Threads.cpp revision 26a4029c95620a2b98187cf003cd3c58eea03747
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// Whether to use fast mixer 113static const enum { 114 FastMixer_Never, // never initialize or use: for debugging only 115 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 116 // normal mixer multiplier is 1 117 FastMixer_Static, // initialize if needed, then use all the time if initialized, 118 // multiplier is calculated based on min & max normal mixer buffer size 119 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 // FIXME for FastMixer_Dynamic: 122 // Supporting this option will require fixing HALs that can't handle large writes. 123 // For example, one HAL implementation returns an error from a large write, 124 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 125 // We could either fix the HAL implementations, or provide a wrapper that breaks 126 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 127} kUseFastMixer = FastMixer_Static; 128 129// Priorities for requestPriority 130static const int kPriorityAudioApp = 2; 131static const int kPriorityFastMixer = 3; 132 133// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 134// for the track. The client then sub-divides this into smaller buffers for its use. 135// Currently the client uses double-buffering by default, but doesn't tell us about that. 136// So for now we just assume that client is double-buffered. 137// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 138// N-buffering, so AudioFlinger could allocate the right amount of memory. 139// See the client's minBufCount and mNotificationFramesAct calculations for details. 140static const int kFastTrackMultiplier = 1; 141 142// ---------------------------------------------------------------------------- 143 144#ifdef ADD_BATTERY_DATA 145// To collect the amplifier usage 146static void addBatteryData(uint32_t params) { 147 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 148 if (service == NULL) { 149 // it already logged 150 return; 151 } 152 153 service->addBatteryData(params); 154} 155#endif 156 157 158// ---------------------------------------------------------------------------- 159// CPU Stats 160// ---------------------------------------------------------------------------- 161 162class CpuStats { 163public: 164 CpuStats(); 165 void sample(const String8 &title); 166#ifdef DEBUG_CPU_USAGE 167private: 168 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 169 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 170 171 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 172 173 int mCpuNum; // thread's current CPU number 174 int mCpukHz; // frequency of thread's current CPU in kHz 175#endif 176}; 177 178CpuStats::CpuStats() 179#ifdef DEBUG_CPU_USAGE 180 : mCpuNum(-1), mCpukHz(-1) 181#endif 182{ 183} 184 185void CpuStats::sample(const String8 &title) { 186#ifdef DEBUG_CPU_USAGE 187 // get current thread's delta CPU time in wall clock ns 188 double wcNs; 189 bool valid = mCpuUsage.sampleAndEnable(wcNs); 190 191 // record sample for wall clock statistics 192 if (valid) { 193 mWcStats.sample(wcNs); 194 } 195 196 // get the current CPU number 197 int cpuNum = sched_getcpu(); 198 199 // get the current CPU frequency in kHz 200 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 201 202 // check if either CPU number or frequency changed 203 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 204 mCpuNum = cpuNum; 205 mCpukHz = cpukHz; 206 // ignore sample for purposes of cycles 207 valid = false; 208 } 209 210 // if no change in CPU number or frequency, then record sample for cycle statistics 211 if (valid && mCpukHz > 0) { 212 double cycles = wcNs * cpukHz * 0.000001; 213 mHzStats.sample(cycles); 214 } 215 216 unsigned n = mWcStats.n(); 217 // mCpuUsage.elapsed() is expensive, so don't call it every loop 218 if ((n & 127) == 1) { 219 long long elapsed = mCpuUsage.elapsed(); 220 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 221 double perLoop = elapsed / (double) n; 222 double perLoop100 = perLoop * 0.01; 223 double perLoop1k = perLoop * 0.001; 224 double mean = mWcStats.mean(); 225 double stddev = mWcStats.stddev(); 226 double minimum = mWcStats.minimum(); 227 double maximum = mWcStats.maximum(); 228 double meanCycles = mHzStats.mean(); 229 double stddevCycles = mHzStats.stddev(); 230 double minCycles = mHzStats.minimum(); 231 double maxCycles = mHzStats.maximum(); 232 mCpuUsage.resetElapsed(); 233 mWcStats.reset(); 234 mHzStats.reset(); 235 ALOGD("CPU usage for %s over past %.1f secs\n" 236 " (%u mixer loops at %.1f mean ms per loop):\n" 237 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 238 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 239 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 240 title.string(), 241 elapsed * .000000001, n, perLoop * .000001, 242 mean * .001, 243 stddev * .001, 244 minimum * .001, 245 maximum * .001, 246 mean / perLoop100, 247 stddev / perLoop100, 248 minimum / perLoop100, 249 maximum / perLoop100, 250 meanCycles / perLoop1k, 251 stddevCycles / perLoop1k, 252 minCycles / perLoop1k, 253 maxCycles / perLoop1k); 254 255 } 256 } 257#endif 258}; 259 260// ---------------------------------------------------------------------------- 261// ThreadBase 262// ---------------------------------------------------------------------------- 263 264AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 265 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 266 : Thread(false /*canCallJava*/), 267 mType(type), 268 mAudioFlinger(audioFlinger), 269 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 270 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 271 mParamStatus(NO_ERROR), 272 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 273 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 274 // mName will be set by concrete (non-virtual) subclass 275 mDeathRecipient(new PMDeathRecipient(this)) 276{ 277} 278 279AudioFlinger::ThreadBase::~ThreadBase() 280{ 281 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 282 for (size_t i = 0; i < mConfigEvents.size(); i++) { 283 delete mConfigEvents[i]; 284 } 285 mConfigEvents.clear(); 286 287 mParamCond.broadcast(); 288 // do not lock the mutex in destructor 289 releaseWakeLock_l(); 290 if (mPowerManager != 0) { 291 sp<IBinder> binder = mPowerManager->asBinder(); 292 binder->unlinkToDeath(mDeathRecipient); 293 } 294} 295 296status_t AudioFlinger::ThreadBase::readyToRun() 297{ 298 status_t status = initCheck(); 299 if (status == NO_ERROR) { 300 ALOGI("AudioFlinger's thread %p ready to run", this); 301 } else { 302 ALOGE("No working audio driver found."); 303 } 304 return status; 305} 306 307void AudioFlinger::ThreadBase::exit() 308{ 309 ALOGV("ThreadBase::exit"); 310 // do any cleanup required for exit to succeed 311 preExit(); 312 { 313 // This lock prevents the following race in thread (uniprocessor for illustration): 314 // if (!exitPending()) { 315 // // context switch from here to exit() 316 // // exit() calls requestExit(), what exitPending() observes 317 // // exit() calls signal(), which is dropped since no waiters 318 // // context switch back from exit() to here 319 // mWaitWorkCV.wait(...); 320 // // now thread is hung 321 // } 322 AutoMutex lock(mLock); 323 requestExit(); 324 mWaitWorkCV.broadcast(); 325 } 326 // When Thread::requestExitAndWait is made virtual and this method is renamed to 327 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 328 requestExitAndWait(); 329} 330 331status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 332{ 333 status_t status; 334 335 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 336 Mutex::Autolock _l(mLock); 337 338 mNewParameters.add(keyValuePairs); 339 mWaitWorkCV.signal(); 340 // wait condition with timeout in case the thread loop has exited 341 // before the request could be processed 342 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 343 status = mParamStatus; 344 mWaitWorkCV.signal(); 345 } else { 346 status = TIMED_OUT; 347 } 348 return status; 349} 350 351void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 352{ 353 Mutex::Autolock _l(mLock); 354 sendIoConfigEvent_l(event, param); 355} 356 357// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 358void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 359{ 360 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 361 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 362 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 363 param); 364 mWaitWorkCV.signal(); 365} 366 367// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 368void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 369{ 370 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 371 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 372 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 373 mConfigEvents.size(), pid, tid, prio); 374 mWaitWorkCV.signal(); 375} 376 377void AudioFlinger::ThreadBase::processConfigEvents() 378{ 379 Mutex::Autolock _l(mLock); 380 processConfigEvents_l(); 381} 382 383void AudioFlinger::ThreadBase::processConfigEvents_l() 384{ 385 while (!mConfigEvents.isEmpty()) { 386 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 387 ConfigEvent *event = mConfigEvents[0]; 388 mConfigEvents.removeAt(0); 389 // release mLock before locking AudioFlinger mLock: lock order is always 390 // AudioFlinger then ThreadBase to avoid cross deadlock 391 mLock.unlock(); 392 switch (event->type()) { 393 case CFG_EVENT_PRIO: { 394 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 395 // FIXME Need to understand why this has be done asynchronously 396 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 397 true /*asynchronous*/); 398 if (err != 0) { 399 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 400 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 401 } 402 } break; 403 case CFG_EVENT_IO: { 404 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 405 { 406 Mutex::Autolock _l(mAudioFlinger->mLock); 407 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 408 } 409 } break; 410 default: 411 ALOGE("processConfigEvents() unknown event type %d", event->type()); 412 break; 413 } 414 delete event; 415 mLock.lock(); 416 } 417} 418 419void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 420{ 421 const size_t SIZE = 256; 422 char buffer[SIZE]; 423 String8 result; 424 425 bool locked = AudioFlinger::dumpTryLock(mLock); 426 if (!locked) { 427 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 428 write(fd, buffer, strlen(buffer)); 429 } 430 431 snprintf(buffer, SIZE, "io handle: %d\n", mId); 432 result.append(buffer); 433 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 434 result.append(buffer); 435 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 436 result.append(buffer); 437 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 438 result.append(buffer); 439 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 440 result.append(buffer); 441 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 442 result.append(buffer); 443 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 444 result.append(buffer); 445 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 446 result.append(buffer); 447 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 448 result.append(buffer); 449 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 450 result.append(buffer); 451 452 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 453 result.append(buffer); 454 result.append(" Index Command"); 455 for (size_t i = 0; i < mNewParameters.size(); ++i) { 456 snprintf(buffer, SIZE, "\n %02d ", i); 457 result.append(buffer); 458 result.append(mNewParameters[i]); 459 } 460 461 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 462 result.append(buffer); 463 for (size_t i = 0; i < mConfigEvents.size(); i++) { 464 mConfigEvents[i]->dump(buffer, SIZE); 465 result.append(buffer); 466 } 467 result.append("\n"); 468 469 write(fd, result.string(), result.size()); 470 471 if (locked) { 472 mLock.unlock(); 473 } 474} 475 476void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 477{ 478 const size_t SIZE = 256; 479 char buffer[SIZE]; 480 String8 result; 481 482 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 483 write(fd, buffer, strlen(buffer)); 484 485 for (size_t i = 0; i < mEffectChains.size(); ++i) { 486 sp<EffectChain> chain = mEffectChains[i]; 487 if (chain != 0) { 488 chain->dump(fd, args); 489 } 490 } 491} 492 493void AudioFlinger::ThreadBase::acquireWakeLock() 494{ 495 Mutex::Autolock _l(mLock); 496 acquireWakeLock_l(); 497} 498 499void AudioFlinger::ThreadBase::acquireWakeLock_l() 500{ 501 if (mPowerManager == 0) { 502 // use checkService() to avoid blocking if power service is not up yet 503 sp<IBinder> binder = 504 defaultServiceManager()->checkService(String16("power")); 505 if (binder == 0) { 506 ALOGW("Thread %s cannot connect to the power manager service", mName); 507 } else { 508 mPowerManager = interface_cast<IPowerManager>(binder); 509 binder->linkToDeath(mDeathRecipient); 510 } 511 } 512 if (mPowerManager != 0) { 513 sp<IBinder> binder = new BBinder(); 514 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 515 binder, 516 String16(mName), 517 String16("media")); 518 if (status == NO_ERROR) { 519 mWakeLockToken = binder; 520 } 521 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 522 } 523} 524 525void AudioFlinger::ThreadBase::releaseWakeLock() 526{ 527 Mutex::Autolock _l(mLock); 528 releaseWakeLock_l(); 529} 530 531void AudioFlinger::ThreadBase::releaseWakeLock_l() 532{ 533 if (mWakeLockToken != 0) { 534 ALOGV("releaseWakeLock_l() %s", mName); 535 if (mPowerManager != 0) { 536 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 537 } 538 mWakeLockToken.clear(); 539 } 540} 541 542void AudioFlinger::ThreadBase::clearPowerManager() 543{ 544 Mutex::Autolock _l(mLock); 545 releaseWakeLock_l(); 546 mPowerManager.clear(); 547} 548 549void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 550{ 551 sp<ThreadBase> thread = mThread.promote(); 552 if (thread != 0) { 553 thread->clearPowerManager(); 554 } 555 ALOGW("power manager service died !!!"); 556} 557 558void AudioFlinger::ThreadBase::setEffectSuspended( 559 const effect_uuid_t *type, bool suspend, int sessionId) 560{ 561 Mutex::Autolock _l(mLock); 562 setEffectSuspended_l(type, suspend, sessionId); 563} 564 565void AudioFlinger::ThreadBase::setEffectSuspended_l( 566 const effect_uuid_t *type, bool suspend, int sessionId) 567{ 568 sp<EffectChain> chain = getEffectChain_l(sessionId); 569 if (chain != 0) { 570 if (type != NULL) { 571 chain->setEffectSuspended_l(type, suspend); 572 } else { 573 chain->setEffectSuspendedAll_l(suspend); 574 } 575 } 576 577 updateSuspendedSessions_l(type, suspend, sessionId); 578} 579 580void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 581{ 582 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 583 if (index < 0) { 584 return; 585 } 586 587 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 588 mSuspendedSessions.valueAt(index); 589 590 for (size_t i = 0; i < sessionEffects.size(); i++) { 591 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 592 for (int j = 0; j < desc->mRefCount; j++) { 593 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 594 chain->setEffectSuspendedAll_l(true); 595 } else { 596 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 597 desc->mType.timeLow); 598 chain->setEffectSuspended_l(&desc->mType, true); 599 } 600 } 601 } 602} 603 604void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 605 bool suspend, 606 int sessionId) 607{ 608 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 609 610 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 611 612 if (suspend) { 613 if (index >= 0) { 614 sessionEffects = mSuspendedSessions.valueAt(index); 615 } else { 616 mSuspendedSessions.add(sessionId, sessionEffects); 617 } 618 } else { 619 if (index < 0) { 620 return; 621 } 622 sessionEffects = mSuspendedSessions.valueAt(index); 623 } 624 625 626 int key = EffectChain::kKeyForSuspendAll; 627 if (type != NULL) { 628 key = type->timeLow; 629 } 630 index = sessionEffects.indexOfKey(key); 631 632 sp<SuspendedSessionDesc> desc; 633 if (suspend) { 634 if (index >= 0) { 635 desc = sessionEffects.valueAt(index); 636 } else { 637 desc = new SuspendedSessionDesc(); 638 if (type != NULL) { 639 desc->mType = *type; 640 } 641 sessionEffects.add(key, desc); 642 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 643 } 644 desc->mRefCount++; 645 } else { 646 if (index < 0) { 647 return; 648 } 649 desc = sessionEffects.valueAt(index); 650 if (--desc->mRefCount == 0) { 651 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 652 sessionEffects.removeItemsAt(index); 653 if (sessionEffects.isEmpty()) { 654 ALOGV("updateSuspendedSessions_l() restore removing session %d", 655 sessionId); 656 mSuspendedSessions.removeItem(sessionId); 657 } 658 } 659 } 660 if (!sessionEffects.isEmpty()) { 661 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 662 } 663} 664 665void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 666 bool enabled, 667 int sessionId) 668{ 669 Mutex::Autolock _l(mLock); 670 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 671} 672 673void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 674 bool enabled, 675 int sessionId) 676{ 677 if (mType != RECORD) { 678 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 679 // another session. This gives the priority to well behaved effect control panels 680 // and applications not using global effects. 