Threads.cpp revision 972a173d7d1de1a3b5a617aae3e2abb6e05ae02d
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 // Do not allow effects with session ID 0 on direct output or duplicating threads 706 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 707 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 708 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 709 desc->name, sessionId); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 // Only Pre processor effects are allowed on input threads and only on input threads 714 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 715 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 716 desc->name, desc->flags, mType); 717 lStatus = BAD_VALUE; 718 goto Exit; 719 } 720 721 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 722 723 { // scope for mLock 724 Mutex::Autolock _l(mLock); 725 726 // check for existing effect chain with the requested audio session 727 chain = getEffectChain_l(sessionId); 728 if (chain == 0) { 729 // create a new chain for this session 730 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 731 chain = new EffectChain(this, sessionId); 732 addEffectChain_l(chain); 733 chain->setStrategy(getStrategyForSession_l(sessionId)); 734 chainCreated = true; 735 } else { 736 effect = chain->getEffectFromDesc_l(desc); 737 } 738 739 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 740 741 if (effect == 0) { 742 int id = mAudioFlinger->nextUniqueId(); 743 // Check CPU and memory usage 744 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 effectRegistered = true; 749 // create a new effect module if none present in the chain 750 effect = new EffectModule(this, chain, desc, id, sessionId); 751 lStatus = effect->status(); 752 if (lStatus != NO_ERROR) { 753 goto Exit; 754 } 755 lStatus = chain->addEffect_l(effect); 756 if (lStatus != NO_ERROR) { 757 goto Exit; 758 } 759 effectCreated = true; 760 761 effect->setDevice(mOutDevice); 762 effect->setDevice(mInDevice); 763 effect->setMode(mAudioFlinger->getMode()); 764 effect->setAudioSource(mAudioSource); 765 } 766 // create effect handle and connect it to effect module 767 handle = new EffectHandle(effect, client, effectClient, priority); 768 lStatus = effect->addHandle(handle.get()); 769 if (enabled != NULL) { 770 *enabled = (int)effect->isEnabled(); 771 } 772 } 773 774Exit: 775 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 776 Mutex::Autolock _l(mLock); 777 if (effectCreated) { 778 chain->removeEffect_l(effect); 779 } 780 if (effectRegistered) { 781 AudioSystem::unregisterEffect(effect->id()); 782 } 783 if (chainCreated) { 784 removeEffectChain_l(chain); 785 } 786 handle.clear(); 787 } 788 789 if (status != NULL) { 790 *status = lStatus; 791 } 792 return handle; 793} 794 795sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 796{ 797 Mutex::Autolock _l(mLock); 798 return getEffect_l(sessionId, effectId); 799} 800 801sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 802{ 803 sp<EffectChain> chain = getEffectChain_l(sessionId); 804 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 805} 806 807// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 808// PlaybackThread::mLock held 809status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 810{ 811 // check for existing effect chain with the requested audio session 812 int sessionId = effect->sessionId(); 813 sp<EffectChain> chain = getEffectChain_l(sessionId); 814 bool chainCreated = false; 815 816 if (chain == 0) { 817 // create a new chain for this session 818 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 819 chain = new EffectChain(this, sessionId); 820 addEffectChain_l(chain); 821 chain->setStrategy(getStrategyForSession_l(sessionId)); 822 chainCreated = true; 823 } 824 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 825 826 if (chain->getEffectFromId_l(effect->id()) != 0) { 827 ALOGW("addEffect_l() %p effect %s already present in chain %p", 828 this, effect->desc().name, chain.get()); 829 return BAD_VALUE; 830 } 831 832 status_t status = chain->addEffect_l(effect); 833 if (status != NO_ERROR) { 834 if (chainCreated) { 835 removeEffectChain_l(chain); 836 } 837 return status; 838 } 839 840 effect->setDevice(mOutDevice); 841 effect->setDevice(mInDevice); 842 effect->setMode(mAudioFlinger->getMode()); 843 effect->setAudioSource(mAudioSource); 844 return NO_ERROR; 845} 846 847void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 848 849 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 850 effect_descriptor_t desc = effect->desc(); 851 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 852 detachAuxEffect_l(effect->id()); 853 } 854 855 sp<EffectChain> chain = effect->chain().promote(); 856 if (chain != 0) { 857 // remove effect chain if removing last effect 858 if (chain->removeEffect_l(effect) == 0) { 859 removeEffectChain_l(chain); 860 } 861 } else { 862 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 863 } 864} 865 866void AudioFlinger::ThreadBase::lockEffectChains_l( 867 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 868{ 869 effectChains = mEffectChains; 870 for (size_t i = 0; i < mEffectChains.size(); i++) { 871 mEffectChains[i]->lock(); 872 } 873} 874 875void AudioFlinger::ThreadBase::unlockEffectChains( 876 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 877{ 878 for (size_t i = 0; i < effectChains.size(); i++) { 879 effectChains[i]->unlock(); 880 } 881} 882 883sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 884{ 885 Mutex::Autolock _l(mLock); 886 return getEffectChain_l(sessionId); 887} 888 889sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 890{ 891 size_t size = mEffectChains.size(); 892 for (size_t i = 0; i < size; i++) { 893 if (mEffectChains[i]->sessionId() == sessionId) { 894 return mEffectChains[i]; 895 } 896 } 897 return 0; 898} 899 900void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 901{ 902 Mutex::Autolock _l(mLock); 903 size_t size = mEffectChains.size(); 904 for (size_t i = 0; i < size; i++) { 905 mEffectChains[i]->setMode_l(mode); 906 } 907} 908 909void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 910 EffectHandle *handle, 911 bool unpinIfLast) { 912 913 Mutex::Autolock _l(mLock); 914 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 915 // delete the effect module if removing last handle on it 916 if (effect->removeHandle(handle) == 0) { 917 if (!effect->isPinned() || unpinIfLast) { 918 removeEffect_l(effect); 919 AudioSystem::unregisterEffect(effect->id()); 920 } 921 } 922} 923 924// ---------------------------------------------------------------------------- 925// Playback 926// ---------------------------------------------------------------------------- 927 928AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 929 AudioStreamOut* output, 930 audio_io_handle_t id, 931 audio_devices_t device, 932 type_t type) 933 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 934 mNormalFrameCount(0), mMixBuffer(NULL), 935 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 936 // mStreamTypes[] initialized in constructor body 937 mOutput(output), 938 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 939 mMixerStatus(MIXER_IDLE), 940 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 941 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 942 mBytesRemaining(0), 943 mCurrentWriteLength(0), 944 mUseAsyncWrite(false), 945 mWriteAckSequence(0), 946 mDrainSequence(0), 947 mScreenState(AudioFlinger::mScreenState), 948 // index 0 is reserved for normal mixer's submix 949 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 950 // mLatchD, mLatchQ, 951 mLatchDValid(false), mLatchQValid(false) 952{ 953 snprintf(mName, kNameLength, "AudioOut_%X", id); 954 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 955 956 // Assumes constructor is called by AudioFlinger with it's mLock held, but 957 // it would be safer to explicitly pass initial masterVolume/masterMute as 958 // parameter. 959 // 960 // If the HAL we are using has support for master volume or master mute, 961 // then do not attenuate or mute during mixing (just leave the volume at 1.0 962 // and the mute set to false). 963 mMasterVolume = audioFlinger->masterVolume_l(); 964 mMasterMute = audioFlinger->masterMute_l(); 965 if (mOutput && mOutput->audioHwDev) { 966 if (mOutput->audioHwDev->canSetMasterVolume()) { 967 mMasterVolume = 1.0; 968 } 969 970 if (mOutput->audioHwDev->canSetMasterMute()) { 971 mMasterMute = false; 972 } 973 } 974 975 readOutputParameters(); 976 977 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 978 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 979 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 980 stream = (audio_stream_type_t) (stream + 1)) { 981 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 982 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 983 } 984 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 985 // because mAudioFlinger doesn't have one to copy from 986} 987 988AudioFlinger::PlaybackThread::~PlaybackThread() 989{ 990 mAudioFlinger->unregisterWriter(mNBLogWriter); 991 delete [] mAllocMixBuffer; 992} 993 994void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 995{ 996 dumpInternals(fd, args); 997 dumpTracks(fd, args); 998 dumpEffectChains(fd, args); 999} 1000 1001void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1002{ 1003 const size_t SIZE = 256; 1004 char buffer[SIZE]; 1005 String8 result; 1006 1007 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1008 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1009 const stream_type_t *st = &mStreamTypes[i]; 1010 if (i > 0) { 1011 result.appendFormat(", "); 1012 } 1013 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1014 if (st->mute) { 1015 result.append("M"); 1016 } 1017 } 1018 result.append("\n"); 1019 write(fd, result.string(), result.length()); 1020 result.clear(); 1021 1022 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1023 result.append(buffer); 1024 Track::appendDumpHeader(result); 1025 for (size_t i = 0; i < mTracks.size(); ++i) { 1026 sp<Track> track = mTracks[i]; 1027 if (track != 0) { 1028 track->dump(buffer, SIZE); 1029 result.append(buffer); 1030 } 1031 } 1032 1033 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1034 result.append(buffer); 1035 Track::appendDumpHeader(result); 1036 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1037 sp<Track> track = mActiveTracks[i].promote(); 1038 if (track != 0) { 1039 track->dump(buffer, SIZE); 1040 result.append(buffer); 1041 } 1042 } 1043 write(fd, result.string(), result.size()); 1044 1045 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1046 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1047 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1048 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1049} 1050 1051void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1052{ 1053 const size_t SIZE = 256; 1054 char buffer[SIZE]; 1055 String8 result; 1056 1057 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1058 result.append(buffer); 1059 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1060 result.append(buffer); 1061 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1062 ns2ms(systemTime() - mLastWriteTime)); 1063 result.append(buffer); 1064 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1065 result.append(buffer); 1066 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1067 result.append(buffer); 1068 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1069 result.append(buffer); 1070 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1071 result.append(buffer); 1072 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1073 result.append(buffer); 1074 write(fd, result.string(), result.size()); 1075 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1076 1077 dumpBase(fd, args); 1078} 1079 1080// Thread virtuals 1081status_t AudioFlinger::PlaybackThread::readyToRun() 1082{ 1083 status_t status = initCheck(); 1084 if (status == NO_ERROR) { 1085 ALOGI("AudioFlinger's thread %p ready to run", this); 1086 } else { 1087 ALOGE("No working audio driver found."); 1088 } 1089 return status; 1090} 1091 1092void AudioFlinger::PlaybackThread::onFirstRef() 1093{ 1094 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1095} 1096 1097// ThreadBase virtuals 1098void AudioFlinger::PlaybackThread::preExit() 1099{ 1100 ALOGV(" preExit()"); 1101 // FIXME this is using hard-coded strings but in the future, this functionality will be 1102 // converted to use audio HAL extensions required to support tunneling 1103 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1104} 1105 1106// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1107sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1108 const sp<AudioFlinger::Client>& client, 1109 audio_stream_type_t streamType, 1110 uint32_t sampleRate, 1111 audio_format_t format, 1112 audio_channel_mask_t channelMask, 1113 size_t frameCount, 1114 const sp<IMemory>& sharedBuffer, 1115 int sessionId, 1116 IAudioFlinger::track_flags_t *flags, 1117 pid_t tid, 1118 status_t *status) 1119{ 1120 sp<Track> track; 1121 status_t lStatus; 1122 1123 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1124 1125 // client expresses a preference for FAST, but we get the final say 1126 if (*flags & IAudioFlinger::TRACK_FAST) { 1127 if ( 1128 // not timed 1129 (!isTimed) && 1130 // either of these use cases: 1131 ( 1132 // use case 1: shared buffer with any frame count 1133 ( 1134 (sharedBuffer != 0) 1135 ) || 1136 // use case 2: callback handler and frame count is default or at least as large as HAL 1137 ( 1138 (tid != -1) && 1139 ((frameCount == 0) || 1140 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1141 ) 1142 ) && 1143 // PCM data 1144 audio_is_linear_pcm(format) && 1145 // mono or stereo 1146 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1147 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1148#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1149 // hardware sample rate 1150 (sampleRate == mSampleRate) && 1151#endif 1152 // normal mixer has an associated fast mixer 1153 hasFastMixer() && 1154 // there are sufficient fast track slots available 1155 (mFastTrackAvailMask != 0) 1156 // FIXME test that MixerThread for this fast track has a capable output HAL 1157 // FIXME add a permission test also? 1158 ) { 1159 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1160 if (frameCount == 0) { 1161 frameCount = mFrameCount * kFastTrackMultiplier; 1162 } 1163 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1164 frameCount, mFrameCount); 1165 } else { 1166 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1167 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1168 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1169 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1170 audio_is_linear_pcm(format), 1171 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1172 *flags &= ~IAudioFlinger::TRACK_FAST; 1173 // For compatibility with AudioTrack calculation, buffer depth is forced 1174 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1175 // This is probably too conservative, but legacy application code may depend on it. 1176 // If you change this calculation, also review the start threshold which is related. 