Threads.cpp revision 377b2ec9a2885f9b6405b07ba900a9e3f4349c38
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: %zu\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: %zu\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 %02zu ", 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- %zu 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: %zu\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 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1328 lStatus = NO_MEMORY; 1329 goto Exit; 1330 } 1331 1332 mTracks.add(track); 1333 1334 sp<EffectChain> chain = getEffectChain_l(sessionId); 1335 if (chain != 0) { 1336 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1337 track->setMainBuffer(chain->inBuffer()); 1338 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1339 chain->incTrackCnt(); 1340 } 1341 1342 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1343 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1344 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1345 // so ask activity manager to do this on our behalf 1346 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1347 } 1348 } 1349 1350 lStatus = NO_ERROR; 1351 1352Exit: 1353 if (status) { 1354 *status = lStatus; 1355 } 1356 return track; 1357} 1358 1359uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1360{ 1361 return latency; 1362} 1363 1364uint32_t AudioFlinger::PlaybackThread::latency() const 1365{ 1366 Mutex::Autolock _l(mLock); 1367 return latency_l(); 1368} 1369uint32_t AudioFlinger::PlaybackThread::latency_l() const 1370{ 1371 if (initCheck() == NO_ERROR) { 1372 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1373 } else { 1374 return 0; 1375 } 1376} 1377 1378void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1379{ 1380 Mutex::Autolock _l(mLock); 1381 // Don't apply master volume in SW if our HAL can do it for us. 1382 if (mOutput && mOutput->audioHwDev && 1383 mOutput->audioHwDev->canSetMasterVolume()) { 1384 mMasterVolume = 1.0; 1385 } else { 1386 mMasterVolume = value; 1387 } 1388} 1389 1390void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1391{ 1392 Mutex::Autolock _l(mLock); 1393 // Don't apply master mute in SW if our HAL can do it for us. 1394 if (mOutput && mOutput->audioHwDev && 1395 mOutput->audioHwDev->canSetMasterMute()) { 1396 mMasterMute = false; 1397 } else { 1398 mMasterMute = muted; 1399 } 1400} 1401 1402void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1403{ 1404 Mutex::Autolock _l(mLock); 1405 mStreamTypes[stream].volume = value; 1406 broadcast_l(); 1407} 1408 1409void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1410{ 1411 Mutex::Autolock _l(mLock); 1412 mStreamTypes[stream].mute = muted; 1413 broadcast_l(); 1414} 1415 1416float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1417{ 1418 Mutex::Autolock _l(mLock); 1419 return mStreamTypes[stream].volume; 1420} 1421 1422// addTrack_l() must be called with ThreadBase::mLock held 1423status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1424{ 1425 status_t status = ALREADY_EXISTS; 1426 1427 // set retry count for buffer fill 1428 track->mRetryCount = kMaxTrackStartupRetries; 1429 if (mActiveTracks.indexOf(track) < 0) { 1430 // the track is newly added, make sure it fills up all its 1431 // buffers before playing. This is to ensure the client will 1432 // effectively get the latency it requested. 1433 if (!track->isOutputTrack()) { 1434 TrackBase::track_state state = track->mState; 1435 mLock.unlock(); 1436 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1437 mLock.lock(); 1438 // abort track was stopped/paused while we released the lock 1439 if (state != track->mState) { 1440 if (status == NO_ERROR) { 1441 mLock.unlock(); 1442 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1443 mLock.lock(); 1444 } 1445 return INVALID_OPERATION; 1446 } 1447 // abort if start is rejected by audio policy manager 1448 if (status != NO_ERROR) { 1449 return PERMISSION_DENIED; 1450 } 1451#ifdef ADD_BATTERY_DATA 1452 // to track the speaker usage 1453 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1454#endif 1455 } 1456 1457 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1458 track->mResetDone = false; 1459 track->mPresentationCompleteFrames = 0; 1460 mActiveTracks.add(track); 1461 mWakeLockUids.add(track->uid()); 1462 mActiveTracksGeneration++; 1463 mLatestActiveTrack = track; 1464 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1465 if (chain != 0) { 1466 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1467 track->sessionId()); 1468 chain->incActiveTrackCnt(); 1469 } 1470 1471 status = NO_ERROR; 1472 } 1473 1474 ALOGV("signal playback thread"); 1475 broadcast_l(); 1476 1477 return status; 1478} 1479 1480bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1481{ 1482 track->terminate(); 1483 // active tracks are removed by threadLoop() 1484 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1485 track->mState = TrackBase::STOPPED; 1486 if (!trackActive) { 1487 removeTrack_l(track); 1488 } else if (track->isFastTrack() || track->isOffloaded()) { 1489 track->mState = TrackBase::STOPPING_1; 1490 } 1491 1492 return trackActive; 1493} 1494 1495void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1496{ 1497 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1498 mTracks.remove(track); 1499 deleteTrackName_l(track->name()); 1500 // redundant as track is about to be destroyed, for dumpsys only 1501 track->mName = -1; 1502 if (track->isFastTrack()) { 1503 int index = track->mFastIndex; 1504 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1505 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1506 mFastTrackAvailMask |= 1 << index; 1507 // redundant as track is about to be destroyed, for dumpsys only 1508 track->mFastIndex = -1; 1509 } 1510 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1511 if (chain != 0) { 1512 chain->decTrackCnt(); 1513 } 1514} 1515 1516void AudioFlinger::PlaybackThread::broadcast_l() 1517{ 1518 // Thread could be blocked waiting for async 1519 // so signal it to handle state changes immediately 1520 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1521 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1522 mSignalPending = true; 1523 mWaitWorkCV.broadcast(); 1524} 1525 1526String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1527{ 1528 Mutex::Autolock _l(mLock); 1529 if (initCheck() != NO_ERROR) { 1530 return String8(); 1531 } 1532 1533 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1534 const String8 out_s8(s); 1535 free(s); 1536 return out_s8; 1537} 1538 1539// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1540void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1541 AudioSystem::OutputDescriptor desc; 1542 void *param2 = NULL; 1543 1544 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1545 param); 1546 1547 switch (event) { 1548 case AudioSystem::OUTPUT_OPENED: 1549 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1550 desc.channelMask = mChannelMask; 1551 desc.samplingRate = mSampleRate; 1552 desc.format = mFormat; 1553 desc.frameCount = mNormalFrameCount; // FIXME see 1554 // AudioFlinger::frameCount(audio_io_handle_t) 1555 desc.latency = latency(); 1556 param2 = &desc; 1557 break; 1558 1559 case AudioSystem::STREAM_CONFIG_CHANGED: 1560 param2 = ¶m; 1561 case AudioSystem::OUTPUT_CLOSED: 1562 default: 1563 break; 1564 } 1565 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1566} 1567 1568void AudioFlinger::PlaybackThread::writeCallback() 1569{ 1570 ALOG_ASSERT(mCallbackThread != 0); 1571 mCallbackThread->resetWriteBlocked(); 1572} 1573 1574void AudioFlinger::PlaybackThread::drainCallback() 1575{ 1576 ALOG_ASSERT(mCallbackThread != 0); 1577 mCallbackThread->resetDraining(); 1578} 1579 1580void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1581{ 1582 Mutex::Autolock _l(mLock); 1583 // reject out of sequence requests 1584 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1585 mWriteAckSequence &= ~1; 1586 mWaitWorkCV.signal(); 1587 } 1588} 1589 1590void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1591{ 1592 Mutex::Autolock _l(mLock); 1593 // reject out of sequence requests 1594 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1595 mDrainSequence &= ~1; 1596 mWaitWorkCV.signal(); 1597 } 1598} 1599 1600// static 1601int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1602 void *param, 1603 void *cookie) 1604{ 1605 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1606 ALOGV("asyncCallback() event %d", event); 1607 switch (event) { 1608 case STREAM_CBK_EVENT_WRITE_READY: 1609 me->writeCallback(); 1610 break; 1611 case STREAM_CBK_EVENT_DRAIN_READY: 1612 me->drainCallback(); 1613 break; 1614 default: 1615 ALOGW("asyncCallback() unknown event %d", event); 1616 break; 1617 } 1618 return 0; 1619} 1620 1621void AudioFlinger::PlaybackThread::readOutputParameters() 1622{ 1623 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1624 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1625 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1626 if (!audio_is_output_channel(mChannelMask)) { 1627 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1628 } 1629 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1630 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1631 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1632 } 1633 mChannelCount = popcount(mChannelMask); 1634 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1635 if (!audio_is_valid_format(mFormat)) { 1636 LOG_FATAL("HAL format %d not valid for output", mFormat); 1637 } 1638 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1639 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1640 mFormat); 1641 } 1642 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1643 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1644 if (mFrameCount & 15) { 1645 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1646 mFrameCount); 1647 } 1648 1649 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1650 (mOutput->stream->set_callback != NULL)) { 1651 if (mOutput->stream->set_callback(mOutput->stream, 1652 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1653 mUseAsyncWrite = true; 1654 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1655 } 1656 } 1657 1658 // Calculate size of normal mix buffer relative to the HAL output buffer size 1659 double multiplier = 1.0; 1660 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1661 kUseFastMixer == FastMixer_Dynamic)) { 1662 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1663 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1664 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1665 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1666 maxNormalFrameCount = maxNormalFrameCount & ~15; 1667 if (maxNormalFrameCount < minNormalFrameCount) { 1668 maxNormalFrameCount = minNormalFrameCount; 1669 } 1670 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1671 if (multiplier <= 1.0) { 1672 multiplier = 1.0; 1673 } else if (multiplier <= 2.0) { 1674 if (2 * mFrameCount <= maxNormalFrameCount) { 1675 multiplier = 2.0; 1676 } else { 1677 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1678 } 1679 } else { 1680 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1681 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1682 // track, but we sometimes have to do this to satisfy the maximum frame count 1683 // constraint) 1684 // FIXME this rounding up should not be done if no HAL SRC 1685 uint32_t truncMult = (uint32_t) multiplier; 1686 if ((truncMult & 1)) { 1687 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1688 ++truncMult; 1689 } 1690 } 1691 multiplier = (double) truncMult; 1692 } 1693 } 1694 mNormalFrameCount = multiplier * mFrameCount; 1695 // round up to nearest 16 frames to satisfy AudioMixer 1696 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1697 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1698 mNormalFrameCount); 1699 1700 delete[] mAllocMixBuffer; 1701 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1702 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1703 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1704 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1705 1706 // force reconfiguration of effect chains and engines to take new buffer size and audio 1707 // parameters into account 1708 // Note that mLock is not held when readOutputParameters() is called from the constructor 1709 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1710 // matter. 1711 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1712 Vector< sp<EffectChain> > effectChains = mEffectChains; 1713 for (size_t i = 0; i < effectChains.size(); i ++) { 1714 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1715 } 1716} 1717 1718 1719status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1720{ 1721 if (halFrames == NULL || dspFrames == NULL) { 1722 return BAD_VALUE; 1723 } 1724 Mutex::Autolock _l(mLock); 1725 if (initCheck() != NO_ERROR) { 1726 return INVALID_OPERATION; 1727 } 1728 size_t framesWritten = mBytesWritten / mFrameSize; 1729 *halFrames = framesWritten; 1730 1731 if (isSuspended()) { 1732 // return an estimation of rendered frames when the output is suspended 1733 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1734 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1735 return NO_ERROR; 1736 } else { 1737 status_t status; 1738 uint32_t frames; 1739 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1740 *dspFrames = (size_t)frames; 1741 return status; 1742 } 1743} 1744 1745uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1746{ 1747 Mutex::Autolock _l(mLock); 1748 uint32_t result = 0; 1749 if (getEffectChain_l(sessionId) != 0) { 1750 result = EFFECT_SESSION; 1751 } 1752 1753 for (size_t i = 0; i < mTracks.size(); ++i) { 1754 sp<Track> track = mTracks[i]; 1755 if (sessionId == track->sessionId() && !track->isInvalid()) { 1756 result |= TRACK_SESSION; 1757 break; 1758 } 1759 } 1760 1761 return result; 1762} 1763 1764uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1765{ 1766 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1767 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1768 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1769 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1770 } 1771 for (size_t i = 0; i < mTracks.size(); i++) { 1772 sp<Track> track = mTracks[i]; 1773 if (sessionId == track->sessionId() && !track->isInvalid()) { 1774 return AudioSystem::getStrategyForStream(track->streamType()); 1775 } 1776 } 1777 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1778} 1779 1780 1781AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1782{ 1783 Mutex::Autolock _l(mLock); 1784 return mOutput; 1785} 1786 1787AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1788{ 1789 Mutex::Autolock _l(mLock); 1790 AudioStreamOut *output = mOutput; 1791 mOutput = NULL; 1792 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1793 // must push a NULL and wait for ack 1794 mOutputSink.clear(); 1795 mPipeSink.clear(); 1796 mNormalSink.clear(); 1797 return output; 1798} 1799 1800// this method must always be called either with ThreadBase mLock held or inside the thread loop 1801audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1802{ 1803 if (mOutput == NULL) { 1804 return NULL; 1805 } 1806 return &mOutput->stream->common; 1807} 1808 1809uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1810{ 1811 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1812} 1813 1814status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1815{ 1816 if (!isValidSyncEvent(event)) { 1817 return BAD_VALUE; 1818 } 1819 1820 Mutex::Autolock _l(mLock); 1821 1822 for (size_t i = 0; i < mTracks.size(); ++i) { 1823 sp<Track> track = mTracks[i]; 1824 if (event->triggerSession() == track->sessionId()) { 1825 (void) track->setSyncEvent(event); 1826 return NO_ERROR; 1827 } 1828 } 1829 1830 return NAME_NOT_FOUND; 1831} 1832 1833bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1834{ 1835 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1836} 1837 1838void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1839 const Vector< sp<Track> >& tracksToRemove) 1840{ 1841 size_t count = tracksToRemove.size(); 1842 if (count) { 1843 for (size_t i = 0 ; i < count ; i++) { 1844 const sp<Track>& track = tracksToRemove.itemAt(i); 1845 if (!track->isOutputTrack()) { 1846 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1847#ifdef ADD_BATTERY_DATA 1848 // to track the speaker usage 1849 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1850#endif 1851 if (track->isTerminated()) { 1852 AudioSystem::releaseOutput(mId); 1853 } 1854 } 1855 } 1856 } 1857} 1858 1859void AudioFlinger::PlaybackThread::checkSilentMode_l() 1860{ 1861 if (!mMasterMute) { 1862 char value[PROPERTY_VALUE_MAX]; 1863 if (property_get("ro.audio.silent", value, "0") > 0) { 1864 char *endptr; 1865 unsigned long ul = strtoul(value, &endptr, 0); 1866 if (*endptr == '\0' && ul != 0) { 1867 ALOGD("Silence is golden"); 1868 // The setprop command will not allow a property to be changed after 1869 // the first time it is set, so we don't have to worry about un-muting. 