Threads.cpp revision 398f21348e5100289f6e5be30c8b5257fa04aaf9
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 //FIXME: mStandby should be true here. Is this some kind of hack? 276 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 277 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 278 // mName will be set by concrete (non-virtual) subclass 279 mDeathRecipient(new PMDeathRecipient(this)) 280{ 281} 282 283AudioFlinger::ThreadBase::~ThreadBase() 284{ 285 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 286 for (size_t i = 0; i < mConfigEvents.size(); i++) { 287 delete mConfigEvents[i]; 288 } 289 mConfigEvents.clear(); 290 291 mParamCond.broadcast(); 292 // do not lock the mutex in destructor 293 releaseWakeLock_l(); 294 if (mPowerManager != 0) { 295 sp<IBinder> binder = mPowerManager->asBinder(); 296 binder->unlinkToDeath(mDeathRecipient); 297 } 298} 299 300void AudioFlinger::ThreadBase::exit() 301{ 302 ALOGV("ThreadBase::exit"); 303 // do any cleanup required for exit to succeed 304 preExit(); 305 { 306 // This lock prevents the following race in thread (uniprocessor for illustration): 307 // if (!exitPending()) { 308 // // context switch from here to exit() 309 // // exit() calls requestExit(), what exitPending() observes 310 // // exit() calls signal(), which is dropped since no waiters 311 // // context switch back from exit() to here 312 // mWaitWorkCV.wait(...); 313 // // now thread is hung 314 // } 315 AutoMutex lock(mLock); 316 requestExit(); 317 mWaitWorkCV.broadcast(); 318 } 319 // When Thread::requestExitAndWait is made virtual and this method is renamed to 320 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 321 requestExitAndWait(); 322} 323 324status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 325{ 326 status_t status; 327 328 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 329 Mutex::Autolock _l(mLock); 330 331 mNewParameters.add(keyValuePairs); 332 mWaitWorkCV.signal(); 333 // wait condition with timeout in case the thread loop has exited 334 // before the request could be processed 335 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 336 status = mParamStatus; 337 mWaitWorkCV.signal(); 338 } else { 339 status = TIMED_OUT; 340 } 341 return status; 342} 343 344void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 345{ 346 Mutex::Autolock _l(mLock); 347 sendIoConfigEvent_l(event, param); 348} 349 350// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 351void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 352{ 353 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 354 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 355 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 356 param); 357 mWaitWorkCV.signal(); 358} 359 360// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 361void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 362{ 363 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 364 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 365 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 366 mConfigEvents.size(), pid, tid, prio); 367 mWaitWorkCV.signal(); 368} 369 370void AudioFlinger::ThreadBase::processConfigEvents() 371{ 372 mLock.lock(); 373 while (!mConfigEvents.isEmpty()) { 374 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 375 ConfigEvent *event = mConfigEvents[0]; 376 mConfigEvents.removeAt(0); 377 // release mLock before locking AudioFlinger mLock: lock order is always 378 // AudioFlinger then ThreadBase to avoid cross deadlock 379 mLock.unlock(); 380 switch(event->type()) { 381 case CFG_EVENT_PRIO: { 382 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 383 // FIXME Need to understand why this has be done asynchronously 384 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 385 true /*asynchronous*/); 386 if (err != 0) { 387 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 388 "error %d", 389 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 390 } 391 } break; 392 case CFG_EVENT_IO: { 393 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 394 mAudioFlinger->mLock.lock(); 395 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 396 mAudioFlinger->mLock.unlock(); 397 } break; 398 default: 399 ALOGE("processConfigEvents() unknown event type %d", event->type()); 400 break; 401 } 402 delete event; 403 mLock.lock(); 404 } 405 mLock.unlock(); 406} 407 408void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 409{ 410 const size_t SIZE = 256; 411 char buffer[SIZE]; 412 String8 result; 413 414 bool locked = AudioFlinger::dumpTryLock(mLock); 415 if (!locked) { 416 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 417 write(fd, buffer, strlen(buffer)); 418 } 419 420 snprintf(buffer, SIZE, "io handle: %d\n", mId); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 437 result.append(buffer); 438 439 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 440 result.append(buffer); 441 result.append(" Index Command"); 442 for (size_t i = 0; i < mNewParameters.size(); ++i) { 443 snprintf(buffer, SIZE, "\n %02d ", i); 444 result.append(buffer); 445 result.append(mNewParameters[i]); 446 } 447 448 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 449 result.append(buffer); 450 for (size_t i = 0; i < mConfigEvents.size(); i++) { 451 mConfigEvents[i]->dump(buffer, SIZE); 452 result.append(buffer); 453 } 454 result.append("\n"); 455 456 write(fd, result.string(), result.size()); 457 458 if (locked) { 459 mLock.unlock(); 460 } 461} 462 463void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 464{ 465 const size_t SIZE = 256; 466 char buffer[SIZE]; 467 String8 result; 468 469 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 470 write(fd, buffer, strlen(buffer)); 471 472 for (size_t i = 0; i < mEffectChains.size(); ++i) { 473 sp<EffectChain> chain = mEffectChains[i]; 474 if (chain != 0) { 475 chain->dump(fd, args); 476 } 477 } 478} 479 480void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 481{ 482 Mutex::Autolock _l(mLock); 483 acquireWakeLock_l(uid); 484} 485 486String16 AudioFlinger::ThreadBase::getWakeLockTag() 487{ 488 switch (mType) { 489 case MIXER: 490 return String16("AudioMix"); 491 case DIRECT: 492 return String16("AudioDirectOut"); 493 case DUPLICATING: 494 return String16("AudioDup"); 495 case RECORD: 496 return String16("AudioIn"); 497 case OFFLOAD: 498 return String16("AudioOffload"); 499 default: 500 ALOG_ASSERT(false); 501 return String16("AudioUnknown"); 502 } 503} 504 505void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 506{ 507 getPowerManager_l(); 508 if (mPowerManager != 0) { 509 sp<IBinder> binder = new BBinder(); 510 status_t status; 511 if (uid >= 0) { 512 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 513 binder, 514 getWakeLockTag(), 515 String16("media"), 516 uid); 517 } else { 518 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 519 binder, 520 getWakeLockTag(), 521 String16("media")); 522 } 523 if (status == NO_ERROR) { 524 mWakeLockToken = binder; 525 } 526 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 527 } 528} 529 530void AudioFlinger::ThreadBase::releaseWakeLock() 531{ 532 Mutex::Autolock _l(mLock); 533 releaseWakeLock_l(); 534} 535 536void AudioFlinger::ThreadBase::releaseWakeLock_l() 537{ 538 if (mWakeLockToken != 0) { 539 ALOGV("releaseWakeLock_l() %s", mName); 540 if (mPowerManager != 0) { 541 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 542 } 543 mWakeLockToken.clear(); 544 } 545} 546 547void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 548 Mutex::Autolock _l(mLock); 549 updateWakeLockUids_l(uids); 550} 551 552void AudioFlinger::ThreadBase::getPowerManager_l() { 553 554 if (mPowerManager == 0) { 555 // use checkService() to avoid blocking if power service is not up yet 556 sp<IBinder> binder = 557 defaultServiceManager()->checkService(String16("power")); 558 if (binder == 0) { 559 ALOGW("Thread %s cannot connect to the power manager service", mName); 560 } else { 561 mPowerManager = interface_cast<IPowerManager>(binder); 562 binder->linkToDeath(mDeathRecipient); 563 } 564 } 565} 566 567void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 568 569 getPowerManager_l(); 570 if (mWakeLockToken == NULL) { 571 ALOGE("no wake lock to update!"); 572 return; 573 } 574 if (mPowerManager != 0) { 575 sp<IBinder> binder = new BBinder(); 576 status_t status; 577 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 578 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 579 } 580} 581 582void AudioFlinger::ThreadBase::clearPowerManager() 583{ 584 Mutex::Autolock _l(mLock); 585 releaseWakeLock_l(); 586 mPowerManager.clear(); 587} 588 589void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 590{ 591 sp<ThreadBase> thread = mThread.promote(); 592 if (thread != 0) { 593 thread->clearPowerManager(); 594 } 595 ALOGW("power manager service died !!!"); 596} 597 598void AudioFlinger::ThreadBase::setEffectSuspended( 599 const effect_uuid_t *type, bool suspend, int sessionId) 600{ 601 Mutex::Autolock _l(mLock); 602 setEffectSuspended_l(type, suspend, sessionId); 603} 604 605void AudioFlinger::ThreadBase::setEffectSuspended_l( 606 const effect_uuid_t *type, bool suspend, int sessionId) 607{ 608 sp<EffectChain> chain = getEffectChain_l(sessionId); 609 if (chain != 0) { 610 if (type != NULL) { 611 chain->setEffectSuspended_l(type, suspend); 612 } else { 613 chain->setEffectSuspendedAll_l(suspend); 614 } 615 } 616 617 updateSuspendedSessions_l(type, suspend, sessionId); 618} 619 620void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 621{ 622 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 623 if (index < 0) { 624 return; 625 } 626 627 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 628 mSuspendedSessions.valueAt(index); 629 630 for (size_t i = 0; i < sessionEffects.size(); i++) { 631 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 632 for (int j = 0; j < desc->mRefCount; j++) { 633 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 634 chain->setEffectSuspendedAll_l(true); 635 } else { 636 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 637 desc->mType.timeLow); 638 chain->setEffectSuspended_l(&desc->mType, true); 639 } 640 } 641 } 642} 643 644void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 645 bool suspend, 646 int sessionId) 647{ 648 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 649 650 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 651 652 if (suspend) { 653 if (index >= 0) { 654 sessionEffects = mSuspendedSessions.valueAt(index); 655 } else { 656 mSuspendedSessions.add(sessionId, sessionEffects); 657 } 658 } else { 659 if (index < 0) { 660 return; 661 } 662 sessionEffects = mSuspendedSessions.valueAt(index); 663 } 664 665 666 int key = EffectChain::kKeyForSuspendAll; 667 if (type != NULL) { 668 key = type->timeLow; 669 } 670 index = sessionEffects.indexOfKey(key); 671 672 sp<SuspendedSessionDesc> desc; 673 if (suspend) { 674 if (index >= 0) { 675 desc = sessionEffects.valueAt(index); 676 } else { 677 desc = new SuspendedSessionDesc(); 678 if (type != NULL) { 679 desc->mType = *type; 680 } 681 sessionEffects.add(key, desc); 682 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 683 } 684 desc->mRefCount++; 685 } else { 686 if (index < 0) { 687 return; 688 } 689 desc = sessionEffects.valueAt(index); 690 if (--desc->mRefCount == 0) { 691 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 692 sessionEffects.removeItemsAt(index); 693 if (sessionEffects.isEmpty()) { 694 ALOGV("updateSuspendedSessions_l() restore removing session %d", 695 sessionId); 696 mSuspendedSessions.removeItem(sessionId); 697 } 698 } 699 } 700 if (!sessionEffects.isEmpty()) { 701 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 702 } 703} 704 705void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 706 bool enabled, 707 int sessionId) 708{ 709 Mutex::Autolock _l(mLock); 710 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 711} 712 713void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 714 bool enabled, 715 int sessionId) 716{ 717 if (mType != RECORD) { 718 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 719 // another session. This gives the priority to well behaved effect control panels 720 // and applications not using global effects. 721 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 722 // global effects 723 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 724 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 725 } 726 } 727 728 sp<EffectChain> chain = getEffectChain_l(sessionId); 729 if (chain != 0) { 730 chain->checkSuspendOnEffectEnabled(effect, enabled); 731 } 732} 733 734// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 735sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 736 const sp<AudioFlinger::Client>& client, 737 const sp<IEffectClient>& effectClient, 738 int32_t priority, 739 int sessionId, 740 effect_descriptor_t *desc, 741 int *enabled, 742 status_t *status 743 ) 744{ 745 sp<EffectModule> effect; 746 sp<EffectHandle> handle; 747 status_t lStatus; 748 sp<EffectChain> chain; 749 bool chainCreated = false; 750 bool effectCreated = false; 751 bool effectRegistered = false; 752 753 lStatus = initCheck(); 754 if (lStatus != NO_ERROR) { 755 ALOGW("createEffect_l() Audio driver not initialized."); 756 goto Exit; 757 } 758 759 // Allow global effects only on offloaded and mixer threads 760 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 761 switch (mType) { 762 case MIXER: 763 case OFFLOAD: 764 break; 765 case DIRECT: 766 case DUPLICATING: 767 case RECORD: 768 default: 769 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 770 lStatus = BAD_VALUE; 771 goto Exit; 772 } 773 } 774 775 // Only Pre processor effects are allowed on input threads and only on input threads 776 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 777 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 778 desc->name, desc->flags, mType); 779 lStatus = BAD_VALUE; 780 goto Exit; 781 } 782 783 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 784 785 { // scope for mLock 786 Mutex::Autolock _l(mLock); 787 788 // check for existing effect chain with the requested audio session 789 chain = getEffectChain_l(sessionId); 790 if (chain == 0) { 791 // create a new chain for this session 792 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 793 chain = new EffectChain(this, sessionId); 794 addEffectChain_l(chain); 795 chain->setStrategy(getStrategyForSession_l(sessionId)); 796 chainCreated = true; 797 } else { 798 effect = chain->getEffectFromDesc_l(desc); 799 } 800 801 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 802 803 if (effect == 0) { 804 int id = mAudioFlinger->nextUniqueId(); 805 // Check CPU and memory usage 806 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 807 if (lStatus != NO_ERROR) { 808 goto Exit; 809 } 810 effectRegistered = true; 811 // create a new effect module if none present in the chain 812 effect = new EffectModule(this, chain, desc, id, sessionId); 813 lStatus = effect->status(); 814 if (lStatus != NO_ERROR) { 815 goto Exit; 816 } 817 effect->setOffloaded(mType == OFFLOAD, mId); 818 819 lStatus = chain->addEffect_l(effect); 820 if (lStatus != NO_ERROR) { 821 goto Exit; 822 } 823 effectCreated = true; 824 825 effect->setDevice(mOutDevice); 826 effect->setDevice(mInDevice); 827 effect->setMode(mAudioFlinger->getMode()); 828 effect->setAudioSource(mAudioSource); 829 } 830 // create effect handle and connect it to effect module 831 handle = new EffectHandle(effect, client, effectClient, priority); 832 lStatus = effect->addHandle(handle.get()); 833 if (enabled != NULL) { 834 *enabled = (int)effect->isEnabled(); 835 } 836 } 837 838Exit: 839 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 840 Mutex::Autolock _l(mLock); 841 if (effectCreated) { 842 chain->removeEffect_l(effect); 843 } 844 if (effectRegistered) { 845 AudioSystem::unregisterEffect(effect->id()); 846 } 847 if (chainCreated) { 848 removeEffectChain_l(chain); 849 } 850 handle.clear(); 851 } 852 853 if (status != NULL) { 854 *status = lStatus; 855 } 856 return handle; 857} 858 859sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 860{ 861 Mutex::Autolock _l(mLock); 862 return getEffect_l(sessionId, effectId); 863} 864 865sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 866{ 867 sp<EffectChain> chain = getEffectChain_l(sessionId); 868 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 869} 870 871// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 872// PlaybackThread::mLock held 873status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 874{ 875 // check for existing effect chain with the requested audio session 876 int sessionId = effect->sessionId(); 877 sp<EffectChain> chain = getEffectChain_l(sessionId); 878 bool chainCreated = false; 879 880 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 881 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 882 this, effect->desc().name, effect->desc().flags); 883 884 if (chain == 0) { 885 // create a new chain for this session 886 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 887 chain = new EffectChain(this, sessionId); 888 addEffectChain_l(chain); 889 chain->setStrategy(getStrategyForSession_l(sessionId)); 890 chainCreated = true; 891 } 892 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 893 894 if (chain->getEffectFromId_l(effect->id()) != 0) { 895 ALOGW("addEffect_l() %p effect %s already present in chain %p", 896 this, effect->desc().name, chain.get()); 897 return BAD_VALUE; 898 } 899 900 effect->setOffloaded(mType == OFFLOAD, mId); 901 902 status_t status = chain->addEffect_l(effect); 903 if (status != NO_ERROR) { 904 if (chainCreated) { 905 removeEffectChain_l(chain); 906 } 907 return status; 908 } 909 910 effect->setDevice(mOutDevice); 911 effect->setDevice(mInDevice); 912 effect->setMode(mAudioFlinger->getMode()); 913 effect->setAudioSource(mAudioSource); 914 return NO_ERROR; 915} 916 917void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 918 919 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 920 effect_descriptor_t desc = effect->desc(); 921 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 922 detachAuxEffect_l(effect->id()); 923 } 924 925 sp<EffectChain> chain = effect->chain().promote(); 926 if (chain != 0) { 927 // remove effect chain if removing last effect 928 if (chain->removeEffect_l(effect) == 0) { 929 removeEffectChain_l(chain); 930 } 931 } else { 932 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 933 } 934} 935 936void AudioFlinger::ThreadBase::lockEffectChains_l( 937 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 938{ 939 effectChains = mEffectChains; 940 for (size_t i = 0; i < mEffectChains.size(); i++) { 941 mEffectChains[i]->lock(); 942 } 943} 944 945void AudioFlinger::ThreadBase::unlockEffectChains( 946 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 947{ 948 for (size_t i = 0; i < effectChains.size(); i++) { 949 effectChains[i]->unlock(); 950 } 951} 952 953sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 954{ 955 Mutex::Autolock _l(mLock); 956 return getEffectChain_l(sessionId); 957} 958 959sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 960{ 961 size_t size = mEffectChains.size(); 962 for (size_t i = 0; i < size; i++) { 963 if (mEffectChains[i]->sessionId() == sessionId) { 964 return mEffectChains[i]; 965 } 966 } 967 return 0; 968} 969 970void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 971{ 972 Mutex::Autolock _l(mLock); 973 size_t size = mEffectChains.size(); 974 for (size_t i = 0; i < size; i++) { 975 mEffectChains[i]->setMode_l(mode); 976 } 977} 978 979void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 980 EffectHandle *handle, 981 bool unpinIfLast) { 982 983 Mutex::Autolock _l(mLock); 984 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 985 // delete the effect module if removing last handle on it 986 if (effect->removeHandle(handle) == 0) { 987 if (!effect->isPinned() || unpinIfLast) { 988 removeEffect_l(effect); 989 AudioSystem::unregisterEffect(effect->id()); 990 } 991 } 992} 993 994// ---------------------------------------------------------------------------- 995// Playback 996// ---------------------------------------------------------------------------- 997 998AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 999 AudioStreamOut* output, 1000 audio_io_handle_t id, 1001 audio_devices_t device, 1002 type_t type) 1003 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1004 mNormalFrameCount(0), mMixBuffer(NULL), 1005 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1006 mActiveTracksGeneration(0), 1007 // mStreamTypes[] initialized in constructor body 1008 mOutput(output), 1009 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1010 mMixerStatus(MIXER_IDLE), 1011 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1012 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1013 mBytesRemaining(0), 1014 mCurrentWriteLength(0), 1015 mUseAsyncWrite(false), 1016 mWriteAckSequence(0), 1017 mDrainSequence(0), 1018 mSignalPending(false), 1019 mScreenState(AudioFlinger::mScreenState), 1020 // index 0 is reserved for normal mixer's submix 1021 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1022 // mLatchD, mLatchQ, 1023 mLatchDValid(false), mLatchQValid(false) 1024{ 1025 snprintf(mName, kNameLength, "AudioOut_%X", id); 1026 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1027 1028 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1029 // it would be safer to explicitly pass initial masterVolume/masterMute as 1030 // parameter. 