioDirectOut"); 492 case DUPLICATING: 493 return String16("AudioDup"); 494 case RECORD: 495 return String16("AudioIn"); 496 case OFFLOAD: 497 return String16("AudioOffload"); 498 default: 499 ALOG_ASSERT(false); 500 return String16("AudioUnknown"); 501 } 502} 503 504void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 505{ 506 getPowerManager_l(); 507 if (mPowerManager != 0) { 508 sp<IBinder> binder = new BBinder(); 509 status_t status; 510 if (uid >= 0) { 511 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 512 binder, 513 getWakeLockTag(), 514 String16("media"), 515 uid); 516 } else { 517 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 518 binder, 519 getWakeLockTag(), 520 String16("media")); 521 } 522 if (status == NO_ERROR) { 523 mWakeLockToken = binder; 524 } 525 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 526 } 527} 528 529void AudioFlinger::ThreadBase::releaseWakeLock() 530{ 531 Mutex::Autolock _l(mLock); 532 releaseWakeLock_l(); 533} 534 535void AudioFlinger::ThreadBase::releaseWakeLock_l() 536{ 537 if (mWakeLockToken != 0) { 538 ALOGV("releaseWakeLock_l() %s", mName); 539 if (mPowerManager != 0) { 540 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 541 } 542 mWakeLockToken.clear(); 543 } 544} 545 546void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 547 Mutex::Autolock _l(mLock); 548 updateWakeLockUids_l(uids); 549} 550 551void AudioFlinger::ThreadBase::getPowerManager_l() { 552 553 if (mPowerManager == 0) { 554 // use checkService() to avoid blocking if power service is not up yet 555 sp<IBinder> binder = 556 defaultServiceManager()->checkService(String16("power")); 557 if (binder == 0) { 558 ALOGW("Thread %s cannot connect to the power manager service", mName); 559 } else { 560 mPowerManager = interface_cast<IPowerManager>(binder); 561 binder->linkToDeath(mDeathRecipient); 562 } 563 } 564} 565 566void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 567 568 getPowerManager_l(); 569 if (mWakeLockToken == NULL) { 570 ALOGE("no wake lock to update!"); 571 return; 572 } 573 if (mPowerManager != 0) { 574 sp<IBinder> binder = new BBinder(); 575 status_t status; 576 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 577 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 578 } 579} 580 581void AudioFlinger::ThreadBase::clearPowerManager() 582{ 583 Mutex::Autolock _l(mLock); 584 releaseWakeLock_l(); 585 mPowerManager.clear(); 586} 587 588void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 589{ 590 sp<ThreadBase> thread = mThread.promote(); 591 if (thread != 0) { 592 thread->clearPowerManager(); 593 } 594 ALOGW("power manager service died !!!"); 595} 596 597void AudioFlinger::ThreadBase::setEffectSuspended( 598 const effect_uuid_t *type, bool suspend, int sessionId) 599{ 600 Mutex::Autolock _l(mLock); 601 setEffectSuspended_l(type, suspend, sessionId); 602} 603 604void AudioFlinger::ThreadBase::setEffectSuspended_l( 605 const effect_uuid_t *type, bool suspend, int sessionId) 606{ 607 sp<EffectChain> chain = getEffectChain_l(sessionId); 608 if (chain != 0) { 609 if (type != NULL) { 610 chain->setEffectSuspended_l(type, suspend); 611 } else { 612 chain->setEffectSuspendedAll_l(suspend); 613 } 614 } 615 616 updateSuspendedSessions_l(type, suspend, sessionId); 617} 618 619void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 620{ 621 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 622 if (index < 0) { 623 return; 624 } 625 626 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 627 mSuspendedSessions.valueAt(index); 628 629 for (size_t i = 0; i < sessionEffects.size(); i++) { 630 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 631 for (int j = 0; j < desc->mRefCount; j++) { 632 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 633 chain->setEffectSuspendedAll_l(true); 634 } else { 635 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 636 desc->mType.timeLow); 637 chain->setEffectSuspended_l(&desc->mType, true); 638 } 639 } 640 } 641} 642 643void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 644 bool suspend, 645 int sessionId) 646{ 647 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 648 649 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 650 651 if (suspend) { 652 if (index >= 0) { 653 sessionEffects = mSuspendedSessions.valueAt(index); 654 } else { 655 mSuspendedSessions.add(sessionId, sessionEffects); 656 } 657 } else { 658 if (index < 0) { 659 return; 660 } 661 sessionEffects = mSuspendedSessions.valueAt(index); 662 } 663 664 665 int key = EffectChain::kKeyForSuspendAll; 666 if (type != NULL) { 667 key = type->timeLow; 668 } 669 index = sessionEffects.indexOfKey(key); 670 671 sp<SuspendedSessionDesc> desc; 672 if (suspend) { 673 if (index >= 0) { 674 desc = sessionEffects.valueAt(index); 675 } else { 676 desc = new SuspendedSessionDesc(); 677 if (type != NULL) { 678 desc->mType = *type; 679 } 680 sessionEffects.add(key, desc); 681 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 682 } 683 desc->mRefCount++; 684 } else { 685 if (index < 0) { 686 return; 687 } 688 desc = sessionEffects.valueAt(index); 689 if (--desc->mRefCount == 0) { 690 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 691 sessionEffects.removeItemsAt(index); 692 if (sessionEffects.isEmpty()) { 693 ALOGV("updateSuspendedSessions_l() restore removing session %d", 694 sessionId); 695 mSuspendedSessions.removeItem(sessionId); 696 } 697 } 698 } 699 if (!sessionEffects.isEmpty()) { 700 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 701 } 702} 703 704void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 705 bool enabled, 706 int sessionId) 707{ 708 Mutex::Autolock _l(mLock); 709 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 710} 711 712void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 713 bool enabled, 714 int sessionId) 715{ 716 if (mType != RECORD) { 717 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 718 // another session. This gives the priority to well behaved effect control panels 719 // and applications not using global effects. 720 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 721 // global effects 722 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 723 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 724 } 725 } 726 727 sp<EffectChain> chain = getEffectChain_l(sessionId); 728 if (chain != 0) { 729 chain->checkSuspendOnEffectEnabled(effect, enabled); 730 } 731} 732 733// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 734sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 735 const sp<AudioFlinger::Client>& client, 736 const sp<IEffectClient>& effectClient, 737 int32_t priority, 738 int sessionId, 739 effect_descriptor_t *desc, 740 int *enabled, 741 status_t *status 742 ) 743{ 744 sp<EffectModule> effect; 745 sp<EffectHandle> handle; 746 status_t lStatus; 747 sp<EffectChain> chain; 748 bool chainCreated = false; 749 bool effectCreated = false; 750 bool effectRegistered = false; 751 752 lStatus = initCheck(); 753 if (lStatus != NO_ERROR) { 754 ALOGW("createEffect_l() Audio driver not initialized."); 755 goto Exit; 756 } 757 758 // Allow global effects only on offloaded and mixer threads 759 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 760 switch (mType) { 761 case MIXER: 762 case OFFLOAD: 763 break; 764 case DIRECT: 765 case DUPLICATING: 766 case RECORD: 767 default: 768 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 769 lStatus = BAD_VALUE; 770 goto Exit; 771 } 772 } 773 774 // Only Pre processor effects are allowed on input threads and only on input threads 775 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 776 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 777 desc->name, desc->flags, mType); 778 lStatus = BAD_VALUE; 779 goto Exit; 780 } 781 782 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 783 784 { // scope for mLock 785 Mutex::Autolock _l(mLock); 786 787 // check for existing effect chain with the requested audio session 788 chain = getEffectChain_l(sessionId); 789 if (chain == 0) { 790 // create a new chain for this session 791 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 792 chain = new EffectChain(this, sessionId); 793 addEffectChain_l(chain); 794 chain->setStrategy(getStrategyForSession_l(sessionId)); 795 chainCreated = true; 796 } else { 797 effect = chain->getEffectFromDesc_l(desc); 798 } 799 800 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 801 802 if (effect == 0) { 803 int id = mAudioFlinger->nextUniqueId(); 804 // Check CPU and memory usage 805 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 806 if (lStatus != NO_ERROR) { 807 goto Exit; 808 } 809 effectRegistered = true; 810 // create a new effect module if none present in the chain 811 effect = new EffectModule(this, chain, desc, id, sessionId); 812 lStatus = effect->status(); 813 if (lStatus != NO_ERROR) { 814 goto Exit; 815 } 816 effect->setOffloaded(mType == OFFLOAD, mId); 817 818 lStatus = chain->addEffect_l(effect); 819 if (lStatus != NO_ERROR) { 820 goto Exit; 821 } 822 effectCreated = true; 823 824 effect->setDevice(mOutDevice); 825 effect->setDevice(mInDevice); 826 effect->setMode(mAudioFlinger->getMode()); 827 effect->setAudioSource(mAudioSource); 828 } 829 // create effect handle and connect it to effect module 830 handle = new EffectHandle(effect, client, effectClient, priority); 831 lStatus = effect->addHandle(handle.get()); 832 if (enabled != NULL) { 833 *enabled = (int)effect->isEnabled(); 834 } 835 } 836 837Exit: 838 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 839 Mutex::Autolock _l(mLock); 840 if (effectCreated) { 841 chain->removeEffect_l(effect); 842 } 843 if (effectRegistered) { 844 AudioSystem::unregisterEffect(effect->id()); 845 } 846 if (chainCreated) { 847 removeEffectChain_l(chain); 848 } 849 handle.clear(); 850 } 851 852 if (status != NULL) { 853 *status = lStatus; 854 } 855 return handle; 856} 857 858sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 859{ 860 Mutex::Autolock _l(mLock); 861 return getEffect_l(sessionId, effectId); 862} 863 864sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 865{ 866 sp<EffectChain> chain = getEffectChain_l(sessionId); 867 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 868} 869 870// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 871// PlaybackThread::mLock held 872status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 873{ 874 // check for existing effect chain with the requested audio session 875 int sessionId = effect->sessionId(); 876 sp<EffectChain> chain = getEffectChain_l(sessionId); 877 bool chainCreated = false; 878 879 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 880 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 881 this, effect->desc().name, effect->desc().flags); 882 883 if (chain == 0) { 884 // create a new chain for this session 885 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 886 chain = new EffectChain(this, sessionId); 887 addEffectChain_l(chain); 888 chain->setStrategy(getStrategyForSession_l(sessionId)); 889 chainCreated = true; 890 } 891 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 892 893 if (chain->getEffectFromId_l(effect->id()) != 0) { 894 ALOGW("addEffect_l() %p effect %s already present in chain %p", 895 this, effect->desc().name, chain.get()); 896 return BAD_VALUE; 897 } 898 899 effect->setOffloaded(mType == OFFLOAD, mId); 900 901 status_t status = chain->addEffect_l(effect); 902 if (status != NO_ERROR) { 903 if (chainCreated) { 904 removeEffectChain_l(chain); 905 } 906 return status; 907 } 908 909 effect->setDevice(mOutDevice); 910 effect->setDevice(mInDevice); 911 effect->setMode(mAudioFlinger->getMode()); 912 effect->setAudioSource(mAudioSource); 913 return NO_ERROR; 914} 915 916void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 917 918 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 919 effect_descriptor_t desc = effect->desc(); 920 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 921 detachAuxEffect_l(effect->id()); 922 } 923 924 sp<EffectChain> chain = effect->chain().promote(); 925 if (chain != 0) { 926 // remove effect chain if removing last effect 927 if (chain->removeEffect_l(effect) == 0) { 928 removeEffectChain_l(chain); 929 } 930 } else { 931 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 932 } 933} 934 935void AudioFlinger::ThreadBase::lockEffectChains_l( 936 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 937{ 938 effectChains = mEffectChains; 939 for (size_t i = 0; i < mEffectChains.size(); i++) { 940 mEffectChains[i]->lock(); 941 } 942} 943 944void AudioFlinger::ThreadBase::unlockEffectChains( 945 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 946{ 947 for (size_t i = 0; i < effectChains.size(); i++) { 948 effectChains[i]->unlock(); 949 } 950} 951 952sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 953{ 954 Mutex::Autolock _l(mLock); 955 return getEffectChain_l(sessionId); 956} 957 958sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 959{ 960 size_t size = mEffectChains.size(); 961 for (size_t i = 0; i < size; i++) { 962 if (mEffectChains[i]->sessionId() == sessionId) { 963 return mEffectChains[i]; 964 } 965 } 966 return 0; 967} 968 969void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 970{ 971 Mutex::Autolock _l(mLock); 972 size_t size = mEffectChains.size(); 973 for (size_t i = 0; i < size; i++) { 974 mEffectChains[i]->setMode_l(mode); 975 } 976} 977 978void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 979 EffectHandle *handle, 980 bool unpinIfLast) { 981 982 Mutex::Autolock _l(mLock); 983 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 984 // delete the effect module if removing last handle on it 985 if (effect->removeHandle(handle) == 0) { 986 if (!effect->isPinned() || unpinIfLast) { 987 removeEffect_l(effect); 988 AudioSystem::unregisterEffect(effect->id()); 989 } 990 } 991} 992 993// ---------------------------------------------------------------------------- 994// Playback 995// ---------------------------------------------------------------------------- 996 997AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 998 AudioStreamOut* output, 999 audio_io_handle_t id, 1000 audio_devices_t device, 1001 type_t type) 1002 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1003 mNormalFrameCount(0), mMixBuffer(NULL), 1004 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1005 mActiveTracksGeneration(0), 1006 // mStreamTypes[] initialized in constructor body 1007 mOutput(output), 1008 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1009 mMixerStatus(MIXER_IDLE), 1010 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1011 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1012 mBytesRemaining(0), 1013 mCurrentWriteLength(0), 1014 mUseAsyncWrite(false), 1015 mWriteAckSequence(0), 1016 mDrainSequence(0), 1017 mSignalPending(false), 1018 mScreenState(AudioFlinger::mScreenState), 1019 // index 0 is reserved for normal mixer's submix 1020 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1021 // mLatchD, mLatchQ, 1022 mLatchDValid(false), mLatchQValid(false) 1023{ 1024 snprintf(mName, kNameLength, "AudioOut_%X", id); 1025 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1026 1027 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1028 // it would be safer to explicitly pass initial masterVolume/masterMute as 1029 // parameter. 1030 // 1031 // If the HAL we are using has support for master volume or master mute, 1032 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1033 // and the mute set to false). 1034 mMasterVolume = audioFlinger->masterVolume_l(); 1035 mMasterMute = audioFlinger->masterMute_l(); 1036 if (mOutput && mOutput->audioHwDev) { 1037 if (mOutput->audioHwDev->canSetMasterVolume()) { 1038 mMasterVolume = 1.0; 1039 } 1040 1041 if (mOutput->audioHwDev->canSetMasterMute()) { 1042 mMasterMute = false; 1043 } 1044 } 1045 1046 readOutputParameters(); 1047 1048 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1049 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1050 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1051 stream = (audio_stream_type_t) (stream + 1)) { 1052 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1053 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1054 } 1055 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1056 // because mAudioFlinger doesn't have one to copy from 1057} 1058 1059AudioFlinger::PlaybackThread::~PlaybackThread() 1060{ 1061 mAudioFlinger->unregisterWriter(mNBLogWriter); 1062 delete [] mAllocMixBuffer; 1063} 1064 1065void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1066{ 1067 dumpInternals(fd, args); 1068 dumpTracks(fd, args); 1069 dumpEffectChains(fd, args); 1070} 1071 1072void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1073{ 1074 const size_t SIZE = 256; 1075 char buffer[SIZE]; 1076 String8 result; 1077 1078 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1079 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1080 const stream_type_t *st = &mStreamTypes[i]; 1081 if (i > 0) { 1082 result.appendFormat(", "); 1083 } 1084 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1085 if (st->mute) { 1086 result.append("M"); 1087 } 1088 } 1089 result.append("\n"); 1090 write(fd, result.string(), result.length()); 1091 result.clear(); 1092 1093 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1094 result.append(buffer); 1095 Track::appendDumpHeader(result); 1096 for (size_t i = 0; i < mTracks.size(); ++i) { 1097 sp<Track> track = mTracks[i]; 1098 if (track != 0) { 1099 track->dump(buffer, SIZE); 1100 result.append(buffer); 1101 } 1102 } 1103 1104 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1105 result.append(buffer); 1106 Track::appendDumpHeader(result); 1107 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1108 sp<Track> track = mActiveTracks[i].promote(); 1109 if (track != 0) { 1110 track->dump(buffer, SIZE); 1111 result.append(buffer); 1112 } 1113 } 1114 write(fd, result.string(), result.size()); 1115 1116 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1117 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1118 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1119 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1120} 1121 1122void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1123{ 1124 const size_t SIZE = 256; 1125 char buffer[SIZE]; 1126 String8 result; 1127 1128 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1129 result.append(buffer); 1130 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1131 result.append(buffer); 1132 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1133 ns2ms(systemTime() - mLastWriteTime)); 1134 result.append(buffer); 1135 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1136 result.append(buffer); 1137 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1138 result.append(buffer); 1139 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1140 