Threads.cpp revision e93cf2ca27ae6f4a81d4ef548bbf10a34db6d98f
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 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 299void AudioFlinger::ThreadBase::exit() 300{ 301 ALOGV("ThreadBase::exit"); 302 // do any cleanup required for exit to succeed 303 preExit(); 304 { 305 // This lock prevents the following race in thread (uniprocessor for illustration): 306 // if (!exitPending()) { 307 // // context switch from here to exit() 308 // // exit() calls requestExit(), what exitPending() observes 309 // // exit() calls signal(), which is dropped since no waiters 310 // // context switch back from exit() to here 311 // mWaitWorkCV.wait(...); 312 // // now thread is hung 313 // } 314 AutoMutex lock(mLock); 315 requestExit(); 316 mWaitWorkCV.broadcast(); 317 } 318 // When Thread::requestExitAndWait is made virtual and this method is renamed to 319 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 320 requestExitAndWait(); 321} 322 323status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 324{ 325 status_t status; 326 327 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 328 Mutex::Autolock _l(mLock); 329 330 mNewParameters.add(keyValuePairs); 331 mWaitWorkCV.signal(); 332 // wait condition with timeout in case the thread loop has exited 333 // before the request could be processed 334 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 335 status = mParamStatus; 336 mWaitWorkCV.signal(); 337 } else { 338 status = TIMED_OUT; 339 } 340 return status; 341} 342 343void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 344{ 345 Mutex::Autolock _l(mLock); 346 sendIoConfigEvent_l(event, param); 347} 348 349// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 350void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 351{ 352 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 353 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 354 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 355 param); 356 mWaitWorkCV.signal(); 357} 358 359// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 360void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 361{ 362 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 363 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 364 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 365 mConfigEvents.size(), pid, tid, prio); 366 mWaitWorkCV.signal(); 367} 368 369void AudioFlinger::ThreadBase::processConfigEvents() 370{ 371 mLock.lock(); 372 while (!mConfigEvents.isEmpty()) { 373 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 374 ConfigEvent *event = mConfigEvents[0]; 375 mConfigEvents.removeAt(0); 376 // release mLock before locking AudioFlinger mLock: lock order is always 377 // AudioFlinger then ThreadBase to avoid cross deadlock 378 mLock.unlock(); 379 switch(event->type()) { 380 case CFG_EVENT_PRIO: { 381 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 382 // FIXME Need to understand why this has be done asynchronously 383 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 384 true /*asynchronous*/); 385 if (err != 0) { 386 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 387 "error %d", 388 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 389 } 390 } break; 391 case CFG_EVENT_IO: { 392 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 393 mAudioFlinger->mLock.lock(); 394 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 395 mAudioFlinger->mLock.unlock(); 396 } break; 397 default: 398 ALOGE("processConfigEvents() unknown event type %d", event->type()); 399 break; 400 } 401 delete event; 402 mLock.lock(); 403 } 404 mLock.unlock(); 405} 406 407void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 408{ 409 const size_t SIZE = 256; 410 char buffer[SIZE]; 411 String8 result; 412 413 bool locked = AudioFlinger::dumpTryLock(mLock); 414 if (!locked) { 415 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 416 write(fd, buffer, strlen(buffer)); 417 } 418 419 snprintf(buffer, SIZE, "io handle: %d\n", mId); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 432 result.append(buffer); 433 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 434 result.append(buffer); 435 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 436 result.append(buffer); 437 438 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 439 result.append(buffer); 440 result.append(" Index Command"); 441 for (size_t i = 0; i < mNewParameters.size(); ++i) { 442 snprintf(buffer, SIZE, "\n %02d ", i); 443 result.append(buffer); 444 result.append(mNewParameters[i]); 445 } 446 447 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 448 result.append(buffer); 449 for (size_t i = 0; i < mConfigEvents.size(); i++) { 450 mConfigEvents[i]->dump(buffer, SIZE); 451 result.append(buffer); 452 } 453 result.append("\n"); 454 455 write(fd, result.string(), result.size()); 456 457 if (locked) { 458 mLock.unlock(); 459 } 460} 461 462void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 463{ 464 const size_t SIZE = 256; 465 char buffer[SIZE]; 466 String8 result; 467 468 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 469 write(fd, buffer, strlen(buffer)); 470 471 for (size_t i = 0; i < mEffectChains.size(); ++i) { 472 sp<EffectChain> chain = mEffectChains[i]; 473 if (chain != 0) { 474 chain->dump(fd, args); 475 } 476 } 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock() 480{ 481 Mutex::Autolock _l(mLock); 482 acquireWakeLock_l(); 483} 484 485void AudioFlinger::ThreadBase::acquireWakeLock_l() 486{ 487 if (mPowerManager == 0) { 488 // use checkService() to avoid blocking if power service is not up yet 489 sp<IBinder> binder = 490 defaultServiceManager()->checkService(String16("power")); 491 if (binder == 0) { 492 ALOGW("Thread %s cannot connect to the power manager service", mName); 493 } else { 494 mPowerManager = interface_cast<IPowerManager>(binder); 495 binder->linkToDeath(mDeathRecipient); 496 } 497 } 498 if (mPowerManager != 0) { 499 sp<IBinder> binder = new BBinder(); 500 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 501 binder, 502 String16(mName), 503 String16("media")); 504 if (status == NO_ERROR) { 505 mWakeLockToken = binder; 506 } 507 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 508 } 509} 510 511void AudioFlinger::ThreadBase::releaseWakeLock() 512{ 513 Mutex::Autolock _l(mLock); 514 releaseWakeLock_l(); 515} 516 517void AudioFlinger::ThreadBase::releaseWakeLock_l() 518{ 519 if (mWakeLockToken != 0) { 520 ALOGV("releaseWakeLock_l() %s", mName); 521 if (mPowerManager != 0) { 522 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 523 } 524 mWakeLockToken.clear(); 525 } 526} 527 528void AudioFlinger::ThreadBase::clearPowerManager() 529{ 530 Mutex::Autolock _l(mLock); 531 releaseWakeLock_l(); 532 mPowerManager.clear(); 533} 534 535void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 536{ 537 sp<ThreadBase> thread = mThread.promote(); 538 if (thread != 0) { 539 thread->clearPowerManager(); 540 } 541 ALOGW("power manager service died !!!"); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 Mutex::Autolock _l(mLock); 548 setEffectSuspended_l(type, suspend, sessionId); 549} 550 551void AudioFlinger::ThreadBase::setEffectSuspended_l( 552 const effect_uuid_t *type, bool suspend, int sessionId) 553{ 554 sp<EffectChain> chain = getEffectChain_l(sessionId); 555 if (chain != 0) { 556 if (type != NULL) { 557 chain->setEffectSuspended_l(type, suspend); 558 } else { 559 chain->setEffectSuspendedAll_l(suspend); 560 } 561 } 562 563 updateSuspendedSessions_l(type, suspend, sessionId); 564} 565 566void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 567{ 568 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 569 if (index < 0) { 570 return; 571 } 572 573 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 574 mSuspendedSessions.valueAt(index); 575 576 for (size_t i = 0; i < sessionEffects.size(); i++) { 577 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 578 for (int j = 0; j < desc->mRefCount; j++) { 579 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 580 chain->setEffectSuspendedAll_l(true); 581 } else { 582 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 583 desc->mType.timeLow); 584 chain->setEffectSuspended_l(&desc->mType, true); 585 } 586 } 587 } 588} 589 590void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 591 bool suspend, 592 int sessionId) 593{ 594 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 595 596 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 597 598 if (suspend) { 599 if (index >= 0) { 600 sessionEffects = mSuspendedSessions.valueAt(index); 601 } else { 602 mSuspendedSessions.add(sessionId, sessionEffects); 603 } 604 } else { 605 if (index < 0) { 606 return; 607 } 608 sessionEffects = mSuspendedSessions.valueAt(index); 609 } 610 611 612 int key = EffectChain::kKeyForSuspendAll; 613 if (type != NULL) { 614 key = type->timeLow; 615 } 616 index = sessionEffects.indexOfKey(key); 617 618 sp<SuspendedSessionDesc> desc; 619 if (suspend) { 620 if (index >= 0) { 621 desc = sessionEffects.valueAt(index); 622 } else { 623 desc = new SuspendedSessionDesc(); 624 if (type != NULL) { 625 desc->mType = *type; 626 } 627 sessionEffects.add(key, desc); 628 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 629 } 630 desc->mRefCount++; 631 } else { 632 if (index < 0) { 633 return; 634 } 635 desc = sessionEffects.valueAt(index); 636 if (--desc->mRefCount == 0) { 637 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 638 sessionEffects.removeItemsAt(index); 639 if (sessionEffects.isEmpty()) { 640 ALOGV("updateSuspendedSessions_l() restore removing session %d", 641 sessionId); 642 mSuspendedSessions.removeItem(sessionId); 643 } 644 } 645 } 646 if (!sessionEffects.isEmpty()) { 647 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 648 } 649} 650 651void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 652 bool enabled, 653 int sessionId) 654{ 655 Mutex::Autolock _l(mLock); 656 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 657} 658 659void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 660 bool enabled, 661 int sessionId) 662{ 663 if (mType != RECORD) { 664 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 665 // another session. This gives the priority to well behaved effect control panels 666 // and applications not using global effects. 667 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 668 // global effects 669 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 670 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 671 } 672 } 673 674 sp<EffectChain> chain = getEffectChain_l(sessionId); 675 if (chain != 0) { 676 chain->checkSuspendOnEffectEnabled(effect, enabled); 677 } 678} 679 680// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 681sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 682 const sp<AudioFlinger::Client>& client, 683 const sp<IEffectClient>& effectClient, 684 int32_t priority, 685 int sessionId, 686 effect_descriptor_t *desc, 687 int *enabled, 688 status_t *status 689 ) 690{ 691 sp<EffectModule> effect; 692 sp<EffectHandle> handle; 693 status_t lStatus; 694 sp<EffectChain> chain; 695 bool chainCreated = false; 696 bool effectCreated = false; 697 bool effectRegistered = false; 698 699 lStatus = initCheck(); 700 if (lStatus != NO_ERROR) { 701 ALOGW("createEffect_l() Audio driver not initialized."); 702 goto Exit; 703 } 704 705 // Allow global effects only on offloaded and mixer threads 706 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 707 switch (mType) { 708 case MIXER: 709 case OFFLOAD: 710 break; 711 case DIRECT: 712 case DUPLICATING: 713 case RECORD: 714 default: 715 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 716 lStatus = BAD_VALUE; 717 goto Exit; 718 } 719 } 720 721 // Only Pre processor effects are allowed on input threads and only on input threads 722 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 723 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 724 desc->name, desc->flags, mType); 725 lStatus = BAD_VALUE; 726 goto Exit; 727 } 728 729 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 730 731 { // scope for mLock 732 Mutex::Autolock _l(mLock); 733 734 // check for existing effect chain with the requested audio session 735 chain = getEffectChain_l(sessionId); 736 if (chain == 0) { 737 // create a new chain for this session 738 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 739 chain = new EffectChain(this, sessionId); 740 addEffectChain_l(chain); 741 chain->setStrategy(getStrategyForSession_l(sessionId)); 742 chainCreated = true; 743 } else { 744 effect = chain->getEffectFromDesc_l(desc); 745 } 746 747 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 748 749 if (effect == 0) { 750 int id = mAudioFlinger->nextUniqueId(); 751 // Check CPU and memory usage 752 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 753 if (lStatus != NO_ERROR) { 754 goto Exit; 755 } 756 effectRegistered = true; 757 // create a new effect module if none present in the chain 758 effect = new EffectModule(this, chain, desc, id, sessionId); 759 lStatus = effect->status(); 760 if (lStatus != NO_ERROR) { 761 goto Exit; 762 } 763 effect->setOffloaded(mType == OFFLOAD, mId); 764 765 lStatus = chain->addEffect_l(effect); 766 if (lStatus != NO_ERROR) { 767 goto Exit; 768 } 769 effectCreated = true; 770 771 effect->setDevice(mOutDevice); 772 effect->setDevice(mInDevice); 773 effect->setMode(mAudioFlinger->getMode()); 774 effect->setAudioSource(mAudioSource); 775 } 776 // create effect handle and connect it to effect module 777 handle = new EffectHandle(effect, client, effectClient, priority); 778 lStatus = effect->addHandle(handle.get()); 779 if (enabled != NULL) { 780 *enabled = (int)effect->isEnabled(); 781 } 782 } 783 784Exit: 785 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 786 Mutex::Autolock _l(mLock); 787 if (effectCreated) { 788 chain->removeEffect_l(effect); 789 } 790 if (effectRegistered) { 791 AudioSystem::unregisterEffect(effect->id()); 792 } 793 if (chainCreated) { 794 removeEffectChain_l(chain); 795 } 796 handle.clear(); 797 } 798 799 if (status != NULL) { 800 *status = lStatus; 801 } 802 return handle; 803} 804 805sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 806{ 807 Mutex::Autolock _l(mLock); 808 return getEffect_l(sessionId, effectId); 809} 810 811sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 812{ 813 sp<EffectChain> chain = getEffectChain_l(sessionId); 814 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 815} 816 817// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 818// PlaybackThread::mLock held 819status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 820{ 821 // check for existing effect chain with the requested audio session 822 int sessionId = effect->sessionId(); 823 sp<EffectChain> chain = getEffectChain_l(sessionId); 824 bool chainCreated = false; 825 826 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 827 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 828 this, effect->desc().name, effect->desc().flags); 829 830 if (chain == 0) { 831 // create a new chain for this session 832 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 833 chain = new EffectChain(this, sessionId); 834 addEffectChain_l(chain); 835 chain->setStrategy(getStrategyForSession_l(sessionId)); 836 chainCreated = true; 837 } 838 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 839 840 if (chain->getEffectFromId_l(effect->id()) != 0) { 841 ALOGW("addEffect_l() %p effect %s already present in chain %p", 842 this, effect->desc().name, chain.get()); 843 return BAD_VALUE; 844 } 845 846 effect->setOffloaded(mType == OFFLOAD, mId); 847 848 status_t status = chain->addEffect_l(effect); 849 if (status != NO_ERROR) { 850 if (chainCreated) { 851 removeEffectChain_l(chain); 852 } 853 return status; 854 } 855 856 effect->setDevice(mOutDevice); 857 effect->setDevice(mInDevice); 858 effect->setMode(mAudioFlinger->getMode()); 859 effect->setAudioSource(mAudioSource); 860 return NO_ERROR; 861} 862 863void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 864 865 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 866 effect_descriptor_t desc = effect->desc(); 867 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 868 detachAuxEffect_l(effect->id()); 869 } 870 871 sp<EffectChain> chain = effect->chain().promote(); 872 if (chain != 0) { 873 // remove effect chain if removing last effect 874 if (chain->removeEffect_l(effect) == 0) { 875 removeEffectChain_l(chain); 876 } 877 } else { 878 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 879 } 880} 881 882void AudioFlinger::ThreadBase::lockEffectChains_l( 883 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 884{ 885 effectChains = mEffectChains; 886 for (size_t i = 0; i < mEffectChains.size(); i++) { 887 mEffectChains[i]->lock(); 888 } 889} 890 891void AudioFlinger::ThreadBase::unlockEffectChains( 892 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 893{ 894 for (size_t i = 0; i < effectChains.size(); i++) { 895 effectChains[i]->unlock(); 896 } 897} 898 899sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 900{ 901 Mutex::Autolock _l(mLock); 902 return getEffectChain_l(sessionId); 903} 904 905sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 906{ 907 size_t size = mEffectChains.size(); 908 for (size_t i = 0; i < size; i++) { 909 if (mEffectChains[i]->sessionId() == sessionId) { 910 return mEffectChains[i]; 911 } 912 } 913 return 0; 914} 915 916void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 917{ 918 Mutex::Autolock _l(mLock); 919 size_t size = mEffectChains.size(); 920 for (size_t i = 0; i < size; i++) { 921 mEffectChains[i]->setMode_l(mode); 922 } 923} 924 925void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 926 EffectHandle *handle, 927 bool unpinIfLast) { 928 929 Mutex::Autolock _l(mLock); 930 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 931 // delete the effect module if removing last handle on it 932 if (effect->removeHandle(handle) == 0) { 933 if (!effect->isPinned() || unpinIfLast) { 934 removeEffect_l(effect); 935 AudioSystem::unregisterEffect(effect->id()); 936 } 937 } 938} 939 940// ---------------------------------------------------------------------------- 941// Playback 942// ---------------------------------------------------------------------------- 943 944AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 945 AudioStreamOut* output, 946 audio_io_handle_t id, 947 audio_devices_t device, 948 type_t type) 949 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 950 mNormalFrameCount(0), mMixBuffer(NULL), 951 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 952 // mStreamTypes[] initialized in constructor body 953 mOutput(output), 954 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 955 mMixerStatus(MIXER_IDLE), 956 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 957 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 958 mBytesRemaining(0), 959 mCurrentWriteLength(0), 960 mUseAsyncWrite(false), 961 mWriteAckSequence(0), 962 mDrainSequence(0), 963 mSignalPending(false), 964 mScreenState(AudioFlinger::mScreenState), 965 // index 0 is reserved for normal mixer's submix 966 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 967 // mLatchD, mLatchQ, 968 mLatchDValid(false), mLatchQValid(false) 969{ 970 snprintf(mName, kNameLength, "AudioOut_%X", id); 971 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 972 973 // Assumes constructor is called by AudioFlinger with it's mLock held, but 974 // it would be safer to explicitly pass initial masterVolume/masterMute as 975 // parameter. 