Threads.cpp revision 7e92abeafb184e8a34213d7149592e95a72601b0
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 //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 * kFastTrackMultiplier))) 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#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1222 // hardware sample rate 1223 (sampleRate == mSampleRate) && 1224#endif 1225 // normal mixer has an associated fast mixer 1226 hasFastMixer() && 1227 // there are sufficient fast track slots available 1228 (mFastTrackAvailMask != 0) 1229 // FIXME test that MixerThread for this fast track has a capable output HAL 1230 // FIXME add a permission test also? 1231 ) { 1232 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1233 if (frameCount == 0) { 1234 frameCount = mFrameCount * kFastTrackMultiplier; 1235 } 1236 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1237 frameCount, mFrameCount); 1238 } else { 1239 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1240 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1241 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1242 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1243 audio_is_linear_pcm(format), 1244 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1245 *flags &= ~IAudioFlinger::TRACK_FAST; 1246 // For compatibility with AudioTrack calculation, buffer depth is forced 1247 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1248 // This is probably too conservative, but legacy application code may depend on it. 1249 // If you change this calculation, also review the start threshold which is related. 1250 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1251 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1252 if (minBufCount < 2) { 1253 minBufCount = 2; 1254 } 1255 size_t minFrameCount = mNormalFrameCount * minBufCount; 1256 if (frameCount < minFrameCount) { 1257 frameCount = minFrameCount; 1258 } 1259 } 1260 } 1261 1262 if (mType == DIRECT) { 1263 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1264 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1265 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1266 "for output %p with format %d", 1267 sampleRate, format, channelMask, mOutput, mFormat); 1268 lStatus = BAD_VALUE; 1269 goto Exit; 1270 } 1271 } 1272 } else if (mType == OFFLOAD) { 1273 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1274 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1275 "for output %p with format %d", 1276 sampleRate, format, channelMask, mOutput, mFormat); 1277 lStatus = BAD_VALUE; 1278 goto Exit; 1279 } 1280 } else { 1281 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1282 ALOGE("createTrack_l() Bad parameter: format %d \"" 1283 "for output %p with format %d", 1284 format, mOutput, mFormat); 1285 lStatus = BAD_VALUE; 1286 goto Exit; 1287 } 1288 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1289 if (sampleRate > mSampleRate*2) { 1290 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1291 lStatus = BAD_VALUE; 1292 goto Exit; 1293 } 1294 } 1295 1296 lStatus = initCheck(); 1297 if (lStatus != NO_ERROR) { 1298 ALOGE("Audio driver not initialized."); 1299 goto Exit; 1300 } 1301 1302 { // scope for mLock 1303 Mutex::Autolock _l(mLock); 1304 1305 // all tracks in same audio session must share the same routing strategy otherwise 1306 // conflicts will happen when tracks are moved from one output to another by audio policy 1307 // manager 1308 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1309 for (size_t i = 0; i < mTracks.size(); ++i) { 1310 sp<Track> t = mTracks[i]; 1311 if (t != 0 && !t->isOutputTrack()) { 1312 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1313 if (sessionId == t->sessionId() && strategy != actual) { 1314 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1315 strategy, actual); 1316 lStatus = BAD_VALUE; 1317 goto Exit; 1318 } 1319 } 1320 } 1321 1322 if (!isTimed) { 1323 track = new Track(this, client, streamType, sampleRate, format, 1324 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1325 } else { 1326 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1327 channelMask, frameCount, sharedBuffer, sessionId, uid); 1328 } 1329 1330 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1331 lStatus = NO_MEMORY; 1332 // track must be cleared from the caller as the caller has the AF lock 1333 goto Exit; 1334 } 1335 1336 mTracks.add(track); 1337 1338 sp<EffectChain> chain = getEffectChain_l(sessionId); 1339 if (chain != 0) { 1340 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1341 track->setMainBuffer(chain->inBuffer()); 1342 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1343 chain->incTrackCnt(); 1344 } 1345 1346 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1347 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1348 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1349 // so ask activity manager to do this on our behalf 1350 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1351 } 1352 } 1353 1354 lStatus = NO_ERROR; 1355 1356Exit: 1357 if (status) { 1358 *status = lStatus; 1359 } 1360 return track; 1361} 1362 1363uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1364{ 1365 return latency; 1366} 1367 1368uint32_t AudioFlinger::PlaybackThread::latency() const 1369{ 1370 Mutex::Autolock _l(mLock); 1371 return latency_l(); 1372} 1373uint32_t AudioFlinger::PlaybackThread::latency_l() const 1374{ 1375 if (initCheck() == NO_ERROR) { 1376 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1377 } else { 1378 return 0; 1379 } 1380} 1381 1382void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1383{ 1384 Mutex::Autolock _l(mLock); 1385 // Don't apply master volume in SW if our HAL can do it for us. 1386 if (mOutput && mOutput->audioHwDev && 1387 mOutput->audioHwDev->canSetMasterVolume()) { 1388 mMasterVolume = 1.0; 1389 } else { 1390 mMasterVolume = value; 1391 } 1392} 1393 1394void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1395{ 1396 Mutex::Autolock _l(mLock); 1397 // Don't apply master mute in SW if our HAL can do it for us. 1398 if (mOutput && mOutput->audioHwDev && 1399 mOutput->audioHwDev->canSetMasterMute()) { 1400 mMasterMute = false; 1401 } else { 1402 mMasterMute = muted; 1403 } 1404} 1405 1406void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1407{ 1408 Mutex::Autolock _l(mLock); 1409 mStreamTypes[stream].volume = value; 1410 broadcast_l(); 1411} 1412 1413void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1414{ 1415 Mutex::Autolock _l(mLock); 1416 mStreamTypes[stream].mute = muted; 1417 broadcast_l(); 1418} 1419 1420float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1421{ 1422 Mutex::Autolock _l(mLock); 1423 return mStreamTypes[stream].volume; 1424} 1425 1426// addTrack_l() must be called with ThreadBase::mLock held 1427status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1428{ 1429 status_t status = ALREADY_EXISTS; 1430 1431 // set retry count for buffer fill 1432 track->mRetryCount = kMaxTrackStartupRetries; 1433 if (mActiveTracks.indexOf(track) < 0) { 1434 // the track is newly added, make sure it fills up all its 1435 // buffers before playing. This is to ensure the client will 1436 // effectively get the latency it requested. 1437 if (!track->isOutputTrack()) { 1438 TrackBase::track_state state = track->mState; 1439 mLock.unlock(); 1440 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1441 mLock.lock(); 1442 // abort track was stopped/paused while we released the lock 1443 if (state != track->mState) { 1444 if (status == NO_ERROR) { 1445 mLock.unlock(); 1446 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1447 mLock.lock(); 1448 } 1449 return INVALID_OPERATION; 1450 } 1451 // abort if start is rejected by audio policy manager 1452 if (status != NO_ERROR) { 1453 return PERMISSION_DENIED; 1454 } 1455#ifdef ADD_BATTERY_DATA 1456 // to track the speaker usage 1457 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1458#endif 1459 } 1460 1461 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1462 track->mResetDone = false; 1463 track->mPresentationCompleteFrames = 0; 1464 mActiveTracks.add(track); 1465 mWakeLockUids.add(track->uid()); 1466 mActiveTracksGeneration++; 1467 mLatestActiveTrack = track; 1468 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1469 if (chain != 0) { 1470 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1471 track->sessionId()); 1472 chain->incActiveTrackCnt(); 1473 } 1474 1475 status = NO_ERROR; 1476 } 1477 1478 ALOGV("signal playback thread"); 1479 broadcast_l(); 1480 1481 return status; 1482} 1483 1484bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1485{ 1486 track->terminate(); 1487 // active tracks are removed by threadLoop() 1488 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1489 track->mState = TrackBase::STOPPED; 1490 if (!trackActive) { 1491 removeTrack_l(track); 1492 } else if (track->isFastTrack() || track->isOffloaded()) { 1493 track->mState = TrackBase::STOPPING_1; 1494 } 1495 1496 return trackActive; 1497} 1498 1499void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1500{ 1501 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1502 mTracks.remove(track); 1503 deleteTrackName_l(track->name()); 1504 // redundant as track is about to be destroyed, for dumpsys only 1505 track->mName = -1; 1506 if (track->isFastTrack()) { 1507 int index = track->mFastIndex; 1508 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1509 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1510 mFastTrackAvailMask |= 1 << index; 1511 // redundant as track is about to be destroyed, for dumpsys only 1512 track->mFastIndex = -1; 1513 } 1514 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1515 if (chain != 0) { 1516 chain->decTrackCnt(); 1517 } 1518} 1519 1520void AudioFlinger::PlaybackThread::broadcast_l() 1521{ 1522 // Thread could be blocked waiting for async 1523 // so signal it to handle state changes immediately 1524 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1525 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1526 mSignalPending = true; 1527 mWaitWorkCV.broadcast(); 1528} 1529 1530String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1531{ 1532 Mutex::Autolock _l(mLock); 1533 if (initCheck() != NO_ERROR) { 1534 return String8(); 1535 } 1536 1537 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1538 const String8 out_s8(s); 1539 free(s); 1540 return out_s8; 1541} 1542 1543// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1544void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1545 AudioSystem::OutputDescriptor desc; 1546 void *param2 = NULL; 1547 1548 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1549 param); 1550 1551 switch (event) { 1552 case AudioSystem::OUTPUT_OPENED: 1553 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1554 desc.channelMask = mChannelMask; 1555 desc.samplingRate = mSampleRate; 1556 desc.format = mFormat; 1557 desc.frameCount = mNormalFrameCount; // FIXME see 1558 // AudioFlinger::frameCount(audio_io_handle_t) 1559 desc.latency = latency(); 1560 param2 = &desc; 1561 break; 1562 1563 case AudioSystem::STREAM_CONFIG_CHANGED: 1564 param2 = ¶m; 1565 case AudioSystem::OUTPUT_CLOSED: 1566 default: 1567 break; 1568 } 1569 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1570} 1571 1572void AudioFlinger::PlaybackThread::writeCallback() 1573{ 1574 ALOG_ASSERT(mCallbackThread != 0); 1575 mCallbackThread->resetWriteBlocked(); 1576} 1577 1578void AudioFlinger::PlaybackThread::drainCallback() 1579{ 1580 ALOG_ASSERT(mCallbackThread != 0); 1581 mCallbackThread->resetDraining(); 1582} 1583 1584void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1585{ 1586 Mutex::Autolock _l(mLock); 1587 // reject out of sequence requests 1588 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1589 mWriteAckSequence &= ~1; 1590 mWaitWorkCV.signal(); 1591 } 1592} 1593 1594void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1595{ 1596 Mutex::Autolock _l(mLock); 1597 // reject out of sequence requests 1598 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1599 mDrainSequence &= ~1; 1600 mWaitWorkCV.signal(); 1601 } 1602} 1603 1604// static 1605int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1606 void *param, 1607 void *cookie) 1608{ 1609 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1610 ALOGV("asyncCallback() event %d", event); 1611 switch (event) { 1612 case STREAM_CBK_EVENT_WRITE_READY: 1613 me->writeCallback(); 1614 break; 1615 case STREAM_CBK_EVENT_DRAIN_READY: 1616 me->drainCallback(); 1617 break; 1618 default: 1619 ALOGW("asyncCallback() unknown event %d", event); 1620 break; 1621 } 1622 return 0; 1623} 1624 1625void AudioFlinger::PlaybackThread::readOutputParameters() 1626{ 1627 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1628 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1629 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1630 if (!audio_is_output_channel(mChannelMask)) { 1631 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1632 } 1633 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1634 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1635 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1636 } 1637 mChannelCount = popcount(mChannelMask); 1638 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1639 if (!audio_is_valid_format(mFormat)) { 1640 LOG_FATAL("HAL format %d not valid for output", mFormat); 1641 } 1642 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1643 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1644 mFormat); 1645 } 1646 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1647 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1648 if (mFrameCount & 15) { 1649 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1650 mFrameCount); 1651 } 1652 1653 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1654 (mOutput->stream->set_callback != NULL)) { 1655 if (mOutput->stream->set_callback(mOutput->stream, 1656 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1657 mUseAsyncWrite = true; 1658 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1659 } 1660 } 1661 1662 // Calculate size of normal mix buffer relative to the HAL output buffer size 1663 double multiplier = 1.0; 1664 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1665 kUseFastMixer == FastMixer_Dynamic)) { 1666 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1667 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1668 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1669 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1670 maxNormalFrameCount = maxNormalFrameCount & ~15; 1671 if (maxNormalFrameCount < minNormalFrameCount) { 1672 maxNormalFrameCount = minNormalFrameCount; 1673 } 1674 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1675 if (multiplier <= 1.0) { 1676 multiplier = 1.0; 1677 } else if (multiplier <= 2.0) { 1678 if (2 * mFrameCount <= maxNormalFrameCount) { 1679 multiplier = 2.0; 1680 } else { 1681 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1682 } 1683 } else { 1684 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1685 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1686 // track, but we sometimes have to do this to satisfy the maximum frame count 1687 // constraint) 1688 // FIXME this rounding up should not be done if no HAL SRC 1689 uint32_t truncMult = (uint32_t) multiplier; 1690 if ((truncMult & 1)) { 1691 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1692 ++truncMult; 1693 } 1694 } 1695 multiplier = (double) truncMult; 1696 } 1697 } 1698 mNormalFrameCount = multiplier * mFrameCount; 1699 // round up to nearest 16 frames to satisfy AudioMixer 1700 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1701 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1702 mNormalFrameCount); 1703 1704 delete[] mAllocMixBuffer; 1705 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1706 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1707 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1708 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1709 1710 // force reconfiguration of effect chains and engines to take new buffer size and audio 1711 // parameters into account 1712 // Note that mLock is not held when readOutputParameters() is called from the constructor 1713 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1714 // matter. 