Threads.cpp revision e857b65c1d3aa055281cb48f59c9b5eb4a062dd0
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 //FIXME: mStandby should be true here. Is this some kind of hack? 276 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 277 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 278 // mName will be set by concrete (non-virtual) subclass 279 mDeathRecipient(new PMDeathRecipient(this)) 280{ 281} 282 283AudioFlinger::ThreadBase::~ThreadBase() 284{ 285 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 286 for (size_t i = 0; i < mConfigEvents.size(); i++) { 287 delete mConfigEvents[i]; 288 } 289 mConfigEvents.clear(); 290 291 mParamCond.broadcast(); 292 // do not lock the mutex in destructor 293 releaseWakeLock_l(); 294 if (mPowerManager != 0) { 295 sp<IBinder> binder = mPowerManager->asBinder(); 296 binder->unlinkToDeath(mDeathRecipient); 297 } 298} 299 300void AudioFlinger::ThreadBase::exit() 301{ 302 ALOGV("ThreadBase::exit"); 303 // do any cleanup required for exit to succeed 304 preExit(); 305 { 306 // This lock prevents the following race in thread (uniprocessor for illustration): 307 // if (!exitPending()) { 308 // // context switch from here to exit() 309 // // exit() calls requestExit(), what exitPending() observes 310 // // exit() calls signal(), which is dropped since no waiters 311 // // context switch back from exit() to here 312 // mWaitWorkCV.wait(...); 313 // // now thread is hung 314 // } 315 AutoMutex lock(mLock); 316 requestExit(); 317 mWaitWorkCV.broadcast(); 318 } 319 // When Thread::requestExitAndWait is made virtual and this method is renamed to 320 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 321 requestExitAndWait(); 322} 323 324status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 325{ 326 status_t status; 327 328 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 329 Mutex::Autolock _l(mLock); 330 331 mNewParameters.add(keyValuePairs); 332 mWaitWorkCV.signal(); 333 // wait condition with timeout in case the thread loop has exited 334 // before the request could be processed 335 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 336 status = mParamStatus; 337 mWaitWorkCV.signal(); 338 } else { 339 status = TIMED_OUT; 340 } 341 return status; 342} 343 344void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 345{ 346 Mutex::Autolock _l(mLock); 347 sendIoConfigEvent_l(event, param); 348} 349 350// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 351void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 352{ 353 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 354 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 355 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 356 param); 357 mWaitWorkCV.signal(); 358} 359 360// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 361void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 362{ 363 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 364 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 365 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 366 mConfigEvents.size(), pid, tid, prio); 367 mWaitWorkCV.signal(); 368} 369 370void AudioFlinger::ThreadBase::processConfigEvents() 371{ 372 mLock.lock(); 373 while (!mConfigEvents.isEmpty()) { 374 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 375 ConfigEvent *event = mConfigEvents[0]; 376 mConfigEvents.removeAt(0); 377 // release mLock before locking AudioFlinger mLock: lock order is always 378 // AudioFlinger then ThreadBase to avoid cross deadlock 379 mLock.unlock(); 380 switch(event->type()) { 381 case CFG_EVENT_PRIO: { 382 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 383 // FIXME Need to understand why this has be done asynchronously 384 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 385 true /*asynchronous*/); 386 if (err != 0) { 387 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 388 "error %d", 389 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 390 } 391 } break; 392 case CFG_EVENT_IO: { 393 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 394 mAudioFlinger->mLock.lock(); 395 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 396 mAudioFlinger->mLock.unlock(); 397 } break; 398 default: 399 ALOGE("processConfigEvents() unknown event type %d", event->type()); 400 break; 401 } 402 delete event; 403 mLock.lock(); 404 } 405 mLock.unlock(); 406} 407 408void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 409{ 410 const size_t SIZE = 256; 411 char buffer[SIZE]; 412 String8 result; 413 414 bool locked = AudioFlinger::dumpTryLock(mLock); 415 if (!locked) { 416 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 417 write(fd, buffer, strlen(buffer)); 418 } 419 420 snprintf(buffer, SIZE, "io handle: %d\n", mId); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "HAL frame count: %zu\n", mFrameCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Frame size: %zu\n", mFrameSize); 437 result.append(buffer); 438 439 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 440 result.append(buffer); 441 result.append(" Index Command"); 442 for (size_t i = 0; i < mNewParameters.size(); ++i) { 443 snprintf(buffer, SIZE, "\n %02zu ", i); 444 result.append(buffer); 445 result.append(mNewParameters[i]); 446 } 447 448 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 449 result.append(buffer); 450 for (size_t i = 0; i < mConfigEvents.size(); i++) { 451 mConfigEvents[i]->dump(buffer, SIZE); 452 result.append(buffer); 453 } 454 result.append("\n"); 455 456 write(fd, result.string(), result.size()); 457 458 if (locked) { 459 mLock.unlock(); 460 } 461} 462 463void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 464{ 465 const size_t SIZE = 256; 466 char buffer[SIZE]; 467 String8 result; 468 469 snprintf(buffer, SIZE, "\n- %zu Effect Chains:\n", mEffectChains.size()); 470 write(fd, buffer, strlen(buffer)); 471 472 for (size_t i = 0; i < mEffectChains.size(); ++i) { 473 sp<EffectChain> chain = mEffectChains[i]; 474 if (chain != 0) { 475 chain->dump(fd, args); 476 } 477 } 478} 479 480void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 481{ 482 Mutex::Autolock _l(mLock); 483 acquireWakeLock_l(uid); 484} 485 486String16 AudioFlinger::ThreadBase::getWakeLockTag() 487{ 488 switch (mType) { 489 case MIXER: 490 return String16("AudioMix"); 491 case DIRECT: 492 return String16("AudioDirectOut"); 493 case DUPLICATING: 494 return String16("AudioDup"); 495 case RECORD: 496 return String16("AudioIn"); 497 case OFFLOAD: 498 return String16("AudioOffload"); 499 default: 500 ALOG_ASSERT(false); 501 return String16("AudioUnknown"); 502 } 503} 504 505void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 506{ 507 getPowerManager_l(); 508 if (mPowerManager != 0) { 509 sp<IBinder> binder = new BBinder(); 510 status_t status; 511 if (uid >= 0) { 512 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 513 binder, 514 getWakeLockTag(), 515 String16("media"), 516 uid); 517 } else { 518 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 519 binder, 520 getWakeLockTag(), 521 String16("media")); 522 } 523 if (status == NO_ERROR) { 524 mWakeLockToken = binder; 525 } 526 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 527 } 528} 529 530void AudioFlinger::ThreadBase::releaseWakeLock() 531{ 532 Mutex::Autolock _l(mLock); 533 releaseWakeLock_l(); 534} 535 536void AudioFlinger::ThreadBase::releaseWakeLock_l() 537{ 538 if (mWakeLockToken != 0) { 539 ALOGV("releaseWakeLock_l() %s", mName); 540 if (mPowerManager != 0) { 541 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 542 } 543 mWakeLockToken.clear(); 544 } 545} 546 547void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 548 Mutex::Autolock _l(mLock); 549 updateWakeLockUids_l(uids); 550} 551 552void AudioFlinger::ThreadBase::getPowerManager_l() { 553 554 if (mPowerManager == 0) { 555 // use checkService() to avoid blocking if power service is not up yet 556 sp<IBinder> binder = 557 defaultServiceManager()->checkService(String16("power")); 558 if (binder == 0) { 559 ALOGW("Thread %s cannot connect to the power manager service", mName); 560 } else { 561 mPowerManager = interface_cast<IPowerManager>(binder); 562 binder->linkToDeath(mDeathRecipient); 563 } 564 } 565} 566 567void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 568 569 getPowerManager_l(); 570 if (mWakeLockToken == NULL) { 571 ALOGE("no wake lock to update!"); 572 return; 573 } 574 if (mPowerManager != 0) { 575 sp<IBinder> binder = new BBinder(); 576 status_t status; 577 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 578 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 579 } 580} 581 582void AudioFlinger::ThreadBase::clearPowerManager() 583{ 584 Mutex::Autolock _l(mLock); 585 releaseWakeLock_l(); 586 mPowerManager.clear(); 587} 588 589void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 590{ 591 sp<ThreadBase> thread = mThread.promote(); 592 if (thread != 0) { 593 thread->clearPowerManager(); 594 } 595 ALOGW("power manager service died !!!"); 596} 597 598void AudioFlinger::ThreadBase::setEffectSuspended( 599 const effect_uuid_t *type, bool suspend, int sessionId) 600{ 601 Mutex::Autolock _l(mLock); 602 setEffectSuspended_l(type, suspend, sessionId); 603} 604 605void AudioFlinger::ThreadBase::setEffectSuspended_l( 606 const effect_uuid_t *type, bool suspend, int sessionId) 607{ 608 sp<EffectChain> chain = getEffectChain_l(sessionId); 609 if (chain != 0) { 610 if (type != NULL) { 611 chain->setEffectSuspended_l(type, suspend); 612 } else { 613 chain->setEffectSuspendedAll_l(suspend); 614 } 615 } 616 617 updateSuspendedSessions_l(type, suspend, sessionId); 618} 619 620void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 621{ 622 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 623 if (index < 0) { 624 return; 625 } 626 627 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 628 mSuspendedSessions.valueAt(index); 629 630 for (size_t i = 0; i < sessionEffects.size(); i++) { 631 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 632 for (int j = 0; j < desc->mRefCount; j++) { 633 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 634 chain->setEffectSuspendedAll_l(true); 635 } else { 636 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 637 desc->mType.timeLow); 638 chain->setEffectSuspended_l(&desc->mType, true); 639 } 640 } 641 } 642} 643 644void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 645 bool suspend, 646 int sessionId) 647{ 648 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 649 650 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 651 652 if (suspend) { 653 if (index >= 0) { 654 sessionEffects = mSuspendedSessions.valueAt(index); 655 } else { 656 mSuspendedSessions.add(sessionId, sessionEffects); 657 } 658 } else { 659 if (index < 0) { 660 return; 661 } 662 sessionEffects = mSuspendedSessions.valueAt(index); 663 } 664 665 666 int key = EffectChain::kKeyForSuspendAll; 667 if (type != NULL) { 668 key = type->timeLow; 669 } 670 index = sessionEffects.indexOfKey(key); 671 672 sp<SuspendedSessionDesc> desc; 673 if (suspend) { 674 if (index >= 0) { 675 desc = sessionEffects.valueAt(index); 676 } else { 677 desc = new SuspendedSessionDesc(); 678 if (type != NULL) { 679 desc->mType = *type; 680 } 681 sessionEffects.add(key, desc); 682 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 683 } 684 desc->mRefCount++; 685 } else { 686 if (index < 0) { 687 return; 688 } 689 desc = sessionEffects.valueAt(index); 690 if (--desc->mRefCount == 0) { 691 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 692 sessionEffects.removeItemsAt(index); 693 if (sessionEffects.isEmpty()) { 694 ALOGV("updateSuspendedSessions_l() restore removing session %d", 695 sessionId); 696 mSuspendedSessions.removeItem(sessionId); 697 } 698 } 699 } 700 if (!sessionEffects.isEmpty()) { 701 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 702 } 703} 704 705void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 706 bool enabled, 707 int sessionId) 708{ 709 Mutex::Autolock _l(mLock); 710 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 711} 712 713void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 714 bool enabled, 715 int sessionId) 716{ 717 if (mType != RECORD) { 718 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 719 // another session. This gives the priority to well behaved effect control panels 720 // and applications not using global effects. 721 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 722 // global effects 723 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 724 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 725 } 726 } 727 728 sp<EffectChain> chain = getEffectChain_l(sessionId); 729 if (chain != 0) { 730 chain->checkSuspendOnEffectEnabled(effect, enabled); 731 } 732} 733 734// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 735sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 736 const sp<AudioFlinger::Client>& client, 737 const sp<IEffectClient>& effectClient, 738 int32_t priority, 739 int sessionId, 740 effect_descriptor_t *desc, 741 int *enabled, 742 status_t *status 743 ) 744{ 745 sp<EffectModule> effect; 746 sp<EffectHandle> handle; 747 status_t lStatus; 748 sp<EffectChain> chain; 749 bool chainCreated = false; 750 bool effectCreated = false; 751 bool effectRegistered = false; 752 753 lStatus = initCheck(); 754 if (lStatus != NO_ERROR) { 755 ALOGW("createEffect_l() Audio driver not initialized."); 756 goto Exit; 757 } 758 759 // Allow global effects only on offloaded and mixer threads 760 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 761 switch (mType) { 762 case MIXER: 763 case OFFLOAD: 764 break; 765 case DIRECT: 766 case DUPLICATING: 767 case RECORD: 768 default: 769 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 770 lStatus = BAD_VALUE; 771 goto Exit; 772 } 773 } 774 775 // Only Pre processor effects are allowed on input threads and only on input threads 776 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 777 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 778 desc->name, desc->flags, mType); 779 lStatus = BAD_VALUE; 780 goto Exit; 781 } 782 783 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 784 785 { // scope for mLock 786 Mutex::Autolock _l(mLock); 787 788 // check for existing effect chain with the requested audio session 789 chain = getEffectChain_l(sessionId); 790 if (chain == 0) { 791 // create a new chain for this session 792 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 793 chain = new EffectChain(this, sessionId); 794 addEffectChain_l(chain); 795 chain->setStrategy(getStrategyForSession_l(sessionId)); 796 chainCreated = true; 797 } else { 798 effect = chain->getEffectFromDesc_l(desc); 799 } 800 801 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 802 803 if (effect == 0) { 804 int id = mAudioFlinger->nextUniqueId(); 805 // Check CPU and memory usage 806 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 807 if (lStatus != NO_ERROR) { 808 goto Exit; 809 } 810 effectRegistered = true; 811 // create a new effect module if none present in the chain 812 effect = new EffectModule(this, chain, desc, id, sessionId); 813 lStatus = effect->status(); 814 if (lStatus != NO_ERROR) { 815 goto Exit; 816 } 817 effect->setOffloaded(mType == OFFLOAD, mId); 818 819 lStatus = chain->addEffect_l(effect); 820 if (lStatus != NO_ERROR) { 821 goto Exit; 822 } 823 effectCreated = true; 824 825 effect->setDevice(mOutDevice); 826 effect->setDevice(mInDevice); 827 effect->setMode(mAudioFlinger->getMode()); 828 effect->setAudioSource(mAudioSource); 829 } 830 // create effect handle and connect it to effect module 831 handle = new EffectHandle(effect, client, effectClient, priority); 832 lStatus = effect->addHandle(handle.get()); 833 if (enabled != NULL) { 834 *enabled = (int)effect->isEnabled(); 835 } 836 } 837 838Exit: 839 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 840 Mutex::Autolock _l(mLock); 841 if (effectCreated) { 842 chain->removeEffect_l(effect); 843 } 844 if (effectRegistered) { 845 AudioSystem::unregisterEffect(effect->id()); 846 } 847 if (chainCreated) { 848 removeEffectChain_l(chain); 849 } 850 handle.clear(); 851 } 852 853 if (status != NULL) { 854 *status = lStatus; 855 } 856 return handle; 857} 858 859sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 860{ 861 Mutex::Autolock _l(mLock); 862 return getEffect_l(sessionId, effectId); 863} 864 865sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 866{ 867 sp<EffectChain> chain = getEffectChain_l(sessionId); 868 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 869} 870 871// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 872// PlaybackThread::mLock held 873status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 874{ 875 // check for existing effect chain with the requested audio session 876 int sessionId = effect->sessionId(); 877 sp<EffectChain> chain = getEffectChain_l(sessionId); 878 bool chainCreated = false; 879 880 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 881 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 882 this, effect->desc().name, effect->desc().flags); 883 884 if (chain == 0) { 885 // create a new chain for this session 886 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 887 chain = new EffectChain(this, sessionId); 888 addEffectChain_l(chain); 889 chain->setStrategy(getStrategyForSession_l(sessionId)); 890 chainCreated = true; 891 } 892 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 893 894 if (chain->getEffectFromId_l(effect->id()) != 0) { 895 ALOGW("addEffect_l() %p effect %s already present in chain %p", 896 this, effect->desc().name, chain.get()); 897 return BAD_VALUE; 898 } 899 900 effect->setOffloaded(mType == OFFLOAD, mId); 901 902 status_t status = chain->addEffect_l(effect); 903 if (status != NO_ERROR) { 904 if (chainCreated) { 905 removeEffectChain_l(chain); 906 } 907 return status; 908 } 909 910 effect->setDevice(mOutDevice); 911 effect->setDevice(mInDevice); 912 effect->setMode(mAudioFlinger->getMode()); 913 effect->setAudioSource(mAudioSource); 914 return NO_ERROR; 915} 916 917void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 918 919 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 920 effect_descriptor_t desc = effect->desc(); 921 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 922 detachAuxEffect_l(effect->id()); 923 } 924 925 sp<EffectChain> chain = effect->chain().promote(); 926 if (chain != 0) { 927 // remove effect chain if removing last effect 928 if (chain->removeEffect_l(effect) == 0) { 929 removeEffectChain_l(chain); 930 } 931 } else { 932 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 933 } 934} 935 936void AudioFlinger::ThreadBase::lockEffectChains_l( 937 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 938{ 939 effectChains = mEffectChains; 940 for (size_t i = 0; i < mEffectChains.size(); i++) { 941 mEffectChains[i]->lock(); 942 } 943} 944 945void AudioFlinger::ThreadBase::unlockEffectChains( 946 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 947{ 948 for (size_t i = 0; i < effectChains.size(); i++) { 949 effectChains[i]->unlock(); 950 } 951} 952 953sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 954{ 955 Mutex::Autolock _l(mLock); 956 return getEffectChain_l(sessionId); 957} 958 959sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 960{ 961 size_t size = mEffectChains.size(); 962 for (size_t i = 0; i < size; i++) { 963 if (mEffectChains[i]->sessionId() == sessionId) { 964 return mEffectChains[i]; 965 } 966 } 967 return 0; 968} 969 970void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 971{ 972 Mutex::Autolock _l(mLock); 973 size_t size = mEffectChains.size(); 974 for (size_t i = 0; i < size; i++) { 975 mEffectChains[i]->setMode_l(mode); 976 } 977} 978 979void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 980 EffectHandle *handle, 981 bool unpinIfLast) { 982 983 Mutex::Autolock _l(mLock); 984 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 985 // delete the effect module if removing last handle on it 986 if (effect->removeHandle(handle) == 0) { 987 if (!effect->isPinned() || unpinIfLast) { 988 removeEffect_l(effect); 989 AudioSystem::unregisterEffect(effect->id()); 990 } 991 } 992} 993 994// ---------------------------------------------------------------------------- 995// Playback 996// ---------------------------------------------------------------------------- 997 998AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 999 AudioStreamOut* output, 1000 audio_io_handle_t id, 1001 audio_devices_t device, 1002 type_t type) 1003 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1004 mNormalFrameCount(0), mMixBuffer(NULL), 1005 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1006 mActiveTracksGeneration(0), 1007 // mStreamTypes[] initialized in constructor body 1008 mOutput(output), 1009 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1010 mMixerStatus(MIXER_IDLE), 1011 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1012 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1013 mBytesRemaining(0), 1014 mCurrentWriteLength(0), 1015 mUseAsyncWrite(false), 1016 mWriteAckSequence(0), 1017 mDrainSequence(0), 1018 mSignalPending(false), 1019 mScreenState(AudioFlinger::mScreenState), 1020 // index 0 is reserved for normal mixer's submix 1021 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1022 // mLatchD, mLatchQ, 1023 mLatchDValid(false), mLatchQValid(false) 1024{ 1025 snprintf(mName, kNameLength, "AudioOut_%X", id); 1026 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1027 1028 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1029 // it would be safer to explicitly pass initial masterVolume/masterMute as 1030 // parameter. 