Threads.cpp revision 56604aa3a56dc8e15532597a0a74b3c7b165e006
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 // track must be cleared from the caller as the caller has the AF lock 1330 goto Exit; 1331 } 1332 1333 mTracks.add(track); 1334 1335 sp<EffectChain> chain = getEffectChain_l(sessionId); 1336 if (chain != 0) { 1337 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1338 track->setMainBuffer(chain->inBuffer()); 1339 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1340 chain->incTrackCnt(); 1341 } 1342 1343 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1344 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1345 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1346 // so ask activity manager to do this on our behalf 1347 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1348 } 1349 } 1350 1351 lStatus = NO_ERROR; 1352 1353Exit: 1354 if (status) { 1355 *status = lStatus; 1356 } 1357 return track; 1358} 1359 1360uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1361{ 1362 return latency; 1363} 1364 1365uint32_t AudioFlinger::PlaybackThread::latency() const 1366{ 1367 Mutex::Autolock _l(mLock); 1368 return latency_l(); 1369} 1370uint32_t AudioFlinger::PlaybackThread::latency_l() const 1371{ 1372 if (initCheck() == NO_ERROR) { 1373 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1374 } else { 1375 return 0; 1376 } 1377} 1378 1379void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1380{ 1381 Mutex::Autolock _l(mLock); 1382 // Don't apply master volume in SW if our HAL can do it for us. 1383 if (mOutput && mOutput->audioHwDev && 1384 mOutput->audioHwDev->canSetMasterVolume()) { 1385 mMasterVolume = 1.0; 1386 } else { 1387 mMasterVolume = value; 1388 } 1389} 1390 1391void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1392{ 1393 Mutex::Autolock _l(mLock); 1394 // Don't apply master mute in SW if our HAL can do it for us. 1395 if (mOutput && mOutput->audioHwDev && 1396 mOutput->audioHwDev->canSetMasterMute()) { 1397 mMasterMute = false; 1398 } else { 1399 mMasterMute = muted; 1400 } 1401} 1402 1403void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1404{ 1405 Mutex::Autolock _l(mLock); 1406 mStreamTypes[stream].volume = value; 1407 broadcast_l(); 1408} 1409 1410void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1411{ 1412 Mutex::Autolock _l(mLock); 1413 mStreamTypes[stream].mute = muted; 1414 broadcast_l(); 1415} 1416 1417float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1418{ 1419 Mutex::Autolock _l(mLock); 1420 return mStreamTypes[stream].volume; 1421} 1422 1423// addTrack_l() must be called with ThreadBase::mLock held 1424status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1425{ 1426 status_t status = ALREADY_EXISTS; 1427 1428 // set retry count for buffer fill 1429 track->mRetryCount = kMaxTrackStartupRetries; 1430 if (mActiveTracks.indexOf(track) < 0) { 1431 // the track is newly added, make sure it fills up all its 1432 // buffers before playing. This is to ensure the client will 1433 // effectively get the latency it requested. 1434 if (!track->isOutputTrack()) { 1435 TrackBase::track_state state = track->mState; 1436 mLock.unlock(); 1437 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1438 mLock.lock(); 1439 // abort track was stopped/paused while we released the lock 1440 if (state != track->mState) { 1441 if (status == NO_ERROR) { 1442 mLock.unlock(); 1443 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1444 mLock.lock(); 1445 } 1446 return INVALID_OPERATION; 1447 } 1448 // abort if start is rejected by audio policy manager 1449 if (status != NO_ERROR) { 1450 return PERMISSION_DENIED; 1451 } 1452#ifdef ADD_BATTERY_DATA 1453 // to track the speaker usage 1454 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1455#endif 1456 } 1457 1458 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1459 track->mResetDone = false; 1460 track->mPresentationCompleteFrames = 0; 1461 mActiveTracks.add(track); 1462 mWakeLockUids.add(track->uid()); 1463 mActiveTracksGeneration++; 1464 mLatestActiveTrack = track; 1465 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1466 if (chain != 0) { 1467 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1468 track->sessionId()); 1469 chain->incActiveTrackCnt(); 1470 } 1471 1472 status = NO_ERROR; 1473 } 1474 1475 ALOGV("signal playback thread"); 1476 broadcast_l(); 1477 1478 return status; 1479} 1480 1481bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1482{ 1483 track->terminate(); 1484 // active tracks are removed by threadLoop() 1485 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1486 track->mState = TrackBase::STOPPED; 1487 if (!trackActive) { 1488 removeTrack_l(track); 1489 } else if (track->isFastTrack() || track->isOffloaded()) { 1490 track->mState = TrackBase::STOPPING_1; 1491 } 1492 1493 return trackActive; 1494} 1495 1496void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1497{ 1498 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1499 mTracks.remove(track); 1500 deleteTrackName_l(track->name()); 1501 // redundant as track is about to be destroyed, for dumpsys only 1502 track->mName = -1; 1503 if (track->isFastTrack()) { 1504 int index = track->mFastIndex; 1505 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1506 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1507 mFastTrackAvailMask |= 1 << index; 1508 // redundant as track is about to be destroyed, for dumpsys only 1509 track->mFastIndex = -1; 1510 } 1511 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1512 if (chain != 0) { 1513 chain->decTrackCnt(); 1514 } 1515} 1516 1517void AudioFlinger::PlaybackThread::broadcast_l() 1518{ 1519 // Thread could be blocked waiting for async 1520 // so signal it to handle state changes immediately 1521 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1522 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1523 mSignalPending = true; 1524 mWaitWorkCV.broadcast(); 1525} 1526 1527String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1528{ 1529 Mutex::Autolock _l(mLock); 1530 if (initCheck() != NO_ERROR) { 1531 return String8(); 1532 } 1533 1534 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1535 const String8 out_s8(s); 1536 free(s); 1537 return out_s8; 1538} 1539 1540// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1541void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1542 AudioSystem::OutputDescriptor desc; 1543 void *param2 = NULL; 1544 1545 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1546 param); 1547 1548 switch (event) { 1549 case AudioSystem::OUTPUT_OPENED: 1550 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1551 desc.channelMask = mChannelMask; 1552 desc.samplingRate = mSampleRate; 1553 desc.format = mFormat; 1554 desc.frameCount = mNormalFrameCount; // FIXME see 1555 // AudioFlinger::frameCount(audio_io_handle_t) 1556 desc.latency = latency(); 1557 param2 = &desc; 1558 break; 1559 1560 case AudioSystem::STREAM_CONFIG_CHANGED: 1561 param2 = ¶m; 1562 case AudioSystem::OUTPUT_CLOSED: 1563 default: 1564 break; 1565 } 1566 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1567} 1568 1569void AudioFlinger::PlaybackThread::writeCallback() 1570{ 1571 ALOG_ASSERT(mCallbackThread != 0); 1572 mCallbackThread->resetWriteBlocked(); 1573} 1574 1575void AudioFlinger::PlaybackThread::drainCallback() 1576{ 1577 ALOG_ASSERT(mCallbackThread != 0); 1578 mCallbackThread->resetDraining(); 1579} 1580 1581void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1582{ 1583 Mutex::Autolock _l(mLock); 1584 // reject out of sequence requests 1585 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1586 mWriteAckSequence &= ~1; 1587 mWaitWorkCV.signal(); 1588 } 1589} 1590 1591void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1592{ 1593 Mutex::Autolock _l(mLock); 1594 // reject out of sequence requests 1595 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1596 mDrainSequence &= ~1; 1597 mWaitWorkCV.signal(); 1598 } 1599} 1600 1601// static 1602int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1603 void *param, 1604 void *cookie) 1605{ 1606 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1607 ALOGV("asyncCallback() event %d", event); 1608 switch (event) { 1609 case STREAM_CBK_EVENT_WRITE_READY: 1610 me->writeCallback(); 1611 break; 1612 case STREAM_CBK_EVENT_DRAIN_READY: 1613 me->drainCallback(); 1614 break; 1615 default: 1616 ALOGW("asyncCallback() unknown event %d", event); 1617 break; 1618 } 1619 return 0; 1620} 1621 1622void AudioFlinger::PlaybackThread::readOutputParameters() 1623{ 1624 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1625 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1626 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1627 if (!audio_is_output_channel(mChannelMask)) { 1628 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1629 } 1630 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1631 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1632 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1633 } 1634 mChannelCount = popcount(mChannelMask); 1635 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1636 if (!audio_is_valid_format(mFormat)) { 1637 LOG_FATAL("HAL format %d not valid for output", mFormat); 1638 } 1639 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1640 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1641 mFormat); 1642 } 1643 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1644 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1645 if (mFrameCount & 15) { 1646 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1647 mFrameCount); 1648 } 1649 1650 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1651 (mOutput->stream->set_callback != NULL)) { 1652 if (mOutput->stream->set_callback(mOutput->stream, 1653 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1654 mUseAsyncWrite = true; 1655 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1656 } 1657 } 1658 1659 // Calculate size of normal mix buffer relative to the HAL output buffer size 1660 double multiplier = 1.0; 1661 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1662 kUseFastMixer == FastMixer_Dynamic)) { 1663 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1664 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1665 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1666 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1667 maxNormalFrameCount = maxNormalFrameCount & ~15; 1668 if (maxNormalFrameCount < minNormalFrameCount) { 1669 maxNormalFrameCount = minNormalFrameCount; 1670 } 1671 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1672 if (multiplier <= 1.0) { 1673 multiplier = 1.0; 1674 } else if (multiplier <= 2.0) { 1675 if (2 * mFrameCount <= maxNormalFrameCount) { 1676 multiplier = 2.0; 1677 } else { 1678 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1679 } 1680 } else { 1681 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1682 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1683 // track, but we sometimes have to do this to satisfy the maximum frame count 1684 // constraint) 1685 // FIXME this rounding up should not be done if no HAL SRC 1686 uint32_t truncMult = (uint32_t) multiplier; 1687 if ((truncMult & 1)) { 1688 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1689 ++truncMult; 1690 } 1691 } 1692 multiplier = (double) truncMult; 1693 } 1694 } 1695 mNormalFrameCount = multiplier * mFrameCount; 1696 // round up to nearest 16 frames to satisfy AudioMixer 1697 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1698 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1699 mNormalFrameCount); 1700 1701 delete[] mAllocMixBuffer; 1702 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1703 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1704 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1705 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1706 1707 // force reconfiguration of effect chains and engines to take new buffer size and audio 1708 // parameters into account 1709 // Note that mLock is not held when readOutputParameters() is called from the constructor 1710 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1711 // matter. 1712 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1713 Vector< sp<EffectChain> > effectChains = mEffectChains; 1714 for (size_t i = 0; i < effectChains.size(); i ++) { 1715 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1716 } 1717} 1718 1719 1720status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1721{ 1722 if (halFrames == NULL || dspFrames == NULL) { 1723 return BAD_VALUE; 1724 } 1725 Mutex::Autolock _l(mLock); 1726 if (initCheck() != NO_ERROR) { 1727 return INVALID_OPERATION; 1728 } 1729 size_t framesWritten = mBytesWritten / mFrameSize; 1730 *halFrames = framesWritten; 1731 1732 if (isSuspended()) { 1733 // return an estimation of rendered frames when the output is suspended 1734 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1735 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1736 return NO_ERROR; 1737 } else { 1738 status_t status; 1739 uint32_t frames; 1740 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1741 *dspFrames = (size_t)frames; 1742 return status; 1743 } 1744} 1745 1746uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1747{ 1748 Mutex::Autolock _l(mLock); 1749 uint32_t result = 0; 1750 if (getEffectChain_l(sessionId) != 0) { 1751 result = EFFECT_SESSION; 1752 } 1753 1754 for (size_t i = 0; i < mTracks.size(); ++i) { 1755 sp<Track> track = mTracks[i]; 1756 if (sessionId == track->sessionId() && !track->isInvalid()) { 1757 result |= TRACK_SESSION; 1758 break; 1759 } 1760 } 1761 1762 return result; 1763} 1764 1765uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1766{ 1767 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1768 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1769 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1770 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1771 } 1772 for (size_t i = 0; i < mTracks.size(); i++) { 1773 sp<Track> track = mTracks[i]; 1774 if (sessionId == track->sessionId() && !track->isInvalid()) { 1775 return AudioSystem::getStrategyForStream(track->streamType()); 1776 } 1777 } 1778 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1779} 1780 1781 1782AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1783{ 1784 Mutex::Autolock _l(mLock); 1785 return mOutput; 1786} 1787 1788AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1789{ 1790 Mutex::Autolock _l(mLock); 1791 AudioStreamOut *output = mOutput; 1792 mOutput = NULL; 1793 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1794 // must push a NULL and wait for ack 1795 mOutputSink.clear(); 1796 mPipeSink.clear(); 1797 mNormalSink.clear(); 1798 return output; 1799} 1800 1801// this method must always be called either with ThreadBase mLock held or inside the thread loop 1802audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1803{ 1804 if (mOutput == NULL) { 1805 return NULL; 1806 } 1807 return &mOutput->stream->common; 1808} 1809 1810uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1811{ 1812 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1813} 1814 1815status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1816{ 1817 if (!isValidSyncEvent(event)) { 1818 return BAD_VALUE; 1819 } 1820 1821 Mutex::Autolock _l(mLock); 1822 1823 for (size_t i = 0; i < mTracks.size(); ++i) { 1824 sp<Track> track = mTracks[i]; 1825 if (event->triggerSession() == track->sessionId()) { 1826 (void) track->setSyncEvent(event); 1827 return NO_ERROR; 1828 } 1829 } 1830 1831 return NAME_NOT_FOUND; 1832} 1833 1834bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1835{ 1836 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1837} 1838 1839void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1840 const Vector< sp<Track> >& tracksToRemove) 1841{ 1842 size_t count = tracksToRemove.size(); 1843 if (count) { 1844 for (size_t i = 0 ; i < count ; i++) { 1845 const sp<Track>& track = tracksToRemove.itemAt(i); 1846 if (!track->isOutputTrack()) { 1847 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1848#ifdef ADD_BATTERY_DATA 1849 // to track the speaker usage 1850 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1851#endif 1852 if (track->isTerminated()) { 1853 AudioSystem::releaseOutput(mId); 1854 } 1855 } 1856 } 1857 } 1858} 1859 1860void AudioFlinger::PlaybackThread::checkSilentMode_l() 1861{ 1862 if (!mMasterMute) { 1863 char value[PROPERTY_VALUE_MAX]; 1864 if (property_get("ro.audio.silent", value, "0") > 0) { 1865 char *endptr; 1866 unsigned long ul = strtoul(value, &endptr, 0); 1867 if (*endptr == '\0' && ul != 0) { 1868 ALOGD("Silence is golden"); 1869 // The setprop command will not allow a property to be changed after 1870 // the first time it is set, so we don't have to worry about un-muting. 