result.append(buffer); 1141 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1142 result.append(buffer); 1143 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1144 result.append(buffer); 1145 write(fd, result.string(), result.size()); 1146 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1147 1148 dumpBase(fd, args); 1149} 1150 1151// Thread virtuals 1152status_t AudioFlinger::PlaybackThread::readyToRun() 1153{ 1154 status_t status = initCheck(); 1155 if (status == NO_ERROR) { 1156 ALOGI("AudioFlinger's thread %p ready to run", this); 1157 } else { 1158 ALOGE("No working audio driver found."); 1159 } 1160 return status; 1161} 1162 1163void AudioFlinger::PlaybackThread::onFirstRef() 1164{ 1165 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1166} 1167 1168// ThreadBase virtuals 1169void AudioFlinger::PlaybackThread::preExit() 1170{ 1171 ALOGV(" preExit()"); 1172 // FIXME this is using hard-coded strings but in the future, this functionality will be 1173 // converted to use audio HAL extensions required to support tunneling 1174 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1175} 1176 1177// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1178sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1179 const sp<AudioFlinger::Client>& client, 1180 audio_stream_type_t streamType, 1181 uint32_t sampleRate, 1182 audio_format_t format, 1183 audio_channel_mask_t channelMask, 1184 size_t frameCount, 1185 const sp<IMemory>& sharedBuffer, 1186 int sessionId, 1187 IAudioFlinger::track_flags_t *flags, 1188 pid_t tid, 1189 int uid, 1190 status_t *status) 1191{ 1192 sp<Track> track; 1193 status_t lStatus; 1194 1195 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1196 1197 // client expresses a preference for FAST, but we get the final say 1198 if (*flags & IAudioFlinger::TRACK_FAST) { 1199 if ( 1200 // not timed 1201 (!isTimed) && 1202 // either of these use cases: 1203 ( 1204 // use case 1: shared buffer with any frame count 1205 ( 1206 (sharedBuffer != 0) 1207 ) || 1208 // use case 2: callback handler and frame count is default or at least as large as HAL 1209 ( 1210 (tid != -1) && 1211 ((frameCount == 0) || 1212 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1213 ) 1214 ) && 1215 // PCM data 1216 audio_is_linear_pcm(format) && 1217 // mono or stereo 1218 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1219 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1220#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1221 // hardware sample rate 1222 (sampleRate == mSampleRate) && 1223#endif 1224 // normal mixer has an associated fast mixer 1225 hasFastMixer() && 1226 // there are sufficient fast track slots available 1227 (mFastTrackAvailMask != 0) 1228 // FIXME test that MixerThread for this fast track has a capable output HAL 1229 // FIXME add a permission test also? 1230 ) { 1231 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1232 if (frameCount == 0) { 1233 frameCount = mFrameCount * kFastTrackMultiplier; 1234 } 1235 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1236 frameCount, mFrameCount); 1237 } else { 1238 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1239 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1240 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1241 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1242 audio_is_linear_pcm(format), 1243 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1244 *flags &= ~IAudioFlinger::TRACK_FAST; 1245 // For compatibility with AudioTrack calculation, buffer depth is forced 1246 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1247 // This is probably too conservative, but legacy application code may depend on it. 1248 // If you change this calculation, also review the start threshold which is related. 1249 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1250 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1251 if (minBufCount < 2) { 1252 minBufCount = 2; 1253 } 1254 size_t minFrameCount = mNormalFrameCount * minBufCount; 1255 if (frameCount < minFrameCount) { 1256 frameCount = minFrameCount; 1257 } 1258 } 1259 } 1260 1261 if (mType == DIRECT) { 1262 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1263 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1264 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1265 "for output %p with format %d", 1266 sampleRate, format, channelMask, mOutput, mFormat); 1267 lStatus = BAD_VALUE; 1268 goto Exit; 1269 } 1270 } 1271 } else if (mType == OFFLOAD) { 1272 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1273 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1274 "for output %p with format %d", 1275 sampleRate, format, channelMask, mOutput, mFormat); 1276 lStatus = BAD_VALUE; 1277 goto Exit; 1278 } 1279 } else { 1280 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1281 ALOGE("createTrack_l() Bad parameter: format %d \"" 1282 "for output %p with format %d", 1283 format, mOutput, mFormat); 1284 lStatus = BAD_VALUE; 1285 goto Exit; 1286 } 1287 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1288 if (sampleRate > mSampleRate*2) { 1289 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1290 lStatus = BAD_VALUE; 1291 goto Exit; 1292 } 1293 } 1294 1295 lStatus = initCheck(); 1296 if (lStatus != NO_ERROR) { 1297 ALOGE("Audio driver not initialized."); 1298 goto Exit; 1299 } 1300 1301 { // scope for mLock 1302 Mutex::Autolock _l(mLock); 1303 1304 // all tracks in same audio session must share the same routing strategy otherwise 1305 // conflicts will happen when tracks are moved from one output to another by audio policy 1306 // manager 1307 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1308 for (size_t i = 0; i < mTracks.size(); ++i) { 1309 sp<Track> t = mTracks[i]; 1310 if (t != 0 && !t->isOutputTrack()) { 1311 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1312 if (sessionId == t->sessionId() && strategy != actual) { 1313 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1314 strategy, actual); 1315 lStatus = BAD_VALUE; 1316 goto Exit; 1317 } 1318 } 1319 } 1320 1321 if (!isTimed) { 1322 track = new Track(this, client, streamType, sampleRate, format, 1323 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1324 } else { 1325 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1326 channelMask, frameCount, sharedBuffer, sessionId, uid); 1327 } 1328 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1329 lStatus = NO_MEMORY; 1330 goto Exit; 1331 } 1332 1333 mTracks.add(track); 1334 1335 sp<EffectChain> chain = getEffectChain_l(sessionId); 1336 if (chain != 0) { 1337 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1338 track->setMainBuffer(chain->inBuffer()); 1339 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1340 chain->incTrackCnt(); 1341 } 1342 1343 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1344 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1345 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1346 // so ask activity manager to do this on our behalf 1347 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1348 } 1349 } 1350 1351 lStatus = NO_ERROR; 1352 1353Exit: 1354 if (status) { 1355 *status = lStatus; 1356 } 1357 return track; 1358} 1359 1360uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1361{ 1362 return latency; 1363} 1364 1365uint32_t AudioFlinger::PlaybackThread::latency() const 1366{ 1367 Mutex::Autolock _l(mLock); 1368 return latency_l(); 1369} 1370uint32_t AudioFlinger::PlaybackThread::latency_l() const 1371{ 1372 if (initCheck() == NO_ERROR) { 1373 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1374 } else { 1375 return 0; 1376 } 1377} 1378 1379void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1380{ 1381 Mutex::Autolock _l(mLock); 1382 // Don't apply master volume in SW if our HAL can do it for us. 1383 if (mOutput && mOutput->audioHwDev && 1384 mOutput->audioHwDev->canSetMasterVolume()) { 1385 mMasterVolume = 1.0; 1386 } else { 1387 mMasterVolume = value; 1388 } 1389} 1390 1391void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1392{ 1393 Mutex::Autolock _l(mLock); 1394 // Don't apply master mute in SW if our HAL can do it for us. 1395 if (mOutput && mOutput->audioHwDev && 1396 mOutput->audioHwDev->canSetMasterMute()) { 1397 mMasterMute = false; 1398 } else { 1399 mMasterMute = muted; 1400 } 1401} 1402 1403void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1404{ 1405 Mutex::Autolock _l(mLock); 1406 mStreamTypes[stream].volume = value; 1407 broadcast_l(); 1408} 1409 1410void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1411{ 1412 Mutex::Autolock _l(mLock); 1413 mStreamTypes[stream].mute = muted; 1414 broadcast_l(); 1415} 1416 1417float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1418{ 1419 Mutex::Autolock _l(mLock); 1420 return mStreamTypes[stream].volume; 1421} 1422 1423// addTrack_l() must be called with ThreadBase::mLock held 1424status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1425{ 1426 status_t status = ALREADY_EXISTS; 1427 1428 // set retry count for buffer fill 1429 track->mRetryCount = kMaxTrackStartupRetries; 1430 if (mActiveTracks.indexOf(track) < 0) { 1431 // the track is newly added, make sure it fills up all its 1432 // buffers before playing. This is to ensure the client will 1433 // effectively get the latency it requested. 1434 if (!track->isOutputTrack()) { 1435 TrackBase::track_state state = track->mState; 1436 mLock.unlock(); 1437 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1438 mLock.lock(); 1439 // abort track was stopped/paused while we released the lock 1440 if (state != track->mState) { 1441 if (status == NO_ERROR) { 1442 mLock.unlock(); 1443 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1444 mLock.lock(); 1445 } 1446 return INVALID_OPERATION; 1447 } 1448 // abort if start is rejected by audio policy manager 1449 if (status != NO_ERROR) { 1450 return PERMISSION_DENIED; 1451 } 1452#ifdef ADD_BATTERY_DATA 1453 // to track the speaker usage 1454 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1455#endif 1456 } 1457 1458 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1459 track->mResetDone = false; 1460 track->mPresentationCompleteFrames = 0; 1461 mActiveTracks.add(track); 1462 mWakeLockUids.add(track->uid()); 1463 mActiveTracksGeneration++; 1464 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1465 if (chain != 0) { 1466 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1467 track->sessionId()); 1468 chain->incActiveTrackCnt(); 1469 } 1470 1471 status = NO_ERROR; 1472 } 1473 1474 ALOGV("signal playback thread"); 1475 broadcast_l(); 1476 1477 return status; 1478} 1479 1480bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1481{ 1482 track->terminate(); 1483 // active tracks are removed by threadLoop() 1484 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1485 track->mState = TrackBase::STOPPED; 1486 if (!trackActive) { 1487 removeTrack_l(track); 1488 } else if (track->isFastTrack() || track->isOffloaded()) { 1489 track->mState = TrackBase::STOPPING_1; 1490 } 1491 1492 return trackActive; 1493} 1494 1495void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1496{ 1497 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1498 mTracks.remove(track); 1499 deleteTrackName_l(track->name()); 1500 // redundant as track is about to be destroyed, for dumpsys only 1501 track->mName = -1; 1502 if (track->isFastTrack()) { 1503 int index = track->mFastIndex; 1504 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1505 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1506 mFastTrackAvailMask |= 1 << index; 1507 // redundant as track is about to be destroyed, for dumpsys only 1508 track->mFastIndex = -1; 1509 } 1510 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1511 if (chain != 0) { 1512 chain->decTrackCnt(); 1513 } 1514} 1515 1516void AudioFlinger::PlaybackThread::broadcast_l() 1517{ 1518 // Thread could be blocked waiting for async 1519 // so signal it to handle state changes immediately 1520 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1521 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1522 mSignalPending = true; 1523 mWaitWorkCV.broadcast(); 1524} 1525 1526String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1527{ 1528 Mutex::Autolock _l(mLock); 1529 if (initCheck() != NO_ERROR) { 1530 return String8(); 1531 } 1532 1533 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1534 const String8 out_s8(s); 1535 free(s); 1536 return out_s8; 1537} 1538 1539// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1540void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1541 AudioSystem::OutputDescriptor desc; 1542 void *param2 = NULL; 1543 1544 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1545 param); 1546 1547 switch (event) { 1548 case AudioSystem::OUTPUT_OPENED: 1549 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1550 desc.channelMask = mChannelMask; 1551 desc.samplingRate = mSampleRate; 1552 desc.format = mFormat; 1553 desc.frameCount = mNormalFrameCount; // FIXME see 1554 // AudioFlinger::frameCount(audio_io_handle_t) 1555 desc.latency = latency(); 1556 param2 = &desc; 1557 break; 1558 1559 case AudioSystem::STREAM_CONFIG_CHANGED: 1560 param2 = ¶m; 1561 case AudioSystem::OUTPUT_CLOSED: 1562 default: 1563 break; 1564 } 1565 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1566} 1567 1568void AudioFlinger::PlaybackThread::writeCallback() 1569{ 1570 ALOG_ASSERT(mCallbackThread != 0); 1571 mCallbackThread->resetWriteBlocked(); 1572} 1573 1574void AudioFlinger::PlaybackThread::drainCallback() 1575{ 1576 ALOG_ASSERT(mCallbackThread != 0); 1577 mCallbackThread->resetDraining(); 1578} 1579 1580void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1581{ 1582 Mutex::Autolock _l(mLock); 1583 // reject out of sequence requests 1584 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1585 mWriteAckSequence &= ~1; 1586 mWaitWorkCV.signal(); 1587 } 1588} 1589 1590void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1591{ 1592 Mutex::Autolock _l(mLock); 1593 // reject out of sequence requests 1594 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1595 mDrainSequence &= ~1; 1596 mWaitWorkCV.signal(); 1597 } 1598} 1599 1600// static 1601int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1602 void *param, 1603 void *cookie) 1604{ 1605 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1606 ALOGV("asyncCallback() event %d", event); 1607 switch (event) { 1608 case STREAM_CBK_EVENT_WRITE_READY: 1609 me->writeCallback(); 1610 break; 1611 case STREAM_CBK_EVENT_DRAIN_READY: 1612 me->drainCallback(); 1613 break; 1614 default: 1615 ALOGW("asyncCallback() unknown event %d", event); 1616 break; 1617 } 1618 return 0; 1619} 1620 1621void AudioFlinger::PlaybackThread::readOutputParameters() 1622{ 1623 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1624 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1625 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1626 if (!audio_is_output_channel(mChannelMask)) { 1627 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1628 } 1629 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1630 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1631 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1632 } 1633 mChannelCount = popcount(mChannelMask); 1634 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1635 if (!audio_is_valid_format(mFormat)) { 1636 LOG_FATAL("HAL format %d not valid for output", mFormat); 1637 } 1638 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1639 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1640 mFormat); 1641 } 1642 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1643 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1644 if (mFrameCount & 15) { 1645 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1646 mFrameCount); 1647 } 1648 1649 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1650 (mOutput->stream->set_callback != NULL)) { 1651 if (mOutput->stream->set_callback(mOutput->stream, 1652 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1653 mUseAsyncWrite = true; 1654 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1655 } 1656 } 1657 1658 // Calculate size of normal mix buffer relative to the HAL output buffer size 1659 double multiplier = 1.0; 1660 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1661 kUseFastMixer == FastMixer_Dynamic)) { 1662 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1663 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1664 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1665 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1666 maxNormalFrameCount = maxNormalFrameCount & ~15; 1667 if (maxNormalFrameCount < minNormalFrameCount) { 1668 maxNormalFrameCount = minNormalFrameCount; 1669 } 1670 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1671 if (multiplier <= 1.0) { 1672 multiplier = 1.0; 1673 } else if (multiplier <= 2.0) { 1674 if (2 * mFrameCount <= maxNormalFrameCount) { 1675 multiplier = 2.0; 1676 } else { 1677 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1678 } 1679 } else { 1680 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1681 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1682 // track, but we sometimes have to do this to satisfy the maximum frame count 1683 // constraint) 1684 // FIXME this rounding up should not be done if no HAL SRC 1685 uint32_t truncMult = (uint32_t) multiplier; 1686 if ((truncMult & 1)) { 1687 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1688 ++truncMult; 1689 } 1690 } 1691 multiplier = (double) truncMult; 1692 } 1693 } 1694 mNormalFrameCount = multiplier * mFrameCount; 1695 // round up to nearest 16 frames to satisfy AudioMixer 1696 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1697 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1698 mNormalFrameCount); 1699 1700 delete[] mAllocMixBuffer; 1701 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1702 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1703 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1704 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1705 1706 // force reconfiguration of effect chains and engines to take new buffer size and audio 1707 // parameters into account 1708 // Note that mLock is not held when readOutputParameters() is called from the constructor 1709 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1710 // matter. 