Threads.cpp revision dc8cae8c118e4aef4ef1f7b2c6f79becc1df4a05
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 273 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 285 for (size_t i = 0; i < mConfigEvents.size(); i++) { 286 delete mConfigEvents[i]; 287 } 288 mConfigEvents.clear(); 289 290 mParamCond.broadcast(); 291 // do not lock the mutex in destructor 292 releaseWakeLock_l(); 293 if (mPowerManager != 0) { 294 sp<IBinder> binder = mPowerManager->asBinder(); 295 binder->unlinkToDeath(mDeathRecipient); 296 } 297} 298 299status_t AudioFlinger::ThreadBase::readyToRun() 300{ 301 status_t status = initCheck(); 302 if (status == NO_ERROR) { 303 ALOGI("AudioFlinger's thread %p ready to run", this); 304 } else { 305 ALOGE("No working audio driver found."); 306 } 307 return status; 308} 309 310void AudioFlinger::ThreadBase::exit() 311{ 312 ALOGV("ThreadBase::exit"); 313 // do any cleanup required for exit to succeed 314 preExit(); 315 { 316 // This lock prevents the following race in thread (uniprocessor for illustration): 317 // if (!exitPending()) { 318 // // context switch from here to exit() 319 // // exit() calls requestExit(), what exitPending() observes 320 // // exit() calls signal(), which is dropped since no waiters 321 // // context switch back from exit() to here 322 // mWaitWorkCV.wait(...); 323 // // now thread is hung 324 // } 325 AutoMutex lock(mLock); 326 requestExit(); 327 mWaitWorkCV.broadcast(); 328 } 329 // When Thread::requestExitAndWait is made virtual and this method is renamed to 330 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 331 requestExitAndWait(); 332} 333 334status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 335{ 336 status_t status; 337 338 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 339 Mutex::Autolock _l(mLock); 340 341 mNewParameters.add(keyValuePairs); 342 mWaitWorkCV.signal(); 343 // wait condition with timeout in case the thread loop has exited 344 // before the request could be processed 345 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 346 status = mParamStatus; 347 mWaitWorkCV.signal(); 348 } else { 349 status = TIMED_OUT; 350 } 351 return status; 352} 353 354void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 355{ 356 Mutex::Autolock _l(mLock); 357 sendIoConfigEvent_l(event, param); 358} 359 360// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 361void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 362{ 363 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 364 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 365 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 366 param); 367 mWaitWorkCV.signal(); 368} 369 370// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 371void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 372{ 373 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 374 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 375 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 376 mConfigEvents.size(), pid, tid, prio); 377 mWaitWorkCV.signal(); 378} 379 380void AudioFlinger::ThreadBase::processConfigEvents() 381{ 382 Mutex::Autolock _l(mLock); 383 processConfigEvents_l(); 384} 385 386// post condition: mConfigEvents.isEmpty() 387void AudioFlinger::ThreadBase::processConfigEvents_l() 388{ 389 while (!mConfigEvents.isEmpty()) { 390 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 391 ConfigEvent *event = mConfigEvents[0]; 392 mConfigEvents.removeAt(0); 393 // release mLock before locking AudioFlinger mLock: lock order is always 394 // AudioFlinger then ThreadBase to avoid cross deadlock 395 mLock.unlock(); 396 switch (event->type()) { 397 case CFG_EVENT_PRIO: { 398 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 399 // FIXME Need to understand why this has be done asynchronously 400 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 401 true /*asynchronous*/); 402 if (err != 0) { 403 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 404 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 405 } 406 } break; 407 case CFG_EVENT_IO: { 408 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 409 { 410 Mutex::Autolock _l(mAudioFlinger->mLock); 411 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 412 } 413 } break; 414 default: 415 ALOGE("processConfigEvents() unknown event type %d", event->type()); 416 break; 417 } 418 delete event; 419 mLock.lock(); 420 } 421} 422 423void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 424{ 425 const size_t SIZE = 256; 426 char buffer[SIZE]; 427 String8 result; 428 429 bool locked = AudioFlinger::dumpTryLock(mLock); 430 if (!locked) { 431 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 432 write(fd, buffer, strlen(buffer)); 433 } 434 435 snprintf(buffer, SIZE, "io handle: %d\n", mId); 436 result.append(buffer); 437 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 438 result.append(buffer); 439 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 440 result.append(buffer); 441 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 442 result.append(buffer); 443 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 444 result.append(buffer); 445 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 446 result.append(buffer); 447 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 448 result.append(buffer); 449 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 450 result.append(buffer); 451 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 452 result.append(buffer); 453 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 454 result.append(buffer); 455 456 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 457 result.append(buffer); 458 result.append(" Index Command"); 459 for (size_t i = 0; i < mNewParameters.size(); ++i) { 460 snprintf(buffer, SIZE, "\n %02d ", i); 461 result.append(buffer); 462 result.append(mNewParameters[i]); 463 } 464 465 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 466 result.append(buffer); 467 for (size_t i = 0; i < mConfigEvents.size(); i++) { 468 mConfigEvents[i]->dump(buffer, SIZE); 469 result.append(buffer); 470 } 471 result.append("\n"); 472 473 write(fd, result.string(), result.size()); 474 475 if (locked) { 476 mLock.unlock(); 477 } 478} 479 480void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 481{ 482 const size_t SIZE = 256; 483 char buffer[SIZE]; 484 String8 result; 485 486 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 487 write(fd, buffer, strlen(buffer)); 488 489 for (size_t i = 0; i < mEffectChains.size(); ++i) { 490 sp<EffectChain> chain = mEffectChains[i]; 491 if (chain != 0) { 492 chain->dump(fd, args); 493 } 494 } 495} 496 497void AudioFlinger::ThreadBase::acquireWakeLock() 498{ 499 Mutex::Autolock _l(mLock); 500 acquireWakeLock_l(); 501} 502 503void AudioFlinger::ThreadBase::acquireWakeLock_l() 504{ 505 if (mPowerManager == 0) { 506 // use checkService() to avoid blocking if power service is not up yet 507 sp<IBinder> binder = 508 defaultServiceManager()->checkService(String16("power")); 509 if (binder == 0) { 510 ALOGW("Thread %s cannot connect to the power manager service", mName); 511 } else { 512 mPowerManager = interface_cast<IPowerManager>(binder); 513 binder->linkToDeath(mDeathRecipient); 514 } 515 } 516 if (mPowerManager != 0) { 517 sp<IBinder> binder = new BBinder(); 518 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 519 binder, 520 String16(mName), 521 String16("media")); 522 if (status == NO_ERROR) { 523 mWakeLockToken = binder; 524 } 525 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 526 } 527} 528 529void AudioFlinger::ThreadBase::releaseWakeLock() 530{ 531 Mutex::Autolock _l(mLock); 532 releaseWakeLock_l(); 533} 534 535void AudioFlinger::ThreadBase::releaseWakeLock_l() 536{ 537 if (mWakeLockToken != 0) { 538 ALOGV("releaseWakeLock_l() %s", mName); 539 if (mPowerManager != 0) { 540 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 541 } 542 mWakeLockToken.clear(); 543 } 544} 545 546void AudioFlinger::ThreadBase::clearPowerManager() 547{ 548 Mutex::Autolock _l(mLock); 549 releaseWakeLock_l(); 550 mPowerManager.clear(); 551} 552 553void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 554{ 555 sp<ThreadBase> thread = mThread.promote(); 556 if (thread != 0) { 557 thread->clearPowerManager(); 558 } 559 ALOGW("power manager service died !!!"); 560} 561 562void AudioFlinger::ThreadBase::setEffectSuspended( 563 const effect_uuid_t *type, bool suspend, int sessionId) 564{ 565 Mutex::Autolock _l(mLock); 566 setEffectSuspended_l(type, suspend, sessionId); 567} 568 569void AudioFlinger::ThreadBase::setEffectSuspended_l( 570 const effect_uuid_t *type, bool suspend, int sessionId) 571{ 572 sp<EffectChain> chain = getEffectChain_l(sessionId); 573 if (chain != 0) { 574 if (type != NULL) { 575 chain->setEffectSuspended_l(type, suspend); 576 } else { 577 chain->setEffectSuspendedAll_l(suspend); 578 } 579 } 580 581 updateSuspendedSessions_l(type, suspend, sessionId); 582} 583 584void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 585{ 586 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 587 if (index < 0) { 588 return; 589 } 590 591 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 592 mSuspendedSessions.valueAt(index); 593 594 for (size_t i = 0; i < sessionEffects.size(); i++) { 595 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 596 for (int j = 0; j < desc->mRefCount; j++) { 597 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 598 chain->setEffectSuspendedAll_l(true); 599 } else { 600 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 601 desc->mType.timeLow); 602 chain->setEffectSuspended_l(&desc->mType, true); 603 } 604 } 605 } 606} 607 608void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 609 bool suspend, 610 int sessionId) 611{ 612 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 613 614 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 615 616 if (suspend) { 617 if (index >= 0) { 618 sessionEffects = mSuspendedSessions.valueAt(index); 619 } else { 620 mSuspendedSessions.add(sessionId, sessionEffects); 621 } 622 } else { 623 if (index < 0) { 624 return; 625 } 626 sessionEffects = mSuspendedSessions.valueAt(index); 627 } 628 629 630 int key = EffectChain::kKeyForSuspendAll; 631 if (type != NULL) { 632 key = type->timeLow; 633 } 634 index = sessionEffects.indexOfKey(key); 635 636 sp<SuspendedSessionDesc> desc; 637 if (suspend) { 638 if (index >= 0) { 639 desc = sessionEffects.valueAt(index); 640 } else { 641 desc = new SuspendedSessionDesc(); 642 if (type != NULL) { 643 desc->mType = *type; 644 } 645 sessionEffects.add(key, desc); 646 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 647 } 648 desc->mRefCount++; 649 } else { 650 if (index < 0) { 651 return; 652 } 653 desc = sessionEffects.valueAt(index); 654 if (--desc->mRefCount == 0) { 655 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 656 sessionEffects.removeItemsAt(index); 657 if (sessionEffects.isEmpty()) { 658 ALOGV("updateSuspendedSessions_l() restore removing session %d", 659 sessionId); 660 mSuspendedSessions.removeItem(sessionId); 661 } 662 } 663 } 664 if (!sessionEffects.isEmpty()) { 665 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 666 } 667} 668 669void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 670 bool enabled, 671 int sessionId) 672{ 673 Mutex::Autolock _l(mLock); 674 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 675} 676 677void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 678 bool enabled, 679 int sessionId) 680{ 681 if (mType != RECORD) { 682 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 683 // another session. This gives the priority to well behaved effect control panels 684 // and applications not using global effects. 685 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 686 // global effects 687 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 688 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 689 } 690 } 691 692 sp<EffectChain> chain = getEffectChain_l(sessionId); 693 if (chain != 0) { 694 chain->checkSuspendOnEffectEnabled(effect, enabled); 695 } 696} 697 698// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 699sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 700 const sp<AudioFlinger::Client>& client, 701 const sp<IEffectClient>& effectClient, 702 int32_t priority, 703 int sessionId, 704 effect_descriptor_t *desc, 705 int *enabled, 706 status_t *status) 707{ 708 sp<EffectModule> effect; 709 sp<EffectHandle> handle; 710 status_t lStatus; 711 sp<EffectChain> chain; 712 bool chainCreated = false; 713 bool effectCreated = false; 714 bool effectRegistered = false; 715 716 lStatus = initCheck(); 717 if (lStatus != NO_ERROR) { 718 ALOGW("createEffect_l() Audio driver not initialized."); 719 goto Exit; 720 } 721 722 // Allow global effects only on offloaded and mixer threads 723 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 724 switch (mType) { 725 case MIXER: 726 case OFFLOAD: 727 break; 728 case DIRECT: 729 case DUPLICATING: 730 case RECORD: 731 default: 732 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 733 lStatus = BAD_VALUE; 734 goto Exit; 735 } 736 } 737 738 // Only Pre processor effects are allowed on input threads and only on input threads 739 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 740 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 741 desc->name, desc->flags, mType); 742 lStatus = BAD_VALUE; 743 goto Exit; 744 } 745 746 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 747 748 { // scope for mLock 749 Mutex::Autolock _l(mLock); 750 751 // check for existing effect chain with the requested audio session 752 chain = getEffectChain_l(sessionId); 753 if (chain == 0) { 754 // create a new chain for this session 755 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 756 chain = new EffectChain(this, sessionId); 757 addEffectChain_l(chain); 758 chain->setStrategy(getStrategyForSession_l(sessionId)); 759 chainCreated = true; 760 } else { 761 effect = chain->getEffectFromDesc_l(desc); 762 } 763 764 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 765 766 if (effect == 0) { 767 int id = mAudioFlinger->nextUniqueId(); 768 // Check CPU and memory usage 769 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 770 if (lStatus != NO_ERROR) { 771 goto Exit; 772 } 773 effectRegistered = true; 774 // create a new effect module if none present in the chain 775 effect = new EffectModule(this, chain, desc, id, sessionId); 776 lStatus = effect->status(); 777 if (lStatus != NO_ERROR) { 778 goto Exit; 779 } 780 effect->setOffloaded(mType == OFFLOAD, mId); 781 782 lStatus = chain->addEffect_l(effect); 783 if (lStatus != NO_ERROR) { 784 goto Exit; 785 } 786 effectCreated = true; 787 788 effect->setDevice(mOutDevice); 789 effect->setDevice(mInDevice); 790 effect->setMode(mAudioFlinger->getMode()); 791 effect->setAudioSource(mAudioSource); 792 } 793 // create effect handle and connect it to effect module 794 handle = new EffectHandle(effect, client, effectClient, priority); 795 lStatus = effect->addHandle(handle.get()); 796 if (enabled != NULL) { 797 *enabled = (int)effect->isEnabled(); 798 } 799 } 800 801Exit: 802 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 803 Mutex::Autolock _l(mLock); 804 if (effectCreated) { 805 chain->removeEffect_l(effect); 806 } 807 if (effectRegistered) { 808 AudioSystem::unregisterEffect(effect->id()); 809 } 810 if (chainCreated) { 811 removeEffectChain_l(chain); 812 } 813 handle.clear(); 814 } 815 816 *status = lStatus; 817 return handle; 818} 819 820sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 821{ 822 Mutex::Autolock _l(mLock); 823 return getEffect_l(sessionId, effectId); 824} 825 826sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 827{ 828 sp<EffectChain> chain = getEffectChain_l(sessionId); 829 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 830} 831 832// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 833// PlaybackThread::mLock held 834status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 835{ 836 // check for existing effect chain with the requested audio session 837 int sessionId = effect->sessionId(); 838 sp<EffectChain> chain = getEffectChain_l(sessionId); 839 bool chainCreated = false; 840 841 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 842 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 843 this, effect->desc().name, effect->desc().flags); 844 845 if (chain == 0) { 846 // create a new chain for this session 847 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 848 chain = new EffectChain(this, sessionId); 849 addEffectChain_l(chain); 850 chain->setStrategy(getStrategyForSession_l(sessionId)); 851 chainCreated = true; 852 } 853 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 854 855 if (chain->getEffectFromId_l(effect->id()) != 0) { 856 ALOGW("addEffect_l() %p effect %s already present in chain %p", 857 this, effect->desc().name, chain.get()); 858 return BAD_VALUE; 859 } 860 861 effect->setOffloaded(mType == OFFLOAD, mId); 862 863 status_t status = chain->addEffect_l(effect); 864 if (status != NO_ERROR) { 865 if (chainCreated) { 866 removeEffectChain_l(chain); 867 } 868 return status; 869 } 870 871 effect->setDevice(mOutDevice); 872 effect->setDevice(mInDevice); 873 effect->setMode(mAudioFlinger->getMode()); 874 effect->setAudioSource(mAudioSource); 875 return NO_ERROR; 876} 877 878void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 879 880 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 881 effect_descriptor_t desc = effect->desc(); 882 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 883 detachAuxEffect_l(effect->id()); 884 } 885 886 sp<EffectChain> chain = effect->chain().promote(); 887 if (chain != 0) { 888 // remove effect chain if removing last effect 889 if (chain->removeEffect_l(effect) == 0) { 890 removeEffectChain_l(chain); 891 } 892 } else { 893 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 894 } 895} 896 897void AudioFlinger::ThreadBase::lockEffectChains_l( 898 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 899{ 900 effectChains = mEffectChains; 901 for (size_t i = 0; i < mEffectChains.size(); i++) { 902 mEffectChains[i]->lock(); 903 } 904} 905 906void AudioFlinger::ThreadBase::unlockEffectChains( 907 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 908{ 909 for (size_t i = 0; i < effectChains.size(); i++) { 910 effectChains[i]->unlock(); 911 } 912} 913 914sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 915{ 916 Mutex::Autolock _l(mLock); 917 return getEffectChain_l(sessionId); 918} 919 920sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 921{ 922 size_t size = mEffectChains.size(); 923 for (size_t i = 0; i < size; i++) { 924 if (mEffectChains[i]->sessionId() == sessionId) { 925 return mEffectChains[i]; 926 } 927 } 928 return 0; 929} 930 931void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 932{ 933 Mutex::Autolock _l(mLock); 934 size_t size = mEffectChains.size(); 935 for (size_t i = 0; i < size; i++) { 936 mEffectChains[i]->setMode_l(mode); 937 } 938} 939 940void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 941 EffectHandle *handle, 942 bool unpinIfLast) { 943 944 Mutex::Autolock _l(mLock); 945 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 946 // delete the effect module if removing last handle on it 947 if (effect->removeHandle(handle) == 0) { 948 if (!effect->isPinned() || unpinIfLast) { 949 removeEffect_l(effect); 950 AudioSystem::unregisterEffect(effect->id()); 951 } 952 } 953} 954 955// ---------------------------------------------------------------------------- 956// Playback 957// ---------------------------------------------------------------------------- 958 959AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 960 AudioStreamOut* output, 961 audio_io_handle_t id, 962 audio_devices_t device, 963 type_t type) 964 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 965 mNormalFrameCount(0), mMixBuffer(NULL), 966 mSuspended(0), mBytesWritten(0), 967 // mStreamTypes[] initialized in constructor body 968 mOutput(output), 969 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 970 mMixerStatus(MIXER_IDLE), 971 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 972 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 973 mBytesRemaining(0), 974 mCurrentWriteLength(0), 975 mUseAsyncWrite(false), 976 mWriteAckSequence(0), 977 mDrainSequence(0), 978 mScreenState(AudioFlinger::mScreenState), 979 // index 0 is reserved for normal mixer's submix 980 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 981 // mLatchD, mLatchQ, 982 mLatchDValid(false), mLatchQValid(false) 983{ 984 snprintf(mName, kNameLength, "AudioOut_%X", id); 985 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 986 987 // Assumes constructor is called by AudioFlinger with it's mLock held, but 988 // it would be safer to explicitly pass initial masterVolume/masterMute as 989 // parameter. 