1871 setMasterMute_l(true); 1872 } 1873 } 1874 } 1875} 1876 1877// shared by MIXER and DIRECT, overridden by DUPLICATING 1878ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1879{ 1880 // FIXME rewrite to reduce number of system calls 1881 mLastWriteTime = systemTime(); 1882 mInWrite = true; 1883 ssize_t bytesWritten; 1884 1885 // If an NBAIO sink is present, use it to write the normal mixer's submix 1886 if (mNormalSink != 0) { 1887#define mBitShift 2 // FIXME 1888 size_t count = mBytesRemaining >> mBitShift; 1889 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1890 ATRACE_BEGIN("write"); 1891 // update the setpoint when AudioFlinger::mScreenState changes 1892 uint32_t screenState = AudioFlinger::mScreenState; 1893 if (screenState != mScreenState) { 1894 mScreenState = screenState; 1895 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1896 if (pipe != NULL) { 1897 pipe->setAvgFrames((mScreenState & 1) ? 1898 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1899 } 1900 } 1901 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1902 ATRACE_END(); 1903 if (framesWritten > 0) { 1904 bytesWritten = framesWritten << mBitShift; 1905 } else { 1906 bytesWritten = framesWritten; 1907 } 1908 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1909 if (status == NO_ERROR) { 1910 size_t totalFramesWritten = mNormalSink->framesWritten(); 1911 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1912 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1913 mLatchDValid = true; 1914 } 1915 } 1916 // otherwise use the HAL / AudioStreamOut directly 1917 } else { 1918 // Direct output and offload threads 1919 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1920 if (mUseAsyncWrite) { 1921 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1922 mWriteAckSequence += 2; 1923 mWriteAckSequence |= 1; 1924 ALOG_ASSERT(mCallbackThread != 0); 1925 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1926 } 1927 // FIXME We should have an implementation of timestamps for direct output threads. 1928 // They are used e.g for multichannel PCM playback over HDMI. 1929 bytesWritten = mOutput->stream->write(mOutput->stream, 1930 mMixBuffer + offset, mBytesRemaining); 1931 if (mUseAsyncWrite && 1932 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1933 // do not wait for async callback in case of error of full write 1934 mWriteAckSequence &= ~1; 1935 ALOG_ASSERT(mCallbackThread != 0); 1936 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1937 } 1938 } 1939 1940 mNumWrites++; 1941 mInWrite = false; 1942 mStandby = false; 1943 return bytesWritten; 1944} 1945 1946void AudioFlinger::PlaybackThread::threadLoop_drain() 1947{ 1948 if (mOutput->stream->drain) { 1949 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1950 if (mUseAsyncWrite) { 1951 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1952 mDrainSequence |= 1; 1953 ALOG_ASSERT(mCallbackThread != 0); 1954 mCallbackThread->setDraining(mDrainSequence); 1955 } 1956 mOutput->stream->drain(mOutput->stream, 1957 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1958 : AUDIO_DRAIN_ALL); 1959 } 1960} 1961 1962void AudioFlinger::PlaybackThread::threadLoop_exit() 1963{ 1964 // Default implementation has nothing to do 1965} 1966 1967/* 1968The derived values that are cached: 1969 - mixBufferSize from frame count * frame size 1970 - activeSleepTime from activeSleepTimeUs() 1971 - idleSleepTime from idleSleepTimeUs() 1972 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1973 - maxPeriod from frame count and sample rate (MIXER only) 1974 1975The parameters that affect these derived values are: 1976 - frame count 1977 - frame size 1978 - sample rate 1979 - device type: A2DP or not 1980 - device latency 1981 - format: PCM or not 1982 - active sleep time 1983 - idle sleep time 1984*/ 1985 1986void AudioFlinger::PlaybackThread::cacheParameters_l() 1987{ 1988 mixBufferSize = mNormalFrameCount * mFrameSize; 1989 activeSleepTime = activeSleepTimeUs(); 1990 idleSleepTime = idleSleepTimeUs(); 1991} 1992 1993void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1994{ 1995 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1996 this, streamType, mTracks.size()); 1997 Mutex::Autolock _l(mLock); 1998 1999 size_t size = mTracks.size(); 2000 for (size_t i = 0; i < size; i++) { 2001 sp<Track> t = mTracks[i]; 2002 if (t->streamType() == streamType) { 2003 t->invalidate(); 2004 } 2005 } 2006} 2007 2008status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2009{ 2010 int session = chain->sessionId(); 2011 int16_t *buffer = mMixBuffer; 2012 bool ownsBuffer = false; 2013 2014 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2015 if (session > 0) { 2016 // Only one effect chain can be present in direct output thread and it uses 2017 // the mix buffer as input 2018 if (mType != DIRECT) { 2019 size_t numSamples = mNormalFrameCount * mChannelCount; 2020 buffer = new int16_t[numSamples]; 2021 memset(buffer, 0, numSamples * sizeof(int16_t)); 2022 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2023 ownsBuffer = true; 2024 } 2025 2026 // Attach all tracks with same session ID to this chain. 2027 for (size_t i = 0; i < mTracks.size(); ++i) { 2028 sp<Track> track = mTracks[i]; 2029 if (session == track->sessionId()) { 2030 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2031 buffer); 2032 track->setMainBuffer(buffer); 2033 chain->incTrackCnt(); 2034 } 2035 } 2036 2037 // indicate all active tracks in the chain 2038 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2039 sp<Track> track = mActiveTracks[i].promote(); 2040 if (track == 0) { 2041 continue; 2042 } 2043 if (session == track->sessionId()) { 2044 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2045 chain->incActiveTrackCnt(); 2046 } 2047 } 2048 } 2049 2050 chain->setInBuffer(buffer, ownsBuffer); 2051 chain->setOutBuffer(mMixBuffer); 2052 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2053 // chains list in order to be processed last as it contains output stage effects 2054 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2055 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2056 // after track specific effects and before output stage 2057 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2058 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2059 // Effect chain for other sessions are inserted at beginning of effect 2060 // chains list to be processed before output mix effects. Relative order between other 2061 // sessions is not important 2062 size_t size = mEffectChains.size(); 2063 size_t i = 0; 2064 for (i = 0; i < size; i++) { 2065 if (mEffectChains[i]->sessionId() < session) { 2066 break; 2067 } 2068 } 2069 mEffectChains.insertAt(chain, i); 2070 checkSuspendOnAddEffectChain_l(chain); 2071 2072 return NO_ERROR; 2073} 2074 2075size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2076{ 2077 int session = chain->sessionId(); 2078 2079 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2080 2081 for (size_t i = 0; i < mEffectChains.size(); i++) { 2082 if (chain == mEffectChains[i]) { 2083 mEffectChains.removeAt(i); 2084 // detach all active tracks from the chain 2085 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2086 sp<Track> track = mActiveTracks[i].promote(); 2087 if (track == 0) { 2088 continue; 2089 } 2090 if (session == track->sessionId()) { 2091 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2092 chain.get(), session); 2093 chain->decActiveTrackCnt(); 2094 } 2095 } 2096 2097 // detach all tracks with same session ID from this chain 2098 for (size_t i = 0; i < mTracks.size(); ++i) { 2099 sp<Track> track = mTracks[i]; 2100 if (session == track->sessionId()) { 2101 track->setMainBuffer(mMixBuffer); 2102 chain->decTrackCnt(); 2103 } 2104 } 2105 break; 2106 } 2107 } 2108 return mEffectChains.size(); 2109} 2110 2111status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2112 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2113{ 2114 Mutex::Autolock _l(mLock); 2115 return attachAuxEffect_l(track, EffectId); 2116} 2117 2118status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2119 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2120{ 2121 status_t status = NO_ERROR; 2122 2123 if (EffectId == 0) { 2124 track->setAuxBuffer(0, NULL); 2125 } else { 2126 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2127 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2128 if (effect != 0) { 2129 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2130 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2131 } else { 2132 status = INVALID_OPERATION; 2133 } 2134 } else { 2135 status = BAD_VALUE; 2136 } 2137 } 2138 return status; 2139} 2140 2141void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2142{ 2143 for (size_t i = 0; i < mTracks.size(); ++i) { 2144 sp<Track> track = mTracks[i]; 2145 if (track->auxEffectId() == effectId) { 2146 attachAuxEffect_l(track, 0); 2147 } 2148 } 2149} 2150 2151bool AudioFlinger::PlaybackThread::threadLoop() 2152{ 2153 Vector< sp<Track> > tracksToRemove; 2154 2155 standbyTime = systemTime(); 2156 2157 // MIXER 2158 nsecs_t lastWarning = 0; 2159 2160 // DUPLICATING 2161 // FIXME could this be made local to while loop? 2162 writeFrames = 0; 2163 2164 int lastGeneration = 0; 2165 2166 cacheParameters_l(); 2167 sleepTime = idleSleepTime; 2168 2169 if (mType == MIXER) { 2170 sleepTimeShift = 0; 2171 } 2172 2173 CpuStats cpuStats; 2174 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2175 2176 acquireWakeLock(); 2177 2178 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2179 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2180 // and then that string will be logged at the next convenient opportunity. 2181 const char *logString = NULL; 2182 2183 checkSilentMode_l(); 2184 2185 while (!exitPending()) 2186 { 2187 cpuStats.sample(myName); 2188 2189 Vector< sp<EffectChain> > effectChains; 2190 2191 processConfigEvents(); 2192 2193 { // scope for mLock 2194 2195 Mutex::Autolock _l(mLock); 2196 2197 if (logString != NULL) { 2198 mNBLogWriter->logTimestamp(); 2199 mNBLogWriter->log(logString); 2200 logString = NULL; 2201 } 2202 2203 if (mLatchDValid) { 2204 mLatchQ = mLatchD; 2205 mLatchDValid = false; 2206 mLatchQValid = true; 2207 } 2208 2209 if (checkForNewParameters_l()) { 2210 cacheParameters_l(); 2211 } 2212 2213 saveOutputTracks(); 2214 if (mSignalPending) { 2215 // A signal was raised while we were unlocked 2216 mSignalPending = false; 2217 } else if (waitingAsyncCallback_l()) { 2218 if (exitPending()) { 2219 break; 2220 } 2221 releaseWakeLock_l(); 2222 mWakeLockUids.clear(); 2223 mActiveTracksGeneration++; 2224 ALOGV("wait async completion"); 2225 mWaitWorkCV.wait(mLock); 2226 ALOGV("async completion/wake"); 2227 acquireWakeLock_l(); 2228 standbyTime = systemTime() + standbyDelay; 2229 sleepTime = 0; 2230 2231 continue; 2232 } 2233 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2234 isSuspended()) { 2235 // put audio hardware into standby after short delay 2236 if (shouldStandby_l()) { 2237 2238 threadLoop_standby(); 2239 2240 mStandby = true; 2241 } 2242 2243 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2244 // we're about to wait, flush the binder command buffer 2245 IPCThreadState::self()->flushCommands(); 2246 2247 clearOutputTracks(); 2248 2249 if (exitPending()) { 2250 break; 2251 } 2252 2253 releaseWakeLock_l(); 2254 mWakeLockUids.clear(); 2255 mActiveTracksGeneration++; 2256 // wait until we have something to do... 2257 ALOGV("%s going to sleep", myName.string()); 2258 mWaitWorkCV.wait(mLock); 2259 ALOGV("%s waking up", myName.string()); 2260 acquireWakeLock_l(); 2261 2262 mMixerStatus = MIXER_IDLE; 2263 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2264 mBytesWritten = 0; 2265 mBytesRemaining = 0; 2266 checkSilentMode_l(); 2267 2268 standbyTime = systemTime() + standbyDelay; 2269 sleepTime = idleSleepTime; 2270 if (mType == MIXER) { 2271 sleepTimeShift = 0; 2272 } 2273 2274 continue; 2275 } 2276 } 2277 // mMixerStatusIgnoringFastTracks is also updated internally 2278 mMixerStatus = prepareTracks_l(&tracksToRemove); 2279 2280 // compare with previously applied list 2281 if (lastGeneration != mActiveTracksGeneration) { 2282 // update wakelock 2283 updateWakeLockUids_l(mWakeLockUids); 2284 lastGeneration = mActiveTracksGeneration; 2285 } 2286 2287 // prevent any changes in effect chain list and in each effect chain 2288 // during mixing and effect process as the audio buffers could be deleted 2289 // or modified if an effect is created or deleted 2290 lockEffectChains_l(effectChains); 2291 } // mLock scope ends 2292 2293 if (mBytesRemaining == 0) { 2294 mCurrentWriteLength = 0; 2295 if (mMixerStatus == MIXER_TRACKS_READY) { 2296 // threadLoop_mix() sets mCurrentWriteLength 2297 threadLoop_mix(); 2298 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2299 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2300 // threadLoop_sleepTime sets sleepTime to 0 if data 2301 // must be written to HAL 2302 threadLoop_sleepTime(); 2303 if (sleepTime == 0) { 2304 mCurrentWriteLength = mixBufferSize; 2305 } 2306 } 2307 mBytesRemaining = mCurrentWriteLength; 2308 if (isSuspended()) { 2309 sleepTime = suspendSleepTimeUs(); 2310 // simulate write to HAL when suspended 2311 mBytesWritten += mixBufferSize; 2312 mBytesRemaining = 0; 2313 } 2314 2315 // only process effects if we're going to write 2316 if (sleepTime == 0 && mType != OFFLOAD) { 2317 for (size_t i = 0; i < effectChains.size(); i ++) { 2318 effectChains[i]->process_l(); 2319 } 2320 } 2321 } 2322 // Process effect chains for offloaded thread even if no audio 2323 // was read from audio track: process only updates effect state 2324 // and thus does have to be synchronized with audio writes but may have 2325 // to be called while waiting for async write callback 2326 if (mType == OFFLOAD) { 2327 for (size_t i = 0; i < effectChains.size(); i ++) { 2328 effectChains[i]->process_l(); 2329 } 2330 } 2331 2332 // enable changes in effect chain 2333 unlockEffectChains(effectChains); 2334 2335 if (!waitingAsyncCallback()) { 2336 // sleepTime == 0 means we must write to audio hardware 2337 if (sleepTime == 0) { 2338 if (mBytesRemaining) { 2339 ssize_t ret = threadLoop_write(); 2340 if (ret < 0) { 2341 mBytesRemaining = 0; 2342 } else { 2343 mBytesWritten += ret; 2344 mBytesRemaining -= ret; 2345 } 2346 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2347 (mMixerStatus == MIXER_DRAIN_ALL)) { 2348 threadLoop_drain(); 2349 } 2350if (mType == MIXER) { 2351 // write blocked detection 2352 nsecs_t now = systemTime(); 2353 nsecs_t delta = now - mLastWriteTime; 2354 if (!mStandby && delta > maxPeriod) { 2355 mNumDelayedWrites++; 2356 if ((now - lastWarning) > kWarningThrottleNs) { 2357 ATRACE_NAME("underrun"); 2358 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2359 ns2ms(delta), mNumDelayedWrites, this); 2360 lastWarning = now; 2361 } 2362 } 2363} 2364 2365 } else { 2366 usleep(sleepTime); 2367 } 2368 } 2369 2370 // Finally let go of removed track(s), without the lock held 2371 // since we can't guarantee the destructors won't acquire that 2372 // same lock. This will also mutate and push a new fast mixer state. 