1870 setMasterMute_l(true); 1871 } 1872 } 1873 } 1874} 1875 1876// shared by MIXER and DIRECT, overridden by DUPLICATING 1877ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1878{ 1879 // FIXME rewrite to reduce number of system calls 1880 mLastWriteTime = systemTime(); 1881 mInWrite = true; 1882 ssize_t bytesWritten; 1883 1884 // If an NBAIO sink is present, use it to write the normal mixer's submix 1885 if (mNormalSink != 0) { 1886#define mBitShift 2 // FIXME 1887 size_t count = mBytesRemaining >> mBitShift; 1888 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1889 ATRACE_BEGIN("write"); 1890 // update the setpoint when AudioFlinger::mScreenState changes 1891 uint32_t screenState = AudioFlinger::mScreenState; 1892 if (screenState != mScreenState) { 1893 mScreenState = screenState; 1894 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1895 if (pipe != NULL) { 1896 pipe->setAvgFrames((mScreenState & 1) ? 1897 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1898 } 1899 } 1900 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1901 ATRACE_END(); 1902 if (framesWritten > 0) { 1903 bytesWritten = framesWritten << mBitShift; 1904 } else { 1905 bytesWritten = framesWritten; 1906 } 1907 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1908 if (status == NO_ERROR) { 1909 size_t totalFramesWritten = mNormalSink->framesWritten(); 1910 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1911 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1912 mLatchDValid = true; 1913 } 1914 } 1915 // otherwise use the HAL / AudioStreamOut directly 1916 } else { 1917 // Direct output and offload threads 1918 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1919 if (mUseAsyncWrite) { 1920 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1921 mWriteAckSequence += 2; 1922 mWriteAckSequence |= 1; 1923 ALOG_ASSERT(mCallbackThread != 0); 1924 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1925 } 1926 // FIXME We should have an implementation of timestamps for direct output threads. 1927 // They are used e.g for multichannel PCM playback over HDMI. 1928 bytesWritten = mOutput->stream->write(mOutput->stream, 1929 mMixBuffer + offset, mBytesRemaining); 1930 if (mUseAsyncWrite && 1931 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1932 // do not wait for async callback in case of error of full write 1933 mWriteAckSequence &= ~1; 1934 ALOG_ASSERT(mCallbackThread != 0); 1935 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1936 } 1937 } 1938 1939 mNumWrites++; 1940 mInWrite = false; 1941 mStandby = false; 1942 return bytesWritten; 1943} 1944 1945void AudioFlinger::PlaybackThread::threadLoop_drain() 1946{ 1947 if (mOutput->stream->drain) { 1948 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1949 if (mUseAsyncWrite) { 1950 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1951 mDrainSequence |= 1; 1952 ALOG_ASSERT(mCallbackThread != 0); 1953 mCallbackThread->setDraining(mDrainSequence); 1954 } 1955 mOutput->stream->drain(mOutput->stream, 1956 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1957 : AUDIO_DRAIN_ALL); 1958 } 1959} 1960 1961void AudioFlinger::PlaybackThread::threadLoop_exit() 1962{ 1963 // Default implementation has nothing to do 1964} 1965 1966/* 1967The derived values that are cached: 1968 - mixBufferSize from frame count * frame size 1969 - activeSleepTime from activeSleepTimeUs() 1970 - idleSleepTime from idleSleepTimeUs() 1971 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1972 - maxPeriod from frame count and sample rate (MIXER only) 1973 1974The parameters that affect these derived values are: 1975 - frame count 1976 - frame size 1977 - sample rate 1978 - device type: A2DP or not 1979 - device latency 1980 - format: PCM or not 1981 - active sleep time 1982 - idle sleep time 1983*/ 1984 1985void AudioFlinger::PlaybackThread::cacheParameters_l() 1986{ 1987 mixBufferSize = mNormalFrameCount * mFrameSize; 1988 activeSleepTime = activeSleepTimeUs(); 1989 idleSleepTime = idleSleepTimeUs(); 1990} 1991 1992void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1993{ 1994 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1995 this, streamType, mTracks.size()); 1996 Mutex::Autolock _l(mLock); 1997 1998 size_t size = mTracks.size(); 1999 for (size_t i = 0; i < size; i++) { 2000 sp<Track> t = mTracks[i]; 2001 if (t->streamType() == streamType) { 2002 t->invalidate(); 2003 } 2004 } 2005} 2006 2007status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2008{ 2009 int session = chain->sessionId(); 2010 int16_t *buffer = mMixBuffer; 2011 bool ownsBuffer = false; 2012 2013 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2014 if (session > 0) { 2015 // Only one effect chain can be present in direct output thread and it uses 2016 // the mix buffer as input 2017 if (mType != DIRECT) { 2018 size_t numSamples = mNormalFrameCount * mChannelCount; 2019 buffer = new int16_t[numSamples]; 2020 memset(buffer, 0, numSamples * sizeof(int16_t)); 2021 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2022 ownsBuffer = true; 2023 } 2024 2025 // Attach all tracks with same session ID to this chain. 2026 for (size_t i = 0; i < mTracks.size(); ++i) { 2027 sp<Track> track = mTracks[i]; 2028 if (session == track->sessionId()) { 2029 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2030 buffer); 2031 track->setMainBuffer(buffer); 2032 chain->incTrackCnt(); 2033 } 2034 } 2035 2036 // indicate all active tracks in the chain 2037 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2038 sp<Track> track = mActiveTracks[i].promote(); 2039 if (track == 0) { 2040 continue; 2041 } 2042 if (session == track->sessionId()) { 2043 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2044 chain->incActiveTrackCnt(); 2045 } 2046 } 2047 } 2048 2049 chain->setInBuffer(buffer, ownsBuffer); 2050 chain->setOutBuffer(mMixBuffer); 2051 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2052 // chains list in order to be processed last as it contains output stage effects 2053 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2054 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2055 // after track specific effects and before output stage 2056 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2057 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2058 // Effect chain for other sessions are inserted at beginning of effect 2059 // chains list to be processed before output mix effects. Relative order between other 2060 // sessions is not important 2061 size_t size = mEffectChains.size(); 2062 size_t i = 0; 2063 for (i = 0; i < size; i++) { 2064 if (mEffectChains[i]->sessionId() < session) { 2065 break; 2066 } 2067 } 2068 mEffectChains.insertAt(chain, i); 2069 checkSuspendOnAddEffectChain_l(chain); 2070 2071 return NO_ERROR; 2072} 2073 2074size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2075{ 2076 int session = chain->sessionId(); 2077 2078 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2079 2080 for (size_t i = 0; i < mEffectChains.size(); i++) { 2081 if (chain == mEffectChains[i]) { 2082 mEffectChains.removeAt(i); 2083 // detach all active tracks from the chain 2084 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2085 sp<Track> track = mActiveTracks[i].promote(); 2086 if (track == 0) { 2087 continue; 2088 } 2089 if (session == track->sessionId()) { 2090 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2091 chain.get(), session); 2092 chain->decActiveTrackCnt(); 2093 } 2094 } 2095 2096 // detach all tracks with same session ID from this chain 2097 for (size_t i = 0; i < mTracks.size(); ++i) { 2098 sp<Track> track = mTracks[i]; 2099 if (session == track->sessionId()) { 2100 track->setMainBuffer(mMixBuffer); 2101 chain->decTrackCnt(); 2102 } 2103 } 2104 break; 2105 } 2106 } 2107 return mEffectChains.size(); 2108} 2109 2110status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2111 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2112{ 2113 Mutex::Autolock _l(mLock); 2114 return attachAuxEffect_l(track, EffectId); 2115} 2116 2117status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2118 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2119{ 2120 status_t status = NO_ERROR; 2121 2122 if (EffectId == 0) { 2123 track->setAuxBuffer(0, NULL); 2124 } else { 2125 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2126 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2127 if (effect != 0) { 2128 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2129 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2130 } else { 2131 status = INVALID_OPERATION; 2132 } 2133 } else { 2134 status = BAD_VALUE; 2135 } 2136 } 2137 return status; 2138} 2139 2140void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2141{ 2142 for (size_t i = 0; i < mTracks.size(); ++i) { 2143 sp<Track> track = mTracks[i]; 2144 if (track->auxEffectId() == effectId) { 2145 attachAuxEffect_l(track, 0); 2146 } 2147 } 2148} 2149 2150bool AudioFlinger::PlaybackThread::threadLoop() 2151{ 2152 Vector< sp<Track> > tracksToRemove; 2153 2154 standbyTime = systemTime(); 2155 2156 // MIXER 2157 nsecs_t lastWarning = 0; 2158 2159 // DUPLICATING 2160 // FIXME could this be made local to while loop? 2161 writeFrames = 0; 2162 2163 int lastGeneration = 0; 2164 2165 cacheParameters_l(); 2166 sleepTime = idleSleepTime; 2167 2168 if (mType == MIXER) { 2169 sleepTimeShift = 0; 2170 } 2171 2172 CpuStats cpuStats; 2173 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2174 2175 acquireWakeLock(); 2176 2177 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2178 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2179 // and then that string will be logged at the next convenient opportunity. 2180 const char *logString = NULL; 2181 2182 checkSilentMode_l(); 2183 2184 while (!exitPending()) 2185 { 2186 cpuStats.sample(myName); 2187 2188 Vector< sp<EffectChain> > effectChains; 2189 2190 processConfigEvents(); 2191 2192 { // scope for mLock 2193 2194 Mutex::Autolock _l(mLock); 2195 2196 if (logString != NULL) { 2197 mNBLogWriter->logTimestamp(); 2198 mNBLogWriter->log(logString); 2199 logString = NULL; 2200 } 2201 2202 if (mLatchDValid) { 2203 mLatchQ = mLatchD; 2204 mLatchDValid = false; 2205 mLatchQValid = true; 2206 } 2207 2208 if (checkForNewParameters_l()) { 2209 cacheParameters_l(); 2210 } 2211 2212 saveOutputTracks(); 2213 if (mSignalPending) { 2214 // A signal was raised while we were unlocked 2215 mSignalPending = false; 2216 } else if (waitingAsyncCallback_l()) { 2217 if (exitPending()) { 2218 break; 2219 } 2220 releaseWakeLock_l(); 2221 mWakeLockUids.clear(); 2222 mActiveTracksGeneration++; 2223 ALOGV("wait async completion"); 2224 mWaitWorkCV.wait(mLock); 2225 ALOGV("async completion/wake"); 2226 acquireWakeLock_l(); 2227 standbyTime = systemTime() + standbyDelay; 2228 sleepTime = 0; 2229 2230 continue; 2231 } 2232 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2233 isSuspended()) { 2234 // put audio hardware into standby after short delay 2235 if (shouldStandby_l()) { 2236 2237 threadLoop_standby(); 2238 2239 mStandby = true; 2240 } 2241 2242 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2243 // we're about to wait, flush the binder command buffer 2244 IPCThreadState::self()->flushCommands(); 2245 2246 clearOutputTracks(); 2247 2248 if (exitPending()) { 2249 break; 2250 } 2251 2252 releaseWakeLock_l(); 2253 mWakeLockUids.clear(); 2254 mActiveTracksGeneration++; 2255 // wait until we have something to do... 2256 ALOGV("%s going to sleep", myName.string()); 2257 mWaitWorkCV.wait(mLock); 2258 ALOGV("%s waking up", myName.string()); 2259 acquireWakeLock_l(); 2260 2261 mMixerStatus = MIXER_IDLE; 2262 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2263 mBytesWritten = 0; 2264 mBytesRemaining = 0; 2265 checkSilentMode_l(); 2266 2267 standbyTime = systemTime() + standbyDelay; 2268 sleepTime = idleSleepTime; 2269 if (mType == MIXER) { 2270 sleepTimeShift = 0; 2271 } 2272 2273 continue; 2274 } 2275 } 2276 // mMixerStatusIgnoringFastTracks is also updated internally 2277 mMixerStatus = prepareTracks_l(&tracksToRemove); 2278 2279 // compare with previously applied list 2280 if (lastGeneration != mActiveTracksGeneration) { 2281 // update wakelock 2282 updateWakeLockUids_l(mWakeLockUids); 2283 lastGeneration = mActiveTracksGeneration; 2284 } 2285 2286 // prevent any changes in effect chain list and in each effect chain 2287 // during mixing and effect process as the audio buffers could be deleted 2288 // or modified if an effect is created or deleted 2289 lockEffectChains_l(effectChains); 2290 } // mLock scope ends 2291 2292 if (mBytesRemaining == 0) { 2293 mCurrentWriteLength = 0; 2294 if (mMixerStatus == MIXER_TRACKS_READY) { 2295 // threadLoop_mix() sets mCurrentWriteLength 2296 threadLoop_mix(); 2297 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2298 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2299 // threadLoop_sleepTime sets sleepTime to 0 if data 2300 // must be written to HAL 2301 threadLoop_sleepTime(); 2302 if (sleepTime == 0) { 2303 mCurrentWriteLength = mixBufferSize; 2304 } 2305 } 2306 mBytesRemaining = mCurrentWriteLength; 2307 if (isSuspended()) { 2308 sleepTime = suspendSleepTimeUs(); 2309 // simulate write to HAL when suspended 2310 mBytesWritten += mixBufferSize; 2311 mBytesRemaining = 0; 2312 } 2313 2314 // only process effects if we're going to write 2315 if (sleepTime == 0 && mType != OFFLOAD) { 2316 for (size_t i = 0; i < effectChains.size(); i ++) { 2317 effectChains[i]->process_l(); 2318 } 2319 } 2320 } 2321 // Process effect chains for offloaded thread even if no audio 2322 // was read from audio track: process only updates effect state 2323 // and thus does have to be synchronized with audio writes but may have 2324 // to be called while waiting for async write callback 2325 if (mType == OFFLOAD) { 2326 for (size_t i = 0; i < effectChains.size(); i ++) { 2327 effectChains[i]->process_l(); 2328 } 2329 } 2330 2331 // enable changes in effect chain 2332 unlockEffectChains(effectChains); 2333 2334 if (!waitingAsyncCallback()) { 2335 // sleepTime == 0 means we must write to audio hardware 2336 if (sleepTime == 0) { 2337 if (mBytesRemaining) { 2338 ssize_t ret = threadLoop_write(); 2339 if (ret < 0) { 2340 mBytesRemaining = 0; 2341 } else { 2342 mBytesWritten += ret; 2343 mBytesRemaining -= ret; 2344 } 2345 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2346 (mMixerStatus == MIXER_DRAIN_ALL)) { 2347 threadLoop_drain(); 2348 } 2349if (mType == MIXER) { 2350 // write blocked detection 2351 nsecs_t now = systemTime(); 2352 nsecs_t delta = now - mLastWriteTime; 2353 if (!mStandby && delta > maxPeriod) { 2354 mNumDelayedWrites++; 2355 if ((now - lastWarning) > kWarningThrottleNs) { 2356 ATRACE_NAME("underrun"); 2357 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2358 ns2ms(delta), mNumDelayedWrites, this); 2359 lastWarning = now; 2360 } 2361 } 2362} 2363 2364 } else { 2365 usleep(sleepTime); 2366 } 2367 } 2368 2369 // Finally let go of removed track(s), without the lock held 2370 // since we can't guarantee the destructors won't acquire that 2371 // same lock. This will also mutate and push a new fast mixer state. 