1711 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1712 Vector< sp<EffectChain> > effectChains = mEffectChains; 1713 for (size_t i = 0; i < effectChains.size(); i ++) { 1714 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1715 } 1716} 1717 1718 1719status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1720{ 1721 if (halFrames == NULL || dspFrames == NULL) { 1722 return BAD_VALUE; 1723 } 1724 Mutex::Autolock _l(mLock); 1725 if (initCheck() != NO_ERROR) { 1726 return INVALID_OPERATION; 1727 } 1728 size_t framesWritten = mBytesWritten / mFrameSize; 1729 *halFrames = framesWritten; 1730 1731 if (isSuspended()) { 1732 // return an estimation of rendered frames when the output is suspended 1733 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1734 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1735 return NO_ERROR; 1736 } else { 1737 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1738 } 1739} 1740 1741uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1742{ 1743 Mutex::Autolock _l(mLock); 1744 uint32_t result = 0; 1745 if (getEffectChain_l(sessionId) != 0) { 1746 result = EFFECT_SESSION; 1747 } 1748 1749 for (size_t i = 0; i < mTracks.size(); ++i) { 1750 sp<Track> track = mTracks[i]; 1751 if (sessionId == track->sessionId() && !track->isInvalid()) { 1752 result |= TRACK_SESSION; 1753 break; 1754 } 1755 } 1756 1757 return result; 1758} 1759 1760uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1761{ 1762 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1763 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1764 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1765 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1766 } 1767 for (size_t i = 0; i < mTracks.size(); i++) { 1768 sp<Track> track = mTracks[i]; 1769 if (sessionId == track->sessionId() && !track->isInvalid()) { 1770 return AudioSystem::getStrategyForStream(track->streamType()); 1771 } 1772 } 1773 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1774} 1775 1776 1777AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1778{ 1779 Mutex::Autolock _l(mLock); 1780 return mOutput; 1781} 1782 1783AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1784{ 1785 Mutex::Autolock _l(mLock); 1786 AudioStreamOut *output = mOutput; 1787 mOutput = NULL; 1788 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1789 // must push a NULL and wait for ack 1790 mOutputSink.clear(); 1791 mPipeSink.clear(); 1792 mNormalSink.clear(); 1793 return output; 1794} 1795 1796// this method must always be called either with ThreadBase mLock held or inside the thread loop 1797audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1798{ 1799 if (mOutput == NULL) { 1800 return NULL; 1801 } 1802 return &mOutput->stream->common; 1803} 1804 1805uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1806{ 1807 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1808} 1809 1810status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1811{ 1812 if (!isValidSyncEvent(event)) { 1813 return BAD_VALUE; 1814 } 1815 1816 Mutex::Autolock _l(mLock); 1817 1818 for (size_t i = 0; i < mTracks.size(); ++i) { 1819 sp<Track> track = mTracks[i]; 1820 if (event->triggerSession() == track->sessionId()) { 1821 (void) track->setSyncEvent(event); 1822 return NO_ERROR; 1823 } 1824 } 1825 1826 return NAME_NOT_FOUND; 1827} 1828 1829bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1830{ 1831 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1832} 1833 1834void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1835 const Vector< sp<Track> >& tracksToRemove) 1836{ 1837 size_t count = tracksToRemove.size(); 1838 if (count) { 1839 for (size_t i = 0 ; i < count ; i++) { 1840 const sp<Track>& track = tracksToRemove.itemAt(i); 1841 if (!track->isOutputTrack()) { 1842 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1843#ifdef ADD_BATTERY_DATA 1844 // to track the speaker usage 1845 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1846#endif 1847 if (track->isTerminated()) { 1848 AudioSystem::releaseOutput(mId); 1849 } 1850 } 1851 } 1852 } 1853} 1854 1855void AudioFlinger::PlaybackThread::checkSilentMode_l() 1856{ 1857 if (!mMasterMute) { 1858 char value[PROPERTY_VALUE_MAX]; 1859 if (property_get("ro.audio.silent", value, "0") > 0) { 1860 char *endptr; 1861 unsigned long ul = strtoul(value, &endptr, 0); 1862 if (*endptr == '\0' && ul != 0) { 1863 ALOGD("Silence is golden"); 1864 // The setprop command will not allow a property to be changed after 1865 // the first time it is set, so we don't have to worry about un-muting. 1866 setMasterMute_l(true); 1867 } 1868 } 1869 } 1870} 1871 1872// shared by MIXER and DIRECT, overridden by DUPLICATING 1873ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1874{ 1875 // FIXME rewrite to reduce number of system calls 1876 mLastWriteTime = systemTime(); 1877 mInWrite = true; 1878 ssize_t bytesWritten; 1879 1880 // If an NBAIO sink is present, use it to write the normal mixer's submix 1881 if (mNormalSink != 0) { 1882#define mBitShift 2 // FIXME 1883 size_t count = mBytesRemaining >> mBitShift; 1884 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1885 ATRACE_BEGIN("write"); 1886 // update the setpoint when AudioFlinger::mScreenState changes 1887 uint32_t screenState = AudioFlinger::mScreenState; 1888 if (screenState != mScreenState) { 1889 mScreenState = screenState; 1890 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1891 if (pipe != NULL) { 1892 pipe->setAvgFrames((mScreenState & 1) ? 1893 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1894 } 1895 } 1896 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1897 ATRACE_END(); 1898 if (framesWritten > 0) { 1899 bytesWritten = framesWritten << mBitShift; 1900 } else { 1901 bytesWritten = framesWritten; 1902 } 1903 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1904 if (status == NO_ERROR) { 1905 size_t totalFramesWritten = mNormalSink->framesWritten(); 1906 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1907 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1908 mLatchDValid = true; 1909 } 1910 } 1911 // otherwise use the HAL / AudioStreamOut directly 1912 } else { 1913 // Direct output and offload threads 1914 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1915 if (mUseAsyncWrite) { 1916 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1917 mWriteAckSequence += 2; 1918 mWriteAckSequence |= 1; 1919 ALOG_ASSERT(mCallbackThread != 0); 1920 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1921 } 1922 // FIXME We should have an implementation of timestamps for direct output threads. 1923 // They are used e.g for multichannel PCM playback over HDMI. 1924 bytesWritten = mOutput->stream->write(mOutput->stream, 1925 mMixBuffer + offset, mBytesRemaining); 1926 if (mUseAsyncWrite && 1927 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1928 // do not wait for async callback in case of error of full write 1929 mWriteAckSequence &= ~1; 1930 ALOG_ASSERT(mCallbackThread != 0); 1931 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1932 } 1933 } 1934 1935 mNumWrites++; 1936 mInWrite = false; 1937 1938 return bytesWritten; 1939} 1940 1941void AudioFlinger::PlaybackThread::threadLoop_drain() 1942{ 1943 if (mOutput->stream->drain) { 1944 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1945 if (mUseAsyncWrite) { 1946 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1947 mDrainSequence |= 1; 1948 ALOG_ASSERT(mCallbackThread != 0); 1949 mCallbackThread->setDraining(mDrainSequence); 1950 } 1951 mOutput->stream->drain(mOutput->stream, 1952 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1953 : AUDIO_DRAIN_ALL); 1954 } 1955} 1956 1957void AudioFlinger::PlaybackThread::threadLoop_exit() 1958{ 1959 // Default implementation has nothing to do 1960} 1961 1962/* 1963The derived values that are cached: 1964 - mixBufferSize from frame count * frame size 1965 - activeSleepTime from activeSleepTimeUs() 1966 - idleSleepTime from idleSleepTimeUs() 1967 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1968 - maxPeriod from frame count and sample rate (MIXER only) 1969 1970The parameters that affect these derived values are: 1971 - frame count 1972 - frame size 1973 - sample rate 1974 - device type: A2DP or not 1975 - device latency 1976 - format: PCM or not 1977 - active sleep time 1978 - idle sleep time 1979*/ 1980 1981void AudioFlinger::PlaybackThread::cacheParameters_l() 1982{ 1983 mixBufferSize = mNormalFrameCount * mFrameSize; 1984 activeSleepTime = activeSleepTimeUs(); 1985 idleSleepTime = idleSleepTimeUs(); 1986} 1987 1988void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1989{ 1990 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1991 this, streamType, mTracks.size()); 1992 Mutex::Autolock _l(mLock); 1993 1994 size_t size = mTracks.size(); 1995 for (size_t i = 0; i < size; i++) { 1996 sp<Track> t = mTracks[i]; 1997 if (t->streamType() == streamType) { 1998 t->invalidate(); 1999 } 2000 } 2001} 2002 2003status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2004{ 2005 int session = chain->sessionId(); 2006 int16_t *buffer = mMixBuffer; 2007 bool ownsBuffer = false; 2008 2009 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2010 if (session > 0) { 2011 // Only one effect chain can be present in direct output thread and it uses 2012 // the mix buffer as input 2013 if (mType != DIRECT) { 2014 size_t numSamples = mNormalFrameCount * mChannelCount; 2015 buffer = new int16_t[numSamples]; 2016 memset(buffer, 0, numSamples * sizeof(int16_t)); 2017 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2018 ownsBuffer = true; 2019 } 2020 2021 // Attach all tracks with same session ID to this chain. 2022 for (size_t i = 0; i < mTracks.size(); ++i) { 2023 sp<Track> track = mTracks[i]; 2024 if (session == track->sessionId()) { 2025 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2026 buffer); 2027 track->setMainBuffer(buffer); 2028 chain->incTrackCnt(); 2029 } 2030 } 2031 2032 // indicate all active tracks in the chain 2033 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2034 sp<Track> track = mActiveTracks[i].promote(); 2035 if (track == 0) { 2036 continue; 2037 } 2038 if (session == track->sessionId()) { 2039 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2040 chain->incActiveTrackCnt(); 2041 } 2042 } 2043 } 2044 2045 chain->setInBuffer(buffer, ownsBuffer); 2046 chain->setOutBuffer(mMixBuffer); 2047 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2048 // chains list in order to be processed last as it contains output stage effects 2049 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2050 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2051 // after track specific effects and before output stage 2052 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2053 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2054 // Effect chain for other sessions are inserted at beginning of effect 2055 // chains list to be processed before output mix effects. Relative order between other 2056 // sessions is not important 2057 size_t size = mEffectChains.size(); 2058 size_t i = 0; 2059 for (i = 0; i < size; i++) { 2060 if (mEffectChains[i]->sessionId() < session) { 2061 break; 2062 } 2063 } 2064 mEffectChains.insertAt(chain, i); 2065 checkSuspendOnAddEffectChain_l(chain); 2066 2067 return NO_ERROR; 2068} 2069 2070size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2071{ 2072 int session = chain->sessionId(); 2073 2074 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2075 2076 for (size_t i = 0; i < mEffectChains.size(); i++) { 2077 if (chain == mEffectChains[i]) { 2078 mEffectChains.removeAt(i); 2079 // detach all active tracks from the chain 2080 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2081 sp<Track> track = mActiveTracks[i].promote(); 2082 if (track == 0) { 2083 continue; 2084 } 2085 if (session == track->sessionId()) { 2086 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2087 chain.get(), session); 2088 chain->decActiveTrackCnt(); 2089 } 2090 } 2091 2092 // detach all tracks with same session ID from this chain 2093 for (size_t i = 0; i < mTracks.size(); ++i) { 2094 sp<Track> track = mTracks[i]; 2095 if (session == track->sessionId()) { 2096 track->setMainBuffer(mMixBuffer); 2097 chain->decTrackCnt(); 2098 } 2099 } 2100 break; 2101 } 2102 } 2103 return mEffectChains.size(); 2104} 2105 2106status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2107 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2108{ 2109 Mutex::Autolock _l(mLock); 2110 return attachAuxEffect_l(track, EffectId); 2111} 2112 2113status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2114 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2115{ 2116 status_t status = NO_ERROR; 2117 2118 if (EffectId == 0) { 2119 track->setAuxBuffer(0, NULL); 2120 } else { 2121 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2122 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2123 if (effect != 0) { 2124 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2125 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2126 } else { 2127 status = INVALID_OPERATION; 2128 } 2129 } else { 2130 status = BAD_VALUE; 2131 } 2132 } 2133 return status; 2134} 2135 2136void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2137{ 2138 for (size_t i = 0; i < mTracks.size(); ++i) { 2139 sp<Track> track = mTracks[i]; 2140 if (track->auxEffectId() == effectId) { 2141 attachAuxEffect_l(track, 0); 2142 } 2143 } 2144} 2145 2146bool AudioFlinger::PlaybackThread::threadLoop() 2147{ 2148 Vector< sp<Track> > tracksToRemove; 2149 2150 standbyTime = systemTime(); 2151 2152 // MIXER 2153 nsecs_t lastWarning = 0; 2154 2155 // DUPLICATING 2156 // FIXME could this be made local to while loop? 2157 writeFrames = 0; 2158 2159 int lastGeneration = 0; 2160 2161 cacheParameters_l(); 2162 sleepTime = idleSleepTime; 2163 2164 if (mType == MIXER) { 2165 sleepTimeShift = 0; 2166 } 2167 2168 CpuStats cpuStats; 2169 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2170 2171 acquireWakeLock(); 2172 2173 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2174 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2175 // and then that string will be logged at the next convenient opportunity. 2176 const char *logString = NULL; 2177 2178 checkSilentMode_l(); 2179 2180 while (!exitPending()) 2181 { 2182 cpuStats.sample(myName); 2183 2184 Vector< sp<EffectChain> > effectChains; 2185 2186 processConfigEvents(); 2187 2188 { // scope for mLock 2189 2190 Mutex::Autolock _l(mLock); 2191 2192 if (logString != NULL) { 2193 mNBLogWriter->logTimestamp(); 2194 mNBLogWriter->log(logString); 2195 logString = NULL; 2196 } 2197 2198 if (mLatchDValid) { 2199 mLatchQ = mLatchD; 2200 mLatchDValid = false; 2201 mLatchQValid = true; 2202 } 2203 2204 if (checkForNewParameters_l()) { 2205 cacheParameters_l(); 2206 } 2207 2208 saveOutputTracks(); 2209 if (mSignalPending) { 2210 // A signal was raised while we were unlocked 2211 mSignalPending = false; 2212 } else if (waitingAsyncCallback_l()) { 2213 if (exitPending()) { 2214 break; 2215 } 2216 releaseWakeLock_l(); 2217 mWakeLockUids.clear(); 2218 mActiveTracksGeneration++; 2219 ALOGV("wait async completion"); 2220 mWaitWorkCV.wait(mLock); 2221 ALOGV("async completion/wake"); 2222 acquireWakeLock_l(); 2223 standbyTime = systemTime() + standbyDelay; 2224 sleepTime = 0; 2225 2226 continue; 2227 } 2228 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2229 isSuspended()) { 2230 // put audio hardware into standby after short delay 2231 if (shouldStandby_l()) { 2232 2233 threadLoop_standby(); 2234 2235 mStandby = true; 2236 } 2237 2238 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2239 // we're about to wait, flush the binder command buffer 2240 IPCThreadState::self()->flushCommands(); 2241 2242 clearOutputTracks(); 2243 2244 if (exitPending()) { 2245 break; 2246 } 2247 2248 releaseWakeLock_l(); 2249 mWakeLockUids.clear(); 2250 mActiveTracksGeneration++; 2251 // wait until we have something to do... 2252 ALOGV("%s going to sleep", myName.string()); 2253 mWaitWorkCV.wait(mLock); 2254 ALOGV("%s waking up", myName.string()); 2255 acquireWakeLock_l(); 2256 2257 mMixerStatus = MIXER_IDLE; 2258 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2259 mBytesWritten = 0; 2260 mBytesRemaining = 0; 2261 checkSilentMode_l(); 2262 2263 standbyTime = systemTime() + standbyDelay; 2264 sleepTime = idleSleepTime; 2265 if (mType == MIXER) { 2266 sleepTimeShift = 0; 2267 } 2268 2269 continue; 2270 } 2271 } 2272 // mMixerStatusIgnoringFastTracks is also updated internally 2273 mMixerStatus = prepareTracks_l(&tracksToRemove); 2274 2275 // compare with previously applied list 2276 if (lastGeneration != mActiveTracksGeneration) { 2277 // update wakelock 2278 updateWakeLockUids_l(mWakeLockUids); 2279 lastGeneration = mActiveTracksGeneration; 2280 } 2281 2282 // prevent any changes in effect chain list and in each effect chain 2283 // during mixing and effect process as the audio buffers could be deleted 2284 // or modified if an effect is created or deleted 2285 lockEffectChains_l(effectChains); 2286 } // mLock scope ends 2287 2288 if (mBytesRemaining == 0) { 2289 mCurrentWriteLength = 0; 2290 if (mMixerStatus == MIXER_TRACKS_READY) { 2291 // threadLoop_mix() sets mCurrentWriteLength 2292 threadLoop_mix(); 2293 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2294 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2295 // threadLoop_sleepTime sets sleepTime to 0 if data 2296 // must be written to HAL 2297 threadLoop_sleepTime(); 2298 if (sleepTime == 0) { 2299 mCurrentWriteLength = mixBufferSize; 2300 } 2301 } 2302 mBytesRemaining = mCurrentWriteLength; 2303 if (isSuspended()) { 2304 sleepTime = suspendSleepTimeUs(); 2305 // simulate write to HAL when suspended 2306 mBytesWritten += mixBufferSize; 2307 mBytesRemaining = 0; 2308 } 2309 2310 // only process effects if we're going to write 2311 if (sleepTime == 0 && mType != OFFLOAD) { 2312 for (size_t i = 0; i < effectChains.size(); i ++) { 2313 effectChains[i]->process_l(); 2314 } 2315 } 2316 } 2317 // Process effect chains for offloaded thread even if no audio 2318 // was read from audio track: process only updates effect state 2319 // and thus does have to be synchronized with audio writes but may have 2320 // to be called while waiting for async write callback 2321 if (mType == OFFLOAD) { 2322 for (size_t i = 0; i < effectChains.size(); i ++) { 2323 effectChains[i]->process_l(); 2324 } 2325 } 2326 2327 // enable changes in effect chain 2328 unlockEffectChains(effectChains); 2329 2330 if (!waitingAsyncCallback()) { 2331 // sleepTime == 0 means we must write to audio hardware 2332 if (sleepTime == 0) { 2333 if (mBytesRemaining) { 2334 ssize_t ret = threadLoop_write(); 2335 if (ret < 0) { 2336 mBytesRemaining = 0; 2337 } else { 2338 mBytesWritten += ret; 2339 mBytesRemaining -= ret; 2340 } 2341 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2342 (mMixerStatus == MIXER_DRAIN_ALL)) { 2343 threadLoop_drain(); 2344 } 2345if (mType == MIXER) { 2346 // write blocked detection 2347 nsecs_t now = systemTime(); 2348 nsecs_t delta = now - mLastWriteTime; 2349 if (!mStandby && delta > maxPeriod) { 2350 mNumDelayedWrites++; 2351 if ((now - lastWarning) > kWarningThrottleNs) { 2352 ATRACE_NAME("underrun"); 2353 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2354 ns2ms(delta), mNumDelayedWrites, this); 2355 lastWarning = now; 2356 } 2357 } 2358} 2359 2360 mStandby = false; 2361 } else { 2362 usleep(sleepTime); 2363 } 2364 } 2365 2366 // Finally let go of removed track(s), without the lock held 2367 // since we can't guarantee the destructors won't acquire that 2368 // same lock. This will also mutate and push a new fast mixer state. 