1715 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1716 Vector< sp<EffectChain> > effectChains = mEffectChains; 1717 for (size_t i = 0; i < effectChains.size(); i ++) { 1718 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1719 } 1720} 1721 1722 1723status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1724{ 1725 if (halFrames == NULL || dspFrames == NULL) { 1726 return BAD_VALUE; 1727 } 1728 Mutex::Autolock _l(mLock); 1729 if (initCheck() != NO_ERROR) { 1730 return INVALID_OPERATION; 1731 } 1732 size_t framesWritten = mBytesWritten / mFrameSize; 1733 *halFrames = framesWritten; 1734 1735 if (isSuspended()) { 1736 // return an estimation of rendered frames when the output is suspended 1737 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1738 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1739 return NO_ERROR; 1740 } else { 1741 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1742 } 1743} 1744 1745uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1746{ 1747 Mutex::Autolock _l(mLock); 1748 uint32_t result = 0; 1749 if (getEffectChain_l(sessionId) != 0) { 1750 result = EFFECT_SESSION; 1751 } 1752 1753 for (size_t i = 0; i < mTracks.size(); ++i) { 1754 sp<Track> track = mTracks[i]; 1755 if (sessionId == track->sessionId() && !track->isInvalid()) { 1756 result |= TRACK_SESSION; 1757 break; 1758 } 1759 } 1760 1761 return result; 1762} 1763 1764uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1765{ 1766 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1767 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1768 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1769 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1770 } 1771 for (size_t i = 0; i < mTracks.size(); i++) { 1772 sp<Track> track = mTracks[i]; 1773 if (sessionId == track->sessionId() && !track->isInvalid()) { 1774 return AudioSystem::getStrategyForStream(track->streamType()); 1775 } 1776 } 1777 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1778} 1779 1780 1781AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1782{ 1783 Mutex::Autolock _l(mLock); 1784 return mOutput; 1785} 1786 1787AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1788{ 1789 Mutex::Autolock _l(mLock); 1790 AudioStreamOut *output = mOutput; 1791 mOutput = NULL; 1792 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1793 // must push a NULL and wait for ack 1794 mOutputSink.clear(); 1795 mPipeSink.clear(); 1796 mNormalSink.clear(); 1797 return output; 1798} 1799 1800// this method must always be called either with ThreadBase mLock held or inside the thread loop 1801audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1802{ 1803 if (mOutput == NULL) { 1804 return NULL; 1805 } 1806 return &mOutput->stream->common; 1807} 1808 1809uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1810{ 1811 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1812} 1813 1814status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1815{ 1816 if (!isValidSyncEvent(event)) { 1817 return BAD_VALUE; 1818 } 1819 1820 Mutex::Autolock _l(mLock); 1821 1822 for (size_t i = 0; i < mTracks.size(); ++i) { 1823 sp<Track> track = mTracks[i]; 1824 if (event->triggerSession() == track->sessionId()) { 1825 (void) track->setSyncEvent(event); 1826 return NO_ERROR; 1827 } 1828 } 1829 1830 return NAME_NOT_FOUND; 1831} 1832 1833bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1834{ 1835 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1836} 1837 1838void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1839 const Vector< sp<Track> >& tracksToRemove) 1840{ 1841 size_t count = tracksToRemove.size(); 1842 if (count) { 1843 for (size_t i = 0 ; i < count ; i++) { 1844 const sp<Track>& track = tracksToRemove.itemAt(i); 1845 if (!track->isOutputTrack()) { 1846 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1847#ifdef ADD_BATTERY_DATA 1848 // to track the speaker usage 1849 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1850#endif 1851 if (track->isTerminated()) { 1852 AudioSystem::releaseOutput(mId); 1853 } 1854 } 1855 } 1856 } 1857} 1858 1859void AudioFlinger::PlaybackThread::checkSilentMode_l() 1860{ 1861 if (!mMasterMute) { 1862 char value[PROPERTY_VALUE_MAX]; 1863 if (property_get("ro.audio.silent", value, "0") > 0) { 1864 char *endptr; 1865 unsigned long ul = strtoul(value, &endptr, 0); 1866 if (*endptr == '\0' && ul != 0) { 1867 ALOGD("Silence is golden"); 1868 // The setprop command will not allow a property to be changed after 1869 // the first time it is set, so we don't have to worry about un-muting. 1870 setMasterMute_l(true); 1871 } 1872 } 1873 } 1874} 1875 1876// shared by MIXER and DIRECT, overridden by DUPLICATING 1877ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1878{ 1879 // FIXME rewrite to reduce number of system calls 1880 mLastWriteTime = systemTime(); 1881 mInWrite = true; 1882 ssize_t bytesWritten; 1883 1884 // If an NBAIO sink is present, use it to write the normal mixer's submix 1885 if (mNormalSink != 0) { 1886#define mBitShift 2 // FIXME 1887 size_t count = mBytesRemaining >> mBitShift; 1888 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1889 ATRACE_BEGIN("write"); 1890 // update the setpoint when AudioFlinger::mScreenState changes 1891 uint32_t screenState = AudioFlinger::mScreenState; 1892 if (screenState != mScreenState) { 1893 mScreenState = screenState; 1894 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1895 if (pipe != NULL) { 1896 pipe->setAvgFrames((mScreenState & 1) ? 1897 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1898 } 1899 } 1900 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1901 ATRACE_END(); 1902 if (framesWritten > 0) { 1903 bytesWritten = framesWritten << mBitShift; 1904 } else { 1905 bytesWritten = framesWritten; 1906 } 1907 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1908 if (status == NO_ERROR) { 1909 size_t totalFramesWritten = mNormalSink->framesWritten(); 1910 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1911 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1912 mLatchDValid = true; 1913 } 1914 } 1915 // otherwise use the HAL / AudioStreamOut directly 1916 } else { 1917 // Direct output and offload threads 1918 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1919 if (mUseAsyncWrite) { 1920 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1921 mWriteAckSequence += 2; 1922 mWriteAckSequence |= 1; 1923 ALOG_ASSERT(mCallbackThread != 0); 1924 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1925 } 1926 // FIXME We should have an implementation of timestamps for direct output threads. 1927 // They are used e.g for multichannel PCM playback over HDMI. 1928 bytesWritten = mOutput->stream->write(mOutput->stream, 1929 (char *)mMixBuffer + offset, mBytesRemaining); 1930 if (mUseAsyncWrite && 1931 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1932 // do not wait for async callback in case of error of full write 1933 mWriteAckSequence &= ~1; 1934 ALOG_ASSERT(mCallbackThread != 0); 1935 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1936 } 1937 } 1938 1939 mNumWrites++; 1940 mInWrite = false; 1941 mStandby = false; 1942 return bytesWritten; 1943} 1944 1945void AudioFlinger::PlaybackThread::threadLoop_drain() 1946{ 1947 if (mOutput->stream->drain) { 1948 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1949 if (mUseAsyncWrite) { 1950 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1951 mDrainSequence |= 1; 1952 ALOG_ASSERT(mCallbackThread != 0); 1953 mCallbackThread->setDraining(mDrainSequence); 1954 } 1955 mOutput->stream->drain(mOutput->stream, 1956 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1957 : AUDIO_DRAIN_ALL); 1958 } 1959} 1960 1961void AudioFlinger::PlaybackThread::threadLoop_exit() 1962{ 1963 // Default implementation has nothing to do 1964} 1965 1966/* 1967The derived values that are cached: 1968 - mixBufferSize from frame count * frame size 1969 - activeSleepTime from activeSleepTimeUs() 1970 - idleSleepTime from idleSleepTimeUs() 1971 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1972 - maxPeriod from frame count and sample rate (MIXER only) 1973 1974The parameters that affect these derived values are: 1975 - frame count 1976 - frame size 1977 - sample rate 1978 - device type: A2DP or not 1979 - device latency 1980 - format: PCM or not 1981 - active sleep time 1982 - idle sleep time 1983*/ 1984 1985void AudioFlinger::PlaybackThread::cacheParameters_l() 1986{ 1987 mixBufferSize = mNormalFrameCount * mFrameSize; 1988 activeSleepTime = activeSleepTimeUs(); 1989 idleSleepTime = idleSleepTimeUs(); 1990} 1991 1992void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1993{ 1994 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1995 this, streamType, mTracks.size()); 1996 Mutex::Autolock _l(mLock); 1997 1998 size_t size = mTracks.size(); 1999 for (size_t i = 0; i < size; i++) { 2000 sp<Track> t = mTracks[i]; 2001 if (t->streamType() == streamType) { 2002 t->invalidate(); 2003 } 2004 } 2005} 2006 2007status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2008{ 2009 int session = chain->sessionId(); 2010 int16_t *buffer = mMixBuffer; 2011 bool ownsBuffer = false; 2012 2013 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2014 if (session > 0) { 2015 // Only one effect chain can be present in direct output thread and it uses 2016 // the mix buffer as input 2017 if (mType != DIRECT) { 2018 size_t numSamples = mNormalFrameCount * mChannelCount; 2019 buffer = new int16_t[numSamples]; 2020 memset(buffer, 0, numSamples * sizeof(int16_t)); 2021 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2022 ownsBuffer = true; 2023 } 2024 2025 // Attach all tracks with same session ID to this chain. 2026 for (size_t i = 0; i < mTracks.size(); ++i) { 2027 sp<Track> track = mTracks[i]; 2028 if (session == track->sessionId()) { 2029 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2030 buffer); 2031 track->setMainBuffer(buffer); 2032 chain->incTrackCnt(); 2033 } 2034 } 2035 2036 // indicate all active tracks in the chain 2037 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2038 sp<Track> track = mActiveTracks[i].promote(); 2039 if (track == 0) { 2040 continue; 2041 } 2042 if (session == track->sessionId()) { 2043 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2044 chain->incActiveTrackCnt(); 2045 } 2046 } 2047 } 2048 2049 chain->setInBuffer(buffer, ownsBuffer); 2050 chain->setOutBuffer(mMixBuffer); 2051 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2052 // chains list in order to be processed last as it contains output stage effects 2053 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2054 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2055 // after track specific effects and before output stage 2056 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2057 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2058 // Effect chain for other sessions are inserted at beginning of effect 2059 // chains list to be processed before output mix effects. Relative order between other 2060 // sessions is not important 2061 size_t size = mEffectChains.size(); 2062 size_t i = 0; 2063 for (i = 0; i < size; i++) { 2064 if (mEffectChains[i]->sessionId() < session) { 2065 break; 2066 } 2067 } 2068 mEffectChains.insertAt(chain, i); 2069 checkSuspendOnAddEffectChain_l(chain); 2070 2071 return NO_ERROR; 2072} 2073 2074size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2075{ 2076 int session = chain->sessionId(); 2077 2078 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2079 2080 for (size_t i = 0; i < mEffectChains.size(); i++) { 2081 if (chain == mEffectChains[i]) { 2082 mEffectChains.removeAt(i); 2083 // detach all active tracks from the chain 2084 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2085 sp<Track> track = mActiveTracks[i].promote(); 2086 if (track == 0) { 2087 continue; 2088 } 2089 if (session == track->sessionId()) { 2090 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2091 chain.get(), session); 2092 chain->decActiveTrackCnt(); 2093 } 2094 } 2095 2096 // detach all tracks with same session ID from this chain 2097 for (size_t i = 0; i < mTracks.size(); ++i) { 2098 sp<Track> track = mTracks[i]; 2099 if (session == track->sessionId()) { 2100 track->setMainBuffer(mMixBuffer); 2101 chain->decTrackCnt(); 2102 } 2103 } 2104 break; 2105 } 2106 } 2107 return mEffectChains.size(); 2108} 2109 2110status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2111 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2112{ 2113 Mutex::Autolock _l(mLock); 2114 return attachAuxEffect_l(track, EffectId); 2115} 2116 2117status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2118 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2119{ 2120 status_t status = NO_ERROR; 2121 2122 if (EffectId == 0) { 2123 track->setAuxBuffer(0, NULL); 2124 } else { 2125 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2126 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2127 if (effect != 0) { 2128 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2129 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2130 } else { 2131 status = INVALID_OPERATION; 2132 } 2133 } else { 2134 status = BAD_VALUE; 2135 } 2136 } 2137 return status; 2138} 2139 2140void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2141{ 2142 for (size_t i = 0; i < mTracks.size(); ++i) { 2143 sp<Track> track = mTracks[i]; 2144 if (track->auxEffectId() == effectId) { 2145 attachAuxEffect_l(track, 0); 2146 } 2147 } 2148} 2149 2150bool AudioFlinger::PlaybackThread::threadLoop() 2151{ 2152 Vector< sp<Track> > tracksToRemove; 2153 2154 standbyTime = systemTime(); 2155 2156 // MIXER 2157 nsecs_t lastWarning = 0; 2158 2159 // DUPLICATING 2160 // FIXME could this be made local to while loop? 2161 writeFrames = 0; 2162 2163 int lastGeneration = 0; 2164 2165 cacheParameters_l(); 2166 sleepTime = idleSleepTime; 2167 2168 if (mType == MIXER) { 2169 sleepTimeShift = 0; 2170 } 2171 2172 CpuStats cpuStats; 2173 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2174 2175 acquireWakeLock(); 2176 2177 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2178 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2179 // and then that string will be logged at the next convenient opportunity. 