976 // 977 // If the HAL we are using has support for master volume or master mute, 978 // then do not attenuate or mute during mixing (just leave the volume at 1.0 979 // and the mute set to false). 980 mMasterVolume = audioFlinger->masterVolume_l(); 981 mMasterMute = audioFlinger->masterMute_l(); 982 if (mOutput && mOutput->audioHwDev) { 983 if (mOutput->audioHwDev->canSetMasterVolume()) { 984 mMasterVolume = 1.0; 985 } 986 987 if (mOutput->audioHwDev->canSetMasterMute()) { 988 mMasterMute = false; 989 } 990 } 991 992 readOutputParameters(); 993 994 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 995 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 996 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 997 stream = (audio_stream_type_t) (stream + 1)) { 998 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 999 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1000 } 1001 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1002 // because mAudioFlinger doesn't have one to copy from 1003} 1004 1005AudioFlinger::PlaybackThread::~PlaybackThread() 1006{ 1007 mAudioFlinger->unregisterWriter(mNBLogWriter); 1008 delete [] mAllocMixBuffer; 1009} 1010 1011void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1012{ 1013 dumpInternals(fd, args); 1014 dumpTracks(fd, args); 1015 dumpEffectChains(fd, args); 1016} 1017 1018void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1019{ 1020 const size_t SIZE = 256; 1021 char buffer[SIZE]; 1022 String8 result; 1023 1024 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1025 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1026 const stream_type_t *st = &mStreamTypes[i]; 1027 if (i > 0) { 1028 result.appendFormat(", "); 1029 } 1030 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1031 if (st->mute) { 1032 result.append("M"); 1033 } 1034 } 1035 result.append("\n"); 1036 write(fd, result.string(), result.length()); 1037 result.clear(); 1038 1039 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1040 result.append(buffer); 1041 Track::appendDumpHeader(result); 1042 for (size_t i = 0; i < mTracks.size(); ++i) { 1043 sp<Track> track = mTracks[i]; 1044 if (track != 0) { 1045 track->dump(buffer, SIZE); 1046 result.append(buffer); 1047 } 1048 } 1049 1050 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1051 result.append(buffer); 1052 Track::appendDumpHeader(result); 1053 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1054 sp<Track> track = mActiveTracks[i].promote(); 1055 if (track != 0) { 1056 track->dump(buffer, SIZE); 1057 result.append(buffer); 1058 } 1059 } 1060 write(fd, result.string(), result.size()); 1061 1062 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1063 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1064 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1065 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1066} 1067 1068void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1069{ 1070 const size_t SIZE = 256; 1071 char buffer[SIZE]; 1072 String8 result; 1073 1074 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1079 ns2ms(systemTime() - mLastWriteTime)); 1080 result.append(buffer); 1081 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1082 result.append(buffer); 1083 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1084 result.append(buffer); 1085 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1086 result.append(buffer); 1087 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1088 result.append(buffer); 1089 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1090 result.append(buffer); 1091 write(fd, result.string(), result.size()); 1092 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1093 1094 dumpBase(fd, args); 1095} 1096 1097// Thread virtuals 1098status_t AudioFlinger::PlaybackThread::readyToRun() 1099{ 1100 status_t status = initCheck(); 1101 if (status == NO_ERROR) { 1102 ALOGI("AudioFlinger's thread %p ready to run", this); 1103 } else { 1104 ALOGE("No working audio driver found."); 1105 } 1106 return status; 1107} 1108 1109void AudioFlinger::PlaybackThread::onFirstRef() 1110{ 1111 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1112} 1113 1114// ThreadBase virtuals 1115void AudioFlinger::PlaybackThread::preExit() 1116{ 1117 ALOGV(" preExit()"); 1118 // FIXME this is using hard-coded strings but in the future, this functionality will be 1119 // converted to use audio HAL extensions required to support tunneling 1120 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1121} 1122 1123// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1124sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1125 const sp<AudioFlinger::Client>& client, 1126 audio_stream_type_t streamType, 1127 uint32_t sampleRate, 1128 audio_format_t format, 1129 audio_channel_mask_t channelMask, 1130 size_t frameCount, 1131 const sp<IMemory>& sharedBuffer, 1132 int sessionId, 1133 IAudioFlinger::track_flags_t *flags, 1134 pid_t tid, 1135 status_t *status) 1136{ 1137 sp<Track> track; 1138 status_t lStatus; 1139 1140 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1141 1142 // client expresses a preference for FAST, but we get the final say 1143 if (*flags & IAudioFlinger::TRACK_FAST) { 1144 if ( 1145 // not timed 1146 (!isTimed) && 1147 // either of these use cases: 1148 ( 1149 // use case 1: shared buffer with any frame count 1150 ( 1151 (sharedBuffer != 0) 1152 ) || 1153 // use case 2: callback handler and frame count is default or at least as large as HAL 1154 ( 1155 (tid != -1) && 1156 ((frameCount == 0) || 1157 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1158 ) 1159 ) && 1160 // PCM data 1161 audio_is_linear_pcm(format) && 1162 // mono or stereo 1163 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1164 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1165#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1166 // hardware sample rate 1167 (sampleRate == mSampleRate) && 1168#endif 1169 // normal mixer has an associated fast mixer 1170 hasFastMixer() && 1171 // there are sufficient fast track slots available 1172 (mFastTrackAvailMask != 0) 1173 // FIXME test that MixerThread for this fast track has a capable output HAL 1174 // FIXME add a permission test also? 1175 ) { 1176 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1177 if (frameCount == 0) { 1178 frameCount = mFrameCount * kFastTrackMultiplier; 1179 } 1180 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1181 frameCount, mFrameCount); 1182 } else { 1183 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1184 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1185 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1186 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1187 audio_is_linear_pcm(format), 1188 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1189 *flags &= ~IAudioFlinger::TRACK_FAST; 1190 // For compatibility with AudioTrack calculation, buffer depth is forced 1191 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1192 // This is probably too conservative, but legacy application code may depend on it. 1193 // If you change this calculation, also review the start threshold which is related. 1194 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1195 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1196 if (minBufCount < 2) { 1197 minBufCount = 2; 1198 } 1199 size_t minFrameCount = mNormalFrameCount * minBufCount; 1200 if (frameCount < minFrameCount) { 1201 frameCount = minFrameCount; 1202 } 1203 } 1204 } 1205 1206 if (mType == DIRECT) { 1207 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1208 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1209 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1210 "for output %p with format %d", 1211 sampleRate, format, channelMask, mOutput, mFormat); 1212 lStatus = BAD_VALUE; 1213 goto Exit; 1214 } 1215 } 1216 } else if (mType == OFFLOAD) { 1217 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1218 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1219 "for output %p with format %d", 1220 sampleRate, format, channelMask, mOutput, mFormat); 1221 lStatus = BAD_VALUE; 1222 goto Exit; 1223 } 1224 } else { 1225 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1226 ALOGE("createTrack_l() Bad parameter: format %d \"" 1227 "for output %p with format %d", 1228 format, mOutput, mFormat); 1229 lStatus = BAD_VALUE; 1230 goto Exit; 1231 } 1232 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1233 if (sampleRate > mSampleRate*2) { 1234 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1235 lStatus = BAD_VALUE; 1236 goto Exit; 1237 } 1238 } 1239 1240 lStatus = initCheck(); 1241 if (lStatus != NO_ERROR) { 1242 ALOGE("Audio driver not initialized."); 1243 goto Exit; 1244 } 1245 1246 { // scope for mLock 1247 Mutex::Autolock _l(mLock); 1248 1249 // all tracks in same audio session must share the same routing strategy otherwise 1250 // conflicts will happen when tracks are moved from one output to another by audio policy 1251 // manager 1252 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1253 for (size_t i = 0; i < mTracks.size(); ++i) { 1254 sp<Track> t = mTracks[i]; 1255 if (t != 0 && !t->isOutputTrack()) { 1256 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1257 if (sessionId == t->sessionId() && strategy != actual) { 1258 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1259 strategy, actual); 1260 lStatus = BAD_VALUE; 1261 goto Exit; 1262 } 1263 } 1264 } 1265 1266 if (!isTimed) { 1267 track = new Track(this, client, streamType, sampleRate, format, 1268 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1269 } else { 1270 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1271 channelMask, frameCount, sharedBuffer, sessionId); 1272 } 1273 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1274 lStatus = NO_MEMORY; 1275 goto Exit; 1276 } 1277 1278 mTracks.add(track); 1279 1280 sp<EffectChain> chain = getEffectChain_l(sessionId); 1281 if (chain != 0) { 1282 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1283 track->setMainBuffer(chain->inBuffer()); 1284 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1285 chain->incTrackCnt(); 1286 } 1287 1288 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1289 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1290 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1291 // so ask activity manager to do this on our behalf 1292 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1293 } 1294 } 1295 1296 lStatus = NO_ERROR; 1297 1298Exit: 1299 if (status) { 1300 *status = lStatus; 1301 } 1302 return track; 1303} 1304 1305uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1306{ 1307 return latency; 1308} 1309 1310uint32_t AudioFlinger::PlaybackThread::latency() const 1311{ 1312 Mutex::Autolock _l(mLock); 1313 return latency_l(); 1314} 1315uint32_t AudioFlinger::PlaybackThread::latency_l() const 1316{ 1317 if (initCheck() == NO_ERROR) { 1318 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1319 } else { 1320 return 0; 1321 } 1322} 1323 1324void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1325{ 1326 Mutex::Autolock _l(mLock); 1327 // Don't apply master volume in SW if our HAL can do it for us. 1328 if (mOutput && mOutput->audioHwDev && 1329 mOutput->audioHwDev->canSetMasterVolume()) { 1330 mMasterVolume = 1.0; 1331 } else { 1332 mMasterVolume = value; 1333 } 1334} 1335 1336void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1337{ 1338 Mutex::Autolock _l(mLock); 1339 // Don't apply master mute in SW if our HAL can do it for us. 1340 if (mOutput && mOutput->audioHwDev && 1341 mOutput->audioHwDev->canSetMasterMute()) { 1342 mMasterMute = false; 1343 } else { 1344 mMasterMute = muted; 1345 } 1346} 1347 1348void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 mStreamTypes[stream].volume = value; 1352 broadcast_l(); 1353} 1354 1355void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1356{ 1357 Mutex::Autolock _l(mLock); 1358 mStreamTypes[stream].mute = muted; 1359 broadcast_l(); 1360} 1361 1362float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1363{ 1364 Mutex::Autolock _l(mLock); 1365 return mStreamTypes[stream].volume; 1366} 1367 1368// addTrack_l() must be called with ThreadBase::mLock held 1369status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1370{ 1371 status_t status = ALREADY_EXISTS; 1372 1373 // set retry count for buffer fill 1374 track->mRetryCount = kMaxTrackStartupRetries; 1375 if (mActiveTracks.indexOf(track) < 0) { 1376 // the track is newly added, make sure it fills up all its 1377 // buffers before playing. This is to ensure the client will 1378 // effectively get the latency it requested. 1379 if (!track->isOutputTrack()) { 1380 TrackBase::track_state state = track->mState; 1381 mLock.unlock(); 1382 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1383 mLock.lock(); 1384 // abort track was stopped/paused while we released the lock 1385 if (state != track->mState) { 1386 if (status == NO_ERROR) { 1387 mLock.unlock(); 1388 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1389 mLock.lock(); 1390 } 1391 return INVALID_OPERATION; 1392 } 1393 // abort if start is rejected by audio policy manager 1394 if (status != NO_ERROR) { 1395 return PERMISSION_DENIED; 1396 } 1397#ifdef ADD_BATTERY_DATA 1398 // to track the speaker usage 1399 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1400#endif 1401 } 1402 1403 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1404 track->mResetDone = false; 1405 track->mPresentationCompleteFrames = 0; 1406 mActiveTracks.add(track); 1407 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1408 if (chain != 0) { 1409 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1410 track->sessionId()); 1411 chain->incActiveTrackCnt(); 1412 } 1413 1414 status = NO_ERROR; 1415 } 1416 1417 ALOGV("signal playback thread"); 1418 broadcast_l(); 1419 1420 return status; 1421} 1422 1423bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1424{ 1425 track->terminate(); 1426 // active tracks are removed by threadLoop() 1427 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1428 track->mState = TrackBase::STOPPED; 1429 if (!trackActive) { 1430 removeTrack_l(track); 1431 } else if (track->isFastTrack() || track->isOffloaded()) { 1432 track->mState = TrackBase::STOPPING_1; 1433 } 1434 1435 return trackActive; 1436} 1437 1438void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1439{ 1440 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1441 mTracks.remove(track); 1442 deleteTrackName_l(track->name()); 1443 // redundant as track is about to be destroyed, for dumpsys only 1444 track->mName = -1; 1445 if (track->isFastTrack()) { 1446 int index = track->mFastIndex; 1447 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1448 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1449 mFastTrackAvailMask |= 1 << index; 1450 // redundant as track is about to be destroyed, for dumpsys only 1451 track->mFastIndex = -1; 1452 } 1453 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1454 if (chain != 0) { 1455 chain->decTrackCnt(); 1456 } 1457} 1458 1459void AudioFlinger::PlaybackThread::broadcast_l() 1460{ 1461 // Thread could be blocked waiting for async 1462 // so signal it to handle state changes immediately 1463 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1464 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1465 mSignalPending = true; 1466 mWaitWorkCV.broadcast(); 1467} 1468 1469String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1470{ 1471 Mutex::Autolock _l(mLock); 1472 if (initCheck() != NO_ERROR) { 1473 return String8(); 1474 } 1475 1476 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1477 const String8 out_s8(s); 1478 free(s); 1479 return out_s8; 1480} 1481 1482// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1483void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1484 AudioSystem::OutputDescriptor desc; 1485 void *param2 = NULL; 1486 1487 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1488 param); 1489 1490 switch (event) { 1491 case AudioSystem::OUTPUT_OPENED: 1492 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1493 desc.channelMask = mChannelMask; 1494 desc.samplingRate = mSampleRate; 1495 desc.format = mFormat; 1496 desc.frameCount = mNormalFrameCount; // FIXME see 1497 // AudioFlinger::frameCount(audio_io_handle_t) 1498 desc.latency = latency(); 1499 param2 = &desc; 1500 break; 1501 1502 case AudioSystem::STREAM_CONFIG_CHANGED: 1503 param2 = ¶m; 1504 case AudioSystem::OUTPUT_CLOSED: 1505 default: 1506 break; 1507 } 1508 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1509} 1510 1511void AudioFlinger::PlaybackThread::writeCallback() 1512{ 1513 ALOG_ASSERT(mCallbackThread != 0); 1514 mCallbackThread->resetWriteBlocked(); 1515} 1516 1517void AudioFlinger::PlaybackThread::drainCallback() 1518{ 1519 ALOG_ASSERT(mCallbackThread != 0); 1520 mCallbackThread->resetDraining(); 1521} 1522 1523void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1524{ 1525 Mutex::Autolock _l(mLock); 1526 // reject out of sequence requests 1527 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1528 mWriteAckSequence &= ~1; 1529 mWaitWorkCV.signal(); 1530 } 1531} 1532 1533void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1534{ 1535 Mutex::Autolock _l(mLock); 1536 // reject out of sequence requests 1537 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1538 mDrainSequence &= ~1; 1539 mWaitWorkCV.signal(); 1540 } 1541} 1542 1543// static 1544int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1545 void *param, 1546 void *cookie) 1547{ 1548 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1549 ALOGV("asyncCallback() event %d", event); 1550 switch (event) { 1551 case STREAM_CBK_EVENT_WRITE_READY: 1552 me->writeCallback(); 1553 break; 1554 case STREAM_CBK_EVENT_DRAIN_READY: 1555 me->drainCallback(); 1556 break; 1557 default: 1558 ALOGW("asyncCallback() unknown event %d", event); 1559 break; 1560 } 1561 return 0; 1562} 1563 1564void AudioFlinger::PlaybackThread::readOutputParameters() 1565{ 1566 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1567 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1568 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1569 if (!audio_is_output_channel(mChannelMask)) { 1570 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1571 } 1572 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1573 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1574 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1575 } 1576 mChannelCount = popcount(mChannelMask); 1577 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1578 if (!audio_is_valid_format(mFormat)) { 1579 LOG_FATAL("HAL format %d not valid for