990 // 991 // If the HAL we are using has support for master volume or master mute, 992 // then do not attenuate or mute during mixing (just leave the volume at 1.0 993 // and the mute set to false). 994 mMasterVolume = audioFlinger->masterVolume_l(); 995 mMasterMute = audioFlinger->masterMute_l(); 996 if (mOutput && mOutput->audioHwDev) { 997 if (mOutput->audioHwDev->canSetMasterVolume()) { 998 mMasterVolume = 1.0; 999 } 1000 1001 if (mOutput->audioHwDev->canSetMasterMute()) { 1002 mMasterMute = false; 1003 } 1004 } 1005 1006 readOutputParameters(); 1007 1008 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1009 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1010 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1011 stream = (audio_stream_type_t) (stream + 1)) { 1012 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1013 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1014 } 1015 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1016 // because mAudioFlinger doesn't have one to copy from 1017} 1018 1019AudioFlinger::PlaybackThread::~PlaybackThread() 1020{ 1021 mAudioFlinger->unregisterWriter(mNBLogWriter); 1022 delete[] mMixBuffer; 1023} 1024 1025void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1026{ 1027 dumpInternals(fd, args); 1028 dumpTracks(fd, args); 1029 dumpEffectChains(fd, args); 1030} 1031 1032void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1033{ 1034 const size_t SIZE = 256; 1035 char buffer[SIZE]; 1036 String8 result; 1037 1038 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1039 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1040 const stream_type_t *st = &mStreamTypes[i]; 1041 if (i > 0) { 1042 result.appendFormat(", "); 1043 } 1044 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1045 if (st->mute) { 1046 result.append("M"); 1047 } 1048 } 1049 result.append("\n"); 1050 write(fd, result.string(), result.length()); 1051 result.clear(); 1052 1053 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1054 result.append(buffer); 1055 Track::appendDumpHeader(result); 1056 for (size_t i = 0; i < mTracks.size(); ++i) { 1057 sp<Track> track = mTracks[i]; 1058 if (track != 0) { 1059 track->dump(buffer, SIZE); 1060 result.append(buffer); 1061 } 1062 } 1063 1064 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1065 result.append(buffer); 1066 Track::appendDumpHeader(result); 1067 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1068 sp<Track> track = mActiveTracks[i].promote(); 1069 if (track != 0) { 1070 track->dump(buffer, SIZE); 1071 result.append(buffer); 1072 } 1073 } 1074 write(fd, result.string(), result.size()); 1075 1076 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1077 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1078 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1079 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1080} 1081 1082void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1083{ 1084 const size_t SIZE = 256; 1085 char buffer[SIZE]; 1086 String8 result; 1087 1088 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1089 result.append(buffer); 1090 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1091 result.append(buffer); 1092 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1093 ns2ms(systemTime() - mLastWriteTime)); 1094 result.append(buffer); 1095 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1096 result.append(buffer); 1097 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1098 result.append(buffer); 1099 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1100 result.append(buffer); 1101 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1102 result.append(buffer); 1103 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1104 result.append(buffer); 1105 write(fd, result.string(), result.size()); 1106 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1107 1108 dumpBase(fd, args); 1109} 1110 1111// Thread virtuals 1112 1113void AudioFlinger::PlaybackThread::onFirstRef() 1114{ 1115 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1116} 1117 1118// ThreadBase virtuals 1119void AudioFlinger::PlaybackThread::preExit() 1120{ 1121 ALOGV(" preExit()"); 1122 // FIXME this is using hard-coded strings but in the future, this functionality will be 1123 // converted to use audio HAL extensions required to support tunneling 1124 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1125} 1126 1127// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1128sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1129 const sp<AudioFlinger::Client>& client, 1130 audio_stream_type_t streamType, 1131 uint32_t sampleRate, 1132 audio_format_t format, 1133 audio_channel_mask_t channelMask, 1134 size_t frameCount, 1135 const sp<IMemory>& sharedBuffer, 1136 int sessionId, 1137 IAudioFlinger::track_flags_t *flags, 1138 pid_t tid, 1139 status_t *status) 1140{ 1141 sp<Track> track; 1142 status_t lStatus; 1143 1144 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1145 1146 // client expresses a preference for FAST, but we get the final say 1147 if (*flags & IAudioFlinger::TRACK_FAST) { 1148 if ( 1149 // not timed 1150 (!isTimed) && 1151 // either of these use cases: 1152 ( 1153 // use case 1: shared buffer with any frame count 1154 ( 1155 (sharedBuffer != 0) 1156 ) || 1157 // use case 2: callback handler and frame count is default or at least as large as HAL 1158 ( 1159 (tid != -1) && 1160 ((frameCount == 0) || 1161 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1162 ) 1163 ) && 1164 // PCM data 1165 audio_is_linear_pcm(format) && 1166 // mono or stereo 1167 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1168 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1169#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1170 // hardware sample rate 1171 (sampleRate == mSampleRate) && 1172#endif 1173 // normal mixer has an associated fast mixer 1174 hasFastMixer() && 1175 // there are sufficient fast track slots available 1176 (mFastTrackAvailMask != 0) 1177 // FIXME test that MixerThread for this fast track has a capable output HAL 1178 // FIXME add a permission test also? 1179 ) { 1180 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1181 if (frameCount == 0) { 1182 frameCount = mFrameCount * kFastTrackMultiplier; 1183 } 1184 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1185 frameCount, mFrameCount); 1186 } else { 1187 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1188 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1189 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1190 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1191 audio_is_linear_pcm(format), 1192 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1193 *flags &= ~IAudioFlinger::TRACK_FAST; 1194 // For compatibility with AudioTrack calculation, buffer depth is forced 1195 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1196 // This is probably too conservative, but legacy application code may depend on it. 1197 // If you change this calculation, also review the start threshold which is related. 1198 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1199 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1200 if (minBufCount < 2) { 1201 minBufCount = 2; 1202 } 1203 size_t minFrameCount = mNormalFrameCount * minBufCount; 1204 if (frameCount < minFrameCount) { 1205 frameCount = minFrameCount; 1206 } 1207 } 1208 } 1209 1210 if (mType == DIRECT) { 1211 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1212 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1213 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1214 "for output %p with format %d", 1215 sampleRate, format, channelMask, mOutput, mFormat); 1216 lStatus = BAD_VALUE; 1217 goto Exit; 1218 } 1219 } 1220 } else if (mType == OFFLOAD) { 1221 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1222 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1223 "for output %p with format %d", 1224 sampleRate, format, channelMask, mOutput, mFormat); 1225 lStatus = BAD_VALUE; 1226 goto Exit; 1227 } 1228 } else { 1229 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1230 ALOGE("createTrack_l() Bad parameter: format %d \"" 1231 "for output %p with format %d", 1232 format, mOutput, mFormat); 1233 lStatus = BAD_VALUE; 1234 goto Exit; 1235 } 1236 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1237 if (sampleRate > mSampleRate*2) { 1238 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1239 lStatus = BAD_VALUE; 1240 goto Exit; 1241 } 1242 } 1243 1244 lStatus = initCheck(); 1245 if (lStatus != NO_ERROR) { 1246 ALOGE("Audio driver not initialized."); 1247 goto Exit; 1248 } 1249 1250 { // scope for mLock 1251 Mutex::Autolock _l(mLock); 1252 1253 // all tracks in same audio session must share the same routing strategy otherwise 1254 // conflicts will happen when tracks are moved from one output to another by audio policy 1255 // manager 1256 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1257 for (size_t i = 0; i < mTracks.size(); ++i) { 1258 sp<Track> t = mTracks[i]; 1259 if (t != 0 && !t->isOutputTrack()) { 1260 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1261 if (sessionId == t->sessionId() && strategy != actual) { 1262 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1263 strategy, actual); 1264 lStatus = BAD_VALUE; 1265 goto Exit; 1266 } 1267 } 1268 } 1269 1270 if (!isTimed) { 1271 track = new Track(this, client, streamType, sampleRate, format, 1272 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1273 } else { 1274 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1275 channelMask, frameCount, sharedBuffer, sessionId); 1276 } 1277 1278 // new Track always returns non-NULL, 1279 // but TimedTrack::create() is a factory that could fail by returning NULL 1280 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1281 if (lStatus != NO_ERROR) { 1282 track.clear(); 1283 goto Exit; 1284 } 1285 1286 mTracks.add(track); 1287 1288 sp<EffectChain> chain = getEffectChain_l(sessionId); 1289 if (chain != 0) { 1290 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1291 track->setMainBuffer(chain->inBuffer()); 1292 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1293 chain->incTrackCnt(); 1294 } 1295 1296 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1297 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1298 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1299 // so ask activity manager to do this on our behalf 1300 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1301 } 1302 } 1303 1304 lStatus = NO_ERROR; 1305 1306Exit: 1307 *status = lStatus; 1308 return track; 1309} 1310 1311uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1312{ 1313 return latency; 1314} 1315 1316uint32_t AudioFlinger::PlaybackThread::latency() const 1317{ 1318 Mutex::Autolock _l(mLock); 1319 return latency_l(); 1320} 1321uint32_t AudioFlinger::PlaybackThread::latency_l() const 1322{ 1323 if (initCheck() == NO_ERROR) { 1324 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1325 } else { 1326 return 0; 1327 } 1328} 1329 1330void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1331{ 1332 Mutex::Autolock _l(mLock); 1333 // Don't apply master volume in SW if our HAL can do it for us. 1334 if (mOutput && mOutput->audioHwDev && 1335 mOutput->audioHwDev->canSetMasterVolume()) { 1336 mMasterVolume = 1.0; 1337 } else { 1338 mMasterVolume = value; 1339 } 1340} 1341 1342void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1343{ 1344 Mutex::Autolock _l(mLock); 1345 // Don't apply master mute in SW if our HAL can do it for us. 1346 if (mOutput && mOutput->audioHwDev && 1347 mOutput->audioHwDev->canSetMasterMute()) { 1348 mMasterMute = false; 1349 } else { 1350 mMasterMute = muted; 1351 } 1352} 1353 1354void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1355{ 1356 Mutex::Autolock _l(mLock); 1357 mStreamTypes[stream].volume = value; 1358 signal_l(); 1359} 1360 1361void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1362{ 1363 Mutex::Autolock _l(mLock); 1364 mStreamTypes[stream].mute = muted; 1365 signal_l(); 1366} 1367 1368float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1369{ 1370 Mutex::Autolock _l(mLock); 1371 return mStreamTypes[stream].volume; 1372} 1373 1374// addTrack_l() must be called with ThreadBase::mLock held 1375status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1376{ 1377 status_t status = ALREADY_EXISTS; 1378 1379 // set retry count for buffer fill 1380 track->mRetryCount = kMaxTrackStartupRetries; 1381 if (mActiveTracks.indexOf(track) < 0) { 1382 // the track is newly added, make sure it fills up all its 1383 // buffers before playing. This is to ensure the client will 1384 // effectively get the latency it requested. 1385 if (!track->isOutputTrack()) { 1386 TrackBase::track_state state = track->mState; 1387 mLock.unlock(); 1388 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1389 mLock.lock(); 1390 // abort track was stopped/paused while we released the lock 1391 if (state != track->mState) { 1392 if (status == NO_ERROR) { 1393 mLock.unlock(); 1394 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1395 mLock.lock(); 1396 } 1397 return INVALID_OPERATION; 1398 } 1399 // abort if start is rejected by audio policy manager 1400 if (status != NO_ERROR) { 1401 return PERMISSION_DENIED; 1402 } 1403#ifdef ADD_BATTERY_DATA 1404 // to track the speaker usage 1405 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1406#endif 1407 } 1408 1409 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1410 track->mResetDone = false; 1411 track->mPresentationCompleteFrames = 0; 1412 mActiveTracks.add(track); 1413 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1414 if (chain != 0) { 1415 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1416 track->sessionId()); 1417 chain->incActiveTrackCnt(); 1418 } 1419 1420 status = NO_ERROR; 1421 } 1422 1423 ALOGV("mWaitWorkCV.broadcast"); 1424 mWaitWorkCV.broadcast(); 1425 1426 return status; 1427} 1428 1429bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1430{ 1431 track->terminate(); 1432 // active tracks are removed by threadLoop() 1433 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1434 track->mState = TrackBase::STOPPED; 1435 if (!trackActive) { 1436 removeTrack_l(track); 1437 } else if (track->isFastTrack() || track->isOffloaded()) { 1438 track->mState = TrackBase::STOPPING_1; 1439 } 1440 1441 return trackActive; 1442} 1443 1444void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1445{ 1446 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1447 mTracks.remove(track); 1448 deleteTrackName_l(track->name()); 1449 // redundant as track is about to be destroyed, for dumpsys only 1450 track->mName = -1; 1451 if (track->isFastTrack()) { 1452 int index = track->mFastIndex; 1453 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1454 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1455 mFastTrackAvailMask |= 1 << index; 1456 // redundant as track is about to be destroyed, for dumpsys only 1457 track->mFastIndex = -1; 1458 } 1459 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1460 if (chain != 0) { 1461 chain->decTrackCnt(); 1462 } 1463} 1464 1465void AudioFlinger::PlaybackThread::signal_l() 1466{ 1467 // Thread could be blocked waiting for async 1468 // so signal it to handle state changes immediately 1469 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1470 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1471 mSignalPending = true; 1472 mWaitWorkCV.signal(); 1473} 1474 1475String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1476{ 1477 Mutex::Autolock _l(mLock); 1478 if (initCheck() != NO_ERROR) { 1479 return String8(); 1480 } 1481 1482 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1483 const String8 out_s8(s); 1484 free(s); 1485 return out_s8; 1486} 1487 1488// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1489void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1490 AudioSystem::OutputDescriptor desc; 1491 void *param2 = NULL; 1492 1493 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1494 param); 1495 1496 switch (event) { 1497 case AudioSystem::OUTPUT_OPENED: 1498 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1499 desc.channelMask = mChannelMask; 1500 desc.samplingRate = mSampleRate; 1501 desc.format = mFormat; 1502 desc.frameCount = mNormalFrameCount; // FIXME see 1503 // AudioFlinger::frameCount(audio_io_handle_t) 1504 desc.latency = latency(); 1505 param2 = &desc; 1506 break; 1507 1508 case AudioSystem::STREAM_CONFIG_CHANGED: 1509 param2 = ¶m; 1510 case AudioSystem::OUTPUT_CLOSED: 1511 default: 1512 break; 1513 } 1514 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1515} 1516 1517void AudioFlinger::PlaybackThread::writeCallback() 1518{ 1519 ALOG_ASSERT(mCallbackThread != 0); 1520 mCallbackThread->resetWriteBlocked(); 1521} 1522 1523void AudioFlinger::PlaybackThread::drainCallback() 1524{ 1525 ALOG_ASSERT(mCallbackThread != 0); 1526 mCallbackThread->resetDraining(); 1527} 1528 1529void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1530{ 1531 Mutex::Autolock _l(mLock); 1532 // reject out of sequence requests 1533 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1534 mWriteAckSequence &= ~1; 1535 mWaitWorkCV.signal(); 1536 } 1537} 1538 1539void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1540{ 1541 Mutex::Autolock _l(mLock); 1542 // reject out of sequence requests 1543 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1544 mDrainSequence &= ~1; 1545 mWaitWorkCV.signal(); 1546 } 1547} 1548 1549// static 1550int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1551 void *param, 1552 void *cookie) 1553{ 1554 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1555 ALOGV("asyncCallback() event %d", event); 1556 switch (event) { 1557 case STREAM_CBK_EVENT_WRITE_READY: 1558 me->writeCallback(); 1559 break; 1560 case STREAM_CBK_EVENT_DRAIN_READY: 1561 me->drainCallback(); 1562 break; 1563 default: 1564 ALOGW("asyncCallback() unknown event %d", event); 1565 break; 