Threads.cpp revision 1ab85ec401801ef9a9184650d0f5a1639b45eeb9
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 <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#undef ADD_BATTERY_DATA 58 59#ifdef ADD_BATTERY_DATA 60#include <media/IMediaPlayerService.h> 61#include <media/IMediaDeathNotifier.h> 62#endif 63 64// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 65#ifdef DEBUG_CPU_USAGE 66#include <cpustats/CentralTendencyStatistics.h> 67#include <cpustats/ThreadCpuUsage.h> 68#endif 69 70// ---------------------------------------------------------------------------- 71 72// Note: the following macro is used for extremely verbose logging message. In 73// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 74// 0; but one side effect of this is to turn all LOGV's as well. Some messages 75// are so verbose that we want to suppress them even when we have ALOG_ASSERT 76// turned on. Do not uncomment the #def below unless you really know what you 77// are doing and want to see all of the extremely verbose messages. 78//#define VERY_VERY_VERBOSE_LOGGING 79#ifdef VERY_VERY_VERBOSE_LOGGING 80#define ALOGVV ALOGV 81#else 82#define ALOGVV(a...) do { } while(0) 83#endif 84 85namespace android { 86 87// retry counts for buffer fill timeout 88// 50 * ~20msecs = 1 second 89static const int8_t kMaxTrackRetries = 50; 90static const int8_t kMaxTrackStartupRetries = 50; 91// allow less retry attempts on direct output thread. 92// direct outputs can be a scarce resource in audio hardware and should 93// be released as quickly as possible. 94static const int8_t kMaxTrackRetriesDirect = 2; 95 96// don't warn about blocked writes or record buffer overflows more often than this 97static const nsecs_t kWarningThrottleNs = seconds(5); 98 99// RecordThread loop sleep time upon application overrun or audio HAL read error 100static const int kRecordThreadSleepUs = 5000; 101 102// maximum time to wait for setParameters to complete 103static const nsecs_t kSetParametersTimeoutNs = seconds(2); 104 105// minimum sleep time for the mixer thread loop when tracks are active but in underrun 106static const uint32_t kMinThreadSleepTimeUs = 5000; 107// maximum divider applied to the active sleep time in the mixer thread loop 108static const uint32_t kMaxThreadSleepTimeShift = 2; 109 110// minimum normal mix buffer size, expressed in milliseconds rather than frames 111static const uint32_t kMinNormalMixBufferSizeMs = 20; 112// maximum normal mix buffer size 113static const uint32_t kMaxNormalMixBufferSizeMs = 24; 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 272 // mChannelMask 273 mChannelCount(0), 274 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 275 mParamStatus(NO_ERROR), 276 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 277 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 278 // mName will be set by concrete (non-virtual) subclass 279 mDeathRecipient(new PMDeathRecipient(this)) 280{ 281} 282 283AudioFlinger::ThreadBase::~ThreadBase() 284{ 285 mParamCond.broadcast(); 286 // do not lock the mutex in destructor 287 releaseWakeLock_l(); 288 if (mPowerManager != 0) { 289 sp<IBinder> binder = mPowerManager->asBinder(); 290 binder->unlinkToDeath(mDeathRecipient); 291 } 292} 293 294void AudioFlinger::ThreadBase::exit() 295{ 296 ALOGV("ThreadBase::exit"); 297 // do any cleanup required for exit to succeed 298 preExit(); 299 { 300 // This lock prevents the following race in thread (uniprocessor for illustration): 301 // if (!exitPending()) { 302 // // context switch from here to exit() 303 // // exit() calls requestExit(), what exitPending() observes 304 // // exit() calls signal(), which is dropped since no waiters 305 // // context switch back from exit() to here 306 // mWaitWorkCV.wait(...); 307 // // now thread is hung 308 // } 309 AutoMutex lock(mLock); 310 requestExit(); 311 mWaitWorkCV.broadcast(); 312 } 313 // When Thread::requestExitAndWait is made virtual and this method is renamed to 314 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 315 requestExitAndWait(); 316} 317 318status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 319{ 320 status_t status; 321 322 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 323 Mutex::Autolock _l(mLock); 324 325 mNewParameters.add(keyValuePairs); 326 mWaitWorkCV.signal(); 327 // wait condition with timeout in case the thread loop has exited 328 // before the request could be processed 329 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 330 status = mParamStatus; 331 mWaitWorkCV.signal(); 332 } else { 333 status = TIMED_OUT; 334 } 335 return status; 336} 337 338void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 339{ 340 Mutex::Autolock _l(mLock); 341 sendIoConfigEvent_l(event, param); 342} 343 344// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 345void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 346{ 347 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 348 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 349 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 350 param); 351 mWaitWorkCV.signal(); 352} 353 354// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 355void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 356{ 357 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 358 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 359 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 360 mConfigEvents.size(), pid, tid, prio); 361 mWaitWorkCV.signal(); 362} 363 364void AudioFlinger::ThreadBase::processConfigEvents() 365{ 366 mLock.lock(); 367 while (!mConfigEvents.isEmpty()) { 368 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 369 ConfigEvent *event = mConfigEvents[0]; 370 mConfigEvents.removeAt(0); 371 // release mLock before locking AudioFlinger mLock: lock order is always 372 // AudioFlinger then ThreadBase to avoid cross deadlock 373 mLock.unlock(); 374 switch(event->type()) { 375 case CFG_EVENT_PRIO: { 376 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 377 // FIXME Need to understand why this has be done asynchronously 378 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 379 true /*asynchronous*/); 380 if (err != 0) { 381 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 382 "error %d", 383 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 384 } 385 } break; 386 case CFG_EVENT_IO: { 387 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 388 mAudioFlinger->mLock.lock(); 389 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 390 mAudioFlinger->mLock.unlock(); 391 } break; 392 default: 393 ALOGE("processConfigEvents() unknown event type %d", event->type()); 394 break; 395 } 396 delete event; 397 mLock.lock(); 398 } 399 mLock.unlock(); 400} 401 402void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 403{ 404 const size_t SIZE = 256; 405 char buffer[SIZE]; 406 String8 result; 407 408 bool locked = AudioFlinger::dumpTryLock(mLock); 409 if (!locked) { 410 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 411 write(fd, buffer, strlen(buffer)); 412 } 413 414 snprintf(buffer, SIZE, "io handle: %d\n", mId); 415 result.append(buffer); 416 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 417 result.append(buffer); 418 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 419 result.append(buffer); 420 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 433 result.append(buffer); 434 435 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 436 result.append(buffer); 437 result.append(" Index Command"); 438 for (size_t i = 0; i < mNewParameters.size(); ++i) { 439 snprintf(buffer, SIZE, "\n %02d ", i); 440 result.append(buffer); 441 result.append(mNewParameters[i]); 442 } 443 444 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 445 result.append(buffer); 446 for (size_t i = 0; i < mConfigEvents.size(); i++) { 447 mConfigEvents[i]->dump(buffer, SIZE); 448 result.append(buffer); 449 } 450 result.append("\n"); 451 452 write(fd, result.string(), result.size()); 453 454 if (locked) { 455 mLock.unlock(); 456 } 457} 458 459void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 460{ 461 const size_t SIZE = 256; 462 char buffer[SIZE]; 463 String8 result; 464 465 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 466 write(fd, buffer, strlen(buffer)); 467 468 for (size_t i = 0; i < mEffectChains.size(); ++i) { 469 sp<EffectChain> chain = mEffectChains[i]; 470 if (chain != 0) { 471 chain->dump(fd, args); 472 } 473 } 474} 475 476void AudioFlinger::ThreadBase::acquireWakeLock() 477{ 478 Mutex::Autolock _l(mLock); 479 acquireWakeLock_l(); 480} 481 482void AudioFlinger::ThreadBase::acquireWakeLock_l() 483{ 484 if (mPowerManager == 0) { 485 // use checkService() to avoid blocking if power service is not up yet 486 sp<IBinder> binder = 487 defaultServiceManager()->checkService(String16("power")); 488 if (binder == 0) { 489 ALOGW("Thread %s cannot connect to the power manager service", mName); 490 } else { 491 mPowerManager = interface_cast<IPowerManager>(binder); 492 binder->linkToDeath(mDeathRecipient); 493 } 494 } 495 if (mPowerManager != 0) { 496 sp<IBinder> binder = new BBinder(); 497 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 498 binder, 499 String16(mName), 500 String16("media")); 501 if (status == NO_ERROR) { 502 mWakeLockToken = binder; 503 } 504 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 505 } 506} 507 508void AudioFlinger::ThreadBase::releaseWakeLock() 509{ 510 Mutex::Autolock _l(mLock); 511 releaseWakeLock_l(); 512} 513 514void AudioFlinger::ThreadBase::releaseWakeLock_l() 515{ 516 if (mWakeLockToken != 0) { 517 ALOGV("releaseWakeLock_l() %s", mName); 518 if (mPowerManager != 0) { 519 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 520 } 521 mWakeLockToken.clear(); 522 } 523} 524 525void AudioFlinger::ThreadBase::clearPowerManager() 526{ 527 Mutex::Autolock _l(mLock); 528 releaseWakeLock_l(); 529 mPowerManager.clear(); 530} 531 532void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 533{ 534 sp<ThreadBase> thread = mThread.promote(); 535 if (thread != 0) { 536 thread->clearPowerManager(); 537 } 538 ALOGW("power manager service died !!!"); 539} 540 541void AudioFlinger::ThreadBase::setEffectSuspended( 542 const effect_uuid_t *type, bool suspend, int sessionId) 543{ 544 Mutex::Autolock _l(mLock); 545 setEffectSuspended_l(type, suspend, sessionId); 546} 547 548void AudioFlinger::ThreadBase::setEffectSuspended_l( 549 const effect_uuid_t *type, bool suspend, int sessionId) 550{ 551 sp<EffectChain> chain = getEffectChain_l(sessionId); 552 if (chain != 0) { 553 if (type != NULL) { 554 chain->setEffectSuspended_l(type, suspend); 555 } else { 556 chain->setEffectSuspendedAll_l(suspend); 557 } 558 } 559 560 updateSuspendedSessions_l(type, suspend, sessionId); 561} 562 563void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 564{ 565 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 566 if (index < 0) { 567 return; 568 } 569 570 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 571 mSuspendedSessions.valueAt(index); 572 573 for (size_t i = 0; i < sessionEffects.size(); i++) { 574 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 575 for (int j = 0; j < desc->mRefCount; j++) { 576 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 577 chain->setEffectSuspendedAll_l(true); 578 } else { 579 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 580 desc->mType.timeLow); 581 chain->setEffectSuspended_l(&desc->mType, true); 582 } 583 } 584 } 585} 586 587void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 588 bool suspend, 589 int sessionId) 590{ 591 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 592 593 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 594 595 if (suspend) { 596 if (index >= 0) { 597 sessionEffects = mSuspendedSessions.valueAt(index); 598 } else { 599 mSuspendedSessions.add(sessionId, sessionEffects); 600 } 601 } else { 602 if (index < 0) { 603 return; 604 } 605 sessionEffects = mSuspendedSessions.valueAt(index); 606 } 607 608 609 int key = EffectChain::kKeyForSuspendAll; 610 if (type != NULL) { 611 key = type->timeLow; 612 } 613 index = sessionEffects.indexOfKey(key); 614 615 sp<SuspendedSessionDesc> desc; 616 if (suspend) { 617 if (index >= 0) { 618 desc = sessionEffects.valueAt(index); 619 } else { 620 desc = new SuspendedSessionDesc(); 621 if (type != NULL) { 622 desc->mType = *type; 623 } 624 sessionEffects.add(key, desc); 625 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 626 } 627 desc->mRefCount++; 628 } else { 629 if (index < 0) { 630 return; 631 } 632 desc = sessionEffects.valueAt(index); 633 if (--desc->mRefCount == 0) { 634 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 635 sessionEffects.removeItemsAt(index); 636 if (sessionEffects.isEmpty()) { 637 ALOGV("updateSuspendedSessions_l() restore removing session %d", 638 sessionId); 639 mSuspendedSessions.removeItem(sessionId); 640 } 641 } 642 } 643 if (!sessionEffects.isEmpty()) { 644 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 645 } 646} 647 648void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 649 bool enabled, 650 int sessionId) 651{ 652 Mutex::Autolock _l(mLock); 653 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 654} 655 656void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 657 bool enabled, 658 int sessionId) 659{ 660 if (mType != RECORD) { 661 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 662 // another session. This gives the priority to well behaved effect control panels 663 // and applications not using global effects. 664 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 665 // global effects 666 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 667 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 668 } 669 } 670 671 sp<EffectChain> chain = getEffectChain_l(sessionId); 672 if (chain != 0) { 673 chain->checkSuspendOnEffectEnabled(effect, enabled); 674 } 675} 676 677// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 678sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 679 const sp<AudioFlinger::Client>& client, 680 const sp<IEffectClient>& effectClient, 681 int32_t priority, 682 int sessionId, 683 effect_descriptor_t *desc, 684 int *enabled, 685 status_t *status 686 ) 687{ 688 sp<EffectModule> effect; 689 sp<EffectHandle> handle; 690 status_t lStatus; 691 sp<EffectChain> chain; 692 bool chainCreated = false; 693 bool effectCreated = false; 694 bool effectRegistered = false; 695 696 lStatus = initCheck(); 697 if (lStatus != NO_ERROR) { 698 ALOGW("createEffect_l() Audio driver not initialized."); 699 goto Exit; 700 } 701 702 // Do not allow effects with session ID 0 on direct output or duplicating threads 703 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 704 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 705 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 706 desc->name, sessionId); 707 lStatus = BAD_VALUE; 708 goto Exit; 709 } 710 // Only Pre processor effects are allowed on input threads and only on input threads 711 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 712 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 713 desc->name, desc->flags, mType); 714 lStatus = BAD_VALUE; 715 goto Exit; 716 } 717 718 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 719 720 { // scope for mLock 721 Mutex::Autolock _l(mLock); 722 723 // check for existing effect chain with the requested audio session 724 chain = getEffectChain_l(sessionId); 725 if (chain == 0) { 726 // create a new chain for this session 727 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 728 chain = new EffectChain(this, sessionId); 729 addEffectChain_l(chain); 730 chain->setStrategy(getStrategyForSession_l(sessionId)); 731 chainCreated = true; 732 } else { 733 effect = chain->getEffectFromDesc_l(desc); 734 } 735 736 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 737 738 if (effect == 0) { 739 int id = mAudioFlinger->nextUniqueId(); 740 // Check CPU and memory usage 741 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 742 if (lStatus != NO_ERROR) { 743 goto Exit; 744 } 745 effectRegistered = true; 746 // create a new effect module if none present in the chain 747 effect = new EffectModule(this, chain, desc, id, sessionId); 748 lStatus = effect->status(); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 lStatus = chain->addEffect_l(effect); 753 if (lStatus != NO_ERROR) { 754 goto Exit; 755 } 756 effectCreated = true; 757 758 effect->setDevice(mOutDevice); 759 effect->setDevice(mInDevice); 760 effect->setMode(mAudioFlinger->getMode()); 761 effect->setAudioSource(mAudioSource); 762 } 763 // create effect handle and connect it to effect module 764 handle = new EffectHandle(effect, client, effectClient, priority); 765 lStatus = effect->addHandle(handle.get()); 766 if (enabled != NULL) { 767 *enabled = (int)effect->isEnabled(); 768 } 769 } 770 771Exit: 772 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 773 Mutex::Autolock _l(mLock); 774 if (effectCreated) { 775 chain->removeEffect_l(effect); 776 } 777 if (effectRegistered) { 778 AudioSystem::unregisterEffect(effect->id()); 779 } 780 if (chainCreated) { 781 removeEffectChain_l(chain); 782 } 783 handle.clear(); 784 } 785 786 if (status != NULL) { 787 *status = lStatus; 788 } 789 return handle; 790} 791 792sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 793{ 794 Mutex::Autolock _l(mLock); 795 return getEffect_l(sessionId, effectId); 796} 797 798sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 799{ 800 sp<EffectChain> chain = getEffectChain_l(sessionId); 801 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 802} 803 804// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 805// PlaybackThread::mLock held 806status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 807{ 808 // check for existing effect chain with the requested audio session 809 int sessionId = effect->sessionId(); 810 sp<EffectChain> chain = getEffectChain_l(sessionId); 811 bool chainCreated = false; 812 813 if (chain == 0) { 814 // create a new chain for this session 815 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 816 chain = new EffectChain(this, sessionId); 817 addEffectChain_l(chain); 818 chain->setStrategy(getStrategyForSession_l(sessionId)); 819 chainCreated = true; 820 } 821 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 822 823 if (chain->getEffectFromId_l(effect->id()) != 0) { 824 ALOGW("addEffect_l() %p effect %s already present in chain %p", 825 this, effect->desc().name, chain.get()); 826 return BAD_VALUE; 827 } 828 829 status_t status = chain->addEffect_l(effect); 830 if (status != NO_ERROR) { 831 if (chainCreated) { 832 removeEffectChain_l(chain); 833 } 834 return status; 835 } 836 837 effect->setDevice(mOutDevice); 838 effect->setDevice(mInDevice); 839 effect->setMode(mAudioFlinger->getMode()); 840 effect->setAudioSource(mAudioSource); 841 return NO_ERROR; 842} 843 844void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 845 846 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 847 effect_descriptor_t desc = effect->desc(); 848 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 849 detachAuxEffect_l(effect->id()); 850 } 851 852 sp<EffectChain> chain = effect->chain().promote(); 853 if (chain != 0) { 854 // remove effect chain if removing last effect 855 if (chain->removeEffect_l(effect) == 0) { 856 removeEffectChain_l(chain); 857 } 858 } else { 859 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 860 } 861} 862 863void AudioFlinger::ThreadBase::lockEffectChains_l( 864 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 865{ 866 effectChains = mEffectChains; 867 for (size_t i = 0; i < mEffectChains.size(); i++) { 868 mEffectChains[i]->lock(); 869 } 870} 871 872void AudioFlinger::ThreadBase::unlockEffectChains( 873 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 874{ 875 for (size_t i = 0; i < effectChains.size(); i++) { 876 effectChains[i]->unlock(); 877 } 878} 879 880sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 881{ 882 Mutex::Autolock _l(mLock); 883 return getEffectChain_l(sessionId); 884} 885 886sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 887{ 888 size_t size = mEffectChains.size(); 889 for (size_t i = 0; i < size; i++) { 890 if (mEffectChains[i]->sessionId() == sessionId) { 891 return mEffectChains[i]; 892 } 893 } 894 return 0; 895} 896 897void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 898{ 899 Mutex::Autolock _l(mLock); 900 size_t size = mEffectChains.size(); 901 for (size_t i = 0; i < size; i++) { 902 mEffectChains[i]->setMode_l(mode); 903 } 904} 905 906void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 907 EffectHandle *handle, 908 bool unpinIfLast) { 909 910 Mutex::Autolock _l(mLock); 911 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 912 // delete the effect module if removing last handle on it 913 if (effect->removeHandle(handle) == 0) { 914 if (!effect->isPinned() || unpinIfLast) { 915 removeEffect_l(effect); 916 AudioSystem::unregisterEffect(effect->id()); 917 } 918 } 919} 920 921// ---------------------------------------------------------------------------- 922// Playback 923// ---------------------------------------------------------------------------- 924 925AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 926 AudioStreamOut* output, 927 audio_io_handle_t id, 928 audio_devices_t device, 929 type_t type) 930 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 931 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 932 // mStreamTypes[] initialized in constructor body 933 mOutput(output), 934 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 935 mMixerStatus(MIXER_IDLE), 936 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 937 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 938 mScreenState(AudioFlinger::mScreenState), 939 // index 0 is reserved for normal mixer's submix 940 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 941{ 942 snprintf(mName, kNameLength, "AudioOut_%X", id); 943 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 944 945 // Assumes constructor is called by AudioFlinger with it's mLock held, but 946 // it would be safer to explicitly pass initial masterVolume/masterMute as 947 // parameter. 948 // 949 // If the HAL we are using has support for master volume or master mute, 950 // then do not attenuate or mute during mixing (just leave the volume at 1.0 951 // and the mute set to false). 952 mMasterVolume = audioFlinger->masterVolume_l(); 953 mMasterMute = audioFlinger->masterMute_l(); 954 if (mOutput && mOutput->audioHwDev) { 955 if (mOutput->audioHwDev->canSetMasterVolume()) { 956 mMasterVolume = 1.0; 957 } 958 959 if (mOutput->audioHwDev->canSetMasterMute()) { 960 mMasterMute = false; 961 } 962 } 963 964 readOutputParameters(); 965 966 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 967 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 968 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 969 stream = (audio_stream_type_t) (stream + 1)) { 970 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 971 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 972 } 973 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 974 // because mAudioFlinger doesn't have one to copy from 975} 976 977AudioFlinger::PlaybackThread::~PlaybackThread() 978{ 979 mAudioFlinger->unregisterWriter(mNBLogWriter); 980 delete [] mMixBuffer; 981} 982 983void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 984{ 985 dumpInternals(fd, args); 986 dumpTracks(fd, args); 987 dumpEffectChains(fd, args); 988} 989 990void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 991{ 992 const size_t SIZE = 256; 993 char buffer[SIZE]; 994 String8 result; 995 996 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 997 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 998 const stream_type_t *st = &mStreamTypes[i]; 999 if (i > 0) { 1000 result.appendFormat(", "); 1001 } 1002 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1003 if (st->mute) { 1004 result.append("M"); 1005 } 1006 } 1007 result.append("\n"); 1008 write(fd, result.string(), result.length()); 1009 result.clear(); 1010 1011 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1012 result.append(buffer); 1013 Track::appendDumpHeader(result); 1014 for (size_t i = 0; i < mTracks.size(); ++i) { 1015 sp<Track> track = mTracks[i]; 1016 if (track != 0) { 1017 track->dump(buffer, SIZE); 1018 result.append(buffer); 1019 } 1020 } 1021 1022 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1023 result.append(buffer); 1024 Track::appendDumpHeader(result); 1025 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1026 sp<Track> track = mActiveTracks[i].promote(); 1027 if (track != 0) { 1028 track->dump(buffer, SIZE); 1029 result.append(buffer); 1030 } 1031 } 1032 write(fd, result.string(), result.size()); 1033 1034 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1035 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1036 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1037 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1038} 1039 1040void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1041{ 1042 const size_t SIZE = 256; 1043 char buffer[SIZE]; 1044 String8 result; 1045 1046 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1047 result.append(buffer); 1048 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1049 ns2ms(systemTime() - mLastWriteTime)); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1056 result.append(buffer); 1057 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1058 result.append(buffer); 1059 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1060 result.append(buffer); 1061 write(fd, result.string(), result.size()); 1062 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1063 1064 dumpBase(fd, args); 1065} 1066 1067// Thread virtuals 1068status_t AudioFlinger::PlaybackThread::readyToRun() 1069{ 1070 status_t status = initCheck(); 1071 if (status == NO_ERROR) { 1072 ALOGI("AudioFlinger's thread %p ready to run", this); 1073 } else { 1074 ALOGE("No working audio driver found."); 1075 } 1076 return status; 1077} 1078 1079void AudioFlinger::PlaybackThread::onFirstRef() 1080{ 1081 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1082} 1083 1084// ThreadBase virtuals 1085void AudioFlinger::PlaybackThread::preExit() 1086{ 1087 ALOGV(" preExit()"); 1088 // FIXME this is using hard-coded strings but in the future, this functionality will be 1089 // converted to use audio HAL extensions required to support tunneling 1090 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1091} 1092 1093// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1094sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1095 const sp<AudioFlinger::Client>& client, 1096 audio_stream_type_t streamType, 1097 uint32_t sampleRate, 1098 audio_format_t format, 1099 audio_channel_mask_t channelMask, 1100 size_t frameCount, 1101 const sp<IMemory>& sharedBuffer, 1102 int sessionId, 1103 IAudioFlinger::track_flags_t *flags, 1104 pid_t tid, 1105 status_t *status) 1106{ 1107 sp<Track> track; 1108 status_t lStatus; 1109 1110 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1111 1112 // client expresses a preference for FAST, but we get the final say 1113 if (*flags & IAudioFlinger::TRACK_FAST) { 1114 if ( 1115 // not timed 1116 (!isTimed) && 1117 // either of these use cases: 1118 ( 1119 // use case 1: shared buffer with any frame count 1120 ( 1121 (sharedBuffer != 0) 1122 ) || 1123 // use case 2: callback handler and frame count is default or at least as large as HAL 1124 ( 1125 (tid != -1) && 1126 ((frameCount == 0) || 1127 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1128 ) 1129 ) && 1130 // PCM data 1131 audio_is_linear_pcm(format) && 1132 // mono or stereo 1133 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1134 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1135#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1136 // hardware sample rate 1137 (sampleRate == mSampleRate) && 1138#endif 1139 // normal mixer has an associated fast mixer 1140 hasFastMixer() && 1141 // there are sufficient fast track slots available 1142 (mFastTrackAvailMask != 0) 1143 // FIXME test that MixerThread for this fast track has a capable output HAL 1144 // FIXME add a permission test also? 1145 ) { 1146 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1147 if (frameCount == 0) { 1148 frameCount = mFrameCount * kFastTrackMultiplier; 1149 } 1150 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1151 frameCount, mFrameCount); 1152 } else { 1153 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1154 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1155 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1156 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1157 audio_is_linear_pcm(format), 1158 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1159 *flags &= ~IAudioFlinger::TRACK_FAST; 1160 // For compatibility with AudioTrack calculation, buffer depth is forced 1161 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1162 // This is probably too conservative, but legacy application code may depend on it. 1163 // If you change this calculation, also review the start threshold which is related. 1164 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1165 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1166 if (minBufCount < 2) { 1167 minBufCount = 2; 1168 } 1169 size_t minFrameCount = mNormalFrameCount * minBufCount; 1170 if (frameCount < minFrameCount) { 1171 frameCount = minFrameCount; 1172 } 1173 } 1174 } 1175 1176 if (mType == DIRECT) { 1177 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1178 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1179 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1180 "for output %p with format %d", 1181 sampleRate, format, channelMask, mOutput, mFormat); 1182 lStatus = BAD_VALUE; 1183 goto Exit; 1184 } 1185 } 1186 } else { 1187 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1188 if (sampleRate > mSampleRate*2) { 1189 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1190 lStatus = BAD_VALUE; 1191 goto Exit; 1192 } 1193 } 1194 1195 lStatus = initCheck(); 1196 if (lStatus != NO_ERROR) { 1197 ALOGE("Audio driver not initialized."); 1198 goto Exit; 1199 } 1200 1201 { // scope for mLock 1202 Mutex::Autolock _l(mLock); 1203 1204 // all tracks in same audio session must share the same routing strategy otherwise 1205 // conflicts will happen when tracks are moved from one output to another by audio policy 1206 // manager 1207 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1208 for (size_t i = 0; i < mTracks.size(); ++i) { 1209 sp<Track> t = mTracks[i]; 1210 if (t != 0 && !t->isOutputTrack()) { 1211 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1212 if (sessionId == t->sessionId() && strategy != actual) { 1213 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1214 strategy, actual); 1215 lStatus = BAD_VALUE; 1216 goto Exit; 1217 } 1218 } 1219 } 1220 1221 if (!isTimed) { 1222 track = new Track(this, client, streamType, sampleRate, format, 1223 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1224 } else { 1225 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1226 channelMask, frameCount, sharedBuffer, sessionId); 1227 } 1228 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1229 lStatus = NO_MEMORY; 1230 goto Exit; 1231 } 1232 mTracks.add(track); 1233 1234 sp<EffectChain> chain = getEffectChain_l(sessionId); 1235 if (chain != 0) { 1236 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1237 track->setMainBuffer(chain->inBuffer()); 1238 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1239 chain->incTrackCnt(); 1240 } 1241 1242 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1243 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1244 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1245 // so ask activity manager to do this on our behalf 1246 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1247 } 1248 } 1249 1250 lStatus = NO_ERROR; 1251 1252Exit: 1253 if (status) { 1254 *status = lStatus; 1255 } 1256 return track; 1257} 1258 1259uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1260{ 1261 return latency; 1262} 1263 1264uint32_t AudioFlinger::PlaybackThread::latency() const 1265{ 1266 Mutex::Autolock _l(mLock); 1267 return latency_l(); 1268} 1269uint32_t AudioFlinger::PlaybackThread::latency_l() const 1270{ 1271 if (initCheck() == NO_ERROR) { 1272 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1273 } else { 1274 return 0; 1275 } 1276} 1277 1278void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1279{ 1280 Mutex::Autolock _l(mLock); 1281 // Don't apply master volume in SW if our HAL can do it for us. 1282 if (mOutput && mOutput->audioHwDev && 1283 mOutput->audioHwDev->canSetMasterVolume()) { 1284 mMasterVolume = 1.0; 1285 } else { 1286 mMasterVolume = value; 1287 } 1288} 1289 1290void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1291{ 1292 Mutex::Autolock _l(mLock); 1293 // Don't apply master mute in SW if our HAL can do it for us. 1294 if (mOutput && mOutput->audioHwDev && 1295 mOutput->audioHwDev->canSetMasterMute()) { 1296 mMasterMute = false; 1297 } else { 1298 mMasterMute = muted; 1299 } 1300} 1301 1302void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1303{ 1304 Mutex::Autolock _l(mLock); 1305 mStreamTypes[stream].volume = value; 1306} 1307 1308void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1309{ 1310 Mutex::Autolock _l(mLock); 1311 mStreamTypes[stream].mute = muted; 1312} 1313 1314float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1315{ 1316 Mutex::Autolock _l(mLock); 1317 return mStreamTypes[stream].volume; 1318} 1319 1320// addTrack_l() must be called with ThreadBase::mLock held 1321status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1322{ 1323 status_t status = ALREADY_EXISTS; 1324 1325 // set retry count for buffer fill 1326 track->mRetryCount = kMaxTrackStartupRetries; 1327 if (mActiveTracks.indexOf(track) < 0) { 1328 // the track is newly added, make sure it fills up all its 1329 // buffers before playing. This is to ensure the client will 1330 // effectively get the latency it requested. 