output", mFormat); 1580 } 1581 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1582 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1583 mFormat); 1584 } 1585 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1586 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1587 if (mFrameCount & 15) { 1588 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1589 mFrameCount); 1590 } 1591 1592 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1593 (mOutput->stream->set_callback != NULL)) { 1594 if (mOutput->stream->set_callback(mOutput->stream, 1595 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1596 mUseAsyncWrite = true; 1597 } 1598 } 1599 1600 // Calculate size of normal mix buffer relative to the HAL output buffer size 1601 double multiplier = 1.0; 1602 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1603 kUseFastMixer == FastMixer_Dynamic)) { 1604 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1605 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1606 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1607 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1608 maxNormalFrameCount = maxNormalFrameCount & ~15; 1609 if (maxNormalFrameCount < minNormalFrameCount) { 1610 maxNormalFrameCount = minNormalFrameCount; 1611 } 1612 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1613 if (multiplier <= 1.0) { 1614 multiplier = 1.0; 1615 } else if (multiplier <= 2.0) { 1616 if (2 * mFrameCount <= maxNormalFrameCount) { 1617 multiplier = 2.0; 1618 } else { 1619 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1620 } 1621 } else { 1622 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1623 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1624 // track, but we sometimes have to do this to satisfy the maximum frame count 1625 // constraint) 1626 // FIXME this rounding up should not be done if no HAL SRC 1627 uint32_t truncMult = (uint32_t) multiplier; 1628 if ((truncMult & 1)) { 1629 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1630 ++truncMult; 1631 } 1632 } 1633 multiplier = (double) truncMult; 1634 } 1635 } 1636 mNormalFrameCount = multiplier * mFrameCount; 1637 // round up to nearest 16 frames to satisfy AudioMixer 1638 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1639 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1640 mNormalFrameCount); 1641 1642 delete[] mAllocMixBuffer; 1643 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1644 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1645 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1646 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1647 1648 // force reconfiguration of effect chains and engines to take new buffer size and audio 1649 // parameters into account 1650 // Note that mLock is not held when readOutputParameters() is called from the constructor 1651 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1652 // matter. 1653 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1654 Vector< sp<EffectChain> > effectChains = mEffectChains; 1655 for (size_t i = 0; i < effectChains.size(); i ++) { 1656 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1657 } 1658} 1659 1660 1661status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1662{ 1663 if (halFrames == NULL || dspFrames == NULL) { 1664 return BAD_VALUE; 1665 } 1666 Mutex::Autolock _l(mLock); 1667 if (initCheck() != NO_ERROR) { 1668 return INVALID_OPERATION; 1669 } 1670 size_t framesWritten = mBytesWritten / mFrameSize; 1671 *halFrames = framesWritten; 1672 1673 if (isSuspended()) { 1674 // return an estimation of rendered frames when the output is suspended 1675 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1676 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1677 return NO_ERROR; 1678 } else { 1679 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1680 } 1681} 1682 1683uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1684{ 1685 Mutex::Autolock _l(mLock); 1686 uint32_t result = 0; 1687 if (getEffectChain_l(sessionId) != 0) { 1688 result = EFFECT_SESSION; 1689 } 1690 1691 for (size_t i = 0; i < mTracks.size(); ++i) { 1692 sp<Track> track = mTracks[i]; 1693 if (sessionId == track->sessionId() && !track->isInvalid()) { 1694 result |= TRACK_SESSION; 1695 break; 1696 } 1697 } 1698 1699 return result; 1700} 1701 1702uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1703{ 1704 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1705 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1706 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1707 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1708 } 1709 for (size_t i = 0; i < mTracks.size(); i++) { 1710 sp<Track> track = mTracks[i]; 1711 if (sessionId == track->sessionId() && !track->isInvalid()) { 1712 return AudioSystem::getStrategyForStream(track->streamType()); 1713 } 1714 } 1715 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1716} 1717 1718 1719AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1720{ 1721 Mutex::Autolock _l(mLock); 1722 return mOutput; 1723} 1724 1725AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1726{ 1727 Mutex::Autolock _l(mLock); 1728 AudioStreamOut *output = mOutput; 1729 mOutput = NULL; 1730 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1731 // must push a NULL and wait for ack 1732 mOutputSink.clear(); 1733 mPipeSink.clear(); 1734 mNormalSink.clear(); 1735 return output; 1736} 1737 1738// this method must always be called either with ThreadBase mLock held or inside the thread loop 1739audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1740{ 1741 if (mOutput == NULL) { 1742 return NULL; 1743 } 1744 return &mOutput->stream->common; 1745} 1746 1747uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1748{ 1749 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1750} 1751 1752status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1753{ 1754 if (!isValidSyncEvent(event)) { 1755 return BAD_VALUE; 1756 } 1757 1758 Mutex::Autolock _l(mLock); 1759 1760 for (size_t i = 0; i < mTracks.size(); ++i) { 1761 sp<Track> track = mTracks[i]; 1762 if (event->triggerSession() == track->sessionId()) { 1763 (void) track->setSyncEvent(event); 1764 return NO_ERROR; 1765 } 1766 } 1767 1768 return NAME_NOT_FOUND; 1769} 1770 1771bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1772{ 1773 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1774} 1775 1776void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1777 const Vector< sp<Track> >& tracksToRemove) 1778{ 1779 size_t count = tracksToRemove.size(); 1780 if (count) { 1781 for (size_t i = 0 ; i < count ; i++) { 1782 const sp<Track>& track = tracksToRemove.itemAt(i); 1783 if (!track->isOutputTrack()) { 1784 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1785#ifdef ADD_BATTERY_DATA 1786 // to track the speaker usage 1787 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1788#endif 1789 if (track->isTerminated()) { 1790 AudioSystem::releaseOutput(mId); 1791 } 1792 } 1793 } 1794 } 1795} 1796 1797void AudioFlinger::PlaybackThread::checkSilentMode_l() 1798{ 1799 if (!mMasterMute) { 1800 char value[PROPERTY_VALUE_MAX]; 1801 if (property_get("ro.audio.silent", value, "0") > 0) { 1802 char *endptr; 1803 unsigned long ul = strtoul(value, &endptr, 0); 1804 if (*endptr == '\0' && ul != 0) { 1805 ALOGD("Silence is golden"); 1806 // The setprop command will not allow a property to be changed after 1807 // the first time it is set, so we don't have to worry about un-muting. 1808 setMasterMute_l(true); 1809 } 1810 } 1811 } 1812} 1813 1814// shared by MIXER and DIRECT, overridden by DUPLICATING 1815ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1816{ 1817 // FIXME rewrite to reduce number of system calls 1818 mLastWriteTime = systemTime(); 1819 mInWrite = true; 1820 ssize_t bytesWritten; 1821 1822 // If an NBAIO sink is present, use it to write the normal mixer's submix 1823 if (mNormalSink != 0) { 1824#define mBitShift 2 // FIXME 1825 size_t count = mBytesRemaining >> mBitShift; 1826 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1827 ATRACE_BEGIN("write"); 1828 // update the setpoint when AudioFlinger::mScreenState changes 1829 uint32_t screenState = AudioFlinger::mScreenState; 1830 if (screenState != mScreenState) { 1831 mScreenState = screenState; 1832 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1833 if (pipe != NULL) { 1834 pipe->setAvgFrames((mScreenState & 1) ? 1835 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1836 } 1837 } 1838 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1839 ATRACE_END(); 1840 if (framesWritten > 0) { 1841 bytesWritten = framesWritten << mBitShift; 1842 } else { 1843 bytesWritten = framesWritten; 1844 } 1845 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1846 if (status == NO_ERROR) { 1847 size_t totalFramesWritten = mNormalSink->framesWritten(); 1848 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1849 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1850 mLatchDValid = true; 1851 } 1852 } 1853 // otherwise use the HAL / AudioStreamOut directly 1854 } else { 1855 // Direct output and offload threads 1856 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1857 if (mUseAsyncWrite) { 1858 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1859 mWriteAckSequence += 2; 1860 mWriteAckSequence |= 1; 1861 ALOG_ASSERT(mCallbackThread != 0); 1862 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1863 } 1864 // FIXME We should have an implementation of timestamps for direct output threads. 1865 // They are used e.g for multichannel PCM playback over HDMI. 1866 bytesWritten = mOutput->stream->write(mOutput->stream, 1867 mMixBuffer + offset, mBytesRemaining); 1868 if (mUseAsyncWrite && 1869 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1870 // do not wait for async callback in case of error of full write 1871 mWriteAckSequence &= ~1; 1872 ALOG_ASSERT(mCallbackThread != 0); 1873 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1874 } 1875 } 1876 1877 mNumWrites++; 1878 mInWrite = false; 1879 1880 return bytesWritten; 1881} 1882 1883void AudioFlinger::PlaybackThread::threadLoop_drain() 1884{ 1885 if (mOutput->stream->drain) { 1886 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1887 if (mUseAsyncWrite) { 1888 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1889 mDrainSequence |= 1; 1890 ALOG_ASSERT(mCallbackThread != 0); 1891 mCallbackThread->setDraining(mDrainSequence); 1892 } 1893 mOutput->stream->drain(mOutput->stream, 1894 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1895 : AUDIO_DRAIN_ALL); 1896 } 1897} 1898 1899void AudioFlinger::PlaybackThread::threadLoop_exit() 1900{ 1901 // Default implementation has nothing to do 1902} 1903 1904/* 1905The derived values that are cached: 1906 - mixBufferSize from frame count * frame size 1907 - activeSleepTime from activeSleepTimeUs() 1908 - idleSleepTime from idleSleepTimeUs() 1909 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1910 - maxPeriod from frame count and sample rate (MIXER only) 1911 1912The parameters that affect these derived values are: 1913 - frame count 1914 - frame size 1915 - sample rate 1916 - device type: A2DP or not 1917 - device latency 1918 - format: PCM or not 1919 - active sleep time 1920 - idle sleep time 1921*/ 1922 1923void AudioFlinger::PlaybackThread::cacheParameters_l() 1924{ 1925 mixBufferSize = mNormalFrameCount * mFrameSize; 1926 activeSleepTime = activeSleepTimeUs(); 1927 idleSleepTime = idleSleepTimeUs(); 1928} 1929 1930void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1931{ 1932 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1933 this, streamType, mTracks.size()); 1934 Mutex::Autolock _l(mLock); 1935 1936 size_t size = mTracks.size(); 1937 for (size_t i = 0; i < size; i++) { 1938 sp<Track> t = mTracks[i]; 1939 if (t->streamType() == streamType) { 1940 t->invalidate(); 1941 } 1942 } 1943} 1944 1945status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1946{ 1947 int session = chain->sessionId(); 1948 int16_t *buffer = mMixBuffer; 1949 bool ownsBuffer = false; 1950 1951 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1952 if (session > 0) { 1953 // Only one effect chain can be present in direct output thread and it uses 1954 // the mix buffer as input 1955 if (mType != DIRECT) { 1956 size_t numSamples = mNormalFrameCount * mChannelCount; 1957 buffer = new int16_t[numSamples]; 1958 memset(buffer, 0, numSamples * sizeof(int16_t)); 1959 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1960 ownsBuffer = true; 1961 } 1962 1963 // Attach all tracks with same session ID to this chain. 1964 for (size_t i = 0; i < mTracks.size(); ++i) { 1965 sp<Track> track = mTracks[i]; 1966 if (session == track->sessionId()) { 1967 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1968 buffer); 1969 track->setMainBuffer(buffer); 1970 chain->incTrackCnt(); 1971 } 1972 } 1973 1974 // indicate all active tracks in the chain 1975 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1976 sp<Track> track = mActiveTracks[i].promote(); 1977 if (track == 0) { 1978 continue; 1979 } 1980 if (session == track->sessionId()) { 1981 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1982 chain->incActiveTrackCnt(); 1983 } 1984 } 1985 } 1986 1987 chain->setInBuffer(buffer, ownsBuffer); 1988 chain->setOutBuffer(mMixBuffer); 1989 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1990 // chains list in order to be processed last as it contains output stage effects 1991 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1992 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1993 // after track specific effects and before output stage 1994 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1995 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1996 // Effect chain for other sessions are inserted at beginning of effect 1997 // chains list to be processed before output mix effects. Relative order between other 1998 // sessions is not important 1999 size_t size = mEffectChains.size(); 2000 size_t i = 0; 2001 for (i = 0; i < size; i++) { 2002 if (mEffectChains[i]->sessionId() < session) { 2003 break; 2004 } 2005 } 2006 mEffectChains.insertAt(chain, i); 2007 checkSuspendOnAddEffectChain_l(chain); 2008 2009 return NO_ERROR; 2010} 2011 2012size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2013{ 2014 int session = chain->sessionId(); 2015 2016 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2017 2018 for (size_t i = 0; i < mEffectChains.size(); i++) { 2019 if (chain == mEffectChains[i]) { 2020 mEffectChains.removeAt(i); 2021 // detach all active tracks from the chain 2022 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2023 sp<Track> track = mActiveTracks[i].promote(); 2024 if (track == 0) { 2025 continue; 2026 } 2027 if (session == track->sessionId()) { 2028 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2029 chain.get(), session); 2030 chain->decActiveTrackCnt(); 2031 } 2032 } 2033 2034 // detach all tracks with same session ID from this chain 2035 for (size_t i = 0; i < mTracks.size(); ++i) { 2036 sp<Track> track = mTracks[i]; 2037 if (session == track->sessionId()) { 2038 track->setMainBuffer(mMixBuffer); 2039 chain->decTrackCnt(); 2040 } 2041 } 2042 break; 2043 } 2044 } 2045 return mEffectChains.size(); 2046} 2047 2048status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2049 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2050{ 2051 Mutex::Autolock _l(mLock); 2052 return attachAuxEffect_l(track, EffectId); 2053} 2054 2055status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2056 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2057{ 2058 status_t status = NO_ERROR; 2059 2060 if (EffectId == 0) { 2061 track->setAuxBuffer(0, NULL); 2062 } else { 2063 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2064 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2065 if (effect != 0) { 2066 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2067 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2068 } else { 2069 status = INVALID_OPERATION; 2070 } 2071 } else { 2072 status = BAD_VALUE; 2073 } 2074 } 2075 return status; 2076} 2077 2078void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2079{ 2080 for (size_t i = 0; i < mTracks.size(); ++i) { 2081 sp<Track> track = mTracks[i]; 2082 if (track->auxEffectId() == effectId) { 2083 attachAuxEffect_l(track, 0); 2084 } 2085 } 2086} 2087 2088bool AudioFlinger::PlaybackThread::threadLoop() 2089{ 2090 Vector< sp<Track> > tracksToRemove; 2091 2092 standbyTime = systemTime(); 2093 2094 // MIXER 2095 nsecs_t lastWarning = 0; 2096 2097 // DUPLICATING 2098 // FIXME could this be made local to while loop? 2099 writeFrames = 0; 2100 2101 cacheParameters_l(); 2102 sleepTime = idleSleepTime; 2103 2104 if (mType == MIXER) { 2105 sleepTimeShift = 0; 2106 } 2107 2108 CpuStats cpuStats; 2109 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2110 2111 acquireWakeLock(); 2112 2113 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2114 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2115 // and then that string will be logged at the next convenient opportunity. 