1031 // 1032 // If the HAL we are using has support for master volume or master mute, 1033 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1034 // and the mute set to false). 1035 mMasterVolume = audioFlinger->masterVolume_l(); 1036 mMasterMute = audioFlinger->masterMute_l(); 1037 if (mOutput && mOutput->audioHwDev) { 1038 if (mOutput->audioHwDev->canSetMasterVolume()) { 1039 mMasterVolume = 1.0; 1040 } 1041 1042 if (mOutput->audioHwDev->canSetMasterMute()) { 1043 mMasterMute = false; 1044 } 1045 } 1046 1047 readOutputParameters(); 1048 1049 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1050 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1051 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1052 stream = (audio_stream_type_t) (stream + 1)) { 1053 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1054 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1055 } 1056 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1057 // because mAudioFlinger doesn't have one to copy from 1058} 1059 1060AudioFlinger::PlaybackThread::~PlaybackThread() 1061{ 1062 mAudioFlinger->unregisterWriter(mNBLogWriter); 1063 delete [] mAllocMixBuffer; 1064} 1065 1066void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1067{ 1068 dumpInternals(fd, args); 1069 dumpTracks(fd, args); 1070 dumpEffectChains(fd, args); 1071} 1072 1073void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1074{ 1075 const size_t SIZE = 256; 1076 char buffer[SIZE]; 1077 String8 result; 1078 1079 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1080 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1081 const stream_type_t *st = &mStreamTypes[i]; 1082 if (i > 0) { 1083 result.appendFormat(", "); 1084 } 1085 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1086 if (st->mute) { 1087 result.append("M"); 1088 } 1089 } 1090 result.append("\n"); 1091 write(fd, result.string(), result.length()); 1092 result.clear(); 1093 1094 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1095 result.append(buffer); 1096 Track::appendDumpHeader(result); 1097 for (size_t i = 0; i < mTracks.size(); ++i) { 1098 sp<Track> track = mTracks[i]; 1099 if (track != 0) { 1100 track->dump(buffer, SIZE); 1101 result.append(buffer); 1102 } 1103 } 1104 1105 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1106 result.append(buffer); 1107 Track::appendDumpHeader(result); 1108 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1109 sp<Track> track = mActiveTracks[i].promote(); 1110 if (track != 0) { 1111 track->dump(buffer, SIZE); 1112 result.append(buffer); 1113 } 1114 } 1115 write(fd, result.string(), result.size()); 1116 1117 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1118 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1119 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1120 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1121} 1122 1123void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1124{ 1125 const size_t SIZE = 256; 1126 char buffer[SIZE]; 1127 String8 result; 1128 1129 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1130 result.append(buffer); 1131 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1132 result.append(buffer); 1133 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1134 ns2ms(systemTime() - mLastWriteTime)); 1135 result.append(buffer); 1136 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1137 result.append(buffer); 1138 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1139 result.append(buffer); 1140 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1141 result.append(buffer); 1142 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1143 result.append(buffer); 1144 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1145 result.append(buffer); 1146 write(fd, result.string(), result.size()); 1147 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1148 1149 dumpBase(fd, args); 1150} 1151 1152// Thread virtuals 1153status_t AudioFlinger::PlaybackThread::readyToRun() 1154{ 1155 status_t status = initCheck(); 1156 if (status == NO_ERROR) { 1157 ALOGI("AudioFlinger's thread %p ready to run", this); 1158 } else { 1159 ALOGE("No working audio driver found."); 1160 } 1161 return status; 1162} 1163 1164void AudioFlinger::PlaybackThread::onFirstRef() 1165{ 1166 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1167} 1168 1169// ThreadBase virtuals 1170void AudioFlinger::PlaybackThread::preExit() 1171{ 1172 ALOGV(" preExit()"); 1173 // FIXME this is using hard-coded strings but in the future, this functionality will be 1174 // converted to use audio HAL extensions required to support tunneling 1175 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1176} 1177 1178// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1179sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1180 const sp<AudioFlinger::Client>& client, 1181 audio_stream_type_t streamType, 1182 uint32_t sampleRate, 1183 audio_format_t format, 1184 audio_channel_mask_t channelMask, 1185 size_t frameCount, 1186 const sp<IMemory>& sharedBuffer, 1187 int sessionId, 1188 IAudioFlinger::track_flags_t *flags, 1189 pid_t tid, 1190 int uid, 1191 status_t *status) 1192{ 1193 sp<Track> track; 1194 status_t lStatus; 1195 1196 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1197 1198 // client expresses a preference for FAST, but we get the final say 1199 if (*flags & IAudioFlinger::TRACK_FAST) { 1200 if ( 1201 // not timed 1202 (!isTimed) && 1203 // either of these use cases: 1204 ( 1205 // use case 1: shared buffer with any frame count 1206 ( 1207 (sharedBuffer != 0) 1208 ) || 1209 // use case 2: callback handler and frame count is default or at least as large as HAL 1210 ( 1211 (tid != -1) && 1212 ((frameCount == 0) || 1213 (frameCount >= mFrameCount)) 1214 ) 1215 ) && 1216 // PCM data 1217 audio_is_linear_pcm(format) && 1218 // mono or stereo 1219 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1220 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1221 // hardware sample rate 1222 (sampleRate == mSampleRate) && 1223 // normal mixer has an associated fast mixer 1224 hasFastMixer() && 1225 // there are sufficient fast track slots available 1226 (mFastTrackAvailMask != 0) 1227 // FIXME test that MixerThread for this fast track has a capable output HAL 1228 // FIXME add a permission test also? 1229 ) { 1230 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1231 if (frameCount == 0) { 1232 frameCount = mFrameCount * kFastTrackMultiplier; 1233 } 1234 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1235 frameCount, mFrameCount); 1236 } else { 1237 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1238 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1239 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1240 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1241 audio_is_linear_pcm(format), 1242 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1243 *flags &= ~IAudioFlinger::TRACK_FAST; 1244 // For compatibility with AudioTrack calculation, buffer depth is forced 1245 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1246 // This is probably too conservative, but legacy application code may depend on it. 1247 // If you change this calculation, also review the start threshold which is related. 1248 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1249 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1250 if (minBufCount < 2) { 1251 minBufCount = 2; 1252 } 1253 size_t minFrameCount = mNormalFrameCount * minBufCount; 1254 if (frameCount < minFrameCount) { 1255 frameCount = minFrameCount; 1256 } 1257 } 1258 } 1259 1260 if (mType == DIRECT) { 1261 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1262 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1263 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1264 "for output %p with format %d", 1265 sampleRate, format, channelMask, mOutput, mFormat); 1266 lStatus = BAD_VALUE; 1267 goto Exit; 1268 } 1269 } 1270 } else if (mType == OFFLOAD) { 1271 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1272 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1273 "for output %p with format %d", 1274 sampleRate, format, channelMask, mOutput, mFormat); 1275 lStatus = BAD_VALUE; 1276 goto Exit; 1277 } 1278 } else { 1279 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1280 ALOGE("createTrack_l() Bad parameter: format %d \"" 1281 "for output %p with format %d", 1282 format, mOutput, mFormat); 1283 lStatus = BAD_VALUE; 1284 goto Exit; 1285 } 1286 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1287 if (sampleRate > mSampleRate*2) { 1288 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1289 lStatus = BAD_VALUE; 1290 goto Exit; 1291 } 1292 } 1293 1294 lStatus = initCheck(); 1295 if (lStatus != NO_ERROR) { 1296 ALOGE("Audio driver not initialized."); 1297 goto Exit; 1298 } 1299 1300 { // scope for mLock 1301 Mutex::Autolock _l(mLock); 1302 1303 // all tracks in same audio session must share the same routing strategy otherwise 1304 // conflicts will happen when tracks are moved from one output to another by audio policy 1305 // manager 1306 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1307 for (size_t i = 0; i < mTracks.size(); ++i) { 1308 sp<Track> t = mTracks[i]; 1309 if (t != 0 && !t->isOutputTrack()) { 1310 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1311 if (sessionId == t->sessionId() && strategy != actual) { 1312 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1313 strategy, actual); 1314 lStatus = BAD_VALUE; 1315 goto Exit; 1316 } 1317 } 1318 } 1319 1320 if (!isTimed) { 1321 track = new Track(this, client, streamType, sampleRate, format, 1322 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1323 } else { 1324 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1325 channelMask, frameCount, sharedBuffer, sessionId, uid); 1326 } 1327 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1328 lStatus = NO_MEMORY; 1329 goto Exit; 1330 } 1331 1332 mTracks.add(track); 1333 1334 sp<EffectChain> chain = getEffectChain_l(sessionId); 1335 if (chain != 0) { 1336 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1337 track->setMainBuffer(chain->inBuffer()); 1338 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1339 chain->incTrackCnt(); 1340 } 1341 1342 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1343 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1344 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1345 // so ask activity manager to do this on our behalf 1346 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1347 } 1348 } 1349 1350 lStatus = NO_ERROR; 1351 1352Exit: 1353 if (status) { 1354 *status = lStatus; 1355 } 1356 return track; 1357} 1358 1359uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1360{ 1361 return latency; 1362} 1363 1364uint32_t AudioFlinger::PlaybackThread::latency() const 1365{ 1366 Mutex::Autolock _l(mLock); 1367 return latency_l(); 1368} 1369uint32_t AudioFlinger::PlaybackThread::latency_l() const 1370{ 1371 if (initCheck() == NO_ERROR) { 1372 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1373 } else { 1374 return 0; 1375 } 1376} 1377 1378void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1379{ 1380 Mutex::Autolock _l(mLock); 1381 // Don't apply master volume in SW if our HAL can do it for us. 1382 if (mOutput && mOutput->audioHwDev && 1383 mOutput->audioHwDev->canSetMasterVolume()) { 1384 mMasterVolume = 1.0; 1385 } else { 1386 mMasterVolume = value; 1387 } 1388} 1389 1390void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1391{ 1392 Mutex::Autolock _l(mLock); 1393 // Don't apply master mute in SW if our HAL can do it for us. 1394 if (mOutput && mOutput->audioHwDev && 1395 mOutput->audioHwDev->canSetMasterMute()) { 1396 mMasterMute = false; 1397 } else { 1398 mMasterMute = muted; 1399 } 1400} 1401 1402void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1403{ 1404 Mutex::Autolock _l(mLock); 1405 mStreamTypes[stream].volume = value; 1406 broadcast_l(); 1407} 1408 1409void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1410{ 1411 Mutex::Autolock _l(mLock); 1412 mStreamTypes[stream].mute = muted; 1413 broadcast_l(); 1414} 1415 1416float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1417{ 1418 Mutex::Autolock _l(mLock); 1419 return mStreamTypes[stream].volume; 1420} 1421 1422// addTrack_l() must be called with ThreadBase::mLock held 1423status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1424{ 1425 status_t status = ALREADY_EXISTS; 1426 1427 // set retry count for buffer fill 1428 track->mRetryCount = kMaxTrackStartupRetries; 1429 if (mActiveTracks.indexOf(track) < 0) { 1430 // the track is newly added, make sure it fills up all its 1431 // buffers before playing. This is to ensure the client will 1432 // effectively get the latency it requested. 1433 if (!track->isOutputTrack()) { 1434 TrackBase::track_state state = track->mState; 1435 mLock.unlock(); 1436 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1437 mLock.lock(); 1438 // abort track was stopped/paused while we released the lock 1439 if (state != track->mState) { 1440 if (status == NO_ERROR) { 1441 mLock.unlock(); 1442 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1443 mLock.lock(); 1444 } 1445 return INVALID_OPERATION; 1446 } 1447 // abort if start is rejected by audio policy manager 1448 if (status != NO_ERROR) { 1449 return PERMISSION_DENIED; 1450 } 1451#ifdef ADD_BATTERY_DATA 1452 // to track the speaker usage 1453 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1454#endif 1455 } 1456 1457 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1458 track->mResetDone = false; 1459 track->mPresentationCompleteFrames = 0; 1460 mActiveTracks.add(track); 1461 mWakeLockUids.add(track->uid()); 1462 mActiveTracksGeneration++; 1463 mLatestActiveTrack = track; 1464 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1465 if (chain != 0) { 1466 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1467 track->sessionId()); 1468 chain->incActiveTrackCnt(); 1469 } 1470 1471 status = NO_ERROR; 1472 } 1473 1474 ALOGV("signal playback thread"); 1475 broadcast_l(); 1476 1477 return status; 1478} 1479 1480bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1481{ 1482 track->terminate(); 1483 // active tracks are removed by threadLoop() 1484 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1485 track->mState = TrackBase::STOPPED; 1486 if (!trackActive) { 1487 removeTrack_l(track); 1488 } else if (track->isFastTrack() || track->isOffloaded()) { 1489 track->mState = TrackBase::STOPPING_1; 1490 } 1491 1492 return trackActive; 1493} 1494 1495void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1496{ 1497 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1498 mTracks.remove(track); 1499 deleteTrackName_l(track->name()); 1500 // redundant as track is about to be destroyed, for dumpsys only 1501 track->mName = -1; 1502 if (track->isFastTrack()) { 1503 int index = track->mFastIndex; 1504 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1505 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1506 mFastTrackAvailMask |= 1 << index; 1507 // redundant as track is about to be destroyed, for dumpsys only 1508 track->mFastIndex = -1; 1509 } 1510 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1511 if (chain != 0) { 1512 chain->decTrackCnt(); 1513 } 1514} 1515 1516void AudioFlinger::PlaybackThread::broadcast_l() 1517{ 1518 // Thread could be blocked waiting for async 1519 // so signal it to handle state changes immediately 1520 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1521 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1522 mSignalPending = true; 1523 mWaitWorkCV.broadcast(); 1524} 1525 1526String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1527{ 1528 Mutex::Autolock _l(mLock); 1529 if (initCheck() != NO_ERROR) { 1530 return String8(); 1531 } 1532 1533 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1534 const String8 out_s8(s); 1535 free(s); 1536 return out_s8; 1537} 1538 1539// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1540void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1541 AudioSystem::OutputDescriptor desc; 1542 void *param2 = NULL; 1543 1544 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1545 param); 1546 1547 switch (event) { 1548 case AudioSystem::OUTPUT_OPENED: 1549 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1550 desc.channelMask = mChannelMask; 1551 desc.samplingRate = mSampleRate; 1552 desc.format = mFormat; 1553 desc.frameCount = mNormalFrameCount; // FIXME see 1554 // AudioFlinger::frameCount(audio_io_handle_t) 1555 desc.latency = latency(); 1556 param2 = &desc; 1557 break; 1558 1559 case AudioSystem::STREAM_CONFIG_CHANGED: 1560 param2 = ¶m; 1561 case AudioSystem::OUTPUT_CLOSED: 1562 default: 1563 break; 1564 } 1565 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1566} 1567 1568void AudioFlinger::PlaybackThread::writeCallback() 1569{ 1570 ALOG_ASSERT(mCallbackThread != 0); 1571 mCallbackThread->resetWriteBlocked(); 1572} 1573 1574void AudioFlinger::PlaybackThread::drainCallback() 1575{ 1576 ALOG_ASSERT(mCallbackThread != 0); 1577 mCallbackThread->resetDraining(); 1578} 1579 1580void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1581{ 1582 Mutex::Autolock _l(mLock); 1583 // reject out of sequence requests 1584 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1585 mWriteAckSequence &= ~1; 1586 mWaitWorkCV.signal(); 1587 } 1588} 1589 1590void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1591{ 1592 Mutex::Autolock _l(mLock); 1593 // reject out of sequence requests 1594 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1595 mDrainSequence &= ~1; 1596 mWaitWorkCV.signal(); 1597 } 1598} 1599 1600// static 1601int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1602 void *param, 1603 void *cookie) 1604{ 1605 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1606 ALOGV("asyncCallback() event %d", event); 1607 switch (event) { 1608 case STREAM_CBK_EVENT_WRITE_READY: 1609 me->writeCallback(); 1610 break; 1611 case STREAM_CBK_EVENT_DRAIN_READY: 1612 me->drainCallback(); 1613 break; 1614 