1331 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1332 track->mResetDone = false; 1333 track->mPresentationCompleteFrames = 0; 1334 mActiveTracks.add(track); 1335 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1336 if (chain != 0) { 1337 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1338 track->sessionId()); 1339 chain->incActiveTrackCnt(); 1340 } 1341 1342 status = NO_ERROR; 1343 } 1344 1345 ALOGV("mWaitWorkCV.broadcast"); 1346 mWaitWorkCV.broadcast(); 1347 1348 return status; 1349} 1350 1351// destroyTrack_l() must be called with ThreadBase::mLock held 1352void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1353{ 1354 track->mState = TrackBase::TERMINATED; 1355 // active tracks are removed by threadLoop() 1356 if (mActiveTracks.indexOf(track) < 0) { 1357 removeTrack_l(track); 1358 } 1359} 1360 1361void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1362{ 1363 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1364 mTracks.remove(track); 1365 deleteTrackName_l(track->name()); 1366 // redundant as track is about to be destroyed, for dumpsys only 1367 track->mName = -1; 1368 if (track->isFastTrack()) { 1369 int index = track->mFastIndex; 1370 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1371 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1372 mFastTrackAvailMask |= 1 << index; 1373 // redundant as track is about to be destroyed, for dumpsys only 1374 track->mFastIndex = -1; 1375 } 1376 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1377 if (chain != 0) { 1378 chain->decTrackCnt(); 1379 } 1380} 1381 1382String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1383{ 1384 String8 out_s8 = String8(""); 1385 char *s; 1386 1387 Mutex::Autolock _l(mLock); 1388 if (initCheck() != NO_ERROR) { 1389 return out_s8; 1390 } 1391 1392 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1393 out_s8 = String8(s); 1394 free(s); 1395 return out_s8; 1396} 1397 1398// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1399void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1400 AudioSystem::OutputDescriptor desc; 1401 void *param2 = NULL; 1402 1403 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1404 param); 1405 1406 switch (event) { 1407 case AudioSystem::OUTPUT_OPENED: 1408 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1409 desc.channels = mChannelMask; 1410 desc.samplingRate = mSampleRate; 1411 desc.format = mFormat; 1412 desc.frameCount = mNormalFrameCount; // FIXME see 1413 // AudioFlinger::frameCount(audio_io_handle_t) 1414 desc.latency = latency(); 1415 param2 = &desc; 1416 break; 1417 1418 case AudioSystem::STREAM_CONFIG_CHANGED: 1419 param2 = ¶m; 1420 case AudioSystem::OUTPUT_CLOSED: 1421 default: 1422 break; 1423 } 1424 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1425} 1426 1427void AudioFlinger::PlaybackThread::readOutputParameters() 1428{ 1429 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1430 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1431 mChannelCount = (uint16_t)popcount(mChannelMask); 1432 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1433 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1434 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1435 if (mFrameCount & 15) { 1436 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1437 mFrameCount); 1438 } 1439 1440 // Calculate size of normal mix buffer relative to the HAL output buffer size 1441 double multiplier = 1.0; 1442 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1443 kUseFastMixer == FastMixer_Dynamic)) { 1444 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1445 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1446 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1447 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1448 maxNormalFrameCount = maxNormalFrameCount & ~15; 1449 if (maxNormalFrameCount < minNormalFrameCount) { 1450 maxNormalFrameCount = minNormalFrameCount; 1451 } 1452 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1453 if (multiplier <= 1.0) { 1454 multiplier = 1.0; 1455 } else if (multiplier <= 2.0) { 1456 if (2 * mFrameCount <= maxNormalFrameCount) { 1457 multiplier = 2.0; 1458 } else { 1459 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1460 } 1461 } else { 1462 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1463 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1464 // track, but we sometimes have to do this to satisfy the maximum frame count 1465 // constraint) 1466 // FIXME this rounding up should not be done if no HAL SRC 1467 uint32_t truncMult = (uint32_t) multiplier; 1468 if ((truncMult & 1)) { 1469 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1470 ++truncMult; 1471 } 1472 } 1473 multiplier = (double) truncMult; 1474 } 1475 } 1476 mNormalFrameCount = multiplier * mFrameCount; 1477 // round up to nearest 16 frames to satisfy AudioMixer 1478 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1479 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1480 mNormalFrameCount); 1481 1482 delete[] mMixBuffer; 1483 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1484 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1485 1486 // force reconfiguration of effect chains and engines to take new buffer size and audio 1487 // parameters into account 1488 // Note that mLock is not held when readOutputParameters() is called from the constructor 1489 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1490 // matter. 1491 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1492 Vector< sp<EffectChain> > effectChains = mEffectChains; 1493 for (size_t i = 0; i < effectChains.size(); i ++) { 1494 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1495 } 1496} 1497 1498 1499status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1500{ 1501 if (halFrames == NULL || dspFrames == NULL) { 1502 return BAD_VALUE; 1503 } 1504 Mutex::Autolock _l(mLock); 1505 if (initCheck() != NO_ERROR) { 1506 return INVALID_OPERATION; 1507 } 1508 size_t framesWritten = mBytesWritten / mFrameSize; 1509 *halFrames = framesWritten; 1510 1511 if (isSuspended()) { 1512 // return an estimation of rendered frames when the output is suspended 1513 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1514 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1515 return NO_ERROR; 1516 } else { 1517 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1518 } 1519} 1520 1521uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1522{ 1523 Mutex::Autolock _l(mLock); 1524 uint32_t result = 0; 1525 if (getEffectChain_l(sessionId) != 0) { 1526 result = EFFECT_SESSION; 1527 } 1528 1529 for (size_t i = 0; i < mTracks.size(); ++i) { 1530 sp<Track> track = mTracks[i]; 1531 if (sessionId == track->sessionId() && !track->isInvalid()) { 1532 result |= TRACK_SESSION; 1533 break; 1534 } 1535 } 1536 1537 return result; 1538} 1539 1540uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1541{ 1542 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1543 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1544 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1545 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1546 } 1547 for (size_t i = 0; i < mTracks.size(); i++) { 1548 sp<Track> track = mTracks[i]; 1549 if (sessionId == track->sessionId() && !track->isInvalid()) { 1550 return AudioSystem::getStrategyForStream(track->streamType()); 1551 } 1552 } 1553 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1554} 1555 1556 1557AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1558{ 1559 Mutex::Autolock _l(mLock); 1560 return mOutput; 1561} 1562 1563AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1564{ 1565 Mutex::Autolock _l(mLock); 1566 AudioStreamOut *output = mOutput; 1567 mOutput = NULL; 1568 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1569 // must push a NULL and wait for ack 1570 mOutputSink.clear(); 1571 mPipeSink.clear(); 1572 mNormalSink.clear(); 1573 return output; 1574} 1575 1576// this method must always be called either with ThreadBase mLock held or inside the thread loop 1577audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1578{ 1579 if (mOutput == NULL) { 1580 return NULL; 1581 } 1582 return &mOutput->stream->common; 1583} 1584 1585uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1586{ 1587 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1588} 1589 1590status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1591{ 1592 if (!isValidSyncEvent(event)) { 1593 return BAD_VALUE; 1594 } 1595 1596 Mutex::Autolock _l(mLock); 1597 1598 for (size_t i = 0; i < mTracks.size(); ++i) { 1599 sp<Track> track = mTracks[i]; 1600 if (event->triggerSession() == track->sessionId()) { 1601 (void) track->setSyncEvent(event); 1602 return NO_ERROR; 1603 } 1604 } 1605 1606 return NAME_NOT_FOUND; 1607} 1608 1609bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1610{ 1611 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1612} 1613 1614void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1615 const Vector< sp<Track> >& tracksToRemove) 1616{ 1617 size_t count = tracksToRemove.size(); 1618 if (CC_UNLIKELY(count)) { 1619 for (size_t i = 0 ; i < count ; i++) { 1620 const sp<Track>& track = tracksToRemove.itemAt(i); 1621 if ((track->sharedBuffer() != 0) && 1622 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1623 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1624 } 1625 } 1626 } 1627 1628} 1629 1630void AudioFlinger::PlaybackThread::checkSilentMode_l() 1631{ 1632 if (!mMasterMute) { 1633 char value[PROPERTY_VALUE_MAX]; 1634 if (property_get("ro.audio.silent", value, "0") > 0) { 1635 char *endptr; 1636 unsigned long ul = strtoul(value, &endptr, 0); 1637 if (*endptr == '\0' && ul != 0) { 1638 ALOGD("Silence is golden"); 1639 // The setprop command will not allow a property to be changed after 1640 // the first time it is set, so we don't have to worry about un-muting. 1641 setMasterMute_l(true); 1642 } 1643 } 1644 } 1645} 1646 1647// shared by MIXER and DIRECT, overridden by DUPLICATING 1648void AudioFlinger::PlaybackThread::threadLoop_write() 1649{ 1650 // FIXME rewrite to reduce number of system calls 1651 mLastWriteTime = systemTime(); 1652 mInWrite = true; 1653 int bytesWritten; 1654 1655 // If an NBAIO sink is present, use it to write the normal mixer's submix 1656 if (mNormalSink != 0) { 1657#define mBitShift 2 // FIXME 1658 size_t count = mixBufferSize >> mBitShift; 1659 ATRACE_BEGIN("write"); 1660 // update the setpoint when AudioFlinger::mScreenState changes 1661 uint32_t screenState = AudioFlinger::mScreenState; 1662 if (screenState != mScreenState) { 1663 mScreenState = screenState; 1664 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1665 if (pipe != NULL) { 1666 pipe->setAvgFrames((mScreenState & 1) ? 1667 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1668 } 1669 } 1670 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1671 ATRACE_END(); 1672 if (framesWritten > 0) { 1673 bytesWritten = framesWritten << mBitShift; 1674 } else { 1675 bytesWritten = framesWritten; 1676 } 1677 // otherwise use the HAL / AudioStreamOut directly 1678 } else { 1679 // Direct output thread. 1680 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1681 } 1682 1683 if (bytesWritten > 0) { 1684 mBytesWritten += mixBufferSize; 1685 } 1686 mNumWrites++; 1687 mInWrite = false; 1688} 1689 1690/* 1691The derived values that are cached: 1692 - mixBufferSize from frame count * frame size 1693 - activeSleepTime from activeSleepTimeUs() 1694 - idleSleepTime from idleSleepTimeUs() 1695 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1696 - maxPeriod from frame count and sample rate (MIXER only) 1697 1698The parameters that affect these derived values are: 1699 - frame count 1700 - frame size 1701 - sample rate 1702 - device type: A2DP or not 1703 - device latency 1704 - format: PCM or not 1705 - active sleep time 1706 - idle sleep time 1707*/ 1708 1709void AudioFlinger::PlaybackThread::cacheParameters_l() 1710{ 1711 mixBufferSize = mNormalFrameCount * mFrameSize; 1712 activeSleepTime = activeSleepTimeUs(); 1713 idleSleepTime = idleSleepTimeUs(); 1714} 1715 1716void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1717{ 1718 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1719 this, streamType, mTracks.size()); 1720 Mutex::Autolock _l(mLock); 1721 1722 size_t size = mTracks.size(); 1723 for (size_t i = 0; i < size; i++) { 1724 sp<Track> t = mTracks[i]; 1725 if (t->streamType() == streamType) { 1726 t->invalidate(); 1727 } 1728 } 1729} 1730 1731status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1732{ 1733 int session = chain->sessionId(); 1734 int16_t *buffer = mMixBuffer; 1735 bool ownsBuffer = false; 1736 1737 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1738 if (session > 0) { 1739 // Only one effect chain can be present in direct output thread and it uses 1740 // the mix buffer as input 1741 if (mType != DIRECT) { 1742 size_t numSamples = mNormalFrameCount * mChannelCount; 1743 buffer = new int16_t[numSamples]; 1744 memset(buffer, 0, numSamples * sizeof(int16_t)); 1745 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1746 ownsBuffer = true; 1747 } 1748 1749 // Attach all tracks with same session ID to this chain. 1750 for (size_t i = 0; i < mTracks.size(); ++i) { 1751 sp<Track> track = mTracks[i]; 1752 if (session == track->sessionId()) { 1753 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1754 buffer); 1755 track->setMainBuffer(buffer); 1756 chain->incTrackCnt(); 1757 } 1758 } 1759 1760 // indicate all active tracks in the chain 1761 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1762 sp<Track> track = mActiveTracks[i].promote(); 1763 if (track == 0) { 1764 continue; 1765 } 1766 if (session == track->sessionId()) { 1767 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1768 chain->incActiveTrackCnt(); 1769 } 1770 } 1771 } 1772 1773 chain->setInBuffer(buffer, ownsBuffer); 1774 chain->setOutBuffer(mMixBuffer); 1775 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1776 // chains list in order to be processed last as it contains output stage effects 1777 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1778 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1779 // after track specific effects and before output stage 1780 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1781 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1782 // Effect chain for other sessions are inserted at beginning of effect 1783 // chains list to be processed before output mix effects. Relative order between other 1784 // sessions is not important 1785 size_t size = mEffectChains.size(); 1786 size_t i = 0; 1787 for (i = 0; i < size; i++) { 1788 if (mEffectChains[i]->sessionId() < session) { 1789 break; 1790 } 1791 } 1792 mEffectChains.insertAt(chain, i); 1793 checkSuspendOnAddEffectChain_l(chain); 1794 1795 return NO_ERROR; 1796} 1797 1798size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1799{ 1800 int session = chain->sessionId(); 1801 1802 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1803 1804 for (size_t i = 0; i < mEffectChains.size(); i++) { 1805 if (chain == mEffectChains[i]) { 1806 mEffectChains.removeAt(i); 1807 // detach all active tracks from the chain 1808 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1809 sp<Track> track = mActiveTracks[i].promote(); 1810 if (track == 0) { 1811 continue; 1812 } 1813 if (session == track->sessionId()) { 1814 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1815 chain.get(), session); 1816 chain->decActiveTrackCnt(); 1817 } 1818 } 1819 1820 // detach all tracks with same session ID from this chain 1821 for (size_t i = 0; i < mTracks.size(); ++i) { 1822 sp<Track> track = mTracks[i]; 1823 if (session == track->sessionId()) { 1824 track->setMainBuffer(mMixBuffer); 1825 chain->decTrackCnt(); 1826 } 1827 } 1828 break; 1829 } 1830 } 1831 return mEffectChains.size(); 1832} 1833 1834status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1835 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1836{ 1837 Mutex::Autolock _l(mLock); 1838 return attachAuxEffect_l(track, EffectId); 1839} 1840 1841status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1842 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1843{ 1844 status_t status = NO_ERROR; 1845 1846 if (EffectId == 0) { 1847 track->setAuxBuffer(0, NULL); 1848 } else { 1849 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1850 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1851 if (effect != 0) { 1852 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1853 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1854 } else { 1855 status = INVALID_OPERATION; 1856 } 1857 } else { 1858 status = BAD_VALUE; 1859 } 1860 } 1861 return status; 1862} 1863 1864void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1865{ 1866 for (size_t i = 0; i < mTracks.size(); ++i) { 1867 sp<Track> track = mTracks[i]; 1868 if (track->auxEffectId() == effectId) { 1869 attachAuxEffect_l(track, 0); 1870 } 1871 } 1872} 1873 1874bool AudioFlinger::PlaybackThread::threadLoop() 1875{ 1876 Vector< sp<Track> > tracksToRemove; 1877 1878 standbyTime = systemTime(); 1879 1880 // MIXER 1881 nsecs_t lastWarning = 0; 1882 1883 // DUPLICATING 1884 // FIXME could this be made local to while loop? 1885 writeFrames = 0; 1886 1887 cacheParameters_l(); 1888 sleepTime = idleSleepTime; 1889 1890 if (mType == MIXER) { 1891 sleepTimeShift = 0; 1892 } 1893 1894 CpuStats cpuStats; 1895 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1896 1897 acquireWakeLock(); 1898 1899 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1900 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1901 // and then that string will be logged at the next convenient opportunity. 