2180 const char *logString = NULL; 2181 2182 checkSilentMode_l(); 2183 2184 while (!exitPending()) 2185 { 2186 cpuStats.sample(myName); 2187 2188 Vector< sp<EffectChain> > effectChains; 2189 2190 processConfigEvents(); 2191 2192 { // scope for mLock 2193 2194 Mutex::Autolock _l(mLock); 2195 2196 if (logString != NULL) { 2197 mNBLogWriter->logTimestamp(); 2198 mNBLogWriter->log(logString); 2199 logString = NULL; 2200 } 2201 2202 if (mLatchDValid) { 2203 mLatchQ = mLatchD; 2204 mLatchDValid = false; 2205 mLatchQValid = true; 2206 } 2207 2208 if (checkForNewParameters_l()) { 2209 cacheParameters_l(); 2210 } 2211 2212 saveOutputTracks(); 2213 if (mSignalPending) { 2214 // A signal was raised while we were unlocked 2215 mSignalPending = false; 2216 } else if (waitingAsyncCallback_l()) { 2217 if (exitPending()) { 2218 break; 2219 } 2220 releaseWakeLock_l(); 2221 mWakeLockUids.clear(); 2222 mActiveTracksGeneration++; 2223 ALOGV("wait async completion"); 2224 mWaitWorkCV.wait(mLock); 2225 ALOGV("async completion/wake"); 2226 acquireWakeLock_l(); 2227 standbyTime = systemTime() + standbyDelay; 2228 sleepTime = 0; 2229 2230 continue; 2231 } 2232 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2233 isSuspended()) { 2234 // put audio hardware into standby after short delay 2235 if (shouldStandby_l()) { 2236 2237 threadLoop_standby(); 2238 2239 mStandby = true; 2240 } 2241 2242 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2243 // we're about to wait, flush the binder command buffer 2244 IPCThreadState::self()->flushCommands(); 2245 2246 clearOutputTracks(); 2247 2248 if (exitPending()) { 2249 break; 2250 } 2251 2252 releaseWakeLock_l(); 2253 mWakeLockUids.clear(); 2254 mActiveTracksGeneration++; 2255 // wait until we have something to do... 2256 ALOGV("%s going to sleep", myName.string()); 2257 mWaitWorkCV.wait(mLock); 2258 ALOGV("%s waking up", myName.string()); 2259 acquireWakeLock_l(); 2260 2261 mMixerStatus = MIXER_IDLE; 2262 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2263 mBytesWritten = 0; 2264 mBytesRemaining = 0; 2265 checkSilentMode_l(); 2266 2267 standbyTime = systemTime() + standbyDelay; 2268 sleepTime = idleSleepTime; 2269 if (mType == MIXER) { 2270 sleepTimeShift = 0; 2271 } 2272 2273 continue; 2274 } 2275 } 2276 // mMixerStatusIgnoringFastTracks is also updated internally 2277 mMixerStatus = prepareTracks_l(&tracksToRemove); 2278 2279 // compare with previously applied list 2280 if (lastGeneration != mActiveTracksGeneration) { 2281 // update wakelock 2282 updateWakeLockUids_l(mWakeLockUids); 2283 lastGeneration = mActiveTracksGeneration; 2284 } 2285 2286 // prevent any changes in effect chain list and in each effect chain 2287 // during mixing and effect process as the audio buffers could be deleted 2288 // or modified if an effect is created or deleted 2289 lockEffectChains_l(effectChains); 2290 } // mLock scope ends 2291 2292 if (mBytesRemaining == 0) { 2293 mCurrentWriteLength = 0; 2294 if (mMixerStatus == MIXER_TRACKS_READY) { 2295 // threadLoop_mix() sets mCurrentWriteLength 2296 threadLoop_mix(); 2297 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2298 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2299 // threadLoop_sleepTime sets sleepTime to 0 if data 2300 // must be written to HAL 2301 threadLoop_sleepTime(); 2302 if (sleepTime == 0) { 2303 mCurrentWriteLength = mixBufferSize; 2304 } 2305 } 2306 mBytesRemaining = mCurrentWriteLength; 2307 if (isSuspended()) { 2308 sleepTime = suspendSleepTimeUs(); 2309 // simulate write to HAL when suspended 2310 mBytesWritten += mixBufferSize; 2311 mBytesRemaining = 0; 2312 } 2313 2314 // only process effects if we're going to write 2315 if (sleepTime == 0 && mType != OFFLOAD) { 2316 for (size_t i = 0; i < effectChains.size(); i ++) { 2317 effectChains[i]->process_l(); 2318 } 2319 } 2320 } 2321 // Process effect chains for offloaded thread even if no audio 2322 // was read from audio track: process only updates effect state 2323 // and thus does have to be synchronized with audio writes but may have 2324 // to be called while waiting for async write callback 2325 if (mType == OFFLOAD) { 2326 for (size_t i = 0; i < effectChains.size(); i ++) { 2327 effectChains[i]->process_l(); 2328 } 2329 } 2330 2331 // enable changes in effect chain 2332 unlockEffectChains(effectChains); 2333 2334 if (!waitingAsyncCallback()) { 2335 // sleepTime == 0 means we must write to audio hardware 2336 if (sleepTime == 0) { 2337 if (mBytesRemaining) { 2338 ssize_t ret = threadLoop_write(); 2339 if (ret < 0) { 2340 mBytesRemaining = 0; 2341 } else { 2342 mBytesWritten += ret; 2343 mBytesRemaining -= ret; 2344 } 2345 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2346 (mMixerStatus == MIXER_DRAIN_ALL)) { 2347 threadLoop_drain(); 2348 } 2349if (mType == MIXER) { 2350 // write blocked detection 2351 nsecs_t now = systemTime(); 2352 nsecs_t delta = now - mLastWriteTime; 2353 if (!mStandby && delta > maxPeriod) { 2354 mNumDelayedWrites++; 2355 if ((now - lastWarning) > kWarningThrottleNs) { 2356 ATRACE_NAME("underrun"); 2357 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2358 ns2ms(delta), mNumDelayedWrites, this); 2359 lastWarning = now; 2360 } 2361 } 2362} 2363 2364 } else { 2365 usleep(sleepTime); 2366 } 2367 } 2368 2369 // Finally let go of removed track(s), without the lock held 2370 // since we can't guarantee the destructors won't acquire that 2371 // same lock. This will also mutate and push a new fast mixer state. 2372 threadLoop_removeTracks(tracksToRemove); 2373 tracksToRemove.clear(); 2374 2375 // FIXME I don't understand the need for this here; 2376 // it was in the original code but maybe the 2377 // assignment in saveOutputTracks() makes this unnecessary? 2378 clearOutputTracks(); 2379 2380 // Effect chains will be actually deleted here if they were removed from 2381 // mEffectChains list during mixing or effects processing 2382 effectChains.clear(); 2383 2384 // FIXME Note that the above .clear() is no longer necessary since effectChains 2385 // is now local to this block, but will keep it for now (at least until merge done). 2386 } 2387 2388 threadLoop_exit(); 2389 2390 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2391 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2392 // put output stream into standby mode 2393 if (!mStandby) { 2394 mOutput->stream->common.standby(&mOutput->stream->common); 2395 } 2396 } 2397 2398 releaseWakeLock(); 2399 mWakeLockUids.clear(); 2400 mActiveTracksGeneration++; 2401 2402 ALOGV("Thread %p type %d exiting", this, mType); 2403 return false; 2404} 2405 2406// removeTracks_l() must be called with ThreadBase::mLock held 2407void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2408{ 2409 size_t count = tracksToRemove.size(); 2410 if (count) { 2411 for (size_t i=0 ; i<count ; i++) { 2412 const sp<Track>& track = tracksToRemove.itemAt(i); 2413 mActiveTracks.remove(track); 2414 mWakeLockUids.remove(track->uid()); 2415 mActiveTracksGeneration++; 2416 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2417 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2418 if (chain != 0) { 2419 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2420 track->sessionId()); 2421 chain->decActiveTrackCnt(); 2422 } 2423 if (track->isTerminated()) { 2424 removeTrack_l(track); 2425 } 2426 } 2427 } 2428 2429} 2430 2431status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2432{ 2433 if (mNormalSink != 0) { 2434 return mNormalSink->getTimestamp(timestamp); 2435 } 2436 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2437 uint64_t position64; 2438 int ret = mOutput->stream->get_presentation_position( 2439 mOutput->stream, &position64, ×tamp.mTime); 2440 if (ret == 0) { 2441 timestamp.mPosition = (uint32_t)position64; 2442 return NO_ERROR; 2443 } 2444 } 2445 return INVALID_OPERATION; 2446} 2447// ---------------------------------------------------------------------------- 2448 2449AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2450 audio_io_handle_t id, audio_devices_t device, type_t type) 2451 : PlaybackThread(audioFlinger, output, id, device, type), 2452 // mAudioMixer below 2453 // mFastMixer below 2454 mFastMixerFutex(0) 2455 // mOutputSink below 2456 // mPipeSink below 2457 // mNormalSink below 2458{ 2459 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2460 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2461 "mFrameCount=%d, mNormalFrameCount=%d", 2462 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2463 mNormalFrameCount); 2464 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2465 2466 // FIXME - Current mixer implementation only supports stereo output 2467 if (mChannelCount != FCC_2) { 2468 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2469 } 2470 2471 // create an NBAIO sink for the HAL output stream, and negotiate 2472 mOutputSink = new AudioStreamOutSink(output->stream); 2473 size_t numCounterOffers = 0; 2474 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2475 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2476 ALOG_ASSERT(index == 0); 2477 2478 // initialize fast mixer depending on configuration 2479 bool initFastMixer; 2480 switch (kUseFastMixer) { 2481 case FastMixer_Never: 2482 initFastMixer = false; 2483 break; 2484 case FastMixer_Always: 2485 initFastMixer = true; 2486 break; 2487 case FastMixer_Static: 2488 case FastMixer_Dynamic: 2489 initFastMixer = mFrameCount < mNormalFrameCount; 2490 break; 2491 } 2492 if (initFastMixer) { 2493 2494 // create a MonoPipe to connect our submix to FastMixer 2495 NBAIO_Format format = mOutputSink->format(); 2496 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2497 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2498 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2499 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2500 const NBAIO_Format offers[1] = {format}; 2501 size_t numCounterOffers = 0; 2502 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2503 ALOG_ASSERT(index == 0); 2504 monoPipe->setAvgFrames((mScreenState & 1) ? 2505 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2506 mPipeSink = monoPipe; 2507 2508#ifdef TEE_SINK 2509 if (mTeeSinkOutputEnabled) { 2510 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2511 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2512 numCounterOffers = 0; 2513 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2514 ALOG_ASSERT(index == 0); 2515 mTeeSink = teeSink; 2516 PipeReader *teeSource = new PipeReader(*teeSink); 2517 numCounterOffers = 0; 2518 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2519 ALOG_ASSERT(index == 0); 2520 mTeeSource = teeSource; 2521 } 2522#endif 2523 2524 // create fast mixer and configure it initially with just one fast track for our submix 2525 mFastMixer = new FastMixer(); 2526 FastMixerStateQueue *sq = mFastMixer->sq(); 2527#ifdef STATE_QUEUE_DUMP 2528 sq->setObserverDump(&mStateQueueObserverDump); 2529 sq->setMutatorDump(&mStateQueueMutatorDump); 2530#endif 2531 FastMixerState *state = sq->begin(); 2532 FastTrack *fastTrack = &state->mFastTracks[0]; 2533 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2534 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2535 fastTrack->mVolumeProvider = NULL; 2536 fastTrack->mGeneration++; 2537 state->mFastTracksGen++; 2538 state->mTrackMask = 1; 2539 // fast mixer will use the HAL output sink 2540 state->mOutputSink = mOutputSink.get(); 2541 state->mOutputSinkGen++; 2542 state->mFrameCount = mFrameCount; 2543 state->mCommand = FastMixerState::COLD_IDLE; 2544 // already done in constructor initialization list 2545 //mFastMixerFutex = 0; 2546 state->mColdFutexAddr = &mFastMixerFutex; 2547 state->mColdGen++; 2548 state->mDumpState = &mFastMixerDumpState; 2549#ifdef TEE_SINK 2550 state->mTeeSink = mTeeSink.get(); 2551#endif 2552 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2553 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2554 sq->end(); 2555 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2556 2557 // start the fast mixer 2558 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2559 pid_t tid = mFastMixer->getTid(); 2560 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2561 if (err != 0) { 2562 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2563 kPriorityFastMixer, getpid_cached, tid, err); 2564 } 2565 2566#ifdef AUDIO_WATCHDOG 2567 // create and start the watchdog 2568 mAudioWatchdog = new AudioWatchdog(); 2569 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2570 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2571 tid = mAudioWatchdog->getTid(); 2572 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2573 if (err != 0) { 2574 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2575 kPriorityFastMixer, getpid_cached, tid, err); 2576 } 2577#endif 2578 2579 } else { 2580 mFastMixer = NULL; 2581 } 2582 2583 switch (kUseFastMixer) { 2584 case FastMixer_Never: 2585 case FastMixer_Dynamic: 2586 mNormalSink = mOutputSink; 2587 break; 2588 case FastMixer_Always: 2589 mNormalSink = mPipeSink; 2590 break; 2591 case FastMixer_Static: 2592 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2593 break; 2594 } 2595} 2596 2597AudioFlinger::MixerThread::~MixerThread() 2598{ 2599 if (mFastMixer != NULL) { 2600 FastMixerStateQueue *sq = mFastMixer->sq(); 2601 FastMixerState *state = sq->begin(); 2602 if (state->mCommand == FastMixerState::COLD_IDLE) { 2603 int32_t old = android_atomic_inc(&mFastMixerFutex); 2604 if (old == -1) { 2605 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2606 } 2607 } 2608 state->mCommand = FastMixerState::EXIT; 2609 sq->end(); 2610 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2611 mFastMixer->join(); 2612 // Though the fast mixer thread has exited, it's state queue is still valid. 2613 // We'll use that extract the final state which contains one remaining fast track 2614 // corresponding to our sub-mix. 2615 state = sq->begin(); 2616 ALOG_ASSERT(state->mTrackMask == 1); 2617 FastTrack *fastTrack = &state->mFastTracks[0]; 2618 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2619 delete fastTrack->mBufferProvider; 2620 sq->end(false /*didModify*/); 2621 delete mFastMixer; 2622#ifdef AUDIO_WATCHDOG 2623 if (mAudioWatchdog != 0) { 2624 mAudioWatchdog->requestExit(); 2625 mAudioWatchdog->requestExitAndWait(); 2626 mAudioWatchdog.clear(); 2627 } 2628#endif 2629 } 2630 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2631 delete mAudioMixer; 2632} 2633 2634 2635uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2636{ 2637 if (mFastMixer != NULL) { 2638 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2639 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2640 } 2641 return latency; 2642} 2643 2644 2645void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2646{ 2647 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2648} 2649 2650ssize_t AudioFlinger::MixerThread::threadLoop_write() 2651{ 2652 // FIXME we should only do one push per cycle; confirm this is true 2653 // Start the fast mixer if it's not already running 2654 if (mFastMixer != NULL) { 2655 FastMixerStateQueue *sq = mFastMixer->sq(); 2656 FastMixerState *state = sq->begin(); 2657 if (state->mCommand != FastMixerState::MIX_WRITE && 2658 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2659 if (state->mCommand == FastMixerState::COLD_IDLE) { 2660 int32_t old = android_atomic_inc(&mFastMixerFutex); 2661 if (old == -1) { 2662 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2663 } 2664#ifdef AUDIO_WATCHDOG 2665 if (mAudioWatchdog != 0) { 2666 mAudioWatchdog->resume(); 2667 } 2668#endif 2669 } 2670 state->mCommand = FastMixerState::MIX_WRITE; 2671 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2672 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2673 sq->end(); 2674 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2675 if (kUseFastMixer == FastMixer_Dynamic) { 2676 mNormalSink = mPipeSink; 2677 } 2678 } else { 2679 sq->end(false /*didModify*/); 2680 } 2681 } 2682 return PlaybackThread::threadLoop_write(); 2683} 2684 2685void AudioFlinger::MixerThread::threadLoop_standby() 2686{ 2687 // Idle the fast mixer if it's currently running 2688 if (mFastMixer != NULL) { 2689 FastMixerStateQueue *sq = mFastMixer->sq(); 2690 FastMixerState *state = sq->begin(); 2691 if (!(state->mCommand & FastMixerState::IDLE)) { 2692 state->mCommand = FastMixerState::COLD_IDLE; 2693 state->mColdFutexAddr = &mFastMixerFutex; 2694 state->mColdGen++; 2695 mFastMixerFutex = 0; 2696 sq->end(); 2697 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2698 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2699 if (kUseFastMixer == FastMixer_Dynamic) { 2700 mNormalSink = mOutputSink; 2701 } 2702#ifdef AUDIO_WATCHDOG 2703 if (mAudioWatchdog != 0) { 2704 mAudioWatchdog->pause(); 2705 } 2706#endif 2707 } else { 2708 sq->end(false /*didModify*/); 2709 } 2710 } 2711 PlaybackThread::threadLoop_standby(); 2712} 2713 2714// Empty implementation for standard mixer 2715// Overridden for offloaded playback 2716void AudioFlinger::PlaybackThread::flushOutput_l() 2717{ 2718} 2719 2720bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2721{ 2722 return false; 2723} 2724 2725bool AudioFlinger::PlaybackThread::shouldStandby_l() 2726{ 2727 return !mStandby; 2728} 2729 2730bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2731{ 2732 Mutex::Autolock _l(mLock); 2733 return waitingAsyncCallback_l(); 2734} 2735 2736// shared by MIXER and DIRECT, overridden by DUPLICATING 2737void AudioFlinger::PlaybackThread::threadLoop_standby() 2738{ 2739 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2740 mOutput->stream->common.standby(&mOutput->stream->common); 2741 if (mUseAsyncWrite != 0) { 2742 // discard any pending drain or write ack by incrementing sequence 2743 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2744 mDrainSequence = (mDrainSequence + 2) & ~1; 2745 ALOG_ASSERT(mCallbackThread != 0); 2746 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2747 mCallbackThread->setDraining(mDrainSequence); 2748 } 2749} 2750 2751void AudioFlinger::MixerThread::threadLoop_mix() 2752{ 2753 // obtain the presentation timestamp of the next output buffer 2754 int64_t pts; 2755 status_t status = INVALID_OPERATION; 2756 2757 if (mNormalSink != 0) { 2758 status = mNormalSink->getNextWriteTimestamp(&pts); 2759 } else { 2760 status = mOutputSink->getNextWriteTimestamp(&pts); 2761 } 2762 2763 if (status != NO_ERROR) { 2764 pts = AudioBufferProvider::kInvalidPTS; 2765 } 2766 2767 // mix buffers... 2768 mAudioMixer->process(pts); 2769 mCurrentWriteLength = mixBufferSize; 2770 // increase sleep time progressively when application underrun condition clears. 