2116 const char *logString = NULL; 2117 2118 while (!exitPending()) 2119 { 2120 cpuStats.sample(myName); 2121 2122 Vector< sp<EffectChain> > effectChains; 2123 2124 processConfigEvents(); 2125 2126 { // scope for mLock 2127 2128 Mutex::Autolock _l(mLock); 2129 2130 if (logString != NULL) { 2131 mNBLogWriter->logTimestamp(); 2132 mNBLogWriter->log(logString); 2133 logString = NULL; 2134 } 2135 2136 if (mLatchDValid) { 2137 mLatchQ = mLatchD; 2138 mLatchDValid = false; 2139 mLatchQValid = true; 2140 } 2141 2142 if (checkForNewParameters_l()) { 2143 cacheParameters_l(); 2144 } 2145 2146 saveOutputTracks(); 2147 if (mSignalPending) { 2148 // A signal was raised while we were unlocked 2149 mSignalPending = false; 2150 } else if (waitingAsyncCallback_l()) { 2151 if (exitPending()) { 2152 break; 2153 } 2154 releaseWakeLock_l(); 2155 ALOGV("wait async completion"); 2156 mWaitWorkCV.wait(mLock); 2157 ALOGV("async completion/wake"); 2158 acquireWakeLock_l(); 2159 standbyTime = systemTime() + standbyDelay; 2160 sleepTime = 0; 2161 2162 continue; 2163 } 2164 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2165 isSuspended()) { 2166 // put audio hardware into standby after short delay 2167 if (shouldStandby_l()) { 2168 2169 threadLoop_standby(); 2170 2171 mStandby = true; 2172 } 2173 2174 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2175 // we're about to wait, flush the binder command buffer 2176 IPCThreadState::self()->flushCommands(); 2177 2178 clearOutputTracks(); 2179 2180 if (exitPending()) { 2181 break; 2182 } 2183 2184 releaseWakeLock_l(); 2185 // wait until we have something to do... 2186 ALOGV("%s going to sleep", myName.string()); 2187 mWaitWorkCV.wait(mLock); 2188 ALOGV("%s waking up", myName.string()); 2189 acquireWakeLock_l(); 2190 2191 mMixerStatus = MIXER_IDLE; 2192 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2193 mBytesWritten = 0; 2194 mBytesRemaining = 0; 2195 checkSilentMode_l(); 2196 2197 standbyTime = systemTime() + standbyDelay; 2198 sleepTime = idleSleepTime; 2199 if (mType == MIXER) { 2200 sleepTimeShift = 0; 2201 } 2202 2203 continue; 2204 } 2205 } 2206 // mMixerStatusIgnoringFastTracks is also updated internally 2207 mMixerStatus = prepareTracks_l(&tracksToRemove); 2208 2209 // prevent any changes in effect chain list and in each effect chain 2210 // during mixing and effect process as the audio buffers could be deleted 2211 // or modified if an effect is created or deleted 2212 lockEffectChains_l(effectChains); 2213 } 2214 2215 if (mBytesRemaining == 0) { 2216 mCurrentWriteLength = 0; 2217 if (mMixerStatus == MIXER_TRACKS_READY) { 2218 // threadLoop_mix() sets mCurrentWriteLength 2219 threadLoop_mix(); 2220 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2221 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2222 // threadLoop_sleepTime sets sleepTime to 0 if data 2223 // must be written to HAL 2224 threadLoop_sleepTime(); 2225 if (sleepTime == 0) { 2226 mCurrentWriteLength = mixBufferSize; 2227 } 2228 } 2229 mBytesRemaining = mCurrentWriteLength; 2230 if (isSuspended()) { 2231 sleepTime = suspendSleepTimeUs(); 2232 // simulate write to HAL when suspended 2233 mBytesWritten += mixBufferSize; 2234 mBytesRemaining = 0; 2235 } 2236 2237 // only process effects if we're going to write 2238 if (sleepTime == 0) { 2239 for (size_t i = 0; i < effectChains.size(); i ++) { 2240 effectChains[i]->process_l(); 2241 } 2242 } 2243 } 2244 2245 // enable changes in effect chain 2246 unlockEffectChains(effectChains); 2247 2248 if (!waitingAsyncCallback()) { 2249 // sleepTime == 0 means we must write to audio hardware 2250 if (sleepTime == 0) { 2251 if (mBytesRemaining) { 2252 ssize_t ret = threadLoop_write(); 2253 if (ret < 0) { 2254 mBytesRemaining = 0; 2255 } else { 2256 mBytesWritten += ret; 2257 mBytesRemaining -= ret; 2258 } 2259 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2260 (mMixerStatus == MIXER_DRAIN_ALL)) { 2261 threadLoop_drain(); 2262 } 2263if (mType == MIXER) { 2264 // write blocked detection 2265 nsecs_t now = systemTime(); 2266 nsecs_t delta = now - mLastWriteTime; 2267 if (!mStandby && delta > maxPeriod) { 2268 mNumDelayedWrites++; 2269 if ((now - lastWarning) > kWarningThrottleNs) { 2270 ATRACE_NAME("underrun"); 2271 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2272 ns2ms(delta), mNumDelayedWrites, this); 2273 lastWarning = now; 2274 } 2275 } 2276} 2277 2278 mStandby = false; 2279 } else { 2280 usleep(sleepTime); 2281 } 2282 } 2283 2284 // Finally let go of removed track(s), without the lock held 2285 // since we can't guarantee the destructors won't acquire that 2286 // same lock. This will also mutate and push a new fast mixer state. 2287 threadLoop_removeTracks(tracksToRemove); 2288 tracksToRemove.clear(); 2289 2290 // FIXME I don't understand the need for this here; 2291 // it was in the original code but maybe the 2292 // assignment in saveOutputTracks() makes this unnecessary? 2293 clearOutputTracks(); 2294 2295 // Effect chains will be actually deleted here if they were removed from 2296 // mEffectChains list during mixing or effects processing 2297 effectChains.clear(); 2298 2299 // FIXME Note that the above .clear() is no longer necessary since effectChains 2300 // is now local to this block, but will keep it for now (at least until merge done). 2301 } 2302 2303 threadLoop_exit(); 2304 2305 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2306 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2307 // put output stream into standby mode 2308 if (!mStandby) { 2309 mOutput->stream->common.standby(&mOutput->stream->common); 2310 } 2311 } 2312 2313 releaseWakeLock(); 2314 2315 ALOGV("Thread %p type %d exiting", this, mType); 2316 return false; 2317} 2318 2319// removeTracks_l() must be called with ThreadBase::mLock held 2320void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2321{ 2322 size_t count = tracksToRemove.size(); 2323 if (count) { 2324 for (size_t i=0 ; i<count ; i++) { 2325 const sp<Track>& track = tracksToRemove.itemAt(i); 2326 mActiveTracks.remove(track); 2327 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2328 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2329 if (chain != 0) { 2330 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2331 track->sessionId()); 2332 chain->decActiveTrackCnt(); 2333 } 2334 if (track->isTerminated()) { 2335 removeTrack_l(track); 2336 } 2337 } 2338 } 2339 2340} 2341 2342status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2343{ 2344 if (mNormalSink != 0) { 2345 return mNormalSink->getTimestamp(timestamp); 2346 } 2347 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2348 uint64_t position64; 2349 int ret = mOutput->stream->get_presentation_position( 2350 mOutput->stream, &position64, ×tamp.mTime); 2351 if (ret == 0) { 2352 timestamp.mPosition = (uint32_t)position64; 2353 return NO_ERROR; 2354 } 2355 } 2356 return INVALID_OPERATION; 2357} 2358// ---------------------------------------------------------------------------- 2359 2360AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2361 audio_io_handle_t id, audio_devices_t device, type_t type) 2362 : PlaybackThread(audioFlinger, output, id, device, type), 2363 // mAudioMixer below 2364 // mFastMixer below 2365 mFastMixerFutex(0) 2366 // mOutputSink below 2367 // mPipeSink below 2368 // mNormalSink below 2369{ 2370 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2371 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2372 "mFrameCount=%d, mNormalFrameCount=%d", 2373 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2374 mNormalFrameCount); 2375 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2376 2377 // FIXME - Current mixer implementation only supports stereo output 2378 if (mChannelCount != FCC_2) { 2379 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2380 } 2381 2382 // create an NBAIO sink for the HAL output stream, and negotiate 2383 mOutputSink = new AudioStreamOutSink(output->stream); 2384 size_t numCounterOffers = 0; 2385 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2386 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2387 ALOG_ASSERT(index == 0); 2388 2389 // initialize fast mixer depending on configuration 2390 bool initFastMixer; 2391 switch (kUseFastMixer) { 2392 case FastMixer_Never: 2393 initFastMixer = false; 2394 break; 2395 case FastMixer_Always: 2396 initFastMixer = true; 2397 break; 2398 case FastMixer_Static: 2399 case FastMixer_Dynamic: 2400 initFastMixer = mFrameCount < mNormalFrameCount; 2401 break; 2402 } 2403 if (initFastMixer) { 2404 2405 // create a MonoPipe to connect our submix to FastMixer 2406 NBAIO_Format format = mOutputSink->format(); 2407 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2408 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2409 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2410 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2411 const NBAIO_Format offers[1] = {format}; 2412 size_t numCounterOffers = 0; 2413 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2414 ALOG_ASSERT(index == 0); 2415 monoPipe->setAvgFrames((mScreenState & 1) ? 2416 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2417 mPipeSink = monoPipe; 2418 2419#ifdef TEE_SINK 2420 if (mTeeSinkOutputEnabled) { 2421 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2422 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2423 numCounterOffers = 0; 2424 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2425 ALOG_ASSERT(index == 0); 2426 mTeeSink = teeSink; 2427 PipeReader *teeSource = new PipeReader(*teeSink); 2428 numCounterOffers = 0; 2429 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2430 ALOG_ASSERT(index == 0); 2431 mTeeSource = teeSource; 2432 } 2433#endif 2434 2435 // create fast mixer and configure it initially with just one fast track for our submix 2436 mFastMixer = new FastMixer(); 2437 FastMixerStateQueue *sq = mFastMixer->sq(); 2438#ifdef STATE_QUEUE_DUMP 2439 sq->setObserverDump(&mStateQueueObserverDump); 2440 sq->setMutatorDump(&mStateQueueMutatorDump); 2441#endif 2442 FastMixerState *state = sq->begin(); 2443 FastTrack *fastTrack = &state->mFastTracks[0]; 2444 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2445 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2446 fastTrack->mVolumeProvider = NULL; 2447 fastTrack->mGeneration++; 2448 state->mFastTracksGen++; 2449 state->mTrackMask = 1; 2450 // fast mixer will use the HAL output sink 2451 state->mOutputSink = mOutputSink.get(); 2452 state->mOutputSinkGen++; 2453 state->mFrameCount = mFrameCount; 2454 state->mCommand = FastMixerState::COLD_IDLE; 2455 // already done in constructor initialization list 2456 //mFastMixerFutex = 0; 2457 state->mColdFutexAddr = &mFastMixerFutex; 2458 state->mColdGen++; 2459 state->mDumpState = &mFastMixerDumpState; 2460#ifdef TEE_SINK 2461 state->mTeeSink = mTeeSink.get(); 2462#endif 2463 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2464 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2465 sq->end(); 2466 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2467 2468 // start the fast mixer 2469 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2470 pid_t tid = mFastMixer->getTid(); 2471 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2472 if (err != 0) { 2473 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2474 kPriorityFastMixer, getpid_cached, tid, err); 2475 } 2476 2477#ifdef AUDIO_WATCHDOG 2478 // create and start the watchdog 2479 mAudioWatchdog = new AudioWatchdog(); 2480 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2481 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2482 tid = mAudioWatchdog->getTid(); 2483 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2484 if (err != 0) { 2485 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2486 kPriorityFastMixer, getpid_cached, tid, err); 2487 } 2488#endif 2489 2490 } else { 2491 mFastMixer = NULL; 2492 } 2493 2494 switch (kUseFastMixer) { 2495 case FastMixer_Never: 2496 case FastMixer_Dynamic: 2497 mNormalSink = mOutputSink; 2498 break; 2499 case FastMixer_Always: 2500 mNormalSink = mPipeSink; 2501 break; 2502 case FastMixer_Static: 2503 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2504 break; 2505 } 2506} 2507 2508AudioFlinger::MixerThread::~MixerThread() 2509{ 2510 if (mFastMixer != NULL) { 2511 FastMixerStateQueue *sq = mFastMixer->sq(); 2512 FastMixerState *state = sq->begin(); 2513 if (state->mCommand == FastMixerState::COLD_IDLE) { 2514 int32_t old = android_atomic_inc(&mFastMixerFutex); 2515 if (old == -1) { 2516 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2517 } 2518 } 2519 state->mCommand = FastMixerState::EXIT; 2520 sq->end(); 2521 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2522 mFastMixer->join(); 2523 // Though the fast mixer thread has exited, it's state queue is still valid. 2524 // We'll use that extract the final state which contains one remaining fast track 2525 // corresponding to our sub-mix. 2526 state = sq->begin(); 2527 ALOG_ASSERT(state->mTrackMask == 1); 2528 FastTrack *fastTrack = &state->mFastTracks[0]; 2529 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2530 delete fastTrack->mBufferProvider; 2531 sq->end(false /*didModify*/); 2532 delete mFastMixer; 2533#ifdef AUDIO_WATCHDOG 2534 if (mAudioWatchdog != 0) { 2535 mAudioWatchdog->requestExit(); 2536 mAudioWatchdog->requestExitAndWait(); 2537 mAudioWatchdog.clear(); 2538 } 2539#endif 2540 } 2541 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2542 delete mAudioMixer; 2543} 2544 2545 2546uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2547{ 2548 if (mFastMixer != NULL) { 2549 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2550 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2551 } 2552 return latency; 2553} 2554 2555 2556void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2557{ 2558 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2559} 2560 2561ssize_t AudioFlinger::MixerThread::threadLoop_write() 2562{ 2563 // FIXME we should only do one push per cycle; confirm this is true 2564 // Start the fast mixer if it's not already running 2565 if (mFastMixer != NULL) { 2566 FastMixerStateQueue *sq = mFastMixer->sq(); 2567 FastMixerState *state = sq->begin(); 2568 if (state->mCommand != FastMixerState::MIX_WRITE && 2569 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2570 if (state->mCommand == FastMixerState::COLD_IDLE) { 2571 int32_t old = android_atomic_inc(&mFastMixerFutex); 2572 if (old == -1) { 2573 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2574 } 2575#ifdef AUDIO_WATCHDOG 2576 if (mAudioWatchdog != 0) { 2577 mAudioWatchdog->resume(); 2578 } 2579#endif 2580 } 2581 state->mCommand = FastMixerState::MIX_WRITE; 2582 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2583 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2584 sq->end(); 2585 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2586 if (kUseFastMixer == FastMixer_Dynamic) { 2587 mNormalSink = mPipeSink; 2588 } 2589 } else { 2590 sq->end(false /*didModify*/); 2591 } 2592 } 2593 return PlaybackThread::threadLoop_write(); 2594} 2595 2596void AudioFlinger::MixerThread::threadLoop_standby() 2597{ 2598 // Idle the fast mixer if it's currently running 2599 if (mFastMixer != NULL) { 2600 FastMixerStateQueue *sq = mFastMixer->sq(); 2601 FastMixerState *state = sq->begin(); 2602 if (!(state->mCommand & FastMixerState::IDLE)) { 2603 state->mCommand = FastMixerState::COLD_IDLE; 2604 state->mColdFutexAddr = &mFastMixerFutex; 2605 state->mColdGen++; 2606 mFastMixerFutex = 0; 2607 sq->end(); 2608 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2609 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2610 if (kUseFastMixer == FastMixer_Dynamic) { 2611 mNormalSink = mOutputSink; 2612 } 2613#ifdef AUDIO_WATCHDOG 2614 if (mAudioWatchdog != 0) { 2615 mAudioWatchdog->pause(); 2616 } 2617#endif 2618 } else { 2619 sq->end(false /*didModify*/); 2620 } 2621 } 2622 PlaybackThread::threadLoop_standby(); 2623} 2624 2625// Empty implementation for standard mixer 2626// Overridden for offloaded playback 2627void AudioFlinger::PlaybackThread::flushOutput_l() 2628{ 2629} 2630 2631bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2632{ 2633 return false; 2634} 2635 2636bool AudioFlinger::PlaybackThread::shouldStandby_l() 2637{ 2638 return !mStandby; 2639} 2640 2641bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2642{ 2643 Mutex::Autolock _l(mLock); 2644 return waitingAsyncCallback_l(); 2645} 2646 2647// shared by MIXER and DIRECT, overridden by DUPLICATING 2648void AudioFlinger::PlaybackThread::threadLoop_standby() 2649{ 2650 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2651 mOutput->stream->common.standby(&mOutput->stream->common); 2652 if (mUseAsyncWrite != 0) { 2653 // discard any pending drain or write ack by incrementing sequence 2654 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2655 mDrainSequence = (mDrainSequence + 2) & ~1; 2656 ALOG_ASSERT(mCallbackThread != 0); 2657 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2658 mCallbackThread->setDraining(mDrainSequence); 2659 } 2660} 2661 2662void AudioFlinger::MixerThread::threadLoop_mix() 2663{ 2664 // obtain the presentation timestamp of the next output buffer 2665 int64_t pts; 2666 status_t status = INVALID_OPERATION; 2667 2668 if (mNormalSink != 0) { 2669 status = mNormalSink->getNextWriteTimestamp(&pts); 2670 } else { 2671 status = mOutputSink->getNextWriteTimestamp(&pts); 2672 } 2673 2674 if (status != NO_ERROR) { 2675 pts = AudioBufferProvider::kInvalidPTS; 2676 } 2677 2678 // mix buffers... 2679 mAudioMixer->process(pts); 2680 mCurrentWriteLength = mixBufferSize; 2681 // increase sleep time progressively when application underrun condition clears. 2682 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2683 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2684 // such that we would underrun the audio HAL. 2685 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2686 sleepTimeShift--; 2687 } 2688 sleepTime = 0; 2689 standbyTime = systemTime() + standbyDelay; 2690 //TODO: delay standby when effects have a tail 2691} 2692 2693void AudioFlinger::MixerThread::threadLoop_sleepTime() 2694{ 2695 // If no tracks are ready, sleep once for the duration of an output 2696 // buffer size, then write 0s to the output 2697 if (sleepTime == 0) { 2698 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2699 sleepTime = activeSleepTime >> sleepTimeShift; 2700 if (sleepTime < kMinThreadSleepTimeUs) { 2701 sleepTime = kMinThreadSleepTimeUs; 2702 } 2703 // reduce sleep time in case of consecutive application underruns to avoid 2704 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2705 // duration we would end up writing less data than needed by the audio HAL if 2706 // the condition persists. 2707 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2708 sleepTimeShift++; 2709 } 2710 } else { 2711 sleepTime = idleSleepTime; 2712 } 2713 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2714 memset (mMixBuffer, 0, mixBufferSize); 2715 sleepTime = 0; 2716 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2717 "anticipated start"); 2718 } 2719 // TODO add standby time extension fct of effect tail 2720} 2721 2722// prepareTracks_l() must be called with ThreadBase::mLock held 2723AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2724 Vector< sp<Track> > *tracksToRemove) 2725{ 2726 2727 mixer_state mixerStatus = MIXER_IDLE; 2728 // find out which tracks need to be processed 2729 size_t count = mActiveTracks.size(); 2730 size_t mixedTracks = 0; 2731 size_t tracksWithEffect = 0; 2732 // counts only _active_ fast tracks 2733 size_t fastTracks = 0; 2734 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2735 2736 float masterVolume = mMasterVolume; 2737 bool masterMute = mMasterMute; 2738 2739 if (masterMute) { 2740 masterVolume = 0; 2741 } 2742 // Delegate master volume control to effect in output mix effect chain if needed 2743 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2744 if (chain != 0) { 2745 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2746 chain->setVolume_l(&v, &v); 2747 masterVolume = (float)((v + (1 << 23)) >> 24); 2748 chain.clear(); 2749 } 2750 2751 // prepare a new state to push 2752 FastMixerStateQueue *sq = NULL; 2753 FastMixerState *state = NULL; 2754 bool didModify = false; 2755 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2756 if (mFastMixer != NULL) { 2757 sq = mFastMixer->sq(); 2758 state = sq->begin(); 2759 } 2760 2761 for (size_t i=0 ; i<count ; i++) { 2762 const sp<Track> t = mActiveTracks[i].promote(); 2763 if (t == 0) { 2764 continue; 2765 } 2766 2767 // this const just means the local variable doesn't change 2768 Track* const track = t.get(); 2769 2770 // process fast tracks 2771 if (track->isFastTrack()) { 2772 2773 // It's theoretically possible (though unlikely) for a fast track to be created 2774 // and then removed within the same normal mix cycle. This is not a problem, as 2775 // the track never becomes active so it's fast mixer slot is never touched. 2776 // The converse, of removing an (active) track and then creating a new track 2777 // at the identical fast mixer slot within the same normal mix cycle, 2778 // is impossible because the slot isn't marked available until the end of each cycle. 