Threads.cpp revision d5577f26de1ae3a0dc6fbea9c60a07d585f894bf
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Whether to use fast mixer 113static const enum { 114 FastMixer_Never, // never initialize or use: for debugging only 115 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 116 // normal mixer multiplier is 1 117 FastMixer_Static, // initialize if needed, then use all the time if initialized, 118 // multiplier is calculated based on min & max normal mixer buffer size 119 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 // FIXME for FastMixer_Dynamic: 122 // Supporting this option will require fixing HALs that can't handle large writes. 123 // For example, one HAL implementation returns an error from a large write, 124 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 125 // We could either fix the HAL implementations, or provide a wrapper that breaks 126 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 127} kUseFastMixer = FastMixer_Static; 128 129// Priorities for requestPriority 130static const int kPriorityAudioApp = 2; 131static const int kPriorityFastMixer = 3; 132 133// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 134// for the track. The client then sub-divides this into smaller buffers for its use. 135// Currently the client uses double-buffering by default, but doesn't tell us about that. 136// So for now we just assume that client is double-buffered. 137// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 138// N-buffering, so AudioFlinger could allocate the right amount of memory. 139// See the client's minBufCount and mNotificationFramesAct calculations for details. 140static const int kFastTrackMultiplier = 1; 141 142// ---------------------------------------------------------------------------- 143 144#ifdef ADD_BATTERY_DATA 145// To collect the amplifier usage 146static void addBatteryData(uint32_t params) { 147 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 148 if (service == NULL) { 149 // it already logged 150 return; 151 } 152 153 service->addBatteryData(params); 154} 155#endif 156 157 158// ---------------------------------------------------------------------------- 159// CPU Stats 160// ---------------------------------------------------------------------------- 161 162class CpuStats { 163public: 164 CpuStats(); 165 void sample(const String8 &title); 166#ifdef DEBUG_CPU_USAGE 167private: 168 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 169 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 170 171 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 172 173 int mCpuNum; // thread's current CPU number 174 int mCpukHz; // frequency of thread's current CPU in kHz 175#endif 176}; 177 178CpuStats::CpuStats() 179#ifdef DEBUG_CPU_USAGE 180 : mCpuNum(-1), mCpukHz(-1) 181#endif 182{ 183} 184 185void CpuStats::sample(const String8 &title) { 186#ifdef DEBUG_CPU_USAGE 187 // get current thread's delta CPU time in wall clock ns 188 double wcNs; 189 bool valid = mCpuUsage.sampleAndEnable(wcNs); 190 191 // record sample for wall clock statistics 192 if (valid) { 193 mWcStats.sample(wcNs); 194 } 195 196 // get the current CPU number 197 int cpuNum = sched_getcpu(); 198 199 // get the current CPU frequency in kHz 200 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 201 202 // check if either CPU number or frequency changed 203 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 204 mCpuNum = cpuNum; 205 mCpukHz = cpukHz; 206 // ignore sample for purposes of cycles 207 valid = false; 208 } 209 210 // if no change in CPU number or frequency, then record sample for cycle statistics 211 if (valid && mCpukHz > 0) { 212 double cycles = wcNs * cpukHz * 0.000001; 213 mHzStats.sample(cycles); 214 } 215 216 unsigned n = mWcStats.n(); 217 // mCpuUsage.elapsed() is expensive, so don't call it every loop 218 if ((n & 127) == 1) { 219 long long elapsed = mCpuUsage.elapsed(); 220 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 221 double perLoop = elapsed / (double) n; 222 double perLoop100 = perLoop * 0.01; 223 double perLoop1k = perLoop * 0.001; 224 double mean = mWcStats.mean(); 225 double stddev = mWcStats.stddev(); 226 double minimum = mWcStats.minimum(); 227 double maximum = mWcStats.maximum(); 228 double meanCycles = mHzStats.mean(); 229 double stddevCycles = mHzStats.stddev(); 230 double minCycles = mHzStats.minimum(); 231 double maxCycles = mHzStats.maximum(); 232 mCpuUsage.resetElapsed(); 233 mWcStats.reset(); 234 mHzStats.reset(); 235 ALOGD("CPU usage for %s over past %.1f secs\n" 236 " (%u mixer loops at %.1f mean ms per loop):\n" 237 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 238 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 239 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 240 title.string(), 241 elapsed * .000000001, n, perLoop * .000001, 242 mean * .001, 243 stddev * .001, 244 minimum * .001, 245 maximum * .001, 246 mean / perLoop100, 247 stddev / perLoop100, 248 minimum / perLoop100, 249 maximum / perLoop100, 250 meanCycles / perLoop1k, 251 stddevCycles / perLoop1k, 252 minCycles / perLoop1k, 253 maxCycles / perLoop1k); 254 255 } 256 } 257#endif 258}; 259 260// ---------------------------------------------------------------------------- 261// ThreadBase 262// ---------------------------------------------------------------------------- 263 264AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 265 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 266 : Thread(false /*canCallJava*/), 267 mType(type), 268 mAudioFlinger(audioFlinger), 269 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 270 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 271 mParamStatus(NO_ERROR), 272 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 273 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 274 // mName will be set by concrete (non-virtual) subclass 275 mDeathRecipient(new PMDeathRecipient(this)) 276{ 277} 278 279AudioFlinger::ThreadBase::~ThreadBase() 280{ 281 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 282 for (size_t i = 0; i < mConfigEvents.size(); i++) { 283 delete mConfigEvents[i]; 284 } 285 mConfigEvents.clear(); 286 287 mParamCond.broadcast(); 288 // do not lock the mutex in destructor 289 releaseWakeLock_l(); 290 if (mPowerManager != 0) { 291 sp<IBinder> binder = mPowerManager->asBinder(); 292 binder->unlinkToDeath(mDeathRecipient); 293 } 294} 295 296status_t AudioFlinger::ThreadBase::readyToRun() 297{ 298 status_t status = initCheck(); 299 if (status == NO_ERROR) { 300 ALOGI("AudioFlinger's thread %p ready to run", this); 301 } else { 302 ALOGE("No working audio driver found."); 303 } 304 return status; 305} 306 307void AudioFlinger::ThreadBase::exit() 308{ 309 ALOGV("ThreadBase::exit"); 310 // do any cleanup required for exit to succeed 311 preExit(); 312 { 313 // This lock prevents the following race in thread (uniprocessor for illustration): 314 // if (!exitPending()) { 315 // // context switch from here to exit() 316 // // exit() calls requestExit(), what exitPending() observes 317 // // exit() calls signal(), which is dropped since no waiters 318 // // context switch back from exit() to here 319 // mWaitWorkCV.wait(...); 320 // // now thread is hung 321 // } 322 AutoMutex lock(mLock); 323 requestExit(); 324 mWaitWorkCV.broadcast(); 325 } 326 // When Thread::requestExitAndWait is made virtual and this method is renamed to 327 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 328 requestExitAndWait(); 329} 330 331status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 332{ 333 status_t status; 334 335 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 336 Mutex::Autolock _l(mLock); 337 338 mNewParameters.add(keyValuePairs); 339 mWaitWorkCV.signal(); 340 // wait condition with timeout in case the thread loop has exited 341 // before the request could be processed 342 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 343 status = mParamStatus; 344 mWaitWorkCV.signal(); 345 } else { 346 status = TIMED_OUT; 347 } 348 return status; 349} 350 351void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 352{ 353 Mutex::Autolock _l(mLock); 354 sendIoConfigEvent_l(event, param); 355} 356 357// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 358void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 359{ 360 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 361 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 362 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 363 param); 364 mWaitWorkCV.signal(); 365} 366 367// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 368void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 369{ 370 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 371 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 372 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 373 mConfigEvents.size(), pid, tid, prio); 374 mWaitWorkCV.signal(); 375} 376 377void AudioFlinger::ThreadBase::processConfigEvents() 378{ 379 Mutex::Autolock _l(mLock); 380 processConfigEvents_l(); 381} 382 383// post condition: mConfigEvents.isEmpty() 384void AudioFlinger::ThreadBase::processConfigEvents_l() 385{ 386 while (!mConfigEvents.isEmpty()) { 387 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 388 ConfigEvent *event = mConfigEvents[0]; 389 mConfigEvents.removeAt(0); 390 // release mLock before locking AudioFlinger mLock: lock order is always 391 // AudioFlinger then ThreadBase to avoid cross deadlock 392 mLock.unlock(); 393 switch (event->type()) { 394 case CFG_EVENT_PRIO: { 395 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 396 // FIXME Need to understand why this has be done asynchronously 397 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 398 true /*asynchronous*/); 399 if (err != 0) { 400 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 401 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 402 } 403 } break; 404 case CFG_EVENT_IO: { 405 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 406 { 407 Mutex::Autolock _l(mAudioFlinger->mLock); 408 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 409 } 410 } break; 411 default: 412 ALOGE("processConfigEvents() unknown event type %d", event->type()); 413 break; 414 } 415 delete event; 416 mLock.lock(); 417 } 418} 419 420void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 421{ 422 const size_t SIZE = 256; 423 char buffer[SIZE]; 424 String8 result; 425 426 bool locked = AudioFlinger::dumpTryLock(mLock); 427 if (!locked) { 428 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 429 write(fd, buffer, strlen(buffer)); 430 } 431 432 snprintf(buffer, SIZE, "io handle: %d\n", mId); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 437 result.append(buffer); 438 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 439 result.append(buffer); 440 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 441 result.append(buffer); 442 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 443 result.append(buffer); 444 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 445 result.append(buffer); 446 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 447 result.append(buffer); 448 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 449 result.append(buffer); 450 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 451 result.append(buffer); 452 453 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 454 result.append(buffer); 455 result.append(" Index Command"); 456 for (size_t i = 0; i < mNewParameters.size(); ++i) { 457 snprintf(buffer, SIZE, "\n %02d ", i); 458 result.append(buffer); 459 result.append(mNewParameters[i]); 460 } 461 462 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 463 result.append(buffer); 464 for (size_t i = 0; i < mConfigEvents.size(); i++) { 465 mConfigEvents[i]->dump(buffer, SIZE); 466 result.append(buffer); 467 } 468 result.append("\n"); 469 470 write(fd, result.string(), result.size()); 471 472 if (locked) { 473 mLock.unlock(); 474 } 475} 476 477void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 478{ 479 const size_t SIZE = 256; 480 char buffer[SIZE]; 481 String8 result; 482 483 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 484 write(fd, buffer, strlen(buffer)); 485 486 for (size_t i = 0; i < mEffectChains.size(); ++i) { 487 sp<EffectChain> chain = mEffectChains[i]; 488 if (chain != 0) { 489 chain->dump(fd, args); 490 } 491 } 492} 493 494void AudioFlinger::ThreadBase::acquireWakeLock() 495{ 496 Mutex::Autolock _l(mLock); 497 acquireWakeLock_l(); 498} 499 500void AudioFlinger::ThreadBase::acquireWakeLock_l() 501{ 502 if (mPowerManager == 0) { 503 // use checkService() to avoid blocking if power service is not up yet 504 sp<IBinder> binder = 505 defaultServiceManager()->checkService(String16("power")); 506 if (binder == 0) { 507 ALOGW("Thread %s cannot connect to the power manager service", mName); 508 } else { 509 mPowerManager = interface_cast<IPowerManager>(binder); 510 binder->linkToDeath(mDeathRecipient); 511 } 512 } 513 if (mPowerManager != 0) { 514 sp<IBinder> binder = new BBinder(); 515 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 516 binder, 517 String16(mName), 518 String16("media")); 519 if (status == NO_ERROR) { 520 mWakeLockToken = binder; 521 } 522 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 523 } 524} 525 526void AudioFlinger::ThreadBase::releaseWakeLock() 527{ 528 Mutex::Autolock _l(mLock); 529 releaseWakeLock_l(); 530} 531 532void AudioFlinger::ThreadBase::releaseWakeLock_l() 533{ 534 if (mWakeLockToken != 0) { 535 ALOGV("releaseWakeLock_l() %s", mName); 536 if (mPowerManager != 0) { 537 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 538 } 539 mWakeLockToken.clear(); 540 } 541} 542 543void AudioFlinger::ThreadBase::clearPowerManager() 544{ 545 Mutex::Autolock _l(mLock); 546 releaseWakeLock_l(); 547 mPowerManager.clear(); 548} 549 550void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 551{ 552 sp<ThreadBase> thread = mThread.promote(); 553 if (thread != 0) { 554 thread->clearPowerManager(); 555 } 556 ALOGW("power manager service died !!!"); 557} 558 559void AudioFlinger::ThreadBase::setEffectSuspended( 560 const effect_uuid_t *type, bool suspend, int sessionId) 561{ 562 Mutex::Autolock _l(mLock); 563 setEffectSuspended_l(type, suspend, sessionId); 564} 565 566void AudioFlinger::ThreadBase::setEffectSuspended_l( 567 const effect_uuid_t *type, bool suspend, int sessionId) 568{ 569 sp<EffectChain> chain = getEffectChain_l(sessionId); 570 if (chain != 0) { 571 if (type != NULL) { 572 chain->setEffectSuspended_l(type, suspend); 573 } else { 574 chain->setEffectSuspendedAll_l(suspend); 575 } 576 } 577 578 updateSuspendedSessions_l(type, suspend, sessionId); 579} 580 581void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 582{ 583 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 584 if (index < 0) { 585 return; 586 } 587 588 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 589 mSuspendedSessions.valueAt(index); 590 591 for (size_t i = 0; i < sessionEffects.size(); i++) { 592 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 593 for (int j = 0; j < desc->mRefCount; j++) { 594 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 595 chain->setEffectSuspendedAll_l(true); 596 } else { 597 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 598 desc->mType.timeLow); 599 chain->setEffectSuspended_l(&desc->mType, true); 600 } 601 } 602 } 603} 604 605void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 606 bool suspend, 607 int sessionId) 608{ 609 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 610 611 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 612 613 if (suspend) { 614 if (index >= 0) { 615 sessionEffects = mSuspendedSessions.valueAt(index); 616 } else { 617 mSuspendedSessions.add(sessionId, sessionEffects); 618 } 619 } else { 620 if (index < 0) { 621 return; 622 } 623 sessionEffects = mSuspendedSessions.valueAt(index); 624 } 625 626 627 int key = EffectChain::kKeyForSuspendAll; 628 if (type != NULL) { 629 key = type->timeLow; 630 } 631 index = sessionEffects.indexOfKey(key); 632 633 sp<SuspendedSessionDesc> desc; 634 if (suspend) { 635 if (index >= 0) { 636 desc = sessionEffects.valueAt(index); 637 } else { 638 desc = new SuspendedSessionDesc(); 639 if (type != NULL) { 640 desc->mType = *type; 641 } 642 sessionEffects.add(key, desc); 643 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 644 } 645 desc->mRefCount++; 646 } else { 647 if (index < 0) { 648 return; 649 } 650 desc = sessionEffects.valueAt(index); 651 if (--desc->mRefCount == 0) { 652 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 653 sessionEffects.removeItemsAt(index); 654 if (sessionEffects.isEmpty()) { 655 ALOGV("updateSuspendedSessions_l() restore removing session %d", 656 sessionId); 657 mSuspendedSessions.removeItem(sessionId); 658 } 659 } 660 } 661 if (!sessionEffects.isEmpty()) { 662 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 663 } 664} 665 666void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 667 bool enabled, 668 int sessionId) 669{ 670 Mutex::Autolock _l(mLock); 671 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 672} 673 674void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 675 bool enabled, 676 int sessionId) 677{ 678 if (mType != RECORD) { 679 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 680 // another session. This gives the priority to well behaved effect control panels 681 // and applications not using global effects. 