2771 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2772 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2773 // such that we would underrun the audio HAL. 2774 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2775 sleepTimeShift--; 2776 } 2777 sleepTime = 0; 2778 standbyTime = systemTime() + standbyDelay; 2779 //TODO: delay standby when effects have a tail 2780} 2781 2782void AudioFlinger::MixerThread::threadLoop_sleepTime() 2783{ 2784 // If no tracks are ready, sleep once for the duration of an output 2785 // buffer size, then write 0s to the output 2786 if (sleepTime == 0) { 2787 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2788 sleepTime = activeSleepTime >> sleepTimeShift; 2789 if (sleepTime < kMinThreadSleepTimeUs) { 2790 sleepTime = kMinThreadSleepTimeUs; 2791 } 2792 // reduce sleep time in case of consecutive application underruns to avoid 2793 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2794 // duration we would end up writing less data than needed by the audio HAL if 2795 // the condition persists. 2796 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2797 sleepTimeShift++; 2798 } 2799 } else { 2800 sleepTime = idleSleepTime; 2801 } 2802 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2803 memset (mMixBuffer, 0, mixBufferSize); 2804 sleepTime = 0; 2805 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2806 "anticipated start"); 2807 } 2808 // TODO add standby time extension fct of effect tail 2809} 2810 2811// prepareTracks_l() must be called with ThreadBase::mLock held 2812AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2813 Vector< sp<Track> > *tracksToRemove) 2814{ 2815 2816 mixer_state mixerStatus = MIXER_IDLE; 2817 // find out which tracks need to be processed 2818 size_t count = mActiveTracks.size(); 2819 size_t mixedTracks = 0; 2820 size_t tracksWithEffect = 0; 2821 // counts only _active_ fast tracks 2822 size_t fastTracks = 0; 2823 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2824 2825 float masterVolume = mMasterVolume; 2826 bool masterMute = mMasterMute; 2827 2828 if (masterMute) { 2829 masterVolume = 0; 2830 } 2831 // Delegate master volume control to effect in output mix effect chain if needed 2832 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2833 if (chain != 0) { 2834 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2835 chain->setVolume_l(&v, &v); 2836 masterVolume = (float)((v + (1 << 23)) >> 24); 2837 chain.clear(); 2838 } 2839 2840 // prepare a new state to push 2841 FastMixerStateQueue *sq = NULL; 2842 FastMixerState *state = NULL; 2843 bool didModify = false; 2844 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2845 if (mFastMixer != NULL) { 2846 sq = mFastMixer->sq(); 2847 state = sq->begin(); 2848 } 2849 2850 for (size_t i=0 ; i<count ; i++) { 2851 const sp<Track> t = mActiveTracks[i].promote(); 2852 if (t == 0) { 2853 continue; 2854 } 2855 2856 // this const just means the local variable doesn't change 2857 Track* const track = t.get(); 2858 2859 // process fast tracks 2860 if (track->isFastTrack()) { 2861 2862 // It's theoretically possible (though unlikely) for a fast track to be created 2863 // and then removed within the same normal mix cycle. This is not a problem, as 2864 // the track never becomes active so it's fast mixer slot is never touched. 2865 // The converse, of removing an (active) track and then creating a new track 2866 // at the identical fast mixer slot within the same normal mix cycle, 2867 // is impossible because the slot isn't marked available until the end of each cycle. 2868 int j = track->mFastIndex; 2869 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2870 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2871 FastTrack *fastTrack = &state->mFastTracks[j]; 2872 2873 // Determine whether the track is currently in underrun condition, 2874 // and whether it had a recent underrun. 2875 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2876 FastTrackUnderruns underruns = ftDump->mUnderruns; 2877 uint32_t recentFull = (underruns.mBitFields.mFull - 2878 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2879 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2880 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2881 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2882 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2883 uint32_t recentUnderruns = recentPartial + recentEmpty; 2884 track->mObservedUnderruns = underruns; 2885 // don't count underruns that occur while stopping or pausing 2886 // or stopped which can occur when flush() is called while active 2887 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2888 recentUnderruns > 0) { 2889 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2890 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2891 } 2892 2893 // This is similar to the state machine for normal tracks, 2894 // with a few modifications for fast tracks. 2895 bool isActive = true; 2896 switch (track->mState) { 2897 case TrackBase::STOPPING_1: 2898 // track stays active in STOPPING_1 state until first underrun 2899 if (recentUnderruns > 0 || track->isTerminated()) { 2900 track->mState = TrackBase::STOPPING_2; 2901 } 2902 break; 2903 case TrackBase::PAUSING: 2904 // ramp down is not yet implemented 2905 track->setPaused(); 2906 break; 2907 case TrackBase::RESUMING: 2908 // ramp up is not yet implemented 2909 track->mState = TrackBase::ACTIVE; 2910 break; 2911 case TrackBase::ACTIVE: 2912 if (recentFull > 0 || recentPartial > 0) { 2913 // track has provided at least some frames recently: reset retry count 2914 track->mRetryCount = kMaxTrackRetries; 2915 } 2916 if (recentUnderruns == 0) { 2917 // no recent underruns: stay active 2918 break; 2919 } 2920 // there has recently been an underrun of some kind 2921 if (track->sharedBuffer() == 0) { 2922 // were any of the recent underruns "empty" (no frames available)? 2923 if (recentEmpty == 0) { 2924 // no, then ignore the partial underruns as they are allowed indefinitely 2925 break; 2926 } 2927 // there has recently been an "empty" underrun: decrement the retry counter 2928 if (--(track->mRetryCount) > 0) { 2929 break; 2930 } 2931 // indicate to client process that the track was disabled because of underrun; 2932 // it will then automatically call start() when data is available 2933 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2934 // remove from active list, but state remains ACTIVE [confusing but true] 2935 isActive = false; 2936 break; 2937 } 2938 // fall through 2939 case TrackBase::STOPPING_2: 2940 case TrackBase::PAUSED: 2941 case TrackBase::STOPPED: 2942 case TrackBase::FLUSHED: // flush() while active 2943 // Check for presentation complete if track is inactive 2944 // We have consumed all the buffers of this track. 2945 // This would be incomplete if we auto-paused on underrun 2946 { 2947 size_t audioHALFrames = 2948 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2949 size_t framesWritten = mBytesWritten / mFrameSize; 2950 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2951 // track stays in active list until presentation is complete 2952 break; 2953 } 2954 } 2955 if (track->isStopping_2()) { 2956 track->mState = TrackBase::STOPPED; 2957 } 2958 if (track->isStopped()) { 2959 // Can't reset directly, as fast mixer is still polling this track 2960 // track->reset(); 2961 // So instead mark this track as needing to be reset after push with ack 2962 resetMask |= 1 << i; 2963 } 2964 isActive = false; 2965 break; 2966 case TrackBase::IDLE: 2967 default: 2968 LOG_FATAL("unexpected track state %d", track->mState); 2969 } 2970 2971 if (isActive) { 2972 // was it previously inactive? 2973 if (!(state->mTrackMask & (1 << j))) { 2974 ExtendedAudioBufferProvider *eabp = track; 2975 VolumeProvider *vp = track; 2976 fastTrack->mBufferProvider = eabp; 2977 fastTrack->mVolumeProvider = vp; 2978 fastTrack->mSampleRate = track->mSampleRate; 2979 fastTrack->mChannelMask = track->mChannelMask; 2980 fastTrack->mGeneration++; 2981 state->mTrackMask |= 1 << j; 2982 didModify = true; 2983 // no acknowledgement required for newly active tracks 2984 } 2985 // cache the combined master volume and stream type volume for fast mixer; this 2986 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2987 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2988 ++fastTracks; 2989 } else { 2990 // was it previously active? 2991 if (state->mTrackMask & (1 << j)) { 2992 fastTrack->mBufferProvider = NULL; 2993 fastTrack->mGeneration++; 2994 state->mTrackMask &= ~(1 << j); 2995 didModify = true; 2996 // If any fast tracks were removed, we must wait for acknowledgement 2997 // because we're about to decrement the last sp<> on those tracks. 2998 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2999 } else { 3000 LOG_FATAL("fast track %d should have been active", j); 3001 } 3002 tracksToRemove->add(track); 3003 // Avoids a misleading display in dumpsys 3004 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3005 } 3006 continue; 3007 } 3008 3009 { // local variable scope to avoid goto warning 3010 3011 audio_track_cblk_t* cblk = track->cblk(); 3012 3013 // The first time a track is added we wait 3014 // for all its buffers to be filled before processing it 3015 int name = track->name(); 3016 // make sure that we have enough frames to mix one full buffer. 3017 // enforce this condition only once to enable draining the buffer in case the client 3018 // app does not call stop() and relies on underrun to stop: 3019 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3020 // during last round 3021 size_t desiredFrames; 3022 uint32_t sr = track->sampleRate(); 3023 if (sr == mSampleRate) { 3024 desiredFrames = mNormalFrameCount; 3025 } else { 3026 // +1 for rounding and +1 for additional sample needed for interpolation 3027 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3028 // add frames already consumed but not yet released by the resampler 3029 // because cblk->framesReady() will include these frames 3030 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3031 // the minimum track buffer size is normally twice the number of frames necessary 3032 // to fill one buffer and the resampler should not leave more than one buffer worth 3033 // of unreleased frames after each pass, but just in case... 3034 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3035 } 3036 uint32_t minFrames = 1; 3037 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3038 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3039 minFrames = desiredFrames; 3040 } 3041 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 3042 size_t framesReady; 3043 if (track->sharedBuffer() == 0) { 3044 framesReady = track->framesReady(); 3045 } else if (track->isStopped()) { 3046 framesReady = 0; 3047 } else { 3048 framesReady = 1; 3049 } 3050 if ((framesReady >= minFrames) && track->isReady() && 3051 !track->isPaused() && !track->isTerminated()) 3052 { 3053 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3054 3055 mixedTracks++; 3056 3057 // track->mainBuffer() != mMixBuffer means there is an effect chain 3058 // connected to the track 3059 chain.clear(); 3060 if (track->mainBuffer() != mMixBuffer) { 3061 chain = getEffectChain_l(track->sessionId()); 3062 // Delegate volume control to effect in track effect chain if needed 3063 if (chain != 0) { 3064 tracksWithEffect++; 3065 } else { 3066 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3067 "session %d", 3068 name, track->sessionId()); 3069 } 3070 } 3071 3072 3073 int param = AudioMixer::VOLUME; 3074 if (track->mFillingUpStatus == Track::FS_FILLED) { 3075 // no ramp for the first volume setting 3076 track->mFillingUpStatus = Track::FS_ACTIVE; 3077 if (track->mState == TrackBase::RESUMING) { 3078 track->mState = TrackBase::ACTIVE; 3079 param = AudioMixer::RAMP_VOLUME; 3080 } 3081 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3082 // FIXME should not make a decision based on mServer 3083 } else if (cblk->mServer != 0) { 3084 // If the track is stopped before the first frame was mixed, 3085 // do not apply ramp 3086 param = AudioMixer::RAMP_VOLUME; 3087 } 3088 3089 // compute volume for this track 3090 uint32_t vl, vr, va; 3091 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3092 vl = vr = va = 0; 3093 if (track->isPausing()) { 3094 track->setPaused(); 3095 } 3096 } else { 3097 3098 // read original volumes with volume control 3099 float typeVolume = mStreamTypes[track->streamType()].volume; 3100 float v = masterVolume * typeVolume; 3101 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3102 uint32_t vlr = proxy->getVolumeLR(); 3103 vl = vlr & 0xFFFF; 3104 vr = vlr >> 16; 3105 // track volumes come from shared memory, so can't be trusted and must be clamped 3106 if (vl > MAX_GAIN_INT) { 3107 ALOGV("Track left volume out of range: %04X", vl); 3108 vl = MAX_GAIN_INT; 3109 } 3110 if (vr > MAX_GAIN_INT) { 3111 ALOGV("Track right volume out of range: %04X", vr); 3112 vr = MAX_GAIN_INT; 3113 } 3114 // now apply the master volume and stream type volume 3115 vl = (uint32_t)(v * vl) << 12; 3116 vr = (uint32_t)(v * vr) << 12; 3117 // assuming master volume and stream type volume each go up to 1.0, 3118 // vl and vr are now in 8.24 format 3119 3120 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3121 // send level comes from shared memory and so may be corrupt 3122 if (sendLevel > MAX_GAIN_INT) { 3123 ALOGV("Track send level out of range: %04X", sendLevel); 3124 sendLevel = MAX_GAIN_INT; 3125 } 3126 va = (uint32_t)(v * sendLevel); 3127 } 3128 3129 // Delegate volume control to effect in track effect chain if needed 3130 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3131 // Do not ramp volume if volume is controlled by effect 3132 param = AudioMixer::VOLUME; 3133 track->mHasVolumeController = true; 3134 } else { 3135 // force no volume ramp when volume controller was just disabled or removed 3136 // from effect chain to avoid volume spike 3137 if (track->mHasVolumeController) { 3138 param = AudioMixer::VOLUME; 3139 } 3140 track->mHasVolumeController = false; 3141 } 3142 3143 // Convert volumes from 8.24 to 4.12 format 3144 // This additional clamping is needed in case chain->setVolume_l() overshot 3145 vl = (vl + (1 << 11)) >> 12; 3146 if (vl > MAX_GAIN_INT) { 3147 vl = MAX_GAIN_INT; 3148 } 3149 vr = (vr + (1 << 11)) >> 12; 3150 if (vr > MAX_GAIN_INT) { 3151 vr = MAX_GAIN_INT; 3152 } 3153 3154 if (va > MAX_GAIN_INT) { 3155 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3156 } 3157 3158 // XXX: these things DON'T need to be done each time 3159 mAudioMixer->setBufferProvider(name, track); 3160 mAudioMixer->enable(name); 3161 3162 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3163 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3164 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3165 mAudioMixer->setParameter( 3166 name, 3167 AudioMixer::TRACK, 3168 AudioMixer::FORMAT, (void *)track->format()); 3169 mAudioMixer->setParameter( 3170 name, 3171 AudioMixer::TRACK, 3172 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3173 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3174 uint32_t maxSampleRate = mSampleRate * 2; 3175 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3176 if (reqSampleRate == 0) { 3177 reqSampleRate = mSampleRate; 3178 } else if (reqSampleRate > maxSampleRate) { 3179 reqSampleRate = maxSampleRate; 3180 } 3181 mAudioMixer->setParameter( 3182 name, 3183 AudioMixer::RESAMPLE, 3184 AudioMixer::SAMPLE_RATE, 3185 (void *)reqSampleRate); 3186 mAudioMixer->setParameter( 3187 name, 3188 AudioMixer::TRACK, 3189 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3190 mAudioMixer->setParameter( 3191 name, 3192 AudioMixer::TRACK, 3193 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3194 3195 // reset retry count 3196 track->mRetryCount = kMaxTrackRetries; 3197 3198 // If one track is ready, set the mixer ready if: 3199 // - the mixer was not ready during previous round OR 3200 // - no other track is not ready 3201 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3202 mixerStatus != MIXER_TRACKS_ENABLED) { 3203 mixerStatus = MIXER_TRACKS_READY; 3204 } 3205 } else { 3206 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3207 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3208 } 3209 // clear effect chain input buffer if an active track underruns to avoid sending 3210 // previous audio buffer again to effects 3211 chain = getEffectChain_l(track->sessionId()); 3212 if (chain != 0) { 3213 chain->clearInputBuffer(); 3214 } 3215 3216 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3217 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3218 track->isStopped() || track->isPaused()) { 3219 // We have consumed all the buffers of this track. 