2369 threadLoop_removeTracks(tracksToRemove); 2370 tracksToRemove.clear(); 2371 2372 // FIXME I don't understand the need for this here; 2373 // it was in the original code but maybe the 2374 // assignment in saveOutputTracks() makes this unnecessary? 2375 clearOutputTracks(); 2376 2377 // Effect chains will be actually deleted here if they were removed from 2378 // mEffectChains list during mixing or effects processing 2379 effectChains.clear(); 2380 2381 // FIXME Note that the above .clear() is no longer necessary since effectChains 2382 // is now local to this block, but will keep it for now (at least until merge done). 2383 } 2384 2385 threadLoop_exit(); 2386 2387 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2388 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2389 // put output stream into standby mode 2390 if (!mStandby) { 2391 mOutput->stream->common.standby(&mOutput->stream->common); 2392 } 2393 } 2394 2395 releaseWakeLock(); 2396 mWakeLockUids.clear(); 2397 mActiveTracksGeneration++; 2398 2399 ALOGV("Thread %p type %d exiting", this, mType); 2400 return false; 2401} 2402 2403// removeTracks_l() must be called with ThreadBase::mLock held 2404void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2405{ 2406 size_t count = tracksToRemove.size(); 2407 if (count) { 2408 for (size_t i=0 ; i<count ; i++) { 2409 const sp<Track>& track = tracksToRemove.itemAt(i); 2410 mActiveTracks.remove(track); 2411 mWakeLockUids.remove(track->uid()); 2412 mActiveTracksGeneration++; 2413 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2414 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2415 if (chain != 0) { 2416 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2417 track->sessionId()); 2418 chain->decActiveTrackCnt(); 2419 } 2420 if (track->isTerminated()) { 2421 removeTrack_l(track); 2422 } 2423 } 2424 } 2425 2426} 2427 2428status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2429{ 2430 if (mNormalSink != 0) { 2431 return mNormalSink->getTimestamp(timestamp); 2432 } 2433 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2434 uint64_t position64; 2435 int ret = mOutput->stream->get_presentation_position( 2436 mOutput->stream, &position64, ×tamp.mTime); 2437 if (ret == 0) { 2438 timestamp.mPosition = (uint32_t)position64; 2439 return NO_ERROR; 2440 } 2441 } 2442 return INVALID_OPERATION; 2443} 2444// ---------------------------------------------------------------------------- 2445 2446AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2447 audio_io_handle_t id, audio_devices_t device, type_t type) 2448 : PlaybackThread(audioFlinger, output, id, device, type), 2449 // mAudioMixer below 2450 // mFastMixer below 2451 mFastMixerFutex(0) 2452 // mOutputSink below 2453 // mPipeSink below 2454 // mNormalSink below 2455{ 2456 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2457 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2458 "mFrameCount=%d, mNormalFrameCount=%d", 2459 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2460 mNormalFrameCount); 2461 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2462 2463 // FIXME - Current mixer implementation only supports stereo output 2464 if (mChannelCount != FCC_2) { 2465 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2466 } 2467 2468 // create an NBAIO sink for the HAL output stream, and negotiate 2469 mOutputSink = new AudioStreamOutSink(output->stream); 2470 size_t numCounterOffers = 0; 2471 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2472 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2473 ALOG_ASSERT(index == 0); 2474 2475 // initialize fast mixer depending on configuration 2476 bool initFastMixer; 2477 switch (kUseFastMixer) { 2478 case FastMixer_Never: 2479 initFastMixer = false; 2480 break; 2481 case FastMixer_Always: 2482 initFastMixer = true; 2483 break; 2484 case FastMixer_Static: 2485 case FastMixer_Dynamic: 2486 initFastMixer = mFrameCount < mNormalFrameCount; 2487 break; 2488 } 2489 if (initFastMixer) { 2490 2491 // create a MonoPipe to connect our submix to FastMixer 2492 NBAIO_Format format = mOutputSink->format(); 2493 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2494 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2495 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2496 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2497 const NBAIO_Format offers[1] = {format}; 2498 size_t numCounterOffers = 0; 2499 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2500 ALOG_ASSERT(index == 0); 2501 monoPipe->setAvgFrames((mScreenState & 1) ? 2502 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2503 mPipeSink = monoPipe; 2504 2505#ifdef TEE_SINK 2506 if (mTeeSinkOutputEnabled) { 2507 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2508 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2509 numCounterOffers = 0; 2510 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2511 ALOG_ASSERT(index == 0); 2512 mTeeSink = teeSink; 2513 PipeReader *teeSource = new PipeReader(*teeSink); 2514 numCounterOffers = 0; 2515 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2516 ALOG_ASSERT(index == 0); 2517 mTeeSource = teeSource; 2518 } 2519#endif 2520 2521 // create fast mixer and configure it initially with just one fast track for our submix 2522 mFastMixer = new FastMixer(); 2523 FastMixerStateQueue *sq = mFastMixer->sq(); 2524#ifdef STATE_QUEUE_DUMP 2525 sq->setObserverDump(&mStateQueueObserverDump); 2526 sq->setMutatorDump(&mStateQueueMutatorDump); 2527#endif 2528 FastMixerState *state = sq->begin(); 2529 FastTrack *fastTrack = &state->mFastTracks[0]; 2530 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2531 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2532 fastTrack->mVolumeProvider = NULL; 2533 fastTrack->mGeneration++; 2534 state->mFastTracksGen++; 2535 state->mTrackMask = 1; 2536 // fast mixer will use the HAL output sink 2537 state->mOutputSink = mOutputSink.get(); 2538 state->mOutputSinkGen++; 2539 state->mFrameCount = mFrameCount; 2540 state->mCommand = FastMixerState::COLD_IDLE; 2541 // already done in constructor initialization list 2542 //mFastMixerFutex = 0; 2543 state->mColdFutexAddr = &mFastMixerFutex; 2544 state->mColdGen++; 2545 state->mDumpState = &mFastMixerDumpState; 2546#ifdef TEE_SINK 2547 state->mTeeSink = mTeeSink.get(); 2548#endif 2549 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2550 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2551 sq->end(); 2552 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2553 2554 // start the fast mixer 2555 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2556 pid_t tid = mFastMixer->getTid(); 2557 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2558 if (err != 0) { 2559 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2560 kPriorityFastMixer, getpid_cached, tid, err); 2561 } 2562 2563#ifdef AUDIO_WATCHDOG 2564 // create and start the watchdog 2565 mAudioWatchdog = new AudioWatchdog(); 2566 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2567 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2568 tid = mAudioWatchdog->getTid(); 2569 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2570 if (err != 0) { 2571 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2572 kPriorityFastMixer, getpid_cached, tid, err); 2573 } 2574#endif 2575 2576 } else { 2577 mFastMixer = NULL; 2578 } 2579 2580 switch (kUseFastMixer) { 2581 case FastMixer_Never: 2582 case FastMixer_Dynamic: 2583 mNormalSink = mOutputSink; 2584 break; 2585 case FastMixer_Always: 2586 mNormalSink = mPipeSink; 2587 break; 2588 case FastMixer_Static: 2589 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2590 break; 2591 } 2592} 2593 2594AudioFlinger::MixerThread::~MixerThread() 2595{ 2596 if (mFastMixer != NULL) { 2597 FastMixerStateQueue *sq = mFastMixer->sq(); 2598 FastMixerState *state = sq->begin(); 2599 if (state->mCommand == FastMixerState::COLD_IDLE) { 2600 int32_t old = android_atomic_inc(&mFastMixerFutex); 2601 if (old == -1) { 2602 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2603 } 2604 } 2605 state->mCommand = FastMixerState::EXIT; 2606 sq->end(); 2607 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2608 mFastMixer->join(); 2609 // Though the fast mixer thread has exited, it's state queue is still valid. 2610 // We'll use that extract the final state which contains one remaining fast track 2611 // corresponding to our sub-mix. 2612 state = sq->begin(); 2613 ALOG_ASSERT(state->mTrackMask == 1); 2614 FastTrack *fastTrack = &state->mFastTracks[0]; 2615 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2616 delete fastTrack->mBufferProvider; 2617 sq->end(false /*didModify*/); 2618 delete mFastMixer; 2619#ifdef AUDIO_WATCHDOG 2620 if (mAudioWatchdog != 0) { 2621 mAudioWatchdog->requestExit(); 2622 mAudioWatchdog->requestExitAndWait(); 2623 mAudioWatchdog.clear(); 2624 } 2625#endif 2626 } 2627 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2628 delete mAudioMixer; 2629} 2630 2631 2632uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2633{ 2634 if (mFastMixer != NULL) { 2635 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2636 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2637 } 2638 return latency; 2639} 2640 2641 2642void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2643{ 2644 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2645} 2646 2647ssize_t AudioFlinger::MixerThread::threadLoop_write() 2648{ 2649 // FIXME we should only do one push per cycle; confirm this is true 2650 // Start the fast mixer if it's not already running 2651 if (mFastMixer != NULL) { 2652 FastMixerStateQueue *sq = mFastMixer->sq(); 2653 FastMixerState *state = sq->begin(); 2654 if (state->mCommand != FastMixerState::MIX_WRITE && 2655 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2656 if (state->mCommand == FastMixerState::COLD_IDLE) { 2657 int32_t old = android_atomic_inc(&mFastMixerFutex); 2658 if (old == -1) { 2659 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2660 } 2661#ifdef AUDIO_WATCHDOG 2662 if (mAudioWatchdog != 0) { 2663 mAudioWatchdog->resume(); 2664 } 2665#endif 2666 } 2667 state->mCommand = FastMixerState::MIX_WRITE; 2668 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2669 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2670 sq->end(); 2671 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2672 if (kUseFastMixer == FastMixer_Dynamic) { 2673 mNormalSink = mPipeSink; 2674 } 2675 } else { 2676 sq->end(false /*didModify*/); 2677 } 2678 } 2679 return PlaybackThread::threadLoop_write(); 2680} 2681 2682void AudioFlinger::MixerThread::threadLoop_standby() 2683{ 2684 // Idle the fast mixer if it's currently running 2685 if (mFastMixer != NULL) { 2686 FastMixerStateQueue *sq = mFastMixer->sq(); 2687 FastMixerState *state = sq->begin(); 2688 if (!(state->mCommand & FastMixerState::IDLE)) { 2689 state->mCommand = FastMixerState::COLD_IDLE; 2690 state->mColdFutexAddr = &mFastMixerFutex; 2691 state->mColdGen++; 2692 mFastMixerFutex = 0; 2693 sq->end(); 2694 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2695 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2696 if (kUseFastMixer == FastMixer_Dynamic) { 2697 mNormalSink = mOutputSink; 2698 } 2699#ifdef AUDIO_WATCHDOG 2700 if (mAudioWatchdog != 0) { 2701 mAudioWatchdog->pause(); 2702 } 2703#endif 2704 } else { 2705 sq->end(false /*didModify*/); 2706 } 2707 } 2708 PlaybackThread::threadLoop_standby(); 2709} 2710 2711// Empty implementation for standard mixer 2712// Overridden for offloaded playback 2713void AudioFlinger::PlaybackThread::flushOutput_l() 2714{ 2715} 2716 2717bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2718{ 2719 return false; 2720} 2721 2722bool AudioFlinger::PlaybackThread::shouldStandby_l() 2723{ 2724 return !mStandby; 2725} 2726 2727bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2728{ 2729 Mutex::Autolock _l(mLock); 2730 return waitingAsyncCallback_l(); 2731} 2732 2733// shared by MIXER and DIRECT, overridden by DUPLICATING 2734void AudioFlinger::PlaybackThread::threadLoop_standby() 2735{ 2736 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2737 mOutput->stream->common.standby(&mOutput->stream->common); 2738 if (mUseAsyncWrite != 0) { 2739 // discard any pending drain or write ack by incrementing sequence 2740 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2741 mDrainSequence = (mDrainSequence + 2) & ~1; 2742 ALOG_ASSERT(mCallbackThread != 0); 2743 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2744 mCallbackThread->setDraining(mDrainSequence); 2745 } 2746} 2747 2748void AudioFlinger::MixerThread::threadLoop_mix() 2749{ 2750 // obtain the presentation timestamp of the next output buffer 2751 int64_t pts; 2752 status_t status = INVALID_OPERATION; 2753 2754 if (mNormalSink != 0) { 2755 status = mNormalSink->getNextWriteTimestamp(&pts); 2756 } else { 2757 status = mOutputSink->getNextWriteTimestamp(&pts); 2758 } 2759 2760 if (status != NO_ERROR) { 2761 pts = AudioBufferProvider::kInvalidPTS; 2762 } 2763 2764 // mix buffers... 2765 mAudioMixer->process(pts); 2766 mCurrentWriteLength = mixBufferSize; 2767 // increase sleep time progressively when application underrun condition clears. 2768 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2769 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2770 // such that we would underrun the audio HAL. 2771 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2772 sleepTimeShift--; 2773 } 2774 sleepTime = 0; 2775 standbyTime = systemTime() + standbyDelay; 2776 //TODO: delay standby when effects have a tail 2777} 2778 2779void AudioFlinger::MixerThread::threadLoop_sleepTime() 2780{ 2781 // If no tracks are ready, sleep once for the duration of an output 2782 // buffer size, then write 0s to the output 2783 if (sleepTime == 0) { 2784 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2785 sleepTime = activeSleepTime >> sleepTimeShift; 2786 if (sleepTime < kMinThreadSleepTimeUs) { 2787 sleepTime = kMinThreadSleepTimeUs; 2788 } 2789 // reduce sleep time in case of consecutive application underruns to avoid 2790 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2791 // duration we would end up writing less data than needed by the audio HAL if 2792 // the condition persists. 2793 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2794 sleepTimeShift++; 2795 } 2796 } else { 2797 sleepTime = idleSleepTime; 2798 } 2799 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2800 memset (mMixBuffer, 0, mixBufferSize); 2801 sleepTime = 0; 2802 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2803 "anticipated start"); 2804 } 2805 // TODO add standby time extension fct of effect tail 2806} 2807 2808// prepareTracks_l() must be called with ThreadBase::mLock held 2809AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2810 Vector< sp<Track> > *tracksToRemove) 2811{ 2812 2813 mixer_state mixerStatus = MIXER_IDLE; 2814 // find out which tracks need to be processed 2815 size_t count = mActiveTracks.size(); 2816 size_t mixedTracks = 0; 2817 size_t tracksWithEffect = 0; 2818 // counts only _active_ fast tracks 2819 size_t fastTracks = 0; 2820 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2821 2822 float masterVolume = mMasterVolume; 2823 bool masterMute = mMasterMute; 2824 2825 if (masterMute) { 2826 masterVolume = 0; 2827 } 2828 // Delegate master volume control to effect in output mix effect chain if needed 2829 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2830 if (chain != 0) { 2831 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2832 chain->setVolume_l(&v, &v); 2833 masterVolume = (float)((v + (1 << 23)) >> 24); 2834 chain.clear(); 2835 } 2836 2837 // prepare a new state to push 2838 FastMixerStateQueue *sq = NULL; 2839 FastMixerState *state = NULL; 2840 bool didModify = false; 2841 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2842 if (mFastMixer != NULL) { 2843 sq = mFastMixer->sq(); 2844 state = sq->begin(); 2845 } 2846 2847 for (size_t i=0 ; i<count ; i++) { 2848 const sp<Track> t = mActiveTracks[i].promote(); 2849 if (t == 0) { 2850 continue; 2851 } 2852 2853 // this const just means the local variable doesn't change 2854 Track* const track = t.get(); 2855 2856 // process fast tracks 2857 if (track->isFastTrack()) { 2858 2859 // It's theoretically possible (though unlikely) for a fast track to be created 2860 // and then removed within the same normal mix cycle. This is not a problem, as 2861 // the track never becomes active so it's fast mixer slot is never touched. 2862 // The converse, of removing an (active) track and then creating a new track 2863 // at the identical fast mixer slot within the same normal mix cycle, 2864 // is impossible because the slot isn't marked available until the end of each cycle. 2865 int j = track->mFastIndex; 2866 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2867 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2868 FastTrack *fastTrack = &state->mFastTracks[j]; 2869 2870 // Determine whether the track is currently in underrun condition, 2871 // and whether it had a recent underrun. 2872 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2873 FastTrackUnderruns underruns = ftDump->mUnderruns; 2874 uint32_t recentFull = (underruns.mBitFields.mFull - 2875 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2876 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2877 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2878 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2879 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2880 uint32_t recentUnderruns = recentPartial + recentEmpty; 2881 track->mObservedUnderruns = underruns; 2882 // don't count underruns that occur while stopping or pausing 2883 // or stopped which can occur when flush() is called while active 2884 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2885 recentUnderruns > 0) { 2886 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2887 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2888 } 2889 2890 // This is similar to the state machine for normal tracks, 2891 // with a few modifications for fast tracks. 