681 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 682 // global effects 683 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 684 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 685 } 686 } 687 688 sp<EffectChain> chain = getEffectChain_l(sessionId); 689 if (chain != 0) { 690 chain->checkSuspendOnEffectEnabled(effect, enabled); 691 } 692} 693 694// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 695sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 696 const sp<AudioFlinger::Client>& client, 697 const sp<IEffectClient>& effectClient, 698 int32_t priority, 699 int sessionId, 700 effect_descriptor_t *desc, 701 int *enabled, 702 status_t *status) 703{ 704 sp<EffectModule> effect; 705 sp<EffectHandle> handle; 706 status_t lStatus; 707 sp<EffectChain> chain; 708 bool chainCreated = false; 709 bool effectCreated = false; 710 bool effectRegistered = false; 711 712 lStatus = initCheck(); 713 if (lStatus != NO_ERROR) { 714 ALOGW("createEffect_l() Audio driver not initialized."); 715 goto Exit; 716 } 717 718 // Do not allow effects with session ID 0 on direct output or duplicating threads 719 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 720 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 721 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 722 desc->name, sessionId); 723 lStatus = BAD_VALUE; 724 goto Exit; 725 } 726 // Only Pre processor effects are allowed on input threads and only on input threads 727 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 728 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 729 desc->name, desc->flags, mType); 730 lStatus = BAD_VALUE; 731 goto Exit; 732 } 733 734 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 735 736 { // scope for mLock 737 Mutex::Autolock _l(mLock); 738 739 // check for existing effect chain with the requested audio session 740 chain = getEffectChain_l(sessionId); 741 if (chain == 0) { 742 // create a new chain for this session 743 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 744 chain = new EffectChain(this, sessionId); 745 addEffectChain_l(chain); 746 chain->setStrategy(getStrategyForSession_l(sessionId)); 747 chainCreated = true; 748 } else { 749 effect = chain->getEffectFromDesc_l(desc); 750 } 751 752 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 753 754 if (effect == 0) { 755 int id = mAudioFlinger->nextUniqueId(); 756 // Check CPU and memory usage 757 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 758 if (lStatus != NO_ERROR) { 759 goto Exit; 760 } 761 effectRegistered = true; 762 // create a new effect module if none present in the chain 763 effect = new EffectModule(this, chain, desc, id, sessionId); 764 lStatus = effect->status(); 765 if (lStatus != NO_ERROR) { 766 goto Exit; 767 } 768 lStatus = chain->addEffect_l(effect); 769 if (lStatus != NO_ERROR) { 770 goto Exit; 771 } 772 effectCreated = true; 773 774 effect->setDevice(mOutDevice); 775 effect->setDevice(mInDevice); 776 effect->setMode(mAudioFlinger->getMode()); 777 effect->setAudioSource(mAudioSource); 778 } 779 // create effect handle and connect it to effect module 780 handle = new EffectHandle(effect, client, effectClient, priority); 781 lStatus = effect->addHandle(handle.get()); 782 if (enabled != NULL) { 783 *enabled = (int)effect->isEnabled(); 784 } 785 } 786 787Exit: 788 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 789 Mutex::Autolock _l(mLock); 790 if (effectCreated) { 791 chain->removeEffect_l(effect); 792 } 793 if (effectRegistered) { 794 AudioSystem::unregisterEffect(effect->id()); 795 } 796 if (chainCreated) { 797 removeEffectChain_l(chain); 798 } 799 handle.clear(); 800 } 801 802 *status = lStatus; 803 return handle; 804} 805 806sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 807{ 808 Mutex::Autolock _l(mLock); 809 return getEffect_l(sessionId, effectId); 810} 811 812sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 813{ 814 sp<EffectChain> chain = getEffectChain_l(sessionId); 815 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 816} 817 818// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 819// PlaybackThread::mLock held 820status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 821{ 822 // check for existing effect chain with the requested audio session 823 int sessionId = effect->sessionId(); 824 sp<EffectChain> chain = getEffectChain_l(sessionId); 825 bool chainCreated = false; 826 827 if (chain == 0) { 828 // create a new chain for this session 829 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 830 chain = new EffectChain(this, sessionId); 831 addEffectChain_l(chain); 832 chain->setStrategy(getStrategyForSession_l(sessionId)); 833 chainCreated = true; 834 } 835 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 836 837 if (chain->getEffectFromId_l(effect->id()) != 0) { 838 ALOGW("addEffect_l() %p effect %s already present in chain %p", 839 this, effect->desc().name, chain.get()); 840 return BAD_VALUE; 841 } 842 843 status_t status = chain->addEffect_l(effect); 844 if (status != NO_ERROR) { 845 if (chainCreated) { 846 removeEffectChain_l(chain); 847 } 848 return status; 849 } 850 851 effect->setDevice(mOutDevice); 852 effect->setDevice(mInDevice); 853 effect->setMode(mAudioFlinger->getMode()); 854 effect->setAudioSource(mAudioSource); 855 return NO_ERROR; 856} 857 858void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 859 860 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 861 effect_descriptor_t desc = effect->desc(); 862 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 863 detachAuxEffect_l(effect->id()); 864 } 865 866 sp<EffectChain> chain = effect->chain().promote(); 867 if (chain != 0) { 868 // remove effect chain if removing last effect 869 if (chain->removeEffect_l(effect) == 0) { 870 removeEffectChain_l(chain); 871 } 872 } else { 873 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 874 } 875} 876 877void AudioFlinger::ThreadBase::lockEffectChains_l( 878 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 879{ 880 effectChains = mEffectChains; 881 for (size_t i = 0; i < mEffectChains.size(); i++) { 882 mEffectChains[i]->lock(); 883 } 884} 885 886void AudioFlinger::ThreadBase::unlockEffectChains( 887 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 888{ 889 for (size_t i = 0; i < effectChains.size(); i++) { 890 effectChains[i]->unlock(); 891 } 892} 893 894sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 895{ 896 Mutex::Autolock _l(mLock); 897 return getEffectChain_l(sessionId); 898} 899 900sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 901{ 902 size_t size = mEffectChains.size(); 903 for (size_t i = 0; i < size; i++) { 904 if (mEffectChains[i]->sessionId() == sessionId) { 905 return mEffectChains[i]; 906 } 907 } 908 return 0; 909} 910 911void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 912{ 913 Mutex::Autolock _l(mLock); 914 size_t size = mEffectChains.size(); 915 for (size_t i = 0; i < size; i++) { 916 mEffectChains[i]->setMode_l(mode); 917 } 918} 919 920void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 921 EffectHandle *handle, 922 bool unpinIfLast) { 923 924 Mutex::Autolock _l(mLock); 925 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 926 // delete the effect module if removing last handle on it 927 if (effect->removeHandle(handle) == 0) { 928 if (!effect->isPinned() || unpinIfLast) { 929 removeEffect_l(effect); 930 AudioSystem::unregisterEffect(effect->id()); 931 } 932 } 933} 934 935// ---------------------------------------------------------------------------- 936// Playback 937// ---------------------------------------------------------------------------- 938 939AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 940 AudioStreamOut* output, 941 audio_io_handle_t id, 942 audio_devices_t device, 943 type_t type) 944 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 945 mNormalFrameCount(0), mMixBuffer(NULL), 946 mSuspended(0), mBytesWritten(0), 947 // mStreamTypes[] initialized in constructor body 948 mOutput(output), 949 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 950 mMixerStatus(MIXER_IDLE), 951 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 952 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 953 mBytesRemaining(0), 954 mCurrentWriteLength(0), 955 mUseAsyncWrite(false), 956 mWriteBlocked(false), 957 mDraining(false), 958 mScreenState(AudioFlinger::mScreenState), 959 // index 0 is reserved for normal mixer's submix 960 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 961{ 962 snprintf(mName, kNameLength, "AudioOut_%X", id); 963 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 964 965 // Assumes constructor is called by AudioFlinger with it's mLock held, but 966 // it would be safer to explicitly pass initial masterVolume/masterMute as 967 // parameter. 968 // 969 // If the HAL we are using has support for master volume or master mute, 970 // then do not attenuate or mute during mixing (just leave the volume at 1.0 971 // and the mute set to false). 972 mMasterVolume = audioFlinger->masterVolume_l(); 973 mMasterMute = audioFlinger->masterMute_l(); 974 if (mOutput && mOutput->audioHwDev) { 975 if (mOutput->audioHwDev->canSetMasterVolume()) { 976 mMasterVolume = 1.0; 977 } 978 979 if (mOutput->audioHwDev->canSetMasterMute()) { 980 mMasterMute = false; 981 } 982 } 983 984 readOutputParameters(); 985 986 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 987 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 988 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 989 stream = (audio_stream_type_t) (stream + 1)) { 990 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 991 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 992 } 993 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 994 // because mAudioFlinger doesn't have one to copy from 995} 996 997AudioFlinger::PlaybackThread::~PlaybackThread() 998{ 999 mAudioFlinger->unregisterWriter(mNBLogWriter); 1000 delete[] mMixBuffer; 1001} 1002 1003void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1004{ 1005 dumpInternals(fd, args); 1006 dumpTracks(fd, args); 1007 dumpEffectChains(fd, args); 1008} 1009 1010void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1011{ 1012 const size_t SIZE = 256; 1013 char buffer[SIZE]; 1014 String8 result; 1015 1016 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1017 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1018 const stream_type_t *st = &mStreamTypes[i]; 1019 if (i > 0) { 1020 result.appendFormat(", "); 1021 } 1022 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1023 if (st->mute) { 1024 result.append("M"); 1025 } 1026 } 1027 result.append("\n"); 1028 write(fd, result.string(), result.length()); 1029 result.clear(); 1030 1031 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1032 result.append(buffer); 1033 Track::appendDumpHeader(result); 1034 for (size_t i = 0; i < mTracks.size(); ++i) { 1035 sp<Track> track = mTracks[i]; 1036 if (track != 0) { 1037 track->dump(buffer, SIZE); 1038 result.append(buffer); 1039 } 1040 } 1041 1042 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1043 result.append(buffer); 1044 Track::appendDumpHeader(result); 1045 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1046 sp<Track> track = mActiveTracks[i].promote(); 1047 if (track != 0) { 1048 track->dump(buffer, SIZE); 1049 result.append(buffer); 1050 } 1051 } 1052 write(fd, result.string(), result.size()); 1053 1054 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1055 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1056 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1057 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1058} 1059 1060void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1061{ 1062 const size_t SIZE = 256; 1063 char buffer[SIZE]; 1064 String8 result; 1065 1066 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1067 result.append(buffer); 1068 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1069 result.append(buffer); 1070 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1071 ns2ms(systemTime() - mLastWriteTime)); 1072 result.append(buffer); 1073 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1074 result.append(buffer); 1075 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1076 result.append(buffer); 1077 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1078 result.append(buffer); 1079 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1080 result.append(buffer); 1081 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1082 result.append(buffer); 1083 write(fd, result.string(), result.size()); 1084 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1085 1086 dumpBase(fd, args); 1087} 1088 1089// Thread virtuals 1090 1091void AudioFlinger::PlaybackThread::onFirstRef() 1092{ 1093 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1094} 1095 1096// ThreadBase virtuals 1097void AudioFlinger::PlaybackThread::preExit() 1098{ 1099 ALOGV(" preExit()"); 1100 // FIXME this is using hard-coded strings but in the future, this functionality will be 1101 // converted to use audio HAL extensions required to support tunneling 1102 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1103} 1104 1105// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1106sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1107 const sp<AudioFlinger::Client>& client, 1108 audio_stream_type_t streamType, 1109 uint32_t sampleRate, 1110 audio_format_t format, 1111 audio_channel_mask_t channelMask, 1112 size_t frameCount, 1113 const sp<IMemory>& sharedBuffer, 1114 int sessionId, 1115 IAudioFlinger::track_flags_t *flags, 1116 pid_t tid, 1117 status_t *status) 1118{ 1119 sp<Track> track; 1120 status_t lStatus; 1121 1122 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1123 1124 // client expresses a preference for FAST, but we get the final say 1125 if (*flags & IAudioFlinger::TRACK_FAST) { 1126 if ( 1127 // not timed 1128 (!isTimed) && 1129 // either of these use cases: 1130 ( 1131 // use case 1: shared buffer with any frame count 1132 ( 1133 (sharedBuffer != 0) 1134 ) || 1135 // use case 2: callback handler and frame count is default or at least as large as HAL 1136 ( 1137 (tid != -1) && 1138 ((frameCount == 0) || 1139 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1140 ) 1141 ) && 1142 // PCM data 1143 audio_is_linear_pcm(format) && 1144 // mono or stereo 1145 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1146 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1147#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1148 // hardware sample rate 1149 (sampleRate == mSampleRate) && 1150#endif 1151 // normal mixer has an associated fast mixer 1152 hasFastMixer() && 1153 // there are sufficient fast track slots available 1154 (mFastTrackAvailMask != 0) 1155 // FIXME test that MixerThread for this fast track has a capable output HAL 1156 // FIXME add a permission test also? 1157 ) { 1158 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1159 if (frameCount == 0) { 1160 frameCount = mFrameCount * kFastTrackMultiplier; 1161 } 1162 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1163 frameCount, mFrameCount); 1164 } else { 1165 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1166 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1167 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1168 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1169 audio_is_linear_pcm(format), 1170 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1171 *flags &= ~IAudioFlinger::TRACK_FAST; 1172 // For compatibility with AudioTrack calculation, buffer depth is forced 1173 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1174 // This is probably too conservative, but legacy application code may depend on it. 1175 // If you change this calculation, also review the start threshold which is related. 1176 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1177 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1178 if (minBufCount < 2) { 1179 minBufCount = 2; 1180 } 1181 size_t minFrameCount = mNormalFrameCount * minBufCount; 1182 if (frameCount < minFrameCount) { 1183 frameCount = minFrameCount; 1184 } 1185 } 1186 } 1187 1188 if (mType == DIRECT) { 1189 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1190 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1191 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1192 "for output %p with format %d", 1193 sampleRate, format, channelMask, mOutput, mFormat); 1194 lStatus = BAD_VALUE; 1195 goto Exit; 1196 } 1197 } 1198 } else if (mType == OFFLOAD) { 1199 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1200 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1201 "for output %p with format %d", 1202 sampleRate, format, channelMask, mOutput, mFormat); 1203 lStatus = BAD_VALUE; 1204 goto Exit; 1205 } 1206 } else { 1207 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1208 ALOGE("createTrack_l() Bad parameter: format %d \"" 1209 "for output %p with format %d", 1210 format, mOutput, mFormat); 1211 lStatus = BAD_VALUE; 1212 goto Exit; 1213 } 1214 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1215 if (sampleRate > mSampleRate*2) { 1216 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1217 lStatus = BAD_VALUE; 1218 goto Exit; 1219 } 1220 } 1221 1222 lStatus = initCheck(); 1223 if (lStatus != NO_ERROR) { 1224 ALOGE("Audio driver not initialized."); 1225 goto Exit; 1226 } 1227 1228 { // scope for mLock 1229 Mutex::Autolock _l(mLock); 1230 1231 // all tracks in same audio session must share the same routing strategy otherwise 1232 // conflicts will happen when tracks are moved from one output to another by audio policy 1233 // manager 1234 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1235 for (size_t i = 0; i < mTracks.size(); ++i) { 1236 sp<Track> t = mTracks[i]; 1237 if (t != 0 && !t->isOutputTrack()) { 1238 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1239 if (sessionId == t->sessionId() && strategy != actual) { 1240 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1241 strategy, actual); 1242 lStatus = BAD_VALUE; 1243 goto Exit; 1244 } 1245 } 1246 } 1247 1248 if (!isTimed) { 1249 track = new Track(this, client, streamType, sampleRate, format, 1250 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1251 } else { 1252 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1253 channelMask, frameCount, sharedBuffer, sessionId); 1254 } 1255 1256 // new Track always returns non-NULL, 1257 // but TimedTrack::create() is a factory that could fail by returning NULL 1258 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1259 if (lStatus != NO_ERROR) { 1260 track.clear(); 1261 goto Exit; 1262 } 1263 1264 mTracks.add(track); 1265 1266 sp<EffectChain> chain = getEffectChain_l(sessionId); 1267 if (chain != 0) { 1268 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1269 track->setMainBuffer(chain->inBuffer()); 1270 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1271 chain->incTrackCnt(); 1272 } 1273 1274 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1275 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1276 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1277 // so ask activity manager to do this on our behalf 1278 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1279 } 1280 } 1281 1282 lStatus = NO_ERROR; 1283 1284Exit: 1285 *status = lStatus; 1286 return track; 1287} 1288 1289uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1290{ 1291 return latency; 1292} 1293 1294uint32_t AudioFlinger::PlaybackThread::latency() const 1295{ 1296 Mutex::Autolock _l(mLock); 1297 return latency_l(); 1298} 1299uint32_t AudioFlinger::PlaybackThread::latency_l() const 1300{ 1301 if (initCheck() == NO_ERROR) { 1302 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1303 } else { 1304 return 0; 1305 } 1306} 1307 1308void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1309{ 1310 Mutex::Autolock _l(mLock); 1311 // Don't apply master volume in SW if our HAL can do it for us. 1312 if (mOutput && mOutput->audioHwDev && 1313 mOutput->audioHwDev->canSetMasterVolume()) { 1314 mMasterVolume = 1.0; 1315 } else { 1316 mMasterVolume = value; 1317 } 1318} 1319 1320void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1321{ 1322 Mutex::Autolock _l(mLock); 1323 // Don't apply master mute in SW if our HAL can do it for us. 1324 if (mOutput && mOutput->audioHwDev && 1325 mOutput->audioHwDev->canSetMasterMute()) { 1326 mMasterMute = false; 1327 } else { 1328 mMasterMute = muted; 1329 } 1330} 1331 1332void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1333{ 1334 Mutex::Autolock _l(mLock); 1335 mStreamTypes[stream].volume = value; 1336 signal_l(); 1337} 1338 1339void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1340{ 1341 Mutex::Autolock _l(mLock); 1342 mStreamTypes[stream].mute = muted; 1343 signal_l(); 1344} 1345 1346float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1347{ 1348 Mutex::Autolock _l(mLock); 1349 return mStreamTypes[stream].volume; 1350} 1351 1352// addTrack_l() must be called with ThreadBase::mLock held 1353status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1354{ 1355 status_t status = ALREADY_EXISTS; 1356 1357 // set retry count for buffer fill 1358 track->mRetryCount = kMaxTrackStartupRetries; 1359 if (mActiveTracks.indexOf(track) < 0) { 1360 // the track is newly added, make sure it fills up all its 1361 // buffers before playing. This is to ensure the client will 1362 // effectively get the latency it requested. 