682 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 683 // global effects 684 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 685 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 686 } 687 } 688 689 sp<EffectChain> chain = getEffectChain_l(sessionId); 690 if (chain != 0) { 691 chain->checkSuspendOnEffectEnabled(effect, enabled); 692 } 693} 694 695// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 696sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 697 const sp<AudioFlinger::Client>& client, 698 const sp<IEffectClient>& effectClient, 699 int32_t priority, 700 int sessionId, 701 effect_descriptor_t *desc, 702 int *enabled, 703 status_t *status) 704{ 705 sp<EffectModule> effect; 706 sp<EffectHandle> handle; 707 status_t lStatus; 708 sp<EffectChain> chain; 709 bool chainCreated = false; 710 bool effectCreated = false; 711 bool effectRegistered = false; 712 713 lStatus = initCheck(); 714 if (lStatus != NO_ERROR) { 715 ALOGW("createEffect_l() Audio driver not initialized."); 716 goto Exit; 717 } 718 719 // Do not allow effects with session ID 0 on direct output or duplicating threads 720 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 721 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 722 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 723 desc->name, sessionId); 724 lStatus = BAD_VALUE; 725 goto Exit; 726 } 727 // Only Pre processor effects are allowed on input threads and only on input threads 728 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 729 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 730 desc->name, desc->flags, mType); 731 lStatus = BAD_VALUE; 732 goto Exit; 733 } 734 735 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 736 737 { // scope for mLock 738 Mutex::Autolock _l(mLock); 739 740 // check for existing effect chain with the requested audio session 741 chain = getEffectChain_l(sessionId); 742 if (chain == 0) { 743 // create a new chain for this session 744 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 745 chain = new EffectChain(this, sessionId); 746 addEffectChain_l(chain); 747 chain->setStrategy(getStrategyForSession_l(sessionId)); 748 chainCreated = true; 749 } else { 750 effect = chain->getEffectFromDesc_l(desc); 751 } 752 753 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 754 755 if (effect == 0) { 756 int id = mAudioFlinger->nextUniqueId(); 757 // Check CPU and memory usage 758 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 759 if (lStatus != NO_ERROR) { 760 goto Exit; 761 } 762 effectRegistered = true; 763 // create a new effect module if none present in the chain 764 effect = new EffectModule(this, chain, desc, id, sessionId); 765 lStatus = effect->status(); 766 if (lStatus != NO_ERROR) { 767 goto Exit; 768 } 769 lStatus = chain->addEffect_l(effect); 770 if (lStatus != NO_ERROR) { 771 goto Exit; 772 } 773 effectCreated = true; 774 775 effect->setDevice(mOutDevice); 776 effect->setDevice(mInDevice); 777 effect->setMode(mAudioFlinger->getMode()); 778 effect->setAudioSource(mAudioSource); 779 } 780 // create effect handle and connect it to effect module 781 handle = new EffectHandle(effect, client, effectClient, priority); 782 lStatus = effect->addHandle(handle.get()); 783 if (enabled != NULL) { 784 *enabled = (int)effect->isEnabled(); 785 } 786 } 787 788Exit: 789 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 790 Mutex::Autolock _l(mLock); 791 if (effectCreated) { 792 chain->removeEffect_l(effect); 793 } 794 if (effectRegistered) { 795 AudioSystem::unregisterEffect(effect->id()); 796 } 797 if (chainCreated) { 798 removeEffectChain_l(chain); 799 } 800 handle.clear(); 801 } 802 803 *status = lStatus; 804 return handle; 805} 806 807sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 808{ 809 Mutex::Autolock _l(mLock); 810 return getEffect_l(sessionId, effectId); 811} 812 813sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 814{ 815 sp<EffectChain> chain = getEffectChain_l(sessionId); 816 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 817} 818 819// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 820// PlaybackThread::mLock held 821status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 822{ 823 // check for existing effect chain with the requested audio session 824 int sessionId = effect->sessionId(); 825 sp<EffectChain> chain = getEffectChain_l(sessionId); 826 bool chainCreated = false; 827 828 if (chain == 0) { 829 // create a new chain for this session 830 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 831 chain = new EffectChain(this, sessionId); 832 addEffectChain_l(chain); 833 chain->setStrategy(getStrategyForSession_l(sessionId)); 834 chainCreated = true; 835 } 836 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 837 838 if (chain->getEffectFromId_l(effect->id()) != 0) { 839 ALOGW("addEffect_l() %p effect %s already present in chain %p", 840 this, effect->desc().name, chain.get()); 841 return BAD_VALUE; 842 } 843 844 status_t status = chain->addEffect_l(effect); 845 if (status != NO_ERROR) { 846 if (chainCreated) { 847 removeEffectChain_l(chain); 848 } 849 return status; 850 } 851 852 effect->setDevice(mOutDevice); 853 effect->setDevice(mInDevice); 854 effect->setMode(mAudioFlinger->getMode()); 855 effect->setAudioSource(mAudioSource); 856 return NO_ERROR; 857} 858 859void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 860 861 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 862 effect_descriptor_t desc = effect->desc(); 863 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 864 detachAuxEffect_l(effect->id()); 865 } 866 867 sp<EffectChain> chain = effect->chain().promote(); 868 if (chain != 0) { 869 // remove effect chain if removing last effect 870 if (chain->removeEffect_l(effect) == 0) { 871 removeEffectChain_l(chain); 872 } 873 } else { 874 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 875 } 876} 877 878void AudioFlinger::ThreadBase::lockEffectChains_l( 879 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 880{ 881 effectChains = mEffectChains; 882 for (size_t i = 0; i < mEffectChains.size(); i++) { 883 mEffectChains[i]->lock(); 884 } 885} 886 887void AudioFlinger::ThreadBase::unlockEffectChains( 888 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 889{ 890 for (size_t i = 0; i < effectChains.size(); i++) { 891 effectChains[i]->unlock(); 892 } 893} 894 895sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 896{ 897 Mutex::Autolock _l(mLock); 898 return getEffectChain_l(sessionId); 899} 900 901sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 902{ 903 size_t size = mEffectChains.size(); 904 for (size_t i = 0; i < size; i++) { 905 if (mEffectChains[i]->sessionId() == sessionId) { 906 return mEffectChains[i]; 907 } 908 } 909 return 0; 910} 911 912void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 913{ 914 Mutex::Autolock _l(mLock); 915 size_t size = mEffectChains.size(); 916 for (size_t i = 0; i < size; i++) { 917 mEffectChains[i]->setMode_l(mode); 918 } 919} 920 921void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 922 EffectHandle *handle, 923 bool unpinIfLast) { 924 925 Mutex::Autolock _l(mLock); 926 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 927 // delete the effect module if removing last handle on it 928 if (effect->removeHandle(handle) == 0) { 929 if (!effect->isPinned() || unpinIfLast) { 930 removeEffect_l(effect); 931 AudioSystem::unregisterEffect(effect->id()); 932 } 933 } 934} 935 936// ---------------------------------------------------------------------------- 937// Playback 938// ---------------------------------------------------------------------------- 939 940AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 941 AudioStreamOut* output, 942 audio_io_handle_t id, 943 audio_devices_t device, 944 type_t type) 945 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 946 mNormalFrameCount(0), mMixBuffer(NULL), 947 mSuspended(0), mBytesWritten(0), 948 // mStreamTypes[] initialized in constructor body 949 mOutput(output), 950 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 951 mMixerStatus(MIXER_IDLE), 952 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 953 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 954 mBytesRemaining(0), 955 mCurrentWriteLength(0), 956 mUseAsyncWrite(false), 957 mWriteAckSequence(0), 958 mDrainSequence(0), 959 mScreenState(AudioFlinger::mScreenState), 960 // index 0 is reserved for normal mixer's submix 961 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 962 // mLatchD, mLatchQ, 963 mLatchDValid(false), mLatchQValid(false) 964{ 965 snprintf(mName, kNameLength, "AudioOut_%X", id); 966 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 967 968 // Assumes constructor is called by AudioFlinger with it's mLock held, but 969 // it would be safer to explicitly pass initial masterVolume/masterMute as 970 // parameter. 971 // 972 // If the HAL we are using has support for master volume or master mute, 973 // then do not attenuate or mute during mixing (just leave the volume at 1.0 974 // and the mute set to false). 975 mMasterVolume = audioFlinger->masterVolume_l(); 976 mMasterMute = audioFlinger->masterMute_l(); 977 if (mOutput && mOutput->audioHwDev) { 978 if (mOutput->audioHwDev->canSetMasterVolume()) { 979 mMasterVolume = 1.0; 980 } 981 982 if (mOutput->audioHwDev->canSetMasterMute()) { 983 mMasterMute = false; 984 } 985 } 986 987 readOutputParameters(); 988 989 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 990 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 991 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 992 stream = (audio_stream_type_t) (stream + 1)) { 993 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 994 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 995 } 996 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 997 // because mAudioFlinger doesn't have one to copy from 998} 999 1000AudioFlinger::PlaybackThread::~PlaybackThread() 1001{ 1002 mAudioFlinger->unregisterWriter(mNBLogWriter); 1003 delete[] mMixBuffer; 1004} 1005 1006void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1007{ 1008 dumpInternals(fd, args); 1009 dumpTracks(fd, args); 1010 dumpEffectChains(fd, args); 1011} 1012 1013void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1014{ 1015 const size_t SIZE = 256; 1016 char buffer[SIZE]; 1017 String8 result; 1018 1019 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1020 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1021 const stream_type_t *st = &mStreamTypes[i]; 1022 if (i > 0) { 1023 result.appendFormat(", "); 1024 } 1025 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1026 if (st->mute) { 1027 result.append("M"); 1028 } 1029 } 1030 result.append("\n"); 1031 write(fd, result.string(), result.length()); 1032 result.clear(); 1033 1034 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1035 result.append(buffer); 1036 Track::appendDumpHeader(result); 1037 for (size_t i = 0; i < mTracks.size(); ++i) { 1038 sp<Track> track = mTracks[i]; 1039 if (track != 0) { 1040 track->dump(buffer, SIZE); 1041 result.append(buffer); 1042 } 1043 } 1044 1045 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1046 result.append(buffer); 1047 Track::appendDumpHeader(result); 1048 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1049 sp<Track> track = mActiveTracks[i].promote(); 1050 if (track != 0) { 1051 track->dump(buffer, SIZE); 1052 result.append(buffer); 1053 } 1054 } 1055 write(fd, result.string(), result.size()); 1056 1057 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1058 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1059 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1060 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1061} 1062 1063void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1064{ 1065 const size_t SIZE = 256; 1066 char buffer[SIZE]; 1067 String8 result; 1068 1069 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1070 result.append(buffer); 1071 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1072 result.append(buffer); 1073 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1074 ns2ms(systemTime() - mLastWriteTime)); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1079 result.append(buffer); 1080 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1081 result.append(buffer); 1082 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1083 result.append(buffer); 1084 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1085 result.append(buffer); 1086 write(fd, result.string(), result.size()); 1087 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1088 1089 dumpBase(fd, args); 1090} 1091 1092// Thread virtuals 1093 1094void AudioFlinger::PlaybackThread::onFirstRef() 1095{ 1096 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1097} 1098 1099// ThreadBase virtuals 1100void AudioFlinger::PlaybackThread::preExit() 1101{ 1102 ALOGV(" preExit()"); 1103 // FIXME this is using hard-coded strings but in the future, this functionality will be 1104 // converted to use audio HAL extensions required to support tunneling 1105 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1106} 1107 1108// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1109sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1110 const sp<AudioFlinger::Client>& client, 1111 audio_stream_type_t streamType, 1112 uint32_t sampleRate, 1113 audio_format_t format, 1114 audio_channel_mask_t channelMask, 1115 size_t frameCount, 1116 const sp<IMemory>& sharedBuffer, 1117 int sessionId, 1118 IAudioFlinger::track_flags_t *flags, 1119 pid_t tid, 1120 status_t *status) 1121{ 1122 sp<Track> track; 1123 status_t lStatus; 1124 1125 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1126 1127 // client expresses a preference for FAST, but we get the final say 1128 if (*flags & IAudioFlinger::TRACK_FAST) { 1129 if ( 1130 // not timed 1131 (!isTimed) && 1132 // either of these use cases: 1133 ( 1134 // use case 1: shared buffer with any frame count 1135 ( 1136 (sharedBuffer != 0) 1137 ) || 1138 // use case 2: callback handler and frame count is default or at least as large as HAL 1139 ( 1140 (tid != -1) && 1141 ((frameCount == 0) || 1142 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1143 ) 1144 ) && 1145 // PCM data 1146 audio_is_linear_pcm(format) && 1147 // mono or stereo 1148 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1149 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1150#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1151 // hardware sample rate 1152 (sampleRate == mSampleRate) && 1153#endif 1154 // normal mixer has an associated fast mixer 1155 hasFastMixer() && 1156 // there are sufficient fast track slots available 1157 (mFastTrackAvailMask != 0) 1158 // FIXME test that MixerThread for this fast track has a capable output HAL 1159 // FIXME add a permission test also? 1160 ) { 1161 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1162 if (frameCount == 0) { 1163 frameCount = mFrameCount * kFastTrackMultiplier; 1164 } 1165 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1166 frameCount, mFrameCount); 1167 } else { 1168 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1169 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1170 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1171 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1172 audio_is_linear_pcm(format), 1173 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1174 *flags &= ~IAudioFlinger::TRACK_FAST; 1175 // For compatibility with AudioTrack calculation, buffer depth is forced 1176 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1177 // This is probably too conservative, but legacy application code may depend on it. 1178 // If you change this calculation, also review the start threshold which is related. 1179 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1180 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1181 if (minBufCount < 2) { 1182 minBufCount = 2; 1183 } 1184 size_t minFrameCount = mNormalFrameCount * minBufCount; 1185 if (frameCount < minFrameCount) { 1186 frameCount = minFrameCount; 1187 } 1188 } 1189 } 1190 1191 if (mType == DIRECT) { 1192 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1193 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1194 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1195 "for output %p with format %d", 1196 sampleRate, format, channelMask, mOutput, mFormat); 1197 lStatus = BAD_VALUE; 1198 goto Exit; 1199 } 1200 } 1201 } else if (mType == OFFLOAD) { 1202 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1203 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1204 "for output %p with format %d", 1205 sampleRate, format, channelMask, mOutput, mFormat); 1206 lStatus = BAD_VALUE; 1207 goto Exit; 1208 } 1209 } else { 1210 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1211 ALOGE("createTrack_l() Bad parameter: format %d \"" 1212 "for output %p with format %d", 1213 format, mOutput, mFormat); 1214 lStatus = BAD_VALUE; 1215 goto Exit; 1216 } 1217 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1218 if (sampleRate > mSampleRate*2) { 1219 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1220 lStatus = BAD_VALUE; 1221 goto Exit; 1222 } 1223 } 1224 1225 lStatus = initCheck(); 1226 if (lStatus != NO_ERROR) { 1227 ALOGE("Audio driver not initialized."); 1228 goto Exit; 1229 } 1230 1231 { // scope for mLock 1232 Mutex::Autolock _l(mLock); 1233 1234 // all tracks in same audio session must share the same routing strategy otherwise 1235 // conflicts will happen when tracks are moved from one output to another by audio policy 1236 // manager 1237 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1238 for (size_t i = 0; i < mTracks.size(); ++i) { 1239 sp<Track> t = mTracks[i]; 1240 if (t != 0 && !t->isOutputTrack()) { 1241 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1242 if (sessionId == t->sessionId() && strategy != actual) { 1243 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1244 strategy, actual); 1245 lStatus = BAD_VALUE; 1246 goto Exit; 1247 } 1248 } 1249 } 1250 1251 if (!isTimed) { 1252 track = new Track(this, client, streamType, sampleRate, format, 1253 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1254 } else { 1255 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1256 channelMask, frameCount, sharedBuffer, sessionId); 1257 } 1258 1259 // new Track always returns non-NULL, 1260 // but TimedTrack::create() is a factory that could fail by returning NULL 1261 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1262 if (lStatus != NO_ERROR) { 1263 track.clear(); 1264 goto Exit; 1265 } 1266 1267 mTracks.add(track); 1268 1269 sp<EffectChain> chain = getEffectChain_l(sessionId); 1270 if (chain != 0) { 1271 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1272 track->setMainBuffer(chain->inBuffer()); 1273 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1274 chain->incTrackCnt(); 1275 } 1276 1277 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1278 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1279 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1280 // so ask activity manager to do this on our behalf 1281 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1282 } 1283 } 1284 1285 lStatus = NO_ERROR; 1286 1287Exit: 1288 *status = lStatus; 1289 return track; 1290} 1291 1292uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1293{ 1294 return latency; 1295} 1296 1297uint32_t AudioFlinger::PlaybackThread::latency() const 1298{ 1299 Mutex::Autolock _l(mLock); 1300 return latency_l(); 1301} 1302uint32_t AudioFlinger::PlaybackThread::latency_l() const 1303{ 1304 if (initCheck() == NO_ERROR) { 1305 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1306 } else { 1307 return 0; 1308 } 1309} 1310 1311void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1312{ 1313 Mutex::Autolock _l(mLock); 1314 // Don't apply master volume in SW if our HAL can do it for us. 1315 if (mOutput && mOutput->audioHwDev && 1316 mOutput->audioHwDev->canSetMasterVolume()) { 1317 mMasterVolume = 1.0; 1318 } else { 1319 mMasterVolume = value; 1320 } 1321} 1322 1323void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1324{ 1325 Mutex::Autolock _l(mLock); 1326 // Don't apply master mute in SW if our HAL can do it for us. 1327 if (mOutput && mOutput->audioHwDev && 1328 mOutput->audioHwDev->canSetMasterMute()) { 1329 mMasterMute = false; 1330 } else { 1331 mMasterMute = muted; 1332 } 1333} 1334 1335void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1336{ 1337 Mutex::Autolock _l(mLock); 1338 mStreamTypes[stream].volume = value; 1339 signal_l(); 1340} 1341 1342void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1343{ 1344 Mutex::Autolock _l(mLock); 1345 mStreamTypes[stream].mute = muted; 1346 signal_l(); 1347} 1348 1349float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1350{ 1351 Mutex::Autolock _l(mLock); 1352 return mStreamTypes[stream].volume; 1353} 1354 1355// addTrack_l() must be called with ThreadBase::mLock held 1356status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1357{ 1358 status_t status = ALREADY_EXISTS; 1359 1360 // set retry count for buffer fill 1361 track->mRetryCount = kMaxTrackStartupRetries; 1362 if (mActiveTracks.indexOf(track) < 0) { 1363 // the track is newly added, make sure it fills up all its 1364 // buffers before playing. This is to ensure the client will 1365 // effectively get the latency it requested. 