2892 bool isActive = true; 2893 switch (track->mState) { 2894 case TrackBase::STOPPING_1: 2895 // track stays active in STOPPING_1 state until first underrun 2896 if (recentUnderruns > 0 || track->isTerminated()) { 2897 track->mState = TrackBase::STOPPING_2; 2898 } 2899 break; 2900 case TrackBase::PAUSING: 2901 // ramp down is not yet implemented 2902 track->setPaused(); 2903 break; 2904 case TrackBase::RESUMING: 2905 // ramp up is not yet implemented 2906 track->mState = TrackBase::ACTIVE; 2907 break; 2908 case TrackBase::ACTIVE: 2909 if (recentFull > 0 || recentPartial > 0) { 2910 // track has provided at least some frames recently: reset retry count 2911 track->mRetryCount = kMaxTrackRetries; 2912 } 2913 if (recentUnderruns == 0) { 2914 // no recent underruns: stay active 2915 break; 2916 } 2917 // there has recently been an underrun of some kind 2918 if (track->sharedBuffer() == 0) { 2919 // were any of the recent underruns "empty" (no frames available)? 2920 if (recentEmpty == 0) { 2921 // no, then ignore the partial underruns as they are allowed indefinitely 2922 break; 2923 } 2924 // there has recently been an "empty" underrun: decrement the retry counter 2925 if (--(track->mRetryCount) > 0) { 2926 break; 2927 } 2928 // indicate to client process that the track was disabled because of underrun; 2929 // it will then automatically call start() when data is available 2930 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2931 // remove from active list, but state remains ACTIVE [confusing but true] 2932 isActive = false; 2933 break; 2934 } 2935 // fall through 2936 case TrackBase::STOPPING_2: 2937 case TrackBase::PAUSED: 2938 case TrackBase::STOPPED: 2939 case TrackBase::FLUSHED: // flush() while active 2940 // Check for presentation complete if track is inactive 2941 // We have consumed all the buffers of this track. 2942 // This would be incomplete if we auto-paused on underrun 2943 { 2944 size_t audioHALFrames = 2945 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2946 size_t framesWritten = mBytesWritten / mFrameSize; 2947 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2948 // track stays in active list until presentation is complete 2949 break; 2950 } 2951 } 2952 if (track->isStopping_2()) { 2953 track->mState = TrackBase::STOPPED; 2954 } 2955 if (track->isStopped()) { 2956 // Can't reset directly, as fast mixer is still polling this track 2957 // track->reset(); 2958 // So instead mark this track as needing to be reset after push with ack 2959 resetMask |= 1 << i; 2960 } 2961 isActive = false; 2962 break; 2963 case TrackBase::IDLE: 2964 default: 2965 LOG_FATAL("unexpected track state %d", track->mState); 2966 } 2967 2968 if (isActive) { 2969 // was it previously inactive? 2970 if (!(state->mTrackMask & (1 << j))) { 2971 ExtendedAudioBufferProvider *eabp = track; 2972 VolumeProvider *vp = track; 2973 fastTrack->mBufferProvider = eabp; 2974 fastTrack->mVolumeProvider = vp; 2975 fastTrack->mSampleRate = track->mSampleRate; 2976 fastTrack->mChannelMask = track->mChannelMask; 2977 fastTrack->mGeneration++; 2978 state->mTrackMask |= 1 << j; 2979 didModify = true; 2980 // no acknowledgement required for newly active tracks 2981 } 2982 // cache the combined master volume and stream type volume for fast mixer; this 2983 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2984 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2985 ++fastTracks; 2986 } else { 2987 // was it previously active? 2988 if (state->mTrackMask & (1 << j)) { 2989 fastTrack->mBufferProvider = NULL; 2990 fastTrack->mGeneration++; 2991 state->mTrackMask &= ~(1 << j); 2992 didModify = true; 2993 // If any fast tracks were removed, we must wait for acknowledgement 2994 // because we're about to decrement the last sp<> on those tracks. 2995 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2996 } else { 2997 LOG_FATAL("fast track %d should have been active", j); 2998 } 2999 tracksToRemove->add(track); 3000 // Avoids a misleading display in dumpsys 3001 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3002 } 3003 continue; 3004 } 3005 3006 { // local variable scope to avoid goto warning 3007 3008 audio_track_cblk_t* cblk = track->cblk(); 3009 3010 // The first time a track is added we wait 3011 // for all its buffers to be filled before processing it 3012 int name = track->name(); 3013 // make sure that we have enough frames to mix one full buffer. 3014 // enforce this condition only once to enable draining the buffer in case the client 3015 // app does not call stop() and relies on underrun to stop: 3016 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3017 // during last round 3018 size_t desiredFrames; 3019 uint32_t sr = track->sampleRate(); 3020 if (sr == mSampleRate) { 3021 desiredFrames = mNormalFrameCount; 3022 } else { 3023 // +1 for rounding and +1 for additional sample needed for interpolation 3024 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3025 // add frames already consumed but not yet released by the resampler 3026 // because cblk->framesReady() will include these frames 3027 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3028 // the minimum track buffer size is normally twice the number of frames necessary 3029 // to fill one buffer and the resampler should not leave more than one buffer worth 3030 // of unreleased frames after each pass, but just in case... 3031 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3032 } 3033 uint32_t minFrames = 1; 3034 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3035 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3036 minFrames = desiredFrames; 3037 } 3038 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 3039 size_t framesReady; 3040 if (track->sharedBuffer() == 0) { 3041 framesReady = track->framesReady(); 3042 } else if (track->isStopped()) { 3043 framesReady = 0; 3044 } else { 3045 framesReady = 1; 3046 } 3047 if ((framesReady >= minFrames) && track->isReady() && 3048 !track->isPaused() && !track->isTerminated()) 3049 { 3050 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3051 3052 mixedTracks++; 3053 3054 // track->mainBuffer() != mMixBuffer means there is an effect chain 3055 // connected to the track 3056 chain.clear(); 3057 if (track->mainBuffer() != mMixBuffer) { 3058 chain = getEffectChain_l(track->sessionId()); 3059 // Delegate volume control to effect in track effect chain if needed 3060 if (chain != 0) { 3061 tracksWithEffect++; 3062 } else { 3063 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3064 "session %d", 3065 name, track->sessionId()); 3066 } 3067 } 3068 3069 3070 int param = AudioMixer::VOLUME; 3071 if (track->mFillingUpStatus == Track::FS_FILLED) { 3072 // no ramp for the first volume setting 3073 track->mFillingUpStatus = Track::FS_ACTIVE; 3074 if (track->mState == TrackBase::RESUMING) { 3075 track->mState = TrackBase::ACTIVE; 3076 param = AudioMixer::RAMP_VOLUME; 3077 } 3078 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3079 // FIXME should not make a decision based on mServer 3080 } else if (cblk->mServer != 0) { 3081 // If the track is stopped before the first frame was mixed, 3082 // do not apply ramp 3083 param = AudioMixer::RAMP_VOLUME; 3084 } 3085 3086 // compute volume for this track 3087 uint32_t vl, vr, va; 3088 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3089 vl = vr = va = 0; 3090 if (track->isPausing()) { 3091 track->setPaused(); 3092 } 3093 } else { 3094 3095 // read original volumes with volume control 3096 float typeVolume = mStreamTypes[track->streamType()].volume; 3097 float v = masterVolume * typeVolume; 3098 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3099 uint32_t vlr = proxy->getVolumeLR(); 3100 vl = vlr & 0xFFFF; 3101 vr = vlr >> 16; 3102 // track volumes come from shared memory, so can't be trusted and must be clamped 3103 if (vl > MAX_GAIN_INT) { 3104 ALOGV("Track left volume out of range: %04X", vl); 3105 vl = MAX_GAIN_INT; 3106 } 3107 if (vr > MAX_GAIN_INT) { 3108 ALOGV("Track right volume out of range: %04X", vr); 3109 vr = MAX_GAIN_INT; 3110 } 3111 // now apply the master volume and stream type volume 3112 vl = (uint32_t)(v * vl) << 12; 3113 vr = (uint32_t)(v * vr) << 12; 3114 // assuming master volume and stream type volume each go up to 1.0, 3115 // vl and vr are now in 8.24 format 3116 3117 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3118 // send level comes from shared memory and so may be corrupt 3119 if (sendLevel > MAX_GAIN_INT) { 3120 ALOGV("Track send level out of range: %04X", sendLevel); 3121 sendLevel = MAX_GAIN_INT; 3122 } 3123 va = (uint32_t)(v * sendLevel); 3124 } 3125 3126 // Delegate volume control to effect in track effect chain if needed 3127 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3128 // Do not ramp volume if volume is controlled by effect 3129 param = AudioMixer::VOLUME; 3130 track->mHasVolumeController = true; 3131 } else { 3132 // force no volume ramp when volume controller was just disabled or removed 3133 // from effect chain to avoid volume spike 3134 if (track->mHasVolumeController) { 3135 param = AudioMixer::VOLUME; 3136 } 3137 track->mHasVolumeController = false; 3138 } 3139 3140 // Convert volumes from 8.24 to 4.12 format 3141 // This additional clamping is needed in case chain->setVolume_l() overshot 3142 vl = (vl + (1 << 11)) >> 12; 3143 if (vl > MAX_GAIN_INT) { 3144 vl = MAX_GAIN_INT; 3145 } 3146 vr = (vr + (1 << 11)) >> 12; 3147 if (vr > MAX_GAIN_INT) { 3148 vr = MAX_GAIN_INT; 3149 } 3150 3151 if (va > MAX_GAIN_INT) { 3152 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3153 } 3154 3155 // XXX: these things DON'T need to be done each time 3156 mAudioMixer->setBufferProvider(name, track); 3157 mAudioMixer->enable(name); 3158 3159 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3160 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3161 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3162 mAudioMixer->setParameter( 3163 name, 3164 AudioMixer::TRACK, 3165 AudioMixer::FORMAT, (void *)track->format()); 3166 mAudioMixer->setParameter( 3167 name, 3168 AudioMixer::TRACK, 3169 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3170 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3171 uint32_t maxSampleRate = mSampleRate * 2; 3172 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3173 if (reqSampleRate == 0) { 3174 reqSampleRate = mSampleRate; 3175 } else if (reqSampleRate > maxSampleRate) { 3176 reqSampleRate = maxSampleRate; 3177 } 3178 mAudioMixer->setParameter( 3179 name, 3180 AudioMixer::RESAMPLE, 3181 AudioMixer::SAMPLE_RATE, 3182 (void *)reqSampleRate); 3183 mAudioMixer->setParameter( 3184 name, 3185 AudioMixer::TRACK, 3186 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3187 mAudioMixer->setParameter( 3188 name, 3189 AudioMixer::TRACK, 3190 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3191 3192 // reset retry count 3193 track->mRetryCount = kMaxTrackRetries; 3194 3195 // If one track is ready, set the mixer ready if: 3196 // - the mixer was not ready during previous round OR 3197 // - no other track is not ready 3198 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3199 mixerStatus != MIXER_TRACKS_ENABLED) { 3200 mixerStatus = MIXER_TRACKS_READY; 3201 } 3202 } else { 3203 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3204 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3205 } 3206 // clear effect chain input buffer if an active track underruns to avoid sending 3207 // previous audio buffer again to effects 3208 chain = getEffectChain_l(track->sessionId()); 3209 if (chain != 0) { 3210 chain->clearInputBuffer(); 3211 } 3212 3213 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3214 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3215 track->isStopped() || track->isPaused()) { 3216 // We have consumed all the buffers of this track. 3217 // Remove it from the list of active tracks. 3218 // TODO: use actual buffer filling status instead of latency when available from 3219 // audio HAL 3220 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3221 size_t framesWritten = mBytesWritten / mFrameSize; 3222 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3223 if (track->isStopped()) { 3224 track->reset(); 3225 } 3226 tracksToRemove->add(track); 3227 } 3228 } else { 3229 // No buffers for this track. Give it a few chances to 3230 // fill a buffer, then remove it from active list. 3231 if (--(track->mRetryCount) <= 0) { 3232 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3233 tracksToRemove->add(track); 3234 // indicate to client process that the track was disabled because of underrun; 3235 // it will then automatically call start() when data is available 3236 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3237 // If one track is not ready, mark the mixer also not ready if: 3238 // - the mixer was ready during previous round OR 3239 // - no other track is ready 3240 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3241 mixerStatus != MIXER_TRACKS_READY) { 3242 mixerStatus = MIXER_TRACKS_ENABLED; 3243 } 3244 } 3245 mAudioMixer->disable(name); 3246 } 3247 3248 } // local variable scope to avoid goto warning 3249track_is_ready: ; 3250 3251 } 3252 3253 // Push the new FastMixer state if necessary 3254 bool pauseAudioWatchdog = false; 3255 if (didModify) { 3256 state->mFastTracksGen++; 3257 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3258 if (kUseFastMixer == FastMixer_Dynamic && 3259 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3260 state->mCommand = FastMixerState::COLD_IDLE; 3261 state->mColdFutexAddr = &mFastMixerFutex; 3262 state->mColdGen++; 3263 mFastMixerFutex = 0; 3264 if (kUseFastMixer == FastMixer_Dynamic) { 3265 mNormalSink = mOutputSink; 3266 } 3267 // If we go into cold idle, need to wait for acknowledgement 3268 // so that fast mixer stops doing I/O. 3269 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3270 pauseAudioWatchdog = true; 3271 } 3272 } 3273 if (sq != NULL) { 3274 sq->end(didModify); 3275 sq->push(block); 3276 } 3277#ifdef AUDIO_WATCHDOG 3278 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3279 mAudioWatchdog->pause(); 3280 } 3281#endif 3282 3283 // Now perform the deferred reset on fast tracks that have stopped 3284 while (resetMask != 0) { 3285 size_t i = __builtin_ctz(resetMask); 3286 ALOG_ASSERT(i < count); 3287 resetMask &= ~(1 << i); 3288 sp<Track> t = mActiveTracks[i].promote(); 3289 if (t == 0) { 3290 continue; 3291 } 3292 Track* track = t.get(); 3293 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3294 track->reset(); 3295 } 3296 3297 // remove all the tracks that need to be... 3298 removeTracks_l(*tracksToRemove); 3299 3300 // mix buffer must be cleared if all tracks are connected to an 3301 // effect chain as in this case the mixer will not write to 3302 // mix buffer and track effects will accumulate into it 3303 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3304 (mixedTracks == 0 && fastTracks > 0))) { 3305 // FIXME as a performance optimization, should remember previous zero status 3306 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3307 } 3308 3309 // if any fast tracks, then status is ready 3310 mMixerStatusIgnoringFastTracks = mixerStatus; 3311 if (fastTracks > 0) { 3312 mixerStatus = MIXER_TRACKS_READY; 3313 } 3314 return mixerStatus; 3315} 3316 3317// getTrackName_l() must be called with ThreadBase::mLock held 3318int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3319{ 3320 return mAudioMixer->getTrackName(channelMask, sessionId); 3321} 3322 3323// deleteTrackName_l() must be called with ThreadBase::mLock held 3324void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3325{ 3326 ALOGV("remove track (%d) and delete from mixer", name); 3327 mAudioMixer->deleteTrackName(name); 3328} 3329 3330// checkForNewParameters_l() must be called with ThreadBase::mLock held 3331bool AudioFlinger::MixerThread::checkForNewParameters_l() 3332{ 3333 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3334 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3335 bool reconfig = false; 3336 3337 while (!mNewParameters.isEmpty()) { 3338 3339 if (mFastMixer != NULL) { 3340 FastMixerStateQueue *sq = mFastMixer->sq(); 3341 FastMixerState *state = sq->begin(); 3342 if (!(state->mCommand & FastMixerState::IDLE)) { 3343 previousCommand = state->mCommand; 3344 state->mCommand = FastMixerState::HOT_IDLE; 3345 sq->end(); 3346 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3347 } else { 3348 sq->end(false /*didModify*/); 3349 } 3350 } 3351 3352 status_t status = NO_ERROR; 3353 String8 keyValuePair = mNewParameters[0]; 3354 AudioParameter param = AudioParameter(keyValuePair); 3355 int value; 3356 3357 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3358 reconfig = true; 3359 } 3360 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3361 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3362 status = BAD_VALUE; 3363 } else { 3364 reconfig = true; 3365 } 3366 } 3367 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3368 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3369 status = BAD_VALUE; 3370 } else { 3371 reconfig = true; 3372 } 3373 } 3374 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3375 // do not accept frame count changes if tracks are open as the track buffer 3376 // size depends on frame count and correct behavior would not be guaranteed 3377 // if frame count is changed after track creation 3378 if (!mTracks.isEmpty()) { 3379 status = INVALID_OPERATION; 3380 } else { 3381 reconfig = true; 3382 } 3383 } 3384 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3385#ifdef ADD_BATTERY_DATA 3386 // when changing the audio output device, call addBatteryData to notify 3387 // the change 3388 if (mOutDevice != value) { 3389 uint32_t params = 0; 3390 // check whether speaker is on 3391 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3392 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3393 } 3394 3395 audio_devices_t deviceWithoutSpeaker 3396 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3397 // check if any other device (except speaker) is on 3398 if (value & deviceWithoutSpeaker ) { 3399 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3400 } 3401 3402 if (params != 0) { 3403 addBatteryData(params); 3404 } 3405 } 3406#endif 3407 3408 // forward device change to effects that have requested to be 3409 // aware of attached audio device. 3410 if (value != AUDIO_DEVICE_NONE) { 3411 mOutDevice = value; 3412 for (size_t i = 0; i < mEffectChains.size(); i++) { 3413 mEffectChains[i]->setDevice_l(mOutDevice); 3414 } 3415 } 3416 } 3417 3418 if (status == NO_ERROR) { 3419 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3420 keyValuePair.string()); 3421 if (!mStandby && status == INVALID_OPERATION) { 3422 mOutput->stream->common.standby(&mOutput->stream->common); 3423 mStandby = true; 3424 mBytesWritten = 0; 3425 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3426 keyValuePair.string()); 3427 } 3428 if (status == NO_ERROR && reconfig) { 3429 readOutputParameters(); 3430 delete mAudioMixer; 3431 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3432 for (size_t i = 0; i < mTracks.size() ; i++) { 3433 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3434 if (name < 0) { 3435 break; 3436 } 3437 mTracks[i]->mName = name; 3438 } 3439 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3440 } 3441 } 3442 3443 mNewParameters.removeAt(0); 3444 3445 mParamStatus = status; 3446 mParamCond.signal(); 3447 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3448 // already timed out waiting for the status and will never signal the condition. 