3220 // Remove it from the list of active tracks. 3221 // TODO: use actual buffer filling status instead of latency when available from 3222 // audio HAL 3223 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3224 size_t framesWritten = mBytesWritten / mFrameSize; 3225 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3226 if (track->isStopped()) { 3227 track->reset(); 3228 } 3229 tracksToRemove->add(track); 3230 } 3231 } else { 3232 // No buffers for this track. Give it a few chances to 3233 // fill a buffer, then remove it from active list. 3234 if (--(track->mRetryCount) <= 0) { 3235 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3236 tracksToRemove->add(track); 3237 // indicate to client process that the track was disabled because of underrun; 3238 // it will then automatically call start() when data is available 3239 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3240 // If one track is not ready, mark the mixer also not ready if: 3241 // - the mixer was ready during previous round OR 3242 // - no other track is ready 3243 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3244 mixerStatus != MIXER_TRACKS_READY) { 3245 mixerStatus = MIXER_TRACKS_ENABLED; 3246 } 3247 } 3248 mAudioMixer->disable(name); 3249 } 3250 3251 } // local variable scope to avoid goto warning 3252track_is_ready: ; 3253 3254 } 3255 3256 // Push the new FastMixer state if necessary 3257 bool pauseAudioWatchdog = false; 3258 if (didModify) { 3259 state->mFastTracksGen++; 3260 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3261 if (kUseFastMixer == FastMixer_Dynamic && 3262 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3263 state->mCommand = FastMixerState::COLD_IDLE; 3264 state->mColdFutexAddr = &mFastMixerFutex; 3265 state->mColdGen++; 3266 mFastMixerFutex = 0; 3267 if (kUseFastMixer == FastMixer_Dynamic) { 3268 mNormalSink = mOutputSink; 3269 } 3270 // If we go into cold idle, need to wait for acknowledgement 3271 // so that fast mixer stops doing I/O. 3272 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3273 pauseAudioWatchdog = true; 3274 } 3275 } 3276 if (sq != NULL) { 3277 sq->end(didModify); 3278 sq->push(block); 3279 } 3280#ifdef AUDIO_WATCHDOG 3281 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3282 mAudioWatchdog->pause(); 3283 } 3284#endif 3285 3286 // Now perform the deferred reset on fast tracks that have stopped 3287 while (resetMask != 0) { 3288 size_t i = __builtin_ctz(resetMask); 3289 ALOG_ASSERT(i < count); 3290 resetMask &= ~(1 << i); 3291 sp<Track> t = mActiveTracks[i].promote(); 3292 if (t == 0) { 3293 continue; 3294 } 3295 Track* track = t.get(); 3296 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3297 track->reset(); 3298 } 3299 3300 // remove all the tracks that need to be... 3301 removeTracks_l(*tracksToRemove); 3302 3303 // mix buffer must be cleared if all tracks are connected to an 3304 // effect chain as in this case the mixer will not write to 3305 // mix buffer and track effects will accumulate into it 3306 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3307 (mixedTracks == 0 && fastTracks > 0))) { 3308 // FIXME as a performance optimization, should remember previous zero status 3309 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3310 } 3311 3312 // if any fast tracks, then status is ready 3313 mMixerStatusIgnoringFastTracks = mixerStatus; 3314 if (fastTracks > 0) { 3315 mixerStatus = MIXER_TRACKS_READY; 3316 } 3317 return mixerStatus; 3318} 3319 3320// getTrackName_l() must be called with ThreadBase::mLock held 3321int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3322{ 3323 return mAudioMixer->getTrackName(channelMask, sessionId); 3324} 3325 3326// deleteTrackName_l() must be called with ThreadBase::mLock held 3327void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3328{ 3329 ALOGV("remove track (%d) and delete from mixer", name); 3330 mAudioMixer->deleteTrackName(name); 3331} 3332 3333// checkForNewParameters_l() must be called with ThreadBase::mLock held 3334bool AudioFlinger::MixerThread::checkForNewParameters_l() 3335{ 3336 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3337 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3338 bool reconfig = false; 3339 3340 while (!mNewParameters.isEmpty()) { 3341 3342 if (mFastMixer != NULL) { 3343 FastMixerStateQueue *sq = mFastMixer->sq(); 3344 FastMixerState *state = sq->begin(); 3345 if (!(state->mCommand & FastMixerState::IDLE)) { 3346 previousCommand = state->mCommand; 3347 state->mCommand = FastMixerState::HOT_IDLE; 3348 sq->end(); 3349 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3350 } else { 3351 sq->end(false /*didModify*/); 3352 } 3353 } 3354 3355 status_t status = NO_ERROR; 3356 String8 keyValuePair = mNewParameters[0]; 3357 AudioParameter param = AudioParameter(keyValuePair); 3358 int value; 3359 3360 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3361 reconfig = true; 3362 } 3363 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3364 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3365 status = BAD_VALUE; 3366 } else { 3367 reconfig = true; 3368 } 3369 } 3370 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3371 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3372 status = BAD_VALUE; 3373 } else { 3374 reconfig = true; 3375 } 3376 } 3377 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3378 // do not accept frame count changes if tracks are open as the track buffer 3379 // size depends on frame count and correct behavior would not be guaranteed 3380 // if frame count is changed after track creation 3381 if (!mTracks.isEmpty()) { 3382 status = INVALID_OPERATION; 3383 } else { 3384 reconfig = true; 3385 } 3386 } 3387 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3388#ifdef ADD_BATTERY_DATA 3389 // when changing the audio output device, call addBatteryData to notify 3390 // the change 3391 if (mOutDevice != value) { 3392 uint32_t params = 0; 3393 // check whether speaker is on 3394 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3395 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3396 } 3397 3398 audio_devices_t deviceWithoutSpeaker 3399 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3400 // check if any other device (except speaker) is on 3401 if (value & deviceWithoutSpeaker ) { 3402 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3403 } 3404 3405 if (params != 0) { 3406 addBatteryData(params); 3407 } 3408 } 3409#endif 3410 3411 // forward device change to effects that have requested to be 3412 // aware of attached audio device. 3413 if (value != AUDIO_DEVICE_NONE) { 3414 mOutDevice = value; 3415 for (size_t i = 0; i < mEffectChains.size(); i++) { 3416 mEffectChains[i]->setDevice_l(mOutDevice); 3417 } 3418 } 3419 } 3420 3421 if (status == NO_ERROR) { 3422 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3423 keyValuePair.string()); 3424 if (!mStandby && status == INVALID_OPERATION) { 3425 mOutput->stream->common.standby(&mOutput->stream->common); 3426 mStandby = true; 3427 mBytesWritten = 0; 3428 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3429 keyValuePair.string()); 3430 } 3431 if (status == NO_ERROR && reconfig) { 3432 readOutputParameters(); 3433 delete mAudioMixer; 3434 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3435 for (size_t i = 0; i < mTracks.size() ; i++) { 3436 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3437 if (name < 0) { 3438 break; 3439 } 3440 mTracks[i]->mName = name; 3441 } 3442 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3443 } 3444 } 3445 3446 mNewParameters.removeAt(0); 3447 3448 mParamStatus = status; 3449 mParamCond.signal(); 3450 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3451 // already timed out waiting for the status and will never signal the condition. 3452 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3453 } 3454 3455 if (!(previousCommand & FastMixerState::IDLE)) { 3456 ALOG_ASSERT(mFastMixer != NULL); 3457 FastMixerStateQueue *sq = mFastMixer->sq(); 3458 FastMixerState *state = sq->begin(); 3459 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3460 state->mCommand = previousCommand; 3461 sq->end(); 3462 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3463 } 3464 3465 return reconfig; 3466} 3467 3468 3469void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3470{ 3471 const size_t SIZE = 256; 3472 char buffer[SIZE]; 3473 String8 result; 3474 3475 PlaybackThread::dumpInternals(fd, args); 3476 3477 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3478 result.append(buffer); 3479 write(fd, result.string(), result.size()); 3480 3481 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3482 const FastMixerDumpState copy(mFastMixerDumpState); 3483 copy.dump(fd); 3484 3485#ifdef STATE_QUEUE_DUMP 3486 // Similar for state queue 3487 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3488 observerCopy.dump(fd); 3489 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3490 mutatorCopy.dump(fd); 3491#endif 3492 3493#ifdef TEE_SINK 3494 // Write the tee output to a .wav file 3495 dumpTee(fd, mTeeSource, mId); 3496#endif 3497 3498#ifdef AUDIO_WATCHDOG 3499 if (mAudioWatchdog != 0) { 3500 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3501 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3502 wdCopy.dump(fd); 3503 } 3504#endif 3505} 3506 3507uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3508{ 3509 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3510} 3511 3512uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3513{ 3514 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3515} 3516 3517void AudioFlinger::MixerThread::cacheParameters_l() 3518{ 3519 PlaybackThread::cacheParameters_l(); 3520 3521 // FIXME: Relaxed timing because of a certain device that can't meet latency 3522 // Should be reduced to 2x after the vendor fixes the driver issue 3523 // increase threshold again due to low power audio mode. The way this warning 3524 // threshold is calculated and its usefulness should be reconsidered anyway. 3525 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3526} 3527 3528// ---------------------------------------------------------------------------- 3529 3530AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3531 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3532 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3533 // mLeftVolFloat, mRightVolFloat 3534{ 3535} 3536 3537AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3538 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3539 ThreadBase::type_t type) 3540 : PlaybackThread(audioFlinger, output, id, device, type) 3541 // mLeftVolFloat, mRightVolFloat 3542{ 3543} 3544 3545AudioFlinger::DirectOutputThread::~DirectOutputThread() 3546{ 3547} 3548 3549void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3550{ 3551 audio_track_cblk_t* cblk = track->cblk(); 3552 float left, right; 3553 3554 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3555 left = right = 0; 3556 } else { 3557 float typeVolume = mStreamTypes[track->streamType()].volume; 3558 float v = mMasterVolume * typeVolume; 3559 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3560 uint32_t vlr = proxy->getVolumeLR(); 3561 float v_clamped = v * (vlr & 0xFFFF); 3562 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3563 left = v_clamped/MAX_GAIN; 3564 v_clamped = v * (vlr >> 16); 3565 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3566 right = v_clamped/MAX_GAIN; 3567 } 3568 3569 if (lastTrack) { 3570 if (left != mLeftVolFloat || right != mRightVolFloat) { 3571 mLeftVolFloat = left; 3572 mRightVolFloat = right; 3573 3574 // Convert volumes from float to 8.24 3575 uint32_t vl = (uint32_t)(left * (1 << 24)); 3576 uint32_t vr = (uint32_t)(right * (1 << 24)); 3577 3578 // Delegate volume control to effect in track effect chain if needed 3579 // only one effect chain can be present on DirectOutputThread, so if 3580 // there is one, the track is connected to it 3581 if (!mEffectChains.isEmpty()) { 3582 mEffectChains[0]->setVolume_l(&vl, &vr); 3583 left = (float)vl / (1 << 24); 3584 right = (float)vr / (1 << 24); 3585 } 3586 if (mOutput->stream->set_volume) { 3587 mOutput->stream->set_volume(mOutput->stream, left, right); 3588 } 3589 } 3590 } 3591} 3592 3593 3594AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3595 Vector< sp<Track> > *tracksToRemove 3596) 3597{ 3598 size_t count = mActiveTracks.size(); 3599 mixer_state mixerStatus = MIXER_IDLE; 3600 3601 // find out which tracks need to be processed 3602 for (size_t i = 0; i < count; i++) { 3603 sp<Track> t = mActiveTracks[i].promote(); 3604 // The track died recently 3605 if (t == 0) { 3606 continue; 3607 } 3608 3609 Track* const track = t.get(); 3610 audio_track_cblk_t* cblk = track->cblk(); 3611 // Only consider last track started for volume and mixer state control. 3612 // In theory an older track could underrun and restart after the new one starts 3613 // but as we only care about the transition phase between two tracks on a 3614 // direct output, it is not a problem to ignore the underrun case. 3615 sp<Track> l = mLatestActiveTrack.promote(); 3616 bool last = l.get() == track; 3617 3618 // The first time a track is added we wait 3619 // for all its buffers to be filled before processing it 3620 uint32_t minFrames; 3621 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3622 minFrames = mNormalFrameCount; 3623 } else { 3624 minFrames = 1; 3625 } 3626 3627 if ((track->framesReady() >= minFrames) && track->isReady() && 3628 !track->isPaused() && !track->isTerminated()) 3629 { 3630 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3631 3632 if (track->mFillingUpStatus == Track::FS_FILLED) { 3633 track->mFillingUpStatus = Track::FS_ACTIVE; 3634 // make sure processVolume_l() will apply new volume even if 0 3635 mLeftVolFloat = mRightVolFloat = -1.0; 3636 if (track->mState == TrackBase::RESUMING) { 3637 track->mState = TrackBase::ACTIVE; 3638 } 3639 } 3640 3641 // compute volume for this track 3642 processVolume_l(track, last); 3643 if (last) { 3644 // reset retry count 3645 track->mRetryCount = kMaxTrackRetriesDirect; 3646 mActiveTrack = t; 3647 mixerStatus = MIXER_TRACKS_READY; 3648 } 3649 } else { 3650 // clear effect chain input buffer if the last active track started underruns 3651 // to avoid sending previous audio buffer again to effects 3652 if (!mEffectChains.isEmpty() && last) { 3653 mEffectChains[0]->clearInputBuffer(); 3654 } 3655 3656 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3657 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3658 track->isStopped() || track->isPaused()) { 3659 // We have consumed all the buffers of this track. 3660 // Remove it from the list of active tracks. 