1031 // 1032 // If the HAL we are using has support for master volume or master mute, 1033 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1034 // and the mute set to false). 1035 mMasterVolume = audioFlinger->masterVolume_l(); 1036 mMasterMute = audioFlinger->masterMute_l(); 1037 if (mOutput && mOutput->audioHwDev) { 1038 if (mOutput->audioHwDev->canSetMasterVolume()) { 1039 mMasterVolume = 1.0; 1040 } 1041 1042 if (mOutput->audioHwDev->canSetMasterMute()) { 1043 mMasterMute = false; 1044 } 1045 } 1046 1047 readOutputParameters(); 1048 1049 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1050 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1051 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1052 stream = (audio_stream_type_t) (stream + 1)) { 1053 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1054 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1055 } 1056 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1057 // because mAudioFlinger doesn't have one to copy from 1058} 1059 1060AudioFlinger::PlaybackThread::~PlaybackThread() 1061{ 1062 mAudioFlinger->unregisterWriter(mNBLogWriter); 1063 delete [] mAllocMixBuffer; 1064} 1065 1066void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1067{ 1068 dumpInternals(fd, args); 1069 dumpTracks(fd, args); 1070 dumpEffectChains(fd, args); 1071} 1072 1073void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1074{ 1075 const size_t SIZE = 256; 1076 char buffer[SIZE]; 1077 String8 result; 1078 1079 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1080 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1081 const stream_type_t *st = &mStreamTypes[i]; 1082 if (i > 0) { 1083 result.appendFormat(", "); 1084 } 1085 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1086 if (st->mute) { 1087 result.append("M"); 1088 } 1089 } 1090 result.append("\n"); 1091 write(fd, result.string(), result.length()); 1092 result.clear(); 1093 1094 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1095 result.append(buffer); 1096 Track::appendDumpHeader(result); 1097 for (size_t i = 0; i < mTracks.size(); ++i) { 1098 sp<Track> track = mTracks[i]; 1099 if (track != 0) { 1100 track->dump(buffer, SIZE); 1101 result.append(buffer); 1102 } 1103 } 1104 1105 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1106 result.append(buffer); 1107 Track::appendDumpHeader(result); 1108 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1109 sp<Track> track = mActiveTracks[i].promote(); 1110 if (track != 0) { 1111 track->dump(buffer, SIZE); 1112 result.append(buffer); 1113 } 1114 } 1115 write(fd, result.string(), result.size()); 1116 1117 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1118 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1119 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1120 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1121} 1122 1123void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1124{ 1125 const size_t SIZE = 256; 1126 char buffer[SIZE]; 1127 String8 result; 1128 1129 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1130 result.append(buffer); 1131 snprintf(buffer, SIZE, "Normal frame count: %zu\n", mNormalFrameCount); 1132 result.append(buffer); 1133 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1134 ns2ms(systemTime() - mLastWriteTime)); 1135 result.append(buffer); 1136 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1137 result.append(buffer); 1138 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1139 result.append(buffer); 1140 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1141 result.append(buffer); 1142 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1143 result.append(buffer); 1144 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1145 result.append(buffer); 1146 write(fd, result.string(), result.size()); 1147 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1148 1149 dumpBase(fd, args); 1150} 1151 1152// Thread virtuals 1153status_t AudioFlinger::PlaybackThread::readyToRun() 1154{ 1155 status_t status = initCheck(); 1156 if (status == NO_ERROR) { 1157 ALOGI("AudioFlinger's thread %p ready to run", this); 1158 } else { 1159 ALOGE("No working audio driver found."); 1160 } 1161 return status; 1162} 1163 1164void AudioFlinger::PlaybackThread::onFirstRef() 1165{ 1166 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1167} 1168 1169// ThreadBase virtuals 1170void AudioFlinger::PlaybackThread::preExit() 1171{ 1172 ALOGV(" preExit()"); 1173 // FIXME this is using hard-coded strings but in the future, this functionality will be 1174 // converted to use audio HAL extensions required to support tunneling 1175 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1176} 1177 1178// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1179sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1180 const sp<AudioFlinger::Client>& client, 1181 audio_stream_type_t streamType, 1182 uint32_t sampleRate, 1183 audio_format_t format, 1184 audio_channel_mask_t channelMask, 1185 size_t frameCount, 1186 const sp<IMemory>& sharedBuffer, 1187 int sessionId, 1188 IAudioFlinger::track_flags_t *flags, 1189 pid_t tid, 1190 int uid, 1191 status_t *status) 1192{ 1193 sp<Track> track; 1194 status_t lStatus; 1195 1196 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1197 1198 // client expresses a preference for FAST, but we get the final say 1199 if (*flags & IAudioFlinger::TRACK_FAST) { 1200 if ( 1201 // not timed 1202 (!isTimed) && 1203 // either of these use cases: 1204 ( 1205 // use case 1: shared buffer with any frame count 1206 ( 1207 (sharedBuffer != 0) 1208 ) || 1209 // use case 2: callback handler and frame count is default or at least as large as HAL 1210 ( 1211 (tid != -1) && 1212 ((frameCount == 0) || 1213 (frameCount >= mFrameCount)) 1214 ) 1215 ) && 1216 // PCM data 1217 audio_is_linear_pcm(format) && 1218 // mono or stereo 1219 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1220 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1221 // hardware sample rate 1222 (sampleRate == mSampleRate) && 1223 // normal mixer has an associated fast mixer 1224 hasFastMixer() && 1225 // there are sufficient fast track slots available 1226 (mFastTrackAvailMask != 0) 1227 // FIXME test that MixerThread for this fast track has a capable output HAL 1228 // FIXME add a permission test also? 1229 ) { 1230 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1231 if (frameCount == 0) { 1232 frameCount = mFrameCount * kFastTrackMultiplier; 1233 } 1234 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1235 frameCount, mFrameCount); 1236 } else { 1237 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1238 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1239 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1240 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1241 audio_is_linear_pcm(format), 1242 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1243 *flags &= ~IAudioFlinger::TRACK_FAST; 1244 // For compatibility with AudioTrack calculation, buffer depth is forced 1245 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1246 // This is probably too conservative, but legacy application code may depend on it. 1247 // If you change this calculation, also review the start threshold which is related. 1248 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1249 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1250 if (minBufCount < 2) { 1251 minBufCount = 2; 1252 } 1253 size_t minFrameCount = mNormalFrameCount * minBufCount; 1254 if (frameCount < minFrameCount) { 1255 frameCount = minFrameCount; 1256 } 1257 } 1258 } 1259 1260 if (mType == DIRECT) { 1261 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1262 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1263 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1264 "for output %p with format %d", 1265 sampleRate, format, channelMask, mOutput, mFormat); 1266 lStatus = BAD_VALUE; 1267 goto Exit; 1268 } 1269 } 1270 } else if (mType == OFFLOAD) { 1271 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1272 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1273 "for output %p with format %d", 1274 sampleRate, format, channelMask, mOutput, mFormat); 1275 lStatus = BAD_VALUE; 1276 goto Exit; 1277 } 1278 } else { 1279 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1280 ALOGE("createTrack_l() Bad parameter: format %d \"" 1281 "for output %p with format %d", 1282 format, mOutput, mFormat); 1283 lStatus = BAD_VALUE; 1284 goto Exit; 1285 } 1286 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1287 if (sampleRate > mSampleRate*2) { 1288 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1289 lStatus = BAD_VALUE; 1290 goto Exit; 1291 } 1292 } 1293 1294 lStatus = initCheck(); 1295 if (lStatus != NO_ERROR) { 1296 ALOGE("Audio driver not initialized."); 1297 goto Exit; 1298 } 1299 1300 { // scope for mLock 1301 Mutex::Autolock _l(mLock); 1302 1303 // all tracks in same audio session must share the same routing strategy otherwise 1304 // conflicts will happen when tracks are moved from one output to another by audio policy 1305 // manager 1306 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1307 for (size_t i = 0; i < mTracks.size(); ++i) { 1308 sp<Track> t = mTracks[i]; 1309 if (t != 0 && !t->isOutputTrack()) { 1310 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1311 if (sessionId == t->sessionId() && strategy != actual) { 1312 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1313 strategy, actual); 1314 lStatus = BAD_VALUE; 1315 goto Exit; 1316 } 1317 } 1318 } 1319 1320 if (!isTimed) { 1321 track = new Track(this, client, streamType, sampleRate, format, 1322 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1323 } else { 1324 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1325 channelMask, frameCount, sharedBuffer, sessionId, uid); 1326 } 1327 1328 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1329 lStatus = NO_MEMORY; 1330 // track must be cleared from the caller as the caller has the AF lock 1331 goto Exit; 1332 } 1333 1334 mTracks.add(track); 1335 1336 sp<EffectChain> chain = getEffectChain_l(sessionId); 1337 if (chain != 0) { 1338 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1339 track->setMainBuffer(chain->inBuffer()); 1340 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1341 chain->incTrackCnt(); 1342 } 1343 1344 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1345 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1346 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1347 // so ask activity manager to do this on our behalf 1348 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1349 } 1350 } 1351 1352 lStatus = NO_ERROR; 1353 1354Exit: 1355 if (status) { 1356 *status = lStatus; 1357 } 1358 return track; 1359} 1360 1361uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1362{ 1363 return latency; 1364} 1365 1366uint32_t AudioFlinger::PlaybackThread::latency() const 1367{ 1368 Mutex::Autolock _l(mLock); 1369 return latency_l(); 1370} 1371uint32_t AudioFlinger::PlaybackThread::latency_l() const 1372{ 1373 if (initCheck() == NO_ERROR) { 1374 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1375 } else { 1376 return 0; 1377 } 1378} 1379 1380void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1381{ 1382 Mutex::Autolock _l(mLock); 1383 // Don't apply master volume in SW if our HAL can do it for us. 1384 if (mOutput && mOutput->audioHwDev && 1385 mOutput->audioHwDev->canSetMasterVolume()) { 1386 mMasterVolume = 1.0; 1387 } else { 1388 mMasterVolume = value; 1389 } 1390} 1391 1392void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1393{ 1394 Mutex::Autolock _l(mLock); 1395 // Don't apply master mute in SW if our HAL can do it for us. 1396 if (mOutput && mOutput->audioHwDev && 1397 mOutput->audioHwDev->canSetMasterMute()) { 1398 mMasterMute = false; 1399 } else { 1400 mMasterMute = muted; 1401 } 1402} 1403 1404void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1405{ 1406 Mutex::Autolock _l(mLock); 1407 mStreamTypes[stream].volume = value; 1408 broadcast_l(); 1409} 1410 1411void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1412{ 1413 Mutex::Autolock _l(mLock); 1414 mStreamTypes[stream].mute = muted; 1415 broadcast_l(); 1416} 1417 1418float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1419{ 1420 Mutex::Autolock _l(mLock); 1421 return mStreamTypes[stream].volume; 1422} 1423 1424// addTrack_l() must be called with ThreadBase::mLock held 1425status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1426{ 1427 status_t status = ALREADY_EXISTS; 1428 1429 // set retry count for buffer fill 1430 track->mRetryCount = kMaxTrackStartupRetries; 1431 if (mActiveTracks.indexOf(track) < 0) { 1432 // the track is newly added, make sure it fills up all its 1433 // buffers before playing. This is to ensure the client will 1434 // effectively get the latency it requested. 1435 if (!track->isOutputTrack()) { 1436 TrackBase::track_state state = track->mState; 1437 mLock.unlock(); 1438 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1439 mLock.lock(); 1440 // abort track was stopped/paused while we released the lock 1441 if (state != track->mState) { 1442 if (status == NO_ERROR) { 1443 mLock.unlock(); 1444 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1445 mLock.lock(); 1446 } 1447 return INVALID_OPERATION; 1448 } 1449 // abort if start is rejected by audio policy manager 1450 if (status != NO_ERROR) { 1451 return PERMISSION_DENIED; 1452 } 1453#ifdef ADD_BATTERY_DATA 1454 // to track the speaker usage 1455 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1456#endif 1457 } 1458 1459 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1460 track->mResetDone = false; 1461 track->mPresentationCompleteFrames = 0; 1462 mActiveTracks.add(track); 1463 mWakeLockUids.add(track->uid()); 1464 mActiveTracksGeneration++; 1465 mLatestActiveTrack = track; 1466 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1467 if (chain != 0) { 1468 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1469 track->sessionId()); 1470 chain->incActiveTrackCnt(); 1471 } 1472 1473 status = NO_ERROR; 1474 } 1475 1476 ALOGV("signal playback thread"); 1477 broadcast_l(); 1478 1479 return status; 1480} 1481 1482bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1483{ 1484 track->terminate(); 1485 // active tracks are removed by threadLoop() 1486 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1487 track->mState = TrackBase::STOPPED; 1488 if (!trackActive) { 1489 removeTrack_l(track); 1490 } else if (track->isFastTrack() || track->isOffloaded()) { 1491 track->mState = TrackBase::STOPPING_1; 1492 } 1493 1494 return trackActive; 1495} 1496 1497void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1498{ 1499 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1500 mTracks.remove(track); 1501 deleteTrackName_l(track->name()); 1502 // redundant as track is about to be destroyed, for dumpsys only 1503 track->mName = -1; 1504 if (track->isFastTrack()) { 1505 int index = track->mFastIndex; 1506 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1507 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1508 mFastTrackAvailMask |= 1 << index; 1509 // redundant as track is about to be destroyed, for dumpsys only 1510 track->mFastIndex = -1; 1511 } 1512 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1513 if (chain != 0) { 1514 chain->decTrackCnt(); 1515 } 1516} 1517 1518void AudioFlinger::PlaybackThread::broadcast_l() 1519{ 1520 // Thread could be blocked waiting for async 1521 // so signal it to handle state changes immediately 1522 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1523 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1524 mSignalPending = true; 1525 mWaitWorkCV.broadcast(); 1526} 1527 1528String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1529{ 1530 Mutex::Autolock _l(mLock); 1531 if (initCheck() != NO_ERROR) { 1532 return String8(); 1533 } 1534 1535 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1536 const String8 out_s8(s); 1537 free(s); 1538 return out_s8; 1539} 1540 1541// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1542void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1543 AudioSystem::OutputDescriptor desc; 1544 void *param2 = NULL; 1545 1546 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1547 param); 1548 1549 switch (event) { 1550 case AudioSystem::OUTPUT_OPENED: 1551 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1552 desc.channelMask = mChannelMask; 1553 desc.samplingRate = mSampleRate; 1554 desc.format = mFormat; 1555 desc.frameCount = mNormalFrameCount; // FIXME see 1556 // AudioFlinger::frameCount(audio_io_handle_t) 1557 desc.latency = latency(); 1558 param2 = &desc; 1559 break; 1560 1561 case AudioSystem::STREAM_CONFIG_CHANGED: 1562 param2 = ¶m; 1563 case AudioSystem::OUTPUT_CLOSED: 1564 default: 1565 break; 1566 } 1567 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1568} 1569 1570void AudioFlinger::PlaybackThread::writeCallback() 1571{ 1572 ALOG_ASSERT(mCallbackThread != 0); 1573 mCallbackThread->resetWriteBlocked(); 1574} 1575 1576void AudioFlinger::PlaybackThread::drainCallback() 1577{ 1578 ALOG_ASSERT(mCallbackThread != 0); 1579 mCallbackThread->resetDraining(); 1580} 1581 1582void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1583{ 1584 Mutex::Autolock _l(mLock); 1585 // reject out of sequence requests 1586 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1587 mWriteAckSequence &= ~1; 1588 mWaitWorkCV.signal(); 1589 } 1590} 1591 1592void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1593{ 1594 Mutex::Autolock _l(mLock); 1595 // reject out of sequence requests 1596 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1597 mDrainSequence &= ~1; 1598 mWaitWorkCV.signal(); 1599 } 1600} 1601 1602// static 1603int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1604 void *param, 1605 void *cookie) 1606{ 1607 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1608 ALOGV("asyncCallback() event %d", event); 1609 switch (event) { 1610 case STREAM_CBK_EVENT_WRITE_READY: 1611 me->writeCallback(); 1612 break; 1613 case STREAM_CBK_EVENT_DRAIN_READY: 1614 me->drainCallback(); 1615 break; 1616 default: 1617 ALOGW("asyncCallback() unknown event %d", event); 1618 break; 1619 } 1620 return 0; 1621} 1622 1623void AudioFlinger::PlaybackThread::readOutputParameters() 1624{ 1625 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1626 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1627 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1628 if (!audio_is_output_channel(mChannelMask)) { 1629 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1630 } 1631 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1632 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1633 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1634 } 1635 mChannelCount = popcount(mChannelMask); 1636 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1637 if (!audio_is_valid_format(mFormat)) { 1638 LOG_FATAL("HAL format %d not valid for output", mFormat); 1639 } 1640 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1641 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1642 mFormat); 1643 } 1644 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1645 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1646 if (mFrameCount & 15) { 1647 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1648 mFrameCount); 1649 } 1650 1651 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1652 (mOutput->stream->set_callback != NULL)) { 1653 if (mOutput->stream->set_callback(mOutput->stream, 1654 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1655 mUseAsyncWrite = true; 1656 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1657 } 1658 } 1659 1660 // Calculate size of normal mix buffer relative to the HAL output buffer size 1661 double multiplier = 1.0; 1662 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1663 kUseFastMixer == FastMixer_Dynamic)) { 1664 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1665 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1666 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1667 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1668 maxNormalFrameCount = maxNormalFrameCount & ~15; 1669 if (maxNormalFrameCount < minNormalFrameCount) { 1670 maxNormalFrameCount = minNormalFrameCount; 1671 } 1672 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1673 if (multiplier <= 1.0) { 1674 multiplier = 1.0; 1675 } else if (multiplier <= 2.0) { 1676 if (2 * mFrameCount <= maxNormalFrameCount) { 1677 multiplier = 2.0; 1678 } else { 1679 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1680 } 1681 } else { 1682 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1683 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1684 // track, but we sometimes have to do this to satisfy the maximum frame count 1685 // constraint) 1686 // FIXME this rounding up should not be done if no HAL SRC 1687 uint32_t truncMult = (uint32_t) multiplier; 1688 if ((truncMult & 1)) { 1689 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1690 ++truncMult; 1691 } 1692 } 1693 multiplier = (double) truncMult; 1694 } 1695 } 1696 mNormalFrameCount = multiplier * mFrameCount; 1697 // round up to nearest 16 frames to satisfy AudioMixer 1698 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1699 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1700 mNormalFrameCount); 1701 1702 delete[] mAllocMixBuffer; 1703 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1704 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1705 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1706 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1707 1708 // force reconfiguration of effect chains and engines to take new buffer size and audio 1709 // parameters into account 1710 // Note that mLock is not held when readOutputParameters() is called from the constructor 1711 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1712 // matter. 1713 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1714 Vector< sp<EffectChain> > effectChains = mEffectChains; 1715 for (size_t i = 0; i < effectChains.size(); i ++) { 1716 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1717 } 1718} 1719 1720 1721status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1722{ 1723 if (halFrames == NULL || dspFrames == NULL) { 1724 return BAD_VALUE; 1725 } 1726 Mutex::Autolock _l(mLock); 1727 if (initCheck() != NO_ERROR) { 1728 return INVALID_OPERATION; 1729 } 1730 size_t framesWritten = mBytesWritten / mFrameSize; 1731 *halFrames = framesWritten; 1732 1733 if (isSuspended()) { 1734 // return an estimation of rendered frames when the output is suspended 1735 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1736 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1737 return NO_ERROR; 1738 } else { 1739 status_t status; 1740 uint32_t frames; 1741 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1742 *dspFrames = (size_t)frames; 1743 return status; 1744 } 1745} 1746 1747uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1748{ 1749 Mutex::Autolock _l(mLock); 1750 uint32_t result = 0; 1751 if (getEffectChain_l(sessionId) != 0) { 1752 result = EFFECT_SESSION; 1753 } 1754 1755 for (size_t i = 0; i < mTracks.size(); ++i) { 1756 sp<Track> track = mTracks[i]; 1757 if (sessionId == track->sessionId() && !track->isInvalid()) { 1758 result |= TRACK_SESSION; 1759 break; 1760 } 1761 } 1762 1763 return result; 1764} 1765 1766uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1767{ 1768 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1769 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1770 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1771 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1772 } 1773 for (size_t i = 0; i < mTracks.size(); i++) { 1774 sp<Track> track = mTracks[i]; 1775 if (sessionId == track->sessionId() && !track->isInvalid()) { 1776 return AudioSystem::getStrategyForStream(track->streamType()); 1777 } 1778 } 1779 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1780} 1781 1782 1783AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1784{ 1785 Mutex::Autolock _l(mLock); 1786 return mOutput; 1787} 1788 1789AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1790{ 1791 Mutex::Autolock _l(mLock); 1792 AudioStreamOut *output = mOutput; 1793 mOutput = NULL; 1794 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1795 // must push a NULL and wait for ack 1796 mOutputSink.clear(); 1797 mPipeSink.clear(); 1798 mNormalSink.clear(); 1799 return output; 1800} 1801 1802// this method must always be called either with ThreadBase mLock held or inside the thread loop 1803audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1804{ 1805 if (mOutput == NULL) { 1806 return NULL; 1807 } 1808 return &mOutput->stream->common; 1809} 1810 1811uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1812{ 1813 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1814} 1815 1816status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1817{ 1818 if (!isValidSyncEvent(event)) { 1819 return BAD_VALUE; 1820 } 1821 1822 Mutex::Autolock _l(mLock); 1823 1824 for (size_t i = 0; i < mTracks.size(); ++i) { 1825 sp<Track> track = mTracks[i]; 1826 if (event->triggerSession() == track->sessionId()) { 1827 (void) track->setSyncEvent(event); 1828 return NO_ERROR; 1829 } 1830 } 1831 1832 return NAME_NOT_FOUND; 1833} 1834 1835bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1836{ 1837 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1838} 1839 1840void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1841 const Vector< sp<Track> >& tracksToRemove) 1842{ 1843 size_t count = tracksToRemove.size(); 1844 if (count) { 1845 for (size_t i = 0 ; i < count ; i++) { 1846 const sp<Track>& track = tracksToRemove.itemAt(i); 1847 if (!track->isOutputTrack()) { 1848 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1849#ifdef ADD_BATTERY_DATA 1850 // to track the speaker usage 1851 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1852#endif 1853 if (track->isTerminated()) { 1854 AudioSystem::releaseOutput(mId); 1855 } 1856 } 1857 } 1858 } 1859} 1860 1861void AudioFlinger::PlaybackThread::checkSilentMode_l() 1862{ 1863 if (!mMasterMute) { 1864 char value[PROPERTY_VALUE_MAX]; 1865 if (property_get("ro.audio.silent", value, "0") > 0) { 1866 char *endptr; 1867 unsigned long ul = strtoul(value, &endptr, 0); 1868 if (*endptr == '\0' && ul != 0) { 1869 ALOGD("Silence is golden"); 1870 // The setprop command will not allow a property to be changed after 1871 // the first time it is set, so we don't have to worry about un-muting. 1872 setMasterMute_l(true); 1873 } 1874 } 1875 } 1876} 1877 1878// shared by MIXER and DIRECT, overridden by DUPLICATING 1879ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1880{ 1881 // FIXME rewrite to reduce number of system calls 1882 mLastWriteTime = systemTime(); 1883 mInWrite = true; 1884 ssize_t bytesWritten; 1885 1886 // If an NBAIO sink is present, use it to write the normal mixer's submix 1887 if (mNormalSink != 0) { 1888#define mBitShift 2 // FIXME 1889 size_t count = mBytesRemaining >> mBitShift; 1890 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1891 ATRACE_BEGIN("write"); 1892 // update the setpoint when AudioFlinger::mScreenState changes 1893 uint32_t screenState = AudioFlinger::mScreenState; 1894 if (screenState != mScreenState) { 1895 mScreenState = screenState; 1896 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1897 if (pipe != NULL) { 1898 pipe->setAvgFrames((mScreenState & 1) ? 1899 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1900 } 1901 } 1902 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1903 ATRACE_END(); 1904 if (framesWritten > 0) { 1905 bytesWritten = framesWritten << mBitShift; 1906 } else { 1907 bytesWritten = framesWritten; 1908 } 1909 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1910 if (status == NO_ERROR) { 1911 size_t totalFramesWritten = mNormalSink->framesWritten(); 1912 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1913 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1914 mLatchDValid = true; 1915 } 1916 } 1917 // otherwise use the HAL / AudioStreamOut directly 1918 } else { 1919 // Direct output and offload threads 1920 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1921 if (mUseAsyncWrite) { 1922 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1923 mWriteAckSequence += 2; 1924 mWriteAckSequence |= 1; 1925 ALOG_ASSERT(mCallbackThread != 0); 1926 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1927 } 1928 // FIXME We should have an implementation of timestamps for direct output threads. 1929 // They are used e.g for multichannel PCM playback over HDMI. 1930 bytesWritten = mOutput->stream->write(mOutput->stream, 1931 (char *)mMixBuffer + offset, mBytesRemaining); 1932 if (mUseAsyncWrite && 1933 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1934 // do not wait for async callback in case of error of full write 1935 mWriteAckSequence &= ~1; 1936 ALOG_ASSERT(mCallbackThread != 0); 1937 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1938 } 1939 } 1940 1941 mNumWrites++; 1942 mInWrite = false; 1943 mStandby = false; 1944 return bytesWritten; 1945} 1946 1947void AudioFlinger::PlaybackThread::threadLoop_drain() 1948{ 1949 if (mOutput->stream->drain) { 1950 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1951 if (mUseAsyncWrite) { 1952 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1953 mDrainSequence |= 1; 1954 ALOG_ASSERT(mCallbackThread != 0); 1955 mCallbackThread->setDraining(mDrainSequence); 1956 } 1957 mOutput->stream->drain(mOutput->stream, 1958 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1959 : AUDIO_DRAIN_ALL); 1960 } 1961} 1962 1963void AudioFlinger::PlaybackThread::threadLoop_exit() 1964{ 1965 // Default implementation has nothing to do 1966} 1967 1968/* 1969The derived values that are cached: 1970 - mixBufferSize from frame count * frame size 1971 - activeSleepTime from activeSleepTimeUs() 1972 - idleSleepTime from idleSleepTimeUs() 1973 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1974 - maxPeriod from frame count and sample rate (MIXER only) 1975 1976The parameters that affect these derived values are: 1977 - frame count 1978 - frame size 1979 - sample rate 1980 - device type: A2DP or not 1981 - device latency 1982 - format: PCM or not 1983 - active sleep time 1984 - idle sleep time 1985*/ 1986 1987void AudioFlinger::PlaybackThread::cacheParameters_l() 1988{ 1989 mixBufferSize = mNormalFrameCount * mFrameSize; 1990 activeSleepTime = activeSleepTimeUs(); 1991 idleSleepTime = idleSleepTimeUs(); 1992} 1993 1994void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1995{ 1996 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1997 this, streamType, mTracks.size()); 1998 Mutex::Autolock _l(mLock); 1999 2000 size_t size = mTracks.size(); 2001 for (size_t i = 0; i < size; i++) { 2002 sp<Track> t = mTracks[i]; 2003 if (t->streamType() == streamType) { 2004 t->invalidate(); 2005 } 2006 } 2007} 2008 2009status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2010{ 2011 int session = chain->sessionId(); 2012 int16_t *buffer = mMixBuffer; 2013 bool ownsBuffer = false; 2014 2015 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2016 if (session > 0) { 2017 // Only one effect chain can be present in direct output thread and it uses 2018 // the mix buffer as input 2019 if (mType != DIRECT) { 2020 size_t numSamples = mNormalFrameCount * mChannelCount; 2021 buffer = new int16_t[numSamples]; 2022 memset(buffer, 0, numSamples * sizeof(int16_t)); 2023 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2024 ownsBuffer = true; 2025 } 2026 2027 // Attach all tracks with same session ID to this chain. 2028 for (size_t i = 0; i < mTracks.size(); ++i) { 2029 sp<Track> track = mTracks[i]; 2030 if (session == track->sessionId()) { 2031 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2032 buffer); 2033 track->setMainBuffer(buffer); 2034 chain->incTrackCnt(); 2035 } 2036 } 2037 2038 // indicate all active tracks in the chain 2039 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2040 sp<Track> track = mActiveTracks[i].promote(); 2041 if (track == 0) { 2042 continue; 2043 } 2044 if (session == track->sessionId()) { 2045 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2046 chain->incActiveTrackCnt(); 2047 } 2048 } 2049 } 2050 2051 chain->setInBuffer(buffer, ownsBuffer); 2052 chain->setOutBuffer(mMixBuffer); 2053 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2054 // chains list in order to be processed last as it contains output stage effects 2055 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2056 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2057 // after track specific effects and before output stage 2058 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2059 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2060 // Effect chain for other sessions are inserted at beginning of effect 2061 // chains list to be processed before output mix effects. Relative order between other 2062 // sessions is not important 2063 size_t size = mEffectChains.size(); 2064 size_t i = 0; 2065 for (i = 0; i < size; i++) { 2066 if (mEffectChains[i]->sessionId() < session) { 2067 break; 2068 } 2069 } 2070 mEffectChains.insertAt(chain, i); 2071 checkSuspendOnAddEffectChain_l(chain); 2072 2073 return NO_ERROR; 2074} 2075 2076size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2077{ 2078 int session = chain->sessionId(); 2079 2080 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2081 2082 for (size_t i = 0; i < mEffectChains.size(); i++) { 2083 if (chain == mEffectChains[i]) { 2084 mEffectChains.removeAt(i); 2085 // detach all active tracks from the chain 2086 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2087 sp<Track> track = mActiveTracks[i].promote(); 2088 if (track == 0) { 2089 continue; 2090 } 2091 if (session == track->sessionId()) { 2092 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2093 chain.get(), session); 2094 chain->decActiveTrackCnt(); 2095 } 2096 } 2097 2098 // detach all tracks with same session ID from this chain 2099 for (size_t i = 0; i < mTracks.size(); ++i) { 2100 sp<Track> track = mTracks[i]; 2101 if (session == track->sessionId()) { 2102 track->setMainBuffer(mMixBuffer); 2103 chain->decTrackCnt(); 2104 } 2105 } 2106 break; 2107 } 2108 } 2109 return mEffectChains.size(); 2110} 2111 2112status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2113 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2114{ 2115 Mutex::Autolock _l(mLock); 2116 return attachAuxEffect_l(track, EffectId); 2117} 2118 2119status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2120 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2121{ 2122 status_t status = NO_ERROR; 2123 2124 if (EffectId == 0) { 2125 track->setAuxBuffer(0, NULL); 2126 } else { 2127 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2128 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2129 if (effect != 0) { 2130 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2131 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2132 } else { 2133 status = INVALID_OPERATION; 2134 } 2135 } else { 2136 status = BAD_VALUE; 2137 } 2138 } 2139 return status; 2140} 2141 2142void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2143{ 2144 for (size_t i = 0; i < mTracks.size(); ++i) { 2145 sp<Track> track = mTracks[i]; 2146 if (track->auxEffectId() == effectId) { 2147 attachAuxEffect_l(track, 0); 2148 } 2149 } 2150} 2151 2152bool AudioFlinger::PlaybackThread::threadLoop() 2153{ 2154 Vector< sp<Track> > tracksToRemove; 2155 2156 standbyTime = systemTime(); 2157 2158 // MIXER 2159 nsecs_t lastWarning = 0; 2160 2161 // DUPLICATING 2162 // FIXME could this be made local to while loop? 2163 writeFrames = 0; 2164 2165 int lastGeneration = 0; 2166 2167 cacheParameters_l(); 2168 sleepTime = idleSleepTime; 2169 2170 if (mType == MIXER) { 2171 sleepTimeShift = 0; 2172 } 2173 2174 CpuStats cpuStats; 2175 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2176 2177 acquireWakeLock(); 2178 2179 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2180 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2181 // and then that string will be logged at the next convenient opportunity. 