3449 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3450 } 3451 3452 if (!(previousCommand & FastMixerState::IDLE)) { 3453 ALOG_ASSERT(mFastMixer != NULL); 3454 FastMixerStateQueue *sq = mFastMixer->sq(); 3455 FastMixerState *state = sq->begin(); 3456 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3457 state->mCommand = previousCommand; 3458 sq->end(); 3459 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3460 } 3461 3462 return reconfig; 3463} 3464 3465 3466void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3467{ 3468 const size_t SIZE = 256; 3469 char buffer[SIZE]; 3470 String8 result; 3471 3472 PlaybackThread::dumpInternals(fd, args); 3473 3474 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3475 result.append(buffer); 3476 write(fd, result.string(), result.size()); 3477 3478 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3479 const FastMixerDumpState copy(mFastMixerDumpState); 3480 copy.dump(fd); 3481 3482#ifdef STATE_QUEUE_DUMP 3483 // Similar for state queue 3484 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3485 observerCopy.dump(fd); 3486 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3487 mutatorCopy.dump(fd); 3488#endif 3489 3490#ifdef TEE_SINK 3491 // Write the tee output to a .wav file 3492 dumpTee(fd, mTeeSource, mId); 3493#endif 3494 3495#ifdef AUDIO_WATCHDOG 3496 if (mAudioWatchdog != 0) { 3497 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3498 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3499 wdCopy.dump(fd); 3500 } 3501#endif 3502} 3503 3504uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3505{ 3506 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3507} 3508 3509uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3510{ 3511 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3512} 3513 3514void AudioFlinger::MixerThread::cacheParameters_l() 3515{ 3516 PlaybackThread::cacheParameters_l(); 3517 3518 // FIXME: Relaxed timing because of a certain device that can't meet latency 3519 // Should be reduced to 2x after the vendor fixes the driver issue 3520 // increase threshold again due to low power audio mode. The way this warning 3521 // threshold is calculated and its usefulness should be reconsidered anyway. 3522 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3523} 3524 3525// ---------------------------------------------------------------------------- 3526 3527AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3528 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3529 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3530 // mLeftVolFloat, mRightVolFloat 3531{ 3532} 3533 3534AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3535 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3536 ThreadBase::type_t type) 3537 : PlaybackThread(audioFlinger, output, id, device, type) 3538 // mLeftVolFloat, mRightVolFloat 3539{ 3540} 3541 3542AudioFlinger::DirectOutputThread::~DirectOutputThread() 3543{ 3544} 3545 3546void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3547{ 3548 audio_track_cblk_t* cblk = track->cblk(); 3549 float left, right; 3550 3551 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3552 left = right = 0; 3553 } else { 3554 float typeVolume = mStreamTypes[track->streamType()].volume; 3555 float v = mMasterVolume * typeVolume; 3556 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3557 uint32_t vlr = proxy->getVolumeLR(); 3558 float v_clamped = v * (vlr & 0xFFFF); 3559 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3560 left = v_clamped/MAX_GAIN; 3561 v_clamped = v * (vlr >> 16); 3562 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3563 right = v_clamped/MAX_GAIN; 3564 } 3565 3566 if (lastTrack) { 3567 if (left != mLeftVolFloat || right != mRightVolFloat) { 3568 mLeftVolFloat = left; 3569 mRightVolFloat = right; 3570 3571 // Convert volumes from float to 8.24 3572 uint32_t vl = (uint32_t)(left * (1 << 24)); 3573 uint32_t vr = (uint32_t)(right * (1 << 24)); 3574 3575 // Delegate volume control to effect in track effect chain if needed 3576 // only one effect chain can be present on DirectOutputThread, so if 3577 // there is one, the track is connected to it 3578 if (!mEffectChains.isEmpty()) { 3579 mEffectChains[0]->setVolume_l(&vl, &vr); 3580 left = (float)vl / (1 << 24); 3581 right = (float)vr / (1 << 24); 3582 } 3583 if (mOutput->stream->set_volume) { 3584 mOutput->stream->set_volume(mOutput->stream, left, right); 3585 } 3586 } 3587 } 3588} 3589 3590 3591AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3592 Vector< sp<Track> > *tracksToRemove 3593) 3594{ 3595 size_t count = mActiveTracks.size(); 3596 mixer_state mixerStatus = MIXER_IDLE; 3597 3598 // find out which tracks need to be processed 3599 for (size_t i = 0; i < count; i++) { 3600 sp<Track> t = mActiveTracks[i].promote(); 3601 // The track died recently 3602 if (t == 0) { 3603 continue; 3604 } 3605 3606 Track* const track = t.get(); 3607 audio_track_cblk_t* cblk = track->cblk(); 3608 3609 // The first time a track is added we wait 3610 // for all its buffers to be filled before processing it 3611 uint32_t minFrames; 3612 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3613 minFrames = mNormalFrameCount; 3614 } else { 3615 minFrames = 1; 3616 } 3617 // Only consider last track started for volume and mixer state control. 3618 // This is the last entry in mActiveTracks unless a track underruns. 3619 // As we only care about the transition phase between two tracks on a 3620 // direct output, it is not a problem to ignore the underrun case. 3621 bool last = (i == (count - 1)); 3622 3623 if ((track->framesReady() >= minFrames) && track->isReady() && 3624 !track->isPaused() && !track->isTerminated()) 3625 { 3626 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3627 3628 if (track->mFillingUpStatus == Track::FS_FILLED) { 3629 track->mFillingUpStatus = Track::FS_ACTIVE; 3630 // make sure processVolume_l() will apply new volume even if 0 3631 mLeftVolFloat = mRightVolFloat = -1.0; 3632 if (track->mState == TrackBase::RESUMING) { 3633 track->mState = TrackBase::ACTIVE; 3634 } 3635 } 3636 3637 // compute volume for this track 3638 processVolume_l(track, last); 3639 if (last) { 3640 // reset retry count 3641 track->mRetryCount = kMaxTrackRetriesDirect; 3642 mActiveTrack = t; 3643 mixerStatus = MIXER_TRACKS_READY; 3644 } 3645 } else { 3646 // clear effect chain input buffer if the last active track started underruns 3647 // to avoid sending previous audio buffer again to effects 3648 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3649 mEffectChains[0]->clearInputBuffer(); 3650 } 3651 3652 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3653 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3654 track->isStopped() || track->isPaused()) { 3655 // We have consumed all the buffers of this track. 3656 // Remove it from the list of active tracks. 3657 // TODO: implement behavior for compressed audio 3658 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3659 size_t framesWritten = mBytesWritten / mFrameSize; 3660 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3661 if (track->isStopped()) { 3662 track->reset(); 3663 } 3664 tracksToRemove->add(track); 3665 } 3666 } else { 3667 // No buffers for this track. Give it a few chances to 3668 // fill a buffer, then remove it from active list. 3669 // Only consider last track started for mixer state control 3670 if (--(track->mRetryCount) <= 0) { 3671 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3672 tracksToRemove->add(track); 3673 } else if (last) { 3674 mixerStatus = MIXER_TRACKS_ENABLED; 3675 } 3676 } 3677 } 3678 } 3679 3680 // remove all the tracks that need to be... 3681 removeTracks_l(*tracksToRemove); 3682 3683 return mixerStatus; 3684} 3685 3686void AudioFlinger::DirectOutputThread::threadLoop_mix() 3687{ 3688 size_t frameCount = mFrameCount; 3689 int8_t *curBuf = (int8_t *)mMixBuffer; 3690 // output audio to hardware 3691 while (frameCount) { 3692 AudioBufferProvider::Buffer buffer; 3693 buffer.frameCount = frameCount; 3694 mActiveTrack->getNextBuffer(&buffer); 3695 if (buffer.raw == NULL) { 3696 memset(curBuf, 0, frameCount * mFrameSize); 3697 break; 3698 } 3699 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3700 frameCount -= buffer.frameCount; 3701 curBuf += buffer.frameCount * mFrameSize; 3702 mActiveTrack->releaseBuffer(&buffer); 3703 } 3704 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3705 sleepTime = 0; 3706 standbyTime = systemTime() + standbyDelay; 3707 mActiveTrack.clear(); 3708} 3709 3710void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3711{ 3712 if (sleepTime == 0) { 3713 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3714 sleepTime = activeSleepTime; 3715 } else { 3716 sleepTime = idleSleepTime; 3717 } 3718 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3719 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3720 sleepTime = 0; 3721 } 3722} 3723 3724// getTrackName_l() must be called with ThreadBase::mLock held 3725int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3726 int sessionId) 3727{ 3728 return 0; 3729} 3730 3731// deleteTrackName_l() must be called with ThreadBase::mLock held 3732void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3733{ 3734} 3735 3736// checkForNewParameters_l() must be called with ThreadBase::mLock held 3737bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3738{ 3739 bool reconfig = false; 3740 3741 while (!mNewParameters.isEmpty()) { 3742 status_t status = NO_ERROR; 3743 String8 keyValuePair = mNewParameters[0]; 3744 AudioParameter param = AudioParameter(keyValuePair); 3745 int value; 3746 3747 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3748 // do not accept frame count changes if tracks are open as the track buffer 3749 // size depends on frame count and correct behavior would not be garantied 3750 // if frame count is changed after track creation 3751 if (!mTracks.isEmpty()) { 3752 status = INVALID_OPERATION; 3753 } else { 3754 reconfig = true; 3755 } 3756 } 3757 if (status == NO_ERROR) { 3758 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3759 keyValuePair.string()); 3760 if (!mStandby && status == INVALID_OPERATION) { 3761 mOutput->stream->common.standby(&mOutput->stream->common); 3762 mStandby = true; 3763 mBytesWritten = 0; 3764 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3765 keyValuePair.string()); 3766 } 3767 if (status == NO_ERROR && reconfig) { 3768 readOutputParameters(); 3769 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3770 } 3771 } 3772 3773 mNewParameters.removeAt(0); 3774 3775 mParamStatus = status; 3776 mParamCond.signal(); 3777 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3778 // already timed out waiting for the status and will never signal the condition. 3779 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3780 } 3781 return reconfig; 3782} 3783 3784uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3785{ 3786 uint32_t time; 3787 if (audio_is_linear_pcm(mFormat)) { 3788 time = PlaybackThread::activeSleepTimeUs(); 3789 } else { 3790 time = 10000; 3791 } 3792 return time; 3793} 3794 3795uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3796{ 3797 uint32_t time; 3798 if (audio_is_linear_pcm(mFormat)) { 3799 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3800 } else { 3801 time = 10000; 3802 } 3803 return time; 3804} 3805 3806uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3807{ 3808 uint32_t time; 3809 if (audio_is_linear_pcm(mFormat)) { 3810 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3811 } else { 3812 time = 10000; 3813 } 3814 return time; 3815} 3816 3817void AudioFlinger::DirectOutputThread::cacheParameters_l() 3818{ 3819 PlaybackThread::cacheParameters_l(); 3820 3821 // use shorter standby delay as on normal output to release 3822 // hardware resources as soon as possible 3823 if (audio_is_linear_pcm(mFormat)) { 3824 standbyDelay = microseconds(activeSleepTime*2); 3825 } else { 3826 standbyDelay = kOffloadStandbyDelayNs; 3827 } 3828} 3829 3830// ---------------------------------------------------------------------------- 3831 3832AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3833 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3834 : Thread(false /*canCallJava*/), 3835 mPlaybackThread(playbackThread), 3836 mWriteAckSequence(0), 3837 mDrainSequence(0) 3838{ 3839} 3840 3841AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3842{ 3843} 3844 3845void AudioFlinger::AsyncCallbackThread::onFirstRef() 3846{ 3847 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3848} 3849 3850bool AudioFlinger::AsyncCallbackThread::threadLoop() 3851{ 3852 while (!exitPending()) { 3853 uint32_t writeAckSequence; 3854 uint32_t drainSequence; 3855 3856 { 3857 Mutex::Autolock _l(mLock); 3858 mWaitWorkCV.wait(mLock); 3859 if (exitPending()) { 3860 break; 3861 } 3862 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3863 mWriteAckSequence, mDrainSequence); 3864 writeAckSequence = mWriteAckSequence; 3865 mWriteAckSequence &= ~1; 3866 drainSequence = mDrainSequence; 3867 mDrainSequence &= ~1; 3868 } 3869 { 3870 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3871 if (playbackThread != 0) { 3872 if (writeAckSequence & 1) { 3873 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3874 } 3875 if (drainSequence & 1) { 3876 playbackThread->resetDraining(drainSequence >> 1); 3877 } 3878 } 3879 } 3880 } 3881 return false; 3882} 3883 3884void AudioFlinger::AsyncCallbackThread::exit() 3885{ 3886 ALOGV("AsyncCallbackThread::exit"); 3887 Mutex::Autolock _l(mLock); 3888 requestExit(); 3889 mWaitWorkCV.broadcast(); 3890} 3891 3892void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3893{ 3894 Mutex::Autolock _l(mLock); 3895 // bit 0 is cleared 3896 mWriteAckSequence = sequence << 1; 3897} 3898 3899void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3900{ 3901 Mutex::Autolock _l(mLock); 3902 // ignore unexpected callbacks 3903 if (mWriteAckSequence & 2) { 3904 mWriteAckSequence |= 1; 3905 mWaitWorkCV.signal(); 3906 } 3907} 3908 3909void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3910{ 3911 Mutex::Autolock _l(mLock); 3912 // bit 0 is cleared 3913 mDrainSequence = sequence << 1; 3914} 3915 3916void AudioFlinger::AsyncCallbackThread::resetDraining() 3917{ 3918 Mutex::Autolock _l(mLock); 3919 // ignore unexpected callbacks 3920 if (mDrainSequence & 2) { 3921 mDrainSequence |= 1; 3922 mWaitWorkCV.signal(); 3923 } 3924} 3925 3926 3927// ---------------------------------------------------------------------------- 3928AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3929 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3930 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3931 mHwPaused(false), 3932 mFlushPending(false), 3933 mPausedBytesRemaining(0), 3934 mPreviousTrack(NULL) 3935{ 3936} 3937 3938void AudioFlinger::OffloadThread::threadLoop_exit() 3939{ 3940 if (mFlushPending || mHwPaused) { 3941 // If a flush is pending or track was paused, just discard buffered data 3942 flushHw_l(); 3943 } else { 3944 mMixerStatus = MIXER_DRAIN_ALL; 3945 threadLoop_drain(); 3946 } 3947 mCallbackThread->exit(); 3948 PlaybackThread::threadLoop_exit(); 3949} 3950 3951AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3952 Vector< sp<Track> > *tracksToRemove 3953) 3954{ 3955 size_t count = mActiveTracks.size(); 3956 3957 mixer_state mixerStatus = MIXER_IDLE; 3958 bool doHwPause = false; 3959 bool doHwResume = false; 3960 3961 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3962 3963 // find out which tracks need to be processed 3964 for (size_t i = 0; i < count; i++) { 3965 sp<Track> t = mActiveTracks[i].promote(); 3966 // The track died recently 3967 if (t == 0) { 3968 continue; 3969 } 3970 Track* const track = t.get(); 3971 audio_track_cblk_t* cblk = track->cblk(); 3972 if (mPreviousTrack != NULL) { 3973 if (t.get() != mPreviousTrack) { 3974 // Flush any data still being written from last track 3975 mBytesRemaining = 0; 3976 if (mPausedBytesRemaining) { 3977 // Last track was paused so we also need to flush saved 3978 // mixbuffer state and invalidate track so that it will 3979 // re-submit that unwritten data when it is next resumed 3980 mPausedBytesRemaining = 0; 3981 // Invalidate is a bit drastic - would be more efficient 3982 // to have a flag to tell client that some of the 3983 // previously written data was lost 3984 mPreviousTrack->invalidate(); 3985 } 3986 } 3987 } 3988 mPreviousTrack = t.get(); 3989 bool last = (i == (count - 1)); 3990 if (track->isPausing()) { 3991 track->setPaused(); 3992 if (last) { 3993 if (!mHwPaused) { 3994 doHwPause = true; 3995 mHwPaused = true; 3996 } 3997 // If we were part way through writing the mixbuffer to 3998 // the HAL we must save this until we resume 3999 // BUG - this will be wrong if a different track is made active, 4000 // in that case we want to discard the pending data in the 4001 // mixbuffer and tell the client to present it again when the 4002 // track is resumed 4003 mPausedWriteLength = mCurrentWriteLength; 4004 mPausedBytesRemaining = mBytesRemaining; 4005 mBytesRemaining = 0; // stop writing 4006 } 4007 tracksToRemove->add(track); 4008 } else if (track->framesReady() && track->isReady() && 4009 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4010 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4011 if (track->mFillingUpStatus == Track::FS_FILLED) { 4012 track->mFillingUpStatus = Track::FS_ACTIVE; 4013 // make sure processVolume_l() will apply new volume even if 0 4014 mLeftVolFloat = mRightVolFloat = -1.0; 4015 if (track->mState == TrackBase::RESUMING) { 4016 track->mState = TrackBase::ACTIVE; 4017 if (last) { 4018 if (mPausedBytesRemaining) { 4019 // Need to continue write that was interrupted 4020 mCurrentWriteLength = mPausedWriteLength; 4021 mBytesRemaining = mPausedBytesRemaining; 4022 mPausedBytesRemaining = 0; 4023 } 4024 if (mHwPaused) { 4025 doHwResume = true; 4026 mHwPaused = false; 4027 // threadLoop_mix() will handle the case that we need to 4028 // resume an interrupted write 4029 } 4030 // enable write to audio HAL 4031 sleepTime = 0; 4032 } 4033 } 4034 } 4035 4036 if (last) { 4037 // reset retry count 4038 track->mRetryCount = kMaxTrackRetriesOffload; 4039 mActiveTrack = t; 4040 mixerStatus = MIXER_TRACKS_READY; 4041 } 4042 } else { 4043 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4044 if (track->isStopping_1()) { 4045 // Hardware buffer can hold a large amount of audio so we must 4046 // wait for all current track's data to drain before we say 4047 // that the track is stopped. 