1902 const char *logString = NULL; 1903 1904 while (!exitPending()) 1905 { 1906 cpuStats.sample(myName); 1907 1908 Vector< sp<EffectChain> > effectChains; 1909 1910 processConfigEvents(); 1911 1912 { // scope for mLock 1913 1914 Mutex::Autolock _l(mLock); 1915 1916 if (logString != NULL) { 1917 mNBLogWriter->logTimestamp(); 1918 mNBLogWriter->log(logString); 1919 logString = NULL; 1920 } 1921 1922 if (checkForNewParameters_l()) { 1923 cacheParameters_l(); 1924 } 1925 1926 saveOutputTracks(); 1927 1928 // put audio hardware into standby after short delay 1929 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1930 isSuspended())) { 1931 if (!mStandby) { 1932 1933 threadLoop_standby(); 1934 1935 mStandby = true; 1936 } 1937 1938 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1939 // we're about to wait, flush the binder command buffer 1940 IPCThreadState::self()->flushCommands(); 1941 1942 clearOutputTracks(); 1943 1944 if (exitPending()) { 1945 break; 1946 } 1947 1948 releaseWakeLock_l(); 1949 // wait until we have something to do... 1950 ALOGV("%s going to sleep", myName.string()); 1951 mWaitWorkCV.wait(mLock); 1952 ALOGV("%s waking up", myName.string()); 1953 acquireWakeLock_l(); 1954 1955 mMixerStatus = MIXER_IDLE; 1956 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1957 mBytesWritten = 0; 1958 1959 checkSilentMode_l(); 1960 1961 standbyTime = systemTime() + standbyDelay; 1962 sleepTime = idleSleepTime; 1963 if (mType == MIXER) { 1964 sleepTimeShift = 0; 1965 } 1966 1967 continue; 1968 } 1969 } 1970 1971 // mMixerStatusIgnoringFastTracks is also updated internally 1972 mMixerStatus = prepareTracks_l(&tracksToRemove); 1973 1974 // prevent any changes in effect chain list and in each effect chain 1975 // during mixing and effect process as the audio buffers could be deleted 1976 // or modified if an effect is created or deleted 1977 lockEffectChains_l(effectChains); 1978 } 1979 1980 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1981 threadLoop_mix(); 1982 } else { 1983 threadLoop_sleepTime(); 1984 } 1985 1986 if (isSuspended()) { 1987 sleepTime = suspendSleepTimeUs(); 1988 mBytesWritten += mixBufferSize; 1989 } 1990 1991 // only process effects if we're going to write 1992 if (sleepTime == 0) { 1993 for (size_t i = 0; i < effectChains.size(); i ++) { 1994 effectChains[i]->process_l(); 1995 } 1996 } 1997 1998 // enable changes in effect chain 1999 unlockEffectChains(effectChains); 2000 2001 // sleepTime == 0 means we must write to audio hardware 2002 if (sleepTime == 0) { 2003 2004 threadLoop_write(); 2005 2006if (mType == MIXER) { 2007 // write blocked detection 2008 nsecs_t now = systemTime(); 2009 nsecs_t delta = now - mLastWriteTime; 2010 if (!mStandby && delta > maxPeriod) { 2011 mNumDelayedWrites++; 2012 if ((now - lastWarning) > kWarningThrottleNs) { 2013 ATRACE_NAME("underrun"); 2014 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2015 ns2ms(delta), mNumDelayedWrites, this); 2016 lastWarning = now; 2017 } 2018 } 2019} 2020 2021 mStandby = false; 2022 } else { 2023 usleep(sleepTime); 2024 } 2025 2026 // Finally let go of removed track(s), without the lock held 2027 // since we can't guarantee the destructors won't acquire that 2028 // same lock. This will also mutate and push a new fast mixer state. 2029 threadLoop_removeTracks(tracksToRemove); 2030 tracksToRemove.clear(); 2031 2032 // FIXME I don't understand the need for this here; 2033 // it was in the original code but maybe the 2034 // assignment in saveOutputTracks() makes this unnecessary? 2035 clearOutputTracks(); 2036 2037 // Effect chains will be actually deleted here if they were removed from 2038 // mEffectChains list during mixing or effects processing 2039 effectChains.clear(); 2040 2041 // FIXME Note that the above .clear() is no longer necessary since effectChains 2042 // is now local to this block, but will keep it for now (at least until merge done). 2043 } 2044 2045 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2046 if (mType == MIXER || mType == DIRECT) { 2047 // put output stream into standby mode 2048 if (!mStandby) { 2049 mOutput->stream->common.standby(&mOutput->stream->common); 2050 } 2051 } 2052 2053 releaseWakeLock(); 2054 2055 ALOGV("Thread %p type %d exiting", this, mType); 2056 return false; 2057} 2058 2059 2060// ---------------------------------------------------------------------------- 2061 2062AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2063 audio_io_handle_t id, audio_devices_t device, type_t type) 2064 : PlaybackThread(audioFlinger, output, id, device, type), 2065 // mAudioMixer below 2066 // mFastMixer below 2067 mFastMixerFutex(0) 2068 // mOutputSink below 2069 // mPipeSink below 2070 // mNormalSink below 2071{ 2072 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2073 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2074 "mFrameCount=%d, mNormalFrameCount=%d", 2075 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2076 mNormalFrameCount); 2077 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2078 2079 // FIXME - Current mixer implementation only supports stereo output 2080 if (mChannelCount != FCC_2) { 2081 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2082 } 2083 2084 // create an NBAIO sink for the HAL output stream, and negotiate 2085 mOutputSink = new AudioStreamOutSink(output->stream); 2086 size_t numCounterOffers = 0; 2087 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2088 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2089 ALOG_ASSERT(index == 0); 2090 2091 // initialize fast mixer depending on configuration 2092 bool initFastMixer; 2093 switch (kUseFastMixer) { 2094 case FastMixer_Never: 2095 initFastMixer = false; 2096 break; 2097 case FastMixer_Always: 2098 initFastMixer = true; 2099 break; 2100 case FastMixer_Static: 2101 case FastMixer_Dynamic: 2102 initFastMixer = mFrameCount < mNormalFrameCount; 2103 break; 2104 } 2105 if (initFastMixer) { 2106 2107 // create a MonoPipe to connect our submix to FastMixer 2108 NBAIO_Format format = mOutputSink->format(); 2109 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2110 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2111 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2112 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2113 const NBAIO_Format offers[1] = {format}; 2114 size_t numCounterOffers = 0; 2115 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2116 ALOG_ASSERT(index == 0); 2117 monoPipe->setAvgFrames((mScreenState & 1) ? 2118 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2119 mPipeSink = monoPipe; 2120 2121#ifdef TEE_SINK 2122 if (mTeeSinkOutputEnabled) { 2123 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2124 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2125 numCounterOffers = 0; 2126 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2127 ALOG_ASSERT(index == 0); 2128 mTeeSink = teeSink; 2129 PipeReader *teeSource = new PipeReader(*teeSink); 2130 numCounterOffers = 0; 2131 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2132 ALOG_ASSERT(index == 0); 2133 mTeeSource = teeSource; 2134 } 2135#endif 2136 2137 // create fast mixer and configure it initially with just one fast track for our submix 2138 mFastMixer = new FastMixer(); 2139 FastMixerStateQueue *sq = mFastMixer->sq(); 2140#ifdef STATE_QUEUE_DUMP 2141 sq->setObserverDump(&mStateQueueObserverDump); 2142 sq->setMutatorDump(&mStateQueueMutatorDump); 2143#endif 2144 FastMixerState *state = sq->begin(); 2145 FastTrack *fastTrack = &state->mFastTracks[0]; 2146 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2147 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2148 fastTrack->mVolumeProvider = NULL; 2149 fastTrack->mGeneration++; 2150 state->mFastTracksGen++; 2151 state->mTrackMask = 1; 2152 // fast mixer will use the HAL output sink 2153 state->mOutputSink = mOutputSink.get(); 2154 state->mOutputSinkGen++; 2155 state->mFrameCount = mFrameCount; 2156 state->mCommand = FastMixerState::COLD_IDLE; 2157 // already done in constructor initialization list 2158 //mFastMixerFutex = 0; 2159 state->mColdFutexAddr = &mFastMixerFutex; 2160 state->mColdGen++; 2161 state->mDumpState = &mFastMixerDumpState; 2162#ifdef TEE_SINK 2163 state->mTeeSink = mTeeSink.get(); 2164#endif 2165 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2166 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2167 sq->end(); 2168 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2169 2170 // start the fast mixer 2171 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2172 pid_t tid = mFastMixer->getTid(); 2173 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2174 if (err != 0) { 2175 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2176 kPriorityFastMixer, getpid_cached, tid, err); 2177 } 2178 2179#ifdef AUDIO_WATCHDOG 2180 // create and start the watchdog 2181 mAudioWatchdog = new AudioWatchdog(); 2182 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2183 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2184 tid = mAudioWatchdog->getTid(); 2185 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2186 if (err != 0) { 2187 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2188 kPriorityFastMixer, getpid_cached, tid, err); 2189 } 2190#endif 2191 2192 } else { 2193 mFastMixer = NULL; 2194 } 2195 2196 switch (kUseFastMixer) { 2197 case FastMixer_Never: 2198 case FastMixer_Dynamic: 2199 mNormalSink = mOutputSink; 2200 break; 2201 case FastMixer_Always: 2202 mNormalSink = mPipeSink; 2203 break; 2204 case FastMixer_Static: 2205 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2206 break; 2207 } 2208} 2209 2210AudioFlinger::MixerThread::~MixerThread() 2211{ 2212 if (mFastMixer != NULL) { 2213 FastMixerStateQueue *sq = mFastMixer->sq(); 2214 FastMixerState *state = sq->begin(); 2215 if (state->mCommand == FastMixerState::COLD_IDLE) { 2216 int32_t old = android_atomic_inc(&mFastMixerFutex); 2217 if (old == -1) { 2218 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2219 } 2220 } 2221 state->mCommand = FastMixerState::EXIT; 2222 sq->end(); 2223 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2224 mFastMixer->join(); 2225 // Though the fast mixer thread has exited, it's state queue is still valid. 2226 // We'll use that extract the final state which contains one remaining fast track 2227 // corresponding to our sub-mix. 2228 state = sq->begin(); 2229 ALOG_ASSERT(state->mTrackMask == 1); 2230 FastTrack *fastTrack = &state->mFastTracks[0]; 2231 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2232 delete fastTrack->mBufferProvider; 2233 sq->end(false /*didModify*/); 2234 delete mFastMixer; 2235#ifdef AUDIO_WATCHDOG 2236 if (mAudioWatchdog != 0) { 2237 mAudioWatchdog->requestExit(); 2238 mAudioWatchdog->requestExitAndWait(); 2239 mAudioWatchdog.clear(); 2240 } 2241#endif 2242 } 2243 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2244 delete mAudioMixer; 2245} 2246 2247 2248uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2249{ 2250 if (mFastMixer != NULL) { 2251 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2252 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2253 } 2254 return latency; 2255} 2256 2257 2258void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2259{ 2260 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2261} 2262 2263void AudioFlinger::MixerThread::threadLoop_write() 2264{ 2265 // FIXME we should only do one push per cycle; confirm this is true 2266 // Start the fast mixer if it's not already running 2267 if (mFastMixer != NULL) { 2268 FastMixerStateQueue *sq = mFastMixer->sq(); 2269 FastMixerState *state = sq->begin(); 2270 if (state->mCommand != FastMixerState::MIX_WRITE && 2271 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2272 if (state->mCommand == FastMixerState::COLD_IDLE) { 2273 int32_t old = android_atomic_inc(&mFastMixerFutex); 2274 if (old == -1) { 2275 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2276 } 2277#ifdef AUDIO_WATCHDOG 2278 if (mAudioWatchdog != 0) { 2279 mAudioWatchdog->resume(); 2280 } 2281#endif 2282 } 2283 state->mCommand = FastMixerState::MIX_WRITE; 2284 sq->end(); 2285 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2286 if (kUseFastMixer == FastMixer_Dynamic) { 2287 mNormalSink = mPipeSink; 2288 } 2289 } else { 2290 sq->end(false /*didModify*/); 2291 } 2292 } 2293 PlaybackThread::threadLoop_write(); 2294} 2295 2296void AudioFlinger::MixerThread::threadLoop_standby() 2297{ 2298 // Idle the fast mixer if it's currently running 2299 if (mFastMixer != NULL) { 2300 FastMixerStateQueue *sq = mFastMixer->sq(); 2301 FastMixerState *state = sq->begin(); 2302 if (!(state->mCommand & FastMixerState::IDLE)) { 2303 state->mCommand = FastMixerState::COLD_IDLE; 2304 state->mColdFutexAddr = &mFastMixerFutex; 2305 state->mColdGen++; 2306 mFastMixerFutex = 0; 2307 sq->end(); 2308 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2309 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2310 if (kUseFastMixer == FastMixer_Dynamic) { 2311 mNormalSink = mOutputSink; 2312 } 2313#ifdef AUDIO_WATCHDOG 2314 if (mAudioWatchdog != 0) { 2315 mAudioWatchdog->pause(); 2316 } 2317#endif 2318 } else { 2319 sq->end(false /*didModify*/); 2320 } 2321 } 2322 PlaybackThread::threadLoop_standby(); 2323} 2324 2325// shared by MIXER and DIRECT, overridden by DUPLICATING 2326void AudioFlinger::PlaybackThread::threadLoop_standby() 2327{ 2328 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2329 mOutput->stream->common.standby(&mOutput->stream->common); 2330} 2331 2332void AudioFlinger::MixerThread::threadLoop_mix() 2333{ 2334 // obtain the presentation timestamp of the next output buffer 2335 int64_t pts; 2336 status_t status = INVALID_OPERATION; 2337 2338 if (mNormalSink != 0) { 2339 status = mNormalSink->getNextWriteTimestamp(&pts); 2340 } else { 2341 status = mOutputSink->getNextWriteTimestamp(&pts); 2342 } 2343 2344 if (status != NO_ERROR) { 2345 pts = AudioBufferProvider::kInvalidPTS; 2346 } 2347 2348 // mix buffers... 2349 mAudioMixer->process(pts); 2350 // increase sleep time progressively when application underrun condition clears. 2351 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2352 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2353 // such that we would underrun the audio HAL. 2354 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2355 sleepTimeShift--; 2356 } 2357 sleepTime = 0; 2358 standbyTime = systemTime() + standbyDelay; 2359 //TODO: delay standby when effects have a tail 2360} 2361 2362void AudioFlinger::MixerThread::threadLoop_sleepTime() 2363{ 2364 // If no tracks are ready, sleep once for the duration of an output 2365 // buffer size, then write 0s to the output 2366 if (sleepTime == 0) { 2367 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2368 sleepTime = activeSleepTime >> sleepTimeShift; 2369 if (sleepTime < kMinThreadSleepTimeUs) { 2370 sleepTime = kMinThreadSleepTimeUs; 2371 } 2372 // reduce sleep time in case of consecutive application underruns to avoid 2373 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2374 // duration we would end up writing less data than needed by the audio HAL if 2375 // the condition persists. 