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