1363 if (!track->isOutputTrack()) { 1364 TrackBase::track_state state = track->mState; 1365 mLock.unlock(); 1366 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1367 mLock.lock(); 1368 // abort track was stopped/paused while we released the lock 1369 if (state != track->mState) { 1370 if (status == NO_ERROR) { 1371 mLock.unlock(); 1372 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1373 mLock.lock(); 1374 } 1375 return INVALID_OPERATION; 1376 } 1377 // abort if start is rejected by audio policy manager 1378 if (status != NO_ERROR) { 1379 return PERMISSION_DENIED; 1380 } 1381#ifdef ADD_BATTERY_DATA 1382 // to track the speaker usage 1383 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1384#endif 1385 } 1386 1387 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1388 track->mResetDone = false; 1389 track->mPresentationCompleteFrames = 0; 1390 mActiveTracks.add(track); 1391 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1392 if (chain != 0) { 1393 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1394 track->sessionId()); 1395 chain->incActiveTrackCnt(); 1396 } 1397 1398 status = NO_ERROR; 1399 } 1400 1401 ALOGV("mWaitWorkCV.broadcast"); 1402 mWaitWorkCV.broadcast(); 1403 1404 return status; 1405} 1406 1407bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1408{ 1409 track->terminate(); 1410 // active tracks are removed by threadLoop() 1411 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1412 track->mState = TrackBase::STOPPED; 1413 if (!trackActive) { 1414 removeTrack_l(track); 1415 } else if (track->isFastTrack() || track->isOffloaded()) { 1416 track->mState = TrackBase::STOPPING_1; 1417 } 1418 1419 return trackActive; 1420} 1421 1422void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1423{ 1424 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1425 mTracks.remove(track); 1426 deleteTrackName_l(track->name()); 1427 // redundant as track is about to be destroyed, for dumpsys only 1428 track->mName = -1; 1429 if (track->isFastTrack()) { 1430 int index = track->mFastIndex; 1431 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1432 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1433 mFastTrackAvailMask |= 1 << index; 1434 // redundant as track is about to be destroyed, for dumpsys only 1435 track->mFastIndex = -1; 1436 } 1437 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1438 if (chain != 0) { 1439 chain->decTrackCnt(); 1440 } 1441} 1442 1443void AudioFlinger::PlaybackThread::signal_l() 1444{ 1445 // Thread could be blocked waiting for async 1446 // so signal it to handle state changes immediately 1447 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1448 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1449 mSignalPending = true; 1450 mWaitWorkCV.signal(); 1451} 1452 1453String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1454{ 1455 Mutex::Autolock _l(mLock); 1456 if (initCheck() != NO_ERROR) { 1457 return String8(); 1458 } 1459 1460 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1461 const String8 out_s8(s); 1462 free(s); 1463 return out_s8; 1464} 1465 1466// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1467void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1468 AudioSystem::OutputDescriptor desc; 1469 void *param2 = NULL; 1470 1471 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1472 param); 1473 1474 switch (event) { 1475 case AudioSystem::OUTPUT_OPENED: 1476 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1477 desc.channelMask = mChannelMask; 1478 desc.samplingRate = mSampleRate; 1479 desc.format = mFormat; 1480 desc.frameCount = mNormalFrameCount; // FIXME see 1481 // AudioFlinger::frameCount(audio_io_handle_t) 1482 desc.latency = latency(); 1483 param2 = &desc; 1484 break; 1485 1486 case AudioSystem::STREAM_CONFIG_CHANGED: 1487 param2 = ¶m; 1488 case AudioSystem::OUTPUT_CLOSED: 1489 default: 1490 break; 1491 } 1492 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1493} 1494 1495void AudioFlinger::PlaybackThread::writeCallback() 1496{ 1497 ALOG_ASSERT(mCallbackThread != 0); 1498 mCallbackThread->setWriteBlocked(false); 1499} 1500 1501void AudioFlinger::PlaybackThread::drainCallback() 1502{ 1503 ALOG_ASSERT(mCallbackThread != 0); 1504 mCallbackThread->setDraining(false); 1505} 1506 1507void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1508{ 1509 Mutex::Autolock _l(mLock); 1510 mWriteBlocked = value; 1511 if (!value) { 1512 mWaitWorkCV.signal(); 1513 } 1514} 1515 1516void AudioFlinger::PlaybackThread::setDraining(bool value) 1517{ 1518 Mutex::Autolock _l(mLock); 1519 mDraining = value; 1520 if (!value) { 1521 mWaitWorkCV.signal(); 1522 } 1523} 1524 1525// static 1526int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1527 void *param, 1528 void *cookie) 1529{ 1530 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1531 ALOGV("asyncCallback() event %d", event); 1532 switch (event) { 1533 case STREAM_CBK_EVENT_WRITE_READY: 1534 me->writeCallback(); 1535 break; 1536 case STREAM_CBK_EVENT_DRAIN_READY: 1537 me->drainCallback(); 1538 break; 1539 default: 1540 ALOGW("asyncCallback() unknown event %d", event); 1541 break; 1542 } 1543 return 0; 1544} 1545 1546void AudioFlinger::PlaybackThread::readOutputParameters() 1547{ 1548 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1549 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1550 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1551 if (!audio_is_output_channel(mChannelMask)) { 1552 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1553 } 1554 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1555 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1556 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1557 } 1558 mChannelCount = popcount(mChannelMask); 1559 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1560 if (!audio_is_valid_format(mFormat)) { 1561 LOG_FATAL("HAL format %d not valid for output", mFormat); 1562 } 1563 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1564 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1565 mFormat); 1566 } 1567 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1568 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1569 mFrameCount = mBufferSize / mFrameSize; 1570 if (mFrameCount & 15) { 1571 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1572 mFrameCount); 1573 } 1574 1575 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1576 (mOutput->stream->set_callback != NULL)) { 1577 if (mOutput->stream->set_callback(mOutput->stream, 1578 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1579 mUseAsyncWrite = true; 1580 } 1581 } 1582 1583 // Calculate size of normal mix buffer relative to the HAL output buffer size 1584 double multiplier = 1.0; 1585 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1586 kUseFastMixer == FastMixer_Dynamic)) { 1587 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1588 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1589 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1590 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1591 maxNormalFrameCount = maxNormalFrameCount & ~15; 1592 if (maxNormalFrameCount < minNormalFrameCount) { 1593 maxNormalFrameCount = minNormalFrameCount; 1594 } 1595 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1596 if (multiplier <= 1.0) { 1597 multiplier = 1.0; 1598 } else if (multiplier <= 2.0) { 1599 if (2 * mFrameCount <= maxNormalFrameCount) { 1600 multiplier = 2.0; 1601 } else { 1602 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1603 } 1604 } else { 1605 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1606 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1607 // track, but we sometimes have to do this to satisfy the maximum frame count 1608 // constraint) 1609 // FIXME this rounding up should not be done if no HAL SRC 1610 uint32_t truncMult = (uint32_t) multiplier; 1611 if ((truncMult & 1)) { 1612 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1613 ++truncMult; 1614 } 1615 } 1616 multiplier = (double) truncMult; 1617 } 1618 } 1619 mNormalFrameCount = multiplier * mFrameCount; 1620 // round up to nearest 16 frames to satisfy AudioMixer 1621 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1622 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1623 mNormalFrameCount); 1624 1625 delete[] mMixBuffer; 1626 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1627 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1628 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1629 memset(mMixBuffer, 0, normalBufferSize); 1630 1631 // force reconfiguration of effect chains and engines to take new buffer size and audio 1632 // parameters into account 1633 // Note that mLock is not held when readOutputParameters() is called from the constructor 1634 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1635 // matter. 1636 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1637 Vector< sp<EffectChain> > effectChains = mEffectChains; 1638 for (size_t i = 0; i < effectChains.size(); i ++) { 1639 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1640 } 1641} 1642 1643 1644status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1645{ 1646 if (halFrames == NULL || dspFrames == NULL) { 1647 return BAD_VALUE; 1648 } 1649 Mutex::Autolock _l(mLock); 1650 if (initCheck() != NO_ERROR) { 1651 return INVALID_OPERATION; 1652 } 1653 size_t framesWritten = mBytesWritten / mFrameSize; 1654 *halFrames = framesWritten; 1655 1656 if (isSuspended()) { 1657 // return an estimation of rendered frames when the output is suspended 1658 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1659 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1660 return NO_ERROR; 1661 } else { 1662 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1663 } 1664} 1665 1666uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1667{ 1668 Mutex::Autolock _l(mLock); 1669 uint32_t result = 0; 1670 if (getEffectChain_l(sessionId) != 0) { 1671 result = EFFECT_SESSION; 1672 } 1673 1674 for (size_t i = 0; i < mTracks.size(); ++i) { 1675 sp<Track> track = mTracks[i]; 1676 if (sessionId == track->sessionId() && !track->isInvalid()) { 1677 result |= TRACK_SESSION; 1678 break; 1679 } 1680 } 1681 1682 return result; 1683} 1684 1685uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1686{ 1687 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1688 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1689 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1690 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1691 } 1692 for (size_t i = 0; i < mTracks.size(); i++) { 1693 sp<Track> track = mTracks[i]; 1694 if (sessionId == track->sessionId() && !track->isInvalid()) { 1695 return AudioSystem::getStrategyForStream(track->streamType()); 1696 } 1697 } 1698 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1699} 1700 1701 1702AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1703{ 1704 Mutex::Autolock _l(mLock); 1705 return mOutput; 1706} 1707 1708AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1709{ 1710 Mutex::Autolock _l(mLock); 1711 AudioStreamOut *output = mOutput; 1712 mOutput = NULL; 1713 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1714 // must push a NULL and wait for ack 1715 mOutputSink.clear(); 1716 mPipeSink.clear(); 1717 mNormalSink.clear(); 1718 return output; 1719} 1720 1721// this method must always be called either with ThreadBase mLock held or inside the thread loop 1722audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1723{ 1724 if (mOutput == NULL) { 1725 return NULL; 1726 } 1727 return &mOutput->stream->common; 1728} 1729 1730uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1731{ 1732 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1733} 1734 1735status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1736{ 1737 if (!isValidSyncEvent(event)) { 1738 return BAD_VALUE; 1739 } 1740 1741 Mutex::Autolock _l(mLock); 1742 1743 for (size_t i = 0; i < mTracks.size(); ++i) { 1744 sp<Track> track = mTracks[i]; 1745 if (event->triggerSession() == track->sessionId()) { 1746 (void) track->setSyncEvent(event); 1747 return NO_ERROR; 1748 } 1749 } 1750 1751 return NAME_NOT_FOUND; 1752} 1753 1754bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1755{ 1756 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1757} 1758 1759void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1760 const Vector< sp<Track> >& tracksToRemove) 1761{ 1762 size_t count = tracksToRemove.size(); 1763 if (count > 0) { 1764 for (size_t i = 0 ; i < count ; i++) { 1765 const sp<Track>& track = tracksToRemove.itemAt(i); 1766 if (!track->isOutputTrack()) { 1767 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1768#ifdef ADD_BATTERY_DATA 1769 // to track the speaker usage 1770 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1771#endif 1772 if (track->isTerminated()) { 1773 AudioSystem::releaseOutput(mId); 1774 } 1775 } 1776 } 1777 } 1778} 1779 1780void AudioFlinger::PlaybackThread::checkSilentMode_l() 1781{ 1782 if (!mMasterMute) { 1783 char value[PROPERTY_VALUE_MAX]; 1784 if (property_get("ro.audio.silent", value, "0") > 0) { 1785 char *endptr; 1786 unsigned long ul = strtoul(value, &endptr, 0); 1787 if (*endptr == '\0' && ul != 0) { 1788 ALOGD("Silence is golden"); 1789 // The setprop command will not allow a property to be changed after 1790 // the first time it is set, so we don't have to worry about un-muting. 1791 setMasterMute_l(true); 1792 } 1793 } 1794 } 1795} 1796 1797// shared by MIXER and DIRECT, overridden by DUPLICATING 1798ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1799{ 1800 // FIXME rewrite to reduce number of system calls 1801 mLastWriteTime = systemTime(); 1802 mInWrite = true; 1803 ssize_t bytesWritten; 1804 1805 // If an NBAIO sink is present, use it to write the normal mixer's submix 1806 if (mNormalSink != 0) { 1807#define mBitShift 2 // FIXME 1808 size_t count = mBytesRemaining >> mBitShift; 1809 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1810 ATRACE_BEGIN("write"); 1811 // update the setpoint when AudioFlinger::mScreenState changes 1812 uint32_t screenState = AudioFlinger::mScreenState; 1813 if (screenState != mScreenState) { 1814 mScreenState = screenState; 1815 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1816 if (pipe != NULL) { 1817 pipe->setAvgFrames((mScreenState & 1) ? 1818 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1819 } 1820 } 1821 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1822 ATRACE_END(); 1823 if (framesWritten > 0) { 1824 bytesWritten = framesWritten << mBitShift; 1825 } else { 1826 bytesWritten = framesWritten; 1827 } 1828 // otherwise use the HAL / AudioStreamOut directly 1829 } else { 1830 // Direct output and offload threads 1831 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1832 if (mUseAsyncWrite) { 1833 mWriteBlocked = true; 1834 ALOG_ASSERT(mCallbackThread != 0); 1835 mCallbackThread->setWriteBlocked(true); 1836 } 1837 bytesWritten = mOutput->stream->write(mOutput->stream, 1838 mMixBuffer + offset, mBytesRemaining); 1839 if (mUseAsyncWrite && 1840 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1841 // do not wait for async callback in case of error of full write 1842 mWriteBlocked = false; 1843 ALOG_ASSERT(mCallbackThread != 0); 1844 mCallbackThread->setWriteBlocked(false); 1845 } 1846 } 1847 1848 mNumWrites++; 1849 mInWrite = false; 1850 1851 return bytesWritten; 1852} 1853 1854void AudioFlinger::PlaybackThread::threadLoop_drain() 1855{ 1856 if (mOutput->stream->drain) { 1857 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1858 if (mUseAsyncWrite) { 1859 mDraining = true; 1860 ALOG_ASSERT(mCallbackThread != 0); 1861 mCallbackThread->setDraining(true); 1862 } 1863 mOutput->stream->drain(mOutput->stream, 1864 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1865 : AUDIO_DRAIN_ALL); 1866 } 1867} 1868 1869void AudioFlinger::PlaybackThread::threadLoop_exit() 1870{ 1871 // Default implementation has nothing to do 1872} 1873 1874/* 1875The derived values that are cached: 1876 - mixBufferSize from frame count * frame size 1877 - activeSleepTime from activeSleepTimeUs() 1878 - idleSleepTime from idleSleepTimeUs() 1879 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1880 - maxPeriod from frame count and sample rate (MIXER only) 1881 1882The parameters that affect these derived values are: 1883 - frame count 1884 - frame size 1885 - sample rate 1886 - device type: A2DP or not 1887 - device latency 1888 - format: PCM or not 1889 - active sleep time 1890 - idle sleep time 1891*/ 1892 1893void AudioFlinger::PlaybackThread::cacheParameters_l() 1894{ 1895 mixBufferSize = mNormalFrameCount * mFrameSize; 1896 activeSleepTime = activeSleepTimeUs(); 1897 idleSleepTime = idleSleepTimeUs(); 1898} 1899 1900void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1901{ 1902 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1903 this, streamType, mTracks.size()); 1904 Mutex::Autolock _l(mLock); 1905 1906 size_t size = mTracks.size(); 1907 for (size_t i = 0; i < size; i++) { 1908 sp<Track> t = mTracks[i]; 1909 if (t->streamType() == streamType) { 1910 t->invalidate(); 1911 } 1912 } 1913} 1914 1915status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1916{ 1917 int session = chain->sessionId(); 1918 int16_t *buffer = mMixBuffer; 1919 bool ownsBuffer = false; 1920 1921 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1922 if (session > 0) { 1923 // Only one effect chain can be present in direct output thread and it uses 1924 // the mix buffer as input 1925 if (mType != DIRECT) { 1926 size_t numSamples = mNormalFrameCount * mChannelCount; 1927 buffer = new int16_t[numSamples]; 1928 memset(buffer, 0, numSamples * sizeof(int16_t)); 1929 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1930 ownsBuffer = true; 1931 } 1932 1933 // Attach all tracks with same session ID to this chain. 1934 for (size_t i = 0; i < mTracks.size(); ++i) { 1935 sp<Track> track = mTracks[i]; 1936 if (session == track->sessionId()) { 1937 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1938 buffer); 1939 track->setMainBuffer(buffer); 1940 chain->incTrackCnt(); 1941 } 1942 } 1943 1944 // indicate all active tracks in the chain 1945 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1946 sp<Track> track = mActiveTracks[i].promote(); 1947 if (track == 0) { 1948 continue; 1949 } 1950 if (session == track->sessionId()) { 1951 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1952 chain->incActiveTrackCnt(); 1953 } 1954 } 1955 } 1956 1957 chain->setInBuffer(buffer, ownsBuffer); 1958 chain->setOutBuffer(mMixBuffer); 1959 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1960 // chains list in order to be processed last as it contains output stage effects 1961 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1962 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1963 // after track specific effects and before output stage 1964 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1965 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1966 // Effect chain for other sessions are inserted at beginning of effect 1967 // chains list to be processed before output mix effects. Relative order between other 1968 // sessions is not important 1969 size_t size = mEffectChains.size(); 1970 size_t i = 0; 1971 for (i = 0; i < size; i++) { 1972 if (mEffectChains[i]->sessionId() < session) { 1973 break; 1974 } 1975 } 1976 mEffectChains.insertAt(chain, i); 1977 checkSuspendOnAddEffectChain_l(chain); 1978 1979 return NO_ERROR; 1980} 1981 1982size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1983{ 1984 int session = chain->sessionId(); 1985 1986 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1987 1988 for (size_t i = 0; i < mEffectChains.size(); i++) { 1989 if (chain == mEffectChains[i]) { 1990 mEffectChains.removeAt(i); 1991 // detach all active tracks from the chain 1992 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1993 sp<Track> track = mActiveTracks[i].promote(); 1994 if (track == 0) { 1995 continue; 1996 } 1997 if (session == track->sessionId()) { 1998 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1999 chain.get(), session); 2000 chain->decActiveTrackCnt(); 2001 } 2002 } 2003 2004 // detach all tracks with same session ID from this chain 2005 for (size_t i = 0; i < mTracks.size(); ++i) { 2006 sp<Track> track = mTracks[i]; 2007 if (session == track->sessionId()) { 2008 track->setMainBuffer(mMixBuffer); 2009 chain->decTrackCnt(); 2010 } 2011 } 2012 break; 2013 } 2014 } 2015 return mEffectChains.size(); 2016} 2017 2018status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2019 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2020{ 2021 Mutex::Autolock _l(mLock); 2022 return attachAuxEffect_l(track, EffectId); 2023} 2024 2025status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2026 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2027{ 2028 status_t status = NO_ERROR; 2029 2030 if (EffectId == 0) { 2031 track->setAuxBuffer(0, NULL); 2032 } else { 2033 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2034 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2035 if (effect != 0) { 2036 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2037 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2038 } else { 2039 status = INVALID_OPERATION; 2040 } 2041 } else { 2042 status = BAD_VALUE; 2043 } 2044 } 2045 return status; 2046} 2047 2048void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2049{ 2050 for (size_t i = 0; i < mTracks.size(); ++i) { 2051 sp<Track> track = mTracks[i]; 2052 if (track->auxEffectId() == effectId) { 2053 attachAuxEffect_l(track, 0); 2054 } 2055 } 2056} 2057 2058bool AudioFlinger::PlaybackThread::threadLoop() 2059{ 2060 Vector< sp<Track> > tracksToRemove; 2061 2062 standbyTime = systemTime(); 2063 2064 // MIXER 2065 nsecs_t lastWarning = 0; 2066 2067 // DUPLICATING 2068 // FIXME could this be made local to while loop? 2069 writeFrames = 0; 2070 2071 cacheParameters_l(); 2072 sleepTime = idleSleepTime; 2073 2074 if (mType == MIXER) { 2075 sleepTimeShift = 0; 2076 } 2077 2078 CpuStats cpuStats; 2079 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2080 2081 acquireWakeLock(); 2082 2083 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2084 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2085 // and then that string will be logged at the next convenient opportunity. 