default: 1615 ALOGW("asyncCallback() unknown event %d", event); 1616 break; 1617 } 1618 return 0; 1619} 1620 1621void AudioFlinger::PlaybackThread::readOutputParameters() 1622{ 1623 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1624 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1625 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1626 if (!audio_is_output_channel(mChannelMask)) { 1627 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1628 } 1629 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1630 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1631 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1632 } 1633 mChannelCount = popcount(mChannelMask); 1634 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1635 if (!audio_is_valid_format(mFormat)) { 1636 LOG_FATAL("HAL format %d not valid for output", mFormat); 1637 } 1638 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1639 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1640 mFormat); 1641 } 1642 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1643 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1644 if (mFrameCount & 15) { 1645 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1646 mFrameCount); 1647 } 1648 1649 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1650 (mOutput->stream->set_callback != NULL)) { 1651 if (mOutput->stream->set_callback(mOutput->stream, 1652 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1653 mUseAsyncWrite = true; 1654 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1655 } 1656 } 1657 1658 // Calculate size of normal mix buffer relative to the HAL output buffer size 1659 double multiplier = 1.0; 1660 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1661 kUseFastMixer == FastMixer_Dynamic)) { 1662 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1663 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1664 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1665 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1666 maxNormalFrameCount = maxNormalFrameCount & ~15; 1667 if (maxNormalFrameCount < minNormalFrameCount) { 1668 maxNormalFrameCount = minNormalFrameCount; 1669 } 1670 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1671 if (multiplier <= 1.0) { 1672 multiplier = 1.0; 1673 } else if (multiplier <= 2.0) { 1674 if (2 * mFrameCount <= maxNormalFrameCount) { 1675 multiplier = 2.0; 1676 } else { 1677 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1678 } 1679 } else { 1680 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1681 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1682 // track, but we sometimes have to do this to satisfy the maximum frame count 1683 // constraint) 1684 // FIXME this rounding up should not be done if no HAL SRC 1685 uint32_t truncMult = (uint32_t) multiplier; 1686 if ((truncMult & 1)) { 1687 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1688 ++truncMult; 1689 } 1690 } 1691 multiplier = (double) truncMult; 1692 } 1693 } 1694 mNormalFrameCount = multiplier * mFrameCount; 1695 // round up to nearest 16 frames to satisfy AudioMixer 1696 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1697 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1698 mNormalFrameCount); 1699 1700 delete[] mAllocMixBuffer; 1701 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1702 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1703 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1704 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1705 1706 // force reconfiguration of effect chains and engines to take new buffer size and audio 1707 // parameters into account 1708 // Note that mLock is not held when readOutputParameters() is called from the constructor 1709 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1710 // matter. 1711 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1712 Vector< sp<EffectChain> > effectChains = mEffectChains; 1713 for (size_t i = 0; i < effectChains.size(); i ++) { 1714 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1715 } 1716} 1717 1718 1719status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1720{ 1721 if (halFrames == NULL || dspFrames == NULL) { 1722 return BAD_VALUE; 1723 } 1724 Mutex::Autolock _l(mLock); 1725 if (initCheck() != NO_ERROR) { 1726 return INVALID_OPERATION; 1727 } 1728 size_t framesWritten = mBytesWritten / mFrameSize; 1729 *halFrames = framesWritten; 1730 1731 if (isSuspended()) { 1732 // return an estimation of rendered frames when the output is suspended 1733 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1734 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1735 return NO_ERROR; 1736 } else { 1737 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1738 } 1739} 1740 1741uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1742{ 1743 Mutex::Autolock _l(mLock); 1744 uint32_t result = 0; 1745 if (getEffectChain_l(sessionId) != 0) { 1746 result = EFFECT_SESSION; 1747 } 1748 1749 for (size_t i = 0; i < mTracks.size(); ++i) { 1750 sp<Track> track = mTracks[i]; 1751 if (sessionId == track->sessionId() && !track->isInvalid()) { 1752 result |= TRACK_SESSION; 1753 break; 1754 } 1755 } 1756 1757 return result; 1758} 1759 1760uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1761{ 1762 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1763 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1764 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1765 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1766 } 1767 for (size_t i = 0; i < mTracks.size(); i++) { 1768 sp<Track> track = mTracks[i]; 1769 if (sessionId == track->sessionId() && !track->isInvalid()) { 1770 return AudioSystem::getStrategyForStream(track->streamType()); 1771 } 1772 } 1773 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1774} 1775 1776 1777AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1778{ 1779 Mutex::Autolock _l(mLock); 1780 return mOutput; 1781} 1782 1783AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1784{ 1785 Mutex::Autolock _l(mLock); 1786 AudioStreamOut *output = mOutput; 1787 mOutput = NULL; 1788 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1789 // must push a NULL and wait for ack 1790 mOutputSink.clear(); 1791 mPipeSink.clear(); 1792 mNormalSink.clear(); 1793 return output; 1794} 1795 1796// this method must always be called either with ThreadBase mLock held or inside the thread loop 1797audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1798{ 1799 if (mOutput == NULL) { 1800 return NULL; 1801 } 1802 return &mOutput->stream->common; 1803} 1804 1805uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1806{ 1807 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1808} 1809 1810status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1811{ 1812 if (!isValidSyncEvent(event)) { 1813 return BAD_VALUE; 1814 } 1815 1816 Mutex::Autolock _l(mLock); 1817 1818 for (size_t i = 0; i < mTracks.size(); ++i) { 1819 sp<Track> track = mTracks[i]; 1820 if (event->triggerSession() == track->sessionId()) { 1821 (void) track->setSyncEvent(event); 1822 return NO_ERROR; 1823 } 1824 } 1825 1826 return NAME_NOT_FOUND; 1827} 1828 1829bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1830{ 1831 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1832} 1833 1834void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1835 const Vector< sp<Track> >& tracksToRemove) 1836{ 1837 size_t count = tracksToRemove.size(); 1838 if (count) { 1839 for (size_t i = 0 ; i < count ; i++) { 1840 const sp<Track>& track = tracksToRemove.itemAt(i); 1841 if (!track->isOutputTrack()) { 1842 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1843#ifdef ADD_BATTERY_DATA 1844 // to track the speaker usage 1845 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1846#endif 1847 if (track->isTerminated()) { 1848 AudioSystem::releaseOutput(mId); 1849 } 1850 } 1851 } 1852 } 1853} 1854 1855void AudioFlinger::PlaybackThread::checkSilentMode_l() 1856{ 1857 if (!mMasterMute) { 1858 char value[PROPERTY_VALUE_MAX]; 1859 if (property_get("ro.audio.silent", value, "0") > 0) { 1860 char *endptr; 1861 unsigned long ul = strtoul(value, &endptr, 0); 1862 if (*endptr == '\0' && ul != 0) { 1863 ALOGD("Silence is golden"); 1864 // The setprop command will not allow a property to be changed after 1865 // the first time it is set, so we don't have to worry about un-muting. 1866 setMasterMute_l(true); 1867 } 1868 } 1869 } 1870} 1871 1872// shared by MIXER and DIRECT, overridden by DUPLICATING 1873ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1874{ 1875 // FIXME rewrite to reduce number of system calls 1876 mLastWriteTime = systemTime(); 1877 mInWrite = true; 1878 ssize_t bytesWritten; 1879 1880 // If an NBAIO sink is present, use it to write the normal mixer's submix 1881 if (mNormalSink != 0) { 1882#define mBitShift 2 // FIXME 1883 size_t count = mBytesRemaining >> mBitShift; 1884 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1885 ATRACE_BEGIN("write"); 1886 // update the setpoint when AudioFlinger::mScreenState changes 1887 uint32_t screenState = AudioFlinger::mScreenState; 1888 if (screenState != mScreenState) { 1889 mScreenState = screenState; 1890 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1891 if (pipe != NULL) { 1892 pipe->setAvgFrames((mScreenState & 1) ? 1893 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1894 } 1895 } 1896 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1897 ATRACE_END(); 1898 if (framesWritten > 0) { 1899 bytesWritten = framesWritten << mBitShift; 1900 } else { 1901 bytesWritten = framesWritten; 1902 } 1903 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1904 if (status == NO_ERROR) { 1905 size_t totalFramesWritten = mNormalSink->framesWritten(); 1906 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1907 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1908 mLatchDValid = true; 1909 } 1910 } 1911 // otherwise use the HAL / AudioStreamOut directly 1912 } else { 1913 // Direct output and offload threads 1914 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1915 if (mUseAsyncWrite) { 1916 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1917 mWriteAckSequence += 2; 1918 mWriteAckSequence |= 1; 1919 ALOG_ASSERT(mCallbackThread != 0); 1920 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1921 } 1922 // FIXME We should have an implementation of timestamps for direct output threads. 1923 // They are used e.g for multichannel PCM playback over HDMI. 1924 bytesWritten = mOutput->stream->write(mOutput->stream, 1925 mMixBuffer + offset, mBytesRemaining); 1926 if (mUseAsyncWrite && 1927 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1928 // do not wait for async callback in case of error of full write 1929 mWriteAckSequence &= ~1; 1930 ALOG_ASSERT(mCallbackThread != 0); 1931 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1932 } 1933 } 1934 1935 mNumWrites++; 1936 mInWrite = false; 1937 mStandby = false; 1938 return bytesWritten; 1939} 1940 1941void AudioFlinger::PlaybackThread::threadLoop_drain() 1942{ 1943 if (mOutput->stream->drain) { 1944 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1945 if (mUseAsyncWrite) { 1946 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1947 mDrainSequence |= 1; 1948 ALOG_ASSERT(mCallbackThread != 0); 1949 mCallbackThread->setDraining(mDrainSequence); 1950 } 1951 mOutput->stream->drain(mOutput->stream, 1952 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1953 : AUDIO_DRAIN_ALL); 1954 } 1955} 1956 1957void AudioFlinger::PlaybackThread::threadLoop_exit() 1958{ 1959 // Default implementation has nothing to do 1960} 1961 1962/* 1963The derived values that are cached: 1964 - mixBufferSize from frame count * frame size 1965 - activeSleepTime from activeSleepTimeUs() 1966 - idleSleepTime from idleSleepTimeUs() 1967 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1968 - maxPeriod from frame count and sample rate (MIXER only) 1969 1970The parameters that affect these derived values are: 1971 - frame count 1972 - frame size 1973 - sample rate 1974 - device type: A2DP or not 1975 - device latency 1976 - format: PCM or not 1977 - active sleep time 1978 - idle sleep time 1979*/ 1980 1981void AudioFlinger::PlaybackThread::cacheParameters_l() 1982{ 1983 mixBufferSize = mNormalFrameCount * mFrameSize; 1984 activeSleepTime = activeSleepTimeUs(); 1985 idleSleepTime = idleSleepTimeUs(); 1986} 1987 1988void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1989{ 1990 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1991 this, streamType, mTracks.size()); 1992 Mutex::Autolock _l(mLock); 1993 1994 size_t size = mTracks.size(); 1995 for (size_t i = 0; i < size; i++) { 1996 sp<Track> t = mTracks[i]; 1997 if (t->streamType() == streamType) { 1998 t->invalidate(); 1999 } 2000 } 2001} 2002 2003status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2004{ 2005 int session = chain->sessionId(); 2006 int16_t *buffer = mMixBuffer; 2007 bool ownsBuffer = false; 2008 2009 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2010 if (session > 0) { 2011 // Only one effect chain can be present in direct output thread and it uses 2012 // the mix buffer as input 2013 if (mType != DIRECT) { 2014 size_t numSamples = mNormalFrameCount * mChannelCount; 2015 buffer = new int16_t[numSamples]; 2016 memset(buffer, 0, numSamples * sizeof(int16_t)); 2017 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2018 ownsBuffer = true; 2019 } 2020 2021 // Attach all tracks with same session ID to this chain. 2022 for (size_t i = 0; i < mTracks.size(); ++i) { 2023 sp<Track> track = mTracks[i]; 2024 if (session == track->sessionId()) { 2025 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2026 buffer); 2027 track->setMainBuffer(buffer); 2028 chain->incTrackCnt(); 2029 } 2030 } 2031 2032 // indicate all active tracks in the chain 2033 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2034 sp<Track> track = mActiveTracks[i].promote(); 2035 if (track == 0) { 2036 continue; 2037 } 2038 if (session == track->sessionId()) { 2039 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2040 chain->incActiveTrackCnt(); 2041 } 2042 } 2043 } 2044 2045 chain->setInBuffer(buffer, ownsBuffer); 2046 chain->setOutBuffer(mMixBuffer); 2047 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2048 // chains list in order to be processed last as it contains output stage effects 2049 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2050 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2051 // after track specific effects and before output stage 2052 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2053 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2054 // Effect chain for other sessions are inserted at beginning of effect 2055 // chains list to be processed before output mix effects. Relative order between other 2056 // sessions is not important 2057 size_t size = mEffectChains.size(); 2058 size_t i = 0; 2059 for (i = 0; i < size; i++) { 2060 if (mEffectChains[i]->sessionId() < session) { 2061 break; 2062 } 2063 } 2064 mEffectChains.insertAt(chain, i); 2065 checkSuspendOnAddEffectChain_l(chain); 2066 2067 return NO_ERROR; 2068} 2069 2070size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2071{ 2072 int session = chain->sessionId(); 2073 2074 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2075 2076 for (size_t i = 0; i < mEffectChains.size(); i++) { 2077 if (chain == mEffectChains[i]) { 2078 mEffectChains.removeAt(i); 2079 // detach all active tracks from the chain 2080 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2081 sp<Track> track = mActiveTracks[i].promote(); 2082 if (track == 0) { 2083 continue; 2084 } 2085 if (session == track->sessionId()) { 2086 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2087 chain.get(), session); 2088 chain->decActiveTrackCnt(); 2089 } 2090 } 2091 2092 // detach all tracks with same session ID from this chain 2093 for (size_t i = 0; i < mTracks.size(); ++i) { 2094 sp<Track> track = mTracks[i]; 2095 if (session == track->sessionId()) { 2096 track->setMainBuffer(mMixBuffer); 2097 chain->decTrackCnt(); 2098 } 2099 } 2100 break; 2101 } 2102 } 2103 return mEffectChains.size(); 2104} 2105 2106status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2107 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2108{ 2109 Mutex::Autolock _l(mLock); 2110 return attachAuxEffect_l(track, EffectId); 2111} 2112 2113status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2114 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2115{ 2116 status_t status = NO_ERROR; 2117 2118 if (EffectId == 0) { 2119 track->setAuxBuffer(0, NULL); 2120 } else { 2121 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2122 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2123 if (effect != 0) { 2124 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2125 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2126 } else { 2127 status = INVALID_OPERATION; 2128 } 2129 } else { 2130 status = BAD_VALUE; 2131 } 2132 } 2133 return status; 2134} 2135 2136void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2137{ 2138 for (size_t i = 0; i < mTracks.size(); ++i) { 2139 sp<Track> track = mTracks[i]; 2140 if (track->auxEffectId() == effectId) { 2141 attachAuxEffect_l(track, 0); 2142 } 2143 } 2144} 2145 2146bool AudioFlinger::PlaybackThread::threadLoop() 2147{ 2148 Vector< sp<Track> > tracksToRemove; 2149 2150 standbyTime = systemTime(); 2151 2152 // MIXER 2153 nsecs_t lastWarning = 0; 2154 2155 // DUPLICATING 2156 // FIXME could this be made local to while loop? 2157 writeFrames = 0; 2158 2159 int lastGeneration = 0; 2160 2161 cacheParameters_l(); 2162 sleepTime = idleSleepTime; 2163 2164 if (mType == MIXER) { 2165 sleepTimeShift = 0; 2166 } 2167 2168 CpuStats cpuStats; 2169 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2170 2171 acquireWakeLock(); 2172 2173 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2174 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2175 // and then that string will be logged at the next convenient opportunity. 