1566 } 1567 return 0; 1568} 1569 1570void AudioFlinger::PlaybackThread::readOutputParameters() 1571{ 1572 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1573 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1574 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1575 if (!audio_is_output_channel(mChannelMask)) { 1576 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1577 } 1578 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1579 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1580 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1581 } 1582 mChannelCount = popcount(mChannelMask); 1583 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1584 if (!audio_is_valid_format(mFormat)) { 1585 LOG_FATAL("HAL format %d not valid for output", mFormat); 1586 } 1587 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1588 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1589 mFormat); 1590 } 1591 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1592 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1593 mFrameCount = mBufferSize / mFrameSize; 1594 if (mFrameCount & 15) { 1595 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1596 mFrameCount); 1597 } 1598 1599 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1600 (mOutput->stream->set_callback != NULL)) { 1601 if (mOutput->stream->set_callback(mOutput->stream, 1602 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1603 mUseAsyncWrite = true; 1604 } 1605 } 1606 1607 // Calculate size of normal mix buffer relative to the HAL output buffer size 1608 double multiplier = 1.0; 1609 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1610 kUseFastMixer == FastMixer_Dynamic)) { 1611 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1612 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1613 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1614 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1615 maxNormalFrameCount = maxNormalFrameCount & ~15; 1616 if (maxNormalFrameCount < minNormalFrameCount) { 1617 maxNormalFrameCount = minNormalFrameCount; 1618 } 1619 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1620 if (multiplier <= 1.0) { 1621 multiplier = 1.0; 1622 } else if (multiplier <= 2.0) { 1623 if (2 * mFrameCount <= maxNormalFrameCount) { 1624 multiplier = 2.0; 1625 } else { 1626 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1627 } 1628 } else { 1629 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1630 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1631 // track, but we sometimes have to do this to satisfy the maximum frame count 1632 // constraint) 1633 // FIXME this rounding up should not be done if no HAL SRC 1634 uint32_t truncMult = (uint32_t) multiplier; 1635 if ((truncMult & 1)) { 1636 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1637 ++truncMult; 1638 } 1639 } 1640 multiplier = (double) truncMult; 1641 } 1642 } 1643 mNormalFrameCount = multiplier * mFrameCount; 1644 // round up to nearest 16 frames to satisfy AudioMixer 1645 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1646 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1647 mNormalFrameCount); 1648 1649 delete[] mMixBuffer; 1650 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1651 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1652 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1653 memset(mMixBuffer, 0, normalBufferSize); 1654 1655 // force reconfiguration of effect chains and engines to take new buffer size and audio 1656 // parameters into account 1657 // Note that mLock is not held when readOutputParameters() is called from the constructor 1658 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1659 // matter. 1660 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1661 Vector< sp<EffectChain> > effectChains = mEffectChains; 1662 for (size_t i = 0; i < effectChains.size(); i ++) { 1663 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1664 } 1665} 1666 1667 1668status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1669{ 1670 if (halFrames == NULL || dspFrames == NULL) { 1671 return BAD_VALUE; 1672 } 1673 Mutex::Autolock _l(mLock); 1674 if (initCheck() != NO_ERROR) { 1675 return INVALID_OPERATION; 1676 } 1677 size_t framesWritten = mBytesWritten / mFrameSize; 1678 *halFrames = framesWritten; 1679 1680 if (isSuspended()) { 1681 // return an estimation of rendered frames when the output is suspended 1682 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1683 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1684 return NO_ERROR; 1685 } else { 1686 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1687 } 1688} 1689 1690uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1691{ 1692 Mutex::Autolock _l(mLock); 1693 uint32_t result = 0; 1694 if (getEffectChain_l(sessionId) != 0) { 1695 result = EFFECT_SESSION; 1696 } 1697 1698 for (size_t i = 0; i < mTracks.size(); ++i) { 1699 sp<Track> track = mTracks[i]; 1700 if (sessionId == track->sessionId() && !track->isInvalid()) { 1701 result |= TRACK_SESSION; 1702 break; 1703 } 1704 } 1705 1706 return result; 1707} 1708 1709uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1710{ 1711 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1712 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1713 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1714 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1715 } 1716 for (size_t i = 0; i < mTracks.size(); i++) { 1717 sp<Track> track = mTracks[i]; 1718 if (sessionId == track->sessionId() && !track->isInvalid()) { 1719 return AudioSystem::getStrategyForStream(track->streamType()); 1720 } 1721 } 1722 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1723} 1724 1725 1726AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1727{ 1728 Mutex::Autolock _l(mLock); 1729 return mOutput; 1730} 1731 1732AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1733{ 1734 Mutex::Autolock _l(mLock); 1735 AudioStreamOut *output = mOutput; 1736 mOutput = NULL; 1737 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1738 // must push a NULL and wait for ack 1739 mOutputSink.clear(); 1740 mPipeSink.clear(); 1741 mNormalSink.clear(); 1742 return output; 1743} 1744 1745// this method must always be called either with ThreadBase mLock held or inside the thread loop 1746audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1747{ 1748 if (mOutput == NULL) { 1749 return NULL; 1750 } 1751 return &mOutput->stream->common; 1752} 1753 1754uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1755{ 1756 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1757} 1758 1759status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1760{ 1761 if (!isValidSyncEvent(event)) { 1762 return BAD_VALUE; 1763 } 1764 1765 Mutex::Autolock _l(mLock); 1766 1767 for (size_t i = 0; i < mTracks.size(); ++i) { 1768 sp<Track> track = mTracks[i]; 1769 if (event->triggerSession() == track->sessionId()) { 1770 (void) track->setSyncEvent(event); 1771 return NO_ERROR; 1772 } 1773 } 1774 1775 return NAME_NOT_FOUND; 1776} 1777 1778bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1779{ 1780 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1781} 1782 1783void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1784 const Vector< sp<Track> >& tracksToRemove) 1785{ 1786 size_t count = tracksToRemove.size(); 1787 if (count > 0) { 1788 for (size_t i = 0 ; i < count ; i++) { 1789 const sp<Track>& track = tracksToRemove.itemAt(i); 1790 if (!track->isOutputTrack()) { 1791 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1792#ifdef ADD_BATTERY_DATA 1793 // to track the speaker usage 1794 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1795#endif 1796 if (track->isTerminated()) { 1797 AudioSystem::releaseOutput(mId); 1798 } 1799 } 1800 } 1801 } 1802} 1803 1804void AudioFlinger::PlaybackThread::checkSilentMode_l() 1805{ 1806 if (!mMasterMute) { 1807 char value[PROPERTY_VALUE_MAX]; 1808 if (property_get("ro.audio.silent", value, "0") > 0) { 1809 char *endptr; 1810 unsigned long ul = strtoul(value, &endptr, 0); 1811 if (*endptr == '\0' && ul != 0) { 1812 ALOGD("Silence is golden"); 1813 // The setprop command will not allow a property to be changed after 1814 // the first time it is set, so we don't have to worry about un-muting. 1815 setMasterMute_l(true); 1816 } 1817 } 1818 } 1819} 1820 1821// shared by MIXER and DIRECT, overridden by DUPLICATING 1822ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1823{ 1824 // FIXME rewrite to reduce number of system calls 1825 mLastWriteTime = systemTime(); 1826 mInWrite = true; 1827 ssize_t bytesWritten; 1828 1829 // If an NBAIO sink is present, use it to write the normal mixer's submix 1830 if (mNormalSink != 0) { 1831#define mBitShift 2 // FIXME 1832 size_t count = mBytesRemaining >> mBitShift; 1833 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1834 ATRACE_BEGIN("write"); 1835 // update the setpoint when AudioFlinger::mScreenState changes 1836 uint32_t screenState = AudioFlinger::mScreenState; 1837 if (screenState != mScreenState) { 1838 mScreenState = screenState; 1839 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1840 if (pipe != NULL) { 1841 pipe->setAvgFrames((mScreenState & 1) ? 1842 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1843 } 1844 } 1845 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1846 ATRACE_END(); 1847 if (framesWritten > 0) { 1848 bytesWritten = framesWritten << mBitShift; 1849 } else { 1850 bytesWritten = framesWritten; 1851 } 1852 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1853 if (status == NO_ERROR) { 1854 size_t totalFramesWritten = mNormalSink->framesWritten(); 1855 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1856 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1857 mLatchDValid = true; 1858 } 1859 } 1860 // otherwise use the HAL / AudioStreamOut directly 1861 } else { 1862 // Direct output and offload threads 1863 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1864 if (mUseAsyncWrite) { 1865 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1866 mWriteAckSequence += 2; 1867 mWriteAckSequence |= 1; 1868 ALOG_ASSERT(mCallbackThread != 0); 1869 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1870 } 1871 // FIXME We should have an implementation of timestamps for direct output threads. 1872 // They are used e.g for multichannel PCM playback over HDMI. 1873 bytesWritten = mOutput->stream->write(mOutput->stream, 1874 mMixBuffer + offset, mBytesRemaining); 1875 if (mUseAsyncWrite && 1876 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1877 // do not wait for async callback in case of error of full write 1878 mWriteAckSequence &= ~1; 1879 ALOG_ASSERT(mCallbackThread != 0); 1880 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1881 } 1882 } 1883 1884 mNumWrites++; 1885 mInWrite = false; 1886 1887 return bytesWritten; 1888} 1889 1890void AudioFlinger::PlaybackThread::threadLoop_drain() 1891{ 1892 if (mOutput->stream->drain) { 1893 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1894 if (mUseAsyncWrite) { 1895 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1896 mDrainSequence |= 1; 1897 ALOG_ASSERT(mCallbackThread != 0); 1898 mCallbackThread->setDraining(mDrainSequence); 1899 } 1900 mOutput->stream->drain(mOutput->stream, 1901 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1902 : AUDIO_DRAIN_ALL); 1903 } 1904} 1905 1906void AudioFlinger::PlaybackThread::threadLoop_exit() 1907{ 1908 // Default implementation has nothing to do 1909} 1910 1911/* 1912The derived values that are cached: 1913 - mixBufferSize from frame count * frame size 1914 - activeSleepTime from activeSleepTimeUs() 1915 - idleSleepTime from idleSleepTimeUs() 1916 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1917 - maxPeriod from frame count and sample rate (MIXER only) 1918 1919The parameters that affect these derived values are: 1920 - frame count 1921 - frame size 1922 - sample rate 1923 - device type: A2DP or not 1924 - device latency 1925 - format: PCM or not 1926 - active sleep time 1927 - idle sleep time 1928*/ 1929 1930void AudioFlinger::PlaybackThread::cacheParameters_l() 1931{ 1932 mixBufferSize = mNormalFrameCount * mFrameSize; 1933 activeSleepTime = activeSleepTimeUs(); 1934 idleSleepTime = idleSleepTimeUs(); 1935} 1936 1937void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1938{ 1939 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1940 this, streamType, mTracks.size()); 1941 Mutex::Autolock _l(mLock); 1942 1943 size_t size = mTracks.size(); 1944 for (size_t i = 0; i < size; i++) { 1945 sp<Track> t = mTracks[i]; 1946 if (t->streamType() == streamType) { 1947 t->invalidate(); 1948 } 1949 } 1950} 1951 1952status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1953{ 1954 int session = chain->sessionId(); 1955 int16_t *buffer = mMixBuffer; 1956 bool ownsBuffer = false; 1957 1958 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1959 if (session > 0) { 1960 // Only one effect chain can be present in direct output thread and it uses 1961 // the mix buffer as input 1962 if (mType != DIRECT) { 1963 size_t numSamples = mNormalFrameCount * mChannelCount; 1964 buffer = new int16_t[numSamples]; 1965 memset(buffer, 0, numSamples * sizeof(int16_t)); 1966 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1967 ownsBuffer = true; 1968 } 1969 1970 // Attach all tracks with same session ID to this chain. 1971 for (size_t i = 0; i < mTracks.size(); ++i) { 1972 sp<Track> track = mTracks[i]; 1973 if (session == track->sessionId()) { 1974 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1975 buffer); 1976 track->setMainBuffer(buffer); 1977 chain->incTrackCnt(); 1978 } 1979 } 1980 1981 // indicate all active tracks in the chain 1982 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1983 sp<Track> track = mActiveTracks[i].promote(); 1984 if (track == 0) { 1985 continue; 1986 } 1987 if (session == track->sessionId()) { 1988 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1989 chain->incActiveTrackCnt(); 1990 } 1991 } 1992 } 1993 1994 chain->setInBuffer(buffer, ownsBuffer); 1995 chain->setOutBuffer(mMixBuffer); 1996 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1997 // chains list in order to be processed last as it contains output stage effects 1998 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1999 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2000 // after track specific effects and before output stage 2001 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2002 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2003 // Effect chain for other sessions are inserted at beginning of effect 2004 // chains list to be processed before output mix effects. Relative order between other 2005 // sessions is not important 2006 size_t size = mEffectChains.size(); 2007 size_t i = 0; 2008 for (i = 0; i < size; i++) { 2009 if (mEffectChains[i]->sessionId() < session) { 2010 break; 2011 } 2012 } 2013 mEffectChains.insertAt(chain, i); 2014 checkSuspendOnAddEffectChain_l(chain); 2015 2016 return NO_ERROR; 2017} 2018 2019size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2020{ 2021 int session = chain->sessionId(); 2022 2023 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2024 2025 for (size_t i = 0; i < mEffectChains.size(); i++) { 2026 if (chain == mEffectChains[i]) { 2027 mEffectChains.removeAt(i); 2028 // detach all active tracks from the chain 2029 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2030 sp<Track> track = mActiveTracks[i].promote(); 2031 if (track == 0) { 2032 continue; 2033 } 2034 if (session == track->sessionId()) { 2035 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2036 chain.get(), session); 2037 chain->decActiveTrackCnt(); 2038 } 2039 } 2040 2041 // detach all tracks with same session ID from this chain 2042 for (size_t i = 0; i < mTracks.size(); ++i) { 2043 sp<Track> track = mTracks[i]; 2044 if (session == track->sessionId()) { 2045 track->setMainBuffer(mMixBuffer); 2046 chain->decTrackCnt(); 2047 } 2048 } 2049 break; 2050 } 2051 } 2052 return mEffectChains.size(); 2053} 2054 2055status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2056 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2057{ 2058 Mutex::Autolock _l(mLock); 2059 return attachAuxEffect_l(track, EffectId); 2060} 2061 2062status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2063 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2064{ 2065 status_t status = NO_ERROR; 2066 2067 if (EffectId == 0) { 2068 track->setAuxBuffer(0, NULL); 2069 } else { 2070 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2071 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2072 if (effect != 0) { 2073 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2074 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2075 } else { 2076 status = INVALID_OPERATION; 2077 } 2078 } else { 2079 status = BAD_VALUE; 2080 } 2081 } 2082 return status; 2083} 2084 2085void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2086{ 2087 for (size_t i = 0; i < mTracks.size(); ++i) { 2088 sp<Track> track = mTracks[i]; 2089 if (track->auxEffectId() == effectId) { 2090 attachAuxEffect_l(track, 0); 2091 } 2092 } 2093} 2094 2095bool AudioFlinger::PlaybackThread::threadLoop() 2096{ 2097 Vector< sp<Track> > tracksToRemove; 2098 2099 standbyTime = systemTime(); 2100 2101 // MIXER 2102 nsecs_t lastWarning = 0; 2103 2104 // DUPLICATING 2105 // FIXME could this be made local to while loop? 2106 writeFrames = 0; 2107 2108 cacheParameters_l(); 2109 sleepTime = idleSleepTime; 2110 2111 if (mType == MIXER) { 2112 sleepTimeShift = 0; 2113 } 2114 2115 CpuStats cpuStats; 2116 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2117 2118 acquireWakeLock(); 2119 2120 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2121 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2122 // and then that string will be logged at the next convenient opportunity. 