2086 const char *logString = NULL; 2087 2088 while (!exitPending()) 2089 { 2090 cpuStats.sample(myName); 2091 2092 Vector< sp<EffectChain> > effectChains; 2093 2094 processConfigEvents(); 2095 2096 { // scope for mLock 2097 2098 Mutex::Autolock _l(mLock); 2099 2100 if (logString != NULL) { 2101 mNBLogWriter->logTimestamp(); 2102 mNBLogWriter->log(logString); 2103 logString = NULL; 2104 } 2105 2106 if (checkForNewParameters_l()) { 2107 cacheParameters_l(); 2108 } 2109 2110 saveOutputTracks(); 2111 2112 if (mSignalPending) { 2113 // A signal was raised while we were unlocked 2114 mSignalPending = false; 2115 } else if (waitingAsyncCallback_l()) { 2116 if (exitPending()) { 2117 break; 2118 } 2119 releaseWakeLock_l(); 2120 ALOGV("wait async completion"); 2121 mWaitWorkCV.wait(mLock); 2122 ALOGV("async completion/wake"); 2123 acquireWakeLock_l(); 2124 if (exitPending()) { 2125 break; 2126 } 2127 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2128 continue; 2129 } 2130 sleepTime = 0; 2131 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2132 isSuspended()) { 2133 // put audio hardware into standby after short delay 2134 if (shouldStandby_l()) { 2135 2136 threadLoop_standby(); 2137 2138 mStandby = true; 2139 } 2140 2141 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2142 // we're about to wait, flush the binder command buffer 2143 IPCThreadState::self()->flushCommands(); 2144 2145 clearOutputTracks(); 2146 2147 if (exitPending()) { 2148 break; 2149 } 2150 2151 releaseWakeLock_l(); 2152 // wait until we have something to do... 2153 ALOGV("%s going to sleep", myName.string()); 2154 mWaitWorkCV.wait(mLock); 2155 ALOGV("%s waking up", myName.string()); 2156 acquireWakeLock_l(); 2157 2158 mMixerStatus = MIXER_IDLE; 2159 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2160 mBytesWritten = 0; 2161 mBytesRemaining = 0; 2162 checkSilentMode_l(); 2163 2164 standbyTime = systemTime() + standbyDelay; 2165 sleepTime = idleSleepTime; 2166 if (mType == MIXER) { 2167 sleepTimeShift = 0; 2168 } 2169 2170 continue; 2171 } 2172 } 2173 2174 // mMixerStatusIgnoringFastTracks is also updated internally 2175 mMixerStatus = prepareTracks_l(&tracksToRemove); 2176 2177 // prevent any changes in effect chain list and in each effect chain 2178 // during mixing and effect process as the audio buffers could be deleted 2179 // or modified if an effect is created or deleted 2180 lockEffectChains_l(effectChains); 2181 } 2182 2183 if (mBytesRemaining == 0) { 2184 mCurrentWriteLength = 0; 2185 if (mMixerStatus == MIXER_TRACKS_READY) { 2186 // threadLoop_mix() sets mCurrentWriteLength 2187 threadLoop_mix(); 2188 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2189 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2190 // threadLoop_sleepTime sets sleepTime to 0 if data 2191 // must be written to HAL 2192 threadLoop_sleepTime(); 2193 if (sleepTime == 0) { 2194 mCurrentWriteLength = mixBufferSize; 2195 } 2196 } 2197 mBytesRemaining = mCurrentWriteLength; 2198 if (isSuspended()) { 2199 sleepTime = suspendSleepTimeUs(); 2200 // simulate write to HAL when suspended 2201 mBytesWritten += mixBufferSize; 2202 mBytesRemaining = 0; 2203 } 2204 2205 // only process effects if we're going to write 2206 if (sleepTime == 0) { 2207 for (size_t i = 0; i < effectChains.size(); i ++) { 2208 effectChains[i]->process_l(); 2209 } 2210 } 2211 } 2212 2213 // enable changes in effect chain 2214 unlockEffectChains(effectChains); 2215 2216 if (!waitingAsyncCallback()) { 2217 // sleepTime == 0 means we must write to audio hardware 2218 if (sleepTime == 0) { 2219 if (mBytesRemaining) { 2220 ssize_t ret = threadLoop_write(); 2221 if (ret < 0) { 2222 mBytesRemaining = 0; 2223 } else { 2224 mBytesWritten += ret; 2225 mBytesRemaining -= ret; 2226 } 2227 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2228 (mMixerStatus == MIXER_DRAIN_ALL)) { 2229 threadLoop_drain(); 2230 } 2231if (mType == MIXER) { 2232 // write blocked detection 2233 nsecs_t now = systemTime(); 2234 nsecs_t delta = now - mLastWriteTime; 2235 if (!mStandby && delta > maxPeriod) { 2236 mNumDelayedWrites++; 2237 if ((now - lastWarning) > kWarningThrottleNs) { 2238 ATRACE_NAME("underrun"); 2239 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2240 ns2ms(delta), mNumDelayedWrites, this); 2241 lastWarning = now; 2242 } 2243 } 2244} 2245 2246 mStandby = false; 2247 } else { 2248 usleep(sleepTime); 2249 } 2250 } 2251 2252 // Finally let go of removed track(s), without the lock held 2253 // since we can't guarantee the destructors won't acquire that 2254 // same lock. This will also mutate and push a new fast mixer state. 2255 threadLoop_removeTracks(tracksToRemove); 2256 tracksToRemove.clear(); 2257 2258 // FIXME I don't understand the need for this here; 2259 // it was in the original code but maybe the 2260 // assignment in saveOutputTracks() makes this unnecessary? 2261 clearOutputTracks(); 2262 2263 // Effect chains will be actually deleted here if they were removed from 2264 // mEffectChains list during mixing or effects processing 2265 effectChains.clear(); 2266 2267 // FIXME Note that the above .clear() is no longer necessary since effectChains 2268 // is now local to this block, but will keep it for now (at least until merge done). 2269 } 2270 2271 threadLoop_exit(); 2272 2273 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2274 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2275 // put output stream into standby mode 2276 if (!mStandby) { 2277 mOutput->stream->common.standby(&mOutput->stream->common); 2278 } 2279 } 2280 2281 releaseWakeLock(); 2282 2283 ALOGV("Thread %p type %d exiting", this, mType); 2284 return false; 2285} 2286 2287// removeTracks_l() must be called with ThreadBase::mLock held 2288void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2289{ 2290 size_t count = tracksToRemove.size(); 2291 if (count > 0) { 2292 for (size_t i=0 ; i<count ; i++) { 2293 const sp<Track>& track = tracksToRemove.itemAt(i); 2294 mActiveTracks.remove(track); 2295 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2296 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2297 if (chain != 0) { 2298 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2299 track->sessionId()); 2300 chain->decActiveTrackCnt(); 2301 } 2302 if (track->isTerminated()) { 2303 removeTrack_l(track); 2304 } 2305 } 2306 } 2307 2308} 2309 2310// ---------------------------------------------------------------------------- 2311 2312AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2313 audio_io_handle_t id, audio_devices_t device, type_t type) 2314 : PlaybackThread(audioFlinger, output, id, device, type), 2315 // mAudioMixer below 2316 // mFastMixer below 2317 mFastMixerFutex(0) 2318 // mOutputSink below 2319 // mPipeSink below 2320 // mNormalSink below 2321{ 2322 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2323 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2324 "mFrameCount=%d, mNormalFrameCount=%d", 2325 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2326 mNormalFrameCount); 2327 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2328 2329 // FIXME - Current mixer implementation only supports stereo output 2330 if (mChannelCount != FCC_2) { 2331 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2332 } 2333 2334 // create an NBAIO sink for the HAL output stream, and negotiate 2335 mOutputSink = new AudioStreamOutSink(output->stream); 2336 size_t numCounterOffers = 0; 2337 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2338 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2339 ALOG_ASSERT(index == 0); 2340 2341 // initialize fast mixer depending on configuration 2342 bool initFastMixer; 2343 switch (kUseFastMixer) { 2344 case FastMixer_Never: 2345 initFastMixer = false; 2346 break; 2347 case FastMixer_Always: 2348 initFastMixer = true; 2349 break; 2350 case FastMixer_Static: 2351 case FastMixer_Dynamic: 2352 initFastMixer = mFrameCount < mNormalFrameCount; 2353 break; 2354 } 2355 if (initFastMixer) { 2356 2357 // create a MonoPipe to connect our submix to FastMixer 2358 NBAIO_Format format = mOutputSink->format(); 2359 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2360 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2361 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2362 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2363 const NBAIO_Format offers[1] = {format}; 2364 size_t numCounterOffers = 0; 2365 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2366 ALOG_ASSERT(index == 0); 2367 monoPipe->setAvgFrames((mScreenState & 1) ? 2368 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2369 mPipeSink = monoPipe; 2370 2371#ifdef TEE_SINK 2372 if (mTeeSinkOutputEnabled) { 2373 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2374 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2375 numCounterOffers = 0; 2376 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2377 ALOG_ASSERT(index == 0); 2378 mTeeSink = teeSink; 2379 PipeReader *teeSource = new PipeReader(*teeSink); 2380 numCounterOffers = 0; 2381 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2382 ALOG_ASSERT(index == 0); 2383 mTeeSource = teeSource; 2384 } 2385#endif 2386 2387 // create fast mixer and configure it initially with just one fast track for our submix 2388 mFastMixer = new FastMixer(); 2389 FastMixerStateQueue *sq = mFastMixer->sq(); 2390#ifdef STATE_QUEUE_DUMP 2391 sq->setObserverDump(&mStateQueueObserverDump); 2392 sq->setMutatorDump(&mStateQueueMutatorDump); 2393#endif 2394 FastMixerState *state = sq->begin(); 2395 FastTrack *fastTrack = &state->mFastTracks[0]; 2396 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2397 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2398 fastTrack->mVolumeProvider = NULL; 2399 fastTrack->mGeneration++; 2400 state->mFastTracksGen++; 2401 state->mTrackMask = 1; 2402 // fast mixer will use the HAL output sink 2403 state->mOutputSink = mOutputSink.get(); 2404 state->mOutputSinkGen++; 2405 state->mFrameCount = mFrameCount; 2406 state->mCommand = FastMixerState::COLD_IDLE; 2407 // already done in constructor initialization list 2408 //mFastMixerFutex = 0; 2409 state->mColdFutexAddr = &mFastMixerFutex; 2410 state->mColdGen++; 2411 state->mDumpState = &mFastMixerDumpState; 2412#ifdef TEE_SINK 2413 state->mTeeSink = mTeeSink.get(); 2414#endif 2415 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2416 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2417 sq->end(); 2418 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2419 2420 // start the fast mixer 2421 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2422 pid_t tid = mFastMixer->getTid(); 2423 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2424 if (err != 0) { 2425 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2426 kPriorityFastMixer, getpid_cached, tid, err); 2427 } 2428 2429#ifdef AUDIO_WATCHDOG 2430 // create and start the watchdog 2431 mAudioWatchdog = new AudioWatchdog(); 2432 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2433 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2434 tid = mAudioWatchdog->getTid(); 2435 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2436 if (err != 0) { 2437 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2438 kPriorityFastMixer, getpid_cached, tid, err); 2439 } 2440#endif 2441 2442 } else { 2443 mFastMixer = NULL; 2444 } 2445 2446 switch (kUseFastMixer) { 2447 case FastMixer_Never: 2448 case FastMixer_Dynamic: 2449 mNormalSink = mOutputSink; 2450 break; 2451 case FastMixer_Always: 2452 mNormalSink = mPipeSink; 2453 break; 2454 case FastMixer_Static: 2455 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2456 break; 2457 } 2458} 2459 2460AudioFlinger::MixerThread::~MixerThread() 2461{ 2462 if (mFastMixer != NULL) { 2463 FastMixerStateQueue *sq = mFastMixer->sq(); 2464 FastMixerState *state = sq->begin(); 2465 if (state->mCommand == FastMixerState::COLD_IDLE) { 2466 int32_t old = android_atomic_inc(&mFastMixerFutex); 2467 if (old == -1) { 2468 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2469 } 2470 } 2471 state->mCommand = FastMixerState::EXIT; 2472 sq->end(); 2473 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2474 mFastMixer->join(); 2475 // Though the fast mixer thread has exited, it's state queue is still valid. 2476 // We'll use that extract the final state which contains one remaining fast track 2477 // corresponding to our sub-mix. 2478 state = sq->begin(); 2479 ALOG_ASSERT(state->mTrackMask == 1); 2480 FastTrack *fastTrack = &state->mFastTracks[0]; 2481 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2482 delete fastTrack->mBufferProvider; 2483 sq->end(false /*didModify*/); 2484 delete mFastMixer; 2485#ifdef AUDIO_WATCHDOG 2486 if (mAudioWatchdog != 0) { 2487 mAudioWatchdog->requestExit(); 2488 mAudioWatchdog->requestExitAndWait(); 2489 mAudioWatchdog.clear(); 2490 } 2491#endif 2492 } 2493 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2494 delete mAudioMixer; 2495} 2496 2497 2498uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2499{ 2500 if (mFastMixer != NULL) { 2501 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2502 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2503 } 2504 return latency; 2505} 2506 2507 2508void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2509{ 2510 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2511} 2512 2513ssize_t AudioFlinger::MixerThread::threadLoop_write() 2514{ 2515 // FIXME we should only do one push per cycle; confirm this is true 2516 // Start the fast mixer if it's not already running 2517 if (mFastMixer != NULL) { 2518 FastMixerStateQueue *sq = mFastMixer->sq(); 2519 FastMixerState *state = sq->begin(); 2520 if (state->mCommand != FastMixerState::MIX_WRITE && 2521 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2522 if (state->mCommand == FastMixerState::COLD_IDLE) { 2523 int32_t old = android_atomic_inc(&mFastMixerFutex); 2524 if (old == -1) { 2525 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2526 } 2527#ifdef AUDIO_WATCHDOG 2528 if (mAudioWatchdog != 0) { 2529 mAudioWatchdog->resume(); 2530 } 2531#endif 2532 } 2533 state->mCommand = FastMixerState::MIX_WRITE; 2534 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2535 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2536 sq->end(); 2537 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2538 if (kUseFastMixer == FastMixer_Dynamic) { 2539 mNormalSink = mPipeSink; 2540 } 2541 } else { 2542 sq->end(false /*didModify*/); 2543 } 2544 } 2545 return PlaybackThread::threadLoop_write(); 2546} 2547 2548void AudioFlinger::MixerThread::threadLoop_standby() 2549{ 2550 // Idle the fast mixer if it's currently running 2551 if (mFastMixer != NULL) { 2552 FastMixerStateQueue *sq = mFastMixer->sq(); 2553 FastMixerState *state = sq->begin(); 2554 if (!(state->mCommand & FastMixerState::IDLE)) { 2555 state->mCommand = FastMixerState::COLD_IDLE; 2556 state->mColdFutexAddr = &mFastMixerFutex; 2557 state->mColdGen++; 2558 mFastMixerFutex = 0; 2559 sq->end(); 2560 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2561 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2562 if (kUseFastMixer == FastMixer_Dynamic) { 2563 mNormalSink = mOutputSink; 2564 } 2565#ifdef AUDIO_WATCHDOG 2566 if (mAudioWatchdog != 0) { 2567 mAudioWatchdog->pause(); 2568 } 2569#endif 2570 } else { 2571 sq->end(false /*didModify*/); 2572 } 2573 } 2574 PlaybackThread::threadLoop_standby(); 2575} 2576 2577// Empty implementation for standard mixer 2578// Overridden for offloaded playback 2579void AudioFlinger::PlaybackThread::flushOutput_l() 2580{ 2581} 2582 2583bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2584{ 2585 return false; 2586} 2587 2588bool AudioFlinger::PlaybackThread::shouldStandby_l() 2589{ 2590 return !mStandby; 2591} 2592 2593bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2594{ 2595 Mutex::Autolock _l(mLock); 2596 return waitingAsyncCallback_l(); 2597} 2598 2599// shared by MIXER and DIRECT, overridden by DUPLICATING 2600void AudioFlinger::PlaybackThread::threadLoop_standby() 2601{ 2602 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2603 mOutput->stream->common.standby(&mOutput->stream->common); 2604 if (mUseAsyncWrite != 0) { 2605 mWriteBlocked = false; 2606 mDraining = false; 2607 ALOG_ASSERT(mCallbackThread != 0); 2608 mCallbackThread->setWriteBlocked(false); 2609 mCallbackThread->setDraining(false); 2610 } 2611} 2612 2613void AudioFlinger::MixerThread::threadLoop_mix() 2614{ 2615 // obtain the presentation timestamp of the next output buffer 2616 int64_t pts; 2617 status_t status = INVALID_OPERATION; 2618 2619 if (mNormalSink != 0) { 2620 status = mNormalSink->getNextWriteTimestamp(&pts); 2621 } else { 2622 status = mOutputSink->getNextWriteTimestamp(&pts); 2623 } 2624 2625 if (status != NO_ERROR) { 2626 pts = AudioBufferProvider::kInvalidPTS; 2627 } 2628 2629 // mix buffers... 2630 mAudioMixer->process(pts); 2631 mCurrentWriteLength = mixBufferSize; 2632 // increase sleep time progressively when application underrun condition clears. 2633 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2634 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2635 // such that we would underrun the audio HAL. 2636 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2637 sleepTimeShift--; 2638 } 2639 sleepTime = 0; 2640 standbyTime = systemTime() + standbyDelay; 2641 //TODO: delay standby when effects have a tail 2642} 2643 2644void AudioFlinger::MixerThread::threadLoop_sleepTime() 2645{ 2646 // If no tracks are ready, sleep once for the duration of an output 2647 // buffer size, then write 0s to the output 2648 if (sleepTime == 0) { 2649 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2650 sleepTime = activeSleepTime >> sleepTimeShift; 2651 if (sleepTime < kMinThreadSleepTimeUs) { 2652 sleepTime = kMinThreadSleepTimeUs; 2653 } 2654 // reduce sleep time in case of consecutive application underruns to avoid 2655 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2656 // duration we would end up writing less data than needed by the audio HAL if 2657 // the condition persists. 