2373 threadLoop_removeTracks(tracksToRemove); 2374 tracksToRemove.clear(); 2375 2376 // FIXME I don't understand the need for this here; 2377 // it was in the original code but maybe the 2378 // assignment in saveOutputTracks() makes this unnecessary? 2379 clearOutputTracks(); 2380 2381 // Effect chains will be actually deleted here if they were removed from 2382 // mEffectChains list during mixing or effects processing 2383 effectChains.clear(); 2384 2385 // FIXME Note that the above .clear() is no longer necessary since effectChains 2386 // is now local to this block, but will keep it for now (at least until merge done). 2387 } 2388 2389 threadLoop_exit(); 2390 2391 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2392 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2393 // put output stream into standby mode 2394 if (!mStandby) { 2395 mOutput->stream->common.standby(&mOutput->stream->common); 2396 } 2397 } 2398 2399 releaseWakeLock(); 2400 mWakeLockUids.clear(); 2401 mActiveTracksGeneration++; 2402 2403 ALOGV("Thread %p type %d exiting", this, mType); 2404 return false; 2405} 2406 2407// removeTracks_l() must be called with ThreadBase::mLock held 2408void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2409{ 2410 size_t count = tracksToRemove.size(); 2411 if (count) { 2412 for (size_t i=0 ; i<count ; i++) { 2413 const sp<Track>& track = tracksToRemove.itemAt(i); 2414 mActiveTracks.remove(track); 2415 mWakeLockUids.remove(track->uid()); 2416 mActiveTracksGeneration++; 2417 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2418 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2419 if (chain != 0) { 2420 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2421 track->sessionId()); 2422 chain->decActiveTrackCnt(); 2423 } 2424 if (track->isTerminated()) { 2425 removeTrack_l(track); 2426 } 2427 } 2428 } 2429 2430} 2431 2432status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2433{ 2434 if (mNormalSink != 0) { 2435 return mNormalSink->getTimestamp(timestamp); 2436 } 2437 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2438 uint64_t position64; 2439 int ret = mOutput->stream->get_presentation_position( 2440 mOutput->stream, &position64, ×tamp.mTime); 2441 if (ret == 0) { 2442 timestamp.mPosition = (uint32_t)position64; 2443 return NO_ERROR; 2444 } 2445 } 2446 return INVALID_OPERATION; 2447} 2448// ---------------------------------------------------------------------------- 2449 2450AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2451 audio_io_handle_t id, audio_devices_t device, type_t type) 2452 : PlaybackThread(audioFlinger, output, id, device, type), 2453 // mAudioMixer below 2454 // mFastMixer below 2455 mFastMixerFutex(0) 2456 // mOutputSink below 2457 // mPipeSink below 2458 // mNormalSink below 2459{ 2460 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2461 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2462 "mFrameCount=%d, mNormalFrameCount=%d", 2463 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2464 mNormalFrameCount); 2465 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2466 2467 // FIXME - Current mixer implementation only supports stereo output 2468 if (mChannelCount != FCC_2) { 2469 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2470 } 2471 2472 // create an NBAIO sink for the HAL output stream, and negotiate 2473 mOutputSink = new AudioStreamOutSink(output->stream); 2474 size_t numCounterOffers = 0; 2475 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2476 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2477 ALOG_ASSERT(index == 0); 2478 2479 // initialize fast mixer depending on configuration 2480 bool initFastMixer; 2481 switch (kUseFastMixer) { 2482 case FastMixer_Never: 2483 initFastMixer = false; 2484 break; 2485 case FastMixer_Always: 2486 initFastMixer = true; 2487 break; 2488 case FastMixer_Static: 2489 case FastMixer_Dynamic: 2490 initFastMixer = mFrameCount < mNormalFrameCount; 2491 break; 2492 } 2493 if (initFastMixer) { 2494 2495 // create a MonoPipe to connect our submix to FastMixer 2496 NBAIO_Format format = mOutputSink->format(); 2497 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2498 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2499 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2500 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2501 const NBAIO_Format offers[1] = {format}; 2502 size_t numCounterOffers = 0; 2503 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2504 ALOG_ASSERT(index == 0); 2505 monoPipe->setAvgFrames((mScreenState & 1) ? 2506 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2507 mPipeSink = monoPipe; 2508 2509#ifdef TEE_SINK 2510 if (mTeeSinkOutputEnabled) { 2511 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2512 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2513 numCounterOffers = 0; 2514 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2515 ALOG_ASSERT(index == 0); 2516 mTeeSink = teeSink; 2517 PipeReader *teeSource = new PipeReader(*teeSink); 2518 numCounterOffers = 0; 2519 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2520 ALOG_ASSERT(index == 0); 2521 mTeeSource = teeSource; 2522 } 2523#endif 2524 2525 // create fast mixer and configure it initially with just one fast track for our submix 2526 mFastMixer = new FastMixer(); 2527 FastMixerStateQueue *sq = mFastMixer->sq(); 2528#ifdef STATE_QUEUE_DUMP 2529 sq->setObserverDump(&mStateQueueObserverDump); 2530 sq->setMutatorDump(&mStateQueueMutatorDump); 2531#endif 2532 FastMixerState *state = sq->begin(); 2533 FastTrack *fastTrack = &state->mFastTracks[0]; 2534 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2535 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2536 fastTrack->mVolumeProvider = NULL; 2537 fastTrack->mGeneration++; 2538 state->mFastTracksGen++; 2539 state->mTrackMask = 1; 2540 // fast mixer will use the HAL output sink 2541 state->mOutputSink = mOutputSink.get(); 2542 state->mOutputSinkGen++; 2543 state->mFrameCount = mFrameCount; 2544 state->mCommand = FastMixerState::COLD_IDLE; 2545 // already done in constructor initialization list 2546 //mFastMixerFutex = 0; 2547 state->mColdFutexAddr = &mFastMixerFutex; 2548 state->mColdGen++; 2549 state->mDumpState = &mFastMixerDumpState; 2550#ifdef TEE_SINK 2551 state->mTeeSink = mTeeSink.get(); 2552#endif 2553 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2554 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2555 sq->end(); 2556 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2557 2558 // start the fast mixer 2559 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2560 pid_t tid = mFastMixer->getTid(); 2561 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2562 if (err != 0) { 2563 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2564 kPriorityFastMixer, getpid_cached, tid, err); 2565 } 2566 2567#ifdef AUDIO_WATCHDOG 2568 // create and start the watchdog 2569 mAudioWatchdog = new AudioWatchdog(); 2570 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2571 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2572 tid = mAudioWatchdog->getTid(); 2573 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2574 if (err != 0) { 2575 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2576 kPriorityFastMixer, getpid_cached, tid, err); 2577 } 2578#endif 2579 2580 } else { 2581 mFastMixer = NULL; 2582 } 2583 2584 switch (kUseFastMixer) { 2585 case FastMixer_Never: 2586 case FastMixer_Dynamic: 2587 mNormalSink = mOutputSink; 2588 break; 2589 case FastMixer_Always: 2590 mNormalSink = mPipeSink; 2591 break; 2592 case FastMixer_Static: 2593 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2594 break; 2595 } 2596} 2597 2598AudioFlinger::MixerThread::~MixerThread() 2599{ 2600 if (mFastMixer != NULL) { 2601 FastMixerStateQueue *sq = mFastMixer->sq(); 2602 FastMixerState *state = sq->begin(); 2603 if (state->mCommand == FastMixerState::COLD_IDLE) { 2604 int32_t old = android_atomic_inc(&mFastMixerFutex); 2605 if (old == -1) { 2606 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2607 } 2608 } 2609 state->mCommand = FastMixerState::EXIT; 2610 sq->end(); 2611 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2612 mFastMixer->join(); 2613 // Though the fast mixer thread has exited, it's state queue is still valid. 2614 // We'll use that extract the final state which contains one remaining fast track 2615 // corresponding to our sub-mix. 2616 state = sq->begin(); 2617 ALOG_ASSERT(state->mTrackMask == 1); 2618 FastTrack *fastTrack = &state->mFastTracks[0]; 2619 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2620 delete fastTrack->mBufferProvider; 2621 sq->end(false /*didModify*/); 2622 delete mFastMixer; 2623#ifdef AUDIO_WATCHDOG 2624 if (mAudioWatchdog != 0) { 2625 mAudioWatchdog->requestExit(); 2626 mAudioWatchdog->requestExitAndWait(); 2627 mAudioWatchdog.clear(); 2628 } 2629#endif 2630 } 2631 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2632 delete mAudioMixer; 2633} 2634 2635 2636uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2637{ 2638 if (mFastMixer != NULL) { 2639 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2640 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2641 } 2642 return latency; 2643} 2644 2645 2646void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2647{ 2648 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2649} 2650 2651ssize_t AudioFlinger::MixerThread::threadLoop_write() 2652{ 2653 // FIXME we should only do one push per cycle; confirm this is true 2654 // Start the fast mixer if it's not already running 2655 if (mFastMixer != NULL) { 2656 FastMixerStateQueue *sq = mFastMixer->sq(); 2657 FastMixerState *state = sq->begin(); 2658 if (state->mCommand != FastMixerState::MIX_WRITE && 2659 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2660 if (state->mCommand == FastMixerState::COLD_IDLE) { 2661 int32_t old = android_atomic_inc(&mFastMixerFutex); 2662 if (old == -1) { 2663 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2664 } 2665#ifdef AUDIO_WATCHDOG 2666 if (mAudioWatchdog != 0) { 2667 mAudioWatchdog->resume(); 2668 } 2669#endif 2670 } 2671 state->mCommand = FastMixerState::MIX_WRITE; 2672 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2673 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2674 sq->end(); 2675 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2676 if (kUseFastMixer == FastMixer_Dynamic) { 2677 mNormalSink = mPipeSink; 2678 } 2679 } else { 2680 sq->end(false /*didModify*/); 2681 } 2682 } 2683 return PlaybackThread::threadLoop_write(); 2684} 2685 2686void AudioFlinger::MixerThread::threadLoop_standby() 2687{ 2688 // Idle the fast mixer if it's currently running 2689 if (mFastMixer != NULL) { 2690 FastMixerStateQueue *sq = mFastMixer->sq(); 2691 FastMixerState *state = sq->begin(); 2692 if (!(state->mCommand & FastMixerState::IDLE)) { 2693 state->mCommand = FastMixerState::COLD_IDLE; 2694 state->mColdFutexAddr = &mFastMixerFutex; 2695 state->mColdGen++; 2696 mFastMixerFutex = 0; 2697 sq->end(); 2698 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2699 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2700 if (kUseFastMixer == FastMixer_Dynamic) { 2701 mNormalSink = mOutputSink; 2702 } 2703#ifdef AUDIO_WATCHDOG 2704 if (mAudioWatchdog != 0) { 2705 mAudioWatchdog->pause(); 2706 } 2707#endif 2708 } else { 2709 sq->end(false /*didModify*/); 2710 } 2711 } 2712 PlaybackThread::threadLoop_standby(); 2713} 2714 2715// Empty implementation for standard mixer 2716// Overridden for offloaded playback 2717void AudioFlinger::PlaybackThread::flushOutput_l() 2718{ 2719} 2720 2721bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2722{ 2723 return false; 2724} 2725 2726bool AudioFlinger::PlaybackThread::shouldStandby_l() 2727{ 2728 return !mStandby; 2729} 2730 2731bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2732{ 2733 Mutex::Autolock _l(mLock); 2734 return waitingAsyncCallback_l(); 2735} 2736 2737// shared by MIXER and DIRECT, overridden by DUPLICATING 2738void AudioFlinger::PlaybackThread::threadLoop_standby() 2739{ 2740 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2741 mOutput->stream->common.standby(&mOutput->stream->common); 2742 if (mUseAsyncWrite != 0) { 2743 // discard any pending drain or write ack by incrementing sequence 2744 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2745 mDrainSequence = (mDrainSequence + 2) & ~1; 2746 ALOG_ASSERT(mCallbackThread != 0); 2747 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2748 mCallbackThread->setDraining(mDrainSequence); 2749 } 2750} 2751 2752void AudioFlinger::MixerThread::threadLoop_mix() 2753{ 2754 // obtain the presentation timestamp of the next output buffer 2755 int64_t pts; 2756 status_t status = INVALID_OPERATION; 2757 2758 if (mNormalSink != 0) { 2759 status = mNormalSink->getNextWriteTimestamp(&pts); 2760 } else { 2761 status = mOutputSink->getNextWriteTimestamp(&pts); 2762 } 2763 2764 if (status != NO_ERROR) { 2765 pts = AudioBufferProvider::kInvalidPTS; 2766 } 2767 2768 // mix buffers... 2769 mAudioMixer->process(pts); 2770 mCurrentWriteLength = mixBufferSize; 2771 // increase sleep time progressively when application underrun condition clears. 2772 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2773 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2774 // such that we would underrun the audio HAL. 2775 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2776 sleepTimeShift--; 2777 } 2778 sleepTime = 0; 2779 standbyTime = systemTime() + standbyDelay; 2780 //TODO: delay standby when effects have a tail 2781} 2782 2783void AudioFlinger::MixerThread::threadLoop_sleepTime() 2784{ 2785 // If no tracks are ready, sleep once for the duration of an output 2786 // buffer size, then write 0s to the output 2787 if (sleepTime == 0) { 2788 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2789 sleepTime = activeSleepTime >> sleepTimeShift; 2790 if (sleepTime < kMinThreadSleepTimeUs) { 2791 sleepTime = kMinThreadSleepTimeUs; 2792 } 2793 // reduce sleep time in case of consecutive application underruns to avoid 2794 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2795 // duration we would end up writing less data than needed by the audio HAL if 2796 // the condition persists. 2797 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2798 sleepTimeShift++; 2799 } 2800 } else { 2801 sleepTime = idleSleepTime; 2802 } 2803 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2804 memset (mMixBuffer, 0, mixBufferSize); 2805 sleepTime = 0; 2806 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2807 "anticipated start"); 2808 } 2809 // TODO add standby time extension fct of effect tail 2810} 2811 2812// prepareTracks_l() must be called with ThreadBase::mLock held 2813AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2814 Vector< sp<Track> > *tracksToRemove) 2815{ 2816 2817 mixer_state mixerStatus = MIXER_IDLE; 2818 // find out which tracks need to be processed 2819 size_t count = mActiveTracks.size(); 2820 size_t mixedTracks = 0; 2821 size_t tracksWithEffect = 0; 2822 // counts only _active_ fast tracks 2823 size_t fastTracks = 0; 2824 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2825 2826 float masterVolume = mMasterVolume; 2827 bool masterMute = mMasterMute; 2828 2829 if (masterMute) { 2830 masterVolume = 0; 2831 } 2832 // Delegate master volume control to effect in output mix effect chain if needed 2833 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2834 if (chain != 0) { 2835 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2836 chain->setVolume_l(&v, &v); 2837 masterVolume = (float)((v + (1 << 23)) >> 24); 2838 chain.clear(); 2839 } 2840 2841 // prepare a new state to push 2842 FastMixerStateQueue *sq = NULL; 2843 FastMixerState *state = NULL; 2844 bool didModify = false; 2845 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2846 if (mFastMixer != NULL) { 2847 sq = mFastMixer->sq(); 2848 state = sq->begin(); 2849 } 2850 2851 for (size_t i=0 ; i<count ; i++) { 2852 const sp<Track> t = mActiveTracks[i].promote(); 2853 if (t == 0) { 2854 continue; 2855 } 2856 2857 // this const just means the local variable doesn't change 2858 Track* const track = t.get(); 2859 2860 // process fast tracks 2861 if (track->isFastTrack()) { 2862 2863 // It's theoretically possible (though unlikely) for a fast track to be created 2864 // and then removed within the same normal mix cycle. This is not a problem, as 2865 // the track never becomes active so it's fast mixer slot is never touched. 