2182 const char *logString = NULL; 2183 2184 checkSilentMode_l(); 2185 2186 while (!exitPending()) 2187 { 2188 cpuStats.sample(myName); 2189 2190 Vector< sp<EffectChain> > effectChains; 2191 2192 processConfigEvents(); 2193 2194 { // scope for mLock 2195 2196 Mutex::Autolock _l(mLock); 2197 2198 if (logString != NULL) { 2199 mNBLogWriter->logTimestamp(); 2200 mNBLogWriter->log(logString); 2201 logString = NULL; 2202 } 2203 2204 if (mLatchDValid) { 2205 mLatchQ = mLatchD; 2206 mLatchDValid = false; 2207 mLatchQValid = true; 2208 } 2209 2210 if (checkForNewParameters_l()) { 2211 cacheParameters_l(); 2212 } 2213 2214 saveOutputTracks(); 2215 if (mSignalPending) { 2216 // A signal was raised while we were unlocked 2217 mSignalPending = false; 2218 } else if (waitingAsyncCallback_l()) { 2219 if (exitPending()) { 2220 break; 2221 } 2222 releaseWakeLock_l(); 2223 mWakeLockUids.clear(); 2224 mActiveTracksGeneration++; 2225 ALOGV("wait async completion"); 2226 mWaitWorkCV.wait(mLock); 2227 ALOGV("async completion/wake"); 2228 acquireWakeLock_l(); 2229 standbyTime = systemTime() + standbyDelay; 2230 sleepTime = 0; 2231 2232 continue; 2233 } 2234 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2235 isSuspended()) { 2236 // put audio hardware into standby after short delay 2237 if (shouldStandby_l()) { 2238 2239 threadLoop_standby(); 2240 2241 mStandby = true; 2242 } 2243 2244 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2245 // we're about to wait, flush the binder command buffer 2246 IPCThreadState::self()->flushCommands(); 2247 2248 clearOutputTracks(); 2249 2250 if (exitPending()) { 2251 break; 2252 } 2253 2254 releaseWakeLock_l(); 2255 mWakeLockUids.clear(); 2256 mActiveTracksGeneration++; 2257 // wait until we have something to do... 2258 ALOGV("%s going to sleep", myName.string()); 2259 mWaitWorkCV.wait(mLock); 2260 ALOGV("%s waking up", myName.string()); 2261 acquireWakeLock_l(); 2262 2263 mMixerStatus = MIXER_IDLE; 2264 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2265 mBytesWritten = 0; 2266 mBytesRemaining = 0; 2267 checkSilentMode_l(); 2268 2269 standbyTime = systemTime() + standbyDelay; 2270 sleepTime = idleSleepTime; 2271 if (mType == MIXER) { 2272 sleepTimeShift = 0; 2273 } 2274 2275 continue; 2276 } 2277 } 2278 // mMixerStatusIgnoringFastTracks is also updated internally 2279 mMixerStatus = prepareTracks_l(&tracksToRemove); 2280 2281 // compare with previously applied list 2282 if (lastGeneration != mActiveTracksGeneration) { 2283 // update wakelock 2284 updateWakeLockUids_l(mWakeLockUids); 2285 lastGeneration = mActiveTracksGeneration; 2286 } 2287 2288 // prevent any changes in effect chain list and in each effect chain 2289 // during mixing and effect process as the audio buffers could be deleted 2290 // or modified if an effect is created or deleted 2291 lockEffectChains_l(effectChains); 2292 } // mLock scope ends 2293 2294 if (mBytesRemaining == 0) { 2295 mCurrentWriteLength = 0; 2296 if (mMixerStatus == MIXER_TRACKS_READY) { 2297 // threadLoop_mix() sets mCurrentWriteLength 2298 threadLoop_mix(); 2299 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2300 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2301 // threadLoop_sleepTime sets sleepTime to 0 if data 2302 // must be written to HAL 2303 threadLoop_sleepTime(); 2304 if (sleepTime == 0) { 2305 mCurrentWriteLength = mixBufferSize; 2306 } 2307 } 2308 mBytesRemaining = mCurrentWriteLength; 2309 if (isSuspended()) { 2310 sleepTime = suspendSleepTimeUs(); 2311 // simulate write to HAL when suspended 2312 mBytesWritten += mixBufferSize; 2313 mBytesRemaining = 0; 2314 } 2315 2316 // only process effects if we're going to write 2317 if (sleepTime == 0 && mType != OFFLOAD) { 2318 for (size_t i = 0; i < effectChains.size(); i ++) { 2319 effectChains[i]->process_l(); 2320 } 2321 } 2322 } 2323 // Process effect chains for offloaded thread even if no audio 2324 // was read from audio track: process only updates effect state 2325 // and thus does have to be synchronized with audio writes but may have 2326 // to be called while waiting for async write callback 2327 if (mType == OFFLOAD) { 2328 for (size_t i = 0; i < effectChains.size(); i ++) { 2329 effectChains[i]->process_l(); 2330 } 2331 } 2332 2333 // enable changes in effect chain 2334 unlockEffectChains(effectChains); 2335 2336 if (!waitingAsyncCallback()) { 2337 // sleepTime == 0 means we must write to audio hardware 2338 if (sleepTime == 0) { 2339 if (mBytesRemaining) { 2340 ssize_t ret = threadLoop_write(); 2341 if (ret < 0) { 2342 mBytesRemaining = 0; 2343 } else { 2344 mBytesWritten += ret; 2345 mBytesRemaining -= ret; 2346 } 2347 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2348 (mMixerStatus == MIXER_DRAIN_ALL)) { 2349 threadLoop_drain(); 2350 } 2351if (mType == MIXER) { 2352 // write blocked detection 2353 nsecs_t now = systemTime(); 2354 nsecs_t delta = now - mLastWriteTime; 2355 if (!mStandby && delta > maxPeriod) { 2356 mNumDelayedWrites++; 2357 if ((now - lastWarning) > kWarningThrottleNs) { 2358 ATRACE_NAME("underrun"); 2359 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2360 ns2ms(delta), mNumDelayedWrites, this); 2361 lastWarning = now; 2362 } 2363 } 2364} 2365 2366 } else { 2367 usleep(sleepTime); 2368 } 2369 } 2370 2371 // Finally let go of removed track(s), without the lock held 2372 // since we can't guarantee the destructors won't acquire that 2373 // same lock. This will also mutate and push a new fast mixer state. 2374 threadLoop_removeTracks(tracksToRemove); 2375 tracksToRemove.clear(); 2376 2377 // FIXME I don't understand the need for this here; 2378 // it was in the original code but maybe the 2379 // assignment in saveOutputTracks() makes this unnecessary? 2380 clearOutputTracks(); 2381 2382 // Effect chains will be actually deleted here if they were removed from 2383 // mEffectChains list during mixing or effects processing 2384 effectChains.clear(); 2385 2386 // FIXME Note that the above .clear() is no longer necessary since effectChains 2387 // is now local to this block, but will keep it for now (at least until merge done). 2388 } 2389 2390 threadLoop_exit(); 2391 2392 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2393 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2394 // put output stream into standby mode 2395 if (!mStandby) { 2396 mOutput->stream->common.standby(&mOutput->stream->common); 2397 } 2398 } 2399 2400 releaseWakeLock(); 2401 mWakeLockUids.clear(); 2402 mActiveTracksGeneration++; 2403 2404 ALOGV("Thread %p type %d exiting", this, mType); 2405 return false; 2406} 2407 2408// removeTracks_l() must be called with ThreadBase::mLock held 2409void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2410{ 2411 size_t count = tracksToRemove.size(); 2412 if (count) { 2413 for (size_t i=0 ; i<count ; i++) { 2414 const sp<Track>& track = tracksToRemove.itemAt(i); 2415 mActiveTracks.remove(track); 2416 mWakeLockUids.remove(track->uid()); 2417 mActiveTracksGeneration++; 2418 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2419 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2420 if (chain != 0) { 2421 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2422 track->sessionId()); 2423 chain->decActiveTrackCnt(); 2424 } 2425 if (track->isTerminated()) { 2426 removeTrack_l(track); 2427 } 2428 } 2429 } 2430 2431} 2432 2433status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2434{ 2435 if (mNormalSink != 0) { 2436 return mNormalSink->getTimestamp(timestamp); 2437 } 2438 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2439 uint64_t position64; 2440 int ret = mOutput->stream->get_presentation_position( 2441 mOutput->stream, &position64, ×tamp.mTime); 2442 if (ret == 0) { 2443 timestamp.mPosition = (uint32_t)position64; 2444 return NO_ERROR; 2445 } 2446 } 2447 return INVALID_OPERATION; 2448} 2449// ---------------------------------------------------------------------------- 2450 2451AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2452 audio_io_handle_t id, audio_devices_t device, type_t type) 2453 : PlaybackThread(audioFlinger, output, id, device, type), 2454 // mAudioMixer below 2455 // mFastMixer below 2456 mFastMixerFutex(0) 2457 // mOutputSink below 2458 // mPipeSink below 2459 // mNormalSink below 2460{ 2461 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2462 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2463 "mFrameCount=%d, mNormalFrameCount=%d", 2464 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2465 mNormalFrameCount); 2466 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2467 2468 // FIXME - Current mixer implementation only supports stereo output 2469 if (mChannelCount != FCC_2) { 2470 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2471 } 2472 2473 // create an NBAIO sink for the HAL output stream, and negotiate 2474 mOutputSink = new AudioStreamOutSink(output->stream); 2475 size_t numCounterOffers = 0; 2476 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2477 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2478 ALOG_ASSERT(index == 0); 2479 2480 // initialize fast mixer depending on configuration 2481 bool initFastMixer; 2482 switch (kUseFastMixer) { 2483 case FastMixer_Never: 2484 initFastMixer = false; 2485 break; 2486 case FastMixer_Always: 2487 initFastMixer = true; 2488 break; 2489 case FastMixer_Static: 2490 case FastMixer_Dynamic: 2491 initFastMixer = mFrameCount < mNormalFrameCount; 2492 break; 2493 } 2494 if (initFastMixer) { 2495 2496 // create a MonoPipe to connect our submix to FastMixer 2497 NBAIO_Format format = mOutputSink->format(); 2498 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2499 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2500 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2501 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2502 const NBAIO_Format offers[1] = {format}; 2503 size_t numCounterOffers = 0; 2504 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2505 ALOG_ASSERT(index == 0); 2506 monoPipe->setAvgFrames((mScreenState & 1) ? 2507 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2508 mPipeSink = monoPipe; 2509 2510#ifdef TEE_SINK 2511 if (mTeeSinkOutputEnabled) { 2512 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2513 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2514 numCounterOffers = 0; 2515 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2516 ALOG_ASSERT(index == 0); 2517 mTeeSink = teeSink; 2518 PipeReader *teeSource = new PipeReader(*teeSink); 2519 numCounterOffers = 0; 2520 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2521 ALOG_ASSERT(index == 0); 2522 mTeeSource = teeSource; 2523 } 2524#endif 2525 2526 // create fast mixer and configure it initially with just one fast track for our submix 2527 mFastMixer = new FastMixer(); 2528 FastMixerStateQueue *sq = mFastMixer->sq(); 2529#ifdef STATE_QUEUE_DUMP 2530 sq->setObserverDump(&mStateQueueObserverDump); 2531 sq->setMutatorDump(&mStateQueueMutatorDump); 2532#endif 2533 FastMixerState *state = sq->begin(); 2534 FastTrack *fastTrack = &state->mFastTracks[0]; 2535 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2536 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2537 fastTrack->mVolumeProvider = NULL; 2538 fastTrack->mGeneration++; 2539 state->mFastTracksGen++; 2540 state->mTrackMask = 1; 2541 // fast mixer will use the HAL output sink 2542 state->mOutputSink = mOutputSink.get(); 2543 state->mOutputSinkGen++; 2544 state->mFrameCount = mFrameCount; 2545 state->mCommand = FastMixerState::COLD_IDLE; 2546 // already done in constructor initialization list 2547 //mFastMixerFutex = 0; 2548 state->mColdFutexAddr = &mFastMixerFutex; 2549 state->mColdGen++; 2550 state->mDumpState = &mFastMixerDumpState; 2551#ifdef TEE_SINK 2552 state->mTeeSink = mTeeSink.get(); 2553#endif 2554 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2555 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2556 sq->end(); 2557 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2558 2559 // start the fast mixer 2560 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2561 pid_t tid = mFastMixer->getTid(); 2562 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2563 if (err != 0) { 2564 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2565 kPriorityFastMixer, getpid_cached, tid, err); 2566 } 2567 2568#ifdef AUDIO_WATCHDOG 2569 // create and start the watchdog 2570 mAudioWatchdog = new AudioWatchdog(); 2571 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2572 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2573 tid = mAudioWatchdog->getTid(); 2574 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#endif 2580 2581 } else { 2582 mFastMixer = NULL; 2583 } 2584 2585 switch (kUseFastMixer) { 2586 case FastMixer_Never: 2587 case FastMixer_Dynamic: 2588 mNormalSink = mOutputSink; 2589 break; 2590 case FastMixer_Always: 2591 mNormalSink = mPipeSink; 2592 break; 2593 case FastMixer_Static: 2594 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2595 break; 2596 } 2597} 2598 2599AudioFlinger::MixerThread::~MixerThread() 2600{ 2601 if (mFastMixer != NULL) { 2602 FastMixerStateQueue *sq = mFastMixer->sq(); 2603 FastMixerState *state = sq->begin(); 2604 if (state->mCommand == FastMixerState::COLD_IDLE) { 2605 int32_t old = android_atomic_inc(&mFastMixerFutex); 2606 if (old == -1) { 2607 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2608 } 2609 } 2610 state->mCommand = FastMixerState::EXIT; 2611 sq->end(); 2612 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2613 mFastMixer->join(); 2614 // Though the fast mixer thread has exited, it's state queue is still valid. 2615 // We'll use that extract the final state which contains one remaining fast track 2616 // corresponding to our sub-mix. 2617 state = sq->begin(); 2618 ALOG_ASSERT(state->mTrackMask == 1); 2619 FastTrack *fastTrack = &state->mFastTracks[0]; 2620 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2621 delete fastTrack->mBufferProvider; 2622 sq->end(false /*didModify*/); 2623 delete mFastMixer; 2624#ifdef AUDIO_WATCHDOG 2625 if (mAudioWatchdog != 0) { 2626 mAudioWatchdog->requestExit(); 2627 mAudioWatchdog->requestExitAndWait(); 2628 mAudioWatchdog.clear(); 2629 } 2630#endif 2631 } 2632 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2633 delete mAudioMixer; 2634} 2635 2636 2637uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2638{ 2639 if (mFastMixer != NULL) { 2640 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2641 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2642 } 2643 return latency; 2644} 2645 2646 2647void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2648{ 2649 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2650} 2651 2652ssize_t AudioFlinger::MixerThread::threadLoop_write() 2653{ 2654 // FIXME we should only do one push per cycle; confirm this is true 2655 // Start the fast mixer if it's not already running 2656 if (mFastMixer != NULL) { 2657 FastMixerStateQueue *sq = mFastMixer->sq(); 2658 FastMixerState *state = sq->begin(); 2659 if (state->mCommand != FastMixerState::MIX_WRITE && 2660 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2661 if (state->mCommand == FastMixerState::COLD_IDLE) { 2662 int32_t old = android_atomic_inc(&mFastMixerFutex); 2663 if (old == -1) { 2664 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2665 } 2666#ifdef AUDIO_WATCHDOG 2667 if (mAudioWatchdog != 0) { 2668 mAudioWatchdog->resume(); 2669 } 2670#endif 2671 } 2672 state->mCommand = FastMixerState::MIX_WRITE; 2673 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2674 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2675 sq->end(); 2676 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2677 if (kUseFastMixer == FastMixer_Dynamic) { 2678 mNormalSink = mPipeSink; 2679 } 2680 } else { 2681 sq->end(false /*didModify*/); 2682 } 2683 } 2684 return PlaybackThread::threadLoop_write(); 2685} 2686 2687void AudioFlinger::MixerThread::threadLoop_standby() 2688{ 2689 // Idle the fast mixer if it's currently running 2690 if (mFastMixer != NULL) { 2691 FastMixerStateQueue *sq = mFastMixer->sq(); 2692 FastMixerState *state = sq->begin(); 2693 if (!(state->mCommand & FastMixerState::IDLE)) { 2694 state->mCommand = FastMixerState::COLD_IDLE; 2695 state->mColdFutexAddr = &mFastMixerFutex; 2696 state->mColdGen++; 2697 mFastMixerFutex = 0; 2698 sq->end(); 2699 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2700 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2701 if (kUseFastMixer == FastMixer_Dynamic) { 2702 mNormalSink = mOutputSink; 2703 } 2704#ifdef AUDIO_WATCHDOG 2705 if (mAudioWatchdog != 0) { 2706 mAudioWatchdog->pause(); 2707 } 2708#endif 2709 } else { 2710 sq->end(false /*didModify*/); 2711 } 2712 } 2713 PlaybackThread::threadLoop_standby(); 2714} 2715 2716// Empty implementation for standard mixer 2717// Overridden for offloaded playback 2718void AudioFlinger::PlaybackThread::flushOutput_l() 2719{ 2720} 2721 2722bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2723{ 2724 return false; 2725} 2726 2727bool AudioFlinger::PlaybackThread::shouldStandby_l() 2728{ 2729 return !mStandby; 2730} 2731 2732bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2733{ 2734 Mutex::Autolock _l(mLock); 2735 return waitingAsyncCallback_l(); 2736} 2737 2738// shared by MIXER and DIRECT, overridden by DUPLICATING 2739void AudioFlinger::PlaybackThread::threadLoop_standby() 2740{ 2741 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2742 mOutput->stream->common.standby(&mOutput->stream->common); 2743 if (mUseAsyncWrite != 0) { 2744 // discard any pending drain or write ack by incrementing sequence 2745 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2746 mDrainSequence = (mDrainSequence + 2) & ~1; 2747 ALOG_ASSERT(mCallbackThread != 0); 2748 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2749 mCallbackThread->setDraining(mDrainSequence); 2750 } 2751} 2752 2753void AudioFlinger::MixerThread::threadLoop_mix() 2754{ 2755 // obtain the presentation timestamp of the next output buffer 2756 int64_t pts; 2757 status_t status = INVALID_OPERATION; 2758 2759 if (mNormalSink != 0) { 2760 status = mNormalSink->getNextWriteTimestamp(&pts); 2761 } else { 2762 status = mOutputSink->getNextWriteTimestamp(&pts); 2763 } 2764 2765 if (status != NO_ERROR) { 2766 pts = AudioBufferProvider::kInvalidPTS; 2767 } 2768 2769 // mix buffers... 2770 mAudioMixer->process(pts); 2771 mCurrentWriteLength = mixBufferSize; 2772 // increase sleep time progressively when application underrun condition clears. 2773 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2774 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2775 // such that we would underrun the audio HAL. 2776 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2777 sleepTimeShift--; 2778 } 2779 sleepTime = 0; 2780 standbyTime = systemTime() + standbyDelay; 2781 //TODO: delay standby when effects have a tail 2782} 2783 2784void AudioFlinger::MixerThread::threadLoop_sleepTime() 2785{ 2786 // If no tracks are ready, sleep once for the duration of an output 2787 // buffer size, then write 0s to the output 2788 if (sleepTime == 0) { 2789 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2790 sleepTime = activeSleepTime >> sleepTimeShift; 2791 if (sleepTime < kMinThreadSleepTimeUs) { 2792 sleepTime = kMinThreadSleepTimeUs; 2793 } 2794 // reduce sleep time in case of consecutive application underruns to avoid 2795 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2796 // duration we would end up writing less data than needed by the audio HAL if 2797 // the condition persists. 2798 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2799 sleepTimeShift++; 2800 } 2801 } else { 2802 sleepTime = idleSleepTime; 2803 } 2804 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2805 memset (mMixBuffer, 0, mixBufferSize); 2806 sleepTime = 0; 2807 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2808 "anticipated start"); 2809 } 2810 // TODO add standby time extension fct of effect tail 2811} 2812 2813// prepareTracks_l() must be called with ThreadBase::mLock held 2814AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2815 Vector< sp<Track> > *tracksToRemove) 2816{ 2817 2818 mixer_state mixerStatus = MIXER_IDLE; 2819 // find out which tracks need to be processed 2820 size_t count = mActiveTracks.size(); 2821 size_t mixedTracks = 0; 2822 size_t tracksWithEffect = 0; 2823 // counts only _active_ fast tracks 2824 size_t fastTracks = 0; 2825 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2826 2827 float masterVolume = mMasterVolume; 2828 bool masterMute = mMasterMute; 2829 2830 if (masterMute) { 2831 masterVolume = 0; 2832 } 2833 // Delegate master volume control to effect in output mix effect chain if needed 2834 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2835 if (chain != 0) { 2836 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2837 chain->setVolume_l(&v, &v); 2838 masterVolume = (float)((v + (1 << 23)) >> 24); 2839 chain.clear(); 2840 } 2841 2842 // prepare a new state to push 2843 FastMixerStateQueue *sq = NULL; 2844 FastMixerState *state = NULL; 2845 bool didModify = false; 2846 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2847 if (mFastMixer != NULL) { 2848 sq = mFastMixer->sq(); 2849 state = sq->begin(); 2850 } 2851 2852 for (size_t i=0 ; i<count ; i++) { 2853 const sp<Track> t = mActiveTracks[i].promote(); 2854 if (t == 0) { 2855 continue; 2856 } 2857 2858 // this const just means the local variable doesn't change 2859 Track* const track = t.get(); 2860 2861 // process fast tracks 2862 if (track->isFastTrack()) { 2863 2864 // It's theoretically possible (though unlikely) for a fast track to be created 2865 // and then removed within the same normal mix cycle. This is not a problem, as 2866 // the track never becomes active so it's fast mixer slot is never touched. 