Threads.cpp revision 4c6a433d74d5ae8b9bc0557207e3ced43bf34a25
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title 189#ifndef DEBUG_CPU_USAGE 190 __unused 191#endif 192 ) { 193#ifdef DEBUG_CPU_USAGE 194 // get current thread's delta CPU time in wall clock ns 195 double wcNs; 196 bool valid = mCpuUsage.sampleAndEnable(wcNs); 197 198 // record sample for wall clock statistics 199 if (valid) { 200 mWcStats.sample(wcNs); 201 } 202 203 // get the current CPU number 204 int cpuNum = sched_getcpu(); 205 206 // get the current CPU frequency in kHz 207 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 208 209 // check if either CPU number or frequency changed 210 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 211 mCpuNum = cpuNum; 212 mCpukHz = cpukHz; 213 // ignore sample for purposes of cycles 214 valid = false; 215 } 216 217 // if no change in CPU number or frequency, then record sample for cycle statistics 218 if (valid && mCpukHz > 0) { 219 double cycles = wcNs * cpukHz * 0.000001; 220 mHzStats.sample(cycles); 221 } 222 223 unsigned n = mWcStats.n(); 224 // mCpuUsage.elapsed() is expensive, so don't call it every loop 225 if ((n & 127) == 1) { 226 long long elapsed = mCpuUsage.elapsed(); 227 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 228 double perLoop = elapsed / (double) n; 229 double perLoop100 = perLoop * 0.01; 230 double perLoop1k = perLoop * 0.001; 231 double mean = mWcStats.mean(); 232 double stddev = mWcStats.stddev(); 233 double minimum = mWcStats.minimum(); 234 double maximum = mWcStats.maximum(); 235 double meanCycles = mHzStats.mean(); 236 double stddevCycles = mHzStats.stddev(); 237 double minCycles = mHzStats.minimum(); 238 double maxCycles = mHzStats.maximum(); 239 mCpuUsage.resetElapsed(); 240 mWcStats.reset(); 241 mHzStats.reset(); 242 ALOGD("CPU usage for %s over past %.1f secs\n" 243 " (%u mixer loops at %.1f mean ms per loop):\n" 244 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 245 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 246 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 247 title.string(), 248 elapsed * .000000001, n, perLoop * .000001, 249 mean * .001, 250 stddev * .001, 251 minimum * .001, 252 maximum * .001, 253 mean / perLoop100, 254 stddev / perLoop100, 255 minimum / perLoop100, 256 maximum / perLoop100, 257 meanCycles / perLoop1k, 258 stddevCycles / perLoop1k, 259 minCycles / perLoop1k, 260 maxCycles / perLoop1k); 261 262 } 263 } 264#endif 265}; 266 267// ---------------------------------------------------------------------------- 268// ThreadBase 269// ---------------------------------------------------------------------------- 270 271AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 272 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 273 : Thread(false /*canCallJava*/), 274 mType(type), 275 mAudioFlinger(audioFlinger), 276 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 277 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 278 mParamStatus(NO_ERROR), 279 //FIXME: mStandby should be true here. Is this some kind of hack? 280 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 281 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 282 // mName will be set by concrete (non-virtual) subclass 283 mDeathRecipient(new PMDeathRecipient(this)) 284{ 285} 286 287AudioFlinger::ThreadBase::~ThreadBase() 288{ 289 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 290 for (size_t i = 0; i < mConfigEvents.size(); i++) { 291 delete mConfigEvents[i]; 292 } 293 mConfigEvents.clear(); 294 295 mParamCond.broadcast(); 296 // do not lock the mutex in destructor 297 releaseWakeLock_l(); 298 if (mPowerManager != 0) { 299 sp<IBinder> binder = mPowerManager->asBinder(); 300 binder->unlinkToDeath(mDeathRecipient); 301 } 302} 303 304status_t AudioFlinger::ThreadBase::readyToRun() 305{ 306 status_t status = initCheck(); 307 if (status == NO_ERROR) { 308 ALOGI("AudioFlinger's thread %p ready to run", this); 309 } else { 310 ALOGE("No working audio driver found."); 311 } 312 return status; 313} 314 315void AudioFlinger::ThreadBase::exit() 316{ 317 ALOGV("ThreadBase::exit"); 318 // do any cleanup required for exit to succeed 319 preExit(); 320 { 321 // This lock prevents the following race in thread (uniprocessor for illustration): 322 // if (!exitPending()) { 323 // // context switch from here to exit() 324 // // exit() calls requestExit(), what exitPending() observes 325 // // exit() calls signal(), which is dropped since no waiters 326 // // context switch back from exit() to here 327 // mWaitWorkCV.wait(...); 328 // // now thread is hung 329 // } 330 AutoMutex lock(mLock); 331 requestExit(); 332 mWaitWorkCV.broadcast(); 333 } 334 // When Thread::requestExitAndWait is made virtual and this method is renamed to 335 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 336 requestExitAndWait(); 337} 338 339status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 340{ 341 status_t status; 342 343 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 344 Mutex::Autolock _l(mLock); 345 346 mNewParameters.add(keyValuePairs); 347 mWaitWorkCV.signal(); 348 // wait condition with timeout in case the thread loop has exited 349 // before the request could be processed 350 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 351 status = mParamStatus; 352 mWaitWorkCV.signal(); 353 } else { 354 status = TIMED_OUT; 355 } 356 return status; 357} 358 359void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 360{ 361 Mutex::Autolock _l(mLock); 362 sendIoConfigEvent_l(event, param); 363} 364 365// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 366void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 367{ 368 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 369 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 370 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 371 param); 372 mWaitWorkCV.signal(); 373} 374 375// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 376void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 377{ 378 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 379 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 380 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 381 mConfigEvents.size(), pid, tid, prio); 382 mWaitWorkCV.signal(); 383} 384 385void AudioFlinger::ThreadBase::processConfigEvents() 386{ 387 Mutex::Autolock _l(mLock); 388 processConfigEvents_l(); 389} 390 391// post condition: mConfigEvents.isEmpty() 392void AudioFlinger::ThreadBase::processConfigEvents_l() 393{ 394 while (!mConfigEvents.isEmpty()) { 395 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 396 ConfigEvent *event = mConfigEvents[0]; 397 mConfigEvents.removeAt(0); 398 // release mLock before locking AudioFlinger mLock: lock order is always 399 // AudioFlinger then ThreadBase to avoid cross deadlock 400 mLock.unlock(); 401 switch (event->type()) { 402 case CFG_EVENT_PRIO: { 403 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 404 // FIXME Need to understand why this has be done asynchronously 405 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 406 true /*asynchronous*/); 407 if (err != 0) { 408 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 409 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 410 } 411 } break; 412 case CFG_EVENT_IO: { 413 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 414 { 415 Mutex::Autolock _l(mAudioFlinger->mLock); 416 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 417 } 418 } break; 419 default: 420 ALOGE("processConfigEvents() unknown event type %d", event->type()); 421 break; 422 } 423 delete event; 424 mLock.lock(); 425 } 426} 427 428void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 429{ 430 const size_t SIZE = 256; 431 char buffer[SIZE]; 432 String8 result; 433 434 bool locked = AudioFlinger::dumpTryLock(mLock); 435 if (!locked) { 436 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 437 write(fd, buffer, strlen(buffer)); 438 } 439 440 snprintf(buffer, SIZE, "io handle: %d\n", mId); 441 result.append(buffer); 442 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 443 result.append(buffer); 444 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 445 result.append(buffer); 446 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 447 result.append(buffer); 448 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 449 result.append(buffer); 450 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 451 result.append(buffer); 452 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 453 result.append(buffer); 454 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 455 result.append(buffer); 456 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 457 result.append(buffer); 458 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 459 result.append(buffer); 460 461 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 462 result.append(buffer); 463 result.append(" Index Command"); 464 for (size_t i = 0; i < mNewParameters.size(); ++i) { 465 snprintf(buffer, SIZE, "\n %02d ", i); 466 result.append(buffer); 467 result.append(mNewParameters[i]); 468 } 469 470 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 471 result.append(buffer); 472 for (size_t i = 0; i < mConfigEvents.size(); i++) { 473 mConfigEvents[i]->dump(buffer, SIZE); 474 result.append(buffer); 475 } 476 result.append("\n"); 477 478 write(fd, result.string(), result.size()); 479 480 if (locked) { 481 mLock.unlock(); 482 } 483} 484 485void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 486{ 487 const size_t SIZE = 256; 488 char buffer[SIZE]; 489 String8 result; 490 491 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 492 write(fd, buffer, strlen(buffer)); 493 494 for (size_t i = 0; i < mEffectChains.size(); ++i) { 495 sp<EffectChain> chain = mEffectChains[i]; 496 if (chain != 0) { 497 chain->dump(fd, args); 498 } 499 } 500} 501 502void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 503{ 504 Mutex::Autolock _l(mLock); 505 acquireWakeLock_l(uid); 506} 507 508String16 AudioFlinger::ThreadBase::getWakeLockTag() 509{ 510 switch (mType) { 511 case MIXER: 512 return String16("AudioMix"); 513 case DIRECT: 514 return String16("AudioDirectOut"); 515 case DUPLICATING: 516 return String16("AudioDup"); 517 case RECORD: 518 return String16("AudioIn"); 519 case OFFLOAD: 520 return String16("AudioOffload"); 521 default: 522 ALOG_ASSERT(false); 523 return String16("AudioUnknown"); 524 } 525} 526 527void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 528{ 529 getPowerManager_l(); 530 if (mPowerManager != 0) { 531 sp<IBinder> binder = new BBinder(); 532 status_t status; 533 if (uid >= 0) { 534 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 535 binder, 536 getWakeLockTag(), 537 String16("media"), 538 uid); 539 } else { 540 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 541 binder, 542 getWakeLockTag(), 543 String16("media")); 544 } 545 if (status == NO_ERROR) { 546 mWakeLockToken = binder; 547 } 548 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 549 } 550} 551 552void AudioFlinger::ThreadBase::releaseWakeLock() 553{ 554 Mutex::Autolock _l(mLock); 555 releaseWakeLock_l(); 556} 557 558void AudioFlinger::ThreadBase::releaseWakeLock_l() 559{ 560 if (mWakeLockToken != 0) { 561 ALOGV("releaseWakeLock_l() %s", mName); 562 if (mPowerManager != 0) { 563 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 564 } 565 mWakeLockToken.clear(); 566 } 567} 568 569void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 570 Mutex::Autolock _l(mLock); 571 updateWakeLockUids_l(uids); 572} 573 574void AudioFlinger::ThreadBase::getPowerManager_l() { 575 576 if (mPowerManager == 0) { 577 // use checkService() to avoid blocking if power service is not up yet 578 sp<IBinder> binder = 579 defaultServiceManager()->checkService(String16("power")); 580 if (binder == 0) { 581 ALOGW("Thread %s cannot connect to the power manager service", mName); 582 } else { 583 mPowerManager = interface_cast<IPowerManager>(binder); 584 binder->linkToDeath(mDeathRecipient); 585 } 586 } 587} 588 589void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 590 591 getPowerManager_l(); 592 if (mWakeLockToken == NULL) { 593 ALOGE("no wake lock to update!"); 594 return; 595 } 596 if (mPowerManager != 0) { 597 sp<IBinder> binder = new BBinder(); 598 status_t status; 599 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 600 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 601 } 602} 603 604void AudioFlinger::ThreadBase::clearPowerManager() 605{ 606 Mutex::Autolock _l(mLock); 607 releaseWakeLock_l(); 608 mPowerManager.clear(); 609} 610 611void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 612{ 613 sp<ThreadBase> thread = mThread.promote(); 614 if (thread != 0) { 615 thread->clearPowerManager(); 616 } 617 ALOGW("power manager service died !!!"); 618} 619 620void AudioFlinger::ThreadBase::setEffectSuspended( 621 const effect_uuid_t *type, bool suspend, int sessionId) 622{ 623 Mutex::Autolock _l(mLock); 624 setEffectSuspended_l(type, suspend, sessionId); 625} 626 627void AudioFlinger::ThreadBase::setEffectSuspended_l( 628 const effect_uuid_t *type, bool suspend, int sessionId) 629{ 630 sp<EffectChain> chain = getEffectChain_l(sessionId); 631 if (chain != 0) { 632 if (type != NULL) { 633 chain->setEffectSuspended_l(type, suspend); 634 } else { 635 chain->setEffectSuspendedAll_l(suspend); 636 } 637 } 638 639 updateSuspendedSessions_l(type, suspend, sessionId); 640} 641 642void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 643{ 644 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 645 if (index < 0) { 646 return; 647 } 648 649 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 650 mSuspendedSessions.valueAt(index); 651 652 for (size_t i = 0; i < sessionEffects.size(); i++) { 653 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 654 for (int j = 0; j < desc->mRefCount; j++) { 655 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 656 chain->setEffectSuspendedAll_l(true); 657 } else { 658 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 659 desc->mType.timeLow); 660 chain->setEffectSuspended_l(&desc->mType, true); 661 } 662 } 663 } 664} 665 666void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 667 bool suspend, 668 int sessionId) 669{ 670 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 671 672 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 673 674 if (suspend) { 675 if (index >= 0) { 676 sessionEffects = mSuspendedSessions.valueAt(index); 677 } else { 678 mSuspendedSessions.add(sessionId, sessionEffects); 679 } 680 } else { 681 if (index < 0) { 682 return; 683 } 684 sessionEffects = mSuspendedSessions.valueAt(index); 685 } 686 687 688 int key = EffectChain::kKeyForSuspendAll; 689 if (type != NULL) { 690 key = type->timeLow; 691 } 692 index = sessionEffects.indexOfKey(key); 693 694 sp<SuspendedSessionDesc> desc; 695 if (suspend) { 696 if (index >= 0) { 697 desc = sessionEffects.valueAt(index); 698 } else { 699 desc = new SuspendedSessionDesc(); 700 if (type != NULL) { 701 desc->mType = *type; 702 } 703 sessionEffects.add(key, desc); 704 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 705 } 706 desc->mRefCount++; 707 } else { 708 if (index < 0) { 709 return; 710 } 711 desc = sessionEffects.valueAt(index); 712 if (--desc->mRefCount == 0) { 713 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 714 sessionEffects.removeItemsAt(index); 715 if (sessionEffects.isEmpty()) { 716 ALOGV("updateSuspendedSessions_l() restore removing session %d", 717 sessionId); 718 mSuspendedSessions.removeItem(sessionId); 719 } 720 } 721 } 722 if (!sessionEffects.isEmpty()) { 723 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 724 } 725} 726 727void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 728 bool enabled, 729 int sessionId) 730{ 731 Mutex::Autolock _l(mLock); 732 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 733} 734 735void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 736 bool enabled, 737 int sessionId) 738{ 739 if (mType != RECORD) { 740 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 741 // another session. This gives the priority to well behaved effect control panels 742 // and applications not using global effects. 743 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 744 // global effects 745 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 746 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 747 } 748 } 749 750 sp<EffectChain> chain = getEffectChain_l(sessionId); 751 if (chain != 0) { 752 chain->checkSuspendOnEffectEnabled(effect, enabled); 753 } 754} 755 756// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 757sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 758 const sp<AudioFlinger::Client>& client, 759 const sp<IEffectClient>& effectClient, 760 int32_t priority, 761 int sessionId, 762 effect_descriptor_t *desc, 763 int *enabled, 764 status_t *status) 765{ 766 sp<EffectModule> effect; 767 sp<EffectHandle> handle; 768 status_t lStatus; 769 sp<EffectChain> chain; 770 bool chainCreated = false; 771 bool effectCreated = false; 772 bool effectRegistered = false; 773 774 lStatus = initCheck(); 775 if (lStatus != NO_ERROR) { 776 ALOGW("createEffect_l() Audio driver not initialized."); 777 goto Exit; 778 } 779 780 // Allow global effects only on offloaded and mixer threads 781 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 782 switch (mType) { 783 case MIXER: 784 case OFFLOAD: 785 break; 786 case DIRECT: 787 case DUPLICATING: 788 case RECORD: 789 default: 790 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 791 lStatus = BAD_VALUE; 792 goto Exit; 793 } 794 } 795 796 // Only Pre processor effects are allowed on input threads and only on input threads 797 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 798 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 799 desc->name, desc->flags, mType); 800 lStatus = BAD_VALUE; 801 goto Exit; 802 } 803 804 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 805 806 { // scope for mLock 807 Mutex::Autolock _l(mLock); 808 809 // check for existing effect chain with the requested audio session 810 chain = getEffectChain_l(sessionId); 811 if (chain == 0) { 812 // create a new chain for this session 813 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 814 chain = new EffectChain(this, sessionId); 815 addEffectChain_l(chain); 816 chain->setStrategy(getStrategyForSession_l(sessionId)); 817 chainCreated = true; 818 } else { 819 effect = chain->getEffectFromDesc_l(desc); 820 } 821 822 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 823 824 if (effect == 0) { 825 int id = mAudioFlinger->nextUniqueId(); 826 // Check CPU and memory usage 827 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 828 if (lStatus != NO_ERROR) { 829 goto Exit; 830 } 831 effectRegistered = true; 832 // create a new effect module if none present in the chain 833 effect = new EffectModule(this, chain, desc, id, sessionId); 834 lStatus = effect->status(); 835 if (lStatus != NO_ERROR) { 836 goto Exit; 837 } 838 effect->setOffloaded(mType == OFFLOAD, mId); 839 840 lStatus = chain->addEffect_l(effect); 841 if (lStatus != NO_ERROR) { 842 goto Exit; 843 } 844 effectCreated = true; 845 846 effect->setDevice(mOutDevice); 847 effect->setDevice(mInDevice); 848 effect->setMode(mAudioFlinger->getMode()); 849 effect->setAudioSource(mAudioSource); 850 } 851 // create effect handle and connect it to effect module 852 handle = new EffectHandle(effect, client, effectClient, priority); 853 lStatus = handle->initCheck(); 854 if (lStatus == OK) { 855 lStatus = effect->addHandle(handle.get()); 856 } 857 if (enabled != NULL) { 858 *enabled = (int)effect->isEnabled(); 859 } 860 } 861 862Exit: 863 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 864 Mutex::Autolock _l(mLock); 865 if (effectCreated) { 866 chain->removeEffect_l(effect); 867 } 868 if (effectRegistered) { 869 AudioSystem::unregisterEffect(effect->id()); 870 } 871 if (chainCreated) { 872 removeEffectChain_l(chain); 873 } 874 handle.clear(); 875 } 876 877 *status = lStatus; 878 return handle; 879} 880 881sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 882{ 883 Mutex::Autolock _l(mLock); 884 return getEffect_l(sessionId, effectId); 885} 886 887sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 888{ 889 sp<EffectChain> chain = getEffectChain_l(sessionId); 890 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 891} 892 893// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 894// PlaybackThread::mLock held 895status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 896{ 897 // check for existing effect chain with the requested audio session 898 int sessionId = effect->sessionId(); 899 sp<EffectChain> chain = getEffectChain_l(sessionId); 900 bool chainCreated = false; 901 902 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 903 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 904 this, effect->desc().name, effect->desc().flags); 905 906 if (chain == 0) { 907 // create a new chain for