2376 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2377 sleepTimeShift++; 2378 } 2379 } else { 2380 sleepTime = idleSleepTime; 2381 } 2382 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2383 memset (mMixBuffer, 0, mixBufferSize); 2384 sleepTime = 0; 2385 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2386 "anticipated start"); 2387 } 2388 // TODO add standby time extension fct of effect tail 2389} 2390 2391// prepareTracks_l() must be called with ThreadBase::mLock held 2392AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2393 Vector< sp<Track> > *tracksToRemove) 2394{ 2395 2396 mixer_state mixerStatus = MIXER_IDLE; 2397 // find out which tracks need to be processed 2398 size_t count = mActiveTracks.size(); 2399 size_t mixedTracks = 0; 2400 size_t tracksWithEffect = 0; 2401 // counts only _active_ fast tracks 2402 size_t fastTracks = 0; 2403 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2404 2405 float masterVolume = mMasterVolume; 2406 bool masterMute = mMasterMute; 2407 2408 if (masterMute) { 2409 masterVolume = 0; 2410 } 2411 // Delegate master volume control to effect in output mix effect chain if needed 2412 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2413 if (chain != 0) { 2414 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2415 chain->setVolume_l(&v, &v); 2416 masterVolume = (float)((v + (1 << 23)) >> 24); 2417 chain.clear(); 2418 } 2419 2420 // prepare a new state to push 2421 FastMixerStateQueue *sq = NULL; 2422 FastMixerState *state = NULL; 2423 bool didModify = false; 2424 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2425 if (mFastMixer != NULL) { 2426 sq = mFastMixer->sq(); 2427 state = sq->begin(); 2428 } 2429 2430 for (size_t i=0 ; i<count ; i++) { 2431 sp<Track> t = mActiveTracks[i].promote(); 2432 if (t == 0) { 2433 continue; 2434 } 2435 2436 // this const just means the local variable doesn't change 2437 Track* const track = t.get(); 2438 2439 // process fast tracks 2440 if (track->isFastTrack()) { 2441 2442 // It's theoretically possible (though unlikely) for a fast track to be created 2443 // and then removed within the same normal mix cycle. This is not a problem, as 2444 // the track never becomes active so it's fast mixer slot is never touched. 2445 // The converse, of removing an (active) track and then creating a new track 2446 // at the identical fast mixer slot within the same normal mix cycle, 2447 // is impossible because the slot isn't marked available until the end of each cycle. 2448 int j = track->mFastIndex; 2449 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2450 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2451 FastTrack *fastTrack = &state->mFastTracks[j]; 2452 2453 // Determine whether the track is currently in underrun condition, 2454 // and whether it had a recent underrun. 2455 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2456 FastTrackUnderruns underruns = ftDump->mUnderruns; 2457 uint32_t recentFull = (underruns.mBitFields.mFull - 2458 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2459 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2460 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2461 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2462 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2463 uint32_t recentUnderruns = recentPartial + recentEmpty; 2464 track->mObservedUnderruns = underruns; 2465 // don't count underruns that occur while stopping or pausing 2466 // or stopped which can occur when flush() is called while active 2467 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2468 track->mUnderrunCount += recentUnderruns; 2469 } 2470 2471 // This is similar to the state machine for normal tracks, 2472 // with a few modifications for fast tracks. 2473 bool isActive = true; 2474 switch (track->mState) { 2475 case TrackBase::STOPPING_1: 2476 // track stays active in STOPPING_1 state until first underrun 2477 if (recentUnderruns > 0) { 2478 track->mState = TrackBase::STOPPING_2; 2479 } 2480 break; 2481 case TrackBase::PAUSING: 2482 // ramp down is not yet implemented 2483 track->setPaused(); 2484 break; 2485 case TrackBase::RESUMING: 2486 // ramp up is not yet implemented 2487 track->mState = TrackBase::ACTIVE; 2488 break; 2489 case TrackBase::ACTIVE: 2490 if (recentFull > 0 || recentPartial > 0) { 2491 // track has provided at least some frames recently: reset retry count 2492 track->mRetryCount = kMaxTrackRetries; 2493 } 2494 if (recentUnderruns == 0) { 2495 // no recent underruns: stay active 2496 break; 2497 } 2498 // there has recently been an underrun of some kind 2499 if (track->sharedBuffer() == 0) { 2500 // were any of the recent underruns "empty" (no frames available)? 2501 if (recentEmpty == 0) { 2502 // no, then ignore the partial underruns as they are allowed indefinitely 2503 break; 2504 } 2505 // there has recently been an "empty" underrun: decrement the retry counter 2506 if (--(track->mRetryCount) > 0) { 2507 break; 2508 } 2509 // indicate to client process that the track was disabled because of underrun; 2510 // it will then automatically call start() when data is available 2511 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2512 // remove from active list, but state remains ACTIVE [confusing but true] 2513 isActive = false; 2514 break; 2515 } 2516 // fall through 2517 case TrackBase::STOPPING_2: 2518 case TrackBase::PAUSED: 2519 case TrackBase::TERMINATED: 2520 case TrackBase::STOPPED: 2521 case TrackBase::FLUSHED: // flush() while active 2522 // Check for presentation complete if track is inactive 2523 // We have consumed all the buffers of this track. 2524 // This would be incomplete if we auto-paused on underrun 2525 { 2526 size_t audioHALFrames = 2527 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2528 size_t framesWritten = mBytesWritten / mFrameSize; 2529 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2530 // track stays in active list until presentation is complete 2531 break; 2532 } 2533 } 2534 if (track->isStopping_2()) { 2535 track->mState = TrackBase::STOPPED; 2536 } 2537 if (track->isStopped()) { 2538 // Can't reset directly, as fast mixer is still polling this track 2539 // track->reset(); 2540 // So instead mark this track as needing to be reset after push with ack 2541 resetMask |= 1 << i; 2542 } 2543 isActive = false; 2544 break; 2545 case TrackBase::IDLE: 2546 default: 2547 LOG_FATAL("unexpected track state %d", track->mState); 2548 } 2549 2550 if (isActive) { 2551 // was it previously inactive? 2552 if (!(state->mTrackMask & (1 << j))) { 2553 ExtendedAudioBufferProvider *eabp = track; 2554 VolumeProvider *vp = track; 2555 fastTrack->mBufferProvider = eabp; 2556 fastTrack->mVolumeProvider = vp; 2557 fastTrack->mSampleRate = track->mSampleRate; 2558 fastTrack->mChannelMask = track->mChannelMask; 2559 fastTrack->mGeneration++; 2560 state->mTrackMask |= 1 << j; 2561 didModify = true; 2562 // no acknowledgement required for newly active tracks 2563 } 2564 // cache the combined master volume and stream type volume for fast mixer; this 2565 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2566 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2567 ++fastTracks; 2568 } else { 2569 // was it previously active? 2570 if (state->mTrackMask & (1 << j)) { 2571 fastTrack->mBufferProvider = NULL; 2572 fastTrack->mGeneration++; 2573 state->mTrackMask &= ~(1 << j); 2574 didModify = true; 2575 // If any fast tracks were removed, we must wait for acknowledgement 2576 // because we're about to decrement the last sp<> on those tracks. 2577 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2578 } else { 2579 LOG_FATAL("fast track %d should have been active", j); 2580 } 2581 tracksToRemove->add(track); 2582 // Avoids a misleading display in dumpsys 2583 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2584 } 2585 continue; 2586 } 2587 2588 { // local variable scope to avoid goto warning 2589 2590 audio_track_cblk_t* cblk = track->cblk(); 2591 2592 // The first time a track is added we wait 2593 // for all its buffers to be filled before processing it 2594 int name = track->name(); 2595 // make sure that we have enough frames to mix one full buffer. 2596 // enforce this condition only once to enable draining the buffer in case the client 2597 // app does not call stop() and relies on underrun to stop: 2598 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2599 // during last round 2600 size_t desiredFrames; 2601 if (t->sampleRate() == mSampleRate) { 2602 desiredFrames = mNormalFrameCount; 2603 } else { 2604 // +1 for rounding and +1 for additional sample needed for interpolation 2605 desiredFrames = (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 desiredFrames += 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(desiredFrames <= cblk->frameCount_); 2613 } 2614 uint32_t minFrames = 1; 2615 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2616 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2617 minFrames = desiredFrames; 2618 } 2619 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2620 size_t framesReady; 2621 if (track->sharedBuffer() == 0) { 2622 framesReady = track->framesReady(); 2623 } else if (track->isStopped()) { 2624 framesReady = 0; 2625 } else { 2626 framesReady = 1; 2627 } 2628 if ((framesReady >= minFrames) && track->isReady() && 2629 !track->isPaused() && !track->isTerminated()) 2630 { 2631 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2632 this); 2633 2634 mixedTracks++; 2635 2636 // track->mainBuffer() != mMixBuffer means there is an effect chain 2637 // connected to the track 2638 chain.clear(); 2639 if (track->mainBuffer() != mMixBuffer) { 2640 chain = getEffectChain_l(track->sessionId()); 2641 // Delegate volume control to effect in track effect chain if needed 2642 if (chain != 0) { 2643 tracksWithEffect++; 2644 } else { 2645 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2646 "session %d", 2647 name, track->sessionId()); 2648 } 2649 } 2650 2651 2652 int param = AudioMixer::VOLUME; 2653 if (track->mFillingUpStatus == Track::FS_FILLED) { 2654 // no ramp for the first volume setting 2655 track->mFillingUpStatus = Track::FS_ACTIVE; 2656 if (track->mState == TrackBase::RESUMING) { 2657 track->mState = TrackBase::ACTIVE; 2658 param = AudioMixer::RAMP_VOLUME; 2659 } 2660 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2661 } else if (cblk->server != 0) { 2662 // If the track is stopped before the first frame was mixed, 2663 // do not apply ramp 2664 param = AudioMixer::RAMP_VOLUME; 2665 } 2666 2667 // compute volume for this track 2668 uint32_t vl, vr, va; 2669 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2670 vl = vr = va = 0; 2671 if (track->isPausing()) { 2672 track->setPaused(); 2673 } 2674 } else { 2675 2676 // read original volumes with volume control 2677 float typeVolume = mStreamTypes[track->streamType()].volume; 2678 float v = masterVolume * typeVolume; 2679 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2680 uint32_t vlr = proxy->getVolumeLR(); 2681 vl = vlr & 0xFFFF; 2682 vr = vlr >> 16; 2683 // track volumes come from shared memory, so can't be trusted and must be clamped 2684 if (vl > MAX_GAIN_INT) { 2685 ALOGV("Track left volume out of range: %04X", vl); 2686 vl = MAX_GAIN_INT; 2687 } 2688 if (vr > MAX_GAIN_INT) { 2689 ALOGV("Track right volume out of range: %04X", vr); 2690 vr = MAX_GAIN_INT; 2691 } 2692 // now apply the master volume and stream type volume 2693 vl = (uint32_t)(v * vl) << 12; 2694 vr = (uint32_t)(v * vr) << 12; 2695 // assuming master volume and stream type volume each go up to 1.0, 2696 // vl and vr are now in 8.24 format 2697 2698 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2699 // send level comes from shared memory and so may be corrupt 2700 if (sendLevel > MAX_GAIN_INT) { 2701 ALOGV("Track send level out of range: %04X", sendLevel); 2702 sendLevel = MAX_GAIN_INT; 2703 } 2704 va = (uint32_t)(v * sendLevel); 2705 } 2706 // Delegate volume control to effect in track effect chain if needed 2707 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2708 // Do not ramp volume if volume is controlled by effect 2709 param = AudioMixer::VOLUME; 2710 track->mHasVolumeController = true; 2711 } else { 2712 // force no volume ramp when volume controller was just disabled or removed 2713 // from effect chain to avoid volume spike 2714 if (track->mHasVolumeController) { 2715 param = AudioMixer::VOLUME; 2716 } 2717 track->mHasVolumeController = false; 2718 } 2719 2720 // Convert volumes from 8.24 to 4.12 format 2721 // This additional clamping is needed in case chain->setVolume_l() overshot 2722 vl = (vl + (1 << 11)) >> 12; 2723 if (vl > MAX_GAIN_INT) { 2724 vl = MAX_GAIN_INT; 2725 } 2726 vr = (vr + (1 << 11)) >> 12; 2727 if (vr > MAX_GAIN_INT) { 2728 vr = MAX_GAIN_INT; 2729 } 2730 2731 if (va > MAX_GAIN_INT) { 2732 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2733 } 2734 2735 // XXX: these things DON'T need to be done each time 2736 mAudioMixer->setBufferProvider(name, track); 2737 mAudioMixer->enable(name); 2738 2739 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2740 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2741 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2742 mAudioMixer->setParameter( 2743 name, 2744 AudioMixer::TRACK, 2745 AudioMixer::FORMAT, (void *)track->format()); 2746 mAudioMixer->setParameter( 2747 name, 2748 AudioMixer::TRACK, 2749 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2750 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2751 uint32_t maxSampleRate = mSampleRate * 2; 2752 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 2753 if (reqSampleRate == 0) { 2754 reqSampleRate = mSampleRate; 2755 } else if (reqSampleRate > maxSampleRate) { 2756 reqSampleRate = maxSampleRate; 2757 } 2758 mAudioMixer->setParameter( 2759 name, 2760 AudioMixer::RESAMPLE, 2761 AudioMixer::SAMPLE_RATE, 2762 (void *)reqSampleRate); 2763 mAudioMixer->setParameter( 2764 name, 2765 AudioMixer::TRACK, 2766 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2767 mAudioMixer->setParameter( 2768 name, 2769 AudioMixer::TRACK, 2770 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2771 2772 // reset retry count 2773 track->mRetryCount = kMaxTrackRetries; 2774 2775 // If one track is ready, set the mixer ready if: 2776 // - the mixer was not ready during previous round OR 2777 // - no other track is not ready 2778 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2779 mixerStatus != MIXER_TRACKS_ENABLED) { 2780 mixerStatus = MIXER_TRACKS_READY; 2781 } 2782 } else { 2783 // only implemented for normal tracks, not fast tracks 2784 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 2785 // we missed desiredFrames whatever the actual number of frames missing was 2786 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 2787 // FIXME also wake futex so that underrun is noticed more quickly 2788 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 2789 } 2790 // clear effect chain input buffer if an active track underruns to avoid sending 2791 // previous audio buffer again to effects 2792 chain = getEffectChain_l(track->sessionId()); 2793 if (chain != 0) { 2794 chain->clearInputBuffer(); 2795 } 2796 2797 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2798 cblk->server, this); 2799 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2800 track->isStopped() || track->isPaused()) { 2801 // We have consumed all the buffers of this track. 2802 // Remove it from the list of active tracks. 