2867 // The converse, of removing an (active) track and then creating a new track 2868 // at the identical fast mixer slot within the same normal mix cycle, 2869 // is impossible because the slot isn't marked available until the end of each cycle. 2870 int j = track->mFastIndex; 2871 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2872 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2873 FastTrack *fastTrack = &state->mFastTracks[j]; 2874 2875 // Determine whether the track is currently in underrun condition, 2876 // and whether it had a recent underrun. 2877 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2878 FastTrackUnderruns underruns = ftDump->mUnderruns; 2879 uint32_t recentFull = (underruns.mBitFields.mFull - 2880 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2881 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2882 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2883 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2884 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2885 uint32_t recentUnderruns = recentPartial + recentEmpty; 2886 track->mObservedUnderruns = underruns; 2887 // don't count underruns that occur while stopping or pausing 2888 // or stopped which can occur when flush() is called while active 2889 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2890 recentUnderruns > 0) { 2891 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2892 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2893 } 2894 2895 // This is similar to the state machine for normal tracks, 2896 // with a few modifications for fast tracks. 2897 bool isActive = true; 2898 switch (track->mState) { 2899 case TrackBase::STOPPING_1: 2900 // track stays active in STOPPING_1 state until first underrun 2901 if (recentUnderruns > 0 || track->isTerminated()) { 2902 track->mState = TrackBase::STOPPING_2; 2903 } 2904 break; 2905 case TrackBase::PAUSING: 2906 // ramp down is not yet implemented 2907 track->setPaused(); 2908 break; 2909 case TrackBase::RESUMING: 2910 // ramp up is not yet implemented 2911 track->mState = TrackBase::ACTIVE; 2912 break; 2913 case TrackBase::ACTIVE: 2914 if (recentFull > 0 || recentPartial > 0) { 2915 // track has provided at least some frames recently: reset retry count 2916 track->mRetryCount = kMaxTrackRetries; 2917 } 2918 if (recentUnderruns == 0) { 2919 // no recent underruns: stay active 2920 break; 2921 } 2922 // there has recently been an underrun of some kind 2923 if (track->sharedBuffer() == 0) { 2924 // were any of the recent underruns "empty" (no frames available)? 2925 if (recentEmpty == 0) { 2926 // no, then ignore the partial underruns as they are allowed indefinitely 2927 break; 2928 } 2929 // there has recently been an "empty" underrun: decrement the retry counter 2930 if (--(track->mRetryCount) > 0) { 2931 break; 2932 } 2933 // indicate to client process that the track was disabled because of underrun; 2934 // it will then automatically call start() when data is available 2935 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2936 // remove from active list, but state remains ACTIVE [confusing but true] 2937 isActive = false; 2938 break; 2939 } 2940 // fall through 2941 case TrackBase::STOPPING_2: 2942 case TrackBase::PAUSED: 2943 case TrackBase::STOPPED: 2944 case TrackBase::FLUSHED: // flush() while active 2945 // Check for presentation complete if track is inactive 2946 // We have consumed all the buffers of this track. 2947 // This would be incomplete if we auto-paused on underrun 2948 { 2949 size_t audioHALFrames = 2950 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2951 size_t framesWritten = mBytesWritten / mFrameSize; 2952 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2953 // track stays in active list until presentation is complete 2954 break; 2955 } 2956 } 2957 if (track->isStopping_2()) { 2958 track->mState = TrackBase::STOPPED; 2959 } 2960 if (track->isStopped()) { 2961 // Can't reset directly, as fast mixer is still polling this track 2962 // track->reset(); 2963 // So instead mark this track as needing to be reset after push with ack 2964 resetMask |= 1 << i; 2965 } 2966 isActive = false; 2967 break; 2968 case TrackBase::IDLE: 2969 default: 2970 LOG_FATAL("unexpected track state %d", track->mState); 2971 } 2972 2973 if (isActive) { 2974 // was it previously inactive? 2975 if (!(state->mTrackMask & (1 << j))) { 2976 ExtendedAudioBufferProvider *eabp = track; 2977 VolumeProvider *vp = track; 2978 fastTrack->mBufferProvider = eabp; 2979 fastTrack->mVolumeProvider = vp; 2980 fastTrack->mChannelMask = track->mChannelMask; 2981 fastTrack->mGeneration++; 2982 state->mTrackMask |= 1 << j; 2983 didModify = true; 2984 // no acknowledgement required for newly active tracks 2985 } 2986 // cache the combined master volume and stream type volume for fast mixer; this 2987 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2988 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2989 ++fastTracks; 2990 } else { 2991 // was it previously active? 2992 if (state->mTrackMask & (1 << j)) { 2993 fastTrack->mBufferProvider = NULL; 2994 fastTrack->mGeneration++; 2995 state->mTrackMask &= ~(1 << j); 2996 didModify = true; 2997 // If any fast tracks were removed, we must wait for acknowledgement 2998 // because we're about to decrement the last sp<> on those tracks. 2999 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3000 } else { 3001 LOG_FATAL("fast track %d should have been active", j); 3002 } 3003 tracksToRemove->add(track); 3004 // Avoids a misleading display in dumpsys 3005 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3006 } 3007 continue; 3008 } 3009 3010 { // local variable scope to avoid goto warning 3011 3012 audio_track_cblk_t* cblk = track->cblk(); 3013 3014 // The first time a track is added we wait 3015 // for all its buffers to be filled before processing it 3016 int name = track->name(); 3017 // make sure that we have enough frames to mix one full buffer. 3018 // enforce this condition only once to enable draining the buffer in case the client 3019 // app does not call stop() and relies on underrun to stop: 3020 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3021 // during last round 3022 size_t desiredFrames; 3023 uint32_t sr = track->sampleRate(); 3024 if (sr == mSampleRate) { 3025 desiredFrames = mNormalFrameCount; 3026 } else { 3027 // +1 for rounding and +1 for additional sample needed for interpolation 3028 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3029 // add frames already consumed but not yet released by the resampler 3030 // because cblk->framesReady() will include these frames 3031 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3032 // the minimum track buffer size is normally twice the number of frames necessary 3033 // to fill one buffer and the resampler should not leave more than one buffer worth 3034 // of unreleased frames after each pass, but just in case... 3035 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3036 } 3037 uint32_t minFrames = 1; 3038 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3039 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3040 minFrames = desiredFrames; 3041 } 3042 3043 size_t framesReady = track->framesReady(); 3044 if ((framesReady >= minFrames) && track->isReady() && 3045 !track->isPaused() && !track->isTerminated()) 3046 { 3047 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3048 3049 mixedTracks++; 3050 3051 // track->mainBuffer() != mMixBuffer means there is an effect chain 3052 // connected to the track 3053 chain.clear(); 3054 if (track->mainBuffer() != mMixBuffer) { 3055 chain = getEffectChain_l(track->sessionId()); 3056 // Delegate volume control to effect in track effect chain if needed 3057 if (chain != 0) { 3058 tracksWithEffect++; 3059 } else { 3060 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3061 "session %d", 3062 name, track->sessionId()); 3063 } 3064 } 3065 3066 3067 int param = AudioMixer::VOLUME; 3068 if (track->mFillingUpStatus == Track::FS_FILLED) { 3069 // no ramp for the first volume setting 3070 track->mFillingUpStatus = Track::FS_ACTIVE; 3071 if (track->mState == TrackBase::RESUMING) { 3072 track->mState = TrackBase::ACTIVE; 3073 param = AudioMixer::RAMP_VOLUME; 3074 } 3075 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3076 // FIXME should not make a decision based on mServer 3077 } else if (cblk->mServer != 0) { 3078 // If the track is stopped before the first frame was mixed, 3079 // do not apply ramp 3080 param = AudioMixer::RAMP_VOLUME; 3081 } 3082 3083 // compute volume for this track 3084 uint32_t vl, vr, va; 3085 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3086 vl = vr = va = 0; 3087 if (track->isPausing()) { 3088 track->setPaused(); 3089 } 3090 } else { 3091 3092 // read original volumes with volume control 3093 float typeVolume = mStreamTypes[track->streamType()].volume; 3094 float v = masterVolume * typeVolume; 3095 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3096 uint32_t vlr = proxy->getVolumeLR(); 3097 vl = vlr & 0xFFFF; 3098 vr = vlr >> 16; 3099 // track volumes come from shared memory, so can't be trusted and must be clamped 3100 if (vl > MAX_GAIN_INT) { 3101 ALOGV("Track left volume out of range: %04X", vl); 3102 vl = MAX_GAIN_INT; 3103 } 3104 if (vr > MAX_GAIN_INT) { 3105 ALOGV("Track right volume out of range: %04X", vr); 3106 vr = MAX_GAIN_INT; 3107 } 3108 // now apply the master volume and stream type volume 3109 vl = (uint32_t)(v * vl) << 12; 3110 vr = (uint32_t)(v * vr) << 12; 3111 // assuming master volume and stream type volume each go up to 1.0, 3112 // vl and vr are now in 8.24 format 3113 3114 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3115 // send level comes from shared memory and so may be corrupt 3116 if (sendLevel > MAX_GAIN_INT) { 3117 ALOGV("Track send level out of range: %04X", sendLevel); 3118 sendLevel = MAX_GAIN_INT; 3119 } 3120 va = (uint32_t)(v * sendLevel); 3121 } 3122 3123 // Delegate volume control to effect in track effect chain if needed 3124 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3125 // Do not ramp volume if volume is controlled by effect 3126 param = AudioMixer::VOLUME; 3127 track->mHasVolumeController = true; 3128 } else { 3129 // force no volume ramp when volume controller was just disabled or removed 3130 // from effect chain to avoid volume spike 3131 if (track->mHasVolumeController) { 3132 param = AudioMixer::VOLUME; 3133 } 3134 track->mHasVolumeController = false; 3135 } 3136 3137 // Convert volumes from 8.24 to 4.12 format 3138 // This additional clamping is needed in case chain->setVolume_l() overshot 3139 vl = (vl + (1 << 11)) >> 12; 3140 if (vl > MAX_GAIN_INT) { 3141 vl = MAX_GAIN_INT; 3142 } 3143 vr = (vr + (1 << 11)) >> 12; 3144 if (vr > MAX_GAIN_INT) { 3145 vr = MAX_GAIN_INT; 3146 } 3147 3148 if (va > MAX_GAIN_INT) { 3149 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3150 } 3151 3152 // XXX: these things DON'T need to be done each time 3153 mAudioMixer->setBufferProvider(name, track); 3154 mAudioMixer->enable(name); 3155 3156 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3157 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3158 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3159 mAudioMixer->setParameter( 3160 name, 3161 AudioMixer::TRACK, 3162 AudioMixer::FORMAT, (void *)track->format()); 3163 mAudioMixer->setParameter( 3164 name, 3165 AudioMixer::TRACK, 3166 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3167 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3168 uint32_t maxSampleRate = mSampleRate * 2; 3169 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3170 if (reqSampleRate == 0) { 3171 reqSampleRate = mSampleRate; 3172 } else if (reqSampleRate > maxSampleRate) { 3173 reqSampleRate = maxSampleRate; 3174 } 3175 mAudioMixer->setParameter( 3176 name, 3177 AudioMixer::RESAMPLE, 3178 AudioMixer::SAMPLE_RATE, 3179 (void *)(uintptr_t)reqSampleRate); 3180 mAudioMixer->setParameter( 3181 name, 3182 AudioMixer::TRACK, 3183 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3184 mAudioMixer->setParameter( 3185 name, 3186 AudioMixer::TRACK, 3187 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3188 3189 // reset retry count 3190 track->mRetryCount = kMaxTrackRetries; 3191 3192 // If one track is ready, set the mixer ready if: 3193 // - the mixer was not ready during previous round OR 3194 // - no other track is not ready 3195 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3196 mixerStatus != MIXER_TRACKS_ENABLED) { 3197 mixerStatus = MIXER_TRACKS_READY; 3198 } 3199 } else { 3200 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3201 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3202 } 3203 // clear effect chain input buffer if an active track underruns to avoid sending 3204 // previous audio buffer again to effects 3205 chain = getEffectChain_l(track->sessionId()); 3206 if (chain != 0) { 3207 chain->clearInputBuffer(); 3208 } 3209 3210 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3211 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3212 track->isStopped() || track->isPaused()) { 3213 // We have consumed all the buffers of this track. 3214 // Remove it from the list of active tracks. 3215 // TODO: use actual buffer filling status instead of latency when available from 3216 // audio HAL 3217 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3218 size_t framesWritten = mBytesWritten / mFrameSize; 3219 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3220 if (track->isStopped()) { 3221 track->reset(); 3222 } 3223 tracksToRemove->add(track); 3224 } 3225 } else { 3226 // No buffers for this track. Give it a few chances to 3227 // fill a buffer, then remove it from active list. 3228 if (--(track->mRetryCount) <= 0) { 3229 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3230 tracksToRemove->add(track); 3231 // indicate to client process that the track was disabled because of underrun; 3232 // it will then automatically call start() when data is available 3233 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3234 // If one track is not ready, mark the mixer also not ready if: 3235 // - the mixer was ready during previous round OR 3236 // - no other track is ready 3237 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3238 mixerStatus != MIXER_TRACKS_READY) { 3239 mixerStatus = MIXER_TRACKS_ENABLED; 3240 } 3241 } 3242 mAudioMixer->disable(name); 3243 } 3244 3245 } // local variable scope to avoid goto warning 3246track_is_ready: ; 3247 3248 } 3249 3250 // Push the new FastMixer state if necessary 3251 bool pauseAudioWatchdog = false; 3252 if (didModify) { 3253 state->mFastTracksGen++; 3254 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3255 if (kUseFastMixer == FastMixer_Dynamic && 3256 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3257 state->mCommand = FastMixerState::COLD_IDLE; 3258 state->mColdFutexAddr = &mFastMixerFutex; 3259 state->mColdGen++; 3260 mFastMixerFutex = 0; 3261 if (kUseFastMixer == FastMixer_Dynamic) { 3262 mNormalSink = mOutputSink; 3263 } 3264 // If we go into cold idle, need to wait for acknowledgement 3265 // so that fast mixer stops doing I/O. 3266 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3267 pauseAudioWatchdog = true; 3268 } 3269 } 3270 if (sq != NULL) { 3271 sq->end(didModify); 3272 sq->push(block); 3273 } 3274#ifdef AUDIO_WATCHDOG 3275 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3276 mAudioWatchdog->pause(); 3277 } 3278#endif 3279 3280 // Now perform the deferred reset on fast tracks that have stopped 3281 while (resetMask != 0) { 3282 size_t i = __builtin_ctz(resetMask); 3283 ALOG_ASSERT(i < count); 3284 resetMask &= ~(1 << i); 3285 sp<Track> t = mActiveTracks[i].promote(); 3286 if (t == 0) { 3287 continue; 3288 } 3289 Track* track = t.get(); 3290 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3291 track->reset(); 3292 } 3293 3294 // remove all the tracks that need to be... 3295 removeTracks_l(*tracksToRemove); 3296 3297 // mix buffer must be cleared if all tracks are connected to an 3298 // effect chain as in this case the mixer will not write to 3299 // mix buffer and track effects will accumulate into it 3300 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3301 (mixedTracks == 0 && fastTracks > 0))) { 3302 // FIXME as a performance optimization, should remember previous zero status 3303 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3304 } 3305 3306 // if any fast tracks, then status is ready 3307 mMixerStatusIgnoringFastTracks = mixerStatus; 3308 if (fastTracks > 0) { 3309 mixerStatus = MIXER_TRACKS_READY; 3310 } 3311 return mixerStatus; 3312} 3313 3314// getTrackName_l() must be called with ThreadBase::mLock held 3315int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3316{ 3317 return mAudioMixer->getTrackName(channelMask, sessionId); 3318} 3319 3320// deleteTrackName_l() must be called with ThreadBase::mLock held 3321void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3322{ 3323 ALOGV("remove track (%d) and delete from mixer", name); 3324 mAudioMixer->deleteTrackName(name); 3325} 3326 3327// checkForNewParameters_l() must be called with ThreadBase::mLock held 3328bool AudioFlinger::MixerThread::checkForNewParameters_l() 3329{ 3330 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3331 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3332 bool reconfig = false; 3333 3334 while (!mNewParameters.isEmpty()) { 3335 3336 if (mFastMixer != NULL) { 3337 FastMixerStateQueue *sq = mFastMixer->sq(); 3338 FastMixerState *state = sq->begin(); 3339 if (!(state->mCommand & FastMixerState::IDLE)) { 3340 previousCommand = state->mCommand; 3341 state->mCommand = FastMixerState::HOT_IDLE; 3342 sq->end(); 3343 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3344 } else { 3345 sq->end(false /*didModify*/); 3346 } 3347 } 3348 3349 status_t status = NO_ERROR; 3350 String8 keyValuePair = mNewParameters[0]; 3351 AudioParameter param = AudioParameter(keyValuePair); 3352 int value; 3353 3354 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3355 reconfig = true; 3356 } 3357 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3358 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3359 status = BAD_VALUE; 3360 } else { 3361 reconfig = true; 3362 } 3363 } 3364 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3365 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3366 status = BAD_VALUE; 3367 } else { 3368 reconfig = true; 3369 } 3370 } 3371 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3372 // do not accept frame count changes if tracks are open as the track buffer 3373 // size depends on frame count and correct behavior would not be guaranteed 3374 // if frame count is changed after track creation 3375 if (!mTracks.isEmpty()) { 3376 status = INVALID_OPERATION; 3377 } else { 3378 reconfig = true; 3379 } 3380 } 3381 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3382#ifdef ADD_BATTERY_DATA 3383 // when changing the audio output device, call addBatteryData to notify 3384 // the change 3385 if (mOutDevice != value) { 3386 uint32_t params = 0; 3387 // check whether speaker is on 3388 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3389 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3390 } 3391 3392 audio_devices_t deviceWithoutSpeaker 3393 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3394 // check if any other device (except speaker) is on 3395 if (value & deviceWithoutSpeaker ) { 3396 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3397 } 3398 3399 if (params != 0) { 3400 addBatteryData(params); 3401 } 3402 } 3403#endif 3404 3405 // forward device change to effects that have requested to be 3406 // aware of attached audio device. 