1177 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1178 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1179 if (minBufCount < 2) { 1180 minBufCount = 2; 1181 } 1182 size_t minFrameCount = mNormalFrameCount * minBufCount; 1183 if (frameCount < minFrameCount) { 1184 frameCount = minFrameCount; 1185 } 1186 } 1187 } 1188 1189 if (mType == DIRECT) { 1190 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1191 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1192 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1193 "for output %p with format %d", 1194 sampleRate, format, channelMask, mOutput, mFormat); 1195 lStatus = BAD_VALUE; 1196 goto Exit; 1197 } 1198 } 1199 } else if (mType == OFFLOAD) { 1200 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1201 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1202 "for output %p with format %d", 1203 sampleRate, format, channelMask, mOutput, mFormat); 1204 lStatus = BAD_VALUE; 1205 goto Exit; 1206 } 1207 } else { 1208 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1209 ALOGE("createTrack_l() Bad parameter: format %d \"" 1210 "for output %p with format %d", 1211 format, mOutput, mFormat); 1212 lStatus = BAD_VALUE; 1213 goto Exit; 1214 } 1215 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1216 if (sampleRate > mSampleRate*2) { 1217 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1218 lStatus = BAD_VALUE; 1219 goto Exit; 1220 } 1221 } 1222 1223 lStatus = initCheck(); 1224 if (lStatus != NO_ERROR) { 1225 ALOGE("Audio driver not initialized."); 1226 goto Exit; 1227 } 1228 1229 { // scope for mLock 1230 Mutex::Autolock _l(mLock); 1231 1232 // all tracks in same audio session must share the same routing strategy otherwise 1233 // conflicts will happen when tracks are moved from one output to another by audio policy 1234 // manager 1235 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1236 for (size_t i = 0; i < mTracks.size(); ++i) { 1237 sp<Track> t = mTracks[i]; 1238 if (t != 0 && !t->isOutputTrack()) { 1239 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1240 if (sessionId == t->sessionId() && strategy != actual) { 1241 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1242 strategy, actual); 1243 lStatus = BAD_VALUE; 1244 goto Exit; 1245 } 1246 } 1247 } 1248 1249 if (!isTimed) { 1250 track = new Track(this, client, streamType, sampleRate, format, 1251 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1252 } else { 1253 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1254 channelMask, frameCount, sharedBuffer, sessionId); 1255 } 1256 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1257 lStatus = NO_MEMORY; 1258 goto Exit; 1259 } 1260 1261 mTracks.add(track); 1262 1263 sp<EffectChain> chain = getEffectChain_l(sessionId); 1264 if (chain != 0) { 1265 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1266 track->setMainBuffer(chain->inBuffer()); 1267 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1268 chain->incTrackCnt(); 1269 } 1270 1271 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1272 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1273 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1274 // so ask activity manager to do this on our behalf 1275 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1276 } 1277 } 1278 1279 lStatus = NO_ERROR; 1280 1281Exit: 1282 if (status) { 1283 *status = lStatus; 1284 } 1285 return track; 1286} 1287 1288uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1289{ 1290 return latency; 1291} 1292 1293uint32_t AudioFlinger::PlaybackThread::latency() const 1294{ 1295 Mutex::Autolock _l(mLock); 1296 return latency_l(); 1297} 1298uint32_t AudioFlinger::PlaybackThread::latency_l() const 1299{ 1300 if (initCheck() == NO_ERROR) { 1301 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1302 } else { 1303 return 0; 1304 } 1305} 1306 1307void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1308{ 1309 Mutex::Autolock _l(mLock); 1310 // Don't apply master volume in SW if our HAL can do it for us. 1311 if (mOutput && mOutput->audioHwDev && 1312 mOutput->audioHwDev->canSetMasterVolume()) { 1313 mMasterVolume = 1.0; 1314 } else { 1315 mMasterVolume = value; 1316 } 1317} 1318 1319void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1320{ 1321 Mutex::Autolock _l(mLock); 1322 // Don't apply master mute in SW if our HAL can do it for us. 1323 if (mOutput && mOutput->audioHwDev && 1324 mOutput->audioHwDev->canSetMasterMute()) { 1325 mMasterMute = false; 1326 } else { 1327 mMasterMute = muted; 1328 } 1329} 1330 1331void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1332{ 1333 Mutex::Autolock _l(mLock); 1334 mStreamTypes[stream].volume = value; 1335 signal_l(); 1336} 1337 1338void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1339{ 1340 Mutex::Autolock _l(mLock); 1341 mStreamTypes[stream].mute = muted; 1342 signal_l(); 1343} 1344 1345float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1346{ 1347 Mutex::Autolock _l(mLock); 1348 return mStreamTypes[stream].volume; 1349} 1350 1351// addTrack_l() must be called with ThreadBase::mLock held 1352status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1353{ 1354 status_t status = ALREADY_EXISTS; 1355 1356 // set retry count for buffer fill 1357 track->mRetryCount = kMaxTrackStartupRetries; 1358 if (mActiveTracks.indexOf(track) < 0) { 1359 // the track is newly added, make sure it fills up all its 1360 // buffers before playing. This is to ensure the client will 1361 // effectively get the latency it requested. 1362 if (!track->isOutputTrack()) { 1363 TrackBase::track_state state = track->mState; 1364 mLock.unlock(); 1365 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1366 mLock.lock(); 1367 // abort track was stopped/paused while we released the lock 1368 if (state != track->mState) { 1369 if (status == NO_ERROR) { 1370 mLock.unlock(); 1371 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1372 mLock.lock(); 1373 } 1374 return INVALID_OPERATION; 1375 } 1376 // abort if start is rejected by audio policy manager 1377 if (status != NO_ERROR) { 1378 return PERMISSION_DENIED; 1379 } 1380#ifdef ADD_BATTERY_DATA 1381 // to track the speaker usage 1382 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1383#endif 1384 } 1385 1386 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1387 track->mResetDone = false; 1388 track->mPresentationCompleteFrames = 0; 1389 mActiveTracks.add(track); 1390 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1391 if (chain != 0) { 1392 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1393 track->sessionId()); 1394 chain->incActiveTrackCnt(); 1395 } 1396 1397 status = NO_ERROR; 1398 } 1399 1400 ALOGV("mWaitWorkCV.broadcast"); 1401 mWaitWorkCV.broadcast(); 1402 1403 return status; 1404} 1405 1406bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1407{ 1408 track->terminate(); 1409 // active tracks are removed by threadLoop() 1410 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1411 track->mState = TrackBase::STOPPED; 1412 if (!trackActive) { 1413 removeTrack_l(track); 1414 } else if (track->isFastTrack() || track->isOffloaded()) { 1415 track->mState = TrackBase::STOPPING_1; 1416 } 1417 1418 return trackActive; 1419} 1420 1421void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1422{ 1423 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1424 mTracks.remove(track); 1425 deleteTrackName_l(track->name()); 1426 // redundant as track is about to be destroyed, for dumpsys only 1427 track->mName = -1; 1428 if (track->isFastTrack()) { 1429 int index = track->mFastIndex; 1430 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1431 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1432 mFastTrackAvailMask |= 1 << index; 1433 // redundant as track is about to be destroyed, for dumpsys only 1434 track->mFastIndex = -1; 1435 } 1436 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1437 if (chain != 0) { 1438 chain->decTrackCnt(); 1439 } 1440} 1441 1442void AudioFlinger::PlaybackThread::signal_l() 1443{ 1444 // Thread could be blocked waiting for async 1445 // so signal it to handle state changes immediately 1446 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1447 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1448 mSignalPending = true; 1449 mWaitWorkCV.signal(); 1450} 1451 1452String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1453{ 1454 Mutex::Autolock _l(mLock); 1455 if (initCheck() != NO_ERROR) { 1456 return String8(); 1457 } 1458 1459 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1460 const String8 out_s8(s); 1461 free(s); 1462 return out_s8; 1463} 1464 1465// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1466void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1467 AudioSystem::OutputDescriptor desc; 1468 void *param2 = NULL; 1469 1470 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1471 param); 1472 1473 switch (event) { 1474 case AudioSystem::OUTPUT_OPENED: 1475 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1476 desc.channelMask = mChannelMask; 1477 desc.samplingRate = mSampleRate; 1478 desc.format = mFormat; 1479 desc.frameCount = mNormalFrameCount; // FIXME see 1480 // AudioFlinger::frameCount(audio_io_handle_t) 1481 desc.latency = latency(); 1482 param2 = &desc; 1483 break; 1484 1485 case AudioSystem::STREAM_CONFIG_CHANGED: 1486 param2 = ¶m; 1487 case AudioSystem::OUTPUT_CLOSED: 1488 default: 1489 break; 1490 } 1491 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1492} 1493 1494void AudioFlinger::PlaybackThread::writeCallback() 1495{ 1496 ALOG_ASSERT(mCallbackThread != 0); 1497 mCallbackThread->resetWriteBlocked(); 1498} 1499 1500void AudioFlinger::PlaybackThread::drainCallback() 1501{ 1502 ALOG_ASSERT(mCallbackThread != 0); 1503 mCallbackThread->resetDraining(); 1504} 1505 1506void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1507{ 1508 Mutex::Autolock _l(mLock); 1509 // reject out of sequence requests 1510 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1511 mWriteAckSequence &= ~1; 1512 mWaitWorkCV.signal(); 1513 } 1514} 1515 1516void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1517{ 1518 Mutex::Autolock _l(mLock); 1519 // reject out of sequence requests 1520 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1521 mDrainSequence &= ~1; 1522 mWaitWorkCV.signal(); 1523 } 1524} 1525 1526// static 1527int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1528 void *param, 1529 void *cookie) 1530{ 1531 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1532 ALOGV("asyncCallback() event %d", event); 1533 switch (event) { 1534 case STREAM_CBK_EVENT_WRITE_READY: 1535 me->writeCallback(); 1536 break; 1537 case STREAM_CBK_EVENT_DRAIN_READY: 1538 me->drainCallback(); 1539 break; 1540 default: 1541 ALOGW("asyncCallback() unknown event %d", event); 1542 break; 1543 } 1544 return 0; 1545} 1546 1547void AudioFlinger::PlaybackThread::readOutputParameters() 1548{ 1549 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1550 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1551 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1552 if (!audio_is_output_channel(mChannelMask)) { 1553 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1554 } 1555 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1556 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1557 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1558 } 1559 mChannelCount = popcount(mChannelMask); 1560 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1561 if (!audio_is_valid_format(mFormat)) { 1562 LOG_FATAL("HAL format %d not valid for output", mFormat); 1563 } 1564 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1565 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1566 mFormat); 1567 } 1568 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1569 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1570 if (mFrameCount & 15) { 1571 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1572 mFrameCount); 1573 } 1574 1575 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1576 (mOutput->stream->set_callback != NULL)) { 1577 if (mOutput->stream->set_callback(mOutput->stream, 1578 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1579 mUseAsyncWrite = true; 1580 } 1581 } 1582 1583 // Calculate size of normal mix buffer relative to the HAL output buffer size 1584 double multiplier = 1.0; 1585 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1586 kUseFastMixer == FastMixer_Dynamic)) { 1587 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1588 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1589 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1590 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1591 maxNormalFrameCount = maxNormalFrameCount & ~15; 1592 if (maxNormalFrameCount < minNormalFrameCount) { 1593 maxNormalFrameCount = minNormalFrameCount; 1594 } 1595 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1596 if (multiplier <= 1.0) { 1597 multiplier = 1.0; 1598 } else if (multiplier <= 2.0) { 1599 if (2 * mFrameCount <= maxNormalFrameCount) { 1600 multiplier = 2.0; 1601 } else { 1602 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1603 } 1604 } else { 1605 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1606 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1607 // track, but we sometimes have to do this to satisfy the maximum frame count 1608 // constraint) 1609 // FIXME this rounding up should not be done if no HAL SRC 1610 uint32_t truncMult = (uint32_t) multiplier; 1611 if ((truncMult & 1)) { 1612 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1613 ++truncMult; 1614 } 1615 } 1616 multiplier = (double) truncMult; 1617 } 1618 } 1619 mNormalFrameCount = multiplier * mFrameCount; 1620 // round up to nearest 16 frames to satisfy AudioMixer 1621 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1622 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1623 mNormalFrameCount); 1624 1625 delete[] mAllocMixBuffer; 1626 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1627 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1628 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1629 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1630 1631 // force reconfiguration of effect chains and engines to take new buffer size and audio 1632 // parameters into account 1633 // Note that mLock is not held when readOutputParameters() is called from the constructor 1634 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1635 // matter. 1636 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1637 Vector< sp<EffectChain> > effectChains = mEffectChains; 1638 for (size_t i = 0; i < effectChains.size(); i ++) { 1639 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1640 } 1641} 1642 1643 1644status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1645{ 1646 if (halFrames == NULL || dspFrames == NULL) { 1647 return BAD_VALUE; 1648 } 1649 Mutex::Autolock _l(mLock); 1650 if (initCheck() != NO_ERROR) { 1651 return INVALID_OPERATION; 1652 } 1653 size_t framesWritten = mBytesWritten / mFrameSize; 1654 *halFrames = framesWritten; 1655 1656 if (isSuspended()) { 1657 // return an estimation of rendered frames when the output is suspended 1658 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1659 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1660 return NO_ERROR; 1661 } else { 1662 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1663 } 1664} 1665 1666uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1667{ 1668 Mutex::Autolock _l(mLock); 1669 uint32_t result = 0; 1670 if (getEffectChain_l(sessionId) != 0) { 1671 result = EFFECT_SESSION; 1672 } 1673 1674 for (size_t i = 0; i < mTracks.size(); ++i) { 1675 sp<Track> track = mTracks[i]; 1676 if (sessionId == track->sessionId() && !track->isInvalid()) { 1677 result |= TRACK_SESSION; 1678 break; 1679 } 1680 } 1681 1682 return result; 1683} 1684 1685uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1686{ 1687 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1688 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1689 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1690 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1691 } 1692 for (size_t i = 0; i < mTracks.size(); i++) { 1693 sp<Track> track = mTracks[i]; 1694 if (sessionId == track->sessionId() && !track->isInvalid()) { 1695 return AudioSystem::getStrategyForStream(track->streamType()); 1696 } 1697 } 1698 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1699} 1700 1701 1702AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1703{ 1704 Mutex::Autolock _l(mLock); 1705 return mOutput; 1706} 1707 1708AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1709{ 1710 Mutex::Autolock _l(mLock); 1711 AudioStreamOut *output = mOutput; 1712 mOutput = NULL; 1713 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1714 // must push a NULL and wait for ack 1715 mOutputSink.clear(); 1716 mPipeSink.clear(); 1717 mNormalSink.clear(); 1718 return output; 1719} 1720 1721// this method must always be called either with ThreadBase mLock held or inside the thread loop 1722audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1723{ 1724 if (mOutput == NULL) { 1725 return NULL; 1726 } 1727 return &mOutput->stream->common; 1728} 1729 1730uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1731{ 1732 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1733} 1734 1735status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1736{ 1737 if (!isValidSyncEvent(event)) { 1738 return BAD_VALUE; 1739 } 1740 1741 Mutex::Autolock _l(mLock); 1742 1743 for (size_t i = 0; i < mTracks.size(); ++i) { 1744 sp<Track> track = mTracks[i]; 1745 if (event->triggerSession() == track->sessionId()) { 1746 (void) track->setSyncEvent(event); 1747 return NO_ERROR; 1748 } 1749 } 1750 1751 return NAME_NOT_FOUND; 1752} 1753 1754bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1755{ 1756 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1757} 1758 1759void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1760 const Vector< sp<Track> >& tracksToRemove) 1761{ 1762 size_t count = tracksToRemove.size(); 1763 if (count) { 1764 for (size_t i = 0 ; i < count ; i++) { 1765 const sp<Track>& track = tracksToRemove.itemAt(i); 1766 if (!track->isOutputTrack()) { 1767 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1768#ifdef ADD_BATTERY_DATA 1769 // to track the speaker usage 1770 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1771#endif 1772 if (track->isTerminated()) { 1773 AudioSystem::releaseOutput(mId); 1774 } 1775 } 1776 } 1777 } 1778} 1779 1780void AudioFlinger::PlaybackThread::checkSilentMode_l() 1781{ 1782 if (!mMasterMute) { 1783 char value[PROPERTY_VALUE_MAX]; 1784 if (property_get("ro.audio.silent", value, "0") > 0) { 1785 char *endptr; 1786 unsigned long ul = strtoul(value, &endptr, 0); 1787 if (*endptr == '\0' && ul != 0) { 1788 ALOGD("Silence is golden"); 1789 // The setprop command will not allow a property to be changed after 1790 // the first time it is set, so we don't have to worry about un-muting. 1791 setMasterMute_l(true); 1792 } 1793 } 1794 } 1795} 1796 1797// shared by MIXER and DIRECT, overridden by DUPLICATING 1798ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1799{ 1800 // FIXME rewrite to reduce number of system calls 1801 mLastWriteTime = systemTime(); 1802 mInWrite = true; 1803 ssize_t bytesWritten; 1804 1805 // If an NBAIO sink is present, use it to write the normal mixer's submix 1806 if (mNormalSink != 0) { 1807#define mBitShift 2 // FIXME 1808 size_t count = mBytesRemaining >> mBitShift; 1809 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1810 ATRACE_BEGIN("write"); 1811 // update the setpoint when AudioFlinger::mScreenState changes 1812 uint32_t screenState = AudioFlinger::mScreenState; 1813 if (screenState != mScreenState) { 1814 mScreenState = screenState; 1815 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1816 if (pipe != NULL) { 1817 pipe->setAvgFrames((mScreenState & 1) ? 