2866 // The converse, of removing an (active) track and then creating a new track 2867 // at the identical fast mixer slot within the same normal mix cycle, 2868 // is impossible because the slot isn't marked available until the end of each cycle. 2869 int j = track->mFastIndex; 2870 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2871 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2872 FastTrack *fastTrack = &state->mFastTracks[j]; 2873 2874 // Determine whether the track is currently in underrun condition, 2875 // and whether it had a recent underrun. 2876 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2877 FastTrackUnderruns underruns = ftDump->mUnderruns; 2878 uint32_t recentFull = (underruns.mBitFields.mFull - 2879 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2880 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2881 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2882 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2883 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2884 uint32_t recentUnderruns = recentPartial + recentEmpty; 2885 track->mObservedUnderruns = underruns; 2886 // don't count underruns that occur while stopping or pausing 2887 // or stopped which can occur when flush() is called while active 2888 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2889 recentUnderruns > 0) { 2890 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2891 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2892 } 2893 2894 // This is similar to the state machine for normal tracks, 2895 // with a few modifications for fast tracks. 2896 bool isActive = true; 2897 switch (track->mState) { 2898 case TrackBase::STOPPING_1: 2899 // track stays active in STOPPING_1 state until first underrun 2900 if (recentUnderruns > 0 || track->isTerminated()) { 2901 track->mState = TrackBase::STOPPING_2; 2902 } 2903 break; 2904 case TrackBase::PAUSING: 2905 // ramp down is not yet implemented 2906 track->setPaused(); 2907 break; 2908 case TrackBase::RESUMING: 2909 // ramp up is not yet implemented 2910 track->mState = TrackBase::ACTIVE; 2911 break; 2912 case TrackBase::ACTIVE: 2913 if (recentFull > 0 || recentPartial > 0) { 2914 // track has provided at least some frames recently: reset retry count 2915 track->mRetryCount = kMaxTrackRetries; 2916 } 2917 if (recentUnderruns == 0) { 2918 // no recent underruns: stay active 2919 break; 2920 } 2921 // there has recently been an underrun of some kind 2922 if (track->sharedBuffer() == 0) { 2923 // were any of the recent underruns "empty" (no frames available)? 2924 if (recentEmpty == 0) { 2925 // no, then ignore the partial underruns as they are allowed indefinitely 2926 break; 2927 } 2928 // there has recently been an "empty" underrun: decrement the retry counter 2929 if (--(track->mRetryCount) > 0) { 2930 break; 2931 } 2932 // indicate to client process that the track was disabled because of underrun; 2933 // it will then automatically call start() when data is available 2934 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2935 // remove from active list, but state remains ACTIVE [confusing but true] 2936 isActive = false; 2937 break; 2938 } 2939 // fall through 2940 case TrackBase::STOPPING_2: 2941 case TrackBase::PAUSED: 2942 case TrackBase::STOPPED: 2943 case TrackBase::FLUSHED: // flush() while active 2944 // Check for presentation complete if track is inactive 2945 // We have consumed all the buffers of this track. 2946 // This would be incomplete if we auto-paused on underrun 2947 { 2948 size_t audioHALFrames = 2949 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2950 size_t framesWritten = mBytesWritten / mFrameSize; 2951 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2952 // track stays in active list until presentation is complete 2953 break; 2954 } 2955 } 2956 if (track->isStopping_2()) { 2957 track->mState = TrackBase::STOPPED; 2958 } 2959 if (track->isStopped()) { 2960 // Can't reset directly, as fast mixer is still polling this track 2961 // track->reset(); 2962 // So instead mark this track as needing to be reset after push with ack 2963 resetMask |= 1 << i; 2964 } 2965 isActive = false; 2966 break; 2967 case TrackBase::IDLE: 2968 default: 2969 LOG_FATAL("unexpected track state %d", track->mState); 2970 } 2971 2972 if (isActive) { 2973 // was it previously inactive? 2974 if (!(state->mTrackMask & (1 << j))) { 2975 ExtendedAudioBufferProvider *eabp = track; 2976 VolumeProvider *vp = track; 2977 fastTrack->mBufferProvider = eabp; 2978 fastTrack->mVolumeProvider = vp; 2979 fastTrack->mChannelMask = track->mChannelMask; 2980 fastTrack->mGeneration++; 2981 state->mTrackMask |= 1 << j; 2982 didModify = true; 2983 // no acknowledgement required for newly active tracks 2984 } 2985 // cache the combined master volume and stream type volume for fast mixer; this 2986 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2987 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2988 ++fastTracks; 2989 } else { 2990 // was it previously active? 2991 if (state->mTrackMask & (1 << j)) { 2992 fastTrack->mBufferProvider = NULL; 2993 fastTrack->mGeneration++; 2994 state->mTrackMask &= ~(1 << j); 2995 didModify = true; 2996 // If any fast tracks were removed, we must wait for acknowledgement 2997 // because we're about to decrement the last sp<> on those tracks. 2998 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2999 } else { 3000 LOG_FATAL("fast track %d should have been active", j); 3001 } 3002 tracksToRemove->add(track); 3003 // Avoids a misleading display in dumpsys 3004 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3005 } 3006 continue; 3007 } 3008 3009 { // local variable scope to avoid goto warning 3010 3011 audio_track_cblk_t* cblk = track->cblk(); 3012 3013 // The first time a track is added we wait 3014 // for all its buffers to be filled before processing it 3015 int name = track->name(); 3016 // make sure that we have enough frames to mix one full buffer. 3017 // enforce this condition only once to enable draining the buffer in case the client 3018 // app does not call stop() and relies on underrun to stop: 3019 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3020 // during last round 3021 size_t desiredFrames; 3022 uint32_t sr = track->sampleRate(); 3023 if (sr == mSampleRate) { 3024 desiredFrames = mNormalFrameCount; 3025 } else { 3026 // +1 for rounding and +1 for additional sample needed for interpolation 3027 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3028 // add frames already consumed but not yet released by the resampler 3029 // because cblk->framesReady() will include these frames 3030 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3031 // the minimum track buffer size is normally twice the number of frames necessary 3032 // to fill one buffer and the resampler should not leave more than one buffer worth 3033 // of unreleased frames after each pass, but just in case... 3034 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3035 } 3036 uint32_t minFrames = 1; 3037 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3038 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3039 minFrames = desiredFrames; 3040 } 3041 3042 size_t framesReady = track->framesReady(); 3043 if ((framesReady >= minFrames) && track->isReady() && 3044 !track->isPaused() && !track->isTerminated()) 3045 { 3046 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3047 3048 mixedTracks++; 3049 3050 // track->mainBuffer() != mMixBuffer means there is an effect chain 3051 // connected to the track 3052 chain.clear(); 3053 if (track->mainBuffer() != mMixBuffer) { 3054 chain = getEffectChain_l(track->sessionId()); 3055 // Delegate volume control to effect in track effect chain if needed 3056 if (chain != 0) { 3057 tracksWithEffect++; 3058 } else { 3059 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3060 "session %d", 3061 name, track->sessionId()); 3062 } 3063 } 3064 3065 3066 int param = AudioMixer::VOLUME; 3067 if (track->mFillingUpStatus == Track::FS_FILLED) { 3068 // no ramp for the first volume setting 3069 track->mFillingUpStatus = Track::FS_ACTIVE; 3070 if (track->mState == TrackBase::RESUMING) { 3071 track->mState = TrackBase::ACTIVE; 3072 param = AudioMixer::RAMP_VOLUME; 3073 } 3074 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3075 // FIXME should not make a decision based on mServer 3076 } else if (cblk->mServer != 0) { 3077 // If the track is stopped before the first frame was mixed, 3078 // do not apply ramp 3079 param = AudioMixer::RAMP_VOLUME; 3080 } 3081 3082 // compute volume for this track 3083 uint32_t vl, vr, va; 3084 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3085 vl = vr = va = 0; 3086 if (track->isPausing()) { 3087 track->setPaused(); 3088 } 3089 } else { 3090 3091 // read original volumes with volume control 3092 float typeVolume = mStreamTypes[track->streamType()].volume; 3093 float v = masterVolume * typeVolume; 3094 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3095 uint32_t vlr = proxy->getVolumeLR(); 3096 vl = vlr & 0xFFFF; 3097 vr = vlr >> 16; 3098 // track volumes come from shared memory, so can't be trusted and must be clamped 3099 if (vl > MAX_GAIN_INT) { 3100 ALOGV("Track left volume out of range: %04X", vl); 3101 vl = MAX_GAIN_INT; 3102 } 3103 if (vr > MAX_GAIN_INT) { 3104 ALOGV("Track right volume out of range: %04X", vr); 3105 vr = MAX_GAIN_INT; 3106 } 3107 // now apply the master volume and stream type volume 3108 vl = (uint32_t)(v * vl) << 12; 3109 vr = (uint32_t)(v * vr) << 12; 3110 // assuming master volume and stream type volume each go up to 1.0, 3111 // vl and vr are now in 8.24 format 3112 3113 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3114 // send level comes from shared memory and so may be corrupt 3115 if (sendLevel > MAX_GAIN_INT) { 3116 ALOGV("Track send level out of range: %04X", sendLevel); 3117 sendLevel = MAX_GAIN_INT; 3118 } 3119 va = (uint32_t)(v * sendLevel); 3120 } 3121 3122 // Delegate volume control to effect in track effect chain if needed 3123 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3124 // Do not ramp volume if volume is controlled by effect 3125 param = AudioMixer::VOLUME; 3126 track->mHasVolumeController = true; 3127 } else { 3128 // force no volume ramp when volume controller was just disabled or removed 3129 // from effect chain to avoid volume spike 3130 if (track->mHasVolumeController) { 3131 param = AudioMixer::VOLUME; 3132 } 3133 track->mHasVolumeController = false; 3134 } 3135 3136 // Convert volumes from 8.24 to 4.12 format 3137 // This additional clamping is needed in case chain->setVolume_l() overshot 3138 vl = (vl + (1 << 11)) >> 12; 3139 if (vl > MAX_GAIN_INT) { 3140 vl = MAX_GAIN_INT; 3141 } 3142 vr = (vr + (1 << 11)) >> 12; 3143 if (vr > MAX_GAIN_INT) { 3144 vr = MAX_GAIN_INT; 3145 } 3146 3147 if (va > MAX_GAIN_INT) { 3148 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3149 } 3150 3151 // XXX: these things DON'T need to be done each time 3152 mAudioMixer->setBufferProvider(name, track); 3153 mAudioMixer->enable(name); 3154 3155 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3156 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3157 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3158 mAudioMixer->setParameter( 3159 name, 3160 AudioMixer::TRACK, 3161 AudioMixer::FORMAT, (void *)track->format()); 3162 mAudioMixer->setParameter( 3163 name, 3164 AudioMixer::TRACK, 3165 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3166 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3167 uint32_t maxSampleRate = mSampleRate * 2; 3168 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3169 if (reqSampleRate == 0) { 3170 reqSampleRate = mSampleRate; 3171 } else if (reqSampleRate > maxSampleRate) { 3172 reqSampleRate = maxSampleRate; 3173 } 3174 mAudioMixer->setParameter( 3175 name, 3176 AudioMixer::RESAMPLE, 3177 AudioMixer::SAMPLE_RATE, 3178 (void *)(uintptr_t)reqSampleRate); 3179 mAudioMixer->setParameter( 3180 name, 3181 AudioMixer::TRACK, 3182 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3183 mAudioMixer->setParameter( 3184 name, 3185 AudioMixer::TRACK, 3186 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3187 3188 // reset retry count 3189 track->mRetryCount = kMaxTrackRetries; 3190 3191 // If one track is ready, set the mixer ready if: 3192 // - the mixer was not ready during previous round OR 3193 // - no other track is not ready 3194 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3195 mixerStatus != MIXER_TRACKS_ENABLED) { 3196 mixerStatus = MIXER_TRACKS_READY; 3197 } 3198 } else { 3199 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3200 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3201 } 3202 // clear effect chain input buffer if an active track underruns to avoid sending 3203 // previous audio buffer again to effects 3204 chain = getEffectChain_l(track->sessionId()); 3205 if (chain != 0) { 3206 chain->clearInputBuffer(); 3207 } 3208 3209 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3210 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3211 track->isStopped() || track->isPaused()) { 3212 // We have consumed all the buffers of this track. 3213 // Remove it from the list of active tracks. 