2176 const char *logString = NULL; 2177 2178 checkSilentMode_l(); 2179 2180 while (!exitPending()) 2181 { 2182 cpuStats.sample(myName); 2183 2184 Vector< sp<EffectChain> > effectChains; 2185 2186 processConfigEvents(); 2187 2188 { // scope for mLock 2189 2190 Mutex::Autolock _l(mLock); 2191 2192 if (logString != NULL) { 2193 mNBLogWriter->logTimestamp(); 2194 mNBLogWriter->log(logString); 2195 logString = NULL; 2196 } 2197 2198 if (mLatchDValid) { 2199 mLatchQ = mLatchD; 2200 mLatchDValid = false; 2201 mLatchQValid = true; 2202 } 2203 2204 if (checkForNewParameters_l()) { 2205 cacheParameters_l(); 2206 } 2207 2208 saveOutputTracks(); 2209 if (mSignalPending) { 2210 // A signal was raised while we were unlocked 2211 mSignalPending = false; 2212 } else if (waitingAsyncCallback_l()) { 2213 if (exitPending()) { 2214 break; 2215 } 2216 releaseWakeLock_l(); 2217 mWakeLockUids.clear(); 2218 mActiveTracksGeneration++; 2219 ALOGV("wait async completion"); 2220 mWaitWorkCV.wait(mLock); 2221 ALOGV("async completion/wake"); 2222 acquireWakeLock_l(); 2223 standbyTime = systemTime() + standbyDelay; 2224 sleepTime = 0; 2225 2226 continue; 2227 } 2228 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2229 isSuspended()) { 2230 // put audio hardware into standby after short delay 2231 if (shouldStandby_l()) { 2232 2233 threadLoop_standby(); 2234 2235 mStandby = true; 2236 } 2237 2238 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2239 // we're about to wait, flush the binder command buffer 2240 IPCThreadState::self()->flushCommands(); 2241 2242 clearOutputTracks(); 2243 2244 if (exitPending()) { 2245 break; 2246 } 2247 2248 releaseWakeLock_l(); 2249 mWakeLockUids.clear(); 2250 mActiveTracksGeneration++; 2251 // wait until we have something to do... 2252 ALOGV("%s going to sleep", myName.string()); 2253 mWaitWorkCV.wait(mLock); 2254 ALOGV("%s waking up", myName.string()); 2255 acquireWakeLock_l(); 2256 2257 mMixerStatus = MIXER_IDLE; 2258 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2259 mBytesWritten = 0; 2260 mBytesRemaining = 0; 2261 checkSilentMode_l(); 2262 2263 standbyTime = systemTime() + standbyDelay; 2264 sleepTime = idleSleepTime; 2265 if (mType == MIXER) { 2266 sleepTimeShift = 0; 2267 } 2268 2269 continue; 2270 } 2271 } 2272 // mMixerStatusIgnoringFastTracks is also updated internally 2273 mMixerStatus = prepareTracks_l(&tracksToRemove); 2274 2275 // compare with previously applied list 2276 if (lastGeneration != mActiveTracksGeneration) { 2277 // update wakelock 2278 updateWakeLockUids_l(mWakeLockUids); 2279 lastGeneration = mActiveTracksGeneration; 2280 } 2281 2282 // prevent any changes in effect chain list and in each effect chain 2283 // during mixing and effect process as the audio buffers could be deleted 2284 // or modified if an effect is created or deleted 2285 lockEffectChains_l(effectChains); 2286 } // mLock scope ends 2287 2288 if (mBytesRemaining == 0) { 2289 mCurrentWriteLength = 0; 2290 if (mMixerStatus == MIXER_TRACKS_READY) { 2291 // threadLoop_mix() sets mCurrentWriteLength 2292 threadLoop_mix(); 2293 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2294 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2295 // threadLoop_sleepTime sets sleepTime to 0 if data 2296 // must be written to HAL 2297 threadLoop_sleepTime(); 2298 if (sleepTime == 0) { 2299 mCurrentWriteLength = mixBufferSize; 2300 } 2301 } 2302 mBytesRemaining = mCurrentWriteLength; 2303 if (isSuspended()) { 2304 sleepTime = suspendSleepTimeUs(); 2305 // simulate write to HAL when suspended 2306 mBytesWritten += mixBufferSize; 2307 mBytesRemaining = 0; 2308 } 2309 2310 // only process effects if we're going to write 2311 if (sleepTime == 0 && mType != OFFLOAD) { 2312 for (size_t i = 0; i < effectChains.size(); i ++) { 2313 effectChains[i]->process_l(); 2314 } 2315 } 2316 } 2317 // Process effect chains for offloaded thread even if no audio 2318 // was read from audio track: process only updates effect state 2319 // and thus does have to be synchronized with audio writes but may have 2320 // to be called while waiting for async write callback 2321 if (mType == OFFLOAD) { 2322 for (size_t i = 0; i < effectChains.size(); i ++) { 2323 effectChains[i]->process_l(); 2324 } 2325 } 2326 2327 // enable changes in effect chain 2328 unlockEffectChains(effectChains); 2329 2330 if (!waitingAsyncCallback()) { 2331 // sleepTime == 0 means we must write to audio hardware 2332 if (sleepTime == 0) { 2333 if (mBytesRemaining) { 2334 ssize_t ret = threadLoop_write(); 2335 if (ret < 0) { 2336 mBytesRemaining = 0; 2337 } else { 2338 mBytesWritten += ret; 2339 mBytesRemaining -= ret; 2340 } 2341 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2342 (mMixerStatus == MIXER_DRAIN_ALL)) { 2343 threadLoop_drain(); 2344 } 2345if (mType == MIXER) { 2346 // write blocked detection 2347 nsecs_t now = systemTime(); 2348 nsecs_t delta = now - mLastWriteTime; 2349 if (!mStandby && delta > maxPeriod) { 2350 mNumDelayedWrites++; 2351 if ((now - lastWarning) > kWarningThrottleNs) { 2352 ATRACE_NAME("underrun"); 2353 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2354 ns2ms(delta), mNumDelayedWrites, this); 2355 lastWarning = now; 2356 } 2357 } 2358} 2359 2360 } else { 2361 usleep(sleepTime); 2362 } 2363 } 2364 2365 // Finally let go of removed track(s), without the lock held 2366 // since we can't guarantee the destructors won't acquire that 2367 // same lock. This will also mutate and push a new fast mixer state. 2368 threadLoop_removeTracks(tracksToRemove); 2369 tracksToRemove.clear(); 2370 2371 // FIXME I don't understand the need for this here; 2372 // it was in the original code but maybe the 2373 // assignment in saveOutputTracks() makes this unnecessary? 2374 clearOutputTracks(); 2375 2376 // Effect chains will be actually deleted here if they were removed from 2377 // mEffectChains list during mixing or effects processing 2378 effectChains.clear(); 2379 2380 // FIXME Note that the above .clear() is no longer necessary since effectChains 2381 // is now local to this block, but will keep it for now (at least until merge done). 2382 } 2383 2384 threadLoop_exit(); 2385 2386 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2387 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2388 // put output stream into standby mode 2389 if (!mStandby) { 2390 mOutput->stream->common.standby(&mOutput->stream->common); 2391 } 2392 } 2393 2394 releaseWakeLock(); 2395 mWakeLockUids.clear(); 2396 mActiveTracksGeneration++; 2397 2398 ALOGV("Thread %p type %d exiting", this, mType); 2399 return false; 2400} 2401 2402// removeTracks_l() must be called with ThreadBase::mLock held 2403void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2404{ 2405 size_t count = tracksToRemove.size(); 2406 if (count) { 2407 for (size_t i=0 ; i<count ; i++) { 2408 const sp<Track>& track = tracksToRemove.itemAt(i); 2409 mActiveTracks.remove(track); 2410 mWakeLockUids.remove(track->uid()); 2411 mActiveTracksGeneration++; 2412 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2413 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2414 if (chain != 0) { 2415 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2416 track->sessionId()); 2417 chain->decActiveTrackCnt(); 2418 } 2419 if (track->isTerminated()) { 2420 removeTrack_l(track); 2421 } 2422 } 2423 } 2424 2425} 2426 2427status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2428{ 2429 if (mNormalSink != 0) { 2430 return mNormalSink->getTimestamp(timestamp); 2431 } 2432 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2433 uint64_t position64; 2434 int ret = mOutput->stream->get_presentation_position( 2435 mOutput->stream, &position64, ×tamp.mTime); 2436 if (ret == 0) { 2437 timestamp.mPosition = (uint32_t)position64; 2438 return NO_ERROR; 2439 } 2440 } 2441 return INVALID_OPERATION; 2442} 2443// ---------------------------------------------------------------------------- 2444 2445AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2446 audio_io_handle_t id, audio_devices_t device, type_t type) 2447 : PlaybackThread(audioFlinger, output, id, device, type), 2448 // mAudioMixer below 2449 // mFastMixer below 2450 mFastMixerFutex(0) 2451 // mOutputSink below 2452 // mPipeSink below 2453 // mNormalSink below 2454{ 2455 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2456 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2457 "mFrameCount=%d, mNormalFrameCount=%d", 2458 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2459 mNormalFrameCount); 2460 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2461 2462 // FIXME - Current mixer implementation only supports stereo output 2463 if (mChannelCount != FCC_2) { 2464 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2465 } 2466 2467 // create an NBAIO sink for the HAL output stream, and negotiate 2468 mOutputSink = new AudioStreamOutSink(output->stream); 2469 size_t numCounterOffers = 0; 2470 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2471 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2472 ALOG_ASSERT(index == 0); 2473 2474 // initialize fast mixer depending on configuration 2475 bool initFastMixer; 2476 switch (kUseFastMixer) { 2477 case FastMixer_Never: 2478 initFastMixer = false; 2479 break; 2480 case FastMixer_Always: 2481 initFastMixer = true; 2482 break; 2483 case FastMixer_Static: 2484 case FastMixer_Dynamic: 2485 initFastMixer = mFrameCount < mNormalFrameCount; 2486 break; 2487 } 2488 if (initFastMixer) { 2489 2490 // create a MonoPipe to connect our submix to FastMixer 2491 NBAIO_Format format = mOutputSink->format(); 2492 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2493 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2494 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2495 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2496 const NBAIO_Format offers[1] = {format}; 2497 size_t numCounterOffers = 0; 2498 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2499 ALOG_ASSERT(index == 0); 2500 monoPipe->setAvgFrames((mScreenState & 1) ? 2501 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2502 mPipeSink = monoPipe; 2503 2504#ifdef TEE_SINK 2505 if (mTeeSinkOutputEnabled) { 2506 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2507 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2508 numCounterOffers = 0; 2509 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2510 ALOG_ASSERT(index == 0); 2511 mTeeSink = teeSink; 2512 PipeReader *teeSource = new PipeReader(*teeSink); 2513 numCounterOffers = 0; 2514 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2515 ALOG_ASSERT(index == 0); 2516 mTeeSource = teeSource; 2517 } 2518#endif 2519 2520 // create fast mixer and configure it initially with just one fast track for our submix 2521 mFastMixer = new FastMixer(); 2522 FastMixerStateQueue *sq = mFastMixer->sq(); 2523#ifdef STATE_QUEUE_DUMP 2524 sq->setObserverDump(&mStateQueueObserverDump); 2525 sq->setMutatorDump(&mStateQueueMutatorDump); 2526#endif 2527 FastMixerState *state = sq->begin(); 2528 FastTrack *fastTrack = &state->mFastTracks[0]; 2529 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2530 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2531 fastTrack->mVolumeProvider = NULL; 2532 fastTrack->mGeneration++; 2533 state->mFastTracksGen++; 2534 state->mTrackMask = 1; 2535 // fast mixer will use the HAL output sink 2536 state->mOutputSink = mOutputSink.get(); 2537 state->mOutputSinkGen++; 2538 state->mFrameCount = mFrameCount; 2539 state->mCommand = FastMixerState::COLD_IDLE; 2540 // already done in constructor initialization list 2541 //mFastMixerFutex = 0; 2542 state->mColdFutexAddr = &mFastMixerFutex; 2543 state->mColdGen++; 2544 state->mDumpState = &mFastMixerDumpState; 2545#ifdef TEE_SINK 2546 state->mTeeSink = mTeeSink.get(); 2547#endif 2548 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2549 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2550 sq->end(); 2551 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2552 2553 // start the fast mixer 2554 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2555 pid_t tid = mFastMixer->getTid(); 2556 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2557 if (err != 0) { 2558 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2559 kPriorityFastMixer, getpid_cached, tid, err); 2560 } 2561 2562#ifdef AUDIO_WATCHDOG 2563 // create and start the watchdog 2564 mAudioWatchdog = new AudioWatchdog(); 2565 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2566 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2567 tid = mAudioWatchdog->getTid(); 2568 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2569 if (err != 0) { 2570 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2571 kPriorityFastMixer, getpid_cached, tid, err); 2572 } 2573#endif 2574 2575 } else { 2576 mFastMixer = NULL; 2577 } 2578 2579 switch (kUseFastMixer) { 2580 case FastMixer_Never: 2581 case FastMixer_Dynamic: 2582 mNormalSink = mOutputSink; 2583 break; 2584 case FastMixer_Always: 2585 mNormalSink = mPipeSink; 2586 break; 2587 case FastMixer_Static: 2588 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2589 break; 2590 } 2591} 2592 2593AudioFlinger::MixerThread::~MixerThread() 2594{ 2595 if (mFastMixer != NULL) { 2596 FastMixerStateQueue *sq = mFastMixer->sq(); 2597 FastMixerState *state = sq->begin(); 2598 if (state->mCommand == FastMixerState::COLD_IDLE) { 2599 int32_t old = android_atomic_inc(&mFastMixerFutex); 2600 if (old == -1) { 2601 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2602 } 2603 } 2604 state->mCommand = FastMixerState::EXIT; 2605 sq->end(); 2606 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2607 mFastMixer->join(); 2608 // Though the fast mixer thread has exited, it's state queue is still valid. 2609 // We'll use that extract the final state which contains one remaining fast track 2610 // corresponding to our sub-mix. 2611 state = sq->begin(); 2612 ALOG_ASSERT(state->mTrackMask == 1); 2613 FastTrack *fastTrack = &state->mFastTracks[0]; 2614 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2615 delete fastTrack->mBufferProvider; 2616 sq->end(false /*didModify*/); 2617 delete mFastMixer; 2618#ifdef AUDIO_WATCHDOG 2619 if (mAudioWatchdog != 0) { 2620 mAudioWatchdog->requestExit(); 2621 mAudioWatchdog->requestExitAndWait(); 2622 mAudioWatchdog.clear(); 2623 } 2624#endif 2625 } 2626 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2627 delete mAudioMixer; 2628} 2629 2630 2631uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2632{ 2633 if (mFastMixer != NULL) { 2634 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2635 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2636 } 2637 return latency; 2638} 2639 2640 2641void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2642{ 2643 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2644} 2645 2646ssize_t AudioFlinger::MixerThread::threadLoop_write() 2647{ 2648 // FIXME we should only do one push per cycle; confirm this is true 2649 // Start the fast mixer if it's not already running 2650 if (mFastMixer != NULL) { 2651 FastMixerStateQueue *sq = mFastMixer->sq(); 2652 FastMixerState *state = sq->begin(); 2653 if (state->mCommand != FastMixerState::MIX_WRITE && 2654 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2655 if (state->mCommand == FastMixerState::COLD_IDLE) { 2656 int32_t old = android_atomic_inc(&mFastMixerFutex); 2657 if (old == -1) { 2658 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2659 } 2660#ifdef AUDIO_WATCHDOG 2661 if (mAudioWatchdog != 0) { 2662 mAudioWatchdog->resume(); 2663 } 2664#endif 2665 } 2666 state->mCommand = FastMixerState::MIX_WRITE; 2667 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2668 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2669 sq->end(); 2670 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2671 if (kUseFastMixer == FastMixer_Dynamic) { 2672 mNormalSink = mPipeSink; 2673 } 2674 } else { 2675 sq->end(false /*didModify*/); 2676 } 2677 } 2678 return PlaybackThread::threadLoop_write(); 2679} 2680 2681void AudioFlinger::MixerThread::threadLoop_standby() 2682{ 2683 // Idle the fast mixer if it's currently running 2684 if (mFastMixer != NULL) { 2685 FastMixerStateQueue *sq = mFastMixer->sq(); 2686 FastMixerState *state = sq->begin(); 2687 if (!(state->mCommand & FastMixerState::IDLE)) { 2688 state->mCommand = FastMixerState::COLD_IDLE; 2689 state->mColdFutexAddr = &mFastMixerFutex; 2690 state->mColdGen++; 2691 mFastMixerFutex = 0; 2692 sq->end(); 2693 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2694 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2695 if (kUseFastMixer == FastMixer_Dynamic) { 2696 mNormalSink = mOutputSink; 2697 } 2698#ifdef AUDIO_WATCHDOG 2699 if (mAudioWatchdog != 0) { 2700 mAudioWatchdog->pause(); 2701 } 2702#endif 2703 } else { 2704 sq->end(false /*didModify*/); 2705 } 2706 } 2707 PlaybackThread::threadLoop_standby(); 2708} 2709 2710// Empty implementation for standard mixer 2711// Overridden for offloaded playback 2712void AudioFlinger::PlaybackThread::flushOutput_l() 2713{ 2714} 2715 2716bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2717{ 2718 return false; 2719} 2720 2721bool AudioFlinger::PlaybackThread::shouldStandby_l() 2722{ 2723 return !mStandby; 2724} 2725 2726bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2727{ 2728 Mutex::Autolock _l(mLock); 2729 return waitingAsyncCallback_l(); 2730} 2731 2732// shared by MIXER and DIRECT, overridden by DUPLICATING 2733void AudioFlinger::PlaybackThread::threadLoop_standby() 2734{ 2735 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2736 mOutput->stream->common.standby(&mOutput->stream->common); 2737 if (mUseAsyncWrite != 0) { 2738 // discard any pending drain or write ack by incrementing sequence 2739 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2740 mDrainSequence = (mDrainSequence + 2) & ~1; 2741 ALOG_ASSERT(mCallbackThread != 0); 2742 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2743 mCallbackThread->setDraining(mDrainSequence); 2744 } 2745} 2746 2747void AudioFlinger::MixerThread::threadLoop_mix() 2748{ 2749 // obtain the presentation timestamp of the next output buffer 2750 int64_t pts; 2751 status_t status = INVALID_OPERATION; 2752 2753 if (mNormalSink != 0) { 2754 status = mNormalSink->getNextWriteTimestamp(&pts); 2755 } else { 2756 status = mOutputSink->getNextWriteTimestamp(&pts); 2757 } 2758 2759 if (status != NO_ERROR) { 2760 pts = AudioBufferProvider::kInvalidPTS; 2761 } 2762 2763 // mix buffers... 2764 mAudioMixer->process(pts); 2765 mCurrentWriteLength = mixBufferSize; 2766 // increase sleep time progressively when application underrun condition clears. 2767 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2768 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2769 // such that we would underrun the audio HAL. 2770 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2771 sleepTimeShift--; 2772 } 2773 sleepTime = 0; 2774 standbyTime = systemTime() + standbyDelay; 2775 //TODO: delay standby when effects have a tail 2776} 2777 2778void AudioFlinger::MixerThread::threadLoop_sleepTime() 2779{ 2780 // If no tracks are ready, sleep once for the duration of an output 2781 // buffer size, then write 0s to the output 2782 if (sleepTime == 0) { 2783 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2784 sleepTime = activeSleepTime >> sleepTimeShift; 2785 if (sleepTime < kMinThreadSleepTimeUs) { 2786 sleepTime = kMinThreadSleepTimeUs; 2787 } 2788 // reduce sleep time in case of consecutive application underruns to avoid 2789 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2790 // duration we would end up writing less data than needed by the audio HAL if 2791 // the condition persists. 2792 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2793 sleepTimeShift++; 2794 } 2795 } else { 2796 sleepTime = idleSleepTime; 2797 } 2798 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2799 memset (mMixBuffer, 0, mixBufferSize); 2800 sleepTime = 0; 2801 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2802 "anticipated start"); 2803 } 2804 // TODO add standby time extension fct of effect tail 2805} 2806 2807// prepareTracks_l() must be called with ThreadBase::mLock held 2808AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2809 Vector< sp<Track> > *tracksToRemove) 2810{ 2811 2812 mixer_state mixerStatus = MIXER_IDLE; 2813 // find out which tracks need to be processed 2814 size_t count = mActiveTracks.size(); 2815 size_t mixedTracks = 0; 2816 size_t tracksWithEffect = 0; 2817 // counts only _active_ fast tracks 2818 size_t fastTracks = 0; 2819 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2820 2821 float masterVolume = mMasterVolume; 2822 bool masterMute = mMasterMute; 2823 2824 if (masterMute) { 2825 masterVolume = 0; 2826 } 2827 // Delegate master volume control to effect in output mix effect chain if needed 2828 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2829 if (chain != 0) { 2830 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2831 chain->setVolume_l(&v, &v); 2832 masterVolume = (float)((v + (1 << 23)) >> 24); 2833 chain.clear(); 2834 } 2835 2836 // prepare a new state to push 2837 FastMixerStateQueue *sq = NULL; 2838 FastMixerState *state = NULL; 2839 bool didModify = false; 2840 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2841 if (mFastMixer != NULL) { 2842 sq = mFastMixer->sq(); 2843 state = sq->begin(); 2844 } 2845 2846 for (size_t i=0 ; i<count ; i++) { 2847 const sp<Track> t = mActiveTracks[i].promote(); 2848 if (t == 0) { 2849 continue; 2850 } 2851 2852 // this const just means the local variable doesn't change 2853 Track* const track = t.get(); 2854 2855 // process fast tracks 2856 if (track->isFastTrack()) { 2857 2858 // It's theoretically possible (though unlikely) for a fast track to be created 2859 // and then removed within the same normal mix cycle. This is not a problem, as 2860 // the track never becomes active so it's fast mixer slot is never touched. 