1818 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1819 } 1820 } 1821 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1822 ATRACE_END(); 1823 if (framesWritten > 0) { 1824 bytesWritten = framesWritten << mBitShift; 1825 } else { 1826 bytesWritten = framesWritten; 1827 } 1828 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1829 if (status == NO_ERROR) { 1830 size_t totalFramesWritten = mNormalSink->framesWritten(); 1831 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1832 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1833 mLatchDValid = true; 1834 } 1835 } 1836 // otherwise use the HAL / AudioStreamOut directly 1837 } else { 1838 // Direct output and offload threads 1839 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1840 if (mUseAsyncWrite) { 1841 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1842 mWriteAckSequence += 2; 1843 mWriteAckSequence |= 1; 1844 ALOG_ASSERT(mCallbackThread != 0); 1845 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1846 } 1847 // FIXME We should have an implementation of timestamps for direct output threads. 1848 // They are used e.g for multichannel PCM playback over HDMI. 1849 bytesWritten = mOutput->stream->write(mOutput->stream, 1850 mMixBuffer + offset, mBytesRemaining); 1851 if (mUseAsyncWrite && 1852 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1853 // do not wait for async callback in case of error of full write 1854 mWriteAckSequence &= ~1; 1855 ALOG_ASSERT(mCallbackThread != 0); 1856 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1857 } 1858 } 1859 1860 mNumWrites++; 1861 mInWrite = false; 1862 1863 return bytesWritten; 1864} 1865 1866void AudioFlinger::PlaybackThread::threadLoop_drain() 1867{ 1868 if (mOutput->stream->drain) { 1869 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1870 if (mUseAsyncWrite) { 1871 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1872 mDrainSequence |= 1; 1873 ALOG_ASSERT(mCallbackThread != 0); 1874 mCallbackThread->setDraining(mDrainSequence); 1875 } 1876 mOutput->stream->drain(mOutput->stream, 1877 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1878 : AUDIO_DRAIN_ALL); 1879 } 1880} 1881 1882void AudioFlinger::PlaybackThread::threadLoop_exit() 1883{ 1884 // Default implementation has nothing to do 1885} 1886 1887/* 1888The derived values that are cached: 1889 - mixBufferSize from frame count * frame size 1890 - activeSleepTime from activeSleepTimeUs() 1891 - idleSleepTime from idleSleepTimeUs() 1892 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1893 - maxPeriod from frame count and sample rate (MIXER only) 1894 1895The parameters that affect these derived values are: 1896 - frame count 1897 - frame size 1898 - sample rate 1899 - device type: A2DP or not 1900 - device latency 1901 - format: PCM or not 1902 - active sleep time 1903 - idle sleep time 1904*/ 1905 1906void AudioFlinger::PlaybackThread::cacheParameters_l() 1907{ 1908 mixBufferSize = mNormalFrameCount * mFrameSize; 1909 activeSleepTime = activeSleepTimeUs(); 1910 idleSleepTime = idleSleepTimeUs(); 1911} 1912 1913void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1914{ 1915 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1916 this, streamType, mTracks.size()); 1917 Mutex::Autolock _l(mLock); 1918 1919 size_t size = mTracks.size(); 1920 for (size_t i = 0; i < size; i++) { 1921 sp<Track> t = mTracks[i]; 1922 if (t->streamType() == streamType) { 1923 t->invalidate(); 1924 } 1925 } 1926} 1927 1928status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1929{ 1930 int session = chain->sessionId(); 1931 int16_t *buffer = mMixBuffer; 1932 bool ownsBuffer = false; 1933 1934 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1935 if (session > 0) { 1936 // Only one effect chain can be present in direct output thread and it uses 1937 // the mix buffer as input 1938 if (mType != DIRECT) { 1939 size_t numSamples = mNormalFrameCount * mChannelCount; 1940 buffer = new int16_t[numSamples]; 1941 memset(buffer, 0, numSamples * sizeof(int16_t)); 1942 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1943 ownsBuffer = true; 1944 } 1945 1946 // Attach all tracks with same session ID to this chain. 1947 for (size_t i = 0; i < mTracks.size(); ++i) { 1948 sp<Track> track = mTracks[i]; 1949 if (session == track->sessionId()) { 1950 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1951 buffer); 1952 track->setMainBuffer(buffer); 1953 chain->incTrackCnt(); 1954 } 1955 } 1956 1957 // indicate all active tracks in the chain 1958 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1959 sp<Track> track = mActiveTracks[i].promote(); 1960 if (track == 0) { 1961 continue; 1962 } 1963 if (session == track->sessionId()) { 1964 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1965 chain->incActiveTrackCnt(); 1966 } 1967 } 1968 } 1969 1970 chain->setInBuffer(buffer, ownsBuffer); 1971 chain->setOutBuffer(mMixBuffer); 1972 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1973 // chains list in order to be processed last as it contains output stage effects 1974 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1975 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1976 // after track specific effects and before output stage 1977 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1978 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1979 // Effect chain for other sessions are inserted at beginning of effect 1980 // chains list to be processed before output mix effects. Relative order between other 1981 // sessions is not important 1982 size_t size = mEffectChains.size(); 1983 size_t i = 0; 1984 for (i = 0; i < size; i++) { 1985 if (mEffectChains[i]->sessionId() < session) { 1986 break; 1987 } 1988 } 1989 mEffectChains.insertAt(chain, i); 1990 checkSuspendOnAddEffectChain_l(chain); 1991 1992 return NO_ERROR; 1993} 1994 1995size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1996{ 1997 int session = chain->sessionId(); 1998 1999 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2000 2001 for (size_t i = 0; i < mEffectChains.size(); i++) { 2002 if (chain == mEffectChains[i]) { 2003 mEffectChains.removeAt(i); 2004 // detach all active tracks from the chain 2005 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2006 sp<Track> track = mActiveTracks[i].promote(); 2007 if (track == 0) { 2008 continue; 2009 } 2010 if (session == track->sessionId()) { 2011 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2012 chain.get(), session); 2013 chain->decActiveTrackCnt(); 2014 } 2015 } 2016 2017 // detach all tracks with same session ID from this chain 2018 for (size_t i = 0; i < mTracks.size(); ++i) { 2019 sp<Track> track = mTracks[i]; 2020 if (session == track->sessionId()) { 2021 track->setMainBuffer(mMixBuffer); 2022 chain->decTrackCnt(); 2023 } 2024 } 2025 break; 2026 } 2027 } 2028 return mEffectChains.size(); 2029} 2030 2031status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2032 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2033{ 2034 Mutex::Autolock _l(mLock); 2035 return attachAuxEffect_l(track, EffectId); 2036} 2037 2038status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2039 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2040{ 2041 status_t status = NO_ERROR; 2042 2043 if (EffectId == 0) { 2044 track->setAuxBuffer(0, NULL); 2045 } else { 2046 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2047 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2048 if (effect != 0) { 2049 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2050 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2051 } else { 2052 status = INVALID_OPERATION; 2053 } 2054 } else { 2055 status = BAD_VALUE; 2056 } 2057 } 2058 return status; 2059} 2060 2061void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2062{ 2063 for (size_t i = 0; i < mTracks.size(); ++i) { 2064 sp<Track> track = mTracks[i]; 2065 if (track->auxEffectId() == effectId) { 2066 attachAuxEffect_l(track, 0); 2067 } 2068 } 2069} 2070 2071bool AudioFlinger::PlaybackThread::threadLoop() 2072{ 2073 Vector< sp<Track> > tracksToRemove; 2074 2075 standbyTime = systemTime(); 2076 2077 // MIXER 2078 nsecs_t lastWarning = 0; 2079 2080 // DUPLICATING 2081 // FIXME could this be made local to while loop? 2082 writeFrames = 0; 2083 2084 cacheParameters_l(); 2085 sleepTime = idleSleepTime; 2086 2087 if (mType == MIXER) { 2088 sleepTimeShift = 0; 2089 } 2090 2091 CpuStats cpuStats; 2092 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2093 2094 acquireWakeLock(); 2095 2096 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2097 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2098 // and then that string will be logged at the next convenient opportunity. 2099 const char *logString = NULL; 2100 2101 while (!exitPending()) 2102 { 2103 cpuStats.sample(myName); 2104 2105 Vector< sp<EffectChain> > effectChains; 2106 2107 processConfigEvents(); 2108 2109 { // scope for mLock 2110 2111 Mutex::Autolock _l(mLock); 2112 2113 if (logString != NULL) { 2114 mNBLogWriter->logTimestamp(); 2115 mNBLogWriter->log(logString); 2116 logString = NULL; 2117 } 2118 2119 if (mLatchDValid) { 2120 mLatchQ = mLatchD; 2121 mLatchDValid = false; 2122 mLatchQValid = true; 2123 } 2124 2125 if (checkForNewParameters_l()) { 2126 cacheParameters_l(); 2127 } 2128 2129 saveOutputTracks(); 2130 2131 if (mSignalPending) { 2132 // A signal was raised while we were unlocked 2133 mSignalPending = false; 2134 } else if (waitingAsyncCallback_l()) { 2135 if (exitPending()) { 2136 break; 2137 } 2138 releaseWakeLock_l(); 2139 ALOGV("wait async completion"); 2140 mWaitWorkCV.wait(mLock); 2141 ALOGV("async completion/wake"); 2142 acquireWakeLock_l(); 2143 standbyTime = systemTime() + standbyDelay; 2144 sleepTime = 0; 2145 if (exitPending()) { 2146 break; 2147 } 2148 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2149 isSuspended()) { 2150 // put audio hardware into standby after short delay 2151 if (shouldStandby_l()) { 2152 2153 threadLoop_standby(); 2154 2155 mStandby = true; 2156 } 2157 2158 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2159 // we're about to wait, flush the binder command buffer 2160 IPCThreadState::self()->flushCommands(); 2161 2162 clearOutputTracks(); 2163 2164 if (exitPending()) { 2165 break; 2166 } 2167 2168 releaseWakeLock_l(); 2169 // wait until we have something to do... 2170 ALOGV("%s going to sleep", myName.string()); 2171 mWaitWorkCV.wait(mLock); 2172 ALOGV("%s waking up", myName.string()); 2173 acquireWakeLock_l(); 2174 2175 mMixerStatus = MIXER_IDLE; 2176 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2177 mBytesWritten = 0; 2178 mBytesRemaining = 0; 2179 checkSilentMode_l(); 2180 2181 standbyTime = systemTime() + standbyDelay; 2182 sleepTime = idleSleepTime; 2183 if (mType == MIXER) { 2184 sleepTimeShift = 0; 2185 } 2186 2187 continue; 2188 } 2189 } 2190 2191 // mMixerStatusIgnoringFastTracks is also updated internally 2192 mMixerStatus = prepareTracks_l(&tracksToRemove); 2193 2194 // prevent any changes in effect chain list and in each effect chain 2195 // during mixing and effect process as the audio buffers could be deleted 2196 // or modified if an effect is created or deleted 2197 lockEffectChains_l(effectChains); 2198 } 2199 2200 if (mBytesRemaining == 0) { 2201 mCurrentWriteLength = 0; 2202 if (mMixerStatus == MIXER_TRACKS_READY) { 2203 // threadLoop_mix() sets mCurrentWriteLength 2204 threadLoop_mix(); 2205 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2206 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2207 // threadLoop_sleepTime sets sleepTime to 0 if data 2208 // must be written to HAL 2209 threadLoop_sleepTime(); 2210 if (sleepTime == 0) { 2211 mCurrentWriteLength = mixBufferSize; 2212 } 2213 } 2214 mBytesRemaining = mCurrentWriteLength; 2215 if (isSuspended()) { 2216 sleepTime = suspendSleepTimeUs(); 2217 // simulate write to HAL when suspended 2218 mBytesWritten += mixBufferSize; 2219 mBytesRemaining = 0; 2220 } 2221 2222 // only process effects if we're going to write 2223 if (sleepTime == 0) { 2224 for (size_t i = 0; i < effectChains.size(); i ++) { 2225 effectChains[i]->process_l(); 2226 } 2227 } 2228 } 2229 2230 // enable changes in effect chain 2231 unlockEffectChains(effectChains); 2232 2233 if (!waitingAsyncCallback()) { 2234 // sleepTime == 0 means we must write to audio hardware 2235 if (sleepTime == 0) { 2236 if (mBytesRemaining) { 2237 ssize_t ret = threadLoop_write(); 2238 if (ret < 0) { 2239 mBytesRemaining = 0; 2240 } else { 2241 mBytesWritten += ret; 2242 mBytesRemaining -= ret; 2243 } 2244 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2245 (mMixerStatus == MIXER_DRAIN_ALL)) { 2246 threadLoop_drain(); 2247 } 2248if (mType == MIXER) { 2249 // write blocked detection 2250 nsecs_t now = systemTime(); 2251 nsecs_t delta = now - mLastWriteTime; 2252 if (!mStandby && delta > maxPeriod) { 2253 mNumDelayedWrites++; 2254 if ((now - lastWarning) > kWarningThrottleNs) { 2255 ATRACE_NAME("underrun"); 2256 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2257 ns2ms(delta), mNumDelayedWrites, this); 2258 lastWarning = now; 2259 } 2260 } 2261} 2262 2263 mStandby = false; 2264 } else { 2265 usleep(sleepTime); 2266 } 2267 } 2268 2269 // Finally let go of removed track(s), without the lock held 2270 // since we can't guarantee the destructors won't acquire that 2271 // same lock. This will also mutate and push a new fast mixer state. 2272 threadLoop_removeTracks(tracksToRemove); 2273 tracksToRemove.clear(); 2274 2275 // FIXME I don't understand the need for this here; 2276 // it was in the original code but maybe the 2277 // assignment in saveOutputTracks() makes this unnecessary? 2278 clearOutputTracks(); 2279 2280 // Effect chains will be actually deleted here if they were removed from 2281 // mEffectChains list during mixing or effects processing 2282 effectChains.clear(); 2283 2284 // FIXME Note that the above .clear() is no longer necessary since effectChains 2285 // is now local to this block, but will keep it for now (at least until merge done). 2286 } 2287 2288 threadLoop_exit(); 2289 2290 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2291 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2292 // put output stream into standby mode 2293 if (!mStandby) { 2294 mOutput->stream->common.standby(&mOutput->stream->common); 2295 } 2296 } 2297 2298 releaseWakeLock(); 2299 2300 ALOGV("Thread %p type %d exiting", this, mType); 2301 return false; 2302} 2303 2304// removeTracks_l() must be called with ThreadBase::mLock held 2305void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2306{ 2307 size_t count = tracksToRemove.size(); 2308 if (count) { 2309 for (size_t i=0 ; i<count ; i++) { 2310 const sp<Track>& track = tracksToRemove.itemAt(i); 2311 mActiveTracks.remove(track); 2312 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2313 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2314 if (chain != 0) { 2315 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2316 track->sessionId()); 2317 chain->decActiveTrackCnt(); 2318 } 2319 if (track->isTerminated()) { 2320 removeTrack_l(track); 2321 } 2322 } 2323 } 2324 2325} 2326 2327// ---------------------------------------------------------------------------- 2328 2329AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2330 audio_io_handle_t id, audio_devices_t device, type_t type) 2331 : PlaybackThread(audioFlinger, output, id, device, type), 2332 // mAudioMixer below 2333 // mFastMixer below 2334 mFastMixerFutex(0) 2335 // mOutputSink below 2336 // mPipeSink below 2337 // mNormalSink below 2338{ 2339 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2340 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2341 "mFrameCount=%d, mNormalFrameCount=%d", 2342 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2343 mNormalFrameCount); 2344 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2345 2346 // FIXME - Current mixer implementation only supports stereo output 2347 if (mChannelCount != FCC_2) { 2348 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2349 } 2350 2351 // create an NBAIO sink for the HAL output stream, and negotiate 2352 mOutputSink = new AudioStreamOutSink(output->stream); 2353 size_t numCounterOffers = 0; 2354 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2355 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2356 ALOG_ASSERT(index == 0); 2357 2358 // initialize fast mixer depending on configuration 2359 bool initFastMixer; 2360 switch (kUseFastMixer) { 2361 case FastMixer_Never: 2362 initFastMixer = false; 2363 break; 2364 case FastMixer_Always: 2365 initFastMixer = true; 2366 break; 2367 case FastMixer_Static: 2368 case FastMixer_Dynamic: 2369 initFastMixer = mFrameCount < mNormalFrameCount; 2370 break; 2371 } 2372 if (initFastMixer) { 2373 2374 // create a MonoPipe to connect our submix to FastMixer 2375 NBAIO_Format format = mOutputSink->format(); 2376 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2377 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2378 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2379 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2380 const NBAIO_Format offers[1] = {format}; 2381 size_t numCounterOffers = 0; 2382 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2383 ALOG_ASSERT(index == 0); 2384 monoPipe->setAvgFrames((mScreenState & 1) ? 