3214 // TODO: use actual buffer filling status instead of latency when available from 3215 // audio HAL 3216 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3217 size_t framesWritten = mBytesWritten / mFrameSize; 3218 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3219 if (track->isStopped()) { 3220 track->reset(); 3221 } 3222 tracksToRemove->add(track); 3223 } 3224 } else { 3225 // No buffers for this track. Give it a few chances to 3226 // fill a buffer, then remove it from active list. 3227 if (--(track->mRetryCount) <= 0) { 3228 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3229 tracksToRemove->add(track); 3230 // indicate to client process that the track was disabled because of underrun; 3231 // it will then automatically call start() when data is available 3232 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3233 // If one track is not ready, mark the mixer also not ready if: 3234 // - the mixer was ready during previous round OR 3235 // - no other track is ready 3236 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3237 mixerStatus != MIXER_TRACKS_READY) { 3238 mixerStatus = MIXER_TRACKS_ENABLED; 3239 } 3240 } 3241 mAudioMixer->disable(name); 3242 } 3243 3244 } // local variable scope to avoid goto warning 3245track_is_ready: ; 3246 3247 } 3248 3249 // Push the new FastMixer state if necessary 3250 bool pauseAudioWatchdog = false; 3251 if (didModify) { 3252 state->mFastTracksGen++; 3253 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3254 if (kUseFastMixer == FastMixer_Dynamic && 3255 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3256 state->mCommand = FastMixerState::COLD_IDLE; 3257 state->mColdFutexAddr = &mFastMixerFutex; 3258 state->mColdGen++; 3259 mFastMixerFutex = 0; 3260 if (kUseFastMixer == FastMixer_Dynamic) { 3261 mNormalSink = mOutputSink; 3262 } 3263 // If we go into cold idle, need to wait for acknowledgement 3264 // so that fast mixer stops doing I/O. 3265 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3266 pauseAudioWatchdog = true; 3267 } 3268 } 3269 if (sq != NULL) { 3270 sq->end(didModify); 3271 sq->push(block); 3272 } 3273#ifdef AUDIO_WATCHDOG 3274 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3275 mAudioWatchdog->pause(); 3276 } 3277#endif 3278 3279 // Now perform the deferred reset on fast tracks that have stopped 3280 while (resetMask != 0) { 3281 size_t i = __builtin_ctz(resetMask); 3282 ALOG_ASSERT(i < count); 3283 resetMask &= ~(1 << i); 3284 sp<Track> t = mActiveTracks[i].promote(); 3285 if (t == 0) { 3286 continue; 3287 } 3288 Track* track = t.get(); 3289 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3290 track->reset(); 3291 } 3292 3293 // remove all the tracks that need to be... 3294 removeTracks_l(*tracksToRemove); 3295 3296 // mix buffer must be cleared if all tracks are connected to an 3297 // effect chain as in this case the mixer will not write to 3298 // mix buffer and track effects will accumulate into it 3299 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3300 (mixedTracks == 0 && fastTracks > 0))) { 3301 // FIXME as a performance optimization, should remember previous zero status 3302 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3303 } 3304 3305 // if any fast tracks, then status is ready 3306 mMixerStatusIgnoringFastTracks = mixerStatus; 3307 if (fastTracks > 0) { 3308 mixerStatus = MIXER_TRACKS_READY; 3309 } 3310 return mixerStatus; 3311} 3312 3313// getTrackName_l() must be called with ThreadBase::mLock held 3314int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3315{ 3316 return mAudioMixer->getTrackName(channelMask, sessionId); 3317} 3318 3319// deleteTrackName_l() must be called with ThreadBase::mLock held 3320void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3321{ 3322 ALOGV("remove track (%d) and delete from mixer", name); 3323 mAudioMixer->deleteTrackName(name); 3324} 3325 3326// checkForNewParameters_l() must be called with ThreadBase::mLock held 3327bool AudioFlinger::MixerThread::checkForNewParameters_l() 3328{ 3329 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3330 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3331 bool reconfig = false; 3332 3333 while (!mNewParameters.isEmpty()) { 3334 3335 if (mFastMixer != NULL) { 3336 FastMixerStateQueue *sq = mFastMixer->sq(); 3337 FastMixerState *state = sq->begin(); 3338 if (!(state->mCommand & FastMixerState::IDLE)) { 3339 previousCommand = state->mCommand; 3340 state->mCommand = FastMixerState::HOT_IDLE; 3341 sq->end(); 3342 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3343 } else { 3344 sq->end(false /*didModify*/); 3345 } 3346 } 3347 3348 status_t status = NO_ERROR; 3349 String8 keyValuePair = mNewParameters[0]; 3350 AudioParameter param = AudioParameter(keyValuePair); 3351 int value; 3352 3353 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3354 reconfig = true; 3355 } 3356 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3357 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3358 status = BAD_VALUE; 3359 } else { 3360 reconfig = true; 3361 } 3362 } 3363 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3364 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3365 status = BAD_VALUE; 3366 } else { 3367 reconfig = true; 3368 } 3369 } 3370 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3371 // do not accept frame count changes if tracks are open as the track buffer 3372 // size depends on frame count and correct behavior would not be guaranteed 3373 // if frame count is changed after track creation 3374 if (!mTracks.isEmpty()) { 3375 status = INVALID_OPERATION; 3376 } else { 3377 reconfig = true; 3378 } 3379 } 3380 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3381#ifdef ADD_BATTERY_DATA 3382 // when changing the audio output device, call addBatteryData to notify 3383 // the change 3384 if (mOutDevice != value) { 3385 uint32_t params = 0; 3386 // check whether speaker is on 3387 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3388 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3389 } 3390 3391 audio_devices_t deviceWithoutSpeaker 3392 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3393 // check if any other device (except speaker) is on 3394 if (value & deviceWithoutSpeaker ) { 3395 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3396 } 3397 3398 if (params != 0) { 3399 addBatteryData(params); 3400 } 3401 } 3402#endif 3403 3404 // forward device change to effects that have requested to be 3405 // aware of attached audio device. 3406 if (value != AUDIO_DEVICE_NONE) { 3407 mOutDevice = value; 3408 for (size_t i = 0; i < mEffectChains.size(); i++) { 3409 mEffectChains[i]->setDevice_l(mOutDevice); 3410 } 3411 } 3412 } 3413 3414 if (status == NO_ERROR) { 3415 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3416 keyValuePair.string()); 3417 if (!mStandby && status == INVALID_OPERATION) { 3418 mOutput->stream->common.standby(&mOutput->stream->common); 3419 mStandby = true; 3420 mBytesWritten = 0; 3421 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3422 keyValuePair.string()); 3423 } 3424 if (status == NO_ERROR && reconfig) { 3425 readOutputParameters(); 3426 delete mAudioMixer; 3427 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3428 for (size_t i = 0; i < mTracks.size() ; i++) { 3429 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3430 if (name < 0) { 3431 break; 3432 } 3433 mTracks[i]->mName = name; 3434 } 3435 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3436 } 3437 } 3438 3439 mNewParameters.removeAt(0); 3440 3441 mParamStatus = status; 3442 mParamCond.signal(); 3443 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3444 // already timed out waiting for the status and will never signal the condition. 3445 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3446 } 3447 3448 if (!(previousCommand & FastMixerState::IDLE)) { 3449 ALOG_ASSERT(mFastMixer != NULL); 3450 FastMixerStateQueue *sq = mFastMixer->sq(); 3451 FastMixerState *state = sq->begin(); 3452 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3453 state->mCommand = previousCommand; 3454 sq->end(); 3455 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3456 } 3457 3458 return reconfig; 3459} 3460 3461 3462void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3463{ 3464 const size_t SIZE = 256; 3465 char buffer[SIZE]; 3466 String8 result; 3467 3468 PlaybackThread::dumpInternals(fd, args); 3469 3470 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3471 result.append(buffer); 3472 write(fd, result.string(), result.size()); 3473 3474 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3475 const FastMixerDumpState copy(mFastMixerDumpState); 3476 copy.dump(fd); 3477 3478#ifdef STATE_QUEUE_DUMP 3479 // Similar for state queue 3480 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3481 observerCopy.dump(fd); 3482 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3483 mutatorCopy.dump(fd); 3484#endif 3485 3486#ifdef TEE_SINK 3487 // Write the tee output to a .wav file 3488 dumpTee(fd, mTeeSource, mId); 3489#endif 3490 3491#ifdef AUDIO_WATCHDOG 3492 if (mAudioWatchdog != 0) { 3493 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3494 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3495 wdCopy.dump(fd); 3496 } 3497#endif 3498} 3499 3500uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3501{ 3502 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3503} 3504 3505uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3506{ 3507 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3508} 3509 3510void AudioFlinger::MixerThread::cacheParameters_l() 3511{ 3512 PlaybackThread::cacheParameters_l(); 3513 3514 // FIXME: Relaxed timing because of a certain device that can't meet latency 3515 // Should be reduced to 2x after the vendor fixes the driver issue 3516 // increase threshold again due to low power audio mode. The way this warning 3517 // threshold is calculated and its usefulness should be reconsidered anyway. 3518 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3519} 3520 3521// ---------------------------------------------------------------------------- 3522 3523AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3524 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3525 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3526 // mLeftVolFloat, mRightVolFloat 3527{ 3528} 3529 3530AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3531 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3532 ThreadBase::type_t type) 3533 : PlaybackThread(audioFlinger, output, id, device, type) 3534 // mLeftVolFloat, mRightVolFloat 3535{ 3536} 3537 3538AudioFlinger::DirectOutputThread::~DirectOutputThread() 3539{ 3540} 3541 3542void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3543{ 3544 audio_track_cblk_t* cblk = track->cblk(); 3545 float left, right; 3546 3547 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3548 left = right = 0; 3549 } else { 3550 float typeVolume = mStreamTypes[track->streamType()].volume; 3551 float v = mMasterVolume * typeVolume; 3552 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3553 uint32_t vlr = proxy->getVolumeLR(); 3554 float v_clamped = v * (vlr & 0xFFFF); 3555 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3556 left = v_clamped/MAX_GAIN; 3557 v_clamped = v * (vlr >> 16); 3558 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3559 right = v_clamped/MAX_GAIN; 3560 } 3561 3562 if (lastTrack) { 3563 if (left != mLeftVolFloat || right != mRightVolFloat) { 3564 mLeftVolFloat = left; 3565 mRightVolFloat = right; 3566 3567 // Convert volumes from float to 8.24 3568 uint32_t vl = (uint32_t)(left * (1 << 24)); 3569 uint32_t vr = (uint32_t)(right * (1 << 24)); 3570 3571 // Delegate volume control to effect in track effect chain if needed 3572 // only one effect chain can be present on DirectOutputThread, so if 3573 // there is one, the track is connected to it 3574 if (!mEffectChains.isEmpty()) { 3575 mEffectChains[0]->setVolume_l(&vl, &vr); 3576 left = (float)vl / (1 << 24); 3577 right = (float)vr / (1 << 24); 3578 } 3579 if (mOutput->stream->set_volume) { 3580 mOutput->stream->set_volume(mOutput->stream, left, right); 3581 } 3582 } 3583 } 3584} 3585 3586 3587AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3588 Vector< sp<Track> > *tracksToRemove 3589) 3590{ 3591 size_t count = mActiveTracks.size(); 3592 mixer_state mixerStatus = MIXER_IDLE; 3593 3594 // find out which tracks need to be processed 3595 for (size_t i = 0; i < count; i++) { 3596 sp<Track> t = mActiveTracks[i].promote(); 3597 // The track died recently 3598 if (t == 0) { 3599 continue; 3600 } 3601 3602 Track* const track = t.get(); 3603 audio_track_cblk_t* cblk = track->cblk(); 3604 // Only consider last track started for volume and mixer state control. 3605 // In theory an older track could underrun and restart after the new one starts 3606 // but as we only care about the transition phase between two tracks on a 3607 // direct output, it is not a problem to ignore the underrun case. 3608 sp<Track> l = mLatestActiveTrack.promote(); 3609 bool last = l.get() == track; 3610 3611 // The first time a track is added we wait 3612 // for all its buffers to be filled before processing it 3613 uint32_t minFrames; 3614 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3615 minFrames = mNormalFrameCount; 3616 } else { 3617 minFrames = 1; 3618 } 3619 3620 if ((track->framesReady() >= minFrames) && track->isReady() && 3621 !track->isPaused() && !track->isTerminated()) 3622 { 3623 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3624 3625 if (track->mFillingUpStatus == Track::FS_FILLED) { 3626 track->mFillingUpStatus = Track::FS_ACTIVE; 3627 // make sure processVolume_l() will apply new volume even if 0 3628 mLeftVolFloat = mRightVolFloat = -1.0; 3629 if (track->mState == TrackBase::RESUMING) { 3630 track->mState = TrackBase::ACTIVE; 3631 } 3632 } 3633 3634 // compute volume for this track 3635 processVolume_l(track, last); 3636 if (last) { 3637 // reset retry count 3638 track->mRetryCount = kMaxTrackRetriesDirect; 3639 mActiveTrack = t; 3640 mixerStatus = MIXER_TRACKS_READY; 3641 } 3642 } else { 3643 // clear effect chain input buffer if the last active track started underruns 3644 // to avoid sending previous audio buffer again to effects 3645 if (!mEffectChains.isEmpty() && last) { 3646 mEffectChains[0]->clearInputBuffer(); 3647 } 3648 3649 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3650 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3651 track->isStopped() || track->isPaused()) { 3652 // We have consumed all the buffers of this track. 3653 // Remove it from the list of active tracks. 3654 // TODO: implement behavior for compressed audio 3655 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3656 size_t framesWritten = mBytesWritten / mFrameSize; 3657 if (mStandby || !last || 3658 track->presentationComplete(framesWritten, audioHALFrames)) { 3659 if (track->isStopped()) { 3660 track->reset(); 3661 } 3662 tracksToRemove->add(track); 3663 } 3664 } else { 3665 // No buffers for this track. Give it a few chances to 3666 // fill a buffer, then remove it from active list. 3667 // Only consider last track started for mixer state control 3668 if (--(track->mRetryCount) <= 0) { 3669 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3670 tracksToRemove->add(track); 3671 // indicate to client process that the track was disabled because of underrun; 3672 // it will then automatically call start() when data is available 3673 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3674 } else if (last) { 3675 mixerStatus = MIXER_TRACKS_ENABLED; 3676 } 3677 } 3678 } 3679 } 3680 3681 // remove all the tracks that need to be... 3682 removeTracks_l(*tracksToRemove); 3683 3684 return mixerStatus; 3685} 3686 3687void AudioFlinger::DirectOutputThread::threadLoop_mix() 3688{ 3689 size_t frameCount = mFrameCount; 3690 int8_t *curBuf = (int8_t *)mMixBuffer; 3691 // output audio to hardware 3692 while (frameCount) { 3693 AudioBufferProvider::Buffer buffer; 3694 buffer.frameCount = frameCount; 3695 mActiveTrack->getNextBuffer(&buffer); 3696 if (buffer.raw == NULL) { 3697 memset(curBuf, 0, frameCount * mFrameSize); 3698 break; 3699 } 3700 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3701 frameCount -= buffer.frameCount; 3702 curBuf += buffer.frameCount * mFrameSize; 3703 mActiveTrack->releaseBuffer(&buffer); 3704 } 3705 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3706 sleepTime = 0; 3707 standbyTime = systemTime() + standbyDelay; 3708 mActiveTrack.clear(); 3709} 3710 3711void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3712{ 3713 if (sleepTime == 0) { 3714 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3715 sleepTime = activeSleepTime; 3716 } else { 3717 sleepTime = idleSleepTime; 3718 } 3719 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3720 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3721 sleepTime = 0; 3722 } 3723} 3724 3725// getTrackName_l() must be called with ThreadBase::mLock held 3726int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3727 int sessionId) 3728{ 3729 return 0; 3730} 3731 3732// deleteTrackName_l() must be called with ThreadBase::mLock held 3733void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3734{ 3735} 3736 3737// checkForNewParameters_l() must be called with ThreadBase::mLock held 3738bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3739{ 3740 bool reconfig = false; 3741 3742 while (!mNewParameters.isEmpty()) { 3743 status_t status = NO_ERROR; 3744 String8 keyValuePair = mNewParameters[0]; 3745 AudioParameter param = AudioParameter(keyValuePair); 3746 int value; 3747 3748 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3749 // do not accept frame count changes if tracks are open as the track buffer 3750 // size depends on frame count and correct behavior would not be garantied 3751 // if frame count is changed after track creation 3752 if (!mTracks.isEmpty()) { 3753 status = INVALID_OPERATION; 3754 } else { 3755 reconfig = true; 3756 } 3757 } 3758 if (status == NO_ERROR) { 3759 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3760 keyValuePair.string()); 3761 if (!mStandby && status == INVALID_OPERATION) { 3762 mOutput->stream->common.standby(&mOutput->stream->common); 3763 mStandby = true; 3764 mBytesWritten = 0; 3765 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3766 keyValuePair.string()); 3767 } 3768 if (status == NO_ERROR && reconfig) { 3769 readOutputParameters(); 3770 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3771 } 3772 } 3773 3774 mNewParameters.removeAt(0); 3775 3776 mParamStatus = status; 3777 mParamCond.signal(); 3778 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3779 // already timed out waiting for the status and will never signal the condition. 