2123 const char *logString = NULL; 2124 2125 while (!exitPending()) 2126 { 2127 cpuStats.sample(myName); 2128 2129 Vector< sp<EffectChain> > effectChains; 2130 2131 processConfigEvents(); 2132 2133 { // scope for mLock 2134 2135 Mutex::Autolock _l(mLock); 2136 2137 if (logString != NULL) { 2138 mNBLogWriter->logTimestamp(); 2139 mNBLogWriter->log(logString); 2140 logString = NULL; 2141 } 2142 2143 if (mLatchDValid) { 2144 mLatchQ = mLatchD; 2145 mLatchDValid = false; 2146 mLatchQValid = true; 2147 } 2148 2149 if (checkForNewParameters_l()) { 2150 cacheParameters_l(); 2151 } 2152 2153 saveOutputTracks(); 2154 2155 if (mSignalPending) { 2156 // A signal was raised while we were unlocked 2157 mSignalPending = false; 2158 } else if (waitingAsyncCallback_l()) { 2159 if (exitPending()) { 2160 break; 2161 } 2162 releaseWakeLock_l(); 2163 ALOGV("wait async completion"); 2164 mWaitWorkCV.wait(mLock); 2165 ALOGV("async completion/wake"); 2166 acquireWakeLock_l(); 2167 standbyTime = systemTime() + standbyDelay; 2168 sleepTime = 0; 2169 if (exitPending()) { 2170 break; 2171 } 2172 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2173 isSuspended()) { 2174 // put audio hardware into standby after short delay 2175 if (shouldStandby_l()) { 2176 2177 threadLoop_standby(); 2178 2179 mStandby = true; 2180 } 2181 2182 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2183 // we're about to wait, flush the binder command buffer 2184 IPCThreadState::self()->flushCommands(); 2185 2186 clearOutputTracks(); 2187 2188 if (exitPending()) { 2189 break; 2190 } 2191 2192 releaseWakeLock_l(); 2193 // wait until we have something to do... 2194 ALOGV("%s going to sleep", myName.string()); 2195 mWaitWorkCV.wait(mLock); 2196 ALOGV("%s waking up", myName.string()); 2197 acquireWakeLock_l(); 2198 2199 mMixerStatus = MIXER_IDLE; 2200 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2201 mBytesWritten = 0; 2202 mBytesRemaining = 0; 2203 checkSilentMode_l(); 2204 2205 standbyTime = systemTime() + standbyDelay; 2206 sleepTime = idleSleepTime; 2207 if (mType == MIXER) { 2208 sleepTimeShift = 0; 2209 } 2210 2211 continue; 2212 } 2213 } 2214 2215 // mMixerStatusIgnoringFastTracks is also updated internally 2216 mMixerStatus = prepareTracks_l(&tracksToRemove); 2217 2218 // prevent any changes in effect chain list and in each effect chain 2219 // during mixing and effect process as the audio buffers could be deleted 2220 // or modified if an effect is created or deleted 2221 lockEffectChains_l(effectChains); 2222 } 2223 2224 if (mBytesRemaining == 0) { 2225 mCurrentWriteLength = 0; 2226 if (mMixerStatus == MIXER_TRACKS_READY) { 2227 // threadLoop_mix() sets mCurrentWriteLength 2228 threadLoop_mix(); 2229 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2230 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2231 // threadLoop_sleepTime sets sleepTime to 0 if data 2232 // must be written to HAL 2233 threadLoop_sleepTime(); 2234 if (sleepTime == 0) { 2235 mCurrentWriteLength = mixBufferSize; 2236 } 2237 } 2238 mBytesRemaining = mCurrentWriteLength; 2239 if (isSuspended()) { 2240 sleepTime = suspendSleepTimeUs(); 2241 // simulate write to HAL when suspended 2242 mBytesWritten += mixBufferSize; 2243 mBytesRemaining = 0; 2244 } 2245 2246 // only process effects if we're going to write 2247 if (sleepTime == 0) { 2248 for (size_t i = 0; i < effectChains.size(); i ++) { 2249 effectChains[i]->process_l(); 2250 } 2251 } 2252 } 2253 2254 // enable changes in effect chain 2255 unlockEffectChains(effectChains); 2256 2257 if (!waitingAsyncCallback()) { 2258 // sleepTime == 0 means we must write to audio hardware 2259 if (sleepTime == 0) { 2260 if (mBytesRemaining) { 2261 ssize_t ret = threadLoop_write(); 2262 if (ret < 0) { 2263 mBytesRemaining = 0; 2264 } else { 2265 mBytesWritten += ret; 2266 mBytesRemaining -= ret; 2267 } 2268 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2269 (mMixerStatus == MIXER_DRAIN_ALL)) { 2270 threadLoop_drain(); 2271 } 2272if (mType == MIXER) { 2273 // write blocked detection 2274 nsecs_t now = systemTime(); 2275 nsecs_t delta = now - mLastWriteTime; 2276 if (!mStandby && delta > maxPeriod) { 2277 mNumDelayedWrites++; 2278 if ((now - lastWarning) > kWarningThrottleNs) { 2279 ATRACE_NAME("underrun"); 2280 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2281 ns2ms(delta), mNumDelayedWrites, this); 2282 lastWarning = now; 2283 } 2284 } 2285} 2286 2287 mStandby = false; 2288 } else { 2289 usleep(sleepTime); 2290 } 2291 } 2292 2293 // Finally let go of removed track(s), without the lock held 2294 // since we can't guarantee the destructors won't acquire that 2295 // same lock. This will also mutate and push a new fast mixer state. 2296 threadLoop_removeTracks(tracksToRemove); 2297 tracksToRemove.clear(); 2298 2299 // FIXME I don't understand the need for this here; 2300 // it was in the original code but maybe the 2301 // assignment in saveOutputTracks() makes this unnecessary? 2302 clearOutputTracks(); 2303 2304 // Effect chains will be actually deleted here if they were removed from 2305 // mEffectChains list during mixing or effects processing 2306 effectChains.clear(); 2307 2308 // FIXME Note that the above .clear() is no longer necessary since effectChains 2309 // is now local to this block, but will keep it for now (at least until merge done). 2310 } 2311 2312 threadLoop_exit(); 2313 2314 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2315 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2316 // put output stream into standby mode 2317 if (!mStandby) { 2318 mOutput->stream->common.standby(&mOutput->stream->common); 2319 } 2320 } 2321 2322 releaseWakeLock(); 2323 2324 ALOGV("Thread %p type %d exiting", this, mType); 2325 return false; 2326} 2327 2328// removeTracks_l() must be called with ThreadBase::mLock held 2329void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2330{ 2331 size_t count = tracksToRemove.size(); 2332 if (count > 0) { 2333 for (size_t i=0 ; i<count ; i++) { 2334 const sp<Track>& track = tracksToRemove.itemAt(i); 2335 mActiveTracks.remove(track); 2336 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2337 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2338 if (chain != 0) { 2339 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2340 track->sessionId()); 2341 chain->decActiveTrackCnt(); 2342 } 2343 if (track->isTerminated()) { 2344 removeTrack_l(track); 2345 } 2346 } 2347 } 2348 2349} 2350 2351// ---------------------------------------------------------------------------- 2352 2353AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2354 audio_io_handle_t id, audio_devices_t device, type_t type) 2355 : PlaybackThread(audioFlinger, output, id, device, type), 2356 // mAudioMixer below 2357 // mFastMixer below 2358 mFastMixerFutex(0) 2359 // mOutputSink below 2360 // mPipeSink below 2361 // mNormalSink below 2362{ 2363 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2364 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2365 "mFrameCount=%d, mNormalFrameCount=%d", 2366 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2367 mNormalFrameCount); 2368 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2369 2370 // FIXME - Current mixer implementation only supports stereo output 2371 if (mChannelCount != FCC_2) { 2372 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2373 } 2374 2375 // create an NBAIO sink for the HAL output stream, and negotiate 2376 mOutputSink = new AudioStreamOutSink(output->stream); 2377 size_t numCounterOffers = 0; 2378 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2379 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2380 ALOG_ASSERT(index == 0); 2381 2382 // initialize fast mixer depending on configuration 2383 bool initFastMixer; 2384 switch (kUseFastMixer) { 2385 case FastMixer_Never: 2386 initFastMixer = false; 2387 break; 2388 case FastMixer_Always: 2389 initFastMixer = true; 2390 break; 2391 case FastMixer_Static: 2392 case FastMixer_Dynamic: 2393 initFastMixer = mFrameCount < mNormalFrameCount; 2394 break; 2395 } 2396 if (initFastMixer) { 2397 2398 // create a MonoPipe to connect our submix to FastMixer 2399 NBAIO_Format format = mOutputSink->format(); 2400 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2401 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2402 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2403 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2404 const NBAIO_Format offers[1] = {format}; 2405 size_t numCounterOffers = 0; 2406 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2407 ALOG_ASSERT(index == 0); 2408 monoPipe->setAvgFrames((mScreenState & 1) ? 2409 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2410 mPipeSink = monoPipe; 2411 2412#ifdef TEE_SINK 2413 if (mTeeSinkOutputEnabled) { 2414 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2415 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2416 numCounterOffers = 0; 2417 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2418 ALOG_ASSERT(index == 0); 2419 mTeeSink = teeSink; 2420 PipeReader *teeSource = new PipeReader(*teeSink); 2421 numCounterOffers = 0; 2422 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2423 ALOG_ASSERT(index == 0); 2424 mTeeSource = teeSource; 2425 } 2426#endif 2427 2428 // create fast mixer and configure it initially with just one fast track for our submix 2429 mFastMixer = new FastMixer(); 2430 FastMixerStateQueue *sq = mFastMixer->sq(); 2431#ifdef STATE_QUEUE_DUMP 2432 sq->setObserverDump(&mStateQueueObserverDump); 2433 sq->setMutatorDump(&mStateQueueMutatorDump); 2434#endif 2435 FastMixerState *state = sq->begin(); 2436 FastTrack *fastTrack = &state->mFastTracks[0]; 2437 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2438 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2439 fastTrack->mVolumeProvider = NULL; 2440 fastTrack->mGeneration++; 2441 state->mFastTracksGen++; 2442 state->mTrackMask = 1; 2443 // fast mixer will use the HAL output sink 2444 state->mOutputSink = mOutputSink.get(); 2445 state->mOutputSinkGen++; 2446 state->mFrameCount = mFrameCount; 2447 state->mCommand = FastMixerState::COLD_IDLE; 2448 // already done in constructor initialization list 2449 //mFastMixerFutex = 0; 2450 state->mColdFutexAddr = &mFastMixerFutex; 2451 state->mColdGen++; 2452 state->mDumpState = &mFastMixerDumpState; 2453#ifdef TEE_SINK 2454 state->mTeeSink = mTeeSink.get(); 2455#endif 2456 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2457 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2458 sq->end(); 2459 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2460 2461 // start the fast mixer 2462 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2463 pid_t tid = mFastMixer->getTid(); 2464 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2465 if (err != 0) { 2466 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2467 kPriorityFastMixer, getpid_cached, tid, err); 2468 } 2469 2470#ifdef AUDIO_WATCHDOG 2471 // create and start the watchdog 2472 mAudioWatchdog = new AudioWatchdog(); 2473 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2474 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2475 tid = mAudioWatchdog->getTid(); 2476 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2477 if (err != 0) { 2478 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2479 kPriorityFastMixer, getpid_cached, tid, err); 2480 } 2481#endif 2482 2483 } else { 2484 mFastMixer = NULL; 2485 } 2486 2487 switch (kUseFastMixer) { 2488 case FastMixer_Never: 2489 case FastMixer_Dynamic: 2490 mNormalSink = mOutputSink; 2491 break; 2492 case FastMixer_Always: 2493 mNormalSink = mPipeSink; 2494 break; 2495 case FastMixer_Static: 2496 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2497 break; 2498 } 2499} 2500 2501AudioFlinger::MixerThread::~MixerThread() 2502{ 2503 if (mFastMixer != NULL) { 2504 FastMixerStateQueue *sq = mFastMixer->sq(); 2505 FastMixerState *state = sq->begin(); 2506 if (state->mCommand == FastMixerState::COLD_IDLE) { 2507 int32_t old = android_atomic_inc(&mFastMixerFutex); 2508 if (old == -1) { 2509 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2510 } 2511 } 2512 state->mCommand = FastMixerState::EXIT; 2513 sq->end(); 2514 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2515 mFastMixer->join(); 2516 // Though the fast mixer thread has exited, it's state queue is still valid. 2517 // We'll use that extract the final state which contains one remaining fast track 2518 // corresponding to our sub-mix. 2519 state = sq->begin(); 2520 ALOG_ASSERT(state->mTrackMask == 1); 2521 FastTrack *fastTrack = &state->mFastTracks[0]; 2522 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2523 delete fastTrack->mBufferProvider; 2524 sq->end(false /*didModify*/); 2525 delete mFastMixer; 2526#ifdef AUDIO_WATCHDOG 2527 if (mAudioWatchdog != 0) { 2528 mAudioWatchdog->requestExit(); 2529 mAudioWatchdog->requestExitAndWait(); 2530 mAudioWatchdog.clear(); 2531 } 2532#endif 2533 } 2534 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2535 delete mAudioMixer; 2536} 2537 2538 2539uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2540{ 2541 if (mFastMixer != NULL) { 2542 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2543 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2544 } 2545 return latency; 2546} 2547 2548 2549void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2550{ 2551 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2552} 2553 2554ssize_t AudioFlinger::MixerThread::threadLoop_write() 2555{ 2556 // FIXME we should only do one push per cycle; confirm this is true 2557 // Start the fast mixer if it's not already running 2558 if (mFastMixer != NULL) { 2559 FastMixerStateQueue *sq = mFastMixer->sq(); 2560 FastMixerState *state = sq->begin(); 2561 if (state->mCommand != FastMixerState::MIX_WRITE && 2562 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2563 if (state->mCommand == FastMixerState::COLD_IDLE) { 2564 int32_t old = android_atomic_inc(&mFastMixerFutex); 2565 if (old == -1) { 2566 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2567 } 2568#ifdef AUDIO_WATCHDOG 2569 if (mAudioWatchdog != 0) { 2570 mAudioWatchdog->resume(); 2571 } 2572#endif 2573 } 2574 state->mCommand = FastMixerState::MIX_WRITE; 2575 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2576 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2577 sq->end(); 2578 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2579 if (kUseFastMixer == FastMixer_Dynamic) { 2580 mNormalSink = mPipeSink; 2581 } 2582 } else { 2583 sq->end(false /*didModify*/); 2584 } 2585 } 2586 return PlaybackThread::threadLoop_write(); 2587} 2588 2589void AudioFlinger::MixerThread::threadLoop_standby() 2590{ 2591 // Idle the fast mixer if it's currently running 2592 if (mFastMixer != NULL) { 2593 FastMixerStateQueue *sq = mFastMixer->sq(); 2594 FastMixerState *state = sq->begin(); 2595 if (!(state->mCommand & FastMixerState::IDLE)) { 2596 state->mCommand = FastMixerState::COLD_IDLE; 2597 state->mColdFutexAddr = &mFastMixerFutex; 2598 state->mColdGen++; 2599 mFastMixerFutex = 0; 2600 sq->end(); 2601 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2602 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2603 if (kUseFastMixer == FastMixer_Dynamic) { 2604 mNormalSink = mOutputSink; 2605 } 2606#ifdef AUDIO_WATCHDOG 2607 if (mAudioWatchdog != 0) { 2608 mAudioWatchdog->pause(); 2609 } 2610#endif 2611 } else { 2612 sq->end(false /*didModify*/); 2613 } 2614 } 2615 PlaybackThread::threadLoop_standby(); 2616} 2617 2618// Empty implementation for standard mixer 2619// Overridden for offloaded playback 2620void AudioFlinger::PlaybackThread::flushOutput_l() 2621{ 2622} 2623 2624bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2625{ 2626 return false; 2627} 2628 2629bool AudioFlinger::PlaybackThread::shouldStandby_l() 2630{ 2631 return !mStandby; 2632} 2633 2634bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2635{ 2636 Mutex::Autolock _l(mLock); 2637 return waitingAsyncCallback_l(); 2638} 2639 2640// shared by MIXER and DIRECT, overridden by DUPLICATING 2641void AudioFlinger::PlaybackThread::threadLoop_standby() 2642{ 2643 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2644 mOutput->stream->common.standby(&mOutput->stream->common); 2645 if (mUseAsyncWrite != 0) { 2646 // discard any pending drain or write ack by incrementing sequence 2647 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2648 mDrainSequence = (mDrainSequence + 2) & ~1; 2649 ALOG_ASSERT(mCallbackThread != 0); 2650 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2651 mCallbackThread->setDraining(mDrainSequence); 2652 } 2653} 2654 2655void AudioFlinger::MixerThread::threadLoop_mix() 2656{ 2657 // obtain the presentation timestamp of the next output buffer 2658 int64_t pts; 2659 status_t status = INVALID_OPERATION; 2660 2661 if (mNormalSink != 0) { 2662 status = mNormalSink->getNextWriteTimestamp(&pts); 2663 } else { 2664 status = mOutputSink->getNextWriteTimestamp(&pts); 2665 } 2666 2667 if (status != NO_ERROR) { 2668 pts = AudioBufferProvider::kInvalidPTS; 2669 } 2670 2671 // mix buffers... 2672 mAudioMixer->process(pts); 2673 mCurrentWriteLength = mixBufferSize; 2674 // increase sleep time progressively when application underrun condition clears. 2675 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2676 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2677 // such that we would underrun the audio HAL. 2678 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2679 sleepTimeShift--; 2680 } 2681 sleepTime = 0; 2682 standbyTime = systemTime() + standbyDelay; 2683 //TODO: delay standby when effects have a tail 2684} 2685 2686void AudioFlinger::MixerThread::threadLoop_sleepTime() 2687{ 2688 // If no tracks are ready, sleep once for the duration of an output 2689 // buffer size, then write 0s to the output 2690 if (sleepTime == 0) { 2691 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2692 sleepTime = activeSleepTime >> sleepTimeShift; 2693 if (sleepTime < kMinThreadSleepTimeUs) { 2694 sleepTime = kMinThreadSleepTimeUs; 2695 } 2696 // reduce sleep time in case of consecutive application underruns to avoid 2697 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2698 // duration we would end up writing less data than needed by the audio HAL if 2699 // the condition persists. 2700 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2701 sleepTimeShift++; 2702 } 2703 } else { 2704 sleepTime = idleSleepTime; 2705 } 2706 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2707 memset(mMixBuffer, 0, mixBufferSize); 2708 sleepTime = 0; 2709 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2710 "anticipated start"); 2711 } 2712 // TODO add standby time extension fct of effect tail 2713} 2714 2715// prepareTracks_l() must be called with ThreadBase::mLock held 2716AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2717 Vector< sp<Track> > *tracksToRemove) 2718{ 2719 2720 mixer_state mixerStatus = MIXER_IDLE; 2721 // find out which tracks need to be processed 2722 size_t count = mActiveTracks.size(); 2723 size_t mixedTracks = 0; 2724 size_t tracksWithEffect = 0; 2725 // counts only _active_ fast tracks 2726 size_t fastTracks = 0; 2727 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2728 2729 float masterVolume = mMasterVolume; 2730 bool masterMute = mMasterMute; 2731 2732 if (masterMute) { 2733 masterVolume = 0; 2734 } 2735 // Delegate master volume control to effect in output mix effect chain if needed 2736 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2737 if (chain != 0) { 2738 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2739 chain->setVolume_l(&v, &v); 2740 masterVolume = (float)((v + (1 << 23)) >> 24); 2741 chain.clear(); 2742 } 2743 2744 // prepare a new state to push 2745 FastMixerStateQueue *sq = NULL; 2746 FastMixerState *state = NULL; 2747 bool didModify = false; 2748 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2749 if (mFastMixer != NULL) { 2750 sq = mFastMixer->sq(); 2751 state = sq->begin(); 2752 } 2753 2754 for (size_t i=0 ; i<count ; i++) { 2755 const sp<Track> t = mActiveTracks[i].promote(); 2756 if (t == 0) { 2757 continue; 2758 } 2759 2760 // this const just means the local variable doesn't change 2761 Track* const track = t.get(); 2762 2763 // process fast tracks 2764 if (track->isFastTrack()) { 2765 2766 // It's theoretically possible (though unlikely) for a fast track to be created 2767 // and then removed within the same normal mix cycle. This is not a problem, as 2768 // the track never becomes active so it's fast mixer slot is never touched. 2769 // The converse, of removing an (active) track and then creating a new track 2770 // at the identical fast mixer slot within the same normal mix cycle, 2771 // is impossible because the slot isn't marked available until the end of each cycle. 