2658 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2659 sleepTimeShift++; 2660 } 2661 } else { 2662 sleepTime = idleSleepTime; 2663 } 2664 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2665 memset(mMixBuffer, 0, mixBufferSize); 2666 sleepTime = 0; 2667 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2668 "anticipated start"); 2669 } 2670 // TODO add standby time extension fct of effect tail 2671} 2672 2673// prepareTracks_l() must be called with ThreadBase::mLock held 2674AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2675 Vector< sp<Track> > *tracksToRemove) 2676{ 2677 2678 mixer_state mixerStatus = MIXER_IDLE; 2679 // find out which tracks need to be processed 2680 size_t count = mActiveTracks.size(); 2681 size_t mixedTracks = 0; 2682 size_t tracksWithEffect = 0; 2683 // counts only _active_ fast tracks 2684 size_t fastTracks = 0; 2685 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2686 2687 float masterVolume = mMasterVolume; 2688 bool masterMute = mMasterMute; 2689 2690 if (masterMute) { 2691 masterVolume = 0; 2692 } 2693 // Delegate master volume control to effect in output mix effect chain if needed 2694 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2695 if (chain != 0) { 2696 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2697 chain->setVolume_l(&v, &v); 2698 masterVolume = (float)((v + (1 << 23)) >> 24); 2699 chain.clear(); 2700 } 2701 2702 // prepare a new state to push 2703 FastMixerStateQueue *sq = NULL; 2704 FastMixerState *state = NULL; 2705 bool didModify = false; 2706 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2707 if (mFastMixer != NULL) { 2708 sq = mFastMixer->sq(); 2709 state = sq->begin(); 2710 } 2711 2712 for (size_t i=0 ; i<count ; i++) { 2713 const sp<Track> t = mActiveTracks[i].promote(); 2714 if (t == 0) { 2715 continue; 2716 } 2717 2718 // this const just means the local variable doesn't change 2719 Track* const track = t.get(); 2720 2721 // process fast tracks 2722 if (track->isFastTrack()) { 2723 2724 // It's theoretically possible (though unlikely) for a fast track to be created 2725 // and then removed within the same normal mix cycle. This is not a problem, as 2726 // the track never becomes active so it's fast mixer slot is never touched. 2727 // The converse, of removing an (active) track and then creating a new track 2728 // at the identical fast mixer slot within the same normal mix cycle, 2729 // is impossible because the slot isn't marked available until the end of each cycle. 2730 int j = track->mFastIndex; 2731 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2732 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2733 FastTrack *fastTrack = &state->mFastTracks[j]; 2734 2735 // Determine whether the track is currently in underrun condition, 2736 // and whether it had a recent underrun. 2737 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2738 FastTrackUnderruns underruns = ftDump->mUnderruns; 2739 uint32_t recentFull = (underruns.mBitFields.mFull - 2740 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2741 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2742 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2743 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2744 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2745 uint32_t recentUnderruns = recentPartial + recentEmpty; 2746 track->mObservedUnderruns = underruns; 2747 // don't count underruns that occur while stopping or pausing 2748 // or stopped which can occur when flush() is called while active 2749 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2750 recentUnderruns > 0) { 2751 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2752 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2753 } 2754 2755 // This is similar to the state machine for normal tracks, 2756 // with a few modifications for fast tracks. 2757 bool isActive = true; 2758 switch (track->mState) { 2759 case TrackBase::STOPPING_1: 2760 // track stays active in STOPPING_1 state until first underrun 2761 if (recentUnderruns > 0 || track->isTerminated()) { 2762 track->mState = TrackBase::STOPPING_2; 2763 } 2764 break; 2765 case TrackBase::PAUSING: 2766 // ramp down is not yet implemented 2767 track->setPaused(); 2768 break; 2769 case TrackBase::RESUMING: 2770 // ramp up is not yet implemented 2771 track->mState = TrackBase::ACTIVE; 2772 break; 2773 case TrackBase::ACTIVE: 2774 if (recentFull > 0 || recentPartial > 0) { 2775 // track has provided at least some frames recently: reset retry count 2776 track->mRetryCount = kMaxTrackRetries; 2777 } 2778 if (recentUnderruns == 0) { 2779 // no recent underruns: stay active 2780 break; 2781 } 2782 // there has recently been an underrun of some kind 2783 if (track->sharedBuffer() == 0) { 2784 // were any of the recent underruns "empty" (no frames available)? 2785 if (recentEmpty == 0) { 2786 // no, then ignore the partial underruns as they are allowed indefinitely 2787 break; 2788 } 2789 // there has recently been an "empty" underrun: decrement the retry counter 2790 if (--(track->mRetryCount) > 0) { 2791 break; 2792 } 2793 // indicate to client process that the track was disabled because of underrun; 2794 // it will then automatically call start() when data is available 2795 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2796 // remove from active list, but state remains ACTIVE [confusing but true] 2797 isActive = false; 2798 break; 2799 } 2800 // fall through 2801 case TrackBase::STOPPING_2: 2802 case TrackBase::PAUSED: 2803 case TrackBase::STOPPED: 2804 case TrackBase::FLUSHED: // flush() while active 2805 // Check for presentation complete if track is inactive 2806 // We have consumed all the buffers of this track. 2807 // This would be incomplete if we auto-paused on underrun 2808 { 2809 size_t audioHALFrames = 2810 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2811 size_t framesWritten = mBytesWritten / mFrameSize; 2812 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2813 // track stays in active list until presentation is complete 2814 break; 2815 } 2816 } 2817 if (track->isStopping_2()) { 2818 track->mState = TrackBase::STOPPED; 2819 } 2820 if (track->isStopped()) { 2821 // Can't reset directly, as fast mixer is still polling this track 2822 // track->reset(); 2823 // So instead mark this track as needing to be reset after push with ack 2824 resetMask |= 1 << i; 2825 } 2826 isActive = false; 2827 break; 2828 case TrackBase::IDLE: 2829 default: 2830 LOG_FATAL("unexpected track state %d", track->mState); 2831 } 2832 2833 if (isActive) { 2834 // was it previously inactive? 2835 if (!(state->mTrackMask & (1 << j))) { 2836 ExtendedAudioBufferProvider *eabp = track; 2837 VolumeProvider *vp = track; 2838 fastTrack->mBufferProvider = eabp; 2839 fastTrack->mVolumeProvider = vp; 2840 fastTrack->mSampleRate = track->mSampleRate; 2841 fastTrack->mChannelMask = track->mChannelMask; 2842 fastTrack->mGeneration++; 2843 state->mTrackMask |= 1 << j; 2844 didModify = true; 2845 // no acknowledgement required for newly active tracks 2846 } 2847 // cache the combined master volume and stream type volume for fast mixer; this 2848 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2849 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2850 ++fastTracks; 2851 } else { 2852 // was it previously active? 2853 if (state->mTrackMask & (1 << j)) { 2854 fastTrack->mBufferProvider = NULL; 2855 fastTrack->mGeneration++; 2856 state->mTrackMask &= ~(1 << j); 2857 didModify = true; 2858 // If any fast tracks were removed, we must wait for acknowledgement 2859 // because we're about to decrement the last sp<> on those tracks. 2860 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2861 } else { 2862 LOG_FATAL("fast track %d should have been active", j); 2863 } 2864 tracksToRemove->add(track); 2865 // Avoids a misleading display in dumpsys 2866 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2867 } 2868 continue; 2869 } 2870 2871 { // local variable scope to avoid goto warning 2872 2873 audio_track_cblk_t* cblk = track->cblk(); 2874 2875 // The first time a track is added we wait 2876 // for all its buffers to be filled before processing it 2877 int name = track->name(); 2878 // make sure that we have enough frames to mix one full buffer. 2879 // enforce this condition only once to enable draining the buffer in case the client 2880 // app does not call stop() and relies on underrun to stop: 2881 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2882 // during last round 2883 size_t desiredFrames; 2884 uint32_t sr = track->sampleRate(); 2885 if (sr == mSampleRate) { 2886 desiredFrames = mNormalFrameCount; 2887 } else { 2888 // +1 for rounding and +1 for additional sample needed for interpolation 2889 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2890 // add frames already consumed but not yet released by the resampler 2891 // because mAudioTrackServerProxy->framesReady() will include these frames 2892 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2893 // the minimum track buffer size is normally twice the number of frames necessary 2894 // to fill one buffer and the resampler should not leave more than one buffer worth 2895 // of unreleased frames after each pass, but just in case... 2896 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2897 } 2898 uint32_t minFrames = 1; 2899 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2900 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2901 minFrames = desiredFrames; 2902 } 2903 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2904 size_t framesReady; 2905 if (track->sharedBuffer() == 0) { 2906 framesReady = track->framesReady(); 2907 } else if (track->isStopped()) { 2908 framesReady = 0; 2909 } else { 2910 framesReady = 1; 2911 } 2912 if ((framesReady >= minFrames) && track->isReady() && 2913 !track->isPaused() && !track->isTerminated()) 2914 { 2915 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2916 2917 mixedTracks++; 2918 2919 // track->mainBuffer() != mMixBuffer means there is an effect chain 2920 // connected to the track 2921 chain.clear(); 2922 if (track->mainBuffer() != mMixBuffer) { 2923 chain = getEffectChain_l(track->sessionId()); 2924 // Delegate volume control to effect in track effect chain if needed 2925 if (chain != 0) { 2926 tracksWithEffect++; 2927 } else { 2928 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2929 "session %d", 2930 name, track->sessionId()); 2931 } 2932 } 2933 2934 2935 int param = AudioMixer::VOLUME; 2936 if (track->mFillingUpStatus == Track::FS_FILLED) { 2937 // no ramp for the first volume setting 2938 track->mFillingUpStatus = Track::FS_ACTIVE; 2939 if (track->mState == TrackBase::RESUMING) { 2940 track->mState = TrackBase::ACTIVE; 2941 param = AudioMixer::RAMP_VOLUME; 2942 } 2943 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2944 // FIXME should not make a decision based on mServer 2945 } else if (cblk->mServer != 0) { 2946 // If the track is stopped before the first frame was mixed, 2947 // do not apply ramp 2948 param = AudioMixer::RAMP_VOLUME; 2949 } 2950 2951 // compute volume for this track 2952 uint32_t vl, vr, va; 2953 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2954 vl = vr = va = 0; 2955 if (track->isPausing()) { 2956 track->setPaused(); 2957 } 2958 } else { 2959 2960 // read original volumes with volume control 2961 float typeVolume = mStreamTypes[track->streamType()].volume; 2962 float v = masterVolume * typeVolume; 2963 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2964 uint32_t vlr = proxy->getVolumeLR(); 2965 vl = vlr & 0xFFFF; 2966 vr = vlr >> 16; 2967 // track volumes come from shared memory, so can't be trusted and must be clamped 2968 if (vl > MAX_GAIN_INT) { 2969 ALOGV("Track left volume out of range: %04X", vl); 2970 vl = MAX_GAIN_INT; 2971 } 2972 if (vr > MAX_GAIN_INT) { 2973 ALOGV("Track right volume out of range: %04X", vr); 2974 vr = MAX_GAIN_INT; 2975 } 2976 // now apply the master volume and stream type volume 2977 vl = (uint32_t)(v * vl) << 12; 2978 vr = (uint32_t)(v * vr) << 12; 2979 // assuming master volume and stream type volume each go up to 1.0, 2980 // vl and vr are now in 8.24 format 2981 2982 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2983 // send level comes from shared memory and so may be corrupt 2984 if (sendLevel > MAX_GAIN_INT) { 2985 ALOGV("Track send level out of range: %04X", sendLevel); 2986 sendLevel = MAX_GAIN_INT; 2987 } 2988 va = (uint32_t)(v * sendLevel); 2989 } 2990 2991 // Delegate volume control to effect in track effect chain if needed 2992 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2993 // Do not ramp volume if volume is controlled by effect 2994 param = AudioMixer::VOLUME; 2995 track->mHasVolumeController = true; 2996 } else { 2997 // force no volume ramp when volume controller was just disabled or removed 2998 // from effect chain to avoid volume spike 2999 if (track->mHasVolumeController) { 3000 param = AudioMixer::VOLUME; 3001 } 3002 track->mHasVolumeController = false; 3003 } 3004 3005 // Convert volumes from 8.24 to 4.12 format 3006 // This additional clamping is needed in case chain->setVolume_l() overshot 3007 vl = (vl + (1 << 11)) >> 12; 3008 if (vl > MAX_GAIN_INT) { 3009 vl = MAX_GAIN_INT; 3010 } 3011 vr = (vr + (1 << 11)) >> 12; 3012 if (vr > MAX_GAIN_INT) { 3013 vr = MAX_GAIN_INT; 3014 } 3015 3016 if (va > MAX_GAIN_INT) { 3017 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3018 } 3019 3020 // XXX: these things DON'T need to be done each time 3021 mAudioMixer->setBufferProvider(name, track); 3022 mAudioMixer->enable(name); 3023 3024 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3025 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3026 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3027 mAudioMixer->setParameter( 3028 name, 3029 AudioMixer::TRACK, 3030 AudioMixer::FORMAT, (void *)track->format()); 3031 mAudioMixer->setParameter( 3032 name, 3033 AudioMixer::TRACK, 3034 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3035 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3036 uint32_t maxSampleRate = mSampleRate * 2; 3037 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3038 if (reqSampleRate == 0) { 3039 reqSampleRate = mSampleRate; 3040 } else if (reqSampleRate > maxSampleRate) { 3041 reqSampleRate = maxSampleRate; 3042 } 3043 mAudioMixer->setParameter( 3044 name, 3045 AudioMixer::RESAMPLE, 3046 AudioMixer::SAMPLE_RATE, 3047 (void *)reqSampleRate); 3048 mAudioMixer->setParameter( 3049 name, 3050 AudioMixer::TRACK, 3051 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3052 mAudioMixer->setParameter( 3053 name, 3054 AudioMixer::TRACK, 3055 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3056 3057 // reset retry count 3058 track->mRetryCount = kMaxTrackRetries; 3059 3060 // If one track is ready, set the mixer ready if: 3061 // - the mixer was not ready during previous round OR 3062 // - no other track is not ready 3063 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3064 mixerStatus != MIXER_TRACKS_ENABLED) { 3065 mixerStatus = MIXER_TRACKS_READY; 3066 } 3067 } else { 3068 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3069 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3070 } 3071 // clear effect chain input buffer if an active track underruns to avoid sending 3072 // previous audio buffer again to effects 3073 chain = getEffectChain_l(track->sessionId()); 3074 if (chain != 0) { 3075 chain->clearInputBuffer(); 3076 } 3077 3078 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3079 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3080 track->isStopped() || track->isPaused()) { 3081 // We have consumed all the buffers of this track. 3082 // Remove it from the list of active tracks. 3083 // TODO: use actual buffer filling status instead of latency when available from 3084 // audio HAL 3085 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3086 size_t framesWritten = mBytesWritten / mFrameSize; 3087 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3088 if (track->isStopped()) { 3089 track->reset(); 3090 } 3091 tracksToRemove->add(track); 3092 } 3093 } else { 3094 // No buffers for this track. Give it a few chances to 3095 // fill a buffer, then remove it from active list. 3096 if (--(track->mRetryCount) <= 0) { 3097 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3098 tracksToRemove->add(track); 3099 // indicate to client process that the track was disabled because of underrun; 3100 // it will then automatically call start() when data is available 3101 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3102 // If one track is not ready, mark the mixer also not ready if: 3103 // - the mixer was ready during previous round OR 3104 // - no other track is ready 3105 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3106 mixerStatus != MIXER_TRACKS_READY) { 3107 mixerStatus = MIXER_TRACKS_ENABLED; 3108 } 3109 } 3110 mAudioMixer->disable(name); 3111 } 3112 3113 } // local variable scope to avoid goto warning 3114track_is_ready: ; 3115 3116 } 3117 3118 // Push the new FastMixer state if necessary 3119 bool pauseAudioWatchdog = false; 3120 if (didModify) { 3121 state->mFastTracksGen++; 3122 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3123 if (kUseFastMixer == FastMixer_Dynamic && 3124 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3125 state->mCommand = FastMixerState::COLD_IDLE; 3126 state->mColdFutexAddr = &mFastMixerFutex; 3127 state->mColdGen++; 3128 mFastMixerFutex = 0; 3129 if (kUseFastMixer == FastMixer_Dynamic) { 3130 mNormalSink = mOutputSink; 3131 } 3132 // If we go into cold idle, need to wait for acknowledgement 3133 // so that fast mixer stops doing I/O. 3134 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3135 pauseAudioWatchdog = true; 3136 } 3137 } 3138 if (sq != NULL) { 3139 sq->end(didModify); 3140 sq->push(block); 3141 } 3142#ifdef AUDIO_WATCHDOG 3143 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3144 mAudioWatchdog->pause(); 3145 } 3146#endif 3147 3148 // Now perform the deferred reset on fast tracks that have stopped 3149 while (resetMask != 0) { 3150 size_t i = __builtin_ctz(resetMask); 3151 ALOG_ASSERT(i < count); 3152 resetMask &= ~(1 << i); 3153 sp<Track> t = mActiveTracks[i].promote(); 3154 if (t == 0) { 3155 continue; 3156 } 3157 Track* track = t.get(); 3158 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3159 track->reset(); 3160 } 3161 3162 // remove all the tracks that need to be... 3163 removeTracks_l(*tracksToRemove); 3164 3165 // mix buffer must be cleared if all tracks are connected to an 3166 // effect chain as in this case the mixer will not write to 3167 // mix buffer and track effects will accumulate into it 3168 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3169 (mixedTracks == 0 && fastTracks > 0))) { 3170 // FIXME as a performance optimization, should remember previous zero status 3171 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3172 } 3173 3174 // if any fast tracks, then status is ready 3175 mMixerStatusIgnoringFastTracks = mixerStatus; 3176 if (fastTracks > 0) { 3177 mixerStatus = MIXER_TRACKS_READY; 3178 } 3179 return mixerStatus; 3180} 3181 3182// getTrackName_l() must be called with ThreadBase::mLock held 3183int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3184{ 3185 return mAudioMixer->getTrackName(channelMask, sessionId); 3186} 3187 3188// deleteTrackName_l() must be called with ThreadBase::mLock held 3189void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3190{ 3191 ALOGV("remove track (%d) and delete from mixer", name); 3192 mAudioMixer->deleteTrackName(name); 3193} 3194 3195// checkForNewParameters_l() must be called with ThreadBase::mLock held 3196bool AudioFlinger::MixerThread::checkForNewParameters_l() 3197{ 3198 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3199 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3200 bool reconfig = false; 3201 3202 while (!mNewParameters.isEmpty()) { 3203 3204 if (mFastMixer != NULL) { 3205 FastMixerStateQueue *sq = mFastMixer->sq(); 3206 FastMixerState *state = sq->begin(); 3207 if (!