1366 if (!track->isOutputTrack()) { 1367 TrackBase::track_state state = track->mState; 1368 mLock.unlock(); 1369 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1370 mLock.lock(); 1371 // abort track was stopped/paused while we released the lock 1372 if (state != track->mState) { 1373 if (status == NO_ERROR) { 1374 mLock.unlock(); 1375 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1376 mLock.lock(); 1377 } 1378 return INVALID_OPERATION; 1379 } 1380 // abort if start is rejected by audio policy manager 1381 if (status != NO_ERROR) { 1382 return PERMISSION_DENIED; 1383 } 1384#ifdef ADD_BATTERY_DATA 1385 // to track the speaker usage 1386 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1387#endif 1388 } 1389 1390 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1391 track->mResetDone = false; 1392 track->mPresentationCompleteFrames = 0; 1393 mActiveTracks.add(track); 1394 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1395 if (chain != 0) { 1396 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1397 track->sessionId()); 1398 chain->incActiveTrackCnt(); 1399 } 1400 1401 status = NO_ERROR; 1402 } 1403 1404 ALOGV("mWaitWorkCV.broadcast"); 1405 mWaitWorkCV.broadcast(); 1406 1407 return status; 1408} 1409 1410bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1411{ 1412 track->terminate(); 1413 // active tracks are removed by threadLoop() 1414 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1415 track->mState = TrackBase::STOPPED; 1416 if (!trackActive) { 1417 removeTrack_l(track); 1418 } else if (track->isFastTrack() || track->isOffloaded()) { 1419 track->mState = TrackBase::STOPPING_1; 1420 } 1421 1422 return trackActive; 1423} 1424 1425void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1426{ 1427 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1428 mTracks.remove(track); 1429 deleteTrackName_l(track->name()); 1430 // redundant as track is about to be destroyed, for dumpsys only 1431 track->mName = -1; 1432 if (track->isFastTrack()) { 1433 int index = track->mFastIndex; 1434 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1435 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1436 mFastTrackAvailMask |= 1 << index; 1437 // redundant as track is about to be destroyed, for dumpsys only 1438 track->mFastIndex = -1; 1439 } 1440 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1441 if (chain != 0) { 1442 chain->decTrackCnt(); 1443 } 1444} 1445 1446void AudioFlinger::PlaybackThread::signal_l() 1447{ 1448 // Thread could be blocked waiting for async 1449 // so signal it to handle state changes immediately 1450 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1451 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1452 mSignalPending = true; 1453 mWaitWorkCV.signal(); 1454} 1455 1456String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1457{ 1458 Mutex::Autolock _l(mLock); 1459 if (initCheck() != NO_ERROR) { 1460 return String8(); 1461 } 1462 1463 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1464 const String8 out_s8(s); 1465 free(s); 1466 return out_s8; 1467} 1468 1469// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1470void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1471 AudioSystem::OutputDescriptor desc; 1472 void *param2 = NULL; 1473 1474 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1475 param); 1476 1477 switch (event) { 1478 case AudioSystem::OUTPUT_OPENED: 1479 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1480 desc.channelMask = mChannelMask; 1481 desc.samplingRate = mSampleRate; 1482 desc.format = mFormat; 1483 desc.frameCount = mNormalFrameCount; // FIXME see 1484 // AudioFlinger::frameCount(audio_io_handle_t) 1485 desc.latency = latency(); 1486 param2 = &desc; 1487 break; 1488 1489 case AudioSystem::STREAM_CONFIG_CHANGED: 1490 param2 = ¶m; 1491 case AudioSystem::OUTPUT_CLOSED: 1492 default: 1493 break; 1494 } 1495 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1496} 1497 1498void AudioFlinger::PlaybackThread::writeCallback() 1499{ 1500 ALOG_ASSERT(mCallbackThread != 0); 1501 mCallbackThread->resetWriteBlocked(); 1502} 1503 1504void AudioFlinger::PlaybackThread::drainCallback() 1505{ 1506 ALOG_ASSERT(mCallbackThread != 0); 1507 mCallbackThread->resetDraining(); 1508} 1509 1510void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1511{ 1512 Mutex::Autolock _l(mLock); 1513 // reject out of sequence requests 1514 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1515 mWriteAckSequence &= ~1; 1516 mWaitWorkCV.signal(); 1517 } 1518} 1519 1520void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1521{ 1522 Mutex::Autolock _l(mLock); 1523 // reject out of sequence requests 1524 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1525 mDrainSequence &= ~1; 1526 mWaitWorkCV.signal(); 1527 } 1528} 1529 1530// static 1531int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1532 void *param, 1533 void *cookie) 1534{ 1535 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1536 ALOGV("asyncCallback() event %d", event); 1537 switch (event) { 1538 case STREAM_CBK_EVENT_WRITE_READY: 1539 me->writeCallback(); 1540 break; 1541 case STREAM_CBK_EVENT_DRAIN_READY: 1542 me->drainCallback(); 1543 break; 1544 default: 1545 ALOGW("asyncCallback() unknown event %d", event); 1546 break; 1547 } 1548 return 0; 1549} 1550 1551void AudioFlinger::PlaybackThread::readOutputParameters() 1552{ 1553 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1554 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1555 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1556 if (!audio_is_output_channel(mChannelMask)) { 1557 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1558 } 1559 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1560 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1561 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1562 } 1563 mChannelCount = popcount(mChannelMask); 1564 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1565 if (!audio_is_valid_format(mFormat)) { 1566 LOG_FATAL("HAL format %d not valid for output", mFormat); 1567 } 1568 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1569 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1570 mFormat); 1571 } 1572 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1573 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1574 mFrameCount = mBufferSize / mFrameSize; 1575 if (mFrameCount & 15) { 1576 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1577 mFrameCount); 1578 } 1579 1580 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1581 (mOutput->stream->set_callback != NULL)) { 1582 if (mOutput->stream->set_callback(mOutput->stream, 1583 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1584 mUseAsyncWrite = true; 1585 } 1586 } 1587 1588 // Calculate size of normal mix buffer relative to the HAL output buffer size 1589 double multiplier = 1.0; 1590 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1591 kUseFastMixer == FastMixer_Dynamic)) { 1592 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1593 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1594 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1595 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1596 maxNormalFrameCount = maxNormalFrameCount & ~15; 1597 if (maxNormalFrameCount < minNormalFrameCount) { 1598 maxNormalFrameCount = minNormalFrameCount; 1599 } 1600 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1601 if (multiplier <= 1.0) { 1602 multiplier = 1.0; 1603 } else if (multiplier <= 2.0) { 1604 if (2 * mFrameCount <= maxNormalFrameCount) { 1605 multiplier = 2.0; 1606 } else { 1607 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1608 } 1609 } else { 1610 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1611 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1612 // track, but we sometimes have to do this to satisfy the maximum frame count 1613 // constraint) 1614 // FIXME this rounding up should not be done if no HAL SRC 1615 uint32_t truncMult = (uint32_t) multiplier; 1616 if ((truncMult & 1)) { 1617 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1618 ++truncMult; 1619 } 1620 } 1621 multiplier = (double) truncMult; 1622 } 1623 } 1624 mNormalFrameCount = multiplier * mFrameCount; 1625 // round up to nearest 16 frames to satisfy AudioMixer 1626 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1627 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1628 mNormalFrameCount); 1629 1630 delete[] mMixBuffer; 1631 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1632 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1633 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1634 memset(mMixBuffer, 0, normalBufferSize); 1635 1636 // force reconfiguration of effect chains and engines to take new buffer size and audio 1637 // parameters into account 1638 // Note that mLock is not held when readOutputParameters() is called from the constructor 1639 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1640 // matter. 1641 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1642 Vector< sp<EffectChain> > effectChains = mEffectChains; 1643 for (size_t i = 0; i < effectChains.size(); i ++) { 1644 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1645 } 1646} 1647 1648 1649status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1650{ 1651 if (halFrames == NULL || dspFrames == NULL) { 1652 return BAD_VALUE; 1653 } 1654 Mutex::Autolock _l(mLock); 1655 if (initCheck() != NO_ERROR) { 1656 return INVALID_OPERATION; 1657 } 1658 size_t framesWritten = mBytesWritten / mFrameSize; 1659 *halFrames = framesWritten; 1660 1661 if (isSuspended()) { 1662 // return an estimation of rendered frames when the output is suspended 1663 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1664 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1665 return NO_ERROR; 1666 } else { 1667 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1668 } 1669} 1670 1671uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1672{ 1673 Mutex::Autolock _l(mLock); 1674 uint32_t result = 0; 1675 if (getEffectChain_l(sessionId) != 0) { 1676 result = EFFECT_SESSION; 1677 } 1678 1679 for (size_t i = 0; i < mTracks.size(); ++i) { 1680 sp<Track> track = mTracks[i]; 1681 if (sessionId == track->sessionId() && !track->isInvalid()) { 1682 result |= TRACK_SESSION; 1683 break; 1684 } 1685 } 1686 1687 return result; 1688} 1689 1690uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1691{ 1692 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1693 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1694 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1695 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1696 } 1697 for (size_t i = 0; i < mTracks.size(); i++) { 1698 sp<Track> track = mTracks[i]; 1699 if (sessionId == track->sessionId() && !track->isInvalid()) { 1700 return AudioSystem::getStrategyForStream(track->streamType()); 1701 } 1702 } 1703 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1704} 1705 1706 1707AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1708{ 1709 Mutex::Autolock _l(mLock); 1710 return mOutput; 1711} 1712 1713AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1714{ 1715 Mutex::Autolock _l(mLock); 1716 AudioStreamOut *output = mOutput; 1717 mOutput = NULL; 1718 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1719 // must push a NULL and wait for ack 1720 mOutputSink.clear(); 1721 mPipeSink.clear(); 1722 mNormalSink.clear(); 1723 return output; 1724} 1725 1726// this method must always be called either with ThreadBase mLock held or inside the thread loop 1727audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1728{ 1729 if (mOutput == NULL) { 1730 return NULL; 1731 } 1732 return &mOutput->stream->common; 1733} 1734 1735uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1736{ 1737 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1738} 1739 1740status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1741{ 1742 if (!isValidSyncEvent(event)) { 1743 return BAD_VALUE; 1744 } 1745 1746 Mutex::Autolock _l(mLock); 1747 1748 for (size_t i = 0; i < mTracks.size(); ++i) { 1749 sp<Track> track = mTracks[i]; 1750 if (event->triggerSession() == track->sessionId()) { 1751 (void) track->setSyncEvent(event); 1752 return NO_ERROR; 1753 } 1754 } 1755 1756 return NAME_NOT_FOUND; 1757} 1758 1759bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1760{ 1761 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1762} 1763 1764void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1765 const Vector< sp<Track> >& tracksToRemove) 1766{ 1767 size_t count = tracksToRemove.size(); 1768 if (count > 0) { 1769 for (size_t i = 0 ; i < count ; i++) { 1770 const sp<Track>& track = tracksToRemove.itemAt(i); 1771 if (!track->isOutputTrack()) { 1772 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1773#ifdef ADD_BATTERY_DATA 1774 // to track the speaker usage 1775 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1776#endif 1777 if (track->isTerminated()) { 1778 AudioSystem::releaseOutput(mId); 1779 } 1780 } 1781 } 1782 } 1783} 1784 1785void AudioFlinger::PlaybackThread::checkSilentMode_l() 1786{ 1787 if (!mMasterMute) { 1788 char value[PROPERTY_VALUE_MAX]; 1789 if (property_get("ro.audio.silent", value, "0") > 0) { 1790 char *endptr; 1791 unsigned long ul = strtoul(value, &endptr, 0); 1792 if (*endptr == '\0' && ul != 0) { 1793 ALOGD("Silence is golden"); 1794 // The setprop command will not allow a property to be changed after 1795 // the first time it is set, so we don't have to worry about un-muting. 1796 setMasterMute_l(true); 1797 } 1798 } 1799 } 1800} 1801 1802// shared by MIXER and DIRECT, overridden by DUPLICATING 1803ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1804{ 1805 // FIXME rewrite to reduce number of system calls 1806 mLastWriteTime = systemTime(); 1807 mInWrite = true; 1808 ssize_t bytesWritten; 1809 1810 // If an NBAIO sink is present, use it to write the normal mixer's submix 1811 if (mNormalSink != 0) { 1812#define mBitShift 2 // FIXME 1813 size_t count = mBytesRemaining >> mBitShift; 1814 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1815 ATRACE_BEGIN("write"); 1816 // update the setpoint when AudioFlinger::mScreenState changes 1817 uint32_t screenState = AudioFlinger::mScreenState; 1818 if (screenState != mScreenState) { 1819 mScreenState = screenState; 1820 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1821 if (pipe != NULL) { 1822 pipe->setAvgFrames((mScreenState & 1) ? 1823 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1824 } 1825 } 1826 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1827 ATRACE_END(); 1828 if (framesWritten > 0) { 1829 bytesWritten = framesWritten << mBitShift; 1830 } else { 1831 bytesWritten = framesWritten; 1832 } 1833 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1834 if (status == NO_ERROR) { 1835 size_t totalFramesWritten = mNormalSink->framesWritten(); 1836 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1837 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1838 mLatchDValid = true; 1839 } 1840 } 1841 // otherwise use the HAL / AudioStreamOut directly 1842 } else { 1843 // Direct output and offload threads 1844 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1845 if (mUseAsyncWrite) { 1846 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1847 mWriteAckSequence += 2; 1848 mWriteAckSequence |= 1; 1849 ALOG_ASSERT(mCallbackThread != 0); 1850 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1851 } 1852 // FIXME We should have an implementation of timestamps for direct output threads. 1853 // They are used e.g for multichannel PCM playback over HDMI. 1854 bytesWritten = mOutput->stream->write(mOutput->stream, 1855 mMixBuffer + offset, mBytesRemaining); 1856 if (mUseAsyncWrite && 1857 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1858 // do not wait for async callback in case of error of full write 1859 mWriteAckSequence &= ~1; 1860 ALOG_ASSERT(mCallbackThread != 0); 1861 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1862 } 1863 } 1864 1865 mNumWrites++; 1866 mInWrite = false; 1867 1868 return bytesWritten; 1869} 1870 1871void AudioFlinger::PlaybackThread::threadLoop_drain() 1872{ 1873 if (mOutput->stream->drain) { 1874 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1875 if (mUseAsyncWrite) { 1876 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1877 mDrainSequence |= 1; 1878 ALOG_ASSERT(mCallbackThread != 0); 1879 mCallbackThread->setDraining(mDrainSequence); 1880 } 1881 mOutput->stream->drain(mOutput->stream, 1882 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1883 : AUDIO_DRAIN_ALL); 1884 } 1885} 1886 1887void AudioFlinger::PlaybackThread::threadLoop_exit() 1888{ 1889 // Default implementation has nothing to do 1890} 1891 1892/* 1893The derived values that are cached: 1894 - mixBufferSize from frame count * frame size 1895 - activeSleepTime from activeSleepTimeUs() 1896 - idleSleepTime from idleSleepTimeUs() 1897 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1898 - maxPeriod from frame count and sample rate (MIXER only) 1899 1900The parameters that affect these derived values are: 1901 - frame count 1902 - frame size 1903 - sample rate 1904 - device type: A2DP or not 1905 - device latency 1906 - format: PCM or not 1907 - active sleep time 1908 - idle sleep time 1909*/ 1910 1911void AudioFlinger::PlaybackThread::cacheParameters_l() 1912{ 1913 mixBufferSize = mNormalFrameCount * mFrameSize; 1914 activeSleepTime = activeSleepTimeUs(); 1915 idleSleepTime = idleSleepTimeUs(); 1916} 1917 1918void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1919{ 1920 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1921 this, streamType, mTracks.size()); 1922 Mutex::Autolock _l(mLock); 1923 1924 size_t size = mTracks.size(); 1925 for (size_t i = 0; i < size; i++) { 1926 sp<Track> t = mTracks[i]; 1927 if (t->streamType() == streamType) { 1928 t->invalidate(); 1929 } 1930 } 1931} 1932 1933status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1934{ 1935 int session = chain->sessionId(); 1936 int16_t *buffer = mMixBuffer; 1937 bool ownsBuffer = false; 1938 1939 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1940 if (session > 0) { 1941 // Only one effect chain can be present in direct output thread and it uses 1942 // the mix buffer as input 1943 if (mType != DIRECT) { 1944 size_t numSamples = mNormalFrameCount * mChannelCount; 1945 buffer = new int16_t[numSamples]; 1946 memset(buffer, 0, numSamples * sizeof(int16_t)); 1947 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1948 ownsBuffer = true; 1949 } 1950 1951 // Attach all tracks with same session ID to this chain. 1952 for (size_t i = 0; i < mTracks.size(); ++i) { 1953 sp<Track> track = mTracks[i]; 1954 if (session == track->sessionId()) { 1955 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1956 buffer); 1957 track->setMainBuffer(buffer); 1958 chain->incTrackCnt(); 1959 } 1960 } 1961 1962 // indicate all active tracks in the chain 1963 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1964 sp<Track> track = mActiveTracks[i].promote(); 1965 if (track == 0) { 1966 continue; 1967 } 1968 if (session == track->sessionId()) { 1969 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1970 chain->incActiveTrackCnt(); 1971 } 1972 } 1973 } 1974 1975 chain->setInBuffer(buffer, ownsBuffer); 1976 chain->setOutBuffer(mMixBuffer); 1977 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1978 // chains list in order to be processed last as it contains output stage effects 1979 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1980 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1981 // after track specific effects and before output stage 1982 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1983 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1984 // Effect chain for other sessions are inserted at beginning of effect 1985 // chains list to be processed before output mix effects. Relative order between other 1986 // sessions is not important 1987 size_t size = mEffectChains.size(); 1988 size_t i = 0; 1989 for (i = 0; i < size; i++) { 1990 if (mEffectChains[i]->sessionId() < session) { 1991 break; 1992 } 1993 } 1994 mEffectChains.insertAt(chain, i); 1995 checkSuspendOnAddEffectChain_l(chain); 1996 1997 return NO_ERROR; 1998} 1999 2000size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2001{ 2002 int session = chain->sessionId(); 2003 2004 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2005 2006 for (size_t i = 0; i < mEffectChains.size(); i++) { 2007 if (chain == mEffectChains[i]) { 2008 mEffectChains.removeAt(i); 2009 // detach all active tracks from the chain 2010 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2011 sp<Track> track = mActiveTracks[i].promote(); 2012 if (track == 0) { 2013 continue; 2014 } 2015 if (session == track->sessionId()) { 2016 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2017 chain.get(), session); 2018 chain->decActiveTrackCnt(); 2019 } 2020 } 2021 2022 // detach all tracks with same session ID from this chain 2023 for (size_t i = 0; i < mTracks.size(); ++i) { 2024 sp<Track> track = mTracks[i]; 2025 if (session == track->sessionId()) { 2026 track->setMainBuffer(mMixBuffer); 2027 chain->decTrackCnt(); 2028 } 2029 } 2030 break; 2031 } 2032 } 2033 return mEffectChains.size(); 2034} 2035 2036status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2037 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2038{ 2039 Mutex::Autolock _l(mLock); 2040 return attachAuxEffect_l(track, EffectId); 2041} 2042 2043status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2044 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2045{ 2046 status_t status = NO_ERROR; 2047 2048 if (EffectId == 0) { 2049 track->setAuxBuffer(0, NULL); 2050 } else { 2051 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2052 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2053 if (effect != 0) { 2054 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2055 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2056 } else { 2057 status = INVALID_OPERATION; 2058 } 2059 } else { 2060 status = BAD_VALUE; 2061 } 2062 } 2063 return status; 2064} 2065 2066void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2067{ 2068 for (size_t i = 0; i < mTracks.size(); ++i) { 2069 sp<Track> track = mTracks[i]; 2070 if (track->auxEffectId() == effectId) { 2071 attachAuxEffect_l(track, 0); 2072 } 2073 } 2074} 2075 2076bool AudioFlinger::PlaybackThread::threadLoop() 2077{ 2078 Vector< sp<Track> > tracksToRemove; 2079 2080 standbyTime = systemTime(); 2081 2082 // MIXER 2083 nsecs_t lastWarning = 0; 2084 2085 // DUPLICATING 2086 // FIXME could this be made local to while loop? 2087 writeFrames = 0; 2088 2089 cacheParameters_l(); 2090 sleepTime = idleSleepTime; 2091 2092 if (mType == MIXER) { 2093 sleepTimeShift = 0; 2094 } 2095 2096 CpuStats cpuStats; 2097 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2098 2099 acquireWakeLock(); 2100 2101 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2102 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2103 // and then that string will be logged at the next convenient opportunity. 