2372 threadLoop_removeTracks(tracksToRemove); 2373 tracksToRemove.clear(); 2374 2375 // FIXME I don't understand the need for this here; 2376 // it was in the original code but maybe the 2377 // assignment in saveOutputTracks() makes this unnecessary? 2378 clearOutputTracks(); 2379 2380 // Effect chains will be actually deleted here if they were removed from 2381 // mEffectChains list during mixing or effects processing 2382 effectChains.clear(); 2383 2384 // FIXME Note that the above .clear() is no longer necessary since effectChains 2385 // is now local to this block, but will keep it for now (at least until merge done). 2386 } 2387 2388 threadLoop_exit(); 2389 2390 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2391 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2392 // put output stream into standby mode 2393 if (!mStandby) { 2394 mOutput->stream->common.standby(&mOutput->stream->common); 2395 } 2396 } 2397 2398 releaseWakeLock(); 2399 mWakeLockUids.clear(); 2400 mActiveTracksGeneration++; 2401 2402 ALOGV("Thread %p type %d exiting", this, mType); 2403 return false; 2404} 2405 2406// removeTracks_l() must be called with ThreadBase::mLock held 2407void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2408{ 2409 size_t count = tracksToRemove.size(); 2410 if (count) { 2411 for (size_t i=0 ; i<count ; i++) { 2412 const sp<Track>& track = tracksToRemove.itemAt(i); 2413 mActiveTracks.remove(track); 2414 mWakeLockUids.remove(track->uid()); 2415 mActiveTracksGeneration++; 2416 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2417 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2418 if (chain != 0) { 2419 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2420 track->sessionId()); 2421 chain->decActiveTrackCnt(); 2422 } 2423 if (track->isTerminated()) { 2424 removeTrack_l(track); 2425 } 2426 } 2427 } 2428 2429} 2430 2431status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2432{ 2433 if (mNormalSink != 0) { 2434 return mNormalSink->getTimestamp(timestamp); 2435 } 2436 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2437 uint64_t position64; 2438 int ret = mOutput->stream->get_presentation_position( 2439 mOutput->stream, &position64, ×tamp.mTime); 2440 if (ret == 0) { 2441 timestamp.mPosition = (uint32_t)position64; 2442 return NO_ERROR; 2443 } 2444 } 2445 return INVALID_OPERATION; 2446} 2447// ---------------------------------------------------------------------------- 2448 2449AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2450 audio_io_handle_t id, audio_devices_t device, type_t type) 2451 : PlaybackThread(audioFlinger, output, id, device, type), 2452 // mAudioMixer below 2453 // mFastMixer below 2454 mFastMixerFutex(0) 2455 // mOutputSink below 2456 // mPipeSink below 2457 // mNormalSink below 2458{ 2459 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2460 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2461 "mFrameCount=%d, mNormalFrameCount=%d", 2462 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2463 mNormalFrameCount); 2464 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2465 2466 // FIXME - Current mixer implementation only supports stereo output 2467 if (mChannelCount != FCC_2) { 2468 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2469 } 2470 2471 // create an NBAIO sink for the HAL output stream, and negotiate 2472 mOutputSink = new AudioStreamOutSink(output->stream); 2473 size_t numCounterOffers = 0; 2474 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2475 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2476 ALOG_ASSERT(index == 0); 2477 2478 // initialize fast mixer depending on configuration 2479 bool initFastMixer; 2480 switch (kUseFastMixer) { 2481 case FastMixer_Never: 2482 initFastMixer = false; 2483 break; 2484 case FastMixer_Always: 2485 initFastMixer = true; 2486 break; 2487 case FastMixer_Static: 2488 case FastMixer_Dynamic: 2489 initFastMixer = mFrameCount < mNormalFrameCount; 2490 break; 2491 } 2492 if (initFastMixer) { 2493 2494 // create a MonoPipe to connect our submix to FastMixer 2495 NBAIO_Format format = mOutputSink->format(); 2496 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2497 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2498 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2499 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2500 const NBAIO_Format offers[1] = {format}; 2501 size_t numCounterOffers = 0; 2502 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2503 ALOG_ASSERT(index == 0); 2504 monoPipe->setAvgFrames((mScreenState & 1) ? 2505 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2506 mPipeSink = monoPipe; 2507 2508#ifdef TEE_SINK 2509 if (mTeeSinkOutputEnabled) { 2510 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2511 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2512 numCounterOffers = 0; 2513 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2514 ALOG_ASSERT(index == 0); 2515 mTeeSink = teeSink; 2516 PipeReader *teeSource = new PipeReader(*teeSink); 2517 numCounterOffers = 0; 2518 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2519 ALOG_ASSERT(index == 0); 2520 mTeeSource = teeSource; 2521 } 2522#endif 2523 2524 // create fast mixer and configure it initially with just one fast track for our submix 2525 mFastMixer = new FastMixer(); 2526 FastMixerStateQueue *sq = mFastMixer->sq(); 2527#ifdef STATE_QUEUE_DUMP 2528 sq->setObserverDump(&mStateQueueObserverDump); 2529 sq->setMutatorDump(&mStateQueueMutatorDump); 2530#endif 2531 FastMixerState *state = sq->begin(); 2532 FastTrack *fastTrack = &state->mFastTracks[0]; 2533 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2534 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2535 fastTrack->mVolumeProvider = NULL; 2536 fastTrack->mGeneration++; 2537 state->mFastTracksGen++; 2538 state->mTrackMask = 1; 2539 // fast mixer will use the HAL output sink 2540 state->mOutputSink = mOutputSink.get(); 2541 state->mOutputSinkGen++; 2542 state->mFrameCount = mFrameCount; 2543 state->mCommand = FastMixerState::COLD_IDLE; 2544 // already done in constructor initialization list 2545 //mFastMixerFutex = 0; 2546 state->mColdFutexAddr = &mFastMixerFutex; 2547 state->mColdGen++; 2548 state->mDumpState = &mFastMixerDumpState; 2549#ifdef TEE_SINK 2550 state->mTeeSink = mTeeSink.get(); 2551#endif 2552 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2553 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2554 sq->end(); 2555 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2556 2557 // start the fast mixer 2558 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2559 pid_t tid = mFastMixer->getTid(); 2560 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2561 if (err != 0) { 2562 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2563 kPriorityFastMixer, getpid_cached, tid, err); 2564 } 2565 2566#ifdef AUDIO_WATCHDOG 2567 // create and start the watchdog 2568 mAudioWatchdog = new AudioWatchdog(); 2569 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2570 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2571 tid = mAudioWatchdog->getTid(); 2572 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2573 if (err != 0) { 2574 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2575 kPriorityFastMixer, getpid_cached, tid, err); 2576 } 2577#endif 2578 2579 } else { 2580 mFastMixer = NULL; 2581 } 2582 2583 switch (kUseFastMixer) { 2584 case FastMixer_Never: 2585 case FastMixer_Dynamic: 2586 mNormalSink = mOutputSink; 2587 break; 2588 case FastMixer_Always: 2589 mNormalSink = mPipeSink; 2590 break; 2591 case FastMixer_Static: 2592 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2593 break; 2594 } 2595} 2596 2597AudioFlinger::MixerThread::~MixerThread() 2598{ 2599 if (mFastMixer != NULL) { 2600 FastMixerStateQueue *sq = mFastMixer->sq(); 2601 FastMixerState *state = sq->begin(); 2602 if (state->mCommand == FastMixerState::COLD_IDLE) { 2603 int32_t old = android_atomic_inc(&mFastMixerFutex); 2604 if (old == -1) { 2605 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2606 } 2607 } 2608 state->mCommand = FastMixerState::EXIT; 2609 sq->end(); 2610 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2611 mFastMixer->join(); 2612 // Though the fast mixer thread has exited, it's state queue is still valid. 2613 // We'll use that extract the final state which contains one remaining fast track 2614 // corresponding to our sub-mix. 2615 state = sq->begin(); 2616 ALOG_ASSERT(state->mTrackMask == 1); 2617 FastTrack *fastTrack = &state->mFastTracks[0]; 2618 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2619 delete fastTrack->mBufferProvider; 2620 sq->end(false /*didModify*/); 2621 delete mFastMixer; 2622#ifdef AUDIO_WATCHDOG 2623 if (mAudioWatchdog != 0) { 2624 mAudioWatchdog->requestExit(); 2625 mAudioWatchdog->requestExitAndWait(); 2626 mAudioWatchdog.clear(); 2627 } 2628#endif 2629 } 2630 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2631 delete mAudioMixer; 2632} 2633 2634 2635uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2636{ 2637 if (mFastMixer != NULL) { 2638 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2639 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2640 } 2641 return latency; 2642} 2643 2644 2645void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2646{ 2647 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2648} 2649 2650ssize_t AudioFlinger::MixerThread::threadLoop_write() 2651{ 2652 // FIXME we should only do one push per cycle; confirm this is true 2653 // Start the fast mixer if it's not already running 2654 if (mFastMixer != NULL) { 2655 FastMixerStateQueue *sq = mFastMixer->sq(); 2656 FastMixerState *state = sq->begin(); 2657 if (state->mCommand != FastMixerState::MIX_WRITE && 2658 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2659 if (state->mCommand == FastMixerState::COLD_IDLE) { 2660 int32_t old = android_atomic_inc(&mFastMixerFutex); 2661 if (old == -1) { 2662 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2663 } 2664#ifdef AUDIO_WATCHDOG 2665 if (mAudioWatchdog != 0) { 2666 mAudioWatchdog->resume(); 2667 } 2668#endif 2669 } 2670 state->mCommand = FastMixerState::MIX_WRITE; 2671 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2672 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2673 sq->end(); 2674 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2675 if (kUseFastMixer == FastMixer_Dynamic) { 2676 mNormalSink = mPipeSink; 2677 } 2678 } else { 2679 sq->end(false /*didModify*/); 2680 } 2681 } 2682 return PlaybackThread::threadLoop_write(); 2683} 2684 2685void AudioFlinger::MixerThread::threadLoop_standby() 2686{ 2687 // Idle the fast mixer if it's currently running 2688 if (mFastMixer != NULL) { 2689 FastMixerStateQueue *sq = mFastMixer->sq(); 2690 FastMixerState *state = sq->begin(); 2691 if (!(state->mCommand & FastMixerState::IDLE)) { 2692 state->mCommand = FastMixerState::COLD_IDLE; 2693 state->mColdFutexAddr = &mFastMixerFutex; 2694 state->mColdGen++; 2695 mFastMixerFutex = 0; 2696 sq->end(); 2697 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2698 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2699 if (kUseFastMixer == FastMixer_Dynamic) { 2700 mNormalSink = mOutputSink; 2701 } 2702#ifdef AUDIO_WATCHDOG 2703 if (mAudioWatchdog != 0) { 2704 mAudioWatchdog->pause(); 2705 } 2706#endif 2707 } else { 2708 sq->end(false /*didModify*/); 2709 } 2710 } 2711 PlaybackThread::threadLoop_standby(); 2712} 2713 2714// Empty implementation for standard mixer 2715// Overridden for offloaded playback 2716void AudioFlinger::PlaybackThread::flushOutput_l() 2717{ 2718} 2719 2720bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2721{ 2722 return false; 2723} 2724 2725bool AudioFlinger::PlaybackThread::shouldStandby_l() 2726{ 2727 return !mStandby; 2728} 2729 2730bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2731{ 2732 Mutex::Autolock _l(mLock); 2733 return waitingAsyncCallback_l(); 2734} 2735 2736// shared by MIXER and DIRECT, overridden by DUPLICATING 2737void AudioFlinger::PlaybackThread::threadLoop_standby() 2738{ 2739 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2740 mOutput->stream->common.standby(&mOutput->stream->common); 2741 if (mUseAsyncWrite != 0) { 2742 // discard any pending drain or write ack by incrementing sequence 2743 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2744 mDrainSequence = (mDrainSequence + 2) & ~1; 2745 ALOG_ASSERT(mCallbackThread != 0); 2746 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2747 mCallbackThread->setDraining(mDrainSequence); 2748 } 2749} 2750 2751void AudioFlinger::MixerThread::threadLoop_mix() 2752{ 2753 // obtain the presentation timestamp of the next output buffer 2754 int64_t pts; 2755 status_t status = INVALID_OPERATION; 2756 2757 if (mNormalSink != 0) { 2758 status = mNormalSink->getNextWriteTimestamp(&pts); 2759 } else { 2760 status = mOutputSink->getNextWriteTimestamp(&pts); 2761 } 2762 2763 if (status != NO_ERROR) { 2764 pts = AudioBufferProvider::kInvalidPTS; 2765 } 2766 2767 // mix buffers... 2768 mAudioMixer->process(pts); 2769 mCurrentWriteLength = mixBufferSize; 2770 // increase sleep time progressively when application underrun condition clears. 2771 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2772 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2773 // such that we would underrun the audio HAL. 2774 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2775 sleepTimeShift--; 2776 } 2777 sleepTime = 0; 2778 standbyTime = systemTime() + standbyDelay; 2779 //TODO: delay standby when effects have a tail 2780} 2781 2782void AudioFlinger::MixerThread::threadLoop_sleepTime() 2783{ 2784 // If no tracks are ready, sleep once for the duration of an output 2785 // buffer size, then write 0s to the output 2786 if (sleepTime == 0) { 2787 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2788 sleepTime = activeSleepTime >> sleepTimeShift; 2789 if (sleepTime < kMinThreadSleepTimeUs) { 2790 sleepTime = kMinThreadSleepTimeUs; 2791 } 2792 // reduce sleep time in case of consecutive application underruns to avoid 2793 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2794 // duration we would end up writing less data than needed by the audio HAL if 2795 // the condition persists. 2796 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2797 sleepTimeShift++; 2798 } 2799 } else { 2800 sleepTime = idleSleepTime; 2801 } 2802 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2803 memset (mMixBuffer, 0, mixBufferSize); 2804 sleepTime = 0; 2805 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2806 "anticipated start"); 2807 } 2808 // TODO add standby time extension fct of effect tail 2809} 2810 2811// prepareTracks_l() must be called with ThreadBase::mLock held 2812AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2813 Vector< sp<Track> > *tracksToRemove) 2814{ 2815 2816 mixer_state mixerStatus = MIXER_IDLE; 2817 // find out which tracks need to be processed 2818 size_t count = mActiveTracks.size(); 2819 size_t mixedTracks = 0; 2820 size_t tracksWithEffect = 0; 2821 // counts only _active_ fast tracks 2822 size_t fastTracks = 0; 2823 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2824 2825 float masterVolume = mMasterVolume; 2826 bool masterMute = mMasterMute; 2827 2828 if (masterMute) { 2829 masterVolume = 0; 2830 } 2831 // Delegate master volume control to effect in output mix effect chain if needed 2832 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2833 if (chain != 0) { 2834 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2835 chain->setVolume_l(&v, &v); 2836 masterVolume = (float)((v + (1 << 23)) >> 24); 2837 chain.clear(); 2838 } 2839 2840 // prepare a new state to push 2841 FastMixerStateQueue *sq = NULL; 2842 FastMixerState *state = NULL; 2843 bool didModify = false; 2844 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2845 if (mFastMixer != NULL) { 2846 sq = mFastMixer->sq(); 2847 state = sq->begin(); 2848 } 2849 2850 for (size_t i=0 ; i<count ; i++) { 2851 const sp<Track> t = mActiveTracks[i].promote(); 2852 if (t == 0) { 2853 continue; 2854 } 2855 2856 // this const just means the local variable doesn't change 2857 Track* const track = t.get(); 2858 2859 // process fast tracks 2860 if (track->isFastTrack()) { 2861 2862 // It's theoretically possible (though unlikely) for a fast track to be created 2863 // and then removed within the same normal mix cycle. This is not a problem, as 2864 // the track never becomes active so it's fast mixer slot is never touched. 