2772 int j = track->mFastIndex; 2773 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2774 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2775 FastTrack *fastTrack = &state->mFastTracks[j]; 2776 2777 // Determine whether the track is currently in underrun condition, 2778 // and whether it had a recent underrun. 2779 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2780 FastTrackUnderruns underruns = ftDump->mUnderruns; 2781 uint32_t recentFull = (underruns.mBitFields.mFull - 2782 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2783 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2784 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2785 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2786 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2787 uint32_t recentUnderruns = recentPartial + recentEmpty; 2788 track->mObservedUnderruns = underruns; 2789 // don't count underruns that occur while stopping or pausing 2790 // or stopped which can occur when flush() is called while active 2791 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2792 recentUnderruns > 0) { 2793 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2794 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2795 } 2796 2797 // This is similar to the state machine for normal tracks, 2798 // with a few modifications for fast tracks. 2799 bool isActive = true; 2800 switch (track->mState) { 2801 case TrackBase::STOPPING_1: 2802 // track stays active in STOPPING_1 state until first underrun 2803 if (recentUnderruns > 0 || track->isTerminated()) { 2804 track->mState = TrackBase::STOPPING_2; 2805 } 2806 break; 2807 case TrackBase::PAUSING: 2808 // ramp down is not yet implemented 2809 track->setPaused(); 2810 break; 2811 case TrackBase::RESUMING: 2812 // ramp up is not yet implemented 2813 track->mState = TrackBase::ACTIVE; 2814 break; 2815 case TrackBase::ACTIVE: 2816 if (recentFull > 0 || recentPartial > 0) { 2817 // track has provided at least some frames recently: reset retry count 2818 track->mRetryCount = kMaxTrackRetries; 2819 } 2820 if (recentUnderruns == 0) { 2821 // no recent underruns: stay active 2822 break; 2823 } 2824 // there has recently been an underrun of some kind 2825 if (track->sharedBuffer() == 0) { 2826 // were any of the recent underruns "empty" (no frames available)? 2827 if (recentEmpty == 0) { 2828 // no, then ignore the partial underruns as they are allowed indefinitely 2829 break; 2830 } 2831 // there has recently been an "empty" underrun: decrement the retry counter 2832 if (--(track->mRetryCount) > 0) { 2833 break; 2834 } 2835 // indicate to client process that the track was disabled because of underrun; 2836 // it will then automatically call start() when data is available 2837 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2838 // remove from active list, but state remains ACTIVE [confusing but true] 2839 isActive = false; 2840 break; 2841 } 2842 // fall through 2843 case TrackBase::STOPPING_2: 2844 case TrackBase::PAUSED: 2845 case TrackBase::STOPPED: 2846 case TrackBase::FLUSHED: // flush() while active 2847 // Check for presentation complete if track is inactive 2848 // We have consumed all the buffers of this track. 2849 // This would be incomplete if we auto-paused on underrun 2850 { 2851 size_t audioHALFrames = 2852 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2853 size_t framesWritten = mBytesWritten / mFrameSize; 2854 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2855 // track stays in active list until presentation is complete 2856 break; 2857 } 2858 } 2859 if (track->isStopping_2()) { 2860 track->mState = TrackBase::STOPPED; 2861 } 2862 if (track->isStopped()) { 2863 // Can't reset directly, as fast mixer is still polling this track 2864 // track->reset(); 2865 // So instead mark this track as needing to be reset after push with ack 2866 resetMask |= 1 << i; 2867 } 2868 isActive = false; 2869 break; 2870 case TrackBase::IDLE: 2871 default: 2872 LOG_FATAL("unexpected track state %d", track->mState); 2873 } 2874 2875 if (isActive) { 2876 // was it previously inactive? 2877 if (!(state->mTrackMask & (1 << j))) { 2878 ExtendedAudioBufferProvider *eabp = track; 2879 VolumeProvider *vp = track; 2880 fastTrack->mBufferProvider = eabp; 2881 fastTrack->mVolumeProvider = vp; 2882 fastTrack->mSampleRate = track->mSampleRate; 2883 fastTrack->mChannelMask = track->mChannelMask; 2884 fastTrack->mGeneration++; 2885 state->mTrackMask |= 1 << j; 2886 didModify = true; 2887 // no acknowledgement required for newly active tracks 2888 } 2889 // cache the combined master volume and stream type volume for fast mixer; this 2890 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2891 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2892 ++fastTracks; 2893 } else { 2894 // was it previously active? 2895 if (state->mTrackMask & (1 << j)) { 2896 fastTrack->mBufferProvider = NULL; 2897 fastTrack->mGeneration++; 2898 state->mTrackMask &= ~(1 << j); 2899 didModify = true; 2900 // If any fast tracks were removed, we must wait for acknowledgement 2901 // because we're about to decrement the last sp<> on those tracks. 2902 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2903 } else { 2904 LOG_FATAL("fast track %d should have been active", j); 2905 } 2906 tracksToRemove->add(track); 2907 // Avoids a misleading display in dumpsys 2908 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2909 } 2910 continue; 2911 } 2912 2913 { // local variable scope to avoid goto warning 2914 2915 audio_track_cblk_t* cblk = track->cblk(); 2916 2917 // The first time a track is added we wait 2918 // for all its buffers to be filled before processing it 2919 int name = track->name(); 2920 // make sure that we have enough frames to mix one full buffer. 2921 // enforce this condition only once to enable draining the buffer in case the client 2922 // app does not call stop() and relies on underrun to stop: 2923 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2924 // during last round 2925 size_t desiredFrames; 2926 uint32_t sr = track->sampleRate(); 2927 if (sr == mSampleRate) { 2928 desiredFrames = mNormalFrameCount; 2929 } else { 2930 // +1 for rounding and +1 for additional sample needed for interpolation 2931 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2932 // add frames already consumed but not yet released by the resampler 2933 // because mAudioTrackServerProxy->framesReady() will include these frames 2934 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2935 // the minimum track buffer size is normally twice the number of frames necessary 2936 // to fill one buffer and the resampler should not leave more than one buffer worth 2937 // of unreleased frames after each pass, but just in case... 2938 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2939 } 2940 uint32_t minFrames = 1; 2941 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2942 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2943 minFrames = desiredFrames; 2944 } 2945 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2946 size_t framesReady; 2947 if (track->sharedBuffer() == 0) { 2948 framesReady = track->framesReady(); 2949 } else if (track->isStopped()) { 2950 framesReady = 0; 2951 } else { 2952 framesReady = 1; 2953 } 2954 if ((framesReady >= minFrames) && track->isReady() && 2955 !track->isPaused() && !track->isTerminated()) 2956 { 2957 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2958 2959 mixedTracks++; 2960 2961 // track->mainBuffer() != mMixBuffer means there is an effect chain 2962 // connected to the track 2963 chain.clear(); 2964 if (track->mainBuffer() != mMixBuffer) { 2965 chain = getEffectChain_l(track->sessionId()); 2966 // Delegate volume control to effect in track effect chain if needed 2967 if (chain != 0) { 2968 tracksWithEffect++; 2969 } else { 2970 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2971 "session %d", 2972 name, track->sessionId()); 2973 } 2974 } 2975 2976 2977 int param = AudioMixer::VOLUME; 2978 if (track->mFillingUpStatus == Track::FS_FILLED) { 2979 // no ramp for the first volume setting 2980 track->mFillingUpStatus = Track::FS_ACTIVE; 2981 if (track->mState == TrackBase::RESUMING) { 2982 track->mState = TrackBase::ACTIVE; 2983 param = AudioMixer::RAMP_VOLUME; 2984 } 2985 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2986 // FIXME should not make a decision based on mServer 2987 } else if (cblk->mServer != 0) { 2988 // If the track is stopped before the first frame was mixed, 2989 // do not apply ramp 2990 param = AudioMixer::RAMP_VOLUME; 2991 } 2992 2993 // compute volume for this track 2994 uint32_t vl, vr, va; 2995 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2996 vl = vr = va = 0; 2997 if (track->isPausing()) { 2998 track->setPaused(); 2999 } 3000 } else { 3001 3002 // read original volumes with volume control 3003 float typeVolume = mStreamTypes[track->streamType()].volume; 3004 float v = masterVolume * typeVolume; 3005 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3006 uint32_t vlr = proxy->getVolumeLR(); 3007 vl = vlr & 0xFFFF; 3008 vr = vlr >> 16; 3009 // track volumes come from shared memory, so can't be trusted and must be clamped 3010 if (vl > MAX_GAIN_INT) { 3011 ALOGV("Track left volume out of range: %04X", vl); 3012 vl = MAX_GAIN_INT; 3013 } 3014 if (vr > MAX_GAIN_INT) { 3015 ALOGV("Track right volume out of range: %04X", vr); 3016 vr = MAX_GAIN_INT; 3017 } 3018 // now apply the master volume and stream type volume 3019 vl = (uint32_t)(v * vl) << 12; 3020 vr = (uint32_t)(v * vr) << 12; 3021 // assuming master volume and stream type volume each go up to 1.0, 3022 // vl and vr are now in 8.24 format 3023 3024 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3025 // send level comes from shared memory and so may be corrupt 3026 if (sendLevel > MAX_GAIN_INT) { 3027 ALOGV("Track send level out of range: %04X", sendLevel); 3028 sendLevel = MAX_GAIN_INT; 3029 } 3030 va = (uint32_t)(v * sendLevel); 3031 } 3032 3033 // Delegate volume control to effect in track effect chain if needed 3034 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3035 // Do not ramp volume if volume is controlled by effect 3036 param = AudioMixer::VOLUME; 3037 track->mHasVolumeController = true; 3038 } else { 3039 // force no volume ramp when volume controller was just disabled or removed 3040 // from effect chain to avoid volume spike 3041 if (track->mHasVolumeController) { 3042 param = AudioMixer::VOLUME; 3043 } 3044 track->mHasVolumeController = false; 3045 } 3046 3047 // Convert volumes from 8.24 to 4.12 format 3048 // This additional clamping is needed in case chain->setVolume_l() overshot 3049 vl = (vl + (1 << 11)) >> 12; 3050 if (vl > MAX_GAIN_INT) { 3051 vl = MAX_GAIN_INT; 3052 } 3053 vr = (vr + (1 << 11)) >> 12; 3054 if (vr > MAX_GAIN_INT) { 3055 vr = MAX_GAIN_INT; 3056 } 3057 3058 if (va > MAX_GAIN_INT) { 3059 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3060 } 3061 3062 // XXX: these things DON'T need to be done each time 3063 mAudioMixer->setBufferProvider(name, track); 3064 mAudioMixer->enable(name); 3065 3066 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3067 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3068 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3069 mAudioMixer->setParameter( 3070 name, 3071 AudioMixer::TRACK, 3072 AudioMixer::FORMAT, (void *)track->format()); 3073 mAudioMixer->setParameter( 3074 name, 3075 AudioMixer::TRACK, 3076 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3077 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3078 uint32_t maxSampleRate = mSampleRate * 2; 3079 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3080 if (reqSampleRate == 0) { 3081 reqSampleRate = mSampleRate; 3082 } else if (reqSampleRate > maxSampleRate) { 3083 reqSampleRate = maxSampleRate; 3084 } 3085 mAudioMixer->setParameter( 3086 name, 3087 AudioMixer::RESAMPLE, 3088 AudioMixer::SAMPLE_RATE, 3089 (void *)reqSampleRate); 3090 mAudioMixer->setParameter( 3091 name, 3092 AudioMixer::TRACK, 3093 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3094 mAudioMixer->setParameter( 3095 name, 3096 AudioMixer::TRACK, 3097 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3098 3099 // reset retry count 3100 track->mRetryCount = kMaxTrackRetries; 3101 3102 // If one track is ready, set the mixer ready if: 3103 // - the mixer was not ready during previous round OR 3104 // - no other track is not ready 3105 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3106 mixerStatus != MIXER_TRACKS_ENABLED) { 3107 mixerStatus = MIXER_TRACKS_READY; 3108 } 3109 } else { 3110 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3111 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3112 } 3113 // clear effect chain input buffer if an active track underruns to avoid sending 3114 // previous audio buffer again to effects 3115 chain = getEffectChain_l(track->sessionId()); 3116 if (chain != 0) { 3117 chain->clearInputBuffer(); 3118 } 3119 3120 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3121 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3122 track->isStopped() || track->isPaused()) { 3123 // We have consumed all the buffers of this track. 3124 // Remove it from the list of active tracks. 3125 // TODO: use actual buffer filling status instead of latency when available from 3126 // audio HAL 3127 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3128 size_t framesWritten = mBytesWritten / mFrameSize; 3129 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3130 if (track->isStopped()) { 3131 track->reset(); 3132 } 3133 tracksToRemove->add(track); 3134 } 3135 } else { 3136 // No buffers for this track. Give it a few chances to 3137 // fill a buffer, then remove it from active list. 3138 if (--(track->mRetryCount) <= 0) { 3139 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3140 tracksToRemove->add(track); 3141 // indicate to client process that the track was disabled because of underrun; 3142 // it will then automatically call start() when data is available 3143 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3144 // If one track is not ready, mark the mixer also not ready if: 3145 // - the mixer was ready during previous round OR 3146 // - no other track is ready 3147 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3148 mixerStatus != MIXER_TRACKS_READY) { 3149 mixerStatus = MIXER_TRACKS_ENABLED; 3150 } 3151 } 3152 mAudioMixer->disable(name); 3153 } 3154 3155 } // local variable scope to avoid goto warning 3156track_is_ready: ; 3157 3158 } 3159 3160 // Push the new FastMixer state if necessary 3161 bool pauseAudioWatchdog = false; 3162 if (didModify) { 3163 state->mFastTracksGen++; 3164 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3165 if (kUseFastMixer == FastMixer_Dynamic && 3166 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3167 state->mCommand = FastMixerState::COLD_IDLE; 3168 state->mColdFutexAddr = &mFastMixerFutex; 3169 state->mColdGen++; 3170 mFastMixerFutex = 0; 3171 if (kUseFastMixer == FastMixer_Dynamic) { 3172 mNormalSink = mOutputSink; 3173 } 3174 // If we go into cold idle, need to wait for acknowledgement 3175 // so that fast mixer stops doing I/O. 3176 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3177 pauseAudioWatchdog = true; 3178 } 3179 } 3180 if (sq != NULL) { 3181 sq->end(didModify); 3182 sq->push(block); 3183 } 3184#ifdef AUDIO_WATCHDOG 3185 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3186 mAudioWatchdog->pause(); 3187 } 3188#endif 3189 3190 // Now perform the deferred reset on fast tracks that have stopped 3191 while (resetMask != 0) { 3192 size_t i = __builtin_ctz(resetMask); 3193 ALOG_ASSERT(i < count); 3194 resetMask &= ~(1 << i); 3195 sp<Track> t = mActiveTracks[i].promote(); 3196 if (t == 0) { 3197 continue; 3198 } 3199 Track* track = t.get(); 3200 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3201 track->reset(); 3202 } 3203 3204 // remove all the tracks that need to be... 3205 removeTracks_l(*tracksToRemove); 3206 3207 // mix buffer must be cleared if all tracks are connected to an 3208 // effect chain as in this case the mixer will not write to 3209 // mix buffer and track effects will accumulate into it 3210 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3211 (mixedTracks == 0 && fastTracks > 0))) { 3212 // FIXME as a performance optimization, should remember previous zero status 3213 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3214 } 3215 3216 // if any fast tracks, then status is ready 3217 mMixerStatusIgnoringFastTracks = mixerStatus; 3218 if (fastTracks > 0) { 3219 mixerStatus = MIXER_TRACKS_READY; 3220 } 3221 return mixerStatus; 3222} 3223 3224// getTrackName_l() must be called with ThreadBase::mLock held 3225int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3226{ 3227 return mAudioMixer->getTrackName(channelMask, sessionId); 3228} 3229 3230// deleteTrackName_l() must be called with ThreadBase::mLock held 3231void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3232{ 3233 ALOGV("remove track (%d) and delete from mixer", name); 3234 mAudioMixer->deleteTrackName(name); 3235} 3236 3237// checkForNewParameters_l() must be called with ThreadBase::mLock held 3238bool AudioFlinger::MixerThread::checkForNewParameters_l() 3239{ 3240 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3241 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3242 bool reconfig = false; 3243 3244 while (!mNewParameters.isEmpty()) { 3245 3246 if (mFastMixer != NULL) { 3247 FastMixerStateQueue *sq = mFastMixer->sq(); 3248 FastMixerState *state = sq->begin(); 3249 if (!(state->mCommand & FastMixerState::IDLE)) { 3250 previousCommand = state->mCommand; 3251 state->mCommand = FastMixerState::HOT_IDLE; 3252 sq->end(); 3253 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3254 } else { 3255 sq->end(false /*didModify*/); 3256 } 3257 } 3258 3259 status_t status = NO_ERROR; 3260 String8 keyValuePair = mNewParameters[0]; 3261 AudioParameter param = AudioParameter(keyValuePair); 3262 int value; 3263 3264 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3265 reconfig = true; 3266 } 3267 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3268 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3269 status = BAD_VALUE; 3270 } else { 3271 // no need to save value, since it's constant 3272 reconfig = true; 3273 } 3274 } 3275 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3276 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3277 status = BAD_VALUE; 3278 } else { 3279 // no need to save value, since it's constant 3280 reconfig = true; 3281 } 3282 } 3283 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3284 // do not accept frame count changes if tracks are open as the track buffer 3285 // size depends on frame count and correct behavior would not be guaranteed 3286 // if frame count is changed after track creation 3287 if (!mTracks.isEmpty()) { 3288 status = INVALID_OPERATION; 3289 } else { 3290 reconfig = true; 3291 } 3292 } 3293 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3294#ifdef ADD_BATTERY_DATA 3295 // when changing the audio output device, call addBatteryData to notify 3296 // the change 3297 if (mOutDevice != value) { 3298 uint32_t params = 0; 3299 // check whether speaker is on 3300 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3301 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3302 } 3303 3304 audio_devices_t deviceWithoutSpeaker 3305 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3306 // check if any other device (except speaker) is on 3307 if (value & deviceWithoutSpeaker ) { 3308 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3309 } 3310 3311 if (params != 0) { 3312 addBatteryData(params); 3313 } 3314 } 3315#endif 3316 3317 // forward device change to effects that have requested to be 3318 // aware of attached audio device. 