2104 const char *logString = NULL; 2105 2106 while (!exitPending()) 2107 { 2108 cpuStats.sample(myName); 2109 2110 Vector< sp<EffectChain> > effectChains; 2111 2112 processConfigEvents(); 2113 2114 { // scope for mLock 2115 2116 Mutex::Autolock _l(mLock); 2117 2118 if (logString != NULL) { 2119 mNBLogWriter->logTimestamp(); 2120 mNBLogWriter->log(logString); 2121 logString = NULL; 2122 } 2123 2124 if (mLatchDValid) { 2125 mLatchQ = mLatchD; 2126 mLatchDValid = false; 2127 mLatchQValid = true; 2128 } 2129 2130 if (checkForNewParameters_l()) { 2131 cacheParameters_l(); 2132 } 2133 2134 saveOutputTracks(); 2135 2136 if (mSignalPending) { 2137 // A signal was raised while we were unlocked 2138 mSignalPending = false; 2139 } else if (waitingAsyncCallback_l()) { 2140 if (exitPending()) { 2141 break; 2142 } 2143 releaseWakeLock_l(); 2144 ALOGV("wait async completion"); 2145 mWaitWorkCV.wait(mLock); 2146 ALOGV("async completion/wake"); 2147 acquireWakeLock_l(); 2148 if (exitPending()) { 2149 break; 2150 } 2151 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2152 continue; 2153 } 2154 sleepTime = 0; 2155 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2156 isSuspended()) { 2157 // put audio hardware into standby after short delay 2158 if (shouldStandby_l()) { 2159 2160 threadLoop_standby(); 2161 2162 mStandby = true; 2163 } 2164 2165 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2166 // we're about to wait, flush the binder command buffer 2167 IPCThreadState::self()->flushCommands(); 2168 2169 clearOutputTracks(); 2170 2171 if (exitPending()) { 2172 break; 2173 } 2174 2175 releaseWakeLock_l(); 2176 // wait until we have something to do... 2177 ALOGV("%s going to sleep", myName.string()); 2178 mWaitWorkCV.wait(mLock); 2179 ALOGV("%s waking up", myName.string()); 2180 acquireWakeLock_l(); 2181 2182 mMixerStatus = MIXER_IDLE; 2183 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2184 mBytesWritten = 0; 2185 mBytesRemaining = 0; 2186 checkSilentMode_l(); 2187 2188 standbyTime = systemTime() + standbyDelay; 2189 sleepTime = idleSleepTime; 2190 if (mType == MIXER) { 2191 sleepTimeShift = 0; 2192 } 2193 2194 continue; 2195 } 2196 } 2197 2198 // mMixerStatusIgnoringFastTracks is also updated internally 2199 mMixerStatus = prepareTracks_l(&tracksToRemove); 2200 2201 // prevent any changes in effect chain list and in each effect chain 2202 // during mixing and effect process as the audio buffers could be deleted 2203 // or modified if an effect is created or deleted 2204 lockEffectChains_l(effectChains); 2205 } 2206 2207 if (mBytesRemaining == 0) { 2208 mCurrentWriteLength = 0; 2209 if (mMixerStatus == MIXER_TRACKS_READY) { 2210 // threadLoop_mix() sets mCurrentWriteLength 2211 threadLoop_mix(); 2212 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2213 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2214 // threadLoop_sleepTime sets sleepTime to 0 if data 2215 // must be written to HAL 2216 threadLoop_sleepTime(); 2217 if (sleepTime == 0) { 2218 mCurrentWriteLength = mixBufferSize; 2219 } 2220 } 2221 mBytesRemaining = mCurrentWriteLength; 2222 if (isSuspended()) { 2223 sleepTime = suspendSleepTimeUs(); 2224 // simulate write to HAL when suspended 2225 mBytesWritten += mixBufferSize; 2226 mBytesRemaining = 0; 2227 } 2228 2229 // only process effects if we're going to write 2230 if (sleepTime == 0) { 2231 for (size_t i = 0; i < effectChains.size(); i ++) { 2232 effectChains[i]->process_l(); 2233 } 2234 } 2235 } 2236 2237 // enable changes in effect chain 2238 unlockEffectChains(effectChains); 2239 2240 if (!waitingAsyncCallback()) { 2241 // sleepTime == 0 means we must write to audio hardware 2242 if (sleepTime == 0) { 2243 if (mBytesRemaining) { 2244 ssize_t ret = threadLoop_write(); 2245 if (ret < 0) { 2246 mBytesRemaining = 0; 2247 } else { 2248 mBytesWritten += ret; 2249 mBytesRemaining -= ret; 2250 } 2251 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2252 (mMixerStatus == MIXER_DRAIN_ALL)) { 2253 threadLoop_drain(); 2254 } 2255if (mType == MIXER) { 2256 // write blocked detection 2257 nsecs_t now = systemTime(); 2258 nsecs_t delta = now - mLastWriteTime; 2259 if (!mStandby && delta > maxPeriod) { 2260 mNumDelayedWrites++; 2261 if ((now - lastWarning) > kWarningThrottleNs) { 2262 ATRACE_NAME("underrun"); 2263 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2264 ns2ms(delta), mNumDelayedWrites, this); 2265 lastWarning = now; 2266 } 2267 } 2268} 2269 2270 mStandby = false; 2271 } else { 2272 usleep(sleepTime); 2273 } 2274 } 2275 2276 // Finally let go of removed track(s), without the lock held 2277 // since we can't guarantee the destructors won't acquire that 2278 // same lock. This will also mutate and push a new fast mixer state. 2279 threadLoop_removeTracks(tracksToRemove); 2280 tracksToRemove.clear(); 2281 2282 // FIXME I don't understand the need for this here; 2283 // it was in the original code but maybe the 2284 // assignment in saveOutputTracks() makes this unnecessary? 2285 clearOutputTracks(); 2286 2287 // Effect chains will be actually deleted here if they were removed from 2288 // mEffectChains list during mixing or effects processing 2289 effectChains.clear(); 2290 2291 // FIXME Note that the above .clear() is no longer necessary since effectChains 2292 // is now local to this block, but will keep it for now (at least until merge done). 2293 } 2294 2295 threadLoop_exit(); 2296 2297 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2298 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2299 // put output stream into standby mode 2300 if (!mStandby) { 2301 mOutput->stream->common.standby(&mOutput->stream->common); 2302 } 2303 } 2304 2305 releaseWakeLock(); 2306 2307 ALOGV("Thread %p type %d exiting", this, mType); 2308 return false; 2309} 2310 2311// removeTracks_l() must be called with ThreadBase::mLock held 2312void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2313{ 2314 size_t count = tracksToRemove.size(); 2315 if (count > 0) { 2316 for (size_t i=0 ; i<count ; i++) { 2317 const sp<Track>& track = tracksToRemove.itemAt(i); 2318 mActiveTracks.remove(track); 2319 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2320 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2321 if (chain != 0) { 2322 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2323 track->sessionId()); 2324 chain->decActiveTrackCnt(); 2325 } 2326 if (track->isTerminated()) { 2327 removeTrack_l(track); 2328 } 2329 } 2330 } 2331 2332} 2333 2334// ---------------------------------------------------------------------------- 2335 2336AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2337 audio_io_handle_t id, audio_devices_t device, type_t type) 2338 : PlaybackThread(audioFlinger, output, id, device, type), 2339 // mAudioMixer below 2340 // mFastMixer below 2341 mFastMixerFutex(0) 2342 // mOutputSink below 2343 // mPipeSink below 2344 // mNormalSink below 2345{ 2346 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2347 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2348 "mFrameCount=%d, mNormalFrameCount=%d", 2349 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2350 mNormalFrameCount); 2351 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2352 2353 // FIXME - Current mixer implementation only supports stereo output 2354 if (mChannelCount != FCC_2) { 2355 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2356 } 2357 2358 // create an NBAIO sink for the HAL output stream, and negotiate 2359 mOutputSink = new AudioStreamOutSink(output->stream); 2360 size_t numCounterOffers = 0; 2361 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2362 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2363 ALOG_ASSERT(index == 0); 2364 2365 // initialize fast mixer depending on configuration 2366 bool initFastMixer; 2367 switch (kUseFastMixer) { 2368 case FastMixer_Never: 2369 initFastMixer = false; 2370 break; 2371 case FastMixer_Always: 2372 initFastMixer = true; 2373 break; 2374 case FastMixer_Static: 2375 case FastMixer_Dynamic: 2376 initFastMixer = mFrameCount < mNormalFrameCount; 2377 break; 2378 } 2379 if (initFastMixer) { 2380 2381 // create a MonoPipe to connect our submix to FastMixer 2382 NBAIO_Format format = mOutputSink->format(); 2383 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2384 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2385 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2386 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2387 const NBAIO_Format offers[1] = {format}; 2388 size_t numCounterOffers = 0; 2389 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2390 ALOG_ASSERT(index == 0); 2391 monoPipe->setAvgFrames((mScreenState & 1) ? 2392 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2393 mPipeSink = monoPipe; 2394 2395#ifdef TEE_SINK 2396 if (mTeeSinkOutputEnabled) { 2397 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2398 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2399 numCounterOffers = 0; 2400 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2401 ALOG_ASSERT(index == 0); 2402 mTeeSink = teeSink; 2403 PipeReader *teeSource = new PipeReader(*teeSink); 2404 numCounterOffers = 0; 2405 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2406 ALOG_ASSERT(index == 0); 2407 mTeeSource = teeSource; 2408 } 2409#endif 2410 2411 // create fast mixer and configure it initially with just one fast track for our submix 2412 mFastMixer = new FastMixer(); 2413 FastMixerStateQueue *sq = mFastMixer->sq(); 2414#ifdef STATE_QUEUE_DUMP 2415 sq->setObserverDump(&mStateQueueObserverDump); 2416 sq->setMutatorDump(&mStateQueueMutatorDump); 2417#endif 2418 FastMixerState *state = sq->begin(); 2419 FastTrack *fastTrack = &state->mFastTracks[0]; 2420 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2421 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2422 fastTrack->mVolumeProvider = NULL; 2423 fastTrack->mGeneration++; 2424 state->mFastTracksGen++; 2425 state->mTrackMask = 1; 2426 // fast mixer will use the HAL output sink 2427 state->mOutputSink = mOutputSink.get(); 2428 state->mOutputSinkGen++; 2429 state->mFrameCount = mFrameCount; 2430 state->mCommand = FastMixerState::COLD_IDLE; 2431 // already done in constructor initialization list 2432 //mFastMixerFutex = 0; 2433 state->mColdFutexAddr = &mFastMixerFutex; 2434 state->mColdGen++; 2435 state->mDumpState = &mFastMixerDumpState; 2436#ifdef TEE_SINK 2437 state->mTeeSink = mTeeSink.get(); 2438#endif 2439 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2440 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2441 sq->end(); 2442 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2443 2444 // start the fast mixer 2445 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2446 pid_t tid = mFastMixer->getTid(); 2447 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2448 if (err != 0) { 2449 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2450 kPriorityFastMixer, getpid_cached, tid, err); 2451 } 2452 2453#ifdef AUDIO_WATCHDOG 2454 // create and start the watchdog 2455 mAudioWatchdog = new AudioWatchdog(); 2456 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2457 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2458 tid = mAudioWatchdog->getTid(); 2459 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2460 if (err != 0) { 2461 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2462 kPriorityFastMixer, getpid_cached, tid, err); 2463 } 2464#endif 2465 2466 } else { 2467 mFastMixer = NULL; 2468 } 2469 2470 switch (kUseFastMixer) { 2471 case FastMixer_Never: 2472 case FastMixer_Dynamic: 2473 mNormalSink = mOutputSink; 2474 break; 2475 case FastMixer_Always: 2476 mNormalSink = mPipeSink; 2477 break; 2478 case FastMixer_Static: 2479 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2480 break; 2481 } 2482} 2483 2484AudioFlinger::MixerThread::~MixerThread() 2485{ 2486 if (mFastMixer != NULL) { 2487 FastMixerStateQueue *sq = mFastMixer->sq(); 2488 FastMixerState *state = sq->begin(); 2489 if (state->mCommand == FastMixerState::COLD_IDLE) { 2490 int32_t old = android_atomic_inc(&mFastMixerFutex); 2491 if (old == -1) { 2492 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2493 } 2494 } 2495 state->mCommand = FastMixerState::EXIT; 2496 sq->end(); 2497 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2498 mFastMixer->join(); 2499 // Though the fast mixer thread has exited, it's state queue is still valid. 2500 // We'll use that extract the final state which contains one remaining fast track 2501 // corresponding to our sub-mix. 2502 state = sq->begin(); 2503 ALOG_ASSERT(state->mTrackMask == 1); 2504 FastTrack *fastTrack = &state->mFastTracks[0]; 2505 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2506 delete fastTrack->mBufferProvider; 2507 sq->end(false /*didModify*/); 2508 delete mFastMixer; 2509#ifdef AUDIO_WATCHDOG 2510 if (mAudioWatchdog != 0) { 2511 mAudioWatchdog->requestExit(); 2512 mAudioWatchdog->requestExitAndWait(); 2513 mAudioWatchdog.clear(); 2514 } 2515#endif 2516 } 2517 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2518 delete mAudioMixer; 2519} 2520 2521 2522uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2523{ 2524 if (mFastMixer != NULL) { 2525 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2526 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2527 } 2528 return latency; 2529} 2530 2531 2532void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2533{ 2534 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2535} 2536 2537ssize_t AudioFlinger::MixerThread::threadLoop_write() 2538{ 2539 // FIXME we should only do one push per cycle; confirm this is true 2540 // Start the fast mixer if it's not already running 2541 if (mFastMixer != NULL) { 2542 FastMixerStateQueue *sq = mFastMixer->sq(); 2543 FastMixerState *state = sq->begin(); 2544 if (state->mCommand != FastMixerState::MIX_WRITE && 2545 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2546 if (state->mCommand == FastMixerState::COLD_IDLE) { 2547 int32_t old = android_atomic_inc(&mFastMixerFutex); 2548 if (old == -1) { 2549 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2550 } 2551#ifdef AUDIO_WATCHDOG 2552 if (mAudioWatchdog != 0) { 2553 mAudioWatchdog->resume(); 2554 } 2555#endif 2556 } 2557 state->mCommand = FastMixerState::MIX_WRITE; 2558 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2559 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2560 sq->end(); 2561 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2562 if (kUseFastMixer == FastMixer_Dynamic) { 2563 mNormalSink = mPipeSink; 2564 } 2565 } else { 2566 sq->end(false /*didModify*/); 2567 } 2568 } 2569 return PlaybackThread::threadLoop_write(); 2570} 2571 2572void AudioFlinger::MixerThread::threadLoop_standby() 2573{ 2574 // Idle the fast mixer if it's currently running 2575 if (mFastMixer != NULL) { 2576 FastMixerStateQueue *sq = mFastMixer->sq(); 2577 FastMixerState *state = sq->begin(); 2578 if (!(state->mCommand & FastMixerState::IDLE)) { 2579 state->mCommand = FastMixerState::COLD_IDLE; 2580 state->mColdFutexAddr = &mFastMixerFutex; 2581 state->mColdGen++; 2582 mFastMixerFutex = 0; 2583 sq->end(); 2584 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2585 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2586 if (kUseFastMixer == FastMixer_Dynamic) { 2587 mNormalSink = mOutputSink; 2588 } 2589#ifdef AUDIO_WATCHDOG 2590 if (mAudioWatchdog != 0) { 2591 mAudioWatchdog->pause(); 2592 } 2593#endif 2594 } else { 2595 sq->end(false /*didModify*/); 2596 } 2597 } 2598 PlaybackThread::threadLoop_standby(); 2599} 2600 2601// Empty implementation for standard mixer 2602// Overridden for offloaded playback 2603void AudioFlinger::PlaybackThread::flushOutput_l() 2604{ 2605} 2606 2607bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2608{ 2609 return false; 2610} 2611 2612bool AudioFlinger::PlaybackThread::shouldStandby_l() 2613{ 2614 return !mStandby; 2615} 2616 2617bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2618{ 2619 Mutex::Autolock _l(mLock); 2620 return waitingAsyncCallback_l(); 2621} 2622 2623// shared by MIXER and DIRECT, overridden by DUPLICATING 2624void AudioFlinger::PlaybackThread::threadLoop_standby() 2625{ 2626 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2627 mOutput->stream->common.standby(&mOutput->stream->common); 2628 if (mUseAsyncWrite != 0) { 2629 // discard any pending drain or write ack by incrementing sequence 2630 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2631 mDrainSequence = (mDrainSequence + 2) & ~1; 2632 ALOG_ASSERT(mCallbackThread != 0); 2633 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2634 mCallbackThread->setDraining(mDrainSequence); 2635 } 2636} 2637 2638void AudioFlinger::MixerThread::threadLoop_mix() 2639{ 2640 // obtain the presentation timestamp of the next output buffer 2641 int64_t pts; 2642 status_t status = INVALID_OPERATION; 2643 2644 if (mNormalSink != 0) { 2645 status = mNormalSink->getNextWriteTimestamp(&pts); 2646 } else { 2647 status = mOutputSink->getNextWriteTimestamp(&pts); 2648 } 2649 2650 if (status != NO_ERROR) { 2651 pts = AudioBufferProvider::kInvalidPTS; 2652 } 2653 2654 // mix buffers... 2655 mAudioMixer->process(pts); 2656 mCurrentWriteLength = mixBufferSize; 2657 // increase sleep time progressively when application underrun condition clears. 2658 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2659 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2660 // such that we would underrun the audio HAL. 2661 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2662 sleepTimeShift--; 2663 } 2664 sleepTime = 0; 2665 standbyTime = systemTime() + standbyDelay; 2666 //TODO: delay standby when effects have a tail 2667} 2668 2669void AudioFlinger::MixerThread::threadLoop_sleepTime() 2670{ 2671 // If no tracks are ready, sleep once for the duration of an output 2672 // buffer size, then write 0s to the output 2673 if (sleepTime == 0) { 2674 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2675 sleepTime = activeSleepTime >> sleepTimeShift; 2676 if (sleepTime < kMinThreadSleepTimeUs) { 2677 sleepTime = kMinThreadSleepTimeUs; 2678 } 2679 // reduce sleep time in case of consecutive application underruns to avoid 2680 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2681 // duration we would end up writing less data than needed by the audio HAL if 2682 // the condition persists. 