Threads.cpp revision 10ac79930120efea20b125f5faedfe89e8674676
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 //FIXME: mStandby should be true here. Is this some kind of hack? 276 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 277 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 278 // mName will be set by concrete (non-virtual) subclass 279 mDeathRecipient(new PMDeathRecipient(this)) 280{ 281} 282 283AudioFlinger::ThreadBase::~ThreadBase() 284{ 285 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 286 for (size_t i = 0; i < mConfigEvents.size(); i++) { 287 delete mConfigEvents[i]; 288 } 289 mConfigEvents.clear(); 290 291 mParamCond.broadcast(); 292 // do not lock the mutex in destructor 293 releaseWakeLock_l(); 294 if (mPowerManager != 0) { 295 sp<IBinder> binder = mPowerManager->asBinder(); 296 binder->unlinkToDeath(mDeathRecipient); 297 } 298} 299 300void AudioFlinger::ThreadBase::exit() 301{ 302 ALOGV("ThreadBase::exit"); 303 // do any cleanup required for exit to succeed 304 preExit(); 305 { 306 // This lock prevents the following race in thread (uniprocessor for illustration): 307 // if (!exitPending()) { 308 // // context switch from here to exit() 309 // // exit() calls requestExit(), what exitPending() observes 310 // // exit() calls signal(), which is dropped since no waiters 311 // // context switch back from exit() to here 312 // mWaitWorkCV.wait(...); 313 // // now thread is hung 314 // } 315 AutoMutex lock(mLock); 316 requestExit(); 317 mWaitWorkCV.broadcast(); 318 } 319 // When Thread::requestExitAndWait is made virtual and this method is renamed to 320 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 321 requestExitAndWait(); 322} 323 324status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 325{ 326 status_t status; 327 328 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 329 Mutex::Autolock _l(mLock); 330 331 mNewParameters.add(keyValuePairs); 332 mWaitWorkCV.signal(); 333 // wait condition with timeout in case the thread loop has exited 334 // before the request could be processed 335 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 336 status = mParamStatus; 337 mWaitWorkCV.signal(); 338 } else { 339 status = TIMED_OUT; 340 } 341 return status; 342} 343 344void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 345{ 346 Mutex::Autolock _l(mLock); 347 sendIoConfigEvent_l(event, param); 348} 349 350// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 351void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 352{ 353 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 354 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 355 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 356 param); 357 mWaitWorkCV.signal(); 358} 359 360// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 361void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 362{ 363 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 364 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 365 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 366 mConfigEvents.size(), pid, tid, prio); 367 mWaitWorkCV.signal(); 368} 369 370void AudioFlinger::ThreadBase::processConfigEvents() 371{ 372 mLock.lock(); 373 while (!mConfigEvents.isEmpty()) { 374 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 375 ConfigEvent *event = mConfigEvents[0]; 376 mConfigEvents.removeAt(0); 377 // release mLock before locking AudioFlinger mLock: lock order is always 378 // AudioFlinger then ThreadBase to avoid cross deadlock 379 mLock.unlock(); 380 switch(event->type()) { 381 case CFG_EVENT_PRIO: { 382 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 383 // FIXME Need to understand why this has be done asynchronously 384 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 385 true /*asynchronous*/); 386 if (err != 0) { 387 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 388 "error %d", 389 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 390 } 391 } break; 392 case CFG_EVENT_IO: { 393 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 394 mAudioFlinger->mLock.lock(); 395 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 396 mAudioFlinger->mLock.unlock(); 397 } break; 398 default: 399 ALOGE("processConfigEvents() unknown event type %d", event->type()); 400 break; 401 } 402 delete event; 403 mLock.lock(); 404 } 405 mLock.unlock(); 406} 407 408void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 409{ 410 const size_t SIZE = 256; 411 char buffer[SIZE]; 412 String8 result; 413 414 bool locked = AudioFlinger::dumpTryLock(mLock); 415 if (!locked) { 416 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 417 write(fd, buffer, strlen(buffer)); 418 } 419 420 snprintf(buffer, SIZE, "io handle: %d\n", mId); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 437 result.append(buffer); 438 439 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 440 result.append(buffer); 441 result.append(" Index Command"); 442 for (size_t i = 0; i < mNewParameters.size(); ++i) { 443 snprintf(buffer, SIZE, "\n %02d ", i); 444 result.append(buffer); 445 result.append(mNewParameters[i]); 446 } 447 448 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 449 result.append(buffer); 450 for (size_t i = 0; i < mConfigEvents.size(); i++) { 451 mConfigEvents[i]->dump(buffer, SIZE); 452 result.append(buffer); 453 } 454 result.append("\n"); 455 456 write(fd, result.string(), result.size()); 457 458 if (locked) { 459 mLock.unlock(); 460 } 461} 462 463void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 464{ 465 const size_t SIZE = 256; 466 char buffer[SIZE]; 467 String8 result; 468 469 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 470 write(fd, buffer, strlen(buffer)); 471 472 for (size_t i = 0; i < mEffectChains.size(); ++i) { 473 sp<EffectChain> chain = mEffectChains[i]; 474 if (chain != 0) { 475 chain->dump(fd, args); 476 } 477 } 478} 479 480void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 481{ 482 Mutex::Autolock _l(mLock); 483 acquireWakeLock_l(uid); 484} 485 486String16 AudioFlinger::ThreadBase::getWakeLockTag() 487{ 488 switch (mType) { 489 case MIXER: 490 return String16("AudioMix"); 491 case DIRECT: 492 return String16("AudioDirectOut"); 493 case DUPLICATING: 494 return String16("AudioDup"); 495 case RECORD: 496 return String16("AudioIn"); 497 case OFFLOAD: 498 return String16("AudioOffload"); 499 default: 500 ALOG_ASSERT(false); 501 return String16("AudioUnknown"); 502 } 503} 504 505void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 506{ 507 getPowerManager_l(); 508 if (mPowerManager != 0) { 509 sp<IBinder> binder = new BBinder(); 510 status_t status; 511 if (uid >= 0) { 512 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 513 binder, 514 getWakeLockTag(), 515 String16("media"), 516 uid); 517 } else { 518 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 519 binder, 520 getWakeLockTag(), 521 String16("media")); 522 } 523 if (status == NO_ERROR) { 524 mWakeLockToken = binder; 525 } 526 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 527 } 528} 529 530void AudioFlinger::ThreadBase::releaseWakeLock() 531{ 532 Mutex::Autolock _l(mLock); 533 releaseWakeLock_l(); 534} 535 536void AudioFlinger::ThreadBase::releaseWakeLock_l() 537{ 538 if (mWakeLockToken != 0) { 539 ALOGV("releaseWakeLock_l() %s", mName); 540 if (mPowerManager != 0) { 541 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 542 } 543 mWakeLockToken.clear(); 544 } 545} 546 547void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 548 Mutex::Autolock _l(mLock); 549 updateWakeLockUids_l(uids); 550} 551 552void AudioFlinger::ThreadBase::getPowerManager_l() { 553 554 if (mPowerManager == 0) { 555 // use checkService() to avoid blocking if power service is not up yet 556 sp<IBinder> binder = 557 defaultServiceManager()->checkService(String16("power")); 558 if (binder == 0) { 559 ALOGW("Thread %s cannot connect to the power manager service", mName); 560 } else { 561 mPowerManager = interface_cast<IPowerManager>(binder); 562 binder->linkToDeath(mDeathRecipient); 563 } 564 } 565} 566 567void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 568 569 getPowerManager_l(); 570 if (mWakeLockToken == NULL) { 571 ALOGE("no wake lock to update!"); 572 return; 573 } 574 if (mPowerManager != 0) { 575 sp<IBinder> binder = new BBinder(); 576 status_t status; 577 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 578 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 579 } 580} 581 582void AudioFlinger::ThreadBase::clearPowerManager() 583{ 584 Mutex::Autolock _l(mLock); 585 releaseWakeLock_l(); 586 mPowerManager.clear(); 587} 588 589void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 590{ 591 sp<ThreadBase> thread = mThread.promote(); 592 if (thread != 0) { 593 thread->clearPowerManager(); 594 } 595 ALOGW("power manager service died !!!"); 596} 597 598void AudioFlinger::ThreadBase::setEffectSuspended( 599 const effect_uuid_t *type, bool suspend, int sessionId) 600{ 601 Mutex::Autolock _l(mLock); 602 setEffectSuspended_l(type, suspend, sessionId); 603} 604 605void AudioFlinger::ThreadBase::setEffectSuspended_l( 606 const effect_uuid_t *type, bool suspend, int sessionId) 607{ 608 sp<EffectChain> chain = getEffectChain_l(sessionId); 609 if (chain != 0) { 610 if (type != NULL) { 611 chain->setEffectSuspended_l(type, suspend); 612 } else { 613 chain->setEffectSuspendedAll_l(suspend); 614 } 615 } 616 617 updateSuspendedSessions_l(type, suspend, sessionId); 618} 619 620void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 621{ 622 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 623 if (index < 0) { 624 return; 625 } 626 627 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 628 mSuspendedSessions.valueAt(index); 629 630 for (size_t i = 0; i < sessionEffects.size(); i++) { 631 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 632 for (int j = 0; j < desc->mRefCount; j++) { 633 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 634 chain->setEffectSuspendedAll_l(true); 635 } else { 636 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 637 desc->mType.timeLow); 638 chain->setEffectSuspended_l(&desc->mType, true); 639 } 640 } 641 } 642} 643 644void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 645 bool suspend, 646 int sessionId) 647{ 648 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 649 650 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 651 652 if (suspend) { 653 if (index >= 0) { 654 sessionEffects = mSuspendedSessions.valueAt(index); 655 } else { 656 mSuspendedSessions.add(sessionId, sessionEffects); 657 } 658 } else { 659 if (index < 0) { 660 return; 661 } 662 sessionEffects = mSuspendedSessions.valueAt(index); 663 } 664 665 666 int key = EffectChain::kKeyForSuspendAll; 667 if (type != NULL) { 668 key = type->timeLow; 669 } 670 index = sessionEffects.indexOfKey(key); 671 672 sp<SuspendedSessionDesc> desc; 673 if (suspend) { 674 if (index >= 0) { 675 desc = sessionEffects.valueAt(index); 676 } else { 677 desc = new SuspendedSessionDesc(); 678 if (type != NULL) { 679 desc->mType = *type; 680 } 681 sessionEffects.add(key, desc); 682 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 683 } 684 desc->mRefCount++; 685 } else { 686 if (index < 0) { 687 return; 688 } 689 desc = sessionEffects.valueAt(index); 690 if (--desc->mRefCount == 0) { 691 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 692 sessionEffects.removeItemsAt(index); 693 if (sessionEffects.isEmpty()) { 694 ALOGV("updateSuspendedSessions_l() restore removing session %d", 695 sessionId); 696 mSuspendedSessions.removeItem(sessionId); 697 } 698 } 699 } 700 if (!sessionEffects.isEmpty()) { 701 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 702 } 703} 704 705void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 706 bool enabled, 707 int sessionId) 708{ 709 Mutex::Autolock _l(mLock); 710 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 711} 712 713void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 714 bool enabled, 715 int sessionId) 716{ 717 if (mType != RECORD) { 718 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 719 // another session. This gives the priority to well behaved effect control panels 720 // and applications not using global effects. 721 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 722 // global effects 723 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 724 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 725 } 726 } 727 728 sp<EffectChain> chain = getEffectChain_l(sessionId); 729 if (chain != 0) { 730 chain->checkSuspendOnEffectEnabled(effect, enabled); 731 } 732} 733 734// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 735sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 736 const sp<AudioFlinger::Client>& client, 737 const sp<IEffectClient>& effectClient, 738 int32_t priority, 739 int sessionId, 740 effect_descriptor_t *desc, 741 int *enabled, 742 status_t *status 743 ) 744{ 745 sp<EffectModule> effect; 746 sp<EffectHandle> handle; 747 status_t lStatus; 748 sp<EffectChain> chain; 749 bool chainCreated = false; 750 bool effectCreated = false; 751 bool effectRegistered = false; 752 753 lStatus = initCheck(); 754 if (lStatus != NO_ERROR) { 755 ALOGW("createEffect_l() Audio driver not initialized."); 756 goto Exit; 757 } 758 759 // Allow global effects only on offloaded and mixer threads 760 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 761 switch (mType) { 762 case MIXER: 763 case OFFLOAD: 764 break; 765 case DIRECT: 766 case DUPLICATING: 767 case RECORD: 768 default: 769 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 770 lStatus = BAD_VALUE; 771 goto Exit; 772 } 773 } 774 775 // Only Pre processor effects are allowed on input threads and only on input threads 776 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 777 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 778 desc->name, desc->flags, mType); 779 lStatus = BAD_VALUE; 780 goto Exit; 781 } 782 783 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 784 785 { // scope for mLock 786 Mutex::Autolock _l(mLock); 787 788 // check for existing effect chain with the requested audio session 789 chain = getEffectChain_l(sessionId); 790 if (chain == 0) { 791 // create a new chain for this session 792 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 793 chain = new EffectChain(this, sessionId); 794 addEffectChain_l(chain); 795 chain->setStrategy(getStrategyForSession_l(sessionId)); 796 chainCreated = true; 797 } else { 798 effect = chain->getEffectFromDesc_l(desc); 799 } 800 801 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 802 803 if (effect == 0) { 804 int id = mAudioFlinger->nextUniqueId(); 805 // Check CPU and memory usage 806 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 807 if (lStatus != NO_ERROR) { 808 goto Exit; 809 } 810 effectRegistered = true; 811 // create a new effect module if none present in the chain 812 effect = new EffectModule(this, chain, desc, id, sessionId); 813 lStatus = effect->status(); 814 if (lStatus != NO_ERROR) { 815 goto Exit; 816 } 817 effect->setOffloaded(mType == OFFLOAD, mId); 818 819 lStatus = chain->addEffect_l(effect); 820 if (lStatus != NO_ERROR) { 821 goto Exit; 822 } 823 effectCreated = true; 824 825 effect->setDevice(mOutDevice); 826 effect->setDevice(mInDevice); 827 effect->setMode(mAudioFlinger->getMode()); 828 effect->setAudioSource(mAudioSource); 829 } 830 // create effect handle and connect it to effect module 831 handle = new EffectHandle(effect, client, effectClient, priority); 832 lStatus = effect->addHandle(handle.get()); 833 if (enabled != NULL) { 834 *enabled = (int)effect->isEnabled(); 835 } 836 } 837 838Exit: 839 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 840 Mutex::Autolock _l(mLock); 841 if (effectCreated) { 842 chain->removeEffect_l(effect); 843 } 844 if (effectRegistered) { 845 AudioSystem::unregisterEffect(effect->id()); 846 } 847 if (chainCreated) { 848 removeEffectChain_l(chain); 849 } 850 handle.clear(); 851 } 852 853 if (status != NULL) { 854 *status = lStatus; 855 } 856 return handle; 857} 858 859sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 860{ 861 Mutex::Autolock _l(mLock); 862 return getEffect_l(sessionId, effectId); 863} 864 865sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 866{ 867 sp<EffectChain> chain = getEffectChain_l(sessionId); 868 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 869} 870 871// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 872// PlaybackThread::mLock held 873status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 874{ 875 // check for existing effect chain with the requested audio session 876 int sessionId = effect->sessionId(); 877 sp<EffectChain> chain = getEffectChain_l(sessionId); 878 bool chainCreated = false; 879 880 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 881 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 882 this, effect->desc().name, effect->desc().flags); 883 884 if (chain == 0) { 885 // create a new chain for this session 886 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 887 chain = new EffectChain(this, sessionId); 888 addEffectChain_l(chain); 889 chain->setStrategy(getStrategyForSession_l(sessionId)); 890 chainCreated = true; 891 } 892 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 893 894 if (chain->getEffectFromId_l(effect->id()) != 0) { 895 ALOGW("addEffect_l() %p effect %s already present in chain %p", 896 this, effect->desc().name, chain.get()); 897 return BAD_VALUE; 898 } 899 900 effect->setOffloaded(mType == OFFLOAD, mId); 901 902 status_t status = chain->addEffect_l(effect); 903 if (status != NO_ERROR) { 904 if (chainCreated) { 905 removeEffectChain_l(chain); 906 } 907 return status; 908 } 909 910 effect->setDevice(mOutDevice); 911 effect->setDevice(mInDevice); 912 effect->setMode(mAudioFlinger->getMode()); 913 effect->setAudioSource(mAudioSource); 914 return NO_ERROR; 915} 916 917void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 918 919 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 920 effect_descriptor_t desc = effect->desc(); 921 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 922 detachAuxEffect_l(effect->id()); 923 } 924 925 sp<EffectChain> chain = effect->chain().promote(); 926 if (chain != 0) { 927 // remove effect chain if removing last effect 928 if (chain->removeEffect_l(effect) == 0) { 929 removeEffectChain_l(chain); 930 } 931 } else { 932 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 933 } 934} 935 936void AudioFlinger::ThreadBase::lockEffectChains_l( 937 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 938{ 939 effectChains = mEffectChains; 940 for (size_t i = 0; i < mEffectChains.size(); i++) { 941 mEffectChains[i]->lock(); 942 } 943} 944 945void AudioFlinger::ThreadBase::unlockEffectChains( 946 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 947{ 948 for (size_t i = 0; i < effectChains.size(); i++) { 949 effectChains[i]->unlock(); 950 } 951} 952 953sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 954{ 955 Mutex::Autolock _l(mLock); 956 return getEffectChain_l(sessionId); 957} 958 959sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 960{ 961 size_t size = mEffectChains.size(); 962 for (size_t i = 0; i < size; i++) { 963 if (mEffectChains[i]->sessionId() == sessionId) { 964 return mEffectChains[i]; 965 } 966 } 967 return 0; 968} 969 970void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 971{ 972 Mutex::Autolock _l(mLock); 973 size_t size = mEffectChains.size(); 974 for (size_t i = 0; i < size; i++) { 975 mEffectChains[i]->setMode_l(mode); 976 } 977} 978 979void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 980 EffectHandle *handle, 981 bool unpinIfLast) { 982 983 Mutex::Autolock _l(mLock); 984 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 985 // delete the effect module if removing last handle on it 986 if (effect->removeHandle(handle) == 0) { 987 if (!effect->isPinned() || unpinIfLast) { 988 removeEffect_l(effect); 989 AudioSystem::unregisterEffect(effect->id()); 990 } 991 } 992} 993 994// ---------------------------------------------------------------------------- 995// Playback 996// ---------------------------------------------------------------------------- 997 998AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 999 AudioStreamOut* output, 1000 audio_io_handle_t id, 1001 audio_devices_t device, 1002 type_t type) 1003 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1004 mNormalFrameCount(0), mMixBuffer(NULL), 1005 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1006 mActiveTracksGeneration(0), 1007 // mStreamTypes[] initialized in constructor body 1008 mOutput(output), 1009 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1010 mMixerStatus(MIXER_IDLE), 1011 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1012 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1013 mBytesRemaining(0), 1014 mCurrentWriteLength(0), 1015 mUseAsyncWrite(false), 1016 mWriteAckSequence(0), 1017 mDrainSequence(0), 1018 mSignalPending(false), 1019 mScreenState(AudioFlinger::mScreenState), 1020 // index 0 is reserved for normal mixer's submix 1021 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1022 // mLatchD, mLatchQ, 1023 mLatchDValid(false), mLatchQValid(false) 1024{ 1025 snprintf(mName, kNameLength, "AudioOut_%X", id); 1026 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1027 1028 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1029 // it would be safer to explicitly pass initial masterVolume/masterMute as 1030 // parameter. 