this session 908 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 909 chain = new EffectChain(this, sessionId); 910 addEffectChain_l(chain); 911 chain->setStrategy(getStrategyForSession_l(sessionId)); 912 chainCreated = true; 913 } 914 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 915 916 if (chain->getEffectFromId_l(effect->id()) != 0) { 917 ALOGW("addEffect_l() %p effect %s already present in chain %p", 918 this, effect->desc().name, chain.get()); 919 return BAD_VALUE; 920 } 921 922 effect->setOffloaded(mType == OFFLOAD, mId); 923 924 status_t status = chain->addEffect_l(effect); 925 if (status != NO_ERROR) { 926 if (chainCreated) { 927 removeEffectChain_l(chain); 928 } 929 return status; 930 } 931 932 effect->setDevice(mOutDevice); 933 effect->setDevice(mInDevice); 934 effect->setMode(mAudioFlinger->getMode()); 935 effect->setAudioSource(mAudioSource); 936 return NO_ERROR; 937} 938 939void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 940 941 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 942 effect_descriptor_t desc = effect->desc(); 943 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 944 detachAuxEffect_l(effect->id()); 945 } 946 947 sp<EffectChain> chain = effect->chain().promote(); 948 if (chain != 0) { 949 // remove effect chain if removing last effect 950 if (chain->removeEffect_l(effect) == 0) { 951 removeEffectChain_l(chain); 952 } 953 } else { 954 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 955 } 956} 957 958void AudioFlinger::ThreadBase::lockEffectChains_l( 959 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 960{ 961 effectChains = mEffectChains; 962 for (size_t i = 0; i < mEffectChains.size(); i++) { 963 mEffectChains[i]->lock(); 964 } 965} 966 967void AudioFlinger::ThreadBase::unlockEffectChains( 968 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 969{ 970 for (size_t i = 0; i < effectChains.size(); i++) { 971 effectChains[i]->unlock(); 972 } 973} 974 975sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 976{ 977 Mutex::Autolock _l(mLock); 978 return getEffectChain_l(sessionId); 979} 980 981sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 982{ 983 size_t size = mEffectChains.size(); 984 for (size_t i = 0; i < size; i++) { 985 if (mEffectChains[i]->sessionId() == sessionId) { 986 return mEffectChains[i]; 987 } 988 } 989 return 0; 990} 991 992void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 993{ 994 Mutex::Autolock _l(mLock); 995 size_t size = mEffectChains.size(); 996 for (size_t i = 0; i < size; i++) { 997 mEffectChains[i]->setMode_l(mode); 998 } 999} 1000 1001void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1002 EffectHandle *handle, 1003 bool unpinIfLast) { 1004 1005 Mutex::Autolock _l(mLock); 1006 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1007 // delete the effect module if removing last handle on it 1008 if (effect->removeHandle(handle) == 0) { 1009 if (!effect->isPinned() || unpinIfLast) { 1010 removeEffect_l(effect); 1011 AudioSystem::unregisterEffect(effect->id()); 1012 } 1013 } 1014} 1015 1016// ---------------------------------------------------------------------------- 1017// Playback 1018// ---------------------------------------------------------------------------- 1019 1020AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1021 AudioStreamOut* output, 1022 audio_io_handle_t id, 1023 audio_devices_t device, 1024 type_t type) 1025 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1026 mNormalFrameCount(0), mMixBuffer(NULL), 1027 mSuspended(0), mBytesWritten(0), 1028 mActiveTracksGeneration(0), 1029 // mStreamTypes[] initialized in constructor body 1030 mOutput(output), 1031 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1032 mMixerStatus(MIXER_IDLE), 1033 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1034 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1035 mBytesRemaining(0), 1036 mCurrentWriteLength(0), 1037 mUseAsyncWrite(false), 1038 mWriteAckSequence(0), 1039 mDrainSequence(0), 1040 mSignalPending(false), 1041 mScreenState(AudioFlinger::mScreenState), 1042 // index 0 is reserved for normal mixer's submix 1043 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1044 // mLatchD, mLatchQ, 1045 mLatchDValid(false), mLatchQValid(false) 1046{ 1047 snprintf(mName, kNameLength, "AudioOut_%X", id); 1048 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1049 1050 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1051 // it would be safer to explicitly pass initial masterVolume/masterMute as 1052 // parameter. 1053 // 1054 // If the HAL we are using has support for master volume or master mute, 1055 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1056 // and the mute set to false). 1057 mMasterVolume = audioFlinger->masterVolume_l(); 1058 mMasterMute = audioFlinger->masterMute_l(); 1059 if (mOutput && mOutput->audioHwDev) { 1060 if (mOutput->audioHwDev->canSetMasterVolume()) { 1061 mMasterVolume = 1.0; 1062 } 1063 1064 if (mOutput->audioHwDev->canSetMasterMute()) { 1065 mMasterMute = false; 1066 } 1067 } 1068 1069 readOutputParameters(); 1070 1071 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1072 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1073 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1074 stream = (audio_stream_type_t) (stream + 1)) { 1075 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1076 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1077 } 1078 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1079 // because mAudioFlinger doesn't have one to copy from 1080} 1081 1082AudioFlinger::PlaybackThread::~PlaybackThread() 1083{ 1084 mAudioFlinger->unregisterWriter(mNBLogWriter); 1085 delete[] mMixBuffer; 1086} 1087 1088void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1089{ 1090 dumpInternals(fd, args); 1091 dumpTracks(fd, args); 1092 dumpEffectChains(fd, args); 1093} 1094 1095void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1096{ 1097 const size_t SIZE = 256; 1098 char buffer[SIZE]; 1099 String8 result; 1100 1101 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1102 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1103 const stream_type_t *st = &mStreamTypes[i]; 1104 if (i > 0) { 1105 result.appendFormat(", "); 1106 } 1107 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1108 if (st->mute) { 1109 result.append("M"); 1110 } 1111 } 1112 result.append("\n"); 1113 write(fd, result.string(), result.length()); 1114 result.clear(); 1115 1116 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1117 result.append(buffer); 1118 Track::appendDumpHeader(result); 1119 for (size_t i = 0; i < mTracks.size(); ++i) { 1120 sp<Track> track = mTracks[i]; 1121 if (track != 0) { 1122 track->dump(buffer, SIZE); 1123 result.append(buffer); 1124 } 1125 } 1126 1127 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1128 result.append(buffer); 1129 Track::appendDumpHeader(result); 1130 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1131 sp<Track> track = mActiveTracks[i].promote(); 1132 if (track != 0) { 1133 track->dump(buffer, SIZE); 1134 result.append(buffer); 1135 } 1136 } 1137 write(fd, result.string(), result.size()); 1138 1139 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1140 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1141 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1142 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1143} 1144 1145void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1146{ 1147 const size_t SIZE = 256; 1148 char buffer[SIZE]; 1149 String8 result; 1150 1151 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1152 result.append(buffer); 1153 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1154 result.append(buffer); 1155 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1156 ns2ms(systemTime() - mLastWriteTime)); 1157 result.append(buffer); 1158 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1159 result.append(buffer); 1160 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1161 result.append(buffer); 1162 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1163 result.append(buffer); 1164 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1165 result.append(buffer); 1166 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1167 result.append(buffer); 1168 write(fd, result.string(), result.size()); 1169 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1170 1171 dumpBase(fd, args); 1172} 1173 1174// Thread virtuals 1175 1176void AudioFlinger::PlaybackThread::onFirstRef() 1177{ 1178 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1179} 1180 1181// ThreadBase virtuals 1182void AudioFlinger::PlaybackThread::preExit() 1183{ 1184 ALOGV(" preExit()"); 1185 // FIXME this is using hard-coded strings but in the future, this functionality will be 1186 // converted to use audio HAL extensions required to support tunneling 1187 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1188} 1189 1190// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1191sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1192 const sp<AudioFlinger::Client>& client, 1193 audio_stream_type_t streamType, 1194 uint32_t sampleRate, 1195 audio_format_t format, 1196 audio_channel_mask_t channelMask, 1197 size_t *pFrameCount, 1198 const sp<IMemory>& sharedBuffer, 1199 int sessionId, 1200 IAudioFlinger::track_flags_t *flags, 1201 pid_t tid, 1202 int uid, 1203 status_t *status) 1204{ 1205 size_t frameCount = *pFrameCount; 1206 sp<Track> track; 1207 status_t lStatus; 1208 1209 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1210 1211 // client expresses a preference for FAST, but we get the final say 1212 if (*flags & IAudioFlinger::TRACK_FAST) { 1213 if ( 1214 // not timed 1215 (!isTimed) && 1216 // either of these use cases: 1217 ( 1218 // use case 1: shared buffer with any frame count 1219 ( 1220 (sharedBuffer != 0) 1221 ) || 1222 // use case 2: callback handler and frame count is default or at least as large as HAL 1223 ( 1224 (tid != -1) && 1225 ((frameCount == 0) || 1226 (frameCount >= mFrameCount)) 1227 ) 1228 ) && 1229 // PCM data 1230 audio_is_linear_pcm(format) && 1231 // mono or stereo 1232 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1233 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1234#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1235 // hardware sample rate 1236 (sampleRate == mSampleRate) && 1237#endif 1238 // normal mixer has an associated fast mixer 1239 hasFastMixer() && 1240 // there are sufficient fast track slots available 1241 (mFastTrackAvailMask != 0) 1242 // FIXME test that MixerThread for this fast track has a capable output HAL 1243 // FIXME add a permission test also? 1244 ) { 1245 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1246 if (frameCount == 0) { 1247 frameCount = mFrameCount * kFastTrackMultiplier; 1248 } 1249 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1250 frameCount, mFrameCount); 1251 } else { 1252 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1253 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1254 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1255 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1256 audio_is_linear_pcm(format), 1257 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1258 *flags &= ~IAudioFlinger::TRACK_FAST; 1259 // For compatibility with AudioTrack calculation, buffer depth is forced 1260 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1261 // This is probably too conservative, but legacy application code may depend on it. 1262 // If you change this calculation, also review the start threshold which is related. 1263 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1264 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1265 if (minBufCount < 2) { 1266 minBufCount = 2; 1267 } 1268 size_t minFrameCount = mNormalFrameCount * minBufCount; 1269 if (frameCount < minFrameCount) { 1270 frameCount = minFrameCount; 1271 } 1272 } 1273 } 1274 *pFrameCount = frameCount; 1275 1276 if (mType == DIRECT) { 1277 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1278 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1279 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1280 "for output %p with format %d", 1281 sampleRate, format, channelMask, mOutput, mFormat); 1282 lStatus = BAD_VALUE; 1283 goto Exit; 1284 } 1285 } 1286 } else if (mType == OFFLOAD) { 1287 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1288 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1289 "for output %p with format %d", 1290 sampleRate, format, channelMask, mOutput, mFormat); 1291 lStatus = BAD_VALUE; 1292 goto Exit; 1293 } 1294 } else { 1295 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1296 ALOGE("createTrack_l() Bad parameter: format %d \"" 1297 "for output %p with format %d", 1298 format, mOutput, mFormat); 1299 lStatus = BAD_VALUE; 1300 goto Exit; 1301 } 1302 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1303 if (sampleRate > mSampleRate*2) { 1304 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1305 lStatus = BAD_VALUE; 1306 goto Exit; 1307 } 1308 } 1309 1310 lStatus = initCheck(); 1311 if (lStatus != NO_ERROR) { 1312 ALOGE("Audio driver not initialized."); 1313 goto Exit; 1314 } 1315 1316 { // scope for mLock 1317 Mutex::Autolock _l(mLock); 1318 1319 // all tracks in same audio session must share the same routing strategy otherwise 1320 // conflicts will happen when tracks are moved from one output to another by audio policy 1321 // manager 1322 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1323 for (size_t i = 0; i < mTracks.size(); ++i) { 1324 sp<Track> t = mTracks[i]; 1325 if (t != 0 && !t->isOutputTrack()) { 1326 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1327 if (sessionId == t->sessionId() && strategy != actual) { 1328 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1329 strategy, actual); 1330 lStatus = BAD_VALUE; 1331 goto Exit; 1332 } 1333 } 1334 } 1335 1336 if (!isTimed) { 1337 track = new Track(this, client, streamType, sampleRate, format, 1338 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1339 } else { 1340 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1341 channelMask, frameCount, sharedBuffer, sessionId, uid); 1342 } 1343 1344 // new Track always returns non-NULL, 1345 // but TimedTrack::create() is a factory that could fail by returning NULL 1346 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1347 if (lStatus != NO_ERROR) { 1348 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1349 // track must be cleared from the caller as the caller has the AF lock 1350 goto Exit; 1351 } 1352 1353 mTracks.add(track); 1354 1355 sp<EffectChain> chain = getEffectChain_l(sessionId); 1356 if (chain != 0) { 1357 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1358 track->setMainBuffer(chain->inBuffer()); 1359 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1360 chain->incTrackCnt(); 1361 } 1362 1363 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1364 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1365 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1366 // so ask activity manager to do this on our behalf 1367 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1368 } 1369 } 1370 1371 lStatus = NO_ERROR; 1372 1373Exit: 1374 *status = lStatus; 1375 return track; 1376} 1377 1378uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1379{ 1380 return latency; 1381} 1382 1383uint32_t AudioFlinger::PlaybackThread::latency() const 1384{ 1385 Mutex::Autolock _l(mLock); 1386 return latency_l(); 1387} 1388uint32_t AudioFlinger::PlaybackThread::latency_l() const 1389{ 1390 if (initCheck() == NO_ERROR) { 1391 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1392 } else { 1393 return 0; 1394 } 1395} 1396 1397void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1398{ 1399 Mutex::Autolock _l(mLock); 1400 // Don't apply master volume in SW if our HAL can do it for us. 1401 if (mOutput && mOutput->audioHwDev && 1402 mOutput->audioHwDev->canSetMasterVolume()) { 1403 mMasterVolume = 1.0; 1404 } else { 1405 mMasterVolume = value; 1406 } 1407} 1408 1409void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1410{ 1411 Mutex::Autolock _l(mLock); 1412 // Don't apply master mute in SW if our HAL can do it for us. 1413 if (mOutput && mOutput->audioHwDev && 1414 mOutput->audioHwDev->canSetMasterMute()) { 1415 mMasterMute = false; 1416 } else { 1417 mMasterMute = muted; 1418 } 1419} 1420 1421void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1422{ 1423 Mutex::Autolock _l(mLock); 1424 mStreamTypes[stream].volume = value; 1425 broadcast_l(); 1426} 1427 1428void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1429{ 1430 Mutex::Autolock _l(mLock); 1431 mStreamTypes[stream].mute = muted; 1432 broadcast_l(); 1433} 1434 1435float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1436{ 1437 Mutex::Autolock _l(mLock); 1438 return mStreamTypes[stream].volume; 1439} 1440 1441// addTrack_l() must be called with ThreadBase::mLock held 1442status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1443{ 1444 status_t status = ALREADY_EXISTS; 1445 1446 // set retry count for buffer fill 1447 track->mRetryCount = kMaxTrackStartupRetries; 1448 if (mActiveTracks.indexOf(track) < 0) { 1449 // the track is newly added, make sure it fills up all its 1450 // buffers before playing. This is to ensure the client will 1451 // effectively get the latency it requested. 1452 if (!track->isOutputTrack()) { 1453 TrackBase::track_state state = track->mState; 1454 mLock.unlock(); 1455 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1456 mLock.lock(); 1457 // abort track was stopped/paused while we released the lock 1458 if (state != track->mState) { 1459 if (status == NO_ERROR) { 1460 mLock.unlock(); 1461 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1462 mLock.lock(); 1463 } 1464 return INVALID_OPERATION; 1465 } 1466 // abort if start is rejected by audio policy manager 1467 if (status != NO_ERROR) { 1468 return PERMISSION_DENIED; 1469 } 1470#ifdef ADD_BATTERY_DATA 1471 // to track the speaker usage 1472 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1473#endif 1474 } 1475 1476 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1477 track->mResetDone = false; 1478 track->mPresentationCompleteFrames = 0; 1479 mActiveTracks.add(track); 1480 mWakeLockUids.add(track->uid()); 1481 mActiveTracksGeneration++; 1482 mLatestActiveTrack = track; 1483 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1484 if (chain != 0) { 1485 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1486 track->sessionId()); 1487 chain->incActiveTrackCnt(); 1488 } 1489 1490 status = NO_ERROR; 1491 } 1492 1493 onAddNewTrack_l(); 1494 return status; 1495} 1496 1497bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1498{ 1499 track->terminate(); 1500 // active tracks are removed by threadLoop() 1501 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1502 track->mState = TrackBase::STOPPED; 1503 if (!trackActive) { 1504 removeTrack_l(track); 1505 } else if (track->isFastTrack() || track->isOffloaded()) { 1506 track->mState = TrackBase::STOPPING_1; 1507 } 1508 1509 return trackActive; 1510} 1511 1512void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1513{ 1514 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1515 mTracks.remove(track); 1516 deleteTrackName_l(track->name()); 1517 // redundant as track is about to be destroyed, for dumpsys only 1518 track->mName = -1; 1519 if (track->isFastTrack()) { 1520 int index = track->mFastIndex; 1521 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1522 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1523 mFastTrackAvailMask |= 1 << index; 1524 // redundant as track is about to be destroyed, for dumpsys only 1525 track->mFastIndex = -1; 1526 } 1527 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1528 if (chain != 0) { 1529 chain->decTrackCnt(); 1530 } 1531} 1532 1533void AudioFlinger::PlaybackThread::broadcast_l() 1534{ 1535 // Thread could be blocked waiting for async 1536 // so signal it to handle state changes immediately 1537 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1538 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1539 mSignalPending = true; 1540 mWaitWorkCV.broadcast(); 1541} 1542 1543String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1544{ 1545 Mutex::Autolock _l(mLock); 1546 if (initCheck() != NO_ERROR) { 1547 return String8(); 1548 } 1549 1550 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1551 const String8 out_s8(s); 1552 free(s); 1553 return out_s8; 1554} 1555 1556// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1557void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1558 AudioSystem::OutputDescriptor desc; 1559 void *param2 = NULL; 1560 1561 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1562 param); 1563 1564 switch (event) { 1565 case AudioSystem::OUTPUT_OPENED: 1566 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1567 desc.channelMask = mChannelMask; 1568 desc.samplingRate = mSampleRate; 1569 desc.format = mFormat; 1570 desc.frameCount = mNormalFrameCount; // FIXME see 1571 // AudioFlinger::frameCount(audio_io_handle_t) 1572 desc.latency = latency(); 1573 param2 = &desc; 1574 break; 1575 1576 case AudioSystem::STREAM_CONFIG_CHANGED: 1577 param2 = ¶m; 1578 case AudioSystem::OUTPUT_CLOSED: 1579 default: 1580 break; 1581 } 1582 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1583} 1584 1585void AudioFlinger::PlaybackThread::writeCallback() 1586{ 1587 ALOG_ASSERT(mCallbackThread != 0); 1588 mCallbackThread->resetWriteBlocked(); 1589} 1590 1591void AudioFlinger::PlaybackThread::drainCallback() 1592{ 1593 ALOG_ASSERT(mCallbackThread != 0); 1594 mCallbackThread->resetDraining(); 1595} 1596 1597void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1598{ 1599 Mutex::Autolock _l(mLock); 1600 // reject out of sequence requests 1601 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1602 mWriteAckSequence &= ~1; 1603 mWaitWorkCV.signal(); 1604 } 1605} 1606 1607void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1608{ 1609 Mutex::Autolock _l(mLock); 1610 // reject out of sequence requests 1611 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1612 mDrainSequence &= ~1; 1613 mWaitWorkCV.signal(); 1614 } 1615} 1616 1617// static 1618int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1619 void *param __unused, 1620 void *cookie) 1621{ 1622 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1623 ALOGV("asyncCallback() event %d", event); 1624 switch (event) { 1625 case STREAM_CBK_EVENT_WRITE_READY: 1626 me->writeCallback(); 1627 break; 1628 case STREAM_CBK_EVENT_DRAIN_READY: 1629 me->drainCallback(); 1630 break; 1631 default: 1632 ALOGW("asyncCallback() unknown event %d", event); 1633 break; 1634 } 1635 return 0; 1636} 1637 1638void AudioFlinger::PlaybackThread::readOutputParameters() 1639{ 1640 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1641 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1642 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1643 if (!audio_is_output_channel(mChannelMask)) { 1644 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1645 } 1646 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1647 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1648 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1649 } 1650 mChannelCount = popcount(mChannelMask); 1651 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1652 if (!audio_is_valid_format(mFormat)) { 1653 LOG_FATAL("HAL format %d not valid for output", mFormat); 1654 } 1655 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1656 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1657 mFormat); 1658 } 1659 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1660 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1661 mFrameCount = mBufferSize / mFrameSize; 1662 if (mFrameCount & 15) { 1663 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1664 mFrameCount); 1665 } 1666 1667 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1668 (mOutput->stream->set_callback != NULL)) { 1669 if (mOutput->stream->set_callback(mOutput->stream, 1670 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1671 mUseAsyncWrite = true; 1672 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1673 } 1674 } 1675 1676 // Calculate size of normal mix buffer relative to the HAL output buffer size 1677 double multiplier = 1.0; 1678 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1679 kUseFastMixer == FastMixer_Dynamic)) { 1680 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1681 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1682 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1683 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1684 maxNormalFrameCount = maxNormalFrameCount & ~15; 1685 if (maxNormalFrameCount < minNormalFrameCount) { 1686 maxNormalFrameCount = minNormalFrameCount; 1687 } 1688 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1689 if (multiplier <= 1.0) { 1690 multiplier = 1.0; 1691 } else if (multiplier <= 2.0) { 1692 if (2 * mFrameCount <= maxNormalFrameCount) { 1693 multiplier = 2.0; 1694 } else { 1695 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1696 } 1697 } else { 1698 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1699 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1700 // track, but we sometimes have to do this to satisfy the maximum frame count 1701 // constraint) 1702 // FIXME this rounding up should not be done if no HAL SRC 1703 uint32_t truncMult = (uint32_t) multiplier; 1704 if ((truncMult & 1)) { 1705 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1706 ++truncMult; 1707 } 1708 } 1709 multiplier = (double) truncMult; 1710 } 1711 } 1712 mNormalFrameCount = multiplier * mFrameCount; 1713 // round up to nearest 16 frames to satisfy AudioMixer 1714 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1715 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1716 mNormalFrameCount); 1717 1718 delete[] mMixBuffer; 1719 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1720 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1721 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1722 memset(mMixBuffer, 0, normalBufferSize); 1723 1724 // force reconfiguration of effect chains and engines to take new buffer size and audio 1725 // parameters into account 1726 // Note that mLock is not held when readOutputParameters() is called from the constructor 1727 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1728 // matter. 1729 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1730 Vector< sp<EffectChain> > effectChains = mEffectChains; 1731 for (size_t i = 0; i < effectChains.size(); i ++) { 1732 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1733 } 1734} 1735 1736 1737status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1738{ 1739 if (halFrames == NULL || dspFrames == NULL) { 1740 return BAD_VALUE; 1741 } 1742 Mutex::Autolock _l(mLock); 1743 if (initCheck() != NO_ERROR) { 1744 return INVALID_OPERATION; 1745 } 1746 size_t framesWritten = mBytesWritten / mFrameSize; 1747 *halFrames = framesWritten; 1748 1749 if (isSuspended()) { 1750 // return an estimation of rendered frames when the output is suspended 1751 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1752 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1753 return NO_ERROR; 1754 } else { 1755 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1756 } 1757} 1758 1759uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1760{ 1761 Mutex::Autolock _l(mLock); 1762 uint32_t result = 0; 1763 if (getEffectChain_l(sessionId) != 0) { 1764 result = EFFECT_SESSION; 1765 } 1766 1767 for (size_t i = 0; i < mTracks.size(); ++i) { 1768 sp<Track> track = mTracks[i]; 1769 if (sessionId == track->sessionId() && !track->isInvalid()) { 1770 result |= TRACK_SESSION; 1771 break; 1772 } 1773 } 1774 1775 return result; 1776} 1777 1778uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1779{ 1780 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1781 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1782 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1783 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1784 } 1785 for (size_t i = 0; i < mTracks.size(); i++) { 1786 sp<Track> track = mTracks[i]; 1787 if (sessionId == track->sessionId() && !track->isInvalid()) { 1788 return AudioSystem::getStrategyForStream(track->streamType()); 1789 } 1790 } 1791 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1792} 1793 1794 1795AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1796{ 1797 Mutex::Autolock _l(mLock); 1798 return mOutput; 1799} 1800 1801AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1802{ 1803 Mutex::Autolock _l(mLock); 1804 AudioStreamOut *output = mOutput; 1805 mOutput = NULL; 1806 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1807 // must push a NULL and wait for ack 1808 mOutputSink.clear(); 1809 mPipeSink.clear(); 1810 mNormalSink.clear(); 1811 return output; 1812} 1813 1814// this method must always be called either with ThreadBase mLock held or inside the thread loop 1815audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1816{ 1817 if (mOutput == NULL) { 1818 return NULL; 1819 } 1820 return &mOutput->stream->common; 1821} 1822 1823uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1824{ 1825 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1826} 1827 1828status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1829{ 1830 if (!isValidSyncEvent(event)) { 1831 return BAD_VALUE; 1832 } 1833 1834 Mutex::Autolock _l(mLock); 1835 1836 for (size_t i = 0; i < mTracks.size(); ++i) { 1837 sp<Track> track = mTracks[i]; 1838 if (event->triggerSession() == track->sessionId()) { 1839 (void) track->setSyncEvent(event); 1840 return NO_ERROR; 1841 } 1842 } 1843 1844 return NAME_NOT_FOUND; 1845} 1846 1847bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1848{ 1849 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1850} 1851 1852void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1853 const Vector< sp<Track> >& tracksToRemove) 1854{ 1855 size_t count = tracksToRemove.size(); 1856 if (count > 0) { 1857 for (size_t i = 0 ; i < count ; i++) { 1858 const sp<Track>& track = tracksToRemove.itemAt(i); 1859 if (!track->isOutputTrack()) { 1860 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1861#ifdef ADD_BATTERY_DATA 1862 // to track the speaker usage 1863 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1864#endif 1865 if (track->isTerminated()) { 1866 AudioSystem::releaseOutput(mId); 1867 } 1868 } 1869 } 1870 } 1871} 1872 1873void AudioFlinger::PlaybackThread::checkSilentMode_l() 1874{ 1875 if (!mMasterMute) { 1876 char value[PROPERTY_VALUE_MAX]; 1877 if (property_get("ro.audio.silent", value, "0") > 0) { 1878 char *endptr; 1879 unsigned long ul = strtoul(value, &endptr, 0); 1880 if (*endptr == '\0' && ul != 0) { 1881 ALOGD("Silence is golden"); 1882 // The setprop command will not allow a property to be changed after 1883 // the first time it is set, so we don't have to worry about un-muting. 1884 setMasterMute_l(true); 1885 } 1886 } 1887 } 1888} 1889 1890// shared by MIXER and DIRECT, overridden by DUPLICATING 1891ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1892{ 1893 // FIXME rewrite to reduce number of system calls 1894 mLastWriteTime = systemTime(); 1895 mInWrite = true; 1896 ssize_t bytesWritten; 1897 1898 // If an NBAIO sink is present, use it to write the normal mixer's submix 1899 if (mNormalSink != 0) { 1900#define mBitShift 2 // FIXME 1901 size_t count = mBytesRemaining >> mBitShift; 1902 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1903 ATRACE_BEGIN("write"); 1904 // update the setpoint when AudioFlinger::mScreenState changes 1905 uint32_t screenState = AudioFlinger::mScreenState; 1906 if (screenState != mScreenState) { 1907 mScreenState = screenState; 1908 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1909 if (pipe != NULL) { 1910 pipe->setAvgFrames((mScreenState & 1) ? 1911 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1912 } 1913 } 1914 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1915 ATRACE_END(); 1916 if (framesWritten > 0) { 1917 bytesWritten = framesWritten << mBitShift; 1918 } else { 1919 bytesWritten = framesWritten; 1920 } 1921 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1922 if (status == NO_ERROR) { 1923 size_t totalFramesWritten = mNormalSink->framesWritten(); 1924 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1925 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1926 mLatchDValid = true; 1927 } 1928 } 1929 // otherwise use the HAL / AudioStreamOut directly 1930 } else { 1931 // Direct output and offload threads 1932 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1933 if (mUseAsyncWrite) { 1934 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1935 mWriteAckSequence += 2; 1936 mWriteAckSequence |= 1; 1937 ALOG_ASSERT(mCallbackThread != 0); 1938 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1939 } 1940 // FIXME We should have an implementation of timestamps for direct output threads. 1941 // They are used e.g for multichannel PCM playback over HDMI. 1942 bytesWritten = mOutput->stream->write(mOutput->stream, 1943 (char *)mMixBuffer + offset, mBytesRemaining); 1944 if (mUseAsyncWrite && 1945 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1946 // do not wait for async callback in case of error of full write 1947 mWriteAckSequence &= ~1; 1948 ALOG_ASSERT(mCallbackThread != 0); 1949 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1950 } 1951 } 1952 1953 mNumWrites++; 1954 mInWrite = false; 1955 mStandby = false; 1956 return bytesWritten; 1957} 1958 1959void AudioFlinger::PlaybackThread::threadLoop_drain() 1960{ 1961 if (mOutput->stream->drain) { 1962 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1963 if (mUseAsyncWrite) { 1964 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1965 mDrainSequence |= 1; 1966 ALOG_ASSERT(mCallbackThread != 0); 1967 mCallbackThread->setDraining(mDrainSequence); 1968 } 1969 mOutput->stream->drain(mOutput->stream, 1970 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1971 : AUDIO_DRAIN_ALL); 1972 } 1973} 1974 1975void AudioFlinger::PlaybackThread::threadLoop_exit() 1976{ 1977 // Default implementation has nothing to do 1978} 1979 1980/* 1981The derived values that are cached: 1982 - mixBufferSize from frame count * frame size 1983 - activeSleepTime from activeSleepTimeUs() 1984 - idleSleepTime from idleSleepTimeUs() 1985 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1986 - maxPeriod from frame count and sample rate (MIXER only) 1987 1988The parameters that affect these derived values are: 1989 - frame count 1990 - frame size 1991 - sample rate 1992 - device type: A2DP or not 1993 - device latency 1994 - format: PCM or not 1995 - active sleep time 1996 - idle sleep time 1997*/ 1998 1999void AudioFlinger::PlaybackThread::cacheParameters_l() 2000{ 2001 mixBufferSize = mNormalFrameCount * mFrameSize; 2002 activeSleepTime = activeSleepTimeUs(); 2003 idleSleepTime = idleSleepTimeUs(); 2004} 2005 2006void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2007{ 2008 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2009 this, streamType, mTracks.size()); 2010 Mutex::Autolock _l(mLock); 2011 2012 size_t size = mTracks.size(); 2013 for (size_t i = 0; i < size; i++) { 2014 sp<Track> t = mTracks[i]; 2015 if (t->streamType() == streamType) { 2016 t->invalidate(); 2017 } 2018 } 2019} 2020 2021status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2022{ 2023 int session = chain->sessionId(); 2024 int16_t *buffer = mMixBuffer; 2025 bool ownsBuffer = false; 2026 2027 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2028 if (session > 0) { 2029 // Only one effect chain can be present in direct output thread and it uses 2030 // the mix buffer as input 2031 if (mType != DIRECT) { 2032 size_t numSamples = mNormalFrameCount * mChannelCount; 2033 buffer = new int16_t[numSamples]; 2034 memset(buffer, 0, numSamples * sizeof(int16_t)); 2035 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2036 ownsBuffer = true; 2037 } 2038 2039 // Attach all tracks with same session ID to this chain. 2040 for (size_t i = 0; i < mTracks.size(); ++i) { 2041 sp<Track> track = mTracks[i]; 2042 if (session == track->sessionId()) { 2043 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2044 buffer); 2045 track->setMainBuffer(buffer); 2046 chain->incTrackCnt(); 2047 } 2048 } 2049 2050 // indicate all active tracks in the chain 2051 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2052 sp<Track> track = mActiveTracks[i].promote(); 2053 if (track == 0) { 2054 continue; 2055 } 2056 if (session == track->sessionId()) { 2057 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2058 chain->incActiveTrackCnt(); 2059 } 2060 } 2061 } 2062 2063 chain->setInBuffer(buffer, ownsBuffer); 2064 chain->setOutBuffer(mMixBuffer); 2065 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2066 // chains list in order to be processed last as it contains output stage effects 2067 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2068 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2069 // after track specific effects and before output stage 2070 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2071 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2072 // Effect chain for other sessions are inserted at beginning of effect 2073 // chains list to be processed before output mix effects. Relative order between other 2074 // sessions is not important 2075 size_t size = mEffectChains.size(); 2076 size_t i = 0; 2077 for (i = 0; i < size; i++) { 2078 if (mEffectChains[i]->sessionId() < session) { 2079 break; 2080 } 2081 } 2082 mEffectChains.insertAt(chain, i); 2083 checkSuspendOnAddEffectChain_l(chain); 2084 2085 return NO_ERROR; 2086} 2087 2088size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2089{ 2090 int session = chain->sessionId(); 2091 2092 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2093 2094 for (size_t i = 0; i < mEffectChains.size(); i++) { 2095 if (chain == mEffectChains[i]) { 2096 mEffectChains.removeAt(i); 2097 // detach all active tracks from the chain 2098 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2099 sp<Track> track = mActiveTracks[i].promote(); 2100 if (track == 0) { 2101 continue; 2102 } 2103 if (session == track->sessionId()) { 2104 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2105 chain.get(), session); 2106 chain->decActiveTrackCnt(); 2107 } 2108 } 2109 2110 // detach all tracks with same session ID from this chain 2111 for (size_t i = 0; i < mTracks.size(); ++i) { 2112 sp<Track> track = mTracks[i]; 2113 if (session == track->sessionId()) { 2114 track->setMainBuffer(mMixBuffer); 2115 chain->decTrackCnt(); 2116 } 2117 } 2118 break; 2119 } 2120 } 2121 return mEffectChains.size(); 2122} 2123 2124status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2125 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2126{ 2127 Mutex::Autolock _l(mLock); 2128 return attachAuxEffect_l(track, EffectId); 2129} 2130 2131status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2132 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2133{ 2134 status_t status = NO_ERROR; 2135 2136 if (EffectId == 0) { 2137 track->setAuxBuffer(0, NULL); 2138 } else { 2139 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2140 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2141 if (effect != 0) { 2142 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2143 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2144 } else { 2145 status = INVALID_OPERATION; 2146 } 2147 } else { 2148 status = BAD_VALUE; 2149 } 2150 } 2151 return status; 2152} 2153 2154void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2155{ 2156 for (size_t i = 0; i < mTracks.size(); ++i) { 2157 sp<Track> track = mTracks[i]; 2158 if (track->auxEffectId() == effectId) { 2159 attachAuxEffect_l(track, 0); 2160 } 2161 } 2162} 2163 2164bool AudioFlinger::PlaybackThread::threadLoop() 2165{ 2166 Vector< sp<Track> > tracksToRemove; 2167 2168 standbyTime = systemTime(); 2169 2170 // MIXER 2171 nsecs_t lastWarning = 0; 2172 2173 // DUPLICATING 2174 // FIXME could this be made local to while loop? 2175 writeFrames = 0; 2176 2177 int lastGeneration = 0; 2178 2179 cacheParameters_l(); 2180 sleepTime = idleSleepTime; 2181 2182 if (mType == MIXER) { 2183 sleepTimeShift = 0; 2184 } 2185 2186 CpuStats cpuStats; 2187 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2188 2189 acquireWakeLock(); 2190 2191 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2192 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2193 // and then that string will be logged at the next convenient opportunity. 