2779 int j = track->mFastIndex; 2780 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2781 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2782 FastTrack *fastTrack = &state->mFastTracks[j]; 2783 2784 // Determine whether the track is currently in underrun condition, 2785 // and whether it had a recent underrun. 2786 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2787 FastTrackUnderruns underruns = ftDump->mUnderruns; 2788 uint32_t recentFull = (underruns.mBitFields.mFull - 2789 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2790 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2791 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2792 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2793 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2794 uint32_t recentUnderruns = recentPartial + recentEmpty; 2795 track->mObservedUnderruns = underruns; 2796 // don't count underruns that occur while stopping or pausing 2797 // or stopped which can occur when flush() is called while active 2798 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2799 recentUnderruns > 0) { 2800 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2801 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2802 } 2803 2804 // This is similar to the state machine for normal tracks, 2805 // with a few modifications for fast tracks. 2806 bool isActive = true; 2807 switch (track->mState) { 2808 case TrackBase::STOPPING_1: 2809 // track stays active in STOPPING_1 state until first underrun 2810 if (recentUnderruns > 0 || track->isTerminated()) { 2811 track->mState = TrackBase::STOPPING_2; 2812 } 2813 break; 2814 case TrackBase::PAUSING: 2815 // ramp down is not yet implemented 2816 track->setPaused(); 2817 break; 2818 case TrackBase::RESUMING: 2819 // ramp up is not yet implemented 2820 track->mState = TrackBase::ACTIVE; 2821 break; 2822 case TrackBase::ACTIVE: 2823 if (recentFull > 0 || recentPartial > 0) { 2824 // track has provided at least some frames recently: reset retry count 2825 track->mRetryCount = kMaxTrackRetries; 2826 } 2827 if (recentUnderruns == 0) { 2828 // no recent underruns: stay active 2829 break; 2830 } 2831 // there has recently been an underrun of some kind 2832 if (track->sharedBuffer() == 0) { 2833 // were any of the recent underruns "empty" (no frames available)? 2834 if (recentEmpty == 0) { 2835 // no, then ignore the partial underruns as they are allowed indefinitely 2836 break; 2837 } 2838 // there has recently been an "empty" underrun: decrement the retry counter 2839 if (--(track->mRetryCount) > 0) { 2840 break; 2841 } 2842 // indicate to client process that the track was disabled because of underrun; 2843 // it will then automatically call start() when data is available 2844 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2845 // remove from active list, but state remains ACTIVE [confusing but true] 2846 isActive = false; 2847 break; 2848 } 2849 // fall through 2850 case TrackBase::STOPPING_2: 2851 case TrackBase::PAUSED: 2852 case TrackBase::STOPPED: 2853 case TrackBase::FLUSHED: // flush() while active 2854 // Check for presentation complete if track is inactive 2855 // We have consumed all the buffers of this track. 2856 // This would be incomplete if we auto-paused on underrun 2857 { 2858 size_t audioHALFrames = 2859 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2860 size_t framesWritten = mBytesWritten / mFrameSize; 2861 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2862 // track stays in active list until presentation is complete 2863 break; 2864 } 2865 } 2866 if (track->isStopping_2()) { 2867 track->mState = TrackBase::STOPPED; 2868 } 2869 if (track->isStopped()) { 2870 // Can't reset directly, as fast mixer is still polling this track 2871 // track->reset(); 2872 // So instead mark this track as needing to be reset after push with ack 2873 resetMask |= 1 << i; 2874 } 2875 isActive = false; 2876 break; 2877 case TrackBase::IDLE: 2878 default: 2879 LOG_FATAL("unexpected track state %d", track->mState); 2880 } 2881 2882 if (isActive) { 2883 // was it previously inactive? 2884 if (!(state->mTrackMask & (1 << j))) { 2885 ExtendedAudioBufferProvider *eabp = track; 2886 VolumeProvider *vp = track; 2887 fastTrack->mBufferProvider = eabp; 2888 fastTrack->mVolumeProvider = vp; 2889 fastTrack->mSampleRate = track->mSampleRate; 2890 fastTrack->mChannelMask = track->mChannelMask; 2891 fastTrack->mGeneration++; 2892 state->mTrackMask |= 1 << j; 2893 didModify = true; 2894 // no acknowledgement required for newly active tracks 2895 } 2896 // cache the combined master volume and stream type volume for fast mixer; this 2897 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2898 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2899 ++fastTracks; 2900 } else { 2901 // was it previously active? 2902 if (state->mTrackMask & (1 << j)) { 2903 fastTrack->mBufferProvider = NULL; 2904 fastTrack->mGeneration++; 2905 state->mTrackMask &= ~(1 << j); 2906 didModify = true; 2907 // If any fast tracks were removed, we must wait for acknowledgement 2908 // because we're about to decrement the last sp<> on those tracks. 2909 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2910 } else { 2911 LOG_FATAL("fast track %d should have been active", j); 2912 } 2913 tracksToRemove->add(track); 2914 // Avoids a misleading display in dumpsys 2915 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2916 } 2917 continue; 2918 } 2919 2920 { // local variable scope to avoid goto warning 2921 2922 audio_track_cblk_t* cblk = track->cblk(); 2923 2924 // The first time a track is added we wait 2925 // for all its buffers to be filled before processing it 2926 int name = track->name(); 2927 // make sure that we have enough frames to mix one full buffer. 2928 // enforce this condition only once to enable draining the buffer in case the client 2929 // app does not call stop() and relies on underrun to stop: 2930 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2931 // during last round 2932 size_t desiredFrames; 2933 uint32_t sr = track->sampleRate(); 2934 if (sr == mSampleRate) { 2935 desiredFrames = mNormalFrameCount; 2936 } else { 2937 // +1 for rounding and +1 for additional sample needed for interpolation 2938 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2939 // add frames already consumed but not yet released by the resampler 2940 // because cblk->framesReady() will include these frames 2941 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2942 // the minimum track buffer size is normally twice the number of frames necessary 2943 // to fill one buffer and the resampler should not leave more than one buffer worth 2944 // of unreleased frames after each pass, but just in case... 2945 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2946 } 2947 uint32_t minFrames = 1; 2948 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2949 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2950 minFrames = desiredFrames; 2951 } 2952 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2953 size_t framesReady; 2954 if (track->sharedBuffer() == 0) { 2955 framesReady = track->framesReady(); 2956 } else if (track->isStopped()) { 2957 framesReady = 0; 2958 } else { 2959 framesReady = 1; 2960 } 2961 if ((framesReady >= minFrames) && track->isReady() && 2962 !track->isPaused() && !track->isTerminated()) 2963 { 2964 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2965 2966 mixedTracks++; 2967 2968 // track->mainBuffer() != mMixBuffer means there is an effect chain 2969 // connected to the track 2970 chain.clear(); 2971 if (track->mainBuffer() != mMixBuffer) { 2972 chain = getEffectChain_l(track->sessionId()); 2973 // Delegate volume control to effect in track effect chain if needed 2974 if (chain != 0) { 2975 tracksWithEffect++; 2976 } else { 2977 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2978 "session %d", 2979 name, track->sessionId()); 2980 } 2981 } 2982 2983 2984 int param = AudioMixer::VOLUME; 2985 if (track->mFillingUpStatus == Track::FS_FILLED) { 2986 // no ramp for the first volume setting 2987 track->mFillingUpStatus = Track::FS_ACTIVE; 2988 if (track->mState == TrackBase::RESUMING) { 2989 track->mState = TrackBase::ACTIVE; 2990 param = AudioMixer::RAMP_VOLUME; 2991 } 2992 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2993 // FIXME should not make a decision based on mServer 2994 } else if (cblk->mServer != 0) { 2995 // If the track is stopped before the first frame was mixed, 2996 // do not apply ramp 2997 param = AudioMixer::RAMP_VOLUME; 2998 } 2999 3000 // compute volume for this track 3001 uint32_t vl, vr, va; 3002 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3003 vl = vr = va = 0; 3004 if (track->isPausing()) { 3005 track->setPaused(); 3006 } 3007 } else { 3008 3009 // read original volumes with volume control 3010 float typeVolume = mStreamTypes[track->streamType()].volume; 3011 float v = masterVolume * typeVolume; 3012 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3013 uint32_t vlr = proxy->getVolumeLR(); 3014 vl = vlr & 0xFFFF; 3015 vr = vlr >> 16; 3016 // track volumes come from shared memory, so can't be trusted and must be clamped 3017 if (vl > MAX_GAIN_INT) { 3018 ALOGV("Track left volume out of range: %04X", vl); 3019 vl = MAX_GAIN_INT; 3020 } 3021 if (vr > MAX_GAIN_INT) { 3022 ALOGV("Track right volume out of range: %04X", vr); 3023 vr = MAX_GAIN_INT; 3024 } 3025 // now apply the master volume and stream type volume 3026 vl = (uint32_t)(v * vl) << 12; 3027 vr = (uint32_t)(v * vr) << 12; 3028 // assuming master volume and stream type volume each go up to 1.0, 3029 // vl and vr are now in 8.24 format 3030 3031 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3032 // send level comes from shared memory and so may be corrupt 3033 if (sendLevel > MAX_GAIN_INT) { 3034 ALOGV("Track send level out of range: %04X", sendLevel); 3035 sendLevel = MAX_GAIN_INT; 3036 } 3037 va = (uint32_t)(v * sendLevel); 3038 } 3039 3040 // Delegate volume control to effect in track effect chain if needed 3041 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3042 // Do not ramp volume if volume is controlled by effect 3043 param = AudioMixer::VOLUME; 3044 track->mHasVolumeController = true; 3045 } else { 3046 // force no volume ramp when volume controller was just disabled or removed 3047 // from effect chain to avoid volume spike 3048 if (track->mHasVolumeController) { 3049 param = AudioMixer::VOLUME; 3050 } 3051 track->mHasVolumeController = false; 3052 } 3053 3054 // Convert volumes from 8.24 to 4.12 format 3055 // This additional clamping is needed in case chain->setVolume_l() overshot 3056 vl = (vl + (1 << 11)) >> 12; 3057 if (vl > MAX_GAIN_INT) { 3058 vl = MAX_GAIN_INT; 3059 } 3060 vr = (vr + (1 << 11)) >> 12; 3061 if (vr > MAX_GAIN_INT) { 3062 vr = MAX_GAIN_INT; 3063 } 3064 3065 if (va > MAX_GAIN_INT) { 3066 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3067 } 3068 3069 // XXX: these things DON'T need to be done each time 3070 mAudioMixer->setBufferProvider(name, track); 3071 mAudioMixer->enable(name); 3072 3073 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3074 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3075 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3076 mAudioMixer->setParameter( 3077 name, 3078 AudioMixer::TRACK, 3079 AudioMixer::FORMAT, (void *)track->format()); 3080 mAudioMixer->setParameter( 3081 name, 3082 AudioMixer::TRACK, 3083 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3084 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3085 uint32_t maxSampleRate = mSampleRate * 2; 3086 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3087 if (reqSampleRate == 0) { 3088 reqSampleRate = mSampleRate; 3089 } else if (reqSampleRate > maxSampleRate) { 3090 reqSampleRate = maxSampleRate; 3091 } 3092 mAudioMixer->setParameter( 3093 name, 3094 AudioMixer::RESAMPLE, 3095 AudioMixer::SAMPLE_RATE, 3096 (void *)reqSampleRate); 3097 mAudioMixer->setParameter( 3098 name, 3099 AudioMixer::TRACK, 3100 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3101 mAudioMixer->setParameter( 3102 name, 3103 AudioMixer::TRACK, 3104 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3105 3106 // reset retry count 3107 track->mRetryCount = kMaxTrackRetries; 3108 3109 // If one track is ready, set the mixer ready if: 3110 // - the mixer was not ready during previous round OR 3111 // - no other track is not ready 3112 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3113 mixerStatus != MIXER_TRACKS_ENABLED) { 3114 mixerStatus = MIXER_TRACKS_READY; 3115 } 3116 } else { 3117 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3118 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3119 } 3120 // clear effect chain input buffer if an active track underruns to avoid sending 3121 // previous audio buffer again to effects 3122 chain = getEffectChain_l(track->sessionId()); 3123 if (chain != 0) { 3124 chain->clearInputBuffer(); 3125 } 3126 3127 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3128 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3129 track->isStopped() || track->isPaused()) { 3130 // We have consumed all the buffers of this track. 3131 // Remove it from the list of active tracks. 3132 // TODO: use actual buffer filling status instead of latency when available from 3133 // audio HAL 3134 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3135 size_t framesWritten = mBytesWritten / mFrameSize; 3136 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3137 if (track->isStopped()) { 3138 track->reset(); 3139 } 3140 tracksToRemove->add(track); 3141 } 3142 } else { 3143 // No buffers for this track. Give it a few chances to 3144 // fill a buffer, then remove it from active list. 3145 if (--(track->mRetryCount) <= 0) { 3146 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3147 tracksToRemove->add(track); 3148 // indicate to client process that the track was disabled because of underrun; 3149 // it will then automatically call start() when data is available 3150 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3151 // If one track is not ready, mark the mixer also not ready if: 3152 // - the mixer was ready during previous round OR 3153 // - no other track is ready 3154 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3155 mixerStatus != MIXER_TRACKS_READY) { 3156 mixerStatus = MIXER_TRACKS_ENABLED; 3157 } 3158 } 3159 mAudioMixer->disable(name); 3160 } 3161 3162 } // local variable scope to avoid goto warning 3163track_is_ready: ; 3164 3165 } 3166 3167 // Push the new FastMixer state if necessary 3168 bool pauseAudioWatchdog = false; 3169 if (didModify) { 3170 state->mFastTracksGen++; 3171 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3172 if (kUseFastMixer == FastMixer_Dynamic && 3173 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3174 state->mCommand = FastMixerState::COLD_IDLE; 3175 state->mColdFutexAddr = &mFastMixerFutex; 3176 state->mColdGen++; 3177 mFastMixerFutex = 0; 3178 if (kUseFastMixer == FastMixer_Dynamic) { 3179 mNormalSink = mOutputSink; 3180 } 3181 // If we go into cold idle, need to wait for acknowledgement 3182 // so that fast mixer stops doing I/O. 3183 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3184 pauseAudioWatchdog = true; 3185 } 3186 } 3187 if (sq != NULL) { 3188 sq->end(didModify); 3189 sq->push(block); 3190 } 3191#ifdef AUDIO_WATCHDOG 3192 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3193 mAudioWatchdog->pause(); 3194 } 3195#endif 3196 3197 // Now perform the deferred reset on fast tracks that have stopped 3198 while (resetMask != 0) { 3199 size_t i = __builtin_ctz(resetMask); 3200 ALOG_ASSERT(i < count); 3201 resetMask &= ~(1 << i); 3202 sp<Track> t = mActiveTracks[i].promote(); 3203 if (t == 0) { 3204 continue; 3205 } 3206 Track* track = t.get(); 3207 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3208 track->reset(); 3209 } 3210 3211 // remove all the tracks that need to be... 3212 removeTracks_l(*tracksToRemove); 3213 3214 // mix buffer must be cleared if all tracks are connected to an 3215 // effect chain as in this case the mixer will not write to 3216 // mix buffer and track effects will accumulate into it 3217 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3218 (mixedTracks == 0 && fastTracks > 0))) { 3219 // FIXME as a performance optimization, should remember previous zero status 3220 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3221 } 3222 3223 // if any fast tracks, then status is ready 3224 mMixerStatusIgnoringFastTracks = mixerStatus; 3225 if (fastTracks > 0) { 3226 mixerStatus = MIXER_TRACKS_READY; 3227 } 3228 return mixerStatus; 3229} 3230 3231// getTrackName_l() must be called with ThreadBase::mLock held 3232int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3233{ 3234 return mAudioMixer->getTrackName(channelMask, sessionId); 3235} 3236 3237// deleteTrackName_l() must be called with ThreadBase::mLock held 3238void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3239{ 3240 ALOGV("remove track (%d) and delete from mixer", name); 3241 mAudioMixer->deleteTrackName(name); 3242} 3243 3244// checkForNewParameters_l() must be called with ThreadBase::mLock held 3245bool AudioFlinger::MixerThread::checkForNewParameters_l() 3246{ 3247 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3248 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3249 bool reconfig = false; 3250 3251 while (!mNewParameters.isEmpty()) { 3252 3253 if (mFastMixer != NULL) { 3254 FastMixerStateQueue *sq = mFastMixer->sq(); 3255 FastMixerState *state = sq->begin(); 3256 if (!(state->mCommand & FastMixerState::IDLE)) { 3257 previousCommand = state->mCommand; 3258 state->mCommand = FastMixerState::HOT_IDLE; 3259 sq->end(); 3260 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3261 } else { 3262 sq->end(false /*didModify*/); 3263 } 3264 } 3265 3266 status_t status = NO_ERROR; 3267 String8 keyValuePair = mNewParameters[0]; 3268 AudioParameter param = AudioParameter(keyValuePair); 3269 int value; 3270 3271 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3272 reconfig = true; 3273 } 3274 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3275 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3276 status = BAD_VALUE; 3277 } else { 3278 reconfig = true; 3279 } 3280 } 3281 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3282 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3283 status = BAD_VALUE; 3284 } else { 3285 reconfig = true; 3286 } 3287 } 3288 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3289 // do not accept frame count changes if tracks are open as the track buffer 3290 // size depends on frame count and correct behavior would not be guaranteed 3291 // if frame count is changed after track creation 3292 if (!mTracks.isEmpty()) { 3293 status = INVALID_OPERATION; 3294 } else { 3295 reconfig = true; 3296 } 3297 } 3298 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3299#ifdef ADD_BATTERY_DATA 3300 // when changing the audio output device, call addBatteryData to notify 3301 // the change 3302 if (mOutDevice != value) { 3303 uint32_t params = 0; 3304 // check whether speaker is on 3305 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3306 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3307 } 3308 3309 audio_devices_t deviceWithoutSpeaker 3310 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3311 // check if any other device (except speaker) is on 3312 if (value & deviceWithoutSpeaker ) { 3313 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3314 } 3315 3316 if (params != 0) { 3317 addBatteryData(params); 3318 } 3319 } 3320#endif 3321 3322 // forward device change to effects that have requested to be 3323 // aware of attached audio device. 