4048 if (mBytesRemaining == 0) { 4049 // Only start draining when all data in mixbuffer 4050 // has been written 4051 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4052 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4053 // do not drain if no data was ever sent to HAL (mStandby == true) 4054 if (last && !mStandby) { 4055 sleepTime = 0; 4056 standbyTime = systemTime() + standbyDelay; 4057 mixerStatus = MIXER_DRAIN_TRACK; 4058 mDrainSequence += 2; 4059 if (mHwPaused) { 4060 // It is possible to move from PAUSED to STOPPING_1 without 4061 // a resume so we must ensure hardware is running 4062 mOutput->stream->resume(mOutput->stream); 4063 mHwPaused = false; 4064 } 4065 } 4066 } 4067 } else if (track->isStopping_2()) { 4068 // Drain has completed or we are in standby, signal presentation complete 4069 if (!(mDrainSequence & 1) || !last || mStandby) { 4070 track->mState = TrackBase::STOPPED; 4071 size_t audioHALFrames = 4072 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4073 size_t framesWritten = 4074 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4075 track->presentationComplete(framesWritten, audioHALFrames); 4076 track->reset(); 4077 tracksToRemove->add(track); 4078 } 4079 } else { 4080 // No buffers for this track. Give it a few chances to 4081 // fill a buffer, then remove it from active list. 4082 if (--(track->mRetryCount) <= 0) { 4083 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4084 track->name()); 4085 tracksToRemove->add(track); 4086 } else if (last){ 4087 mixerStatus = MIXER_TRACKS_ENABLED; 4088 } 4089 } 4090 } 4091 // compute volume for this track 4092 processVolume_l(track, last); 4093 } 4094 4095 // make sure the pause/flush/resume sequence is executed in the right order. 4096 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4097 // before flush and then resume HW. This can happen in case of pause/flush/resume 4098 // if resume is received before pause is executed. 4099 if (doHwPause || (mFlushPending && !mHwPaused && (count != 0))) { 4100 mOutput->stream->pause(mOutput->stream); 4101 if (!doHwPause) { 4102 doHwResume = true; 4103 } 4104 } 4105 if (mFlushPending) { 4106 flushHw_l(); 4107 mFlushPending = false; 4108 } 4109 if (doHwResume) { 4110 mOutput->stream->resume(mOutput->stream); 4111 } 4112 4113 // remove all the tracks that need to be... 4114 removeTracks_l(*tracksToRemove); 4115 4116 return mixerStatus; 4117} 4118 4119void AudioFlinger::OffloadThread::flushOutput_l() 4120{ 4121 mFlushPending = true; 4122} 4123 4124// must be called with thread mutex locked 4125bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4126{ 4127 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4128 mWriteAckSequence, mDrainSequence); 4129 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4130 return true; 4131 } 4132 return false; 4133} 4134 4135// must be called with thread mutex locked 4136bool AudioFlinger::OffloadThread::shouldStandby_l() 4137{ 4138 bool TrackPaused = false; 4139 4140 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4141 // after a timeout and we will enter standby then. 4142 if (mTracks.size() > 0) { 4143 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4144 } 4145 4146 return !mStandby && !TrackPaused; 4147} 4148 4149 4150bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4151{ 4152 Mutex::Autolock _l(mLock); 4153 return waitingAsyncCallback_l(); 4154} 4155 4156void AudioFlinger::OffloadThread::flushHw_l() 4157{ 4158 mOutput->stream->flush(mOutput->stream); 4159 // Flush anything still waiting in the mixbuffer 4160 mCurrentWriteLength = 0; 4161 mBytesRemaining = 0; 4162 mPausedWriteLength = 0; 4163 mPausedBytesRemaining = 0; 4164 if (mUseAsyncWrite) { 4165 // discard any pending drain or write ack by incrementing sequence 4166 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4167 mDrainSequence = (mDrainSequence + 2) & ~1; 4168 ALOG_ASSERT(mCallbackThread != 0); 4169 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4170 mCallbackThread->setDraining(mDrainSequence); 4171 } 4172} 4173 4174// ---------------------------------------------------------------------------- 4175 4176AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4177 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4178 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4179 DUPLICATING), 4180 mWaitTimeMs(UINT_MAX) 4181{ 4182 addOutputTrack(mainThread); 4183} 4184 4185AudioFlinger::DuplicatingThread::~DuplicatingThread() 4186{ 4187 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4188 mOutputTracks[i]->destroy(); 4189 } 4190} 4191 4192void AudioFlinger::DuplicatingThread::threadLoop_mix() 4193{ 4194 // mix buffers... 4195 if (outputsReady(outputTracks)) { 4196 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4197 } else { 4198 memset(mMixBuffer, 0, mixBufferSize); 4199 } 4200 sleepTime = 0; 4201 writeFrames = mNormalFrameCount; 4202 mCurrentWriteLength = mixBufferSize; 4203 standbyTime = systemTime() + standbyDelay; 4204} 4205 4206void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4207{ 4208 if (sleepTime == 0) { 4209 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4210 sleepTime = activeSleepTime; 4211 } else { 4212 sleepTime = idleSleepTime; 4213 } 4214 } else if (mBytesWritten != 0) { 4215 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4216 writeFrames = mNormalFrameCount; 4217 memset(mMixBuffer, 0, mixBufferSize); 4218 } else { 4219 // flush remaining overflow buffers in output tracks 4220 writeFrames = 0; 4221 } 4222 sleepTime = 0; 4223 } 4224} 4225 4226ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4227{ 4228 for (size_t i = 0; i < outputTracks.size(); i++) { 4229 outputTracks[i]->write(mMixBuffer, writeFrames); 4230 } 4231 return (ssize_t)mixBufferSize; 4232} 4233 4234void AudioFlinger::DuplicatingThread::threadLoop_standby() 4235{ 4236 // DuplicatingThread implements standby by stopping all tracks 4237 for (size_t i = 0; i < outputTracks.size(); i++) { 4238 outputTracks[i]->stop(); 4239 } 4240} 4241 4242void AudioFlinger::DuplicatingThread::saveOutputTracks() 4243{ 4244 outputTracks = mOutputTracks; 4245} 4246 4247void AudioFlinger::DuplicatingThread::clearOutputTracks() 4248{ 4249 outputTracks.clear(); 4250} 4251 4252void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4253{ 4254 Mutex::Autolock _l(mLock); 4255 // FIXME explain this formula 4256 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4257 OutputTrack *outputTrack = new OutputTrack(thread, 4258 this, 4259 mSampleRate, 4260 mFormat, 4261 mChannelMask, 4262 frameCount, 4263 IPCThreadState::self()->getCallingUid()); 4264 if (outputTrack->cblk() != NULL) { 4265 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4266 mOutputTracks.add(outputTrack); 4267 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4268 updateWaitTime_l(); 4269 } 4270} 4271 4272void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4273{ 4274 Mutex::Autolock _l(mLock); 4275 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4276 if (mOutputTracks[i]->thread() == thread) { 4277 mOutputTracks[i]->destroy(); 4278 mOutputTracks.removeAt(i); 4279 updateWaitTime_l(); 4280 return; 4281 } 4282 } 4283 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4284} 4285 4286// caller must hold mLock 4287void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4288{ 4289 mWaitTimeMs = UINT_MAX; 4290 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4291 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4292 if (strong != 0) { 4293 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4294 if (waitTimeMs < mWaitTimeMs) { 4295 mWaitTimeMs = waitTimeMs; 4296 } 4297 } 4298 } 4299} 4300 4301 4302bool AudioFlinger::DuplicatingThread::outputsReady( 4303 const SortedVector< sp<OutputTrack> > &outputTracks) 4304{ 4305 for (size_t i = 0; i < outputTracks.size(); i++) { 4306 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4307 if (thread == 0) { 4308 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4309 outputTracks[i].get()); 4310 return false; 4311 } 4312 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4313 // see note at standby() declaration 4314 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4315 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4316 thread.get()); 4317 return false; 4318 } 4319 } 4320 return true; 4321} 4322 4323uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4324{ 4325 return (mWaitTimeMs * 1000) / 2; 4326} 4327 4328void AudioFlinger::DuplicatingThread::cacheParameters_l() 4329{ 4330 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4331 updateWaitTime_l(); 4332 4333 MixerThread::cacheParameters_l(); 4334} 4335 4336// ---------------------------------------------------------------------------- 4337// Record 4338// ---------------------------------------------------------------------------- 4339 4340AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4341 AudioStreamIn *input, 4342 uint32_t sampleRate, 4343 audio_channel_mask_t channelMask, 4344 audio_io_handle_t id, 4345 audio_devices_t outDevice, 4346 audio_devices_t inDevice 4347#ifdef TEE_SINK 4348 , const sp<NBAIO_Sink>& teeSink 4349#endif 4350 ) : 4351 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4352 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4353 // mRsmpInIndex and mBufferSize set by readInputParameters() 4354 mReqChannelCount(popcount(channelMask)), 4355 mReqSampleRate(sampleRate) 4356 // mBytesRead is only meaningful while active, and so is cleared in start() 4357 // (but might be better to also clear here for dump?) 4358#ifdef TEE_SINK 4359 , mTeeSink(teeSink) 4360#endif 4361{ 4362 snprintf(mName, kNameLength, "AudioIn_%X", id); 4363 4364 readInputParameters(); 4365} 4366 4367 4368AudioFlinger::RecordThread::~RecordThread() 4369{ 4370 delete[] mRsmpInBuffer; 4371 delete mResampler; 4372 delete[] mRsmpOutBuffer; 4373} 4374 4375void AudioFlinger::RecordThread::onFirstRef() 4376{ 4377 run(mName, PRIORITY_URGENT_AUDIO); 4378} 4379 4380status_t AudioFlinger::RecordThread::readyToRun() 4381{ 4382 status_t status = initCheck(); 4383 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4384 return status; 4385} 4386 4387bool AudioFlinger::RecordThread::threadLoop() 4388{ 4389 AudioBufferProvider::Buffer buffer; 4390 sp<RecordTrack> activeTrack; 4391 Vector< sp<EffectChain> > effectChains; 4392 4393 nsecs_t lastWarning = 0; 4394 4395 inputStandBy(); 4396 { 4397 Mutex::Autolock _l(mLock); 4398 activeTrack = mActiveTrack; 4399 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4400 } 4401 4402 // used to verify we've read at least once before evaluating how many bytes were read 4403 bool readOnce = false; 4404 4405 // start recording 4406 while (!exitPending()) { 4407 4408 processConfigEvents(); 4409 4410 { // scope for mLock 4411 Mutex::Autolock _l(mLock); 4412 checkForNewParameters_l(); 4413 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4414 SortedVector<int> tmp; 4415 tmp.add(mActiveTrack->uid()); 4416 updateWakeLockUids_l(tmp); 4417 } 4418 activeTrack = mActiveTrack; 4419 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4420 standby(); 4421 4422 if (exitPending()) { 4423 break; 4424 } 4425 4426 releaseWakeLock_l(); 4427 ALOGV("RecordThread: loop stopping"); 4428 // go to sleep 4429 mWaitWorkCV.wait(mLock); 4430 ALOGV("RecordThread: loop starting"); 4431 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4432 continue; 4433 } 4434 if (mActiveTrack != 0) { 4435 if (mActiveTrack->isTerminated()) { 4436 removeTrack_l(mActiveTrack); 4437 mActiveTrack.clear(); 4438 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4439 standby(); 4440 mActiveTrack.clear(); 4441 mStartStopCond.broadcast(); 4442 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4443 if (mReqChannelCount != mActiveTrack->channelCount()) { 4444 mActiveTrack.clear(); 4445 mStartStopCond.broadcast(); 4446 } else if (readOnce) { 4447 // record start succeeds only if first read from audio input 4448 // succeeds 4449 if (mBytesRead >= 0) { 4450 mActiveTrack->mState = TrackBase::ACTIVE; 4451 } else { 4452 mActiveTrack.clear(); 4453 } 4454 mStartStopCond.broadcast(); 4455 } 4456 mStandby = false; 4457 } 4458 } 4459 4460 lockEffectChains_l(effectChains); 4461 } 4462 4463 if (mActiveTrack != 0) { 4464 if (mActiveTrack->mState != TrackBase::ACTIVE && 4465 mActiveTrack->mState != TrackBase::RESUMING) { 4466 unlockEffectChains(effectChains); 4467 usleep(kRecordThreadSleepUs); 4468 continue; 4469 } 4470 for (size_t i = 0; i < effectChains.size(); i ++) { 4471 effectChains[i]->process_l(); 4472 } 4473 4474 buffer.frameCount = mFrameCount; 4475 status_t status = mActiveTrack->getNextBuffer(&buffer); 4476 if (status == NO_ERROR) { 4477 readOnce = true; 4478 size_t framesOut = buffer.frameCount; 4479 if (mResampler == NULL) { 4480 // no resampling 4481 while (framesOut) { 4482 size_t framesIn = mFrameCount - mRsmpInIndex; 4483 if (framesIn) { 4484 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4485 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4486 mActiveTrack->mFrameSize; 4487 if (framesIn > framesOut) 4488 framesIn = framesOut; 4489 mRsmpInIndex += framesIn; 4490 framesOut -= framesIn; 4491 if (mChannelCount == mReqChannelCount) { 4492 memcpy(dst, src, framesIn * mFrameSize); 4493 } else { 4494 if (mChannelCount == 1) { 4495 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4496 (int16_t *)src, framesIn); 4497 } else { 4498 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4499 (int16_t *)src, framesIn); 4500 } 4501 } 4502 } 4503 if (framesOut && mFrameCount == mRsmpInIndex) { 4504 void *readInto; 4505 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4506 readInto = buffer.raw; 4507 framesOut = 0; 4508 } else { 4509 readInto = mRsmpInBuffer; 4510 mRsmpInIndex = 0; 4511 } 4512 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4513 mBufferSize); 4514 if (mBytesRead <= 0) { 4515 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4516 { 4517 ALOGE("Error reading audio input"); 4518 // Force input into standby so that it tries to 4519 // recover at next read attempt 4520 inputStandBy(); 4521 usleep(kRecordThreadSleepUs); 4522 } 4523 mRsmpInIndex = mFrameCount; 4524 framesOut = 0; 4525 buffer.frameCount = 0; 4526 } 4527#ifdef TEE_SINK 4528 else if (mTeeSink != 0) { 4529 (void) mTeeSink->write(readInto, 4530 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4531 } 4532#endif 4533 } 4534 } 4535 } else { 4536 // resampling 4537 4538 // resampler accumulates, but we only have one source track 4539 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4540 // alter output frame count as if we were expecting stereo samples 4541 if (mChannelCount == 1 && mReqChannelCount == 1) { 4542 framesOut >>= 1; 4543 } 4544 mResampler->resample(mRsmpOutBuffer, framesOut, 4545 this /* AudioBufferProvider* */); 4546 // ditherAndClamp() works as long as all buffers returned by 4547 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4548 if (mChannelCount == 2 && mReqChannelCount == 1) { 4549 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4550 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4551 // the resampler always outputs stereo samples: 4552 // do post stereo to mono conversion 4553 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4554 framesOut); 4555 } else { 4556 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4557 } 4558 // now done with mRsmpOutBuffer 4559 4560 } 4561 if (mFramestoDrop == 0) { 4562 mActiveTrack->releaseBuffer(&buffer); 4563 } else { 4564 if (mFramestoDrop > 0) { 4565 mFramestoDrop -= buffer.frameCount; 4566 if (mFramestoDrop <= 0) { 4567 clearSyncStartEvent(); 4568 } 4569 } else { 4570 mFramestoDrop += buffer.frameCount; 4571 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4572 mSyncStartEvent->isCancelled()) { 4573 ALOGW("Synced record %s, session %d, trigger session %d", 4574 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4575 mActiveTrack->sessionId(), 4576 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4577 clearSyncStartEvent(); 4578 } 4579 } 4580 } 4581 mActiveTrack->clearOverflow(); 4582 } 4583 // client isn't retrieving buffers fast enough 4584 else { 4585 if (!mActiveTrack->setOverflow()) { 4586 nsecs_t now = systemTime(); 4587 if ((now - lastWarning) > kWarningThrottleNs) { 4588 ALOGW("RecordThread: buffer overflow"); 4589 lastWarning = now; 4590 } 4591 } 4592 // Release the processor for a while before asking for a new buffer. 4593 // This will give the application more chance to read from the buffer and 4594 // clear the overflow. 4595 usleep(kRecordThreadSleepUs); 4596 } 4597 } 4598 // enable changes in effect chain 4599 unlockEffectChains(effectChains); 4600 effectChains.clear(); 4601 } 4602 4603 standby(); 4604 4605 { 4606 Mutex::Autolock _l(mLock); 4607 for (size_t i = 0; i < mTracks.size(); i++) { 4608 sp<RecordTrack> track = mTracks[i]; 4609 track->invalidate(); 4610 } 4611 mActiveTrack.clear(); 4612 mStartStopCond.broadcast(); 4613 } 4614 4615 releaseWakeLock(); 4616 4617 ALOGV("RecordThread %p exiting", this); 4618 return false; 4619} 4620 4621void AudioFlinger::RecordThread::standby() 4622{ 4623 if (!mStandby) { 4624 inputStandBy(); 4625 mStandby = true; 4626 } 4627} 4628 4629void AudioFlinger::RecordThread::inputStandBy() 4630{ 4631 mInput->stream->common.standby(&mInput->stream->common); 4632} 4633 4634sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4635 const sp<AudioFlinger::Client>& client, 4636 uint32_t sampleRate, 4637 audio_format_t format, 4638 audio_channel_mask_t channelMask, 4639 size_t frameCount, 4640 int sessionId, 4641 int uid, 4642 IAudioFlinger::track_flags_t *flags, 4643 pid_t tid, 4644 status_t *status) 4645{ 4646 sp<RecordTrack> track; 4647 status_t lStatus; 4648 4649 lStatus = initCheck(); 4650 if (lStatus != NO_ERROR) { 4651 ALOGE("createRecordTrack_l() audio driver not initialized"); 4652 goto Exit; 4653 } 4654 // client expresses a preference for FAST, but we get the final say 4655 if (*flags & IAudioFlinger::TRACK_FAST) { 4656 if ( 4657 // use case: callback handler and frame count is default or at least as large as HAL 4658 ( 4659 (tid != -1) && 4660 ((frameCount == 0) || 4661 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4662 ) && 4663 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4664 // mono or stereo 4665 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4666 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4667 // hardware sample rate 4668 (sampleRate == mSampleRate) && 4669 // record thread has an associated fast recorder 4670 hasFastRecorder() 4671 // FIXME test that RecordThread for this fast track has a capable output HAL 4672 // FIXME add a permission test also? 4673 ) { 4674 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4675 if (frameCount == 0) { 4676 frameCount = mFrameCount * kFastTrackMultiplier; 4677 } 4678 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4679 frameCount, mFrameCount); 4680 } else { 4681 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4682 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4683 "hasFastRecorder=%d tid=%d", 4684 frameCount, mFrameCount, format, 4685 audio_is_linear_pcm(format), 4686 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4687 *flags &= ~IAudioFlinger::TRACK_FAST; 4688 // For compatibility with AudioRecord calculation, buffer depth is forced 4689 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4690 // This is probably too conservative, but legacy application code may depend on it. 4691 // If you change this calculation, also review the start threshold which is related. 