3661 // TODO: implement behavior for compressed audio 3662 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3663 size_t framesWritten = mBytesWritten / mFrameSize; 3664 if (mStandby || !last || 3665 track->presentationComplete(framesWritten, audioHALFrames)) { 3666 if (track->isStopped()) { 3667 track->reset(); 3668 } 3669 tracksToRemove->add(track); 3670 } 3671 } else { 3672 // No buffers for this track. Give it a few chances to 3673 // fill a buffer, then remove it from active list. 3674 // Only consider last track started for mixer state control 3675 if (--(track->mRetryCount) <= 0) { 3676 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3677 tracksToRemove->add(track); 3678 // indicate to client process that the track was disabled because of underrun; 3679 // it will then automatically call start() when data is available 3680 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3681 } else if (last) { 3682 mixerStatus = MIXER_TRACKS_ENABLED; 3683 } 3684 } 3685 } 3686 } 3687 3688 // remove all the tracks that need to be... 3689 removeTracks_l(*tracksToRemove); 3690 3691 return mixerStatus; 3692} 3693 3694void AudioFlinger::DirectOutputThread::threadLoop_mix() 3695{ 3696 size_t frameCount = mFrameCount; 3697 int8_t *curBuf = (int8_t *)mMixBuffer; 3698 // output audio to hardware 3699 while (frameCount) { 3700 AudioBufferProvider::Buffer buffer; 3701 buffer.frameCount = frameCount; 3702 mActiveTrack->getNextBuffer(&buffer); 3703 if (buffer.raw == NULL) { 3704 memset(curBuf, 0, frameCount * mFrameSize); 3705 break; 3706 } 3707 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3708 frameCount -= buffer.frameCount; 3709 curBuf += buffer.frameCount * mFrameSize; 3710 mActiveTrack->releaseBuffer(&buffer); 3711 } 3712 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3713 sleepTime = 0; 3714 standbyTime = systemTime() + standbyDelay; 3715 mActiveTrack.clear(); 3716} 3717 3718void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3719{ 3720 if (sleepTime == 0) { 3721 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3722 sleepTime = activeSleepTime; 3723 } else { 3724 sleepTime = idleSleepTime; 3725 } 3726 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3727 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3728 sleepTime = 0; 3729 } 3730} 3731 3732// getTrackName_l() must be called with ThreadBase::mLock held 3733int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3734 int sessionId) 3735{ 3736 return 0; 3737} 3738 3739// deleteTrackName_l() must be called with ThreadBase::mLock held 3740void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3741{ 3742} 3743 3744// checkForNewParameters_l() must be called with ThreadBase::mLock held 3745bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3746{ 3747 bool reconfig = false; 3748 3749 while (!mNewParameters.isEmpty()) { 3750 status_t status = NO_ERROR; 3751 String8 keyValuePair = mNewParameters[0]; 3752 AudioParameter param = AudioParameter(keyValuePair); 3753 int value; 3754 3755 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3756 // do not accept frame count changes if tracks are open as the track buffer 3757 // size depends on frame count and correct behavior would not be garantied 3758 // if frame count is changed after track creation 3759 if (!mTracks.isEmpty()) { 3760 status = INVALID_OPERATION; 3761 } else { 3762 reconfig = true; 3763 } 3764 } 3765 if (status == NO_ERROR) { 3766 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3767 keyValuePair.string()); 3768 if (!mStandby && status == INVALID_OPERATION) { 3769 mOutput->stream->common.standby(&mOutput->stream->common); 3770 mStandby = true; 3771 mBytesWritten = 0; 3772 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3773 keyValuePair.string()); 3774 } 3775 if (status == NO_ERROR && reconfig) { 3776 readOutputParameters(); 3777 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3778 } 3779 } 3780 3781 mNewParameters.removeAt(0); 3782 3783 mParamStatus = status; 3784 mParamCond.signal(); 3785 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3786 // already timed out waiting for the status and will never signal the condition. 3787 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3788 } 3789 return reconfig; 3790} 3791 3792uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3793{ 3794 uint32_t time; 3795 if (audio_is_linear_pcm(mFormat)) { 3796 time = PlaybackThread::activeSleepTimeUs(); 3797 } else { 3798 time = 10000; 3799 } 3800 return time; 3801} 3802 3803uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3804{ 3805 uint32_t time; 3806 if (audio_is_linear_pcm(mFormat)) { 3807 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3808 } else { 3809 time = 10000; 3810 } 3811 return time; 3812} 3813 3814uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3815{ 3816 uint32_t time; 3817 if (audio_is_linear_pcm(mFormat)) { 3818 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3819 } else { 3820 time = 10000; 3821 } 3822 return time; 3823} 3824 3825void AudioFlinger::DirectOutputThread::cacheParameters_l() 3826{ 3827 PlaybackThread::cacheParameters_l(); 3828 3829 // use shorter standby delay as on normal output to release 3830 // hardware resources as soon as possible 3831 if (audio_is_linear_pcm(mFormat)) { 3832 standbyDelay = microseconds(activeSleepTime*2); 3833 } else { 3834 standbyDelay = kOffloadStandbyDelayNs; 3835 } 3836} 3837 3838// ---------------------------------------------------------------------------- 3839 3840AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3841 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3842 : Thread(false /*canCallJava*/), 3843 mPlaybackThread(playbackThread), 3844 mWriteAckSequence(0), 3845 mDrainSequence(0) 3846{ 3847} 3848 3849AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3850{ 3851} 3852 3853void AudioFlinger::AsyncCallbackThread::onFirstRef() 3854{ 3855 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3856} 3857 3858bool AudioFlinger::AsyncCallbackThread::threadLoop() 3859{ 3860 while (!exitPending()) { 3861 uint32_t writeAckSequence; 3862 uint32_t drainSequence; 3863 3864 { 3865 Mutex::Autolock _l(mLock); 3866 while (!((mWriteAckSequence & 1) || 3867 (mDrainSequence & 1) || 3868 exitPending())) { 3869 mWaitWorkCV.wait(mLock); 3870 } 3871 3872 if (exitPending()) { 3873 break; 3874 } 3875 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3876 mWriteAckSequence, mDrainSequence); 3877 writeAckSequence = mWriteAckSequence; 3878 mWriteAckSequence &= ~1; 3879 drainSequence = mDrainSequence; 3880 mDrainSequence &= ~1; 3881 } 3882 { 3883 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3884 if (playbackThread != 0) { 3885 if (writeAckSequence & 1) { 3886 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3887 } 3888 if (drainSequence & 1) { 3889 playbackThread->resetDraining(drainSequence >> 1); 3890 } 3891 } 3892 } 3893 } 3894 return false; 3895} 3896 3897void AudioFlinger::AsyncCallbackThread::exit() 3898{ 3899 ALOGV("AsyncCallbackThread::exit"); 3900 Mutex::Autolock _l(mLock); 3901 requestExit(); 3902 mWaitWorkCV.broadcast(); 3903} 3904 3905void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3906{ 3907 Mutex::Autolock _l(mLock); 3908 // bit 0 is cleared 3909 mWriteAckSequence = sequence << 1; 3910} 3911 3912void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3913{ 3914 Mutex::Autolock _l(mLock); 3915 // ignore unexpected callbacks 3916 if (mWriteAckSequence & 2) { 3917 mWriteAckSequence |= 1; 3918 mWaitWorkCV.signal(); 3919 } 3920} 3921 3922void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3923{ 3924 Mutex::Autolock _l(mLock); 3925 // bit 0 is cleared 3926 mDrainSequence = sequence << 1; 3927} 3928 3929void AudioFlinger::AsyncCallbackThread::resetDraining() 3930{ 3931 Mutex::Autolock _l(mLock); 3932 // ignore unexpected callbacks 3933 if (mDrainSequence & 2) { 3934 mDrainSequence |= 1; 3935 mWaitWorkCV.signal(); 3936 } 3937} 3938 3939 3940// ---------------------------------------------------------------------------- 3941AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3942 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3943 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3944 mHwPaused(false), 3945 mFlushPending(false), 3946 mPausedBytesRemaining(0) 3947{ 3948 //FIXME: mStandby should be set to true by ThreadBase constructor 3949 mStandby = true; 3950} 3951 3952void AudioFlinger::OffloadThread::threadLoop_exit() 3953{ 3954 if (mFlushPending || mHwPaused) { 3955 // If a flush is pending or track was paused, just discard buffered data 3956 flushHw_l(); 3957 } else { 3958 mMixerStatus = MIXER_DRAIN_ALL; 3959 threadLoop_drain(); 3960 } 3961 mCallbackThread->exit(); 3962 PlaybackThread::threadLoop_exit(); 3963} 3964 3965AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3966 Vector< sp<Track> > *tracksToRemove 3967) 3968{ 3969 size_t count = mActiveTracks.size(); 3970 3971 mixer_state mixerStatus = MIXER_IDLE; 3972 bool doHwPause = false; 3973 bool doHwResume = false; 3974 3975 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3976 3977 // find out which tracks need to be processed 3978 for (size_t i = 0; i < count; i++) { 3979 sp<Track> t = mActiveTracks[i].promote(); 3980 // The track died recently 3981 if (t == 0) { 3982 continue; 3983 } 3984 Track* const track = t.get(); 3985 audio_track_cblk_t* cblk = track->cblk(); 3986 // Only consider last track started for volume and mixer state control. 3987 // In theory an older track could underrun and restart after the new one starts 3988 // but as we only care about the transition phase between two tracks on a 3989 // direct output, it is not a problem to ignore the underrun case. 3990 sp<Track> l = mLatestActiveTrack.promote(); 3991 bool last = l.get() == track; 3992 3993 if (track->isPausing()) { 3994 track->setPaused(); 3995 if (last) { 3996 if (!mHwPaused) { 3997 doHwPause = true; 3998 mHwPaused = true; 3999 } 4000 // If we were part way through writing the mixbuffer to 4001 // the HAL we must save this until we resume 4002 // BUG - this will be wrong if a different track is made active, 4003 // in that case we want to discard the pending data in the 4004 // mixbuffer and tell the client to present it again when the 4005 // track is resumed 4006 mPausedWriteLength = mCurrentWriteLength; 4007 mPausedBytesRemaining = mBytesRemaining; 4008 mBytesRemaining = 0; // stop writing 4009 } 4010 tracksToRemove->add(track); 4011 } else if (track->framesReady() && track->isReady() && 4012 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4013 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4014 if (track->mFillingUpStatus == Track::FS_FILLED) { 4015 track->mFillingUpStatus = Track::FS_ACTIVE; 4016 // make sure processVolume_l() will apply new volume even if 0 4017 mLeftVolFloat = mRightVolFloat = -1.0; 4018 if (track->mState == TrackBase::RESUMING) { 4019 track->mState = TrackBase::ACTIVE; 4020 if (last) { 4021 if (mPausedBytesRemaining) { 4022 // Need to continue write that was interrupted 4023 mCurrentWriteLength = mPausedWriteLength; 4024 mBytesRemaining = mPausedBytesRemaining; 4025 mPausedBytesRemaining = 0; 4026 } 4027 if (mHwPaused) { 4028 doHwResume = true; 4029 mHwPaused = false; 4030 // threadLoop_mix() will handle the case that we need to 4031 // resume an interrupted write 4032 } 4033 // enable write to audio HAL 4034 sleepTime = 0; 4035 } 4036 } 4037 } 4038 4039 if (last) { 4040 sp<Track> previousTrack = mPreviousTrack.promote(); 4041 if (previousTrack != 0) { 4042 if (track != previousTrack.get()) { 4043 // Flush any data still being written from last track 4044 mBytesRemaining = 0; 4045 if (mPausedBytesRemaining) { 4046 // Last track was paused so we also need to flush saved 4047 // mixbuffer state and invalidate track so that it will 4048 // re-submit that unwritten data when it is next resumed 4049 mPausedBytesRemaining = 0; 4050 // Invalidate is a bit drastic - would be more efficient 4051 // to have a flag to tell client that some of the 4052 // previously written data was lost 4053 previousTrack->invalidate(); 4054 } 4055 // flush data already sent to the DSP if changing audio session as audio 4056 // comes from a different source. Also invalidate previous track to force a 4057 // seek when resuming. 4058 if (previousTrack->sessionId() != track->sessionId()) { 4059 previousTrack->invalidate(); 4060 mFlushPending = true; 4061 } 4062 } 4063 } 4064 mPreviousTrack = track; 4065 // reset retry count 4066 track->mRetryCount = kMaxTrackRetriesOffload; 4067 mActiveTrack = t; 4068 mixerStatus = MIXER_TRACKS_READY; 4069 } 4070 } else { 4071 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4072 if (track->isStopping_1()) { 4073 // Hardware buffer can hold a large amount of audio so we must 4074 // wait for all current track's data to drain before we say 4075 // that the track is stopped. 4076 if (mBytesRemaining == 0) { 4077 // Only start draining when all data in mixbuffer 4078 // has been written 4079 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4080 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4081 // do not drain if no data was ever sent to HAL (mStandby == true) 4082 if (last && !mStandby) { 4083 // do not modify drain sequence if we are already draining. This happens 4084 // when resuming from pause after drain. 4085 if ((mDrainSequence & 1) == 0) { 4086 sleepTime = 0; 4087 standbyTime = systemTime() + standbyDelay; 4088 mixerStatus = MIXER_DRAIN_TRACK; 4089 mDrainSequence += 2; 4090 } 4091 if (mHwPaused) { 4092 // It is possible to move from PAUSED to STOPPING_1 without 4093 // a resume so we must ensure hardware is running 4094 doHwResume = true; 4095 mHwPaused = false; 4096 } 4097 } 4098 } 4099 } else if (track->isStopping_2()) { 4100 // Drain has completed or we are in standby, signal presentation complete 4101 if (!(mDrainSequence & 1) || !last || mStandby) { 4102 track->mState = TrackBase::STOPPED; 4103 size_t audioHALFrames = 4104 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4105 size_t framesWritten = 4106 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4107 track->presentationComplete(framesWritten, audioHALFrames); 4108 track->reset(); 4109 tracksToRemove->add(track); 4110 } 4111 } else { 4112 // No buffers for this track. Give it a few chances to 4113 // fill a buffer, then remove it from active list. 4114 if (--(track->mRetryCount) <= 0) { 4115 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4116 track->name()); 4117 tracksToRemove->add(track); 4118 // indicate to client process that the track was disabled because of underrun; 4119 // it will then automatically call start() when data is available 4120 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4121 } else if (last){ 4122 mixerStatus = MIXER_TRACKS_ENABLED; 4123 } 4124 } 4125 } 4126 // compute volume for this track 4127 processVolume_l(track, last); 4128 } 4129 4130 // make sure the pause/flush/resume sequence is executed in the right order. 4131 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4132 // before flush and then resume HW. This can happen in case of pause/flush/resume 4133 // if resume is received before pause is executed. 4134 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4135 mOutput->stream->pause(mOutput->stream); 4136 if (!doHwPause) { 4137 doHwResume = true; 4138 } 4139 } 4140 if (mFlushPending) { 4141 flushHw_l(); 4142 mFlushPending = false; 4143 } 4144 if (!mStandby && doHwResume) { 4145 mOutput->stream->resume(mOutput->stream); 4146 } 4147 4148 // remove all the tracks that need to be... 4149 removeTracks_l(*tracksToRemove); 4150 4151 return mixerStatus; 4152} 4153 4154void AudioFlinger::OffloadThread::flushOutput_l() 4155{ 4156 mFlushPending = true; 4157} 4158 4159// must be called with thread mutex locked 4160bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4161{ 4162 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4163 mWriteAckSequence, mDrainSequence); 4164 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4165 return true; 4166 } 4167 return false; 4168} 4169 4170// must be called with thread mutex locked 4171bool AudioFlinger::OffloadThread::shouldStandby_l() 4172{ 4173 bool TrackPaused = false; 4174 4175 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4176 // after a timeout and we will enter standby then. 4177 if (mTracks.size() > 0) { 4178 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4179 } 4180 4181 return !mStandby && !TrackPaused; 4182} 4183 4184 4185bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4186{ 4187 Mutex::Autolock _l(mLock); 4188 return waitingAsyncCallback_l(); 4189} 4190 4191void AudioFlinger::OffloadThread::flushHw_l() 4192{ 4193 mOutput->stream->flush(mOutput->stream); 4194 // Flush anything still waiting in the mixbuffer 4195 mCurrentWriteLength = 0; 4196 mBytesRemaining = 0; 4197 mPausedWriteLength = 0; 4198 mPausedBytesRemaining = 0; 4199 if (mUseAsyncWrite) { 4200 // discard any pending drain or write ack by incrementing sequence 4201 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4202 mDrainSequence = (mDrainSequence + 2) & ~1; 4203 ALOG_ASSERT(mCallbackThread != 0); 4204 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4205 mCallbackThread->setDraining(mDrainSequence); 4206 } 4207} 4208 4209// ---------------------------------------------------------------------------- 4210 4211AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4212 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4213 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4214 DUPLICATING), 4215 mWaitTimeMs(UINT_MAX) 4216{ 4217 addOutputTrack(mainThread); 4218} 4219 4220AudioFlinger::DuplicatingThread::~DuplicatingThread() 4221{ 4222 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4223 mOutputTracks[i]->destroy(); 4224 } 4225} 4226 4227void AudioFlinger::DuplicatingThread::threadLoop_mix() 4228{ 4229 // mix buffers... 