2385 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2386 mPipeSink = monoPipe; 2387 2388#ifdef TEE_SINK 2389 if (mTeeSinkOutputEnabled) { 2390 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2391 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2392 numCounterOffers = 0; 2393 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2394 ALOG_ASSERT(index == 0); 2395 mTeeSink = teeSink; 2396 PipeReader *teeSource = new PipeReader(*teeSink); 2397 numCounterOffers = 0; 2398 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2399 ALOG_ASSERT(index == 0); 2400 mTeeSource = teeSource; 2401 } 2402#endif 2403 2404 // create fast mixer and configure it initially with just one fast track for our submix 2405 mFastMixer = new FastMixer(); 2406 FastMixerStateQueue *sq = mFastMixer->sq(); 2407#ifdef STATE_QUEUE_DUMP 2408 sq->setObserverDump(&mStateQueueObserverDump); 2409 sq->setMutatorDump(&mStateQueueMutatorDump); 2410#endif 2411 FastMixerState *state = sq->begin(); 2412 FastTrack *fastTrack = &state->mFastTracks[0]; 2413 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2414 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2415 fastTrack->mVolumeProvider = NULL; 2416 fastTrack->mGeneration++; 2417 state->mFastTracksGen++; 2418 state->mTrackMask = 1; 2419 // fast mixer will use the HAL output sink 2420 state->mOutputSink = mOutputSink.get(); 2421 state->mOutputSinkGen++; 2422 state->mFrameCount = mFrameCount; 2423 state->mCommand = FastMixerState::COLD_IDLE; 2424 // already done in constructor initialization list 2425 //mFastMixerFutex = 0; 2426 state->mColdFutexAddr = &mFastMixerFutex; 2427 state->mColdGen++; 2428 state->mDumpState = &mFastMixerDumpState; 2429#ifdef TEE_SINK 2430 state->mTeeSink = mTeeSink.get(); 2431#endif 2432 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2433 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2434 sq->end(); 2435 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2436 2437 // start the fast mixer 2438 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2439 pid_t tid = mFastMixer->getTid(); 2440 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2441 if (err != 0) { 2442 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2443 kPriorityFastMixer, getpid_cached, tid, err); 2444 } 2445 2446#ifdef AUDIO_WATCHDOG 2447 // create and start the watchdog 2448 mAudioWatchdog = new AudioWatchdog(); 2449 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2450 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2451 tid = mAudioWatchdog->getTid(); 2452 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2453 if (err != 0) { 2454 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2455 kPriorityFastMixer, getpid_cached, tid, err); 2456 } 2457#endif 2458 2459 } else { 2460 mFastMixer = NULL; 2461 } 2462 2463 switch (kUseFastMixer) { 2464 case FastMixer_Never: 2465 case FastMixer_Dynamic: 2466 mNormalSink = mOutputSink; 2467 break; 2468 case FastMixer_Always: 2469 mNormalSink = mPipeSink; 2470 break; 2471 case FastMixer_Static: 2472 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2473 break; 2474 } 2475} 2476 2477AudioFlinger::MixerThread::~MixerThread() 2478{ 2479 if (mFastMixer != NULL) { 2480 FastMixerStateQueue *sq = mFastMixer->sq(); 2481 FastMixerState *state = sq->begin(); 2482 if (state->mCommand == FastMixerState::COLD_IDLE) { 2483 int32_t old = android_atomic_inc(&mFastMixerFutex); 2484 if (old == -1) { 2485 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2486 } 2487 } 2488 state->mCommand = FastMixerState::EXIT; 2489 sq->end(); 2490 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2491 mFastMixer->join(); 2492 // Though the fast mixer thread has exited, it's state queue is still valid. 2493 // We'll use that extract the final state which contains one remaining fast track 2494 // corresponding to our sub-mix. 2495 state = sq->begin(); 2496 ALOG_ASSERT(state->mTrackMask == 1); 2497 FastTrack *fastTrack = &state->mFastTracks[0]; 2498 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2499 delete fastTrack->mBufferProvider; 2500 sq->end(false /*didModify*/); 2501 delete mFastMixer; 2502#ifdef AUDIO_WATCHDOG 2503 if (mAudioWatchdog != 0) { 2504 mAudioWatchdog->requestExit(); 2505 mAudioWatchdog->requestExitAndWait(); 2506 mAudioWatchdog.clear(); 2507 } 2508#endif 2509 } 2510 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2511 delete mAudioMixer; 2512} 2513 2514 2515uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2516{ 2517 if (mFastMixer != NULL) { 2518 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2519 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2520 } 2521 return latency; 2522} 2523 2524 2525void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2526{ 2527 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2528} 2529 2530ssize_t AudioFlinger::MixerThread::threadLoop_write() 2531{ 2532 // FIXME we should only do one push per cycle; confirm this is true 2533 // Start the fast mixer if it's not already running 2534 if (mFastMixer != NULL) { 2535 FastMixerStateQueue *sq = mFastMixer->sq(); 2536 FastMixerState *state = sq->begin(); 2537 if (state->mCommand != FastMixerState::MIX_WRITE && 2538 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2539 if (state->mCommand == FastMixerState::COLD_IDLE) { 2540 int32_t old = android_atomic_inc(&mFastMixerFutex); 2541 if (old == -1) { 2542 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2543 } 2544#ifdef AUDIO_WATCHDOG 2545 if (mAudioWatchdog != 0) { 2546 mAudioWatchdog->resume(); 2547 } 2548#endif 2549 } 2550 state->mCommand = FastMixerState::MIX_WRITE; 2551 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2552 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2553 sq->end(); 2554 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2555 if (kUseFastMixer == FastMixer_Dynamic) { 2556 mNormalSink = mPipeSink; 2557 } 2558 } else { 2559 sq->end(false /*didModify*/); 2560 } 2561 } 2562 return PlaybackThread::threadLoop_write(); 2563} 2564 2565void AudioFlinger::MixerThread::threadLoop_standby() 2566{ 2567 // Idle the fast mixer if it's currently running 2568 if (mFastMixer != NULL) { 2569 FastMixerStateQueue *sq = mFastMixer->sq(); 2570 FastMixerState *state = sq->begin(); 2571 if (!(state->mCommand & FastMixerState::IDLE)) { 2572 state->mCommand = FastMixerState::COLD_IDLE; 2573 state->mColdFutexAddr = &mFastMixerFutex; 2574 state->mColdGen++; 2575 mFastMixerFutex = 0; 2576 sq->end(); 2577 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2578 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2579 if (kUseFastMixer == FastMixer_Dynamic) { 2580 mNormalSink = mOutputSink; 2581 } 2582#ifdef AUDIO_WATCHDOG 2583 if (mAudioWatchdog != 0) { 2584 mAudioWatchdog->pause(); 2585 } 2586#endif 2587 } else { 2588 sq->end(false /*didModify*/); 2589 } 2590 } 2591 PlaybackThread::threadLoop_standby(); 2592} 2593 2594// Empty implementation for standard mixer 2595// Overridden for offloaded playback 2596void AudioFlinger::PlaybackThread::flushOutput_l() 2597{ 2598} 2599 2600bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2601{ 2602 return false; 2603} 2604 2605bool AudioFlinger::PlaybackThread::shouldStandby_l() 2606{ 2607 return !mStandby; 2608} 2609 2610bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2611{ 2612 Mutex::Autolock _l(mLock); 2613 return waitingAsyncCallback_l(); 2614} 2615 2616// shared by MIXER and DIRECT, overridden by DUPLICATING 2617void AudioFlinger::PlaybackThread::threadLoop_standby() 2618{ 2619 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2620 mOutput->stream->common.standby(&mOutput->stream->common); 2621 if (mUseAsyncWrite != 0) { 2622 // discard any pending drain or write ack by incrementing sequence 2623 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2624 mDrainSequence = (mDrainSequence + 2) & ~1; 2625 ALOG_ASSERT(mCallbackThread != 0); 2626 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2627 mCallbackThread->setDraining(mDrainSequence); 2628 } 2629} 2630 2631void AudioFlinger::MixerThread::threadLoop_mix() 2632{ 2633 // obtain the presentation timestamp of the next output buffer 2634 int64_t pts; 2635 status_t status = INVALID_OPERATION; 2636 2637 if (mNormalSink != 0) { 2638 status = mNormalSink->getNextWriteTimestamp(&pts); 2639 } else { 2640 status = mOutputSink->getNextWriteTimestamp(&pts); 2641 } 2642 2643 if (status != NO_ERROR) { 2644 pts = AudioBufferProvider::kInvalidPTS; 2645 } 2646 2647 // mix buffers... 2648 mAudioMixer->process(pts); 2649 mCurrentWriteLength = mixBufferSize; 2650 // increase sleep time progressively when application underrun condition clears. 2651 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2652 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2653 // such that we would underrun the audio HAL. 2654 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2655 sleepTimeShift--; 2656 } 2657 sleepTime = 0; 2658 standbyTime = systemTime() + standbyDelay; 2659 //TODO: delay standby when effects have a tail 2660} 2661 2662void AudioFlinger::MixerThread::threadLoop_sleepTime() 2663{ 2664 // If no tracks are ready, sleep once for the duration of an output 2665 // buffer size, then write 0s to the output 2666 if (sleepTime == 0) { 2667 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2668 sleepTime = activeSleepTime >> sleepTimeShift; 2669 if (sleepTime < kMinThreadSleepTimeUs) { 2670 sleepTime = kMinThreadSleepTimeUs; 2671 } 2672 // reduce sleep time in case of consecutive application underruns to avoid 2673 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2674 // duration we would end up writing less data than needed by the audio HAL if 2675 // the condition persists. 2676 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2677 sleepTimeShift++; 2678 } 2679 } else { 2680 sleepTime = idleSleepTime; 2681 } 2682 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2683 memset (mMixBuffer, 0, mixBufferSize); 2684 sleepTime = 0; 2685 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2686 "anticipated start"); 2687 } 2688 // TODO add standby time extension fct of effect tail 2689} 2690 2691// prepareTracks_l() must be called with ThreadBase::mLock held 2692AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2693 Vector< sp<Track> > *tracksToRemove) 2694{ 2695 2696 mixer_state mixerStatus = MIXER_IDLE; 2697 // find out which tracks need to be processed 2698 size_t count = mActiveTracks.size(); 2699 size_t mixedTracks = 0; 2700 size_t tracksWithEffect = 0; 2701 // counts only _active_ fast tracks 2702 size_t fastTracks = 0; 2703 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2704 2705 float masterVolume = mMasterVolume; 2706 bool masterMute = mMasterMute; 2707 2708 if (masterMute) { 2709 masterVolume = 0; 2710 } 2711 // Delegate master volume control to effect in output mix effect chain if needed 2712 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2713 if (chain != 0) { 2714 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2715 chain->setVolume_l(&v, &v); 2716 masterVolume = (float)((v + (1 << 23)) >> 24); 2717 chain.clear(); 2718 } 2719 2720 // prepare a new state to push 2721 FastMixerStateQueue *sq = NULL; 2722 FastMixerState *state = NULL; 2723 bool didModify = false; 2724 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2725 if (mFastMixer != NULL) { 2726 sq = mFastMixer->sq(); 2727 state = sq->begin(); 2728 } 2729 2730 for (size_t i=0 ; i<count ; i++) { 2731 const sp<Track> t = mActiveTracks[i].promote(); 2732 if (t == 0) { 2733 continue; 2734 } 2735 2736 // this const just means the local variable doesn't change 2737 Track* const track = t.get(); 2738 2739 // process fast tracks 2740 if (track->isFastTrack()) { 2741 2742 // It's theoretically possible (though unlikely) for a fast track to be created 2743 // and then removed within the same normal mix cycle. This is not a problem, as 2744 // the track never becomes active so it's fast mixer slot is never touched. 2745 // The converse, of removing an (active) track and then creating a new track 2746 // at the identical fast mixer slot within the same normal mix cycle, 2747 // is impossible because the slot isn't marked available until the end of each cycle. 2748 int j = track->mFastIndex; 2749 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2750 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2751 FastTrack *fastTrack = &state->mFastTracks[j]; 2752 2753 // Determine whether the track is currently in underrun condition, 2754 // and whether it had a recent underrun. 2755 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2756 FastTrackUnderruns underruns = ftDump->mUnderruns; 2757 uint32_t recentFull = (underruns.mBitFields.mFull - 2758 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2759 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2760 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2761 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2762 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2763 uint32_t recentUnderruns = recentPartial + recentEmpty; 2764 track->mObservedUnderruns = underruns; 2765 // don't count underruns that occur while stopping or pausing 2766 // or stopped which can occur when flush() is called while active 2767 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2768 recentUnderruns > 0) { 2769 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2770 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2771 } 2772 2773 // This is similar to the state machine for normal tracks, 2774 // with a few modifications for fast tracks. 2775 bool isActive = true; 2776 switch (track->mState) { 2777 case TrackBase::STOPPING_1: 2778 // track stays active in STOPPING_1 state until first underrun 2779 if (recentUnderruns > 0 || track->isTerminated()) { 2780 track->mState = TrackBase::STOPPING_2; 2781 } 2782 break; 2783 case TrackBase::PAUSING: 2784 // ramp down is not yet implemented 2785 track->setPaused(); 2786 break; 2787 case TrackBase::RESUMING: 2788 // ramp up is not yet implemented 2789 track->mState = TrackBase::ACTIVE; 2790 break; 2791 case TrackBase::ACTIVE: 2792 if (recentFull > 0 || recentPartial > 0) { 2793 // track has provided at least some frames recently: reset retry count 2794 track->mRetryCount = kMaxTrackRetries; 2795 } 2796 if (recentUnderruns == 0) { 2797 // no recent underruns: stay active 2798 break; 2799 } 2800 // there has recently been an underrun of some kind 2801 if (track->sharedBuffer() == 0) { 2802 // were any of the recent underruns "empty" (no frames available)? 2803 if (recentEmpty == 0) { 2804 // no, then ignore the partial underruns as they are allowed indefinitely 2805 break; 2806 } 2807 // there has recently been an "empty" underrun: decrement the retry counter 2808 if (--(track->mRetryCount) > 0) { 2809 break; 2810 } 2811 // indicate to client process that the track was disabled because of underrun; 2812 // it will then automatically call start() when data is available 2813 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2814 // remove from active list, but state remains ACTIVE [confusing but true] 2815 isActive = false; 2816 break; 2817 } 2818 // fall through 2819 case TrackBase::STOPPING_2: 2820 case TrackBase::PAUSED: 2821 case TrackBase::STOPPED: 2822 case TrackBase::FLUSHED: // flush() while active 2823 // Check for presentation complete if track is inactive 2824 // We have consumed all the buffers of this track. 2825 // This would be incomplete if we auto-paused on underrun 2826 { 2827 size_t audioHALFrames = 2828 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2829 size_t framesWritten = mBytesWritten / mFrameSize; 2830 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2831 // track stays in active list until presentation is complete 2832 break; 2833 } 2834 } 2835 if (track->isStopping_2()) { 2836 track->mState = TrackBase::STOPPED; 2837 } 2838 if (track->isStopped()) { 2839 // Can't reset directly, as fast mixer is still polling this track 2840 // track->reset(); 2841 // So instead mark this track as needing to be reset after push with ack 2842 resetMask |= 1 << i; 2843 } 2844 isActive = false; 2845 break; 2846 case TrackBase::IDLE: 2847 default: 2848 LOG_FATAL("unexpected track state %d", track->mState); 2849 } 2850 2851 if (isActive) { 2852 // was it previously inactive? 2853 if (!(state->mTrackMask & (1 << j))) { 2854 ExtendedAudioBufferProvider *eabp = track; 2855 VolumeProvider *vp = track; 2856 fastTrack->mBufferProvider = eabp; 2857 fastTrack->mVolumeProvider = vp; 2858 fastTrack->mSampleRate = track->mSampleRate; 2859 fastTrack->mChannelMask = track->mChannelMask; 2860 fastTrack->mGeneration++; 2861 state->mTrackMask |= 1 << j; 2862 didModify = true; 2863 // no acknowledgement required for newly active tracks 2864 } 2865 // cache the combined master volume and stream type volume for fast mixer; this 2866 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2867 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2868 ++fastTracks; 2869 } else { 2870 // was it previously active? 2871 if (state->mTrackMask & (1 << j)) { 2872 fastTrack->mBufferProvider = NULL; 2873 fastTrack->mGeneration++; 2874 state->mTrackMask &= ~(1 << j); 2875 didModify = true; 2876 // If any fast tracks were removed, we must wait for acknowledgement 2877 // because we're about to decrement the last sp<> on those tracks. 2878 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2879 } else { 2880 LOG_FATAL("fast track %d should have been active", j); 2881 } 2882 tracksToRemove->add(track); 2883 // Avoids a misleading display in dumpsys 2884 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2885 } 2886 continue; 2887 } 2888 2889 { // local variable scope to avoid goto warning 2890 2891 audio_track_cblk_t* cblk = track->cblk(); 2892 2893 // The first time a track is added we wait 2894 // for all its buffers to be filled before processing it 2895 int name = track->name(); 2896 // make sure that we have enough frames to mix one full buffer. 