2683 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2684 sleepTimeShift++; 2685 } 2686 } else { 2687 sleepTime = idleSleepTime; 2688 } 2689 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2690 memset(mMixBuffer, 0, mixBufferSize); 2691 sleepTime = 0; 2692 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2693 "anticipated start"); 2694 } 2695 // TODO add standby time extension fct of effect tail 2696} 2697 2698// prepareTracks_l() must be called with ThreadBase::mLock held 2699AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2700 Vector< sp<Track> > *tracksToRemove) 2701{ 2702 2703 mixer_state mixerStatus = MIXER_IDLE; 2704 // find out which tracks need to be processed 2705 size_t count = mActiveTracks.size(); 2706 size_t mixedTracks = 0; 2707 size_t tracksWithEffect = 0; 2708 // counts only _active_ fast tracks 2709 size_t fastTracks = 0; 2710 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2711 2712 float masterVolume = mMasterVolume; 2713 bool masterMute = mMasterMute; 2714 2715 if (masterMute) { 2716 masterVolume = 0; 2717 } 2718 // Delegate master volume control to effect in output mix effect chain if needed 2719 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2720 if (chain != 0) { 2721 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2722 chain->setVolume_l(&v, &v); 2723 masterVolume = (float)((v + (1 << 23)) >> 24); 2724 chain.clear(); 2725 } 2726 2727 // prepare a new state to push 2728 FastMixerStateQueue *sq = NULL; 2729 FastMixerState *state = NULL; 2730 bool didModify = false; 2731 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2732 if (mFastMixer != NULL) { 2733 sq = mFastMixer->sq(); 2734 state = sq->begin(); 2735 } 2736 2737 for (size_t i=0 ; i<count ; i++) { 2738 const sp<Track> t = mActiveTracks[i].promote(); 2739 if (t == 0) { 2740 continue; 2741 } 2742 2743 // this const just means the local variable doesn't change 2744 Track* const track = t.get(); 2745 2746 // process fast tracks 2747 if (track->isFastTrack()) { 2748 2749 // It's theoretically possible (though unlikely) for a fast track to be created 2750 // and then removed within the same normal mix cycle. This is not a problem, as 2751 // the track never becomes active so it's fast mixer slot is never touched. 2752 // The converse, of removing an (active) track and then creating a new track 2753 // at the identical fast mixer slot within the same normal mix cycle, 2754 // is impossible because the slot isn't marked available until the end of each cycle. 2755 int j = track->mFastIndex; 2756 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2757 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2758 FastTrack *fastTrack = &state->mFastTracks[j]; 2759 2760 // Determine whether the track is currently in underrun condition, 2761 // and whether it had a recent underrun. 2762 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2763 FastTrackUnderruns underruns = ftDump->mUnderruns; 2764 uint32_t recentFull = (underruns.mBitFields.mFull - 2765 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2766 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2767 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2768 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2769 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2770 uint32_t recentUnderruns = recentPartial + recentEmpty; 2771 track->mObservedUnderruns = underruns; 2772 // don't count underruns that occur while stopping or pausing 2773 // or stopped which can occur when flush() is called while active 2774 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2775 recentUnderruns > 0) { 2776 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2777 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2778 } 2779 2780 // This is similar to the state machine for normal tracks, 2781 // with a few modifications for fast tracks. 2782 bool isActive = true; 2783 switch (track->mState) { 2784 case TrackBase::STOPPING_1: 2785 // track stays active in STOPPING_1 state until first underrun 2786 if (recentUnderruns > 0 || track->isTerminated()) { 2787 track->mState = TrackBase::STOPPING_2; 2788 } 2789 break; 2790 case TrackBase::PAUSING: 2791 // ramp down is not yet implemented 2792 track->setPaused(); 2793 break; 2794 case TrackBase::RESUMING: 2795 // ramp up is not yet implemented 2796 track->mState = TrackBase::ACTIVE; 2797 break; 2798 case TrackBase::ACTIVE: 2799 if (recentFull > 0 || recentPartial > 0) { 2800 // track has provided at least some frames recently: reset retry count 2801 track->mRetryCount = kMaxTrackRetries; 2802 } 2803 if (recentUnderruns == 0) { 2804 // no recent underruns: stay active 2805 break; 2806 } 2807 // there has recently been an underrun of some kind 2808 if (track->sharedBuffer() == 0) { 2809 // were any of the recent underruns "empty" (no frames available)? 2810 if (recentEmpty == 0) { 2811 // no, then ignore the partial underruns as they are allowed indefinitely 2812 break; 2813 } 2814 // there has recently been an "empty" underrun: decrement the retry counter 2815 if (--(track->mRetryCount) > 0) { 2816 break; 2817 } 2818 // indicate to client process that the track was disabled because of underrun; 2819 // it will then automatically call start() when data is available 2820 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2821 // remove from active list, but state remains ACTIVE [confusing but true] 2822 isActive = false; 2823 break; 2824 } 2825 // fall through 2826 case TrackBase::STOPPING_2: 2827 case TrackBase::PAUSED: 2828 case TrackBase::STOPPED: 2829 case TrackBase::FLUSHED: // flush() while active 2830 // Check for presentation complete if track is inactive 2831 // We have consumed all the buffers of this track. 2832 // This would be incomplete if we auto-paused on underrun 2833 { 2834 size_t audioHALFrames = 2835 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2836 size_t framesWritten = mBytesWritten / mFrameSize; 2837 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2838 // track stays in active list until presentation is complete 2839 break; 2840 } 2841 } 2842 if (track->isStopping_2()) { 2843 track->mState = TrackBase::STOPPED; 2844 } 2845 if (track->isStopped()) { 2846 // Can't reset directly, as fast mixer is still polling this track 2847 // track->reset(); 2848 // So instead mark this track as needing to be reset after push with ack 2849 resetMask |= 1 << i; 2850 } 2851 isActive = false; 2852 break; 2853 case TrackBase::IDLE: 2854 default: 2855 LOG_FATAL("unexpected track state %d", track->mState); 2856 } 2857 2858 if (isActive) { 2859 // was it previously inactive? 2860 if (!(state->mTrackMask & (1 << j))) { 2861 ExtendedAudioBufferProvider *eabp = track; 2862 VolumeProvider *vp = track; 2863 fastTrack->mBufferProvider = eabp; 2864 fastTrack->mVolumeProvider = vp; 2865 fastTrack->mSampleRate = track->mSampleRate; 2866 fastTrack->mChannelMask = track->mChannelMask; 2867 fastTrack->mGeneration++; 2868 state->mTrackMask |= 1 << j; 2869 didModify = true; 2870 // no acknowledgement required for newly active tracks 2871 } 2872 // cache the combined master volume and stream type volume for fast mixer; this 2873 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2874 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2875 ++fastTracks; 2876 } else { 2877 // was it previously active? 2878 if (state->mTrackMask & (1 << j)) { 2879 fastTrack->mBufferProvider = NULL; 2880 fastTrack->mGeneration++; 2881 state->mTrackMask &= ~(1 << j); 2882 didModify = true; 2883 // If any fast tracks were removed, we must wait for acknowledgement 2884 // because we're about to decrement the last sp<> on those tracks. 2885 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2886 } else { 2887 LOG_FATAL("fast track %d should have been active", j); 2888 } 2889 tracksToRemove->add(track); 2890 // Avoids a misleading display in dumpsys 2891 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2892 } 2893 continue; 2894 } 2895 2896 { // local variable scope to avoid goto warning 2897 2898 audio_track_cblk_t* cblk = track->cblk(); 2899 2900 // The first time a track is added we wait 2901 // for all its buffers to be filled before processing it 2902 int name = track->name(); 2903 // make sure that we have enough frames to mix one full buffer. 2904 // enforce this condition only once to enable draining the buffer in case the client 2905 // app does not call stop() and relies on underrun to stop: 2906 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2907 // during last round 2908 size_t desiredFrames; 2909 uint32_t sr = track->sampleRate(); 2910 if (sr == mSampleRate) { 2911 desiredFrames = mNormalFrameCount; 2912 } else { 2913 // +1 for rounding and +1 for additional sample needed for interpolation 2914 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2915 // add frames already consumed but not yet released by the resampler 2916 // because mAudioTrackServerProxy->framesReady() will include these frames 2917 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2918 // the minimum track buffer size is normally twice the number of frames necessary 2919 // to fill one buffer and the resampler should not leave more than one buffer worth 2920 // of unreleased frames after each pass, but just in case... 2921 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2922 } 2923 uint32_t minFrames = 1; 2924 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2925 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2926 minFrames = desiredFrames; 2927 } 2928 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2929 size_t framesReady; 2930 if (track->sharedBuffer() == 0) { 2931 framesReady = track->framesReady(); 2932 } else if (track->isStopped()) { 2933 framesReady = 0; 2934 } else { 2935 framesReady = 1; 2936 } 2937 if ((framesReady >= minFrames) && track->isReady() && 2938 !track->isPaused() && !track->isTerminated()) 2939 { 2940 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2941 2942 mixedTracks++; 2943 2944 // track->mainBuffer() != mMixBuffer means there is an effect chain 2945 // connected to the track 2946 chain.clear(); 2947 if (track->mainBuffer() != mMixBuffer) { 2948 chain = getEffectChain_l(track->sessionId()); 2949 // Delegate volume control to effect in track effect chain if needed 2950 if (chain != 0) { 2951 tracksWithEffect++; 2952 } else { 2953 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2954 "session %d", 2955 name, track->sessionId()); 2956 } 2957 } 2958 2959 2960 int param = AudioMixer::VOLUME; 2961 if (track->mFillingUpStatus == Track::FS_FILLED) { 2962 // no ramp for the first volume setting 2963 track->mFillingUpStatus = Track::FS_ACTIVE; 2964 if (track->mState == TrackBase::RESUMING) { 2965 track->mState = TrackBase::ACTIVE; 2966 param = AudioMixer::RAMP_VOLUME; 2967 } 2968 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2969 // FIXME should not make a decision based on mServer 2970 } else if (cblk->mServer != 0) { 2971 // If the track is stopped before the first frame was mixed, 2972 // do not apply ramp 2973 param = AudioMixer::RAMP_VOLUME; 2974 } 2975 2976 // compute volume for this track 2977 uint32_t vl, vr, va; 2978 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2979 vl = vr = va = 0; 2980 if (track->isPausing()) { 2981 track->setPaused(); 2982 } 2983 } else { 2984 2985 // read original volumes with volume control 2986 float typeVolume = mStreamTypes[track->streamType()].volume; 2987 float v = masterVolume * typeVolume; 2988 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2989 uint32_t vlr = proxy->getVolumeLR(); 2990 vl = vlr & 0xFFFF; 2991 vr = vlr >> 16; 2992 // track volumes come from shared memory, so can't be trusted and must be clamped 2993 if (vl > MAX_GAIN_INT) { 2994 ALOGV("Track left volume out of range: %04X", vl); 2995 vl = MAX_GAIN_INT; 2996 } 2997 if (vr > MAX_GAIN_INT) { 2998 ALOGV("Track right volume out of range: %04X", vr); 2999 vr = MAX_GAIN_INT; 3000 } 3001 // now apply the master volume and stream type volume 3002 vl = (uint32_t)(v * vl) << 12; 3003 vr = (uint32_t)(v * vr) << 12; 3004 // assuming master volume and stream type volume each go up to 1.0, 3005 // vl and vr are now in 8.24 format 3006 3007 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3008 // send level comes from shared memory and so may be corrupt 3009 if (sendLevel > MAX_GAIN_INT) { 3010 ALOGV("Track send level out of range: %04X", sendLevel); 3011 sendLevel = MAX_GAIN_INT; 3012 } 3013 va = (uint32_t)(v * sendLevel); 3014 } 3015 3016 // Delegate volume control to effect in track effect chain if needed 3017 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3018 // Do not ramp volume if volume is controlled by effect 3019 param = AudioMixer::VOLUME; 3020 track->mHasVolumeController = true; 3021 } else { 3022 // force no volume ramp when volume controller was just disabled or removed 3023 // from effect chain to avoid volume spike 3024 if (track->mHasVolumeController) { 3025 param = AudioMixer::VOLUME; 3026 } 3027 track->mHasVolumeController = false; 3028 } 3029 3030 // Convert volumes from 8.24 to 4.12 format 3031 // This additional clamping is needed in case chain->setVolume_l() overshot 3032 vl = (vl + (1 << 11)) >> 12; 3033 if (vl > MAX_GAIN_INT) { 3034 vl = MAX_GAIN_INT; 3035 } 3036 vr = (vr + (1 << 11)) >> 12; 3037 if (vr > MAX_GAIN_INT) { 3038 vr = MAX_GAIN_INT; 3039 } 3040 3041 if (va > MAX_GAIN_INT) { 3042 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3043 } 3044 3045 // XXX: these things DON'T need to be done each time 3046 mAudioMixer->setBufferProvider(name, track); 3047 mAudioMixer->enable(name); 3048 3049 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3050 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3051 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3052 mAudioMixer->setParameter( 3053 name, 3054 AudioMixer::TRACK, 3055 AudioMixer::FORMAT, (void *)track->format()); 3056 mAudioMixer->setParameter( 3057 name, 3058 AudioMixer::TRACK, 3059 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3060 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3061 uint32_t maxSampleRate = mSampleRate * 2; 3062 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3063 if (reqSampleRate == 0) { 3064 reqSampleRate = mSampleRate; 3065 } else if (reqSampleRate > maxSampleRate) { 3066 reqSampleRate = maxSampleRate; 3067 } 3068 mAudioMixer->setParameter( 3069 name, 3070 AudioMixer::RESAMPLE, 3071 AudioMixer::SAMPLE_RATE, 3072 (void *)reqSampleRate); 3073 mAudioMixer->setParameter( 3074 name, 3075 AudioMixer::TRACK, 3076 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3077 mAudioMixer->setParameter( 3078 name, 3079 AudioMixer::TRACK, 3080 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3081 3082 // reset retry count 3083 track->mRetryCount = kMaxTrackRetries; 3084 3085 // If one track is ready, set the mixer ready if: 3086 // - the mixer was not ready during previous round OR 3087 // - no other track is not ready 3088 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3089 mixerStatus != MIXER_TRACKS_ENABLED) { 3090 mixerStatus = MIXER_TRACKS_READY; 3091 } 3092 } else { 3093 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3094 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3095 } 3096 // clear effect chain input buffer if an active track underruns to avoid sending 3097 // previous audio buffer again to effects 3098 chain = getEffectChain_l(track->sessionId()); 3099 if (chain != 0) { 3100 chain->clearInputBuffer(); 3101 } 3102 3103 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3104 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3105 track->isStopped() || track->isPaused()) { 3106 // We have consumed all the buffers of this track. 3107 // Remove it from the list of active tracks. 3108 // TODO: use actual buffer filling status instead of latency when available from 3109 // audio HAL 3110 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3111 size_t framesWritten = mBytesWritten / mFrameSize; 3112 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3113 if (track->isStopped()) { 3114 track->reset(); 3115 } 3116 tracksToRemove->add(track); 3117 } 3118 } else { 3119 // No buffers for this track. Give it a few chances to 3120 // fill a buffer, then remove it from active list. 3121 if (--(track->mRetryCount) <= 0) { 3122 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3123 tracksToRemove->add(track); 3124 // indicate to client process that the track was disabled because of underrun; 3125 // it will then automatically call start() when data is available 3126 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3127 // If one track is not ready, mark the mixer also not ready if: 3128 // - the mixer was ready during previous round OR 3129 // - no other track is ready 3130 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3131 mixerStatus != MIXER_TRACKS_READY) { 3132 mixerStatus = MIXER_TRACKS_ENABLED; 3133 } 3134 } 3135 mAudioMixer->disable(name); 3136 } 3137 3138 } // local variable scope to avoid goto warning 3139track_is_ready: ; 3140 3141 } 3142 3143 // Push the new FastMixer state if necessary 3144 bool pauseAudioWatchdog = false; 3145 if (didModify) { 3146 state->mFastTracksGen++; 3147 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3148 if (kUseFastMixer == FastMixer_Dynamic && 3149 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3150 state->mCommand = FastMixerState::COLD_IDLE; 3151 state->mColdFutexAddr = &mFastMixerFutex; 3152 state->mColdGen++; 3153 mFastMixerFutex = 0; 3154 if (kUseFastMixer == FastMixer_Dynamic) { 3155 mNormalSink = mOutputSink; 3156 } 3157 // If we go into cold idle, need to wait for acknowledgement 3158 // so that fast mixer stops doing I/O. 3159 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3160 pauseAudioWatchdog = true; 3161 } 3162 } 3163 if (sq != NULL) { 3164 sq->end(didModify); 3165 sq->push(block); 3166 } 3167#ifdef AUDIO_WATCHDOG 3168 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3169 mAudioWatchdog->pause(); 3170 } 3171#endif 3172 3173 // Now perform the deferred reset on fast tracks that have stopped 3174 while (resetMask != 0) { 3175 size_t i = __builtin_ctz(resetMask); 3176 ALOG_ASSERT(i < count); 3177 resetMask &= ~(1 << i); 3178 sp<Track> t = mActiveTracks[i].promote(); 3179 if (t == 0) { 3180 continue; 3181 } 3182 Track* track = t.get(); 3183 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3184 track->reset(); 3185 } 3186 3187 // remove all the tracks that need to be... 3188 removeTracks_l(*tracksToRemove); 3189 3190 // mix buffer must be cleared if all tracks are connected to an 3191 // effect chain as in this case the mixer will not write to 3192 // mix buffer and track effects will accumulate into it 3193 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3194 (mixedTracks == 0 && fastTracks > 0))) { 3195 // FIXME as a performance optimization, should remember previous zero status 3196 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3197 } 3198 3199 // if any fast tracks, then status is ready 3200 mMixerStatusIgnoringFastTracks = mixerStatus; 3201 if (fastTracks > 0) { 3202 mixerStatus = MIXER_TRACKS_READY; 3203 } 3204 return mixerStatus; 3205} 3206 3207// getTrackName_l() must be called with ThreadBase::mLock held 3208int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3209{ 3210 return mAudioMixer->getTrackName(channelMask, sessionId); 3211} 3212 3213// deleteTrackName_l() must be called with ThreadBase::mLock held 3214void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3215{ 3216 ALOGV("remove track (%d) and delete from mixer", name); 3217 mAudioMixer->deleteTrackName(name); 3218} 3219 3220// checkForNewParameters_l() must be called with ThreadBase::mLock held 3221bool AudioFlinger::MixerThread::checkForNewParameters_l() 3222{ 3223 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3224 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3225 bool reconfig = false; 3226 3227 while (!mNewParameters.isEmpty()) { 3228 3229 if (mFastMixer != NULL) { 3230 FastMixerStateQueue *sq = mFastMixer->sq(); 3231 FastMixerState *state = sq->begin(); 3232 if (!