2803 // TODO: use actual buffer filling status instead of latency when available from 2804 // audio HAL 2805 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2806 size_t framesWritten = mBytesWritten / mFrameSize; 2807 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2808 if (track->isStopped()) { 2809 track->reset(); 2810 } 2811 tracksToRemove->add(track); 2812 } 2813 } else { 2814 track->mUnderrunCount++; 2815 // No buffers for this track. Give it a few chances to 2816 // fill a buffer, then remove it from active list. 2817 if (--(track->mRetryCount) <= 0) { 2818 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2819 tracksToRemove->add(track); 2820 // indicate to client process that the track was disabled because of underrun; 2821 // it will then automatically call start() when data is available 2822 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2823 // If one track is not ready, mark the mixer also not ready if: 2824 // - the mixer was ready during previous round OR 2825 // - no other track is ready 2826 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2827 mixerStatus != MIXER_TRACKS_READY) { 2828 mixerStatus = MIXER_TRACKS_ENABLED; 2829 } 2830 } 2831 mAudioMixer->disable(name); 2832 } 2833 2834 } // local variable scope to avoid goto warning 2835track_is_ready: ; 2836 2837 } 2838 2839 // Push the new FastMixer state if necessary 2840 bool pauseAudioWatchdog = false; 2841 if (didModify) { 2842 state->mFastTracksGen++; 2843 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2844 if (kUseFastMixer == FastMixer_Dynamic && 2845 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2846 state->mCommand = FastMixerState::COLD_IDLE; 2847 state->mColdFutexAddr = &mFastMixerFutex; 2848 state->mColdGen++; 2849 mFastMixerFutex = 0; 2850 if (kUseFastMixer == FastMixer_Dynamic) { 2851 mNormalSink = mOutputSink; 2852 } 2853 // If we go into cold idle, need to wait for acknowledgement 2854 // so that fast mixer stops doing I/O. 2855 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2856 pauseAudioWatchdog = true; 2857 } 2858 } 2859 if (sq != NULL) { 2860 sq->end(didModify); 2861 sq->push(block); 2862 } 2863#ifdef AUDIO_WATCHDOG 2864 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2865 mAudioWatchdog->pause(); 2866 } 2867#endif 2868 2869 // Now perform the deferred reset on fast tracks that have stopped 2870 while (resetMask != 0) { 2871 size_t i = __builtin_ctz(resetMask); 2872 ALOG_ASSERT(i < count); 2873 resetMask &= ~(1 << i); 2874 sp<Track> t = mActiveTracks[i].promote(); 2875 if (t == 0) { 2876 continue; 2877 } 2878 Track* track = t.get(); 2879 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2880 track->reset(); 2881 } 2882 2883 // remove all the tracks that need to be... 2884 count = tracksToRemove->size(); 2885 if (CC_UNLIKELY(count)) { 2886 for (size_t i=0 ; i<count ; i++) { 2887 const sp<Track>& track = tracksToRemove->itemAt(i); 2888 mActiveTracks.remove(track); 2889 if (track->mainBuffer() != mMixBuffer) { 2890 chain = getEffectChain_l(track->sessionId()); 2891 if (chain != 0) { 2892 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2893 track->sessionId()); 2894 chain->decActiveTrackCnt(); 2895 } 2896 } 2897 if (track->isTerminated()) { 2898 removeTrack_l(track); 2899 } 2900 } 2901 } 2902 2903 // mix buffer must be cleared if all tracks are connected to an 2904 // effect chain as in this case the mixer will not write to 2905 // mix buffer and track effects will accumulate into it 2906 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2907 (mixedTracks == 0 && fastTracks > 0)) { 2908 // FIXME as a performance optimization, should remember previous zero status 2909 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2910 } 2911 2912 // if any fast tracks, then status is ready 2913 mMixerStatusIgnoringFastTracks = mixerStatus; 2914 if (fastTracks > 0) { 2915 mixerStatus = MIXER_TRACKS_READY; 2916 } 2917 return mixerStatus; 2918} 2919 2920// getTrackName_l() must be called with ThreadBase::mLock held 2921int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2922{ 2923 return mAudioMixer->getTrackName(channelMask, sessionId); 2924} 2925 2926// deleteTrackName_l() must be called with ThreadBase::mLock held 2927void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2928{ 2929 ALOGV("remove track (%d) and delete from mixer", name); 2930 mAudioMixer->deleteTrackName(name); 2931} 2932 2933// checkForNewParameters_l() must be called with ThreadBase::mLock held 2934bool AudioFlinger::MixerThread::checkForNewParameters_l() 2935{ 2936 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2937 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2938 bool reconfig = false; 2939 2940 while (!mNewParameters.isEmpty()) { 2941 2942 if (mFastMixer != NULL) { 2943 FastMixerStateQueue *sq = mFastMixer->sq(); 2944 FastMixerState *state = sq->begin(); 2945 if (!(state->mCommand & FastMixerState::IDLE)) { 2946 previousCommand = state->mCommand; 2947 state->mCommand = FastMixerState::HOT_IDLE; 2948 sq->end(); 2949 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2950 } else { 2951 sq->end(false /*didModify*/); 2952 } 2953 } 2954 2955 status_t status = NO_ERROR; 2956 String8 keyValuePair = mNewParameters[0]; 2957 AudioParameter param = AudioParameter(keyValuePair); 2958 int value; 2959 2960 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2961 reconfig = true; 2962 } 2963 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2964 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2965 status = BAD_VALUE; 2966 } else { 2967 reconfig = true; 2968 } 2969 } 2970 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2971 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2972 status = BAD_VALUE; 2973 } else { 2974 reconfig = true; 2975 } 2976 } 2977 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2978 // do not accept frame count changes if tracks are open as the track buffer 2979 // size depends on frame count and correct behavior would not be guaranteed 2980 // if frame count is changed after track creation 2981 if (!mTracks.isEmpty()) { 2982 status = INVALID_OPERATION; 2983 } else { 2984 reconfig = true; 2985 } 2986 } 2987 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2988#ifdef ADD_BATTERY_DATA 2989 // when changing the audio output device, call addBatteryData to notify 2990 // the change 2991 if (mOutDevice != value) { 2992 uint32_t params = 0; 2993 // check whether speaker is on 2994 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2995 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2996 } 2997 2998 audio_devices_t deviceWithoutSpeaker 2999 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3000 // check if any other device (except speaker) is on 3001 if (value & deviceWithoutSpeaker ) { 3002 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3003 } 3004 3005 if (params != 0) { 3006 addBatteryData(params); 3007 } 3008 } 3009#endif 3010 3011 // forward device change to effects that have requested to be 3012 // aware of attached audio device. 3013 if (value != AUDIO_DEVICE_NONE) { 3014 mOutDevice = value; 3015 for (size_t i = 0; i < mEffectChains.size(); i++) { 3016 mEffectChains[i]->setDevice_l(mOutDevice); 3017 } 3018 } 3019 } 3020 3021 if (status == NO_ERROR) { 3022 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3023 keyValuePair.string()); 3024 if (!mStandby && status == INVALID_OPERATION) { 3025 mOutput->stream->common.standby(&mOutput->stream->common); 3026 mStandby = true; 3027 mBytesWritten = 0; 3028 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3029 keyValuePair.string()); 3030 } 3031 if (status == NO_ERROR && reconfig) { 3032 delete mAudioMixer; 3033 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3034 mAudioMixer = NULL; 3035 readOutputParameters(); 3036 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3037 for (size_t i = 0; i < mTracks.size() ; i++) { 3038 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3039 if (name < 0) { 3040 break; 3041 } 3042 mTracks[i]->mName = name; 3043 } 3044 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3045 } 3046 } 3047 3048 mNewParameters.removeAt(0); 3049 3050 mParamStatus = status; 3051 mParamCond.signal(); 3052 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3053 // already timed out waiting for the status and will never signal the condition. 3054 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3055 } 3056 3057 if (!(previousCommand & FastMixerState::IDLE)) { 3058 ALOG_ASSERT(mFastMixer != NULL); 3059 FastMixerStateQueue *sq = mFastMixer->sq(); 3060 FastMixerState *state = sq->begin(); 3061 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3062 state->mCommand = previousCommand; 3063 sq->end(); 3064 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3065 } 3066 3067 return reconfig; 3068} 3069 3070 3071void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3072{ 3073 const size_t SIZE = 256; 3074 char buffer[SIZE]; 3075 String8 result; 3076 3077 PlaybackThread::dumpInternals(fd, args); 3078 3079 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3080 result.append(buffer); 3081 write(fd, result.string(), result.size()); 3082 3083 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3084 FastMixerDumpState copy = mFastMixerDumpState; 3085 copy.dump(fd); 3086 3087#ifdef STATE_QUEUE_DUMP 3088 // Similar for state queue 3089 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3090 observerCopy.dump(fd); 3091 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3092 mutatorCopy.dump(fd); 3093#endif 3094 3095#ifdef TEE_SINK 3096 // Write the tee output to a .wav file 3097 dumpTee(fd, mTeeSource, mId); 3098#endif 3099 3100#ifdef AUDIO_WATCHDOG 3101 if (mAudioWatchdog != 0) { 3102 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3103 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3104 wdCopy.dump(fd); 3105 } 3106#endif 3107} 3108 3109uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3110{ 3111 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3112} 3113 3114uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3115{ 3116 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3117} 3118 3119void AudioFlinger::MixerThread::cacheParameters_l() 3120{ 3121 PlaybackThread::cacheParameters_l(); 3122 3123 // FIXME: Relaxed timing because of a certain device that can't meet latency 3124 // Should be reduced to 2x after the vendor fixes the driver issue 3125 // increase threshold again due to low power audio mode. The way this warning 3126 // threshold is calculated and its usefulness should be reconsidered anyway. 3127 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3128} 3129 3130// ---------------------------------------------------------------------------- 3131 3132AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3133 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3134 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3135 // mLeftVolFloat, mRightVolFloat 3136{ 3137} 3138 3139AudioFlinger::DirectOutputThread::~DirectOutputThread() 3140{ 3141} 3142 3143AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3144 Vector< sp<Track> > *tracksToRemove 3145) 3146{ 3147 size_t count = mActiveTracks.size(); 3148 mixer_state mixerStatus = MIXER_IDLE; 3149 3150 // find out which tracks need to be processed 3151 for (size_t i = 0; i < count; i++) { 3152 sp<Track> t = mActiveTracks[i].promote(); 3153 // The track died recently 3154 if (t == 0) { 3155 continue; 3156 } 3157 3158 Track* const track = t.get(); 3159 audio_track_cblk_t* cblk = track->cblk(); 3160 3161 // The first time a track is added we wait 3162 // for all its buffers to be filled before processing it 3163 uint32_t minFrames; 3164 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3165 minFrames = mNormalFrameCount; 3166 } else { 3167 minFrames = 1; 3168 } 3169 if ((track->framesReady() >= minFrames) && track->isReady() && 3170 !track->isPaused() && !track->isTerminated()) 3171 { 3172 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3173 3174 if (track->mFillingUpStatus == Track::FS_FILLED) { 3175 track->mFillingUpStatus = Track::FS_ACTIVE; 3176 mLeftVolFloat = mRightVolFloat = 0; 3177 if (track->mState == TrackBase::RESUMING) { 3178 track->mState = TrackBase::ACTIVE; 3179 } 3180 } 3181 3182 // compute volume for this track 3183 float left, right; 3184 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3185 left = right = 0; 3186 if (track->isPausing()) { 3187 track->setPaused(); 3188 } 3189 } else { 3190 float typeVolume = mStreamTypes[track->streamType()].volume; 3191 float v = mMasterVolume * typeVolume; 3192 uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); 3193 float v_clamped = v * (vlr & 0xFFFF); 3194 if (v_clamped > MAX_GAIN) { 3195 v_clamped = MAX_GAIN; 3196 } 3197 left = v_clamped/MAX_GAIN; 3198 v_clamped = v * (vlr >> 16); 3199 if (v_clamped > MAX_GAIN) { 3200 v_clamped = MAX_GAIN; 3201 } 3202 right = v_clamped/MAX_GAIN; 3203 } 3204 // Only consider last track started for volume and mixer state control. 3205 // This is the last entry in mActiveTracks unless a track underruns. 3206 // As we only care about the transition phase between two tracks on a 3207 // direct output, it is not a problem to ignore the underrun case. 3208 if (i == (count - 1)) { 3209 if (left != mLeftVolFloat || right != mRightVolFloat) { 3210 mLeftVolFloat = left; 3211 mRightVolFloat = right; 3212 3213 // Convert volumes from float to 8.24 3214 uint32_t vl = (uint32_t)(left * (1 << 24)); 3215 uint32_t vr = (uint32_t)(right * (1 << 24)); 3216 3217 // Delegate volume control to effect in track effect chain if needed 3218 // only one effect chain can be present on DirectOutputThread, so if 3219 // there is one, the track is connected to it 3220 if (!mEffectChains.isEmpty()) { 3221 // Do not ramp volume if volume is controlled by effect 3222 mEffectChains[0]->setVolume_l(&vl, &vr); 3223 left = (float)vl / (1 << 24); 3224 right = (float)vr / (1 << 24); 3225 } 3226 mOutput->stream->set_volume(mOutput->stream, left, right); 3227 } 3228 3229 // reset retry count 3230 track->mRetryCount = kMaxTrackRetriesDirect; 3231 mActiveTrack = t; 3232 mixerStatus = MIXER_TRACKS_READY; 3233 } 3234 } else { 3235 // clear effect chain input buffer if the last active track started underruns 3236 // to avoid sending previous audio buffer again to effects 3237 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3238 mEffectChains[0]->clearInputBuffer(); 3239 } 3240 3241 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3242 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3243 track->isStopped() || track->isPaused()) { 3244 // We have consumed all the buffers of this track. 3245 // Remove it from the list of active tracks. 3246 // TODO: implement behavior for compressed audio 3247 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3248 size_t framesWritten = mBytesWritten / mFrameSize; 3249 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3250 if (track->isStopped()) { 3251 track->reset(); 3252 } 3253 tracksToRemove->add(track); 3254 } 3255 } else { 3256 // No buffers for this track. Give it a few chances to 3257 // fill a buffer, then remove it from active list. 3258 // Only consider last track started for mixer state control 3259 if (--(track->mRetryCount) <= 0) { 3260 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3261 tracksToRemove->add(track); 3262 } else if (i == (count -1)){ 3263 mixerStatus = MIXER_TRACKS_ENABLED; 3264 } 3265 } 3266 } 3267 } 3268 3269 // remove all the tracks that need to be... 3270 count = tracksToRemove->size(); 3271 if (CC_UNLIKELY(count)) { 3272 for (size_t i = 0 ; i < count ; i++) { 3273 const sp<Track>& track = tracksToRemove->itemAt(i); 3274 mActiveTracks.remove(track); 3275 if (!mEffectChains.isEmpty()) { 3276 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3277 track->sessionId()); 3278 mEffectChains[0]->decActiveTrackCnt(); 3279 } 3280 if (track->isTerminated()) { 3281 removeTrack_l(track); 3282 } 3283 } 3284 } 3285 3286 return mixerStatus; 3287} 3288 3289void AudioFlinger::DirectOutputThread::threadLoop_mix() 3290{ 3291 AudioBufferProvider::Buffer buffer; 3292 size_t frameCount = mFrameCount; 3293 int8_t *curBuf = (int8_t *)mMixBuffer; 3294 // output audio to hardware 3295 while (frameCount) { 3296 buffer.frameCount = frameCount; 3297 mActiveTrack->getNextBuffer(&buffer); 3298 if (CC_UNLIKELY(buffer.raw == NULL)) { 3299 memset(curBuf, 0, frameCount * mFrameSize); 3300 break; 3301 } 3302 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3303 frameCount -= buffer.frameCount; 3304 curBuf += buffer.frameCount * mFrameSize; 3305 mActiveTrack->releaseBuffer(&buffer); 3306 } 3307 sleepTime = 0; 3308 standbyTime = systemTime() + standbyDelay; 3309 mActiveTrack.clear(); 3310 3311} 3312 3313void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3314{ 3315 if (sleepTime == 0) { 3316 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3317 sleepTime = activeSleepTime; 3318 } else { 3319 sleepTime = idleSleepTime; 3320 } 3321 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3322 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3323 sleepTime = 0; 3324 } 3325} 3326 3327// getTrackName_l() must be called with ThreadBase::mLock held 3328int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3329 int sessionId) 3330{ 3331 return 0; 3332} 3333 3334// deleteTrackName_l() must be called with ThreadBase::mLock held 3335void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3336{ 3337} 3338 3339// checkForNewParameters_l() must be called with ThreadBase::mLock held 3340bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3341{ 3342 bool reconfig = false; 3343 3344 while (!mNewParameters.isEmpty()) { 3345 status_t status = NO_ERROR; 3346 String8 keyValuePair = mNewParameters[0]; 3347 AudioParameter param = AudioParameter(keyValuePair); 3348 int value; 3349 3350 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3351 // do not accept frame count changes if tracks are open as the track buffer 3352 // size depends on frame count and correct behavior would not be garantied 3353 // if frame count is changed after track creation 3354 if (!mTracks.isEmpty()) { 3355 status = INVALID_OPERATION; 3356 } else { 3357 reconfig = true; 3358 } 3359 } 3360 if (status == NO_ERROR) { 3361 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3362 keyValuePair.string()); 3363 if (!mStandby && status == INVALID_OPERATION) { 3364 mOutput->stream->common.standby(&mOutput->stream->common); 3365 mStandby = true; 3366 mBytesWritten = 0; 3367 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3368 keyValuePair.string()); 3369 } 3370 if (status == NO_ERROR && reconfig) { 3371 readOutputParameters(); 3372 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3373 } 3374 } 3375 3376 mNewParameters.removeAt(0); 3377 3378 mParamStatus = status; 3379 mParamCond.signal(); 3380 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3381 // already timed out waiting for the status and will never signal the condition. 