2897 // enforce this condition only once to enable draining the buffer in case the client 2898 // app does not call stop() and relies on underrun to stop: 2899 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2900 // during last round 2901 size_t desiredFrames; 2902 uint32_t sr = track->sampleRate(); 2903 if (sr == mSampleRate) { 2904 desiredFrames = mNormalFrameCount; 2905 } else { 2906 // +1 for rounding and +1 for additional sample needed for interpolation 2907 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2908 // add frames already consumed but not yet released by the resampler 2909 // because cblk->framesReady() will include these frames 2910 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2911 // the minimum track buffer size is normally twice the number of frames necessary 2912 // to fill one buffer and the resampler should not leave more than one buffer worth 2913 // of unreleased frames after each pass, but just in case... 2914 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2915 } 2916 uint32_t minFrames = 1; 2917 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2918 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2919 minFrames = desiredFrames; 2920 } 2921 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2922 size_t framesReady; 2923 if (track->sharedBuffer() == 0) { 2924 framesReady = track->framesReady(); 2925 } else if (track->isStopped()) { 2926 framesReady = 0; 2927 } else { 2928 framesReady = 1; 2929 } 2930 if ((framesReady >= minFrames) && track->isReady() && 2931 !track->isPaused() && !track->isTerminated()) 2932 { 2933 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2934 2935 mixedTracks++; 2936 2937 // track->mainBuffer() != mMixBuffer means there is an effect chain 2938 // connected to the track 2939 chain.clear(); 2940 if (track->mainBuffer() != mMixBuffer) { 2941 chain = getEffectChain_l(track->sessionId()); 2942 // Delegate volume control to effect in track effect chain if needed 2943 if (chain != 0) { 2944 tracksWithEffect++; 2945 } else { 2946 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2947 "session %d", 2948 name, track->sessionId()); 2949 } 2950 } 2951 2952 2953 int param = AudioMixer::VOLUME; 2954 if (track->mFillingUpStatus == Track::FS_FILLED) { 2955 // no ramp for the first volume setting 2956 track->mFillingUpStatus = Track::FS_ACTIVE; 2957 if (track->mState == TrackBase::RESUMING) { 2958 track->mState = TrackBase::ACTIVE; 2959 param = AudioMixer::RAMP_VOLUME; 2960 } 2961 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2962 // FIXME should not make a decision based on mServer 2963 } else if (cblk->mServer != 0) { 2964 // If the track is stopped before the first frame was mixed, 2965 // do not apply ramp 2966 param = AudioMixer::RAMP_VOLUME; 2967 } 2968 2969 // compute volume for this track 2970 uint32_t vl, vr, va; 2971 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2972 vl = vr = va = 0; 2973 if (track->isPausing()) { 2974 track->setPaused(); 2975 } 2976 } else { 2977 2978 // read original volumes with volume control 2979 float typeVolume = mStreamTypes[track->streamType()].volume; 2980 float v = masterVolume * typeVolume; 2981 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2982 uint32_t vlr = proxy->getVolumeLR(); 2983 vl = vlr & 0xFFFF; 2984 vr = vlr >> 16; 2985 // track volumes come from shared memory, so can't be trusted and must be clamped 2986 if (vl > MAX_GAIN_INT) { 2987 ALOGV("Track left volume out of range: %04X", vl); 2988 vl = MAX_GAIN_INT; 2989 } 2990 if (vr > MAX_GAIN_INT) { 2991 ALOGV("Track right volume out of range: %04X", vr); 2992 vr = MAX_GAIN_INT; 2993 } 2994 // now apply the master volume and stream type volume 2995 vl = (uint32_t)(v * vl) << 12; 2996 vr = (uint32_t)(v * vr) << 12; 2997 // assuming master volume and stream type volume each go up to 1.0, 2998 // vl and vr are now in 8.24 format 2999 3000 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3001 // send level comes from shared memory and so may be corrupt 3002 if (sendLevel > MAX_GAIN_INT) { 3003 ALOGV("Track send level out of range: %04X", sendLevel); 3004 sendLevel = MAX_GAIN_INT; 3005 } 3006 va = (uint32_t)(v * sendLevel); 3007 } 3008 3009 // Delegate volume control to effect in track effect chain if needed 3010 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3011 // Do not ramp volume if volume is controlled by effect 3012 param = AudioMixer::VOLUME; 3013 track->mHasVolumeController = true; 3014 } else { 3015 // force no volume ramp when volume controller was just disabled or removed 3016 // from effect chain to avoid volume spike 3017 if (track->mHasVolumeController) { 3018 param = AudioMixer::VOLUME; 3019 } 3020 track->mHasVolumeController = false; 3021 } 3022 3023 // Convert volumes from 8.24 to 4.12 format 3024 // This additional clamping is needed in case chain->setVolume_l() overshot 3025 vl = (vl + (1 << 11)) >> 12; 3026 if (vl > MAX_GAIN_INT) { 3027 vl = MAX_GAIN_INT; 3028 } 3029 vr = (vr + (1 << 11)) >> 12; 3030 if (vr > MAX_GAIN_INT) { 3031 vr = MAX_GAIN_INT; 3032 } 3033 3034 if (va > MAX_GAIN_INT) { 3035 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3036 } 3037 3038 // XXX: these things DON'T need to be done each time 3039 mAudioMixer->setBufferProvider(name, track); 3040 mAudioMixer->enable(name); 3041 3042 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3043 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3044 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3045 mAudioMixer->setParameter( 3046 name, 3047 AudioMixer::TRACK, 3048 AudioMixer::FORMAT, (void *)track->format()); 3049 mAudioMixer->setParameter( 3050 name, 3051 AudioMixer::TRACK, 3052 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3053 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3054 uint32_t maxSampleRate = mSampleRate * 2; 3055 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3056 if (reqSampleRate == 0) { 3057 reqSampleRate = mSampleRate; 3058 } else if (reqSampleRate > maxSampleRate) { 3059 reqSampleRate = maxSampleRate; 3060 } 3061 mAudioMixer->setParameter( 3062 name, 3063 AudioMixer::RESAMPLE, 3064 AudioMixer::SAMPLE_RATE, 3065 (void *)reqSampleRate); 3066 mAudioMixer->setParameter( 3067 name, 3068 AudioMixer::TRACK, 3069 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3070 mAudioMixer->setParameter( 3071 name, 3072 AudioMixer::TRACK, 3073 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3074 3075 // reset retry count 3076 track->mRetryCount = kMaxTrackRetries; 3077 3078 // If one track is ready, set the mixer ready if: 3079 // - the mixer was not ready during previous round OR 3080 // - no other track is not ready 3081 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3082 mixerStatus != MIXER_TRACKS_ENABLED) { 3083 mixerStatus = MIXER_TRACKS_READY; 3084 } 3085 } else { 3086 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3087 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3088 } 3089 // clear effect chain input buffer if an active track underruns to avoid sending 3090 // previous audio buffer again to effects 3091 chain = getEffectChain_l(track->sessionId()); 3092 if (chain != 0) { 3093 chain->clearInputBuffer(); 3094 } 3095 3096 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3097 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3098 track->isStopped() || track->isPaused()) { 3099 // We have consumed all the buffers of this track. 3100 // Remove it from the list of active tracks. 3101 // TODO: use actual buffer filling status instead of latency when available from 3102 // audio HAL 3103 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3104 size_t framesWritten = mBytesWritten / mFrameSize; 3105 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3106 if (track->isStopped()) { 3107 track->reset(); 3108 } 3109 tracksToRemove->add(track); 3110 } 3111 } else { 3112 // No buffers for this track. Give it a few chances to 3113 // fill a buffer, then remove it from active list. 3114 if (--(track->mRetryCount) <= 0) { 3115 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3116 tracksToRemove->add(track); 3117 // indicate to client process that the track was disabled because of underrun; 3118 // it will then automatically call start() when data is available 3119 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3120 // If one track is not ready, mark the mixer also not ready if: 3121 // - the mixer was ready during previous round OR 3122 // - no other track is ready 3123 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3124 mixerStatus != MIXER_TRACKS_READY) { 3125 mixerStatus = MIXER_TRACKS_ENABLED; 3126 } 3127 } 3128 mAudioMixer->disable(name); 3129 } 3130 3131 } // local variable scope to avoid goto warning 3132track_is_ready: ; 3133 3134 } 3135 3136 // Push the new FastMixer state if necessary 3137 bool pauseAudioWatchdog = false; 3138 if (didModify) { 3139 state->mFastTracksGen++; 3140 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3141 if (kUseFastMixer == FastMixer_Dynamic && 3142 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3143 state->mCommand = FastMixerState::COLD_IDLE; 3144 state->mColdFutexAddr = &mFastMixerFutex; 3145 state->mColdGen++; 3146 mFastMixerFutex = 0; 3147 if (kUseFastMixer == FastMixer_Dynamic) { 3148 mNormalSink = mOutputSink; 3149 } 3150 // If we go into cold idle, need to wait for acknowledgement 3151 // so that fast mixer stops doing I/O. 3152 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3153 pauseAudioWatchdog = true; 3154 } 3155 } 3156 if (sq != NULL) { 3157 sq->end(didModify); 3158 sq->push(block); 3159 } 3160#ifdef AUDIO_WATCHDOG 3161 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3162 mAudioWatchdog->pause(); 3163 } 3164#endif 3165 3166 // Now perform the deferred reset on fast tracks that have stopped 3167 while (resetMask != 0) { 3168 size_t i = __builtin_ctz(resetMask); 3169 ALOG_ASSERT(i < count); 3170 resetMask &= ~(1 << i); 3171 sp<Track> t = mActiveTracks[i].promote(); 3172 if (t == 0) { 3173 continue; 3174 } 3175 Track* track = t.get(); 3176 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3177 track->reset(); 3178 } 3179 3180 // remove all the tracks that need to be... 3181 removeTracks_l(*tracksToRemove); 3182 3183 // mix buffer must be cleared if all tracks are connected to an 3184 // effect chain as in this case the mixer will not write to 3185 // mix buffer and track effects will accumulate into it 3186 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3187 (mixedTracks == 0 && fastTracks > 0))) { 3188 // FIXME as a performance optimization, should remember previous zero status 3189 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3190 } 3191 3192 // if any fast tracks, then status is ready 3193 mMixerStatusIgnoringFastTracks = mixerStatus; 3194 if (fastTracks > 0) { 3195 mixerStatus = MIXER_TRACKS_READY; 3196 } 3197 return mixerStatus; 3198} 3199 3200// getTrackName_l() must be called with ThreadBase::mLock held 3201int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3202{ 3203 return mAudioMixer->getTrackName(channelMask, sessionId); 3204} 3205 3206// deleteTrackName_l() must be called with ThreadBase::mLock held 3207void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3208{ 3209 ALOGV("remove track (%d) and delete from mixer", name); 3210 mAudioMixer->deleteTrackName(name); 3211} 3212 3213// checkForNewParameters_l() must be called with ThreadBase::mLock held 3214bool AudioFlinger::MixerThread::checkForNewParameters_l() 3215{ 3216 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3217 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3218 bool reconfig = false; 3219 3220 while (!mNewParameters.isEmpty()) { 3221 3222 if (mFastMixer != NULL) { 3223 FastMixerStateQueue *sq = mFastMixer->sq(); 3224 FastMixerState *state = sq->begin(); 3225 if (!(state->mCommand & FastMixerState::IDLE)) { 3226 previousCommand = state->mCommand; 3227 state->mCommand = FastMixerState::HOT_IDLE; 3228 sq->end(); 3229 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3230 } else { 3231 sq->end(false /*didModify*/); 3232 } 3233 } 3234 3235 status_t status = NO_ERROR; 3236 String8 keyValuePair = mNewParameters[0]; 3237 AudioParameter param = AudioParameter(keyValuePair); 3238 int value; 3239 3240 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3241 reconfig = true; 3242 } 3243 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3244 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3245 status = BAD_VALUE; 3246 } else { 3247 reconfig = true; 3248 } 3249 } 3250 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3251 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3252 status = BAD_VALUE; 3253 } else { 3254 reconfig = true; 3255 } 3256 } 3257 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3258 // do not accept frame count changes if tracks are open as the track buffer 3259 // size depends on frame count and correct behavior would not be guaranteed 3260 // if frame count is changed after track creation 3261 if (!mTracks.isEmpty()) { 3262 status = INVALID_OPERATION; 3263 } else { 3264 reconfig = true; 3265 } 3266 } 3267 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3268#ifdef ADD_BATTERY_DATA 3269 // when changing the audio output device, call addBatteryData to notify 3270 // the change 3271 if (mOutDevice != value) { 3272 uint32_t params = 0; 3273 // check whether speaker is on 3274 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3275 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3276 } 3277 3278 audio_devices_t deviceWithoutSpeaker 3279 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3280 // check if any other device (except speaker) is on 3281 if (value & deviceWithoutSpeaker ) { 3282 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3283 } 3284 3285 if (params != 0) { 3286 addBatteryData(params); 3287 } 3288 } 3289#endif 3290 3291 // forward device change to effects that have requested to be 3292 // aware of attached audio device. 3293 if (value != AUDIO_DEVICE_NONE) { 3294 mOutDevice = value; 3295 for (size_t i = 0; i < mEffectChains.size(); i++) { 3296 mEffectChains[i]->setDevice_l(mOutDevice); 3297 } 3298 } 3299 } 3300 3301 if (status == NO_ERROR) { 3302 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3303 keyValuePair.string()); 3304 if (!mStandby && status == INVALID_OPERATION) { 3305 mOutput->stream->common.standby(&mOutput->stream->common); 3306 mStandby = true; 3307 mBytesWritten = 0; 3308 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3309 keyValuePair.string()); 3310 } 3311 if (status == NO_ERROR && reconfig) { 3312 readOutputParameters(); 3313 delete mAudioMixer; 3314 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3315 for (size_t i = 0; i < mTracks.size() ; i++) { 3316 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3317 if (name < 0) { 3318 break; 3319 } 3320 mTracks[i]->mName = name; 3321 } 3322 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3323 } 3324 } 3325 3326 mNewParameters.removeAt(0); 3327 3328 mParamStatus = status; 3329 mParamCond.signal(); 3330 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3331 // already timed out waiting for the status and will never signal the condition. 3332 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3333 } 3334 3335 if (!(previousCommand & FastMixerState::IDLE)) { 3336 ALOG_ASSERT(mFastMixer != NULL); 3337 FastMixerStateQueue *sq = mFastMixer->sq(); 3338 FastMixerState *state = sq->begin(); 3339 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3340 state->mCommand = previousCommand; 3341 sq->end(); 3342 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3343 } 3344 3345 return reconfig; 3346} 3347 3348 3349void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3350{ 3351 const size_t SIZE = 256; 3352 char buffer[SIZE]; 3353 String8 result; 3354 3355 PlaybackThread::dumpInternals(fd, args); 3356 3357 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3358 result.append(buffer); 3359 write(fd, result.string(), result.size()); 3360 3361 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3362 const FastMixerDumpState copy(mFastMixerDumpState); 3363 copy.dump(fd); 3364 3365#ifdef STATE_QUEUE_DUMP 3366 // Similar for state queue 3367 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3368 observerCopy.dump(fd); 3369 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3370 mutatorCopy.dump(fd); 3371#endif 3372 3373#ifdef TEE_SINK 3374 // Write the tee output to a .wav file 3375 dumpTee(fd, mTeeSource, mId); 3376#endif 3377 3378#ifdef AUDIO_WATCHDOG 3379 if (mAudioWatchdog != 0) { 3380 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3381 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3382 wdCopy.dump(fd); 3383 } 3384#endif 3385} 3386 3387uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3388{ 3389 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3390} 3391 3392uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3393{ 3394 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3395} 3396 3397void AudioFlinger::MixerThread::cacheParameters_l() 3398{ 3399 PlaybackThread::cacheParameters_l(); 3400 3401 // FIXME: Relaxed timing because of a certain device that can't meet latency 3402 // Should be reduced to 2x after the vendor fixes the driver issue 3403 // increase threshold again due to low power audio mode. The way this warning 3404 // threshold is calculated and its usefulness should be reconsidered anyway. 