3780 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3781 } 3782 return reconfig; 3783} 3784 3785uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3786{ 3787 uint32_t time; 3788 if (audio_is_linear_pcm(mFormat)) { 3789 time = PlaybackThread::activeSleepTimeUs(); 3790 } else { 3791 time = 10000; 3792 } 3793 return time; 3794} 3795 3796uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3797{ 3798 uint32_t time; 3799 if (audio_is_linear_pcm(mFormat)) { 3800 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3801 } else { 3802 time = 10000; 3803 } 3804 return time; 3805} 3806 3807uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3808{ 3809 uint32_t time; 3810 if (audio_is_linear_pcm(mFormat)) { 3811 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3812 } else { 3813 time = 10000; 3814 } 3815 return time; 3816} 3817 3818void AudioFlinger::DirectOutputThread::cacheParameters_l() 3819{ 3820 PlaybackThread::cacheParameters_l(); 3821 3822 // use shorter standby delay as on normal output to release 3823 // hardware resources as soon as possible 3824 if (audio_is_linear_pcm(mFormat)) { 3825 standbyDelay = microseconds(activeSleepTime*2); 3826 } else { 3827 standbyDelay = kOffloadStandbyDelayNs; 3828 } 3829} 3830 3831// ---------------------------------------------------------------------------- 3832 3833AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3834 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3835 : Thread(false /*canCallJava*/), 3836 mPlaybackThread(playbackThread), 3837 mWriteAckSequence(0), 3838 mDrainSequence(0) 3839{ 3840} 3841 3842AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3843{ 3844} 3845 3846void AudioFlinger::AsyncCallbackThread::onFirstRef() 3847{ 3848 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3849} 3850 3851bool AudioFlinger::AsyncCallbackThread::threadLoop() 3852{ 3853 while (!exitPending()) { 3854 uint32_t writeAckSequence; 3855 uint32_t drainSequence; 3856 3857 { 3858 Mutex::Autolock _l(mLock); 3859 while (!((mWriteAckSequence & 1) || 3860 (mDrainSequence & 1) || 3861 exitPending())) { 3862 mWaitWorkCV.wait(mLock); 3863 } 3864 3865 if (exitPending()) { 3866 break; 3867 } 3868 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3869 mWriteAckSequence, mDrainSequence); 3870 writeAckSequence = mWriteAckSequence; 3871 mWriteAckSequence &= ~1; 3872 drainSequence = mDrainSequence; 3873 mDrainSequence &= ~1; 3874 } 3875 { 3876 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3877 if (playbackThread != 0) { 3878 if (writeAckSequence & 1) { 3879 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3880 } 3881 if (drainSequence & 1) { 3882 playbackThread->resetDraining(drainSequence >> 1); 3883 } 3884 } 3885 } 3886 } 3887 return false; 3888} 3889 3890void AudioFlinger::AsyncCallbackThread::exit() 3891{ 3892 ALOGV("AsyncCallbackThread::exit"); 3893 Mutex::Autolock _l(mLock); 3894 requestExit(); 3895 mWaitWorkCV.broadcast(); 3896} 3897 3898void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3899{ 3900 Mutex::Autolock _l(mLock); 3901 // bit 0 is cleared 3902 mWriteAckSequence = sequence << 1; 3903} 3904 3905void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3906{ 3907 Mutex::Autolock _l(mLock); 3908 // ignore unexpected callbacks 3909 if (mWriteAckSequence & 2) { 3910 mWriteAckSequence |= 1; 3911 mWaitWorkCV.signal(); 3912 } 3913} 3914 3915void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3916{ 3917 Mutex::Autolock _l(mLock); 3918 // bit 0 is cleared 3919 mDrainSequence = sequence << 1; 3920} 3921 3922void AudioFlinger::AsyncCallbackThread::resetDraining() 3923{ 3924 Mutex::Autolock _l(mLock); 3925 // ignore unexpected callbacks 3926 if (mDrainSequence & 2) { 3927 mDrainSequence |= 1; 3928 mWaitWorkCV.signal(); 3929 } 3930} 3931 3932 3933// ---------------------------------------------------------------------------- 3934AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3935 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3936 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3937 mHwPaused(false), 3938 mFlushPending(false), 3939 mPausedBytesRemaining(0) 3940{ 3941 //FIXME: mStandby should be set to true by ThreadBase constructor 3942 mStandby = true; 3943} 3944 3945void AudioFlinger::OffloadThread::threadLoop_exit() 3946{ 3947 if (mFlushPending || mHwPaused) { 3948 // If a flush is pending or track was paused, just discard buffered data 3949 flushHw_l(); 3950 } else { 3951 mMixerStatus = MIXER_DRAIN_ALL; 3952 threadLoop_drain(); 3953 } 3954 if (mUseAsyncWrite) { 3955 ALOG_ASSERT(mCallbackThread != 0); 3956 mCallbackThread->exit(); 3957 } 3958 PlaybackThread::threadLoop_exit(); 3959} 3960 3961AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3962 Vector< sp<Track> > *tracksToRemove 3963) 3964{ 3965 size_t count = mActiveTracks.size(); 3966 3967 mixer_state mixerStatus = MIXER_IDLE; 3968 bool doHwPause = false; 3969 bool doHwResume = false; 3970 3971 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3972 3973 // find out which tracks need to be processed 3974 for (size_t i = 0; i < count; i++) { 3975 sp<Track> t = mActiveTracks[i].promote(); 3976 // The track died recently 3977 if (t == 0) { 3978 continue; 3979 } 3980 Track* const track = t.get(); 3981 audio_track_cblk_t* cblk = track->cblk(); 3982 // Only consider last track started for volume and mixer state control. 3983 // In theory an older track could underrun and restart after the new one starts 3984 // but as we only care about the transition phase between two tracks on a 3985 // direct output, it is not a problem to ignore the underrun case. 3986 sp<Track> l = mLatestActiveTrack.promote(); 3987 bool last = l.get() == track; 3988 3989 if (track->isPausing()) { 3990 track->setPaused(); 3991 if (last) { 3992 if (!mHwPaused) { 3993 doHwPause = true; 3994 mHwPaused = true; 3995 } 3996 // If we were part way through writing the mixbuffer to 3997 // the HAL we must save this until we resume 3998 // BUG - this will be wrong if a different track is made active, 3999 // in that case we want to discard the pending data in the 4000 // mixbuffer and tell the client to present it again when the 4001 // track is resumed 4002 mPausedWriteLength = mCurrentWriteLength; 4003 mPausedBytesRemaining = mBytesRemaining; 4004 mBytesRemaining = 0; // stop writing 4005 } 4006 tracksToRemove->add(track); 4007 } else if (track->framesReady() && track->isReady() && 4008 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4009 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4010 if (track->mFillingUpStatus == Track::FS_FILLED) { 4011 track->mFillingUpStatus = Track::FS_ACTIVE; 4012 // make sure processVolume_l() will apply new volume even if 0 4013 mLeftVolFloat = mRightVolFloat = -1.0; 4014 if (track->mState == TrackBase::RESUMING) { 4015 track->mState = TrackBase::ACTIVE; 4016 if (last) { 4017 if (mPausedBytesRemaining) { 4018 // Need to continue write that was interrupted 4019 mCurrentWriteLength = mPausedWriteLength; 4020 mBytesRemaining = mPausedBytesRemaining; 4021 mPausedBytesRemaining = 0; 4022 } 4023 if (mHwPaused) { 4024 doHwResume = true; 4025 mHwPaused = false; 4026 // threadLoop_mix() will handle the case that we need to 4027 // resume an interrupted write 4028 } 4029 // enable write to audio HAL 4030 sleepTime = 0; 4031 } 4032 } 4033 } 4034 4035 if (last) { 4036 sp<Track> previousTrack = mPreviousTrack.promote(); 4037 if (previousTrack != 0) { 4038 if (track != previousTrack.get()) { 4039 // Flush any data still being written from last track 4040 mBytesRemaining = 0; 4041 if (mPausedBytesRemaining) { 4042 // Last track was paused so we also need to flush saved 4043 // mixbuffer state and invalidate track so that it will 4044 // re-submit that unwritten data when it is next resumed 4045 mPausedBytesRemaining = 0; 4046 // Invalidate is a bit drastic - would be more efficient 4047 // to have a flag to tell client that some of the 4048 // previously written data was lost 4049 previousTrack->invalidate(); 4050 } 4051 // flush data already sent to the DSP if changing audio session as audio 4052 // comes from a different source. Also invalidate previous track to force a 4053 // seek when resuming. 4054 if (previousTrack->sessionId() != track->sessionId()) { 4055 previousTrack->invalidate(); 4056 mFlushPending = true; 4057 } 4058 } 4059 } 4060 mPreviousTrack = track; 4061 // reset retry count 4062 track->mRetryCount = kMaxTrackRetriesOffload; 4063 mActiveTrack = t; 4064 mixerStatus = MIXER_TRACKS_READY; 4065 } 4066 } else { 4067 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4068 if (track->isStopping_1()) { 4069 // Hardware buffer can hold a large amount of audio so we must 4070 // wait for all current track's data to drain before we say 4071 // that the track is stopped. 4072 if (mBytesRemaining == 0) { 4073 // Only start draining when all data in mixbuffer 4074 // has been written 4075 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4076 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4077 // do not drain if no data was ever sent to HAL (mStandby == true) 4078 if (last && !mStandby) { 4079 // do not modify drain sequence if we are already draining. This happens 4080 // when resuming from pause after drain. 4081 if ((mDrainSequence & 1) == 0) { 4082 sleepTime = 0; 4083 standbyTime = systemTime() + standbyDelay; 4084 mixerStatus = MIXER_DRAIN_TRACK; 4085 mDrainSequence += 2; 4086 } 4087 if (mHwPaused) { 4088 // It is possible to move from PAUSED to STOPPING_1 without 4089 // a resume so we must ensure hardware is running 4090 doHwResume = true; 4091 mHwPaused = false; 4092 } 4093 } 4094 } 4095 } else if (track->isStopping_2()) { 4096 // Drain has completed or we are in standby, signal presentation complete 4097 if (!(mDrainSequence & 1) || !last || mStandby) { 4098 track->mState = TrackBase::STOPPED; 4099 size_t audioHALFrames = 4100 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4101 size_t framesWritten = 4102 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4103 track->presentationComplete(framesWritten, audioHALFrames); 4104 track->reset(); 4105 tracksToRemove->add(track); 4106 } 4107 } else { 4108 // No buffers for this track. Give it a few chances to 4109 // fill a buffer, then remove it from active list. 4110 if (--(track->mRetryCount) <= 0) { 4111 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4112 track->name()); 4113 tracksToRemove->add(track); 4114 // indicate to client process that the track was disabled because of underrun; 4115 // it will then automatically call start() when data is available 4116 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4117 } else if (last){ 4118 mixerStatus = MIXER_TRACKS_ENABLED; 4119 } 4120 } 4121 } 4122 // compute volume for this track 4123 processVolume_l(track, last); 4124 } 4125 4126 // make sure the pause/flush/resume sequence is executed in the right order. 4127 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4128 // before flush and then resume HW. This can happen in case of pause/flush/resume 4129 // if resume is received before pause is executed. 4130 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4131 mOutput->stream->pause(mOutput->stream); 4132 if (!doHwPause) { 4133 doHwResume = true; 4134 } 4135 } 4136 if (mFlushPending) { 4137 flushHw_l(); 4138 mFlushPending = false; 4139 } 4140 if (!mStandby && doHwResume) { 4141 mOutput->stream->resume(mOutput->stream); 4142 } 4143 4144 // remove all the tracks that need to be... 4145 removeTracks_l(*tracksToRemove); 4146 4147 return mixerStatus; 4148} 4149 4150void AudioFlinger::OffloadThread::flushOutput_l() 4151{ 4152 mFlushPending = true; 4153} 4154 4155// must be called with thread mutex locked 4156bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4157{ 4158 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4159 mWriteAckSequence, mDrainSequence); 4160 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4161 return true; 4162 } 4163 return false; 4164} 4165 4166// must be called with thread mutex locked 4167bool AudioFlinger::OffloadThread::shouldStandby_l() 4168{ 4169 bool TrackPaused = false; 4170 4171 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4172 // after a timeout and we will enter standby then. 4173 if (mTracks.size() > 0) { 4174 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4175 } 4176 4177 return !mStandby && !TrackPaused; 4178} 4179 4180 4181bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4182{ 4183 Mutex::Autolock _l(mLock); 4184 return waitingAsyncCallback_l(); 4185} 4186 4187void AudioFlinger::OffloadThread::flushHw_l() 4188{ 4189 mOutput->stream->flush(mOutput->stream); 4190 // Flush anything still waiting in the mixbuffer 4191 mCurrentWriteLength = 0; 4192 mBytesRemaining = 0; 4193 mPausedWriteLength = 0; 4194 mPausedBytesRemaining = 0; 4195 if (mUseAsyncWrite) { 4196 // discard any pending drain or write ack by incrementing sequence 4197 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4198 mDrainSequence = (mDrainSequence + 2) & ~1; 4199 ALOG_ASSERT(mCallbackThread != 0); 4200 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4201 mCallbackThread->setDraining(mDrainSequence); 4202 } 4203} 4204 4205// ---------------------------------------------------------------------------- 4206 4207AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4208 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4209 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4210 DUPLICATING), 4211 mWaitTimeMs(UINT_MAX) 4212{ 4213 addOutputTrack(mainThread); 4214} 4215 4216AudioFlinger::DuplicatingThread::~DuplicatingThread() 4217{ 4218 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4219 mOutputTracks[i]->destroy(); 4220 } 4221} 4222 4223void AudioFlinger::DuplicatingThread::threadLoop_mix() 4224{ 4225 // mix buffers... 