1031 // 1032 // If the HAL we are using has support for master volume or master mute, 1033 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1034 // and the mute set to false). 1035 mMasterVolume = audioFlinger->masterVolume_l(); 1036 mMasterMute = audioFlinger->masterMute_l(); 1037 if (mOutput && mOutput->audioHwDev) { 1038 if (mOutput->audioHwDev->canSetMasterVolume()) { 1039 mMasterVolume = 1.0; 1040 } 1041 1042 if (mOutput->audioHwDev->canSetMasterMute()) { 1043 mMasterMute = false; 1044 } 1045 } 1046 1047 readOutputParameters(); 1048 1049 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1050 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1051 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1052 stream = (audio_stream_type_t) (stream + 1)) { 1053 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1054 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1055 } 1056 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1057 // because mAudioFlinger doesn't have one to copy from 1058} 1059 1060AudioFlinger::PlaybackThread::~PlaybackThread() 1061{ 1062 mAudioFlinger->unregisterWriter(mNBLogWriter); 1063 delete [] mAllocMixBuffer; 1064} 1065 1066void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1067{ 1068 dumpInternals(fd, args); 1069 dumpTracks(fd, args); 1070 dumpEffectChains(fd, args); 1071} 1072 1073void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1074{ 1075 const size_t SIZE = 256; 1076 char buffer[SIZE]; 1077 String8 result; 1078 1079 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1080 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1081 const stream_type_t *st = &mStreamTypes[i]; 1082 if (i > 0) { 1083 result.appendFormat(", "); 1084 } 1085 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1086 if (st->mute) { 1087 result.append("M"); 1088 } 1089 } 1090 result.append("\n"); 1091 write(fd, result.string(), result.length()); 1092 result.clear(); 1093 1094 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1095 result.append(buffer); 1096 Track::appendDumpHeader(result); 1097 for (size_t i = 0; i < mTracks.size(); ++i) { 1098 sp<Track> track = mTracks[i]; 1099 if (track != 0) { 1100 track->dump(buffer, SIZE); 1101 result.append(buffer); 1102 } 1103 } 1104 1105 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1106 result.append(buffer); 1107 Track::appendDumpHeader(result); 1108 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1109 sp<Track> track = mActiveTracks[i].promote(); 1110 if (track != 0) { 1111 track->dump(buffer, SIZE); 1112 result.append(buffer); 1113 } 1114 } 1115 write(fd, result.string(), result.size()); 1116 1117 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1118 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1119 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1120 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1121} 1122 1123void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1124{ 1125 const size_t SIZE = 256; 1126 char buffer[SIZE]; 1127 String8 result; 1128 1129 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1130 result.append(buffer); 1131 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1132 result.append(buffer); 1133 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1134 ns2ms(systemTime() - mLastWriteTime)); 1135 result.append(buffer); 1136 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1137 result.append(buffer); 1138 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1139 result.append(buffer); 1140 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1141 result.append(buffer); 1142 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1143 result.append(buffer); 1144 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1145 result.append(buffer); 1146 write(fd, result.string(), result.size()); 1147 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1148 1149 dumpBase(fd, args); 1150} 1151 1152// Thread virtuals 1153status_t AudioFlinger::PlaybackThread::readyToRun() 1154{ 1155 status_t status = initCheck(); 1156 if (status == NO_ERROR) { 1157 ALOGI("AudioFlinger's thread %p ready to run", this); 1158 } else { 1159 ALOGE("No working audio driver found."); 1160 } 1161 return status; 1162} 1163 1164void AudioFlinger::PlaybackThread::onFirstRef() 1165{ 1166 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1167} 1168 1169// ThreadBase virtuals 1170void AudioFlinger::PlaybackThread::preExit() 1171{ 1172 ALOGV(" preExit()"); 1173 // FIXME this is using hard-coded strings but in the future, this functionality will be 1174 // converted to use audio HAL extensions required to support tunneling 1175 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1176} 1177 1178// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1179sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1180 const sp<AudioFlinger::Client>& client, 1181 audio_stream_type_t streamType, 1182 uint32_t sampleRate, 1183 audio_format_t format, 1184 audio_channel_mask_t channelMask, 1185 size_t frameCount, 1186 const sp<IMemory>& sharedBuffer, 1187 int sessionId, 1188 IAudioFlinger::track_flags_t *flags, 1189 pid_t tid, 1190 int uid, 1191 status_t *status) 1192{ 1193 sp<Track> track; 1194 status_t lStatus; 1195 1196 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1197 1198 // client expresses a preference for FAST, but we get the final say 1199 if (*flags & IAudioFlinger::TRACK_FAST) { 1200 if ( 1201 // not timed 1202 (!isTimed) && 1203 // either of these use cases: 1204 ( 1205 // use case 1: shared buffer with any frame count 1206 ( 1207 (sharedBuffer != 0) 1208 ) || 1209 // use case 2: callback handler and frame count is default or at least as large as HAL 1210 ( 1211 (tid != -1) && 1212 ((frameCount == 0) || 1213 (frameCount >= mFrameCount)) 1214 ) 1215 ) && 1216 // PCM data 1217 audio_is_linear_pcm(format) && 1218 // mono or stereo 1219 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1220 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1221 // hardware sample rate 1222 (sampleRate == mSampleRate) && 1223 // normal mixer has an associated fast mixer 1224 hasFastMixer() && 1225 // there are sufficient fast track slots available 1226 (mFastTrackAvailMask != 0) 1227 // FIXME test that MixerThread for this fast track has a capable output HAL 1228 // FIXME add a permission test also? 1229 ) { 1230 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1231 if (frameCount == 0) { 1232 frameCount = mFrameCount * kFastTrackMultiplier; 1233 } 1234 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1235 frameCount, mFrameCount); 1236 } else { 1237 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1238 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1239 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1240 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1241 audio_is_linear_pcm(format), 1242 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1243 *flags &= ~IAudioFlinger::TRACK_FAST; 1244 // For compatibility with AudioTrack calculation, buffer depth is forced 1245 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1246 // This is probably too conservative, but legacy application code may depend on it. 1247 // If you change this calculation, also review the start threshold which is related. 1248 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1249 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1250 if (minBufCount < 2) { 1251 minBufCount = 2; 1252 } 1253 size_t minFrameCount = mNormalFrameCount * minBufCount; 1254 if (frameCount < minFrameCount) { 1255 frameCount = minFrameCount; 1256 } 1257 } 1258 } 1259 1260 if (mType == DIRECT) { 1261 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1262 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1263 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1264 "for output %p with format %d", 1265 sampleRate, format, channelMask, mOutput, mFormat); 1266 lStatus = BAD_VALUE; 1267 goto Exit; 1268 } 1269 } 1270 } else if (mType == OFFLOAD) { 1271 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1272 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1273 "for output %p with format %d", 1274 sampleRate, format, channelMask, mOutput, mFormat); 1275 lStatus = BAD_VALUE; 1276 goto Exit; 1277 } 1278 } else { 1279 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1280 ALOGE("createTrack_l() Bad parameter: format %d \"" 1281 "for output %p with format %d", 1282 format, mOutput, mFormat); 1283 lStatus = BAD_VALUE; 1284 goto Exit; 1285 } 1286 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1287 if (sampleRate > mSampleRate*2) { 1288 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1289 lStatus = BAD_VALUE; 1290 goto Exit; 1291 } 1292 } 1293 1294 lStatus = initCheck(); 1295 if (lStatus != NO_ERROR) { 1296 ALOGE("Audio driver not initialized."); 1297 goto Exit; 1298 } 1299 1300 { // scope for mLock 1301 Mutex::Autolock _l(mLock); 1302 1303 // all tracks in same audio session must share the same routing strategy otherwise 1304 // conflicts will happen when tracks are moved from one output to another by audio policy 1305 // manager 1306 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1307 for (size_t i = 0; i < mTracks.size(); ++i) { 1308 sp<Track> t = mTracks[i]; 1309 if (t != 0 && !t->isOutputTrack()) { 1310 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1311 if (sessionId == t->sessionId() && strategy != actual) { 1312 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1313 strategy, actual); 1314 lStatus = BAD_VALUE; 1315 goto Exit; 1316 } 1317 } 1318 } 1319 1320 if (!isTimed) { 1321 track = new Track(this, client, streamType, sampleRate, format, 1322 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1323 } else { 1324 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1325 channelMask, frameCount, sharedBuffer, sessionId, uid); 1326 } 1327 1328 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1329 lStatus = NO_MEMORY; 1330 // track must be cleared from the caller as the caller has the AF lock 1331 goto Exit; 1332 } 1333 1334 mTracks.add(track); 1335 1336 sp<EffectChain> chain = getEffectChain_l(sessionId); 1337 if (chain != 0) { 1338 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1339 track->setMainBuffer(chain->inBuffer()); 1340 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1341 chain->incTrackCnt(); 1342 } 1343 1344 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1345 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1346 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1347 // so ask activity manager to do this on our behalf 1348 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1349 } 1350 } 1351 1352 lStatus = NO_ERROR; 1353 1354Exit: 1355 if (status) { 1356 *status = lStatus; 1357 } 1358 return track; 1359} 1360 1361uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1362{ 1363 return latency; 1364} 1365 1366uint32_t AudioFlinger::PlaybackThread::latency() const 1367{ 1368 Mutex::Autolock _l(mLock); 1369 return latency_l(); 1370} 1371uint32_t AudioFlinger::PlaybackThread::latency_l() const 1372{ 1373 if (initCheck() == NO_ERROR) { 1374 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1375 } else { 1376 return 0; 1377 } 1378} 1379 1380void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1381{ 1382 Mutex::Autolock _l(mLock); 1383 // Don't apply master volume in SW if our HAL can do it for us. 1384 if (mOutput && mOutput->audioHwDev && 1385 mOutput->audioHwDev->canSetMasterVolume()) { 1386 mMasterVolume = 1.0; 1387 } else { 1388 mMasterVolume = value; 1389 } 1390} 1391 1392void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1393{ 1394 Mutex::Autolock _l(mLock); 1395 // Don't apply master mute in SW if our HAL can do it for us. 1396 if (mOutput && mOutput->audioHwDev && 1397 mOutput->audioHwDev->canSetMasterMute()) { 1398 mMasterMute = false; 1399 } else { 1400 mMasterMute = muted; 1401 } 1402} 1403 1404void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1405{ 1406 Mutex::Autolock _l(mLock); 1407 mStreamTypes[stream].volume = value; 1408 broadcast_l(); 1409} 1410 1411void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1412{ 1413 Mutex::Autolock _l(mLock); 1414 mStreamTypes[stream].mute = muted; 1415 broadcast_l(); 1416} 1417 1418float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1419{ 1420 Mutex::Autolock _l(mLock); 1421 return mStreamTypes[stream].volume; 1422} 1423 1424// addTrack_l() must be called with ThreadBase::mLock held 1425status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1426{ 1427 status_t status = ALREADY_EXISTS; 1428 1429 // set retry count for buffer fill 1430 track->mRetryCount = kMaxTrackStartupRetries; 1431 if (mActiveTracks.indexOf(track) < 0) { 1432 // the track is newly added, make sure it fills up all its 1433 // buffers before playing. This is to ensure the client will 1434 // effectively get the latency it requested. 1435 if (!track->isOutputTrack()) { 1436 TrackBase::track_state state = track->mState; 1437 mLock.unlock(); 1438 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1439 mLock.lock(); 1440 // abort track was stopped/paused while we released the lock 1441 if (state != track->mState) { 1442 if (status == NO_ERROR) { 1443 mLock.unlock(); 1444 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1445 mLock.lock(); 1446 } 1447 return INVALID_OPERATION; 1448 } 1449 // abort if start is rejected by audio policy manager 1450 if (status != NO_ERROR) { 1451 return PERMISSION_DENIED; 1452 } 1453#ifdef ADD_BATTERY_DATA 1454 // to track the speaker usage 1455 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1456#endif 1457 } 1458 1459 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1460 track->mResetDone = false; 1461 track->mPresentationCompleteFrames = 0; 1462 mActiveTracks.add(track); 1463 mWakeLockUids.add(track->uid()); 1464 mActiveTracksGeneration++; 1465 mLatestActiveTrack = track; 1466 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1467 if (chain != 0) { 1468 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1469 track->sessionId()); 1470 chain->incActiveTrackCnt(); 1471 } 1472 1473 status = NO_ERROR; 1474 } 1475 1476 ALOGV("signal playback thread"); 1477 broadcast_l(); 1478 1479 return status; 1480} 1481 1482bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1483{ 1484 track->terminate(); 1485 // active tracks are removed by threadLoop() 1486 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1487 track->mState = TrackBase::STOPPED; 1488 if (!trackActive) { 1489 removeTrack_l(track); 1490 } else if (track->isFastTrack() || track->isOffloaded()) { 1491 track->mState = TrackBase::STOPPING_1; 1492 } 1493 1494 return trackActive; 1495} 1496 1497void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1498{ 1499 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1500 mTracks.remove(track); 1501 deleteTrackName_l(track->name()); 1502 // redundant as track is about to be destroyed, for dumpsys only 1503 track->mName = -1; 1504 if (track->isFastTrack()) { 1505 int index = track->mFastIndex; 1506 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1507 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1508 mFastTrackAvailMask |= 1 << index; 1509 // redundant as track is about to be destroyed, for dumpsys only 1510 track->mFastIndex = -1; 1511 } 1512 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1513 if (chain != 0) { 1514 chain->decTrackCnt(); 1515 } 1516} 1517 1518void AudioFlinger::PlaybackThread::broadcast_l() 1519{ 1520 // Thread could be blocked waiting for async 1521 // so signal it to handle state changes immediately 1522 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1523 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1524 mSignalPending = true; 1525 mWaitWorkCV.broadcast(); 1526} 1527 1528String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1529{ 1530 Mutex::Autolock _l(mLock); 1531 if (initCheck() != NO_ERROR) { 1532 return String8(); 1533 } 1534 1535 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1536 const String8 out_s8(s); 1537 free(s); 1538 return out_s8; 1539} 1540 1541// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1542void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1543 AudioSystem::OutputDescriptor desc; 1544 void *param2 = NULL; 1545 1546 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1547 param); 1548 1549 switch (event) { 1550 case AudioSystem::OUTPUT_OPENED: 1551 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1552 desc.channelMask = mChannelMask; 1553 desc.samplingRate = mSampleRate; 1554 desc.format = mFormat; 1555 desc.frameCount = mNormalFrameCount; // FIXME see 1556 // AudioFlinger::frameCount(audio_io_handle_t) 1557 desc.latency = latency(); 1558 param2 = &desc; 1559 break; 1560 1561 case AudioSystem::STREAM_CONFIG_CHANGED: 1562 param2 = ¶m; 1563 case AudioSystem::OUTPUT_CLOSED: 1564 default: 1565 break; 1566 } 1567 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1568} 1569 1570void AudioFlinger::PlaybackThread::writeCallback() 1571{ 1572 ALOG_ASSERT(mCallbackThread != 0); 1573 mCallbackThread->resetWriteBlocked(); 1574} 1575 1576void AudioFlinger::PlaybackThread::drainCallback() 1577{ 1578 ALOG_ASSERT(mCallbackThread != 0); 1579 mCallbackThread->resetDraining(); 1580} 1581 1582void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1583{ 1584 Mutex::Autolock _l(mLock); 1585 // reject out of sequence requests 1586 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1587 mWriteAckSequence &= ~1; 1588 mWaitWorkCV.signal(); 