3319 if (value != AUDIO_DEVICE_NONE) { 3320 mOutDevice = value; 3321 for (size_t i = 0; i < mEffectChains.size(); i++) { 3322 mEffectChains[i]->setDevice_l(mOutDevice); 3323 } 3324 } 3325 } 3326 3327 if (status == NO_ERROR) { 3328 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3329 keyValuePair.string()); 3330 if (!mStandby && status == INVALID_OPERATION) { 3331 mOutput->stream->common.standby(&mOutput->stream->common); 3332 mStandby = true; 3333 mBytesWritten = 0; 3334 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3335 keyValuePair.string()); 3336 } 3337 if (status == NO_ERROR && reconfig) { 3338 readOutputParameters(); 3339 delete mAudioMixer; 3340 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3341 for (size_t i = 0; i < mTracks.size() ; i++) { 3342 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3343 if (name < 0) { 3344 break; 3345 } 3346 mTracks[i]->mName = name; 3347 } 3348 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3349 } 3350 } 3351 3352 mNewParameters.removeAt(0); 3353 3354 mParamStatus = status; 3355 mParamCond.signal(); 3356 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3357 // already timed out waiting for the status and will never signal the condition. 3358 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3359 } 3360 3361 if (!(previousCommand & FastMixerState::IDLE)) { 3362 ALOG_ASSERT(mFastMixer != NULL); 3363 FastMixerStateQueue *sq = mFastMixer->sq(); 3364 FastMixerState *state = sq->begin(); 3365 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3366 state->mCommand = previousCommand; 3367 sq->end(); 3368 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3369 } 3370 3371 return reconfig; 3372} 3373 3374 3375void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3376{ 3377 const size_t SIZE = 256; 3378 char buffer[SIZE]; 3379 String8 result; 3380 3381 PlaybackThread::dumpInternals(fd, args); 3382 3383 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3384 result.append(buffer); 3385 write(fd, result.string(), result.size()); 3386 3387 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3388 const FastMixerDumpState copy(mFastMixerDumpState); 3389 copy.dump(fd); 3390 3391#ifdef STATE_QUEUE_DUMP 3392 // Similar for state queue 3393 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3394 observerCopy.dump(fd); 3395 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3396 mutatorCopy.dump(fd); 3397#endif 3398 3399#ifdef TEE_SINK 3400 // Write the tee output to a .wav file 3401 dumpTee(fd, mTeeSource, mId); 3402#endif 3403 3404#ifdef AUDIO_WATCHDOG 3405 if (mAudioWatchdog != 0) { 3406 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3407 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3408 wdCopy.dump(fd); 3409 } 3410#endif 3411} 3412 3413uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3414{ 3415 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3416} 3417 3418uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3419{ 3420 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3421} 3422 3423void AudioFlinger::MixerThread::cacheParameters_l() 3424{ 3425 PlaybackThread::cacheParameters_l(); 3426 3427 // FIXME: Relaxed timing because of a certain device that can't meet latency 3428 // Should be reduced to 2x after the vendor fixes the driver issue 3429 // increase threshold again due to low power audio mode. The way this warning 3430 // threshold is calculated and its usefulness should be reconsidered anyway. 3431 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3432} 3433 3434// ---------------------------------------------------------------------------- 3435 3436AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3437 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3438 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3439 // mLeftVolFloat, mRightVolFloat 3440{ 3441} 3442 3443AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3444 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3445 ThreadBase::type_t type) 3446 : PlaybackThread(audioFlinger, output, id, device, type) 3447 // mLeftVolFloat, mRightVolFloat 3448{ 3449} 3450 3451AudioFlinger::DirectOutputThread::~DirectOutputThread() 3452{ 3453} 3454 3455void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3456{ 3457 audio_track_cblk_t* cblk = track->cblk(); 3458 float left, right; 3459 3460 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3461 left = right = 0; 3462 } else { 3463 float typeVolume = mStreamTypes[track->streamType()].volume; 3464 float v = mMasterVolume * typeVolume; 3465 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3466 uint32_t vlr = proxy->getVolumeLR(); 3467 float v_clamped = v * (vlr & 0xFFFF); 3468 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3469 left = v_clamped/MAX_GAIN; 3470 v_clamped = v * (vlr >> 16); 3471 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3472 right = v_clamped/MAX_GAIN; 3473 } 3474 3475 if (lastTrack) { 3476 if (left != mLeftVolFloat || right != mRightVolFloat) { 3477 mLeftVolFloat = left; 3478 mRightVolFloat = right; 3479 3480 // Convert volumes from float to 8.24 3481 uint32_t vl = (uint32_t)(left * (1 << 24)); 3482 uint32_t vr = (uint32_t)(right * (1 << 24)); 3483 3484 // Delegate volume control to effect in track effect chain if needed 3485 // only one effect chain can be present on DirectOutputThread, so if 3486 // there is one, the track is connected to it 3487 if (!mEffectChains.isEmpty()) { 3488 mEffectChains[0]->setVolume_l(&vl, &vr); 3489 left = (float)vl / (1 << 24); 3490 right = (float)vr / (1 << 24); 3491 } 3492 if (mOutput->stream->set_volume) { 3493 mOutput->stream->set_volume(mOutput->stream, left, right); 3494 } 3495 } 3496 } 3497} 3498 3499 3500AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3501 Vector< sp<Track> > *tracksToRemove 3502) 3503{ 3504 size_t count = mActiveTracks.size(); 3505 mixer_state mixerStatus = MIXER_IDLE; 3506 3507 // find out which tracks need to be processed 3508 for (size_t i = 0; i < count; i++) { 3509 sp<Track> t = mActiveTracks[i].promote(); 3510 // The track died recently 3511 if (t == 0) { 3512 continue; 3513 } 3514 3515 Track* const track = t.get(); 3516 audio_track_cblk_t* cblk = track->cblk(); 3517 3518 // The first time a track is added we wait 3519 // for all its buffers to be filled before processing it 3520 uint32_t minFrames; 3521 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3522 minFrames = mNormalFrameCount; 3523 } else { 3524 minFrames = 1; 3525 } 3526 // Only consider last track started for volume and mixer state control. 3527 // This is the last entry in mActiveTracks unless a track underruns. 3528 // As we only care about the transition phase between two tracks on a 3529 // direct output, it is not a problem to ignore the underrun case. 3530 bool last = (i == (count - 1)); 3531 3532 if ((track->framesReady() >= minFrames) && track->isReady() && 3533 !track->isPaused() && !track->isTerminated()) 3534 { 3535 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3536 3537 if (track->mFillingUpStatus == Track::FS_FILLED) { 3538 track->mFillingUpStatus = Track::FS_ACTIVE; 3539 // make sure processVolume_l() will apply new volume even if 0 3540 mLeftVolFloat = mRightVolFloat = -1.0; 3541 if (track->mState == TrackBase::RESUMING) { 3542 track->mState = TrackBase::ACTIVE; 3543 } 3544 } 3545 3546 // compute volume for this track 3547 processVolume_l(track, last); 3548 if (last) { 3549 // reset retry count 3550 track->mRetryCount = kMaxTrackRetriesDirect; 3551 mActiveTrack = t; 3552 mixerStatus = MIXER_TRACKS_READY; 3553 } 3554 } else { 3555 // clear effect chain input buffer if the last active track started underruns 3556 // to avoid sending previous audio buffer again to effects 3557 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3558 mEffectChains[0]->clearInputBuffer(); 3559 } 3560 3561 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3562 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3563 track->isStopped() || track->isPaused()) { 3564 // We have consumed all the buffers of this track. 3565 // Remove it from the list of active tracks. 3566 // TODO: implement behavior for compressed audio 3567 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3568 size_t framesWritten = mBytesWritten / mFrameSize; 3569 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3570 if (track->isStopped()) { 3571 track->reset(); 3572 } 3573 tracksToRemove->add(track); 3574 } 3575 } else { 3576 // No buffers for this track. Give it a few chances to 3577 // fill a buffer, then remove it from active list. 3578 // Only consider last track started for mixer state control 3579 if (--(track->mRetryCount) <= 0) { 3580 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3581 tracksToRemove->add(track); 3582 } else if (last) { 3583 mixerStatus = MIXER_TRACKS_ENABLED; 3584 } 3585 } 3586 } 3587 } 3588 3589 // remove all the tracks that need to be... 3590 removeTracks_l(*tracksToRemove); 3591 3592 return mixerStatus; 3593} 3594 3595void AudioFlinger::DirectOutputThread::threadLoop_mix() 3596{ 3597 size_t frameCount = mFrameCount; 3598 int8_t *curBuf = (int8_t *)mMixBuffer; 3599 // output audio to hardware 3600 while (frameCount) { 3601 AudioBufferProvider::Buffer buffer; 3602 buffer.frameCount = frameCount; 3603 mActiveTrack->getNextBuffer(&buffer); 3604 if (buffer.raw == NULL) { 3605 memset(curBuf, 0, frameCount * mFrameSize); 3606 break; 3607 } 3608 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3609 frameCount -= buffer.frameCount; 3610 curBuf += buffer.frameCount * mFrameSize; 3611 mActiveTrack->releaseBuffer(&buffer); 3612 } 3613 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3614 sleepTime = 0; 3615 standbyTime = systemTime() + standbyDelay; 3616 mActiveTrack.clear(); 3617} 3618 3619void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3620{ 3621 if (sleepTime == 0) { 3622 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3623 sleepTime = activeSleepTime; 3624 } else { 3625 sleepTime = idleSleepTime; 3626 } 3627 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3628 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3629 sleepTime = 0; 3630 } 3631} 3632 3633// getTrackName_l() must be called with ThreadBase::mLock held 3634int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3635 int sessionId) 3636{ 3637 return 0; 3638} 3639 3640// deleteTrackName_l() must be called with ThreadBase::mLock held 3641void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3642{ 3643} 3644 3645// checkForNewParameters_l() must be called with ThreadBase::mLock held 3646bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3647{ 3648 bool reconfig = false; 3649 3650 while (!mNewParameters.isEmpty()) { 3651 status_t status = NO_ERROR; 3652 String8 keyValuePair = mNewParameters[0]; 3653 AudioParameter param = AudioParameter(keyValuePair); 3654 int value; 3655 3656 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3657 // do not accept frame count changes if tracks are open as the track buffer 3658 // size depends on frame count and correct behavior would not be garantied 3659 // if frame count is changed after track creation 3660 if (!mTracks.isEmpty()) { 3661 status = INVALID_OPERATION; 3662 } else { 3663 reconfig = true; 3664 } 3665 } 3666 if (status == NO_ERROR) { 3667 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3668 keyValuePair.string()); 3669 if (!mStandby && status == INVALID_OPERATION) { 3670 mOutput->stream->common.standby(&mOutput->stream->common); 3671 mStandby = true; 3672 mBytesWritten = 0; 3673 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3674 keyValuePair.string()); 3675 } 3676 if (status == NO_ERROR && reconfig) { 3677 readOutputParameters(); 3678 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3679 } 3680 } 3681 3682 mNewParameters.removeAt(0); 3683 3684 mParamStatus = status; 3685 mParamCond.signal(); 3686 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3687 // already timed out waiting for the status and will never signal the condition. 3688 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3689 } 3690 return reconfig; 3691} 3692 3693uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3694{ 3695 uint32_t time; 3696 if (audio_is_linear_pcm(mFormat)) { 3697 time = PlaybackThread::activeSleepTimeUs(); 3698 } else { 3699 time = 10000; 3700 } 3701 return time; 3702} 3703 3704uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3705{ 3706 uint32_t time; 3707 if (audio_is_linear_pcm(mFormat)) { 3708 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3709 } else { 3710 time = 10000; 3711 } 3712 return time; 3713} 3714 3715uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3716{ 3717 uint32_t time; 3718 if (audio_is_linear_pcm(mFormat)) { 3719 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3720 } else { 3721 time = 10000; 3722 } 3723 return time; 3724} 3725 3726void AudioFlinger::DirectOutputThread::cacheParameters_l() 3727{ 3728 PlaybackThread::cacheParameters_l(); 3729 3730 // use shorter standby delay as on normal output to release 3731 // hardware resources as soon as possible 3732 if (audio_is_linear_pcm(mFormat)) { 3733 standbyDelay = microseconds(activeSleepTime*2); 3734 } else { 3735 standbyDelay = kOffloadStandbyDelayNs; 3736 } 3737} 3738 3739// ---------------------------------------------------------------------------- 3740 3741AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3742 const sp<AudioFlinger::OffloadThread>& offloadThread) 3743 : Thread(false /*canCallJava*/), 3744 mOffloadThread(offloadThread), 3745 mWriteAckSequence(0), 3746 mDrainSequence(0) 3747{ 3748} 3749 3750AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3751{ 3752} 3753 3754void AudioFlinger::AsyncCallbackThread::onFirstRef() 3755{ 3756 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3757} 3758 3759bool AudioFlinger::AsyncCallbackThread::threadLoop() 3760{ 3761 while (!exitPending()) { 3762 uint32_t writeAckSequence; 3763 uint32_t drainSequence; 3764 3765 { 3766 Mutex::Autolock _l(mLock); 3767 mWaitWorkCV.wait(mLock); 3768 if (exitPending()) { 3769 break; 3770 } 3771 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3772 mWriteAckSequence, mDrainSequence); 3773 writeAckSequence = mWriteAckSequence; 3774 mWriteAckSequence &= ~1; 3775 drainSequence = mDrainSequence; 3776 mDrainSequence &= ~1; 3777 } 3778 { 3779 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3780 if (offloadThread != 0) { 3781 if (writeAckSequence & 1) { 3782 offloadThread->resetWriteBlocked(writeAckSequence >> 1); 3783 } 3784 if (drainSequence & 1) { 3785 offloadThread->resetDraining(drainSequence >> 1); 3786 } 3787 } 3788 } 3789 } 3790 return false; 3791} 3792 3793void AudioFlinger::AsyncCallbackThread::exit() 3794{ 3795 ALOGV("AsyncCallbackThread::exit"); 3796 Mutex::Autolock _l(mLock); 3797 requestExit(); 3798 mWaitWorkCV.broadcast(); 3799} 3800 3801void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3802{ 3803 Mutex::Autolock _l(mLock); 3804 // bit 0 is cleared 3805 mWriteAckSequence = sequence << 1; 3806} 3807 3808void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3809{ 3810 Mutex::Autolock _l(mLock); 3811 // ignore unexpected callbacks 3812 if (mWriteAckSequence & 2) { 3813 mWriteAckSequence |= 1; 3814 mWaitWorkCV.signal(); 3815 } 3816} 3817 3818void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3819{ 3820 Mutex::Autolock _l(mLock); 3821 // bit 0 is cleared 3822 mDrainSequence = sequence << 1; 3823} 3824 3825void AudioFlinger::AsyncCallbackThread::resetDraining() 3826{ 3827 Mutex::Autolock _l(mLock); 3828 // ignore unexpected callbacks 3829 if (mDrainSequence & 2) { 3830 mDrainSequence |= 1; 3831 mWaitWorkCV.signal(); 3832 } 3833} 3834 3835 3836// ---------------------------------------------------------------------------- 3837AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3838 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3839 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3840 mHwPaused(false), 3841 mPausedBytesRemaining(0) 3842{ 3843 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3844} 3845 3846AudioFlinger::OffloadThread::~OffloadThread() 3847{ 3848 mPreviousTrack.clear(); 3849} 3850 3851void AudioFlinger::OffloadThread::threadLoop_exit() 3852{ 3853 if (mFlushPending || mHwPaused) { 3854 // If a flush is pending or track was paused, just discard buffered data 3855 flushHw_l(); 3856 } else { 3857 mMixerStatus = MIXER_DRAIN_ALL; 3858 threadLoop_drain(); 3859 } 3860 mCallbackThread->exit(); 3861 PlaybackThread::threadLoop_exit(); 3862} 3863 3864AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3865 Vector< sp<Track> > *tracksToRemove 3866) 3867{ 3868 ALOGV("OffloadThread::prepareTracks_l"); 3869 size_t count = mActiveTracks.size(); 3870 3871 mixer_state mixerStatus = MIXER_IDLE; 3872 bool doHwPause = false; 3873 bool doHwResume = false; 3874 3875 // find out which tracks need to be processed 3876 for (size_t i = 0; i < count; i++) { 3877 sp<Track> t = mActiveTracks[i].promote(); 3878 // The track died recently 3879 if (t == 0) { 3880 continue; 3881 } 3882 Track* const track = t.get(); 3883 audio_track_cblk_t* cblk = track->cblk(); 3884 if (mPreviousTrack != NULL) { 3885 if (t != mPreviousTrack) { 3886 // Flush any data still being written from last track 3887 mBytesRemaining = 0; 3888 if (mPausedBytesRemaining) { 3889 // Last track was paused so we also need to flush saved 3890 // mixbuffer state and invalidate track so that it will 3891 // re-submit that unwritten data when it is next resumed 3892 mPausedBytesRemaining = 0; 3893 // Invalidate is a bit drastic - would be more efficient 3894 // to have a flag to tell client that some of the 3895 // previously written data was lost 3896 mPreviousTrack->invalidate(); 3897 } 3898 } 3899 } 3900 mPreviousTrack = t; 3901 bool last = (i == (count - 1)); 3902 if (track->isPausing()) { 3903 track->setPaused(); 3904 if (last) { 3905 if (!mHwPaused) { 3906 doHwPause = true; 3907 mHwPaused = true; 3908 } 3909 // If we were part way through writing the mixbuffer to 3910 // the HAL we must save this until we resume 3911 // BUG - this will be wrong if a different track is made active, 3912 // in that case we want to discard the pending data in the 3913 // mixbuffer and tell the client to present it again when the 3914 // track is resumed 3915 mPausedWriteLength = mCurrentWriteLength; 3916 mPausedBytesRemaining = mBytesRemaining; 3917 mBytesRemaining = 0; // stop writing 3918 } 3919 tracksToRemove->add(track); 3920 } else if (track->framesReady() && track->isReady() && 3921 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3922 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3923 if (track->mFillingUpStatus == Track::FS_FILLED) { 3924 track->mFillingUpStatus = Track::FS_ACTIVE; 3925 // make sure processVolume_l() will apply new volume even if 0 3926 mLeftVolFloat = mRightVolFloat = -1.0; 3927 if (track->mState == TrackBase::RESUMING) { 3928 if (mPausedBytesRemaining) { 3929 // Need to continue write that was interrupted 3930 mCurrentWriteLength = mPausedWriteLength; 3931 mBytesRemaining = mPausedBytesRemaining; 3932 mPausedBytesRemaining = 0; 3933 } 3934 track->mState = TrackBase::ACTIVE; 3935 } 3936 } 3937 3938 if (last) { 3939 if (mHwPaused) { 3940 doHwResume = true; 3941 mHwPaused = false; 3942 // threadLoop_mix() will handle the case that we need to 3943 // resume an interrupted write 3944 } 3945 // reset retry count 3946 track->mRetryCount = kMaxTrackRetriesOffload; 3947 mActiveTrack = t; 3948 mixerStatus = MIXER_TRACKS_READY; 3949 } 3950 } else { 3951 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3952 if (track->isStopping_1()) { 3953 // Hardware buffer can hold a large amount of audio so we must 3954 // wait for all current track's data to drain before we say 3955 // that the track is stopped. 