4226 if (outputsReady(outputTracks)) { 4227 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4228 } else { 4229 memset(mMixBuffer, 0, mixBufferSize); 4230 } 4231 sleepTime = 0; 4232 writeFrames = mNormalFrameCount; 4233 mCurrentWriteLength = mixBufferSize; 4234 standbyTime = systemTime() + standbyDelay; 4235} 4236 4237void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4238{ 4239 if (sleepTime == 0) { 4240 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4241 sleepTime = activeSleepTime; 4242 } else { 4243 sleepTime = idleSleepTime; 4244 } 4245 } else if (mBytesWritten != 0) { 4246 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4247 writeFrames = mNormalFrameCount; 4248 memset(mMixBuffer, 0, mixBufferSize); 4249 } else { 4250 // flush remaining overflow buffers in output tracks 4251 writeFrames = 0; 4252 } 4253 sleepTime = 0; 4254 } 4255} 4256 4257ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4258{ 4259 for (size_t i = 0; i < outputTracks.size(); i++) { 4260 outputTracks[i]->write(mMixBuffer, writeFrames); 4261 } 4262 mStandby = false; 4263 return (ssize_t)mixBufferSize; 4264} 4265 4266void AudioFlinger::DuplicatingThread::threadLoop_standby() 4267{ 4268 // DuplicatingThread implements standby by stopping all tracks 4269 for (size_t i = 0; i < outputTracks.size(); i++) { 4270 outputTracks[i]->stop(); 4271 } 4272} 4273 4274void AudioFlinger::DuplicatingThread::saveOutputTracks() 4275{ 4276 outputTracks = mOutputTracks; 4277} 4278 4279void AudioFlinger::DuplicatingThread::clearOutputTracks() 4280{ 4281 outputTracks.clear(); 4282} 4283 4284void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4285{ 4286 Mutex::Autolock _l(mLock); 4287 // FIXME explain this formula 4288 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4289 OutputTrack *outputTrack = new OutputTrack(thread, 4290 this, 4291 mSampleRate, 4292 mFormat, 4293 mChannelMask, 4294 frameCount, 4295 IPCThreadState::self()->getCallingUid()); 4296 if (outputTrack->cblk() != NULL) { 4297 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4298 mOutputTracks.add(outputTrack); 4299 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4300 updateWaitTime_l(); 4301 } 4302} 4303 4304void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4305{ 4306 Mutex::Autolock _l(mLock); 4307 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4308 if (mOutputTracks[i]->thread() == thread) { 4309 mOutputTracks[i]->destroy(); 4310 mOutputTracks.removeAt(i); 4311 updateWaitTime_l(); 4312 return; 4313 } 4314 } 4315 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4316} 4317 4318// caller must hold mLock 4319void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4320{ 4321 mWaitTimeMs = UINT_MAX; 4322 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4323 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4324 if (strong != 0) { 4325 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4326 if (waitTimeMs < mWaitTimeMs) { 4327 mWaitTimeMs = waitTimeMs; 4328 } 4329 } 4330 } 4331} 4332 4333 4334bool AudioFlinger::DuplicatingThread::outputsReady( 4335 const SortedVector< sp<OutputTrack> > &outputTracks) 4336{ 4337 for (size_t i = 0; i < outputTracks.size(); i++) { 4338 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4339 if (thread == 0) { 4340 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4341 outputTracks[i].get()); 4342 return false; 4343 } 4344 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4345 // see note at standby() declaration 4346 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4347 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4348 thread.get()); 4349 return false; 4350 } 4351 } 4352 return true; 4353} 4354 4355uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4356{ 4357 return (mWaitTimeMs * 1000) / 2; 4358} 4359 4360void AudioFlinger::DuplicatingThread::cacheParameters_l() 4361{ 4362 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4363 updateWaitTime_l(); 4364 4365 MixerThread::cacheParameters_l(); 4366} 4367 4368// ---------------------------------------------------------------------------- 4369// Record 4370// ---------------------------------------------------------------------------- 4371 4372AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4373 AudioStreamIn *input, 4374 uint32_t sampleRate, 4375 audio_channel_mask_t channelMask, 4376 audio_io_handle_t id, 4377 audio_devices_t outDevice, 4378 audio_devices_t inDevice 4379#ifdef TEE_SINK 4380 , const sp<NBAIO_Sink>& teeSink 4381#endif 4382 ) : 4383 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4384 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4385 // mRsmpInIndex and mBufferSize set by readInputParameters() 4386 mReqChannelCount(popcount(channelMask)), 4387 mReqSampleRate(sampleRate) 4388 // mBytesRead is only meaningful while active, and so is cleared in start() 4389 // (but might be better to also clear here for dump?) 4390#ifdef TEE_SINK 4391 , mTeeSink(teeSink) 4392#endif 4393{ 4394 snprintf(mName, kNameLength, "AudioIn_%X", id); 4395 4396 readInputParameters(); 4397} 4398 4399 4400AudioFlinger::RecordThread::~RecordThread() 4401{ 4402 delete[] mRsmpInBuffer; 4403 delete mResampler; 4404 delete[] mRsmpOutBuffer; 4405} 4406 4407void AudioFlinger::RecordThread::onFirstRef() 4408{ 4409 run(mName, PRIORITY_URGENT_AUDIO); 4410} 4411 4412status_t AudioFlinger::RecordThread::readyToRun() 4413{ 4414 status_t status = initCheck(); 4415 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4416 return status; 4417} 4418 4419bool AudioFlinger::RecordThread::threadLoop() 4420{ 4421 AudioBufferProvider::Buffer buffer; 4422 sp<RecordTrack> activeTrack; 4423 Vector< sp<EffectChain> > effectChains; 4424 4425 nsecs_t lastWarning = 0; 4426 4427 inputStandBy(); 4428 { 4429 Mutex::Autolock _l(mLock); 4430 activeTrack = mActiveTrack; 4431 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4432 } 4433 4434 // used to verify we've read at least once before evaluating how many bytes were read 4435 bool readOnce = false; 4436 4437 // start recording 4438 while (!exitPending()) { 4439 4440 processConfigEvents(); 4441 4442 { // scope for mLock 4443 Mutex::Autolock _l(mLock); 4444 checkForNewParameters_l(); 4445 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4446 SortedVector<int> tmp; 4447 tmp.add(mActiveTrack->uid()); 4448 updateWakeLockUids_l(tmp); 4449 } 4450 activeTrack = mActiveTrack; 4451 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4452 standby(); 4453 4454 if (exitPending()) { 4455 break; 4456 } 4457 4458 releaseWakeLock_l(); 4459 ALOGV("RecordThread: loop stopping"); 4460 // go to sleep 4461 mWaitWorkCV.wait(mLock); 4462 ALOGV("RecordThread: loop starting"); 4463 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4464 continue; 4465 } 4466 if (mActiveTrack != 0) { 4467 if (mActiveTrack->isTerminated()) { 4468 removeTrack_l(mActiveTrack); 4469 mActiveTrack.clear(); 4470 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4471 standby(); 4472 mActiveTrack.clear(); 4473 mStartStopCond.broadcast(); 4474 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4475 if (mReqChannelCount != mActiveTrack->channelCount()) { 4476 mActiveTrack.clear(); 4477 mStartStopCond.broadcast(); 4478 } else if (readOnce) { 4479 // record start succeeds only if first read from audio input 4480 // succeeds 4481 if (mBytesRead >= 0) { 4482 mActiveTrack->mState = TrackBase::ACTIVE; 4483 } else { 4484 mActiveTrack.clear(); 4485 } 4486 mStartStopCond.broadcast(); 4487 } 4488 mStandby = false; 4489 } 4490 } 4491 4492 lockEffectChains_l(effectChains); 4493 } 4494 4495 if (mActiveTrack != 0) { 4496 if (mActiveTrack->mState != TrackBase::ACTIVE && 4497 mActiveTrack->mState != TrackBase::RESUMING) { 4498 unlockEffectChains(effectChains); 4499 usleep(kRecordThreadSleepUs); 4500 continue; 4501 } 4502 for (size_t i = 0; i < effectChains.size(); i ++) { 4503 effectChains[i]->process_l(); 4504 } 4505 4506 buffer.frameCount = mFrameCount; 4507 status_t status = mActiveTrack->getNextBuffer(&buffer); 4508 if (status == NO_ERROR) { 4509 readOnce = true; 4510 size_t framesOut = buffer.frameCount; 4511 if (mResampler == NULL) { 4512 // no resampling 4513 while (framesOut) { 4514 size_t framesIn = mFrameCount - mRsmpInIndex; 4515 if (framesIn) { 4516 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4517 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4518 mActiveTrack->mFrameSize; 4519 if (framesIn > framesOut) 4520 framesIn = framesOut; 4521 mRsmpInIndex += framesIn; 4522 framesOut -= framesIn; 4523 if (mChannelCount == mReqChannelCount) { 4524 memcpy(dst, src, framesIn * mFrameSize); 4525 } else { 4526 if (mChannelCount == 1) { 4527 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4528 (int16_t *)src, framesIn); 4529 } else { 4530 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4531 (int16_t *)src, framesIn); 4532 } 4533 } 4534 } 4535 if (framesOut && mFrameCount == mRsmpInIndex) { 4536 void *readInto; 4537 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4538 readInto = buffer.raw; 4539 framesOut = 0; 4540 } else { 4541 readInto = mRsmpInBuffer; 4542 mRsmpInIndex = 0; 4543 } 4544 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4545 mBufferSize); 4546 if (mBytesRead <= 0) { 4547 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4548 { 4549 ALOGE("Error reading audio input"); 4550 // Force input into standby so that it tries to 4551 // recover at next read attempt 4552 inputStandBy(); 4553 usleep(kRecordThreadSleepUs); 4554 } 4555 mRsmpInIndex = mFrameCount; 4556 framesOut = 0; 4557 buffer.frameCount = 0; 4558 } 4559#ifdef TEE_SINK 4560 else if (mTeeSink != 0) { 4561 (void) mTeeSink->write(readInto, 4562 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4563 } 4564#endif 4565 } 4566 } 4567 } else { 4568 // resampling 4569 4570 // resampler accumulates, but we only have one source track 4571 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4572 // alter output frame count as if we were expecting stereo samples 4573 if (mChannelCount == 1 && mReqChannelCount == 1) { 4574 framesOut >>= 1; 4575 } 4576 mResampler->resample(mRsmpOutBuffer, framesOut, 4577 this /* AudioBufferProvider* */); 4578 // ditherAndClamp() works as long as all buffers returned by 4579 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4580 if (mChannelCount == 2 && mReqChannelCount == 1) { 4581 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4582 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4583 // the resampler always outputs stereo samples: 4584 // do post stereo to mono conversion 4585 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4586 framesOut); 4587 } else { 4588 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4589 } 4590 // now done with mRsmpOutBuffer 4591 4592 } 4593 if (mFramestoDrop == 0) { 4594 mActiveTrack->releaseBuffer(&buffer); 4595 } else { 4596 if (mFramestoDrop > 0) { 4597 mFramestoDrop -= buffer.frameCount; 4598 if (mFramestoDrop <= 0) { 4599 clearSyncStartEvent(); 4600 } 4601 } else { 4602 mFramestoDrop += buffer.frameCount; 4603 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4604 mSyncStartEvent->isCancelled()) { 4605 ALOGW("Synced record %s, session %d, trigger session %d", 4606 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4607 mActiveTrack->sessionId(), 4608 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4609 clearSyncStartEvent(); 4610 } 4611 } 4612 } 4613 mActiveTrack->clearOverflow(); 4614 } 4615 // client isn't retrieving buffers fast enough 4616 else { 4617 if (!mActiveTrack->setOverflow()) { 4618 nsecs_t now = systemTime(); 4619 if ((now - lastWarning) > kWarningThrottleNs) { 4620 ALOGW("RecordThread: buffer overflow"); 4621 lastWarning = now; 4622 } 4623 } 4624 // Release the processor for a while before asking for a new buffer. 4625 // This will give the application more chance to read from the buffer and 4626 // clear the overflow. 4627 usleep(kRecordThreadSleepUs); 4628 } 4629 } 4630 // enable changes in effect chain 4631 unlockEffectChains(effectChains); 4632 effectChains.clear(); 4633 } 4634 4635 standby(); 4636 4637 { 4638 Mutex::Autolock _l(mLock); 4639 for (size_t i = 0; i < mTracks.size(); i++) { 4640 sp<RecordTrack> track = mTracks[i]; 4641 track->invalidate(); 4642 } 4643 mActiveTrack.clear(); 4644 mStartStopCond.broadcast(); 4645 } 4646 4647 releaseWakeLock(); 4648 4649 ALOGV("RecordThread %p exiting", this); 4650 return false; 4651} 4652 4653void AudioFlinger::RecordThread::standby() 4654{ 4655 if (!mStandby) { 4656 inputStandBy(); 4657 mStandby = true; 4658 } 4659} 4660 4661void AudioFlinger::RecordThread::inputStandBy() 4662{ 4663 mInput->stream->common.standby(&mInput->stream->common); 4664} 4665 4666sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4667 const sp<AudioFlinger::Client>& client, 4668 uint32_t sampleRate, 4669 audio_format_t format, 4670 audio_channel_mask_t channelMask, 4671 size_t frameCount, 4672 int sessionId, 4673 int uid, 4674 IAudioFlinger::track_flags_t *flags, 4675 pid_t tid, 4676 status_t *status) 4677{ 4678 sp<RecordTrack> track; 4679 status_t lStatus; 4680 4681 lStatus = initCheck(); 4682 if (lStatus != NO_ERROR) { 4683 ALOGE("createRecordTrack_l() audio driver not initialized"); 4684 goto Exit; 4685 } 4686 // client expresses a preference for FAST, but we get the final say 4687 if (*flags & IAudioFlinger::TRACK_FAST) { 4688 if ( 4689 // use case: callback handler and frame count is default or at least as large as HAL 4690 ( 4691 (tid != -1) && 4692 ((frameCount == 0) || 4693 (frameCount >= mFrameCount)) 4694 ) && 4695 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4696 // mono or stereo 4697 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4698 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4699 // hardware sample rate 4700 (sampleRate == mSampleRate) && 4701 // record thread has an associated fast recorder 4702 hasFastRecorder() 4703 // FIXME test that RecordThread for this fast track has a capable output HAL 4704 // FIXME add a permission test also? 4705 ) { 4706 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4707 if (frameCount == 0) { 4708 frameCount = mFrameCount * kFastTrackMultiplier; 4709 } 4710 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4711 frameCount, mFrameCount); 4712 } else { 4713 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4714 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4715 "hasFastRecorder=%d tid=%d", 4716 frameCount, mFrameCount, format, 4717 audio_is_linear_pcm(format), 4718 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4719 *flags &= ~IAudioFlinger::TRACK_FAST; 4720 // For compatibility with AudioRecord calculation, buffer depth is forced 4721 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4722 // This is probably too conservative, but legacy application code may depend on it. 4723 // If you change this calculation, also review the start threshold which is related. 