1589 } 1590} 1591 1592void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1593{ 1594 Mutex::Autolock _l(mLock); 1595 // reject out of sequence requests 1596 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1597 mDrainSequence &= ~1; 1598 mWaitWorkCV.signal(); 1599 } 1600} 1601 1602// static 1603int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1604 void *param, 1605 void *cookie) 1606{ 1607 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1608 ALOGV("asyncCallback() event %d", event); 1609 switch (event) { 1610 case STREAM_CBK_EVENT_WRITE_READY: 1611 me->writeCallback(); 1612 break; 1613 case STREAM_CBK_EVENT_DRAIN_READY: 1614 me->drainCallback(); 1615 break; 1616 default: 1617 ALOGW("asyncCallback() unknown event %d", event); 1618 break; 1619 } 1620 return 0; 1621} 1622 1623void AudioFlinger::PlaybackThread::readOutputParameters() 1624{ 1625 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1626 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1627 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1628 if (!audio_is_output_channel(mChannelMask)) { 1629 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1630 } 1631 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1632 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1633 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1634 } 1635 mChannelCount = popcount(mChannelMask); 1636 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1637 if (!audio_is_valid_format(mFormat)) { 1638 LOG_FATAL("HAL format %d not valid for output", mFormat); 1639 } 1640 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1641 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1642 mFormat); 1643 } 1644 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1645 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1646 if (mFrameCount & 15) { 1647 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1648 mFrameCount); 1649 } 1650 1651 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1652 (mOutput->stream->set_callback != NULL)) { 1653 if (mOutput->stream->set_callback(mOutput->stream, 1654 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1655 mUseAsyncWrite = true; 1656 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1657 } 1658 } 1659 1660 // Calculate size of normal mix buffer relative to the HAL output buffer size 1661 double multiplier = 1.0; 1662 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1663 kUseFastMixer == FastMixer_Dynamic)) { 1664 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1665 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1666 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1667 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1668 maxNormalFrameCount = maxNormalFrameCount & ~15; 1669 if (maxNormalFrameCount < minNormalFrameCount) { 1670 maxNormalFrameCount = minNormalFrameCount; 1671 } 1672 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1673 if (multiplier <= 1.0) { 1674 multiplier = 1.0; 1675 } else if (multiplier <= 2.0) { 1676 if (2 * mFrameCount <= maxNormalFrameCount) { 1677 multiplier = 2.0; 1678 } else { 1679 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1680 } 1681 } else { 1682 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1683 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1684 // track, but we sometimes have to do this to satisfy the maximum frame count 1685 // constraint) 1686 // FIXME this rounding up should not be done if no HAL SRC 1687 uint32_t truncMult = (uint32_t) multiplier; 1688 if ((truncMult & 1)) { 1689 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1690 ++truncMult; 1691 } 1692 } 1693 multiplier = (double) truncMult; 1694 } 1695 } 1696 mNormalFrameCount = multiplier * mFrameCount; 1697 // round up to nearest 16 frames to satisfy AudioMixer 1698 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1699 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1700 mNormalFrameCount); 1701 1702 delete[] mAllocMixBuffer; 1703 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1704 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1705 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1706 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1707 1708 // force reconfiguration of effect chains and engines to take new buffer size and audio 1709 // parameters into account 1710 // Note that mLock is not held when readOutputParameters() is called from the constructor 1711 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1712 // matter. 1713 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1714 Vector< sp<EffectChain> > effectChains = mEffectChains; 1715 for (size_t i = 0; i < effectChains.size(); i ++) { 1716 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1717 } 1718} 1719 1720 1721status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1722{ 1723 if (halFrames == NULL || dspFrames == NULL) { 1724 return BAD_VALUE; 1725 } 1726 Mutex::Autolock _l(mLock); 1727 if (initCheck() != NO_ERROR) { 1728 return INVALID_OPERATION; 1729 } 1730 size_t framesWritten = mBytesWritten / mFrameSize; 1731 *halFrames = framesWritten; 1732 1733 if (isSuspended()) { 1734 // return an estimation of rendered frames when the output is suspended 1735 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1736 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1737 return NO_ERROR; 1738 } else { 1739 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1740 } 1741} 1742 1743uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1744{ 1745 Mutex::Autolock _l(mLock); 1746 uint32_t result = 0; 1747 if (getEffectChain_l(sessionId) != 0) { 1748 result = EFFECT_SESSION; 1749 } 1750 1751 for (size_t i = 0; i < mTracks.size(); ++i) { 1752 sp<Track> track = mTracks[i]; 1753 if (sessionId == track->sessionId() && !track->isInvalid()) { 1754 result |= TRACK_SESSION; 1755 break; 1756 } 1757 } 1758 1759 return result; 1760} 1761 1762uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1763{ 1764 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1765 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1766 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1767 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1768 } 1769 for (size_t i = 0; i < mTracks.size(); i++) { 1770 sp<Track> track = mTracks[i]; 1771 if (sessionId == track->sessionId() && !track->isInvalid()) { 1772 return AudioSystem::getStrategyForStream(track->streamType()); 1773 } 1774 } 1775 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1776} 1777 1778 1779AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1780{ 1781 Mutex::Autolock _l(mLock); 1782 return mOutput; 1783} 1784 1785AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1786{ 1787 Mutex::Autolock _l(mLock); 1788 AudioStreamOut *output = mOutput; 1789 mOutput = NULL; 1790 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1791 // must push a NULL and wait for ack 1792 mOutputSink.clear(); 1793 mPipeSink.clear(); 1794 mNormalSink.clear(); 1795 return output; 1796} 1797 1798// this method must always be called either with ThreadBase mLock held or inside the thread loop 1799audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1800{ 1801 if (mOutput == NULL) { 1802 return NULL; 1803 } 1804 return &mOutput->stream->common; 1805} 1806 1807uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1808{ 1809 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1810} 1811 1812status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1813{ 1814 if (!isValidSyncEvent(event)) { 1815 return BAD_VALUE; 1816 } 1817 1818 Mutex::Autolock _l(mLock); 1819 1820 for (size_t i = 0; i < mTracks.size(); ++i) { 1821 sp<Track> track = mTracks[i]; 1822 if (event->triggerSession() == track->sessionId()) { 1823 (void) track->setSyncEvent(event); 1824 return NO_ERROR; 1825 } 1826 } 1827 1828 return NAME_NOT_FOUND; 1829} 1830 1831bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1832{ 1833 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1834} 1835 1836void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1837 const Vector< sp<Track> >& tracksToRemove) 1838{ 1839 size_t count = tracksToRemove.size(); 1840 if (count) { 1841 for (size_t i = 0 ; i < count ; i++) { 1842 const sp<Track>& track = tracksToRemove.itemAt(i); 1843 if (!track->isOutputTrack()) { 1844 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1845#ifdef ADD_BATTERY_DATA 1846 // to track the speaker usage 1847 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1848#endif 1849 if (track->isTerminated()) { 1850 AudioSystem::releaseOutput(mId); 1851 } 1852 } 1853 } 1854 } 1855} 1856 1857void AudioFlinger::PlaybackThread::checkSilentMode_l() 1858{ 1859 if (!mMasterMute) { 1860 char value[PROPERTY_VALUE_MAX]; 1861 if (property_get("ro.audio.silent", value, "0") > 0) { 1862 char *endptr; 1863 unsigned long ul = strtoul(value, &endptr, 0); 1864 if (*endptr == '\0' && ul != 0) { 1865 ALOGD("Silence is golden"); 1866 // The setprop command will not allow a property to be changed after 1867 // the first time it is set, so we don't have to worry about un-muting. 1868 setMasterMute_l(true); 1869 } 1870 } 1871 } 1872} 1873 1874// shared by MIXER and DIRECT, overridden by DUPLICATING 1875ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1876{ 1877 // FIXME rewrite to reduce number of system calls 1878 mLastWriteTime = systemTime(); 1879 mInWrite = true; 1880 ssize_t bytesWritten; 1881 1882 // If an NBAIO sink is present, use it to write the normal mixer's submix 1883 if (mNormalSink != 0) { 1884#define mBitShift 2 // FIXME 1885 size_t count = mBytesRemaining >> mBitShift; 1886 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1887 ATRACE_BEGIN("write"); 1888 // update the setpoint when AudioFlinger::mScreenState changes 1889 uint32_t screenState = AudioFlinger::mScreenState; 1890 if (screenState != mScreenState) { 1891 mScreenState = screenState; 1892 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1893 if (pipe != NULL) { 1894 pipe->setAvgFrames((mScreenState & 1) ? 1895 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1896 } 1897 } 1898 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1899 ATRACE_END(); 1900 if (framesWritten > 0) { 1901 bytesWritten = framesWritten << mBitShift; 1902 } else { 1903 bytesWritten = framesWritten; 1904 } 1905 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1906 if (status == NO_ERROR) { 1907 size_t totalFramesWritten = mNormalSink->framesWritten(); 1908 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1909 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1910 mLatchDValid = true; 1911 } 1912 } 1913 // otherwise use the HAL / AudioStreamOut directly 1914 } else { 1915 // Direct output and offload threads 1916 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1917 if (mUseAsyncWrite) { 1918 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1919 mWriteAckSequence += 2; 1920 mWriteAckSequence |= 1; 1921 ALOG_ASSERT(mCallbackThread != 0); 1922 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1923 } 1924 // FIXME We should have an implementation of timestamps for direct output threads. 1925 // They are used e.g for multichannel PCM playback over HDMI. 1926 bytesWritten = mOutput->stream->write(mOutput->stream, 1927 (char *)mMixBuffer + offset, mBytesRemaining); 1928 if (mUseAsyncWrite && 1929 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1930 // do not wait for async callback in case of error of full write 1931 mWriteAckSequence &= ~1; 1932 ALOG_ASSERT(mCallbackThread != 0); 1933 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1934 } 1935 } 1936 1937 mNumWrites++; 1938 mInWrite = false; 1939 mStandby = false; 1940 return bytesWritten; 1941} 1942 1943void AudioFlinger::PlaybackThread::threadLoop_drain() 1944{ 1945 if (mOutput->stream->drain) { 1946 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1947 if (mUseAsyncWrite) { 1948 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1949 mDrainSequence |= 1; 1950 ALOG_ASSERT(mCallbackThread != 0); 1951 mCallbackThread->setDraining(mDrainSequence); 1952 } 1953 mOutput->stream->drain(mOutput->stream, 1954 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1955 : AUDIO_DRAIN_ALL); 1956 } 1957} 1958 1959void AudioFlinger::PlaybackThread::threadLoop_exit() 1960{ 1961 // Default implementation has nothing to do 1962} 1963 1964/* 1965The derived values that are cached: 1966 - mixBufferSize from frame count * frame size 1967 - activeSleepTime from activeSleepTimeUs() 1968 - idleSleepTime from idleSleepTimeUs() 1969 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1970 - maxPeriod from frame count and sample rate (MIXER only) 1971 1972The parameters that affect these derived values are: 1973 - frame count 1974 - frame size 1975 - sample rate 1976 - device type: A2DP or not 1977 - device latency 1978 - format: PCM or not 1979 - active sleep time 1980 - idle sleep time 1981*/ 1982 1983void AudioFlinger::PlaybackThread::cacheParameters_l() 1984{ 1985 mixBufferSize = mNormalFrameCount * mFrameSize; 1986 activeSleepTime = activeSleepTimeUs(); 1987 idleSleepTime = idleSleepTimeUs(); 1988} 1989 1990void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1991{ 1992 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1993 this, streamType, mTracks.size()); 1994 Mutex::Autolock _l(mLock); 1995 1996 size_t size = mTracks.size(); 1997 for (size_t i = 0; i < size; i++) { 1998 sp<Track> t = mTracks[i]; 1999 if (t->streamType() == streamType) { 2000 t->invalidate(); 2001 } 2002 } 2003} 2004 2005status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2006{ 2007 int session = chain->sessionId(); 2008 int16_t *buffer = mMixBuffer; 2009 bool ownsBuffer = false; 2010 2011 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2012 if (session > 0) { 2013 // Only one effect chain can be present in direct output thread and it uses 2014 // the mix buffer as input 2015 if (mType != DIRECT) { 2016 size_t numSamples = mNormalFrameCount * mChannelCount; 2017 buffer = new int16_t[numSamples]; 2018 memset(buffer, 0, numSamples * sizeof(int16_t)); 2019 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2020 ownsBuffer = true; 2021 } 2022 2023 // Attach all tracks with same session ID to this chain. 2024 for (size_t i = 0; i < mTracks.size(); ++i) { 2025 sp<Track> track = mTracks[i]; 2026 if (session == track->sessionId()) { 2027 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2028 buffer); 2029 track->setMainBuffer(buffer); 2030 chain->incTrackCnt(); 2031 } 2032 } 2033 2034 // indicate all active tracks in the chain 2035 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2036 sp<Track> track = mActiveTracks[i].promote(); 2037 if (track == 0) { 2038 continue; 2039 } 2040 if (session == track->sessionId()) { 2041 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2042 chain->incActiveTrackCnt(); 2043 } 2044 } 2045 } 2046 2047 chain->setInBuffer(buffer, ownsBuffer); 2048 chain->setOutBuffer(mMixBuffer); 2049 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2050 // chains list in order to be processed last as it contains output stage effects 2051 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2052 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2053 // after track specific effects and before output stage 2054 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2055 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2056 // Effect chain for other sessions are inserted at beginning of effect 2057 // chains list to be processed before output mix effects. Relative order between other 2058 // sessions is not important 2059 size_t size = mEffectChains.size(); 2060 size_t i = 0; 2061 for (i = 0; i < size; i++) { 2062 if (mEffectChains[i]->sessionId() < session) { 2063 break; 2064 } 2065 } 2066 mEffectChains.insertAt(chain, i); 2067 checkSuspendOnAddEffectChain_l(chain); 2068 2069 return NO_ERROR; 2070} 2071 2072size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2073{ 2074 int session = chain->sessionId(); 2075 2076 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2077 2078 for (size_t i = 0; i < mEffectChains.size(); i++) { 2079 if (chain == mEffectChains[i]) { 2080 mEffectChains.removeAt(i); 2081 // detach all active tracks from the chain 2082 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2083 sp<Track> track = mActiveTracks[i].promote(); 2084 if (track == 0) { 2085 continue; 2086 } 2087 if (session == track->sessionId()) { 2088 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2089 chain.get(), session); 2090 chain->decActiveTrackCnt(); 2091 } 2092 } 2093 2094 // detach all tracks with same session ID from this chain 2095 for (size_t i = 0; i < mTracks.size(); ++i) { 2096 sp<Track> track = mTracks[i]; 2097 if (session == track->sessionId()) { 2098 track->setMainBuffer(mMixBuffer); 2099 chain->decTrackCnt(); 2100 } 2101 } 2102 break; 2103 } 2104 } 2105 return mEffectChains.size(); 2106} 2107 2108status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2109 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2110{ 2111 Mutex::Autolock _l(mLock); 2112 return attachAuxEffect_l(track, EffectId); 2113} 2114 2115status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2116 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2117{ 2118 status_t status = NO_ERROR; 2119 2120 if (EffectId == 0) { 2121 track->setAuxBuffer(0, NULL); 2122 } else { 2123 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2124 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2125 if (effect != 0) { 2126 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2127 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2128 } else { 2129 status = INVALID_OPERATION; 2130 } 2131 } else { 2132 status = BAD_VALUE; 2133 } 2134 } 2135 return status; 2136} 2137 2138void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2139{ 2140 for (size_t i = 0; i < mTracks.size(); ++i) { 2141 sp<Track> track = mTracks[i]; 2142 if (track->auxEffectId() == effectId) { 2143 attachAuxEffect_l(track, 0); 2144 } 2145 } 2146} 2147 2148bool AudioFlinger::PlaybackThread::threadLoop() 2149{ 2150 Vector< sp<Track> > tracksToRemove; 2151 2152 standbyTime = systemTime(); 2153 2154 // MIXER 2155 nsecs_t lastWarning = 0; 2156 2157 // DUPLICATING 2158 // FIXME could this be made local to while loop? 2159 writeFrames = 0; 2160 2161 int lastGeneration = 0; 2162 2163 cacheParameters_l(); 2164 sleepTime = idleSleepTime; 2165 2166 if (mType == MIXER) { 2167 sleepTimeShift = 0; 2168 } 2169 2170 CpuStats cpuStats; 2171 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2172 2173 acquireWakeLock(); 2174 2175 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2176 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2177 // and then that string will be logged at the next convenient opportunity. 