3382 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3383 } 3384 return reconfig; 3385} 3386 3387uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3388{ 3389 uint32_t time; 3390 if (audio_is_linear_pcm(mFormat)) { 3391 time = PlaybackThread::activeSleepTimeUs(); 3392 } else { 3393 time = 10000; 3394 } 3395 return time; 3396} 3397 3398uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3399{ 3400 uint32_t time; 3401 if (audio_is_linear_pcm(mFormat)) { 3402 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3403 } else { 3404 time = 10000; 3405 } 3406 return time; 3407} 3408 3409uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3410{ 3411 uint32_t time; 3412 if (audio_is_linear_pcm(mFormat)) { 3413 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3414 } else { 3415 time = 10000; 3416 } 3417 return time; 3418} 3419 3420void AudioFlinger::DirectOutputThread::cacheParameters_l() 3421{ 3422 PlaybackThread::cacheParameters_l(); 3423 3424 // use shorter standby delay as on normal output to release 3425 // hardware resources as soon as possible 3426 standbyDelay = microseconds(activeSleepTime*2); 3427} 3428 3429// ---------------------------------------------------------------------------- 3430 3431AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3432 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3433 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3434 DUPLICATING), 3435 mWaitTimeMs(UINT_MAX) 3436{ 3437 addOutputTrack(mainThread); 3438} 3439 3440AudioFlinger::DuplicatingThread::~DuplicatingThread() 3441{ 3442 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3443 mOutputTracks[i]->destroy(); 3444 } 3445} 3446 3447void AudioFlinger::DuplicatingThread::threadLoop_mix() 3448{ 3449 // mix buffers... 3450 if (outputsReady(outputTracks)) { 3451 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3452 } else { 3453 memset(mMixBuffer, 0, mixBufferSize); 3454 } 3455 sleepTime = 0; 3456 writeFrames = mNormalFrameCount; 3457 standbyTime = systemTime() + standbyDelay; 3458} 3459 3460void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3461{ 3462 if (sleepTime == 0) { 3463 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3464 sleepTime = activeSleepTime; 3465 } else { 3466 sleepTime = idleSleepTime; 3467 } 3468 } else if (mBytesWritten != 0) { 3469 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3470 writeFrames = mNormalFrameCount; 3471 memset(mMixBuffer, 0, mixBufferSize); 3472 } else { 3473 // flush remaining overflow buffers in output tracks 3474 writeFrames = 0; 3475 } 3476 sleepTime = 0; 3477 } 3478} 3479 3480void AudioFlinger::DuplicatingThread::threadLoop_write() 3481{ 3482 for (size_t i = 0; i < outputTracks.size(); i++) { 3483 outputTracks[i]->write(mMixBuffer, writeFrames); 3484 } 3485 mBytesWritten += mixBufferSize; 3486} 3487 3488void AudioFlinger::DuplicatingThread::threadLoop_standby() 3489{ 3490 // DuplicatingThread implements standby by stopping all tracks 3491 for (size_t i = 0; i < outputTracks.size(); i++) { 3492 outputTracks[i]->stop(); 3493 } 3494} 3495 3496void AudioFlinger::DuplicatingThread::saveOutputTracks() 3497{ 3498 outputTracks = mOutputTracks; 3499} 3500 3501void AudioFlinger::DuplicatingThread::clearOutputTracks() 3502{ 3503 outputTracks.clear(); 3504} 3505 3506void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3507{ 3508 Mutex::Autolock _l(mLock); 3509 // FIXME explain this formula 3510 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3511 OutputTrack *outputTrack = new OutputTrack(thread, 3512 this, 3513 mSampleRate, 3514 mFormat, 3515 mChannelMask, 3516 frameCount); 3517 if (outputTrack->cblk() != NULL) { 3518 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3519 mOutputTracks.add(outputTrack); 3520 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3521 updateWaitTime_l(); 3522 } 3523} 3524 3525void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3526{ 3527 Mutex::Autolock _l(mLock); 3528 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3529 if (mOutputTracks[i]->thread() == thread) { 3530 mOutputTracks[i]->destroy(); 3531 mOutputTracks.removeAt(i); 3532 updateWaitTime_l(); 3533 return; 3534 } 3535 } 3536 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3537} 3538 3539// caller must hold mLock 3540void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3541{ 3542 mWaitTimeMs = UINT_MAX; 3543 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3544 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3545 if (strong != 0) { 3546 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3547 if (waitTimeMs < mWaitTimeMs) { 3548 mWaitTimeMs = waitTimeMs; 3549 } 3550 } 3551 } 3552} 3553 3554 3555bool AudioFlinger::DuplicatingThread::outputsReady( 3556 const SortedVector< sp<OutputTrack> > &outputTracks) 3557{ 3558 for (size_t i = 0; i < outputTracks.size(); i++) { 3559 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3560 if (thread == 0) { 3561 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3562 outputTracks[i].get()); 3563 return false; 3564 } 3565 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3566 // see note at standby() declaration 3567 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3568 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3569 thread.get()); 3570 return false; 3571 } 3572 } 3573 return true; 3574} 3575 3576uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3577{ 3578 return (mWaitTimeMs * 1000) / 2; 3579} 3580 3581void AudioFlinger::DuplicatingThread::cacheParameters_l() 3582{ 3583 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3584 updateWaitTime_l(); 3585 3586 MixerThread::cacheParameters_l(); 3587} 3588 3589// ---------------------------------------------------------------------------- 3590// Record 3591// ---------------------------------------------------------------------------- 3592 3593AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3594 AudioStreamIn *input, 3595 uint32_t sampleRate, 3596 audio_channel_mask_t channelMask, 3597 audio_io_handle_t id, 3598 audio_devices_t outDevice, 3599 audio_devices_t inDevice 3600#ifdef TEE_SINK 3601 , const sp<NBAIO_Sink>& teeSink 3602#endif 3603 ) : 3604 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3605 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3606 // mRsmpInIndex and mInputBytes set by readInputParameters() 3607 mReqChannelCount(popcount(channelMask)), 3608 mReqSampleRate(sampleRate) 3609 // mBytesRead is only meaningful while active, and so is cleared in start() 3610 // (but might be better to also clear here for dump?) 3611#ifdef TEE_SINK 3612 , mTeeSink(teeSink) 3613#endif 3614{ 3615 snprintf(mName, kNameLength, "AudioIn_%X", id); 3616 3617 readInputParameters(); 3618 3619} 3620 3621 3622AudioFlinger::RecordThread::~RecordThread() 3623{ 3624 delete[] mRsmpInBuffer; 3625 delete mResampler; 3626 delete[] mRsmpOutBuffer; 3627} 3628 3629void AudioFlinger::RecordThread::onFirstRef() 3630{ 3631 run(mName, PRIORITY_URGENT_AUDIO); 3632} 3633 3634status_t AudioFlinger::RecordThread::readyToRun() 3635{ 3636 status_t status = initCheck(); 3637 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3638 return status; 3639} 3640 3641bool AudioFlinger::RecordThread::threadLoop() 3642{ 3643 AudioBufferProvider::Buffer buffer; 3644 sp<RecordTrack> activeTrack; 3645 Vector< sp<EffectChain> > effectChains; 3646 3647 nsecs_t lastWarning = 0; 3648 3649 inputStandBy(); 3650 acquireWakeLock(); 3651 3652 // used to verify we've read at least once before evaluating how many bytes were read 3653 bool readOnce = false; 3654 3655 // start recording 3656 while (!exitPending()) { 3657 3658 processConfigEvents(); 3659 3660 { // scope for mLock 3661 Mutex::Autolock _l(mLock); 3662 checkForNewParameters_l(); 3663 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3664 standby(); 3665 3666 if (exitPending()) { 3667 break; 3668 } 3669 3670 releaseWakeLock_l(); 3671 ALOGV("RecordThread: loop stopping"); 3672 // go to sleep 3673 mWaitWorkCV.wait(mLock); 3674 ALOGV("RecordThread: loop starting"); 3675 acquireWakeLock_l(); 3676 continue; 3677 } 3678 if (mActiveTrack != 0) { 3679 if (mActiveTrack->mState == TrackBase::PAUSING) { 3680 standby(); 3681 mActiveTrack.clear(); 3682 mStartStopCond.broadcast(); 3683 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3684 if (mReqChannelCount != mActiveTrack->channelCount()) { 3685 mActiveTrack.clear(); 3686 mStartStopCond.broadcast(); 3687 } else if (readOnce) { 3688 // record start succeeds only if first read from audio input 3689 // succeeds 3690 if (mBytesRead >= 0) { 3691 mActiveTrack->mState = TrackBase::ACTIVE; 3692 } else { 3693 mActiveTrack.clear(); 3694 } 3695 mStartStopCond.broadcast(); 3696 } 3697 mStandby = false; 3698 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3699 removeTrack_l(mActiveTrack); 3700 mActiveTrack.clear(); 3701 } 3702 } 3703 lockEffectChains_l(effectChains); 3704 } 3705 3706 if (mActiveTrack != 0) { 3707 if (mActiveTrack->mState != TrackBase::ACTIVE && 3708 mActiveTrack->mState != TrackBase::RESUMING) { 3709 unlockEffectChains(effectChains); 3710 usleep(kRecordThreadSleepUs); 3711 continue; 3712 } 3713 for (size_t i = 0; i < effectChains.size(); i ++) { 3714 effectChains[i]->process_l(); 3715 } 3716 3717 buffer.frameCount = mFrameCount; 3718 status_t status = mActiveTrack->getNextBuffer(&buffer); 3719 if (CC_LIKELY(status == NO_ERROR)) { 3720 readOnce = true; 3721 size_t framesOut = buffer.frameCount; 3722 if (mResampler == NULL) { 3723 // no resampling 3724 while (framesOut) { 3725 size_t framesIn = mFrameCount - mRsmpInIndex; 3726 if (framesIn) { 3727 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3728 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3729 mActiveTrack->mFrameSize; 3730 if (framesIn > framesOut) 3731 framesIn = framesOut; 3732 mRsmpInIndex += framesIn; 3733 framesOut -= framesIn; 3734 if (mChannelCount == mReqChannelCount || 3735 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3736 memcpy(dst, src, framesIn * mFrameSize); 3737 } else { 3738 if (mChannelCount == 1) { 3739 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3740 (int16_t *)src, framesIn); 3741 } else { 3742 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3743 (int16_t *)src, framesIn); 3744 } 3745 } 3746 } 3747 if (framesOut && mFrameCount == mRsmpInIndex) { 3748 void *readInto; 3749 if (framesOut == mFrameCount && 3750 (mChannelCount == mReqChannelCount || 3751 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3752 readInto = buffer.raw; 3753 framesOut = 0; 3754 } else { 3755 readInto = mRsmpInBuffer; 3756 mRsmpInIndex = 0; 3757 } 3758 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3759 mInputBytes); 3760 if (mBytesRead <= 0) { 3761 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3762 { 3763 ALOGE("Error reading audio input"); 3764 // Force input into standby so that it tries to 3765 // recover at next read attempt 3766 inputStandBy(); 3767 usleep(kRecordThreadSleepUs); 3768 } 3769 mRsmpInIndex = mFrameCount; 3770 framesOut = 0; 3771 buffer.frameCount = 0; 3772 } 3773#ifdef TEE_SINK 3774 else if (mTeeSink != 0) { 3775 (void) mTeeSink->write(readInto, 3776 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3777 } 3778#endif 3779 } 3780 } 3781 } else { 3782 // resampling 3783 3784 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3785 // alter output frame count as if we were expecting stereo samples 3786 if (mChannelCount == 1 && mReqChannelCount == 1) { 3787 framesOut >>= 1; 3788 } 3789 mResampler->resample(mRsmpOutBuffer, framesOut, 3790 this /* AudioBufferProvider* */); 3791 // ditherAndClamp() works as long as all buffers returned by 3792 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3793 if (mChannelCount == 2 && mReqChannelCount == 1) { 3794 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3795 // the resampler always outputs stereo samples: 3796 // do post stereo to mono conversion 3797 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3798 framesOut); 3799 } else { 3800 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3801 } 3802 3803 } 3804 if (mFramestoDrop == 0) { 3805 mActiveTrack->releaseBuffer(&buffer); 3806 } else { 3807 if (mFramestoDrop > 0) { 3808 mFramestoDrop -= buffer.frameCount; 3809 if (mFramestoDrop <= 0) { 3810 clearSyncStartEvent(); 3811 } 3812 } else { 3813 mFramestoDrop += buffer.frameCount; 3814 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3815 mSyncStartEvent->isCancelled()) { 3816 ALOGW("Synced record %s, session %d, trigger session %d", 3817 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3818 mActiveTrack->sessionId(), 3819 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3820 clearSyncStartEvent(); 3821 } 3822 } 3823 } 3824 mActiveTrack->clearOverflow(); 3825 } 3826 // client isn't retrieving buffers fast enough 3827 else { 3828 if (!mActiveTrack->setOverflow()) { 3829 nsecs_t now = systemTime(); 3830 if ((now - lastWarning) > kWarningThrottleNs) { 3831 ALOGW("RecordThread: buffer overflow"); 3832 lastWarning = now; 3833 } 3834 } 3835 // Release the processor for a while before asking for a new buffer. 3836 // This will give the application more chance to read from the buffer and 3837 // clear the overflow. 