4230 if (outputsReady(outputTracks)) { 4231 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4232 } else { 4233 memset(mMixBuffer, 0, mixBufferSize); 4234 } 4235 sleepTime = 0; 4236 writeFrames = mNormalFrameCount; 4237 mCurrentWriteLength = mixBufferSize; 4238 standbyTime = systemTime() + standbyDelay; 4239} 4240 4241void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4242{ 4243 if (sleepTime == 0) { 4244 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4245 sleepTime = activeSleepTime; 4246 } else { 4247 sleepTime = idleSleepTime; 4248 } 4249 } else if (mBytesWritten != 0) { 4250 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4251 writeFrames = mNormalFrameCount; 4252 memset(mMixBuffer, 0, mixBufferSize); 4253 } else { 4254 // flush remaining overflow buffers in output tracks 4255 writeFrames = 0; 4256 } 4257 sleepTime = 0; 4258 } 4259} 4260 4261ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4262{ 4263 for (size_t i = 0; i < outputTracks.size(); i++) { 4264 outputTracks[i]->write(mMixBuffer, writeFrames); 4265 } 4266 mStandby = false; 4267 return (ssize_t)mixBufferSize; 4268} 4269 4270void AudioFlinger::DuplicatingThread::threadLoop_standby() 4271{ 4272 // DuplicatingThread implements standby by stopping all tracks 4273 for (size_t i = 0; i < outputTracks.size(); i++) { 4274 outputTracks[i]->stop(); 4275 } 4276} 4277 4278void AudioFlinger::DuplicatingThread::saveOutputTracks() 4279{ 4280 outputTracks = mOutputTracks; 4281} 4282 4283void AudioFlinger::DuplicatingThread::clearOutputTracks() 4284{ 4285 outputTracks.clear(); 4286} 4287 4288void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4289{ 4290 Mutex::Autolock _l(mLock); 4291 // FIXME explain this formula 4292 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4293 OutputTrack *outputTrack = new OutputTrack(thread, 4294 this, 4295 mSampleRate, 4296 mFormat, 4297 mChannelMask, 4298 frameCount, 4299 IPCThreadState::self()->getCallingUid()); 4300 if (outputTrack->cblk() != NULL) { 4301 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4302 mOutputTracks.add(outputTrack); 4303 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4304 updateWaitTime_l(); 4305 } 4306} 4307 4308void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4309{ 4310 Mutex::Autolock _l(mLock); 4311 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4312 if (mOutputTracks[i]->thread() == thread) { 4313 mOutputTracks[i]->destroy(); 4314 mOutputTracks.removeAt(i); 4315 updateWaitTime_l(); 4316 return; 4317 } 4318 } 4319 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4320} 4321 4322// caller must hold mLock 4323void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4324{ 4325 mWaitTimeMs = UINT_MAX; 4326 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4327 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4328 if (strong != 0) { 4329 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4330 if (waitTimeMs < mWaitTimeMs) { 4331 mWaitTimeMs = waitTimeMs; 4332 } 4333 } 4334 } 4335} 4336 4337 4338bool AudioFlinger::DuplicatingThread::outputsReady( 4339 const SortedVector< sp<OutputTrack> > &outputTracks) 4340{ 4341 for (size_t i = 0; i < outputTracks.size(); i++) { 4342 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4343 if (thread == 0) { 4344 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4345 outputTracks[i].get()); 4346 return false; 4347 } 4348 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4349 // see note at standby() declaration 4350 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4351 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4352 thread.get()); 4353 return false; 4354 } 4355 } 4356 return true; 4357} 4358 4359uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4360{ 4361 return (mWaitTimeMs * 1000) / 2; 4362} 4363 4364void AudioFlinger::DuplicatingThread::cacheParameters_l() 4365{ 4366 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4367 updateWaitTime_l(); 4368 4369 MixerThread::cacheParameters_l(); 4370} 4371 4372// ---------------------------------------------------------------------------- 4373// Record 4374// ---------------------------------------------------------------------------- 4375 4376AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4377 AudioStreamIn *input, 4378 uint32_t sampleRate, 4379 audio_channel_mask_t channelMask, 4380 audio_io_handle_t id, 4381 audio_devices_t outDevice, 4382 audio_devices_t inDevice 4383#ifdef TEE_SINK 4384 , const sp<NBAIO_Sink>& teeSink 4385#endif 4386 ) : 4387 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4388 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4389 // mRsmpInIndex and mBufferSize set by readInputParameters() 4390 mReqChannelCount(popcount(channelMask)), 4391 mReqSampleRate(sampleRate) 4392 // mBytesRead is only meaningful while active, and so is cleared in start() 4393 // (but might be better to also clear here for dump?) 4394#ifdef TEE_SINK 4395 , mTeeSink(teeSink) 4396#endif 4397{ 4398 snprintf(mName, kNameLength, "AudioIn_%X", id); 4399 4400 readInputParameters(); 4401} 4402 4403 4404AudioFlinger::RecordThread::~RecordThread() 4405{ 4406 delete[] mRsmpInBuffer; 4407 delete mResampler; 4408 delete[] mRsmpOutBuffer; 4409} 4410 4411void AudioFlinger::RecordThread::onFirstRef() 4412{ 4413 run(mName, PRIORITY_URGENT_AUDIO); 4414} 4415 4416status_t AudioFlinger::RecordThread::readyToRun() 4417{ 4418 status_t status = initCheck(); 4419 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4420 return status; 4421} 4422 4423bool AudioFlinger::RecordThread::threadLoop() 4424{ 4425 AudioBufferProvider::Buffer buffer; 4426 sp<RecordTrack> activeTrack; 4427 Vector< sp<EffectChain> > effectChains; 4428 4429 nsecs_t lastWarning = 0; 4430 4431 inputStandBy(); 4432 { 4433 Mutex::Autolock _l(mLock); 4434 activeTrack = mActiveTrack; 4435 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4436 } 4437 4438 // used to verify we've read at least once before evaluating how many bytes were read 4439 bool readOnce = false; 4440 4441 // start recording 4442 while (!exitPending()) { 4443 4444 processConfigEvents(); 4445 4446 { // scope for mLock 4447 Mutex::Autolock _l(mLock); 4448 checkForNewParameters_l(); 4449 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4450 SortedVector<int> tmp; 4451 tmp.add(mActiveTrack->uid()); 4452 updateWakeLockUids_l(tmp); 4453 } 4454 activeTrack = mActiveTrack; 4455 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4456 standby(); 4457 4458 if (exitPending()) { 4459 break; 4460 } 4461 4462 releaseWakeLock_l(); 4463 ALOGV("RecordThread: loop stopping"); 4464 // go to sleep 4465 mWaitWorkCV.wait(mLock); 4466 ALOGV("RecordThread: loop starting"); 4467 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4468 continue; 4469 } 4470 if (mActiveTrack != 0) { 4471 if (mActiveTrack->isTerminated()) { 4472 removeTrack_l(mActiveTrack); 4473 mActiveTrack.clear(); 4474 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4475 standby(); 4476 mActiveTrack.clear(); 4477 mStartStopCond.broadcast(); 4478 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4479 if (mReqChannelCount != mActiveTrack->channelCount()) { 4480 mActiveTrack.clear(); 4481 mStartStopCond.broadcast(); 4482 } else if (readOnce) { 4483 // record start succeeds only if first read from audio input 4484 // succeeds 4485 if (mBytesRead >= 0) { 4486 mActiveTrack->mState = TrackBase::ACTIVE; 4487 } else { 4488 mActiveTrack.clear(); 4489 } 4490 mStartStopCond.broadcast(); 4491 } 4492 mStandby = false; 4493 } 4494 } 4495 4496 lockEffectChains_l(effectChains); 4497 } 4498 4499 if (mActiveTrack != 0) { 4500 if (mActiveTrack->mState != TrackBase::ACTIVE && 4501 mActiveTrack->mState != TrackBase::RESUMING) { 4502 unlockEffectChains(effectChains); 4503 usleep(kRecordThreadSleepUs); 4504 continue; 4505 } 4506 for (size_t i = 0; i < effectChains.size(); i ++) { 4507 effectChains[i]->process_l(); 4508 } 4509 4510 buffer.frameCount = mFrameCount; 4511 status_t status = mActiveTrack->getNextBuffer(&buffer); 4512 if (status == NO_ERROR) { 4513 readOnce = true; 4514 size_t framesOut = buffer.frameCount; 4515 if (mResampler == NULL) { 4516 // no resampling 4517 while (framesOut) { 4518 size_t framesIn = mFrameCount - mRsmpInIndex; 4519 if (framesIn) { 4520 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4521 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4522 mActiveTrack->mFrameSize; 4523 if (framesIn > framesOut) 4524 framesIn = framesOut; 4525 mRsmpInIndex += framesIn; 4526 framesOut -= framesIn; 4527 if (mChannelCount == mReqChannelCount) { 4528 memcpy(dst, src, framesIn * mFrameSize); 4529 } else { 4530 if (mChannelCount == 1) { 4531 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4532 (int16_t *)src, framesIn); 4533 } else { 4534 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4535 (int16_t *)src, framesIn); 4536 } 4537 } 4538 } 4539 if (framesOut && mFrameCount == mRsmpInIndex) { 4540 void *readInto; 4541 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4542 readInto = buffer.raw; 4543 framesOut = 0; 4544 } else { 4545 readInto = mRsmpInBuffer; 4546 mRsmpInIndex = 0; 4547 } 4548 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4549 mBufferSize); 4550 if (mBytesRead <= 0) { 4551 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4552 { 4553 ALOGE("Error reading audio input"); 4554 // Force input into standby so that it tries to 4555 // recover at next read attempt 4556 inputStandBy(); 4557 usleep(kRecordThreadSleepUs); 4558 } 4559 mRsmpInIndex = mFrameCount; 4560 framesOut = 0; 4561 buffer.frameCount = 0; 4562 } 4563#ifdef TEE_SINK 4564 else if (mTeeSink != 0) { 4565 (void) mTeeSink->write(readInto, 4566 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4567 } 4568#endif 4569 } 4570 } 4571 } else { 4572 // resampling 4573 4574 // resampler accumulates, but we only have one source track 4575 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4576 // alter output frame count as if we were expecting stereo samples 4577 if (mChannelCount == 1 && mReqChannelCount == 1) { 4578 framesOut >>= 1; 4579 } 4580 mResampler->resample(mRsmpOutBuffer, framesOut, 4581 this /* AudioBufferProvider* */); 4582 // ditherAndClamp() works as long as all buffers returned by 4583 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4584 if (mChannelCount == 2 && mReqChannelCount == 1) { 4585 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4586 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4587 // the resampler always outputs stereo samples: 4588 // do post stereo to mono conversion 4589 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4590 framesOut); 4591 } else { 4592 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4593 } 4594 // now done with mRsmpOutBuffer 4595 4596 } 4597 if (mFramestoDrop == 0) { 4598 mActiveTrack->releaseBuffer(&buffer); 4599 } else { 4600 if (mFramestoDrop > 0) { 4601 mFramestoDrop -= buffer.frameCount; 4602 if (mFramestoDrop <= 0) { 4603 clearSyncStartEvent(); 4604 } 4605 } else { 4606 mFramestoDrop += buffer.frameCount; 4607 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4608 mSyncStartEvent->isCancelled()) { 4609 ALOGW("Synced record %s, session %d, trigger session %d", 4610 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4611 mActiveTrack->sessionId(), 4612 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4613 clearSyncStartEvent(); 4614 } 4615 } 4616 } 4617 mActiveTrack->clearOverflow(); 4618 } 4619 // client isn't retrieving buffers fast enough 4620 else { 4621 if (!mActiveTrack->setOverflow()) { 4622 nsecs_t now = systemTime(); 4623 if ((now - lastWarning) > kWarningThrottleNs) { 4624 ALOGW("RecordThread: buffer overflow"); 4625 lastWarning = now; 4626 } 4627 } 4628 // Release the processor for a while before asking for a new buffer. 4629 // This will give the application more chance to read from the buffer and 4630 // clear the overflow. 4631 usleep(kRecordThreadSleepUs); 4632 } 4633 } 4634 // enable changes in effect chain 4635 unlockEffectChains(effectChains); 4636 effectChains.clear(); 4637 } 4638 4639 standby(); 4640 4641 { 4642 Mutex::Autolock _l(mLock); 4643 for (size_t i = 0; i < mTracks.size(); i++) { 4644 sp<RecordTrack> track = mTracks[i]; 4645 track->invalidate(); 4646 } 4647 mActiveTrack.clear(); 4648 mStartStopCond.broadcast(); 4649 } 4650 4651 releaseWakeLock(); 4652 4653 ALOGV("RecordThread %p exiting", this); 4654 return false; 4655} 4656 4657void AudioFlinger::RecordThread::standby() 4658{ 4659 if (!mStandby) { 4660 inputStandBy(); 4661 mStandby = true; 4662 } 4663} 4664 4665void AudioFlinger::RecordThread::inputStandBy() 4666{ 4667 mInput->stream->common.standby(&mInput->stream->common); 4668} 4669 4670sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4671 const sp<AudioFlinger::Client>& client, 4672 uint32_t sampleRate, 4673 audio_format_t format, 4674 audio_channel_mask_t channelMask, 4675 size_t frameCount, 4676 int sessionId, 4677 int uid, 4678 IAudioFlinger::track_flags_t *flags, 4679 pid_t tid, 4680 status_t *status) 4681{ 4682 sp<RecordTrack> track; 4683 status_t lStatus; 4684 4685 lStatus = initCheck(); 4686 if (lStatus != NO_ERROR) { 4687 ALOGE("createRecordTrack_l() audio driver not initialized"); 4688 goto Exit; 4689 } 4690 // client expresses a preference for FAST, but we get the final say 4691 if (*flags & IAudioFlinger::TRACK_FAST) { 4692 if ( 4693 // use case: callback handler and frame count is default or at least as large as HAL 4694 ( 4695 (tid != -1) && 4696 ((frameCount == 0) || 4697 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4698 ) && 4699 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4700 // mono or stereo 4701 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4702 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4703 // hardware sample rate 4704 (sampleRate == mSampleRate) && 4705 // record thread has an associated fast recorder 4706 hasFastRecorder() 4707 // FIXME test that RecordThread for this fast track has a capable output HAL 4708 // FIXME add a permission test also? 4709 ) { 4710 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4711 if (frameCount == 0) { 4712 frameCount = mFrameCount * kFastTrackMultiplier; 4713 } 4714 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4715 frameCount, mFrameCount); 4716 } else { 4717 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4718 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4719 "hasFastRecorder=%d tid=%d", 4720 frameCount, mFrameCount, format, 4721 audio_is_linear_pcm(format), 4722 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4723 *flags &= ~IAudioFlinger::TRACK_FAST; 4724 // For compatibility with AudioRecord calculation, buffer depth is forced 4725 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4726 // This is probably too conservative, but legacy application code may depend on it. 4727 // If you change this calculation, also review the start threshold which is related. 