2178 const char *logString = NULL; 2179 2180 checkSilentMode_l(); 2181 2182 while (!exitPending()) 2183 { 2184 cpuStats.sample(myName); 2185 2186 Vector< sp<EffectChain> > effectChains; 2187 2188 processConfigEvents(); 2189 2190 { // scope for mLock 2191 2192 Mutex::Autolock _l(mLock); 2193 2194 if (logString != NULL) { 2195 mNBLogWriter->logTimestamp(); 2196 mNBLogWriter->log(logString); 2197 logString = NULL; 2198 } 2199 2200 if (mLatchDValid) { 2201 mLatchQ = mLatchD; 2202 mLatchDValid = false; 2203 mLatchQValid = true; 2204 } 2205 2206 if (checkForNewParameters_l()) { 2207 cacheParameters_l(); 2208 } 2209 2210 saveOutputTracks(); 2211 if (mSignalPending) { 2212 // A signal was raised while we were unlocked 2213 mSignalPending = false; 2214 } else if (waitingAsyncCallback_l()) { 2215 if (exitPending()) { 2216 break; 2217 } 2218 releaseWakeLock_l(); 2219 mWakeLockUids.clear(); 2220 mActiveTracksGeneration++; 2221 ALOGV("wait async completion"); 2222 mWaitWorkCV.wait(mLock); 2223 ALOGV("async completion/wake"); 2224 acquireWakeLock_l(); 2225 standbyTime = systemTime() + standbyDelay; 2226 sleepTime = 0; 2227 2228 continue; 2229 } 2230 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2231 isSuspended()) { 2232 // put audio hardware into standby after short delay 2233 if (shouldStandby_l()) { 2234 2235 threadLoop_standby(); 2236 2237 mStandby = true; 2238 } 2239 2240 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2241 // we're about to wait, flush the binder command buffer 2242 IPCThreadState::self()->flushCommands(); 2243 2244 clearOutputTracks(); 2245 2246 if (exitPending()) { 2247 break; 2248 } 2249 2250 releaseWakeLock_l(); 2251 mWakeLockUids.clear(); 2252 mActiveTracksGeneration++; 2253 // wait until we have something to do... 2254 ALOGV("%s going to sleep", myName.string()); 2255 mWaitWorkCV.wait(mLock); 2256 ALOGV("%s waking up", myName.string()); 2257 acquireWakeLock_l(); 2258 2259 mMixerStatus = MIXER_IDLE; 2260 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2261 mBytesWritten = 0; 2262 mBytesRemaining = 0; 2263 checkSilentMode_l(); 2264 2265 standbyTime = systemTime() + standbyDelay; 2266 sleepTime = idleSleepTime; 2267 if (mType == MIXER) { 2268 sleepTimeShift = 0; 2269 } 2270 2271 continue; 2272 } 2273 } 2274 // mMixerStatusIgnoringFastTracks is also updated internally 2275 mMixerStatus = prepareTracks_l(&tracksToRemove); 2276 2277 // compare with previously applied list 2278 if (lastGeneration != mActiveTracksGeneration) { 2279 // update wakelock 2280 updateWakeLockUids_l(mWakeLockUids); 2281 lastGeneration = mActiveTracksGeneration; 2282 } 2283 2284 // prevent any changes in effect chain list and in each effect chain 2285 // during mixing and effect process as the audio buffers could be deleted 2286 // or modified if an effect is created or deleted 2287 lockEffectChains_l(effectChains); 2288 } // mLock scope ends 2289 2290 if (mBytesRemaining == 0) { 2291 mCurrentWriteLength = 0; 2292 if (mMixerStatus == MIXER_TRACKS_READY) { 2293 // threadLoop_mix() sets mCurrentWriteLength 2294 threadLoop_mix(); 2295 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2296 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2297 // threadLoop_sleepTime sets sleepTime to 0 if data 2298 // must be written to HAL 2299 threadLoop_sleepTime(); 2300 if (sleepTime == 0) { 2301 mCurrentWriteLength = mixBufferSize; 2302 } 2303 } 2304 mBytesRemaining = mCurrentWriteLength; 2305 if (isSuspended()) { 2306 sleepTime = suspendSleepTimeUs(); 2307 // simulate write to HAL when suspended 2308 mBytesWritten += mixBufferSize; 2309 mBytesRemaining = 0; 2310 } 2311 2312 // only process effects if we're going to write 2313 if (sleepTime == 0 && mType != OFFLOAD) { 2314 for (size_t i = 0; i < effectChains.size(); i ++) { 2315 effectChains[i]->process_l(); 2316 } 2317 } 2318 } 2319 // Process effect chains for offloaded thread even if no audio 2320 // was read from audio track: process only updates effect state 2321 // and thus does have to be synchronized with audio writes but may have 2322 // to be called while waiting for async write callback 2323 if (mType == OFFLOAD) { 2324 for (size_t i = 0; i < effectChains.size(); i ++) { 2325 effectChains[i]->process_l(); 2326 } 2327 } 2328 2329 // enable changes in effect chain 2330 unlockEffectChains(effectChains); 2331 2332 if (!waitingAsyncCallback()) { 2333 // sleepTime == 0 means we must write to audio hardware 2334 if (sleepTime == 0) { 2335 if (mBytesRemaining) { 2336 ssize_t ret = threadLoop_write(); 2337 if (ret < 0) { 2338 mBytesRemaining = 0; 2339 } else { 2340 mBytesWritten += ret; 2341 mBytesRemaining -= ret; 2342 } 2343 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2344 (mMixerStatus == MIXER_DRAIN_ALL)) { 2345 threadLoop_drain(); 2346 } 2347if (mType == MIXER) { 2348 // write blocked detection 2349 nsecs_t now = systemTime(); 2350 nsecs_t delta = now - mLastWriteTime; 2351 if (!mStandby && delta > maxPeriod) { 2352 mNumDelayedWrites++; 2353 if ((now - lastWarning) > kWarningThrottleNs) { 2354 ATRACE_NAME("underrun"); 2355 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2356 ns2ms(delta), mNumDelayedWrites, this); 2357 lastWarning = now; 2358 } 2359 } 2360} 2361 2362 } else { 2363 usleep(sleepTime); 2364 } 2365 } 2366 2367 // Finally let go of removed track(s), without the lock held 2368 // since we can't guarantee the destructors won't acquire that 2369 // same lock. This will also mutate and push a new fast mixer state. 2370 threadLoop_removeTracks(tracksToRemove); 2371 tracksToRemove.clear(); 2372 2373 // FIXME I don't understand the need for this here; 2374 // it was in the original code but maybe the 2375 // assignment in saveOutputTracks() makes this unnecessary? 2376 clearOutputTracks(); 2377 2378 // Effect chains will be actually deleted here if they were removed from 2379 // mEffectChains list during mixing or effects processing 2380 effectChains.clear(); 2381 2382 // FIXME Note that the above .clear() is no longer necessary since effectChains 2383 // is now local to this block, but will keep it for now (at least until merge done). 2384 } 2385 2386 threadLoop_exit(); 2387 2388 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2389 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2390 // put output stream into standby mode 2391 if (!mStandby) { 2392 mOutput->stream->common.standby(&mOutput->stream->common); 2393 } 2394 } 2395 2396 releaseWakeLock(); 2397 mWakeLockUids.clear(); 2398 mActiveTracksGeneration++; 2399 2400 ALOGV("Thread %p type %d exiting", this, mType); 2401 return false; 2402} 2403 2404// removeTracks_l() must be called with ThreadBase::mLock held 2405void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2406{ 2407 size_t count = tracksToRemove.size(); 2408 if (count) { 2409 for (size_t i=0 ; i<count ; i++) { 2410 const sp<Track>& track = tracksToRemove.itemAt(i); 2411 mActiveTracks.remove(track); 2412 mWakeLockUids.remove(track->uid()); 2413 mActiveTracksGeneration++; 2414 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2415 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2416 if (chain != 0) { 2417 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2418 track->sessionId()); 2419 chain->decActiveTrackCnt(); 2420 } 2421 if (track->isTerminated()) { 2422 removeTrack_l(track); 2423 } 2424 } 2425 } 2426 2427} 2428 2429status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2430{ 2431 if (mNormalSink != 0) { 2432 return mNormalSink->getTimestamp(timestamp); 2433 } 2434 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2435 uint64_t position64; 2436 int ret = mOutput->stream->get_presentation_position( 2437 mOutput->stream, &position64, ×tamp.mTime); 2438 if (ret == 0) { 2439 timestamp.mPosition = (uint32_t)position64; 2440 return NO_ERROR; 2441 } 2442 } 2443 return INVALID_OPERATION; 2444} 2445// ---------------------------------------------------------------------------- 2446 2447AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2448 audio_io_handle_t id, audio_devices_t device, type_t type) 2449 : PlaybackThread(audioFlinger, output, id, device, type), 2450 // mAudioMixer below 2451 // mFastMixer below 2452 mFastMixerFutex(0) 2453 // mOutputSink below 2454 // mPipeSink below 2455 // mNormalSink below 2456{ 2457 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2458 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2459 "mFrameCount=%d, mNormalFrameCount=%d", 2460 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2461 mNormalFrameCount); 2462 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2463 2464 // FIXME - Current mixer implementation only supports stereo output 2465 if (mChannelCount != FCC_2) { 2466 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2467 } 2468 2469 // create an NBAIO sink for the HAL output stream, and negotiate 2470 mOutputSink = new AudioStreamOutSink(output->stream); 2471 size_t numCounterOffers = 0; 2472 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2473 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2474 ALOG_ASSERT(index == 0); 2475 2476 // initialize fast mixer depending on configuration 2477 bool initFastMixer; 2478 switch (kUseFastMixer) { 2479 case FastMixer_Never: 2480 initFastMixer = false; 2481 break; 2482 case FastMixer_Always: 2483 initFastMixer = true; 2484 break; 2485 case FastMixer_Static: 2486 case FastMixer_Dynamic: 2487 initFastMixer = mFrameCount < mNormalFrameCount; 2488 break; 2489 } 2490 if (initFastMixer) { 2491 2492 // create a MonoPipe to connect our submix to FastMixer 2493 NBAIO_Format format = mOutputSink->format(); 2494 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2495 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2496 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2497 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2498 const NBAIO_Format offers[1] = {format}; 2499 size_t numCounterOffers = 0; 2500 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2501 ALOG_ASSERT(index == 0); 2502 monoPipe->setAvgFrames((mScreenState & 1) ? 2503 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2504 mPipeSink = monoPipe; 2505 2506#ifdef TEE_SINK 2507 if (mTeeSinkOutputEnabled) { 2508 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2509 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2510 numCounterOffers = 0; 2511 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2512 ALOG_ASSERT(index == 0); 2513 mTeeSink = teeSink; 2514 PipeReader *teeSource = new PipeReader(*teeSink); 2515 numCounterOffers = 0; 2516 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2517 ALOG_ASSERT(index == 0); 2518 mTeeSource = teeSource; 2519 } 2520#endif 2521 2522 // create fast mixer and configure it initially with just one fast track for our submix 2523 mFastMixer = new FastMixer(); 2524 FastMixerStateQueue *sq = mFastMixer->sq(); 2525#ifdef STATE_QUEUE_DUMP 2526 sq->setObserverDump(&mStateQueueObserverDump); 2527 sq->setMutatorDump(&mStateQueueMutatorDump); 2528#endif 2529 FastMixerState *state = sq->begin(); 2530 FastTrack *fastTrack = &state->mFastTracks[0]; 2531 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2532 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2533 fastTrack->mVolumeProvider = NULL; 2534 fastTrack->mGeneration++; 2535 state->mFastTracksGen++; 2536 state->mTrackMask = 1; 2537 // fast mixer will use the HAL output sink 2538 state->mOutputSink = mOutputSink.get(); 2539 state->mOutputSinkGen++; 2540 state->mFrameCount = mFrameCount; 2541 state->mCommand = FastMixerState::COLD_IDLE; 2542 // already done in constructor initialization list 2543 //mFastMixerFutex = 0; 2544 state->mColdFutexAddr = &mFastMixerFutex; 2545 state->mColdGen++; 2546 state->mDumpState = &mFastMixerDumpState; 2547#ifdef TEE_SINK 2548 state->mTeeSink = mTeeSink.get(); 2549#endif 2550 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2551 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2552 sq->end(); 2553 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2554 2555 // start the fast mixer 2556 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2557 pid_t tid = mFastMixer->getTid(); 2558 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2559 if (err != 0) { 2560 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2561 kPriorityFastMixer, getpid_cached, tid, err); 2562 } 2563 2564#ifdef AUDIO_WATCHDOG 2565 // create and start the watchdog 2566 mAudioWatchdog = new AudioWatchdog(); 2567 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2568 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2569 tid = mAudioWatchdog->getTid(); 2570 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2571 if (err != 0) { 2572 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2573 kPriorityFastMixer, getpid_cached, tid, err); 2574 } 2575#endif 2576 2577 } else { 2578 mFastMixer = NULL; 2579 } 2580 2581 switch (kUseFastMixer) { 2582 case FastMixer_Never: 2583 case FastMixer_Dynamic: 2584 mNormalSink = mOutputSink; 2585 break; 2586 case FastMixer_Always: 2587 mNormalSink = mPipeSink; 2588 break; 2589 case FastMixer_Static: 2590 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2591 break; 2592 } 2593} 2594 2595AudioFlinger::MixerThread::~MixerThread() 2596{ 2597 if (mFastMixer != NULL) { 2598 FastMixerStateQueue *sq = mFastMixer->sq(); 2599 FastMixerState *state = sq->begin(); 2600 if (state->mCommand == FastMixerState::COLD_IDLE) { 2601 int32_t old = android_atomic_inc(&mFastMixerFutex); 2602 if (old == -1) { 2603 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2604 } 2605 } 2606 state->mCommand = FastMixerState::EXIT; 2607 sq->end(); 2608 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2609 mFastMixer->join(); 2610 // Though the fast mixer thread has exited, it's state queue is still valid. 2611 // We'll use that extract the final state which contains one remaining fast track 2612 // corresponding to our sub-mix. 2613 state = sq->begin(); 2614 ALOG_ASSERT(state->mTrackMask == 1); 2615 FastTrack *fastTrack = &state->mFastTracks[0]; 2616 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2617 delete fastTrack->mBufferProvider; 2618 sq->end(false /*didModify*/); 2619 delete mFastMixer; 2620#ifdef AUDIO_WATCHDOG 2621 if (mAudioWatchdog != 0) { 2622 mAudioWatchdog->requestExit(); 2623 mAudioWatchdog->requestExitAndWait(); 2624 mAudioWatchdog.clear(); 2625 } 2626#endif 2627 } 2628 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2629 delete mAudioMixer; 2630} 2631 2632 2633uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2634{ 2635 if (mFastMixer != NULL) { 2636 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2637 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2638 } 2639 return latency; 2640} 2641 2642 2643void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2644{ 2645 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2646} 2647 2648ssize_t AudioFlinger::MixerThread::threadLoop_write() 2649{ 2650 // FIXME we should only do one push per cycle; confirm this is true 2651 // Start the fast mixer if it's not already running 2652 if (mFastMixer != NULL) { 2653 FastMixerStateQueue *sq = mFastMixer->sq(); 2654 FastMixerState *state = sq->begin(); 2655 if (state->mCommand != FastMixerState::MIX_WRITE && 2656 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2657 if (state->mCommand == FastMixerState::COLD_IDLE) { 2658 int32_t old = android_atomic_inc(&mFastMixerFutex); 2659 if (old == -1) { 2660 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2661 } 2662#ifdef AUDIO_WATCHDOG 2663 if (mAudioWatchdog != 0) { 2664 mAudioWatchdog->resume(); 2665 } 2666#endif 2667 } 2668 state->mCommand = FastMixerState::MIX_WRITE; 2669 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2670 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2671 sq->end(); 2672 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2673 if (kUseFastMixer == FastMixer_Dynamic) { 2674 mNormalSink = mPipeSink; 2675 } 2676 } else { 2677 sq->end(false /*didModify*/); 2678 } 2679 } 2680 return PlaybackThread::threadLoop_write(); 2681} 2682 2683void AudioFlinger::MixerThread::threadLoop_standby() 2684{ 2685 // Idle the fast mixer if it's currently running 2686 if (mFastMixer != NULL) { 2687 FastMixerStateQueue *sq = mFastMixer->sq(); 2688 FastMixerState *state = sq->begin(); 2689 if (!(state->mCommand & FastMixerState::IDLE)) { 2690 state->mCommand = FastMixerState::COLD_IDLE; 2691 state->mColdFutexAddr = &mFastMixerFutex; 2692 state->mColdGen++; 2693 mFastMixerFutex = 0; 2694 sq->end(); 2695 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2696 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2697 if (kUseFastMixer == FastMixer_Dynamic) { 2698 mNormalSink = mOutputSink; 2699 } 2700#ifdef AUDIO_WATCHDOG 2701 if (mAudioWatchdog != 0) { 2702 mAudioWatchdog->pause(); 2703 } 2704#endif 2705 } else { 2706 sq->end(false /*didModify*/); 2707 } 2708 } 2709 PlaybackThread::threadLoop_standby(); 2710} 2711 2712// Empty implementation for standard mixer 2713// Overridden for offloaded playback 2714void AudioFlinger::PlaybackThread::flushOutput_l() 2715{ 2716} 2717 2718bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2719{ 2720 return false; 2721} 2722 2723bool AudioFlinger::PlaybackThread::shouldStandby_l() 2724{ 2725 return !mStandby; 2726} 2727 2728bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2729{ 2730 Mutex::Autolock _l(mLock); 2731 return waitingAsyncCallback_l(); 2732} 2733 2734// shared by MIXER and DIRECT, overridden by DUPLICATING 2735void AudioFlinger::PlaybackThread::threadLoop_standby() 2736{ 2737 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2738 mOutput->stream->common.standby(&mOutput->stream->common); 2739 if (mUseAsyncWrite != 0) { 2740 // discard any pending drain or write ack by incrementing sequence 2741 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2742 mDrainSequence = (mDrainSequence + 2) & ~1; 2743 ALOG_ASSERT(mCallbackThread != 0); 2744 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2745 mCallbackThread->setDraining(mDrainSequence); 2746 } 2747} 2748 2749void AudioFlinger::MixerThread::threadLoop_mix() 2750{ 2751 // obtain the presentation timestamp of the next output buffer 2752 int64_t pts; 2753 status_t status = INVALID_OPERATION; 2754 2755 if (mNormalSink != 0) { 2756 status = mNormalSink->getNextWriteTimestamp(&pts); 2757 } else { 2758 status = mOutputSink->getNextWriteTimestamp(&pts); 2759 } 2760 2761 if (status != NO_ERROR) { 2762 pts = AudioBufferProvider::kInvalidPTS; 2763 } 2764 2765 // mix buffers... 2766 mAudioMixer->process(pts); 2767 mCurrentWriteLength = mixBufferSize; 2768 // increase sleep time progressively when application underrun condition clears. 2769 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2770 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2771 // such that we would underrun the audio HAL. 2772 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2773 sleepTimeShift--; 2774 } 2775 sleepTime = 0; 2776 standbyTime = systemTime() + standbyDelay; 2777 //TODO: delay standby when effects have a tail 2778} 2779 2780void AudioFlinger::MixerThread::threadLoop_sleepTime() 2781{ 2782 // If no tracks are ready, sleep once for the duration of an output 2783 // buffer size, then write 0s to the output 2784 if (sleepTime == 0) { 2785 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2786 sleepTime = activeSleepTime >> sleepTimeShift; 2787 if (sleepTime < kMinThreadSleepTimeUs) { 2788 sleepTime = kMinThreadSleepTimeUs; 2789 } 2790 // reduce sleep time in case of consecutive application underruns to avoid 2791 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2792 // duration we would end up writing less data than needed by the audio HAL if 2793 // the condition persists. 