3838 usleep(kRecordThreadSleepUs); 3839 } 3840 } 3841 // enable changes in effect chain 3842 unlockEffectChains(effectChains); 3843 effectChains.clear(); 3844 } 3845 3846 standby(); 3847 3848 { 3849 Mutex::Autolock _l(mLock); 3850 mActiveTrack.clear(); 3851 mStartStopCond.broadcast(); 3852 } 3853 3854 releaseWakeLock(); 3855 3856 ALOGV("RecordThread %p exiting", this); 3857 return false; 3858} 3859 3860void AudioFlinger::RecordThread::standby() 3861{ 3862 if (!mStandby) { 3863 inputStandBy(); 3864 mStandby = true; 3865 } 3866} 3867 3868void AudioFlinger::RecordThread::inputStandBy() 3869{ 3870 mInput->stream->common.standby(&mInput->stream->common); 3871} 3872 3873sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3874 const sp<AudioFlinger::Client>& client, 3875 uint32_t sampleRate, 3876 audio_format_t format, 3877 audio_channel_mask_t channelMask, 3878 size_t frameCount, 3879 int sessionId, 3880 IAudioFlinger::track_flags_t flags, 3881 pid_t tid, 3882 status_t *status) 3883{ 3884 sp<RecordTrack> track; 3885 status_t lStatus; 3886 3887 lStatus = initCheck(); 3888 if (lStatus != NO_ERROR) { 3889 ALOGE("Audio driver not initialized."); 3890 goto Exit; 3891 } 3892 3893 // FIXME use flags and tid similar to createTrack_l() 3894 3895 { // scope for mLock 3896 Mutex::Autolock _l(mLock); 3897 3898 track = new RecordTrack(this, client, sampleRate, 3899 format, channelMask, frameCount, sessionId); 3900 3901 if (track->getCblk() == 0) { 3902 lStatus = NO_MEMORY; 3903 goto Exit; 3904 } 3905 mTracks.add(track); 3906 3907 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3908 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3909 mAudioFlinger->btNrecIsOff(); 3910 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3911 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3912 } 3913 lStatus = NO_ERROR; 3914 3915Exit: 3916 if (status) { 3917 *status = lStatus; 3918 } 3919 return track; 3920} 3921 3922status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3923 AudioSystem::sync_event_t event, 3924 int triggerSession) 3925{ 3926 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3927 sp<ThreadBase> strongMe = this; 3928 status_t status = NO_ERROR; 3929 3930 if (event == AudioSystem::SYNC_EVENT_NONE) { 3931 clearSyncStartEvent(); 3932 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3933 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3934 triggerSession, 3935 recordTrack->sessionId(), 3936 syncStartEventCallback, 3937 this); 3938 // Sync event can be cancelled by the trigger session if the track is not in a 3939 // compatible state in which case we start record immediately 3940 if (mSyncStartEvent->isCancelled()) { 3941 clearSyncStartEvent(); 3942 } else { 3943 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3944 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3945 } 3946 } 3947 3948 { 3949 AutoMutex lock(mLock); 3950 if (mActiveTrack != 0) { 3951 if (recordTrack != mActiveTrack.get()) { 3952 status = -EBUSY; 3953 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3954 mActiveTrack->mState = TrackBase::ACTIVE; 3955 } 3956 return status; 3957 } 3958 3959 recordTrack->mState = TrackBase::IDLE; 3960 mActiveTrack = recordTrack; 3961 mLock.unlock(); 3962 status_t status = AudioSystem::startInput(mId); 3963 mLock.lock(); 3964 if (status != NO_ERROR) { 3965 mActiveTrack.clear(); 3966 clearSyncStartEvent(); 3967 return status; 3968 } 3969 mRsmpInIndex = mFrameCount; 3970 mBytesRead = 0; 3971 if (mResampler != NULL) { 3972 mResampler->reset(); 3973 } 3974 mActiveTrack->mState = TrackBase::RESUMING; 3975 // signal thread to start 3976 ALOGV("Signal record thread"); 3977 mWaitWorkCV.broadcast(); 3978 // do not wait for mStartStopCond if exiting 3979 if (exitPending()) { 3980 mActiveTrack.clear(); 3981 status = INVALID_OPERATION; 3982 goto startError; 3983 } 3984 mStartStopCond.wait(mLock); 3985 if (mActiveTrack == 0) { 3986 ALOGV("Record failed to start"); 3987 status = BAD_VALUE; 3988 goto startError; 3989 } 3990 ALOGV("Record started OK"); 3991 return status; 3992 } 3993 3994startError: 3995 AudioSystem::stopInput(mId); 3996 clearSyncStartEvent(); 3997 return status; 3998} 3999 4000void AudioFlinger::RecordThread::clearSyncStartEvent() 4001{ 4002 if (mSyncStartEvent != 0) { 4003 mSyncStartEvent->cancel(); 4004 } 4005 mSyncStartEvent.clear(); 4006 mFramestoDrop = 0; 4007} 4008 4009void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4010{ 4011 sp<SyncEvent> strongEvent = event.promote(); 4012 4013 if (strongEvent != 0) { 4014 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4015 me->handleSyncStartEvent(strongEvent); 4016 } 4017} 4018 4019void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4020{ 4021 if (event == mSyncStartEvent) { 4022 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4023 // from audio HAL 4024 mFramestoDrop = mFrameCount * 2; 4025 } 4026} 4027 4028bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4029 ALOGV("RecordThread::stop"); 4030 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4031 return false; 4032 } 4033 recordTrack->mState = TrackBase::PAUSING; 4034 // do not wait for mStartStopCond if exiting 4035 if (exitPending()) { 4036 return true; 4037 } 4038 mStartStopCond.wait(mLock); 4039 // if we have been restarted, recordTrack == mActiveTrack.get() here 4040 if (exitPending() || recordTrack != mActiveTrack.get()) { 4041 ALOGV("Record stopped OK"); 4042 return true; 4043 } 4044 return false; 4045} 4046 4047bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4048{ 4049 return false; 4050} 4051 4052status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4053{ 4054#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4055 if (!isValidSyncEvent(event)) { 4056 return BAD_VALUE; 4057 } 4058 4059 int eventSession = event->triggerSession(); 4060 status_t ret = NAME_NOT_FOUND; 4061 4062 Mutex::Autolock _l(mLock); 4063 4064 for (size_t i = 0; i < mTracks.size(); i++) { 4065 sp<RecordTrack> track = mTracks[i]; 4066 if (eventSession == track->sessionId()) { 4067 (void) track->setSyncEvent(event); 4068 ret = NO_ERROR; 4069 } 4070 } 4071 return ret; 4072#else 4073 return BAD_VALUE; 4074#endif 4075} 4076 4077// destroyTrack_l() must be called with ThreadBase::mLock held 4078void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4079{ 4080 track->mState = TrackBase::TERMINATED; 4081 // active tracks are removed by threadLoop() 4082 if (mActiveTrack != track) { 4083 removeTrack_l(track); 4084 } 4085} 4086 4087void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4088{ 4089 mTracks.remove(track); 4090 // need anything related to effects here? 4091} 4092 4093void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4094{ 4095 dumpInternals(fd, args); 4096 dumpTracks(fd, args); 4097 dumpEffectChains(fd, args); 4098} 4099 4100void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4101{ 4102 const size_t SIZE = 256; 4103 char buffer[SIZE]; 4104 String8 result; 4105 4106 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4107 result.append(buffer); 4108 4109 if (mActiveTrack != 0) { 4110 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4111 result.append(buffer); 4112 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4113 result.append(buffer); 4114 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4115 result.append(buffer); 4116 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4117 result.append(buffer); 4118 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4119 result.append(buffer); 4120 } else { 4121 result.append("No active record client\n"); 4122 } 4123 4124 write(fd, result.string(), result.size()); 4125 4126 dumpBase(fd, args); 4127} 4128 4129void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4130{ 4131 const size_t SIZE = 256; 4132 char buffer[SIZE]; 4133 String8 result; 4134 4135 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4136 result.append(buffer); 4137 RecordTrack::appendDumpHeader(result); 4138 for (size_t i = 0; i < mTracks.size(); ++i) { 4139 sp<RecordTrack> track = mTracks[i]; 4140 if (track != 0) { 4141 track->dump(buffer, SIZE); 4142 result.append(buffer); 4143 } 4144 } 4145 4146 if (mActiveTrack != 0) { 4147 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4148 result.append(buffer); 4149 RecordTrack::appendDumpHeader(result); 4150 mActiveTrack->dump(buffer, SIZE); 4151 result.append(buffer); 4152 4153 } 4154 write(fd, result.string(), result.size()); 4155} 4156 4157// AudioBufferProvider interface 4158status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4159{ 4160 size_t framesReq = buffer->frameCount; 4161 size_t framesReady = mFrameCount - mRsmpInIndex; 4162 int channelCount; 4163 4164 if (framesReady == 0) { 4165 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4166 if (mBytesRead <= 0) { 4167 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4168 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4169 // Force input into standby so that it tries to 4170 // recover at next read attempt 4171 inputStandBy(); 4172 usleep(kRecordThreadSleepUs); 4173 } 4174 buffer->raw = NULL; 4175 buffer->frameCount = 0; 4176 return NOT_ENOUGH_DATA; 4177 } 4178 mRsmpInIndex = 0; 4179 framesReady = mFrameCount; 4180 } 4181 4182 if (framesReq > framesReady) { 4183 framesReq = framesReady; 4184 } 4185 4186 if (mChannelCount == 1 && mReqChannelCount == 2) { 4187 channelCount = 1; 4188 } else { 4189 channelCount = 2; 4190 } 4191 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4192 buffer->frameCount = framesReq; 4193 return NO_ERROR; 4194} 4195 4196// AudioBufferProvider interface 4197void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4198{ 4199 mRsmpInIndex += buffer->frameCount; 4200 buffer->frameCount = 0; 4201} 4202 4203bool AudioFlinger::RecordThread::checkForNewParameters_l() 4204{ 4205 bool reconfig = false; 4206 4207 while (!mNewParameters.isEmpty()) { 4208 status_t status = NO_ERROR; 4209 String8 keyValuePair = mNewParameters[0]; 4210 AudioParameter param = AudioParameter(keyValuePair); 4211 int value; 4212 audio_format_t reqFormat = mFormat; 4213 uint32_t reqSamplingRate = mReqSampleRate; 4214 uint32_t reqChannelCount = mReqChannelCount; 4215 4216 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4217 reqSamplingRate = value; 4218 reconfig = true; 4219 } 4220 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4221 reqFormat = (audio_format_t) value; 4222 reconfig = true; 4223 } 4224 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4225 reqChannelCount = popcount(value); 4226 reconfig = true; 4227 } 4228 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4229 // do not accept frame count changes if tracks are open as the track buffer 4230 // size depends on frame count and correct behavior would not be guaranteed 4231 // if frame count is changed after track creation 4232 if (mActiveTrack != 0) { 4233 status = INVALID_OPERATION; 4234 } else { 4235 reconfig = true; 4236 } 4237 } 4238 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4239 // forward device change to effects that have requested to be 4240 // aware of attached audio device. 4241 for (size_t i = 0; i < mEffectChains.size(); i++) { 4242 mEffectChains[i]->setDevice_l(value); 4243 } 4244 4245 // store input device and output device but do not forward output device to audio HAL. 4246 // Note that status is ignored by the caller for output device 4247 // (see AudioFlinger::setParameters() 4248 if (audio_is_output_devices(value)) { 4249 mOutDevice = value; 4250 status = BAD_VALUE; 4251 } else { 4252 mInDevice = value; 4253 // disable AEC and NS if the device is a BT SCO headset supporting those 4254 // pre processings 4255 if (mTracks.size() > 0) { 4256 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4257 mAudioFlinger->btNrecIsOff(); 4258 for (size_t i = 0; i < mTracks.size(); i++) { 4259 sp<RecordTrack> track = mTracks[i]; 4260 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4261 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4262 } 4263 } 4264 } 4265 } 4266 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4267 mAudioSource != (audio_source_t)value) { 4268 // forward device change to effects that have requested to be 4269 // aware of attached audio device. 4270 for (size_t i = 0; i < mEffectChains.size(); i++) { 4271 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4272 } 4273 mAudioSource = (audio_source_t)value; 4274 } 4275 if (status == NO_ERROR) { 4276 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4277 keyValuePair.string()); 4278 if (status == INVALID_OPERATION) { 4279 inputStandBy(); 4280 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4281 keyValuePair.string()); 4282 } 4283 if (reconfig) { 4284 if (status == BAD_VALUE && 4285 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4286 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4287 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4288 <= (2 * reqSamplingRate)) && 4289 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4290 <= FCC_2 && 4291 (reqChannelCount <= FCC_2)) { 4292 status = NO_ERROR; 4293 } 4294 if (status == NO_ERROR) { 4295 readInputParameters(); 4296 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4297 } 4298 } 4299 } 4300 4301 mNewParameters.removeAt(0); 4302 4303 mParamStatus = status; 4304 mParamCond.signal(); 4305 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4306 // already timed out waiting for the status and will never signal the condition. 4307 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4308 } 4309 return reconfig; 4310} 4311 4312String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4313{ 4314 char *s; 4315 String8 out_s8 = String8(); 4316 4317 Mutex::Autolock _l(mLock); 4318 if (initCheck() != NO_ERROR) { 4319 return out_s8; 4320 } 4321 4322 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4323 out_s8 = String8(s); 4324 free(s); 4325 return out_s8; 4326} 4327 4328void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4329 AudioSystem::OutputDescriptor desc; 4330 void *param2 = NULL; 4331 4332 switch (event) { 4333 case AudioSystem::INPUT_OPENED: 4334 case AudioSystem::INPUT_CONFIG_CHANGED: 4335 desc.channels = mChannelMask; 4336 desc.samplingRate = mSampleRate; 4337 desc.format = mFormat; 4338 desc.frameCount = mFrameCount; 4339 desc.latency = 0; 4340 param2 = &desc; 4341 break; 4342 4343 case AudioSystem::INPUT_CLOSED: 4344 default: 4345 break; 4346 } 4347 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4348} 4349 4350void AudioFlinger::RecordThread::readInputParameters() 4351{ 4352 delete mRsmpInBuffer; 4353 // mRsmpInBuffer is always assigned a new[] below 4354 delete mRsmpOutBuffer; 4355 mRsmpOutBuffer = NULL; 4356 delete mResampler; 4357 mResampler = NULL; 4358 4359 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4360 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4361 mChannelCount = (uint16_t)popcount(mChannelMask); 4362 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4363 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4364 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4365 mFrameCount = mInputBytes / mFrameSize; 4366 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4367 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4368 4369 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4370 { 4371 int channelCount; 4372 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4373 // stereo to mono post process as the resampler always outputs stereo. 4374 if (mChannelCount == 1 && mReqChannelCount == 2) { 4375 channelCount = 1; 4376 } else { 4377 channelCount = 2; 4378 } 4379 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4380 mResampler->setSampleRate(mSampleRate); 4381 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4382 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4383 4384 // optmization: if mono to mono, alter input frame count as if we were inputing 4385 // stereo samples 4386 if (mChannelCount == 1 && mReqChannelCount == 1) { 4387 mFrameCount >>= 1; 4388 } 4389 4390 } 4391 mRsmpInIndex = mFrameCount; 4392} 4393 4394unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4395{ 4396 Mutex::Autolock _l(mLock); 4397 if (initCheck() != NO_ERROR) { 4398 return 0; 4399 } 4400 4401 return mInput->stream->get_input_frames_lost(mInput->stream); 4402} 4403 4404uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4405{ 4406 Mutex::Autolock _l(mLock); 4407 uint32_t result = 0; 4408 if (getEffectChain_l(sessionId) != 0) { 4409 result = EFFECT_SESSION; 4410 } 4411 4412 for (size_t i = 0; i < mTracks.size(); ++i) { 4413 if (sessionId == mTracks[i]->sessionId()) { 4414 result |= TRACK_SESSION; 4415 break; 4416 } 4417 } 4418 4419 return result; 4420} 4421 4422KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4423{ 4424 KeyedVector<int, bool> ids; 4425 Mutex::Autolock _l(mLock); 4426 for (size_t j = 0; j < mTracks.size(); ++j) { 4427 sp<RecordThread::RecordTrack> track = mTracks[j]; 4428 int sessionId = track->sessionId(); 4429 if (ids.indexOfKey(sessionId) < 0) { 4430 ids.add(sessionId, true); 4431 } 4432 } 4433 return ids; 4434} 4435 4436AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4437{ 4438 Mutex::Autolock _l(mLock); 4439 AudioStreamIn *input = mInput; 4440 mInput = NULL; 4441 return input; 4442} 4443 4444// this method must always be called either with ThreadBase mLock held or inside the thread loop 4445audio_stream_t* AudioFlinger::RecordThread::stream() const 4446{ 4447 if (mInput == NULL) { 4448 return NULL; 4449 } 4450 return &mInput->stream->common; 4451} 4452 4453status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4454{ 4455 // only one chain per input thread 4456 if (mEffectChains.size() != 0) { 4457 return INVALID_OPERATION; 4458 } 4459 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4460 4461 chain->setInBuffer(NULL); 4462 chain->setOutBuffer(NULL); 4463 4464 checkSuspendOnAddEffectChain_l(chain); 4465 4466 mEffectChains.add(chain); 4467 4468 return NO_ERROR; 4469} 4470 4471size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4472{ 4473 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4474 ALOGW_IF(mEffectChains.size() != 1, 4475 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4476 chain.get(), mEffectChains.size(), this); 4477 if (mEffectChains.size() == 1) { 4478 mEffectChains.removeAt(0); 4479 } 4480 return 0; 4481} 4482 4483}; // namespace android 4484