4724 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4725 size_t mNormalFrameCount = 2048; // FIXME 4726 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4727 if (minBufCount < 2) { 4728 minBufCount = 2; 4729 } 4730 size_t minFrameCount = mNormalFrameCount * minBufCount; 4731 if (frameCount < minFrameCount) { 4732 frameCount = minFrameCount; 4733 } 4734 } 4735 } 4736 4737 // FIXME use flags and tid similar to createTrack_l() 4738 4739 { // scope for mLock 4740 Mutex::Autolock _l(mLock); 4741 4742 track = new RecordTrack(this, client, sampleRate, 4743 format, channelMask, frameCount, sessionId, uid); 4744 4745 if (track->getCblk() == 0) { 4746 ALOGE("createRecordTrack_l() no control block"); 4747 lStatus = NO_MEMORY; 4748 // track must be cleared from the caller as the caller has the AF lock 4749 goto Exit; 4750 } 4751 mTracks.add(track); 4752 4753 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4754 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4755 mAudioFlinger->btNrecIsOff(); 4756 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4757 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4758 4759 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4760 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4761 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4762 // so ask activity manager to do this on our behalf 4763 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4764 } 4765 } 4766 lStatus = NO_ERROR; 4767 4768Exit: 4769 if (status) { 4770 *status = lStatus; 4771 } 4772 return track; 4773} 4774 4775status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4776 AudioSystem::sync_event_t event, 4777 int triggerSession) 4778{ 4779 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4780 sp<ThreadBase> strongMe = this; 4781 status_t status = NO_ERROR; 4782 4783 if (event == AudioSystem::SYNC_EVENT_NONE) { 4784 clearSyncStartEvent(); 4785 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4786 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4787 triggerSession, 4788 recordTrack->sessionId(), 4789 syncStartEventCallback, 4790 this); 4791 // Sync event can be cancelled by the trigger session if the track is not in a 4792 // compatible state in which case we start record immediately 4793 if (mSyncStartEvent->isCancelled()) { 4794 clearSyncStartEvent(); 4795 } else { 4796 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4797 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4798 } 4799 } 4800 4801 { 4802 AutoMutex lock(mLock); 4803 if (mActiveTrack != 0) { 4804 if (recordTrack != mActiveTrack.get()) { 4805 status = -EBUSY; 4806 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4807 mActiveTrack->mState = TrackBase::ACTIVE; 4808 } 4809 return status; 4810 } 4811 4812 recordTrack->mState = TrackBase::IDLE; 4813 mActiveTrack = recordTrack; 4814 mLock.unlock(); 4815 status_t status = AudioSystem::startInput(mId); 4816 mLock.lock(); 4817 if (status != NO_ERROR) { 4818 mActiveTrack.clear(); 4819 clearSyncStartEvent(); 4820 return status; 4821 } 4822 mRsmpInIndex = mFrameCount; 4823 mBytesRead = 0; 4824 if (mResampler != NULL) { 4825 mResampler->reset(); 4826 } 4827 mActiveTrack->mState = TrackBase::RESUMING; 4828 // signal thread to start 4829 ALOGV("Signal record thread"); 4830 mWaitWorkCV.broadcast(); 4831 // do not wait for mStartStopCond if exiting 4832 if (exitPending()) { 4833 mActiveTrack.clear(); 4834 status = INVALID_OPERATION; 4835 goto startError; 4836 } 4837 mStartStopCond.wait(mLock); 4838 if (mActiveTrack == 0) { 4839 ALOGV("Record failed to start"); 4840 status = BAD_VALUE; 4841 goto startError; 4842 } 4843 ALOGV("Record started OK"); 4844 return status; 4845 } 4846 4847startError: 4848 AudioSystem::stopInput(mId); 4849 clearSyncStartEvent(); 4850 return status; 4851} 4852 4853void AudioFlinger::RecordThread::clearSyncStartEvent() 4854{ 4855 if (mSyncStartEvent != 0) { 4856 mSyncStartEvent->cancel(); 4857 } 4858 mSyncStartEvent.clear(); 4859 mFramestoDrop = 0; 4860} 4861 4862void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4863{ 4864 sp<SyncEvent> strongEvent = event.promote(); 4865 4866 if (strongEvent != 0) { 4867 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4868 me->handleSyncStartEvent(strongEvent); 4869 } 4870} 4871 4872void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4873{ 4874 if (event == mSyncStartEvent) { 4875 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4876 // from audio HAL 4877 mFramestoDrop = mFrameCount * 2; 4878 } 4879} 4880 4881bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4882 ALOGV("RecordThread::stop"); 4883 AutoMutex _l(mLock); 4884 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4885 return false; 4886 } 4887 recordTrack->mState = TrackBase::PAUSING; 4888 // do not wait for mStartStopCond if exiting 4889 if (exitPending()) { 4890 return true; 4891 } 4892 mStartStopCond.wait(mLock); 4893 // if we have been restarted, recordTrack == mActiveTrack.get() here 4894 if (exitPending() || recordTrack != mActiveTrack.get()) { 4895 ALOGV("Record stopped OK"); 4896 return true; 4897 } 4898 return false; 4899} 4900 4901bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4902{ 4903 return false; 4904} 4905 4906status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4907{ 4908#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4909 if (!isValidSyncEvent(event)) { 4910 return BAD_VALUE; 4911 } 4912 4913 int eventSession = event->triggerSession(); 4914 status_t ret = NAME_NOT_FOUND; 4915 4916 Mutex::Autolock _l(mLock); 4917 4918 for (size_t i = 0; i < mTracks.size(); i++) { 4919 sp<RecordTrack> track = mTracks[i]; 4920 if (eventSession == track->sessionId()) { 4921 (void) track->setSyncEvent(event); 4922 ret = NO_ERROR; 4923 } 4924 } 4925 return ret; 4926#else 4927 return BAD_VALUE; 4928#endif 4929} 4930 4931// destroyTrack_l() must be called with ThreadBase::mLock held 4932void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4933{ 4934 track->terminate(); 4935 track->mState = TrackBase::STOPPED; 4936 // active tracks are removed by threadLoop() 4937 if (mActiveTrack != track) { 4938 removeTrack_l(track); 4939 } 4940} 4941 4942void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4943{ 4944 mTracks.remove(track); 4945 // need anything related to effects here? 4946} 4947 4948void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4949{ 4950 dumpInternals(fd, args); 4951 dumpTracks(fd, args); 4952 dumpEffectChains(fd, args); 4953} 4954 4955void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4956{ 4957 const size_t SIZE = 256; 4958 char buffer[SIZE]; 4959 String8 result; 4960 4961 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4962 result.append(buffer); 4963 4964 if (mActiveTrack != 0) { 4965 snprintf(buffer, SIZE, "In index: %zu\n", mRsmpInIndex); 4966 result.append(buffer); 4967 snprintf(buffer, SIZE, "Buffer size: %zu bytes\n", mBufferSize); 4968 result.append(buffer); 4969 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4970 result.append(buffer); 4971 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4972 result.append(buffer); 4973 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4974 result.append(buffer); 4975 } else { 4976 result.append("No active record client\n"); 4977 } 4978 4979 write(fd, result.string(), result.size()); 4980 4981 dumpBase(fd, args); 4982} 4983 4984void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4985{ 4986 const size_t SIZE = 256; 4987 char buffer[SIZE]; 4988 String8 result; 4989 4990 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4991 result.append(buffer); 4992 RecordTrack::appendDumpHeader(result); 4993 for (size_t i = 0; i < mTracks.size(); ++i) { 4994 sp<RecordTrack> track = mTracks[i]; 4995 if (track != 0) { 4996 track->dump(buffer, SIZE); 4997 result.append(buffer); 4998 } 4999 } 5000 5001 if (mActiveTrack != 0) { 5002 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5003 result.append(buffer); 5004 RecordTrack::appendDumpHeader(result); 5005 mActiveTrack->dump(buffer, SIZE); 5006 result.append(buffer); 5007 5008 } 5009 write(fd, result.string(), result.size()); 5010} 5011 5012// AudioBufferProvider interface 5013status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5014{ 5015 size_t framesReq = buffer->frameCount; 5016 size_t framesReady = mFrameCount - mRsmpInIndex; 5017 int channelCount; 5018 5019 if (framesReady == 0) { 5020 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 5021 if (mBytesRead <= 0) { 5022 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 5023 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5024 // Force input into standby so that it tries to 5025 // recover at next read attempt 5026 inputStandBy(); 5027 usleep(kRecordThreadSleepUs); 5028 } 5029 buffer->raw = NULL; 5030 buffer->frameCount = 0; 5031 return NOT_ENOUGH_DATA; 5032 } 5033 mRsmpInIndex = 0; 5034 framesReady = mFrameCount; 5035 } 5036 5037 if (framesReq > framesReady) { 5038 framesReq = framesReady; 5039 } 5040 5041 if (mChannelCount == 1 && mReqChannelCount == 2) { 5042 channelCount = 1; 5043 } else { 5044 channelCount = 2; 5045 } 5046 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5047 buffer->frameCount = framesReq; 5048 return NO_ERROR; 5049} 5050 5051// AudioBufferProvider interface 5052void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5053{ 5054 mRsmpInIndex += buffer->frameCount; 5055 buffer->frameCount = 0; 5056} 5057 5058bool AudioFlinger::RecordThread::checkForNewParameters_l() 5059{ 5060 bool reconfig = false; 5061 5062 while (!mNewParameters.isEmpty()) { 5063 status_t status = NO_ERROR; 5064 String8 keyValuePair = mNewParameters[0]; 5065 AudioParameter param = AudioParameter(keyValuePair); 5066 int value; 5067 audio_format_t reqFormat = mFormat; 5068 uint32_t reqSamplingRate = mReqSampleRate; 5069 uint32_t reqChannelCount = mReqChannelCount; 5070 5071 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5072 reqSamplingRate = value; 5073 reconfig = true; 5074 } 5075 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5076 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5077 status = BAD_VALUE; 5078 } else { 5079 reqFormat = (audio_format_t) value; 5080 reconfig = true; 5081 } 5082 } 5083 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5084 reqChannelCount = popcount(value); 5085 reconfig = true; 5086 } 5087 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5088 // do not accept frame count changes if tracks are open as the track buffer 5089 // size depends on frame count and correct behavior would not be guaranteed 5090 // if frame count is changed after track creation 5091 if (mActiveTrack != 0) { 5092 status = INVALID_OPERATION; 5093 } else { 5094 reconfig = true; 5095 } 5096 } 5097 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5098 // forward device change to effects that have requested to be 5099 // aware of attached audio device. 5100 for (size_t i = 0; i < mEffectChains.size(); i++) { 5101 mEffectChains[i]->setDevice_l(value); 5102 } 5103 5104 // store input device and output device but do not forward output device to audio HAL. 5105 // Note that status is ignored by the caller for output device 5106 // (see AudioFlinger::setParameters() 5107 if (audio_is_output_devices(value)) { 5108 mOutDevice = value; 5109 status = BAD_VALUE; 5110 } else { 5111 mInDevice = value; 5112 // disable AEC and NS if the device is a BT SCO headset supporting those 5113 // pre processings 5114 if (mTracks.size() > 0) { 5115 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5116 mAudioFlinger->btNrecIsOff(); 5117 for (size_t i = 0; i < mTracks.size(); i++) { 5118 sp<RecordTrack> track = mTracks[i]; 5119 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5120 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5121 } 5122 } 5123 } 5124 } 5125 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5126 mAudioSource != (audio_source_t)value) { 5127 // forward device change to effects that have requested to be 5128 // aware of attached audio device. 5129 for (size_t i = 0; i < mEffectChains.size(); i++) { 5130 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5131 } 5132 mAudioSource = (audio_source_t)value; 5133 } 5134 if (status == NO_ERROR) { 5135 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5136 keyValuePair.string()); 5137 if (status == INVALID_OPERATION) { 5138 inputStandBy(); 5139 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5140 keyValuePair.string()); 5141 } 5142 if (reconfig) { 5143 if (status == BAD_VALUE && 5144 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5145 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5146 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5147 <= (2 * reqSamplingRate)) && 5148 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5149 <= FCC_2 && 5150 (reqChannelCount <= FCC_2)) { 5151 status = NO_ERROR; 5152 } 5153 if (status == NO_ERROR) { 5154 readInputParameters(); 5155 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5156 } 5157 } 5158 } 5159 5160 mNewParameters.removeAt(0); 5161 5162 mParamStatus = status; 5163 mParamCond.signal(); 5164 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5165 // already timed out waiting for the status and will never signal the condition. 5166 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5167 } 5168 return reconfig; 5169} 5170 5171String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5172{ 5173 Mutex::Autolock _l(mLock); 5174 if (initCheck() != NO_ERROR) { 5175 return String8(); 5176 } 5177 5178 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5179 const String8 out_s8(s); 5180 free(s); 5181 return out_s8; 5182} 5183 5184void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5185 AudioSystem::OutputDescriptor desc; 5186 void *param2 = NULL; 5187 5188 switch (event) { 5189 case AudioSystem::INPUT_OPENED: 5190 case AudioSystem::INPUT_CONFIG_CHANGED: 5191 desc.channelMask = mChannelMask; 5192 desc.samplingRate = mSampleRate; 5193 desc.format = mFormat; 5194 desc.frameCount = mFrameCount; 5195 desc.latency = 0; 5196 param2 = &desc; 5197 break; 5198 5199 case AudioSystem::INPUT_CLOSED: 5200 default: 5201 break; 5202 } 5203 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5204} 5205 5206void AudioFlinger::RecordThread::readInputParameters() 5207{ 5208 delete[] mRsmpInBuffer; 5209 // mRsmpInBuffer is always assigned a new[] below 5210 delete[] mRsmpOutBuffer; 5211 mRsmpOutBuffer = NULL; 5212 delete mResampler; 5213 mResampler = NULL; 5214 5215 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5216 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5217 mChannelCount = popcount(mChannelMask); 5218 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5219 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5220 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5221 } 5222 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5223 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5224 mFrameCount = mBufferSize / mFrameSize; 5225 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5226 5227 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5228 { 5229 int channelCount; 5230 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5231 // stereo to mono post process as the resampler always outputs stereo. 5232 if (mChannelCount == 1 && mReqChannelCount == 2) { 5233 channelCount = 1; 5234 } else { 5235 channelCount = 2; 5236 } 5237 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5238 mResampler->setSampleRate(mSampleRate); 5239 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5240 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5241 5242 // optmization: if mono to mono, alter input frame count as if we were inputing 5243 // stereo samples 5244 if (mChannelCount == 1 && mReqChannelCount == 1) { 5245 mFrameCount >>= 1; 5246 } 5247 5248 } 5249 mRsmpInIndex = mFrameCount; 5250} 5251 5252unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5253{ 5254 Mutex::Autolock _l(mLock); 5255 if (initCheck() != NO_ERROR) { 5256 return 0; 5257 } 5258 5259 return mInput->stream->get_input_frames_lost(mInput->stream); 5260} 5261 5262uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5263{ 5264 Mutex::Autolock _l(mLock); 5265 uint32_t result = 0; 5266 if (getEffectChain_l(sessionId) != 0) { 5267 result = EFFECT_SESSION; 5268 } 5269 5270 for (size_t i = 0; i < mTracks.size(); ++i) { 5271 if (sessionId == mTracks[i]->sessionId()) { 5272 result |= TRACK_SESSION; 5273 break; 5274 } 5275 } 5276 5277 return result; 5278} 5279 5280KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5281{ 5282 KeyedVector<int, bool> ids; 5283 Mutex::Autolock _l(mLock); 5284 for (size_t j = 0; j < mTracks.size(); ++j) { 5285 sp<RecordThread::RecordTrack> track = mTracks[j]; 5286 int sessionId = track->sessionId(); 5287 if (ids.indexOfKey(sessionId) < 0) { 5288 ids.add(sessionId, true); 5289 } 5290 } 5291 return ids; 5292} 5293 5294AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5295{ 5296 Mutex::Autolock _l(mLock); 5297 AudioStreamIn *input = mInput; 5298 mInput = NULL; 5299 return input; 5300} 5301 5302// this method must always be called either with ThreadBase mLock held or inside the thread loop 5303audio_stream_t* AudioFlinger::RecordThread::stream() const 5304{ 5305 if (mInput == NULL) { 5306 return NULL; 5307 } 5308 return &mInput->stream->common; 5309} 5310 5311status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5312{ 5313 // only one chain per input thread 5314 if (mEffectChains.size() != 0) { 5315 return INVALID_OPERATION; 5316 } 5317 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5318 5319 chain->setInBuffer(NULL); 5320 chain->setOutBuffer(NULL); 5321 5322 checkSuspendOnAddEffectChain_l(chain); 5323 5324 mEffectChains.add(chain); 5325 5326 return NO_ERROR; 5327} 5328 5329size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5330{ 5331 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5332 ALOGW_IF(mEffectChains.size() != 1, 5333 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5334 chain.get(), mEffectChains.size(), this); 5335 if (mEffectChains.size() == 1) { 5336 mEffectChains.removeAt(0); 5337 } 5338 return 0; 5339} 5340 5341}; // namespace android 5342