2794 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2795 sleepTimeShift++; 2796 } 2797 } else { 2798 sleepTime = idleSleepTime; 2799 } 2800 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2801 memset (mMixBuffer, 0, mixBufferSize); 2802 sleepTime = 0; 2803 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2804 "anticipated start"); 2805 } 2806 // TODO add standby time extension fct of effect tail 2807} 2808 2809// prepareTracks_l() must be called with ThreadBase::mLock held 2810AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2811 Vector< sp<Track> > *tracksToRemove) 2812{ 2813 2814 mixer_state mixerStatus = MIXER_IDLE; 2815 // find out which tracks need to be processed 2816 size_t count = mActiveTracks.size(); 2817 size_t mixedTracks = 0; 2818 size_t tracksWithEffect = 0; 2819 // counts only _active_ fast tracks 2820 size_t fastTracks = 0; 2821 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2822 2823 float masterVolume = mMasterVolume; 2824 bool masterMute = mMasterMute; 2825 2826 if (masterMute) { 2827 masterVolume = 0; 2828 } 2829 // Delegate master volume control to effect in output mix effect chain if needed 2830 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2831 if (chain != 0) { 2832 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2833 chain->setVolume_l(&v, &v); 2834 masterVolume = (float)((v + (1 << 23)) >> 24); 2835 chain.clear(); 2836 } 2837 2838 // prepare a new state to push 2839 FastMixerStateQueue *sq = NULL; 2840 FastMixerState *state = NULL; 2841 bool didModify = false; 2842 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2843 if (mFastMixer != NULL) { 2844 sq = mFastMixer->sq(); 2845 state = sq->begin(); 2846 } 2847 2848 for (size_t i=0 ; i<count ; i++) { 2849 const sp<Track> t = mActiveTracks[i].promote(); 2850 if (t == 0) { 2851 continue; 2852 } 2853 2854 // this const just means the local variable doesn't change 2855 Track* const track = t.get(); 2856 2857 // process fast tracks 2858 if (track->isFastTrack()) { 2859 2860 // It's theoretically possible (though unlikely) for a fast track to be created 2861 // and then removed within the same normal mix cycle. This is not a problem, as 2862 // the track never becomes active so it's fast mixer slot is never touched. 2863 // The converse, of removing an (active) track and then creating a new track 2864 // at the identical fast mixer slot within the same normal mix cycle, 2865 // is impossible because the slot isn't marked available until the end of each cycle. 2866 int j = track->mFastIndex; 2867 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2868 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2869 FastTrack *fastTrack = &state->mFastTracks[j]; 2870 2871 // Determine whether the track is currently in underrun condition, 2872 // and whether it had a recent underrun. 2873 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2874 FastTrackUnderruns underruns = ftDump->mUnderruns; 2875 uint32_t recentFull = (underruns.mBitFields.mFull - 2876 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2877 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2878 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2879 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2880 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2881 uint32_t recentUnderruns = recentPartial + recentEmpty; 2882 track->mObservedUnderruns = underruns; 2883 // don't count underruns that occur while stopping or pausing 2884 // or stopped which can occur when flush() is called while active 2885 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2886 recentUnderruns > 0) { 2887 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2888 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2889 } 2890 2891 // This is similar to the state machine for normal tracks, 2892 // with a few modifications for fast tracks. 2893 bool isActive = true; 2894 switch (track->mState) { 2895 case TrackBase::STOPPING_1: 2896 // track stays active in STOPPING_1 state until first underrun 2897 if (recentUnderruns > 0 || track->isTerminated()) { 2898 track->mState = TrackBase::STOPPING_2; 2899 } 2900 break; 2901 case TrackBase::PAUSING: 2902 // ramp down is not yet implemented 2903 track->setPaused(); 2904 break; 2905 case TrackBase::RESUMING: 2906 // ramp up is not yet implemented 2907 track->mState = TrackBase::ACTIVE; 2908 break; 2909 case TrackBase::ACTIVE: 2910 if (recentFull > 0 || recentPartial > 0) { 2911 // track has provided at least some frames recently: reset retry count 2912 track->mRetryCount = kMaxTrackRetries; 2913 } 2914 if (recentUnderruns == 0) { 2915 // no recent underruns: stay active 2916 break; 2917 } 2918 // there has recently been an underrun of some kind 2919 if (track->sharedBuffer() == 0) { 2920 // were any of the recent underruns "empty" (no frames available)? 2921 if (recentEmpty == 0) { 2922 // no, then ignore the partial underruns as they are allowed indefinitely 2923 break; 2924 } 2925 // there has recently been an "empty" underrun: decrement the retry counter 2926 if (--(track->mRetryCount) > 0) { 2927 break; 2928 } 2929 // indicate to client process that the track was disabled because of underrun; 2930 // it will then automatically call start() when data is available 2931 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2932 // remove from active list, but state remains ACTIVE [confusing but true] 2933 isActive = false; 2934 break; 2935 } 2936 // fall through 2937 case TrackBase::STOPPING_2: 2938 case TrackBase::PAUSED: 2939 case TrackBase::STOPPED: 2940 case TrackBase::FLUSHED: // flush() while active 2941 // Check for presentation complete if track is inactive 2942 // We have consumed all the buffers of this track. 2943 // This would be incomplete if we auto-paused on underrun 2944 { 2945 size_t audioHALFrames = 2946 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2947 size_t framesWritten = mBytesWritten / mFrameSize; 2948 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2949 // track stays in active list until presentation is complete 2950 break; 2951 } 2952 } 2953 if (track->isStopping_2()) { 2954 track->mState = TrackBase::STOPPED; 2955 } 2956 if (track->isStopped()) { 2957 // Can't reset directly, as fast mixer is still polling this track 2958 // track->reset(); 2959 // So instead mark this track as needing to be reset after push with ack 2960 resetMask |= 1 << i; 2961 } 2962 isActive = false; 2963 break; 2964 case TrackBase::IDLE: 2965 default: 2966 LOG_FATAL("unexpected track state %d", track->mState); 2967 } 2968 2969 if (isActive) { 2970 // was it previously inactive? 2971 if (!(state->mTrackMask & (1 << j))) { 2972 ExtendedAudioBufferProvider *eabp = track; 2973 VolumeProvider *vp = track; 2974 fastTrack->mBufferProvider = eabp; 2975 fastTrack->mVolumeProvider = vp; 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 3039 size_t framesReady = track->framesReady(); 3040 if ((framesReady >= minFrames) && track->isReady() && 3041 !track->isPaused() && !track->isTerminated()) 3042 { 3043 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3044 3045 mixedTracks++; 3046 3047 // track->mainBuffer() != mMixBuffer means there is an effect chain 3048 // connected to the track 3049 chain.clear(); 3050 if (track->mainBuffer() != mMixBuffer) { 3051 chain = getEffectChain_l(track->sessionId()); 3052 // Delegate volume control to effect in track effect chain if needed 3053 if (chain != 0) { 3054 tracksWithEffect++; 3055 } else { 3056 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3057 "session %d", 3058 name, track->sessionId()); 3059 } 3060 } 3061 3062 3063 int param = AudioMixer::VOLUME; 3064 if (track->mFillingUpStatus == Track::FS_FILLED) { 3065 // no ramp for the first volume setting 3066 track->mFillingUpStatus = Track::FS_ACTIVE; 3067 if (track->mState == TrackBase::RESUMING) { 3068 track->mState = TrackBase::ACTIVE; 3069 param = AudioMixer::RAMP_VOLUME; 3070 } 3071 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3072 // FIXME should not make a decision based on mServer 3073 } else if (cblk->mServer != 0) { 3074 // If the track is stopped before the first frame was mixed, 3075 // do not apply ramp 3076 param = AudioMixer::RAMP_VOLUME; 3077 } 3078 3079 // compute volume for this track 3080 uint32_t vl, vr, va; 3081 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3082 vl = vr = va = 0; 3083 if (track->isPausing()) { 3084 track->setPaused(); 3085 } 3086 } else { 3087 3088 // read original volumes with volume control 3089 float typeVolume = mStreamTypes[track->streamType()].volume; 3090 float v = masterVolume * typeVolume; 3091 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3092 uint32_t vlr = proxy->getVolumeLR(); 3093 vl = vlr & 0xFFFF; 3094 vr = vlr >> 16; 3095 // track volumes come from shared memory, so can't be trusted and must be clamped 3096 if (vl > MAX_GAIN_INT) { 3097 ALOGV("Track left volume out of range: %04X", vl); 3098 vl = MAX_GAIN_INT; 3099 } 3100 if (vr > MAX_GAIN_INT) { 3101 ALOGV("Track right volume out of range: %04X", vr); 3102 vr = MAX_GAIN_INT; 3103 } 3104 // now apply the master volume and stream type volume 3105 vl = (uint32_t)(v * vl) << 12; 3106 vr = (uint32_t)(v * vr) << 12; 3107 // assuming master volume and stream type volume each go up to 1.0, 3108 // vl and vr are now in 8.24 format 3109 3110 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3111 // send level comes from shared memory and so may be corrupt 3112 if (sendLevel > MAX_GAIN_INT) { 3113 ALOGV("Track send level out of range: %04X", sendLevel); 3114 sendLevel = MAX_GAIN_INT; 3115 } 3116 va = (uint32_t)(v * sendLevel); 3117 } 3118 3119 // Delegate volume control to effect in track effect chain if needed 3120 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3121 // Do not ramp volume if volume is controlled by effect 3122 param = AudioMixer::VOLUME; 3123 track->mHasVolumeController = true; 3124 } else { 3125 // force no volume ramp when volume controller was just disabled or removed 3126 // from effect chain to avoid volume spike 3127 if (track->mHasVolumeController) { 3128 param = AudioMixer::VOLUME; 3129 } 3130 track->mHasVolumeController = false; 3131 } 3132 3133 // Convert volumes from 8.24 to 4.12 format 3134 // This additional clamping is needed in case chain->setVolume_l() overshot 3135 vl = (vl + (1 << 11)) >> 12; 3136 if (vl > MAX_GAIN_INT) { 3137 vl = MAX_GAIN_INT; 3138 } 3139 vr = (vr + (1 << 11)) >> 12; 3140 if (vr > MAX_GAIN_INT) { 3141 vr = MAX_GAIN_INT; 3142 } 3143 3144 if (va > MAX_GAIN_INT) { 3145 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3146 } 3147 3148 // XXX: these things DON'T need to be done each time 3149 mAudioMixer->setBufferProvider(name, track); 3150 mAudioMixer->enable(name); 3151 3152 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3153 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3154 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3155 mAudioMixer->setParameter( 3156 name, 3157 AudioMixer::TRACK, 3158 AudioMixer::FORMAT, (void *)track->format()); 3159 mAudioMixer->setParameter( 3160 name, 3161 AudioMixer::TRACK, 3162 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3163 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3164 uint32_t maxSampleRate = mSampleRate * 2; 3165 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3166 if (reqSampleRate == 0) { 3167 reqSampleRate = mSampleRate; 3168 } else if (reqSampleRate > maxSampleRate) { 3169 reqSampleRate = maxSampleRate; 3170 } 3171 mAudioMixer->setParameter( 3172 name, 3173 AudioMixer::RESAMPLE, 3174 AudioMixer::SAMPLE_RATE, 3175 (void *)reqSampleRate); 3176 mAudioMixer->setParameter( 3177 name, 3178 AudioMixer::TRACK, 3179 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3180 mAudioMixer->setParameter( 3181 name, 3182 AudioMixer::TRACK, 3183 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3184 3185 // reset retry count 3186 track->mRetryCount = kMaxTrackRetries; 3187 3188 // If one track is ready, set the mixer ready if: 3189 // - the mixer was not ready during previous round OR 3190 // - no other track is not ready 3191 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3192 mixerStatus != MIXER_TRACKS_ENABLED) { 3193 mixerStatus = MIXER_TRACKS_READY; 3194 } 3195 } else { 3196 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3197 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3198 } 3199 // clear effect chain input buffer if an active track underruns to avoid sending 3200 // previous audio buffer again to effects 3201 chain = getEffectChain_l(track->sessionId()); 3202 if (chain != 0) { 3203 chain->clearInputBuffer(); 3204 } 3205 3206 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3207 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3208 track->isStopped() || track->isPaused()) { 3209 // We have consumed all the buffers of this track. 3210 // Remove it from the list of active tracks. 3211 // TODO: use actual buffer filling status instead of latency when available from 3212 // audio HAL 3213 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3214 size_t framesWritten = mBytesWritten / mFrameSize; 3215 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3216 if (track->isStopped()) { 3217 track->reset(); 3218 } 3219 tracksToRemove->add(track); 3220 } 3221 } else { 3222 // No buffers for this track. Give it a few chances to 3223 // fill a buffer, then remove it from active list. 3224 if (--(track->mRetryCount) <= 0) { 3225 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3226 tracksToRemove->add(track); 3227 // indicate to client process that the track was disabled because of underrun; 3228 // it will then automatically call start() when data is available 3229 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3230 // If one track is not ready, mark the mixer also not ready if: 3231 // - the mixer was ready during previous round OR 3232 // - no other track is ready 3233 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3234 mixerStatus != MIXER_TRACKS_READY) { 3235 mixerStatus = MIXER_TRACKS_ENABLED; 3236 } 3237 } 3238 mAudioMixer->disable(name); 3239 } 3240 3241 } // local variable scope to avoid goto warning 3242track_is_ready: ; 3243 3244 } 3245 3246 // Push the new FastMixer state if necessary 3247 bool pauseAudioWatchdog = false; 3248 if (didModify) { 3249 state->mFastTracksGen++; 3250 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3251 if (kUseFastMixer == FastMixer_Dynamic && 3252 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3253 state->mCommand = FastMixerState::COLD_IDLE; 3254 state->mColdFutexAddr = &mFastMixerFutex; 3255 state->mColdGen++; 3256 mFastMixerFutex = 0; 3257 if (kUseFastMixer == FastMixer_Dynamic) { 3258 mNormalSink = mOutputSink; 3259 } 3260 // If we go into cold idle, need to wait for acknowledgement 3261 // so that fast mixer stops doing I/O. 3262 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3263 pauseAudioWatchdog = true; 3264 } 3265 } 3266 if (sq != NULL) { 3267 sq->end(didModify); 3268 sq->push(block); 3269 } 3270#ifdef AUDIO_WATCHDOG 3271 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3272 mAudioWatchdog->pause(); 3273 } 3274#endif 3275 3276 // Now perform the deferred reset on fast tracks that have stopped 3277 while (resetMask != 0) { 3278 size_t i = __builtin_ctz(resetMask); 3279 ALOG_ASSERT(i < count); 3280 resetMask &= ~(1 << i); 3281 sp<Track> t = mActiveTracks[i].promote(); 3282 if (t == 0) { 3283 continue; 3284 } 3285 Track* track = t.get(); 3286 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3287 track->reset(); 3288 } 3289 3290 // remove all the tracks that need to be... 3291 removeTracks_l(*tracksToRemove); 3292 3293 // mix buffer must be cleared if all tracks are connected to an 3294 // effect chain as in this case the mixer will not write to 3295 // mix buffer and track effects will accumulate into it 3296 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3297 (mixedTracks == 0 && fastTracks > 0))) { 3298 // FIXME as a performance optimization, should remember previous zero status 3299 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3300 } 3301 3302 // if any fast tracks, then status is ready 3303 mMixerStatusIgnoringFastTracks = mixerStatus; 3304 if (fastTracks > 0) { 3305 mixerStatus = MIXER_TRACKS_READY; 3306 } 3307 return mixerStatus; 3308} 3309 3310// getTrackName_l() must be called with ThreadBase::mLock held 3311int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3312{ 3313 return mAudioMixer->getTrackName(channelMask, sessionId); 3314} 3315 3316// deleteTrackName_l() must be called with ThreadBase::mLock held 3317void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3318{ 3319 ALOGV("remove track (%d) and delete from mixer", name); 3320 mAudioMixer->deleteTrackName(name); 3321} 3322 3323// checkForNewParameters_l() must be called with ThreadBase::mLock held 3324bool AudioFlinger::MixerThread::checkForNewParameters_l() 3325{ 3326 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3327 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3328 bool reconfig = false; 3329 3330 while (!mNewParameters.isEmpty()) { 3331 3332 if (mFastMixer != NULL) { 3333 FastMixerStateQueue *sq = mFastMixer->sq(); 3334 FastMixerState *state = sq->begin(); 3335 if (!(state->mCommand & FastMixerState::IDLE)) { 3336 previousCommand = state->mCommand; 3337 state->mCommand = FastMixerState::HOT_IDLE; 3338 sq->end(); 3339 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3340 } else { 3341 sq->end(false /*didModify*/); 3342 } 3343 } 3344 3345 status_t status = NO_ERROR; 3346 String8 keyValuePair = mNewParameters[0]; 3347 AudioParameter param = AudioParameter(keyValuePair); 3348 int value; 3349 3350 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3351 reconfig = true; 3352 } 3353 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3354 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3355 status = BAD_VALUE; 3356 } else { 3357 reconfig = true; 3358 } 3359 } 3360 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3361 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3362 status = BAD_VALUE; 3363 } else { 3364 reconfig = true; 3365 } 3366 } 3367 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3368 // do not accept frame count changes if tracks are open as the track buffer 3369 // size depends on frame count and correct behavior would not be guaranteed 3370 // if frame count is changed after track creation 3371 if (!mTracks.isEmpty()) { 3372 status = INVALID_OPERATION; 3373 } else { 3374 reconfig = true; 3375 } 3376 } 3377 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3378#ifdef ADD_BATTERY_DATA 3379 // when changing the audio output device, call addBatteryData to notify 3380 // the change 3381 if (mOutDevice != value) { 3382 uint32_t params = 0; 3383 // check whether speaker is on 3384 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3385 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3386 } 3387 3388 audio_devices_t deviceWithoutSpeaker 3389 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3390 // check if any other device (except speaker) is on 3391 if (value & deviceWithoutSpeaker ) { 3392 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3393 } 3394 3395 if (params != 0) { 3396 addBatteryData(params); 3397 } 3398 } 3399#endif 3400 3401 // forward device change to effects that have requested to be 3402 // aware of attached audio device. 