(state->mCommand & FastMixerState::IDLE)) { 3208 previousCommand = state->mCommand; 3209 state->mCommand = FastMixerState::HOT_IDLE; 3210 sq->end(); 3211 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3212 } else { 3213 sq->end(false /*didModify*/); 3214 } 3215 } 3216 3217 status_t status = NO_ERROR; 3218 String8 keyValuePair = mNewParameters[0]; 3219 AudioParameter param = AudioParameter(keyValuePair); 3220 int value; 3221 3222 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3223 reconfig = true; 3224 } 3225 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3226 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3227 status = BAD_VALUE; 3228 } else { 3229 // no need to save value, since it's constant 3230 reconfig = true; 3231 } 3232 } 3233 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3234 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3235 status = BAD_VALUE; 3236 } else { 3237 // no need to save value, since it's constant 3238 reconfig = true; 3239 } 3240 } 3241 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3242 // do not accept frame count changes if tracks are open as the track buffer 3243 // size depends on frame count and correct behavior would not be guaranteed 3244 // if frame count is changed after track creation 3245 if (!mTracks.isEmpty()) { 3246 status = INVALID_OPERATION; 3247 } else { 3248 reconfig = true; 3249 } 3250 } 3251 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3252#ifdef ADD_BATTERY_DATA 3253 // when changing the audio output device, call addBatteryData to notify 3254 // the change 3255 if (mOutDevice != value) { 3256 uint32_t params = 0; 3257 // check whether speaker is on 3258 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3259 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3260 } 3261 3262 audio_devices_t deviceWithoutSpeaker 3263 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3264 // check if any other device (except speaker) is on 3265 if (value & deviceWithoutSpeaker ) { 3266 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3267 } 3268 3269 if (params != 0) { 3270 addBatteryData(params); 3271 } 3272 } 3273#endif 3274 3275 // forward device change to effects that have requested to be 3276 // aware of attached audio device. 3277 if (value != AUDIO_DEVICE_NONE) { 3278 mOutDevice = value; 3279 for (size_t i = 0; i < mEffectChains.size(); i++) { 3280 mEffectChains[i]->setDevice_l(mOutDevice); 3281 } 3282 } 3283 } 3284 3285 if (status == NO_ERROR) { 3286 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3287 keyValuePair.string()); 3288 if (!mStandby && status == INVALID_OPERATION) { 3289 mOutput->stream->common.standby(&mOutput->stream->common); 3290 mStandby = true; 3291 mBytesWritten = 0; 3292 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3293 keyValuePair.string()); 3294 } 3295 if (status == NO_ERROR && reconfig) { 3296 readOutputParameters(); 3297 delete mAudioMixer; 3298 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3299 for (size_t i = 0; i < mTracks.size() ; i++) { 3300 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3301 if (name < 0) { 3302 break; 3303 } 3304 mTracks[i]->mName = name; 3305 } 3306 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3307 } 3308 } 3309 3310 mNewParameters.removeAt(0); 3311 3312 mParamStatus = status; 3313 mParamCond.signal(); 3314 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3315 // already timed out waiting for the status and will never signal the condition. 3316 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3317 } 3318 3319 if (!(previousCommand & FastMixerState::IDLE)) { 3320 ALOG_ASSERT(mFastMixer != NULL); 3321 FastMixerStateQueue *sq = mFastMixer->sq(); 3322 FastMixerState *state = sq->begin(); 3323 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3324 state->mCommand = previousCommand; 3325 sq->end(); 3326 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3327 } 3328 3329 return reconfig; 3330} 3331 3332 3333void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3334{ 3335 const size_t SIZE = 256; 3336 char buffer[SIZE]; 3337 String8 result; 3338 3339 PlaybackThread::dumpInternals(fd, args); 3340 3341 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3342 result.append(buffer); 3343 write(fd, result.string(), result.size()); 3344 3345 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3346 const FastMixerDumpState copy(mFastMixerDumpState); 3347 copy.dump(fd); 3348 3349#ifdef STATE_QUEUE_DUMP 3350 // Similar for state queue 3351 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3352 observerCopy.dump(fd); 3353 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3354 mutatorCopy.dump(fd); 3355#endif 3356 3357#ifdef TEE_SINK 3358 // Write the tee output to a .wav file 3359 dumpTee(fd, mTeeSource, mId); 3360#endif 3361 3362#ifdef AUDIO_WATCHDOG 3363 if (mAudioWatchdog != 0) { 3364 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3365 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3366 wdCopy.dump(fd); 3367 } 3368#endif 3369} 3370 3371uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3372{ 3373 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3374} 3375 3376uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3377{ 3378 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3379} 3380 3381void AudioFlinger::MixerThread::cacheParameters_l() 3382{ 3383 PlaybackThread::cacheParameters_l(); 3384 3385 // FIXME: Relaxed timing because of a certain device that can't meet latency 3386 // Should be reduced to 2x after the vendor fixes the driver issue 3387 // increase threshold again due to low power audio mode. The way this warning 3388 // threshold is calculated and its usefulness should be reconsidered anyway. 3389 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3390} 3391 3392// ---------------------------------------------------------------------------- 3393 3394AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3395 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3396 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3397 // mLeftVolFloat, mRightVolFloat 3398{ 3399} 3400 3401AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3402 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3403 ThreadBase::type_t type) 3404 : PlaybackThread(audioFlinger, output, id, device, type) 3405 // mLeftVolFloat, mRightVolFloat 3406{ 3407} 3408 3409AudioFlinger::DirectOutputThread::~DirectOutputThread() 3410{ 3411} 3412 3413void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3414{ 3415 audio_track_cblk_t* cblk = track->cblk(); 3416 float left, right; 3417 3418 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3419 left = right = 0; 3420 } else { 3421 float typeVolume = mStreamTypes[track->streamType()].volume; 3422 float v = mMasterVolume * typeVolume; 3423 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3424 uint32_t vlr = proxy->getVolumeLR(); 3425 float v_clamped = v * (vlr & 0xFFFF); 3426 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3427 left = v_clamped/MAX_GAIN; 3428 v_clamped = v * (vlr >> 16); 3429 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3430 right = v_clamped/MAX_GAIN; 3431 } 3432 3433 if (lastTrack) { 3434 if (left != mLeftVolFloat || right != mRightVolFloat) { 3435 mLeftVolFloat = left; 3436 mRightVolFloat = right; 3437 3438 // Convert volumes from float to 8.24 3439 uint32_t vl = (uint32_t)(left * (1 << 24)); 3440 uint32_t vr = (uint32_t)(right * (1 << 24)); 3441 3442 // Delegate volume control to effect in track effect chain if needed 3443 // only one effect chain can be present on DirectOutputThread, so if 3444 // there is one, the track is connected to it 3445 if (!mEffectChains.isEmpty()) { 3446 mEffectChains[0]->setVolume_l(&vl, &vr); 3447 left = (float)vl / (1 << 24); 3448 right = (float)vr / (1 << 24); 3449 } 3450 if (mOutput->stream->set_volume) { 3451 mOutput->stream->set_volume(mOutput->stream, left, right); 3452 } 3453 } 3454 } 3455} 3456 3457 3458AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3459 Vector< sp<Track> > *tracksToRemove 3460) 3461{ 3462 size_t count = mActiveTracks.size(); 3463 mixer_state mixerStatus = MIXER_IDLE; 3464 3465 // find out which tracks need to be processed 3466 for (size_t i = 0; i < count; i++) { 3467 sp<Track> t = mActiveTracks[i].promote(); 3468 // The track died recently 3469 if (t == 0) { 3470 continue; 3471 } 3472 3473 Track* const track = t.get(); 3474 audio_track_cblk_t* cblk = track->cblk(); 3475 3476 // The first time a track is added we wait 3477 // for all its buffers to be filled before processing it 3478 uint32_t minFrames; 3479 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3480 minFrames = mNormalFrameCount; 3481 } else { 3482 minFrames = 1; 3483 } 3484 // Only consider last track started for volume and mixer state control. 3485 // This is the last entry in mActiveTracks unless a track underruns. 3486 // As we only care about the transition phase between two tracks on a 3487 // direct output, it is not a problem to ignore the underrun case. 3488 bool last = (i == (count - 1)); 3489 3490 if ((track->framesReady() >= minFrames) && track->isReady() && 3491 !track->isPaused() && !track->isTerminated()) 3492 { 3493 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3494 3495 if (track->mFillingUpStatus == Track::FS_FILLED) { 3496 track->mFillingUpStatus = Track::FS_ACTIVE; 3497 mLeftVolFloat = mRightVolFloat = 0; 3498 if (track->mState == TrackBase::RESUMING) { 3499 track->mState = TrackBase::ACTIVE; 3500 } 3501 } 3502 3503 // compute volume for this track 3504 processVolume_l(track, last); 3505 if (last) { 3506 // reset retry count 3507 track->mRetryCount = kMaxTrackRetriesDirect; 3508 mActiveTrack = t; 3509 mixerStatus = MIXER_TRACKS_READY; 3510 } 3511 } else { 3512 // clear effect chain input buffer if the last active track started underruns 3513 // to avoid sending previous audio buffer again to effects 3514 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3515 mEffectChains[0]->clearInputBuffer(); 3516 } 3517 3518 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3519 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3520 track->isStopped() || track->isPaused()) { 3521 // We have consumed all the buffers of this track. 3522 // Remove it from the list of active tracks. 3523 // TODO: implement behavior for compressed audio 3524 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3525 size_t framesWritten = mBytesWritten / mFrameSize; 3526 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3527 if (track->isStopped()) { 3528 track->reset(); 3529 } 3530 tracksToRemove->add(track); 3531 } 3532 } else { 3533 // No buffers for this track. Give it a few chances to 3534 // fill a buffer, then remove it from active list. 3535 // Only consider last track started for mixer state control 3536 if (--(track->mRetryCount) <= 0) { 3537 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3538 tracksToRemove->add(track); 3539 } else if (last) { 3540 mixerStatus = MIXER_TRACKS_ENABLED; 3541 } 3542 } 3543 } 3544 } 3545 3546 // remove all the tracks that need to be... 3547 removeTracks_l(*tracksToRemove); 3548 3549 return mixerStatus; 3550} 3551 3552void AudioFlinger::DirectOutputThread::threadLoop_mix() 3553{ 3554 size_t frameCount = mFrameCount; 3555 int8_t *curBuf = (int8_t *)mMixBuffer; 3556 // output audio to hardware 3557 while (frameCount) { 3558 AudioBufferProvider::Buffer buffer; 3559 buffer.frameCount = frameCount; 3560 mActiveTrack->getNextBuffer(&buffer); 3561 if (buffer.raw == NULL) { 3562 memset(curBuf, 0, frameCount * mFrameSize); 3563 break; 3564 } 3565 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3566 frameCount -= buffer.frameCount; 3567 curBuf += buffer.frameCount * mFrameSize; 3568 mActiveTrack->releaseBuffer(&buffer); 3569 } 3570 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3571 sleepTime = 0; 3572 standbyTime = systemTime() + standbyDelay; 3573 mActiveTrack.clear(); 3574} 3575 3576void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3577{ 3578 if (sleepTime == 0) { 3579 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3580 sleepTime = activeSleepTime; 3581 } else { 3582 sleepTime = idleSleepTime; 3583 } 3584 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3585 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3586 sleepTime = 0; 3587 } 3588} 3589 3590// getTrackName_l() must be called with ThreadBase::mLock held 3591int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3592 int sessionId) 3593{ 3594 return 0; 3595} 3596 3597// deleteTrackName_l() must be called with ThreadBase::mLock held 3598void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3599{ 3600} 3601 3602// checkForNewParameters_l() must be called with ThreadBase::mLock held 3603bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3604{ 3605 bool reconfig = false; 3606 3607 while (!mNewParameters.isEmpty()) { 3608 status_t status = NO_ERROR; 3609 String8 keyValuePair = mNewParameters[0]; 3610 AudioParameter param = AudioParameter(keyValuePair); 3611 int value; 3612 3613 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3614 // do not accept frame count changes if tracks are open as the track buffer 3615 // size depends on frame count and correct behavior would not be garantied 3616 // if frame count is changed after track creation 3617 if (!mTracks.isEmpty()) { 3618 status = INVALID_OPERATION; 3619 } else { 3620 reconfig = true; 3621 } 3622 } 3623 if (status == NO_ERROR) { 3624 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3625 keyValuePair.string()); 3626 if (!mStandby && status == INVALID_OPERATION) { 3627 mOutput->stream->common.standby(&mOutput->stream->common); 3628 mStandby = true; 3629 mBytesWritten = 0; 3630 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3631 keyValuePair.string()); 3632 } 3633 if (status == NO_ERROR && reconfig) { 3634 readOutputParameters(); 3635 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3636 } 3637 } 3638 3639 mNewParameters.removeAt(0); 3640 3641 mParamStatus = status; 3642 mParamCond.signal(); 3643 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3644 // already timed out waiting for the status and will never signal the condition. 3645 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3646 } 3647 return reconfig; 3648} 3649 3650uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3651{ 3652 uint32_t time; 3653 if (audio_is_linear_pcm(mFormat)) { 3654 time = PlaybackThread::activeSleepTimeUs(); 3655 } else { 3656 time = 10000; 3657 } 3658 return time; 3659} 3660 3661uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3662{ 3663 uint32_t time; 3664 if (audio_is_linear_pcm(mFormat)) { 3665 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3666 } else { 3667 time = 10000; 3668 } 3669 return time; 3670} 3671 3672uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3673{ 3674 uint32_t time; 3675 if (audio_is_linear_pcm(mFormat)) { 3676 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3677 } else { 3678 time = 10000; 3679 } 3680 return time; 3681} 3682 3683void AudioFlinger::DirectOutputThread::cacheParameters_l() 3684{ 3685 PlaybackThread::cacheParameters_l(); 3686 3687 // use shorter standby delay as on normal output to release 3688 // hardware resources as soon as possible 3689 standbyDelay = microseconds(activeSleepTime*2); 3690} 3691 3692// ---------------------------------------------------------------------------- 3693 3694AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3695 const sp<AudioFlinger::OffloadThread>& offloadThread) 3696 : Thread(false /*canCallJava*/), 3697 mOffloadThread(offloadThread), 3698 mWriteBlocked(false), 3699 mDraining(false) 3700{ 3701} 3702 3703AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3704{ 3705} 3706 3707void AudioFlinger::AsyncCallbackThread::onFirstRef() 3708{ 3709 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3710} 3711 3712bool AudioFlinger::AsyncCallbackThread::threadLoop() 3713{ 3714 while (!exitPending()) { 3715 bool writeBlocked; 3716 bool draining; 3717 3718 { 3719 Mutex::Autolock _l(mLock); 3720 mWaitWorkCV.wait(mLock); 3721 if (exitPending()) { 3722 break; 3723 } 3724 writeBlocked = mWriteBlocked; 3725 draining = mDraining; 3726 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3727 } 3728 { 3729 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3730 if (offloadThread != 0) { 3731 if (writeBlocked == false) { 3732 offloadThread->setWriteBlocked(false); 3733 } 3734 if (draining == false) { 3735 offloadThread->setDraining(false); 3736 } 3737 } 3738 } 3739 } 3740 return false; 3741} 3742 3743void AudioFlinger::AsyncCallbackThread::exit() 3744{ 3745 ALOGV("AsyncCallbackThread::exit"); 3746 Mutex::Autolock _l(mLock); 3747 requestExit(); 3748 mWaitWorkCV.broadcast(); 3749} 3750 3751void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3752{ 3753 Mutex::Autolock _l(mLock); 3754 mWriteBlocked = value; 3755 if (!value) { 3756 mWaitWorkCV.signal(); 3757 } 3758} 3759 3760void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3761{ 3762 Mutex::Autolock _l(mLock); 3763 mDraining = value; 3764 if (!value) { 3765 mWaitWorkCV.signal(); 3766 } 3767} 3768 3769 3770// ---------------------------------------------------------------------------- 3771AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3772 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3773 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3774 mHwPaused(false), 3775 mPausedBytesRemaining(0) 3776{ 3777 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3778} 3779 3780AudioFlinger::OffloadThread::~OffloadThread() 3781{ 3782 mPreviousTrack.clear(); 3783} 3784 3785void AudioFlinger::OffloadThread::threadLoop_exit() 3786{ 3787 if (mFlushPending || mHwPaused) { 3788 // If a flush is pending or track was paused, just discard buffered data 3789 flushHw_l(); 3790 } else { 3791 mMixerStatus = MIXER_DRAIN_ALL; 3792 threadLoop_drain(); 3793 } 3794 mCallbackThread->exit(); 3795 PlaybackThread::threadLoop_exit(); 3796} 3797 3798AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3799 Vector< sp<Track> > *tracksToRemove 3800) 3801{ 3802 ALOGV("OffloadThread::prepareTracks_l"); 3803 size_t count = mActiveTracks.size(); 3804 3805 mixer_state mixerStatus = MIXER_IDLE; 3806 if (mFlushPending) { 3807 flushHw_l(); 3808 mFlushPending = false; 3809 } 3810 // find out which tracks need to be processed 3811 for (size_t i = 0; i < count; i++) { 3812 sp<Track> t = mActiveTracks[i].promote(); 3813 // The track died recently 3814 if (t == 0) { 3815 continue; 3816 } 3817 Track* const track = t.get(); 3818 audio_track_cblk_t* cblk = track->cblk(); 3819 if (mPreviousTrack != NULL) { 3820 if (t != mPreviousTrack) { 3821 // Flush any data still being written from last track 3822 mBytesRemaining = 0; 3823 if (mPausedBytesRemaining) { 3824 // Last track was paused so we also need to flush saved 3825 // mixbuffer state and invalidate track so that it will 3826 // re-submit that unwritten data when it is next resumed 3827 mPausedBytesRemaining = 0; 3828 // Invalidate is a bit drastic - would be more efficient 3829 // to have a flag to tell client that some of the 3830 // previously written data was lost 3831 mPreviousTrack->invalidate(); 3832 } 3833 } 3834 } 3835 mPreviousTrack = t; 3836 bool last = (i == (count - 1)); 3837 if (track->isPausing()) { 3838 track->setPaused(); 3839 if (last) { 3840 if (!mHwPaused) { 3841 mOutput->stream->pause(mOutput->stream); 3842 mHwPaused = true; 3843 } 3844 // If we were part way through writing the mixbuffer to 3845 // the HAL we must save this until we resume 3846 // BUG - this will be wrong if a different track is made active, 3847 // in that case we want to discard the pending data in the 3848 // mixbuffer and tell the client to present it again when the 3849 // track is resumed 3850 mPausedWriteLength = mCurrentWriteLength; 3851 mPausedBytesRemaining = mBytesRemaining; 3852 mBytesRemaining = 0; // stop writing 3853 } 3854 tracksToRemove->add(track); 3855 } else if (track->framesReady() && track->isReady() && 3856 !track->isPaused() && !track->isTerminated()) { 3857 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3858 if (track->mFillingUpStatus == Track::FS_FILLED) { 3859 track->mFillingUpStatus = Track::FS_ACTIVE; 3860 mLeftVolFloat = mRightVolFloat = 0; 3861 if (track->mState == TrackBase::RESUMING) { 3862 if (mPausedBytesRemaining) { 3863 // Need to continue write that was interrupted 3864 mCurrentWriteLength = mPausedWriteLength; 3865 mBytesRemaining = mPausedBytesRemaining; 3866 mPausedBytesRemaining = 0; 3867 } 3868 track->mState = TrackBase::ACTIVE; 3869 } 3870 } 3871 3872 if (last) { 3873 if (mHwPaused) { 3874 mOutput->stream->resume(mOutput->stream); 3875 mHwPaused = false; 3876 // threadLoop_mix() will handle the case that we need to 3877 // resume an interrupted write 3878 } 3879 // reset retry count 3880 track->mRetryCount = kMaxTrackRetriesOffload; 3881 mActiveTrack = t; 3882 mixerStatus = MIXER_TRACKS_READY; 3883 } 3884 } else { 3885 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3886 if (track->isStopping_1()) { 3887 // Hardware buffer can hold a large amount of audio so we must 3888 // wait for all current track's data to drain before we say 3889 // that the track is stopped. 