2865 // The converse, of removing an (active) track and then creating a new track 2866 // at the identical fast mixer slot within the same normal mix cycle, 2867 // is impossible because the slot isn't marked available until the end of each cycle. 2868 int j = track->mFastIndex; 2869 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2870 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2871 FastTrack *fastTrack = &state->mFastTracks[j]; 2872 2873 // Determine whether the track is currently in underrun condition, 2874 // and whether it had a recent underrun. 2875 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2876 FastTrackUnderruns underruns = ftDump->mUnderruns; 2877 uint32_t recentFull = (underruns.mBitFields.mFull - 2878 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2879 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2880 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2881 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2882 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2883 uint32_t recentUnderruns = recentPartial + recentEmpty; 2884 track->mObservedUnderruns = underruns; 2885 // don't count underruns that occur while stopping or pausing 2886 // or stopped which can occur when flush() is called while active 2887 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2888 recentUnderruns > 0) { 2889 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2890 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2891 } 2892 2893 // This is similar to the state machine for normal tracks, 2894 // with a few modifications for fast tracks. 2895 bool isActive = true; 2896 switch (track->mState) { 2897 case TrackBase::STOPPING_1: 2898 // track stays active in STOPPING_1 state until first underrun 2899 if (recentUnderruns > 0 || track->isTerminated()) { 2900 track->mState = TrackBase::STOPPING_2; 2901 } 2902 break; 2903 case TrackBase::PAUSING: 2904 // ramp down is not yet implemented 2905 track->setPaused(); 2906 break; 2907 case TrackBase::RESUMING: 2908 // ramp up is not yet implemented 2909 track->mState = TrackBase::ACTIVE; 2910 break; 2911 case TrackBase::ACTIVE: 2912 if (recentFull > 0 || recentPartial > 0) { 2913 // track has provided at least some frames recently: reset retry count 2914 track->mRetryCount = kMaxTrackRetries; 2915 } 2916 if (recentUnderruns == 0) { 2917 // no recent underruns: stay active 2918 break; 2919 } 2920 // there has recently been an underrun of some kind 2921 if (track->sharedBuffer() == 0) { 2922 // were any of the recent underruns "empty" (no frames available)? 2923 if (recentEmpty == 0) { 2924 // no, then ignore the partial underruns as they are allowed indefinitely 2925 break; 2926 } 2927 // there has recently been an "empty" underrun: decrement the retry counter 2928 if (--(track->mRetryCount) > 0) { 2929 break; 2930 } 2931 // indicate to client process that the track was disabled because of underrun; 2932 // it will then automatically call start() when data is available 2933 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2934 // remove from active list, but state remains ACTIVE [confusing but true] 2935 isActive = false; 2936 break; 2937 } 2938 // fall through 2939 case TrackBase::STOPPING_2: 2940 case TrackBase::PAUSED: 2941 case TrackBase::STOPPED: 2942 case TrackBase::FLUSHED: // flush() while active 2943 // Check for presentation complete if track is inactive 2944 // We have consumed all the buffers of this track. 2945 // This would be incomplete if we auto-paused on underrun 2946 { 2947 size_t audioHALFrames = 2948 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2949 size_t framesWritten = mBytesWritten / mFrameSize; 2950 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2951 // track stays in active list until presentation is complete 2952 break; 2953 } 2954 } 2955 if (track->isStopping_2()) { 2956 track->mState = TrackBase::STOPPED; 2957 } 2958 if (track->isStopped()) { 2959 // Can't reset directly, as fast mixer is still polling this track 2960 // track->reset(); 2961 // So instead mark this track as needing to be reset after push with ack 2962 resetMask |= 1 << i; 2963 } 2964 isActive = false; 2965 break; 2966 case TrackBase::IDLE: 2967 default: 2968 LOG_FATAL("unexpected track state %d", track->mState); 2969 } 2970 2971 if (isActive) { 2972 // was it previously inactive? 2973 if (!(state->mTrackMask & (1 << j))) { 2974 ExtendedAudioBufferProvider *eabp = track; 2975 VolumeProvider *vp = track; 2976 fastTrack->mBufferProvider = eabp; 2977 fastTrack->mVolumeProvider = vp; 2978 fastTrack->mChannelMask = track->mChannelMask; 2979 fastTrack->mGeneration++; 2980 state->mTrackMask |= 1 << j; 2981 didModify = true; 2982 // no acknowledgement required for newly active tracks 2983 } 2984 // cache the combined master volume and stream type volume for fast mixer; this 2985 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2986 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2987 ++fastTracks; 2988 } else { 2989 // was it previously active? 2990 if (state->mTrackMask & (1 << j)) { 2991 fastTrack->mBufferProvider = NULL; 2992 fastTrack->mGeneration++; 2993 state->mTrackMask &= ~(1 << j); 2994 didModify = true; 2995 // If any fast tracks were removed, we must wait for acknowledgement 2996 // because we're about to decrement the last sp<> on those tracks. 2997 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2998 } else { 2999 LOG_FATAL("fast track %d should have been active", j); 3000 } 3001 tracksToRemove->add(track); 3002 // Avoids a misleading display in dumpsys 3003 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3004 } 3005 continue; 3006 } 3007 3008 { // local variable scope to avoid goto warning 3009 3010 audio_track_cblk_t* cblk = track->cblk(); 3011 3012 // The first time a track is added we wait 3013 // for all its buffers to be filled before processing it 3014 int name = track->name(); 3015 // make sure that we have enough frames to mix one full buffer. 3016 // enforce this condition only once to enable draining the buffer in case the client 3017 // app does not call stop() and relies on underrun to stop: 3018 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3019 // during last round 3020 size_t desiredFrames; 3021 uint32_t sr = track->sampleRate(); 3022 if (sr == mSampleRate) { 3023 desiredFrames = mNormalFrameCount; 3024 } else { 3025 // +1 for rounding and +1 for additional sample needed for interpolation 3026 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3027 // add frames already consumed but not yet released by the resampler 3028 // because cblk->framesReady() will include these frames 3029 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3030 // the minimum track buffer size is normally twice the number of frames necessary 3031 // to fill one buffer and the resampler should not leave more than one buffer worth 3032 // of unreleased frames after each pass, but just in case... 3033 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3034 } 3035 uint32_t minFrames = 1; 3036 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3037 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3038 minFrames = desiredFrames; 3039 } 3040 3041 size_t framesReady = track->framesReady(); 3042 if ((framesReady >= minFrames) && track->isReady() && 3043 !track->isPaused() && !track->isTerminated()) 3044 { 3045 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3046 3047 mixedTracks++; 3048 3049 // track->mainBuffer() != mMixBuffer means there is an effect chain 3050 // connected to the track 3051 chain.clear(); 3052 if (track->mainBuffer() != mMixBuffer) { 3053 chain = getEffectChain_l(track->sessionId()); 3054 // Delegate volume control to effect in track effect chain if needed 3055 if (chain != 0) { 3056 tracksWithEffect++; 3057 } else { 3058 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3059 "session %d", 3060 name, track->sessionId()); 3061 } 3062 } 3063 3064 3065 int param = AudioMixer::VOLUME; 3066 if (track->mFillingUpStatus == Track::FS_FILLED) { 3067 // no ramp for the first volume setting 3068 track->mFillingUpStatus = Track::FS_ACTIVE; 3069 if (track->mState == TrackBase::RESUMING) { 3070 track->mState = TrackBase::ACTIVE; 3071 param = AudioMixer::RAMP_VOLUME; 3072 } 3073 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3074 // FIXME should not make a decision based on mServer 3075 } else if (cblk->mServer != 0) { 3076 // If the track is stopped before the first frame was mixed, 3077 // do not apply ramp 3078 param = AudioMixer::RAMP_VOLUME; 3079 } 3080 3081 // compute volume for this track 3082 uint32_t vl, vr, va; 3083 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3084 vl = vr = va = 0; 3085 if (track->isPausing()) { 3086 track->setPaused(); 3087 } 3088 } else { 3089 3090 // read original volumes with volume control 3091 float typeVolume = mStreamTypes[track->streamType()].volume; 3092 float v = masterVolume * typeVolume; 3093 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3094 uint32_t vlr = proxy->getVolumeLR(); 3095 vl = vlr & 0xFFFF; 3096 vr = vlr >> 16; 3097 // track volumes come from shared memory, so can't be trusted and must be clamped 3098 if (vl > MAX_GAIN_INT) { 3099 ALOGV("Track left volume out of range: %04X", vl); 3100 vl = MAX_GAIN_INT; 3101 } 3102 if (vr > MAX_GAIN_INT) { 3103 ALOGV("Track right volume out of range: %04X", vr); 3104 vr = MAX_GAIN_INT; 3105 } 3106 // now apply the master volume and stream type volume 3107 vl = (uint32_t)(v * vl) << 12; 3108 vr = (uint32_t)(v * vr) << 12; 3109 // assuming master volume and stream type volume each go up to 1.0, 3110 // vl and vr are now in 8.24 format 3111 3112 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3113 // send level comes from shared memory and so may be corrupt 3114 if (sendLevel > MAX_GAIN_INT) { 3115 ALOGV("Track send level out of range: %04X", sendLevel); 3116 sendLevel = MAX_GAIN_INT; 3117 } 3118 va = (uint32_t)(v * sendLevel); 3119 } 3120 3121 // Delegate volume control to effect in track effect chain if needed 3122 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3123 // Do not ramp volume if volume is controlled by effect 3124 param = AudioMixer::VOLUME; 3125 track->mHasVolumeController = true; 3126 } else { 3127 // force no volume ramp when volume controller was just disabled or removed 3128 // from effect chain to avoid volume spike 3129 if (track->mHasVolumeController) { 3130 param = AudioMixer::VOLUME; 3131 } 3132 track->mHasVolumeController = false; 3133 } 3134 3135 // Convert volumes from 8.24 to 4.12 format 3136 // This additional clamping is needed in case chain->setVolume_l() overshot 3137 vl = (vl + (1 << 11)) >> 12; 3138 if (vl > MAX_GAIN_INT) { 3139 vl = MAX_GAIN_INT; 3140 } 3141 vr = (vr + (1 << 11)) >> 12; 3142 if (vr > MAX_GAIN_INT) { 3143 vr = MAX_GAIN_INT; 3144 } 3145 3146 if (va > MAX_GAIN_INT) { 3147 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3148 } 3149 3150 // XXX: these things DON'T need to be done each time 3151 mAudioMixer->setBufferProvider(name, track); 3152 mAudioMixer->enable(name); 3153 3154 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3155 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3156 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3157 mAudioMixer->setParameter( 3158 name, 3159 AudioMixer::TRACK, 3160 AudioMixer::FORMAT, (void *)track->format()); 3161 mAudioMixer->setParameter( 3162 name, 3163 AudioMixer::TRACK, 3164 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3165 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3166 uint32_t maxSampleRate = mSampleRate * 2; 3167 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3168 if (reqSampleRate == 0) { 3169 reqSampleRate = mSampleRate; 3170 } else if (reqSampleRate > maxSampleRate) { 3171 reqSampleRate = maxSampleRate; 3172 } 3173 mAudioMixer->setParameter( 3174 name, 3175 AudioMixer::RESAMPLE, 3176 AudioMixer::SAMPLE_RATE, 3177 (void *)(uintptr_t)reqSampleRate); 3178 mAudioMixer->setParameter( 3179 name, 3180 AudioMixer::TRACK, 3181 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3182 mAudioMixer->setParameter( 3183 name, 3184 AudioMixer::TRACK, 3185 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3186 3187 // reset retry count 3188 track->mRetryCount = kMaxTrackRetries; 3189 3190 // If one track is ready, set the mixer ready if: 3191 // - the mixer was not ready during previous round OR 3192 // - no other track is not ready 3193 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3194 mixerStatus != MIXER_TRACKS_ENABLED) { 3195 mixerStatus = MIXER_TRACKS_READY; 3196 } 3197 } else { 3198 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3199 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3200 } 3201 // clear effect chain input buffer if an active track underruns to avoid sending 3202 // previous audio buffer again to effects 3203 chain = getEffectChain_l(track->sessionId()); 3204 if (chain != 0) { 3205 chain->clearInputBuffer(); 3206 } 3207 3208 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3209 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3210 track->isStopped() || track->isPaused()) { 3211 // We have consumed all the buffers of this track. 3212 // Remove it from the list of active tracks. 