3403 if (value != AUDIO_DEVICE_NONE) { 3404 mOutDevice = value; 3405 for (size_t i = 0; i < mEffectChains.size(); i++) { 3406 mEffectChains[i]->setDevice_l(mOutDevice); 3407 } 3408 } 3409 } 3410 3411 if (status == NO_ERROR) { 3412 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3413 keyValuePair.string()); 3414 if (!mStandby && status == INVALID_OPERATION) { 3415 mOutput->stream->common.standby(&mOutput->stream->common); 3416 mStandby = true; 3417 mBytesWritten = 0; 3418 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3419 keyValuePair.string()); 3420 } 3421 if (status == NO_ERROR && reconfig) { 3422 readOutputParameters(); 3423 delete mAudioMixer; 3424 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3425 for (size_t i = 0; i < mTracks.size() ; i++) { 3426 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3427 if (name < 0) { 3428 break; 3429 } 3430 mTracks[i]->mName = name; 3431 } 3432 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3433 } 3434 } 3435 3436 mNewParameters.removeAt(0); 3437 3438 mParamStatus = status; 3439 mParamCond.signal(); 3440 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3441 // already timed out waiting for the status and will never signal the condition. 3442 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3443 } 3444 3445 if (!(previousCommand & FastMixerState::IDLE)) { 3446 ALOG_ASSERT(mFastMixer != NULL); 3447 FastMixerStateQueue *sq = mFastMixer->sq(); 3448 FastMixerState *state = sq->begin(); 3449 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3450 state->mCommand = previousCommand; 3451 sq->end(); 3452 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3453 } 3454 3455 return reconfig; 3456} 3457 3458 3459void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3460{ 3461 const size_t SIZE = 256; 3462 char buffer[SIZE]; 3463 String8 result; 3464 3465 PlaybackThread::dumpInternals(fd, args); 3466 3467 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3468 result.append(buffer); 3469 write(fd, result.string(), result.size()); 3470 3471 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3472 const FastMixerDumpState copy(mFastMixerDumpState); 3473 copy.dump(fd); 3474 3475#ifdef STATE_QUEUE_DUMP 3476 // Similar for state queue 3477 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3478 observerCopy.dump(fd); 3479 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3480 mutatorCopy.dump(fd); 3481#endif 3482 3483#ifdef TEE_SINK 3484 // Write the tee output to a .wav file 3485 dumpTee(fd, mTeeSource, mId); 3486#endif 3487 3488#ifdef AUDIO_WATCHDOG 3489 if (mAudioWatchdog != 0) { 3490 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3491 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3492 wdCopy.dump(fd); 3493 } 3494#endif 3495} 3496 3497uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3498{ 3499 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3500} 3501 3502uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3503{ 3504 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3505} 3506 3507void AudioFlinger::MixerThread::cacheParameters_l() 3508{ 3509 PlaybackThread::cacheParameters_l(); 3510 3511 // FIXME: Relaxed timing because of a certain device that can't meet latency 3512 // Should be reduced to 2x after the vendor fixes the driver issue 3513 // increase threshold again due to low power audio mode. The way this warning 3514 // threshold is calculated and its usefulness should be reconsidered anyway. 3515 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3516} 3517 3518// ---------------------------------------------------------------------------- 3519 3520AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3521 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3522 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3523 // mLeftVolFloat, mRightVolFloat 3524{ 3525} 3526 3527AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3528 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3529 ThreadBase::type_t type) 3530 : PlaybackThread(audioFlinger, output, id, device, type) 3531 // mLeftVolFloat, mRightVolFloat 3532{ 3533} 3534 3535AudioFlinger::DirectOutputThread::~DirectOutputThread() 3536{ 3537} 3538 3539void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3540{ 3541 audio_track_cblk_t* cblk = track->cblk(); 3542 float left, right; 3543 3544 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3545 left = right = 0; 3546 } else { 3547 float typeVolume = mStreamTypes[track->streamType()].volume; 3548 float v = mMasterVolume * typeVolume; 3549 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3550 uint32_t vlr = proxy->getVolumeLR(); 3551 float v_clamped = v * (vlr & 0xFFFF); 3552 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3553 left = v_clamped/MAX_GAIN; 3554 v_clamped = v * (vlr >> 16); 3555 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3556 right = v_clamped/MAX_GAIN; 3557 } 3558 3559 if (lastTrack) { 3560 if (left != mLeftVolFloat || right != mRightVolFloat) { 3561 mLeftVolFloat = left; 3562 mRightVolFloat = right; 3563 3564 // Convert volumes from float to 8.24 3565 uint32_t vl = (uint32_t)(left * (1 << 24)); 3566 uint32_t vr = (uint32_t)(right * (1 << 24)); 3567 3568 // Delegate volume control to effect in track effect chain if needed 3569 // only one effect chain can be present on DirectOutputThread, so if 3570 // there is one, the track is connected to it 3571 if (!mEffectChains.isEmpty()) { 3572 mEffectChains[0]->setVolume_l(&vl, &vr); 3573 left = (float)vl / (1 << 24); 3574 right = (float)vr / (1 << 24); 3575 } 3576 if (mOutput->stream->set_volume) { 3577 mOutput->stream->set_volume(mOutput->stream, left, right); 3578 } 3579 } 3580 } 3581} 3582 3583 3584AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3585 Vector< sp<Track> > *tracksToRemove 3586) 3587{ 3588 size_t count = mActiveTracks.size(); 3589 mixer_state mixerStatus = MIXER_IDLE; 3590 3591 // find out which tracks need to be processed 3592 for (size_t i = 0; i < count; i++) { 3593 sp<Track> t = mActiveTracks[i].promote(); 3594 // The track died recently 3595 if (t == 0) { 3596 continue; 3597 } 3598 3599 Track* const track = t.get(); 3600 audio_track_cblk_t* cblk = track->cblk(); 3601 // Only consider last track started for volume and mixer state control. 3602 // In theory an older track could underrun and restart after the new one starts 3603 // but as we only care about the transition phase between two tracks on a 3604 // direct output, it is not a problem to ignore the underrun case. 3605 sp<Track> l = mLatestActiveTrack.promote(); 3606 bool last = l.get() == track; 3607 3608 // The first time a track is added we wait 3609 // for all its buffers to be filled before processing it 3610 uint32_t minFrames; 3611 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3612 minFrames = mNormalFrameCount; 3613 } else { 3614 minFrames = 1; 3615 } 3616 3617 if ((track->framesReady() >= minFrames) && track->isReady() && 3618 !track->isPaused() && !track->isTerminated()) 3619 { 3620 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3621 3622 if (track->mFillingUpStatus == Track::FS_FILLED) { 3623 track->mFillingUpStatus = Track::FS_ACTIVE; 3624 // make sure processVolume_l() will apply new volume even if 0 3625 mLeftVolFloat = mRightVolFloat = -1.0; 3626 if (track->mState == TrackBase::RESUMING) { 3627 track->mState = TrackBase::ACTIVE; 3628 } 3629 } 3630 3631 // compute volume for this track 3632 processVolume_l(track, last); 3633 if (last) { 3634 // reset retry count 3635 track->mRetryCount = kMaxTrackRetriesDirect; 3636 mActiveTrack = t; 3637 mixerStatus = MIXER_TRACKS_READY; 3638 } 3639 } else { 3640 // clear effect chain input buffer if the last active track started underruns 3641 // to avoid sending previous audio buffer again to effects 3642 if (!mEffectChains.isEmpty() && last) { 3643 mEffectChains[0]->clearInputBuffer(); 3644 } 3645 3646 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3647 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3648 track->isStopped() || track->isPaused()) { 3649 // We have consumed all the buffers of this track. 3650 // Remove it from the list of active tracks. 3651 // TODO: implement behavior for compressed audio 3652 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3653 size_t framesWritten = mBytesWritten / mFrameSize; 3654 if (mStandby || !last || 3655 track->presentationComplete(framesWritten, audioHALFrames)) { 3656 if (track->isStopped()) { 3657 track->reset(); 3658 } 3659 tracksToRemove->add(track); 3660 } 3661 } else { 3662 // No buffers for this track. Give it a few chances to 3663 // fill a buffer, then remove it from active list. 3664 // Only consider last track started for mixer state control 3665 if (--(track->mRetryCount) <= 0) { 3666 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3667 tracksToRemove->add(track); 3668 // indicate to client process that the track was disabled because of underrun; 3669 // it will then automatically call start() when data is available 3670 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3671 } else if (last) { 3672 mixerStatus = MIXER_TRACKS_ENABLED; 3673 } 3674 } 3675 } 3676 } 3677 3678 // remove all the tracks that need to be... 3679 removeTracks_l(*tracksToRemove); 3680 3681 return mixerStatus; 3682} 3683 3684void AudioFlinger::DirectOutputThread::threadLoop_mix() 3685{ 3686 size_t frameCount = mFrameCount; 3687 int8_t *curBuf = (int8_t *)mMixBuffer; 3688 // output audio to hardware 3689 while (frameCount) { 3690 AudioBufferProvider::Buffer buffer; 3691 buffer.frameCount = frameCount; 3692 mActiveTrack->getNextBuffer(&buffer); 3693 if (buffer.raw == NULL) { 3694 memset(curBuf, 0, frameCount * mFrameSize); 3695 break; 3696 } 3697 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3698 frameCount -= buffer.frameCount; 3699 curBuf += buffer.frameCount * mFrameSize; 3700 mActiveTrack->releaseBuffer(&buffer); 3701 } 3702 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3703 sleepTime = 0; 3704 standbyTime = systemTime() + standbyDelay; 3705 mActiveTrack.clear(); 3706} 3707 3708void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3709{ 3710 if (sleepTime == 0) { 3711 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3712 sleepTime = activeSleepTime; 3713 } else { 3714 sleepTime = idleSleepTime; 3715 } 3716 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3717 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3718 sleepTime = 0; 3719 } 3720} 3721 3722// getTrackName_l() must be called with ThreadBase::mLock held 3723int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3724 int sessionId) 3725{ 3726 return 0; 3727} 3728 3729// deleteTrackName_l() must be called with ThreadBase::mLock held 3730void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3731{ 3732} 3733 3734// checkForNewParameters_l() must be called with ThreadBase::mLock held 3735bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3736{ 3737 bool reconfig = false; 3738 3739 while (!mNewParameters.isEmpty()) { 3740 status_t status = NO_ERROR; 3741 String8 keyValuePair = mNewParameters[0]; 3742 AudioParameter param = AudioParameter(keyValuePair); 3743 int value; 3744 3745 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3746 // do not accept frame count changes if tracks are open as the track buffer 3747 // size depends on frame count and correct behavior would not be garantied 3748 // if frame count is changed after track creation 3749 if (!mTracks.isEmpty()) { 3750 status = INVALID_OPERATION; 3751 } else { 3752 reconfig = true; 3753 } 3754 } 3755 if (status == NO_ERROR) { 3756 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3757 keyValuePair.string()); 3758 if (!mStandby && status == INVALID_OPERATION) { 3759 mOutput->stream->common.standby(&mOutput->stream->common); 3760 mStandby = true; 3761 mBytesWritten = 0; 3762 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3763 keyValuePair.string()); 3764 } 3765 if (status == NO_ERROR && reconfig) { 3766 readOutputParameters(); 3767 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3768 } 3769 } 3770 3771 mNewParameters.removeAt(0); 3772 3773 mParamStatus = status; 3774 mParamCond.signal(); 3775 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3776 // already timed out waiting for the status and will never signal the condition. 3777 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3778 } 3779 return reconfig; 3780} 3781 3782uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3783{ 3784 uint32_t time; 3785 if (audio_is_linear_pcm(mFormat)) { 3786 time = PlaybackThread::activeSleepTimeUs(); 3787 } else { 3788 time = 10000; 3789 } 3790 return time; 3791} 3792 3793uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3794{ 3795 uint32_t time; 3796 if (audio_is_linear_pcm(mFormat)) { 3797 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3798 } else { 3799 time = 10000; 3800 } 3801 return time; 3802} 3803 3804uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3805{ 3806 uint32_t time; 3807 if (audio_is_linear_pcm(mFormat)) { 3808 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3809 } else { 3810 time = 10000; 3811 } 3812 return time; 3813} 3814 3815void AudioFlinger::DirectOutputThread::cacheParameters_l() 3816{ 3817 PlaybackThread::cacheParameters_l(); 3818 3819 // use shorter standby delay as on normal output to release 3820 // hardware resources as soon as possible 3821 if (audio_is_linear_pcm(mFormat)) { 3822 standbyDelay = microseconds(activeSleepTime*2); 3823 } else { 3824 standbyDelay = kOffloadStandbyDelayNs; 3825 } 3826} 3827 3828// ---------------------------------------------------------------------------- 3829 3830AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3831 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3832 : Thread(false /*canCallJava*/), 3833 mPlaybackThread(playbackThread), 3834 mWriteAckSequence(0), 3835 mDrainSequence(0) 3836{ 3837} 3838 3839AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3840{ 3841} 3842 3843void AudioFlinger::AsyncCallbackThread::onFirstRef() 3844{ 3845 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3846} 3847 3848bool AudioFlinger::AsyncCallbackThread::threadLoop() 3849{ 3850 while (!exitPending()) { 3851 uint32_t writeAckSequence; 3852 uint32_t drainSequence; 3853 3854 { 3855 Mutex::Autolock _l(mLock); 3856 while (!((mWriteAckSequence & 1) || 3857 (mDrainSequence & 1) || 3858 exitPending())) { 3859 mWaitWorkCV.wait(mLock); 3860 } 3861 3862 if (exitPending()) { 3863 break; 3864 } 3865 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3866 mWriteAckSequence, mDrainSequence); 3867 writeAckSequence = mWriteAckSequence; 3868 mWriteAckSequence &= ~1; 3869 drainSequence = mDrainSequence; 3870 mDrainSequence &= ~1; 3871 } 3872 { 3873 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3874 if (playbackThread != 0) { 3875 if (writeAckSequence & 1) { 3876 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3877 } 3878 if (drainSequence & 1) { 3879 playbackThread->resetDraining(drainSequence >> 1); 3880 } 3881 } 3882 } 3883 } 3884 return false; 3885} 3886 3887void AudioFlinger::AsyncCallbackThread::exit() 3888{ 3889 ALOGV("AsyncCallbackThread::exit"); 3890 Mutex::Autolock _l(mLock); 3891 requestExit(); 3892 mWaitWorkCV.broadcast(); 3893} 3894 3895void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3896{ 3897 Mutex::Autolock _l(mLock); 3898 // bit 0 is cleared 3899 mWriteAckSequence = sequence << 1; 3900} 3901 3902void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3903{ 3904 Mutex::Autolock _l(mLock); 3905 // ignore unexpected callbacks 3906 if (mWriteAckSequence & 2) { 3907 mWriteAckSequence |= 1; 3908 mWaitWorkCV.signal(); 3909 } 3910} 3911 3912void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3913{ 3914 Mutex::Autolock _l(mLock); 3915 // bit 0 is cleared 3916 mDrainSequence = sequence << 1; 3917} 3918 3919void AudioFlinger::AsyncCallbackThread::resetDraining() 3920{ 3921 Mutex::Autolock _l(mLock); 3922 // ignore unexpected callbacks 3923 if (mDrainSequence & 2) { 3924 mDrainSequence |= 1; 3925 mWaitWorkCV.signal(); 3926 } 3927} 3928 3929 3930// ---------------------------------------------------------------------------- 3931AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3932 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3933 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3934 mHwPaused(false), 3935 mFlushPending(false), 3936 mPausedBytesRemaining(0) 3937{ 3938 //FIXME: mStandby should be set to true by ThreadBase constructor 3939 mStandby = true; 3940} 3941 3942void AudioFlinger::OffloadThread::threadLoop_exit() 3943{ 3944 if (mFlushPending || mHwPaused) { 3945 // If a flush is pending or track was paused, just discard buffered data 3946 flushHw_l(); 3947 } else { 3948 mMixerStatus = MIXER_DRAIN_ALL; 3949 threadLoop_drain(); 3950 } 3951 mCallbackThread->exit(); 3952 PlaybackThread::threadLoop_exit(); 3953} 3954 3955AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3956 Vector< sp<Track> > *tracksToRemove 3957) 3958{ 3959 size_t count = mActiveTracks.size(); 3960 3961 mixer_state mixerStatus = MIXER_IDLE; 3962 bool doHwPause = false; 3963 bool doHwResume = false; 3964 3965 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3966 3967 // find out which tracks need to be processed 3968 for (size_t i = 0; i < count; i++) { 3969 sp<Track> t = mActiveTracks[i].promote(); 3970 // The track died recently 3971 if (t == 0) { 3972 continue; 3973 } 3974 Track* const track = t.get(); 3975 audio_track_cblk_t* cblk = track->cblk(); 3976 // Only consider last track started for volume and mixer state control. 3977 // In theory an older track could underrun and restart after the new one starts 3978 // but as we only care about the transition phase between two tracks on a 3979 // direct output, it is not a problem to ignore the underrun case. 3980 sp<Track> l = mLatestActiveTrack.promote(); 3981 bool last = l.get() == track; 3982 3983 if (track->isPausing()) { 3984 track->setPaused(); 3985 if (last) { 3986 if (!mHwPaused) { 3987 doHwPause = true; 3988 mHwPaused = true; 3989 } 3990 // If we were part way through writing the mixbuffer to 3991 // the HAL we must save this until we resume 3992 // BUG - this will be wrong if a different track is made active, 3993 // in that case we want to discard the pending data in the 3994 // mixbuffer and tell the client to present it again when the 3995 // track is resumed 3996 mPausedWriteLength = mCurrentWriteLength; 3997 mPausedBytesRemaining = mBytesRemaining; 3998 mBytesRemaining = 0; // stop writing 3999 } 4000 tracksToRemove->add(track); 4001 } else if (track->framesReady() && track->isReady() && 4002 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4003 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4004 if (track->mFillingUpStatus == Track::FS_FILLED) { 4005 track->mFillingUpStatus = Track::FS_ACTIVE; 4006 // make sure processVolume_l() will apply new volume even if 0 4007 mLeftVolFloat = mRightVolFloat = -1.0; 4008 if (track->mState == TrackBase::RESUMING) { 4009 track->mState = TrackBase::ACTIVE; 4010 if (last) { 4011 if (mPausedBytesRemaining) { 4012 // Need to continue write that was interrupted 4013 mCurrentWriteLength = mPausedWriteLength; 4014 mBytesRemaining = mPausedBytesRemaining; 4015 mPausedBytesRemaining = 0; 4016 } 4017 if (mHwPaused) { 4018 doHwResume = true; 4019 mHwPaused = false; 4020 // threadLoop_mix() will handle the case that we need to 4021 // resume an interrupted write 4022 } 4023 // enable write to audio HAL 4024 sleepTime = 0; 4025 } 4026 } 4027 } 4028 4029 if (last) { 4030 sp<Track> previousTrack = mPreviousTrack.promote(); 4031 if (previousTrack != 0) { 4032 if (track != previousTrack.get()) { 4033 // Flush any data still being written from last track 4034 mBytesRemaining = 0; 4035 if (mPausedBytesRemaining) { 4036 // Last track was paused so we also need to flush saved 4037 // mixbuffer state and invalidate track so that it will 4038 // re-submit that unwritten data when it is next resumed 4039 mPausedBytesRemaining = 0; 4040 // Invalidate is a bit drastic - would be more efficient 4041 // to have a flag to tell client that some of the 4042 // previously written data was lost 4043 previousTrack->invalidate(); 4044 } 4045 // flush data already sent to the DSP if changing audio session as audio 4046 // comes from a different source. Also invalidate previous track to force a 4047 // seek when resuming. 