3405 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3406} 3407 3408// ---------------------------------------------------------------------------- 3409 3410AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3411 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3412 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3413 // mLeftVolFloat, mRightVolFloat 3414{ 3415} 3416 3417AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3418 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3419 ThreadBase::type_t type) 3420 : PlaybackThread(audioFlinger, output, id, device, type) 3421 // mLeftVolFloat, mRightVolFloat 3422{ 3423} 3424 3425AudioFlinger::DirectOutputThread::~DirectOutputThread() 3426{ 3427} 3428 3429void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3430{ 3431 audio_track_cblk_t* cblk = track->cblk(); 3432 float left, right; 3433 3434 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3435 left = right = 0; 3436 } else { 3437 float typeVolume = mStreamTypes[track->streamType()].volume; 3438 float v = mMasterVolume * typeVolume; 3439 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3440 uint32_t vlr = proxy->getVolumeLR(); 3441 float v_clamped = v * (vlr & 0xFFFF); 3442 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3443 left = v_clamped/MAX_GAIN; 3444 v_clamped = v * (vlr >> 16); 3445 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3446 right = v_clamped/MAX_GAIN; 3447 } 3448 3449 if (lastTrack) { 3450 if (left != mLeftVolFloat || right != mRightVolFloat) { 3451 mLeftVolFloat = left; 3452 mRightVolFloat = right; 3453 3454 // Convert volumes from float to 8.24 3455 uint32_t vl = (uint32_t)(left * (1 << 24)); 3456 uint32_t vr = (uint32_t)(right * (1 << 24)); 3457 3458 // Delegate volume control to effect in track effect chain if needed 3459 // only one effect chain can be present on DirectOutputThread, so if 3460 // there is one, the track is connected to it 3461 if (!mEffectChains.isEmpty()) { 3462 mEffectChains[0]->setVolume_l(&vl, &vr); 3463 left = (float)vl / (1 << 24); 3464 right = (float)vr / (1 << 24); 3465 } 3466 if (mOutput->stream->set_volume) { 3467 mOutput->stream->set_volume(mOutput->stream, left, right); 3468 } 3469 } 3470 } 3471} 3472 3473 3474AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3475 Vector< sp<Track> > *tracksToRemove 3476) 3477{ 3478 size_t count = mActiveTracks.size(); 3479 mixer_state mixerStatus = MIXER_IDLE; 3480 3481 // find out which tracks need to be processed 3482 for (size_t i = 0; i < count; i++) { 3483 sp<Track> t = mActiveTracks[i].promote(); 3484 // The track died recently 3485 if (t == 0) { 3486 continue; 3487 } 3488 3489 Track* const track = t.get(); 3490 audio_track_cblk_t* cblk = track->cblk(); 3491 3492 // The first time a track is added we wait 3493 // for all its buffers to be filled before processing it 3494 uint32_t minFrames; 3495 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3496 minFrames = mNormalFrameCount; 3497 } else { 3498 minFrames = 1; 3499 } 3500 // Only consider last track started for volume and mixer state control. 3501 // This is the last entry in mActiveTracks unless a track underruns. 3502 // As we only care about the transition phase between two tracks on a 3503 // direct output, it is not a problem to ignore the underrun case. 3504 bool last = (i == (count - 1)); 3505 3506 if ((track->framesReady() >= minFrames) && track->isReady() && 3507 !track->isPaused() && !track->isTerminated()) 3508 { 3509 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3510 3511 if (track->mFillingUpStatus == Track::FS_FILLED) { 3512 track->mFillingUpStatus = Track::FS_ACTIVE; 3513 mLeftVolFloat = mRightVolFloat = 0; 3514 if (track->mState == TrackBase::RESUMING) { 3515 track->mState = TrackBase::ACTIVE; 3516 } 3517 } 3518 3519 // compute volume for this track 3520 processVolume_l(track, last); 3521 if (last) { 3522 // reset retry count 3523 track->mRetryCount = kMaxTrackRetriesDirect; 3524 mActiveTrack = t; 3525 mixerStatus = MIXER_TRACKS_READY; 3526 } 3527 } else { 3528 // clear effect chain input buffer if the last active track started underruns 3529 // to avoid sending previous audio buffer again to effects 3530 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3531 mEffectChains[0]->clearInputBuffer(); 3532 } 3533 3534 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3535 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3536 track->isStopped() || track->isPaused()) { 3537 // We have consumed all the buffers of this track. 3538 // Remove it from the list of active tracks. 3539 // TODO: implement behavior for compressed audio 3540 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3541 size_t framesWritten = mBytesWritten / mFrameSize; 3542 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3543 if (track->isStopped()) { 3544 track->reset(); 3545 } 3546 tracksToRemove->add(track); 3547 } 3548 } else { 3549 // No buffers for this track. Give it a few chances to 3550 // fill a buffer, then remove it from active list. 3551 // Only consider last track started for mixer state control 3552 if (--(track->mRetryCount) <= 0) { 3553 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3554 tracksToRemove->add(track); 3555 } else if (last) { 3556 mixerStatus = MIXER_TRACKS_ENABLED; 3557 } 3558 } 3559 } 3560 } 3561 3562 // remove all the tracks that need to be... 3563 removeTracks_l(*tracksToRemove); 3564 3565 return mixerStatus; 3566} 3567 3568void AudioFlinger::DirectOutputThread::threadLoop_mix() 3569{ 3570 size_t frameCount = mFrameCount; 3571 int8_t *curBuf = (int8_t *)mMixBuffer; 3572 // output audio to hardware 3573 while (frameCount) { 3574 AudioBufferProvider::Buffer buffer; 3575 buffer.frameCount = frameCount; 3576 mActiveTrack->getNextBuffer(&buffer); 3577 if (buffer.raw == NULL) { 3578 memset(curBuf, 0, frameCount * mFrameSize); 3579 break; 3580 } 3581 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3582 frameCount -= buffer.frameCount; 3583 curBuf += buffer.frameCount * mFrameSize; 3584 mActiveTrack->releaseBuffer(&buffer); 3585 } 3586 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3587 sleepTime = 0; 3588 standbyTime = systemTime() + standbyDelay; 3589 mActiveTrack.clear(); 3590} 3591 3592void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3593{ 3594 if (sleepTime == 0) { 3595 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3596 sleepTime = activeSleepTime; 3597 } else { 3598 sleepTime = idleSleepTime; 3599 } 3600 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3601 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3602 sleepTime = 0; 3603 } 3604} 3605 3606// getTrackName_l() must be called with ThreadBase::mLock held 3607int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3608 int sessionId) 3609{ 3610 return 0; 3611} 3612 3613// deleteTrackName_l() must be called with ThreadBase::mLock held 3614void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3615{ 3616} 3617 3618// checkForNewParameters_l() must be called with ThreadBase::mLock held 3619bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3620{ 3621 bool reconfig = false; 3622 3623 while (!mNewParameters.isEmpty()) { 3624 status_t status = NO_ERROR; 3625 String8 keyValuePair = mNewParameters[0]; 3626 AudioParameter param = AudioParameter(keyValuePair); 3627 int value; 3628 3629 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3630 // do not accept frame count changes if tracks are open as the track buffer 3631 // size depends on frame count and correct behavior would not be garantied 3632 // if frame count is changed after track creation 3633 if (!mTracks.isEmpty()) { 3634 status = INVALID_OPERATION; 3635 } else { 3636 reconfig = true; 3637 } 3638 } 3639 if (status == NO_ERROR) { 3640 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3641 keyValuePair.string()); 3642 if (!mStandby && status == INVALID_OPERATION) { 3643 mOutput->stream->common.standby(&mOutput->stream->common); 3644 mStandby = true; 3645 mBytesWritten = 0; 3646 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3647 keyValuePair.string()); 3648 } 3649 if (status == NO_ERROR && reconfig) { 3650 readOutputParameters(); 3651 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3652 } 3653 } 3654 3655 mNewParameters.removeAt(0); 3656 3657 mParamStatus = status; 3658 mParamCond.signal(); 3659 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3660 // already timed out waiting for the status and will never signal the condition. 3661 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3662 } 3663 return reconfig; 3664} 3665 3666uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3667{ 3668 uint32_t time; 3669 if (audio_is_linear_pcm(mFormat)) { 3670 time = PlaybackThread::activeSleepTimeUs(); 3671 } else { 3672 time = 10000; 3673 } 3674 return time; 3675} 3676 3677uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3678{ 3679 uint32_t time; 3680 if (audio_is_linear_pcm(mFormat)) { 3681 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3682 } else { 3683 time = 10000; 3684 } 3685 return time; 3686} 3687 3688uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3689{ 3690 uint32_t time; 3691 if (audio_is_linear_pcm(mFormat)) { 3692 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3693 } else { 3694 time = 10000; 3695 } 3696 return time; 3697} 3698 3699void AudioFlinger::DirectOutputThread::cacheParameters_l() 3700{ 3701 PlaybackThread::cacheParameters_l(); 3702 3703 // use shorter standby delay as on normal output to release 3704 // hardware resources as soon as possible 3705 if (audio_is_linear_pcm(mFormat)) { 3706 standbyDelay = microseconds(activeSleepTime*2); 3707 } else { 3708 standbyDelay = kOffloadStandbyDelayNs; 3709 } 3710} 3711 3712// ---------------------------------------------------------------------------- 3713 3714AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3715 const sp<AudioFlinger::OffloadThread>& offloadThread) 3716 : Thread(false /*canCallJava*/), 3717 mOffloadThread(offloadThread), 3718 mWriteAckSequence(0), 3719 mDrainSequence(0) 3720{ 3721} 3722 3723AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3724{ 3725} 3726 3727void AudioFlinger::AsyncCallbackThread::onFirstRef() 3728{ 3729 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3730} 3731 3732bool AudioFlinger::AsyncCallbackThread::threadLoop() 3733{ 3734 while (!exitPending()) { 3735 uint32_t writeAckSequence; 3736 uint32_t drainSequence; 3737 3738 { 3739 Mutex::Autolock _l(mLock); 3740 mWaitWorkCV.wait(mLock); 3741 if (exitPending()) { 3742 break; 3743 } 3744 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3745 mWriteAckSequence, mDrainSequence); 3746 writeAckSequence = mWriteAckSequence; 3747 mWriteAckSequence &= ~1; 3748 drainSequence = mDrainSequence; 3749 mDrainSequence &= ~1; 3750 } 3751 { 3752 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3753 if (offloadThread != 0) { 3754 if (writeAckSequence & 1) { 3755 offloadThread->resetWriteBlocked(writeAckSequence >> 1); 3756 } 3757 if (drainSequence & 1) { 3758 offloadThread->resetDraining(drainSequence >> 1); 3759 } 3760 } 3761 } 3762 } 3763 return false; 3764} 3765 3766void AudioFlinger::AsyncCallbackThread::exit() 3767{ 3768 ALOGV("AsyncCallbackThread::exit"); 3769 Mutex::Autolock _l(mLock); 3770 requestExit(); 3771 mWaitWorkCV.broadcast(); 3772} 3773 3774void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3775{ 3776 Mutex::Autolock _l(mLock); 3777 // bit 0 is cleared 3778 mWriteAckSequence = sequence << 1; 3779} 3780 3781void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3782{ 3783 Mutex::Autolock _l(mLock); 3784 // ignore unexpected callbacks 3785 if (mWriteAckSequence & 2) { 3786 mWriteAckSequence |= 1; 3787 mWaitWorkCV.signal(); 3788 } 3789} 3790 3791void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3792{ 3793 Mutex::Autolock _l(mLock); 3794 // bit 0 is cleared 3795 mDrainSequence = sequence << 1; 3796} 3797 3798void AudioFlinger::AsyncCallbackThread::resetDraining() 3799{ 3800 Mutex::Autolock _l(mLock); 3801 // ignore unexpected callbacks 3802 if (mDrainSequence & 2) { 3803 mDrainSequence |= 1; 3804 mWaitWorkCV.signal(); 3805 } 3806} 3807 3808 3809// ---------------------------------------------------------------------------- 3810AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3811 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3812 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3813 mHwPaused(false), 3814 mPausedBytesRemaining(0) 3815{ 3816 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3817} 3818 3819AudioFlinger::OffloadThread::~OffloadThread() 3820{ 3821 mPreviousTrack.clear(); 3822} 3823 3824void AudioFlinger::OffloadThread::threadLoop_exit() 3825{ 3826 if (mFlushPending || mHwPaused) { 3827 // If a flush is pending or track was paused, just discard buffered data 3828 flushHw_l(); 3829 } else { 3830 mMixerStatus = MIXER_DRAIN_ALL; 3831 threadLoop_drain(); 3832 } 3833 mCallbackThread->exit(); 3834 PlaybackThread::threadLoop_exit(); 3835} 3836 3837AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3838 Vector< sp<Track> > *tracksToRemove 3839) 3840{ 3841 ALOGV("OffloadThread::prepareTracks_l"); 3842 size_t count = mActiveTracks.size(); 3843 3844 mixer_state mixerStatus = MIXER_IDLE; 3845 bool doHwPause = false; 3846 bool doHwResume = false; 3847 3848 // find out which tracks need to be processed 3849 for (size_t i = 0; i < count; i++) { 3850 sp<Track> t = mActiveTracks[i].promote(); 3851 // The track died recently 3852 if (t == 0) { 3853 continue; 3854 } 3855 Track* const track = t.get(); 3856 audio_track_cblk_t* cblk = track->cblk(); 3857 if (mPreviousTrack != NULL) { 3858 if (t != mPreviousTrack) { 3859 // Flush any data still being written from last track 3860 mBytesRemaining = 0; 3861 if (mPausedBytesRemaining) { 3862 // Last track was paused so we also need to flush saved 3863 // mixbuffer state and invalidate track so that it will 3864 // re-submit that unwritten data when it is next resumed 3865 mPausedBytesRemaining = 0; 3866 // Invalidate is a bit drastic - would be more efficient 3867 // to have a flag to tell client that some of the 3868 // previously written data was lost 3869 mPreviousTrack->invalidate(); 3870 } 3871 } 3872 } 3873 mPreviousTrack = t; 3874 bool last = (i == (count - 1)); 3875 if (track->isPausing()) { 3876 track->setPaused(); 3877 if (last) { 3878 if (!mHwPaused) { 3879 doHwPause = true; 3880 mHwPaused = true; 3881 } 3882 // If we were part way through writing the mixbuffer to 3883 // the HAL we must save this until we resume 3884 // BUG - this will be wrong if a different track is made active, 3885 // in that case we want to discard the pending data in the 3886 // mixbuffer and tell the client to present it again when the 3887 // track is resumed 3888 mPausedWriteLength = mCurrentWriteLength; 3889 mPausedBytesRemaining = mBytesRemaining; 3890 mBytesRemaining = 0; // stop writing 3891 } 3892 tracksToRemove->add(track); 3893 } else if (track->framesReady() && track->isReady() && 3894 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3895 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3896 if (track->mFillingUpStatus == Track::FS_FILLED) { 3897 track->mFillingUpStatus = Track::FS_ACTIVE; 3898 mLeftVolFloat = mRightVolFloat = 0; 3899 if (track->mState == TrackBase::RESUMING) { 3900 if (mPausedBytesRemaining) { 3901 // Need to continue write that was interrupted 3902 mCurrentWriteLength = mPausedWriteLength; 3903 mBytesRemaining = mPausedBytesRemaining; 3904 mPausedBytesRemaining = 0; 3905 } 3906 track->mState = TrackBase::ACTIVE; 3907 } 3908 } 3909 3910 if (last) { 3911 if (mHwPaused) { 3912 doHwResume = true; 3913 mHwPaused = false; 3914 // threadLoop_mix() will handle the case that we need to 3915 // resume an interrupted write 3916 } 3917 // reset retry count 3918 track->mRetryCount = kMaxTrackRetriesOffload; 3919 mActiveTrack = t; 3920 mixerStatus = MIXER_TRACKS_READY; 3921 } 3922 } else { 3923 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3924 if (track->isStopping_1()) { 3925 // Hardware buffer can hold a large amount of audio so we must 3926 // wait for all current track's data to drain before we say 3927 // that the track is stopped. 3928 if (mBytesRemaining == 0) { 3929 // Only start draining when all data in mixbuffer 3930 // has been written 3931 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3932 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3933 sleepTime = 0; 3934 standbyTime = systemTime() + standbyDelay; 3935 if (last) { 3936 mixerStatus = MIXER_DRAIN_TRACK; 3937 mDrainSequence += 2; 3938 if (mHwPaused) { 3939 // It is possible to move from PAUSED to STOPPING_1 without 3940 // a resume so we must ensure hardware is running 3941 mOutput->stream->resume(mOutput->stream); 3942 mHwPaused = false; 3943 } 3944 } 3945 } 3946 } else if (track->isStopping_2()) { 3947 // Drain has completed, signal presentation complete 3948 if (!