3407 if (value != AUDIO_DEVICE_NONE) { 3408 mOutDevice = value; 3409 for (size_t i = 0; i < mEffectChains.size(); i++) { 3410 mEffectChains[i]->setDevice_l(mOutDevice); 3411 } 3412 } 3413 } 3414 3415 if (status == NO_ERROR) { 3416 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3417 keyValuePair.string()); 3418 if (!mStandby && status == INVALID_OPERATION) { 3419 mOutput->stream->common.standby(&mOutput->stream->common); 3420 mStandby = true; 3421 mBytesWritten = 0; 3422 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3423 keyValuePair.string()); 3424 } 3425 if (status == NO_ERROR && reconfig) { 3426 readOutputParameters(); 3427 delete mAudioMixer; 3428 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3429 for (size_t i = 0; i < mTracks.size() ; i++) { 3430 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3431 if (name < 0) { 3432 break; 3433 } 3434 mTracks[i]->mName = name; 3435 } 3436 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3437 } 3438 } 3439 3440 mNewParameters.removeAt(0); 3441 3442 mParamStatus = status; 3443 mParamCond.signal(); 3444 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3445 // already timed out waiting for the status and will never signal the condition. 3446 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3447 } 3448 3449 if (!(previousCommand & FastMixerState::IDLE)) { 3450 ALOG_ASSERT(mFastMixer != NULL); 3451 FastMixerStateQueue *sq = mFastMixer->sq(); 3452 FastMixerState *state = sq->begin(); 3453 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3454 state->mCommand = previousCommand; 3455 sq->end(); 3456 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3457 } 3458 3459 return reconfig; 3460} 3461 3462 3463void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3464{ 3465 const size_t SIZE = 256; 3466 char buffer[SIZE]; 3467 String8 result; 3468 3469 PlaybackThread::dumpInternals(fd, args); 3470 3471 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3472 result.append(buffer); 3473 write(fd, result.string(), result.size()); 3474 3475 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3476 const FastMixerDumpState copy(mFastMixerDumpState); 3477 copy.dump(fd); 3478 3479#ifdef STATE_QUEUE_DUMP 3480 // Similar for state queue 3481 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3482 observerCopy.dump(fd); 3483 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3484 mutatorCopy.dump(fd); 3485#endif 3486 3487#ifdef TEE_SINK 3488 // Write the tee output to a .wav file 3489 dumpTee(fd, mTeeSource, mId); 3490#endif 3491 3492#ifdef AUDIO_WATCHDOG 3493 if (mAudioWatchdog != 0) { 3494 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3495 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3496 wdCopy.dump(fd); 3497 } 3498#endif 3499} 3500 3501uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3502{ 3503 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3504} 3505 3506uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3507{ 3508 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3509} 3510 3511void AudioFlinger::MixerThread::cacheParameters_l() 3512{ 3513 PlaybackThread::cacheParameters_l(); 3514 3515 // FIXME: Relaxed timing because of a certain device that can't meet latency 3516 // Should be reduced to 2x after the vendor fixes the driver issue 3517 // increase threshold again due to low power audio mode. The way this warning 3518 // threshold is calculated and its usefulness should be reconsidered anyway. 3519 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3520} 3521 3522// ---------------------------------------------------------------------------- 3523 3524AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3525 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3526 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3527 // mLeftVolFloat, mRightVolFloat 3528{ 3529} 3530 3531AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3532 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3533 ThreadBase::type_t type) 3534 : PlaybackThread(audioFlinger, output, id, device, type) 3535 // mLeftVolFloat, mRightVolFloat 3536{ 3537} 3538 3539AudioFlinger::DirectOutputThread::~DirectOutputThread() 3540{ 3541} 3542 3543void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3544{ 3545 audio_track_cblk_t* cblk = track->cblk(); 3546 float left, right; 3547 3548 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3549 left = right = 0; 3550 } else { 3551 float typeVolume = mStreamTypes[track->streamType()].volume; 3552 float v = mMasterVolume * typeVolume; 3553 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3554 uint32_t vlr = proxy->getVolumeLR(); 3555 float v_clamped = v * (vlr & 0xFFFF); 3556 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3557 left = v_clamped/MAX_GAIN; 3558 v_clamped = v * (vlr >> 16); 3559 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3560 right = v_clamped/MAX_GAIN; 3561 } 3562 3563 if (lastTrack) { 3564 if (left != mLeftVolFloat || right != mRightVolFloat) { 3565 mLeftVolFloat = left; 3566 mRightVolFloat = right; 3567 3568 // Convert volumes from float to 8.24 3569 uint32_t vl = (uint32_t)(left * (1 << 24)); 3570 uint32_t vr = (uint32_t)(right * (1 << 24)); 3571 3572 // Delegate volume control to effect in track effect chain if needed 3573 // only one effect chain can be present on DirectOutputThread, so if 3574 // there is one, the track is connected to it 3575 if (!mEffectChains.isEmpty()) { 3576 mEffectChains[0]->setVolume_l(&vl, &vr); 3577 left = (float)vl / (1 << 24); 3578 right = (float)vr / (1 << 24); 3579 } 3580 if (mOutput->stream->set_volume) { 3581 mOutput->stream->set_volume(mOutput->stream, left, right); 3582 } 3583 } 3584 } 3585} 3586 3587 3588AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3589 Vector< sp<Track> > *tracksToRemove 3590) 3591{ 3592 size_t count = mActiveTracks.size(); 3593 mixer_state mixerStatus = MIXER_IDLE; 3594 3595 // find out which tracks need to be processed 3596 for (size_t i = 0; i < count; i++) { 3597 sp<Track> t = mActiveTracks[i].promote(); 3598 // The track died recently 3599 if (t == 0) { 3600 continue; 3601 } 3602 3603 Track* const track = t.get(); 3604 audio_track_cblk_t* cblk = track->cblk(); 3605 // Only consider last track started for volume and mixer state control. 3606 // In theory an older track could underrun and restart after the new one starts 3607 // but as we only care about the transition phase between two tracks on a 3608 // direct output, it is not a problem to ignore the underrun case. 3609 sp<Track> l = mLatestActiveTrack.promote(); 3610 bool last = l.get() == track; 3611 3612 // The first time a track is added we wait 3613 // for all its buffers to be filled before processing it 3614 uint32_t minFrames; 3615 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3616 minFrames = mNormalFrameCount; 3617 } else { 3618 minFrames = 1; 3619 } 3620 3621 if ((track->framesReady() >= minFrames) && track->isReady() && 3622 !track->isPaused() && !track->isTerminated()) 3623 { 3624 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3625 3626 if (track->mFillingUpStatus == Track::FS_FILLED) { 3627 track->mFillingUpStatus = Track::FS_ACTIVE; 3628 // make sure processVolume_l() will apply new volume even if 0 3629 mLeftVolFloat = mRightVolFloat = -1.0; 3630 if (track->mState == TrackBase::RESUMING) { 3631 track->mState = TrackBase::ACTIVE; 3632 } 3633 } 3634 3635 // compute volume for this track 3636 processVolume_l(track, last); 3637 if (last) { 3638 // reset retry count 3639 track->mRetryCount = kMaxTrackRetriesDirect; 3640 mActiveTrack = t; 3641 mixerStatus = MIXER_TRACKS_READY; 3642 } 3643 } else { 3644 // clear effect chain input buffer if the last active track started underruns 3645 // to avoid sending previous audio buffer again to effects 3646 if (!mEffectChains.isEmpty() && last) { 3647 mEffectChains[0]->clearInputBuffer(); 3648 } 3649 3650 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3651 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3652 track->isStopped() || track->isPaused()) { 3653 // We have consumed all the buffers of this track. 3654 // Remove it from the list of active tracks. 3655 // TODO: implement behavior for compressed audio 3656 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3657 size_t framesWritten = mBytesWritten / mFrameSize; 3658 if (mStandby || !last || 3659 track->presentationComplete(framesWritten, audioHALFrames)) { 3660 if (track->isStopped()) { 3661 track->reset(); 3662 } 3663 tracksToRemove->add(track); 3664 } 3665 } else { 3666 // No buffers for this track. Give it a few chances to 3667 // fill a buffer, then remove it from active list. 3668 // Only consider last track started for mixer state control 3669 if (--(track->mRetryCount) <= 0) { 3670 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3671 tracksToRemove->add(track); 3672 // indicate to client process that the track was disabled because of underrun; 3673 // it will then automatically call start() when data is available 3674 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3675 } else if (last) { 3676 mixerStatus = MIXER_TRACKS_ENABLED; 3677 } 3678 } 3679 } 3680 } 3681 3682 // remove all the tracks that need to be... 3683 removeTracks_l(*tracksToRemove); 3684 3685 return mixerStatus; 3686} 3687 3688void AudioFlinger::DirectOutputThread::threadLoop_mix() 3689{ 3690 size_t frameCount = mFrameCount; 3691 int8_t *curBuf = (int8_t *)mMixBuffer; 3692 // output audio to hardware 3693 while (frameCount) { 3694 AudioBufferProvider::Buffer buffer; 3695 buffer.frameCount = frameCount; 3696 mActiveTrack->getNextBuffer(&buffer); 3697 if (buffer.raw == NULL) { 3698 memset(curBuf, 0, frameCount * mFrameSize); 3699 break; 3700 } 3701 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3702 frameCount -= buffer.frameCount; 3703 curBuf += buffer.frameCount * mFrameSize; 3704 mActiveTrack->releaseBuffer(&buffer); 3705 } 3706 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3707 sleepTime = 0; 3708 standbyTime = systemTime() + standbyDelay; 3709 mActiveTrack.clear(); 3710} 3711 3712void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3713{ 3714 if (sleepTime == 0) { 3715 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3716 sleepTime = activeSleepTime; 3717 } else { 3718 sleepTime = idleSleepTime; 3719 } 3720 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3721 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3722 sleepTime = 0; 3723 } 3724} 3725 3726// getTrackName_l() must be called with ThreadBase::mLock held 3727int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3728 int sessionId) 3729{ 3730 return 0; 3731} 3732 3733// deleteTrackName_l() must be called with ThreadBase::mLock held 3734void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3735{ 3736} 3737 3738// checkForNewParameters_l() must be called with ThreadBase::mLock held 3739bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3740{ 3741 bool reconfig = false; 3742 3743 while (!mNewParameters.isEmpty()) { 3744 status_t status = NO_ERROR; 3745 String8 keyValuePair = mNewParameters[0]; 3746 AudioParameter param = AudioParameter(keyValuePair); 3747 int value; 3748 3749 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3750 // do not accept frame count changes if tracks are open as the track buffer 3751 // size depends on frame count and correct behavior would not be garantied 3752 // if frame count is changed after track creation 3753 if (!mTracks.isEmpty()) { 3754 status = INVALID_OPERATION; 3755 } else { 3756 reconfig = true; 3757 } 3758 } 3759 if (status == NO_ERROR) { 3760 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3761 keyValuePair.string()); 3762 if (!mStandby && status == INVALID_OPERATION) { 3763 mOutput->stream->common.standby(&mOutput->stream->common); 3764 mStandby = true; 3765 mBytesWritten = 0; 3766 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3767 keyValuePair.string()); 3768 } 3769 if (status == NO_ERROR && reconfig) { 3770 readOutputParameters(); 3771 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3772 } 3773 } 3774 3775 mNewParameters.removeAt(0); 3776 3777 mParamStatus = status; 3778 mParamCond.signal(); 3779 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3780 // already timed out waiting for the status and will never signal the condition. 3781 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3782 } 3783 return reconfig; 3784} 3785 3786uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3787{ 3788 uint32_t time; 3789 if (audio_is_linear_pcm(mFormat)) { 3790 time = PlaybackThread::activeSleepTimeUs(); 3791 } else { 3792 time = 10000; 3793 } 3794 return time; 3795} 3796 3797uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3798{ 3799 uint32_t time; 3800 if (audio_is_linear_pcm(mFormat)) { 3801 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3802 } else { 3803 time = 10000; 3804 } 3805 return time; 3806} 3807 3808uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3809{ 3810 uint32_t time; 3811 if (audio_is_linear_pcm(mFormat)) { 3812 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3813 } else { 3814 time = 10000; 3815 } 3816 return time; 3817} 3818 3819void AudioFlinger::DirectOutputThread::cacheParameters_l() 3820{ 3821 PlaybackThread::cacheParameters_l(); 3822 3823 // use shorter standby delay as on normal output to release 3824 // hardware resources as soon as possible 3825 if (audio_is_linear_pcm(mFormat)) { 3826 standbyDelay = microseconds(activeSleepTime*2); 3827 } else { 3828 standbyDelay = kOffloadStandbyDelayNs; 3829 } 3830} 3831 3832// ---------------------------------------------------------------------------- 3833 3834AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3835 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3836 : Thread(false /*canCallJava*/), 3837 mPlaybackThread(playbackThread), 3838 mWriteAckSequence(0), 3839 mDrainSequence(0) 3840{ 3841} 3842 3843AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3844{ 3845} 3846 3847void AudioFlinger::AsyncCallbackThread::onFirstRef() 3848{ 3849 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3850} 3851 3852bool AudioFlinger::AsyncCallbackThread::threadLoop() 3853{ 3854 while (!exitPending()) { 3855 uint32_t writeAckSequence; 3856 uint32_t drainSequence; 3857 3858 { 3859 Mutex::Autolock _l(mLock); 3860 while (!((mWriteAckSequence & 1) || 3861 (mDrainSequence & 1) || 3862 exitPending())) { 3863 mWaitWorkCV.wait(mLock); 3864 } 3865 3866 if (exitPending()) { 3867 break; 3868 } 3869 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3870 mWriteAckSequence, mDrainSequence); 3871 writeAckSequence = mWriteAckSequence; 3872 mWriteAckSequence &= ~1; 3873 drainSequence = mDrainSequence; 3874 mDrainSequence &= ~1; 3875 } 3876 { 3877 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3878 if (playbackThread != 0) { 3879 if (writeAckSequence & 1) { 3880 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3881 } 3882 if (drainSequence & 1) { 3883 playbackThread->resetDraining(drainSequence >> 1); 3884 } 3885 } 3886 } 3887 } 3888 return false; 3889} 3890 3891void AudioFlinger::AsyncCallbackThread::exit() 3892{ 3893 ALOGV("AsyncCallbackThread::exit"); 3894 Mutex::Autolock _l(mLock); 3895 requestExit(); 3896 mWaitWorkCV.broadcast(); 3897} 3898 3899void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3900{ 3901 Mutex::Autolock _l(mLock); 3902 // bit 0 is cleared 3903 mWriteAckSequence = sequence << 1; 3904} 3905 3906void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3907{ 3908 Mutex::Autolock _l(mLock); 3909 // ignore unexpected callbacks 3910 if (mWriteAckSequence & 2) { 3911 mWriteAckSequence |= 1; 3912 mWaitWorkCV.signal(); 3913 } 3914} 3915 3916void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3917{ 3918 Mutex::Autolock _l(mLock); 3919 // bit 0 is cleared 3920 mDrainSequence = sequence << 1; 3921} 3922 3923void AudioFlinger::AsyncCallbackThread::resetDraining() 3924{ 3925 Mutex::Autolock _l(mLock); 3926 // ignore unexpected callbacks 3927 if (mDrainSequence & 2) { 3928 mDrainSequence |= 1; 3929 mWaitWorkCV.signal(); 3930 } 3931} 3932 3933 3934// ---------------------------------------------------------------------------- 3935AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3936 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3937 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3938 mHwPaused(false), 3939 mFlushPending(false), 3940 mPausedBytesRemaining(0) 3941{ 3942 //FIXME: mStandby should be set to true by ThreadBase constructor 3943 mStandby = true; 3944} 3945 3946void AudioFlinger::OffloadThread::threadLoop_exit() 3947{ 3948 if (mFlushPending || mHwPaused) { 3949 // If a flush is pending or track was paused, just discard buffered data 3950 flushHw_l(); 3951 } else { 3952 mMixerStatus = MIXER_DRAIN_ALL; 3953 threadLoop_drain(); 3954 } 3955 mCallbackThread->exit(); 3956 PlaybackThread::threadLoop_exit(); 3957} 3958 3959AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3960 Vector< sp<Track> > *tracksToRemove 3961) 3962{ 3963 size_t count = mActiveTracks.size(); 3964 3965 mixer_state mixerStatus = MIXER_IDLE; 3966 bool doHwPause = false; 3967 bool doHwResume = false; 3968 3969 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3970 3971 // find out which tracks need to be processed 3972 for (size_t i = 0; i < count; i++) { 3973 sp<Track> t = mActiveTracks[i].promote(); 3974 // The track died recently 3975 if (t == 0) { 3976 continue; 3977 } 3978 Track* const track = t.get(); 3979 audio_track_cblk_t* cblk = track->cblk(); 3980 // Only consider last track started for volume and mixer state control. 3981 // In theory an older track could underrun and restart after the new one starts 3982 // but as we only care about the transition phase between two tracks on a 3983 // direct output, it is not a problem to ignore the underrun case. 3984 sp<Track> l = mLatestActiveTrack.promote(); 3985 bool last = l.get() == track; 3986 3987 if (track->isPausing()) { 3988 track->setPaused(); 3989 if (last) { 3990 if (!mHwPaused) { 3991 doHwPause = true; 3992 mHwPaused = true; 3993 } 3994 // If we were part way through writing the mixbuffer to 3995 // the HAL we must save this until we resume 3996 // BUG - this will be wrong if a different track is made active, 3997 // in that case we want to discard the pending data in the 3998 // mixbuffer and tell the client to present it again when the 3999 // track is resumed 4000 mPausedWriteLength = mCurrentWriteLength; 4001 mPausedBytesRemaining = mBytesRemaining; 4002 mBytesRemaining = 0; // stop writing 4003 } 4004 tracksToRemove->add(track); 4005 } else if (track->framesReady() && track->isReady() && 4006 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4007 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4008 if (track->mFillingUpStatus == Track::FS_FILLED) { 4009 track->mFillingUpStatus = Track::FS_ACTIVE; 4010 // make sure processVolume_l() will apply new volume even if 0 4011 mLeftVolFloat = mRightVolFloat = -1.0; 4012 if (track->mState == TrackBase::RESUMING) { 4013 track->mState = TrackBase::ACTIVE; 4014 if (last) { 4015 if (mPausedBytesRemaining) { 4016 // Need to continue write that was interrupted 4017 mCurrentWriteLength = mPausedWriteLength; 4018 mBytesRemaining = mPausedBytesRemaining; 4019 mPausedBytesRemaining = 0; 4020 } 4021 if (mHwPaused) { 4022 doHwResume = true; 4023 mHwPaused = false; 4024 // threadLoop_mix() will handle the case that we need to 4025 // resume an interrupted write 4026 } 4027 // enable write to audio HAL 4028 sleepTime = 0; 4029 } 4030 } 4031 } 4032 4033 if (last) { 4034 sp<Track> previousTrack = mPreviousTrack.promote(); 4035 if (previousTrack != 0) { 4036 if (track != previousTrack.get()) { 4037 // Flush any data still being written from last track 4038 mBytesRemaining = 0; 4039 if (mPausedBytesRemaining) { 4040 // Last track was paused so we also need to flush saved 4041 // mixbuffer state and invalidate track so that it will 4042 // re-submit that unwritten data when it is next resumed 4043 mPausedBytesRemaining = 0; 4044 // Invalidate is a bit drastic - would be more efficient 4045 // to have a flag to tell client that some of the 4046 // previously written data was lost 4047 previousTrack->invalidate(); 4048 } 4049 // flush data already sent to the DSP if changing audio session as audio 4050 // comes from a different source. Also invalidate previous track to force a 4051 // seek when resuming. 