4728 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4729 size_t mNormalFrameCount = 2048; // FIXME 4730 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4731 if (minBufCount < 2) { 4732 minBufCount = 2; 4733 } 4734 size_t minFrameCount = mNormalFrameCount * minBufCount; 4735 if (frameCount < minFrameCount) { 4736 frameCount = minFrameCount; 4737 } 4738 } 4739 } 4740 4741 // FIXME use flags and tid similar to createTrack_l() 4742 4743 { // scope for mLock 4744 Mutex::Autolock _l(mLock); 4745 4746 track = new RecordTrack(this, client, sampleRate, 4747 format, channelMask, frameCount, sessionId, uid); 4748 4749 if (track->getCblk() == 0) { 4750 ALOGE("createRecordTrack_l() no control block"); 4751 lStatus = NO_MEMORY; 4752 // track must be cleared from the caller as the caller has the AF lock 4753 goto Exit; 4754 } 4755 mTracks.add(track); 4756 4757 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4758 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4759 mAudioFlinger->btNrecIsOff(); 4760 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4761 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4762 4763 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4764 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4765 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4766 // so ask activity manager to do this on our behalf 4767 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4768 } 4769 } 4770 lStatus = NO_ERROR; 4771 4772Exit: 4773 if (status) { 4774 *status = lStatus; 4775 } 4776 return track; 4777} 4778 4779status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4780 AudioSystem::sync_event_t event, 4781 int triggerSession) 4782{ 4783 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4784 sp<ThreadBase> strongMe = this; 4785 status_t status = NO_ERROR; 4786 4787 if (event == AudioSystem::SYNC_EVENT_NONE) { 4788 clearSyncStartEvent(); 4789 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4790 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4791 triggerSession, 4792 recordTrack->sessionId(), 4793 syncStartEventCallback, 4794 this); 4795 // Sync event can be cancelled by the trigger session if the track is not in a 4796 // compatible state in which case we start record immediately 4797 if (mSyncStartEvent->isCancelled()) { 4798 clearSyncStartEvent(); 4799 } else { 4800 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4801 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4802 } 4803 } 4804 4805 { 4806 AutoMutex lock(mLock); 4807 if (mActiveTrack != 0) { 4808 if (recordTrack != mActiveTrack.get()) { 4809 status = -EBUSY; 4810 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4811 mActiveTrack->mState = TrackBase::ACTIVE; 4812 } 4813 return status; 4814 } 4815 4816 recordTrack->mState = TrackBase::IDLE; 4817 mActiveTrack = recordTrack; 4818 mLock.unlock(); 4819 status_t status = AudioSystem::startInput(mId); 4820 mLock.lock(); 4821 if (status != NO_ERROR) { 4822 mActiveTrack.clear(); 4823 clearSyncStartEvent(); 4824 return status; 4825 } 4826 mRsmpInIndex = mFrameCount; 4827 mBytesRead = 0; 4828 if (mResampler != NULL) { 4829 mResampler->reset(); 4830 } 4831 mActiveTrack->mState = TrackBase::RESUMING; 4832 // signal thread to start 4833 ALOGV("Signal record thread"); 4834 mWaitWorkCV.broadcast(); 4835 // do not wait for mStartStopCond if exiting 4836 if (exitPending()) { 4837 mActiveTrack.clear(); 4838 status = INVALID_OPERATION; 4839 goto startError; 4840 } 4841 mStartStopCond.wait(mLock); 4842 if (mActiveTrack == 0) { 4843 ALOGV("Record failed to start"); 4844 status = BAD_VALUE; 4845 goto startError; 4846 } 4847 ALOGV("Record started OK"); 4848 return status; 4849 } 4850 4851startError: 4852 AudioSystem::stopInput(mId); 4853 clearSyncStartEvent(); 4854 return status; 4855} 4856 4857void AudioFlinger::RecordThread::clearSyncStartEvent() 4858{ 4859 if (mSyncStartEvent != 0) { 4860 mSyncStartEvent->cancel(); 4861 } 4862 mSyncStartEvent.clear(); 4863 mFramestoDrop = 0; 4864} 4865 4866void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4867{ 4868 sp<SyncEvent> strongEvent = event.promote(); 4869 4870 if (strongEvent != 0) { 4871 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4872 me->handleSyncStartEvent(strongEvent); 4873 } 4874} 4875 4876void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4877{ 4878 if (event == mSyncStartEvent) { 4879 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4880 // from audio HAL 4881 mFramestoDrop = mFrameCount * 2; 4882 } 4883} 4884 4885bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4886 ALOGV("RecordThread::stop"); 4887 AutoMutex _l(mLock); 4888 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4889 return false; 4890 } 4891 recordTrack->mState = TrackBase::PAUSING; 4892 // do not wait for mStartStopCond if exiting 4893 if (exitPending()) { 4894 return true; 4895 } 4896 mStartStopCond.wait(mLock); 4897 // if we have been restarted, recordTrack == mActiveTrack.get() here 4898 if (exitPending() || recordTrack != mActiveTrack.get()) { 4899 ALOGV("Record stopped OK"); 4900 return true; 4901 } 4902 return false; 4903} 4904 4905bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4906{ 4907 return false; 4908} 4909 4910status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4911{ 4912#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4913 if (!isValidSyncEvent(event)) { 4914 return BAD_VALUE; 4915 } 4916 4917 int eventSession = event->triggerSession(); 4918 status_t ret = NAME_NOT_FOUND; 4919 4920 Mutex::Autolock _l(mLock); 4921 4922 for (size_t i = 0; i < mTracks.size(); i++) { 4923 sp<RecordTrack> track = mTracks[i]; 4924 if (eventSession == track->sessionId()) { 4925 (void) track->setSyncEvent(event); 4926 ret = NO_ERROR; 4927 } 4928 } 4929 return ret; 4930#else 4931 return BAD_VALUE; 4932#endif 4933} 4934 4935// destroyTrack_l() must be called with ThreadBase::mLock held 4936void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4937{ 4938 track->terminate(); 4939 track->mState = TrackBase::STOPPED; 4940 // active tracks are removed by threadLoop() 4941 if (mActiveTrack != track) { 4942 removeTrack_l(track); 4943 } 4944} 4945 4946void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4947{ 4948 mTracks.remove(track); 4949 // need anything related to effects here? 4950} 4951 4952void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4953{ 4954 dumpInternals(fd, args); 4955 dumpTracks(fd, args); 4956 dumpEffectChains(fd, args); 4957} 4958 4959void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4960{ 4961 const size_t SIZE = 256; 4962 char buffer[SIZE]; 4963 String8 result; 4964 4965 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4966 result.append(buffer); 4967 4968 if (mActiveTrack != 0) { 4969 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4970 result.append(buffer); 4971 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4972 result.append(buffer); 4973 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4974 result.append(buffer); 4975 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4976 result.append(buffer); 4977 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4978 result.append(buffer); 4979 } else { 4980 result.append("No active record client\n"); 4981 } 4982 4983 write(fd, result.string(), result.size()); 4984 4985 dumpBase(fd, args); 4986} 4987 4988void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4989{ 4990 const size_t SIZE = 256; 4991 char buffer[SIZE]; 4992 String8 result; 4993 4994 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4995 result.append(buffer); 4996 RecordTrack::appendDumpHeader(result); 4997 for (size_t i = 0; i < mTracks.size(); ++i) { 4998 sp<RecordTrack> track = mTracks[i]; 4999 if (track != 0) { 5000 track->dump(buffer, SIZE); 5001 result.append(buffer); 5002 } 5003 } 5004 5005 if (mActiveTrack != 0) { 5006 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5007 result.append(buffer); 5008 RecordTrack::appendDumpHeader(result); 5009 mActiveTrack->dump(buffer, SIZE); 5010 result.append(buffer); 5011 5012 } 5013 write(fd, result.string(), result.size()); 5014} 5015 5016// AudioBufferProvider interface 5017status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5018{ 5019 size_t framesReq = buffer->frameCount; 5020 size_t framesReady = mFrameCount - mRsmpInIndex; 5021 int channelCount; 5022 5023 if (framesReady == 0) { 5024 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 5025 if (mBytesRead <= 0) { 5026 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 5027 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5028 // Force input into standby so that it tries to 5029 // recover at next read attempt 5030 inputStandBy(); 5031 usleep(kRecordThreadSleepUs); 5032 } 5033 buffer->raw = NULL; 5034 buffer->frameCount = 0; 5035 return NOT_ENOUGH_DATA; 5036 } 5037 mRsmpInIndex = 0; 5038 framesReady = mFrameCount; 5039 } 5040 5041 if (framesReq > framesReady) { 5042 framesReq = framesReady; 5043 } 5044 5045 if (mChannelCount == 1 && mReqChannelCount == 2) { 5046 channelCount = 1; 5047 } else { 5048 channelCount = 2; 5049 } 5050 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5051 buffer->frameCount = framesReq; 5052 return NO_ERROR; 5053} 5054 5055// AudioBufferProvider interface 5056void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5057{ 5058 mRsmpInIndex += buffer->frameCount; 5059 buffer->frameCount = 0; 5060} 5061 5062bool AudioFlinger::RecordThread::checkForNewParameters_l() 5063{ 5064 bool reconfig = false; 5065 5066 while (!mNewParameters.isEmpty()) { 5067 status_t status = NO_ERROR; 5068 String8 keyValuePair = mNewParameters[0]; 5069 AudioParameter param = AudioParameter(keyValuePair); 5070 int value; 5071 audio_format_t reqFormat = mFormat; 5072 uint32_t reqSamplingRate = mReqSampleRate; 5073 uint32_t reqChannelCount = mReqChannelCount; 5074 5075 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5076 reqSamplingRate = value; 5077 reconfig = true; 5078 } 5079 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5080 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5081 status = BAD_VALUE; 5082 } else { 5083 reqFormat = (audio_format_t) value; 5084 reconfig = true; 5085 } 5086 } 5087 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5088 reqChannelCount = popcount(value); 5089 reconfig = true; 5090 } 5091 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5092 // do not accept frame count changes if tracks are open as the track buffer 5093 // size depends on frame count and correct behavior would not be guaranteed 5094 // if frame count is changed after track creation 5095 if (mActiveTrack != 0) { 5096 status = INVALID_OPERATION; 5097 } else { 5098 reconfig = true; 5099 } 5100 } 5101 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5102 // forward device change to effects that have requested to be 5103 // aware of attached audio device. 5104 for (size_t i = 0; i < mEffectChains.size(); i++) { 5105 mEffectChains[i]->setDevice_l(value); 5106 } 5107 5108 // store input device and output device but do not forward output device to audio HAL. 5109 // Note that status is ignored by the caller for output device 5110 // (see AudioFlinger::setParameters() 5111 if (audio_is_output_devices(value)) { 5112 mOutDevice = value; 5113 status = BAD_VALUE; 5114 } else { 5115 mInDevice = value; 5116 // disable AEC and NS if the device is a BT SCO headset supporting those 5117 // pre processings 5118 if (mTracks.size() > 0) { 5119 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5120 mAudioFlinger->btNrecIsOff(); 5121 for (size_t i = 0; i < mTracks.size(); i++) { 5122 sp<RecordTrack> track = mTracks[i]; 5123 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5124 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5125 } 5126 } 5127 } 5128 } 5129 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5130 mAudioSource != (audio_source_t)value) { 5131 // forward device change to effects that have requested to be 5132 // aware of attached audio device. 5133 for (size_t i = 0; i < mEffectChains.size(); i++) { 5134 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5135 } 5136 mAudioSource = (audio_source_t)value; 5137 } 5138 if (status == NO_ERROR) { 5139 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5140 keyValuePair.string()); 5141 if (status == INVALID_OPERATION) { 5142 inputStandBy(); 5143 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5144 keyValuePair.string()); 5145 } 5146 if (reconfig) { 5147 if (status == BAD_VALUE && 5148 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5149 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5150 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5151 <= (2 * reqSamplingRate)) && 5152 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5153 <= FCC_2 && 5154 (reqChannelCount <= FCC_2)) { 5155 status = NO_ERROR; 5156 } 5157 if (status == NO_ERROR) { 5158 readInputParameters(); 5159 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5160 } 5161 } 5162 } 5163 5164 mNewParameters.removeAt(0); 5165 5166 mParamStatus = status; 5167 mParamCond.signal(); 5168 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5169 // already timed out waiting for the status and will never signal the condition. 5170 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5171 } 5172 return reconfig; 5173} 5174 5175String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5176{ 5177 Mutex::Autolock _l(mLock); 5178 if (initCheck() != NO_ERROR) { 5179 return String8(); 5180 } 5181 5182 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5183 const String8 out_s8(s); 5184 free(s); 5185 return out_s8; 5186} 5187 5188void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5189 AudioSystem::OutputDescriptor desc; 5190 void *param2 = NULL; 5191 5192 switch (event) { 5193 case AudioSystem::INPUT_OPENED: 5194 case AudioSystem::INPUT_CONFIG_CHANGED: 5195 desc.channelMask = mChannelMask; 5196 desc.samplingRate = mSampleRate; 5197 desc.format = mFormat; 5198 desc.frameCount = mFrameCount; 5199 desc.latency = 0; 5200 param2 = &desc; 5201 break; 5202 5203 case AudioSystem::INPUT_CLOSED: 5204 default: 5205 break; 5206 } 5207 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5208} 5209 5210void AudioFlinger::RecordThread::readInputParameters() 5211{ 5212 delete[] mRsmpInBuffer; 5213 // mRsmpInBuffer is always assigned a new[] below 5214 delete[] mRsmpOutBuffer; 5215 mRsmpOutBuffer = NULL; 5216 delete mResampler; 5217 mResampler = NULL; 5218 5219 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5220 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5221 mChannelCount = popcount(mChannelMask); 5222 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5223 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5224 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5225 } 5226 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5227 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5228 mFrameCount = mBufferSize / mFrameSize; 5229 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5230 5231 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5232 { 5233 int channelCount; 5234 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5235 // stereo to mono post process as the resampler always outputs stereo. 5236 if (mChannelCount == 1 && mReqChannelCount == 2) { 5237 channelCount = 1; 5238 } else { 5239 channelCount = 2; 5240 } 5241 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5242 mResampler->setSampleRate(mSampleRate); 5243 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5244 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5245 5246 // optmization: if mono to mono, alter input frame count as if we were inputing 5247 // stereo samples 5248 if (mChannelCount == 1 && mReqChannelCount == 1) { 5249 mFrameCount >>= 1; 5250 } 5251 5252 } 5253 mRsmpInIndex = mFrameCount; 5254} 5255 5256unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5257{ 5258 Mutex::Autolock _l(mLock); 5259 if (initCheck() != NO_ERROR) { 5260 return 0; 5261 } 5262 5263 return mInput->stream->get_input_frames_lost(mInput->stream); 5264} 5265 5266uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5267{ 5268 Mutex::Autolock _l(mLock); 5269 uint32_t result = 0; 5270 if (getEffectChain_l(sessionId) != 0) { 5271 result = EFFECT_SESSION; 5272 } 5273 5274 for (size_t i = 0; i < mTracks.size(); ++i) { 5275 if (sessionId == mTracks[i]->sessionId()) { 5276 result |= TRACK_SESSION; 5277 break; 5278 } 5279 } 5280 5281 return result; 5282} 5283 5284KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5285{ 5286 KeyedVector<int, bool> ids; 5287 Mutex::Autolock _l(mLock); 5288 for (size_t j = 0; j < mTracks.size(); ++j) { 5289 sp<RecordThread::RecordTrack> track = mTracks[j]; 5290 int sessionId = track->sessionId(); 5291 if (ids.indexOfKey(sessionId) < 0) { 5292 ids.add(sessionId, true); 5293 } 5294 } 5295 return ids; 5296} 5297 5298AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5299{ 5300 Mutex::Autolock _l(mLock); 5301 AudioStreamIn *input = mInput; 5302 mInput = NULL; 5303 return input; 5304} 5305 5306// this method must always be called either with ThreadBase mLock held or inside the thread loop 5307audio_stream_t* AudioFlinger::RecordThread::stream() const 5308{ 5309 if (mInput == NULL) { 5310 return NULL; 5311 } 5312 return &mInput->stream->common; 5313} 5314 5315status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5316{ 5317 // only one chain per input thread 5318 if (mEffectChains.size() != 0) { 5319 return INVALID_OPERATION; 5320 } 5321 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5322 5323 chain->setInBuffer(NULL); 5324 chain->setOutBuffer(NULL); 5325 5326 checkSuspendOnAddEffectChain_l(chain); 5327 5328 mEffectChains.add(chain); 5329 5330 return NO_ERROR; 5331} 5332 5333size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5334{ 5335 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5336 ALOGW_IF(mEffectChains.size() != 1, 5337 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5338 chain.get(), mEffectChains.size(), this); 5339 if (mEffectChains.size() == 1) { 5340 mEffectChains.removeAt(0); 5341 } 5342 return 0; 5343} 5344 5345}; // namespace android 5346