(mDrainSequence & 1) || !last) { 3949 track->mState = TrackBase::STOPPED; 3950 size_t audioHALFrames = 3951 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3952 size_t framesWritten = 3953 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3954 track->presentationComplete(framesWritten, audioHALFrames); 3955 track->reset(); 3956 tracksToRemove->add(track); 3957 } 3958 } else { 3959 // No buffers for this track. Give it a few chances to 3960 // fill a buffer, then remove it from active list. 3961 if (--(track->mRetryCount) <= 0) { 3962 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3963 track->name()); 3964 tracksToRemove->add(track); 3965 } else if (last){ 3966 mixerStatus = MIXER_TRACKS_ENABLED; 3967 } 3968 } 3969 } 3970 // compute volume for this track 3971 processVolume_l(track, last); 3972 } 3973 3974 // make sure the pause/flush/resume sequence is executed in the right order 3975 if (doHwPause) { 3976 mOutput->stream->pause(mOutput->stream); 3977 } 3978 if (mFlushPending) { 3979 flushHw_l(); 3980 mFlushPending = false; 3981 } 3982 if (doHwResume) { 3983 mOutput->stream->resume(mOutput->stream); 3984 } 3985 3986 // remove all the tracks that need to be... 3987 removeTracks_l(*tracksToRemove); 3988 3989 return mixerStatus; 3990} 3991 3992void AudioFlinger::OffloadThread::flushOutput_l() 3993{ 3994 mFlushPending = true; 3995} 3996 3997// must be called with thread mutex locked 3998bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3999{ 4000 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4001 mWriteAckSequence, mDrainSequence); 4002 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4003 return true; 4004 } 4005 return false; 4006} 4007 4008// must be called with thread mutex locked 4009bool AudioFlinger::OffloadThread::shouldStandby_l() 4010{ 4011 bool TrackPaused = false; 4012 4013 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4014 // after a timeout and we will enter standby then. 4015 if (mTracks.size() > 0) { 4016 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4017 } 4018 4019 return !mStandby && !TrackPaused; 4020} 4021 4022 4023bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4024{ 4025 Mutex::Autolock _l(mLock); 4026 return waitingAsyncCallback_l(); 4027} 4028 4029void AudioFlinger::OffloadThread::flushHw_l() 4030{ 4031 mOutput->stream->flush(mOutput->stream); 4032 // Flush anything still waiting in the mixbuffer 4033 mCurrentWriteLength = 0; 4034 mBytesRemaining = 0; 4035 mPausedWriteLength = 0; 4036 mPausedBytesRemaining = 0; 4037 if (mUseAsyncWrite) { 4038 // discard any pending drain or write ack by incrementing sequence 4039 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4040 mDrainSequence = (mDrainSequence + 2) & ~1; 4041 ALOG_ASSERT(mCallbackThread != 0); 4042 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4043 mCallbackThread->setDraining(mDrainSequence); 4044 } 4045} 4046 4047// ---------------------------------------------------------------------------- 4048 4049AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4050 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4051 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4052 DUPLICATING), 4053 mWaitTimeMs(UINT_MAX) 4054{ 4055 addOutputTrack(mainThread); 4056} 4057 4058AudioFlinger::DuplicatingThread::~DuplicatingThread() 4059{ 4060 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4061 mOutputTracks[i]->destroy(); 4062 } 4063} 4064 4065void AudioFlinger::DuplicatingThread::threadLoop_mix() 4066{ 4067 // mix buffers... 4068 if (outputsReady(outputTracks)) { 4069 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4070 } else { 4071 memset(mMixBuffer, 0, mixBufferSize); 4072 } 4073 sleepTime = 0; 4074 writeFrames = mNormalFrameCount; 4075 mCurrentWriteLength = mixBufferSize; 4076 standbyTime = systemTime() + standbyDelay; 4077} 4078 4079void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4080{ 4081 if (sleepTime == 0) { 4082 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4083 sleepTime = activeSleepTime; 4084 } else { 4085 sleepTime = idleSleepTime; 4086 } 4087 } else if (mBytesWritten != 0) { 4088 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4089 writeFrames = mNormalFrameCount; 4090 memset(mMixBuffer, 0, mixBufferSize); 4091 } else { 4092 // flush remaining overflow buffers in output tracks 4093 writeFrames = 0; 4094 } 4095 sleepTime = 0; 4096 } 4097} 4098 4099ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4100{ 4101 for (size_t i = 0; i < outputTracks.size(); i++) { 4102 outputTracks[i]->write(mMixBuffer, writeFrames); 4103 } 4104 return (ssize_t)mixBufferSize; 4105} 4106 4107void AudioFlinger::DuplicatingThread::threadLoop_standby() 4108{ 4109 // DuplicatingThread implements standby by stopping all tracks 4110 for (size_t i = 0; i < outputTracks.size(); i++) { 4111 outputTracks[i]->stop(); 4112 } 4113} 4114 4115void AudioFlinger::DuplicatingThread::saveOutputTracks() 4116{ 4117 outputTracks = mOutputTracks; 4118} 4119 4120void AudioFlinger::DuplicatingThread::clearOutputTracks() 4121{ 4122 outputTracks.clear(); 4123} 4124 4125void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4126{ 4127 Mutex::Autolock _l(mLock); 4128 // FIXME explain this formula 4129 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4130 OutputTrack *outputTrack = new OutputTrack(thread, 4131 this, 4132 mSampleRate, 4133 mFormat, 4134 mChannelMask, 4135 frameCount); 4136 if (outputTrack->cblk() != NULL) { 4137 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4138 mOutputTracks.add(outputTrack); 4139 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4140 updateWaitTime_l(); 4141 } 4142} 4143 4144void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4145{ 4146 Mutex::Autolock _l(mLock); 4147 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4148 if (mOutputTracks[i]->thread() == thread) { 4149 mOutputTracks[i]->destroy(); 4150 mOutputTracks.removeAt(i); 4151 updateWaitTime_l(); 4152 return; 4153 } 4154 } 4155 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4156} 4157 4158// caller must hold mLock 4159void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4160{ 4161 mWaitTimeMs = UINT_MAX; 4162 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4163 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4164 if (strong != 0) { 4165 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4166 if (waitTimeMs < mWaitTimeMs) { 4167 mWaitTimeMs = waitTimeMs; 4168 } 4169 } 4170 } 4171} 4172 4173 4174bool AudioFlinger::DuplicatingThread::outputsReady( 4175 const SortedVector< sp<OutputTrack> > &outputTracks) 4176{ 4177 for (size_t i = 0; i < outputTracks.size(); i++) { 4178 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4179 if (thread == 0) { 4180 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4181 outputTracks[i].get()); 4182 return false; 4183 } 4184 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4185 // see note at standby() declaration 4186 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4187 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4188 thread.get()); 4189 return false; 4190 } 4191 } 4192 return true; 4193} 4194 4195uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4196{ 4197 return (mWaitTimeMs * 1000) / 2; 4198} 4199 4200void AudioFlinger::DuplicatingThread::cacheParameters_l() 4201{ 4202 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4203 updateWaitTime_l(); 4204 4205 MixerThread::cacheParameters_l(); 4206} 4207 4208// ---------------------------------------------------------------------------- 4209// Record 4210// ---------------------------------------------------------------------------- 4211 4212AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4213 AudioStreamIn *input, 4214 uint32_t sampleRate, 4215 audio_channel_mask_t channelMask, 4216 audio_io_handle_t id, 4217 audio_devices_t outDevice, 4218 audio_devices_t inDevice 4219#ifdef TEE_SINK 4220 , const sp<NBAIO_Sink>& teeSink 4221#endif 4222 ) : 4223 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4224 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4225 // mRsmpInIndex and mBufferSize set by readInputParameters() 4226 mReqChannelCount(popcount(channelMask)), 4227 mReqSampleRate(sampleRate) 4228 // mBytesRead is only meaningful while active, and so is cleared in start() 4229 // (but might be better to also clear here for dump?) 4230#ifdef TEE_SINK 4231 , mTeeSink(teeSink) 4232#endif 4233{ 4234 snprintf(mName, kNameLength, "AudioIn_%X", id); 4235 4236 readInputParameters(); 4237 4238} 4239 4240 4241AudioFlinger::RecordThread::~RecordThread() 4242{ 4243 delete[] mRsmpInBuffer; 4244 delete mResampler; 4245 delete[] mRsmpOutBuffer; 4246} 4247 4248void AudioFlinger::RecordThread::onFirstRef() 4249{ 4250 run(mName, PRIORITY_URGENT_AUDIO); 4251} 4252 4253status_t AudioFlinger::RecordThread::readyToRun() 4254{ 4255 status_t status = initCheck(); 4256 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4257 return status; 4258} 4259 4260bool AudioFlinger::RecordThread::threadLoop() 4261{ 4262 AudioBufferProvider::Buffer buffer; 4263 sp<RecordTrack> activeTrack; 4264 Vector< sp<EffectChain> > effectChains; 4265 4266 nsecs_t lastWarning = 0; 4267 4268 inputStandBy(); 4269 acquireWakeLock(); 4270 4271 // used to verify we've read at least once before evaluating how many bytes were read 4272 bool readOnce = false; 4273 4274 // start recording 4275 while (!exitPending()) { 4276 4277 processConfigEvents(); 4278 4279 { // scope for mLock 4280 Mutex::Autolock _l(mLock); 4281 checkForNewParameters_l(); 4282 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4283 standby(); 4284 4285 if (exitPending()) { 4286 break; 4287 } 4288 4289 releaseWakeLock_l(); 4290 ALOGV("RecordThread: loop stopping"); 4291 // go to sleep 4292 mWaitWorkCV.wait(mLock); 4293 ALOGV("RecordThread: loop starting"); 4294 acquireWakeLock_l(); 4295 continue; 4296 } 4297 if (mActiveTrack != 0) { 4298 if (mActiveTrack->isTerminated()) { 4299 removeTrack_l(mActiveTrack); 4300 mActiveTrack.clear(); 4301 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4302 standby(); 4303 mActiveTrack.clear(); 4304 mStartStopCond.broadcast(); 4305 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4306 if (mReqChannelCount != mActiveTrack->channelCount()) { 4307 mActiveTrack.clear(); 4308 mStartStopCond.broadcast(); 4309 } else if (readOnce) { 4310 // record start succeeds only if first read from audio input 4311 // succeeds 4312 if (mBytesRead >= 0) { 4313 mActiveTrack->mState = TrackBase::ACTIVE; 4314 } else { 4315 mActiveTrack.clear(); 4316 } 4317 mStartStopCond.broadcast(); 4318 } 4319 mStandby = false; 4320 } 4321 } 4322 lockEffectChains_l(effectChains); 4323 } 4324 4325 if (mActiveTrack != 0) { 4326 if (mActiveTrack->mState != TrackBase::ACTIVE && 4327 mActiveTrack->mState != TrackBase::RESUMING) { 4328 unlockEffectChains(effectChains); 4329 usleep(kRecordThreadSleepUs); 4330 continue; 4331 } 4332 for (size_t i = 0; i < effectChains.size(); i ++) { 4333 effectChains[i]->process_l(); 4334 } 4335 4336 buffer.frameCount = mFrameCount; 4337 status_t status = mActiveTrack->getNextBuffer(&buffer); 4338 if (status == NO_ERROR) { 4339 readOnce = true; 4340 size_t framesOut = buffer.frameCount; 4341 if (mResampler == NULL) { 4342 // no resampling 4343 while (framesOut) { 4344 size_t framesIn = mFrameCount - mRsmpInIndex; 4345 if (framesIn) { 4346 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4347 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4348 mActiveTrack->mFrameSize; 4349 if (framesIn > framesOut) 4350 framesIn = framesOut; 4351 mRsmpInIndex += framesIn; 4352 framesOut -= framesIn; 4353 if (mChannelCount == mReqChannelCount) { 4354 memcpy(dst, src, framesIn * mFrameSize); 4355 } else { 4356 if (mChannelCount == 1) { 4357 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4358 (int16_t *)src, framesIn); 4359 } else { 4360 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4361 (int16_t *)src, framesIn); 4362 } 4363 } 4364 } 4365 if (framesOut && mFrameCount == mRsmpInIndex) { 4366 void *readInto; 4367 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4368 readInto = buffer.raw; 4369 framesOut = 0; 4370 } else { 4371 readInto = mRsmpInBuffer; 4372 mRsmpInIndex = 0; 4373 } 4374 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4375 mBufferSize); 4376 if (mBytesRead <= 0) { 4377 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4378 { 4379 ALOGE("Error reading audio input"); 4380 // Force input into standby so that it tries to 4381 // recover at next read attempt 4382 inputStandBy(); 4383 usleep(kRecordThreadSleepUs); 4384 } 4385 mRsmpInIndex = mFrameCount; 4386 framesOut = 0; 4387 buffer.frameCount = 0; 4388 } 4389#ifdef TEE_SINK 4390 else if (mTeeSink != 0) { 4391 (void) mTeeSink->write(readInto, 4392 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4393 } 4394#endif 4395 } 4396 } 4397 } else { 4398 // resampling 4399 4400 // resampler accumulates, but we only have one source track 4401 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4402 // alter output frame count as if we were expecting stereo samples 4403 if (mChannelCount == 1 && mReqChannelCount == 1) { 4404 framesOut >>= 1; 4405 } 4406 mResampler->resample(mRsmpOutBuffer, framesOut, 4407 this /* AudioBufferProvider* */); 4408 // ditherAndClamp() works as long as all buffers returned by 4409 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4410 if (mChannelCount == 2 && mReqChannelCount == 1) { 4411 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4412 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4413 // the resampler always outputs stereo samples: 4414 // do post stereo to mono conversion 4415 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4416 framesOut); 4417 } else { 4418 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4419 } 4420 // now done with mRsmpOutBuffer 4421 4422 } 4423 if (mFramestoDrop == 0) { 4424 mActiveTrack->releaseBuffer(&buffer); 4425 } else { 4426 if (mFramestoDrop > 0) { 4427 mFramestoDrop -= buffer.frameCount; 4428 if (mFramestoDrop <= 0) { 4429 clearSyncStartEvent(); 4430 } 4431 } else { 4432 mFramestoDrop += buffer.frameCount; 4433 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4434 mSyncStartEvent->isCancelled()) { 4435 ALOGW("Synced record %s, session %d, trigger session %d", 4436 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4437 mActiveTrack->sessionId(), 4438 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4439 clearSyncStartEvent(); 4440 } 4441 } 4442 } 4443 mActiveTrack->clearOverflow(); 4444 } 4445 // client isn't retrieving buffers fast enough 4446 else { 4447 if (!mActiveTrack->setOverflow()) { 4448 nsecs_t now = systemTime(); 4449 if ((now - lastWarning) > kWarningThrottleNs) { 4450 ALOGW("RecordThread: buffer overflow"); 4451 lastWarning = now; 4452 } 4453 } 4454 // Release the processor for a while before asking for a new buffer. 4455 // This will give the application more chance to read from the buffer and 4456 // clear the overflow. 4457 usleep(kRecordThreadSleepUs); 4458 } 4459 } 4460 // enable changes in effect chain 4461 unlockEffectChains(effectChains); 4462 effectChains.clear(); 4463 } 4464 4465 standby(); 4466 4467 { 4468 Mutex::Autolock _l(mLock); 4469 for (size_t i = 0; i < mTracks.size(); i++) { 4470 sp<RecordTrack> track = mTracks[i]; 4471 track->invalidate(); 4472 } 4473 mActiveTrack.clear(); 4474 mStartStopCond.broadcast(); 4475 } 4476 4477 releaseWakeLock(); 4478 4479 ALOGV("RecordThread %p exiting", this); 4480 return false; 4481} 4482 4483void AudioFlinger::RecordThread::standby() 4484{ 4485 if (!mStandby) { 4486 inputStandBy(); 4487 mStandby = true; 4488 } 4489} 4490 4491void AudioFlinger::RecordThread::inputStandBy() 4492{ 4493 mInput->stream->common.standby(&mInput->stream->common); 4494} 4495 4496sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4497 const sp<AudioFlinger::Client>& client, 4498 uint32_t sampleRate, 4499 audio_format_t format, 4500 audio_channel_mask_t channelMask, 4501 size_t frameCount, 4502 int sessionId, 4503 IAudioFlinger::track_flags_t *flags, 4504 pid_t tid, 4505 status_t *status) 4506{ 4507 sp<RecordTrack> track; 4508 status_t lStatus; 4509 4510 lStatus = initCheck(); 4511 if (lStatus != NO_ERROR) { 4512 ALOGE("Audio driver not initialized."); 4513 goto Exit; 4514 } 4515 4516 // client expresses a preference for FAST, but we get the final say 4517 if (*flags & IAudioFlinger::TRACK_FAST) { 4518 if ( 4519 // use case: callback handler and frame count is default or at least as large as HAL 4520 ( 4521 (tid != -1) && 4522 ((frameCount == 0) || 4523 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4524 ) && 4525 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4526 // mono or stereo 4527 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4528 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4529 // hardware sample rate 4530 (sampleRate == mSampleRate) && 4531 // record thread has an associated fast recorder 4532 hasFastRecorder() 4533 // FIXME test that RecordThread for this fast track has a capable output HAL 4534 // FIXME add a permission test also? 4535 ) { 4536 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4537 if (frameCount == 0) { 4538 frameCount = mFrameCount * kFastTrackMultiplier; 4539 } 4540 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4541 frameCount, mFrameCount); 4542 } else { 4543 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4544 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4545 "hasFastRecorder=%d tid=%d", 4546 frameCount, mFrameCount, format, 4547 audio_is_linear_pcm(format), 4548 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4549 *flags &= ~IAudioFlinger::TRACK_FAST; 4550 // For compatibility with AudioRecord calculation, buffer depth is forced 4551 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4552 // This is probably too conservative, but legacy application code may depend on it. 4553 // If you change this calculation, also review the start threshold which is related. 