2194 const char *logString = NULL; 2195 2196 checkSilentMode_l(); 2197 2198 while (!exitPending()) 2199 { 2200 cpuStats.sample(myName); 2201 2202 Vector< sp<EffectChain> > effectChains; 2203 2204 processConfigEvents(); 2205 2206 { // scope for mLock 2207 2208 Mutex::Autolock _l(mLock); 2209 2210 if (logString != NULL) { 2211 mNBLogWriter->logTimestamp(); 2212 mNBLogWriter->log(logString); 2213 logString = NULL; 2214 } 2215 2216 if (mLatchDValid) { 2217 mLatchQ = mLatchD; 2218 mLatchDValid = false; 2219 mLatchQValid = true; 2220 } 2221 2222 if (checkForNewParameters_l()) { 2223 cacheParameters_l(); 2224 } 2225 2226 saveOutputTracks(); 2227 if (mSignalPending) { 2228 // A signal was raised while we were unlocked 2229 mSignalPending = false; 2230 } else if (waitingAsyncCallback_l()) { 2231 if (exitPending()) { 2232 break; 2233 } 2234 releaseWakeLock_l(); 2235 mWakeLockUids.clear(); 2236 mActiveTracksGeneration++; 2237 ALOGV("wait async completion"); 2238 mWaitWorkCV.wait(mLock); 2239 ALOGV("async completion/wake"); 2240 acquireWakeLock_l(); 2241 standbyTime = systemTime() + standbyDelay; 2242 sleepTime = 0; 2243 2244 continue; 2245 } 2246 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2247 isSuspended()) { 2248 // put audio hardware into standby after short delay 2249 if (shouldStandby_l()) { 2250 2251 threadLoop_standby(); 2252 2253 mStandby = true; 2254 } 2255 2256 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2257 // we're about to wait, flush the binder command buffer 2258 IPCThreadState::self()->flushCommands(); 2259 2260 clearOutputTracks(); 2261 2262 if (exitPending()) { 2263 break; 2264 } 2265 2266 releaseWakeLock_l(); 2267 mWakeLockUids.clear(); 2268 mActiveTracksGeneration++; 2269 // wait until we have something to do... 2270 ALOGV("%s going to sleep", myName.string()); 2271 mWaitWorkCV.wait(mLock); 2272 ALOGV("%s waking up", myName.string()); 2273 acquireWakeLock_l(); 2274 2275 mMixerStatus = MIXER_IDLE; 2276 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2277 mBytesWritten = 0; 2278 mBytesRemaining = 0; 2279 checkSilentMode_l(); 2280 2281 standbyTime = systemTime() + standbyDelay; 2282 sleepTime = idleSleepTime; 2283 if (mType == MIXER) { 2284 sleepTimeShift = 0; 2285 } 2286 2287 continue; 2288 } 2289 } 2290 // mMixerStatusIgnoringFastTracks is also updated internally 2291 mMixerStatus = prepareTracks_l(&tracksToRemove); 2292 2293 // compare with previously applied list 2294 if (lastGeneration != mActiveTracksGeneration) { 2295 // update wakelock 2296 updateWakeLockUids_l(mWakeLockUids); 2297 lastGeneration = mActiveTracksGeneration; 2298 } 2299 2300 // prevent any changes in effect chain list and in each effect chain 2301 // during mixing and effect process as the audio buffers could be deleted 2302 // or modified if an effect is created or deleted 2303 lockEffectChains_l(effectChains); 2304 } // mLock scope ends 2305 2306 if (mBytesRemaining == 0) { 2307 mCurrentWriteLength = 0; 2308 if (mMixerStatus == MIXER_TRACKS_READY) { 2309 // threadLoop_mix() sets mCurrentWriteLength 2310 threadLoop_mix(); 2311 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2312 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2313 // threadLoop_sleepTime sets sleepTime to 0 if data 2314 // must be written to HAL 2315 threadLoop_sleepTime(); 2316 if (sleepTime == 0) { 2317 mCurrentWriteLength = mixBufferSize; 2318 } 2319 } 2320 mBytesRemaining = mCurrentWriteLength; 2321 if (isSuspended()) { 2322 sleepTime = suspendSleepTimeUs(); 2323 // simulate write to HAL when suspended 2324 mBytesWritten += mixBufferSize; 2325 mBytesRemaining = 0; 2326 } 2327 2328 // only process effects if we're going to write 2329 if (sleepTime == 0 && mType != OFFLOAD) { 2330 for (size_t i = 0; i < effectChains.size(); i ++) { 2331 effectChains[i]->process_l(); 2332 } 2333 } 2334 } 2335 // Process effect chains for offloaded thread even if no audio 2336 // was read from audio track: process only updates effect state 2337 // and thus does have to be synchronized with audio writes but may have 2338 // to be called while waiting for async write callback 2339 if (mType == OFFLOAD) { 2340 for (size_t i = 0; i < effectChains.size(); i ++) { 2341 effectChains[i]->process_l(); 2342 } 2343 } 2344 2345 // enable changes in effect chain 2346 unlockEffectChains(effectChains); 2347 2348 if (!waitingAsyncCallback()) { 2349 // sleepTime == 0 means we must write to audio hardware 2350 if (sleepTime == 0) { 2351 if (mBytesRemaining) { 2352 ssize_t ret = threadLoop_write(); 2353 if (ret < 0) { 2354 mBytesRemaining = 0; 2355 } else { 2356 mBytesWritten += ret; 2357 mBytesRemaining -= ret; 2358 } 2359 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2360 (mMixerStatus == MIXER_DRAIN_ALL)) { 2361 threadLoop_drain(); 2362 } 2363if (mType == MIXER) { 2364 // write blocked detection 2365 nsecs_t now = systemTime(); 2366 nsecs_t delta = now - mLastWriteTime; 2367 if (!mStandby && delta > maxPeriod) { 2368 mNumDelayedWrites++; 2369 if ((now - lastWarning) > kWarningThrottleNs) { 2370 ATRACE_NAME("underrun"); 2371 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2372 ns2ms(delta), mNumDelayedWrites, this); 2373 lastWarning = now; 2374 } 2375 } 2376} 2377 2378 } else { 2379 usleep(sleepTime); 2380 } 2381 } 2382 2383 // Finally let go of removed track(s), without the lock held 2384 // since we can't guarantee the destructors won't acquire that 2385 // same lock. This will also mutate and push a new fast mixer state. 2386 threadLoop_removeTracks(tracksToRemove); 2387 tracksToRemove.clear(); 2388 2389 // FIXME I don't understand the need for this here; 2390 // it was in the original code but maybe the 2391 // assignment in saveOutputTracks() makes this unnecessary? 2392 clearOutputTracks(); 2393 2394 // Effect chains will be actually deleted here if they were removed from 2395 // mEffectChains list during mixing or effects processing 2396 effectChains.clear(); 2397 2398 // FIXME Note that the above .clear() is no longer necessary since effectChains 2399 // is now local to this block, but will keep it for now (at least until merge done). 2400 } 2401 2402 threadLoop_exit(); 2403 2404 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2405 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2406 // put output stream into standby mode 2407 if (!mStandby) { 2408 mOutput->stream->common.standby(&mOutput->stream->common); 2409 } 2410 } 2411 2412 releaseWakeLock(); 2413 mWakeLockUids.clear(); 2414 mActiveTracksGeneration++; 2415 2416 ALOGV("Thread %p type %d exiting", this, mType); 2417 return false; 2418} 2419 2420// removeTracks_l() must be called with ThreadBase::mLock held 2421void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2422{ 2423 size_t count = tracksToRemove.size(); 2424 if (count > 0) { 2425 for (size_t i=0 ; i<count ; i++) { 2426 const sp<Track>& track = tracksToRemove.itemAt(i); 2427 mActiveTracks.remove(track); 2428 mWakeLockUids.remove(track->uid()); 2429 mActiveTracksGeneration++; 2430 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2431 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2432 if (chain != 0) { 2433 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2434 track->sessionId()); 2435 chain->decActiveTrackCnt(); 2436 } 2437 if (track->isTerminated()) { 2438 removeTrack_l(track); 2439 } 2440 } 2441 } 2442 2443} 2444 2445status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2446{ 2447 if (mNormalSink != 0) { 2448 return mNormalSink->getTimestamp(timestamp); 2449 } 2450 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2451 uint64_t position64; 2452 int ret = mOutput->stream->get_presentation_position( 2453 mOutput->stream, &position64, ×tamp.mTime); 2454 if (ret == 0) { 2455 timestamp.mPosition = (uint32_t)position64; 2456 return NO_ERROR; 2457 } 2458 } 2459 return INVALID_OPERATION; 2460} 2461// ---------------------------------------------------------------------------- 2462 2463AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2464 audio_io_handle_t id, audio_devices_t device, type_t type) 2465 : PlaybackThread(audioFlinger, output, id, device, type), 2466 // mAudioMixer below 2467 // mFastMixer below 2468 mFastMixerFutex(0) 2469 // mOutputSink below 2470 // mPipeSink below 2471 // mNormalSink below 2472{ 2473 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2474 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2475 "mFrameCount=%d, mNormalFrameCount=%d", 2476 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2477 mNormalFrameCount); 2478 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2479 2480 // FIXME - Current mixer implementation only supports stereo output 2481 if (mChannelCount != FCC_2) { 2482 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2483 } 2484 2485 // create an NBAIO sink for the HAL output stream, and negotiate 2486 mOutputSink = new AudioStreamOutSink(output->stream); 2487 size_t numCounterOffers = 0; 2488 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2489 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2490 ALOG_ASSERT(index == 0); 2491 2492 // initialize fast mixer depending on configuration 2493 bool initFastMixer; 2494 switch (kUseFastMixer) { 2495 case FastMixer_Never: 2496 initFastMixer = false; 2497 break; 2498 case FastMixer_Always: 2499 initFastMixer = true; 2500 break; 2501 case FastMixer_Static: 2502 case FastMixer_Dynamic: 2503 initFastMixer = mFrameCount < mNormalFrameCount; 2504 break; 2505 } 2506 if (initFastMixer) { 2507 2508 // create a MonoPipe to connect our submix to FastMixer 2509 NBAIO_Format format = mOutputSink->format(); 2510 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2511 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2512 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2513 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2514 const NBAIO_Format offers[1] = {format}; 2515 size_t numCounterOffers = 0; 2516 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2517 ALOG_ASSERT(index == 0); 2518 monoPipe->setAvgFrames((mScreenState & 1) ? 2519 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2520 mPipeSink = monoPipe; 2521 2522#ifdef TEE_SINK 2523 if (mTeeSinkOutputEnabled) { 2524 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2525 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2526 numCounterOffers = 0; 2527 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2528 ALOG_ASSERT(index == 0); 2529 mTeeSink = teeSink; 2530 PipeReader *teeSource = new PipeReader(*teeSink); 2531 numCounterOffers = 0; 2532 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2533 ALOG_ASSERT(index == 0); 2534 mTeeSource = teeSource; 2535 } 2536#endif 2537 2538 // create fast mixer and configure it initially with just one fast track for our submix 2539 mFastMixer = new FastMixer(); 2540 FastMixerStateQueue *sq = mFastMixer->sq(); 2541#ifdef STATE_QUEUE_DUMP 2542 sq->setObserverDump(&mStateQueueObserverDump); 2543 sq->setMutatorDump(&mStateQueueMutatorDump); 2544#endif 2545 FastMixerState *state = sq->begin(); 2546 FastTrack *fastTrack = &state->mFastTracks[0]; 2547 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2548 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2549 fastTrack->mVolumeProvider = NULL; 2550 fastTrack->mGeneration++; 2551 state->mFastTracksGen++; 2552 state->mTrackMask = 1; 2553 // fast mixer will use the HAL output sink 2554 state->mOutputSink = mOutputSink.get(); 2555 state->mOutputSinkGen++; 2556 state->mFrameCount = mFrameCount; 2557 state->mCommand = FastMixerState::COLD_IDLE; 2558 // already done in constructor initialization list 2559 //mFastMixerFutex = 0; 2560 state->mColdFutexAddr = &mFastMixerFutex; 2561 state->mColdGen++; 2562 state->mDumpState = &mFastMixerDumpState; 2563#ifdef TEE_SINK 2564 state->mTeeSink = mTeeSink.get(); 2565#endif 2566 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2567 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2568 sq->end(); 2569 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2570 2571 // start the fast mixer 2572 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2573 pid_t tid = mFastMixer->getTid(); 2574 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2575 if (err != 0) { 2576 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2577 kPriorityFastMixer, getpid_cached, tid, err); 2578 } 2579 2580#ifdef AUDIO_WATCHDOG 2581 // create and start the watchdog 2582 mAudioWatchdog = new AudioWatchdog(); 2583 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2584 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2585 tid = mAudioWatchdog->getTid(); 2586 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2587 if (err != 0) { 2588 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2589 kPriorityFastMixer, getpid_cached, tid, err); 2590 } 2591#endif 2592 2593 } else { 2594 mFastMixer = NULL; 2595 } 2596 2597 switch (kUseFastMixer) { 2598 case FastMixer_Never: 2599 case FastMixer_Dynamic: 2600 mNormalSink = mOutputSink; 2601 break; 2602 case FastMixer_Always: 2603 mNormalSink = mPipeSink; 2604 break; 2605 case FastMixer_Static: 2606 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2607 break; 2608 } 2609} 2610 2611AudioFlinger::MixerThread::~MixerThread() 2612{ 2613 if (mFastMixer != NULL) { 2614 FastMixerStateQueue *sq = mFastMixer->sq(); 2615 FastMixerState *state = sq->begin(); 2616 if (state->mCommand == FastMixerState::COLD_IDLE) { 2617 int32_t old = android_atomic_inc(&mFastMixerFutex); 2618 if (old == -1) { 2619 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2620 } 2621 } 2622 state->mCommand = FastMixerState::EXIT; 2623 sq->end(); 2624 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2625 mFastMixer->join(); 2626 // Though the fast mixer thread has exited, it's state queue is still valid. 2627 // We'll use that extract the final state which contains one remaining fast track 2628 // corresponding to our sub-mix. 2629 state = sq->begin(); 2630 ALOG_ASSERT(state->mTrackMask == 1); 2631 FastTrack *fastTrack = &state->mFastTracks[0]; 2632 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2633 delete fastTrack->mBufferProvider; 2634 sq->end(false /*didModify*/); 2635 delete mFastMixer; 2636#ifdef AUDIO_WATCHDOG 2637 if (mAudioWatchdog != 0) { 2638 mAudioWatchdog->requestExit(); 2639 mAudioWatchdog->requestExitAndWait(); 2640 mAudioWatchdog.clear(); 2641 } 2642#endif 2643 } 2644 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2645 delete mAudioMixer; 2646} 2647 2648 2649uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2650{ 2651 if (mFastMixer != NULL) { 2652 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2653 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2654 } 2655 return latency; 2656} 2657 2658 2659void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2660{ 2661 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2662} 2663 2664ssize_t AudioFlinger::MixerThread::threadLoop_write() 2665{ 2666 // FIXME we should only do one push per cycle; confirm this is true 2667 // Start the fast mixer if it's not already running 2668 if (mFastMixer != NULL) { 2669 FastMixerStateQueue *sq = mFastMixer->sq(); 2670 FastMixerState *state = sq->begin(); 2671 if (state->mCommand != FastMixerState::MIX_WRITE && 2672 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2673 if (state->mCommand == FastMixerState::COLD_IDLE) { 2674 int32_t old = android_atomic_inc(&mFastMixerFutex); 2675 if (old == -1) { 2676 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2677 } 2678#ifdef AUDIO_WATCHDOG 2679 if (mAudioWatchdog != 0) { 2680 mAudioWatchdog->resume(); 2681 } 2682#endif 2683 } 2684 state->mCommand = FastMixerState::MIX_WRITE; 2685 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2686 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2687 sq->end(); 2688 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2689 if (kUseFastMixer == FastMixer_Dynamic) { 2690 mNormalSink = mPipeSink; 2691 } 2692 } else { 2693 sq->end(false /*didModify*/); 2694 } 2695 } 2696 return PlaybackThread::threadLoop_write(); 2697} 2698 2699void AudioFlinger::MixerThread::threadLoop_standby() 2700{ 2701 // Idle the fast mixer if it's currently running 2702 if (mFastMixer != NULL) { 2703 FastMixerStateQueue *sq = mFastMixer->sq(); 2704 FastMixerState *state = sq->begin(); 2705 if (!(state->mCommand & FastMixerState::IDLE)) { 2706 state->mCommand = FastMixerState::COLD_IDLE; 2707 state->mColdFutexAddr = &mFastMixerFutex; 2708 state->mColdGen++; 2709 mFastMixerFutex = 0; 2710 sq->end(); 2711 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2712 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2713 if (kUseFastMixer == FastMixer_Dynamic) { 2714 mNormalSink = mOutputSink; 2715 } 2716#ifdef AUDIO_WATCHDOG 2717 if (mAudioWatchdog != 0) { 2718 mAudioWatchdog->pause(); 2719 } 2720#endif 2721 } else { 2722 sq->end(false /*didModify*/); 2723 } 2724 } 2725 PlaybackThread::threadLoop_standby(); 2726} 2727 2728bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2729{ 2730 return false; 2731} 2732 2733bool AudioFlinger::PlaybackThread::shouldStandby_l() 2734{ 2735 return !mStandby; 2736} 2737 2738bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2739{ 2740 Mutex::Autolock _l(mLock); 2741 return waitingAsyncCallback_l(); 2742} 2743 2744// shared by MIXER and DIRECT, overridden by DUPLICATING 2745void AudioFlinger::PlaybackThread::threadLoop_standby() 2746{ 2747 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2748 mOutput->stream->common.standby(&mOutput->stream->common); 2749 if (mUseAsyncWrite != 0) { 2750 // discard any pending drain or write ack by incrementing sequence 2751 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2752 mDrainSequence = (mDrainSequence + 2) & ~1; 2753 ALOG_ASSERT(mCallbackThread != 0); 2754 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2755 mCallbackThread->setDraining(mDrainSequence); 2756 } 2757} 2758 2759void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2760{ 2761 ALOGV("signal playback thread"); 2762 broadcast_l(); 2763} 2764 2765void AudioFlinger::MixerThread::threadLoop_mix() 2766{ 2767 // obtain the presentation timestamp of the next output buffer 2768 int64_t pts; 2769 status_t status = INVALID_OPERATION; 2770 2771 if (mNormalSink != 0) { 2772 status = mNormalSink->getNextWriteTimestamp(&pts); 2773 } else { 2774 status = mOutputSink->getNextWriteTimestamp(&pts); 2775 } 2776 2777 if (status != NO_ERROR) { 2778 pts = AudioBufferProvider::kInvalidPTS; 2779 } 2780 2781 // mix buffers... 2782 mAudioMixer->process(pts); 2783 mCurrentWriteLength = mixBufferSize; 2784 // increase sleep time progressively when application underrun condition clears. 2785 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2786 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2787 // such that we would underrun the audio HAL. 2788 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2789 sleepTimeShift--; 2790 } 2791 sleepTime = 0; 2792 standbyTime = systemTime() + standbyDelay; 2793 //TODO: delay standby when effects have a tail 2794} 2795 2796void AudioFlinger::MixerThread::threadLoop_sleepTime() 2797{ 2798 // If no tracks are ready, sleep once for the duration of an output 2799 // buffer size, then write 0s to the output 2800 if (sleepTime == 0) { 2801 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2802 sleepTime = activeSleepTime >> sleepTimeShift; 2803 if (sleepTime < kMinThreadSleepTimeUs) { 2804 sleepTime = kMinThreadSleepTimeUs; 2805 } 2806 // reduce sleep time in case of consecutive application underruns to avoid 2807 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2808 // duration we would end up writing less data than needed by the audio HAL if 2809 // the condition persists. 2810 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2811 sleepTimeShift++; 2812 } 2813 } else { 2814 sleepTime = idleSleepTime; 2815 } 2816 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2817 memset(mMixBuffer, 0, mixBufferSize); 2818 sleepTime = 0; 2819 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2820 "anticipated start"); 2821 } 2822 // TODO add standby time extension fct of effect tail 2823} 2824 2825// prepareTracks_l() must be called with ThreadBase::mLock held 2826AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2827 Vector< sp<Track> > *tracksToRemove) 2828{ 2829 2830 mixer_state mixerStatus = MIXER_IDLE; 2831 // find out which tracks need to be processed 2832 size_t count = mActiveTracks.size(); 2833 size_t mixedTracks = 0; 2834 size_t tracksWithEffect = 0; 2835 // counts only _active_ fast tracks 2836 size_t fastTracks = 0; 2837 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2838 2839 float masterVolume = mMasterVolume; 2840 bool masterMute = mMasterMute; 2841 2842 if (masterMute) { 2843 masterVolume = 0; 2844 } 2845 // Delegate master volume control to effect in output mix effect chain if needed 2846 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2847 if (chain != 0) { 2848 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2849 chain->setVolume_l(&v, &v); 2850 masterVolume = (float)((v + (1 << 23)) >> 24); 2851 chain.clear(); 2852 } 2853 2854 // prepare a new state to push 2855 FastMixerStateQueue *sq = NULL; 2856 FastMixerState *state = NULL; 2857 bool didModify = false; 2858 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2859 if (mFastMixer != NULL) { 2860 sq = mFastMixer->sq(); 2861 state = sq->begin(); 2862 } 2863 2864 for (size_t i=0 ; i<count ; i++) { 2865 const sp<Track> t = mActiveTracks[i].promote(); 2866 if (t == 0) { 2867 continue; 2868 } 2869 2870 // this const just means the local variable doesn't change 2871 Track* const track = t.get(); 2872 2873 // process fast tracks 2874 if (track->isFastTrack()) { 2875 2876 // It's theoretically possible (though unlikely) for a fast track to be created 2877 // and then removed within the same normal mix cycle. This is not a problem, as 2878 // the track never becomes active so it's fast mixer slot is never touched. 2879 // The converse, of removing an (active) track and then creating a new track 2880 // at the identical fast mixer slot within the same normal mix cycle, 2881 // is impossible because the slot isn't marked available until the end of each cycle. 