4048 if (previousTrack->sessionId() != track->sessionId()) { 4049 previousTrack->invalidate(); 4050 mFlushPending = true; 4051 } 4052 } 4053 } 4054 mPreviousTrack = track; 4055 // reset retry count 4056 track->mRetryCount = kMaxTrackRetriesOffload; 4057 mActiveTrack = t; 4058 mixerStatus = MIXER_TRACKS_READY; 4059 } 4060 } else { 4061 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4062 if (track->isStopping_1()) { 4063 // Hardware buffer can hold a large amount of audio so we must 4064 // wait for all current track's data to drain before we say 4065 // that the track is stopped. 4066 if (mBytesRemaining == 0) { 4067 // Only start draining when all data in mixbuffer 4068 // has been written 4069 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4070 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4071 // do not drain if no data was ever sent to HAL (mStandby == true) 4072 if (last && !mStandby) { 4073 // do not modify drain sequence if we are already draining. This happens 4074 // when resuming from pause after drain. 4075 if ((mDrainSequence & 1) == 0) { 4076 sleepTime = 0; 4077 standbyTime = systemTime() + standbyDelay; 4078 mixerStatus = MIXER_DRAIN_TRACK; 4079 mDrainSequence += 2; 4080 } 4081 if (mHwPaused) { 4082 // It is possible to move from PAUSED to STOPPING_1 without 4083 // a resume so we must ensure hardware is running 4084 doHwResume = true; 4085 mHwPaused = false; 4086 } 4087 } 4088 } 4089 } else if (track->isStopping_2()) { 4090 // Drain has completed or we are in standby, signal presentation complete 4091 if (!(mDrainSequence & 1) || !last || mStandby) { 4092 track->mState = TrackBase::STOPPED; 4093 size_t audioHALFrames = 4094 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4095 size_t framesWritten = 4096 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4097 track->presentationComplete(framesWritten, audioHALFrames); 4098 track->reset(); 4099 tracksToRemove->add(track); 4100 } 4101 } else { 4102 // No buffers for this track. Give it a few chances to 4103 // fill a buffer, then remove it from active list. 4104 if (--(track->mRetryCount) <= 0) { 4105 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4106 track->name()); 4107 tracksToRemove->add(track); 4108 // indicate to client process that the track was disabled because of underrun; 4109 // it will then automatically call start() when data is available 4110 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4111 } else if (last){ 4112 mixerStatus = MIXER_TRACKS_ENABLED; 4113 } 4114 } 4115 } 4116 // compute volume for this track 4117 processVolume_l(track, last); 4118 } 4119 4120 // make sure the pause/flush/resume sequence is executed in the right order. 4121 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4122 // before flush and then resume HW. This can happen in case of pause/flush/resume 4123 // if resume is received before pause is executed. 4124 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4125 mOutput->stream->pause(mOutput->stream); 4126 if (!doHwPause) { 4127 doHwResume = true; 4128 } 4129 } 4130 if (mFlushPending) { 4131 flushHw_l(); 4132 mFlushPending = false; 4133 } 4134 if (!mStandby && doHwResume) { 4135 mOutput->stream->resume(mOutput->stream); 4136 } 4137 4138 // remove all the tracks that need to be... 4139 removeTracks_l(*tracksToRemove); 4140 4141 return mixerStatus; 4142} 4143 4144void AudioFlinger::OffloadThread::flushOutput_l() 4145{ 4146 mFlushPending = true; 4147} 4148 4149// must be called with thread mutex locked 4150bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4151{ 4152 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4153 mWriteAckSequence, mDrainSequence); 4154 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4155 return true; 4156 } 4157 return false; 4158} 4159 4160// must be called with thread mutex locked 4161bool AudioFlinger::OffloadThread::shouldStandby_l() 4162{ 4163 bool TrackPaused = false; 4164 4165 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4166 // after a timeout and we will enter standby then. 4167 if (mTracks.size() > 0) { 4168 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4169 } 4170 4171 return !mStandby && !TrackPaused; 4172} 4173 4174 4175bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4176{ 4177 Mutex::Autolock _l(mLock); 4178 return waitingAsyncCallback_l(); 4179} 4180 4181void AudioFlinger::OffloadThread::flushHw_l() 4182{ 4183 mOutput->stream->flush(mOutput->stream); 4184 // Flush anything still waiting in the mixbuffer 4185 mCurrentWriteLength = 0; 4186 mBytesRemaining = 0; 4187 mPausedWriteLength = 0; 4188 mPausedBytesRemaining = 0; 4189 if (mUseAsyncWrite) { 4190 // discard any pending drain or write ack by incrementing sequence 4191 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4192 mDrainSequence = (mDrainSequence + 2) & ~1; 4193 ALOG_ASSERT(mCallbackThread != 0); 4194 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4195 mCallbackThread->setDraining(mDrainSequence); 4196 } 4197} 4198 4199// ---------------------------------------------------------------------------- 4200 4201AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4202 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4203 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4204 DUPLICATING), 4205 mWaitTimeMs(UINT_MAX) 4206{ 4207 addOutputTrack(mainThread); 4208} 4209 4210AudioFlinger::DuplicatingThread::~DuplicatingThread() 4211{ 4212 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4213 mOutputTracks[i]->destroy(); 4214 } 4215} 4216 4217void AudioFlinger::DuplicatingThread::threadLoop_mix() 4218{ 4219 // mix buffers... 4220 if (outputsReady(outputTracks)) { 4221 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4222 } else { 4223 memset(mMixBuffer, 0, mixBufferSize); 4224 } 4225 sleepTime = 0; 4226 writeFrames = mNormalFrameCount; 4227 mCurrentWriteLength = mixBufferSize; 4228 standbyTime = systemTime() + standbyDelay; 4229} 4230 4231void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4232{ 4233 if (sleepTime == 0) { 4234 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4235 sleepTime = activeSleepTime; 4236 } else { 4237 sleepTime = idleSleepTime; 4238 } 4239 } else if (mBytesWritten != 0) { 4240 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4241 writeFrames = mNormalFrameCount; 4242 memset(mMixBuffer, 0, mixBufferSize); 4243 } else { 4244 // flush remaining overflow buffers in output tracks 4245 writeFrames = 0; 4246 } 4247 sleepTime = 0; 4248 } 4249} 4250 4251ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4252{ 4253 for (size_t i = 0; i < outputTracks.size(); i++) { 4254 outputTracks[i]->write(mMixBuffer, writeFrames); 4255 } 4256 mStandby = false; 4257 return (ssize_t)mixBufferSize; 4258} 4259 4260void AudioFlinger::DuplicatingThread::threadLoop_standby() 4261{ 4262 // DuplicatingThread implements standby by stopping all tracks 4263 for (size_t i = 0; i < outputTracks.size(); i++) { 4264 outputTracks[i]->stop(); 4265 } 4266} 4267 4268void AudioFlinger::DuplicatingThread::saveOutputTracks() 4269{ 4270 outputTracks = mOutputTracks; 4271} 4272 4273void AudioFlinger::DuplicatingThread::clearOutputTracks() 4274{ 4275 outputTracks.clear(); 4276} 4277 4278void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4279{ 4280 Mutex::Autolock _l(mLock); 4281 // FIXME explain this formula 4282 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4283 OutputTrack *outputTrack = new OutputTrack(thread, 4284 this, 4285 mSampleRate, 4286 mFormat, 4287 mChannelMask, 4288 frameCount, 4289 IPCThreadState::self()->getCallingUid()); 4290 if (outputTrack->cblk() != NULL) { 4291 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4292 mOutputTracks.add(outputTrack); 4293 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4294 updateWaitTime_l(); 4295 } 4296} 4297 4298void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4299{ 4300 Mutex::Autolock _l(mLock); 4301 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4302 if (mOutputTracks[i]->thread() == thread) { 4303 mOutputTracks[i]->destroy(); 4304 mOutputTracks.removeAt(i); 4305 updateWaitTime_l(); 4306 return; 4307 } 4308 } 4309 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4310} 4311 4312// caller must hold mLock 4313void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4314{ 4315 mWaitTimeMs = UINT_MAX; 4316 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4317 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4318 if (strong != 0) { 4319 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4320 if (waitTimeMs < mWaitTimeMs) { 4321 mWaitTimeMs = waitTimeMs; 4322 } 4323 } 4324 } 4325} 4326 4327 4328bool AudioFlinger::DuplicatingThread::outputsReady( 4329 const SortedVector< sp<OutputTrack> > &outputTracks) 4330{ 4331 for (size_t i = 0; i < outputTracks.size(); i++) { 4332 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4333 if (thread == 0) { 4334 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4335 outputTracks[i].get()); 4336 return false; 4337 } 4338 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4339 // see note at standby() declaration 4340 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4341 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4342 thread.get()); 4343 return false; 4344 } 4345 } 4346 return true; 4347} 4348 4349uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4350{ 4351 return (mWaitTimeMs * 1000) / 2; 4352} 4353 4354void AudioFlinger::DuplicatingThread::cacheParameters_l() 4355{ 4356 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4357 updateWaitTime_l(); 4358 4359 MixerThread::cacheParameters_l(); 4360} 4361 4362// ---------------------------------------------------------------------------- 4363// Record 4364// ---------------------------------------------------------------------------- 4365 4366AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4367 AudioStreamIn *input, 4368 uint32_t sampleRate, 4369 audio_channel_mask_t channelMask, 4370 audio_io_handle_t id, 4371 audio_devices_t outDevice, 4372 audio_devices_t inDevice 4373#ifdef TEE_SINK 4374 , const sp<NBAIO_Sink>& teeSink 4375#endif 4376 ) : 4377 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4378 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4379 // mRsmpInIndex and mBufferSize set by readInputParameters() 4380 mReqChannelCount(popcount(channelMask)), 4381 mReqSampleRate(sampleRate) 4382 // mBytesRead is only meaningful while active, and so is cleared in start() 4383 // (but might be better to also clear here for dump?) 4384#ifdef TEE_SINK 4385 , mTeeSink(teeSink) 4386#endif 4387{ 4388 snprintf(mName, kNameLength, "AudioIn_%X", id); 4389 4390 readInputParameters(); 4391} 4392 4393 4394AudioFlinger::RecordThread::~RecordThread() 4395{ 4396 delete[] mRsmpInBuffer; 4397 delete mResampler; 4398 delete[] mRsmpOutBuffer; 4399} 4400 4401void AudioFlinger::RecordThread::onFirstRef() 4402{ 4403 run(mName, PRIORITY_URGENT_AUDIO); 4404} 4405 4406status_t AudioFlinger::RecordThread::readyToRun() 4407{ 4408 status_t status = initCheck(); 4409 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4410 return status; 4411} 4412 4413bool AudioFlinger::RecordThread::threadLoop() 4414{ 4415 AudioBufferProvider::Buffer buffer; 4416 sp<RecordTrack> activeTrack; 4417 Vector< sp<EffectChain> > effectChains; 4418 4419 nsecs_t lastWarning = 0; 4420 4421 inputStandBy(); 4422 { 4423 Mutex::Autolock _l(mLock); 4424 activeTrack = mActiveTrack; 4425 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4426 } 4427 4428 // used to verify we've read at least once before evaluating how many bytes were read 4429 bool readOnce = false; 4430 4431 // start recording 4432 while (!exitPending()) { 4433 4434 processConfigEvents(); 4435 4436 { // scope for mLock 4437 Mutex::Autolock _l(mLock); 4438 checkForNewParameters_l(); 4439 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4440 SortedVector<int> tmp; 4441 tmp.add(mActiveTrack->uid()); 4442 updateWakeLockUids_l(tmp); 4443 } 4444 activeTrack = mActiveTrack; 4445 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4446 standby(); 4447 4448 if (exitPending()) { 4449 break; 4450 } 4451 4452 releaseWakeLock_l(); 4453 ALOGV("RecordThread: loop stopping"); 4454 // go to sleep 4455 mWaitWorkCV.wait(mLock); 4456 ALOGV("RecordThread: loop starting"); 4457 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4458 continue; 4459 } 4460 if (mActiveTrack != 0) { 4461 if (mActiveTrack->isTerminated()) { 4462 removeTrack_l(mActiveTrack); 4463 mActiveTrack.clear(); 4464 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4465 standby(); 4466 mActiveTrack.clear(); 4467 mStartStopCond.broadcast(); 4468 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4469 if (mReqChannelCount != mActiveTrack->channelCount()) { 4470 mActiveTrack.clear(); 4471 mStartStopCond.broadcast(); 4472 } else if (readOnce) { 4473 // record start succeeds only if first read from audio input 4474 // succeeds 4475 if (mBytesRead >= 0) { 4476 mActiveTrack->mState = TrackBase::ACTIVE; 4477 } else { 4478 mActiveTrack.clear(); 4479 } 4480 mStartStopCond.broadcast(); 4481 } 4482 mStandby = false; 4483 } 4484 } 4485 4486 lockEffectChains_l(effectChains); 4487 } 4488 4489 if (mActiveTrack != 0) { 4490 if (mActiveTrack->mState != TrackBase::ACTIVE && 4491 mActiveTrack->mState != TrackBase::RESUMING) { 4492 unlockEffectChains(effectChains); 4493 usleep(kRecordThreadSleepUs); 4494 continue; 4495 } 4496 for (size_t i = 0; i < effectChains.size(); i ++) { 4497 effectChains[i]->process_l(); 4498 } 4499 4500 buffer.frameCount = mFrameCount; 4501 status_t status = mActiveTrack->getNextBuffer(&buffer); 4502 if (status == NO_ERROR) { 4503 readOnce = true; 4504 size_t framesOut = buffer.frameCount; 4505 if (mResampler == NULL) { 4506 // no resampling 4507 while (framesOut) { 4508 size_t framesIn = mFrameCount - mRsmpInIndex; 4509 if (framesIn) { 4510 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4511 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4512 mActiveTrack->mFrameSize; 4513 if (framesIn > framesOut) 4514 framesIn = framesOut; 4515 mRsmpInIndex += framesIn; 4516 framesOut -= framesIn; 4517 if (mChannelCount == mReqChannelCount) { 4518 memcpy(dst, src, framesIn * mFrameSize); 4519 } else { 4520 if (mChannelCount == 1) { 4521 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4522 (int16_t *)src, framesIn); 4523 } else { 4524 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4525 (int16_t *)src, framesIn); 4526 } 4527 } 4528 } 4529 if (framesOut && mFrameCount == mRsmpInIndex) { 4530 void *readInto; 4531 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4532 readInto = buffer.raw; 4533 framesOut = 0; 4534 } else { 4535 readInto = mRsmpInBuffer; 4536 mRsmpInIndex = 0; 4537 } 4538 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4539 mBufferSize); 4540 if (mBytesRead <= 0) { 4541 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4542 { 4543 ALOGE("Error reading audio input"); 4544 // Force input into standby so that it tries to 4545 // recover at next read attempt 4546 inputStandBy(); 4547 usleep(kRecordThreadSleepUs); 4548 } 4549 mRsmpInIndex = mFrameCount; 4550 framesOut = 0; 4551 buffer.frameCount = 0; 4552 } 4553#ifdef TEE_SINK 4554 else if (mTeeSink != 0) { 4555 (void) mTeeSink->write(readInto, 4556 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4557 } 4558#endif 4559 } 4560 } 4561 } else { 4562 // resampling 4563 4564 // resampler accumulates, but we only have one source track 4565 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4566 // alter output frame count as if we were expecting stereo samples 4567 if (mChannelCount == 1 && mReqChannelCount == 1) { 4568 framesOut >>= 1; 4569 } 4570 mResampler->resample(mRsmpOutBuffer, framesOut, 4571 this /* AudioBufferProvider* */); 4572 // ditherAndClamp() works as long as all buffers returned by 4573 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4574 if (mChannelCount == 2 && mReqChannelCount == 1) { 4575 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4576 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4577 // the resampler always outputs stereo samples: 4578 // do post stereo to mono conversion 4579 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4580 framesOut); 4581 } else { 4582 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4583 } 4584 // now done with mRsmpOutBuffer 4585 4586 } 4587 if (mFramestoDrop == 0) { 4588 mActiveTrack->releaseBuffer(&buffer); 4589 } else { 4590 if (mFramestoDrop > 0) { 4591 mFramestoDrop -= buffer.frameCount; 4592 if (mFramestoDrop <= 0) { 4593 clearSyncStartEvent(); 4594 } 4595 } else { 4596 mFramestoDrop += buffer.frameCount; 4597 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4598 mSyncStartEvent->isCancelled()) { 4599 ALOGW("Synced record %s, session %d, trigger session %d", 4600 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4601 mActiveTrack->sessionId(), 4602 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4603 clearSyncStartEvent(); 4604 } 4605 } 4606 } 4607 mActiveTrack->clearOverflow(); 4608 } 4609 // client isn't retrieving buffers fast enough 4610 else { 4611 if (!mActiveTrack->setOverflow()) { 4612 nsecs_t now = systemTime(); 4613 if ((now - lastWarning) > kWarningThrottleNs) { 4614 ALOGW("RecordThread: buffer overflow"); 4615 lastWarning = now; 4616 } 4617 } 4618 // Release the processor for a while before asking for a new buffer. 4619 // This will give the application more chance to read from the buffer and 4620 // clear the overflow. 4621 usleep(kRecordThreadSleepUs); 4622 } 4623 } 4624 // enable changes in effect chain 4625 unlockEffectChains(effectChains); 4626 effectChains.clear(); 4627 } 4628 4629 standby(); 4630 4631 { 4632 Mutex::Autolock _l(mLock); 4633 for (size_t i = 0; i < mTracks.size(); i++) { 4634 sp<RecordTrack> track = mTracks[i]; 4635 track->invalidate(); 4636 } 4637 mActiveTrack.clear(); 4638 mStartStopCond.broadcast(); 4639 } 4640 4641 releaseWakeLock(); 4642 4643 ALOGV("RecordThread %p exiting", this); 4644 return false; 4645} 4646 4647void AudioFlinger::RecordThread::standby() 4648{ 4649 if (!mStandby) { 4650 inputStandBy(); 4651 mStandby = true; 4652 } 4653} 4654 4655void AudioFlinger::RecordThread::inputStandBy() 4656{ 4657 mInput->stream->common.standby(&mInput->stream->common); 4658} 4659 4660sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4661 const sp<AudioFlinger::Client>& client, 4662 uint32_t sampleRate, 4663 audio_format_t format, 4664 audio_channel_mask_t channelMask, 4665 size_t frameCount, 4666 int sessionId, 4667 int uid, 4668 IAudioFlinger::track_flags_t *flags, 4669 pid_t tid, 4670 status_t *status) 4671{ 4672 sp<RecordTrack> track; 4673 status_t lStatus; 4674 4675 lStatus = initCheck(); 4676 if (lStatus != NO_ERROR) { 4677 ALOGE("createRecordTrack_l() audio driver not initialized"); 4678 goto Exit; 4679 } 4680 // client expresses a preference for FAST, but we get the final say 4681 if (*flags & IAudioFlinger::TRACK_FAST) { 4682 if ( 4683 // use case: callback handler and frame count is default or at least as large as HAL 4684 ( 4685 (tid != -1) && 4686 ((frameCount == 0) || 4687 (frameCount >= mFrameCount)) 4688 ) && 4689 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4690 // mono or stereo 4691 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4692 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4693 // hardware sample rate 4694 (sampleRate == mSampleRate) && 4695 // record thread has an associated fast recorder 4696 hasFastRecorder() 4697 // FIXME test that RecordThread for this fast track has a capable output HAL 4698 // FIXME add a permission test also? 4699 ) { 4700 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4701 if (frameCount == 0) { 4702 frameCount = mFrameCount * kFastTrackMultiplier; 4703 } 4704 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4705 frameCount, mFrameCount); 4706 } else { 4707 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4708 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4709 "hasFastRecorder=%d tid=%d", 4710 frameCount, mFrameCount, format, 4711 audio_is_linear_pcm(format), 4712 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4713 *flags &= ~IAudioFlinger::TRACK_FAST; 4714 // For compatibility with AudioRecord calculation, buffer depth is forced 4715 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4716 // This is probably too conservative, but legacy application code may depend on it. 4717 // If you change this calculation, also review the start threshold which is related. 