4554 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4555 size_t mNormalFrameCount = 2048; // FIXME 4556 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4557 if (minBufCount < 2) { 4558 minBufCount = 2; 4559 } 4560 size_t minFrameCount = mNormalFrameCount * minBufCount; 4561 if (frameCount < minFrameCount) { 4562 frameCount = minFrameCount; 4563 } 4564 } 4565 } 4566 4567 // FIXME use flags and tid similar to createTrack_l() 4568 4569 { // scope for mLock 4570 Mutex::Autolock _l(mLock); 4571 4572 track = new RecordTrack(this, client, sampleRate, 4573 format, channelMask, frameCount, sessionId); 4574 4575 if (track->getCblk() == 0) { 4576 lStatus = NO_MEMORY; 4577 goto Exit; 4578 } 4579 mTracks.add(track); 4580 4581 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4582 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4583 mAudioFlinger->btNrecIsOff(); 4584 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4585 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4586 4587 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4588 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4589 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4590 // so ask activity manager to do this on our behalf 4591 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4592 } 4593 } 4594 lStatus = NO_ERROR; 4595 4596Exit: 4597 if (status) { 4598 *status = lStatus; 4599 } 4600 return track; 4601} 4602 4603status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4604 AudioSystem::sync_event_t event, 4605 int triggerSession) 4606{ 4607 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4608 sp<ThreadBase> strongMe = this; 4609 status_t status = NO_ERROR; 4610 4611 if (event == AudioSystem::SYNC_EVENT_NONE) { 4612 clearSyncStartEvent(); 4613 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4614 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4615 triggerSession, 4616 recordTrack->sessionId(), 4617 syncStartEventCallback, 4618 this); 4619 // Sync event can be cancelled by the trigger session if the track is not in a 4620 // compatible state in which case we start record immediately 4621 if (mSyncStartEvent->isCancelled()) { 4622 clearSyncStartEvent(); 4623 } else { 4624 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4625 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4626 } 4627 } 4628 4629 { 4630 AutoMutex lock(mLock); 4631 if (mActiveTrack != 0) { 4632 if (recordTrack != mActiveTrack.get()) { 4633 status = -EBUSY; 4634 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4635 mActiveTrack->mState = TrackBase::ACTIVE; 4636 } 4637 return status; 4638 } 4639 4640 recordTrack->mState = TrackBase::IDLE; 4641 mActiveTrack = recordTrack; 4642 mLock.unlock(); 4643 status_t status = AudioSystem::startInput(mId); 4644 mLock.lock(); 4645 if (status != NO_ERROR) { 4646 mActiveTrack.clear(); 4647 clearSyncStartEvent(); 4648 return status; 4649 } 4650 mRsmpInIndex = mFrameCount; 4651 mBytesRead = 0; 4652 if (mResampler != NULL) { 4653 mResampler->reset(); 4654 } 4655 mActiveTrack->mState = TrackBase::RESUMING; 4656 // signal thread to start 4657 ALOGV("Signal record thread"); 4658 mWaitWorkCV.broadcast(); 4659 // do not wait for mStartStopCond if exiting 4660 if (exitPending()) { 4661 mActiveTrack.clear(); 4662 status = INVALID_OPERATION; 4663 goto startError; 4664 } 4665 mStartStopCond.wait(mLock); 4666 if (mActiveTrack == 0) { 4667 ALOGV("Record failed to start"); 4668 status = BAD_VALUE; 4669 goto startError; 4670 } 4671 ALOGV("Record started OK"); 4672 return status; 4673 } 4674 4675startError: 4676 AudioSystem::stopInput(mId); 4677 clearSyncStartEvent(); 4678 return status; 4679} 4680 4681void AudioFlinger::RecordThread::clearSyncStartEvent() 4682{ 4683 if (mSyncStartEvent != 0) { 4684 mSyncStartEvent->cancel(); 4685 } 4686 mSyncStartEvent.clear(); 4687 mFramestoDrop = 0; 4688} 4689 4690void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4691{ 4692 sp<SyncEvent> strongEvent = event.promote(); 4693 4694 if (strongEvent != 0) { 4695 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4696 me->handleSyncStartEvent(strongEvent); 4697 } 4698} 4699 4700void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4701{ 4702 if (event == mSyncStartEvent) { 4703 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4704 // from audio HAL 4705 mFramestoDrop = mFrameCount * 2; 4706 } 4707} 4708 4709bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4710 ALOGV("RecordThread::stop"); 4711 AutoMutex _l(mLock); 4712 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4713 return false; 4714 } 4715 recordTrack->mState = TrackBase::PAUSING; 4716 // do not wait for mStartStopCond if exiting 4717 if (exitPending()) { 4718 return true; 4719 } 4720 mStartStopCond.wait(mLock); 4721 // if we have been restarted, recordTrack == mActiveTrack.get() here 4722 if (exitPending() || recordTrack != mActiveTrack.get()) { 4723 ALOGV("Record stopped OK"); 4724 return true; 4725 } 4726 return false; 4727} 4728 4729bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4730{ 4731 return false; 4732} 4733 4734status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4735{ 4736#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4737 if (!isValidSyncEvent(event)) { 4738 return BAD_VALUE; 4739 } 4740 4741 int eventSession = event->triggerSession(); 4742 status_t ret = NAME_NOT_FOUND; 4743 4744 Mutex::Autolock _l(mLock); 4745 4746 for (size_t i = 0; i < mTracks.size(); i++) { 4747 sp<RecordTrack> track = mTracks[i]; 4748 if (eventSession == track->sessionId()) { 4749 (void) track->setSyncEvent(event); 4750 ret = NO_ERROR; 4751 } 4752 } 4753 return ret; 4754#else 4755 return BAD_VALUE; 4756#endif 4757} 4758 4759// destroyTrack_l() must be called with ThreadBase::mLock held 4760void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4761{ 4762 track->terminate(); 4763 track->mState = TrackBase::STOPPED; 4764 // active tracks are removed by threadLoop() 4765 if (mActiveTrack != track) { 4766 removeTrack_l(track); 4767 } 4768} 4769 4770void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4771{ 4772 mTracks.remove(track); 4773 // need anything related to effects here? 4774} 4775 4776void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4777{ 4778 dumpInternals(fd, args); 4779 dumpTracks(fd, args); 4780 dumpEffectChains(fd, args); 4781} 4782 4783void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4784{ 4785 const size_t SIZE = 256; 4786 char buffer[SIZE]; 4787 String8 result; 4788 4789 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4790 result.append(buffer); 4791 4792 if (mActiveTrack != 0) { 4793 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4794 result.append(buffer); 4795 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4796 result.append(buffer); 4797 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4798 result.append(buffer); 4799 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4800 result.append(buffer); 4801 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4802 result.append(buffer); 4803 } else { 4804 result.append("No active record client\n"); 4805 } 4806 4807 write(fd, result.string(), result.size()); 4808 4809 dumpBase(fd, args); 4810} 4811 4812void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4813{ 4814 const size_t SIZE = 256; 4815 char buffer[SIZE]; 4816 String8 result; 4817 4818 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4819 result.append(buffer); 4820 RecordTrack::appendDumpHeader(result); 4821 for (size_t i = 0; i < mTracks.size(); ++i) { 4822 sp<RecordTrack> track = mTracks[i]; 4823 if (track != 0) { 4824 track->dump(buffer, SIZE); 4825 result.append(buffer); 4826 } 4827 } 4828 4829 if (mActiveTrack != 0) { 4830 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4831 result.append(buffer); 4832 RecordTrack::appendDumpHeader(result); 4833 mActiveTrack->dump(buffer, SIZE); 4834 result.append(buffer); 4835 4836 } 4837 write(fd, result.string(), result.size()); 4838} 4839 4840// AudioBufferProvider interface 4841status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4842{ 4843 size_t framesReq = buffer->frameCount; 4844 size_t framesReady = mFrameCount - mRsmpInIndex; 4845 int channelCount; 4846 4847 if (framesReady == 0) { 4848 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4849 if (mBytesRead <= 0) { 4850 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4851 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4852 // Force input into standby so that it tries to 4853 // recover at next read attempt 4854 inputStandBy(); 4855 usleep(kRecordThreadSleepUs); 4856 } 4857 buffer->raw = NULL; 4858 buffer->frameCount = 0; 4859 return NOT_ENOUGH_DATA; 4860 } 4861 mRsmpInIndex = 0; 4862 framesReady = mFrameCount; 4863 } 4864 4865 if (framesReq > framesReady) { 4866 framesReq = framesReady; 4867 } 4868 4869 if (mChannelCount == 1 && mReqChannelCount == 2) { 4870 channelCount = 1; 4871 } else { 4872 channelCount = 2; 4873 } 4874 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4875 buffer->frameCount = framesReq; 4876 return NO_ERROR; 4877} 4878 4879// AudioBufferProvider interface 4880void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4881{ 4882 mRsmpInIndex += buffer->frameCount; 4883 buffer->frameCount = 0; 4884} 4885 4886bool AudioFlinger::RecordThread::checkForNewParameters_l() 4887{ 4888 bool reconfig = false; 4889 4890 while (!mNewParameters.isEmpty()) { 4891 status_t status = NO_ERROR; 4892 String8 keyValuePair = mNewParameters[0]; 4893 AudioParameter param = AudioParameter(keyValuePair); 4894 int value; 4895 audio_format_t reqFormat = mFormat; 4896 uint32_t reqSamplingRate = mReqSampleRate; 4897 uint32_t reqChannelCount = mReqChannelCount; 4898 4899 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4900 reqSamplingRate = value; 4901 reconfig = true; 4902 } 4903 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4904 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4905 status = BAD_VALUE; 4906 } else { 4907 reqFormat = (audio_format_t) value; 4908 reconfig = true; 4909 } 4910 } 4911 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4912 reqChannelCount = popcount(value); 4913 reconfig = true; 4914 } 4915 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4916 // do not accept frame count changes if tracks are open as the track buffer 4917 // size depends on frame count and correct behavior would not be guaranteed 4918 // if frame count is changed after track creation 4919 if (mActiveTrack != 0) { 4920 status = INVALID_OPERATION; 4921 } else { 4922 reconfig = true; 4923 } 4924 } 4925 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4926 // forward device change to effects that have requested to be 4927 // aware of attached audio device. 4928 for (size_t i = 0; i < mEffectChains.size(); i++) { 4929 mEffectChains[i]->setDevice_l(value); 4930 } 4931 4932 // store input device and output device but do not forward output device to audio HAL. 4933 // Note that status is ignored by the caller for output device 4934 // (see AudioFlinger::setParameters() 4935 if (audio_is_output_devices(value)) { 4936 mOutDevice = value; 4937 status = BAD_VALUE; 4938 } else { 4939 mInDevice = value; 4940 // disable AEC and NS if the device is a BT SCO headset supporting those 4941 // pre processings 4942 if (mTracks.size() > 0) { 4943 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4944 mAudioFlinger->btNrecIsOff(); 4945 for (size_t i = 0; i < mTracks.size(); i++) { 4946 sp<RecordTrack> track = mTracks[i]; 4947 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4948 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4949 } 4950 } 4951 } 4952 } 4953 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4954 mAudioSource != (audio_source_t)value) { 4955 // forward device change to effects that have requested to be 4956 // aware of attached audio device. 4957 for (size_t i = 0; i < mEffectChains.size(); i++) { 4958 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4959 } 4960 mAudioSource = (audio_source_t)value; 4961 } 4962 if (status == NO_ERROR) { 4963 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4964 keyValuePair.string()); 4965 if (status == INVALID_OPERATION) { 4966 inputStandBy(); 4967 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4968 keyValuePair.string()); 4969 } 4970 if (reconfig) { 4971 if (status == BAD_VALUE && 4972 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4973 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4974 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4975 <= (2 * reqSamplingRate)) && 4976 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4977 <= FCC_2 && 4978 (reqChannelCount <= FCC_2)) { 4979 status = NO_ERROR; 4980 } 4981 if (status == NO_ERROR) { 4982 readInputParameters(); 4983 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4984 } 4985 } 4986 } 4987 4988 mNewParameters.removeAt(0); 4989 4990 mParamStatus = status; 4991 mParamCond.signal(); 4992 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4993 // already timed out waiting for the status and will never signal the condition. 4994 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4995 } 4996 return reconfig; 4997} 4998 4999String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5000{ 5001 Mutex::Autolock _l(mLock); 5002 if (initCheck() != NO_ERROR) { 5003 return String8(); 5004 } 5005 5006 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5007 const String8 out_s8(s); 5008 free(s); 5009 return out_s8; 5010} 5011 5012void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5013 AudioSystem::OutputDescriptor desc; 5014 void *param2 = NULL; 5015 5016 switch (event) { 5017 case AudioSystem::INPUT_OPENED: 5018 case AudioSystem::INPUT_CONFIG_CHANGED: 5019 desc.channelMask = mChannelMask; 5020 desc.samplingRate = mSampleRate; 5021 desc.format = mFormat; 5022 desc.frameCount = mFrameCount; 5023 desc.latency = 0; 5024 param2 = &desc; 5025 break; 5026 5027 case AudioSystem::INPUT_CLOSED: 5028 default: 5029 break; 5030 } 5031 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5032} 5033 5034void AudioFlinger::RecordThread::readInputParameters() 5035{ 5036 delete[] mRsmpInBuffer; 5037 // mRsmpInBuffer is always assigned a new[] below 5038 delete[] mRsmpOutBuffer; 5039 mRsmpOutBuffer = NULL; 5040 delete mResampler; 5041 mResampler = NULL; 5042 5043 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5044 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5045 mChannelCount = popcount(mChannelMask); 5046 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5047 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5048 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5049 } 5050 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5051 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5052 mFrameCount = mBufferSize / mFrameSize; 5053 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5054 5055 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5056 { 5057 int channelCount; 5058 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5059 // stereo to mono post process as the resampler always outputs stereo. 5060 if (mChannelCount == 1 && mReqChannelCount == 2) { 5061 channelCount = 1; 5062 } else { 5063 channelCount = 2; 5064 } 5065 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5066 mResampler->setSampleRate(mSampleRate); 5067 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5068 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5069 5070 // optmization: if mono to mono, alter input frame count as if we were inputing 5071 // stereo samples 5072 if (mChannelCount == 1 && mReqChannelCount == 1) { 5073 mFrameCount >>= 1; 5074 } 5075 5076 } 5077 mRsmpInIndex = mFrameCount; 5078} 5079 5080unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5081{ 5082 Mutex::Autolock _l(mLock); 5083 if (initCheck() != NO_ERROR) { 5084 return 0; 5085 } 5086 5087 return mInput->stream->get_input_frames_lost(mInput->stream); 5088} 5089 5090uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5091{ 5092 Mutex::Autolock _l(mLock); 5093 uint32_t result = 0; 5094 if (getEffectChain_l(sessionId) != 0) { 5095 result = EFFECT_SESSION; 5096 } 5097 5098 for (size_t i = 0; i < mTracks.size(); ++i) { 5099 if (sessionId == mTracks[i]->sessionId()) { 5100 result |= TRACK_SESSION; 5101 break; 5102 } 5103 } 5104 5105 return result; 5106} 5107 5108KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5109{ 5110 KeyedVector<int, bool> ids; 5111 Mutex::Autolock _l(mLock); 5112 for (size_t j = 0; j < mTracks.size(); ++j) { 5113 sp<RecordThread::RecordTrack> track = mTracks[j]; 5114 int sessionId = track->sessionId(); 5115 if (ids.indexOfKey(sessionId) < 0) { 5116 ids.add(sessionId, true); 5117 } 5118 } 5119 return ids; 5120} 5121 5122AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5123{ 5124 Mutex::Autolock _l(mLock); 5125 AudioStreamIn *input = mInput; 5126 mInput = NULL; 5127 return input; 5128} 5129 5130// this method must always be called either with ThreadBase mLock held or inside the thread loop 5131audio_stream_t* AudioFlinger::RecordThread::stream() const 5132{ 5133 if (mInput == NULL) { 5134 return NULL; 5135 } 5136 return &mInput->stream->common; 5137} 5138 5139status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5140{ 5141 // only one chain per input thread 5142 if (mEffectChains.size() != 0) { 5143 return INVALID_OPERATION; 5144 } 5145 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5146 5147 chain->setInBuffer(NULL); 5148 chain->setOutBuffer(NULL); 5149 5150 checkSuspendOnAddEffectChain_l(chain); 5151 5152 mEffectChains.add(chain); 5153 5154 return NO_ERROR; 5155} 5156 5157size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5158{ 5159 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5160 ALOGW_IF(mEffectChains.size() != 1, 5161 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5162 chain.get(), mEffectChains.size(), this); 5163 if (mEffectChains.size() == 1) { 5164 mEffectChains.removeAt(0); 5165 } 5166 return 0; 5167} 5168 5169}; // namespace android 5170