2882 int j = track->mFastIndex; 2883 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2884 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2885 FastTrack *fastTrack = &state->mFastTracks[j]; 2886 2887 // Determine whether the track is currently in underrun condition, 2888 // and whether it had a recent underrun. 2889 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2890 FastTrackUnderruns underruns = ftDump->mUnderruns; 2891 uint32_t recentFull = (underruns.mBitFields.mFull - 2892 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2893 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2894 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2895 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2896 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2897 uint32_t recentUnderruns = recentPartial + recentEmpty; 2898 track->mObservedUnderruns = underruns; 2899 // don't count underruns that occur while stopping or pausing 2900 // or stopped which can occur when flush() is called while active 2901 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2902 recentUnderruns > 0) { 2903 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2904 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2905 } 2906 2907 // This is similar to the state machine for normal tracks, 2908 // with a few modifications for fast tracks. 2909 bool isActive = true; 2910 switch (track->mState) { 2911 case TrackBase::STOPPING_1: 2912 // track stays active in STOPPING_1 state until first underrun 2913 if (recentUnderruns > 0 || track->isTerminated()) { 2914 track->mState = TrackBase::STOPPING_2; 2915 } 2916 break; 2917 case TrackBase::PAUSING: 2918 // ramp down is not yet implemented 2919 track->setPaused(); 2920 break; 2921 case TrackBase::RESUMING: 2922 // ramp up is not yet implemented 2923 track->mState = TrackBase::ACTIVE; 2924 break; 2925 case TrackBase::ACTIVE: 2926 if (recentFull > 0 || recentPartial > 0) { 2927 // track has provided at least some frames recently: reset retry count 2928 track->mRetryCount = kMaxTrackRetries; 2929 } 2930 if (recentUnderruns == 0) { 2931 // no recent underruns: stay active 2932 break; 2933 } 2934 // there has recently been an underrun of some kind 2935 if (track->sharedBuffer() == 0) { 2936 // were any of the recent underruns "empty" (no frames available)? 2937 if (recentEmpty == 0) { 2938 // no, then ignore the partial underruns as they are allowed indefinitely 2939 break; 2940 } 2941 // there has recently been an "empty" underrun: decrement the retry counter 2942 if (--(track->mRetryCount) > 0) { 2943 break; 2944 } 2945 // indicate to client process that the track was disabled because of underrun; 2946 // it will then automatically call start() when data is available 2947 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2948 // remove from active list, but state remains ACTIVE [confusing but true] 2949 isActive = false; 2950 break; 2951 } 2952 // fall through 2953 case TrackBase::STOPPING_2: 2954 case TrackBase::PAUSED: 2955 case TrackBase::STOPPED: 2956 case TrackBase::FLUSHED: // flush() while active 2957 // Check for presentation complete if track is inactive 2958 // We have consumed all the buffers of this track. 2959 // This would be incomplete if we auto-paused on underrun 2960 { 2961 size_t audioHALFrames = 2962 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2963 size_t framesWritten = mBytesWritten / mFrameSize; 2964 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2965 // track stays in active list until presentation is complete 2966 break; 2967 } 2968 } 2969 if (track->isStopping_2()) { 2970 track->mState = TrackBase::STOPPED; 2971 } 2972 if (track->isStopped()) { 2973 // Can't reset directly, as fast mixer is still polling this track 2974 // track->reset(); 2975 // So instead mark this track as needing to be reset after push with ack 2976 resetMask |= 1 << i; 2977 } 2978 isActive = false; 2979 break; 2980 case TrackBase::IDLE: 2981 default: 2982 LOG_FATAL("unexpected track state %d", track->mState); 2983 } 2984 2985 if (isActive) { 2986 // was it previously inactive? 2987 if (!(state->mTrackMask & (1 << j))) { 2988 ExtendedAudioBufferProvider *eabp = track; 2989 VolumeProvider *vp = track; 2990 fastTrack->mBufferProvider = eabp; 2991 fastTrack->mVolumeProvider = vp; 2992 fastTrack->mSampleRate = track->mSampleRate; 2993 fastTrack->mChannelMask = track->mChannelMask; 2994 fastTrack->mGeneration++; 2995 state->mTrackMask |= 1 << j; 2996 didModify = true; 2997 // no acknowledgement required for newly active tracks 2998 } 2999 // cache the combined master volume and stream type volume for fast mixer; this 3000 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3001 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3002 ++fastTracks; 3003 } else { 3004 // was it previously active? 3005 if (state->mTrackMask & (1 << j)) { 3006 fastTrack->mBufferProvider = NULL; 3007 fastTrack->mGeneration++; 3008 state->mTrackMask &= ~(1 << j); 3009 didModify = true; 3010 // If any fast tracks were removed, we must wait for acknowledgement 3011 // because we're about to decrement the last sp<> on those tracks. 3012 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3013 } else { 3014 LOG_FATAL("fast track %d should have been active", j); 3015 } 3016 tracksToRemove->add(track); 3017 // Avoids a misleading display in dumpsys 3018 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3019 } 3020 continue; 3021 } 3022 3023 { // local variable scope to avoid goto warning 3024 3025 audio_track_cblk_t* cblk = track->cblk(); 3026 3027 // The first time a track is added we wait 3028 // for all its buffers to be filled before processing it 3029 int name = track->name(); 3030 // make sure that we have enough frames to mix one full buffer. 3031 // enforce this condition only once to enable draining the buffer in case the client 3032 // app does not call stop() and relies on underrun to stop: 3033 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3034 // during last round 3035 size_t desiredFrames; 3036 uint32_t sr = track->sampleRate(); 3037 if (sr == mSampleRate) { 3038 desiredFrames = mNormalFrameCount; 3039 } else { 3040 // +1 for rounding and +1 for additional sample needed for interpolation 3041 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3042 // add frames already consumed but not yet released by the resampler 3043 // because mAudioTrackServerProxy->framesReady() will include these frames 3044 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3045#if 0 3046 // the minimum track buffer size is normally twice the number of frames necessary 3047 // to fill one buffer and the resampler should not leave more than one buffer worth 3048 // of unreleased frames after each pass, but just in case... 3049 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3050#endif 3051 } 3052 uint32_t minFrames = 1; 3053 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3054 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3055 minFrames = desiredFrames; 3056 } 3057 3058 size_t framesReady = track->framesReady(); 3059 if ((framesReady >= minFrames) && track->isReady() && 3060 !track->isPaused() && !track->isTerminated()) 3061 { 3062 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3063 3064 mixedTracks++; 3065 3066 // track->mainBuffer() != mMixBuffer means there is an effect chain 3067 // connected to the track 3068 chain.clear(); 3069 if (track->mainBuffer() != mMixBuffer) { 3070 chain = getEffectChain_l(track->sessionId()); 3071 // Delegate volume control to effect in track effect chain if needed 3072 if (chain != 0) { 3073 tracksWithEffect++; 3074 } else { 3075 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3076 "session %d", 3077 name, track->sessionId()); 3078 } 3079 } 3080 3081 3082 int param = AudioMixer::VOLUME; 3083 if (track->mFillingUpStatus == Track::FS_FILLED) { 3084 // no ramp for the first volume setting 3085 track->mFillingUpStatus = Track::FS_ACTIVE; 3086 if (track->mState == TrackBase::RESUMING) { 3087 track->mState = TrackBase::ACTIVE; 3088 param = AudioMixer::RAMP_VOLUME; 3089 } 3090 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3091 // FIXME should not make a decision based on mServer 3092 } else if (cblk->mServer != 0) { 3093 // If the track is stopped before the first frame was mixed, 3094 // do not apply ramp 3095 param = AudioMixer::RAMP_VOLUME; 3096 } 3097 3098 // compute volume for this track 3099 uint32_t vl, vr, va; 3100 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3101 vl = vr = va = 0; 3102 if (track->isPausing()) { 3103 track->setPaused(); 3104 } 3105 } else { 3106 3107 // read original volumes with volume control 3108 float typeVolume = mStreamTypes[track->streamType()].volume; 3109 float v = masterVolume * typeVolume; 3110 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3111 uint32_t vlr = proxy->getVolumeLR(); 3112 vl = vlr & 0xFFFF; 3113 vr = vlr >> 16; 3114 // track volumes come from shared memory, so can't be trusted and must be clamped 3115 if (vl > MAX_GAIN_INT) { 3116 ALOGV("Track left volume out of range: %04X", vl); 3117 vl = MAX_GAIN_INT; 3118 } 3119 if (vr > MAX_GAIN_INT) { 3120 ALOGV("Track right volume out of range: %04X", vr); 3121 vr = MAX_GAIN_INT; 3122 } 3123 // now apply the master volume and stream type volume 3124 vl = (uint32_t)(v * vl) << 12; 3125 vr = (uint32_t)(v * vr) << 12; 3126 // assuming master volume and stream type volume each go up to 1.0, 3127 // vl and vr are now in 8.24 format 3128 3129 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3130 // send level comes from shared memory and so may be corrupt 3131 if (sendLevel > MAX_GAIN_INT) { 3132 ALOGV("Track send level out of range: %04X", sendLevel); 3133 sendLevel = MAX_GAIN_INT; 3134 } 3135 va = (uint32_t)(v * sendLevel); 3136 } 3137 3138 // Delegate volume control to effect in track effect chain if needed 3139 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3140 // Do not ramp volume if volume is controlled by effect 3141 param = AudioMixer::VOLUME; 3142 track->mHasVolumeController = true; 3143 } else { 3144 // force no volume ramp when volume controller was just disabled or removed 3145 // from effect chain to avoid volume spike 3146 if (track->mHasVolumeController) { 3147 param = AudioMixer::VOLUME; 3148 } 3149 track->mHasVolumeController = false; 3150 } 3151 3152 // Convert volumes from 8.24 to 4.12 format 3153 // This additional clamping is needed in case chain->setVolume_l() overshot 3154 vl = (vl + (1 << 11)) >> 12; 3155 if (vl > MAX_GAIN_INT) { 3156 vl = MAX_GAIN_INT; 3157 } 3158 vr = (vr + (1 << 11)) >> 12; 3159 if (vr > MAX_GAIN_INT) { 3160 vr = MAX_GAIN_INT; 3161 } 3162 3163 if (va > MAX_GAIN_INT) { 3164 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3165 } 3166 3167 // XXX: these things DON'T need to be done each time 3168 mAudioMixer->setBufferProvider(name, track); 3169 mAudioMixer->enable(name); 3170 3171 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3172 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3173 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3174 mAudioMixer->setParameter( 3175 name, 3176 AudioMixer::TRACK, 3177 AudioMixer::FORMAT, (void *)track->format()); 3178 mAudioMixer->setParameter( 3179 name, 3180 AudioMixer::TRACK, 3181 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3182 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3183 uint32_t maxSampleRate = mSampleRate * 2; 3184 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3185 if (reqSampleRate == 0) { 3186 reqSampleRate = mSampleRate; 3187 } else if (reqSampleRate > maxSampleRate) { 3188 reqSampleRate = maxSampleRate; 3189 } 3190 mAudioMixer->setParameter( 3191 name, 3192 AudioMixer::RESAMPLE, 3193 AudioMixer::SAMPLE_RATE, 3194 (void *)reqSampleRate); 3195 mAudioMixer->setParameter( 3196 name, 3197 AudioMixer::TRACK, 3198 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3199 mAudioMixer->setParameter( 3200 name, 3201 AudioMixer::TRACK, 3202 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3203 3204 // reset retry count 3205 track->mRetryCount = kMaxTrackRetries; 3206 3207 // If one track is ready, set the mixer ready if: 3208 // - the mixer was not ready during previous round OR 3209 // - no other track is not ready 3210 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3211 mixerStatus != MIXER_TRACKS_ENABLED) { 3212 mixerStatus = MIXER_TRACKS_READY; 3213 } 3214 } else { 3215 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3216 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3217 } 3218 // clear effect chain input buffer if an active track underruns to avoid sending 3219 // previous audio buffer again to effects 3220 chain = getEffectChain_l(track->sessionId()); 3221 if (chain != 0) { 3222 chain->clearInputBuffer(); 3223 } 3224 3225 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3226 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3227 track->isStopped() || track->isPaused()) { 3228 // We have consumed all the buffers of this track. 3229 // Remove it from the list of active tracks. 3230 // TODO: use actual buffer filling status instead of latency when available from 3231 // audio HAL 3232 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3233 size_t framesWritten = mBytesWritten / mFrameSize; 3234 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3235 if (track->isStopped()) { 3236 track->reset(); 3237 } 3238 tracksToRemove->add(track); 3239 } 3240 } else { 3241 // No buffers for this track. Give it a few chances to 3242 // fill a buffer, then remove it from active list. 3243 if (--(track->mRetryCount) <= 0) { 3244 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3245 tracksToRemove->add(track); 3246 // indicate to client process that the track was disabled because of underrun; 3247 // it will then automatically call start() when data is available 3248 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3249 // If one track is not ready, mark the mixer also not ready if: 3250 // - the mixer was ready during previous round OR 3251 // - no other track is ready 3252 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3253 mixerStatus != MIXER_TRACKS_READY) { 3254 mixerStatus = MIXER_TRACKS_ENABLED; 3255 } 3256 } 3257 mAudioMixer->disable(name); 3258 } 3259 3260 } // local variable scope to avoid goto warning 3261track_is_ready: ; 3262 3263 } 3264 3265 // Push the new FastMixer state if necessary 3266 bool pauseAudioWatchdog = false; 3267 if (didModify) { 3268 state->mFastTracksGen++; 3269 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3270 if (kUseFastMixer == FastMixer_Dynamic && 3271 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3272 state->mCommand = FastMixerState::COLD_IDLE; 3273 state->mColdFutexAddr = &mFastMixerFutex; 3274 state->mColdGen++; 3275 mFastMixerFutex = 0; 3276 if (kUseFastMixer == FastMixer_Dynamic) { 3277 mNormalSink = mOutputSink; 3278 } 3279 // If we go into cold idle, need to wait for acknowledgement 3280 // so that fast mixer stops doing I/O. 3281 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3282 pauseAudioWatchdog = true; 3283 } 3284 } 3285 if (sq != NULL) { 3286 sq->end(didModify); 3287 sq->push(block); 3288 } 3289#ifdef AUDIO_WATCHDOG 3290 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3291 mAudioWatchdog->pause(); 3292 } 3293#endif 3294 3295 // Now perform the deferred reset on fast tracks that have stopped 3296 while (resetMask != 0) { 3297 size_t i = __builtin_ctz(resetMask); 3298 ALOG_ASSERT(i < count); 3299 resetMask &= ~(1 << i); 3300 sp<Track> t = mActiveTracks[i].promote(); 3301 if (t == 0) { 3302 continue; 3303 } 3304 Track* track = t.get(); 3305 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3306 track->reset(); 3307 } 3308 3309 // remove all the tracks that need to be... 3310 removeTracks_l(*tracksToRemove); 3311 3312 // mix buffer must be cleared if all tracks are connected to an 3313 // effect chain as in this case the mixer will not write to 3314 // mix buffer and track effects will accumulate into it 3315 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3316 (mixedTracks == 0 && fastTracks > 0))) { 3317 // FIXME as a performance optimization, should remember previous zero status 3318 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3319 } 3320 3321 // if any fast tracks, then status is ready 3322 mMixerStatusIgnoringFastTracks = mixerStatus; 3323 if (fastTracks > 0) { 3324 mixerStatus = MIXER_TRACKS_READY; 3325 } 3326 return mixerStatus; 3327} 3328 3329// getTrackName_l() must be called with ThreadBase::mLock held 3330int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3331{ 3332 return mAudioMixer->getTrackName(channelMask, sessionId); 3333} 3334 3335// deleteTrackName_l() must be called with ThreadBase::mLock held 3336void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3337{ 3338 ALOGV("remove track (%d) and delete from mixer", name); 3339 mAudioMixer->deleteTrackName(name); 3340} 3341 3342// checkForNewParameters_l() must be called with ThreadBase::mLock held 3343bool AudioFlinger::MixerThread::checkForNewParameters_l() 3344{ 3345 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3346 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3347 bool reconfig = false; 3348 3349 while (!mNewParameters.isEmpty()) { 3350 3351 if (mFastMixer != NULL) { 3352 FastMixerStateQueue *sq = mFastMixer->sq(); 3353 FastMixerState *state = sq->begin(); 3354 if (!(state->mCommand & FastMixerState::IDLE)) { 3355 previousCommand = state->mCommand; 3356 state->mCommand = FastMixerState::HOT_IDLE; 3357 sq->end(); 3358 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3359 } else { 3360 sq->end(false /*didModify*/); 3361 } 3362 } 3363 3364 status_t status = NO_ERROR; 3365 String8 keyValuePair = mNewParameters[0]; 3366 AudioParameter param = AudioParameter(keyValuePair); 3367 int value; 3368 3369 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3370 reconfig = true; 3371 } 3372 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3373 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3374 status = BAD_VALUE; 3375 } else { 3376 // no need to save value, since it's constant 3377 reconfig = true; 3378 } 3379 } 3380 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3381 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3382 status = BAD_VALUE; 3383 } else { 3384 // no need to save value, since it's constant 3385 reconfig = true; 3386 } 3387 } 3388 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3389 // do not accept frame count changes if tracks are open as the track buffer 3390 // size depends on frame count and correct behavior would not be guaranteed 3391 // if frame count is changed after track creation 3392 if (!mTracks.isEmpty()) { 3393 status = INVALID_OPERATION; 3394 } else { 3395 reconfig = true; 3396 } 3397 } 3398 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3399#ifdef ADD_BATTERY_DATA 3400 // when changing the audio output device, call addBatteryData to notify 3401 // the change 3402 if (mOutDevice != value) { 3403 uint32_t params = 0; 3404 // check whether speaker is on 3405 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3406 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3407 } 3408 3409 audio_devices_t deviceWithoutSpeaker 3410 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3411 // check if any other device (except speaker) is on 3412 if (value & deviceWithoutSpeaker ) { 3413 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3414 } 3415 3416 if (params != 0) { 3417 addBatteryData(params); 3418 } 3419 } 3420#endif 3421 3422 // forward device change to effects that have requested to be 3423 // aware of attached audio device. 3424 if (value != AUDIO_DEVICE_NONE) { 3425 mOutDevice = value; 3426 for (size_t i = 0; i < mEffectChains.size(); i++) { 3427 mEffectChains[i]->setDevice_l(mOutDevice); 3428 } 3429 } 3430 } 3431 3432 if (status == NO_ERROR) { 3433 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3434 keyValuePair.string()); 3435 if (!mStandby && status == INVALID_OPERATION) { 3436 mOutput->stream->common.standby(&mOutput->stream->common); 3437 mStandby = true; 3438 mBytesWritten = 0; 3439 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3440 keyValuePair.string()); 3441 } 3442 if (status == NO_ERROR && reconfig) { 3443 readOutputParameters(); 3444 delete mAudioMixer; 3445 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3446 for (size_t i = 0; i < mTracks.size() ; i++) { 3447 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3448 if (name < 0) { 3449 break; 3450 } 3451 mTracks[i]->mName = name; 3452 } 3453 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3454 } 3455 } 3456 3457 mNewParameters.removeAt(0); 3458 3459 mParamStatus = status; 3460 mParamCond.signal(); 3461 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3462 // already timed out waiting for the status and will never signal the condition. 3463 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3464 } 3465 3466 if (!