2861 // The converse, of removing an (active) track and then creating a new track 2862 // at the identical fast mixer slot within the same normal mix cycle, 2863 // is impossible because the slot isn't marked available until the end of each cycle. 2864 int j = track->mFastIndex; 2865 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2866 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2867 FastTrack *fastTrack = &state->mFastTracks[j]; 2868 2869 // Determine whether the track is currently in underrun condition, 2870 // and whether it had a recent underrun. 2871 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2872 FastTrackUnderruns underruns = ftDump->mUnderruns; 2873 uint32_t recentFull = (underruns.mBitFields.mFull - 2874 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2875 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2876 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2877 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2878 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2879 uint32_t recentUnderruns = recentPartial + recentEmpty; 2880 track->mObservedUnderruns = underruns; 2881 // don't count underruns that occur while stopping or pausing 2882 // or stopped which can occur when flush() is called while active 2883 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2884 recentUnderruns > 0) { 2885 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2886 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2887 } 2888 2889 // This is similar to the state machine for normal tracks, 2890 // with a few modifications for fast tracks. 2891 bool isActive = true; 2892 switch (track->mState) { 2893 case TrackBase::STOPPING_1: 2894 // track stays active in STOPPING_1 state until first underrun 2895 if (recentUnderruns > 0 || track->isTerminated()) { 2896 track->mState = TrackBase::STOPPING_2; 2897 } 2898 break; 2899 case TrackBase::PAUSING: 2900 // ramp down is not yet implemented 2901 track->setPaused(); 2902 break; 2903 case TrackBase::RESUMING: 2904 // ramp up is not yet implemented 2905 track->mState = TrackBase::ACTIVE; 2906 break; 2907 case TrackBase::ACTIVE: 2908 if (recentFull > 0 || recentPartial > 0) { 2909 // track has provided at least some frames recently: reset retry count 2910 track->mRetryCount = kMaxTrackRetries; 2911 } 2912 if (recentUnderruns == 0) { 2913 // no recent underruns: stay active 2914 break; 2915 } 2916 // there has recently been an underrun of some kind 2917 if (track->sharedBuffer() == 0) { 2918 // were any of the recent underruns "empty" (no frames available)? 2919 if (recentEmpty == 0) { 2920 // no, then ignore the partial underruns as they are allowed indefinitely 2921 break; 2922 } 2923 // there has recently been an "empty" underrun: decrement the retry counter 2924 if (--(track->mRetryCount) > 0) { 2925 break; 2926 } 2927 // indicate to client process that the track was disabled because of underrun; 2928 // it will then automatically call start() when data is available 2929 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2930 // remove from active list, but state remains ACTIVE [confusing but true] 2931 isActive = false; 2932 break; 2933 } 2934 // fall through 2935 case TrackBase::STOPPING_2: 2936 case TrackBase::PAUSED: 2937 case TrackBase::STOPPED: 2938 case TrackBase::FLUSHED: // flush() while active 2939 // Check for presentation complete if track is inactive 2940 // We have consumed all the buffers of this track. 2941 // This would be incomplete if we auto-paused on underrun 2942 { 2943 size_t audioHALFrames = 2944 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2945 size_t framesWritten = mBytesWritten / mFrameSize; 2946 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2947 // track stays in active list until presentation is complete 2948 break; 2949 } 2950 } 2951 if (track->isStopping_2()) { 2952 track->mState = TrackBase::STOPPED; 2953 } 2954 if (track->isStopped()) { 2955 // Can't reset directly, as fast mixer is still polling this track 2956 // track->reset(); 2957 // So instead mark this track as needing to be reset after push with ack 2958 resetMask |= 1 << i; 2959 } 2960 isActive = false; 2961 break; 2962 case TrackBase::IDLE: 2963 default: 2964 LOG_FATAL("unexpected track state %d", track->mState); 2965 } 2966 2967 if (isActive) { 2968 // was it previously inactive? 2969 if (!(state->mTrackMask & (1 << j))) { 2970 ExtendedAudioBufferProvider *eabp = track; 2971 VolumeProvider *vp = track; 2972 fastTrack->mBufferProvider = eabp; 2973 fastTrack->mVolumeProvider = vp; 2974 fastTrack->mSampleRate = track->mSampleRate; 2975 fastTrack->mChannelMask = track->mChannelMask; 2976 fastTrack->mGeneration++; 2977 state->mTrackMask |= 1 << j; 2978 didModify = true; 2979 // no acknowledgement required for newly active tracks 2980 } 2981 // cache the combined master volume and stream type volume for fast mixer; this 2982 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2983 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2984 ++fastTracks; 2985 } else { 2986 // was it previously active? 2987 if (state->mTrackMask & (1 << j)) { 2988 fastTrack->mBufferProvider = NULL; 2989 fastTrack->mGeneration++; 2990 state->mTrackMask &= ~(1 << j); 2991 didModify = true; 2992 // If any fast tracks were removed, we must wait for acknowledgement 2993 // because we're about to decrement the last sp<> on those tracks. 2994 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2995 } else { 2996 LOG_FATAL("fast track %d should have been active", j); 2997 } 2998 tracksToRemove->add(track); 2999 // Avoids a misleading display in dumpsys 3000 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3001 } 3002 continue; 3003 } 3004 3005 { // local variable scope to avoid goto warning 3006 3007 audio_track_cblk_t* cblk = track->cblk(); 3008 3009 // The first time a track is added we wait 3010 // for all its buffers to be filled before processing it 3011 int name = track->name(); 3012 // make sure that we have enough frames to mix one full buffer. 3013 // enforce this condition only once to enable draining the buffer in case the client 3014 // app does not call stop() and relies on underrun to stop: 3015 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3016 // during last round 3017 size_t desiredFrames; 3018 uint32_t sr = track->sampleRate(); 3019 if (sr == mSampleRate) { 3020 desiredFrames = mNormalFrameCount; 3021 } else { 3022 // +1 for rounding and +1 for additional sample needed for interpolation 3023 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3024 // add frames already consumed but not yet released by the resampler 3025 // because cblk->framesReady() will include these frames 3026 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3027 // the minimum track buffer size is normally twice the number of frames necessary 3028 // to fill one buffer and the resampler should not leave more than one buffer worth 3029 // of unreleased frames after each pass, but just in case... 3030 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3031 } 3032 uint32_t minFrames = 1; 3033 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3034 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3035 minFrames = desiredFrames; 3036 } 3037 3038 size_t framesReady = track->framesReady(); 3039 if ((framesReady >= minFrames) && track->isReady() && 3040 !track->isPaused() && !track->isTerminated()) 3041 { 3042 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3043 3044 mixedTracks++; 3045 3046 // track->mainBuffer() != mMixBuffer means there is an effect chain 3047 // connected to the track 3048 chain.clear(); 3049 if (track->mainBuffer() != mMixBuffer) { 3050 chain = getEffectChain_l(track->sessionId()); 3051 // Delegate volume control to effect in track effect chain if needed 3052 if (chain != 0) { 3053 tracksWithEffect++; 3054 } else { 3055 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3056 "session %d", 3057 name, track->sessionId()); 3058 } 3059 } 3060 3061 3062 int param = AudioMixer::VOLUME; 3063 if (track->mFillingUpStatus == Track::FS_FILLED) { 3064 // no ramp for the first volume setting 3065 track->mFillingUpStatus = Track::FS_ACTIVE; 3066 if (track->mState == TrackBase::RESUMING) { 3067 track->mState = TrackBase::ACTIVE; 3068 param = AudioMixer::RAMP_VOLUME; 3069 } 3070 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3071 // FIXME should not make a decision based on mServer 3072 } else if (cblk->mServer != 0) { 3073 // If the track is stopped before the first frame was mixed, 3074 // do not apply ramp 3075 param = AudioMixer::RAMP_VOLUME; 3076 } 3077 3078 // compute volume for this track 3079 uint32_t vl, vr, va; 3080 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3081 vl = vr = va = 0; 3082 if (track->isPausing()) { 3083 track->setPaused(); 3084 } 3085 } else { 3086 3087 // read original volumes with volume control 3088 float typeVolume = mStreamTypes[track->streamType()].volume; 3089 float v = masterVolume * typeVolume; 3090 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3091 uint32_t vlr = proxy->getVolumeLR(); 3092 vl = vlr & 0xFFFF; 3093 vr = vlr >> 16; 3094 // track volumes come from shared memory, so can't be trusted and must be clamped 3095 if (vl > MAX_GAIN_INT) { 3096 ALOGV("Track left volume out of range: %04X", vl); 3097 vl = MAX_GAIN_INT; 3098 } 3099 if (vr > MAX_GAIN_INT) { 3100 ALOGV("Track right volume out of range: %04X", vr); 3101 vr = MAX_GAIN_INT; 3102 } 3103 // now apply the master volume and stream type volume 3104 vl = (uint32_t)(v * vl) << 12; 3105 vr = (uint32_t)(v * vr) << 12; 3106 // assuming master volume and stream type volume each go up to 1.0, 3107 // vl and vr are now in 8.24 format 3108 3109 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3110 // send level comes from shared memory and so may be corrupt 3111 if (sendLevel > MAX_GAIN_INT) { 3112 ALOGV("Track send level out of range: %04X", sendLevel); 3113 sendLevel = MAX_GAIN_INT; 3114 } 3115 va = (uint32_t)(v * sendLevel); 3116 } 3117 3118 // Delegate volume control to effect in track effect chain if needed 3119 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3120 // Do not ramp volume if volume is controlled by effect 3121 param = AudioMixer::VOLUME; 3122 track->mHasVolumeController = true; 3123 } else { 3124 // force no volume ramp when volume controller was just disabled or removed 3125 // from effect chain to avoid volume spike 3126 if (track->mHasVolumeController) { 3127 param = AudioMixer::VOLUME; 3128 } 3129 track->mHasVolumeController = false; 3130 } 3131 3132 // Convert volumes from 8.24 to 4.12 format 3133 // This additional clamping is needed in case chain->setVolume_l() overshot 3134 vl = (vl + (1 << 11)) >> 12; 3135 if (vl > MAX_GAIN_INT) { 3136 vl = MAX_GAIN_INT; 3137 } 3138 vr = (vr + (1 << 11)) >> 12; 3139 if (vr > MAX_GAIN_INT) { 3140 vr = MAX_GAIN_INT; 3141 } 3142 3143 if (va > MAX_GAIN_INT) { 3144 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3145 } 3146 3147 // XXX: these things DON'T need to be done each time 3148 mAudioMixer->setBufferProvider(name, track); 3149 mAudioMixer->enable(name); 3150 3151 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3152 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3153 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3154 mAudioMixer->setParameter( 3155 name, 3156 AudioMixer::TRACK, 3157 AudioMixer::FORMAT, (void *)track->format()); 3158 mAudioMixer->setParameter( 3159 name, 3160 AudioMixer::TRACK, 3161 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3162 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3163 uint32_t maxSampleRate = mSampleRate * 2; 3164 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3165 if (reqSampleRate == 0) { 3166 reqSampleRate = mSampleRate; 3167 } else if (reqSampleRate > maxSampleRate) { 3168 reqSampleRate = maxSampleRate; 3169 } 3170 mAudioMixer->setParameter( 3171 name, 3172 AudioMixer::RESAMPLE, 3173 AudioMixer::SAMPLE_RATE, 3174 (void *)reqSampleRate); 3175 mAudioMixer->setParameter( 3176 name, 3177 AudioMixer::TRACK, 3178 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3179 mAudioMixer->setParameter( 3180 name, 3181 AudioMixer::TRACK, 3182 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3183 3184 // reset retry count 3185 track->mRetryCount = kMaxTrackRetries; 3186 3187 // If one track is ready, set the mixer ready if: 3188 // - the mixer was not ready during previous round OR 3189 // - no other track is not ready 3190 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3191 mixerStatus != MIXER_TRACKS_ENABLED) { 3192 mixerStatus = MIXER_TRACKS_READY; 3193 } 3194 } else { 3195 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3196 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3197 } 3198 // clear effect chain input buffer if an active track underruns to avoid sending 3199 // previous audio buffer again to effects 3200 chain = getEffectChain_l(track->sessionId()); 3201 if (chain != 0) { 3202 chain->clearInputBuffer(); 3203 } 3204 3205 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3206 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3207 track->isStopped() || track->isPaused()) { 3208 // We have consumed all the buffers of this track. 3209 // Remove it from the list of active tracks. 3210 // TODO: use actual buffer filling status instead of latency when available from 3211 // audio HAL 3212 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3213 size_t framesWritten = mBytesWritten / mFrameSize; 3214 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3215 if (track->isStopped()) { 3216 track->reset(); 3217 } 3218 tracksToRemove->add(track); 3219 } 3220 } else { 3221 // No buffers for this track. Give it a few chances to 3222 // fill a buffer, then remove it from active list. 3223 if (--(track->mRetryCount) <= 0) { 3224 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3225 tracksToRemove->add(track); 3226 // indicate to client process that the track was disabled because of underrun; 3227 // it will then automatically call start() when data is available 3228 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3229 // If one track is not ready, mark the mixer also not ready if: 3230 // - the mixer was ready during previous round OR 3231 // - no other track is ready 3232 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3233 mixerStatus != MIXER_TRACKS_READY) { 3234 mixerStatus = MIXER_TRACKS_ENABLED; 3235 } 3236 } 3237 mAudioMixer->disable(name); 3238 } 3239 3240 } // local variable scope to avoid goto warning 3241track_is_ready: ; 3242 3243 } 3244 3245 // Push the new FastMixer state if necessary 3246 bool pauseAudioWatchdog = false; 3247 if (didModify) { 3248 state->mFastTracksGen++; 3249 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3250 if (kUseFastMixer == FastMixer_Dynamic && 3251 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3252 state->mCommand = FastMixerState::COLD_IDLE; 3253 state->mColdFutexAddr = &mFastMixerFutex; 3254 state->mColdGen++; 3255 mFastMixerFutex = 0; 3256 if (kUseFastMixer == FastMixer_Dynamic) { 3257 mNormalSink = mOutputSink; 3258 } 3259 // If we go into cold idle, need to wait for acknowledgement 3260 // so that fast mixer stops doing I/O. 3261 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3262 pauseAudioWatchdog = true; 3263 } 3264 } 3265 if (sq != NULL) { 3266 sq->end(didModify); 3267 sq->push(block); 3268 } 3269#ifdef AUDIO_WATCHDOG 3270 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3271 mAudioWatchdog->pause(); 3272 } 3273#endif 3274 3275 // Now perform the deferred reset on fast tracks that have stopped 3276 while (resetMask != 0) { 3277 size_t i = __builtin_ctz(resetMask); 3278 ALOG_ASSERT(i < count); 3279 resetMask &= ~(1 << i); 3280 sp<Track> t = mActiveTracks[i].promote(); 3281 if (t == 0) { 3282 continue; 3283 } 3284 Track* track = t.get(); 3285 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3286 track->reset(); 3287 } 3288 3289 // remove all the tracks that need to be... 3290 removeTracks_l(*tracksToRemove); 3291 3292 // mix buffer must be cleared if all tracks are connected to an 3293 // effect chain as in this case the mixer will not write to 3294 // mix buffer and track effects will accumulate into it 3295 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3296 (mixedTracks == 0 && fastTracks > 0))) { 3297 // FIXME as a performance optimization, should remember previous zero status 3298 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3299 } 3300 3301 // if any fast tracks, then status is ready 3302 mMixerStatusIgnoringFastTracks = mixerStatus; 3303 if (fastTracks > 0) { 3304 mixerStatus = MIXER_TRACKS_READY; 3305 } 3306 return mixerStatus; 3307} 3308 3309// getTrackName_l() must be called with ThreadBase::mLock held 3310int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3311{ 3312 return mAudioMixer->getTrackName(channelMask, sessionId); 3313} 3314 3315// deleteTrackName_l() must be called with ThreadBase::mLock held 3316void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3317{ 3318 ALOGV("remove track (%d) and delete from mixer", name); 3319 mAudioMixer->deleteTrackName(name); 3320} 3321 3322// checkForNewParameters_l() must be called with ThreadBase::mLock held 3323bool AudioFlinger::MixerThread::checkForNewParameters_l() 3324{ 3325 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3326 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3327 bool reconfig = false; 3328 3329 while (!mNewParameters.isEmpty()) { 3330 3331 if (mFastMixer != NULL) { 3332 FastMixerStateQueue *sq = mFastMixer->sq(); 3333 FastMixerState *state = sq->begin(); 3334 if (!(state->mCommand & FastMixerState::IDLE)) { 3335 previousCommand = state->mCommand; 3336 state->mCommand = FastMixerState::HOT_IDLE; 3337 sq->end(); 3338 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3339 } else { 3340 sq->end(false /*didModify*/); 3341 } 3342 } 3343 3344 status_t status = NO_ERROR; 3345 String8 keyValuePair = mNewParameters[0]; 3346 AudioParameter param = AudioParameter(keyValuePair); 3347 int value; 3348 3349 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3350 reconfig = true; 3351 } 3352 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3353 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3354 status = BAD_VALUE; 3355 } else { 3356 reconfig = true; 3357 } 3358 } 3359 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3360 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3361 status = BAD_VALUE; 3362 } else { 3363 reconfig = true; 3364 } 3365 } 3366 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3367 // do not accept frame count changes if tracks are open as the track buffer 3368 // size depends on frame count and correct behavior would not be guaranteed 3369 // if frame count is changed after track creation 3370 if (!mTracks.isEmpty()) { 3371 status = INVALID_OPERATION; 3372 } else { 3373 reconfig = true; 3374 } 3375 } 3376 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3377#ifdef ADD_BATTERY_DATA 3378 // when changing the audio output device, call addBatteryData to notify 3379 // the change 3380 if (mOutDevice != value) { 3381 uint32_t params = 0; 3382 // check whether speaker is on 3383 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3384 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3385 } 3386 3387 audio_devices_t deviceWithoutSpeaker 3388 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3389 // check if any other device (except speaker) is on 3390 if (value & deviceWithoutSpeaker ) { 3391 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3392 } 3393 3394 if (params != 0) { 3395 addBatteryData(params); 3396 } 3397 } 3398#endif 3399 3400 // forward device change to effects that have requested to be 3401 // aware of attached audio device. 