(state->mCommand & FastMixerState::IDLE)) { 3233 previousCommand = state->mCommand; 3234 state->mCommand = FastMixerState::HOT_IDLE; 3235 sq->end(); 3236 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3237 } else { 3238 sq->end(false /*didModify*/); 3239 } 3240 } 3241 3242 status_t status = NO_ERROR; 3243 String8 keyValuePair = mNewParameters[0]; 3244 AudioParameter param = AudioParameter(keyValuePair); 3245 int value; 3246 3247 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3248 reconfig = true; 3249 } 3250 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3251 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3252 status = BAD_VALUE; 3253 } else { 3254 // no need to save value, since it's constant 3255 reconfig = true; 3256 } 3257 } 3258 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3259 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3260 status = BAD_VALUE; 3261 } else { 3262 // no need to save value, since it's constant 3263 reconfig = true; 3264 } 3265 } 3266 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3267 // do not accept frame count changes if tracks are open as the track buffer 3268 // size depends on frame count and correct behavior would not be guaranteed 3269 // if frame count is changed after track creation 3270 if (!mTracks.isEmpty()) { 3271 status = INVALID_OPERATION; 3272 } else { 3273 reconfig = true; 3274 } 3275 } 3276 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3277#ifdef ADD_BATTERY_DATA 3278 // when changing the audio output device, call addBatteryData to notify 3279 // the change 3280 if (mOutDevice != value) { 3281 uint32_t params = 0; 3282 // check whether speaker is on 3283 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3284 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3285 } 3286 3287 audio_devices_t deviceWithoutSpeaker 3288 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3289 // check if any other device (except speaker) is on 3290 if (value & deviceWithoutSpeaker ) { 3291 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3292 } 3293 3294 if (params != 0) { 3295 addBatteryData(params); 3296 } 3297 } 3298#endif 3299 3300 // forward device change to effects that have requested to be 3301 // aware of attached audio device. 3302 if (value != AUDIO_DEVICE_NONE) { 3303 mOutDevice = value; 3304 for (size_t i = 0; i < mEffectChains.size(); i++) { 3305 mEffectChains[i]->setDevice_l(mOutDevice); 3306 } 3307 } 3308 } 3309 3310 if (status == NO_ERROR) { 3311 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3312 keyValuePair.string()); 3313 if (!mStandby && status == INVALID_OPERATION) { 3314 mOutput->stream->common.standby(&mOutput->stream->common); 3315 mStandby = true; 3316 mBytesWritten = 0; 3317 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3318 keyValuePair.string()); 3319 } 3320 if (status == NO_ERROR && reconfig) { 3321 readOutputParameters(); 3322 delete mAudioMixer; 3323 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3324 for (size_t i = 0; i < mTracks.size() ; i++) { 3325 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3326 if (name < 0) { 3327 break; 3328 } 3329 mTracks[i]->mName = name; 3330 } 3331 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3332 } 3333 } 3334 3335 mNewParameters.removeAt(0); 3336 3337 mParamStatus = status; 3338 mParamCond.signal(); 3339 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3340 // already timed out waiting for the status and will never signal the condition. 3341 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3342 } 3343 3344 if (!(previousCommand & FastMixerState::IDLE)) { 3345 ALOG_ASSERT(mFastMixer != NULL); 3346 FastMixerStateQueue *sq = mFastMixer->sq(); 3347 FastMixerState *state = sq->begin(); 3348 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3349 state->mCommand = previousCommand; 3350 sq->end(); 3351 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3352 } 3353 3354 return reconfig; 3355} 3356 3357 3358void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3359{ 3360 const size_t SIZE = 256; 3361 char buffer[SIZE]; 3362 String8 result; 3363 3364 PlaybackThread::dumpInternals(fd, args); 3365 3366 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3367 result.append(buffer); 3368 write(fd, result.string(), result.size()); 3369 3370 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3371 const FastMixerDumpState copy(mFastMixerDumpState); 3372 copy.dump(fd); 3373 3374#ifdef STATE_QUEUE_DUMP 3375 // Similar for state queue 3376 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3377 observerCopy.dump(fd); 3378 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3379 mutatorCopy.dump(fd); 3380#endif 3381 3382#ifdef TEE_SINK 3383 // Write the tee output to a .wav file 3384 dumpTee(fd, mTeeSource, mId); 3385#endif 3386 3387#ifdef AUDIO_WATCHDOG 3388 if (mAudioWatchdog != 0) { 3389 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3390 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3391 wdCopy.dump(fd); 3392 } 3393#endif 3394} 3395 3396uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3397{ 3398 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3399} 3400 3401uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3402{ 3403 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3404} 3405 3406void AudioFlinger::MixerThread::cacheParameters_l() 3407{ 3408 PlaybackThread::cacheParameters_l(); 3409 3410 // FIXME: Relaxed timing because of a certain device that can't meet latency 3411 // Should be reduced to 2x after the vendor fixes the driver issue 3412 // increase threshold again due to low power audio mode. The way this warning 3413 // threshold is calculated and its usefulness should be reconsidered anyway. 3414 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3415} 3416 3417// ---------------------------------------------------------------------------- 3418 3419AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3420 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3421 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3422 // mLeftVolFloat, mRightVolFloat 3423{ 3424} 3425 3426AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3427 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3428 ThreadBase::type_t type) 3429 : PlaybackThread(audioFlinger, output, id, device, type) 3430 // mLeftVolFloat, mRightVolFloat 3431{ 3432} 3433 3434AudioFlinger::DirectOutputThread::~DirectOutputThread() 3435{ 3436} 3437 3438void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3439{ 3440 audio_track_cblk_t* cblk = track->cblk(); 3441 float left, right; 3442 3443 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3444 left = right = 0; 3445 } else { 3446 float typeVolume = mStreamTypes[track->streamType()].volume; 3447 float v = mMasterVolume * typeVolume; 3448 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3449 uint32_t vlr = proxy->getVolumeLR(); 3450 float v_clamped = v * (vlr & 0xFFFF); 3451 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3452 left = v_clamped/MAX_GAIN; 3453 v_clamped = v * (vlr >> 16); 3454 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3455 right = v_clamped/MAX_GAIN; 3456 } 3457 3458 if (lastTrack) { 3459 if (left != mLeftVolFloat || right != mRightVolFloat) { 3460 mLeftVolFloat = left; 3461 mRightVolFloat = right; 3462 3463 // Convert volumes from float to 8.24 3464 uint32_t vl = (uint32_t)(left * (1 << 24)); 3465 uint32_t vr = (uint32_t)(right * (1 << 24)); 3466 3467 // Delegate volume control to effect in track effect chain if needed 3468 // only one effect chain can be present on DirectOutputThread, so if 3469 // there is one, the track is connected to it 3470 if (!mEffectChains.isEmpty()) { 3471 mEffectChains[0]->setVolume_l(&vl, &vr); 3472 left = (float)vl / (1 << 24); 3473 right = (float)vr / (1 << 24); 3474 } 3475 if (mOutput->stream->set_volume) { 3476 mOutput->stream->set_volume(mOutput->stream, left, right); 3477 } 3478 } 3479 } 3480} 3481 3482 3483AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3484 Vector< sp<Track> > *tracksToRemove 3485) 3486{ 3487 size_t count = mActiveTracks.size(); 3488 mixer_state mixerStatus = MIXER_IDLE; 3489 3490 // find out which tracks need to be processed 3491 for (size_t i = 0; i < count; i++) { 3492 sp<Track> t = mActiveTracks[i].promote(); 3493 // The track died recently 3494 if (t == 0) { 3495 continue; 3496 } 3497 3498 Track* const track = t.get(); 3499 audio_track_cblk_t* cblk = track->cblk(); 3500 3501 // The first time a track is added we wait 3502 // for all its buffers to be filled before processing it 3503 uint32_t minFrames; 3504 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3505 minFrames = mNormalFrameCount; 3506 } else { 3507 minFrames = 1; 3508 } 3509 // Only consider last track started for volume and mixer state control. 3510 // This is the last entry in mActiveTracks unless a track underruns. 3511 // As we only care about the transition phase between two tracks on a 3512 // direct output, it is not a problem to ignore the underrun case. 3513 bool last = (i == (count - 1)); 3514 3515 if ((track->framesReady() >= minFrames) && track->isReady() && 3516 !track->isPaused() && !track->isTerminated()) 3517 { 3518 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3519 3520 if (track->mFillingUpStatus == Track::FS_FILLED) { 3521 track->mFillingUpStatus = Track::FS_ACTIVE; 3522 mLeftVolFloat = mRightVolFloat = 0; 3523 if (track->mState == TrackBase::RESUMING) { 3524 track->mState = TrackBase::ACTIVE; 3525 } 3526 } 3527 3528 // compute volume for this track 3529 processVolume_l(track, last); 3530 if (last) { 3531 // reset retry count 3532 track->mRetryCount = kMaxTrackRetriesDirect; 3533 mActiveTrack = t; 3534 mixerStatus = MIXER_TRACKS_READY; 3535 } 3536 } else { 3537 // clear effect chain input buffer if the last active track started underruns 3538 // to avoid sending previous audio buffer again to effects 3539 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3540 mEffectChains[0]->clearInputBuffer(); 3541 } 3542 3543 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3544 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3545 track->isStopped() || track->isPaused()) { 3546 // We have consumed all the buffers of this track. 3547 // Remove it from the list of active tracks. 3548 // TODO: implement behavior for compressed audio 3549 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3550 size_t framesWritten = mBytesWritten / mFrameSize; 3551 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3552 if (track->isStopped()) { 3553 track->reset(); 3554 } 3555 tracksToRemove->add(track); 3556 } 3557 } else { 3558 // No buffers for this track. Give it a few chances to 3559 // fill a buffer, then remove it from active list. 3560 // Only consider last track started for mixer state control 3561 if (--(track->mRetryCount) <= 0) { 3562 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3563 tracksToRemove->add(track); 3564 } else if (last) { 3565 mixerStatus = MIXER_TRACKS_ENABLED; 3566 } 3567 } 3568 } 3569 } 3570 3571 // remove all the tracks that need to be... 3572 removeTracks_l(*tracksToRemove); 3573 3574 return mixerStatus; 3575} 3576 3577void AudioFlinger::DirectOutputThread::threadLoop_mix() 3578{ 3579 size_t frameCount = mFrameCount; 3580 int8_t *curBuf = (int8_t *)mMixBuffer; 3581 // output audio to hardware 3582 while (frameCount) { 3583 AudioBufferProvider::Buffer buffer; 3584 buffer.frameCount = frameCount; 3585 mActiveTrack->getNextBuffer(&buffer); 3586 if (buffer.raw == NULL) { 3587 memset(curBuf, 0, frameCount * mFrameSize); 3588 break; 3589 } 3590 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3591 frameCount -= buffer.frameCount; 3592 curBuf += buffer.frameCount * mFrameSize; 3593 mActiveTrack->releaseBuffer(&buffer); 3594 } 3595 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3596 sleepTime = 0; 3597 standbyTime = systemTime() + standbyDelay; 3598 mActiveTrack.clear(); 3599} 3600 3601void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3602{ 3603 if (sleepTime == 0) { 3604 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3605 sleepTime = activeSleepTime; 3606 } else { 3607 sleepTime = idleSleepTime; 3608 } 3609 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3610 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3611 sleepTime = 0; 3612 } 3613} 3614 3615// getTrackName_l() must be called with ThreadBase::mLock held 3616int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3617 int sessionId) 3618{ 3619 return 0; 3620} 3621 3622// deleteTrackName_l() must be called with ThreadBase::mLock held 3623void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3624{ 3625} 3626 3627// checkForNewParameters_l() must be called with ThreadBase::mLock held 3628bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3629{ 3630 bool reconfig = false; 3631 3632 while (!mNewParameters.isEmpty()) { 3633 status_t status = NO_ERROR; 3634 String8 keyValuePair = mNewParameters[0]; 3635 AudioParameter param = AudioParameter(keyValuePair); 3636 int value; 3637 3638 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3639 // do not accept frame count changes if tracks are open as the track buffer 3640 // size depends on frame count and correct behavior would not be garantied 3641 // if frame count is changed after track creation 3642 if (!mTracks.isEmpty()) { 3643 status = INVALID_OPERATION; 3644 } else { 3645 reconfig = true; 3646 } 3647 } 3648 if (status == NO_ERROR) { 3649 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3650 keyValuePair.string()); 3651 if (!mStandby && status == INVALID_OPERATION) { 3652 mOutput->stream->common.standby(&mOutput->stream->common); 3653 mStandby = true; 3654 mBytesWritten = 0; 3655 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3656 keyValuePair.string()); 3657 } 3658 if (status == NO_ERROR && reconfig) { 3659 readOutputParameters(); 3660 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3661 } 3662 } 3663 3664 mNewParameters.removeAt(0); 3665 3666 mParamStatus = status; 3667 mParamCond.signal(); 3668 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3669 // already timed out waiting for the status and will never signal the condition. 3670 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3671 } 3672 return reconfig; 3673} 3674 3675uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3676{ 3677 uint32_t time; 3678 if (audio_is_linear_pcm(mFormat)) { 3679 time = PlaybackThread::activeSleepTimeUs(); 3680 } else { 3681 time = 10000; 3682 } 3683 return time; 3684} 3685 3686uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3687{ 3688 uint32_t time; 3689 if (audio_is_linear_pcm(mFormat)) { 3690 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3691 } else { 3692 time = 10000; 3693 } 3694 return time; 3695} 3696 3697uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3698{ 3699 uint32_t time; 3700 if (audio_is_linear_pcm(mFormat)) { 3701 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3702 } else { 3703 time = 10000; 3704 } 3705 return time; 3706} 3707 3708void AudioFlinger::DirectOutputThread::cacheParameters_l() 3709{ 3710 PlaybackThread::cacheParameters_l(); 3711 3712 // use shorter standby delay as on normal output to release 3713 // hardware resources as soon as possible 3714 standbyDelay = microseconds(activeSleepTime*2); 3715} 3716 3717// ---------------------------------------------------------------------------- 3718 3719AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3720 const sp<AudioFlinger::OffloadThread>& offloadThread) 3721 : Thread(false /*canCallJava*/), 3722 mOffloadThread(offloadThread), 3723 mWriteAckSequence(0), 3724 mDrainSequence(0) 3725{ 3726} 3727 3728AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3729{ 3730} 3731 3732void AudioFlinger::AsyncCallbackThread::onFirstRef() 3733{ 3734 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3735} 3736 3737bool AudioFlinger::AsyncCallbackThread::threadLoop() 3738{ 3739 while (!exitPending()) { 3740 uint32_t writeAckSequence; 3741 uint32_t drainSequence; 3742 3743 { 3744 Mutex::Autolock _l(mLock); 3745 mWaitWorkCV.wait(mLock); 3746 if (exitPending()) { 3747 break; 3748 } 3749 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3750 mWriteAckSequence, mDrainSequence); 3751 writeAckSequence = mWriteAckSequence; 3752 mWriteAckSequence &= ~1; 3753 drainSequence = mDrainSequence; 3754 mDrainSequence &= ~1; 3755 } 3756 { 3757 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3758 if (offloadThread != 0) { 3759 if (writeAckSequence & 1) { 3760 offloadThread->resetWriteBlocked(writeAckSequence >> 1); 3761 } 3762 if (drainSequence & 1) { 3763 offloadThread->resetDraining(drainSequence >> 1); 3764 } 3765 } 3766 } 3767 } 3768 return false; 3769} 3770 3771void AudioFlinger::AsyncCallbackThread::exit() 3772{ 3773 ALOGV("AsyncCallbackThread::exit"); 3774 Mutex::Autolock _l(mLock); 3775 requestExit(); 3776 mWaitWorkCV.broadcast(); 3777} 3778 3779void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3780{ 3781 Mutex::Autolock _l(mLock); 3782 // bit 0 is cleared 3783 mWriteAckSequence = sequence << 1; 3784} 3785 3786void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3787{ 3788 Mutex::Autolock _l(mLock); 3789 // ignore unexpected callbacks 3790 if (mWriteAckSequence & 2) { 3791 mWriteAckSequence |= 1; 3792 mWaitWorkCV.signal(); 3793 } 3794} 3795 3796void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3797{ 3798 Mutex::Autolock _l(mLock); 3799 // bit 0 is cleared 3800 mDrainSequence = sequence << 1; 3801} 3802 3803void AudioFlinger::AsyncCallbackThread::resetDraining() 3804{ 3805 Mutex::Autolock _l(mLock); 3806 // ignore unexpected callbacks 3807 if (mDrainSequence & 2) { 3808 mDrainSequence |= 1; 3809 mWaitWorkCV.signal(); 3810 } 3811} 3812 3813 3814// ---------------------------------------------------------------------------- 3815AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3816 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3817 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3818 mHwPaused(false), 3819 mPausedBytesRemaining(0) 3820{ 3821 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3822} 3823 3824AudioFlinger::OffloadThread::~OffloadThread() 3825{ 3826 mPreviousTrack.clear(); 3827} 3828 3829void AudioFlinger::OffloadThread::threadLoop_exit() 3830{ 3831 if (mFlushPending || mHwPaused) { 3832 // If a flush is pending or track was paused, just discard buffered data 3833 flushHw_l(); 3834 } else { 3835 mMixerStatus = MIXER_DRAIN_ALL; 3836 threadLoop_drain(); 3837 } 3838 mCallbackThread->exit(); 3839 PlaybackThread::threadLoop_exit(); 3840} 3841 3842AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3843 Vector< sp<Track> > *tracksToRemove 3844) 3845{ 3846 ALOGV("OffloadThread::prepareTracks_l"); 3847 size_t count = mActiveTracks.size(); 3848 3849 mixer_state mixerStatus = MIXER_IDLE; 3850 // find out which tracks need to be processed 3851 for (size_t i = 0; i < count; i++) { 3852 sp<Track> t = mActiveTracks[i].promote(); 3853 // The track died recently 3854 if (t == 0) { 3855 continue; 3856 } 3857 Track* const track = t.get(); 3858 audio_track_cblk_t* cblk = track->cblk(); 3859 if (mPreviousTrack != NULL) { 3860 if (t != mPreviousTrack) { 3861 // Flush any data still being written from last track 3862 mBytesRemaining = 0; 3863 if (mPausedBytesRemaining) { 3864 // Last track was paused so we also need to flush saved 3865 // mixbuffer state and invalidate track so that it will 3866 // re-submit that unwritten data when it is next resumed 3867 mPausedBytesRemaining = 0; 3868 // Invalidate is a bit drastic - would be more efficient 3869 // to have a flag to tell client that some of the 3870 // previously written data was lost 3871 mPreviousTrack->invalidate(); 3872 } 3873 } 3874 } 3875 mPreviousTrack = t; 3876 bool last = (i == (count - 1)); 3877 if (track->isPausing()) { 3878 track->setPaused(); 3879 if (last) { 3880 if (!mHwPaused) { 3881 mOutput->stream->pause(mOutput->stream); 3882 mHwPaused = true; 3883 } 3884 // If we were part way through writing the mixbuffer to 3885 // the HAL we must save this until we resume 3886 // BUG - this will be wrong if a different track is made active, 3887 // in that case we want to discard the pending data in the 3888 // mixbuffer and tell the client to present it again when the 3889 // track is resumed 3890 mPausedWriteLength = mCurrentWriteLength; 3891 mPausedBytesRemaining = mBytesRemaining; 3892 mBytesRemaining = 0; // stop writing 3893 } 3894 tracksToRemove->add(track); 3895 } else if (track->framesReady() && track->isReady() && 3896 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3897 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3898 if (track->mFillingUpStatus == Track::FS_FILLED) { 3899 track->mFillingUpStatus = Track::FS_ACTIVE; 3900 mLeftVolFloat = mRightVolFloat = 0; 3901 if (track->mState == TrackBase::RESUMING) { 3902 if (mPausedBytesRemaining) { 3903 // Need to continue write that was interrupted 3904 mCurrentWriteLength = mPausedWriteLength; 3905 mBytesRemaining = mPausedBytesRemaining; 3906 mPausedBytesRemaining = 0; 3907 } 3908 track->mState = TrackBase::ACTIVE; 3909 } 3910 } 3911 3912 if (last) { 3913 if (mHwPaused) { 3914 mOutput->stream->resume(mOutput->stream); 3915 mHwPaused = false; 3916 // threadLoop_mix() will handle the case that we need to 3917 // resume an interrupted write 3918 } 3919 // reset retry count 3920 track->mRetryCount = kMaxTrackRetriesOffload; 3921 mActiveTrack = t; 3922 mixerStatus = MIXER_TRACKS_READY; 3923 } 3924 } else { 3925 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3926 if (track->isStopping_1()) { 3927 // Hardware buffer can hold a large amount of audio so we must 3928 // wait for all current track's data to drain before we say 3929 // that the track is stopped. 