4718 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4719 size_t mNormalFrameCount = 2048; // FIXME 4720 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4721 if (minBufCount < 2) { 4722 minBufCount = 2; 4723 } 4724 size_t minFrameCount = mNormalFrameCount * minBufCount; 4725 if (frameCount < minFrameCount) { 4726 frameCount = minFrameCount; 4727 } 4728 } 4729 } 4730 4731 // FIXME use flags and tid similar to createTrack_l() 4732 4733 { // scope for mLock 4734 Mutex::Autolock _l(mLock); 4735 4736 track = new RecordTrack(this, client, sampleRate, 4737 format, channelMask, frameCount, sessionId, uid); 4738 4739 if (track->getCblk() == 0) { 4740 ALOGE("createRecordTrack_l() no control block"); 4741 lStatus = NO_MEMORY; 4742 // track must be cleared from the caller as the caller has the AF lock 4743 goto Exit; 4744 } 4745 mTracks.add(track); 4746 4747 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4748 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4749 mAudioFlinger->btNrecIsOff(); 4750 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4751 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4752 4753 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4754 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4755 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4756 // so ask activity manager to do this on our behalf 4757 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4758 } 4759 } 4760 lStatus = NO_ERROR; 4761 4762Exit: 4763 if (status) { 4764 *status = lStatus; 4765 } 4766 return track; 4767} 4768 4769status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4770 AudioSystem::sync_event_t event, 4771 int triggerSession) 4772{ 4773 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4774 sp<ThreadBase> strongMe = this; 4775 status_t status = NO_ERROR; 4776 4777 if (event == AudioSystem::SYNC_EVENT_NONE) { 4778 clearSyncStartEvent(); 4779 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4780 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4781 triggerSession, 4782 recordTrack->sessionId(), 4783 syncStartEventCallback, 4784 this); 4785 // Sync event can be cancelled by the trigger session if the track is not in a 4786 // compatible state in which case we start record immediately 4787 if (mSyncStartEvent->isCancelled()) { 4788 clearSyncStartEvent(); 4789 } else { 4790 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4791 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4792 } 4793 } 4794 4795 { 4796 AutoMutex lock(mLock); 4797 if (mActiveTrack != 0) { 4798 if (recordTrack != mActiveTrack.get()) { 4799 status = -EBUSY; 4800 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4801 mActiveTrack->mState = TrackBase::ACTIVE; 4802 } 4803 return status; 4804 } 4805 4806 recordTrack->mState = TrackBase::IDLE; 4807 mActiveTrack = recordTrack; 4808 mLock.unlock(); 4809 status_t status = AudioSystem::startInput(mId); 4810 mLock.lock(); 4811 if (status != NO_ERROR) { 4812 mActiveTrack.clear(); 4813 clearSyncStartEvent(); 4814 return status; 4815 } 4816 mRsmpInIndex = mFrameCount; 4817 mBytesRead = 0; 4818 if (mResampler != NULL) { 4819 mResampler->reset(); 4820 } 4821 mActiveTrack->mState = TrackBase::RESUMING; 4822 // signal thread to start 4823 ALOGV("Signal record thread"); 4824 mWaitWorkCV.broadcast(); 4825 // do not wait for mStartStopCond if exiting 4826 if (exitPending()) { 4827 mActiveTrack.clear(); 4828 status = INVALID_OPERATION; 4829 goto startError; 4830 } 4831 mStartStopCond.wait(mLock); 4832 if (mActiveTrack == 0) { 4833 ALOGV("Record failed to start"); 4834 status = BAD_VALUE; 4835 goto startError; 4836 } 4837 ALOGV("Record started OK"); 4838 return status; 4839 } 4840 4841startError: 4842 AudioSystem::stopInput(mId); 4843 clearSyncStartEvent(); 4844 return status; 4845} 4846 4847void AudioFlinger::RecordThread::clearSyncStartEvent() 4848{ 4849 if (mSyncStartEvent != 0) { 4850 mSyncStartEvent->cancel(); 4851 } 4852 mSyncStartEvent.clear(); 4853 mFramestoDrop = 0; 4854} 4855 4856void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4857{ 4858 sp<SyncEvent> strongEvent = event.promote(); 4859 4860 if (strongEvent != 0) { 4861 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4862 me->handleSyncStartEvent(strongEvent); 4863 } 4864} 4865 4866void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4867{ 4868 if (event == mSyncStartEvent) { 4869 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4870 // from audio HAL 4871 mFramestoDrop = mFrameCount * 2; 4872 } 4873} 4874 4875bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4876 ALOGV("RecordThread::stop"); 4877 AutoMutex _l(mLock); 4878 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4879 return false; 4880 } 4881 recordTrack->mState = TrackBase::PAUSING; 4882 // do not wait for mStartStopCond if exiting 4883 if (exitPending()) { 4884 return true; 4885 } 4886 mStartStopCond.wait(mLock); 4887 // if we have been restarted, recordTrack == mActiveTrack.get() here 4888 if (exitPending() || recordTrack != mActiveTrack.get()) { 4889 ALOGV("Record stopped OK"); 4890 return true; 4891 } 4892 return false; 4893} 4894 4895bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4896{ 4897 return false; 4898} 4899 4900status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4901{ 4902#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4903 if (!isValidSyncEvent(event)) { 4904 return BAD_VALUE; 4905 } 4906 4907 int eventSession = event->triggerSession(); 4908 status_t ret = NAME_NOT_FOUND; 4909 4910 Mutex::Autolock _l(mLock); 4911 4912 for (size_t i = 0; i < mTracks.size(); i++) { 4913 sp<RecordTrack> track = mTracks[i]; 4914 if (eventSession == track->sessionId()) { 4915 (void) track->setSyncEvent(event); 4916 ret = NO_ERROR; 4917 } 4918 } 4919 return ret; 4920#else 4921 return BAD_VALUE; 4922#endif 4923} 4924 4925// destroyTrack_l() must be called with ThreadBase::mLock held 4926void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4927{ 4928 track->terminate(); 4929 track->mState = TrackBase::STOPPED; 4930 // active tracks are removed by threadLoop() 4931 if (mActiveTrack != track) { 4932 removeTrack_l(track); 4933 } 4934} 4935 4936void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4937{ 4938 mTracks.remove(track); 4939 // need anything related to effects here? 4940} 4941 4942void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4943{ 4944 dumpInternals(fd, args); 4945 dumpTracks(fd, args); 4946 dumpEffectChains(fd, args); 4947} 4948 4949void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4950{ 4951 const size_t SIZE = 256; 4952 char buffer[SIZE]; 4953 String8 result; 4954 4955 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4956 result.append(buffer); 4957 4958 if (mActiveTrack != 0) { 4959 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4960 result.append(buffer); 4961 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4962 result.append(buffer); 4963 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4964 result.append(buffer); 4965 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4966 result.append(buffer); 4967 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4968 result.append(buffer); 4969 } else { 4970 result.append("No active record client\n"); 4971 } 4972 4973 write(fd, result.string(), result.size()); 4974 4975 dumpBase(fd, args); 4976} 4977 4978void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4979{ 4980 const size_t SIZE = 256; 4981 char buffer[SIZE]; 4982 String8 result; 4983 4984 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4985 result.append(buffer); 4986 RecordTrack::appendDumpHeader(result); 4987 for (size_t i = 0; i < mTracks.size(); ++i) { 4988 sp<RecordTrack> track = mTracks[i]; 4989 if (track != 0) { 4990 track->dump(buffer, SIZE); 4991 result.append(buffer); 4992 } 4993 } 4994 4995 if (mActiveTrack != 0) { 4996 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4997 result.append(buffer); 4998 RecordTrack::appendDumpHeader(result); 4999 mActiveTrack->dump(buffer, SIZE); 5000 result.append(buffer); 5001 5002 } 5003 write(fd, result.string(), result.size()); 5004} 5005 5006// AudioBufferProvider interface 5007status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5008{ 5009 size_t framesReq = buffer->frameCount; 5010 size_t framesReady = mFrameCount - mRsmpInIndex; 5011 int channelCount; 5012 5013 if (framesReady == 0) { 5014 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 5015 if (mBytesRead <= 0) { 5016 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 5017 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5018 // Force input into standby so that it tries to 5019 // recover at next read attempt 5020 inputStandBy(); 5021 usleep(kRecordThreadSleepUs); 5022 } 5023 buffer->raw = NULL; 5024 buffer->frameCount = 0; 5025 return NOT_ENOUGH_DATA; 5026 } 5027 mRsmpInIndex = 0; 5028 framesReady = mFrameCount; 5029 } 5030 5031 if (framesReq > framesReady) { 5032 framesReq = framesReady; 5033 } 5034 5035 if (mChannelCount == 1 && mReqChannelCount == 2) { 5036 channelCount = 1; 5037 } else { 5038 channelCount = 2; 5039 } 5040 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5041 buffer->frameCount = framesReq; 5042 return NO_ERROR; 5043} 5044 5045// AudioBufferProvider interface 5046void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5047{ 5048 mRsmpInIndex += buffer->frameCount; 5049 buffer->frameCount = 0; 5050} 5051 5052bool AudioFlinger::RecordThread::checkForNewParameters_l() 5053{ 5054 bool reconfig = false; 5055 5056 while (!mNewParameters.isEmpty()) { 5057 status_t status = NO_ERROR; 5058 String8 keyValuePair = mNewParameters[0]; 5059 AudioParameter param = AudioParameter(keyValuePair); 5060 int value; 5061 audio_format_t reqFormat = mFormat; 5062 uint32_t reqSamplingRate = mReqSampleRate; 5063 uint32_t reqChannelCount = mReqChannelCount; 5064 5065 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5066 reqSamplingRate = value; 5067 reconfig = true; 5068 } 5069 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5070 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5071 status = BAD_VALUE; 5072 } else { 5073 reqFormat = (audio_format_t) value; 5074 reconfig = true; 5075 } 5076 } 5077 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5078 reqChannelCount = popcount(value); 5079 reconfig = true; 5080 } 5081 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5082 // do not accept frame count changes if tracks are open as the track buffer 5083 // size depends on frame count and correct behavior would not be guaranteed 5084 // if frame count is changed after track creation 5085 if (mActiveTrack != 0) { 5086 status = INVALID_OPERATION; 5087 } else { 5088 reconfig = true; 5089 } 5090 } 5091 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5092 // forward device change to effects that have requested to be 5093 // aware of attached audio device. 5094 for (size_t i = 0; i < mEffectChains.size(); i++) { 5095 mEffectChains[i]->setDevice_l(value); 5096 } 5097 5098 // store input device and output device but do not forward output device to audio HAL. 5099 // Note that status is ignored by the caller for output device 5100 // (see AudioFlinger::setParameters() 5101 if (audio_is_output_devices(value)) { 5102 mOutDevice = value; 5103 status = BAD_VALUE; 5104 } else { 5105 mInDevice = value; 5106 // disable AEC and NS if the device is a BT SCO headset supporting those 5107 // pre processings 5108 if (mTracks.size() > 0) { 5109 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5110 mAudioFlinger->btNrecIsOff(); 5111 for (size_t i = 0; i < mTracks.size(); i++) { 5112 sp<RecordTrack> track = mTracks[i]; 5113 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5114 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5115 } 5116 } 5117 } 5118 } 5119 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5120 mAudioSource != (audio_source_t)value) { 5121 // forward device change to effects that have requested to be 5122 // aware of attached audio device. 5123 for (size_t i = 0; i < mEffectChains.size(); i++) { 5124 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5125 } 5126 mAudioSource = (audio_source_t)value; 5127 } 5128 if (status == NO_ERROR) { 5129 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5130 keyValuePair.string()); 5131 if (status == INVALID_OPERATION) { 5132 inputStandBy(); 5133 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5134 keyValuePair.string()); 5135 } 5136 if (reconfig) { 5137 if (status == BAD_VALUE && 5138 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5139 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5140 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5141 <= (2 * reqSamplingRate)) && 5142 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5143 <= FCC_2 && 5144 (reqChannelCount <= FCC_2)) { 5145 status = NO_ERROR; 5146 } 5147 if (status == NO_ERROR) { 5148 readInputParameters(); 5149 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5150 } 5151 } 5152 } 5153 5154 mNewParameters.removeAt(0); 5155 5156 mParamStatus = status; 5157 mParamCond.signal(); 5158 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5159 // already timed out waiting for the status and will never signal the condition. 5160 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5161 } 5162 return reconfig; 5163} 5164 5165String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5166{ 5167 Mutex::Autolock _l(mLock); 5168 if (initCheck() != NO_ERROR) { 5169 return String8(); 5170 } 5171 5172 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5173 const String8 out_s8(s); 5174 free(s); 5175 return out_s8; 5176} 5177 5178void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5179 AudioSystem::OutputDescriptor desc; 5180 void *param2 = NULL; 5181 5182 switch (event) { 5183 case AudioSystem::INPUT_OPENED: 5184 case AudioSystem::INPUT_CONFIG_CHANGED: 5185 desc.channelMask = mChannelMask; 5186 desc.samplingRate = mSampleRate; 5187 desc.format = mFormat; 5188 desc.frameCount = mFrameCount; 5189 desc.latency = 0; 5190 param2 = &desc; 5191 break; 5192 5193 case AudioSystem::INPUT_CLOSED: 5194 default: 5195 break; 5196 } 5197 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5198} 5199 5200void AudioFlinger::RecordThread::readInputParameters() 5201{ 5202 delete[] mRsmpInBuffer; 5203 // mRsmpInBuffer is always assigned a new[] below 5204 delete[] mRsmpOutBuffer; 5205 mRsmpOutBuffer = NULL; 5206 delete mResampler; 5207 mResampler = NULL; 5208 5209 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5210 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5211 mChannelCount = popcount(mChannelMask); 5212 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5213 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5214 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5215 } 5216 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5217 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5218 mFrameCount = mBufferSize / mFrameSize; 5219 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5220 5221 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5222 { 5223 int channelCount; 5224 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5225 // stereo to mono post process as the resampler always outputs stereo. 5226 if (mChannelCount == 1 && mReqChannelCount == 2) { 5227 channelCount = 1; 5228 } else { 5229 channelCount = 2; 5230 } 5231 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5232 mResampler->setSampleRate(mSampleRate); 5233 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5234 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5235 5236 // optmization: if mono to mono, alter input frame count as if we were inputing 5237 // stereo samples 5238 if (mChannelCount == 1 && mReqChannelCount == 1) { 5239 mFrameCount >>= 1; 5240 } 5241 5242 } 5243 mRsmpInIndex = mFrameCount; 5244} 5245 5246unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5247{ 5248 Mutex::Autolock _l(mLock); 5249 if (initCheck() != NO_ERROR) { 5250 return 0; 5251 } 5252 5253 return mInput->stream->get_input_frames_lost(mInput->stream); 5254} 5255 5256uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5257{ 5258 Mutex::Autolock _l(mLock); 5259 uint32_t result = 0; 5260 if (getEffectChain_l(sessionId) != 0) { 5261 result = EFFECT_SESSION; 5262 } 5263 5264 for (size_t i = 0; i < mTracks.size(); ++i) { 5265 if (sessionId == mTracks[i]->sessionId()) { 5266 result |= TRACK_SESSION; 5267 break; 5268 } 5269 } 5270 5271 return result; 5272} 5273 5274KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5275{ 5276 KeyedVector<int, bool> ids; 5277 Mutex::Autolock _l(mLock); 5278 for (size_t j = 0; j < mTracks.size(); ++j) { 5279 sp<RecordThread::RecordTrack> track = mTracks[j]; 5280 int sessionId = track->sessionId(); 5281 if (ids.indexOfKey(sessionId) < 0) { 5282 ids.add(sessionId, true); 5283 } 5284 } 5285 return ids; 5286} 5287 5288AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5289{ 5290 Mutex::Autolock _l(mLock); 5291 AudioStreamIn *input = mInput; 5292 mInput = NULL; 5293 return input; 5294} 5295 5296// this method must always be called either with ThreadBase mLock held or inside the thread loop 5297audio_stream_t* AudioFlinger::RecordThread::stream() const 5298{ 5299 if (mInput == NULL) { 5300 return NULL; 5301 } 5302 return &mInput->stream->common; 5303} 5304 5305status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5306{ 5307 // only one chain per input thread 5308 if (mEffectChains.size() != 0) { 5309 return INVALID_OPERATION; 5310 } 5311 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5312 5313 chain->setInBuffer(NULL); 5314 chain->setOutBuffer(NULL); 5315 5316 checkSuspendOnAddEffectChain_l(chain); 5317 5318 mEffectChains.add(chain); 5319 5320 return NO_ERROR; 5321} 5322 5323size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5324{ 5325 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5326 ALOGW_IF(mEffectChains.size() != 1, 5327 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5328 chain.get(), mEffectChains.size(), this); 5329 if (mEffectChains.size() == 1) { 5330 mEffectChains.removeAt(0); 5331 } 5332 return 0; 5333} 5334 5335}; // namespace android 5336