3213 // TODO: use actual buffer filling status instead of latency when available from 3214 // audio HAL 3215 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3216 size_t framesWritten = mBytesWritten / mFrameSize; 3217 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3218 if (track->isStopped()) { 3219 track->reset(); 3220 } 3221 tracksToRemove->add(track); 3222 } 3223 } else { 3224 // No buffers for this track. Give it a few chances to 3225 // fill a buffer, then remove it from active list. 3226 if (--(track->mRetryCount) <= 0) { 3227 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3228 tracksToRemove->add(track); 3229 // indicate to client process that the track was disabled because of underrun; 3230 // it will then automatically call start() when data is available 3231 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3232 // If one track is not ready, mark the mixer also not ready if: 3233 // - the mixer was ready during previous round OR 3234 // - no other track is ready 3235 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3236 mixerStatus != MIXER_TRACKS_READY) { 3237 mixerStatus = MIXER_TRACKS_ENABLED; 3238 } 3239 } 3240 mAudioMixer->disable(name); 3241 } 3242 3243 } // local variable scope to avoid goto warning 3244track_is_ready: ; 3245 3246 } 3247 3248 // Push the new FastMixer state if necessary 3249 bool pauseAudioWatchdog = false; 3250 if (didModify) { 3251 state->mFastTracksGen++; 3252 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3253 if (kUseFastMixer == FastMixer_Dynamic && 3254 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3255 state->mCommand = FastMixerState::COLD_IDLE; 3256 state->mColdFutexAddr = &mFastMixerFutex; 3257 state->mColdGen++; 3258 mFastMixerFutex = 0; 3259 if (kUseFastMixer == FastMixer_Dynamic) { 3260 mNormalSink = mOutputSink; 3261 } 3262 // If we go into cold idle, need to wait for acknowledgement 3263 // so that fast mixer stops doing I/O. 3264 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3265 pauseAudioWatchdog = true; 3266 } 3267 } 3268 if (sq != NULL) { 3269 sq->end(didModify); 3270 sq->push(block); 3271 } 3272#ifdef AUDIO_WATCHDOG 3273 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3274 mAudioWatchdog->pause(); 3275 } 3276#endif 3277 3278 // Now perform the deferred reset on fast tracks that have stopped 3279 while (resetMask != 0) { 3280 size_t i = __builtin_ctz(resetMask); 3281 ALOG_ASSERT(i < count); 3282 resetMask &= ~(1 << i); 3283 sp<Track> t = mActiveTracks[i].promote(); 3284 if (t == 0) { 3285 continue; 3286 } 3287 Track* track = t.get(); 3288 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3289 track->reset(); 3290 } 3291 3292 // remove all the tracks that need to be... 3293 removeTracks_l(*tracksToRemove); 3294 3295 // mix buffer must be cleared if all tracks are connected to an 3296 // effect chain as in this case the mixer will not write to 3297 // mix buffer and track effects will accumulate into it 3298 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3299 (mixedTracks == 0 && fastTracks > 0))) { 3300 // FIXME as a performance optimization, should remember previous zero status 3301 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3302 } 3303 3304 // if any fast tracks, then status is ready 3305 mMixerStatusIgnoringFastTracks = mixerStatus; 3306 if (fastTracks > 0) { 3307 mixerStatus = MIXER_TRACKS_READY; 3308 } 3309 return mixerStatus; 3310} 3311 3312// getTrackName_l() must be called with ThreadBase::mLock held 3313int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3314{ 3315 return mAudioMixer->getTrackName(channelMask, sessionId); 3316} 3317 3318// deleteTrackName_l() must be called with ThreadBase::mLock held 3319void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3320{ 3321 ALOGV("remove track (%d) and delete from mixer", name); 3322 mAudioMixer->deleteTrackName(name); 3323} 3324 3325// checkForNewParameters_l() must be called with ThreadBase::mLock held 3326bool AudioFlinger::MixerThread::checkForNewParameters_l() 3327{ 3328 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3329 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3330 bool reconfig = false; 3331 3332 while (!mNewParameters.isEmpty()) { 3333 3334 if (mFastMixer != NULL) { 3335 FastMixerStateQueue *sq = mFastMixer->sq(); 3336 FastMixerState *state = sq->begin(); 3337 if (!(state->mCommand & FastMixerState::IDLE)) { 3338 previousCommand = state->mCommand; 3339 state->mCommand = FastMixerState::HOT_IDLE; 3340 sq->end(); 3341 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3342 } else { 3343 sq->end(false /*didModify*/); 3344 } 3345 } 3346 3347 status_t status = NO_ERROR; 3348 String8 keyValuePair = mNewParameters[0]; 3349 AudioParameter param = AudioParameter(keyValuePair); 3350 int value; 3351 3352 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3353 reconfig = true; 3354 } 3355 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3356 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3357 status = BAD_VALUE; 3358 } else { 3359 reconfig = true; 3360 } 3361 } 3362 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3363 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3364 status = BAD_VALUE; 3365 } else { 3366 reconfig = true; 3367 } 3368 } 3369 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3370 // do not accept frame count changes if tracks are open as the track buffer 3371 // size depends on frame count and correct behavior would not be guaranteed 3372 // if frame count is changed after track creation 3373 if (!mTracks.isEmpty()) { 3374 status = INVALID_OPERATION; 3375 } else { 3376 reconfig = true; 3377 } 3378 } 3379 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3380#ifdef ADD_BATTERY_DATA 3381 // when changing the audio output device, call addBatteryData to notify 3382 // the change 3383 if (mOutDevice != value) { 3384 uint32_t params = 0; 3385 // check whether speaker is on 3386 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3387 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3388 } 3389 3390 audio_devices_t deviceWithoutSpeaker 3391 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3392 // check if any other device (except speaker) is on 3393 if (value & deviceWithoutSpeaker ) { 3394 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3395 } 3396 3397 if (params != 0) { 3398 addBatteryData(params); 3399 } 3400 } 3401#endif 3402 3403 // forward device change to effects that have requested to be 3404 // aware of attached audio device. 3405 if (value != AUDIO_DEVICE_NONE) { 3406 mOutDevice = value; 3407 for (size_t i = 0; i < mEffectChains.size(); i++) { 3408 mEffectChains[i]->setDevice_l(mOutDevice); 3409 } 3410 } 3411 } 3412 3413 if (status == NO_ERROR) { 3414 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3415 keyValuePair.string()); 3416 if (!mStandby && status == INVALID_OPERATION) { 3417 mOutput->stream->common.standby(&mOutput->stream->common); 3418 mStandby = true; 3419 mBytesWritten = 0; 3420 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3421 keyValuePair.string()); 3422 } 3423 if (status == NO_ERROR && reconfig) { 3424 readOutputParameters(); 3425 delete mAudioMixer; 3426 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3427 for (size_t i = 0; i < mTracks.size() ; i++) { 3428 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3429 if (name < 0) { 3430 break; 3431 } 3432 mTracks[i]->mName = name; 3433 } 3434 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3435 } 3436 } 3437 3438 mNewParameters.removeAt(0); 3439 3440 mParamStatus = status; 3441 mParamCond.signal(); 3442 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3443 // already timed out waiting for the status and will never signal the condition. 3444 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3445 } 3446 3447 if (!(previousCommand & FastMixerState::IDLE)) { 3448 ALOG_ASSERT(mFastMixer != NULL); 3449 FastMixerStateQueue *sq = mFastMixer->sq(); 3450 FastMixerState *state = sq->begin(); 3451 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3452 state->mCommand = previousCommand; 3453 sq->end(); 3454 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3455 } 3456 3457 return reconfig; 3458} 3459 3460 3461void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3462{ 3463 const size_t SIZE = 256; 3464 char buffer[SIZE]; 3465 String8 result; 3466 3467 PlaybackThread::dumpInternals(fd, args); 3468 3469 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3470 result.append(buffer); 3471 write(fd, result.string(), result.size()); 3472 3473 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3474 const FastMixerDumpState copy(mFastMixerDumpState); 3475 copy.dump(fd); 3476 3477#ifdef STATE_QUEUE_DUMP 3478 // Similar for state queue 3479 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3480 observerCopy.dump(fd); 3481 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3482 mutatorCopy.dump(fd); 3483#endif 3484 3485#ifdef TEE_SINK 3486 // Write the tee output to a .wav file 3487 dumpTee(fd, mTeeSource, mId); 3488#endif 3489 3490#ifdef AUDIO_WATCHDOG 3491 if (mAudioWatchdog != 0) { 3492 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3493 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3494 wdCopy.dump(fd); 3495 } 3496#endif 3497} 3498 3499uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3500{ 3501 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3502} 3503 3504uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3505{ 3506 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3507} 3508 3509void AudioFlinger::MixerThread::cacheParameters_l() 3510{ 3511 PlaybackThread::cacheParameters_l(); 3512 3513 // FIXME: Relaxed timing because of a certain device that can't meet latency 3514 // Should be reduced to 2x after the vendor fixes the driver issue 3515 // increase threshold again due to low power audio mode. The way this warning 3516 // threshold is calculated and its usefulness should be reconsidered anyway. 3517 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3518} 3519 3520// ---------------------------------------------------------------------------- 3521 3522AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3523 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3524 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3525 // mLeftVolFloat, mRightVolFloat 3526{ 3527} 3528 3529AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3530 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3531 ThreadBase::type_t type) 3532 : PlaybackThread(audioFlinger, output, id, device, type) 3533 // mLeftVolFloat, mRightVolFloat 3534{ 3535} 3536 3537AudioFlinger::DirectOutputThread::~DirectOutputThread() 3538{ 3539} 3540 3541void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3542{ 3543 audio_track_cblk_t* cblk = track->cblk(); 3544 float left, right; 3545 3546 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3547 left = right = 0; 3548 } else { 3549 float typeVolume = mStreamTypes[track->streamType()].volume; 3550 float v = mMasterVolume * typeVolume; 3551 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3552 uint32_t vlr = proxy->getVolumeLR(); 3553 float v_clamped = v * (vlr & 0xFFFF); 3554 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3555 left = v_clamped/MAX_GAIN; 3556 v_clamped = v * (vlr >> 16); 3557 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3558 right = v_clamped/MAX_GAIN; 3559 } 3560 3561 if (lastTrack) { 3562 if (left != mLeftVolFloat || right != mRightVolFloat) { 3563 mLeftVolFloat = left; 3564 mRightVolFloat = right; 3565 3566 // Convert volumes from float to 8.24 3567 uint32_t vl = (uint32_t)(left * (1 << 24)); 3568 uint32_t vr = (uint32_t)(right * (1 << 24)); 3569 3570 // Delegate volume control to effect in track effect chain if needed 3571 // only one effect chain can be present on DirectOutputThread, so if 3572 // there is one, the track is connected to it 3573 if (!mEffectChains.isEmpty()) { 3574 mEffectChains[0]->setVolume_l(&vl, &vr); 3575 left = (float)vl / (1 << 24); 3576 right = (float)vr / (1 << 24); 3577 } 3578 if (mOutput->stream->set_volume) { 3579 mOutput->stream->set_volume(mOutput->stream, left, right); 3580 } 3581 } 3582 } 3583} 3584 3585 3586AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3587 Vector< sp<Track> > *tracksToRemove 3588) 3589{ 3590 size_t count = mActiveTracks.size(); 3591 mixer_state mixerStatus = MIXER_IDLE; 3592 3593 // find out which tracks need to be processed 3594 for (size_t i = 0; i < count; i++) { 3595 sp<Track> t = mActiveTracks[i].promote(); 3596 // The track died recently 3597 if (t == 0) { 3598 continue; 3599 } 3600 3601 Track* const track = t.get(); 3602 audio_track_cblk_t* cblk = track->cblk(); 3603 // Only consider last track started for volume and mixer state control. 3604 // In theory an older track could underrun and restart after the new one starts 3605 // but as we only care about the transition phase between two tracks on a 3606 // direct output, it is not a problem to ignore the underrun case. 3607 sp<Track> l = mLatestActiveTrack.promote(); 3608 bool last = l.get() == track; 3609 3610 // The first time a track is added we wait 3611 // for all its buffers to be filled before processing it 3612 uint32_t minFrames; 3613 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3614 minFrames = mNormalFrameCount; 3615 } else { 3616 minFrames = 1; 3617 } 3618 3619 if ((track->framesReady() >= minFrames) && track->isReady() && 3620 !track->isPaused() && !track->isTerminated()) 3621 { 3622 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3623 3624 if (track->mFillingUpStatus == Track::FS_FILLED) { 3625 track->mFillingUpStatus = Track::FS_ACTIVE; 3626 // make sure processVolume_l() will apply new volume even if 0 3627 mLeftVolFloat = mRightVolFloat = -1.0; 3628 if (track->mState == TrackBase::RESUMING) { 3629 track->mState = TrackBase::ACTIVE; 3630 } 3631 } 3632 3633 // compute volume for this track 3634 processVolume_l(track, last); 3635 if (last) { 3636 // reset retry count 3637 track->mRetryCount = kMaxTrackRetriesDirect; 3638 mActiveTrack = t; 3639 mixerStatus = MIXER_TRACKS_READY; 3640 } 3641 } else { 3642 // clear effect chain input buffer if the last active track started underruns 3643 // to avoid sending previous audio buffer again to effects 3644 if (!mEffectChains.isEmpty() && last) { 3645 mEffectChains[0]->clearInputBuffer(); 3646 } 3647 3648 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3649 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3650 track->isStopped() || track->isPaused()) { 3651 // We have consumed all the buffers of this track. 3652 // Remove it from the list of active tracks. 3653 // TODO: implement behavior for compressed audio 3654 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3655 size_t framesWritten = mBytesWritten / mFrameSize; 3656 if (mStandby || !last || 3657 track->presentationComplete(framesWritten, audioHALFrames)) { 3658 if (track->isStopped()) { 3659 track->reset(); 3660 } 3661 tracksToRemove->add(track); 3662 } 3663 } else { 3664 // No buffers for this track. Give it a few chances to 3665 // fill a buffer, then remove it from active list. 3666 // Only consider last track started for mixer state control 3667 if (--(track->mRetryCount) <= 0) { 3668 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3669 tracksToRemove->add(track); 3670 // indicate to client process that the track was disabled because of underrun; 3671 // it will then automatically call start() when data is available 3672 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3673 } else if (last) { 3674 mixerStatus = MIXER_TRACKS_ENABLED; 3675 } 3676 } 3677 } 3678 } 3679 3680 // remove all the tracks that need to be... 3681 removeTracks_l(*tracksToRemove); 3682 3683 return mixerStatus; 3684} 3685 3686void AudioFlinger::DirectOutputThread::threadLoop_mix() 3687{ 3688 size_t frameCount = mFrameCount; 3689 int8_t *curBuf = (int8_t *)mMixBuffer; 3690 // output audio to hardware 3691 while (frameCount) { 3692 AudioBufferProvider::Buffer buffer; 3693 buffer.frameCount = frameCount; 3694 mActiveTrack->getNextBuffer(&buffer); 3695 if (buffer.raw == NULL) { 3696 memset(curBuf, 0, frameCount * mFrameSize); 3697 break; 3698 } 3699 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3700 frameCount -= buffer.frameCount; 3701 curBuf += buffer.frameCount * mFrameSize; 3702 mActiveTrack->releaseBuffer(&buffer); 3703 } 3704 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3705 sleepTime = 0; 3706 standbyTime = systemTime() + standbyDelay; 3707 mActiveTrack.clear(); 3708} 3709 3710void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3711{ 3712 if (sleepTime == 0) { 3713 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3714 sleepTime = activeSleepTime; 3715 } else { 3716 sleepTime = idleSleepTime; 3717 } 3718 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3719 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3720 sleepTime = 0; 3721 } 3722} 3723 3724// getTrackName_l() must be called with ThreadBase::mLock held 3725int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3726 int sessionId) 3727{ 3728 return 0; 3729} 3730 3731// deleteTrackName_l() must be called with ThreadBase::mLock held 3732void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3733{ 3734} 3735 3736// checkForNewParameters_l() must be called with ThreadBase::mLock held 3737bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3738{ 3739 bool reconfig = false; 3740 3741 while (!mNewParameters.isEmpty()) { 3742 status_t status = NO_ERROR; 3743 String8 keyValuePair = mNewParameters[0]; 3744 AudioParameter param = AudioParameter(keyValuePair); 3745 int value; 3746 3747 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3748 // do not accept frame count changes if tracks are open as the track buffer 3749 // size depends on frame count and correct behavior would not be garantied 3750 // if frame count is changed after track creation 3751 if (!mTracks.isEmpty()) { 3752 status = INVALID_OPERATION; 3753 } else { 3754 reconfig = true; 3755 } 3756 } 3757 if (status == NO_ERROR) { 3758 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3759 keyValuePair.string()); 3760 if (!mStandby && status == INVALID_OPERATION) { 3761 mOutput->stream->common.standby(&mOutput->stream->common); 3762 mStandby = true; 3763 mBytesWritten = 0; 3764 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3765 keyValuePair.string()); 3766 } 3767 if (status == NO_ERROR && reconfig) { 3768 readOutputParameters(); 3769 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3770 } 3771 } 3772 3773 mNewParameters.removeAt(0); 3774 3775 mParamStatus = status; 3776 mParamCond.signal(); 3777 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3778 // already timed out waiting for the status and will never signal the condition. 