3402 if (value != AUDIO_DEVICE_NONE) { 3403 mOutDevice = value; 3404 for (size_t i = 0; i < mEffectChains.size(); i++) { 3405 mEffectChains[i]->setDevice_l(mOutDevice); 3406 } 3407 } 3408 } 3409 3410 if (status == NO_ERROR) { 3411 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3412 keyValuePair.string()); 3413 if (!mStandby && status == INVALID_OPERATION) { 3414 mOutput->stream->common.standby(&mOutput->stream->common); 3415 mStandby = true; 3416 mBytesWritten = 0; 3417 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3418 keyValuePair.string()); 3419 } 3420 if (status == NO_ERROR && reconfig) { 3421 readOutputParameters(); 3422 delete mAudioMixer; 3423 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3424 for (size_t i = 0; i < mTracks.size() ; i++) { 3425 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3426 if (name < 0) { 3427 break; 3428 } 3429 mTracks[i]->mName = name; 3430 } 3431 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3432 } 3433 } 3434 3435 mNewParameters.removeAt(0); 3436 3437 mParamStatus = status; 3438 mParamCond.signal(); 3439 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3440 // already timed out waiting for the status and will never signal the condition. 3441 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3442 } 3443 3444 if (!(previousCommand & FastMixerState::IDLE)) { 3445 ALOG_ASSERT(mFastMixer != NULL); 3446 FastMixerStateQueue *sq = mFastMixer->sq(); 3447 FastMixerState *state = sq->begin(); 3448 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3449 state->mCommand = previousCommand; 3450 sq->end(); 3451 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3452 } 3453 3454 return reconfig; 3455} 3456 3457 3458void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3459{ 3460 const size_t SIZE = 256; 3461 char buffer[SIZE]; 3462 String8 result; 3463 3464 PlaybackThread::dumpInternals(fd, args); 3465 3466 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3467 result.append(buffer); 3468 write(fd, result.string(), result.size()); 3469 3470 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3471 const FastMixerDumpState copy(mFastMixerDumpState); 3472 copy.dump(fd); 3473 3474#ifdef STATE_QUEUE_DUMP 3475 // Similar for state queue 3476 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3477 observerCopy.dump(fd); 3478 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3479 mutatorCopy.dump(fd); 3480#endif 3481 3482#ifdef TEE_SINK 3483 // Write the tee output to a .wav file 3484 dumpTee(fd, mTeeSource, mId); 3485#endif 3486 3487#ifdef AUDIO_WATCHDOG 3488 if (mAudioWatchdog != 0) { 3489 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3490 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3491 wdCopy.dump(fd); 3492 } 3493#endif 3494} 3495 3496uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3497{ 3498 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3499} 3500 3501uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3502{ 3503 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3504} 3505 3506void AudioFlinger::MixerThread::cacheParameters_l() 3507{ 3508 PlaybackThread::cacheParameters_l(); 3509 3510 // FIXME: Relaxed timing because of a certain device that can't meet latency 3511 // Should be reduced to 2x after the vendor fixes the driver issue 3512 // increase threshold again due to low power audio mode. The way this warning 3513 // threshold is calculated and its usefulness should be reconsidered anyway. 3514 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3515} 3516 3517// ---------------------------------------------------------------------------- 3518 3519AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3520 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3521 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3522 // mLeftVolFloat, mRightVolFloat 3523{ 3524} 3525 3526AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3527 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3528 ThreadBase::type_t type) 3529 : PlaybackThread(audioFlinger, output, id, device, type) 3530 // mLeftVolFloat, mRightVolFloat 3531{ 3532} 3533 3534AudioFlinger::DirectOutputThread::~DirectOutputThread() 3535{ 3536} 3537 3538void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3539{ 3540 audio_track_cblk_t* cblk = track->cblk(); 3541 float left, right; 3542 3543 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3544 left = right = 0; 3545 } else { 3546 float typeVolume = mStreamTypes[track->streamType()].volume; 3547 float v = mMasterVolume * typeVolume; 3548 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3549 uint32_t vlr = proxy->getVolumeLR(); 3550 float v_clamped = v * (vlr & 0xFFFF); 3551 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3552 left = v_clamped/MAX_GAIN; 3553 v_clamped = v * (vlr >> 16); 3554 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3555 right = v_clamped/MAX_GAIN; 3556 } 3557 3558 if (lastTrack) { 3559 if (left != mLeftVolFloat || right != mRightVolFloat) { 3560 mLeftVolFloat = left; 3561 mRightVolFloat = right; 3562 3563 // Convert volumes from float to 8.24 3564 uint32_t vl = (uint32_t)(left * (1 << 24)); 3565 uint32_t vr = (uint32_t)(right * (1 << 24)); 3566 3567 // Delegate volume control to effect in track effect chain if needed 3568 // only one effect chain can be present on DirectOutputThread, so if 3569 // there is one, the track is connected to it 3570 if (!mEffectChains.isEmpty()) { 3571 mEffectChains[0]->setVolume_l(&vl, &vr); 3572 left = (float)vl / (1 << 24); 3573 right = (float)vr / (1 << 24); 3574 } 3575 if (mOutput->stream->set_volume) { 3576 mOutput->stream->set_volume(mOutput->stream, left, right); 3577 } 3578 } 3579 } 3580} 3581 3582 3583AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3584 Vector< sp<Track> > *tracksToRemove 3585) 3586{ 3587 size_t count = mActiveTracks.size(); 3588 mixer_state mixerStatus = MIXER_IDLE; 3589 3590 // find out which tracks need to be processed 3591 for (size_t i = 0; i < count; i++) { 3592 sp<Track> t = mActiveTracks[i].promote(); 3593 // The track died recently 3594 if (t == 0) { 3595 continue; 3596 } 3597 3598 Track* const track = t.get(); 3599 audio_track_cblk_t* cblk = track->cblk(); 3600 // Only consider last track started for volume and mixer state control. 3601 // In theory an older track could underrun and restart after the new one starts 3602 // but as we only care about the transition phase between two tracks on a 3603 // direct output, it is not a problem to ignore the underrun case. 3604 sp<Track> l = mLatestActiveTrack.promote(); 3605 bool last = l.get() == track; 3606 3607 // The first time a track is added we wait 3608 // for all its buffers to be filled before processing it 3609 uint32_t minFrames; 3610 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3611 minFrames = mNormalFrameCount; 3612 } else { 3613 minFrames = 1; 3614 } 3615 3616 if ((track->framesReady() >= minFrames) && track->isReady() && 3617 !track->isPaused() && !track->isTerminated()) 3618 { 3619 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3620 3621 if (track->mFillingUpStatus == Track::FS_FILLED) { 3622 track->mFillingUpStatus = Track::FS_ACTIVE; 3623 // make sure processVolume_l() will apply new volume even if 0 3624 mLeftVolFloat = mRightVolFloat = -1.0; 3625 if (track->mState == TrackBase::RESUMING) { 3626 track->mState = TrackBase::ACTIVE; 3627 } 3628 } 3629 3630 // compute volume for this track 3631 processVolume_l(track, last); 3632 if (last) { 3633 // reset retry count 3634 track->mRetryCount = kMaxTrackRetriesDirect; 3635 mActiveTrack = t; 3636 mixerStatus = MIXER_TRACKS_READY; 3637 } 3638 } else { 3639 // clear effect chain input buffer if the last active track started underruns 3640 // to avoid sending previous audio buffer again to effects 3641 if (!mEffectChains.isEmpty() && last) { 3642 mEffectChains[0]->clearInputBuffer(); 3643 } 3644 3645 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3646 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3647 track->isStopped() || track->isPaused()) { 3648 // We have consumed all the buffers of this track. 3649 // Remove it from the list of active tracks. 3650 // TODO: implement behavior for compressed audio 3651 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3652 size_t framesWritten = mBytesWritten / mFrameSize; 3653 if (mStandby || !last || 3654 track->presentationComplete(framesWritten, audioHALFrames)) { 3655 if (track->isStopped()) { 3656 track->reset(); 3657 } 3658 tracksToRemove->add(track); 3659 } 3660 } else { 3661 // No buffers for this track. Give it a few chances to 3662 // fill a buffer, then remove it from active list. 3663 // Only consider last track started for mixer state control 3664 if (--(track->mRetryCount) <= 0) { 3665 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3666 tracksToRemove->add(track); 3667 // indicate to client process that the track was disabled because of underrun; 3668 // it will then automatically call start() when data is available 3669 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3670 } else if (last) { 3671 mixerStatus = MIXER_TRACKS_ENABLED; 3672 } 3673 } 3674 } 3675 } 3676 3677 // remove all the tracks that need to be... 3678 removeTracks_l(*tracksToRemove); 3679 3680 return mixerStatus; 3681} 3682 3683void AudioFlinger::DirectOutputThread::threadLoop_mix() 3684{ 3685 size_t frameCount = mFrameCount; 3686 int8_t *curBuf = (int8_t *)mMixBuffer; 3687 // output audio to hardware 3688 while (frameCount) { 3689 AudioBufferProvider::Buffer buffer; 3690 buffer.frameCount = frameCount; 3691 mActiveTrack->getNextBuffer(&buffer); 3692 if (buffer.raw == NULL) { 3693 memset(curBuf, 0, frameCount * mFrameSize); 3694 break; 3695 } 3696 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3697 frameCount -= buffer.frameCount; 3698 curBuf += buffer.frameCount * mFrameSize; 3699 mActiveTrack->releaseBuffer(&buffer); 3700 } 3701 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3702 sleepTime = 0; 3703 standbyTime = systemTime() + standbyDelay; 3704 mActiveTrack.clear(); 3705} 3706 3707void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3708{ 3709 if (sleepTime == 0) { 3710 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3711 sleepTime = activeSleepTime; 3712 } else { 3713 sleepTime = idleSleepTime; 3714 } 3715 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3716 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3717 sleepTime = 0; 3718 } 3719} 3720 3721// getTrackName_l() must be called with ThreadBase::mLock held 3722int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3723 int sessionId) 3724{ 3725 return 0; 3726} 3727 3728// deleteTrackName_l() must be called with ThreadBase::mLock held 3729void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3730{ 3731} 3732 3733// checkForNewParameters_l() must be called with ThreadBase::mLock held 3734bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3735{ 3736 bool reconfig = false; 3737 3738 while (!mNewParameters.isEmpty()) { 3739 status_t status = NO_ERROR; 3740 String8 keyValuePair = mNewParameters[0]; 3741 AudioParameter param = AudioParameter(keyValuePair); 3742 int value; 3743 3744 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3745 // do not accept frame count changes if tracks are open as the track buffer 3746 // size depends on frame count and correct behavior would not be garantied 3747 // if frame count is changed after track creation 3748 if (!mTracks.isEmpty()) { 3749 status = INVALID_OPERATION; 3750 } else { 3751 reconfig = true; 3752 } 3753 } 3754 if (status == NO_ERROR) { 3755 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3756 keyValuePair.string()); 3757 if (!mStandby && status == INVALID_OPERATION) { 3758 mOutput->stream->common.standby(&mOutput->stream->common); 3759 mStandby = true; 3760 mBytesWritten = 0; 3761 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3762 keyValuePair.string()); 3763 } 3764 if (status == NO_ERROR && reconfig) { 3765 readOutputParameters(); 3766 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3767 } 3768 } 3769 3770 mNewParameters.removeAt(0); 3771 3772 mParamStatus = status; 3773 mParamCond.signal(); 3774 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3775 // already timed out waiting for the status and will never signal the condition. 3776 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3777 } 3778 return reconfig; 3779} 3780 3781uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3782{ 3783 uint32_t time; 3784 if (audio_is_linear_pcm(mFormat)) { 3785 time = PlaybackThread::activeSleepTimeUs(); 3786 } else { 3787 time = 10000; 3788 } 3789 return time; 3790} 3791 3792uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3793{ 3794 uint32_t time; 3795 if (audio_is_linear_pcm(mFormat)) { 3796 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3797 } else { 3798 time = 10000; 3799 } 3800 return time; 3801} 3802 3803uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3804{ 3805 uint32_t time; 3806 if (audio_is_linear_pcm(mFormat)) { 3807 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3808 } else { 3809 time = 10000; 3810 } 3811 return time; 3812} 3813 3814void AudioFlinger::DirectOutputThread::cacheParameters_l() 3815{ 3816 PlaybackThread::cacheParameters_l(); 3817 3818 // use shorter standby delay as on normal output to release 3819 // hardware resources as soon as possible 3820 if (audio_is_linear_pcm(mFormat)) { 3821 standbyDelay = microseconds(activeSleepTime*2); 3822 } else { 3823 standbyDelay = kOffloadStandbyDelayNs; 3824 } 3825} 3826 3827// ---------------------------------------------------------------------------- 3828 3829AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3830 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3831 : Thread(false /*canCallJava*/), 3832 mPlaybackThread(playbackThread), 3833 mWriteAckSequence(0), 3834 mDrainSequence(0) 3835{ 3836} 3837 3838AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3839{ 3840} 3841 3842void AudioFlinger::AsyncCallbackThread::onFirstRef() 3843{ 3844 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3845} 3846 3847bool AudioFlinger::AsyncCallbackThread::threadLoop() 3848{ 3849 while (!exitPending()) { 3850 uint32_t writeAckSequence; 3851 uint32_t drainSequence; 3852 3853 { 3854 Mutex::Autolock _l(mLock); 3855 while (!((mWriteAckSequence & 1) || 3856 (mDrainSequence & 1) || 3857 exitPending())) { 3858 mWaitWorkCV.wait(mLock); 3859 } 3860 3861 if (exitPending()) { 3862 break; 3863 } 3864 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3865 mWriteAckSequence, mDrainSequence); 3866 writeAckSequence = mWriteAckSequence; 3867 mWriteAckSequence &= ~1; 3868 drainSequence = mDrainSequence; 3869 mDrainSequence &= ~1; 3870 } 3871 { 3872 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3873 if (playbackThread != 0) { 3874 if (writeAckSequence & 1) { 3875 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3876 } 3877 if (drainSequence & 1) { 3878 playbackThread->resetDraining(drainSequence >> 1); 3879 } 3880 } 3881 } 3882 } 3883 return false; 3884} 3885 3886void AudioFlinger::AsyncCallbackThread::exit() 3887{ 3888 ALOGV("AsyncCallbackThread::exit"); 3889 Mutex::Autolock _l(mLock); 3890 requestExit(); 3891 mWaitWorkCV.broadcast(); 3892} 3893 3894void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3895{ 3896 Mutex::Autolock _l(mLock); 3897 // bit 0 is cleared 3898 mWriteAckSequence = sequence << 1; 3899} 3900 3901void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3902{ 3903 Mutex::Autolock _l(mLock); 3904 // ignore unexpected callbacks 3905 if (mWriteAckSequence & 2) { 3906 mWriteAckSequence |= 1; 3907 mWaitWorkCV.signal(); 3908 } 3909} 3910 3911void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3912{ 3913 Mutex::Autolock _l(mLock); 3914 // bit 0 is cleared 3915 mDrainSequence = sequence << 1; 3916} 3917 3918void AudioFlinger::AsyncCallbackThread::resetDraining() 3919{ 3920 Mutex::Autolock _l(mLock); 3921 // ignore unexpected callbacks 3922 if (mDrainSequence & 2) { 3923 mDrainSequence |= 1; 3924 mWaitWorkCV.signal(); 3925 } 3926} 3927 3928 3929// ---------------------------------------------------------------------------- 3930AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3931 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3932 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3933 mHwPaused(false), 3934 mFlushPending(false), 3935 mPausedBytesRemaining(0) 3936{ 3937 //FIXME: mStandby should be set to true by ThreadBase constructor 3938 mStandby = true; 3939} 3940 3941void AudioFlinger::OffloadThread::threadLoop_exit() 3942{ 3943 if (mFlushPending || mHwPaused) { 3944 // If a flush is pending or track was paused, just discard buffered data 3945 flushHw_l(); 3946 } else { 3947 mMixerStatus = MIXER_DRAIN_ALL; 3948 threadLoop_drain(); 3949 } 3950 mCallbackThread->exit(); 3951 PlaybackThread::threadLoop_exit(); 3952} 3953 3954AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3955 Vector< sp<Track> > *tracksToRemove 3956) 3957{ 3958 size_t count = mActiveTracks.size(); 3959 3960 mixer_state mixerStatus = MIXER_IDLE; 3961 bool doHwPause = false; 3962 bool doHwResume = false; 3963 3964 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3965 3966 // find out which tracks need to be processed 3967 for (size_t i = 0; i < count; i++) { 3968 sp<Track> t = mActiveTracks[i].promote(); 3969 // The track died recently 3970 if (t == 0) { 3971 continue; 3972 } 3973 Track* const track = t.get(); 3974 audio_track_cblk_t* cblk = track->cblk(); 3975 // Only consider last track started for volume and mixer state control. 3976 // In theory an older track could underrun and restart after the new one starts 3977 // but as we only care about the transition phase between two tracks on a 3978 // direct output, it is not a problem to ignore the underrun case. 3979 sp<Track> l = mLatestActiveTrack.promote(); 3980 bool last = l.get() == track; 3981 3982 if (track->isPausing()) { 3983 track->setPaused(); 3984 if (last) { 3985 if (!mHwPaused) { 3986 doHwPause = true; 3987 mHwPaused = true; 3988 } 3989 // If we were part way through writing the mixbuffer to 3990 // the HAL we must save this until we resume 3991 // BUG - this will be wrong if a different track is made active, 3992 // in that case we want to discard the pending data in the 3993 // mixbuffer and tell the client to present it again when the 3994 // track is resumed 3995 mPausedWriteLength = mCurrentWriteLength; 3996 mPausedBytesRemaining = mBytesRemaining; 3997 mBytesRemaining = 0; // stop writing 3998 } 3999 tracksToRemove->add(track); 4000 } else if (track->framesReady() && track->isReady() && 4001 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4002 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4003 if (track->mFillingUpStatus == Track::FS_FILLED) { 4004 track->mFillingUpStatus = Track::FS_ACTIVE; 4005 // make sure processVolume_l() will apply new volume even if 0 4006 mLeftVolFloat = mRightVolFloat = -1.0; 4007 if (track->mState == TrackBase::RESUMING) { 4008 track->mState = TrackBase::ACTIVE; 4009 if (last) { 4010 if (mPausedBytesRemaining) { 4011 // Need to continue write that was interrupted 4012 mCurrentWriteLength = mPausedWriteLength; 4013 mBytesRemaining = mPausedBytesRemaining; 4014 mPausedBytesRemaining = 0; 4015 } 4016 if (mHwPaused) { 4017 doHwResume = true; 4018 mHwPaused = false; 4019 // threadLoop_mix() will handle the case that we need to 4020 // resume an interrupted write 4021 } 4022 // enable write to audio HAL 4023 sleepTime = 0; 4024 } 4025 } 4026 } 4027 4028 if (last) { 4029 sp<Track> previousTrack = mPreviousTrack.promote(); 4030 if (previousTrack != 0) { 4031 if (track != previousTrack.get()) { 4032 // Flush any data still being written from last track 4033 mBytesRemaining = 0; 4034 if (mPausedBytesRemaining) { 4035 // Last track was paused so we also need to flush saved 4036 // mixbuffer state and invalidate track so that it will 4037 // re-submit that unwritten data when it is next resumed 4038 mPausedBytesRemaining = 0; 4039 // Invalidate is a bit drastic - would be more efficient 4040 // to have a flag to tell client that some of the 4041 // previously written data was lost 4042 previousTrack->invalidate(); 4043 } 4044 // flush data already sent to the DSP if changing audio session as audio 4045 // comes from a different source. Also invalidate previous track to force a 4046 // seek when resuming. 