3324 if (value != AUDIO_DEVICE_NONE) { 3325 mOutDevice = value; 3326 for (size_t i = 0; i < mEffectChains.size(); i++) { 3327 mEffectChains[i]->setDevice_l(mOutDevice); 3328 } 3329 } 3330 } 3331 3332 if (status == NO_ERROR) { 3333 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3334 keyValuePair.string()); 3335 if (!mStandby && status == INVALID_OPERATION) { 3336 mOutput->stream->common.standby(&mOutput->stream->common); 3337 mStandby = true; 3338 mBytesWritten = 0; 3339 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3340 keyValuePair.string()); 3341 } 3342 if (status == NO_ERROR && reconfig) { 3343 readOutputParameters(); 3344 delete mAudioMixer; 3345 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3346 for (size_t i = 0; i < mTracks.size() ; i++) { 3347 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3348 if (name < 0) { 3349 break; 3350 } 3351 mTracks[i]->mName = name; 3352 } 3353 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3354 } 3355 } 3356 3357 mNewParameters.removeAt(0); 3358 3359 mParamStatus = status; 3360 mParamCond.signal(); 3361 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3362 // already timed out waiting for the status and will never signal the condition. 3363 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3364 } 3365 3366 if (!(previousCommand & FastMixerState::IDLE)) { 3367 ALOG_ASSERT(mFastMixer != NULL); 3368 FastMixerStateQueue *sq = mFastMixer->sq(); 3369 FastMixerState *state = sq->begin(); 3370 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3371 state->mCommand = previousCommand; 3372 sq->end(); 3373 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3374 } 3375 3376 return reconfig; 3377} 3378 3379 3380void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3381{ 3382 const size_t SIZE = 256; 3383 char buffer[SIZE]; 3384 String8 result; 3385 3386 PlaybackThread::dumpInternals(fd, args); 3387 3388 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3389 result.append(buffer); 3390 write(fd, result.string(), result.size()); 3391 3392 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3393 const FastMixerDumpState copy(mFastMixerDumpState); 3394 copy.dump(fd); 3395 3396#ifdef STATE_QUEUE_DUMP 3397 // Similar for state queue 3398 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3399 observerCopy.dump(fd); 3400 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3401 mutatorCopy.dump(fd); 3402#endif 3403 3404#ifdef TEE_SINK 3405 // Write the tee output to a .wav file 3406 dumpTee(fd, mTeeSource, mId); 3407#endif 3408 3409#ifdef AUDIO_WATCHDOG 3410 if (mAudioWatchdog != 0) { 3411 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3412 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3413 wdCopy.dump(fd); 3414 } 3415#endif 3416} 3417 3418uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3419{ 3420 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3421} 3422 3423uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3424{ 3425 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3426} 3427 3428void AudioFlinger::MixerThread::cacheParameters_l() 3429{ 3430 PlaybackThread::cacheParameters_l(); 3431 3432 // FIXME: Relaxed timing because of a certain device that can't meet latency 3433 // Should be reduced to 2x after the vendor fixes the driver issue 3434 // increase threshold again due to low power audio mode. The way this warning 3435 // threshold is calculated and its usefulness should be reconsidered anyway. 3436 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3437} 3438 3439// ---------------------------------------------------------------------------- 3440 3441AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3442 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3443 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3444 // mLeftVolFloat, mRightVolFloat 3445{ 3446} 3447 3448AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3449 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3450 ThreadBase::type_t type) 3451 : PlaybackThread(audioFlinger, output, id, device, type) 3452 // mLeftVolFloat, mRightVolFloat 3453{ 3454} 3455 3456AudioFlinger::DirectOutputThread::~DirectOutputThread() 3457{ 3458} 3459 3460void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3461{ 3462 audio_track_cblk_t* cblk = track->cblk(); 3463 float left, right; 3464 3465 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3466 left = right = 0; 3467 } else { 3468 float typeVolume = mStreamTypes[track->streamType()].volume; 3469 float v = mMasterVolume * typeVolume; 3470 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3471 uint32_t vlr = proxy->getVolumeLR(); 3472 float v_clamped = v * (vlr & 0xFFFF); 3473 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3474 left = v_clamped/MAX_GAIN; 3475 v_clamped = v * (vlr >> 16); 3476 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3477 right = v_clamped/MAX_GAIN; 3478 } 3479 3480 if (lastTrack) { 3481 if (left != mLeftVolFloat || right != mRightVolFloat) { 3482 mLeftVolFloat = left; 3483 mRightVolFloat = right; 3484 3485 // Convert volumes from float to 8.24 3486 uint32_t vl = (uint32_t)(left * (1 << 24)); 3487 uint32_t vr = (uint32_t)(right * (1 << 24)); 3488 3489 // Delegate volume control to effect in track effect chain if needed 3490 // only one effect chain can be present on DirectOutputThread, so if 3491 // there is one, the track is connected to it 3492 if (!mEffectChains.isEmpty()) { 3493 mEffectChains[0]->setVolume_l(&vl, &vr); 3494 left = (float)vl / (1 << 24); 3495 right = (float)vr / (1 << 24); 3496 } 3497 if (mOutput->stream->set_volume) { 3498 mOutput->stream->set_volume(mOutput->stream, left, right); 3499 } 3500 } 3501 } 3502} 3503 3504 3505AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3506 Vector< sp<Track> > *tracksToRemove 3507) 3508{ 3509 size_t count = mActiveTracks.size(); 3510 mixer_state mixerStatus = MIXER_IDLE; 3511 3512 // find out which tracks need to be processed 3513 for (size_t i = 0; i < count; i++) { 3514 sp<Track> t = mActiveTracks[i].promote(); 3515 // The track died recently 3516 if (t == 0) { 3517 continue; 3518 } 3519 3520 Track* const track = t.get(); 3521 audio_track_cblk_t* cblk = track->cblk(); 3522 3523 // The first time a track is added we wait 3524 // for all its buffers to be filled before processing it 3525 uint32_t minFrames; 3526 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3527 minFrames = mNormalFrameCount; 3528 } else { 3529 minFrames = 1; 3530 } 3531 // Only consider last track started for volume and mixer state control. 3532 // This is the last entry in mActiveTracks unless a track underruns. 3533 // As we only care about the transition phase between two tracks on a 3534 // direct output, it is not a problem to ignore the underrun case. 3535 bool last = (i == (count - 1)); 3536 3537 if ((track->framesReady() >= minFrames) && track->isReady() && 3538 !track->isPaused() && !track->isTerminated()) 3539 { 3540 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3541 3542 if (track->mFillingUpStatus == Track::FS_FILLED) { 3543 track->mFillingUpStatus = Track::FS_ACTIVE; 3544 // make sure processVolume_l() will apply new volume even if 0 3545 mLeftVolFloat = mRightVolFloat = -1.0; 3546 if (track->mState == TrackBase::RESUMING) { 3547 track->mState = TrackBase::ACTIVE; 3548 } 3549 } 3550 3551 // compute volume for this track 3552 processVolume_l(track, last); 3553 if (last) { 3554 // reset retry count 3555 track->mRetryCount = kMaxTrackRetriesDirect; 3556 mActiveTrack = t; 3557 mixerStatus = MIXER_TRACKS_READY; 3558 } 3559 } else { 3560 // clear effect chain input buffer if the last active track started underruns 3561 // to avoid sending previous audio buffer again to effects 3562 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3563 mEffectChains[0]->clearInputBuffer(); 3564 } 3565 3566 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3567 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3568 track->isStopped() || track->isPaused()) { 3569 // We have consumed all the buffers of this track. 3570 // Remove it from the list of active tracks. 3571 // TODO: implement behavior for compressed audio 3572 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3573 size_t framesWritten = mBytesWritten / mFrameSize; 3574 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3575 if (track->isStopped()) { 3576 track->reset(); 3577 } 3578 tracksToRemove->add(track); 3579 } 3580 } else { 3581 // No buffers for this track. Give it a few chances to 3582 // fill a buffer, then remove it from active list. 3583 // Only consider last track started for mixer state control 3584 if (--(track->mRetryCount) <= 0) { 3585 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3586 tracksToRemove->add(track); 3587 } else if (last) { 3588 mixerStatus = MIXER_TRACKS_ENABLED; 3589 } 3590 } 3591 } 3592 } 3593 3594 // remove all the tracks that need to be... 3595 removeTracks_l(*tracksToRemove); 3596 3597 return mixerStatus; 3598} 3599 3600void AudioFlinger::DirectOutputThread::threadLoop_mix() 3601{ 3602 size_t frameCount = mFrameCount; 3603 int8_t *curBuf = (int8_t *)mMixBuffer; 3604 // output audio to hardware 3605 while (frameCount) { 3606 AudioBufferProvider::Buffer buffer; 3607 buffer.frameCount = frameCount; 3608 mActiveTrack->getNextBuffer(&buffer); 3609 if (buffer.raw == NULL) { 3610 memset(curBuf, 0, frameCount * mFrameSize); 3611 break; 3612 } 3613 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3614 frameCount -= buffer.frameCount; 3615 curBuf += buffer.frameCount * mFrameSize; 3616 mActiveTrack->releaseBuffer(&buffer); 3617 } 3618 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3619 sleepTime = 0; 3620 standbyTime = systemTime() + standbyDelay; 3621 mActiveTrack.clear(); 3622} 3623 3624void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3625{ 3626 if (sleepTime == 0) { 3627 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3628 sleepTime = activeSleepTime; 3629 } else { 3630 sleepTime = idleSleepTime; 3631 } 3632 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3633 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3634 sleepTime = 0; 3635 } 3636} 3637 3638// getTrackName_l() must be called with ThreadBase::mLock held 3639int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3640 int sessionId) 3641{ 3642 return 0; 3643} 3644 3645// deleteTrackName_l() must be called with ThreadBase::mLock held 3646void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3647{ 3648} 3649 3650// checkForNewParameters_l() must be called with ThreadBase::mLock held 3651bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3652{ 3653 bool reconfig = false; 3654 3655 while (!mNewParameters.isEmpty()) { 3656 status_t status = NO_ERROR; 3657 String8 keyValuePair = mNewParameters[0]; 3658 AudioParameter param = AudioParameter(keyValuePair); 3659 int value; 3660 3661 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3662 // do not accept frame count changes if tracks are open as the track buffer 3663 // size depends on frame count and correct behavior would not be garantied 3664 // if frame count is changed after track creation 3665 if (!mTracks.isEmpty()) { 3666 status = INVALID_OPERATION; 3667 } else { 3668 reconfig = true; 3669 } 3670 } 3671 if (status == NO_ERROR) { 3672 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3673 keyValuePair.string()); 3674 if (!mStandby && status == INVALID_OPERATION) { 3675 mOutput->stream->common.standby(&mOutput->stream->common); 3676 mStandby = true; 3677 mBytesWritten = 0; 3678 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3679 keyValuePair.string()); 3680 } 3681 if (status == NO_ERROR && reconfig) { 3682 readOutputParameters(); 3683 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3684 } 3685 } 3686 3687 mNewParameters.removeAt(0); 3688 3689 mParamStatus = status; 3690 mParamCond.signal(); 3691 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3692 // already timed out waiting for the status and will never signal the condition. 3693 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3694 } 3695 return reconfig; 3696} 3697 3698uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3699{ 3700 uint32_t time; 3701 if (audio_is_linear_pcm(mFormat)) { 3702 time = PlaybackThread::activeSleepTimeUs(); 3703 } else { 3704 time = 10000; 3705 } 3706 return time; 3707} 3708 3709uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3710{ 3711 uint32_t time; 3712 if (audio_is_linear_pcm(mFormat)) { 3713 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3714 } else { 3715 time = 10000; 3716 } 3717 return time; 3718} 3719 3720uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3721{ 3722 uint32_t time; 3723 if (audio_is_linear_pcm(mFormat)) { 3724 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3725 } else { 3726 time = 10000; 3727 } 3728 return time; 3729} 3730 3731void AudioFlinger::DirectOutputThread::cacheParameters_l() 3732{ 3733 PlaybackThread::cacheParameters_l(); 3734 3735 // use shorter standby delay as on normal output to release 3736 // hardware resources as soon as possible 3737 if (audio_is_linear_pcm(mFormat)) { 3738 standbyDelay = microseconds(activeSleepTime*2); 3739 } else { 3740 standbyDelay = kOffloadStandbyDelayNs; 3741 } 3742} 3743 3744// ---------------------------------------------------------------------------- 3745 3746AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3747 const sp<AudioFlinger::OffloadThread>& offloadThread) 3748 : Thread(false /*canCallJava*/), 3749 mOffloadThread(offloadThread), 3750 mWriteAckSequence(0), 3751 mDrainSequence(0) 3752{ 3753} 3754 3755AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3756{ 3757} 3758 3759void AudioFlinger::AsyncCallbackThread::onFirstRef() 3760{ 3761 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3762} 3763 3764bool AudioFlinger::AsyncCallbackThread::threadLoop() 3765{ 3766 while (!exitPending()) { 3767 uint32_t writeAckSequence; 3768 uint32_t drainSequence; 3769 3770 { 3771 Mutex::Autolock _l(mLock); 3772 mWaitWorkCV.wait(mLock); 3773 if (exitPending()) { 3774 break; 3775 } 3776 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3777 mWriteAckSequence, mDrainSequence); 3778 writeAckSequence = mWriteAckSequence; 3779 mWriteAckSequence &= ~1; 3780 drainSequence = mDrainSequence; 3781 mDrainSequence &= ~1; 3782 } 3783 { 3784 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3785 if (offloadThread != 0) { 3786 if (writeAckSequence & 1) { 3787 offloadThread->resetWriteBlocked(writeAckSequence >> 1); 3788 } 3789 if (drainSequence & 1) { 3790 offloadThread->resetDraining(drainSequence >> 1); 3791 } 3792 } 3793 } 3794 } 3795 return false; 3796} 3797 3798void AudioFlinger::AsyncCallbackThread::exit() 3799{ 3800 ALOGV("AsyncCallbackThread::exit"); 3801 Mutex::Autolock _l(mLock); 3802 requestExit(); 3803 mWaitWorkCV.broadcast(); 3804} 3805 3806void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3807{ 3808 Mutex::Autolock _l(mLock); 3809 // bit 0 is cleared 3810 mWriteAckSequence = sequence << 1; 3811} 3812 3813void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3814{ 3815 Mutex::Autolock _l(mLock); 3816 // ignore unexpected callbacks 3817 if (mWriteAckSequence & 2) { 3818 mWriteAckSequence |= 1; 3819 mWaitWorkCV.signal(); 3820 } 3821} 3822 3823void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3824{ 3825 Mutex::Autolock _l(mLock); 3826 // bit 0 is cleared 3827 mDrainSequence = sequence << 1; 3828} 3829 3830void AudioFlinger::AsyncCallbackThread::resetDraining() 3831{ 3832 Mutex::Autolock _l(mLock); 3833 // ignore unexpected callbacks 3834 if (mDrainSequence & 2) { 3835 mDrainSequence |= 1; 3836 mWaitWorkCV.signal(); 3837 } 3838} 3839 3840 3841// ---------------------------------------------------------------------------- 3842AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3843 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3844 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3845 mHwPaused(false), 3846 mPausedBytesRemaining(0) 3847{ 3848 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3849} 3850 3851AudioFlinger::OffloadThread::~OffloadThread() 3852{ 3853 mPreviousTrack.clear(); 3854} 3855 3856void AudioFlinger::OffloadThread::threadLoop_exit() 3857{ 3858 if (mFlushPending || mHwPaused) { 3859 // If a flush is pending or track was paused, just discard buffered data 3860 flushHw_l(); 3861 } else { 3862 mMixerStatus = MIXER_DRAIN_ALL; 3863 threadLoop_drain(); 3864 } 3865 mCallbackThread->exit(); 3866 PlaybackThread::threadLoop_exit(); 3867} 3868 3869AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3870 Vector< sp<Track> > *tracksToRemove 3871) 3872{ 3873 size_t count = mActiveTracks.size(); 3874 3875 mixer_state mixerStatus = MIXER_IDLE; 3876 bool doHwPause = false; 3877 bool doHwResume = false; 3878 3879 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3880 3881 // find out which tracks need to be processed 3882 for (size_t i = 0; i < count; i++) { 3883 sp<Track> t = mActiveTracks[i].promote(); 3884 // The track died recently 3885 if (t == 0) { 3886 continue; 3887 } 3888 Track* const track = t.get(); 3889 audio_track_cblk_t* cblk = track->cblk(); 3890 if (mPreviousTrack != NULL) { 3891 if (t != mPreviousTrack) { 3892 // Flush any data still being written from last track 3893 mBytesRemaining = 0; 3894 if (mPausedBytesRemaining) { 3895 // Last track was paused so we also need to flush saved 3896 // mixbuffer state and invalidate track so that it will 3897 // re-submit that unwritten data when it is next resumed 3898 mPausedBytesRemaining = 0; 3899 // Invalidate is a bit drastic - would be more efficient 3900 // to have a flag to tell client that some of the 3901 // previously written data was lost 3902 mPreviousTrack->invalidate(); 3903 } 3904 } 3905 } 3906 mPreviousTrack = t; 3907 bool last = (i == (count - 1)); 3908 if (track->isPausing()) { 3909 track->setPaused(); 3910 if (last) { 3911 if (!mHwPaused) { 3912 doHwPause = true; 3913 mHwPaused = true; 3914 } 3915 // If we were part way through writing the mixbuffer to 3916 // the HAL we must save this until we resume 3917 // BUG - this will be wrong if a different track is made active, 3918 // in that case we want to discard the pending data in the 3919 // mixbuffer and tell the client to present it again when the 3920 // track is resumed 3921 mPausedWriteLength = mCurrentWriteLength; 3922 mPausedBytesRemaining = mBytesRemaining; 3923 mBytesRemaining = 0; // stop writing 3924 } 3925 tracksToRemove->add(track); 3926 } else if (track->framesReady() && track->isReady() && 3927 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3928 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3929 if (track->mFillingUpStatus == Track::FS_FILLED) { 3930 track->mFillingUpStatus = Track::FS_ACTIVE; 3931 // make sure processVolume_l() will apply new volume even if 0 3932 mLeftVolFloat = mRightVolFloat = -1.0; 3933 if (track->mState == TrackBase::RESUMING) { 3934 track->mState = TrackBase::ACTIVE; 3935 if (last) { 3936 if (mPausedBytesRemaining) { 3937 // Need to continue write that was interrupted 3938 mCurrentWriteLength = mPausedWriteLength; 3939 mBytesRemaining = mPausedBytesRemaining; 3940 mPausedBytesRemaining = 0; 3941 } 3942 if (mHwPaused) { 3943 doHwResume = true; 3944 mHwPaused = false; 3945 // threadLoop_mix() will handle the case that we need to 3946 // resume an interrupted write 3947 } 3948 // enable write to audio HAL 3949 sleepTime = 0; 3950 } 3951 } 3952 } 3953 3954 if (last) { 3955 // reset retry count 3956 track->mRetryCount = kMaxTrackRetriesOffload; 3957 mActiveTrack = t; 3958 mixerStatus = MIXER_TRACKS_READY; 3959 } 3960 } else { 3961 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3962 if (track->isStopping_1()) { 3963 // Hardware buffer can hold a large amount of audio so we must 3964 // wait for all current track's data to drain before we say 3965 // that the track is stopped. 