3779 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3780 } 3781 return reconfig; 3782} 3783 3784uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3785{ 3786 uint32_t time; 3787 if (audio_is_linear_pcm(mFormat)) { 3788 time = PlaybackThread::activeSleepTimeUs(); 3789 } else { 3790 time = 10000; 3791 } 3792 return time; 3793} 3794 3795uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3796{ 3797 uint32_t time; 3798 if (audio_is_linear_pcm(mFormat)) { 3799 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3800 } else { 3801 time = 10000; 3802 } 3803 return time; 3804} 3805 3806uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3807{ 3808 uint32_t time; 3809 if (audio_is_linear_pcm(mFormat)) { 3810 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3811 } else { 3812 time = 10000; 3813 } 3814 return time; 3815} 3816 3817void AudioFlinger::DirectOutputThread::cacheParameters_l() 3818{ 3819 PlaybackThread::cacheParameters_l(); 3820 3821 // use shorter standby delay as on normal output to release 3822 // hardware resources as soon as possible 3823 if (audio_is_linear_pcm(mFormat)) { 3824 standbyDelay = microseconds(activeSleepTime*2); 3825 } else { 3826 standbyDelay = kOffloadStandbyDelayNs; 3827 } 3828} 3829 3830// ---------------------------------------------------------------------------- 3831 3832AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3833 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3834 : Thread(false /*canCallJava*/), 3835 mPlaybackThread(playbackThread), 3836 mWriteAckSequence(0), 3837 mDrainSequence(0) 3838{ 3839} 3840 3841AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3842{ 3843} 3844 3845void AudioFlinger::AsyncCallbackThread::onFirstRef() 3846{ 3847 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3848} 3849 3850bool AudioFlinger::AsyncCallbackThread::threadLoop() 3851{ 3852 while (!exitPending()) { 3853 uint32_t writeAckSequence; 3854 uint32_t drainSequence; 3855 3856 { 3857 Mutex::Autolock _l(mLock); 3858 while (!((mWriteAckSequence & 1) || 3859 (mDrainSequence & 1) || 3860 exitPending())) { 3861 mWaitWorkCV.wait(mLock); 3862 } 3863 3864 if (exitPending()) { 3865 break; 3866 } 3867 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3868 mWriteAckSequence, mDrainSequence); 3869 writeAckSequence = mWriteAckSequence; 3870 mWriteAckSequence &= ~1; 3871 drainSequence = mDrainSequence; 3872 mDrainSequence &= ~1; 3873 } 3874 { 3875 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3876 if (playbackThread != 0) { 3877 if (writeAckSequence & 1) { 3878 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3879 } 3880 if (drainSequence & 1) { 3881 playbackThread->resetDraining(drainSequence >> 1); 3882 } 3883 } 3884 } 3885 } 3886 return false; 3887} 3888 3889void AudioFlinger::AsyncCallbackThread::exit() 3890{ 3891 ALOGV("AsyncCallbackThread::exit"); 3892 Mutex::Autolock _l(mLock); 3893 requestExit(); 3894 mWaitWorkCV.broadcast(); 3895} 3896 3897void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3898{ 3899 Mutex::Autolock _l(mLock); 3900 // bit 0 is cleared 3901 mWriteAckSequence = sequence << 1; 3902} 3903 3904void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3905{ 3906 Mutex::Autolock _l(mLock); 3907 // ignore unexpected callbacks 3908 if (mWriteAckSequence & 2) { 3909 mWriteAckSequence |= 1; 3910 mWaitWorkCV.signal(); 3911 } 3912} 3913 3914void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3915{ 3916 Mutex::Autolock _l(mLock); 3917 // bit 0 is cleared 3918 mDrainSequence = sequence << 1; 3919} 3920 3921void AudioFlinger::AsyncCallbackThread::resetDraining() 3922{ 3923 Mutex::Autolock _l(mLock); 3924 // ignore unexpected callbacks 3925 if (mDrainSequence & 2) { 3926 mDrainSequence |= 1; 3927 mWaitWorkCV.signal(); 3928 } 3929} 3930 3931 3932// ---------------------------------------------------------------------------- 3933AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3934 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3935 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3936 mHwPaused(false), 3937 mFlushPending(false), 3938 mPausedBytesRemaining(0) 3939{ 3940 //FIXME: mStandby should be set to true by ThreadBase constructor 3941 mStandby = true; 3942} 3943 3944void AudioFlinger::OffloadThread::threadLoop_exit() 3945{ 3946 if (mFlushPending || mHwPaused) { 3947 // If a flush is pending or track was paused, just discard buffered data 3948 flushHw_l(); 3949 } else { 3950 mMixerStatus = MIXER_DRAIN_ALL; 3951 threadLoop_drain(); 3952 } 3953 mCallbackThread->exit(); 3954 PlaybackThread::threadLoop_exit(); 3955} 3956 3957AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3958 Vector< sp<Track> > *tracksToRemove 3959) 3960{ 3961 size_t count = mActiveTracks.size(); 3962 3963 mixer_state mixerStatus = MIXER_IDLE; 3964 bool doHwPause = false; 3965 bool doHwResume = false; 3966 3967 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3968 3969 // find out which tracks need to be processed 3970 for (size_t i = 0; i < count; i++) { 3971 sp<Track> t = mActiveTracks[i].promote(); 3972 // The track died recently 3973 if (t == 0) { 3974 continue; 3975 } 3976 Track* const track = t.get(); 3977 audio_track_cblk_t* cblk = track->cblk(); 3978 // Only consider last track started for volume and mixer state control. 3979 // In theory an older track could underrun and restart after the new one starts 3980 // but as we only care about the transition phase between two tracks on a 3981 // direct output, it is not a problem to ignore the underrun case. 3982 sp<Track> l = mLatestActiveTrack.promote(); 3983 bool last = l.get() == track; 3984 3985 if (track->isPausing()) { 3986 track->setPaused(); 3987 if (last) { 3988 if (!mHwPaused) { 3989 doHwPause = true; 3990 mHwPaused = true; 3991 } 3992 // If we were part way through writing the mixbuffer to 3993 // the HAL we must save this until we resume 3994 // BUG - this will be wrong if a different track is made active, 3995 // in that case we want to discard the pending data in the 3996 // mixbuffer and tell the client to present it again when the 3997 // track is resumed 3998 mPausedWriteLength = mCurrentWriteLength; 3999 mPausedBytesRemaining = mBytesRemaining; 4000 mBytesRemaining = 0; // stop writing 4001 } 4002 tracksToRemove->add(track); 4003 } else if (track->framesReady() && track->isReady() && 4004 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4005 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4006 if (track->mFillingUpStatus == Track::FS_FILLED) { 4007 track->mFillingUpStatus = Track::FS_ACTIVE; 4008 // make sure processVolume_l() will apply new volume even if 0 4009 mLeftVolFloat = mRightVolFloat = -1.0; 4010 if (track->mState == TrackBase::RESUMING) { 4011 track->mState = TrackBase::ACTIVE; 4012 if (last) { 4013 if (mPausedBytesRemaining) { 4014 // Need to continue write that was interrupted 4015 mCurrentWriteLength = mPausedWriteLength; 4016 mBytesRemaining = mPausedBytesRemaining; 4017 mPausedBytesRemaining = 0; 4018 } 4019 if (mHwPaused) { 4020 doHwResume = true; 4021 mHwPaused = false; 4022 // threadLoop_mix() will handle the case that we need to 4023 // resume an interrupted write 4024 } 4025 // enable write to audio HAL 4026 sleepTime = 0; 4027 } 4028 } 4029 } 4030 4031 if (last) { 4032 sp<Track> previousTrack = mPreviousTrack.promote(); 4033 if (previousTrack != 0) { 4034 if (track != previousTrack.get()) { 4035 // Flush any data still being written from last track 4036 mBytesRemaining = 0; 4037 if (mPausedBytesRemaining) { 4038 // Last track was paused so we also need to flush saved 4039 // mixbuffer state and invalidate track so that it will 4040 // re-submit that unwritten data when it is next resumed 4041 mPausedBytesRemaining = 0; 4042 // Invalidate is a bit drastic - would be more efficient 4043 // to have a flag to tell client that some of the 4044 // previously written data was lost 4045 previousTrack->invalidate(); 4046 } 4047 // flush data already sent to the DSP if changing audio session as audio 4048 // comes from a different source. Also invalidate previous track to force a 4049 // seek when resuming. 4050 if (previousTrack->sessionId() != track->sessionId()) { 4051 previousTrack->invalidate(); 4052 mFlushPending = true; 4053 } 4054 } 4055 } 4056 mPreviousTrack = track; 4057 // reset retry count 4058 track->mRetryCount = kMaxTrackRetriesOffload; 4059 mActiveTrack = t; 4060 mixerStatus = MIXER_TRACKS_READY; 4061 } 4062 } else { 4063 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4064 if (track->isStopping_1()) { 4065 // Hardware buffer can hold a large amount of audio so we must 4066 // wait for all current track's data to drain before we say 4067 // that the track is stopped. 4068 if (mBytesRemaining == 0) { 4069 // Only start draining when all data in mixbuffer 4070 // has been written 4071 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4072 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4073 // do not drain if no data was ever sent to HAL (mStandby == true) 4074 if (last && !mStandby) { 4075 // do not modify drain sequence if we are already draining. This happens 4076 // when resuming from pause after drain. 4077 if ((mDrainSequence & 1) == 0) { 4078 sleepTime = 0; 4079 standbyTime = systemTime() + standbyDelay; 4080 mixerStatus = MIXER_DRAIN_TRACK; 4081 mDrainSequence += 2; 4082 } 4083 if (mHwPaused) { 4084 // It is possible to move from PAUSED to STOPPING_1 without 4085 // a resume so we must ensure hardware is running 4086 doHwResume = true; 4087 mHwPaused = false; 4088 } 4089 } 4090 } 4091 } else if (track->isStopping_2()) { 4092 // Drain has completed or we are in standby, signal presentation complete 4093 if (!(mDrainSequence & 1) || !last || mStandby) { 4094 track->mState = TrackBase::STOPPED; 4095 size_t audioHALFrames = 4096 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4097 size_t framesWritten = 4098 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4099 track->presentationComplete(framesWritten, audioHALFrames); 4100 track->reset(); 4101 tracksToRemove->add(track); 4102 } 4103 } else { 4104 // No buffers for this track. Give it a few chances to 4105 // fill a buffer, then remove it from active list. 4106 if (--(track->mRetryCount) <= 0) { 4107 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4108 track->name()); 4109 tracksToRemove->add(track); 4110 // indicate to client process that the track was disabled because of underrun; 4111 // it will then automatically call start() when data is available 4112 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4113 } else if (last){ 4114 mixerStatus = MIXER_TRACKS_ENABLED; 4115 } 4116 } 4117 } 4118 // compute volume for this track 4119 processVolume_l(track, last); 4120 } 4121 4122 // make sure the pause/flush/resume sequence is executed in the right order. 4123 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4124 // before flush and then resume HW. This can happen in case of pause/flush/resume 4125 // if resume is received before pause is executed. 4126 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4127 mOutput->stream->pause(mOutput->stream); 4128 if (!doHwPause) { 4129 doHwResume = true; 4130 } 4131 } 4132 if (mFlushPending) { 4133 flushHw_l(); 4134 mFlushPending = false; 4135 } 4136 if (!mStandby && doHwResume) { 4137 mOutput->stream->resume(mOutput->stream); 4138 } 4139 4140 // remove all the tracks that need to be... 4141 removeTracks_l(*tracksToRemove); 4142 4143 return mixerStatus; 4144} 4145 4146void AudioFlinger::OffloadThread::flushOutput_l() 4147{ 4148 mFlushPending = true; 4149} 4150 4151// must be called with thread mutex locked 4152bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4153{ 4154 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4155 mWriteAckSequence, mDrainSequence); 4156 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4157 return true; 4158 } 4159 return false; 4160} 4161 4162// must be called with thread mutex locked 4163bool AudioFlinger::OffloadThread::shouldStandby_l() 4164{ 4165 bool TrackPaused = false; 4166 4167 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4168 // after a timeout and we will enter standby then. 4169 if (mTracks.size() > 0) { 4170 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4171 } 4172 4173 return !mStandby && !TrackPaused; 4174} 4175 4176 4177bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4178{ 4179 Mutex::Autolock _l(mLock); 4180 return waitingAsyncCallback_l(); 4181} 4182 4183void AudioFlinger::OffloadThread::flushHw_l() 4184{ 4185 mOutput->stream->flush(mOutput->stream); 4186 // Flush anything still waiting in the mixbuffer 4187 mCurrentWriteLength = 0; 4188 mBytesRemaining = 0; 4189 mPausedWriteLength = 0; 4190 mPausedBytesRemaining = 0; 4191 if (mUseAsyncWrite) { 4192 // discard any pending drain or write ack by incrementing sequence 4193 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4194 mDrainSequence = (mDrainSequence + 2) & ~1; 4195 ALOG_ASSERT(mCallbackThread != 0); 4196 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4197 mCallbackThread->setDraining(mDrainSequence); 4198 } 4199} 4200 4201// ---------------------------------------------------------------------------- 4202 4203AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4204 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4205 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4206 DUPLICATING), 4207 mWaitTimeMs(UINT_MAX) 4208{ 4209 addOutputTrack(mainThread); 4210} 4211 4212AudioFlinger::DuplicatingThread::~DuplicatingThread() 4213{ 4214 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4215 mOutputTracks[i]->destroy(); 4216 } 4217} 4218 4219void AudioFlinger::DuplicatingThread::threadLoop_mix() 4220{ 4221 // mix buffers... 