4047 if (previousTrack->sessionId() != track->sessionId()) { 4048 previousTrack->invalidate(); 4049 mFlushPending = true; 4050 } 4051 } 4052 } 4053 mPreviousTrack = track; 4054 // reset retry count 4055 track->mRetryCount = kMaxTrackRetriesOffload; 4056 mActiveTrack = t; 4057 mixerStatus = MIXER_TRACKS_READY; 4058 } 4059 } else { 4060 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4061 if (track->isStopping_1()) { 4062 // Hardware buffer can hold a large amount of audio so we must 4063 // wait for all current track's data to drain before we say 4064 // that the track is stopped. 4065 if (mBytesRemaining == 0) { 4066 // Only start draining when all data in mixbuffer 4067 // has been written 4068 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4069 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4070 // do not drain if no data was ever sent to HAL (mStandby == true) 4071 if (last && !mStandby) { 4072 // do not modify drain sequence if we are already draining. This happens 4073 // when resuming from pause after drain. 4074 if ((mDrainSequence & 1) == 0) { 4075 sleepTime = 0; 4076 standbyTime = systemTime() + standbyDelay; 4077 mixerStatus = MIXER_DRAIN_TRACK; 4078 mDrainSequence += 2; 4079 } 4080 if (mHwPaused) { 4081 // It is possible to move from PAUSED to STOPPING_1 without 4082 // a resume so we must ensure hardware is running 4083 doHwResume = true; 4084 mHwPaused = false; 4085 } 4086 } 4087 } 4088 } else if (track->isStopping_2()) { 4089 // Drain has completed or we are in standby, signal presentation complete 4090 if (!(mDrainSequence & 1) || !last || mStandby) { 4091 track->mState = TrackBase::STOPPED; 4092 size_t audioHALFrames = 4093 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4094 size_t framesWritten = 4095 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4096 track->presentationComplete(framesWritten, audioHALFrames); 4097 track->reset(); 4098 tracksToRemove->add(track); 4099 } 4100 } else { 4101 // No buffers for this track. Give it a few chances to 4102 // fill a buffer, then remove it from active list. 4103 if (--(track->mRetryCount) <= 0) { 4104 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4105 track->name()); 4106 tracksToRemove->add(track); 4107 // indicate to client process that the track was disabled because of underrun; 4108 // it will then automatically call start() when data is available 4109 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4110 } else if (last){ 4111 mixerStatus = MIXER_TRACKS_ENABLED; 4112 } 4113 } 4114 } 4115 // compute volume for this track 4116 processVolume_l(track, last); 4117 } 4118 4119 // make sure the pause/flush/resume sequence is executed in the right order. 4120 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4121 // before flush and then resume HW. This can happen in case of pause/flush/resume 4122 // if resume is received before pause is executed. 4123 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4124 mOutput->stream->pause(mOutput->stream); 4125 if (!doHwPause) { 4126 doHwResume = true; 4127 } 4128 } 4129 if (mFlushPending) { 4130 flushHw_l(); 4131 mFlushPending = false; 4132 } 4133 if (!mStandby && doHwResume) { 4134 mOutput->stream->resume(mOutput->stream); 4135 } 4136 4137 // remove all the tracks that need to be... 4138 removeTracks_l(*tracksToRemove); 4139 4140 return mixerStatus; 4141} 4142 4143void AudioFlinger::OffloadThread::flushOutput_l() 4144{ 4145 mFlushPending = true; 4146} 4147 4148// must be called with thread mutex locked 4149bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4150{ 4151 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4152 mWriteAckSequence, mDrainSequence); 4153 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4154 return true; 4155 } 4156 return false; 4157} 4158 4159// must be called with thread mutex locked 4160bool AudioFlinger::OffloadThread::shouldStandby_l() 4161{ 4162 bool TrackPaused = false; 4163 4164 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4165 // after a timeout and we will enter standby then. 4166 if (mTracks.size() > 0) { 4167 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4168 } 4169 4170 return !mStandby && !TrackPaused; 4171} 4172 4173 4174bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4175{ 4176 Mutex::Autolock _l(mLock); 4177 return waitingAsyncCallback_l(); 4178} 4179 4180void AudioFlinger::OffloadThread::flushHw_l() 4181{ 4182 mOutput->stream->flush(mOutput->stream); 4183 // Flush anything still waiting in the mixbuffer 4184 mCurrentWriteLength = 0; 4185 mBytesRemaining = 0; 4186 mPausedWriteLength = 0; 4187 mPausedBytesRemaining = 0; 4188 if (mUseAsyncWrite) { 4189 // discard any pending drain or write ack by incrementing sequence 4190 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4191 mDrainSequence = (mDrainSequence + 2) & ~1; 4192 ALOG_ASSERT(mCallbackThread != 0); 4193 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4194 mCallbackThread->setDraining(mDrainSequence); 4195 } 4196} 4197 4198// ---------------------------------------------------------------------------- 4199 4200AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4201 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4202 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4203 DUPLICATING), 4204 mWaitTimeMs(UINT_MAX) 4205{ 4206 addOutputTrack(mainThread); 4207} 4208 4209AudioFlinger::DuplicatingThread::~DuplicatingThread() 4210{ 4211 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4212 mOutputTracks[i]->destroy(); 4213 } 4214} 4215 4216void AudioFlinger::DuplicatingThread::threadLoop_mix() 4217{ 4218 // mix buffers... 4219 if (outputsReady(outputTracks)) { 4220 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4221 } else { 4222 memset(mMixBuffer, 0, mixBufferSize); 4223 } 4224 sleepTime = 0; 4225 writeFrames = mNormalFrameCount; 4226 mCurrentWriteLength = mixBufferSize; 4227 standbyTime = systemTime() + standbyDelay; 4228} 4229 4230void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4231{ 4232 if (sleepTime == 0) { 4233 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4234 sleepTime = activeSleepTime; 4235 } else { 4236 sleepTime = idleSleepTime; 4237 } 4238 } else if (mBytesWritten != 0) { 4239 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4240 writeFrames = mNormalFrameCount; 4241 memset(mMixBuffer, 0, mixBufferSize); 4242 } else { 4243 // flush remaining overflow buffers in output tracks 4244 writeFrames = 0; 4245 } 4246 sleepTime = 0; 4247 } 4248} 4249 4250ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4251{ 4252 for (size_t i = 0; i < outputTracks.size(); i++) { 4253 outputTracks[i]->write(mMixBuffer, writeFrames); 4254 } 4255 mStandby = false; 4256 return (ssize_t)mixBufferSize; 4257} 4258 4259void AudioFlinger::DuplicatingThread::threadLoop_standby() 4260{ 4261 // DuplicatingThread implements standby by stopping all tracks 4262 for (size_t i = 0; i < outputTracks.size(); i++) { 4263 outputTracks[i]->stop(); 4264 } 4265} 4266 4267void AudioFlinger::DuplicatingThread::saveOutputTracks() 4268{ 4269 outputTracks = mOutputTracks; 4270} 4271 4272void AudioFlinger::DuplicatingThread::clearOutputTracks() 4273{ 4274 outputTracks.clear(); 4275} 4276 4277void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4278{ 4279 Mutex::Autolock _l(mLock); 4280 // FIXME explain this formula 4281 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4282 OutputTrack *outputTrack = new OutputTrack(thread, 4283 this, 4284 mSampleRate, 4285 mFormat, 4286 mChannelMask, 4287 frameCount, 4288 IPCThreadState::self()->getCallingUid()); 4289 if (outputTrack->cblk() != NULL) { 4290 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4291 mOutputTracks.add(outputTrack); 4292 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4293 updateWaitTime_l(); 4294 } 4295} 4296 4297void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4298{ 4299 Mutex::Autolock _l(mLock); 4300 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4301 if (mOutputTracks[i]->thread() == thread) { 4302 mOutputTracks[i]->destroy(); 4303 mOutputTracks.removeAt(i); 4304 updateWaitTime_l(); 4305 return; 4306 } 4307 } 4308 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4309} 4310 4311// caller must hold mLock 4312void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4313{ 4314 mWaitTimeMs = UINT_MAX; 4315 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4316 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4317 if (strong != 0) { 4318 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4319 if (waitTimeMs < mWaitTimeMs) { 4320 mWaitTimeMs = waitTimeMs; 4321 } 4322 } 4323 } 4324} 4325 4326 4327bool AudioFlinger::DuplicatingThread::outputsReady( 4328 const SortedVector< sp<OutputTrack> > &outputTracks) 4329{ 4330 for (size_t i = 0; i < outputTracks.size(); i++) { 4331 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4332 if (thread == 0) { 4333 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4334 outputTracks[i].get()); 4335 return false; 4336 } 4337 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4338 // see note at standby() declaration 4339 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4340 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4341 thread.get()); 4342 return false; 4343 } 4344 } 4345 return true; 4346} 4347 4348uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4349{ 4350 return (mWaitTimeMs * 1000) / 2; 4351} 4352 4353void AudioFlinger::DuplicatingThread::cacheParameters_l() 4354{ 4355 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4356 updateWaitTime_l(); 4357 4358 MixerThread::cacheParameters_l(); 4359} 4360 4361// ---------------------------------------------------------------------------- 4362// Record 4363// ---------------------------------------------------------------------------- 4364 4365AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4366 AudioStreamIn *input, 4367 uint32_t sampleRate, 4368 audio_channel_mask_t channelMask, 4369 audio_io_handle_t id, 4370 audio_devices_t outDevice, 4371 audio_devices_t inDevice 4372#ifdef TEE_SINK 4373 , const sp<NBAIO_Sink>& teeSink 4374#endif 4375 ) : 4376 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4377 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4378 // mRsmpInIndex and mBufferSize set by readInputParameters() 4379 mReqChannelCount(popcount(channelMask)), 4380 mReqSampleRate(sampleRate) 4381 // mBytesRead is only meaningful while active, and so is cleared in start() 4382 // (but might be better to also clear here for dump?) 4383#ifdef TEE_SINK 4384 , mTeeSink(teeSink) 4385#endif 4386{ 4387 snprintf(mName, kNameLength, "AudioIn_%X", id); 4388 4389 readInputParameters(); 4390} 4391 4392 4393AudioFlinger::RecordThread::~RecordThread() 4394{ 4395 delete[] mRsmpInBuffer; 4396 delete mResampler; 4397 delete[] mRsmpOutBuffer; 4398} 4399 4400void AudioFlinger::RecordThread::onFirstRef() 4401{ 4402 run(mName, PRIORITY_URGENT_AUDIO); 4403} 4404 4405status_t AudioFlinger::RecordThread::readyToRun() 4406{ 4407 status_t status = initCheck(); 4408 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4409 return status; 4410} 4411 4412bool AudioFlinger::RecordThread::threadLoop() 4413{ 4414 AudioBufferProvider::Buffer buffer; 4415 sp<RecordTrack> activeTrack; 4416 Vector< sp<EffectChain> > effectChains; 4417 4418 nsecs_t lastWarning = 0; 4419 4420 inputStandBy(); 4421 { 4422 Mutex::Autolock _l(mLock); 4423 activeTrack = mActiveTrack; 4424 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4425 } 4426 4427 // used to verify we've read at least once before evaluating how many bytes were read 4428 bool readOnce = false; 4429 4430 // start recording 4431 while (!exitPending()) { 4432 4433 processConfigEvents(); 4434 4435 { // scope for mLock 4436 Mutex::Autolock _l(mLock); 4437 checkForNewParameters_l(); 4438 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4439 SortedVector<int> tmp; 4440 tmp.add(mActiveTrack->uid()); 4441 updateWakeLockUids_l(tmp); 4442 } 4443 activeTrack = mActiveTrack; 4444 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4445 standby(); 4446 4447 if (exitPending()) { 4448 break; 4449 } 4450 4451 releaseWakeLock_l(); 4452 ALOGV("RecordThread: loop stopping"); 4453 // go to sleep 4454 mWaitWorkCV.wait(mLock); 4455 ALOGV("RecordThread: loop starting"); 4456 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4457 continue; 4458 } 4459 if (mActiveTrack != 0) { 4460 if (mActiveTrack->isTerminated()) { 4461 removeTrack_l(mActiveTrack); 4462 mActiveTrack.clear(); 4463 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4464 standby(); 4465 mActiveTrack.clear(); 4466 mStartStopCond.broadcast(); 4467 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4468 if (mReqChannelCount != mActiveTrack->channelCount()) { 4469 mActiveTrack.clear(); 4470 mStartStopCond.broadcast(); 4471 } else if (readOnce) { 4472 // record start succeeds only if first read from audio input 4473 // succeeds 4474 if (mBytesRead >= 0) { 4475 mActiveTrack->mState = TrackBase::ACTIVE; 4476 } else { 4477 mActiveTrack.clear(); 4478 } 4479 mStartStopCond.broadcast(); 4480 } 4481 mStandby = false; 4482 } 4483 } 4484 4485 lockEffectChains_l(effectChains); 4486 } 4487 4488 if (mActiveTrack != 0) { 4489 if (mActiveTrack->mState != TrackBase::ACTIVE && 4490 mActiveTrack->mState != TrackBase::RESUMING) { 4491 unlockEffectChains(effectChains); 4492 usleep(kRecordThreadSleepUs); 4493 continue; 4494 } 4495 for (size_t i = 0; i < effectChains.size(); i ++) { 4496 effectChains[i]->process_l(); 4497 } 4498 4499 buffer.frameCount = mFrameCount; 4500 status_t status = mActiveTrack->getNextBuffer(&buffer); 4501 if (status == NO_ERROR) { 4502 readOnce = true; 4503 size_t framesOut = buffer.frameCount; 4504 if (mResampler == NULL) { 4505 // no resampling 4506 while (framesOut) { 4507 size_t framesIn = mFrameCount - mRsmpInIndex; 4508 if (framesIn) { 4509 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4510 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4511 mActiveTrack->mFrameSize; 4512 if (framesIn > framesOut) 4513 framesIn = framesOut; 4514 mRsmpInIndex += framesIn; 4515 framesOut -= framesIn; 4516 if (mChannelCount == mReqChannelCount) { 4517 memcpy(dst, src, framesIn * mFrameSize); 4518 } else { 4519 if (mChannelCount == 1) { 4520 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4521 (int16_t *)src, framesIn); 4522 } else { 4523 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4524 (int16_t *)src, framesIn); 4525 } 4526 } 4527 } 4528 if (framesOut && mFrameCount == mRsmpInIndex) { 4529 void *readInto; 4530 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4531 readInto = buffer.raw; 4532 framesOut = 0; 4533 } else { 4534 readInto = mRsmpInBuffer; 4535 mRsmpInIndex = 0; 4536 } 4537 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4538 mBufferSize); 4539 if (mBytesRead <= 0) { 4540 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4541 { 4542 ALOGE("Error reading audio input"); 4543 // Force input into standby so that it tries to 4544 // recover at next read attempt 4545 inputStandBy(); 4546 usleep(kRecordThreadSleepUs); 4547 } 4548 mRsmpInIndex = mFrameCount; 4549 framesOut = 0; 4550 buffer.frameCount = 0; 4551 } 4552#ifdef TEE_SINK 4553 else if (mTeeSink != 0) { 4554 (void) mTeeSink->write(readInto, 4555 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4556 } 4557#endif 4558 } 4559 } 4560 } else { 4561 // resampling 4562 4563 // resampler accumulates, but we only have one source track 4564 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4565 // alter output frame count as if we were expecting stereo samples 4566 if (mChannelCount == 1 && mReqChannelCount == 1) { 4567 framesOut >>= 1; 4568 } 4569 mResampler->resample(mRsmpOutBuffer, framesOut, 4570 this /* AudioBufferProvider* */); 4571 // ditherAndClamp() works as long as all buffers returned by 4572 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4573 if (mChannelCount == 2 && mReqChannelCount == 1) { 4574 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4575 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4576 // the resampler always outputs stereo samples: 4577 // do post stereo to mono conversion 4578 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4579 framesOut); 4580 } else { 4581 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4582 } 4583 // now done with mRsmpOutBuffer 4584 4585 } 4586 if (mFramestoDrop == 0) { 4587 mActiveTrack->releaseBuffer(&buffer); 4588 } else { 4589 if (mFramestoDrop > 0) { 4590 mFramestoDrop -= buffer.frameCount; 4591 if (mFramestoDrop <= 0) { 4592 clearSyncStartEvent(); 4593 } 4594 } else { 4595 mFramestoDrop += buffer.frameCount; 4596 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4597 mSyncStartEvent->isCancelled()) { 4598 ALOGW("Synced record %s, session %d, trigger session %d", 4599 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4600 mActiveTrack->sessionId(), 4601 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4602 clearSyncStartEvent(); 4603 } 4604 } 4605 } 4606 mActiveTrack->clearOverflow(); 4607 } 4608 // client isn't retrieving buffers fast enough 4609 else { 4610 if (!mActiveTrack->setOverflow()) { 4611 nsecs_t now = systemTime(); 4612 if ((now - lastWarning) > kWarningThrottleNs) { 4613 ALOGW("RecordThread: buffer overflow"); 4614 lastWarning = now; 4615 } 4616 } 4617 // Release the processor for a while before asking for a new buffer. 4618 // This will give the application more chance to read from the buffer and 4619 // clear the overflow. 