4052 if (previousTrack->sessionId() != track->sessionId()) { 4053 previousTrack->invalidate(); 4054 mFlushPending = true; 4055 } 4056 } 4057 } 4058 mPreviousTrack = track; 4059 // reset retry count 4060 track->mRetryCount = kMaxTrackRetriesOffload; 4061 mActiveTrack = t; 4062 mixerStatus = MIXER_TRACKS_READY; 4063 } 4064 } else { 4065 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4066 if (track->isStopping_1()) { 4067 // Hardware buffer can hold a large amount of audio so we must 4068 // wait for all current track's data to drain before we say 4069 // that the track is stopped. 4070 if (mBytesRemaining == 0) { 4071 // Only start draining when all data in mixbuffer 4072 // has been written 4073 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4074 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4075 // do not drain if no data was ever sent to HAL (mStandby == true) 4076 if (last && !mStandby) { 4077 // do not modify drain sequence if we are already draining. This happens 4078 // when resuming from pause after drain. 4079 if ((mDrainSequence & 1) == 0) { 4080 sleepTime = 0; 4081 standbyTime = systemTime() + standbyDelay; 4082 mixerStatus = MIXER_DRAIN_TRACK; 4083 mDrainSequence += 2; 4084 } 4085 if (mHwPaused) { 4086 // It is possible to move from PAUSED to STOPPING_1 without 4087 // a resume so we must ensure hardware is running 4088 doHwResume = true; 4089 mHwPaused = false; 4090 } 4091 } 4092 } 4093 } else if (track->isStopping_2()) { 4094 // Drain has completed or we are in standby, signal presentation complete 4095 if (!(mDrainSequence & 1) || !last || mStandby) { 4096 track->mState = TrackBase::STOPPED; 4097 size_t audioHALFrames = 4098 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4099 size_t framesWritten = 4100 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4101 track->presentationComplete(framesWritten, audioHALFrames); 4102 track->reset(); 4103 tracksToRemove->add(track); 4104 } 4105 } else { 4106 // No buffers for this track. Give it a few chances to 4107 // fill a buffer, then remove it from active list. 4108 if (--(track->mRetryCount) <= 0) { 4109 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4110 track->name()); 4111 tracksToRemove->add(track); 4112 // indicate to client process that the track was disabled because of underrun; 4113 // it will then automatically call start() when data is available 4114 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4115 } else if (last){ 4116 mixerStatus = MIXER_TRACKS_ENABLED; 4117 } 4118 } 4119 } 4120 // compute volume for this track 4121 processVolume_l(track, last); 4122 } 4123 4124 // make sure the pause/flush/resume sequence is executed in the right order. 4125 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4126 // before flush and then resume HW. This can happen in case of pause/flush/resume 4127 // if resume is received before pause is executed. 4128 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4129 mOutput->stream->pause(mOutput->stream); 4130 if (!doHwPause) { 4131 doHwResume = true; 4132 } 4133 } 4134 if (mFlushPending) { 4135 flushHw_l(); 4136 mFlushPending = false; 4137 } 4138 if (!mStandby && doHwResume) { 4139 mOutput->stream->resume(mOutput->stream); 4140 } 4141 4142 // remove all the tracks that need to be... 4143 removeTracks_l(*tracksToRemove); 4144 4145 return mixerStatus; 4146} 4147 4148void AudioFlinger::OffloadThread::flushOutput_l() 4149{ 4150 mFlushPending = true; 4151} 4152 4153// must be called with thread mutex locked 4154bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4155{ 4156 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4157 mWriteAckSequence, mDrainSequence); 4158 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4159 return true; 4160 } 4161 return false; 4162} 4163 4164// must be called with thread mutex locked 4165bool AudioFlinger::OffloadThread::shouldStandby_l() 4166{ 4167 bool TrackPaused = false; 4168 4169 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4170 // after a timeout and we will enter standby then. 4171 if (mTracks.size() > 0) { 4172 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4173 } 4174 4175 return !mStandby && !TrackPaused; 4176} 4177 4178 4179bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4180{ 4181 Mutex::Autolock _l(mLock); 4182 return waitingAsyncCallback_l(); 4183} 4184 4185void AudioFlinger::OffloadThread::flushHw_l() 4186{ 4187 mOutput->stream->flush(mOutput->stream); 4188 // Flush anything still waiting in the mixbuffer 4189 mCurrentWriteLength = 0; 4190 mBytesRemaining = 0; 4191 mPausedWriteLength = 0; 4192 mPausedBytesRemaining = 0; 4193 if (mUseAsyncWrite) { 4194 // discard any pending drain or write ack by incrementing sequence 4195 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4196 mDrainSequence = (mDrainSequence + 2) & ~1; 4197 ALOG_ASSERT(mCallbackThread != 0); 4198 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4199 mCallbackThread->setDraining(mDrainSequence); 4200 } 4201} 4202 4203// ---------------------------------------------------------------------------- 4204 4205AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4206 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4207 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4208 DUPLICATING), 4209 mWaitTimeMs(UINT_MAX) 4210{ 4211 addOutputTrack(mainThread); 4212} 4213 4214AudioFlinger::DuplicatingThread::~DuplicatingThread() 4215{ 4216 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4217 mOutputTracks[i]->destroy(); 4218 } 4219} 4220 4221void AudioFlinger::DuplicatingThread::threadLoop_mix() 4222{ 4223 // mix buffers... 4224 if (outputsReady(outputTracks)) { 4225 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4226 } else { 4227 memset(mMixBuffer, 0, mixBufferSize); 4228 } 4229 sleepTime = 0; 4230 writeFrames = mNormalFrameCount; 4231 mCurrentWriteLength = mixBufferSize; 4232 standbyTime = systemTime() + standbyDelay; 4233} 4234 4235void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4236{ 4237 if (sleepTime == 0) { 4238 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4239 sleepTime = activeSleepTime; 4240 } else { 4241 sleepTime = idleSleepTime; 4242 } 4243 } else if (mBytesWritten != 0) { 4244 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4245 writeFrames = mNormalFrameCount; 4246 memset(mMixBuffer, 0, mixBufferSize); 4247 } else { 4248 // flush remaining overflow buffers in output tracks 4249 writeFrames = 0; 4250 } 4251 sleepTime = 0; 4252 } 4253} 4254 4255ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4256{ 4257 for (size_t i = 0; i < outputTracks.size(); i++) { 4258 outputTracks[i]->write(mMixBuffer, writeFrames); 4259 } 4260 mStandby = false; 4261 return (ssize_t)mixBufferSize; 4262} 4263 4264void AudioFlinger::DuplicatingThread::threadLoop_standby() 4265{ 4266 // DuplicatingThread implements standby by stopping all tracks 4267 for (size_t i = 0; i < outputTracks.size(); i++) { 4268 outputTracks[i]->stop(); 4269 } 4270} 4271 4272void AudioFlinger::DuplicatingThread::saveOutputTracks() 4273{ 4274 outputTracks = mOutputTracks; 4275} 4276 4277void AudioFlinger::DuplicatingThread::clearOutputTracks() 4278{ 4279 outputTracks.clear(); 4280} 4281 4282void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4283{ 4284 Mutex::Autolock _l(mLock); 4285 // FIXME explain this formula 4286 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4287 OutputTrack *outputTrack = new OutputTrack(thread, 4288 this, 4289 mSampleRate, 4290 mFormat, 4291 mChannelMask, 4292 frameCount, 4293 IPCThreadState::self()->getCallingUid()); 4294 if (outputTrack->cblk() != NULL) { 4295 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4296 mOutputTracks.add(outputTrack); 4297 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4298 updateWaitTime_l(); 4299 } 4300} 4301 4302void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4303{ 4304 Mutex::Autolock _l(mLock); 4305 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4306 if (mOutputTracks[i]->thread() == thread) { 4307 mOutputTracks[i]->destroy(); 4308 mOutputTracks.removeAt(i); 4309 updateWaitTime_l(); 4310 return; 4311 } 4312 } 4313 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4314} 4315 4316// caller must hold mLock 4317void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4318{ 4319 mWaitTimeMs = UINT_MAX; 4320 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4321 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4322 if (strong != 0) { 4323 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4324 if (waitTimeMs < mWaitTimeMs) { 4325 mWaitTimeMs = waitTimeMs; 4326 } 4327 } 4328 } 4329} 4330 4331 4332bool AudioFlinger::DuplicatingThread::outputsReady( 4333 const SortedVector< sp<OutputTrack> > &outputTracks) 4334{ 4335 for (size_t i = 0; i < outputTracks.size(); i++) { 4336 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4337 if (thread == 0) { 4338 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4339 outputTracks[i].get()); 4340 return false; 4341 } 4342 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4343 // see note at standby() declaration 4344 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4345 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4346 thread.get()); 4347 return false; 4348 } 4349 } 4350 return true; 4351} 4352 4353uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4354{ 4355 return (mWaitTimeMs * 1000) / 2; 4356} 4357 4358void AudioFlinger::DuplicatingThread::cacheParameters_l() 4359{ 4360 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4361 updateWaitTime_l(); 4362 4363 MixerThread::cacheParameters_l(); 4364} 4365 4366// ---------------------------------------------------------------------------- 4367// Record 4368// ---------------------------------------------------------------------------- 4369 4370AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4371 AudioStreamIn *input, 4372 uint32_t sampleRate, 4373 audio_channel_mask_t channelMask, 4374 audio_io_handle_t id, 4375 audio_devices_t outDevice, 4376 audio_devices_t inDevice 4377#ifdef TEE_SINK 4378 , const sp<NBAIO_Sink>& teeSink 4379#endif 4380 ) : 4381 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4382 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4383 // mRsmpInIndex and mBufferSize set by readInputParameters() 4384 mReqChannelCount(popcount(channelMask)), 4385 mReqSampleRate(sampleRate) 4386 // mBytesRead is only meaningful while active, and so is cleared in start() 4387 // (but might be better to also clear here for dump?) 4388#ifdef TEE_SINK 4389 , mTeeSink(teeSink) 4390#endif 4391{ 4392 snprintf(mName, kNameLength, "AudioIn_%X", id); 4393 4394 readInputParameters(); 4395} 4396 4397 4398AudioFlinger::RecordThread::~RecordThread() 4399{ 4400 delete[] mRsmpInBuffer; 4401 delete mResampler; 4402 delete[] mRsmpOutBuffer; 4403} 4404 4405void AudioFlinger::RecordThread::onFirstRef() 4406{ 4407 run(mName, PRIORITY_URGENT_AUDIO); 4408} 4409 4410status_t AudioFlinger::RecordThread::readyToRun() 4411{ 4412 status_t status = initCheck(); 4413 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4414 return status; 4415} 4416 4417bool AudioFlinger::RecordThread::threadLoop() 4418{ 4419 AudioBufferProvider::Buffer buffer; 4420 sp<RecordTrack> activeTrack; 4421 Vector< sp<EffectChain> > effectChains; 4422 4423 nsecs_t lastWarning = 0; 4424 4425 inputStandBy(); 4426 { 4427 Mutex::Autolock _l(mLock); 4428 activeTrack = mActiveTrack; 4429 acquireWakeLock_l(activeTrack != 0 ? activeTrack->uid() : -1); 4430 } 4431 4432 // used to verify we've read at least once before evaluating how many bytes were read 4433 bool readOnce = false; 4434 4435 // start recording 4436 while (!exitPending()) { 4437 4438 processConfigEvents(); 4439 4440 { // scope for mLock 4441 Mutex::Autolock _l(mLock); 4442 checkForNewParameters_l(); 4443 if (mActiveTrack != 0 && activeTrack != mActiveTrack) { 4444 SortedVector<int> tmp; 4445 tmp.add(mActiveTrack->uid()); 4446 updateWakeLockUids_l(tmp); 4447 } 4448 activeTrack = mActiveTrack; 4449 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4450 standby(); 4451 4452 if (exitPending()) { 4453 break; 4454 } 4455 4456 releaseWakeLock_l(); 4457 ALOGV("RecordThread: loop stopping"); 4458 // go to sleep 4459 mWaitWorkCV.wait(mLock); 4460 ALOGV("RecordThread: loop starting"); 4461 acquireWakeLock_l(mActiveTrack != 0 ? mActiveTrack->uid() : -1); 4462 continue; 4463 } 4464 if (mActiveTrack != 0) { 4465 if (mActiveTrack->isTerminated()) { 4466 removeTrack_l(mActiveTrack); 4467 mActiveTrack.clear(); 4468 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4469 standby(); 4470 mActiveTrack.clear(); 4471 mStartStopCond.broadcast(); 4472 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4473 if (mReqChannelCount != mActiveTrack->channelCount()) { 4474 mActiveTrack.clear(); 4475 mStartStopCond.broadcast(); 4476 } else if (readOnce) { 4477 // record start succeeds only if first read from audio input 4478 // succeeds 4479 if (mBytesRead >= 0) { 4480 mActiveTrack->mState = TrackBase::ACTIVE; 4481 } else { 4482 mActiveTrack.clear(); 4483 } 4484 mStartStopCond.broadcast(); 4485 } 4486 mStandby = false; 4487 } 4488 } 4489 4490 lockEffectChains_l(effectChains); 4491 } 4492 4493 if (mActiveTrack != 0) { 4494 if (mActiveTrack->mState != TrackBase::ACTIVE && 4495 mActiveTrack->mState != TrackBase::RESUMING) { 4496 unlockEffectChains(effectChains); 4497 usleep(kRecordThreadSleepUs); 4498 continue; 4499 } 4500 for (size_t i = 0; i < effectChains.size(); i ++) { 4501 effectChains[i]->process_l(); 4502 } 4503 4504 buffer.frameCount = mFrameCount; 4505 status_t status = mActiveTrack->getNextBuffer(&buffer); 4506 if (status == NO_ERROR) { 4507 readOnce = true; 4508 size_t framesOut = buffer.frameCount; 4509 if (mResampler == NULL) { 4510 // no resampling 4511 while (framesOut) { 4512 size_t framesIn = mFrameCount - mRsmpInIndex; 4513 if (framesIn) { 4514 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4515 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4516 mActiveTrack->mFrameSize; 4517 if (framesIn > framesOut) 4518 framesIn = framesOut; 4519 mRsmpInIndex += framesIn; 4520 framesOut -= framesIn; 4521 if (mChannelCount == mReqChannelCount) { 4522 memcpy(dst, src, framesIn * mFrameSize); 4523 } else { 4524 if (mChannelCount == 1) { 4525 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4526 (int16_t *)src, framesIn); 4527 } else { 4528 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4529 (int16_t *)src, framesIn); 4530 } 4531 } 4532 } 4533 if (framesOut && mFrameCount == mRsmpInIndex) { 4534 void *readInto; 4535 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4536 readInto = buffer.raw; 4537 framesOut = 0; 4538 } else { 4539 readInto = mRsmpInBuffer; 4540 mRsmpInIndex = 0; 4541 } 4542 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4543 mBufferSize); 4544 if (mBytesRead <= 0) { 4545 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4546 { 4547 ALOGE("Error reading audio input"); 4548 // Force input into standby so that it tries to 4549 // recover at next read attempt 4550 inputStandBy(); 4551 usleep(kRecordThreadSleepUs); 4552 } 4553 mRsmpInIndex = mFrameCount; 4554 framesOut = 0; 4555 buffer.frameCount = 0; 4556 } 4557#ifdef TEE_SINK 4558 else if (mTeeSink != 0) { 4559 (void) mTeeSink->write(readInto, 4560 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4561 } 4562#endif 4563 } 4564 } 4565 } else { 4566 // resampling 4567 4568 // resampler accumulates, but we only have one source track 4569 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4570 // alter output frame count as if we were expecting stereo samples 4571 if (mChannelCount == 1 && mReqChannelCount == 1) { 4572 framesOut >>= 1; 4573 } 4574 mResampler->resample(mRsmpOutBuffer, framesOut, 4575 this /* AudioBufferProvider* */); 4576 // ditherAndClamp() works as long as all buffers returned by 4577 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4578 if (mChannelCount == 2 && mReqChannelCount == 1) { 4579 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4580 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4581 // the resampler always outputs stereo samples: 4582 // do post stereo to mono conversion 4583 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4584 framesOut); 4585 } else { 4586 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4587 } 4588 // now done with mRsmpOutBuffer 4589 4590 } 4591 if (mFramestoDrop == 0) { 4592 mActiveTrack->releaseBuffer(&buffer); 4593 } else { 4594 if (mFramestoDrop > 0) { 4595 mFramestoDrop -= buffer.frameCount; 4596 if (mFramestoDrop <= 0) { 4597 clearSyncStartEvent(); 4598 } 4599 } else { 4600 mFramestoDrop += buffer.frameCount; 4601 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4602 mSyncStartEvent->isCancelled()) { 4603 ALOGW("Synced record %s, session %d, trigger session %d", 4604 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4605 mActiveTrack->sessionId(), 4606 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4607 clearSyncStartEvent(); 4608 } 4609 } 4610 } 4611 mActiveTrack->clearOverflow(); 4612 } 4613 // client isn't retrieving buffers fast enough 4614 else { 4615 if (!mActiveTrack->setOverflow()) { 4616 nsecs_t now = systemTime(); 4617 if ((now - lastWarning) > kWarningThrottleNs) { 4618 ALOGW("RecordThread: buffer overflow"); 4619 lastWarning = now; 4620 } 4621 } 4622 // Release the processor for a while before asking for a new buffer. 4623 // This will give the application more chance to read from the buffer and 4624 // clear the overflow. 