4692 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4693 size_t mNormalFrameCount = 2048; // FIXME 4694 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4695 if (minBufCount < 2) { 4696 minBufCount = 2; 4697 } 4698 size_t minFrameCount = mNormalFrameCount * minBufCount; 4699 if (frameCount < minFrameCount) { 4700 frameCount = minFrameCount; 4701 } 4702 } 4703 } 4704 4705 // FIXME use flags and tid similar to createTrack_l() 4706 4707 { // scope for mLock 4708 Mutex::Autolock _l(mLock); 4709 4710 track = new RecordTrack(this, client, sampleRate, 4711 format, channelMask, frameCount, sessionId, uid); 4712 4713 if (track->getCblk() == 0) { 4714 ALOGE("createRecordTrack_l() no control block"); 4715 lStatus = NO_MEMORY; 4716 track.clear(); 4717 goto Exit; 4718 } 4719 mTracks.add(track); 4720 4721 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4722 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4723 mAudioFlinger->btNrecIsOff(); 4724 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4725 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4726 4727 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4728 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4729 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4730 // so ask activity manager to do this on our behalf 4731 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4732 } 4733 } 4734 lStatus = NO_ERROR; 4735 4736Exit: 4737 if (status) { 4738 *status = lStatus; 4739 } 4740 return track; 4741} 4742 4743status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4744 AudioSystem::sync_event_t event, 4745 int triggerSession) 4746{ 4747 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4748 sp<ThreadBase> strongMe = this; 4749 status_t status = NO_ERROR; 4750 4751 if (event == AudioSystem::SYNC_EVENT_NONE) { 4752 clearSyncStartEvent(); 4753 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4754 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4755 triggerSession, 4756 recordTrack->sessionId(), 4757 syncStartEventCallback, 4758 this); 4759 // Sync event can be cancelled by the trigger session if the track is not in a 4760 // compatible state in which case we start record immediately 4761 if (mSyncStartEvent->isCancelled()) { 4762 clearSyncStartEvent(); 4763 } else { 4764 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4765 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4766 } 4767 } 4768 4769 { 4770 AutoMutex lock(mLock); 4771 if (mActiveTrack != 0) { 4772 if (recordTrack != mActiveTrack.get()) { 4773 status = -EBUSY; 4774 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4775 mActiveTrack->mState = TrackBase::ACTIVE; 4776 } 4777 return status; 4778 } 4779 4780 recordTrack->mState = TrackBase::IDLE; 4781 mActiveTrack = recordTrack; 4782 mLock.unlock(); 4783 status_t status = AudioSystem::startInput(mId); 4784 mLock.lock(); 4785 if (status != NO_ERROR) { 4786 mActiveTrack.clear(); 4787 clearSyncStartEvent(); 4788 return status; 4789 } 4790 mRsmpInIndex = mFrameCount; 4791 mBytesRead = 0; 4792 if (mResampler != NULL) { 4793 mResampler->reset(); 4794 } 4795 mActiveTrack->mState = TrackBase::RESUMING; 4796 // signal thread to start 4797 ALOGV("Signal record thread"); 4798 mWaitWorkCV.broadcast(); 4799 // do not wait for mStartStopCond if exiting 4800 if (exitPending()) { 4801 mActiveTrack.clear(); 4802 status = INVALID_OPERATION; 4803 goto startError; 4804 } 4805 mStartStopCond.wait(mLock); 4806 if (mActiveTrack == 0) { 4807 ALOGV("Record failed to start"); 4808 status = BAD_VALUE; 4809 goto startError; 4810 } 4811 ALOGV("Record started OK"); 4812 return status; 4813 } 4814 4815startError: 4816 AudioSystem::stopInput(mId); 4817 clearSyncStartEvent(); 4818 return status; 4819} 4820 4821void AudioFlinger::RecordThread::clearSyncStartEvent() 4822{ 4823 if (mSyncStartEvent != 0) { 4824 mSyncStartEvent->cancel(); 4825 } 4826 mSyncStartEvent.clear(); 4827 mFramestoDrop = 0; 4828} 4829 4830void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4831{ 4832 sp<SyncEvent> strongEvent = event.promote(); 4833 4834 if (strongEvent != 0) { 4835 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4836 me->handleSyncStartEvent(strongEvent); 4837 } 4838} 4839 4840void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4841{ 4842 if (event == mSyncStartEvent) { 4843 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4844 // from audio HAL 4845 mFramestoDrop = mFrameCount * 2; 4846 } 4847} 4848 4849bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4850 ALOGV("RecordThread::stop"); 4851 AutoMutex _l(mLock); 4852 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4853 return false; 4854 } 4855 recordTrack->mState = TrackBase::PAUSING; 4856 // do not wait for mStartStopCond if exiting 4857 if (exitPending()) { 4858 return true; 4859 } 4860 mStartStopCond.wait(mLock); 4861 // if we have been restarted, recordTrack == mActiveTrack.get() here 4862 if (exitPending() || recordTrack != mActiveTrack.get()) { 4863 ALOGV("Record stopped OK"); 4864 return true; 4865 } 4866 return false; 4867} 4868 4869bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4870{ 4871 return false; 4872} 4873 4874status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4875{ 4876#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4877 if (!isValidSyncEvent(event)) { 4878 return BAD_VALUE; 4879 } 4880 4881 int eventSession = event->triggerSession(); 4882 status_t ret = NAME_NOT_FOUND; 4883 4884 Mutex::Autolock _l(mLock); 4885 4886 for (size_t i = 0; i < mTracks.size(); i++) { 4887 sp<RecordTrack> track = mTracks[i]; 4888 if (eventSession == track->sessionId()) { 4889 (void) track->setSyncEvent(event); 4890 ret = NO_ERROR; 4891 } 4892 } 4893 return ret; 4894#else 4895 return BAD_VALUE; 4896#endif 4897} 4898 4899// destroyTrack_l() must be called with ThreadBase::mLock held 4900void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4901{ 4902 track->terminate(); 4903 track->mState = TrackBase::STOPPED; 4904 // active tracks are removed by threadLoop() 4905 if (mActiveTrack != track) { 4906 removeTrack_l(track); 4907 } 4908} 4909 4910void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4911{ 4912 mTracks.remove(track); 4913 // need anything related to effects here? 4914} 4915 4916void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4917{ 4918 dumpInternals(fd, args); 4919 dumpTracks(fd, args); 4920 dumpEffectChains(fd, args); 4921} 4922 4923void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4924{ 4925 const size_t SIZE = 256; 4926 char buffer[SIZE]; 4927 String8 result; 4928 4929 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4930 result.append(buffer); 4931 4932 if (mActiveTrack != 0) { 4933 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4934 result.append(buffer); 4935 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4936 result.append(buffer); 4937 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4938 result.append(buffer); 4939 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4940 result.append(buffer); 4941 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4942 result.append(buffer); 4943 } else { 4944 result.append("No active record client\n"); 4945 } 4946 4947 write(fd, result.string(), result.size()); 4948 4949 dumpBase(fd, args); 4950} 4951 4952void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4953{ 4954 const size_t SIZE = 256; 4955 char buffer[SIZE]; 4956 String8 result; 4957 4958 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4959 result.append(buffer); 4960 RecordTrack::appendDumpHeader(result); 4961 for (size_t i = 0; i < mTracks.size(); ++i) { 4962 sp<RecordTrack> track = mTracks[i]; 4963 if (track != 0) { 4964 track->dump(buffer, SIZE); 4965 result.append(buffer); 4966 } 4967 } 4968 4969 if (mActiveTrack != 0) { 4970 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4971 result.append(buffer); 4972 RecordTrack::appendDumpHeader(result); 4973 mActiveTrack->dump(buffer, SIZE); 4974 result.append(buffer); 4975 4976 } 4977 write(fd, result.string(), result.size()); 4978} 4979 4980// AudioBufferProvider interface 4981status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4982{ 4983 size_t framesReq = buffer->frameCount; 4984 size_t framesReady = mFrameCount - mRsmpInIndex; 4985 int channelCount; 4986 4987 if (framesReady == 0) { 4988 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4989 if (mBytesRead <= 0) { 4990 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4991 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4992 // Force input into standby so that it tries to 4993 // recover at next read attempt 4994 inputStandBy(); 4995 usleep(kRecordThreadSleepUs); 4996 } 4997 buffer->raw = NULL; 4998 buffer->frameCount = 0; 4999 return NOT_ENOUGH_DATA; 5000 } 5001 mRsmpInIndex = 0; 5002 framesReady = mFrameCount; 5003 } 5004 5005 if (framesReq > framesReady) { 5006 framesReq = framesReady; 5007 } 5008 5009 if (mChannelCount == 1 && mReqChannelCount == 2) { 5010 channelCount = 1; 5011 } else { 5012 channelCount = 2; 5013 } 5014 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5015 buffer->frameCount = framesReq; 5016 return NO_ERROR; 5017} 5018 5019// AudioBufferProvider interface 5020void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5021{ 5022 mRsmpInIndex += buffer->frameCount; 5023 buffer->frameCount = 0; 5024} 5025 5026bool AudioFlinger::RecordThread::checkForNewParameters_l() 5027{ 5028 bool reconfig = false; 5029 5030 while (!mNewParameters.isEmpty()) { 5031 status_t status = NO_ERROR; 5032 String8 keyValuePair = mNewParameters[0]; 5033 AudioParameter param = AudioParameter(keyValuePair); 5034 int value; 5035 audio_format_t reqFormat = mFormat; 5036 uint32_t reqSamplingRate = mReqSampleRate; 5037 uint32_t reqChannelCount = mReqChannelCount; 5038 5039 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5040 reqSamplingRate = value; 5041 reconfig = true; 5042 } 5043 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5044 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5045 status = BAD_VALUE; 5046 } else { 5047 reqFormat = (audio_format_t) value; 5048 reconfig = true; 5049 } 5050 } 5051 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5052 reqChannelCount = popcount(value); 5053 reconfig = true; 5054 } 5055 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5056 // do not accept frame count changes if tracks are open as the track buffer 5057 // size depends on frame count and correct behavior would not be guaranteed 5058 // if frame count is changed after track creation 5059 if (mActiveTrack != 0) { 5060 status = INVALID_OPERATION; 5061 } else { 5062 reconfig = true; 5063 } 5064 } 5065 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5066 // forward device change to effects that have requested to be 5067 // aware of attached audio device. 5068 for (size_t i = 0; i < mEffectChains.size(); i++) { 5069 mEffectChains[i]->setDevice_l(value); 5070 } 5071 5072 // store input device and output device but do not forward output device to audio HAL. 5073 // Note that status is ignored by the caller for output device 5074 // (see AudioFlinger::setParameters() 5075 if (audio_is_output_devices(value)) { 5076 mOutDevice = value; 5077 status = BAD_VALUE; 5078 } else { 5079 mInDevice = value; 5080 // disable AEC and NS if the device is a BT SCO headset supporting those 5081 // pre processings 5082 if (mTracks.size() > 0) { 5083 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5084 mAudioFlinger->btNrecIsOff(); 5085 for (size_t i = 0; i < mTracks.size(); i++) { 5086 sp<RecordTrack> track = mTracks[i]; 5087 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5088 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5089 } 5090 } 5091 } 5092 } 5093 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5094 mAudioSource != (audio_source_t)value) { 5095 // forward device change to effects that have requested to be 5096 // aware of attached audio device. 5097 for (size_t i = 0; i < mEffectChains.size(); i++) { 5098 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5099 } 5100 mAudioSource = (audio_source_t)value; 5101 } 5102 if (status == NO_ERROR) { 5103 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5104 keyValuePair.string()); 5105 if (status == INVALID_OPERATION) { 5106 inputStandBy(); 5107 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5108 keyValuePair.string()); 5109 } 5110 if (reconfig) { 5111 if (status == BAD_VALUE && 5112 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5113 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5114 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5115 <= (2 * reqSamplingRate)) && 5116 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5117 <= FCC_2 && 5118 (reqChannelCount <= FCC_2)) { 5119 status = NO_ERROR; 5120 } 5121 if (status == NO_ERROR) { 5122 readInputParameters(); 5123 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5124 } 5125 } 5126 } 5127 5128 mNewParameters.removeAt(0); 5129 5130 mParamStatus = status; 5131 mParamCond.signal(); 5132 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5133 // already timed out waiting for the status and will never signal the condition. 5134 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5135 } 5136 return reconfig; 5137} 5138 5139String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5140{ 5141 Mutex::Autolock _l(mLock); 5142 if (initCheck() != NO_ERROR) { 5143 return String8(); 5144 } 5145 5146 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5147 const String8 out_s8(s); 5148 free(s); 5149 return out_s8; 5150} 5151 5152void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5153 AudioSystem::OutputDescriptor desc; 5154 void *param2 = NULL; 5155 5156 switch (event) { 5157 case AudioSystem::INPUT_OPENED: 5158 case AudioSystem::INPUT_CONFIG_CHANGED: 5159 desc.channelMask = mChannelMask; 5160 desc.samplingRate = mSampleRate; 5161 desc.format = mFormat; 5162 desc.frameCount = mFrameCount; 5163 desc.latency = 0; 5164 param2 = &desc; 5165 break; 5166 5167 case AudioSystem::INPUT_CLOSED: 5168 default: 5169 break; 5170 } 5171 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5172} 5173 5174void AudioFlinger::RecordThread::readInputParameters() 5175{ 5176 delete[] mRsmpInBuffer; 5177 // mRsmpInBuffer is always assigned a new[] below 5178 delete[] mRsmpOutBuffer; 5179 mRsmpOutBuffer = NULL; 5180 delete mResampler; 5181 mResampler = NULL; 5182 5183 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5184 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5185 mChannelCount = popcount(mChannelMask); 5186 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5187 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5188 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5189 } 5190 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5191 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5192 mFrameCount = mBufferSize / mFrameSize; 5193 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5194 5195 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5196 { 5197 int channelCount; 5198 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5199 // stereo to mono post process as the resampler always outputs stereo. 5200 if (mChannelCount == 1 && mReqChannelCount == 2) { 5201 channelCount = 1; 5202 } else { 5203 channelCount = 2; 5204 } 5205 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5206 mResampler->setSampleRate(mSampleRate); 5207 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5208 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5209 5210 // optmization: if mono to mono, alter input frame count as if we were inputing 5211 // stereo samples 5212 if (mChannelCount == 1 && mReqChannelCount == 1) { 5213 mFrameCount >>= 1; 5214 } 5215 5216 } 5217 mRsmpInIndex = mFrameCount; 5218} 5219 5220unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5221{ 5222 Mutex::Autolock _l(mLock); 5223 if (initCheck() != NO_ERROR) { 5224 return 0; 5225 } 5226 5227 return mInput->stream->get_input_frames_lost(mInput->stream); 5228} 5229 5230uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5231{ 5232 Mutex::Autolock _l(mLock); 5233 uint32_t result = 0; 5234 if (getEffectChain_l(sessionId) != 0) { 5235 result = EFFECT_SESSION; 5236 } 5237 5238 for (size_t i = 0; i < mTracks.size(); ++i) { 5239 if (sessionId == mTracks[i]->sessionId()) { 5240 result |= TRACK_SESSION; 5241 break; 5242 } 5243 } 5244 5245 return result; 5246} 5247 5248KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5249{ 5250 KeyedVector<int, bool> ids; 5251 Mutex::Autolock _l(mLock); 5252 for (size_t j = 0; j < mTracks.size(); ++j) { 5253 sp<RecordThread::RecordTrack> track = mTracks[j]; 5254 int sessionId = track->sessionId(); 5255 if (ids.indexOfKey(sessionId) < 0) { 5256 ids.add(sessionId, true); 5257 } 5258 } 5259 return ids; 5260} 5261 5262AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5263{ 5264 Mutex::Autolock _l(mLock); 5265 AudioStreamIn *input = mInput; 5266 mInput = NULL; 5267 return input; 5268} 5269 5270// this method must always be called either with ThreadBase mLock held or inside the thread loop 5271audio_stream_t* AudioFlinger::RecordThread::stream() const 5272{ 5273 if (mInput == NULL) { 5274 return NULL; 5275 } 5276 return &mInput->stream->common; 5277} 5278 5279status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5280{ 5281 // only one chain per input thread 5282 if (mEffectChains.size() != 0) { 5283 return INVALID_OPERATION; 5284 } 5285 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5286 5287 chain->setInBuffer(NULL); 5288 chain->setOutBuffer(NULL); 5289 5290 checkSuspendOnAddEffectChain_l(chain); 5291 5292 mEffectChains.add(chain); 5293 5294 return NO_ERROR; 5295} 5296 5297size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5298{ 5299 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5300 ALOGW_IF(mEffectChains.size() != 1, 5301 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5302 chain.get(), mEffectChains.size(), this); 5303 if (mEffectChains.size() == 1) { 5304 mEffectChains.removeAt(0); 5305 } 5306 return 0; 5307} 5308 5309}; // namespace android 5310