4620 usleep(kRecordThreadSleepUs); 4621 } 4622 } 4623 // enable changes in effect chain 4624 unlockEffectChains(effectChains); 4625 effectChains.clear(); 4626 } 4627 4628 standby(); 4629 4630 { 4631 Mutex::Autolock _l(mLock); 4632 for (size_t i = 0; i < mTracks.size(); i++) { 4633 sp<RecordTrack> track = mTracks[i]; 4634 track->invalidate(); 4635 } 4636 mActiveTrack.clear(); 4637 mStartStopCond.broadcast(); 4638 } 4639 4640 releaseWakeLock(); 4641 4642 ALOGV("RecordThread %p exiting", this); 4643 return false; 4644} 4645 4646void AudioFlinger::RecordThread::standby() 4647{ 4648 if (!mStandby) { 4649 inputStandBy(); 4650 mStandby = true; 4651 } 4652} 4653 4654void AudioFlinger::RecordThread::inputStandBy() 4655{ 4656 mInput->stream->common.standby(&mInput->stream->common); 4657} 4658 4659sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4660 const sp<AudioFlinger::Client>& client, 4661 uint32_t sampleRate, 4662 audio_format_t format, 4663 audio_channel_mask_t channelMask, 4664 size_t frameCount, 4665 int sessionId, 4666 int uid, 4667 IAudioFlinger::track_flags_t *flags, 4668 pid_t tid, 4669 status_t *status) 4670{ 4671 sp<RecordTrack> track; 4672 status_t lStatus; 4673 4674 lStatus = initCheck(); 4675 if (lStatus != NO_ERROR) { 4676 ALOGE("createRecordTrack_l() audio driver not initialized"); 4677 goto Exit; 4678 } 4679 // client expresses a preference for FAST, but we get the final say 4680 if (*flags & IAudioFlinger::TRACK_FAST) { 4681 if ( 4682 // use case: callback handler and frame count is default or at least as large as HAL 4683 ( 4684 (tid != -1) && 4685 ((frameCount == 0) || 4686 (frameCount >= mFrameCount)) 4687 ) && 4688 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4689 // mono or stereo 4690 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4691 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4692 // hardware sample rate 4693 (sampleRate == mSampleRate) && 4694 // record thread has an associated fast recorder 4695 hasFastRecorder() 4696 // FIXME test that RecordThread for this fast track has a capable output HAL 4697 // FIXME add a permission test also? 4698 ) { 4699 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4700 if (frameCount == 0) { 4701 frameCount = mFrameCount * kFastTrackMultiplier; 4702 } 4703 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4704 frameCount, mFrameCount); 4705 } else { 4706 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4707 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4708 "hasFastRecorder=%d tid=%d", 4709 frameCount, mFrameCount, format, 4710 audio_is_linear_pcm(format), 4711 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4712 *flags &= ~IAudioFlinger::TRACK_FAST; 4713 // For compatibility with AudioRecord calculation, buffer depth is forced 4714 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4715 // This is probably too conservative, but legacy application code may depend on it. 4716 // If you change this calculation, also review the start threshold which is related. 4717 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4718 size_t mNormalFrameCount = 2048; // FIXME 4719 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4720 if (minBufCount < 2) { 4721 minBufCount = 2; 4722 } 4723 size_t minFrameCount = mNormalFrameCount * minBufCount; 4724 if (frameCount < minFrameCount) { 4725 frameCount = minFrameCount; 4726 } 4727 } 4728 } 4729 4730 // FIXME use flags and tid similar to createTrack_l() 4731 4732 { // scope for mLock 4733 Mutex::Autolock _l(mLock); 4734 4735 track = new RecordTrack(this, client, sampleRate, 4736 format, channelMask, frameCount, sessionId, uid); 4737 4738 if (track->getCblk() == 0) { 4739 ALOGE("createRecordTrack_l() no control block"); 4740 lStatus = NO_MEMORY; 4741 track.clear(); 4742 goto Exit; 4743 } 4744 mTracks.add(track); 4745 4746 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4747 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4748 mAudioFlinger->btNrecIsOff(); 4749 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4750 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4751 4752 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4753 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4754 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4755 // so ask activity manager to do this on our behalf 4756 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4757 } 4758 } 4759 lStatus = NO_ERROR; 4760 4761Exit: 4762 if (status) { 4763 *status = lStatus; 4764 } 4765 return track; 4766} 4767 4768status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4769 AudioSystem::sync_event_t event, 4770 int triggerSession) 4771{ 4772 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4773 sp<ThreadBase> strongMe = this; 4774 status_t status = NO_ERROR; 4775 4776 if (event == AudioSystem::SYNC_EVENT_NONE) { 4777 clearSyncStartEvent(); 4778 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4779 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4780 triggerSession, 4781 recordTrack->sessionId(), 4782 syncStartEventCallback, 4783 this); 4784 // Sync event can be cancelled by the trigger session if the track is not in a 4785 // compatible state in which case we start record immediately 4786 if (mSyncStartEvent->isCancelled()) { 4787 clearSyncStartEvent(); 4788 } else { 4789 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4790 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4791 } 4792 } 4793 4794 { 4795 AutoMutex lock(mLock); 4796 if (mActiveTrack != 0) { 4797 if (recordTrack != mActiveTrack.get()) { 4798 status = -EBUSY; 4799 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4800 mActiveTrack->mState = TrackBase::ACTIVE; 4801 } 4802 return status; 4803 } 4804 4805 recordTrack->mState = TrackBase::IDLE; 4806 mActiveTrack = recordTrack; 4807 mLock.unlock(); 4808 status_t status = AudioSystem::startInput(mId); 4809 mLock.lock(); 4810 if (status != NO_ERROR) { 4811 mActiveTrack.clear(); 4812 clearSyncStartEvent(); 4813 return status; 4814 } 4815 mRsmpInIndex = mFrameCount; 4816 mBytesRead = 0; 4817 if (mResampler != NULL) { 4818 mResampler->reset(); 4819 } 4820 mActiveTrack->mState = TrackBase::RESUMING; 4821 // signal thread to start 4822 ALOGV("Signal record thread"); 4823 mWaitWorkCV.broadcast(); 4824 // do not wait for mStartStopCond if exiting 4825 if (exitPending()) { 4826 mActiveTrack.clear(); 4827 status = INVALID_OPERATION; 4828 goto startError; 4829 } 4830 mStartStopCond.wait(mLock); 4831 if (mActiveTrack == 0) { 4832 ALOGV("Record failed to start"); 4833 status = BAD_VALUE; 4834 goto startError; 4835 } 4836 ALOGV("Record started OK"); 4837 return status; 4838 } 4839 4840startError: 4841 AudioSystem::stopInput(mId); 4842 clearSyncStartEvent(); 4843 return status; 4844} 4845 4846void AudioFlinger::RecordThread::clearSyncStartEvent() 4847{ 4848 if (mSyncStartEvent != 0) { 4849 mSyncStartEvent->cancel(); 4850 } 4851 mSyncStartEvent.clear(); 4852 mFramestoDrop = 0; 4853} 4854 4855void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4856{ 4857 sp<SyncEvent> strongEvent = event.promote(); 4858 4859 if (strongEvent != 0) { 4860 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4861 me->handleSyncStartEvent(strongEvent); 4862 } 4863} 4864 4865void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4866{ 4867 if (event == mSyncStartEvent) { 4868 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4869 // from audio HAL 4870 mFramestoDrop = mFrameCount * 2; 4871 } 4872} 4873 4874bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4875 ALOGV("RecordThread::stop"); 4876 AutoMutex _l(mLock); 4877 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4878 return false; 4879 } 4880 recordTrack->mState = TrackBase::PAUSING; 4881 // do not wait for mStartStopCond if exiting 4882 if (exitPending()) { 4883 return true; 4884 } 4885 mStartStopCond.wait(mLock); 4886 // if we have been restarted, recordTrack == mActiveTrack.get() here 4887 if (exitPending() || recordTrack != mActiveTrack.get()) { 4888 ALOGV("Record stopped OK"); 4889 return true; 4890 } 4891 return false; 4892} 4893 4894bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4895{ 4896 return false; 4897} 4898 4899status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4900{ 4901#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4902 if (!isValidSyncEvent(event)) { 4903 return BAD_VALUE; 4904 } 4905 4906 int eventSession = event->triggerSession(); 4907 status_t ret = NAME_NOT_FOUND; 4908 4909 Mutex::Autolock _l(mLock); 4910 4911 for (size_t i = 0; i < mTracks.size(); i++) { 4912 sp<RecordTrack> track = mTracks[i]; 4913 if (eventSession == track->sessionId()) { 4914 (void) track->setSyncEvent(event); 4915 ret = NO_ERROR; 4916 } 4917 } 4918 return ret; 4919#else 4920 return BAD_VALUE; 4921#endif 4922} 4923 4924// destroyTrack_l() must be called with ThreadBase::mLock held 4925void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4926{ 4927 track->terminate(); 4928 track->mState = TrackBase::STOPPED; 4929 // active tracks are removed by threadLoop() 4930 if (mActiveTrack != track) { 4931 removeTrack_l(track); 4932 } 4933} 4934 4935void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4936{ 4937 mTracks.remove(track); 4938 // need anything related to effects here? 4939} 4940 4941void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4942{ 4943 dumpInternals(fd, args); 4944 dumpTracks(fd, args); 4945 dumpEffectChains(fd, args); 4946} 4947 4948void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4949{ 4950 const size_t SIZE = 256; 4951 char buffer[SIZE]; 4952 String8 result; 4953 4954 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4955 result.append(buffer); 4956 4957 if (mActiveTrack != 0) { 4958 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4959 result.append(buffer); 4960 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4961 result.append(buffer); 4962 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4963 result.append(buffer); 4964 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4965 result.append(buffer); 4966 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4967 result.append(buffer); 4968 } else { 4969 result.append("No active record client\n"); 4970 } 4971 4972 write(fd, result.string(), result.size()); 4973 4974 dumpBase(fd, args); 4975} 4976 4977void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4978{ 4979 const size_t SIZE = 256; 4980 char buffer[SIZE]; 4981 String8 result; 4982 4983 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4984 result.append(buffer); 4985 RecordTrack::appendDumpHeader(result); 4986 for (size_t i = 0; i < mTracks.size(); ++i) { 4987 sp<RecordTrack> track = mTracks[i]; 4988 if (track != 0) { 4989 track->dump(buffer, SIZE); 4990 result.append(buffer); 4991 } 4992 } 4993 4994 if (mActiveTrack != 0) { 4995 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4996 result.append(buffer); 4997 RecordTrack::appendDumpHeader(result); 4998 mActiveTrack->dump(buffer, SIZE); 4999 result.append(buffer); 5000 5001 } 5002 write(fd, result.string(), result.size()); 5003} 5004 5005// AudioBufferProvider interface 5006status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5007{ 5008 size_t framesReq = buffer->frameCount; 5009 size_t framesReady = mFrameCount - mRsmpInIndex; 5010 int channelCount; 5011 5012 if (framesReady == 0) { 5013 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 5014 if (mBytesRead <= 0) { 5015 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 5016 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5017 // Force input into standby so that it tries to 5018 // recover at next read attempt 5019 inputStandBy(); 5020 usleep(kRecordThreadSleepUs); 5021 } 5022 buffer->raw = NULL; 5023 buffer->frameCount = 0; 5024 return NOT_ENOUGH_DATA; 5025 } 5026 mRsmpInIndex = 0; 5027 framesReady = mFrameCount; 5028 } 5029 5030 if (framesReq > framesReady) { 5031 framesReq = framesReady; 5032 } 5033 5034 if (mChannelCount == 1 && mReqChannelCount == 2) { 5035 channelCount = 1; 5036 } else { 5037 channelCount = 2; 5038 } 5039 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5040 buffer->frameCount = framesReq; 5041 return NO_ERROR; 5042} 5043 5044// AudioBufferProvider interface 5045void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5046{ 5047 mRsmpInIndex += buffer->frameCount; 5048 buffer->frameCount = 0; 5049} 5050 5051bool AudioFlinger::RecordThread::checkForNewParameters_l() 5052{ 5053 bool reconfig = false; 5054 5055 while (!mNewParameters.isEmpty()) { 5056 status_t status = NO_ERROR; 5057 String8 keyValuePair = mNewParameters[0]; 5058 AudioParameter param = AudioParameter(keyValuePair); 5059 int value; 5060 audio_format_t reqFormat = mFormat; 5061 uint32_t reqSamplingRate = mReqSampleRate; 5062 uint32_t reqChannelCount = mReqChannelCount; 5063 5064 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5065 reqSamplingRate = value; 5066 reconfig = true; 5067 } 5068 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5069 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5070 status = BAD_VALUE; 5071 } else { 5072 reqFormat = (audio_format_t) value; 5073 reconfig = true; 5074 } 5075 } 5076 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5077 reqChannelCount = popcount(value); 5078 reconfig = true; 5079 } 5080 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5081 // do not accept frame count changes if tracks are open as the track buffer 5082 // size depends on frame count and correct behavior would not be guaranteed 5083 // if frame count is changed after track creation 5084 if (mActiveTrack != 0) { 5085 status = INVALID_OPERATION; 5086 } else { 5087 reconfig = true; 5088 } 5089 } 5090 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5091 // forward device change to effects that have requested to be 5092 // aware of attached audio device. 5093 for (size_t i = 0; i < mEffectChains.size(); i++) { 5094 mEffectChains[i]->setDevice_l(value); 5095 } 5096 5097 // store input device and output device but do not forward output device to audio HAL. 5098 // Note that status is ignored by the caller for output device 5099 // (see AudioFlinger::setParameters() 5100 if (audio_is_output_devices(value)) { 5101 mOutDevice = value; 5102 status = BAD_VALUE; 5103 } else { 5104 mInDevice = value; 5105 // disable AEC and NS if the device is a BT SCO headset supporting those 5106 // pre processings 5107 if (mTracks.size() > 0) { 5108 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5109 mAudioFlinger->btNrecIsOff(); 5110 for (size_t i = 0; i < mTracks.size(); i++) { 5111 sp<RecordTrack> track = mTracks[i]; 5112 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5113 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5114 } 5115 } 5116 } 5117 } 5118 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5119 mAudioSource != (audio_source_t)value) { 5120 // forward device change to effects that have requested to be 5121 // aware of attached audio device. 5122 for (size_t i = 0; i < mEffectChains.size(); i++) { 5123 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5124 } 5125 mAudioSource = (audio_source_t)value; 5126 } 5127 if (status == NO_ERROR) { 5128 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5129 keyValuePair.string()); 5130 if (status == INVALID_OPERATION) { 5131 inputStandBy(); 5132 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5133 keyValuePair.string()); 5134 } 5135 if (reconfig) { 5136 if (status == BAD_VALUE && 5137 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5138 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5139 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5140 <= (2 * reqSamplingRate)) && 5141 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5142 <= FCC_2 && 5143 (reqChannelCount <= FCC_2)) { 5144 status = NO_ERROR; 5145 } 5146 if (status == NO_ERROR) { 5147 readInputParameters(); 5148 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5149 } 5150 } 5151 } 5152 5153 mNewParameters.removeAt(0); 5154 5155 mParamStatus = status; 5156 mParamCond.signal(); 5157 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5158 // already timed out waiting for the status and will never signal the condition. 5159 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5160 } 5161 return reconfig; 5162} 5163 5164String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5165{ 5166 Mutex::Autolock _l(mLock); 5167 if (initCheck() != NO_ERROR) { 5168 return String8(); 5169 } 5170 5171 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5172 const String8 out_s8(s); 5173 free(s); 5174 return out_s8; 5175} 5176 5177void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5178 AudioSystem::OutputDescriptor desc; 5179 void *param2 = NULL; 5180 5181 switch (event) { 5182 case AudioSystem::INPUT_OPENED: 5183 case AudioSystem::INPUT_CONFIG_CHANGED: 5184 desc.channelMask = mChannelMask; 5185 desc.samplingRate = mSampleRate; 5186 desc.format = mFormat; 5187 desc.frameCount = mFrameCount; 5188 desc.latency = 0; 5189 param2 = &desc; 5190 break; 5191 5192 case AudioSystem::INPUT_CLOSED: 5193 default: 5194 break; 5195 } 5196 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5197} 5198 5199void AudioFlinger::RecordThread::readInputParameters() 5200{ 5201 delete[] mRsmpInBuffer; 5202 // mRsmpInBuffer is always assigned a new[] below 5203 delete[] mRsmpOutBuffer; 5204 mRsmpOutBuffer = NULL; 5205 delete mResampler; 5206 mResampler = NULL; 5207 5208 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5209 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5210 mChannelCount = popcount(mChannelMask); 5211 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5212 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5213 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5214 } 5215 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5216 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5217 mFrameCount = mBufferSize / mFrameSize; 5218 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5219 5220 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5221 { 5222 int channelCount; 5223 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5224 // stereo to mono post process as the resampler always outputs stereo. 5225 if (mChannelCount == 1 && mReqChannelCount == 2) { 5226 channelCount = 1; 5227 } else { 5228 channelCount = 2; 5229 } 5230 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5231 mResampler->setSampleRate(mSampleRate); 5232 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5233 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5234 5235 // optmization: if mono to mono, alter input frame count as if we were inputing 5236 // stereo samples 5237 if (mChannelCount == 1 && mReqChannelCount == 1) { 5238 mFrameCount >>= 1; 5239 } 5240 5241 } 5242 mRsmpInIndex = mFrameCount; 5243} 5244 5245unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5246{ 5247 Mutex::Autolock _l(mLock); 5248 if (initCheck() != NO_ERROR) { 5249 return 0; 5250 } 5251 5252 return mInput->stream->get_input_frames_lost(mInput->stream); 5253} 5254 5255uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5256{ 5257 Mutex::Autolock _l(mLock); 5258 uint32_t result = 0; 5259 if (getEffectChain_l(sessionId) != 0) { 5260 result = EFFECT_SESSION; 5261 } 5262 5263 for (size_t i = 0; i < mTracks.size(); ++i) { 5264 if (sessionId == mTracks[i]->sessionId()) { 5265 result |= TRACK_SESSION; 5266 break; 5267 } 5268 } 5269 5270 return result; 5271} 5272 5273KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5274{ 5275 KeyedVector<int, bool> ids; 5276 Mutex::Autolock _l(mLock); 5277 for (size_t j = 0; j < mTracks.size(); ++j) { 5278 sp<RecordThread::RecordTrack> track = mTracks[j]; 5279 int sessionId = track->sessionId(); 5280 if (ids.indexOfKey(sessionId) < 0) { 5281 ids.add(sessionId, true); 5282 } 5283 } 5284 return ids; 5285} 5286 5287AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5288{ 5289 Mutex::Autolock _l(mLock); 5290 AudioStreamIn *input = mInput; 5291 mInput = NULL; 5292 return input; 5293} 5294 5295// this method must always be called either with ThreadBase mLock held or inside the thread loop 5296audio_stream_t* AudioFlinger::RecordThread::stream() const 5297{ 5298 if (mInput == NULL) { 5299 return NULL; 5300 } 5301 return &mInput->stream->common; 5302} 5303 5304status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5305{ 5306 // only one chain per input thread 5307 if (mEffectChains.size() != 0) { 5308 return INVALID_OPERATION; 5309 } 5310 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5311 5312 chain->setInBuffer(NULL); 5313 chain->setOutBuffer(NULL); 5314 5315 checkSuspendOnAddEffectChain_l(chain); 5316 5317 mEffectChains.add(chain); 5318 5319 return NO_ERROR; 5320} 5321 5322size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5323{ 5324 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5325 ALOGW_IF(mEffectChains.size() != 1, 5326 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5327 chain.get(), mEffectChains.size(), this); 5328 if (mEffectChains.size() == 1) { 5329 mEffectChains.removeAt(0); 5330 } 5331 return 0; 5332} 5333 5334}; // namespace android 5335