3966 if (mBytesRemaining == 0) { 3967 // Only start draining when all data in mixbuffer 3968 // has been written 3969 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3970 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3971 if (last) { 3972 sleepTime = 0; 3973 standbyTime = systemTime() + standbyDelay; 3974 mixerStatus = MIXER_DRAIN_TRACK; 3975 mDrainSequence += 2; 3976 if (mHwPaused) { 3977 // It is possible to move from PAUSED to STOPPING_1 without 3978 // a resume so we must ensure hardware is running 3979 mOutput->stream->resume(mOutput->stream); 3980 mHwPaused = false; 3981 } 3982 } 3983 } 3984 } else if (track->isStopping_2()) { 3985 // Drain has completed, signal presentation complete 3986 if (!(mDrainSequence & 1) || !last) { 3987 track->mState = TrackBase::STOPPED; 3988 size_t audioHALFrames = 3989 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3990 size_t framesWritten = 3991 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3992 track->presentationComplete(framesWritten, audioHALFrames); 3993 track->reset(); 3994 tracksToRemove->add(track); 3995 } 3996 } else { 3997 // No buffers for this track. Give it a few chances to 3998 // fill a buffer, then remove it from active list. 3999 if (--(track->mRetryCount) <= 0) { 4000 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4001 track->name()); 4002 tracksToRemove->add(track); 4003 } else if (last){ 4004 mixerStatus = MIXER_TRACKS_ENABLED; 4005 } 4006 } 4007 } 4008 // compute volume for this track 4009 processVolume_l(track, last); 4010 } 4011 4012 // make sure the pause/flush/resume sequence is executed in the right order 4013 if (doHwPause) { 4014 mOutput->stream->pause(mOutput->stream); 4015 } 4016 if (mFlushPending) { 4017 flushHw_l(); 4018 mFlushPending = false; 4019 } 4020 if (doHwResume) { 4021 mOutput->stream->resume(mOutput->stream); 4022 } 4023 4024 // remove all the tracks that need to be... 4025 removeTracks_l(*tracksToRemove); 4026 4027 return mixerStatus; 4028} 4029 4030void AudioFlinger::OffloadThread::flushOutput_l() 4031{ 4032 mFlushPending = true; 4033} 4034 4035// must be called with thread mutex locked 4036bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4037{ 4038 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4039 mWriteAckSequence, mDrainSequence); 4040 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4041 return true; 4042 } 4043 return false; 4044} 4045 4046// must be called with thread mutex locked 4047bool AudioFlinger::OffloadThread::shouldStandby_l() 4048{ 4049 bool TrackPaused = false; 4050 4051 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4052 // after a timeout and we will enter standby then. 4053 if (mTracks.size() > 0) { 4054 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4055 } 4056 4057 return !mStandby && !TrackPaused; 4058} 4059 4060 4061bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4062{ 4063 Mutex::Autolock _l(mLock); 4064 return waitingAsyncCallback_l(); 4065} 4066 4067void AudioFlinger::OffloadThread::flushHw_l() 4068{ 4069 mOutput->stream->flush(mOutput->stream); 4070 // Flush anything still waiting in the mixbuffer 4071 mCurrentWriteLength = 0; 4072 mBytesRemaining = 0; 4073 mPausedWriteLength = 0; 4074 mPausedBytesRemaining = 0; 4075 if (mUseAsyncWrite) { 4076 // discard any pending drain or write ack by incrementing sequence 4077 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4078 mDrainSequence = (mDrainSequence + 2) & ~1; 4079 ALOG_ASSERT(mCallbackThread != 0); 4080 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4081 mCallbackThread->setDraining(mDrainSequence); 4082 } 4083} 4084 4085// ---------------------------------------------------------------------------- 4086 4087AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4088 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4089 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4090 DUPLICATING), 4091 mWaitTimeMs(UINT_MAX) 4092{ 4093 addOutputTrack(mainThread); 4094} 4095 4096AudioFlinger::DuplicatingThread::~DuplicatingThread() 4097{ 4098 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4099 mOutputTracks[i]->destroy(); 4100 } 4101} 4102 4103void AudioFlinger::DuplicatingThread::threadLoop_mix() 4104{ 4105 // mix buffers... 4106 if (outputsReady(outputTracks)) { 4107 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4108 } else { 4109 memset(mMixBuffer, 0, mixBufferSize); 4110 } 4111 sleepTime = 0; 4112 writeFrames = mNormalFrameCount; 4113 mCurrentWriteLength = mixBufferSize; 4114 standbyTime = systemTime() + standbyDelay; 4115} 4116 4117void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4118{ 4119 if (sleepTime == 0) { 4120 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4121 sleepTime = activeSleepTime; 4122 } else { 4123 sleepTime = idleSleepTime; 4124 } 4125 } else if (mBytesWritten != 0) { 4126 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4127 writeFrames = mNormalFrameCount; 4128 memset(mMixBuffer, 0, mixBufferSize); 4129 } else { 4130 // flush remaining overflow buffers in output tracks 4131 writeFrames = 0; 4132 } 4133 sleepTime = 0; 4134 } 4135} 4136 4137ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4138{ 4139 for (size_t i = 0; i < outputTracks.size(); i++) { 4140 outputTracks[i]->write(mMixBuffer, writeFrames); 4141 } 4142 return (ssize_t)mixBufferSize; 4143} 4144 4145void AudioFlinger::DuplicatingThread::threadLoop_standby() 4146{ 4147 // DuplicatingThread implements standby by stopping all tracks 4148 for (size_t i = 0; i < outputTracks.size(); i++) { 4149 outputTracks[i]->stop(); 4150 } 4151} 4152 4153void AudioFlinger::DuplicatingThread::saveOutputTracks() 4154{ 4155 outputTracks = mOutputTracks; 4156} 4157 4158void AudioFlinger::DuplicatingThread::clearOutputTracks() 4159{ 4160 outputTracks.clear(); 4161} 4162 4163void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4164{ 4165 Mutex::Autolock _l(mLock); 4166 // FIXME explain this formula 4167 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4168 OutputTrack *outputTrack = new OutputTrack(thread, 4169 this, 4170 mSampleRate, 4171 mFormat, 4172 mChannelMask, 4173 frameCount); 4174 if (outputTrack->cblk() != NULL) { 4175 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4176 mOutputTracks.add(outputTrack); 4177 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4178 updateWaitTime_l(); 4179 } 4180} 4181 4182void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4183{ 4184 Mutex::Autolock _l(mLock); 4185 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4186 if (mOutputTracks[i]->thread() == thread) { 4187 mOutputTracks[i]->destroy(); 4188 mOutputTracks.removeAt(i); 4189 updateWaitTime_l(); 4190 return; 4191 } 4192 } 4193 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4194} 4195 4196// caller must hold mLock 4197void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4198{ 4199 mWaitTimeMs = UINT_MAX; 4200 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4201 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4202 if (strong != 0) { 4203 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4204 if (waitTimeMs < mWaitTimeMs) { 4205 mWaitTimeMs = waitTimeMs; 4206 } 4207 } 4208 } 4209} 4210 4211 4212bool AudioFlinger::DuplicatingThread::outputsReady( 4213 const SortedVector< sp<OutputTrack> > &outputTracks) 4214{ 4215 for (size_t i = 0; i < outputTracks.size(); i++) { 4216 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4217 if (thread == 0) { 4218 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4219 outputTracks[i].get()); 4220 return false; 4221 } 4222 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4223 // see note at standby() declaration 4224 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4225 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4226 thread.get()); 4227 return false; 4228 } 4229 } 4230 return true; 4231} 4232 4233uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4234{ 4235 return (mWaitTimeMs * 1000) / 2; 4236} 4237 4238void AudioFlinger::DuplicatingThread::cacheParameters_l() 4239{ 4240 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4241 updateWaitTime_l(); 4242 4243 MixerThread::cacheParameters_l(); 4244} 4245 4246// ---------------------------------------------------------------------------- 4247// Record 4248// ---------------------------------------------------------------------------- 4249 4250AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4251 AudioStreamIn *input, 4252 uint32_t sampleRate, 4253 audio_channel_mask_t channelMask, 4254 audio_io_handle_t id, 4255 audio_devices_t outDevice, 4256 audio_devices_t inDevice 4257#ifdef TEE_SINK 4258 , const sp<NBAIO_Sink>& teeSink 4259#endif 4260 ) : 4261 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4262 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4263 // mRsmpInIndex and mBufferSize set by readInputParameters() 4264 mReqChannelCount(popcount(channelMask)), 4265 mReqSampleRate(sampleRate) 4266 // mBytesRead is only meaningful while active, and so is cleared in start() 4267 // (but might be better to also clear here for dump?) 4268#ifdef TEE_SINK 4269 , mTeeSink(teeSink) 4270#endif 4271{ 4272 snprintf(mName, kNameLength, "AudioIn_%X", id); 4273 4274 readInputParameters(); 4275 4276} 4277 4278 4279AudioFlinger::RecordThread::~RecordThread() 4280{ 4281 delete[] mRsmpInBuffer; 4282 delete mResampler; 4283 delete[] mRsmpOutBuffer; 4284} 4285 4286void AudioFlinger::RecordThread::onFirstRef() 4287{ 4288 run(mName, PRIORITY_URGENT_AUDIO); 4289} 4290 4291status_t AudioFlinger::RecordThread::readyToRun() 4292{ 4293 status_t status = initCheck(); 4294 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4295 return status; 4296} 4297 4298bool AudioFlinger::RecordThread::threadLoop() 4299{ 4300 AudioBufferProvider::Buffer buffer; 4301 sp<RecordTrack> activeTrack; 4302 Vector< sp<EffectChain> > effectChains; 4303 4304 nsecs_t lastWarning = 0; 4305 4306 inputStandBy(); 4307 acquireWakeLock(); 4308 4309 // used to verify we've read at least once before evaluating how many bytes were read 4310 bool readOnce = false; 4311 4312 // start recording 4313 while (!exitPending()) { 4314 4315 processConfigEvents(); 4316 4317 { // scope for mLock 4318 Mutex::Autolock _l(mLock); 4319 checkForNewParameters_l(); 4320 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4321 standby(); 4322 4323 if (exitPending()) { 4324 break; 4325 } 4326 4327 releaseWakeLock_l(); 4328 ALOGV("RecordThread: loop stopping"); 4329 // go to sleep 4330 mWaitWorkCV.wait(mLock); 4331 ALOGV("RecordThread: loop starting"); 4332 acquireWakeLock_l(); 4333 continue; 4334 } 4335 if (mActiveTrack != 0) { 4336 if (mActiveTrack->isTerminated()) { 4337 removeTrack_l(mActiveTrack); 4338 mActiveTrack.clear(); 4339 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4340 standby(); 4341 mActiveTrack.clear(); 4342 mStartStopCond.broadcast(); 4343 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4344 if (mReqChannelCount != mActiveTrack->channelCount()) { 4345 mActiveTrack.clear(); 4346 mStartStopCond.broadcast(); 4347 } else if (readOnce) { 4348 // record start succeeds only if first read from audio input 4349 // succeeds 4350 if (mBytesRead >= 0) { 4351 mActiveTrack->mState = TrackBase::ACTIVE; 4352 } else { 4353 mActiveTrack.clear(); 4354 } 4355 mStartStopCond.broadcast(); 4356 } 4357 mStandby = false; 4358 } 4359 } 4360 4361 lockEffectChains_l(effectChains); 4362 } 4363 4364 if (mActiveTrack != 0) { 4365 if (mActiveTrack->mState != TrackBase::ACTIVE && 4366 mActiveTrack->mState != TrackBase::RESUMING) { 4367 unlockEffectChains(effectChains); 4368 usleep(kRecordThreadSleepUs); 4369 continue; 4370 } 4371 for (size_t i = 0; i < effectChains.size(); i ++) { 4372 effectChains[i]->process_l(); 4373 } 4374 4375 buffer.frameCount = mFrameCount; 4376 status_t status = mActiveTrack->getNextBuffer(&buffer); 4377 if (status == NO_ERROR) { 4378 readOnce = true; 4379 size_t framesOut = buffer.frameCount; 4380 if (mResampler == NULL) { 4381 // no resampling 4382 while (framesOut) { 4383 size_t framesIn = mFrameCount - mRsmpInIndex; 4384 if (framesIn) { 4385 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4386 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4387 mActiveTrack->mFrameSize; 4388 if (framesIn > framesOut) 4389 framesIn = framesOut; 4390 mRsmpInIndex += framesIn; 4391 framesOut -= framesIn; 4392 if (mChannelCount == mReqChannelCount) { 4393 memcpy(dst, src, framesIn * mFrameSize); 4394 } else { 4395 if (mChannelCount == 1) { 4396 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4397 (int16_t *)src, framesIn); 4398 } else { 4399 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4400 (int16_t *)src, framesIn); 4401 } 4402 } 4403 } 4404 if (framesOut && mFrameCount == mRsmpInIndex) { 4405 void *readInto; 4406 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4407 readInto = buffer.raw; 4408 framesOut = 0; 4409 } else { 4410 readInto = mRsmpInBuffer; 4411 mRsmpInIndex = 0; 4412 } 4413 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4414 mBufferSize); 4415 if (mBytesRead <= 0) { 4416 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4417 { 4418 ALOGE("Error reading audio input"); 4419 // Force input into standby so that it tries to 4420 // recover at next read attempt 4421 inputStandBy(); 4422 usleep(kRecordThreadSleepUs); 4423 } 4424 mRsmpInIndex = mFrameCount; 4425 framesOut = 0; 4426 buffer.frameCount = 0; 4427 } 4428#ifdef TEE_SINK 4429 else if (mTeeSink != 0) { 4430 (void) mTeeSink->write(readInto, 4431 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4432 } 4433#endif 4434 } 4435 } 4436 } else { 4437 // resampling 4438 4439 // resampler accumulates, but we only have one source track 4440 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4441 // alter output frame count as if we were expecting stereo samples 4442 if (mChannelCount == 1 && mReqChannelCount == 1) { 4443 framesOut >>= 1; 4444 } 4445 mResampler->resample(mRsmpOutBuffer, framesOut, 4446 this /* AudioBufferProvider* */); 4447 // ditherAndClamp() works as long as all buffers returned by 4448 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4449 if (mChannelCount == 2 && mReqChannelCount == 1) { 4450 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4451 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4452 // the resampler always outputs stereo samples: 4453 // do post stereo to mono conversion 4454 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4455 framesOut); 4456 } else { 4457 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4458 } 4459 // now done with mRsmpOutBuffer 4460 4461 } 4462 if (mFramestoDrop == 0) { 4463 mActiveTrack->releaseBuffer(&buffer); 4464 } else { 4465 if (mFramestoDrop > 0) { 4466 mFramestoDrop -= buffer.frameCount; 4467 if (mFramestoDrop <= 0) { 4468 clearSyncStartEvent(); 4469 } 4470 } else { 4471 mFramestoDrop += buffer.frameCount; 4472 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4473 mSyncStartEvent->isCancelled()) { 4474 ALOGW("Synced record %s, session %d, trigger session %d", 4475 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4476 mActiveTrack->sessionId(), 4477 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4478 clearSyncStartEvent(); 4479 } 4480 } 4481 } 4482 mActiveTrack->clearOverflow(); 4483 } 4484 // client isn't retrieving buffers fast enough 4485 else { 4486 if (!mActiveTrack->setOverflow()) { 4487 nsecs_t now = systemTime(); 4488 if ((now - lastWarning) > kWarningThrottleNs) { 4489 ALOGW("RecordThread: buffer overflow"); 4490 lastWarning = now; 4491 } 4492 } 4493 // Release the processor for a while before asking for a new buffer. 4494 // This will give the application more chance to read from the buffer and 4495 // clear the overflow. 4496 usleep(kRecordThreadSleepUs); 4497 } 4498 } 4499 // enable changes in effect chain 4500 unlockEffectChains(effectChains); 4501 effectChains.clear(); 4502 } 4503 4504 standby(); 4505 4506 { 4507 Mutex::Autolock _l(mLock); 4508 for (size_t i = 0; i < mTracks.size(); i++) { 4509 sp<RecordTrack> track = mTracks[i]; 4510 track->invalidate(); 4511 } 4512 mActiveTrack.clear(); 4513 mStartStopCond.broadcast(); 4514 } 4515 4516 releaseWakeLock(); 4517 4518 ALOGV("RecordThread %p exiting", this); 4519 return false; 4520} 4521 4522void AudioFlinger::RecordThread::standby() 4523{ 4524 if (!mStandby) { 4525 inputStandBy(); 4526 mStandby = true; 4527 } 4528} 4529 4530void AudioFlinger::RecordThread::inputStandBy() 4531{ 4532 mInput->stream->common.standby(&mInput->stream->common); 4533} 4534 4535sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4536 const sp<AudioFlinger::Client>& client, 4537 uint32_t sampleRate, 4538 audio_format_t format, 4539 audio_channel_mask_t channelMask, 4540 size_t frameCount, 4541 int sessionId, 4542 IAudioFlinger::track_flags_t *flags, 4543 pid_t tid, 4544 status_t *status) 4545{ 4546 sp<RecordTrack> track; 4547 status_t lStatus; 4548 4549 lStatus = initCheck(); 4550 if (lStatus != NO_ERROR) { 4551 ALOGE("createRecordTrack_l() audio driver not initialized"); 4552 goto Exit; 4553 } 4554 4555 // client expresses a preference for FAST, but we get the final say 4556 if (*flags & IAudioFlinger::TRACK_FAST) { 4557 if ( 4558 // use case: callback handler and frame count is default or at least as large as HAL 4559 ( 4560 (tid != -1) && 4561 ((frameCount == 0) || 4562 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4563 ) && 4564 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4565 // mono or stereo 4566 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4567 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4568 // hardware sample rate 4569 (sampleRate == mSampleRate) && 4570 // record thread has an associated fast recorder 4571 hasFastRecorder() 4572 // FIXME test that RecordThread for this fast track has a capable output HAL 4573 // FIXME add a permission test also? 4574 ) { 4575 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4576 if (frameCount == 0) { 4577 frameCount = mFrameCount * kFastTrackMultiplier; 4578 } 4579 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4580 frameCount, mFrameCount); 4581 } else { 4582 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4583 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4584 "hasFastRecorder=%d tid=%d", 4585 frameCount, mFrameCount, format, 4586 audio_is_linear_pcm(format), 4587 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4588 *flags &= ~IAudioFlinger::TRACK_FAST; 4589 // For compatibility with AudioRecord calculation, buffer depth is forced 4590 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4591 // This is probably too conservative, but legacy application code may depend on it. 4592 // If you change this calculation, also review the start threshold which is related. 