3930 if (mBytesRemaining == 0) { 3931 // Only start draining when all data in mixbuffer 3932 // has been written 3933 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3934 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3935 sleepTime = 0; 3936 standbyTime = systemTime() + standbyDelay; 3937 if (last) { 3938 mixerStatus = MIXER_DRAIN_TRACK; 3939 mDrainSequence += 2; 3940 if (mHwPaused) { 3941 // It is possible to move from PAUSED to STOPPING_1 without 3942 // a resume so we must ensure hardware is running 3943 mOutput->stream->resume(mOutput->stream); 3944 mHwPaused = false; 3945 } 3946 } 3947 } 3948 } else if (track->isStopping_2()) { 3949 // Drain has completed, signal presentation complete 3950 if (!(mDrainSequence & 1) || !last) { 3951 track->mState = TrackBase::STOPPED; 3952 size_t audioHALFrames = 3953 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3954 size_t framesWritten = 3955 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3956 track->presentationComplete(framesWritten, audioHALFrames); 3957 track->reset(); 3958 tracksToRemove->add(track); 3959 } 3960 } else { 3961 // No buffers for this track. Give it a few chances to 3962 // fill a buffer, then remove it from active list. 3963 if (--(track->mRetryCount) <= 0) { 3964 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3965 track->name()); 3966 tracksToRemove->add(track); 3967 } else if (last){ 3968 mixerStatus = MIXER_TRACKS_ENABLED; 3969 } 3970 } 3971 } 3972 // compute volume for this track 3973 processVolume_l(track, last); 3974 } 3975 3976 if (mFlushPending) { 3977 flushHw_l(); 3978 mFlushPending = false; 3979 } 3980 3981 // remove all the tracks that need to be... 3982 removeTracks_l(*tracksToRemove); 3983 3984 return mixerStatus; 3985} 3986 3987void AudioFlinger::OffloadThread::flushOutput_l() 3988{ 3989 mFlushPending = true; 3990} 3991 3992// must be called with thread mutex locked 3993bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3994{ 3995 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 3996 mWriteAckSequence, mDrainSequence); 3997 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 3998 return true; 3999 } 4000 return false; 4001} 4002 4003// must be called with thread mutex locked 4004bool AudioFlinger::OffloadThread::shouldStandby_l() 4005{ 4006 bool TrackPaused = false; 4007 4008 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4009 // after a timeout and we will enter standby then. 4010 if (mTracks.size() > 0) { 4011 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4012 } 4013 4014 return !mStandby && !TrackPaused; 4015} 4016 4017 4018bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4019{ 4020 Mutex::Autolock _l(mLock); 4021 return waitingAsyncCallback_l(); 4022} 4023 4024void AudioFlinger::OffloadThread::flushHw_l() 4025{ 4026 mOutput->stream->flush(mOutput->stream); 4027 // Flush anything still waiting in the mixbuffer 4028 mCurrentWriteLength = 0; 4029 mBytesRemaining = 0; 4030 mPausedWriteLength = 0; 4031 mPausedBytesRemaining = 0; 4032 if (mUseAsyncWrite) { 4033 // discard any pending drain or write ack by incrementing sequence 4034 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4035 mDrainSequence = (mDrainSequence + 2) & ~1; 4036 ALOG_ASSERT(mCallbackThread != 0); 4037 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4038 mCallbackThread->setDraining(mDrainSequence); 4039 } 4040} 4041 4042// ---------------------------------------------------------------------------- 4043 4044AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4045 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4046 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4047 DUPLICATING), 4048 mWaitTimeMs(UINT_MAX) 4049{ 4050 addOutputTrack(mainThread); 4051} 4052 4053AudioFlinger::DuplicatingThread::~DuplicatingThread() 4054{ 4055 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4056 mOutputTracks[i]->destroy(); 4057 } 4058} 4059 4060void AudioFlinger::DuplicatingThread::threadLoop_mix() 4061{ 4062 // mix buffers... 4063 if (outputsReady(outputTracks)) { 4064 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4065 } else { 4066 memset(mMixBuffer, 0, mixBufferSize); 4067 } 4068 sleepTime = 0; 4069 writeFrames = mNormalFrameCount; 4070 mCurrentWriteLength = mixBufferSize; 4071 standbyTime = systemTime() + standbyDelay; 4072} 4073 4074void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4075{ 4076 if (sleepTime == 0) { 4077 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4078 sleepTime = activeSleepTime; 4079 } else { 4080 sleepTime = idleSleepTime; 4081 } 4082 } else if (mBytesWritten != 0) { 4083 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4084 writeFrames = mNormalFrameCount; 4085 memset(mMixBuffer, 0, mixBufferSize); 4086 } else { 4087 // flush remaining overflow buffers in output tracks 4088 writeFrames = 0; 4089 } 4090 sleepTime = 0; 4091 } 4092} 4093 4094ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4095{ 4096 for (size_t i = 0; i < outputTracks.size(); i++) { 4097 outputTracks[i]->write(mMixBuffer, writeFrames); 4098 } 4099 return (ssize_t)mixBufferSize; 4100} 4101 4102void AudioFlinger::DuplicatingThread::threadLoop_standby() 4103{ 4104 // DuplicatingThread implements standby by stopping all tracks 4105 for (size_t i = 0; i < outputTracks.size(); i++) { 4106 outputTracks[i]->stop(); 4107 } 4108} 4109 4110void AudioFlinger::DuplicatingThread::saveOutputTracks() 4111{ 4112 outputTracks = mOutputTracks; 4113} 4114 4115void AudioFlinger::DuplicatingThread::clearOutputTracks() 4116{ 4117 outputTracks.clear(); 4118} 4119 4120void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4121{ 4122 Mutex::Autolock _l(mLock); 4123 // FIXME explain this formula 4124 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4125 OutputTrack *outputTrack = new OutputTrack(thread, 4126 this, 4127 mSampleRate, 4128 mFormat, 4129 mChannelMask, 4130 frameCount); 4131 if (outputTrack->cblk() != NULL) { 4132 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4133 mOutputTracks.add(outputTrack); 4134 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4135 updateWaitTime_l(); 4136 } 4137} 4138 4139void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4140{ 4141 Mutex::Autolock _l(mLock); 4142 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4143 if (mOutputTracks[i]->thread() == thread) { 4144 mOutputTracks[i]->destroy(); 4145 mOutputTracks.removeAt(i); 4146 updateWaitTime_l(); 4147 return; 4148 } 4149 } 4150 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4151} 4152 4153// caller must hold mLock 4154void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4155{ 4156 mWaitTimeMs = UINT_MAX; 4157 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4158 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4159 if (strong != 0) { 4160 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4161 if (waitTimeMs < mWaitTimeMs) { 4162 mWaitTimeMs = waitTimeMs; 4163 } 4164 } 4165 } 4166} 4167 4168 4169bool AudioFlinger::DuplicatingThread::outputsReady( 4170 const SortedVector< sp<OutputTrack> > &outputTracks) 4171{ 4172 for (size_t i = 0; i < outputTracks.size(); i++) { 4173 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4174 if (thread == 0) { 4175 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4176 outputTracks[i].get()); 4177 return false; 4178 } 4179 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4180 // see note at standby() declaration 4181 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4182 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4183 thread.get()); 4184 return false; 4185 } 4186 } 4187 return true; 4188} 4189 4190uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4191{ 4192 return (mWaitTimeMs * 1000) / 2; 4193} 4194 4195void AudioFlinger::DuplicatingThread::cacheParameters_l() 4196{ 4197 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4198 updateWaitTime_l(); 4199 4200 MixerThread::cacheParameters_l(); 4201} 4202 4203// ---------------------------------------------------------------------------- 4204// Record 4205// ---------------------------------------------------------------------------- 4206 4207AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4208 AudioStreamIn *input, 4209 uint32_t sampleRate, 4210 audio_channel_mask_t channelMask, 4211 audio_io_handle_t id, 4212 audio_devices_t outDevice, 4213 audio_devices_t inDevice 4214#ifdef TEE_SINK 4215 , const sp<NBAIO_Sink>& teeSink 4216#endif 4217 ) : 4218 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4219 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4220 // mRsmpInIndex set by readInputParameters() 4221 mReqChannelCount(popcount(channelMask)), 4222 mReqSampleRate(sampleRate) 4223 // mBytesRead is only meaningful while active, and so is cleared in start() 4224 // (but might be better to also clear here for dump?) 4225#ifdef TEE_SINK 4226 , mTeeSink(teeSink) 4227#endif 4228{ 4229 snprintf(mName, kNameLength, "AudioIn_%X", id); 4230 4231 readInputParameters(); 4232 4233} 4234 4235 4236AudioFlinger::RecordThread::~RecordThread() 4237{ 4238 delete[] mRsmpInBuffer; 4239 delete mResampler; 4240 delete[] mRsmpOutBuffer; 4241} 4242 4243void AudioFlinger::RecordThread::onFirstRef() 4244{ 4245 run(mName, PRIORITY_URGENT_AUDIO); 4246} 4247 4248bool AudioFlinger::RecordThread::threadLoop() 4249{ 4250 AudioBufferProvider::Buffer buffer; 4251 4252 nsecs_t lastWarning = 0; 4253 4254 inputStandBy(); 4255 acquireWakeLock(); 4256 4257 // used to verify we've read at least once before evaluating how many bytes were read 4258 bool readOnce = false; 4259 4260 // used to request a deferred sleep, to be executed later while mutex is unlocked 4261 bool doSleep = false; 4262 4263 // start recording 4264 for (;;) { 4265 sp<RecordTrack> activeTrack; 4266 TrackBase::track_state activeTrackState; 4267 Vector< sp<EffectChain> > effectChains; 4268 4269 // sleep with mutex unlocked 4270 if (doSleep) { 4271 doSleep = false; 4272 usleep(kRecordThreadSleepUs); 4273 } 4274 4275 { // scope for mLock 4276 Mutex::Autolock _l(mLock); 4277 if (exitPending()) { 4278 break; 4279 } 4280 processConfigEvents_l(); 4281 // return value 'reconfig' is currently unused 4282 bool reconfig = checkForNewParameters_l(); 4283 // make a stable copy of mActiveTrack 4284 activeTrack = mActiveTrack; 4285 if (activeTrack == 0) { 4286 standby(); 4287 // exitPending() can't become true here 4288 releaseWakeLock_l(); 4289 ALOGV("RecordThread: loop stopping"); 4290 // go to sleep 4291 mWaitWorkCV.wait(mLock); 4292 ALOGV("RecordThread: loop starting"); 4293 acquireWakeLock_l(); 4294 continue; 4295 } 4296 4297 if (activeTrack->isTerminated()) { 4298 removeTrack_l(activeTrack); 4299 mActiveTrack.clear(); 4300 continue; 4301 } 4302 4303 activeTrackState = activeTrack->mState; 4304 switch (activeTrackState) { 4305 case TrackBase::PAUSING: 4306 standby(); 4307 mActiveTrack.clear(); 4308 mStartStopCond.broadcast(); 4309 doSleep = true; 4310 continue; 4311 4312 case TrackBase::RESUMING: 4313 mStandby = false; 4314 if (mReqChannelCount != activeTrack->channelCount()) { 4315 mActiveTrack.clear(); 4316 mStartStopCond.broadcast(); 4317 continue; 4318 } 4319 if (readOnce) { 4320 mStartStopCond.broadcast(); 4321 // record start succeeds only if first read from audio input succeeds 4322 if (mBytesRead < 0) { 4323 mActiveTrack.clear(); 4324 continue; 4325 } 4326 activeTrack->mState = TrackBase::ACTIVE; 4327 } 4328 break; 4329 4330 case TrackBase::ACTIVE: 4331 break; 4332 4333 case TrackBase::IDLE: 4334 doSleep = true; 4335 continue; 4336 4337 default: 4338 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4339 } 4340 4341 lockEffectChains_l(effectChains); 4342 } 4343 4344 // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable 4345 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4346 4347 for (size_t i = 0; i < effectChains.size(); i ++) { 4348 // thread mutex is not locked, but effect chain is locked 4349 effectChains[i]->process_l(); 4350 } 4351 4352 buffer.frameCount = mFrameCount; 4353 status_t status = activeTrack->getNextBuffer(&buffer); 4354 if (status == NO_ERROR) { 4355 readOnce = true; 4356 size_t framesOut = buffer.frameCount; 4357 if (mResampler == NULL) { 4358 // no resampling 4359 while (framesOut) { 4360 size_t framesIn = mFrameCount - mRsmpInIndex; 4361 if (framesIn > 0) { 4362 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4363 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4364 activeTrack->mFrameSize; 4365 if (framesIn > framesOut) { 4366 framesIn = framesOut; 4367 } 4368 mRsmpInIndex += framesIn; 4369 framesOut -= framesIn; 4370 if (mChannelCount == mReqChannelCount) { 4371 memcpy(dst, src, framesIn * mFrameSize); 4372 } else { 4373 if (mChannelCount == 1) { 4374 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4375 (int16_t *)src, framesIn); 4376 } else { 4377 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4378 (int16_t *)src, framesIn); 4379 } 4380 } 4381 } 4382 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4383 void *readInto; 4384 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4385 readInto = buffer.raw; 4386 framesOut = 0; 4387 } else { 4388 readInto = mRsmpInBuffer; 4389 mRsmpInIndex = 0; 4390 } 4391 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4392 mBufferSize); 4393 if (mBytesRead <= 0) { 4394 // TODO: verify that it's benign to use a stale track state 4395 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4396 { 4397 ALOGE("Error reading audio input"); 4398 // Force input into standby so that it tries to 4399 // recover at next read attempt 4400 inputStandBy(); 4401 doSleep = true; 4402 } 4403 mRsmpInIndex = mFrameCount; 4404 framesOut = 0; 4405 buffer.frameCount = 0; 4406 } 4407#ifdef TEE_SINK 4408 else if (mTeeSink != 0) { 4409 (void) mTeeSink->write(readInto, 4410 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4411 } 4412#endif 4413 } 4414 } 4415 } else { 4416 // resampling 4417 4418 // resampler accumulates, but we only have one source track 4419 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4420 // alter output frame count as if we were expecting stereo samples 4421 if (mChannelCount == 1 && mReqChannelCount == 1) { 4422 framesOut >>= 1; 4423 } 4424 mResampler->resample(mRsmpOutBuffer, framesOut, 4425 this /* AudioBufferProvider* */); 4426 // ditherAndClamp() works as long as all buffers returned by 4427 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4428 if (mChannelCount == 2 && mReqChannelCount == 1) { 4429 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4430 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4431 // the resampler always outputs stereo samples: 4432 // do post stereo to mono conversion 4433 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4434 framesOut); 4435 } else { 4436 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4437 } 4438 // now done with mRsmpOutBuffer 4439 4440 } 4441 if (mFramestoDrop == 0) { 4442 activeTrack->releaseBuffer(&buffer); 4443 } else { 4444 if (mFramestoDrop > 0) { 4445 mFramestoDrop -= buffer.frameCount; 4446 if (mFramestoDrop <= 0) { 4447 clearSyncStartEvent(); 4448 } 4449 } else { 4450 mFramestoDrop += buffer.frameCount; 4451 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4452 mSyncStartEvent->isCancelled()) { 4453 ALOGW("Synced record %s, session %d, trigger session %d", 4454 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4455 activeTrack->sessionId(), 4456 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4457 clearSyncStartEvent(); 4458 } 4459 } 4460 } 4461 activeTrack->clearOverflow(); 4462 } 4463 // client isn't retrieving buffers fast enough 4464 else { 4465 if (!activeTrack->setOverflow()) { 4466 nsecs_t now = systemTime(); 4467 if ((now - lastWarning) > kWarningThrottleNs) { 4468 ALOGW("RecordThread: buffer overflow"); 4469 lastWarning = now; 4470 } 4471 } 4472 // Release the processor for a while before asking for a new buffer. 4473 // This will give the application more chance to read from the buffer and 4474 // clear the overflow. 4475 doSleep = true; 4476 } 4477 4478 // enable changes in effect chain 4479 unlockEffectChains(effectChains); 4480 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4481 } 4482 4483 standby(); 4484 4485 { 4486 Mutex::Autolock _l(mLock); 4487 for (size_t i = 0; i < mTracks.size(); i++) { 4488 sp<RecordTrack> track = mTracks[i]; 4489 track->invalidate(); 4490 } 4491 mActiveTrack.clear(); 4492 mStartStopCond.broadcast(); 4493 } 4494 4495 releaseWakeLock(); 4496 4497 ALOGV("RecordThread %p exiting", this); 4498 return false; 4499} 4500 4501void AudioFlinger::RecordThread::standby() 4502{ 4503 if (!mStandby) { 4504 inputStandBy(); 4505 mStandby = true; 4506 } 4507} 4508 4509void AudioFlinger::RecordThread::inputStandBy() 4510{ 4511 mInput->stream->common.standby(&mInput->stream->common); 4512} 4513 4514sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4515 const sp<AudioFlinger::Client>& client, 4516 uint32_t sampleRate, 4517 audio_format_t format, 4518 audio_channel_mask_t channelMask, 4519 size_t frameCount, 4520 int sessionId, 4521 IAudioFlinger::track_flags_t *flags, 4522 pid_t tid, 4523 status_t *status) 4524{ 4525 sp<RecordTrack> track; 4526 status_t lStatus; 4527 4528 lStatus = initCheck(); 4529 if (lStatus != NO_ERROR) { 4530 ALOGE("Audio driver not initialized."); 4531 goto Exit; 4532 } 4533 4534 // client expresses a preference for FAST, but we get the final say 4535 if (*flags & IAudioFlinger::TRACK_FAST) { 4536 if ( 4537 // use case: callback handler and frame count is default or at least as large as HAL 4538 ( 4539 (tid != -1) && 4540 ((frameCount == 0) || 4541 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4542 ) && 4543 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4544 // mono or stereo 4545 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4546 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4547 // hardware sample rate 4548 (sampleRate == mSampleRate) && 4549 // record thread has an associated fast recorder 4550 hasFastRecorder() 4551 // FIXME test that RecordThread for this fast track has a capable output HAL 4552 // FIXME add a permission test also? 4553 ) { 4554 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4555 if (frameCount == 0) { 4556 frameCount = mFrameCount * kFastTrackMultiplier; 4557 } 4558 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4559 frameCount, mFrameCount); 4560 } else { 4561 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4562 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4563 "hasFastRecorder=%d tid=%d", 4564 frameCount, mFrameCount, format, 4565 audio_is_linear_pcm(format), 4566 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4567 *flags &= ~IAudioFlinger::TRACK_FAST; 4568 // For compatibility with AudioRecord calculation, buffer depth is forced 4569 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4570 // This is probably too conservative, but legacy application code may depend on it. 4571 // If you change this calculation, also review the start threshold which is related. 