3956 if (mBytesRemaining == 0) { 3957 // Only start draining when all data in mixbuffer 3958 // has been written 3959 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3960 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3961 sleepTime = 0; 3962 standbyTime = systemTime() + standbyDelay; 3963 if (last) { 3964 mixerStatus = MIXER_DRAIN_TRACK; 3965 mDrainSequence += 2; 3966 if (mHwPaused) { 3967 // It is possible to move from PAUSED to STOPPING_1 without 3968 // a resume so we must ensure hardware is running 3969 mOutput->stream->resume(mOutput->stream); 3970 mHwPaused = false; 3971 } 3972 } 3973 } 3974 } else if (track->isStopping_2()) { 3975 // Drain has completed, signal presentation complete 3976 if (!(mDrainSequence & 1) || !last) { 3977 track->mState = TrackBase::STOPPED; 3978 size_t audioHALFrames = 3979 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3980 size_t framesWritten = 3981 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3982 track->presentationComplete(framesWritten, audioHALFrames); 3983 track->reset(); 3984 tracksToRemove->add(track); 3985 } 3986 } else { 3987 // No buffers for this track. Give it a few chances to 3988 // fill a buffer, then remove it from active list. 3989 if (--(track->mRetryCount) <= 0) { 3990 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3991 track->name()); 3992 tracksToRemove->add(track); 3993 } else if (last){ 3994 mixerStatus = MIXER_TRACKS_ENABLED; 3995 } 3996 } 3997 } 3998 // compute volume for this track 3999 processVolume_l(track, last); 4000 } 4001 4002 // make sure the pause/flush/resume sequence is executed in the right order 4003 if (doHwPause) { 4004 mOutput->stream->pause(mOutput->stream); 4005 } 4006 if (mFlushPending) { 4007 flushHw_l(); 4008 mFlushPending = false; 4009 } 4010 if (doHwResume) { 4011 mOutput->stream->resume(mOutput->stream); 4012 } 4013 4014 // remove all the tracks that need to be... 4015 removeTracks_l(*tracksToRemove); 4016 4017 return mixerStatus; 4018} 4019 4020void AudioFlinger::OffloadThread::flushOutput_l() 4021{ 4022 mFlushPending = true; 4023} 4024 4025// must be called with thread mutex locked 4026bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4027{ 4028 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4029 mWriteAckSequence, mDrainSequence); 4030 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4031 return true; 4032 } 4033 return false; 4034} 4035 4036// must be called with thread mutex locked 4037bool AudioFlinger::OffloadThread::shouldStandby_l() 4038{ 4039 bool TrackPaused = false; 4040 4041 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4042 // after a timeout and we will enter standby then. 4043 if (mTracks.size() > 0) { 4044 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4045 } 4046 4047 return !mStandby && !TrackPaused; 4048} 4049 4050 4051bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4052{ 4053 Mutex::Autolock _l(mLock); 4054 return waitingAsyncCallback_l(); 4055} 4056 4057void AudioFlinger::OffloadThread::flushHw_l() 4058{ 4059 mOutput->stream->flush(mOutput->stream); 4060 // Flush anything still waiting in the mixbuffer 4061 mCurrentWriteLength = 0; 4062 mBytesRemaining = 0; 4063 mPausedWriteLength = 0; 4064 mPausedBytesRemaining = 0; 4065 if (mUseAsyncWrite) { 4066 // discard any pending drain or write ack by incrementing sequence 4067 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4068 mDrainSequence = (mDrainSequence + 2) & ~1; 4069 ALOG_ASSERT(mCallbackThread != 0); 4070 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4071 mCallbackThread->setDraining(mDrainSequence); 4072 } 4073} 4074 4075// ---------------------------------------------------------------------------- 4076 4077AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4078 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4079 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4080 DUPLICATING), 4081 mWaitTimeMs(UINT_MAX) 4082{ 4083 addOutputTrack(mainThread); 4084} 4085 4086AudioFlinger::DuplicatingThread::~DuplicatingThread() 4087{ 4088 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4089 mOutputTracks[i]->destroy(); 4090 } 4091} 4092 4093void AudioFlinger::DuplicatingThread::threadLoop_mix() 4094{ 4095 // mix buffers... 4096 if (outputsReady(outputTracks)) { 4097 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4098 } else { 4099 memset(mMixBuffer, 0, mixBufferSize); 4100 } 4101 sleepTime = 0; 4102 writeFrames = mNormalFrameCount; 4103 mCurrentWriteLength = mixBufferSize; 4104 standbyTime = systemTime() + standbyDelay; 4105} 4106 4107void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4108{ 4109 if (sleepTime == 0) { 4110 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4111 sleepTime = activeSleepTime; 4112 } else { 4113 sleepTime = idleSleepTime; 4114 } 4115 } else if (mBytesWritten != 0) { 4116 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4117 writeFrames = mNormalFrameCount; 4118 memset(mMixBuffer, 0, mixBufferSize); 4119 } else { 4120 // flush remaining overflow buffers in output tracks 4121 writeFrames = 0; 4122 } 4123 sleepTime = 0; 4124 } 4125} 4126 4127ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4128{ 4129 for (size_t i = 0; i < outputTracks.size(); i++) { 4130 outputTracks[i]->write(mMixBuffer, writeFrames); 4131 } 4132 return (ssize_t)mixBufferSize; 4133} 4134 4135void AudioFlinger::DuplicatingThread::threadLoop_standby() 4136{ 4137 // DuplicatingThread implements standby by stopping all tracks 4138 for (size_t i = 0; i < outputTracks.size(); i++) { 4139 outputTracks[i]->stop(); 4140 } 4141} 4142 4143void AudioFlinger::DuplicatingThread::saveOutputTracks() 4144{ 4145 outputTracks = mOutputTracks; 4146} 4147 4148void AudioFlinger::DuplicatingThread::clearOutputTracks() 4149{ 4150 outputTracks.clear(); 4151} 4152 4153void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4154{ 4155 Mutex::Autolock _l(mLock); 4156 // FIXME explain this formula 4157 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4158 OutputTrack *outputTrack = new OutputTrack(thread, 4159 this, 4160 mSampleRate, 4161 mFormat, 4162 mChannelMask, 4163 frameCount); 4164 if (outputTrack->cblk() != NULL) { 4165 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4166 mOutputTracks.add(outputTrack); 4167 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4168 updateWaitTime_l(); 4169 } 4170} 4171 4172void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4173{ 4174 Mutex::Autolock _l(mLock); 4175 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4176 if (mOutputTracks[i]->thread() == thread) { 4177 mOutputTracks[i]->destroy(); 4178 mOutputTracks.removeAt(i); 4179 updateWaitTime_l(); 4180 return; 4181 } 4182 } 4183 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4184} 4185 4186// caller must hold mLock 4187void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4188{ 4189 mWaitTimeMs = UINT_MAX; 4190 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4191 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4192 if (strong != 0) { 4193 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4194 if (waitTimeMs < mWaitTimeMs) { 4195 mWaitTimeMs = waitTimeMs; 4196 } 4197 } 4198 } 4199} 4200 4201 4202bool AudioFlinger::DuplicatingThread::outputsReady( 4203 const SortedVector< sp<OutputTrack> > &outputTracks) 4204{ 4205 for (size_t i = 0; i < outputTracks.size(); i++) { 4206 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4207 if (thread == 0) { 4208 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4209 outputTracks[i].get()); 4210 return false; 4211 } 4212 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4213 // see note at standby() declaration 4214 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4215 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4216 thread.get()); 4217 return false; 4218 } 4219 } 4220 return true; 4221} 4222 4223uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4224{ 4225 return (mWaitTimeMs * 1000) / 2; 4226} 4227 4228void AudioFlinger::DuplicatingThread::cacheParameters_l() 4229{ 4230 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4231 updateWaitTime_l(); 4232 4233 MixerThread::cacheParameters_l(); 4234} 4235 4236// ---------------------------------------------------------------------------- 4237// Record 4238// ---------------------------------------------------------------------------- 4239 4240AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4241 AudioStreamIn *input, 4242 uint32_t sampleRate, 4243 audio_channel_mask_t channelMask, 4244 audio_io_handle_t id, 4245 audio_devices_t outDevice, 4246 audio_devices_t inDevice 4247#ifdef TEE_SINK 4248 , const sp<NBAIO_Sink>& teeSink 4249#endif 4250 ) : 4251 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4252 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4253 // mRsmpInIndex set by readInputParameters() 4254 mReqChannelCount(popcount(channelMask)), 4255 mReqSampleRate(sampleRate) 4256 // mBytesRead is only meaningful while active, and so is cleared in start() 4257 // (but might be better to also clear here for dump?) 4258#ifdef TEE_SINK 4259 , mTeeSink(teeSink) 4260#endif 4261{ 4262 snprintf(mName, kNameLength, "AudioIn_%X", id); 4263 4264 readInputParameters(); 4265 4266} 4267 4268 4269AudioFlinger::RecordThread::~RecordThread() 4270{ 4271 delete[] mRsmpInBuffer; 4272 delete mResampler; 4273 delete[] mRsmpOutBuffer; 4274} 4275 4276void AudioFlinger::RecordThread::onFirstRef() 4277{ 4278 run(mName, PRIORITY_URGENT_AUDIO); 4279} 4280 4281bool AudioFlinger::RecordThread::threadLoop() 4282{ 4283 AudioBufferProvider::Buffer buffer; 4284 4285 nsecs_t lastWarning = 0; 4286 4287 inputStandBy(); 4288 acquireWakeLock(); 4289 4290 // used to verify we've read at least once before evaluating how many bytes were read 4291 bool readOnce = false; 4292 4293 // used to request a deferred sleep, to be executed later while mutex is unlocked 4294 bool doSleep = false; 4295 4296 // start recording 4297 for (;;) { 4298 sp<RecordTrack> activeTrack; 4299 TrackBase::track_state activeTrackState; 4300 Vector< sp<EffectChain> > effectChains; 4301 4302 // sleep with mutex unlocked 4303 if (doSleep) { 4304 doSleep = false; 4305 usleep(kRecordThreadSleepUs); 4306 } 4307 4308 { // scope for mLock 4309 Mutex::Autolock _l(mLock); 4310 if (exitPending()) { 4311 break; 4312 } 4313 processConfigEvents_l(); 4314 // return value 'reconfig' is currently unused 4315 bool reconfig = checkForNewParameters_l(); 4316 // make a stable copy of mActiveTrack 4317 activeTrack = mActiveTrack; 4318 if (activeTrack == 0) { 4319 standby(); 4320 // exitPending() can't become true here 4321 releaseWakeLock_l(); 4322 ALOGV("RecordThread: loop stopping"); 4323 // go to sleep 4324 mWaitWorkCV.wait(mLock); 4325 ALOGV("RecordThread: loop starting"); 4326 acquireWakeLock_l(); 4327 continue; 4328 } 4329 4330 if (activeTrack->isTerminated()) { 4331 removeTrack_l(activeTrack); 4332 mActiveTrack.clear(); 4333 continue; 4334 } 4335 4336 activeTrackState = activeTrack->mState; 4337 switch (activeTrackState) { 4338 case TrackBase::PAUSING: 4339 standby(); 4340 mActiveTrack.clear(); 4341 mStartStopCond.broadcast(); 4342 doSleep = true; 4343 continue; 4344 4345 case TrackBase::RESUMING: 4346 mStandby = false; 4347 if (mReqChannelCount != activeTrack->channelCount()) { 4348 mActiveTrack.clear(); 4349 mStartStopCond.broadcast(); 4350 continue; 4351 } 4352 if (readOnce) { 4353 mStartStopCond.broadcast(); 4354 // record start succeeds only if first read from audio input succeeds 4355 if (mBytesRead < 0) { 4356 mActiveTrack.clear(); 4357 continue; 4358 } 4359 activeTrack->mState = TrackBase::ACTIVE; 4360 } 4361 break; 4362 4363 case TrackBase::ACTIVE: 4364 break; 4365 4366 case TrackBase::IDLE: 4367 doSleep = true; 4368 continue; 4369 4370 default: 4371 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4372 } 4373 4374 lockEffectChains_l(effectChains); 4375 } 4376 4377 // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable 4378 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4379 4380 for (size_t i = 0; i < effectChains.size(); i ++) { 4381 // thread mutex is not locked, but effect chain is locked 4382 effectChains[i]->process_l(); 4383 } 4384 4385 buffer.frameCount = mFrameCount; 4386 status_t status = activeTrack->getNextBuffer(&buffer); 4387 if (status == NO_ERROR) { 4388 readOnce = true; 4389 size_t framesOut = buffer.frameCount; 4390 if (mResampler == NULL) { 4391 // no resampling 4392 while (framesOut) { 4393 size_t framesIn = mFrameCount - mRsmpInIndex; 4394 if (framesIn > 0) { 4395 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4396 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4397 activeTrack->mFrameSize; 4398 if (framesIn > framesOut) { 4399 framesIn = framesOut; 4400 } 4401 mRsmpInIndex += framesIn; 4402 framesOut -= framesIn; 4403 if (mChannelCount == mReqChannelCount) { 4404 memcpy(dst, src, framesIn * mFrameSize); 4405 } else { 4406 if (mChannelCount == 1) { 4407 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4408 (int16_t *)src, framesIn); 4409 } else { 4410 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4411 (int16_t *)src, framesIn); 4412 } 4413 } 4414 } 4415 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4416 void *readInto; 4417 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4418 readInto = buffer.raw; 4419 framesOut = 0; 4420 } else { 4421 readInto = mRsmpInBuffer; 4422 mRsmpInIndex = 0; 4423 } 4424 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4425 mBufferSize); 4426 if (mBytesRead <= 0) { 4427 // TODO: verify that it's benign to use a stale track state 4428 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4429 { 4430 ALOGE("Error reading audio input"); 4431 // Force input into standby so that it tries to 4432 // recover at next read attempt 4433 inputStandBy(); 4434 doSleep = true; 4435 } 4436 mRsmpInIndex = mFrameCount; 4437 framesOut = 0; 4438 buffer.frameCount = 0; 4439 } 4440#ifdef TEE_SINK 4441 else if (mTeeSink != 0) { 4442 (void) mTeeSink->write(readInto, 4443 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4444 } 4445#endif 4446 } 4447 } 4448 } else { 4449 // resampling 4450 4451 // resampler accumulates, but we only have one source track 4452 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4453 // alter output frame count as if we were expecting stereo samples 4454 if (mChannelCount == 1 && mReqChannelCount == 1) { 4455 framesOut >>= 1; 4456 } 4457 mResampler->resample(mRsmpOutBuffer, framesOut, 4458 this /* AudioBufferProvider* */); 4459 // ditherAndClamp() works as long as all buffers returned by 4460 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4461 if (mChannelCount == 2 && mReqChannelCount == 1) { 4462 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4463 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4464 // the resampler always outputs stereo samples: 4465 // do post stereo to mono conversion 4466 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4467 framesOut); 4468 } else { 4469 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4470 } 4471 // now done with mRsmpOutBuffer 4472 4473 } 4474 if (mFramestoDrop == 0) { 4475 activeTrack->releaseBuffer(&buffer); 4476 } else { 4477 if (mFramestoDrop > 0) { 4478 mFramestoDrop -= buffer.frameCount; 4479 if (mFramestoDrop <= 0) { 4480 clearSyncStartEvent(); 4481 } 4482 } else { 4483 mFramestoDrop += buffer.frameCount; 4484 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4485 mSyncStartEvent->isCancelled()) { 4486 ALOGW("Synced record %s, session %d, trigger session %d", 4487 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4488 activeTrack->sessionId(), 4489 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4490 clearSyncStartEvent(); 4491 } 4492 } 4493 } 4494 activeTrack->clearOverflow(); 4495 } 4496 // client isn't retrieving buffers fast enough 4497 else { 4498 if (!activeTrack->setOverflow()) { 4499 nsecs_t now = systemTime(); 4500 if ((now - lastWarning) > kWarningThrottleNs) { 4501 ALOGW("RecordThread: buffer overflow"); 4502 lastWarning = now; 4503 } 4504 } 4505 // Release the processor for a while before asking for a new buffer. 4506 // This will give the application more chance to read from the buffer and 4507 // clear the overflow. 4508 doSleep = true; 4509 } 4510 4511 // enable changes in effect chain 4512 unlockEffectChains(effectChains); 4513 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4514 } 4515 4516 standby(); 4517 4518 { 4519 Mutex::Autolock _l(mLock); 4520 for (size_t i = 0; i < mTracks.size(); i++) { 4521 sp<RecordTrack> track = mTracks[i]; 4522 track->invalidate(); 4523 } 4524 mActiveTrack.clear(); 4525 mStartStopCond.broadcast(); 4526 } 4527 4528 releaseWakeLock(); 4529 4530 ALOGV("RecordThread %p exiting", this); 4531 return false; 4532} 4533 4534void AudioFlinger::RecordThread::standby() 4535{ 4536 if (!mStandby) { 4537 inputStandBy(); 4538 mStandby = true; 4539 } 4540} 4541 4542void AudioFlinger::RecordThread::inputStandBy() 4543{ 4544 mInput->stream->common.standby(&mInput->stream->common); 4545} 4546 4547sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4548 const sp<AudioFlinger::Client>& client, 4549 uint32_t sampleRate, 4550 audio_format_t format, 4551 audio_channel_mask_t channelMask, 4552 size_t frameCount, 4553 int sessionId, 4554 IAudioFlinger::track_flags_t *flags, 4555 pid_t tid, 4556 status_t *status) 4557{ 4558 sp<RecordTrack> track; 4559 status_t lStatus; 4560 4561 lStatus = initCheck(); 4562 if (lStatus != NO_ERROR) { 4563 ALOGE("Audio driver not initialized."); 4564 goto Exit; 4565 } 4566 4567 // client expresses a preference for FAST, but we get the final say 4568 if (*flags & IAudioFlinger::TRACK_FAST) { 4569 if ( 4570 // use case: callback handler and frame count is default or at least as large as HAL 4571 ( 4572 (tid != -1) && 4573 ((frameCount == 0) || 4574 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4575 ) && 4576 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4577 // mono or stereo 4578 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4579 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4580 // hardware sample rate 4581 (sampleRate == mSampleRate) && 4582 // record thread has an associated fast recorder 4583 hasFastRecorder() 4584 // FIXME test that RecordThread for this fast track has a capable output HAL 4585 // FIXME add a permission test also? 4586 ) { 4587 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4588 if (frameCount == 0) { 4589 frameCount = mFrameCount * kFastTrackMultiplier; 4590 } 4591 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4592 frameCount, mFrameCount); 4593 } else { 4594 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4595 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4596 "hasFastRecorder=%d tid=%d", 4597 frameCount, mFrameCount, format, 4598 audio_is_linear_pcm(format), 4599 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4600 *flags &= ~IAudioFlinger::TRACK_FAST; 4601 // For compatibility with AudioRecord calculation, buffer depth is forced 4602 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4603 // This is probably too conservative, but legacy application code may depend on it. 4604 // If you change this calculation, also review the start threshold which is related. 