4625 usleep(kRecordThreadSleepUs); 4626 } 4627 } 4628 // enable changes in effect chain 4629 unlockEffectChains(effectChains); 4630 effectChains.clear(); 4631 } 4632 4633 standby(); 4634 4635 { 4636 Mutex::Autolock _l(mLock); 4637 for (size_t i = 0; i < mTracks.size(); i++) { 4638 sp<RecordTrack> track = mTracks[i]; 4639 track->invalidate(); 4640 } 4641 mActiveTrack.clear(); 4642 mStartStopCond.broadcast(); 4643 } 4644 4645 releaseWakeLock(); 4646 4647 ALOGV("RecordThread %p exiting", this); 4648 return false; 4649} 4650 4651void AudioFlinger::RecordThread::standby() 4652{ 4653 if (!mStandby) { 4654 inputStandBy(); 4655 mStandby = true; 4656 } 4657} 4658 4659void AudioFlinger::RecordThread::inputStandBy() 4660{ 4661 mInput->stream->common.standby(&mInput->stream->common); 4662} 4663 4664sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4665 const sp<AudioFlinger::Client>& client, 4666 uint32_t sampleRate, 4667 audio_format_t format, 4668 audio_channel_mask_t channelMask, 4669 size_t frameCount, 4670 int sessionId, 4671 int uid, 4672 IAudioFlinger::track_flags_t *flags, 4673 pid_t tid, 4674 status_t *status) 4675{ 4676 sp<RecordTrack> track; 4677 status_t lStatus; 4678 4679 lStatus = initCheck(); 4680 if (lStatus != NO_ERROR) { 4681 ALOGE("createRecordTrack_l() audio driver not initialized"); 4682 goto Exit; 4683 } 4684 // client expresses a preference for FAST, but we get the final say 4685 if (*flags & IAudioFlinger::TRACK_FAST) { 4686 if ( 4687 // use case: callback handler and frame count is default or at least as large as HAL 4688 ( 4689 (tid != -1) && 4690 ((frameCount == 0) || 4691 (frameCount >= mFrameCount)) 4692 ) && 4693 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4694 // mono or stereo 4695 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4696 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4697 // hardware sample rate 4698 (sampleRate == mSampleRate) && 4699 // record thread has an associated fast recorder 4700 hasFastRecorder() 4701 // FIXME test that RecordThread for this fast track has a capable output HAL 4702 // FIXME add a permission test also? 4703 ) { 4704 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4705 if (frameCount == 0) { 4706 frameCount = mFrameCount * kFastTrackMultiplier; 4707 } 4708 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4709 frameCount, mFrameCount); 4710 } else { 4711 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4712 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4713 "hasFastRecorder=%d tid=%d", 4714 frameCount, mFrameCount, format, 4715 audio_is_linear_pcm(format), 4716 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4717 *flags &= ~IAudioFlinger::TRACK_FAST; 4718 // For compatibility with AudioRecord calculation, buffer depth is forced 4719 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4720 // This is probably too conservative, but legacy application code may depend on it. 4721 // If you change this calculation, also review the start threshold which is related. 4722 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4723 size_t mNormalFrameCount = 2048; // FIXME 4724 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4725 if (minBufCount < 2) { 4726 minBufCount = 2; 4727 } 4728 size_t minFrameCount = mNormalFrameCount * minBufCount; 4729 if (frameCount < minFrameCount) { 4730 frameCount = minFrameCount; 4731 } 4732 } 4733 } 4734 4735 // FIXME use flags and tid similar to createTrack_l() 4736 4737 { // scope for mLock 4738 Mutex::Autolock _l(mLock); 4739 4740 track = new RecordTrack(this, client, sampleRate, 4741 format, channelMask, frameCount, sessionId, uid); 4742 4743 if (track->getCblk() == 0) { 4744 ALOGE("createRecordTrack_l() no control block"); 4745 lStatus = NO_MEMORY; 4746 // track must be cleared from the caller as the caller has the AF lock 4747 goto Exit; 4748 } 4749 mTracks.add(track); 4750 4751 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4752 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4753 mAudioFlinger->btNrecIsOff(); 4754 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4755 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4756 4757 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4758 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4759 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4760 // so ask activity manager to do this on our behalf 4761 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4762 } 4763 } 4764 lStatus = NO_ERROR; 4765 4766Exit: 4767 if (status) { 4768 *status = lStatus; 4769 } 4770 return track; 4771} 4772 4773status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4774 AudioSystem::sync_event_t event, 4775 int triggerSession) 4776{ 4777 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4778 sp<ThreadBase> strongMe = this; 4779 status_t status = NO_ERROR; 4780 4781 if (event == AudioSystem::SYNC_EVENT_NONE) { 4782 clearSyncStartEvent(); 4783 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4784 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4785 triggerSession, 4786 recordTrack->sessionId(), 4787 syncStartEventCallback, 4788 this); 4789 // Sync event can be cancelled by the trigger session if the track is not in a 4790 // compatible state in which case we start record immediately 4791 if (mSyncStartEvent->isCancelled()) { 4792 clearSyncStartEvent(); 4793 } else { 4794 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4795 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4796 } 4797 } 4798 4799 { 4800 AutoMutex lock(mLock); 4801 if (mActiveTrack != 0) { 4802 if (recordTrack != mActiveTrack.get()) { 4803 status = -EBUSY; 4804 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4805 mActiveTrack->mState = TrackBase::ACTIVE; 4806 } 4807 return status; 4808 } 4809 4810 recordTrack->mState = TrackBase::IDLE; 4811 mActiveTrack = recordTrack; 4812 mLock.unlock(); 4813 status_t status = AudioSystem::startInput(mId); 4814 mLock.lock(); 4815 if (status != NO_ERROR) { 4816 mActiveTrack.clear(); 4817 clearSyncStartEvent(); 4818 return status; 4819 } 4820 mRsmpInIndex = mFrameCount; 4821 mBytesRead = 0; 4822 if (mResampler != NULL) { 4823 mResampler->reset(); 4824 } 4825 mActiveTrack->mState = TrackBase::RESUMING; 4826 // signal thread to start 4827 ALOGV("Signal record thread"); 4828 mWaitWorkCV.broadcast(); 4829 // do not wait for mStartStopCond if exiting 4830 if (exitPending()) { 4831 mActiveTrack.clear(); 4832 status = INVALID_OPERATION; 4833 goto startError; 4834 } 4835 mStartStopCond.wait(mLock); 4836 if (mActiveTrack == 0) { 4837 ALOGV("Record failed to start"); 4838 status = BAD_VALUE; 4839 goto startError; 4840 } 4841 ALOGV("Record started OK"); 4842 return status; 4843 } 4844 4845startError: 4846 AudioSystem::stopInput(mId); 4847 clearSyncStartEvent(); 4848 return status; 4849} 4850 4851void AudioFlinger::RecordThread::clearSyncStartEvent() 4852{ 4853 if (mSyncStartEvent != 0) { 4854 mSyncStartEvent->cancel(); 4855 } 4856 mSyncStartEvent.clear(); 4857 mFramestoDrop = 0; 4858} 4859 4860void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4861{ 4862 sp<SyncEvent> strongEvent = event.promote(); 4863 4864 if (strongEvent != 0) { 4865 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4866 me->handleSyncStartEvent(strongEvent); 4867 } 4868} 4869 4870void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4871{ 4872 if (event == mSyncStartEvent) { 4873 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4874 // from audio HAL 4875 mFramestoDrop = mFrameCount * 2; 4876 } 4877} 4878 4879bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4880 ALOGV("RecordThread::stop"); 4881 AutoMutex _l(mLock); 4882 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4883 return false; 4884 } 4885 recordTrack->mState = TrackBase::PAUSING; 4886 // do not wait for mStartStopCond if exiting 4887 if (exitPending()) { 4888 return true; 4889 } 4890 mStartStopCond.wait(mLock); 4891 // if we have been restarted, recordTrack == mActiveTrack.get() here 4892 if (exitPending() || recordTrack != mActiveTrack.get()) { 4893 ALOGV("Record stopped OK"); 4894 return true; 4895 } 4896 return false; 4897} 4898 4899bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4900{ 4901 return false; 4902} 4903 4904status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4905{ 4906#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4907 if (!isValidSyncEvent(event)) { 4908 return BAD_VALUE; 4909 } 4910 4911 int eventSession = event->triggerSession(); 4912 status_t ret = NAME_NOT_FOUND; 4913 4914 Mutex::Autolock _l(mLock); 4915 4916 for (size_t i = 0; i < mTracks.size(); i++) { 4917 sp<RecordTrack> track = mTracks[i]; 4918 if (eventSession == track->sessionId()) { 4919 (void) track->setSyncEvent(event); 4920 ret = NO_ERROR; 4921 } 4922 } 4923 return ret; 4924#else 4925 return BAD_VALUE; 4926#endif 4927} 4928 4929// destroyTrack_l() must be called with ThreadBase::mLock held 4930void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4931{ 4932 track->terminate(); 4933 track->mState = TrackBase::STOPPED; 4934 // active tracks are removed by threadLoop() 4935 if (mActiveTrack != track) { 4936 removeTrack_l(track); 4937 } 4938} 4939 4940void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4941{ 4942 mTracks.remove(track); 4943 // need anything related to effects here? 4944} 4945 4946void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4947{ 4948 dumpInternals(fd, args); 4949 dumpTracks(fd, args); 4950 dumpEffectChains(fd, args); 4951} 4952 4953void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4954{ 4955 const size_t SIZE = 256; 4956 char buffer[SIZE]; 4957 String8 result; 4958 4959 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4960 result.append(buffer); 4961 4962 if (mActiveTrack != 0) { 4963 snprintf(buffer, SIZE, "In index: %zu\n", mRsmpInIndex); 4964 result.append(buffer); 4965 snprintf(buffer, SIZE, "Buffer size: %zu bytes\n", mBufferSize); 4966 result.append(buffer); 4967 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4968 result.append(buffer); 4969 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4970 result.append(buffer); 4971 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4972 result.append(buffer); 4973 } else { 4974 result.append("No active record client\n"); 4975 } 4976 4977 write(fd, result.string(), result.size()); 4978 4979 dumpBase(fd, args); 4980} 4981 4982void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4983{ 4984 const size_t SIZE = 256; 4985 char buffer[SIZE]; 4986 String8 result; 4987 4988 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4989 result.append(buffer); 4990 RecordTrack::appendDumpHeader(result); 4991 for (size_t i = 0; i < mTracks.size(); ++i) { 4992 sp<RecordTrack> track = mTracks[i]; 4993 if (track != 0) { 4994 track->dump(buffer, SIZE); 4995 result.append(buffer); 4996 } 4997 } 4998 4999 if (mActiveTrack != 0) { 5000 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 5001 result.append(buffer); 5002 RecordTrack::appendDumpHeader(result); 5003 mActiveTrack->dump(buffer, SIZE); 5004 result.append(buffer); 5005 5006 } 5007 write(fd, result.string(), result.size()); 5008} 5009 5010// AudioBufferProvider interface 5011status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5012{ 5013 size_t framesReq = buffer->frameCount; 5014 size_t framesReady = mFrameCount - mRsmpInIndex; 5015 int channelCount; 5016 5017 if (framesReady == 0) { 5018 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 5019 if (mBytesRead <= 0) { 5020 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 5021 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5022 // Force input into standby so that it tries to 5023 // recover at next read attempt 5024 inputStandBy(); 5025 usleep(kRecordThreadSleepUs); 5026 } 5027 buffer->raw = NULL; 5028 buffer->frameCount = 0; 5029 return NOT_ENOUGH_DATA; 5030 } 5031 mRsmpInIndex = 0; 5032 framesReady = mFrameCount; 5033 } 5034 5035 if (framesReq > framesReady) { 5036 framesReq = framesReady; 5037 } 5038 5039 if (mChannelCount == 1 && mReqChannelCount == 2) { 5040 channelCount = 1; 5041 } else { 5042 channelCount = 2; 5043 } 5044 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5045 buffer->frameCount = framesReq; 5046 return NO_ERROR; 5047} 5048 5049// AudioBufferProvider interface 5050void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5051{ 5052 mRsmpInIndex += buffer->frameCount; 5053 buffer->frameCount = 0; 5054} 5055 5056bool AudioFlinger::RecordThread::checkForNewParameters_l() 5057{ 5058 bool reconfig = false; 5059 5060 while (!mNewParameters.isEmpty()) { 5061 status_t status = NO_ERROR; 5062 String8 keyValuePair = mNewParameters[0]; 5063 AudioParameter param = AudioParameter(keyValuePair); 5064 int value; 5065 audio_format_t reqFormat = mFormat; 5066 uint32_t reqSamplingRate = mReqSampleRate; 5067 uint32_t reqChannelCount = mReqChannelCount; 5068 5069 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5070 reqSamplingRate = value; 5071 reconfig = true; 5072 } 5073 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5074 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5075 status = BAD_VALUE; 5076 } else { 5077 reqFormat = (audio_format_t) value; 5078 reconfig = true; 5079 } 5080 } 5081 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5082 reqChannelCount = popcount(value); 5083 reconfig = true; 5084 } 5085 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5086 // do not accept frame count changes if tracks are open as the track buffer 5087 // size depends on frame count and correct behavior would not be guaranteed 5088 // if frame count is changed after track creation 5089 if (mActiveTrack != 0) { 5090 status = INVALID_OPERATION; 5091 } else { 5092 reconfig = true; 5093 } 5094 } 5095 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5096 // forward device change to effects that have requested to be 5097 // aware of attached audio device. 5098 for (size_t i = 0; i < mEffectChains.size(); i++) { 5099 mEffectChains[i]->setDevice_l(value); 5100 } 5101 5102 // store input device and output device but do not forward output device to audio HAL. 5103 // Note that status is ignored by the caller for output device 5104 // (see AudioFlinger::setParameters() 5105 if (audio_is_output_devices(value)) { 5106 mOutDevice = value; 5107 status = BAD_VALUE; 5108 } else { 5109 mInDevice = value; 5110 // disable AEC and NS if the device is a BT SCO headset supporting those 5111 // pre processings 5112 if (mTracks.size() > 0) { 5113 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5114 mAudioFlinger->btNrecIsOff(); 5115 for (size_t i = 0; i < mTracks.size(); i++) { 5116 sp<RecordTrack> track = mTracks[i]; 5117 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5118 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5119 } 5120 } 5121 } 5122 } 5123 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5124 mAudioSource != (audio_source_t)value) { 5125 // forward device change to effects that have requested to be 5126 // aware of attached audio device. 5127 for (size_t i = 0; i < mEffectChains.size(); i++) { 5128 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5129 } 5130 mAudioSource = (audio_source_t)value; 5131 } 5132 if (status == NO_ERROR) { 5133 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5134 keyValuePair.string()); 5135 if (status == INVALID_OPERATION) { 5136 inputStandBy(); 5137 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5138 keyValuePair.string()); 5139 } 5140 if (reconfig) { 5141 if (status == BAD_VALUE && 5142 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5143 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5144 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5145 <= (2 * reqSamplingRate)) && 5146 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5147 <= FCC_2 && 5148 (reqChannelCount <= FCC_2)) { 5149 status = NO_ERROR; 5150 } 5151 if (status == NO_ERROR) { 5152 readInputParameters(); 5153 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5154 } 5155 } 5156 } 5157 5158 mNewParameters.removeAt(0); 5159 5160 mParamStatus = status; 5161 mParamCond.signal(); 5162 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5163 // already timed out waiting for the status and will never signal the condition. 5164 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5165 } 5166 return reconfig; 5167} 5168 5169String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5170{ 5171 Mutex::Autolock _l(mLock); 5172 if (initCheck() != NO_ERROR) { 5173 return String8(); 5174 } 5175 5176 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5177 const String8 out_s8(s); 5178 free(s); 5179 return out_s8; 5180} 5181 5182void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5183 AudioSystem::OutputDescriptor desc; 5184 void *param2 = NULL; 5185 5186 switch (event) { 5187 case AudioSystem::INPUT_OPENED: 5188 case AudioSystem::INPUT_CONFIG_CHANGED: 5189 desc.channelMask = mChannelMask; 5190 desc.samplingRate = mSampleRate; 5191 desc.format = mFormat; 5192 desc.frameCount = mFrameCount; 5193 desc.latency = 0; 5194 param2 = &desc; 5195 break; 5196 5197 case AudioSystem::INPUT_CLOSED: 5198 default: 5199 break; 5200 } 5201 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5202} 5203 5204void AudioFlinger::RecordThread::readInputParameters() 5205{ 5206 delete[] mRsmpInBuffer; 5207 // mRsmpInBuffer is always assigned a new[] below 5208 delete[] mRsmpOutBuffer; 5209 mRsmpOutBuffer = NULL; 5210 delete mResampler; 5211 mResampler = NULL; 5212 5213 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5214 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5215 mChannelCount = popcount(mChannelMask); 5216 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5217 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5218 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5219 } 5220 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5221 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5222 mFrameCount = mBufferSize / mFrameSize; 5223 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5224 5225 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5226 { 5227 int channelCount; 5228 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5229 // stereo to mono post process as the resampler always outputs stereo. 5230 if (mChannelCount == 1 && mReqChannelCount == 2) { 5231 channelCount = 1; 5232 } else { 5233 channelCount = 2; 5234 } 5235 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5236 mResampler->setSampleRate(mSampleRate); 5237 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5238 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5239 5240 // optmization: if mono to mono, alter input frame count as if we were inputing 5241 // stereo samples 5242 if (mChannelCount == 1 && mReqChannelCount == 1) { 5243 mFrameCount >>= 1; 5244 } 5245 5246 } 5247 mRsmpInIndex = mFrameCount; 5248} 5249 5250unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5251{ 5252 Mutex::Autolock _l(mLock); 5253 if (initCheck() != NO_ERROR) { 5254 return 0; 5255 } 5256 5257 return mInput->stream->get_input_frames_lost(mInput->stream); 5258} 5259 5260uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5261{ 5262 Mutex::Autolock _l(mLock); 5263 uint32_t result = 0; 5264 if (getEffectChain_l(sessionId) != 0) { 5265 result = EFFECT_SESSION; 5266 } 5267 5268 for (size_t i = 0; i < mTracks.size(); ++i) { 5269 if (sessionId == mTracks[i]->sessionId()) { 5270 result |= TRACK_SESSION; 5271 break; 5272 } 5273 } 5274 5275 return result; 5276} 5277 5278KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5279{ 5280 KeyedVector<int, bool> ids; 5281 Mutex::Autolock _l(mLock); 5282 for (size_t j = 0; j < mTracks.size(); ++j) { 5283 sp<RecordThread::RecordTrack> track = mTracks[j]; 5284 int sessionId = track->sessionId(); 5285 if (ids.indexOfKey(sessionId) < 0) { 5286 ids.add(sessionId, true); 5287 } 5288 } 5289 return ids; 5290} 5291 5292AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5293{ 5294 Mutex::Autolock _l(mLock); 5295 AudioStreamIn *input = mInput; 5296 mInput = NULL; 5297 return input; 5298} 5299 5300// this method must always be called either with ThreadBase mLock held or inside the thread loop 5301audio_stream_t* AudioFlinger::RecordThread::stream() const 5302{ 5303 if (mInput == NULL) { 5304 return NULL; 5305 } 5306 return &mInput->stream->common; 5307} 5308 5309status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5310{ 5311 // only one chain per input thread 5312 if (mEffectChains.size() != 0) { 5313 return INVALID_OPERATION; 5314 } 5315 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5316 5317 chain->setInBuffer(NULL); 5318 chain->setOutBuffer(NULL); 5319 5320 checkSuspendOnAddEffectChain_l(chain); 5321 5322 mEffectChains.add(chain); 5323 5324 return NO_ERROR; 5325} 5326 5327size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5328{ 5329 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5330 ALOGW_IF(mEffectChains.size() != 1, 5331 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5332 chain.get(), mEffectChains.size(), this); 5333 if (mEffectChains.size() == 1) { 5334 mEffectChains.removeAt(0); 5335 } 5336 return 0; 5337} 5338 5339}; // namespace android 5340