(previousCommand & FastMixerState::IDLE)) { 3467 ALOG_ASSERT(mFastMixer != NULL); 3468 FastMixerStateQueue *sq = mFastMixer->sq(); 3469 FastMixerState *state = sq->begin(); 3470 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3471 state->mCommand = previousCommand; 3472 sq->end(); 3473 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3474 } 3475 3476 return reconfig; 3477} 3478 3479 3480void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3481{ 3482 const size_t SIZE = 256; 3483 char buffer[SIZE]; 3484 String8 result; 3485 3486 PlaybackThread::dumpInternals(fd, args); 3487 3488 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3489 result.append(buffer); 3490 write(fd, result.string(), result.size()); 3491 3492 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3493 const FastMixerDumpState copy(mFastMixerDumpState); 3494 copy.dump(fd); 3495 3496#ifdef STATE_QUEUE_DUMP 3497 // Similar for state queue 3498 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3499 observerCopy.dump(fd); 3500 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3501 mutatorCopy.dump(fd); 3502#endif 3503 3504#ifdef TEE_SINK 3505 // Write the tee output to a .wav file 3506 dumpTee(fd, mTeeSource, mId); 3507#endif 3508 3509#ifdef AUDIO_WATCHDOG 3510 if (mAudioWatchdog != 0) { 3511 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3512 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3513 wdCopy.dump(fd); 3514 } 3515#endif 3516} 3517 3518uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3519{ 3520 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3521} 3522 3523uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3524{ 3525 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3526} 3527 3528void AudioFlinger::MixerThread::cacheParameters_l() 3529{ 3530 PlaybackThread::cacheParameters_l(); 3531 3532 // FIXME: Relaxed timing because of a certain device that can't meet latency 3533 // Should be reduced to 2x after the vendor fixes the driver issue 3534 // increase threshold again due to low power audio mode. The way this warning 3535 // threshold is calculated and its usefulness should be reconsidered anyway. 3536 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3537} 3538 3539// ---------------------------------------------------------------------------- 3540 3541AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3542 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3543 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3544 // mLeftVolFloat, mRightVolFloat 3545{ 3546} 3547 3548AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3549 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3550 ThreadBase::type_t type) 3551 : PlaybackThread(audioFlinger, output, id, device, type) 3552 // mLeftVolFloat, mRightVolFloat 3553{ 3554} 3555 3556AudioFlinger::DirectOutputThread::~DirectOutputThread() 3557{ 3558} 3559 3560void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3561{ 3562 audio_track_cblk_t* cblk = track->cblk(); 3563 float left, right; 3564 3565 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3566 left = right = 0; 3567 } else { 3568 float typeVolume = mStreamTypes[track->streamType()].volume; 3569 float v = mMasterVolume * typeVolume; 3570 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3571 uint32_t vlr = proxy->getVolumeLR(); 3572 float v_clamped = v * (vlr & 0xFFFF); 3573 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3574 left = v_clamped/MAX_GAIN; 3575 v_clamped = v * (vlr >> 16); 3576 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3577 right = v_clamped/MAX_GAIN; 3578 } 3579 3580 if (lastTrack) { 3581 if (left != mLeftVolFloat || right != mRightVolFloat) { 3582 mLeftVolFloat = left; 3583 mRightVolFloat = right; 3584 3585 // Convert volumes from float to 8.24 3586 uint32_t vl = (uint32_t)(left * (1 << 24)); 3587 uint32_t vr = (uint32_t)(right * (1 << 24)); 3588 3589 // Delegate volume control to effect in track effect chain if needed 3590 // only one effect chain can be present on DirectOutputThread, so if 3591 // there is one, the track is connected to it 3592 if (!mEffectChains.isEmpty()) { 3593 mEffectChains[0]->setVolume_l(&vl, &vr); 3594 left = (float)vl / (1 << 24); 3595 right = (float)vr / (1 << 24); 3596 } 3597 if (mOutput->stream->set_volume) { 3598 mOutput->stream->set_volume(mOutput->stream, left, right); 3599 } 3600 } 3601 } 3602} 3603 3604 3605AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3606 Vector< sp<Track> > *tracksToRemove 3607) 3608{ 3609 size_t count = mActiveTracks.size(); 3610 mixer_state mixerStatus = MIXER_IDLE; 3611 3612 // find out which tracks need to be processed 3613 for (size_t i = 0; i < count; i++) { 3614 sp<Track> t = mActiveTracks[i].promote(); 3615 // The track died recently 3616 if (t == 0) { 3617 continue; 3618 } 3619 3620 Track* const track = t.get(); 3621 audio_track_cblk_t* cblk = track->cblk(); 3622 // Only consider last track started for volume and mixer state control. 3623 // In theory an older track could underrun and restart after the new one starts 3624 // but as we only care about the transition phase between two tracks on a 3625 // direct output, it is not a problem to ignore the underrun case. 3626 sp<Track> l = mLatestActiveTrack.promote(); 3627 bool last = l.get() == track; 3628 3629 // The first time a track is added we wait 3630 // for all its buffers to be filled before processing it 3631 uint32_t minFrames; 3632 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3633 minFrames = mNormalFrameCount; 3634 } else { 3635 minFrames = 1; 3636 } 3637 3638 if ((track->framesReady() >= minFrames) && track->isReady() && 3639 !track->isPaused() && !track->isTerminated()) 3640 { 3641 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3642 3643 if (track->mFillingUpStatus == Track::FS_FILLED) { 3644 track->mFillingUpStatus = Track::FS_ACTIVE; 3645 // make sure processVolume_l() will apply new volume even if 0 3646 mLeftVolFloat = mRightVolFloat = -1.0; 3647 if (track->mState == TrackBase::RESUMING) { 3648 track->mState = TrackBase::ACTIVE; 3649 } 3650 } 3651 3652 // compute volume for this track 3653 processVolume_l(track, last); 3654 if (last) { 3655 // reset retry count 3656 track->mRetryCount = kMaxTrackRetriesDirect; 3657 mActiveTrack = t; 3658 mixerStatus = MIXER_TRACKS_READY; 3659 } 3660 } else { 3661 // clear effect chain input buffer if the last active track started underruns 3662 // to avoid sending previous audio buffer again to effects 3663 if (!mEffectChains.isEmpty() && last) { 3664 mEffectChains[0]->clearInputBuffer(); 3665 } 3666 3667 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3668 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3669 track->isStopped() || track->isPaused()) { 3670 // We have consumed all the buffers of this track. 3671 // Remove it from the list of active tracks. 3672 // TODO: implement behavior for compressed audio 3673 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3674 size_t framesWritten = mBytesWritten / mFrameSize; 3675 if (mStandby || !last || 3676 track->presentationComplete(framesWritten, audioHALFrames)) { 3677 if (track->isStopped()) { 3678 track->reset(); 3679 } 3680 tracksToRemove->add(track); 3681 } 3682 } else { 3683 // No buffers for this track. Give it a few chances to 3684 // fill a buffer, then remove it from active list. 3685 // Only consider last track started for mixer state control 3686 if (--(track->mRetryCount) <= 0) { 3687 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3688 tracksToRemove->add(track); 3689 // indicate to client process that the track was disabled because of underrun; 3690 // it will then automatically call start() when data is available 3691 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3692 } else if (last) { 3693 mixerStatus = MIXER_TRACKS_ENABLED; 3694 } 3695 } 3696 } 3697 } 3698 3699 // remove all the tracks that need to be... 3700 removeTracks_l(*tracksToRemove); 3701 3702 return mixerStatus; 3703} 3704 3705void AudioFlinger::DirectOutputThread::threadLoop_mix() 3706{ 3707 size_t frameCount = mFrameCount; 3708 int8_t *curBuf = (int8_t *)mMixBuffer; 3709 // output audio to hardware 3710 while (frameCount) { 3711 AudioBufferProvider::Buffer buffer; 3712 buffer.frameCount = frameCount; 3713 mActiveTrack->getNextBuffer(&buffer); 3714 if (buffer.raw == NULL) { 3715 memset(curBuf, 0, frameCount * mFrameSize); 3716 break; 3717 } 3718 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3719 frameCount -= buffer.frameCount; 3720 curBuf += buffer.frameCount * mFrameSize; 3721 mActiveTrack->releaseBuffer(&buffer); 3722 } 3723 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3724 sleepTime = 0; 3725 standbyTime = systemTime() + standbyDelay; 3726 mActiveTrack.clear(); 3727} 3728 3729void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3730{ 3731 if (sleepTime == 0) { 3732 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3733 sleepTime = activeSleepTime; 3734 } else { 3735 sleepTime = idleSleepTime; 3736 } 3737 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3738 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3739 sleepTime = 0; 3740 } 3741} 3742 3743// getTrackName_l() must be called with ThreadBase::mLock held 3744int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3745 int sessionId __unused) 3746{ 3747 return 0; 3748} 3749 3750// deleteTrackName_l() must be called with ThreadBase::mLock held 3751void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3752{ 3753} 3754 3755// checkForNewParameters_l() must be called with ThreadBase::mLock held 3756bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3757{ 3758 bool reconfig = false; 3759 3760 while (!mNewParameters.isEmpty()) { 3761 status_t status = NO_ERROR; 3762 String8 keyValuePair = mNewParameters[0]; 3763 AudioParameter param = AudioParameter(keyValuePair); 3764 int value; 3765 3766 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3767 // do not accept frame count changes if tracks are open as the track buffer 3768 // size depends on frame count and correct behavior would not be garantied 3769 // if frame count is changed after track creation 3770 if (!mTracks.isEmpty()) { 3771 status = INVALID_OPERATION; 3772 } else { 3773 reconfig = true; 3774 } 3775 } 3776 if (status == NO_ERROR) { 3777 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3778 keyValuePair.string()); 3779 if (!mStandby && status == INVALID_OPERATION) { 3780 mOutput->stream->common.standby(&mOutput->stream->common); 3781 mStandby = true; 3782 mBytesWritten = 0; 3783 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3784 keyValuePair.string()); 3785 } 3786 if (status == NO_ERROR && reconfig) { 3787 readOutputParameters(); 3788 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3789 } 3790 } 3791 3792 mNewParameters.removeAt(0); 3793 3794 mParamStatus = status; 3795 mParamCond.signal(); 3796 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3797 // already timed out waiting for the status and will never signal the condition. 3798 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3799 } 3800 return reconfig; 3801} 3802 3803uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3804{ 3805 uint32_t time; 3806 if (audio_is_linear_pcm(mFormat)) { 3807 time = PlaybackThread::activeSleepTimeUs(); 3808 } else { 3809 time = 10000; 3810 } 3811 return time; 3812} 3813 3814uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3815{ 3816 uint32_t time; 3817 if (audio_is_linear_pcm(mFormat)) { 3818 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3819 } else { 3820 time = 10000; 3821 } 3822 return time; 3823} 3824 3825uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3826{ 3827 uint32_t time; 3828 if (audio_is_linear_pcm(mFormat)) { 3829 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3830 } else { 3831 time = 10000; 3832 } 3833 return time; 3834} 3835 3836void AudioFlinger::DirectOutputThread::cacheParameters_l() 3837{ 3838 PlaybackThread::cacheParameters_l(); 3839 3840 // use shorter standby delay as on normal output to release 3841 // hardware resources as soon as possible 3842 if (audio_is_linear_pcm(mFormat)) { 3843 standbyDelay = microseconds(activeSleepTime*2); 3844 } else { 3845 standbyDelay = kOffloadStandbyDelayNs; 3846 } 3847} 3848 3849// ---------------------------------------------------------------------------- 3850 3851AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3852 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3853 : Thread(false /*canCallJava*/), 3854 mPlaybackThread(playbackThread), 3855 mWriteAckSequence(0), 3856 mDrainSequence(0) 3857{ 3858} 3859 3860AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3861{ 3862} 3863 3864void AudioFlinger::AsyncCallbackThread::onFirstRef() 3865{ 3866 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3867} 3868 3869bool AudioFlinger::AsyncCallbackThread::threadLoop() 3870{ 3871 while (!exitPending()) { 3872 uint32_t writeAckSequence; 3873 uint32_t drainSequence; 3874 3875 { 3876 Mutex::Autolock _l(mLock); 3877 while (!((mWriteAckSequence & 1) || 3878 (mDrainSequence & 1) || 3879 exitPending())) { 3880 mWaitWorkCV.wait(mLock); 3881 } 3882 3883 if (exitPending()) { 3884 break; 3885 } 3886 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3887 mWriteAckSequence, mDrainSequence); 3888 writeAckSequence = mWriteAckSequence; 3889 mWriteAckSequence &= ~1; 3890 drainSequence = mDrainSequence; 3891 mDrainSequence &= ~1; 3892 } 3893 { 3894 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3895 if (playbackThread != 0) { 3896 if (writeAckSequence & 1) { 3897 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3898 } 3899 if (drainSequence & 1) { 3900 playbackThread->resetDraining(drainSequence >> 1); 3901 } 3902 } 3903 } 3904 } 3905 return false; 3906} 3907 3908void AudioFlinger::AsyncCallbackThread::exit() 3909{ 3910 ALOGV("AsyncCallbackThread::exit"); 3911 Mutex::Autolock _l(mLock); 3912 requestExit(); 3913 mWaitWorkCV.broadcast(); 3914} 3915 3916void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3917{ 3918 Mutex::Autolock _l(mLock); 3919 // bit 0 is cleared 3920 mWriteAckSequence = sequence << 1; 3921} 3922 3923void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3924{ 3925 Mutex::Autolock _l(mLock); 3926 // ignore unexpected callbacks 3927 if (mWriteAckSequence & 2) { 3928 mWriteAckSequence |= 1; 3929 mWaitWorkCV.signal(); 3930 } 3931} 3932 3933void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3934{ 3935 Mutex::Autolock _l(mLock); 3936 // bit 0 is cleared 3937 mDrainSequence = sequence << 1; 3938} 3939 3940void AudioFlinger::AsyncCallbackThread::resetDraining() 3941{ 3942 Mutex::Autolock _l(mLock); 3943 // ignore unexpected callbacks 3944 if (mDrainSequence & 2) { 3945 mDrainSequence |= 1; 3946 mWaitWorkCV.signal(); 3947 } 3948} 3949 3950 3951// ---------------------------------------------------------------------------- 3952AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3953 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3954 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3955 mHwPaused(false), 3956 mFlushPending(false), 3957 mPausedBytesRemaining(0) 3958{ 3959 //FIXME: mStandby should be set to true by ThreadBase constructor 3960 mStandby = true; 3961} 3962 3963void AudioFlinger::OffloadThread::threadLoop_exit() 3964{ 3965 if (mFlushPending || mHwPaused) { 3966 // If a flush is pending or track was paused, just discard buffered data 3967 flushHw_l(); 3968 } else { 3969 mMixerStatus = MIXER_DRAIN_ALL; 3970 threadLoop_drain(); 3971 } 3972 mCallbackThread->exit(); 3973 PlaybackThread::threadLoop_exit(); 3974} 3975 3976AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3977 Vector< sp<Track> > *tracksToRemove 3978) 3979{ 3980 size_t count = mActiveTracks.size(); 3981 3982 mixer_state mixerStatus = MIXER_IDLE; 3983 bool doHwPause = false; 3984 bool doHwResume = false; 3985 3986 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3987 3988 // find out which tracks need to be processed 3989 for (size_t i = 0; i < count; i++) { 3990 sp<Track> t = mActiveTracks[i].promote(); 3991 // The track died recently 3992 if (t == 0) { 3993 continue; 3994 } 3995 Track* const track = t.get(); 3996 audio_track_cblk_t* cblk = track->cblk(); 3997 // Only consider last track started for volume and mixer state control. 3998 // In theory an older track could underrun and restart after the new one starts 3999 // but as we only care about the transition phase between two tracks on a 4000 // direct output, it is not a problem to ignore the underrun case. 4001 sp<Track> l = mLatestActiveTrack.promote(); 4002 bool last = l.get() == track; 4003 4004 if (track->isInvalid()) { 4005 ALOGW("An invalidated track shouldn't be in active list"); 4006 tracksToRemove->add(track); 4007 continue; 4008 } 4009 4010 if (track->mState == TrackBase::IDLE) { 4011 ALOGW("An idle track shouldn't be in active list"); 4012 continue; 4013 } 4014 4015 if (track->isPausing()) { 4016 track->setPaused(); 4017 if (last) { 4018 if (!mHwPaused) { 4019 doHwPause = true; 4020 mHwPaused = true; 4021 } 4022 // If we were part way through writing the mixbuffer to 4023 // the HAL we must save this until we resume 4024 // BUG - this will be wrong if a different track is made active, 4025 // in that case we want to discard the pending data in the 4026 // mixbuffer and tell the client to present it again when the 4027 // track is resumed 4028 mPausedWriteLength = mCurrentWriteLength; 4029 mPausedBytesRemaining = mBytesRemaining; 4030 mBytesRemaining = 0; // stop writing 4031 } 4032 tracksToRemove->add(track); 4033 } else if (track->isFlushPending()) { 4034 track->flushAck(); 4035 if (last) { 4036 mFlushPending = true; 4037 } 4038 } else if (track->framesReady() && track->isReady() && 4039 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4040 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4041 if (track->mFillingUpStatus == Track::FS_FILLED) { 4042 track->mFillingUpStatus = Track::FS_ACTIVE; 4043 // make sure processVolume_l() will apply new volume even if 0 4044 mLeftVolFloat = mRightVolFloat = -1.0; 4045 if (track->mState == TrackBase::RESUMING) { 4046 track->mState = TrackBase::ACTIVE; 4047 if (last) { 4048 if (mPausedBytesRemaining) { 4049 // Need to continue write that was interrupted 4050 mCurrentWriteLength = mPausedWriteLength; 4051 mBytesRemaining = mPausedBytesRemaining; 4052 mPausedBytesRemaining = 0; 4053 } 4054 if (mHwPaused) { 4055 doHwResume = true; 4056 mHwPaused = false; 4057 // threadLoop_mix() will handle the case that we need to 4058 // resume an interrupted write 4059 } 4060 // enable write to audio HAL 4061 sleepTime = 0; 4062 } 4063 } 4064 } 4065 4066 if (last) { 4067 sp<Track> previousTrack = mPreviousTrack.promote(); 4068 if (previousTrack != 0) { 4069 if (track != previousTrack.get()) { 4070 // Flush any data still being written from last track 4071 mBytesRemaining = 0; 4072 if (mPausedBytesRemaining) { 4073 // Last track was paused so we also need to flush saved 4074 // mixbuffer state and invalidate track so that it will 4075 // re-submit that unwritten data when it is next resumed 4076 mPausedBytesRemaining = 0; 4077 // Invalidate is a bit drastic - would be more efficient 4078 // to have a flag to tell client that some of the 4079 // previously written data was lost 4080 previousTrack->invalidate(); 4081 } 4082 // flush data already sent to the DSP if changing audio session as audio 4083 // comes from a different source. Also invalidate previous track to force a 4084 // seek when resuming. 4085 if (previousTrack->sessionId() != track->sessionId()) { 4086 previousTrack->invalidate(); 4087 } 4088 } 4089 } 4090 mPreviousTrack = track; 4091 // reset retry count 4092 track->mRetryCount = kMaxTrackRetriesOffload; 4093 mActiveTrack = t; 4094 mixerStatus = MIXER_TRACKS_READY; 4095 } 4096 } else { 4097 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4098 if (track->isStopping_1()) { 4099 // Hardware buffer can hold a large amount of audio so we must 4100 // wait for all current track's data to drain before we say 4101 // that the track is stopped. 4102 if (mBytesRemaining == 0) { 4103 // Only start draining when all data in mixbuffer 4104 // has been written 4105 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4106 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4107 // do not drain if no data was ever sent to HAL (mStandby == true) 4108 if (last && !mStandby) { 4109 // do not modify drain sequence if we are already draining. This happens 4110 // when resuming from pause after drain. 4111 if ((mDrainSequence & 1) == 0) { 4112 sleepTime = 0; 4113 standbyTime = systemTime() + standbyDelay; 4114 mixerStatus = MIXER_DRAIN_TRACK; 4115 mDrainSequence += 2; 4116 } 4117 if (mHwPaused) { 4118 // It is possible to move from PAUSED to STOPPING_1 without 4119 // a resume so we must ensure hardware is running 4120 doHwResume = true; 4121 mHwPaused = false; 4122 } 4123 } 4124 } 4125 } else if (track->isStopping_2()) { 4126 // Drain has completed or we are in standby, signal presentation complete 4127 if (!