4605 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4606 size_t mNormalFrameCount = 2048; // FIXME 4607 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4608 if (minBufCount < 2) { 4609 minBufCount = 2; 4610 } 4611 size_t minFrameCount = mNormalFrameCount * minBufCount; 4612 if (frameCount < minFrameCount) { 4613 frameCount = minFrameCount; 4614 } 4615 } 4616 } 4617 4618 // FIXME use flags and tid similar to createTrack_l() 4619 4620 { // scope for mLock 4621 Mutex::Autolock _l(mLock); 4622 4623 track = new RecordTrack(this, client, sampleRate, 4624 format, channelMask, frameCount, sessionId); 4625 4626 lStatus = track->initCheck(); 4627 if (lStatus != NO_ERROR) { 4628 track.clear(); 4629 goto Exit; 4630 } 4631 mTracks.add(track); 4632 4633 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4634 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4635 mAudioFlinger->btNrecIsOff(); 4636 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4637 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4638 4639 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4640 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4641 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4642 // so ask activity manager to do this on our behalf 4643 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4644 } 4645 } 4646 lStatus = NO_ERROR; 4647 4648Exit: 4649 *status = lStatus; 4650 return track; 4651} 4652 4653status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4654 AudioSystem::sync_event_t event, 4655 int triggerSession) 4656{ 4657 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4658 sp<ThreadBase> strongMe = this; 4659 status_t status = NO_ERROR; 4660 4661 if (event == AudioSystem::SYNC_EVENT_NONE) { 4662 clearSyncStartEvent(); 4663 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4664 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4665 triggerSession, 4666 recordTrack->sessionId(), 4667 syncStartEventCallback, 4668 this); 4669 // Sync event can be cancelled by the trigger session if the track is not in a 4670 // compatible state in which case we start record immediately 4671 if (mSyncStartEvent->isCancelled()) { 4672 clearSyncStartEvent(); 4673 } else { 4674 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4675 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4676 } 4677 } 4678 4679 { 4680 // This section is a rendezvous between binder thread executing start() and RecordThread 4681 AutoMutex lock(mLock); 4682 if (mActiveTrack != 0) { 4683 if (recordTrack != mActiveTrack.get()) { 4684 status = -EBUSY; 4685 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4686 mActiveTrack->mState = TrackBase::ACTIVE; 4687 } 4688 return status; 4689 } 4690 4691 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4692 recordTrack->mState = TrackBase::IDLE; 4693 mActiveTrack = recordTrack; 4694 mLock.unlock(); 4695 status_t status = AudioSystem::startInput(mId); 4696 mLock.lock(); 4697 // FIXME should verify that mActiveTrack is still == recordTrack 4698 if (status != NO_ERROR) { 4699 mActiveTrack.clear(); 4700 clearSyncStartEvent(); 4701 return status; 4702 } 4703 mRsmpInIndex = mFrameCount; 4704 mBytesRead = 0; 4705 if (mResampler != NULL) { 4706 mResampler->reset(); 4707 } 4708 // FIXME hijacking a playback track state name which was intended for start after pause; 4709 // here 'STARTING_2' would be more accurate 4710 mActiveTrack->mState = TrackBase::RESUMING; 4711 // signal thread to start 4712 ALOGV("Signal record thread"); 4713 mWaitWorkCV.broadcast(); 4714 // do not wait for mStartStopCond if exiting 4715 if (exitPending()) { 4716 mActiveTrack.clear(); 4717 status = INVALID_OPERATION; 4718 goto startError; 4719 } 4720 // FIXME incorrect usage of wait: no explicit predicate or loop 4721 mStartStopCond.wait(mLock); 4722 if (mActiveTrack == 0) { 4723 ALOGV("Record failed to start"); 4724 status = BAD_VALUE; 4725 goto startError; 4726 } 4727 ALOGV("Record started OK"); 4728 return status; 4729 } 4730 4731startError: 4732 AudioSystem::stopInput(mId); 4733 clearSyncStartEvent(); 4734 return status; 4735} 4736 4737void AudioFlinger::RecordThread::clearSyncStartEvent() 4738{ 4739 if (mSyncStartEvent != 0) { 4740 mSyncStartEvent->cancel(); 4741 } 4742 mSyncStartEvent.clear(); 4743 mFramestoDrop = 0; 4744} 4745 4746void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4747{ 4748 sp<SyncEvent> strongEvent = event.promote(); 4749 4750 if (strongEvent != 0) { 4751 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4752 me->handleSyncStartEvent(strongEvent); 4753 } 4754} 4755 4756void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4757{ 4758 if (event == mSyncStartEvent) { 4759 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4760 // from audio HAL 4761 mFramestoDrop = mFrameCount * 2; 4762 } 4763} 4764 4765bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4766 ALOGV("RecordThread::stop"); 4767 AutoMutex _l(mLock); 4768 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4769 return false; 4770 } 4771 // note that threadLoop may still be processing the track at this point [without lock] 4772 recordTrack->mState = TrackBase::PAUSING; 4773 // do not wait for mStartStopCond if exiting 4774 if (exitPending()) { 4775 return true; 4776 } 4777 // FIXME incorrect usage of wait: no explicit predicate or loop 4778 mStartStopCond.wait(mLock); 4779 // if we have been restarted, recordTrack == mActiveTrack.get() here 4780 if (exitPending() || recordTrack != mActiveTrack.get()) { 4781 ALOGV("Record stopped OK"); 4782 return true; 4783 } 4784 return false; 4785} 4786 4787bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4788{ 4789 return false; 4790} 4791 4792status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4793{ 4794#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4795 if (!isValidSyncEvent(event)) { 4796 return BAD_VALUE; 4797 } 4798 4799 int eventSession = event->triggerSession(); 4800 status_t ret = NAME_NOT_FOUND; 4801 4802 Mutex::Autolock _l(mLock); 4803 4804 for (size_t i = 0; i < mTracks.size(); i++) { 4805 sp<RecordTrack> track = mTracks[i]; 4806 if (eventSession == track->sessionId()) { 4807 (void) track->setSyncEvent(event); 4808 ret = NO_ERROR; 4809 } 4810 } 4811 return ret; 4812#else 4813 return BAD_VALUE; 4814#endif 4815} 4816 4817// destroyTrack_l() must be called with ThreadBase::mLock held 4818void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4819{ 4820 track->terminate(); 4821 track->mState = TrackBase::STOPPED; 4822 // active tracks are removed by threadLoop() 4823 if (mActiveTrack != track) { 4824 removeTrack_l(track); 4825 } 4826} 4827 4828void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4829{ 4830 mTracks.remove(track); 4831 // need anything related to effects here? 4832} 4833 4834void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4835{ 4836 dumpInternals(fd, args); 4837 dumpTracks(fd, args); 4838 dumpEffectChains(fd, args); 4839} 4840 4841void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4842{ 4843 const size_t SIZE = 256; 4844 char buffer[SIZE]; 4845 String8 result; 4846 4847 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4848 result.append(buffer); 4849 4850 if (mActiveTrack != 0) { 4851 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4852 result.append(buffer); 4853 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4854 result.append(buffer); 4855 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4856 result.append(buffer); 4857 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4858 result.append(buffer); 4859 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4860 result.append(buffer); 4861 } else { 4862 result.append("No active record client\n"); 4863 } 4864 4865 write(fd, result.string(), result.size()); 4866 4867 dumpBase(fd, args); 4868} 4869 4870void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4871{ 4872 const size_t SIZE = 256; 4873 char buffer[SIZE]; 4874 String8 result; 4875 4876 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4877 result.append(buffer); 4878 RecordTrack::appendDumpHeader(result); 4879 for (size_t i = 0; i < mTracks.size(); ++i) { 4880 sp<RecordTrack> track = mTracks[i]; 4881 if (track != 0) { 4882 track->dump(buffer, SIZE); 4883 result.append(buffer); 4884 } 4885 } 4886 4887 if (mActiveTrack != 0) { 4888 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4889 result.append(buffer); 4890 RecordTrack::appendDumpHeader(result); 4891 mActiveTrack->dump(buffer, SIZE); 4892 result.append(buffer); 4893 4894 } 4895 write(fd, result.string(), result.size()); 4896} 4897 4898// AudioBufferProvider interface 4899status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4900{ 4901 size_t framesReq = buffer->frameCount; 4902 size_t framesReady = mFrameCount - mRsmpInIndex; 4903 int channelCount; 4904 4905 if (framesReady == 0) { 4906 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4907 if (mBytesRead <= 0) { 4908 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4909 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4910 // Force input into standby so that it tries to 4911 // recover at next read attempt 4912 inputStandBy(); 4913 // FIXME an awkward place to sleep, consider using doSleep when this is pulled up 4914 usleep(kRecordThreadSleepUs); 4915 } 4916 buffer->raw = NULL; 4917 buffer->frameCount = 0; 4918 return NOT_ENOUGH_DATA; 4919 } 4920 mRsmpInIndex = 0; 4921 framesReady = mFrameCount; 4922 } 4923 4924 if (framesReq > framesReady) { 4925 framesReq = framesReady; 4926 } 4927 4928 if (mChannelCount == 1 && mReqChannelCount == 2) { 4929 channelCount = 1; 4930 } else { 4931 channelCount = 2; 4932 } 4933 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4934 buffer->frameCount = framesReq; 4935 return NO_ERROR; 4936} 4937 4938// AudioBufferProvider interface 4939void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4940{ 4941 mRsmpInIndex += buffer->frameCount; 4942 buffer->frameCount = 0; 4943} 4944 4945bool AudioFlinger::RecordThread::checkForNewParameters_l() 4946{ 4947 bool reconfig = false; 4948 4949 while (!mNewParameters.isEmpty()) { 4950 status_t status = NO_ERROR; 4951 String8 keyValuePair = mNewParameters[0]; 4952 AudioParameter param = AudioParameter(keyValuePair); 4953 int value; 4954 audio_format_t reqFormat = mFormat; 4955 uint32_t reqSamplingRate = mReqSampleRate; 4956 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 4957 4958 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4959 reqSamplingRate = value; 4960 reconfig = true; 4961 } 4962 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4963 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4964 status = BAD_VALUE; 4965 } else { 4966 reqFormat = (audio_format_t) value; 4967 reconfig = true; 4968 } 4969 } 4970 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4971 audio_channel_mask_t mask = (audio_channel_mask_t) value; 4972 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 4973 status = BAD_VALUE; 4974 } else { 4975 reqChannelMask = mask; 4976 reconfig = true; 4977 } 4978 } 4979 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4980 // do not accept frame count changes if tracks are open as the track buffer 4981 // size depends on frame count and correct behavior would not be guaranteed 4982 // if frame count is changed after track creation 4983 if (mActiveTrack != 0) { 4984 status = INVALID_OPERATION; 4985 } else { 4986 reconfig = true; 4987 } 4988 } 4989 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4990 // forward device change to effects that have requested to be 4991 // aware of attached audio device. 4992 for (size_t i = 0; i < mEffectChains.size(); i++) { 4993 mEffectChains[i]->setDevice_l(value); 4994 } 4995 4996 // store input device and output device but do not forward output device to audio HAL. 4997 // Note that status is ignored by the caller for output device 4998 // (see AudioFlinger::setParameters() 4999 if (audio_is_output_devices(value)) { 5000 mOutDevice = value; 5001 status = BAD_VALUE; 5002 } else { 5003 mInDevice = value; 5004 // disable AEC and NS if the device is a BT SCO headset supporting those 5005 // pre processings 5006 if (mTracks.size() > 0) { 5007 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5008 mAudioFlinger->btNrecIsOff(); 5009 for (size_t i = 0; i < mTracks.size(); i++) { 5010 sp<RecordTrack> track = mTracks[i]; 5011 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5012 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5013 } 5014 } 5015 } 5016 } 5017 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5018 mAudioSource != (audio_source_t)value) { 5019 // forward device change to effects that have requested to be 5020 // aware of attached audio device. 5021 for (size_t i = 0; i < mEffectChains.size(); i++) { 5022 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5023 } 5024 mAudioSource = (audio_source_t)value; 5025 } 5026 5027 if (status == NO_ERROR) { 5028 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5029 keyValuePair.string()); 5030 if (status == INVALID_OPERATION) { 5031 inputStandBy(); 5032 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5033 keyValuePair.string()); 5034 } 5035 if (reconfig) { 5036 if (status == BAD_VALUE && 5037 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5038 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5039 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5040 <= (2 * reqSamplingRate)) && 5041 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5042 <= FCC_2 && 5043 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5044 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5045 status = NO_ERROR; 5046 } 5047 if (status == NO_ERROR) { 5048 readInputParameters(); 5049 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5050 } 5051 } 5052 } 5053 5054 mNewParameters.removeAt(0); 5055 5056 mParamStatus = status; 5057 mParamCond.signal(); 5058 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5059 // already timed out waiting for the status and will never signal the condition. 5060 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5061 } 5062 return reconfig; 5063} 5064 5065String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5066{ 5067 Mutex::Autolock _l(mLock); 5068 if (initCheck() != NO_ERROR) { 5069 return String8(); 5070 } 5071 5072 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5073 const String8 out_s8(s); 5074 free(s); 5075 return out_s8; 5076} 5077 5078void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5079 AudioSystem::OutputDescriptor desc; 5080 void *param2 = NULL; 5081 5082 switch (event) { 5083 case AudioSystem::INPUT_OPENED: 5084 case AudioSystem::INPUT_CONFIG_CHANGED: 5085 desc.channelMask = mChannelMask; 5086 desc.samplingRate = mSampleRate; 5087 desc.format = mFormat; 5088 desc.frameCount = mFrameCount; 5089 desc.latency = 0; 5090 param2 = &desc; 5091 break; 5092 5093 case AudioSystem::INPUT_CLOSED: 5094 default: 5095 break; 5096 } 5097 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5098} 5099 5100void AudioFlinger::RecordThread::readInputParameters() 5101{ 5102 delete[] mRsmpInBuffer; 5103 // mRsmpInBuffer is always assigned a new[] below 5104 delete[] mRsmpOutBuffer; 5105 mRsmpOutBuffer = NULL; 5106 delete mResampler; 5107 mResampler = NULL; 5108 5109 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5110 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5111 mChannelCount = popcount(mChannelMask); 5112 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5113 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5114 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5115 } 5116 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5117 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5118 mFrameCount = mBufferSize / mFrameSize; 5119 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5120 5121 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5122 int channelCount; 5123 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5124 // stereo to mono post process as the resampler always outputs stereo. 5125 if (mChannelCount == 1 && mReqChannelCount == 2) { 5126 channelCount = 1; 5127 } else { 5128 channelCount = 2; 5129 } 5130 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5131 mResampler->setSampleRate(mSampleRate); 5132 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5133 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5134 5135 // optmization: if mono to mono, alter input frame count as if we were inputing 5136 // stereo samples 5137 if (mChannelCount == 1 && mReqChannelCount == 1) { 5138 mFrameCount >>= 1; 5139 } 5140 5141 } 5142 mRsmpInIndex = mFrameCount; 5143} 5144 5145unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5146{ 5147 Mutex::Autolock _l(mLock); 5148 if (initCheck() != NO_ERROR) { 5149 return 0; 5150 } 5151 5152 return mInput->stream->get_input_frames_lost(mInput->stream); 5153} 5154 5155uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5156{ 5157 Mutex::Autolock _l(mLock); 5158 uint32_t result = 0; 5159 if (getEffectChain_l(sessionId) != 0) { 5160 result = EFFECT_SESSION; 5161 } 5162 5163 for (size_t i = 0; i < mTracks.size(); ++i) { 5164 if (sessionId == mTracks[i]->sessionId()) { 5165 result |= TRACK_SESSION; 5166 break; 5167 } 5168 } 5169 5170 return result; 5171} 5172 5173KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5174{ 5175 KeyedVector<int, bool> ids; 5176 Mutex::Autolock _l(mLock); 5177 for (size_t j = 0; j < mTracks.size(); ++j) { 5178 sp<RecordThread::RecordTrack> track = mTracks[j]; 5179 int sessionId = track->sessionId(); 5180 if (ids.indexOfKey(sessionId) < 0) { 5181 ids.add(sessionId, true); 5182 } 5183 } 5184 return ids; 5185} 5186 5187AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5188{ 5189 Mutex::Autolock _l(mLock); 5190 AudioStreamIn *input = mInput; 5191 mInput = NULL; 5192 return input; 5193} 5194 5195// this method must always be called either with ThreadBase mLock held or inside the thread loop 5196audio_stream_t* AudioFlinger::RecordThread::stream() const 5197{ 5198 if (mInput == NULL) { 5199 return NULL; 5200 } 5201 return &mInput->stream->common; 5202} 5203 5204status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5205{ 5206 // only one chain per input thread 5207 if (mEffectChains.size() != 0) { 5208 return INVALID_OPERATION; 5209 } 5210 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5211 5212 chain->setInBuffer(NULL); 5213 chain->setOutBuffer(NULL); 5214 5215 checkSuspendOnAddEffectChain_l(chain); 5216 5217 mEffectChains.add(chain); 5218 5219 return NO_ERROR; 5220} 5221 5222size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5223{ 5224 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5225 ALOGW_IF(mEffectChains.size() != 1, 5226 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5227 chain.get(), mEffectChains.size(), this); 5228 if (mEffectChains.size() == 1) { 5229 mEffectChains.removeAt(0); 5230 } 5231 return 0; 5232} 5233 5234}; // namespace android 5235