3890 if (mBytesRemaining == 0) { 3891 // Only start draining when all data in mixbuffer 3892 // has been written 3893 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3894 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3895 sleepTime = 0; 3896 standbyTime = systemTime() + standbyDelay; 3897 if (last) { 3898 mixerStatus = MIXER_DRAIN_TRACK; 3899 if (mHwPaused) { 3900 // It is possible to move from PAUSED to STOPPING_1 without 3901 // a resume so we must ensure hardware is running 3902 mOutput->stream->resume(mOutput->stream); 3903 mHwPaused = false; 3904 } 3905 } 3906 } 3907 } else if (track->isStopping_2()) { 3908 // Drain has completed, signal presentation complete 3909 if (!mDraining || !last) { 3910 track->mState = TrackBase::STOPPED; 3911 size_t audioHALFrames = 3912 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3913 size_t framesWritten = 3914 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3915 track->presentationComplete(framesWritten, audioHALFrames); 3916 track->reset(); 3917 tracksToRemove->add(track); 3918 } 3919 } else { 3920 // No buffers for this track. Give it a few chances to 3921 // fill a buffer, then remove it from active list. 3922 if (--(track->mRetryCount) <= 0) { 3923 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3924 track->name()); 3925 tracksToRemove->add(track); 3926 } else if (last){ 3927 mixerStatus = MIXER_TRACKS_ENABLED; 3928 } 3929 } 3930 } 3931 // compute volume for this track 3932 processVolume_l(track, last); 3933 } 3934 // remove all the tracks that need to be... 3935 removeTracks_l(*tracksToRemove); 3936 3937 return mixerStatus; 3938} 3939 3940void AudioFlinger::OffloadThread::flushOutput_l() 3941{ 3942 mFlushPending = true; 3943} 3944 3945// must be called with thread mutex locked 3946bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3947{ 3948 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3949 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3950 return true; 3951 } 3952 return false; 3953} 3954 3955// must be called with thread mutex locked 3956bool AudioFlinger::OffloadThread::shouldStandby_l() 3957{ 3958 bool TrackPaused = false; 3959 3960 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3961 // after a timeout and we will enter standby then. 3962 if (mTracks.size() > 0) { 3963 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3964 } 3965 3966 return !mStandby && !TrackPaused; 3967} 3968 3969 3970bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3971{ 3972 Mutex::Autolock _l(mLock); 3973 return waitingAsyncCallback_l(); 3974} 3975 3976void AudioFlinger::OffloadThread::flushHw_l() 3977{ 3978 mOutput->stream->flush(mOutput->stream); 3979 // Flush anything still waiting in the mixbuffer 3980 mCurrentWriteLength = 0; 3981 mBytesRemaining = 0; 3982 mPausedWriteLength = 0; 3983 mPausedBytesRemaining = 0; 3984 if (mUseAsyncWrite) { 3985 mWriteBlocked = false; 3986 mDraining = false; 3987 ALOG_ASSERT(mCallbackThread != 0); 3988 mCallbackThread->setWriteBlocked(false); 3989 mCallbackThread->setDraining(false); 3990 } 3991} 3992 3993// ---------------------------------------------------------------------------- 3994 3995AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3996 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3997 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3998 DUPLICATING), 3999 mWaitTimeMs(UINT_MAX) 4000{ 4001 addOutputTrack(mainThread); 4002} 4003 4004AudioFlinger::DuplicatingThread::~DuplicatingThread() 4005{ 4006 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4007 mOutputTracks[i]->destroy(); 4008 } 4009} 4010 4011void AudioFlinger::DuplicatingThread::threadLoop_mix() 4012{ 4013 // mix buffers... 4014 if (outputsReady(outputTracks)) { 4015 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4016 } else { 4017 memset(mMixBuffer, 0, mixBufferSize); 4018 } 4019 sleepTime = 0; 4020 writeFrames = mNormalFrameCount; 4021 mCurrentWriteLength = mixBufferSize; 4022 standbyTime = systemTime() + standbyDelay; 4023} 4024 4025void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4026{ 4027 if (sleepTime == 0) { 4028 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4029 sleepTime = activeSleepTime; 4030 } else { 4031 sleepTime = idleSleepTime; 4032 } 4033 } else if (mBytesWritten != 0) { 4034 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4035 writeFrames = mNormalFrameCount; 4036 memset(mMixBuffer, 0, mixBufferSize); 4037 } else { 4038 // flush remaining overflow buffers in output tracks 4039 writeFrames = 0; 4040 } 4041 sleepTime = 0; 4042 } 4043} 4044 4045ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4046{ 4047 for (size_t i = 0; i < outputTracks.size(); i++) { 4048 outputTracks[i]->write(mMixBuffer, writeFrames); 4049 } 4050 return (ssize_t)mixBufferSize; 4051} 4052 4053void AudioFlinger::DuplicatingThread::threadLoop_standby() 4054{ 4055 // DuplicatingThread implements standby by stopping all tracks 4056 for (size_t i = 0; i < outputTracks.size(); i++) { 4057 outputTracks[i]->stop(); 4058 } 4059} 4060 4061void AudioFlinger::DuplicatingThread::saveOutputTracks() 4062{ 4063 outputTracks = mOutputTracks; 4064} 4065 4066void AudioFlinger::DuplicatingThread::clearOutputTracks() 4067{ 4068 outputTracks.clear(); 4069} 4070 4071void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4072{ 4073 Mutex::Autolock _l(mLock); 4074 // FIXME explain this formula 4075 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4076 OutputTrack *outputTrack = new OutputTrack(thread, 4077 this, 4078 mSampleRate, 4079 mFormat, 4080 mChannelMask, 4081 frameCount); 4082 if (outputTrack->cblk() != NULL) { 4083 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4084 mOutputTracks.add(outputTrack); 4085 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4086 updateWaitTime_l(); 4087 } 4088} 4089 4090void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4091{ 4092 Mutex::Autolock _l(mLock); 4093 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4094 if (mOutputTracks[i]->thread() == thread) { 4095 mOutputTracks[i]->destroy(); 4096 mOutputTracks.removeAt(i); 4097 updateWaitTime_l(); 4098 return; 4099 } 4100 } 4101 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4102} 4103 4104// caller must hold mLock 4105void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4106{ 4107 mWaitTimeMs = UINT_MAX; 4108 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4109 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4110 if (strong != 0) { 4111 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4112 if (waitTimeMs < mWaitTimeMs) { 4113 mWaitTimeMs = waitTimeMs; 4114 } 4115 } 4116 } 4117} 4118 4119 4120bool AudioFlinger::DuplicatingThread::outputsReady( 4121 const SortedVector< sp<OutputTrack> > &outputTracks) 4122{ 4123 for (size_t i = 0; i < outputTracks.size(); i++) { 4124 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4125 if (thread == 0) { 4126 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4127 outputTracks[i].get()); 4128 return false; 4129 } 4130 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4131 // see note at standby() declaration 4132 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4133 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4134 thread.get()); 4135 return false; 4136 } 4137 } 4138 return true; 4139} 4140 4141uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4142{ 4143 return (mWaitTimeMs * 1000) / 2; 4144} 4145 4146void AudioFlinger::DuplicatingThread::cacheParameters_l() 4147{ 4148 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4149 updateWaitTime_l(); 4150 4151 MixerThread::cacheParameters_l(); 4152} 4153 4154// ---------------------------------------------------------------------------- 4155// Record 4156// ---------------------------------------------------------------------------- 4157 4158AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4159 AudioStreamIn *input, 4160 uint32_t sampleRate, 4161 audio_channel_mask_t channelMask, 4162 audio_io_handle_t id, 4163 audio_devices_t outDevice, 4164 audio_devices_t inDevice 4165#ifdef TEE_SINK 4166 , const sp<NBAIO_Sink>& teeSink 4167#endif 4168 ) : 4169 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4170 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4171 // mRsmpInIndex set by readInputParameters() 4172 mReqChannelCount(popcount(channelMask)), 4173 mReqSampleRate(sampleRate) 4174 // mBytesRead is only meaningful while active, and so is cleared in start() 4175 // (but might be better to also clear here for dump?) 4176#ifdef TEE_SINK 4177 , mTeeSink(teeSink) 4178#endif 4179{ 4180 snprintf(mName, kNameLength, "AudioIn_%X", id); 4181 4182 readInputParameters(); 4183 4184} 4185 4186 4187AudioFlinger::RecordThread::~RecordThread() 4188{ 4189 delete[] mRsmpInBuffer; 4190 delete mResampler; 4191 delete[] mRsmpOutBuffer; 4192} 4193 4194void AudioFlinger::RecordThread::onFirstRef() 4195{ 4196 run(mName, PRIORITY_URGENT_AUDIO); 4197} 4198 4199bool AudioFlinger::RecordThread::threadLoop() 4200{ 4201 AudioBufferProvider::Buffer buffer; 4202 sp<RecordTrack> activeTrack; 4203 4204 nsecs_t lastWarning = 0; 4205 4206 inputStandBy(); 4207 acquireWakeLock(); 4208 4209 // used to verify we've read at least once before evaluating how many bytes were read 4210 bool readOnce = false; 4211 4212 // start recording 4213 // FIXME Race here: exitPending could become true immediately after testing. 4214 // It is only set to true while mLock held, but we don't hold mLock yet. 4215 // Probably a benign race, but it would be safer to check exitPending with mLock held. 4216 while (!exitPending()) { 4217 4218 processConfigEvents(); 4219 4220 Vector< sp<EffectChain> > effectChains; 4221 { // scope for mLock 4222 Mutex::Autolock _l(mLock); 4223 // return value 'reconfig' is currently unused 4224 bool reconfig = checkForNewParameters_l(); 4225 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4226 standby(); 4227 4228 if (exitPending()) { 4229 break; 4230 } 4231 4232 releaseWakeLock_l(); 4233 ALOGV("RecordThread: loop stopping"); 4234 // go to sleep 4235 mWaitWorkCV.wait(mLock); 4236 ALOGV("RecordThread: loop starting"); 4237 acquireWakeLock_l(); 4238 continue; 4239 } 4240 if (mActiveTrack != 0) { 4241 if (mActiveTrack->isTerminated()) { 4242 removeTrack_l(mActiveTrack); 4243 mActiveTrack.clear(); 4244 } else { 4245 switch (mActiveTrack->mState) { 4246 case TrackBase::PAUSING: 4247 standby(); 4248 mActiveTrack.clear(); 4249 mStartStopCond.broadcast(); 4250 break; 4251 4252 case TrackBase::RESUMING: 4253 if (mReqChannelCount != mActiveTrack->channelCount()) { 4254 mActiveTrack.clear(); 4255 mStartStopCond.broadcast(); 4256 } else if (readOnce) { 4257 // record start succeeds only if first read from audio input 4258 // succeeds 4259 if (mBytesRead >= 0) { 4260 mActiveTrack->mState = TrackBase::ACTIVE; 4261 } else { 4262 mActiveTrack.clear(); 4263 } 4264 mStartStopCond.broadcast(); 4265 } 4266 mStandby = false; 4267 break; 4268 4269 case TrackBase::ACTIVE: 4270 break; 4271 4272 case TrackBase::IDLE: 4273 break; 4274 4275 default: 4276 LOG_FATAL("Unexpected mActiveTrack->mState %d", mActiveTrack->mState); 4277 } 4278 4279 } 4280 } 4281 lockEffectChains_l(effectChains); 4282 } 4283 4284 // thread mutex is now unlocked 4285 // FIXME RecordThread::start assigns to mActiveTrack under lock, but we read without lock 4286 if (mActiveTrack != 0) { 4287 // FIXME RecordThread::stop assigns to mState under lock, but we read without lock 4288 if (mActiveTrack->mState != TrackBase::ACTIVE && 4289 mActiveTrack->mState != TrackBase::RESUMING) { 4290 unlockEffectChains(effectChains); 4291 usleep(kRecordThreadSleepUs); 4292 continue; 4293 } 4294 for (size_t i = 0; i < effectChains.size(); i ++) { 4295 // thread mutex is not locked, but effect chain is locked 4296 effectChains[i]->process_l(); 4297 } 4298 4299 buffer.frameCount = mFrameCount; 4300 status_t status = mActiveTrack->getNextBuffer(&buffer); 4301 if (status == NO_ERROR) { 4302 readOnce = true; 4303 size_t framesOut = buffer.frameCount; 4304 if (mResampler == NULL) { 4305 // no resampling 4306 while (framesOut) { 4307 size_t framesIn = mFrameCount - mRsmpInIndex; 4308 if (framesIn > 0) { 4309 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4310 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4311 mActiveTrack->mFrameSize; 4312 if (framesIn > framesOut) { 4313 framesIn = framesOut; 4314 } 4315 mRsmpInIndex += framesIn; 4316 framesOut -= framesIn; 4317 if (mChannelCount == mReqChannelCount) { 4318 memcpy(dst, src, framesIn * mFrameSize); 4319 } else { 4320 if (mChannelCount == 1) { 4321 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4322 (int16_t *)src, framesIn); 4323 } else { 4324 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4325 (int16_t *)src, framesIn); 4326 } 4327 } 4328 } 4329 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4330 void *readInto; 4331 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4332 readInto = buffer.raw; 4333 framesOut = 0; 4334 } else { 4335 readInto = mRsmpInBuffer; 4336 mRsmpInIndex = 0; 4337 } 4338 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4339 mBufferSize); 4340 if (mBytesRead <= 0) { 4341 // FIXME read mState without lock 4342 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4343 { 4344 ALOGE("Error reading audio input"); 4345 // Force input into standby so that it tries to 4346 // recover at next read attempt 4347 inputStandBy(); 4348 // FIXME sleep with effect chains locked 4349 usleep(kRecordThreadSleepUs); 4350 } 4351 mRsmpInIndex = mFrameCount; 4352 framesOut = 0; 4353 buffer.frameCount = 0; 4354 } 4355#ifdef TEE_SINK 4356 else if (mTeeSink != 0) { 4357 (void) mTeeSink->write(readInto, 4358 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4359 } 4360#endif 4361 } 4362 } 4363 } else { 4364 // resampling 4365 4366 // resampler accumulates, but we only have one source track 4367 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4368 // alter output frame count as if we were expecting stereo samples 4369 if (mChannelCount == 1 && mReqChannelCount == 1) { 4370 framesOut >>= 1; 4371 } 4372 mResampler->resample(mRsmpOutBuffer, framesOut, 4373 this /* AudioBufferProvider* */); 4374 // ditherAndClamp() works as long as all buffers returned by 4375 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4376 if (mChannelCount == 2 && mReqChannelCount == 1) { 4377 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4378 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4379 // the resampler always outputs stereo samples: 4380 // do post stereo to mono conversion 4381 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4382 framesOut); 4383 } else { 4384 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4385 } 4386 // now done with mRsmpOutBuffer 4387 4388 } 4389 if (mFramestoDrop == 0) { 4390 mActiveTrack->releaseBuffer(&buffer); 4391 } else { 4392 if (mFramestoDrop > 0) { 4393 mFramestoDrop -= buffer.frameCount; 4394 if (mFramestoDrop <= 0) { 4395 clearSyncStartEvent(); 4396 } 4397 } else { 4398 mFramestoDrop += buffer.frameCount; 4399 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4400 mSyncStartEvent->isCancelled()) { 4401 ALOGW("Synced record %s, session %d, trigger session %d", 4402 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4403 mActiveTrack->sessionId(), 4404 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4405 clearSyncStartEvent(); 4406 } 4407 } 4408 } 4409 mActiveTrack->clearOverflow(); 4410 } 4411 // client isn't retrieving buffers fast enough 4412 else { 4413 if (!mActiveTrack->setOverflow()) { 4414 nsecs_t now = systemTime(); 4415 if ((now - lastWarning) > kWarningThrottleNs) { 4416 ALOGW("RecordThread: buffer overflow"); 4417 lastWarning = now; 4418 } 4419 } 4420 // Release the processor for a while before asking for a new buffer. 4421 // This will give the application more chance to read from the buffer and 4422 // clear the overflow. 4423 // FIXME sleep with effect chains locked 4424 usleep(kRecordThreadSleepUs); 4425 } 4426 } 4427 // enable changes in effect chain 4428 unlockEffectChains(effectChains); 4429 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4430 } 4431 4432 standby(); 4433 4434 { 4435 Mutex::Autolock _l(mLock); 4436 mActiveTrack.clear(); 4437 mStartStopCond.broadcast(); 4438 } 4439 4440 releaseWakeLock(); 4441 4442 ALOGV("RecordThread %p exiting", this); 4443 return false; 4444} 4445 4446void AudioFlinger::RecordThread::standby() 4447{ 4448 if (!mStandby) { 4449 inputStandBy(); 4450 mStandby = true; 4451 } 4452} 4453 4454void AudioFlinger::RecordThread::inputStandBy() 4455{ 4456 mInput->stream->common.standby(&mInput->stream->common); 4457} 4458 4459sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4460 const sp<AudioFlinger::Client>& client, 4461 uint32_t sampleRate, 4462 audio_format_t format, 4463 audio_channel_mask_t channelMask, 4464 size_t frameCount, 4465 int sessionId, 4466 IAudioFlinger::track_flags_t *flags, 4467 pid_t tid, 4468 status_t *status) 4469{ 4470 sp<RecordTrack> track; 4471 status_t lStatus; 4472 4473 lStatus = initCheck(); 4474 if (lStatus != NO_ERROR) { 4475 ALOGE("Audio driver not initialized."); 4476 goto Exit; 4477 } 4478 4479 // client expresses a preference for FAST, but we get the final say 4480 if (*flags & IAudioFlinger::TRACK_FAST) { 4481 if ( 4482 // use case: callback handler and frame count is default or at least as large as HAL 4483 ( 4484 (tid != -1) && 4485 ((frameCount == 0) || 4486 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4487 ) && 4488 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4489 // mono or stereo 4490 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4491 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4492 // hardware sample rate 4493 (sampleRate == mSampleRate) && 4494 // record thread has an associated fast recorder 4495 hasFastRecorder() 4496 // FIXME test that RecordThread for this fast track has a capable output HAL 4497 // FIXME add a permission test also? 4498 ) { 4499 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4500 if (frameCount == 0) { 4501 frameCount = mFrameCount * kFastTrackMultiplier; 4502 } 4503 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4504 frameCount, mFrameCount); 4505 } else { 4506 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4507 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4508 "hasFastRecorder=%d tid=%d", 4509 frameCount, mFrameCount, format, 4510 audio_is_linear_pcm(format), 4511 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4512 *flags &= ~IAudioFlinger::TRACK_FAST; 4513 // For compatibility with AudioRecord calculation, buffer depth is forced 4514 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4515 // This is probably too conservative, but legacy application code may depend on it. 4516 // If you change this calculation, also review the start threshold which is related. 