4222 if (outputsReady(outputTracks)) { 4223 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4224 } else { 4225 memset(mMixBuffer, 0, mixBufferSize); 4226 } 4227 sleepTime = 0; 4228 writeFrames = mNormalFrameCount; 4229 mCurrentWriteLength = mixBufferSize; 4230 standbyTime = systemTime() + standbyDelay; 4231} 4232 4233void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4234{ 4235 if (sleepTime == 0) { 4236 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4237 sleepTime = activeSleepTime; 4238 } else { 4239 sleepTime = idleSleepTime; 4240 } 4241 } else if (mBytesWritten != 0) { 4242 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4243 writeFrames = mNormalFrameCount; 4244 memset(mMixBuffer, 0, mixBufferSize); 4245 } else { 4246 // flush remaining overflow buffers in output tracks 4247 writeFrames = 0; 4248 } 4249 sleepTime = 0; 4250 } 4251} 4252 4253ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4254{ 4255 for (size_t i = 0; i < outputTracks.size(); i++) { 4256 outputTracks[i]->write(mMixBuffer, writeFrames); 4257 } 4258 mStandby = false; 4259 return (ssize_t)mixBufferSize; 4260} 4261 4262void AudioFlinger::DuplicatingThread::threadLoop_standby() 4263{ 4264 // DuplicatingThread implements standby by stopping all tracks 4265 for (size_t i = 0; i < outputTracks.size(); i++) { 4266 outputTracks[i]->stop(); 4267 } 4268} 4269 4270void AudioFlinger::DuplicatingThread::saveOutputTracks() 4271{ 4272 outputTracks = mOutputTracks; 4273} 4274 4275void AudioFlinger::DuplicatingThread::clearOutputTracks() 4276{ 4277 outputTracks.clear(); 4278} 4279 4280void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4281{ 4282 Mutex::Autolock _l(mLock); 4283 // FIXME explain this formula 4284 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4285 OutputTrack *outputTrack = new OutputTrack(thread, 4286 this, 4287 mSampleRate, 4288 mFormat, 4289 mChannelMask, 4290 frameCount, 4291 IPCThreadState::self()->getCallingUid()); 4292 if (outputTrack->cblk() != NULL) { 4293 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4294 mOutputTracks.add(outputTrack); 4295 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4296 updateWaitTime_l(); 4297 } 4298} 4299 4300void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4301{ 4302 Mutex::Autolock _l(mLock); 4303 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4304 if (mOutputTracks[i]->thread() == thread) { 4305 mOutputTracks[i]->destroy(); 4306 mOutputTracks.removeAt(i); 4307 updateWaitTime_l(); 4308 return; 4309 } 4310 } 4311 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4312} 4313 4314// caller must hold mLock 4315void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4316{ 4317 mWaitTimeMs = UINT_MAX; 4318 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4319 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4320 if (strong != 0) { 4321 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4322 if (waitTimeMs < mWaitTimeMs) { 4323 mWaitTimeMs = waitTimeMs; 4324 } 4325 } 4326 } 4327} 4328 4329 4330bool AudioFlinger::DuplicatingThread::outputsReady( 4331 const SortedVector< sp<OutputTrack> > &outputTracks) 4332{ 4333 for (size_t i = 0; i < outputTracks.size(); i++) { 4334 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4335 if (thread == 0) { 4336 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4337 outputTracks[i].get()); 4338 return false; 4339 } 4340 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4341 // see note at standby() declaration 4342 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4343 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4344 thread.get()); 4345 return false; 4346 } 4347 } 4348 return true; 4349} 4350 4351uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4352{ 4353 return (mWaitTimeMs * 1000) / 2; 4354} 4355 4356void AudioFlinger::DuplicatingThread::cacheParameters_l() 4357{ 4358 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4359 updateWaitTime_l(); 4360 4361 MixerThread::cacheParameters_l(); 4362} 4363 4364// ---------------------------------------------------------------------------- 4365// Record 4366// ---------------------------------------------------------------------------- 4367 4368AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4369 AudioStreamIn *input, 4370 uint32_t sampleRate, 4371 audio_channel_mask_t channelMask, 4372 audio_io_handle_t id, 4373 audio_devices_t outDevice, 4374 audio_devices_t inDevice 4375#ifdef TEE_SINK 4376 , const sp<NBAIO_Sink>& teeSink 4377#endif 4378 ) : 4379 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4380 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4381 // mRsmpInIndex and mBufferSize set by readInputParameters() 4382 mReqChannelCount(popcount(channelMask)), 4383 mReqSampleRate(sampleRate) 4384 // mBytesRead is only meaningful while active, and so is cleared in start() 4385 // (but might be better to also clear here for dump?) 4386#ifdef TEE_SINK 4387 , mTeeSink(teeSink) 4388#endif 4389{ 4390 snprintf(mName, kNameLength, "AudioIn_%X", id); 4391 4392 readInputParameters(); 4393} 4394 4395 4396AudioFlinger::RecordThread::~RecordThread() 4397{ 4398 delete[] mRsmpInBuffer; 4399 delete mResampler; 4400 delete[] mRsmpOutBuffer; 4401} 4402 4403void AudioFlinger::RecordThread::onFirstRef() 4404{ 4405 run(mName, PRIORITY_URGENT_AUDIO); 4406} 4407 4408status_t AudioFlinger::RecordThread::readyToRun() 4409{ 4410 status_t status = initCheck(); 4411 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4412 return status; 4413} 4414 4415bool AudioFlinger::RecordThread::threadLoop() 4416{ 4417 AudioBufferProvider::Buffer buffer; 4418 sp<RecordTrack> activeTrack; 4419 Vector< sp<EffectChain> > effectChains; 4420 4421 nsecs_t lastWarning = 0; 4422 4423 inputStandBy(); 4424 { 4425 Mutex::Autolock _l(mLock); 4426 activeTrack = mActiveTrack; 4427 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4428 } 4429 4430 // used to verify we've read at least once before evaluating how many bytes were read 4431 bool readOnce = false; 4432 4433 // start recording 4434 while (!exitPending()) { 4435 4436 processConfigEvents(); 4437 4438 { // scope for mLock 4439 Mutex::Autolock _l(mLock); 4440 checkForNewParameters_l(); 4441 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4442 SortedVector<int> tmp; 4443 tmp.add(mActiveTrack->uid()); 4444 updateWakeLockUids_l(tmp); 4445 } 4446 activeTrack = mActiveTrack; 4447 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4448 standby(); 4449 4450 if (exitPending()) { 4451 break; 4452 } 4453 4454 releaseWakeLock_l(); 4455 ALOGV("RecordThread: loop stopping"); 4456 // go to sleep 4457 mWaitWorkCV.wait(mLock); 4458 ALOGV("RecordThread: loop starting"); 4459 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4460 continue; 4461 } 4462 if (mActiveTrack != 0) { 4463 if (mActiveTrack->isTerminated()) { 4464 removeTrack_l(mActiveTrack); 4465 mActiveTrack.clear(); 4466 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4467 standby(); 4468 mActiveTrack.clear(); 4469 mStartStopCond.broadcast(); 4470 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4471 if (mReqChannelCount != mActiveTrack->channelCount()) { 4472 mActiveTrack.clear(); 4473 mStartStopCond.broadcast(); 4474 } else if (readOnce) { 4475 // record start succeeds only if first read from audio input 4476 // succeeds 4477 if (mBytesRead >= 0) { 4478 mActiveTrack->mState = TrackBase::ACTIVE; 4479 } else { 4480 mActiveTrack.clear(); 4481 } 4482 mStartStopCond.broadcast(); 4483 } 4484 mStandby = false; 4485 } 4486 } 4487 4488 lockEffectChains_l(effectChains); 4489 } 4490 4491 if (mActiveTrack != 0) { 4492 if (mActiveTrack->mState != TrackBase::ACTIVE && 4493 mActiveTrack->mState != TrackBase::RESUMING) { 4494 unlockEffectChains(effectChains); 4495 usleep(kRecordThreadSleepUs); 4496 continue; 4497 } 4498 for (size_t i = 0; i < effectChains.size(); i ++) { 4499 effectChains[i]->process_l(); 4500 } 4501 4502 buffer.frameCount = mFrameCount; 4503 status_t status = mActiveTrack->getNextBuffer(&buffer); 4504 if (status == NO_ERROR) { 4505 readOnce = true; 4506 size_t framesOut = buffer.frameCount; 4507 if (mResampler == NULL) { 4508 // no resampling 4509 while (framesOut) { 4510 size_t framesIn = mFrameCount - mRsmpInIndex; 4511 if (framesIn) { 4512 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4513 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4514 mActiveTrack->mFrameSize; 4515 if (framesIn > framesOut) 4516 framesIn = framesOut; 4517 mRsmpInIndex += framesIn; 4518 framesOut -= framesIn; 4519 if (mChannelCount == mReqChannelCount) { 4520 memcpy(dst, src, framesIn * mFrameSize); 4521 } else { 4522 if (mChannelCount == 1) { 4523 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4524 (int16_t *)src, framesIn); 4525 } else { 4526 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4527 (int16_t *)src, framesIn); 4528 } 4529 } 4530 } 4531 if (framesOut && mFrameCount == mRsmpInIndex) { 4532 void *readInto; 4533 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4534 readInto = buffer.raw; 4535 framesOut = 0; 4536 } else { 4537 readInto = mRsmpInBuffer; 4538 mRsmpInIndex = 0; 4539 } 4540 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4541 mBufferSize); 4542 if (mBytesRead <= 0) { 4543 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4544 { 4545 ALOGE("Error reading audio input"); 4546 // Force input into standby so that it tries to 4547 // recover at next read attempt 4548 inputStandBy(); 4549 usleep(kRecordThreadSleepUs); 4550 } 4551 mRsmpInIndex = mFrameCount; 4552 framesOut = 0; 4553 buffer.frameCount = 0; 4554 } 4555#ifdef TEE_SINK 4556 else if (mTeeSink != 0) { 4557 (void) mTeeSink->write(readInto, 4558 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4559 } 4560#endif 4561 } 4562 } 4563 } else { 4564 // resampling 4565 4566 // resampler accumulates, but we only have one source track 4567 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4568 // alter output frame count as if we were expecting stereo samples 4569 if (mChannelCount == 1 && mReqChannelCount == 1) { 4570 framesOut >>= 1; 4571 } 4572 mResampler->resample(mRsmpOutBuffer, framesOut, 4573 this /* AudioBufferProvider* */); 4574 // ditherAndClamp() works as long as all buffers returned by 4575 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4576 if (mChannelCount == 2 && mReqChannelCount == 1) { 4577 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4578 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4579 // the resampler always outputs stereo samples: 4580 // do post stereo to mono conversion 4581 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4582 framesOut); 4583 } else { 4584 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4585 } 4586 // now done with mRsmpOutBuffer 4587 4588 } 4589 if (mFramestoDrop == 0) { 4590 mActiveTrack->releaseBuffer(&buffer); 4591 } else { 4592 if (mFramestoDrop > 0) { 4593 mFramestoDrop -= buffer.frameCount; 4594 if (mFramestoDrop <= 0) { 4595 clearSyncStartEvent(); 4596 } 4597 } else { 4598 mFramestoDrop += buffer.frameCount; 4599 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4600 mSyncStartEvent->isCancelled()) { 4601 ALOGW("Synced record %s, session %d, trigger session %d", 4602 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4603 mActiveTrack->sessionId(), 4604 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4605 clearSyncStartEvent(); 4606 } 4607 } 4608 } 4609 mActiveTrack->clearOverflow(); 4610 } 4611 // client isn't retrieving buffers fast enough 4612 else { 4613 if (!mActiveTrack->setOverflow()) { 4614 nsecs_t now = systemTime(); 4615 if ((now - lastWarning) > kWarningThrottleNs) { 4616 ALOGW("RecordThread: buffer overflow"); 4617 lastWarning = now; 4618 } 4619 } 4620 // Release the processor for a while before asking for a new buffer. 4621 // This will give the application more chance to read from the buffer and 4622 // clear the overflow. 4623 usleep(kRecordThreadSleepUs); 4624 } 4625 } 4626 // enable changes in effect chain 4627 unlockEffectChains(effectChains); 4628 effectChains.clear(); 4629 } 4630 4631 standby(); 4632 4633 { 4634 Mutex::Autolock _l(mLock); 4635 for (size_t i = 0; i < mTracks.size(); i++) { 4636 sp<RecordTrack> track = mTracks[i]; 4637 track->invalidate(); 4638 } 4639 mActiveTrack.clear(); 4640 mStartStopCond.broadcast(); 4641 } 4642 4643 releaseWakeLock(); 4644 4645 ALOGV("RecordThread %p exiting", this); 4646 return false; 4647} 4648 4649void AudioFlinger::RecordThread::standby() 4650{ 4651 if (!mStandby) { 4652 inputStandBy(); 4653 mStandby = true; 4654 } 4655} 4656 4657void AudioFlinger::RecordThread::inputStandBy() 4658{ 4659 mInput->stream->common.standby(&mInput->stream->common); 4660} 4661 4662sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4663 const sp<AudioFlinger::Client>& client, 4664 uint32_t sampleRate, 4665 audio_format_t format, 4666 audio_channel_mask_t channelMask, 4667 size_t frameCount, 4668 int sessionId, 4669 int uid, 4670 IAudioFlinger::track_flags_t *flags, 4671 pid_t tid, 4672 status_t *status) 4673{ 4674 sp<RecordTrack> track; 4675 status_t lStatus; 4676 4677 lStatus = initCheck(); 4678 if (lStatus != NO_ERROR) { 4679 ALOGE("createRecordTrack_l() audio driver not initialized"); 4680 goto Exit; 4681 } 4682 // client expresses a preference for FAST, but we get the final say 4683 if (*flags & IAudioFlinger::TRACK_FAST) { 4684 if ( 4685 // use case: callback handler and frame count is default or at least as large as HAL 4686 ( 4687 (tid != -1) && 4688 ((frameCount == 0) || 4689 (frameCount >= mFrameCount)) 4690 ) && 4691 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4692 // mono or stereo 4693 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4694 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4695 // hardware sample rate 4696 (sampleRate == mSampleRate) && 4697 // record thread has an associated fast recorder 4698 hasFastRecorder() 4699 // FIXME test that RecordThread for this fast track has a capable output HAL 4700 // FIXME add a permission test also? 4701 ) { 4702 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4703 if (frameCount == 0) { 4704 frameCount = mFrameCount * kFastTrackMultiplier; 4705 } 4706 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4707 frameCount, mFrameCount); 4708 } else { 4709 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4710 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4711 "hasFastRecorder=%d tid=%d", 4712 frameCount, mFrameCount, format, 4713 audio_is_linear_pcm(format), 4714 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4715 *flags &= ~IAudioFlinger::TRACK_FAST; 4716 // For compatibility with AudioRecord calculation, buffer depth is forced 4717 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4718 // This is probably too conservative, but legacy application code may depend on it. 4719 // If you change this calculation, also review the start threshold which is related. 