4572 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4573 size_t mNormalFrameCount = 2048; // FIXME 4574 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4575 if (minBufCount < 2) { 4576 minBufCount = 2; 4577 } 4578 size_t minFrameCount = mNormalFrameCount * minBufCount; 4579 if (frameCount < minFrameCount) { 4580 frameCount = minFrameCount; 4581 } 4582 } 4583 } 4584 4585 // FIXME use flags and tid similar to createTrack_l() 4586 4587 { // scope for mLock 4588 Mutex::Autolock _l(mLock); 4589 4590 track = new RecordTrack(this, client, sampleRate, 4591 format, channelMask, frameCount, sessionId); 4592 4593 lStatus = track->initCheck(); 4594 if (lStatus != NO_ERROR) { 4595 track.clear(); 4596 goto Exit; 4597 } 4598 mTracks.add(track); 4599 4600 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4601 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4602 mAudioFlinger->btNrecIsOff(); 4603 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4604 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4605 4606 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4607 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4608 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4609 // so ask activity manager to do this on our behalf 4610 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4611 } 4612 } 4613 lStatus = NO_ERROR; 4614 4615Exit: 4616 *status = lStatus; 4617 return track; 4618} 4619 4620status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4621 AudioSystem::sync_event_t event, 4622 int triggerSession) 4623{ 4624 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4625 sp<ThreadBase> strongMe = this; 4626 status_t status = NO_ERROR; 4627 4628 if (event == AudioSystem::SYNC_EVENT_NONE) { 4629 clearSyncStartEvent(); 4630 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4631 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4632 triggerSession, 4633 recordTrack->sessionId(), 4634 syncStartEventCallback, 4635 this); 4636 // Sync event can be cancelled by the trigger session if the track is not in a 4637 // compatible state in which case we start record immediately 4638 if (mSyncStartEvent->isCancelled()) { 4639 clearSyncStartEvent(); 4640 } else { 4641 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4642 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4643 } 4644 } 4645 4646 { 4647 // This section is a rendezvous between binder thread executing start() and RecordThread 4648 AutoMutex lock(mLock); 4649 if (mActiveTrack != 0) { 4650 if (recordTrack != mActiveTrack.get()) { 4651 status = -EBUSY; 4652 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4653 mActiveTrack->mState = TrackBase::ACTIVE; 4654 } 4655 return status; 4656 } 4657 4658 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4659 recordTrack->mState = TrackBase::IDLE; 4660 mActiveTrack = recordTrack; 4661 mLock.unlock(); 4662 status_t status = AudioSystem::startInput(mId); 4663 mLock.lock(); 4664 // FIXME should verify that mActiveTrack is still == recordTrack 4665 if (status != NO_ERROR) { 4666 mActiveTrack.clear(); 4667 clearSyncStartEvent(); 4668 return status; 4669 } 4670 mRsmpInIndex = mFrameCount; 4671 mBytesRead = 0; 4672 if (mResampler != NULL) { 4673 mResampler->reset(); 4674 } 4675 // FIXME hijacking a playback track state name which was intended for start after pause; 4676 // here 'STARTING_2' would be more accurate 4677 mActiveTrack->mState = TrackBase::RESUMING; 4678 // signal thread to start 4679 ALOGV("Signal record thread"); 4680 mWaitWorkCV.broadcast(); 4681 // do not wait for mStartStopCond if exiting 4682 if (exitPending()) { 4683 mActiveTrack.clear(); 4684 status = INVALID_OPERATION; 4685 goto startError; 4686 } 4687 // FIXME incorrect usage of wait: no explicit predicate or loop 4688 mStartStopCond.wait(mLock); 4689 if (mActiveTrack == 0) { 4690 ALOGV("Record failed to start"); 4691 status = BAD_VALUE; 4692 goto startError; 4693 } 4694 ALOGV("Record started OK"); 4695 return status; 4696 } 4697 4698startError: 4699 AudioSystem::stopInput(mId); 4700 clearSyncStartEvent(); 4701 return status; 4702} 4703 4704void AudioFlinger::RecordThread::clearSyncStartEvent() 4705{ 4706 if (mSyncStartEvent != 0) { 4707 mSyncStartEvent->cancel(); 4708 } 4709 mSyncStartEvent.clear(); 4710 mFramestoDrop = 0; 4711} 4712 4713void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4714{ 4715 sp<SyncEvent> strongEvent = event.promote(); 4716 4717 if (strongEvent != 0) { 4718 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4719 me->handleSyncStartEvent(strongEvent); 4720 } 4721} 4722 4723void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4724{ 4725 if (event == mSyncStartEvent) { 4726 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4727 // from audio HAL 4728 mFramestoDrop = mFrameCount * 2; 4729 } 4730} 4731 4732bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4733 ALOGV("RecordThread::stop"); 4734 AutoMutex _l(mLock); 4735 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4736 return false; 4737 } 4738 // note that threadLoop may still be processing the track at this point [without lock] 4739 recordTrack->mState = TrackBase::PAUSING; 4740 // do not wait for mStartStopCond if exiting 4741 if (exitPending()) { 4742 return true; 4743 } 4744 // FIXME incorrect usage of wait: no explicit predicate or loop 4745 mStartStopCond.wait(mLock); 4746 // if we have been restarted, recordTrack == mActiveTrack.get() here 4747 if (exitPending() || recordTrack != mActiveTrack.get()) { 4748 ALOGV("Record stopped OK"); 4749 return true; 4750 } 4751 return false; 4752} 4753 4754bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4755{ 4756 return false; 4757} 4758 4759status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4760{ 4761#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4762 if (!isValidSyncEvent(event)) { 4763 return BAD_VALUE; 4764 } 4765 4766 int eventSession = event->triggerSession(); 4767 status_t ret = NAME_NOT_FOUND; 4768 4769 Mutex::Autolock _l(mLock); 4770 4771 for (size_t i = 0; i < mTracks.size(); i++) { 4772 sp<RecordTrack> track = mTracks[i]; 4773 if (eventSession == track->sessionId()) { 4774 (void) track->setSyncEvent(event); 4775 ret = NO_ERROR; 4776 } 4777 } 4778 return ret; 4779#else 4780 return BAD_VALUE; 4781#endif 4782} 4783 4784// destroyTrack_l() must be called with ThreadBase::mLock held 4785void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4786{ 4787 track->terminate(); 4788 track->mState = TrackBase::STOPPED; 4789 // active tracks are removed by threadLoop() 4790 if (mActiveTrack != track) { 4791 removeTrack_l(track); 4792 } 4793} 4794 4795void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4796{ 4797 mTracks.remove(track); 4798 // need anything related to effects here? 4799} 4800 4801void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4802{ 4803 dumpInternals(fd, args); 4804 dumpTracks(fd, args); 4805 dumpEffectChains(fd, args); 4806} 4807 4808void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4809{ 4810 const size_t SIZE = 256; 4811 char buffer[SIZE]; 4812 String8 result; 4813 4814 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4815 result.append(buffer); 4816 4817 if (mActiveTrack != 0) { 4818 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4819 result.append(buffer); 4820 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4821 result.append(buffer); 4822 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4823 result.append(buffer); 4824 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4825 result.append(buffer); 4826 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4827 result.append(buffer); 4828 } else { 4829 result.append("No active record client\n"); 4830 } 4831 4832 write(fd, result.string(), result.size()); 4833 4834 dumpBase(fd, args); 4835} 4836 4837void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4838{ 4839 const size_t SIZE = 256; 4840 char buffer[SIZE]; 4841 String8 result; 4842 4843 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4844 result.append(buffer); 4845 RecordTrack::appendDumpHeader(result); 4846 for (size_t i = 0; i < mTracks.size(); ++i) { 4847 sp<RecordTrack> track = mTracks[i]; 4848 if (track != 0) { 4849 track->dump(buffer, SIZE); 4850 result.append(buffer); 4851 } 4852 } 4853 4854 if (mActiveTrack != 0) { 4855 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4856 result.append(buffer); 4857 RecordTrack::appendDumpHeader(result); 4858 mActiveTrack->dump(buffer, SIZE); 4859 result.append(buffer); 4860 4861 } 4862 write(fd, result.string(), result.size()); 4863} 4864 4865// AudioBufferProvider interface 4866status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4867{ 4868 size_t framesReq = buffer->frameCount; 4869 size_t framesReady = mFrameCount - mRsmpInIndex; 4870 int channelCount; 4871 4872 if (framesReady == 0) { 4873 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4874 if (mBytesRead <= 0) { 4875 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4876 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4877 // Force input into standby so that it tries to 4878 // recover at next read attempt 4879 inputStandBy(); 4880 // FIXME an awkward place to sleep, consider using doSleep when this is pulled up 4881 usleep(kRecordThreadSleepUs); 4882 } 4883 buffer->raw = NULL; 4884 buffer->frameCount = 0; 4885 return NOT_ENOUGH_DATA; 4886 } 4887 mRsmpInIndex = 0; 4888 framesReady = mFrameCount; 4889 } 4890 4891 if (framesReq > framesReady) { 4892 framesReq = framesReady; 4893 } 4894 4895 if (mChannelCount == 1 && mReqChannelCount == 2) { 4896 channelCount = 1; 4897 } else { 4898 channelCount = 2; 4899 } 4900 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4901 buffer->frameCount = framesReq; 4902 return NO_ERROR; 4903} 4904 4905// AudioBufferProvider interface 4906void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4907{ 4908 mRsmpInIndex += buffer->frameCount; 4909 buffer->frameCount = 0; 4910} 4911 4912bool AudioFlinger::RecordThread::checkForNewParameters_l() 4913{ 4914 bool reconfig = false; 4915 4916 while (!mNewParameters.isEmpty()) { 4917 status_t status = NO_ERROR; 4918 String8 keyValuePair = mNewParameters[0]; 4919 AudioParameter param = AudioParameter(keyValuePair); 4920 int value; 4921 audio_format_t reqFormat = mFormat; 4922 uint32_t reqSamplingRate = mReqSampleRate; 4923 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 4924 4925 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4926 reqSamplingRate = value; 4927 reconfig = true; 4928 } 4929 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4930 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4931 status = BAD_VALUE; 4932 } else { 4933 reqFormat = (audio_format_t) value; 4934 reconfig = true; 4935 } 4936 } 4937 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4938 audio_channel_mask_t mask = (audio_channel_mask_t) value; 4939 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 4940 status = BAD_VALUE; 4941 } else { 4942 reqChannelMask = mask; 4943 reconfig = true; 4944 } 4945 } 4946 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4947 // do not accept frame count changes if tracks are open as the track buffer 4948 // size depends on frame count and correct behavior would not be guaranteed 4949 // if frame count is changed after track creation 4950 if (mActiveTrack != 0) { 4951 status = INVALID_OPERATION; 4952 } else { 4953 reconfig = true; 4954 } 4955 } 4956 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4957 // forward device change to effects that have requested to be 4958 // aware of attached audio device. 4959 for (size_t i = 0; i < mEffectChains.size(); i++) { 4960 mEffectChains[i]->setDevice_l(value); 4961 } 4962 4963 // store input device and output device but do not forward output device to audio HAL. 4964 // Note that status is ignored by the caller for output device 4965 // (see AudioFlinger::setParameters() 4966 if (audio_is_output_devices(value)) { 4967 mOutDevice = value; 4968 status = BAD_VALUE; 4969 } else { 4970 mInDevice = value; 4971 // disable AEC and NS if the device is a BT SCO headset supporting those 4972 // pre processings 4973 if (mTracks.size() > 0) { 4974 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4975 mAudioFlinger->btNrecIsOff(); 4976 for (size_t i = 0; i < mTracks.size(); i++) { 4977 sp<RecordTrack> track = mTracks[i]; 4978 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4979 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4980 } 4981 } 4982 } 4983 } 4984 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4985 mAudioSource != (audio_source_t)value) { 4986 // forward device change to effects that have requested to be 4987 // aware of attached audio device. 4988 for (size_t i = 0; i < mEffectChains.size(); i++) { 4989 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4990 } 4991 mAudioSource = (audio_source_t)value; 4992 } 4993 4994 if (status == NO_ERROR) { 4995 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4996 keyValuePair.string()); 4997 if (status == INVALID_OPERATION) { 4998 inputStandBy(); 4999 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5000 keyValuePair.string()); 5001 } 5002 if (reconfig) { 5003 if (status == BAD_VALUE && 5004 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5005 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5006 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5007 <= (2 * reqSamplingRate)) && 5008 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5009 <= FCC_2 && 5010 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5011 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5012 status = NO_ERROR; 5013 } 5014 if (status == NO_ERROR) { 5015 readInputParameters(); 5016 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5017 } 5018 } 5019 } 5020 5021 mNewParameters.removeAt(0); 5022 5023 mParamStatus = status; 5024 mParamCond.signal(); 5025 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5026 // already timed out waiting for the status and will never signal the condition. 5027 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5028 } 5029 return reconfig; 5030} 5031 5032String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5033{ 5034 Mutex::Autolock _l(mLock); 5035 if (initCheck() != NO_ERROR) { 5036 return String8(); 5037 } 5038 5039 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5040 const String8 out_s8(s); 5041 free(s); 5042 return out_s8; 5043} 5044 5045void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5046 AudioSystem::OutputDescriptor desc; 5047 void *param2 = NULL; 5048 5049 switch (event) { 5050 case AudioSystem::INPUT_OPENED: 5051 case AudioSystem::INPUT_CONFIG_CHANGED: 5052 desc.channelMask = mChannelMask; 5053 desc.samplingRate = mSampleRate; 5054 desc.format = mFormat; 5055 desc.frameCount = mFrameCount; 5056 desc.latency = 0; 5057 param2 = &desc; 5058 break; 5059 5060 case AudioSystem::INPUT_CLOSED: 5061 default: 5062 break; 5063 } 5064 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5065} 5066 5067void AudioFlinger::RecordThread::readInputParameters() 5068{ 5069 delete[] mRsmpInBuffer; 5070 // mRsmpInBuffer is always assigned a new[] below 5071 delete[] mRsmpOutBuffer; 5072 mRsmpOutBuffer = NULL; 5073 delete mResampler; 5074 mResampler = NULL; 5075 5076 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5077 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5078 mChannelCount = popcount(mChannelMask); 5079 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5080 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5081 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5082 } 5083 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5084 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5085 mFrameCount = mBufferSize / mFrameSize; 5086 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5087 5088 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5089 int channelCount; 5090 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5091 // stereo to mono post process as the resampler always outputs stereo. 5092 if (mChannelCount == 1 && mReqChannelCount == 2) { 5093 channelCount = 1; 5094 } else { 5095 channelCount = 2; 5096 } 5097 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5098 mResampler->setSampleRate(mSampleRate); 5099 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5100 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5101 5102 // optmization: if mono to mono, alter input frame count as if we were inputing 5103 // stereo samples 5104 if (mChannelCount == 1 && mReqChannelCount == 1) { 5105 mFrameCount >>= 1; 5106 } 5107 5108 } 5109 mRsmpInIndex = mFrameCount; 5110} 5111 5112unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5113{ 5114 Mutex::Autolock _l(mLock); 5115 if (initCheck() != NO_ERROR) { 5116 return 0; 5117 } 5118 5119 return mInput->stream->get_input_frames_lost(mInput->stream); 5120} 5121 5122uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5123{ 5124 Mutex::Autolock _l(mLock); 5125 uint32_t result = 0; 5126 if (getEffectChain_l(sessionId) != 0) { 5127 result = EFFECT_SESSION; 5128 } 5129 5130 for (size_t i = 0; i < mTracks.size(); ++i) { 5131 if (sessionId == mTracks[i]->sessionId()) { 5132 result |= TRACK_SESSION; 5133 break; 5134 } 5135 } 5136 5137 return result; 5138} 5139 5140KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5141{ 5142 KeyedVector<int, bool> ids; 5143 Mutex::Autolock _l(mLock); 5144 for (size_t j = 0; j < mTracks.size(); ++j) { 5145 sp<RecordThread::RecordTrack> track = mTracks[j]; 5146 int sessionId = track->sessionId(); 5147 if (ids.indexOfKey(sessionId) < 0) { 5148 ids.add(sessionId, true); 5149 } 5150 } 5151 return ids; 5152} 5153 5154AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5155{ 5156 Mutex::Autolock _l(mLock); 5157 AudioStreamIn *input = mInput; 5158 mInput = NULL; 5159 return input; 5160} 5161 5162// this method must always be called either with ThreadBase mLock held or inside the thread loop 5163audio_stream_t* AudioFlinger::RecordThread::stream() const 5164{ 5165 if (mInput == NULL) { 5166 return NULL; 5167 } 5168 return &mInput->stream->common; 5169} 5170 5171status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5172{ 5173 // only one chain per input thread 5174 if (mEffectChains.size() != 0) { 5175 return INVALID_OPERATION; 5176 } 5177 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5178 5179 chain->setInBuffer(NULL); 5180 chain->setOutBuffer(NULL); 5181 5182 checkSuspendOnAddEffectChain_l(chain); 5183 5184 mEffectChains.add(chain); 5185 5186 return NO_ERROR; 5187} 5188 5189size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5190{ 5191 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5192 ALOGW_IF(mEffectChains.size() != 1, 5193 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5194 chain.get(), mEffectChains.size(), this); 5195 if (mEffectChains.size() == 1) { 5196 mEffectChains.removeAt(0); 5197 } 5198 return 0; 5199} 5200 5201}; // namespace android 5202