4517 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4518 size_t mNormalFrameCount = 2048; // FIXME 4519 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4520 if (minBufCount < 2) { 4521 minBufCount = 2; 4522 } 4523 size_t minFrameCount = mNormalFrameCount * minBufCount; 4524 if (frameCount < minFrameCount) { 4525 frameCount = minFrameCount; 4526 } 4527 } 4528 } 4529 4530 // FIXME use flags and tid similar to createTrack_l() 4531 4532 { // scope for mLock 4533 Mutex::Autolock _l(mLock); 4534 4535 track = new RecordTrack(this, client, sampleRate, 4536 format, channelMask, frameCount, sessionId); 4537 4538 lStatus = track->initCheck(); 4539 if (lStatus != NO_ERROR) { 4540 track.clear(); 4541 goto Exit; 4542 } 4543 mTracks.add(track); 4544 4545 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4546 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4547 mAudioFlinger->btNrecIsOff(); 4548 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4549 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4550 4551 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4552 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4553 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4554 // so ask activity manager to do this on our behalf 4555 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4556 } 4557 } 4558 lStatus = NO_ERROR; 4559 4560Exit: 4561 *status = lStatus; 4562 return track; 4563} 4564 4565status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4566 AudioSystem::sync_event_t event, 4567 int triggerSession) 4568{ 4569 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4570 sp<ThreadBase> strongMe = this; 4571 status_t status = NO_ERROR; 4572 4573 if (event == AudioSystem::SYNC_EVENT_NONE) { 4574 clearSyncStartEvent(); 4575 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4576 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4577 triggerSession, 4578 recordTrack->sessionId(), 4579 syncStartEventCallback, 4580 this); 4581 // Sync event can be cancelled by the trigger session if the track is not in a 4582 // compatible state in which case we start record immediately 4583 if (mSyncStartEvent->isCancelled()) { 4584 clearSyncStartEvent(); 4585 } else { 4586 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4587 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4588 } 4589 } 4590 4591 { 4592 // This section is a rendezvous between binder thread executing start() and RecordThread 4593 AutoMutex lock(mLock); 4594 if (mActiveTrack != 0) { 4595 if (recordTrack != mActiveTrack.get()) { 4596 status = -EBUSY; 4597 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4598 mActiveTrack->mState = TrackBase::ACTIVE; 4599 } 4600 return status; 4601 } 4602 4603 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4604 recordTrack->mState = TrackBase::IDLE; 4605 mActiveTrack = recordTrack; 4606 mLock.unlock(); 4607 status_t status = AudioSystem::startInput(mId); 4608 mLock.lock(); 4609 // FIXME should verify that mActiveTrack is still == recordTrack 4610 if (status != NO_ERROR) { 4611 mActiveTrack.clear(); 4612 clearSyncStartEvent(); 4613 return status; 4614 } 4615 mRsmpInIndex = mFrameCount; 4616 mBytesRead = 0; 4617 if (mResampler != NULL) { 4618 mResampler->reset(); 4619 } 4620 // FIXME hijacking a playback track state name which was intended for start after pause; 4621 // here 'STARTING_2' would be more accurate 4622 mActiveTrack->mState = TrackBase::RESUMING; 4623 // signal thread to start 4624 ALOGV("Signal record thread"); 4625 mWaitWorkCV.broadcast(); 4626 // do not wait for mStartStopCond if exiting 4627 if (exitPending()) { 4628 mActiveTrack.clear(); 4629 status = INVALID_OPERATION; 4630 goto startError; 4631 } 4632 // FIXME incorrect usage of wait: no explicit predicate or loop 4633 mStartStopCond.wait(mLock); 4634 if (mActiveTrack == 0) { 4635 ALOGV("Record failed to start"); 4636 status = BAD_VALUE; 4637 goto startError; 4638 } 4639 ALOGV("Record started OK"); 4640 return status; 4641 } 4642 4643startError: 4644 AudioSystem::stopInput(mId); 4645 clearSyncStartEvent(); 4646 return status; 4647} 4648 4649void AudioFlinger::RecordThread::clearSyncStartEvent() 4650{ 4651 if (mSyncStartEvent != 0) { 4652 mSyncStartEvent->cancel(); 4653 } 4654 mSyncStartEvent.clear(); 4655 mFramestoDrop = 0; 4656} 4657 4658void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4659{ 4660 sp<SyncEvent> strongEvent = event.promote(); 4661 4662 if (strongEvent != 0) { 4663 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4664 me->handleSyncStartEvent(strongEvent); 4665 } 4666} 4667 4668void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4669{ 4670 if (event == mSyncStartEvent) { 4671 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4672 // from audio HAL 4673 mFramestoDrop = mFrameCount * 2; 4674 } 4675} 4676 4677bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4678 ALOGV("RecordThread::stop"); 4679 AutoMutex _l(mLock); 4680 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4681 return false; 4682 } 4683 // note that threadLoop may still be processing the track at this point [without lock] 4684 recordTrack->mState = TrackBase::PAUSING; 4685 // do not wait for mStartStopCond if exiting 4686 if (exitPending()) { 4687 return true; 4688 } 4689 // FIXME incorrect usage of wait: no explicit predicate or loop 4690 mStartStopCond.wait(mLock); 4691 // if we have been restarted, recordTrack == mActiveTrack.get() here 4692 if (exitPending() || recordTrack != mActiveTrack.get()) { 4693 ALOGV("Record stopped OK"); 4694 return true; 4695 } 4696 return false; 4697} 4698 4699bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4700{ 4701 return false; 4702} 4703 4704status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4705{ 4706#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4707 if (!isValidSyncEvent(event)) { 4708 return BAD_VALUE; 4709 } 4710 4711 int eventSession = event->triggerSession(); 4712 status_t ret = NAME_NOT_FOUND; 4713 4714 Mutex::Autolock _l(mLock); 4715 4716 for (size_t i = 0; i < mTracks.size(); i++) { 4717 sp<RecordTrack> track = mTracks[i]; 4718 if (eventSession == track->sessionId()) { 4719 (void) track->setSyncEvent(event); 4720 ret = NO_ERROR; 4721 } 4722 } 4723 return ret; 4724#else 4725 return BAD_VALUE; 4726#endif 4727} 4728 4729// destroyTrack_l() must be called with ThreadBase::mLock held 4730void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4731{ 4732 track->terminate(); 4733 track->mState = TrackBase::STOPPED; 4734 // active tracks are removed by threadLoop() 4735 if (mActiveTrack != track) { 4736 removeTrack_l(track); 4737 } 4738} 4739 4740void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4741{ 4742 mTracks.remove(track); 4743 // need anything related to effects here? 4744} 4745 4746void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4747{ 4748 dumpInternals(fd, args); 4749 dumpTracks(fd, args); 4750 dumpEffectChains(fd, args); 4751} 4752 4753void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4754{ 4755 const size_t SIZE = 256; 4756 char buffer[SIZE]; 4757 String8 result; 4758 4759 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4760 result.append(buffer); 4761 4762 if (mActiveTrack != 0) { 4763 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4764 result.append(buffer); 4765 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4766 result.append(buffer); 4767 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4768 result.append(buffer); 4769 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4770 result.append(buffer); 4771 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4772 result.append(buffer); 4773 } else { 4774 result.append("No active record client\n"); 4775 } 4776 4777 write(fd, result.string(), result.size()); 4778 4779 dumpBase(fd, args); 4780} 4781 4782void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4783{ 4784 const size_t SIZE = 256; 4785 char buffer[SIZE]; 4786 String8 result; 4787 4788 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4789 result.append(buffer); 4790 RecordTrack::appendDumpHeader(result); 4791 for (size_t i = 0; i < mTracks.size(); ++i) { 4792 sp<RecordTrack> track = mTracks[i]; 4793 if (track != 0) { 4794 track->dump(buffer, SIZE); 4795 result.append(buffer); 4796 } 4797 } 4798 4799 if (mActiveTrack != 0) { 4800 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4801 result.append(buffer); 4802 RecordTrack::appendDumpHeader(result); 4803 mActiveTrack->dump(buffer, SIZE); 4804 result.append(buffer); 4805 4806 } 4807 write(fd, result.string(), result.size()); 4808} 4809 4810// AudioBufferProvider interface 4811status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4812{ 4813 size_t framesReq = buffer->frameCount; 4814 size_t framesReady = mFrameCount - mRsmpInIndex; 4815 int channelCount; 4816 4817 if (framesReady == 0) { 4818 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4819 if (mBytesRead <= 0) { 4820 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4821 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4822 // Force input into standby so that it tries to 4823 // recover at next read attempt 4824 inputStandBy(); 4825 usleep(kRecordThreadSleepUs); 4826 } 4827 buffer->raw = NULL; 4828 buffer->frameCount = 0; 4829 return NOT_ENOUGH_DATA; 4830 } 4831 mRsmpInIndex = 0; 4832 framesReady = mFrameCount; 4833 } 4834 4835 if (framesReq > framesReady) { 4836 framesReq = framesReady; 4837 } 4838 4839 if (mChannelCount == 1 && mReqChannelCount == 2) { 4840 channelCount = 1; 4841 } else { 4842 channelCount = 2; 4843 } 4844 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4845 buffer->frameCount = framesReq; 4846 return NO_ERROR; 4847} 4848 4849// AudioBufferProvider interface 4850void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4851{ 4852 mRsmpInIndex += buffer->frameCount; 4853 buffer->frameCount = 0; 4854} 4855 4856bool AudioFlinger::RecordThread::checkForNewParameters_l() 4857{ 4858 bool reconfig = false; 4859 4860 while (!mNewParameters.isEmpty()) { 4861 status_t status = NO_ERROR; 4862 String8 keyValuePair = mNewParameters[0]; 4863 AudioParameter param = AudioParameter(keyValuePair); 4864 int value; 4865 audio_format_t reqFormat = mFormat; 4866 uint32_t reqSamplingRate = mReqSampleRate; 4867 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 4868 4869 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4870 reqSamplingRate = value; 4871 reconfig = true; 4872 } 4873 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4874 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4875 status = BAD_VALUE; 4876 } else { 4877 reqFormat = (audio_format_t) value; 4878 reconfig = true; 4879 } 4880 } 4881 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4882 audio_channel_mask_t mask = (audio_channel_mask_t) value; 4883 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 4884 status = BAD_VALUE; 4885 } else { 4886 reqChannelMask = mask; 4887 reconfig = true; 4888 } 4889 } 4890 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4891 // do not accept frame count changes if tracks are open as the track buffer 4892 // size depends on frame count and correct behavior would not be guaranteed 4893 // if frame count is changed after track creation 4894 if (mActiveTrack != 0) { 4895 status = INVALID_OPERATION; 4896 } else { 4897 reconfig = true; 4898 } 4899 } 4900 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4901 // forward device change to effects that have requested to be 4902 // aware of attached audio device. 4903 for (size_t i = 0; i < mEffectChains.size(); i++) { 4904 mEffectChains[i]->setDevice_l(value); 4905 } 4906 4907 // store input device and output device but do not forward output device to audio HAL. 4908 // Note that status is ignored by the caller for output device 4909 // (see AudioFlinger::setParameters() 4910 if (audio_is_output_devices(value)) { 4911 mOutDevice = value; 4912 status = BAD_VALUE; 4913 } else { 4914 mInDevice = value; 4915 // disable AEC and NS if the device is a BT SCO headset supporting those 4916 // pre processings 4917 if (mTracks.size() > 0) { 4918 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4919 mAudioFlinger->btNrecIsOff(); 4920 for (size_t i = 0; i < mTracks.size(); i++) { 4921 sp<RecordTrack> track = mTracks[i]; 4922 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4923 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4924 } 4925 } 4926 } 4927 } 4928 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4929 mAudioSource != (audio_source_t)value) { 4930 // forward device change to effects that have requested to be 4931 // aware of attached audio device. 4932 for (size_t i = 0; i < mEffectChains.size(); i++) { 4933 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4934 } 4935 mAudioSource = (audio_source_t)value; 4936 } 4937 4938 if (status == NO_ERROR) { 4939 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4940 keyValuePair.string()); 4941 if (status == INVALID_OPERATION) { 4942 inputStandBy(); 4943 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4944 keyValuePair.string()); 4945 } 4946 if (reconfig) { 4947 if (status == BAD_VALUE && 4948 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4949 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4950 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4951 <= (2 * reqSamplingRate)) && 4952 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4953 <= FCC_2 && 4954 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 4955 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 4956 status = NO_ERROR; 4957 } 4958 if (status == NO_ERROR) { 4959 readInputParameters(); 4960 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4961 } 4962 } 4963 } 4964 4965 mNewParameters.removeAt(0); 4966 4967 mParamStatus = status; 4968 mParamCond.signal(); 4969 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4970 // already timed out waiting for the status and will never signal the condition. 4971 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4972 } 4973 return reconfig; 4974} 4975 4976String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4977{ 4978 Mutex::Autolock _l(mLock); 4979 if (initCheck() != NO_ERROR) { 4980 return String8(); 4981 } 4982 4983 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4984 const String8 out_s8(s); 4985 free(s); 4986 return out_s8; 4987} 4988 4989void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4990 AudioSystem::OutputDescriptor desc; 4991 void *param2 = NULL; 4992 4993 switch (event) { 4994 case AudioSystem::INPUT_OPENED: 4995 case AudioSystem::INPUT_CONFIG_CHANGED: 4996 desc.channelMask = mChannelMask; 4997 desc.samplingRate = mSampleRate; 4998 desc.format = mFormat; 4999 desc.frameCount = mFrameCount; 5000 desc.latency = 0; 5001 param2 = &desc; 5002 break; 5003 5004 case AudioSystem::INPUT_CLOSED: 5005 default: 5006 break; 5007 } 5008 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5009} 5010 5011void AudioFlinger::RecordThread::readInputParameters() 5012{ 5013 delete[] mRsmpInBuffer; 5014 // mRsmpInBuffer is always assigned a new[] below 5015 delete[] mRsmpOutBuffer; 5016 mRsmpOutBuffer = NULL; 5017 delete mResampler; 5018 mResampler = NULL; 5019 5020 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5021 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5022 mChannelCount = popcount(mChannelMask); 5023 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5024 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5025 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5026 } 5027 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5028 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5029 mFrameCount = mBufferSize / mFrameSize; 5030 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5031 5032 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5033 int channelCount; 5034 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5035 // stereo to mono post process as the resampler always outputs stereo. 5036 if (mChannelCount == 1 && mReqChannelCount == 2) { 5037 channelCount = 1; 5038 } else { 5039 channelCount = 2; 5040 } 5041 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5042 mResampler->setSampleRate(mSampleRate); 5043 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5044 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5045 5046 // optmization: if mono to mono, alter input frame count as if we were inputing 5047 // stereo samples 5048 if (mChannelCount == 1 && mReqChannelCount == 1) { 5049 mFrameCount >>= 1; 5050 } 5051 5052 } 5053 mRsmpInIndex = mFrameCount; 5054} 5055 5056unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5057{ 5058 Mutex::Autolock _l(mLock); 5059 if (initCheck() != NO_ERROR) { 5060 return 0; 5061 } 5062 5063 return mInput->stream->get_input_frames_lost(mInput->stream); 5064} 5065 5066uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5067{ 5068 Mutex::Autolock _l(mLock); 5069 uint32_t result = 0; 5070 if (getEffectChain_l(sessionId) != 0) { 5071 result = EFFECT_SESSION; 5072 } 5073 5074 for (size_t i = 0; i < mTracks.size(); ++i) { 5075 if (sessionId == mTracks[i]->sessionId()) { 5076 result |= TRACK_SESSION; 5077 break; 5078 } 5079 } 5080 5081 return result; 5082} 5083 5084KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5085{ 5086 KeyedVector<int, bool> ids; 5087 Mutex::Autolock _l(mLock); 5088 for (size_t j = 0; j < mTracks.size(); ++j) { 5089 sp<RecordThread::RecordTrack> track = mTracks[j]; 5090 int sessionId = track->sessionId(); 5091 if (ids.indexOfKey(sessionId) < 0) { 5092 ids.add(sessionId, true); 5093 } 5094 } 5095 return ids; 5096} 5097 5098AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5099{ 5100 Mutex::Autolock _l(mLock); 5101 AudioStreamIn *input = mInput; 5102 mInput = NULL; 5103 return input; 5104} 5105 5106// this method must always be called either with ThreadBase mLock held or inside the thread loop 5107audio_stream_t* AudioFlinger::RecordThread::stream() const 5108{ 5109 if (mInput == NULL) { 5110 return NULL; 5111 } 5112 return &mInput->stream->common; 5113} 5114 5115status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5116{ 5117 // only one chain per input thread 5118 if (mEffectChains.size() != 0) { 5119 return INVALID_OPERATION; 5120 } 5121 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5122 5123 chain->setInBuffer(NULL); 5124 chain->setOutBuffer(NULL); 5125 5126 checkSuspendOnAddEffectChain_l(chain); 5127 5128 mEffectChains.add(chain); 5129 5130 return NO_ERROR; 5131} 5132 5133size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5134{ 5135 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5136 ALOGW_IF(mEffectChains.size() != 1, 5137 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5138 chain.get(), mEffectChains.size(), this); 5139 if (mEffectChains.size() == 1) { 5140 mEffectChains.removeAt(0); 5141 } 5142 return 0; 5143} 5144 5145}; // namespace android 5146