(mDrainSequence & 1) || !last || mStandby) { 4128 track->mState = TrackBase::STOPPED; 4129 size_t audioHALFrames = 4130 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4131 size_t framesWritten = 4132 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4133 track->presentationComplete(framesWritten, audioHALFrames); 4134 track->reset(); 4135 tracksToRemove->add(track); 4136 } 4137 } else { 4138 // No buffers for this track. Give it a few chances to 4139 // fill a buffer, then remove it from active list. 4140 if (--(track->mRetryCount) <= 0) { 4141 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4142 track->name()); 4143 tracksToRemove->add(track); 4144 // indicate to client process that the track was disabled because of underrun; 4145 // it will then automatically call start() when data is available 4146 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4147 } else if (last){ 4148 mixerStatus = MIXER_TRACKS_ENABLED; 4149 } 4150 } 4151 } 4152 // compute volume for this track 4153 processVolume_l(track, last); 4154 } 4155 4156 // make sure the pause/flush/resume sequence is executed in the right order. 4157 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4158 // before flush and then resume HW. This can happen in case of pause/flush/resume 4159 // if resume is received before pause is executed. 4160 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4161 mOutput->stream->pause(mOutput->stream); 4162 } 4163 if (mFlushPending) { 4164 flushHw_l(); 4165 mFlushPending = false; 4166 } 4167 if (!mStandby && doHwResume) { 4168 mOutput->stream->resume(mOutput->stream); 4169 } 4170 4171 // remove all the tracks that need to be... 4172 removeTracks_l(*tracksToRemove); 4173 4174 return mixerStatus; 4175} 4176 4177// must be called with thread mutex locked 4178bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4179{ 4180 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4181 mWriteAckSequence, mDrainSequence); 4182 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4183 return true; 4184 } 4185 return false; 4186} 4187 4188// must be called with thread mutex locked 4189bool AudioFlinger::OffloadThread::shouldStandby_l() 4190{ 4191 bool trackPaused = false; 4192 4193 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4194 // after a timeout and we will enter standby then. 4195 if (mTracks.size() > 0) { 4196 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4197 } 4198 4199 return !mStandby && !trackPaused; 4200} 4201 4202 4203bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4204{ 4205 Mutex::Autolock _l(mLock); 4206 return waitingAsyncCallback_l(); 4207} 4208 4209void AudioFlinger::OffloadThread::flushHw_l() 4210{ 4211 mOutput->stream->flush(mOutput->stream); 4212 // Flush anything still waiting in the mixbuffer 4213 mCurrentWriteLength = 0; 4214 mBytesRemaining = 0; 4215 mPausedWriteLength = 0; 4216 mPausedBytesRemaining = 0; 4217 mHwPaused = false; 4218 4219 if (mUseAsyncWrite) { 4220 // discard any pending drain or write ack by incrementing sequence 4221 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4222 mDrainSequence = (mDrainSequence + 2) & ~1; 4223 ALOG_ASSERT(mCallbackThread != 0); 4224 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4225 mCallbackThread->setDraining(mDrainSequence); 4226 } 4227} 4228 4229void AudioFlinger::OffloadThread::onAddNewTrack_l() 4230{ 4231 sp<Track> previousTrack = mPreviousTrack.promote(); 4232 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4233 4234 if (previousTrack != 0 && latestTrack != 0 && 4235 (previousTrack->sessionId() != latestTrack->sessionId())) { 4236 mFlushPending = true; 4237 } 4238 PlaybackThread::onAddNewTrack_l(); 4239} 4240 4241// ---------------------------------------------------------------------------- 4242 4243AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4244 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4245 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4246 DUPLICATING), 4247 mWaitTimeMs(UINT_MAX) 4248{ 4249 addOutputTrack(mainThread); 4250} 4251 4252AudioFlinger::DuplicatingThread::~DuplicatingThread() 4253{ 4254 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4255 mOutputTracks[i]->destroy(); 4256 } 4257} 4258 4259void AudioFlinger::DuplicatingThread::threadLoop_mix() 4260{ 4261 // mix buffers... 4262 if (outputsReady(outputTracks)) { 4263 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4264 } else { 4265 memset(mMixBuffer, 0, mixBufferSize); 4266 } 4267 sleepTime = 0; 4268 writeFrames = mNormalFrameCount; 4269 mCurrentWriteLength = mixBufferSize; 4270 standbyTime = systemTime() + standbyDelay; 4271} 4272 4273void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4274{ 4275 if (sleepTime == 0) { 4276 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4277 sleepTime = activeSleepTime; 4278 } else { 4279 sleepTime = idleSleepTime; 4280 } 4281 } else if (mBytesWritten != 0) { 4282 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4283 writeFrames = mNormalFrameCount; 4284 memset(mMixBuffer, 0, mixBufferSize); 4285 } else { 4286 // flush remaining overflow buffers in output tracks 4287 writeFrames = 0; 4288 } 4289 sleepTime = 0; 4290 } 4291} 4292 4293ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4294{ 4295 for (size_t i = 0; i < outputTracks.size(); i++) { 4296 outputTracks[i]->write(mMixBuffer, writeFrames); 4297 } 4298 mStandby = false; 4299 return (ssize_t)mixBufferSize; 4300} 4301 4302void AudioFlinger::DuplicatingThread::threadLoop_standby() 4303{ 4304 // DuplicatingThread implements standby by stopping all tracks 4305 for (size_t i = 0; i < outputTracks.size(); i++) { 4306 outputTracks[i]->stop(); 4307 } 4308} 4309 4310void AudioFlinger::DuplicatingThread::saveOutputTracks() 4311{ 4312 outputTracks = mOutputTracks; 4313} 4314 4315void AudioFlinger::DuplicatingThread::clearOutputTracks() 4316{ 4317 outputTracks.clear(); 4318} 4319 4320void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4321{ 4322 Mutex::Autolock _l(mLock); 4323 // FIXME explain this formula 4324 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4325 OutputTrack *outputTrack = new OutputTrack(thread, 4326 this, 4327 mSampleRate, 4328 mFormat, 4329 mChannelMask, 4330 frameCount, 4331 IPCThreadState::self()->getCallingUid()); 4332 if (outputTrack->cblk() != NULL) { 4333 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4334 mOutputTracks.add(outputTrack); 4335 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4336 updateWaitTime_l(); 4337 } 4338} 4339 4340void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4341{ 4342 Mutex::Autolock _l(mLock); 4343 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4344 if (mOutputTracks[i]->thread() == thread) { 4345 mOutputTracks[i]->destroy(); 4346 mOutputTracks.removeAt(i); 4347 updateWaitTime_l(); 4348 return; 4349 } 4350 } 4351 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4352} 4353 4354// caller must hold mLock 4355void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4356{ 4357 mWaitTimeMs = UINT_MAX; 4358 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4359 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4360 if (strong != 0) { 4361 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4362 if (waitTimeMs < mWaitTimeMs) { 4363 mWaitTimeMs = waitTimeMs; 4364 } 4365 } 4366 } 4367} 4368 4369 4370bool AudioFlinger::DuplicatingThread::outputsReady( 4371 const SortedVector< sp<OutputTrack> > &outputTracks) 4372{ 4373 for (size_t i = 0; i < outputTracks.size(); i++) { 4374 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4375 if (thread == 0) { 4376 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4377 outputTracks[i].get()); 4378 return false; 4379 } 4380 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4381 // see note at standby() declaration 4382 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4383 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4384 thread.get()); 4385 return false; 4386 } 4387 } 4388 return true; 4389} 4390 4391uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4392{ 4393 return (mWaitTimeMs * 1000) / 2; 4394} 4395 4396void AudioFlinger::DuplicatingThread::cacheParameters_l() 4397{ 4398 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4399 updateWaitTime_l(); 4400 4401 MixerThread::cacheParameters_l(); 4402} 4403 4404// ---------------------------------------------------------------------------- 4405// Record 4406// ---------------------------------------------------------------------------- 4407 4408AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4409 AudioStreamIn *input, 4410 uint32_t sampleRate, 4411 audio_channel_mask_t channelMask, 4412 audio_io_handle_t id, 4413 audio_devices_t outDevice, 4414 audio_devices_t inDevice 4415#ifdef TEE_SINK 4416 , const sp<NBAIO_Sink>& teeSink 4417#endif 4418 ) : 4419 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4420 mInput(input), mActiveTracksGen(0), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4421 // mRsmpInFrames, mRsmpInFramesP2, mRsmpInUnrel, mRsmpInFront, and mRsmpInRear 4422 // are set by readInputParameters() 4423 // mRsmpInIndex LEGACY 4424 mReqChannelCount(popcount(channelMask)), 4425 mReqSampleRate(sampleRate) 4426 // mBytesRead is only meaningful while active, and so is cleared in start() 4427 // (but might be better to also clear here for dump?) 4428#ifdef TEE_SINK 4429 , mTeeSink(teeSink) 4430#endif 4431{ 4432 snprintf(mName, kNameLength, "AudioIn_%X", id); 4433 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4434 4435 readInputParameters(); 4436} 4437 4438 4439AudioFlinger::RecordThread::~RecordThread() 4440{ 4441 mAudioFlinger->unregisterWriter(mNBLogWriter); 4442 delete[] mRsmpInBuffer; 4443 delete mResampler; 4444 delete[] mRsmpOutBuffer; 4445} 4446 4447void AudioFlinger::RecordThread::onFirstRef() 4448{ 4449 run(mName, PRIORITY_URGENT_AUDIO); 4450} 4451 4452bool AudioFlinger::RecordThread::threadLoop() 4453{ 4454 nsecs_t lastWarning = 0; 4455 4456 inputStandBy(); 4457 4458 // used to verify we've read at least once before evaluating how many bytes were read 4459 bool readOnce = false; 4460 4461 // used to request a deferred sleep, to be executed later while mutex is unlocked 4462 bool doSleep = false; 4463 4464reacquire_wakelock: 4465 sp<RecordTrack> activeTrack; 4466 int activeTracksGen; 4467 { 4468 Mutex::Autolock _l(mLock); 4469 size_t size = mActiveTracks.size(); 4470 activeTracksGen = mActiveTracksGen; 4471 if (size > 0) { 4472 // FIXME an arbitrary choice 4473 activeTrack = mActiveTracks[0]; 4474 acquireWakeLock_l(activeTrack->uid()); 4475 if (size > 1) { 4476 SortedVector<int> tmp; 4477 for (size_t i = 0; i < size; i++) { 4478 tmp.add(mActiveTracks[i]->uid()); 4479 } 4480 updateWakeLockUids_l(tmp); 4481 } 4482 } else { 4483 acquireWakeLock_l(-1); 4484 } 4485 } 4486 4487 // start recording 4488 for (;;) { 4489 TrackBase::track_state activeTrackState; 4490 Vector< sp<EffectChain> > effectChains; 4491 4492 // sleep with mutex unlocked 4493 if (doSleep) { 4494 doSleep = false; 4495 usleep(kRecordThreadSleepUs); 4496 } 4497 4498 { // scope for mLock 4499 Mutex::Autolock _l(mLock); 4500 4501 processConfigEvents_l(); 4502 // return value 'reconfig' is currently unused 4503 bool reconfig = checkForNewParameters_l(); 4504 4505 // check exitPending here because checkForNewParameters_l() and 4506 // checkForNewParameters_l() can temporarily release mLock 4507 if (exitPending()) { 4508 break; 4509 } 4510 4511 // if no active track(s), then standby and release wakelock 4512 size_t size = mActiveTracks.size(); 4513 if (size == 0) { 4514 standbyIfNotAlreadyInStandby(); 4515 // exitPending() can't become true here 4516 releaseWakeLock_l(); 4517 ALOGV("RecordThread: loop stopping"); 4518 // go to sleep 4519 mWaitWorkCV.wait(mLock); 4520 ALOGV("RecordThread: loop starting"); 4521 goto reacquire_wakelock; 4522 } 4523 4524 if (mActiveTracksGen != activeTracksGen) { 4525 activeTracksGen = mActiveTracksGen; 4526 SortedVector<int> tmp; 4527 for (size_t i = 0; i < size; i++) { 4528 tmp.add(mActiveTracks[i]->uid()); 4529 } 4530 updateWakeLockUids_l(tmp); 4531 // FIXME an arbitrary choice 4532 activeTrack = mActiveTracks[0]; 4533 } 4534 4535 if (activeTrack->isTerminated()) { 4536 removeTrack_l(activeTrack); 4537 mActiveTracks.remove(activeTrack); 4538 mActiveTracksGen++; 4539 continue; 4540 } 4541 4542 activeTrackState = activeTrack->mState; 4543 switch (activeTrackState) { 4544 case TrackBase::PAUSING: 4545 standbyIfNotAlreadyInStandby(); 4546 mActiveTracks.remove(activeTrack); 4547 mActiveTracksGen++; 4548 mStartStopCond.broadcast(); 4549 doSleep = true; 4550 continue; 4551 4552 case TrackBase::RESUMING: 4553 mStandby = false; 4554 if (mReqChannelCount != activeTrack->channelCount()) { 4555 mActiveTracks.remove(activeTrack); 4556 mActiveTracksGen++; 4557 mStartStopCond.broadcast(); 4558 continue; 4559 } 4560 if (readOnce) { 4561 mStartStopCond.broadcast(); 4562 // record start succeeds only if first read from audio input succeeds 4563 if (mBytesRead < 0) { 4564 mActiveTracks.remove(activeTrack); 4565 mActiveTracksGen++; 4566 continue; 4567 } 4568 activeTrack->mState = TrackBase::ACTIVE; 4569 } 4570 break; 4571 4572 case TrackBase::ACTIVE: 4573 break; 4574 4575 case TrackBase::IDLE: 4576 doSleep = true; 4577 continue; 4578 4579 default: 4580 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4581 } 4582 4583 lockEffectChains_l(effectChains); 4584 } 4585 4586 // thread mutex is now unlocked, mActiveTracks unknown, activeTrack != 0, kept, immutable 4587 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4588 4589 for (size_t i = 0; i < effectChains.size(); i ++) { 4590 // thread mutex is not locked, but effect chain is locked 4591 effectChains[i]->process_l(); 4592 } 4593 4594 AudioBufferProvider::Buffer buffer; 4595 buffer.frameCount = mFrameCount; 4596 status_t status = activeTrack->getNextBuffer(&buffer); 4597 if (status == NO_ERROR) { 4598 readOnce = true; 4599 size_t framesOut = buffer.frameCount; 4600 if (mResampler == NULL) { 4601 // no resampling 4602 while (framesOut) { 4603 size_t framesIn = mFrameCount - mRsmpInIndex; 4604 if (framesIn > 0) { 4605 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4606 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4607 activeTrack->mFrameSize; 4608 if (framesIn > framesOut) { 4609 framesIn = framesOut; 4610 } 4611 mRsmpInIndex += framesIn; 4612 framesOut -= framesIn; 4613 if (mChannelCount == mReqChannelCount) { 4614 memcpy(dst, src, framesIn * mFrameSize); 4615 } else { 4616 if (mChannelCount == 1) { 4617 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4618 (int16_t *)src, framesIn); 4619 } else { 4620 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4621 (int16_t *)src, framesIn); 4622 } 4623 } 4624 } 4625 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4626 void *readInto; 4627 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4628 readInto = buffer.raw; 4629 framesOut = 0; 4630 } else { 4631 readInto = mRsmpInBuffer; 4632 mRsmpInIndex = 0; 4633 } 4634 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4635 mBufferSize); 4636 if (mBytesRead <= 0) { 4637 // TODO: verify that it's benign to use a stale track state 4638 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4639 { 4640 ALOGE("Error reading audio input"); 4641 // Force input into standby so that it tries to 4642 // recover at next read attempt 4643 inputStandBy(); 4644 doSleep = true; 4645 } 4646 mRsmpInIndex = mFrameCount; 4647 framesOut = 0; 4648 buffer.frameCount = 0; 4649 } 4650#ifdef TEE_SINK 4651 else if (mTeeSink != 0) { 4652 (void) mTeeSink->write(readInto, 4653 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4654 } 4655#endif 4656 } 4657 } 4658 } else { 4659 // resampling 4660 4661 // avoid busy-waiting if client doesn't keep up 4662 bool madeProgress = false; 4663 4664 // keep mRsmpInBuffer full so resampler always has sufficient input 4665 for (;;) { 4666 int32_t rear = mRsmpInRear; 4667 ssize_t filled = rear - mRsmpInFront; 4668 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 4669 // exit once there is enough data in buffer for resampler 4670 if ((size_t) filled >= mRsmpInFrames) { 4671 break; 4672 } 4673 size_t avail = mRsmpInFramesP2 - filled; 4674 // Only try to read full HAL buffers. 4675 // But if the HAL read returns a partial buffer, use it. 4676 if (avail < mFrameCount) { 4677 ALOGE("insufficient space to read: avail %d < mFrameCount %d", 4678 avail, mFrameCount); 4679 break; 4680 } 4681 // If 'avail' is non-contiguous, first read past the nominal end of buffer, then 4682 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4683 rear &= mRsmpInFramesP2 - 1; 4684 mBytesRead = mInput->stream->read(mInput->stream, 4685 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4686 if (mBytesRead <= 0) { 4687 ALOGE("read failed: mBytesRead=%d < %u", mBytesRead, mBufferSize); 4688 break; 4689 } 4690 ALOG_ASSERT((size_t) mBytesRead <= mBufferSize); 4691 size_t framesRead = mBytesRead / mFrameSize; 4692 ALOG_ASSERT(framesRead > 0); 4693 madeProgress = true; 4694 // If 'avail' was non-contiguous, we now correct for reading past end of buffer. 4695 size_t part1 = mRsmpInFramesP2 - rear; 4696 if (framesRead > part1) { 4697 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4698 (framesRead - part1) * mFrameSize); 4699 } 4700 mRsmpInRear += framesRead; 4701 } 4702 4703 if (!madeProgress) { 4704 ALOGV("Did not make progress"); 4705 usleep(((mFrameCount * 1000) / mSampleRate) * 1000); 4706 } 4707 4708 // resampler accumulates, but we only have one source track 4709 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4710 mResampler->resample(mRsmpOutBuffer, framesOut, 4711 this /* AudioBufferProvider* */); 4712 // ditherAndClamp() works as long as all buffers returned by 4713 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4714 if (mReqChannelCount == 1) { 4715 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4716 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4717 // the resampler always outputs stereo samples: 4718 // do post stereo to mono conversion 4719 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4720 framesOut); 4721 } else { 4722 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4723 } 4724 // now done with mRsmpOutBuffer 4725 4726 } 4727 if (mFramestoDrop == 0) { 4728 activeTrack->releaseBuffer(&buffer); 4729 } else { 4730 if (mFramestoDrop > 0) { 4731 mFramestoDrop -= buffer.frameCount; 4732 if (mFramestoDrop <= 0) { 4733 clearSyncStartEvent(); 4734 } 4735 } else { 4736 mFramestoDrop += buffer.frameCount; 4737 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4738 mSyncStartEvent->isCancelled()) { 4739 ALOGW("Synced record %s, session %d, trigger session %d", 4740 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4741 activeTrack->sessionId(), 4742 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4743 clearSyncStartEvent(); 4744 } 4745 } 4746 } 4747 activeTrack->clearOverflow(); 4748 } 4749 // client isn't retrieving buffers fast enough 4750 else { 4751 if (!activeTrack->setOverflow()) { 4752 nsecs_t now = systemTime(); 4753 if ((now - lastWarning) > kWarningThrottleNs) { 4754 ALOGW("RecordThread: buffer overflow"); 4755 lastWarning = now; 4756 } 4757 } 4758 // Release the processor for a while before asking for a new buffer. 4759 // This will give the application more chance to read from the buffer and 4760 // clear the overflow. 4761 doSleep = true; 4762 } 4763 4764 // enable changes in effect chain 4765 unlockEffectChains(effectChains); 4766 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4767 } 4768 4769 standbyIfNotAlreadyInStandby(); 4770 4771 { 4772 Mutex::Autolock _l(mLock); 4773 for (size_t i = 0; i < mTracks.size(); i++) { 4774 sp<RecordTrack> track = mTracks[i]; 4775 track->invalidate(); 4776 } 4777 mActiveTracks.clear(); 4778 mActiveTracksGen++; 4779 mStartStopCond.broadcast(); 4780 } 4781 4782 releaseWakeLock(); 4783 4784 ALOGV("RecordThread %p exiting", this); 4785 return false; 4786} 4787 4788void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4789{ 4790 if (!mStandby) { 4791 inputStandBy(); 4792 mStandby = true; 4793 } 4794} 4795 4796void AudioFlinger::RecordThread::inputStandBy() 4797{ 4798 mInput->stream->common.standby(&mInput->stream->common); 4799} 4800 4801sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4802 const sp<AudioFlinger::Client>& client, 4803 uint32_t sampleRate, 4804 audio_format_t format, 4805 audio_channel_mask_t channelMask, 4806 size_t *pFrameCount, 4807 int sessionId, 4808 int uid, 4809 IAudioFlinger::track_flags_t *flags, 4810 pid_t tid, 4811 status_t *status) 4812{ 4813 size_t frameCount = *pFrameCount; 4814 sp<RecordTrack> track; 4815 status_t lStatus; 4816 4817 lStatus = initCheck(); 4818 if (lStatus != NO_ERROR) { 4819 ALOGE("createRecordTrack_l() audio driver not initialized"); 4820 goto Exit; 4821 } 4822 // client expresses a preference for FAST, but we get the final say 4823 if (*flags & IAudioFlinger::TRACK_FAST) { 4824 if ( 4825 // use case: callback handler and frame count is default or at least as large as HAL 4826 ( 4827 (tid != -1) && 4828 ((frameCount == 0) || 4829 (frameCount >= mFrameCount)) 4830 ) && 4831 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4832 // mono or stereo 4833 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4834 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4835 // hardware sample rate 4836 (sampleRate == mSampleRate) && 4837 // record thread has an associated fast recorder 4838 hasFastRecorder() 4839 // FIXME test that RecordThread for this fast track has a capable output HAL 4840 // FIXME add a permission test also? 4841 ) { 4842 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4843 if (frameCount == 0) { 4844 frameCount = mFrameCount * kFastTrackMultiplier; 4845 } 4846 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4847 frameCount, mFrameCount); 4848 } else { 4849 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4850 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4851 "hasFastRecorder=%d tid=%d", 4852 frameCount, mFrameCount, format, 4853 audio_is_linear_pcm(format), 4854 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4855 *flags &= ~IAudioFlinger::TRACK_FAST; 4856 // For compatibility with AudioRecord calculation, buffer depth is forced 4857 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4858 // This is probably too conservative, but legacy application code may depend on it. 4859 // If you change this calculation, also review the start threshold which is related. 