4720 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4721 size_t mNormalFrameCount = 2048; // FIXME 4722 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4723 if (minBufCount < 2) { 4724 minBufCount = 2; 4725 } 4726 size_t minFrameCount = mNormalFrameCount * minBufCount; 4727 if (frameCount < minFrameCount) { 4728 frameCount = minFrameCount; 4729 } 4730 } 4731 } 4732 4733 // FIXME use flags and tid similar to createTrack_l() 4734 4735 { // scope for mLock 4736 Mutex::Autolock _l(mLock); 4737 4738 track = new RecordTrack(this, client, sampleRate, 4739 format, channelMask, frameCount, sessionId, uid); 4740 4741 if (track->getCblk() == 0) { 4742 ALOGE("createRecordTrack_l() no control block"); 4743 lStatus = NO_MEMORY; 4744 track.clear(); 4745 goto Exit; 4746 } 4747 mTracks.add(track); 4748 4749 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4750 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4751 mAudioFlinger->btNrecIsOff(); 4752 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4753 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4754 4755 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4756 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4757 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4758 // so ask activity manager to do this on our behalf 4759 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4760 } 4761 } 4762 lStatus = NO_ERROR; 4763 4764Exit: 4765 if (status) { 4766 *status = lStatus; 4767 } 4768 return track; 4769} 4770 4771status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4772 AudioSystem::sync_event_t event, 4773 int triggerSession) 4774{ 4775 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4776 sp<ThreadBase> strongMe = this; 4777 status_t status = NO_ERROR; 4778 4779 if (event == AudioSystem::SYNC_EVENT_NONE) { 4780 clearSyncStartEvent(); 4781 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4782 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4783 triggerSession, 4784 recordTrack->sessionId(), 4785 syncStartEventCallback, 4786 this); 4787 // Sync event can be cancelled by the trigger session if the track is not in a 4788 // compatible state in which case we start record immediately 4789 if (mSyncStartEvent->isCancelled()) { 4790 clearSyncStartEvent(); 4791 } else { 4792 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4793 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4794 } 4795 } 4796 4797 { 4798 AutoMutex lock(mLock); 4799 if (mActiveTrack != 0) { 4800 if (recordTrack != mActiveTrack.get()) { 4801 status = -EBUSY; 4802 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4803 mActiveTrack->mState = TrackBase::ACTIVE; 4804 } 4805 return status; 4806 } 4807 4808 recordTrack->mState = TrackBase::IDLE; 4809 mActiveTrack = recordTrack; 4810 mLock.unlock(); 4811 status_t status = AudioSystem::startInput(mId); 4812 mLock.lock(); 4813 if (status != NO_ERROR) { 4814 mActiveTrack.clear(); 4815 clearSyncStartEvent(); 4816 return status; 4817 } 4818 mRsmpInIndex = mFrameCount; 4819 mBytesRead = 0; 4820 if (mResampler != NULL) { 4821 mResampler->reset(); 4822 } 4823 mActiveTrack->mState = TrackBase::RESUMING; 4824 // signal thread to start 4825 ALOGV("Signal record thread"); 4826 mWaitWorkCV.broadcast(); 4827 // do not wait for mStartStopCond if exiting 4828 if (exitPending()) { 4829 mActiveTrack.clear(); 4830 status = INVALID_OPERATION; 4831 goto startError; 4832 } 4833 mStartStopCond.wait(mLock); 4834 if (mActiveTrack == 0) { 4835 ALOGV("Record failed to start"); 4836 status = BAD_VALUE; 4837 goto startError; 4838 } 4839 ALOGV("Record started OK"); 4840 return status; 4841 } 4842 4843startError: 4844 AudioSystem::stopInput(mId); 4845 clearSyncStartEvent(); 4846 return status; 4847} 4848 4849void AudioFlinger::RecordThread::clearSyncStartEvent() 4850{ 4851 if (mSyncStartEvent != 0) { 4852 mSyncStartEvent->cancel(); 4853 } 4854 mSyncStartEvent.clear(); 4855 mFramestoDrop = 0; 4856} 4857 4858void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4859{ 4860 sp<SyncEvent> strongEvent = event.promote(); 4861 4862 if (strongEvent != 0) { 4863 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4864 me->handleSyncStartEvent(strongEvent); 4865 } 4866} 4867 4868void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4869{ 4870 if (event == mSyncStartEvent) { 4871 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4872 // from audio HAL 4873 mFramestoDrop = mFrameCount * 2; 4874 } 4875} 4876 4877bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4878 ALOGV("RecordThread::stop"); 4879 AutoMutex _l(mLock); 4880 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4881 return false; 4882 } 4883 recordTrack->mState = TrackBase::PAUSING; 4884 // do not wait for mStartStopCond if exiting 4885 if (exitPending()) { 4886 return true; 4887 } 4888 mStartStopCond.wait(mLock); 4889 // if we have been restarted, recordTrack == mActiveTrack.get() here 4890 if (exitPending() || recordTrack != mActiveTrack.get()) { 4891 ALOGV("Record stopped OK"); 4892 return true; 4893 } 4894 return false; 4895} 4896 4897bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4898{ 4899 return false; 4900} 4901 4902status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4903{ 4904#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4905 if (!isValidSyncEvent(event)) { 4906 return BAD_VALUE; 4907 } 4908 4909 int eventSession = event->triggerSession(); 4910 status_t ret = NAME_NOT_FOUND; 4911 4912 Mutex::Autolock _l(mLock); 4913 4914 for (size_t i = 0; i < mTracks.size(); i++) { 4915 sp<RecordTrack> track = mTracks[i]; 4916 if (eventSession == track->sessionId()) { 4917 (void) track->setSyncEvent(event); 4918 ret = NO_ERROR; 4919 } 4920 } 4921 return ret; 4922#else 4923 return BAD_VALUE; 4924#endif 4925} 4926 4927// destroyTrack_l() must be called with ThreadBase::mLock held 4928void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4929{ 4930 track->terminate(); 4931 track->mState = TrackBase::STOPPED; 4932 // active tracks are removed by threadLoop() 4933 if (mActiveTrack != track) { 4934 removeTrack_l(track); 4935 } 4936} 4937 4938void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4939{ 4940 mTracks.remove(track); 4941 // need anything related to effects here? 4942} 4943 4944void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4945{ 4946 dumpInternals(fd, args); 4947 dumpTracks(fd, args); 4948 dumpEffectChains(fd, args); 4949} 4950 4951void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4952{ 4953 const size_t SIZE = 256; 4954 char buffer[SIZE]; 4955 String8 result; 4956 4957 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4958 result.append(buffer); 4959 4960 if (mActiveTrack != 0) { 4961 snprintf(buffer, SIZE, "In index: %zu\n", mRsmpInIndex); 4962 result.append(buffer); 4963 snprintf(buffer, SIZE, "Buffer size: %zu bytes\n", mBufferSize); 4964 result.append(buffer); 4965 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4966 result.append(buffer); 4967 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4968 result.append(buffer); 4969 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4970 result.append(buffer); 4971 } else { 4972 result.append("No active record client\n"); 4973 } 4974 4975 write(fd, result.string(), result.size()); 4976 4977 dumpBase(fd, args); 4978} 4979 4980void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4981{ 4982 const size_t SIZE = 256; 4983 char buffer[SIZE]; 4984 String8 result; 4985 4986 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4987 result.append(buffer); 4988 RecordTrack::appendDumpHeader(result); 4989 for (size_t i = 0; i < mTracks.size(); ++i) { 4990 sp<RecordTrack> track = mTracks[i]; 4991 if (track != 0) { 4992 track->dump(buffer, SIZE); 4993 result.append(buffer); 4994 } 4995 } 4996 4997 if (mActiveTrack != 0) { 4998 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4999 result.append(buffer); 5000 RecordTrack::appendDumpHeader(result); 5001 mActiveTrack->dump(buffer, SIZE); 5002 result.append(buffer); 5003 5004 } 5005 write(fd, result.string(), result.size()); 5006} 5007 5008// AudioBufferProvider interface 5009status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5010{ 5011 size_t framesReq = buffer->frameCount; 5012 size_t framesReady = mFrameCount - mRsmpInIndex; 5013 int channelCount; 5014 5015 if (framesReady == 0) { 5016 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 5017 if (mBytesRead <= 0) { 5018 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 5019 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5020 // Force input into standby so that it tries to 5021 // recover at next read attempt 5022 inputStandBy(); 5023 usleep(kRecordThreadSleepUs); 5024 } 5025 buffer->raw = NULL; 5026 buffer->frameCount = 0; 5027 return NOT_ENOUGH_DATA; 5028 } 5029 mRsmpInIndex = 0; 5030 framesReady = mFrameCount; 5031 } 5032 5033 if (framesReq > framesReady) { 5034 framesReq = framesReady; 5035 } 5036 5037 if (mChannelCount == 1 && mReqChannelCount == 2) { 5038 channelCount = 1; 5039 } else { 5040 channelCount = 2; 5041 } 5042 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5043 buffer->frameCount = framesReq; 5044 return NO_ERROR; 5045} 5046 5047// AudioBufferProvider interface 5048void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5049{ 5050 mRsmpInIndex += buffer->frameCount; 5051 buffer->frameCount = 0; 5052} 5053 5054bool AudioFlinger::RecordThread::checkForNewParameters_l() 5055{ 5056 bool reconfig = false; 5057 5058 while (!mNewParameters.isEmpty()) { 5059 status_t status = NO_ERROR; 5060 String8 keyValuePair = mNewParameters[0]; 5061 AudioParameter param = AudioParameter(keyValuePair); 5062 int value; 5063 audio_format_t reqFormat = mFormat; 5064 uint32_t reqSamplingRate = mReqSampleRate; 5065 uint32_t reqChannelCount = mReqChannelCount; 5066 5067 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5068 reqSamplingRate = value; 5069 reconfig = true; 5070 } 5071 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5072 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5073 status = BAD_VALUE; 5074 } else { 5075 reqFormat = (audio_format_t) value; 5076 reconfig = true; 5077 } 5078 } 5079 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5080 reqChannelCount = popcount(value); 5081 reconfig = true; 5082 } 5083 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5084 // do not accept frame count changes if tracks are open as the track buffer 5085 // size depends on frame count and correct behavior would not be guaranteed 5086 // if frame count is changed after track creation 5087 if (mActiveTrack != 0) { 5088 status = INVALID_OPERATION; 5089 } else { 5090 reconfig = true; 5091 } 5092 } 5093 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5094 // forward device change to effects that have requested to be 5095 // aware of attached audio device. 5096 for (size_t i = 0; i < mEffectChains.size(); i++) { 5097 mEffectChains[i]->setDevice_l(value); 5098 } 5099 5100 // store input device and output device but do not forward output device to audio HAL. 5101 // Note that status is ignored by the caller for output device 5102 // (see AudioFlinger::setParameters() 5103 if (audio_is_output_devices(value)) { 5104 mOutDevice = value; 5105 status = BAD_VALUE; 5106 } else { 5107 mInDevice = value; 5108 // disable AEC and NS if the device is a BT SCO headset supporting those 5109 // pre processings 5110 if (mTracks.size() > 0) { 5111 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5112 mAudioFlinger->btNrecIsOff(); 5113 for (size_t i = 0; i < mTracks.size(); i++) { 5114 sp<RecordTrack> track = mTracks[i]; 5115 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5116 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5117 } 5118 } 5119 } 5120 } 5121 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5122 mAudioSource != (audio_source_t)value) { 5123 // forward device change to effects that have requested to be 5124 // aware of attached audio device. 5125 for (size_t i = 0; i < mEffectChains.size(); i++) { 5126 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5127 } 5128 mAudioSource = (audio_source_t)value; 5129 } 5130 if (status == NO_ERROR) { 5131 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5132 keyValuePair.string()); 5133 if (status == INVALID_OPERATION) { 5134 inputStandBy(); 5135 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5136 keyValuePair.string()); 5137 } 5138 if (reconfig) { 5139 if (status == BAD_VALUE && 5140 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5141 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5142 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5143 <= (2 * reqSamplingRate)) && 5144 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5145 <= FCC_2 && 5146 (reqChannelCount <= FCC_2)) { 5147 status = NO_ERROR; 5148 } 5149 if (status == NO_ERROR) { 5150 readInputParameters(); 5151 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5152 } 5153 } 5154 } 5155 5156 mNewParameters.removeAt(0); 5157 5158 mParamStatus = status; 5159 mParamCond.signal(); 5160 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5161 // already timed out waiting for the status and will never signal the condition. 5162 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5163 } 5164 return reconfig; 5165} 5166 5167String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5168{ 5169 Mutex::Autolock _l(mLock); 5170 if (initCheck() != NO_ERROR) { 5171 return String8(); 5172 } 5173 5174 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5175 const String8 out_s8(s); 5176 free(s); 5177 return out_s8; 5178} 5179 5180void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5181 AudioSystem::OutputDescriptor desc; 5182 void *param2 = NULL; 5183 5184 switch (event) { 5185 case AudioSystem::INPUT_OPENED: 5186 case AudioSystem::INPUT_CONFIG_CHANGED: 5187 desc.channelMask = mChannelMask; 5188 desc.samplingRate = mSampleRate; 5189 desc.format = mFormat; 5190 desc.frameCount = mFrameCount; 5191 desc.latency = 0; 5192 param2 = &desc; 5193 break; 5194 5195 case AudioSystem::INPUT_CLOSED: 5196 default: 5197 break; 5198 } 5199 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5200} 5201 5202void AudioFlinger::RecordThread::readInputParameters() 5203{ 5204 delete[] mRsmpInBuffer; 5205 // mRsmpInBuffer is always assigned a new[] below 5206 delete[] mRsmpOutBuffer; 5207 mRsmpOutBuffer = NULL; 5208 delete mResampler; 5209 mResampler = NULL; 5210 5211 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5212 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5213 mChannelCount = popcount(mChannelMask); 5214 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5215 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5216 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5217 } 5218 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5219 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5220 mFrameCount = mBufferSize / mFrameSize; 5221 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5222 5223 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5224 { 5225 int channelCount; 5226 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5227 // stereo to mono post process as the resampler always outputs stereo. 5228 if (mChannelCount == 1 && mReqChannelCount == 2) { 5229 channelCount = 1; 5230 } else { 5231 channelCount = 2; 5232 } 5233 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5234 mResampler->setSampleRate(mSampleRate); 5235 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5236 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5237 5238 // optmization: if mono to mono, alter input frame count as if we were inputing 5239 // stereo samples 5240 if (mChannelCount == 1 && mReqChannelCount == 1) { 5241 mFrameCount >>= 1; 5242 } 5243 5244 } 5245 mRsmpInIndex = mFrameCount; 5246} 5247 5248unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5249{ 5250 Mutex::Autolock _l(mLock); 5251 if (initCheck() != NO_ERROR) { 5252 return 0; 5253 } 5254 5255 return mInput->stream->get_input_frames_lost(mInput->stream); 5256} 5257 5258uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5259{ 5260 Mutex::Autolock _l(mLock); 5261 uint32_t result = 0; 5262 if (getEffectChain_l(sessionId) != 0) { 5263 result = EFFECT_SESSION; 5264 } 5265 5266 for (size_t i = 0; i < mTracks.size(); ++i) { 5267 if (sessionId == mTracks[i]->sessionId()) { 5268 result |= TRACK_SESSION; 5269 break; 5270 } 5271 } 5272 5273 return result; 5274} 5275 5276KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5277{ 5278 KeyedVector<int, bool> ids; 5279 Mutex::Autolock _l(mLock); 5280 for (size_t j = 0; j < mTracks.size(); ++j) { 5281 sp<RecordThread::RecordTrack> track = mTracks[j]; 5282 int sessionId = track->sessionId(); 5283 if (ids.indexOfKey(sessionId) < 0) { 5284 ids.add(sessionId, true); 5285 } 5286 } 5287 return ids; 5288} 5289 5290AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5291{ 5292 Mutex::Autolock _l(mLock); 5293 AudioStreamIn *input = mInput; 5294 mInput = NULL; 5295 return input; 5296} 5297 5298// this method must always be called either with ThreadBase mLock held or inside the thread loop 5299audio_stream_t* AudioFlinger::RecordThread::stream() const 5300{ 5301 if (mInput == NULL) { 5302 return NULL; 5303 } 5304 return &mInput->stream->common; 5305} 5306 5307status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5308{ 5309 // only one chain per input thread 5310 if (mEffectChains.size() != 0) { 5311 return INVALID_OPERATION; 5312 } 5313 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5314 5315 chain->setInBuffer(NULL); 5316 chain->setOutBuffer(NULL); 5317 5318 checkSuspendOnAddEffectChain_l(chain); 5319 5320 mEffectChains.add(chain); 5321 5322 return NO_ERROR; 5323} 5324 5325size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5326{ 5327 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5328 ALOGW_IF(mEffectChains.size() != 1, 5329 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5330 chain.get(), mEffectChains.size(), this); 5331 if (mEffectChains.size() == 1) { 5332 mEffectChains.removeAt(0); 5333 } 5334 return 0; 5335} 5336 5337}; // namespace android 5338