4593 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4594 size_t mNormalFrameCount = 2048; // FIXME 4595 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4596 if (minBufCount < 2) { 4597 minBufCount = 2; 4598 } 4599 size_t minFrameCount = mNormalFrameCount * minBufCount; 4600 if (frameCount < minFrameCount) { 4601 frameCount = minFrameCount; 4602 } 4603 } 4604 } 4605 4606 // FIXME use flags and tid similar to createTrack_l() 4607 4608 { // scope for mLock 4609 Mutex::Autolock _l(mLock); 4610 4611 track = new RecordTrack(this, client, sampleRate, 4612 format, channelMask, frameCount, sessionId); 4613 4614 if (track->getCblk() == 0) { 4615 ALOGE("createRecordTrack_l() no control block"); 4616 lStatus = NO_MEMORY; 4617 track.clear(); 4618 goto Exit; 4619 } 4620 mTracks.add(track); 4621 4622 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4623 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4624 mAudioFlinger->btNrecIsOff(); 4625 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4626 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4627 4628 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4629 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4630 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4631 // so ask activity manager to do this on our behalf 4632 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4633 } 4634 } 4635 lStatus = NO_ERROR; 4636 4637Exit: 4638 if (status) { 4639 *status = lStatus; 4640 } 4641 return track; 4642} 4643 4644status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4645 AudioSystem::sync_event_t event, 4646 int triggerSession) 4647{ 4648 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4649 sp<ThreadBase> strongMe = this; 4650 status_t status = NO_ERROR; 4651 4652 if (event == AudioSystem::SYNC_EVENT_NONE) { 4653 clearSyncStartEvent(); 4654 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4655 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4656 triggerSession, 4657 recordTrack->sessionId(), 4658 syncStartEventCallback, 4659 this); 4660 // Sync event can be cancelled by the trigger session if the track is not in a 4661 // compatible state in which case we start record immediately 4662 if (mSyncStartEvent->isCancelled()) { 4663 clearSyncStartEvent(); 4664 } else { 4665 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4666 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4667 } 4668 } 4669 4670 { 4671 AutoMutex lock(mLock); 4672 if (mActiveTrack != 0) { 4673 if (recordTrack != mActiveTrack.get()) { 4674 status = -EBUSY; 4675 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4676 mActiveTrack->mState = TrackBase::ACTIVE; 4677 } 4678 return status; 4679 } 4680 4681 recordTrack->mState = TrackBase::IDLE; 4682 mActiveTrack = recordTrack; 4683 mLock.unlock(); 4684 status_t status = AudioSystem::startInput(mId); 4685 mLock.lock(); 4686 if (status != NO_ERROR) { 4687 mActiveTrack.clear(); 4688 clearSyncStartEvent(); 4689 return status; 4690 } 4691 mRsmpInIndex = mFrameCount; 4692 mBytesRead = 0; 4693 if (mResampler != NULL) { 4694 mResampler->reset(); 4695 } 4696 mActiveTrack->mState = TrackBase::RESUMING; 4697 // signal thread to start 4698 ALOGV("Signal record thread"); 4699 mWaitWorkCV.broadcast(); 4700 // do not wait for mStartStopCond if exiting 4701 if (exitPending()) { 4702 mActiveTrack.clear(); 4703 status = INVALID_OPERATION; 4704 goto startError; 4705 } 4706 mStartStopCond.wait(mLock); 4707 if (mActiveTrack == 0) { 4708 ALOGV("Record failed to start"); 4709 status = BAD_VALUE; 4710 goto startError; 4711 } 4712 ALOGV("Record started OK"); 4713 return status; 4714 } 4715 4716startError: 4717 AudioSystem::stopInput(mId); 4718 clearSyncStartEvent(); 4719 return status; 4720} 4721 4722void AudioFlinger::RecordThread::clearSyncStartEvent() 4723{ 4724 if (mSyncStartEvent != 0) { 4725 mSyncStartEvent->cancel(); 4726 } 4727 mSyncStartEvent.clear(); 4728 mFramestoDrop = 0; 4729} 4730 4731void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4732{ 4733 sp<SyncEvent> strongEvent = event.promote(); 4734 4735 if (strongEvent != 0) { 4736 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4737 me->handleSyncStartEvent(strongEvent); 4738 } 4739} 4740 4741void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4742{ 4743 if (event == mSyncStartEvent) { 4744 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4745 // from audio HAL 4746 mFramestoDrop = mFrameCount * 2; 4747 } 4748} 4749 4750bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4751 ALOGV("RecordThread::stop"); 4752 AutoMutex _l(mLock); 4753 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4754 return false; 4755 } 4756 recordTrack->mState = TrackBase::PAUSING; 4757 // do not wait for mStartStopCond if exiting 4758 if (exitPending()) { 4759 return true; 4760 } 4761 mStartStopCond.wait(mLock); 4762 // if we have been restarted, recordTrack == mActiveTrack.get() here 4763 if (exitPending() || recordTrack != mActiveTrack.get()) { 4764 ALOGV("Record stopped OK"); 4765 return true; 4766 } 4767 return false; 4768} 4769 4770bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4771{ 4772 return false; 4773} 4774 4775status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4776{ 4777#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4778 if (!isValidSyncEvent(event)) { 4779 return BAD_VALUE; 4780 } 4781 4782 int eventSession = event->triggerSession(); 4783 status_t ret = NAME_NOT_FOUND; 4784 4785 Mutex::Autolock _l(mLock); 4786 4787 for (size_t i = 0; i < mTracks.size(); i++) { 4788 sp<RecordTrack> track = mTracks[i]; 4789 if (eventSession == track->sessionId()) { 4790 (void) track->setSyncEvent(event); 4791 ret = NO_ERROR; 4792 } 4793 } 4794 return ret; 4795#else 4796 return BAD_VALUE; 4797#endif 4798} 4799 4800// destroyTrack_l() must be called with ThreadBase::mLock held 4801void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4802{ 4803 track->terminate(); 4804 track->mState = TrackBase::STOPPED; 4805 // active tracks are removed by threadLoop() 4806 if (mActiveTrack != track) { 4807 removeTrack_l(track); 4808 } 4809} 4810 4811void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4812{ 4813 mTracks.remove(track); 4814 // need anything related to effects here? 4815} 4816 4817void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4818{ 4819 dumpInternals(fd, args); 4820 dumpTracks(fd, args); 4821 dumpEffectChains(fd, args); 4822} 4823 4824void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4825{ 4826 const size_t SIZE = 256; 4827 char buffer[SIZE]; 4828 String8 result; 4829 4830 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4831 result.append(buffer); 4832 4833 if (mActiveTrack != 0) { 4834 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4835 result.append(buffer); 4836 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4837 result.append(buffer); 4838 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4839 result.append(buffer); 4840 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4841 result.append(buffer); 4842 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4843 result.append(buffer); 4844 } else { 4845 result.append("No active record client\n"); 4846 } 4847 4848 write(fd, result.string(), result.size()); 4849 4850 dumpBase(fd, args); 4851} 4852 4853void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4854{ 4855 const size_t SIZE = 256; 4856 char buffer[SIZE]; 4857 String8 result; 4858 4859 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4860 result.append(buffer); 4861 RecordTrack::appendDumpHeader(result); 4862 for (size_t i = 0; i < mTracks.size(); ++i) { 4863 sp<RecordTrack> track = mTracks[i]; 4864 if (track != 0) { 4865 track->dump(buffer, SIZE); 4866 result.append(buffer); 4867 } 4868 } 4869 4870 if (mActiveTrack != 0) { 4871 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4872 result.append(buffer); 4873 RecordTrack::appendDumpHeader(result); 4874 mActiveTrack->dump(buffer, SIZE); 4875 result.append(buffer); 4876 4877 } 4878 write(fd, result.string(), result.size()); 4879} 4880 4881// AudioBufferProvider interface 4882status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4883{ 4884 size_t framesReq = buffer->frameCount; 4885 size_t framesReady = mFrameCount - mRsmpInIndex; 4886 int channelCount; 4887 4888 if (framesReady == 0) { 4889 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4890 if (mBytesRead <= 0) { 4891 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4892 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4893 // Force input into standby so that it tries to 4894 // recover at next read attempt 4895 inputStandBy(); 4896 usleep(kRecordThreadSleepUs); 4897 } 4898 buffer->raw = NULL; 4899 buffer->frameCount = 0; 4900 return NOT_ENOUGH_DATA; 4901 } 4902 mRsmpInIndex = 0; 4903 framesReady = mFrameCount; 4904 } 4905 4906 if (framesReq > framesReady) { 4907 framesReq = framesReady; 4908 } 4909 4910 if (mChannelCount == 1 && mReqChannelCount == 2) { 4911 channelCount = 1; 4912 } else { 4913 channelCount = 2; 4914 } 4915 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4916 buffer->frameCount = framesReq; 4917 return NO_ERROR; 4918} 4919 4920// AudioBufferProvider interface 4921void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4922{ 4923 mRsmpInIndex += buffer->frameCount; 4924 buffer->frameCount = 0; 4925} 4926 4927bool AudioFlinger::RecordThread::checkForNewParameters_l() 4928{ 4929 bool reconfig = false; 4930 4931 while (!mNewParameters.isEmpty()) { 4932 status_t status = NO_ERROR; 4933 String8 keyValuePair = mNewParameters[0]; 4934 AudioParameter param = AudioParameter(keyValuePair); 4935 int value; 4936 audio_format_t reqFormat = mFormat; 4937 uint32_t reqSamplingRate = mReqSampleRate; 4938 uint32_t reqChannelCount = mReqChannelCount; 4939 4940 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4941 reqSamplingRate = value; 4942 reconfig = true; 4943 } 4944 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4945 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4946 status = BAD_VALUE; 4947 } else { 4948 reqFormat = (audio_format_t) value; 4949 reconfig = true; 4950 } 4951 } 4952 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4953 reqChannelCount = popcount(value); 4954 reconfig = true; 4955 } 4956 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4957 // do not accept frame count changes if tracks are open as the track buffer 4958 // size depends on frame count and correct behavior would not be guaranteed 4959 // if frame count is changed after track creation 4960 if (mActiveTrack != 0) { 4961 status = INVALID_OPERATION; 4962 } else { 4963 reconfig = true; 4964 } 4965 } 4966 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4967 // forward device change to effects that have requested to be 4968 // aware of attached audio device. 4969 for (size_t i = 0; i < mEffectChains.size(); i++) { 4970 mEffectChains[i]->setDevice_l(value); 4971 } 4972 4973 // store input device and output device but do not forward output device to audio HAL. 4974 // Note that status is ignored by the caller for output device 4975 // (see AudioFlinger::setParameters() 4976 if (audio_is_output_devices(value)) { 4977 mOutDevice = value; 4978 status = BAD_VALUE; 4979 } else { 4980 mInDevice = value; 4981 // disable AEC and NS if the device is a BT SCO headset supporting those 4982 // pre processings 4983 if (mTracks.size() > 0) { 4984 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4985 mAudioFlinger->btNrecIsOff(); 4986 for (size_t i = 0; i < mTracks.size(); i++) { 4987 sp<RecordTrack> track = mTracks[i]; 4988 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4989 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4990 } 4991 } 4992 } 4993 } 4994 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4995 mAudioSource != (audio_source_t)value) { 4996 // forward device change to effects that have requested to be 4997 // aware of attached audio device. 4998 for (size_t i = 0; i < mEffectChains.size(); i++) { 4999 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5000 } 5001 mAudioSource = (audio_source_t)value; 5002 } 5003 if (status == NO_ERROR) { 5004 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5005 keyValuePair.string()); 5006 if (status == INVALID_OPERATION) { 5007 inputStandBy(); 5008 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5009 keyValuePair.string()); 5010 } 5011 if (reconfig) { 5012 if (status == BAD_VALUE && 5013 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5014 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5015 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5016 <= (2 * reqSamplingRate)) && 5017 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5018 <= FCC_2 && 5019 (reqChannelCount <= FCC_2)) { 5020 status = NO_ERROR; 5021 } 5022 if (status == NO_ERROR) { 5023 readInputParameters(); 5024 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5025 } 5026 } 5027 } 5028 5029 mNewParameters.removeAt(0); 5030 5031 mParamStatus = status; 5032 mParamCond.signal(); 5033 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5034 // already timed out waiting for the status and will never signal the condition. 5035 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5036 } 5037 return reconfig; 5038} 5039 5040String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5041{ 5042 Mutex::Autolock _l(mLock); 5043 if (initCheck() != NO_ERROR) { 5044 return String8(); 5045 } 5046 5047 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5048 const String8 out_s8(s); 5049 free(s); 5050 return out_s8; 5051} 5052 5053void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5054 AudioSystem::OutputDescriptor desc; 5055 void *param2 = NULL; 5056 5057 switch (event) { 5058 case AudioSystem::INPUT_OPENED: 5059 case AudioSystem::INPUT_CONFIG_CHANGED: 5060 desc.channelMask = mChannelMask; 5061 desc.samplingRate = mSampleRate; 5062 desc.format = mFormat; 5063 desc.frameCount = mFrameCount; 5064 desc.latency = 0; 5065 param2 = &desc; 5066 break; 5067 5068 case AudioSystem::INPUT_CLOSED: 5069 default: 5070 break; 5071 } 5072 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5073} 5074 5075void AudioFlinger::RecordThread::readInputParameters() 5076{ 5077 delete[] mRsmpInBuffer; 5078 // mRsmpInBuffer is always assigned a new[] below 5079 delete[] mRsmpOutBuffer; 5080 mRsmpOutBuffer = NULL; 5081 delete mResampler; 5082 mResampler = NULL; 5083 5084 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5085 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5086 mChannelCount = popcount(mChannelMask); 5087 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5088 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5089 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5090 } 5091 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5092 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5093 mFrameCount = mBufferSize / mFrameSize; 5094 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5095 5096 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5097 { 5098 int channelCount; 5099 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5100 // stereo to mono post process as the resampler always outputs stereo. 5101 if (mChannelCount == 1 && mReqChannelCount == 2) { 5102 channelCount = 1; 5103 } else { 5104 channelCount = 2; 5105 } 5106 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5107 mResampler->setSampleRate(mSampleRate); 5108 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5109 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5110 5111 // optmization: if mono to mono, alter input frame count as if we were inputing 5112 // stereo samples 5113 if (mChannelCount == 1 && mReqChannelCount == 1) { 5114 mFrameCount >>= 1; 5115 } 5116 5117 } 5118 mRsmpInIndex = mFrameCount; 5119} 5120 5121unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5122{ 5123 Mutex::Autolock _l(mLock); 5124 if (initCheck() != NO_ERROR) { 5125 return 0; 5126 } 5127 5128 return mInput->stream->get_input_frames_lost(mInput->stream); 5129} 5130 5131uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5132{ 5133 Mutex::Autolock _l(mLock); 5134 uint32_t result = 0; 5135 if (getEffectChain_l(sessionId) != 0) { 5136 result = EFFECT_SESSION; 5137 } 5138 5139 for (size_t i = 0; i < mTracks.size(); ++i) { 5140 if (sessionId == mTracks[i]->sessionId()) { 5141 result |= TRACK_SESSION; 5142 break; 5143 } 5144 } 5145 5146 return result; 5147} 5148 5149KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5150{ 5151 KeyedVector<int, bool> ids; 5152 Mutex::Autolock _l(mLock); 5153 for (size_t j = 0; j < mTracks.size(); ++j) { 5154 sp<RecordThread::RecordTrack> track = mTracks[j]; 5155 int sessionId = track->sessionId(); 5156 if (ids.indexOfKey(sessionId) < 0) { 5157 ids.add(sessionId, true); 5158 } 5159 } 5160 return ids; 5161} 5162 5163AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5164{ 5165 Mutex::Autolock _l(mLock); 5166 AudioStreamIn *input = mInput; 5167 mInput = NULL; 5168 return input; 5169} 5170 5171// this method must always be called either with ThreadBase mLock held or inside the thread loop 5172audio_stream_t* AudioFlinger::RecordThread::stream() const 5173{ 5174 if (mInput == NULL) { 5175 return NULL; 5176 } 5177 return &mInput->stream->common; 5178} 5179 5180status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5181{ 5182 // only one chain per input thread 5183 if (mEffectChains.size() != 0) { 5184 return INVALID_OPERATION; 5185 } 5186 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5187 5188 chain->setInBuffer(NULL); 5189 chain->setOutBuffer(NULL); 5190 5191 checkSuspendOnAddEffectChain_l(chain); 5192 5193 mEffectChains.add(chain); 5194 5195 return NO_ERROR; 5196} 5197 5198size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5199{ 5200 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5201 ALOGW_IF(mEffectChains.size() != 1, 5202 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5203 chain.get(), mEffectChains.size(), this); 5204 if (mEffectChains.size() == 1) { 5205 mEffectChains.removeAt(0); 5206 } 5207 return 0; 5208} 5209 5210}; // namespace android 5211