4860 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4861 size_t mNormalFrameCount = 2048; // FIXME 4862 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4863 if (minBufCount < 2) { 4864 minBufCount = 2; 4865 } 4866 size_t minFrameCount = mNormalFrameCount * minBufCount; 4867 if (frameCount < minFrameCount) { 4868 frameCount = minFrameCount; 4869 } 4870 } 4871 } 4872 *pFrameCount = frameCount; 4873 4874 // FIXME use flags and tid similar to createTrack_l() 4875 4876 { // scope for mLock 4877 Mutex::Autolock _l(mLock); 4878 4879 track = new RecordTrack(this, client, sampleRate, 4880 format, channelMask, frameCount, sessionId, uid); 4881 4882 lStatus = track->initCheck(); 4883 if (lStatus != NO_ERROR) { 4884 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 4885 // track must be cleared from the caller as the caller has the AF lock 4886 goto Exit; 4887 } 4888 mTracks.add(track); 4889 4890 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4891 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4892 mAudioFlinger->btNrecIsOff(); 4893 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4894 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4895 4896 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4897 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4898 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4899 // so ask activity manager to do this on our behalf 4900 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4901 } 4902 } 4903 lStatus = NO_ERROR; 4904 4905Exit: 4906 *status = lStatus; 4907 return track; 4908} 4909 4910status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4911 AudioSystem::sync_event_t event, 4912 int triggerSession) 4913{ 4914 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4915 sp<ThreadBase> strongMe = this; 4916 status_t status = NO_ERROR; 4917 4918 if (event == AudioSystem::SYNC_EVENT_NONE) { 4919 clearSyncStartEvent(); 4920 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4921 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4922 triggerSession, 4923 recordTrack->sessionId(), 4924 syncStartEventCallback, 4925 this); 4926 // Sync event can be cancelled by the trigger session if the track is not in a 4927 // compatible state in which case we start record immediately 4928 if (mSyncStartEvent->isCancelled()) { 4929 clearSyncStartEvent(); 4930 } else { 4931 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4932 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4933 } 4934 } 4935 4936 { 4937 // This section is a rendezvous between binder thread executing start() and RecordThread 4938 AutoMutex lock(mLock); 4939 if (mActiveTracks.size() > 0) { 4940 // FIXME does not work for multiple active tracks 4941 if (mActiveTracks.indexOf(recordTrack) != 0) { 4942 status = -EBUSY; 4943 } else if (recordTrack->mState == TrackBase::PAUSING) { 4944 recordTrack->mState = TrackBase::ACTIVE; 4945 } 4946 return status; 4947 } 4948 4949 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4950 recordTrack->mState = TrackBase::IDLE; 4951 mActiveTracks.add(recordTrack); 4952 mActiveTracksGen++; 4953 mLock.unlock(); 4954 status_t status = AudioSystem::startInput(mId); 4955 mLock.lock(); 4956 // FIXME should verify that mActiveTrack is still == recordTrack 4957 if (status != NO_ERROR) { 4958 mActiveTracks.remove(recordTrack); 4959 mActiveTracksGen++; 4960 clearSyncStartEvent(); 4961 return status; 4962 } 4963 // FIXME LEGACY 4964 mRsmpInIndex = mFrameCount; 4965 mRsmpInFront = 0; 4966 mRsmpInRear = 0; 4967 mRsmpInUnrel = 0; 4968 mBytesRead = 0; 4969 if (mResampler != NULL) { 4970 mResampler->reset(); 4971 } 4972 // FIXME hijacking a playback track state name which was intended for start after pause; 4973 // here 'STARTING_2' would be more accurate 4974 recordTrack->mState = TrackBase::RESUMING; 4975 // signal thread to start 4976 ALOGV("Signal record thread"); 4977 mWaitWorkCV.broadcast(); 4978 // do not wait for mStartStopCond if exiting 4979 if (exitPending()) { 4980 mActiveTracks.remove(recordTrack); 4981 mActiveTracksGen++; 4982 status = INVALID_OPERATION; 4983 goto startError; 4984 } 4985 // FIXME incorrect usage of wait: no explicit predicate or loop 4986 mStartStopCond.wait(mLock); 4987 if (mActiveTracks.indexOf(recordTrack) < 0) { 4988 ALOGV("Record failed to start"); 4989 status = BAD_VALUE; 4990 goto startError; 4991 } 4992 ALOGV("Record started OK"); 4993 return status; 4994 } 4995 4996startError: 4997 AudioSystem::stopInput(mId); 4998 clearSyncStartEvent(); 4999 return status; 5000} 5001 5002void AudioFlinger::RecordThread::clearSyncStartEvent() 5003{ 5004 if (mSyncStartEvent != 0) { 5005 mSyncStartEvent->cancel(); 5006 } 5007 mSyncStartEvent.clear(); 5008 mFramestoDrop = 0; 5009} 5010 5011void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5012{ 5013 sp<SyncEvent> strongEvent = event.promote(); 5014 5015 if (strongEvent != 0) { 5016 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5017 me->handleSyncStartEvent(strongEvent); 5018 } 5019} 5020 5021void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5022{ 5023 if (event == mSyncStartEvent) { 5024 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5025 // from audio HAL 5026 mFramestoDrop = mFrameCount * 2; 5027 } 5028} 5029 5030bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5031 ALOGV("RecordThread::stop"); 5032 AutoMutex _l(mLock); 5033 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5034 return false; 5035 } 5036 // note that threadLoop may still be processing the track at this point [without lock] 5037 recordTrack->mState = TrackBase::PAUSING; 5038 // do not wait for mStartStopCond if exiting 5039 if (exitPending()) { 5040 return true; 5041 } 5042 // FIXME incorrect usage of wait: no explicit predicate or loop 5043 mStartStopCond.wait(mLock); 5044 // if we have been restarted, recordTrack is in mActiveTracks here 5045 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5046 ALOGV("Record stopped OK"); 5047 return true; 5048 } 5049 return false; 5050} 5051 5052bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5053{ 5054 return false; 5055} 5056 5057status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5058{ 5059#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5060 if (!isValidSyncEvent(event)) { 5061 return BAD_VALUE; 5062 } 5063 5064 int eventSession = event->triggerSession(); 5065 status_t ret = NAME_NOT_FOUND; 5066 5067 Mutex::Autolock _l(mLock); 5068 5069 for (size_t i = 0; i < mTracks.size(); i++) { 5070 sp<RecordTrack> track = mTracks[i]; 5071 if (eventSession == track->sessionId()) { 5072 (void) track->setSyncEvent(event); 5073 ret = NO_ERROR; 5074 } 5075 } 5076 return ret; 5077#else 5078 return BAD_VALUE; 5079#endif 5080} 5081 5082// destroyTrack_l() must be called with ThreadBase::mLock held 5083void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5084{ 5085 track->terminate(); 5086 track->mState = TrackBase::STOPPED; 5087 // active tracks are removed by threadLoop() 5088 if (mActiveTracks.indexOf(track) < 0) { 5089 removeTrack_l(track); 5090 } 5091} 5092 5093void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5094{ 5095 mTracks.remove(track); 5096 // need anything related to effects here? 5097} 5098 5099void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5100{ 5101 dumpInternals(fd, args); 5102 dumpTracks(fd, args); 5103 dumpEffectChains(fd, args); 5104} 5105 5106void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5107{ 5108 const size_t SIZE = 256; 5109 char buffer[SIZE]; 5110 String8 result; 5111 5112 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5113 result.append(buffer); 5114 5115 if (mActiveTracks.size() > 0) { 5116 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5117 result.append(buffer); 5118 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 5119 result.append(buffer); 5120 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5121 result.append(buffer); 5122 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 5123 result.append(buffer); 5124 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 5125 result.append(buffer); 5126 } else { 5127 result.append("No active record client\n"); 5128 } 5129 5130 write(fd, result.string(), result.size()); 5131 5132 dumpBase(fd, args); 5133} 5134 5135void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5136{ 5137 const size_t SIZE = 256; 5138 char buffer[SIZE]; 5139 String8 result; 5140 5141 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 5142 result.append(buffer); 5143 RecordTrack::appendDumpHeader(result); 5144 for (size_t i = 0; i < mTracks.size(); ++i) { 5145 sp<RecordTrack> track = mTracks[i]; 5146 if (track != 0) { 5147 track->dump(buffer, SIZE); 5148 result.append(buffer); 5149 } 5150 } 5151 5152 size_t size = mActiveTracks.size(); 5153 if (size > 0) { 5154 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5155 result.append(buffer); 5156 RecordTrack::appendDumpHeader(result); 5157 for (size_t i = 0; i < size; ++i) { 5158 sp<RecordTrack> track = mActiveTracks[i]; 5159 track->dump(buffer, SIZE); 5160 result.append(buffer); 5161 } 5162 5163 } 5164 write(fd, result.string(), result.size()); 5165} 5166 5167// AudioBufferProvider interface 5168status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5169{ 5170 int32_t rear = mRsmpInRear; 5171 int32_t front = mRsmpInFront; 5172 ssize_t filled = rear - front; 5173 ALOG_ASSERT(0 <= filled && (size_t) filled <= mRsmpInFramesP2); 5174 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5175 front &= mRsmpInFramesP2 - 1; 5176 size_t part1 = mRsmpInFramesP2 - front; 5177 if (part1 > (size_t) filled) { 5178 part1 = filled; 5179 } 5180 size_t ask = buffer->frameCount; 5181 ALOG_ASSERT(ask > 0); 5182 if (part1 > ask) { 5183 part1 = ask; 5184 } 5185 if (part1 == 0) { 5186 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5187 ALOGE("RecordThread::getNextBuffer() starved"); 5188 buffer->raw = NULL; 5189 buffer->frameCount = 0; 5190 mRsmpInUnrel = 0; 5191 return NOT_ENOUGH_DATA; 5192 } 5193 5194 buffer->raw = mRsmpInBuffer + front * mChannelCount; 5195 buffer->frameCount = part1; 5196 mRsmpInUnrel = part1; 5197 return NO_ERROR; 5198} 5199 5200// AudioBufferProvider interface 5201void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5202{ 5203 size_t stepCount = buffer->frameCount; 5204 if (stepCount == 0) { 5205 return; 5206 } 5207 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 5208 mRsmpInUnrel -= stepCount; 5209 mRsmpInFront += stepCount; 5210 buffer->raw = NULL; 5211 buffer->frameCount = 0; 5212} 5213 5214bool AudioFlinger::RecordThread::checkForNewParameters_l() 5215{ 5216 bool reconfig = false; 5217 5218 while (!mNewParameters.isEmpty()) { 5219 status_t status = NO_ERROR; 5220 String8 keyValuePair = mNewParameters[0]; 5221 AudioParameter param = AudioParameter(keyValuePair); 5222 int value; 5223 audio_format_t reqFormat = mFormat; 5224 uint32_t reqSamplingRate = mReqSampleRate; 5225 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 5226 5227 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5228 reqSamplingRate = value; 5229 reconfig = true; 5230 } 5231 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5232 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5233 status = BAD_VALUE; 5234 } else { 5235 reqFormat = (audio_format_t) value; 5236 reconfig = true; 5237 } 5238 } 5239 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5240 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5241 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5242 status = BAD_VALUE; 5243 } else { 5244 reqChannelMask = mask; 5245 reconfig = true; 5246 } 5247 } 5248 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5249 // do not accept frame count changes if tracks are open as the track buffer 5250 // size depends on frame count and correct behavior would not be guaranteed 5251 // if frame count is changed after track creation 5252 if (mActiveTracks.size() > 0) { 5253 status = INVALID_OPERATION; 5254 } else { 5255 reconfig = true; 5256 } 5257 } 5258 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5259 // forward device change to effects that have requested to be 5260 // aware of attached audio device. 5261 for (size_t i = 0; i < mEffectChains.size(); i++) { 5262 mEffectChains[i]->setDevice_l(value); 5263 } 5264 5265 // store input device and output device but do not forward output device to audio HAL. 5266 // Note that status is ignored by the caller for output device 5267 // (see AudioFlinger::setParameters() 5268 if (audio_is_output_devices(value)) { 5269 mOutDevice = value; 5270 status = BAD_VALUE; 5271 } else { 5272 mInDevice = value; 5273 // disable AEC and NS if the device is a BT SCO headset supporting those 5274 // pre processings 5275 if (mTracks.size() > 0) { 5276 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5277 mAudioFlinger->btNrecIsOff(); 5278 for (size_t i = 0; i < mTracks.size(); i++) { 5279 sp<RecordTrack> track = mTracks[i]; 5280 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5281 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5282 } 5283 } 5284 } 5285 } 5286 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5287 mAudioSource != (audio_source_t)value) { 5288 // forward device change to effects that have requested to be 5289 // aware of attached audio device. 5290 for (size_t i = 0; i < mEffectChains.size(); i++) { 5291 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5292 } 5293 mAudioSource = (audio_source_t)value; 5294 } 5295 5296 if (status == NO_ERROR) { 5297 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5298 keyValuePair.string()); 5299 if (status == INVALID_OPERATION) { 5300 inputStandBy(); 5301 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5302 keyValuePair.string()); 5303 } 5304 if (reconfig) { 5305 if (status == BAD_VALUE && 5306 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5307 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5308 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5309 <= (2 * reqSamplingRate)) && 5310 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5311 <= FCC_2 && 5312 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5313 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5314 status = NO_ERROR; 5315 } 5316 if (status == NO_ERROR) { 5317 readInputParameters(); 5318 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5319 } 5320 } 5321 } 5322 5323 mNewParameters.removeAt(0); 5324 5325 mParamStatus = status; 5326 mParamCond.signal(); 5327 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5328 // already timed out waiting for the status and will never signal the condition. 5329 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5330 } 5331 return reconfig; 5332} 5333 5334String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5335{ 5336 Mutex::Autolock _l(mLock); 5337 if (initCheck() != NO_ERROR) { 5338 return String8(); 5339 } 5340 5341 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5342 const String8 out_s8(s); 5343 free(s); 5344 return out_s8; 5345} 5346 5347void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5348 AudioSystem::OutputDescriptor desc; 5349 const void *param2 = NULL; 5350 5351 switch (event) { 5352 case AudioSystem::INPUT_OPENED: 5353 case AudioSystem::INPUT_CONFIG_CHANGED: 5354 desc.channelMask = mChannelMask; 5355 desc.samplingRate = mSampleRate; 5356 desc.format = mFormat; 5357 desc.frameCount = mFrameCount; 5358 desc.latency = 0; 5359 param2 = &desc; 5360 break; 5361 5362 case AudioSystem::INPUT_CLOSED: 5363 default: 5364 break; 5365 } 5366 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5367} 5368 5369void AudioFlinger::RecordThread::readInputParameters() 5370{ 5371 delete[] mRsmpInBuffer; 5372 // mRsmpInBuffer is always assigned a new[] below 5373 delete[] mRsmpOutBuffer; 5374 mRsmpOutBuffer = NULL; 5375 delete mResampler; 5376 mResampler = NULL; 5377 5378 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5379 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5380 mChannelCount = popcount(mChannelMask); 5381 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5382 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5383 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5384 } 5385 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5386 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5387 mFrameCount = mBufferSize / mFrameSize; 5388 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5389 // 1 full output buffer, regardless of the alignment of the available input. 5390 mRsmpInFrames = mFrameCount * 3; 5391 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5392 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5393 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5394 mRsmpInFront = 0; 5395 mRsmpInRear = 0; 5396 mRsmpInUnrel = 0; 5397 5398 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5399 mResampler = AudioResampler::create(16, (int) mChannelCount, mReqSampleRate); 5400 mResampler->setSampleRate(mSampleRate); 5401 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5402 // resampler always outputs stereo 5403 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5404 } 5405 mRsmpInIndex = mFrameCount; 5406} 5407 5408uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5409{ 5410 Mutex::Autolock _l(mLock); 5411 if (initCheck() != NO_ERROR) { 5412 return 0; 5413 } 5414 5415 return mInput->stream->get_input_frames_lost(mInput->stream); 5416} 5417 5418uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5419{ 5420 Mutex::Autolock _l(mLock); 5421 uint32_t result = 0; 5422 if (getEffectChain_l(sessionId) != 0) { 5423 result = EFFECT_SESSION; 5424 } 5425 5426 for (size_t i = 0; i < mTracks.size(); ++i) { 5427 if (sessionId == mTracks[i]->sessionId()) { 5428 result |= TRACK_SESSION; 5429 break; 5430 } 5431 } 5432 5433 return result; 5434} 5435 5436KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5437{ 5438 KeyedVector<int, bool> ids; 5439 Mutex::Autolock _l(mLock); 5440 for (size_t j = 0; j < mTracks.size(); ++j) { 5441 sp<RecordThread::RecordTrack> track = mTracks[j]; 5442 int sessionId = track->sessionId(); 5443 if (ids.indexOfKey(sessionId) < 0) { 5444 ids.add(sessionId, true); 5445 } 5446 } 5447 return ids; 5448} 5449 5450AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5451{ 5452 Mutex::Autolock _l(mLock); 5453 AudioStreamIn *input = mInput; 5454 mInput = NULL; 5455 return input; 5456} 5457 5458// this method must always be called either with ThreadBase mLock held or inside the thread loop 5459audio_stream_t* AudioFlinger::RecordThread::stream() const 5460{ 5461 if (mInput == NULL) { 5462 return NULL; 5463 } 5464 return &mInput->stream->common; 5465} 5466 5467status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5468{ 5469 // only one chain per input thread 5470 if (mEffectChains.size() != 0) { 5471 return INVALID_OPERATION; 5472 } 5473 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5474 5475 chain->setInBuffer(NULL); 5476 chain->setOutBuffer(NULL); 5477 5478 checkSuspendOnAddEffectChain_l(chain); 5479 5480 mEffectChains.add(chain); 5481 5482 return NO_ERROR; 5483} 5484 5485size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5486{ 5487 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5488 ALOGW_IF(mEffectChains.size() != 1, 5489 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5490 chain.get(), mEffectChains.size(), this); 5491 if (mEffectChains.size() == 1) { 5492 mEffectChains.removeAt(0); 5493 } 5494 return 0; 5495} 5496 5497}; // namespace android 5498