Threads.cpp revision c6ae3c8a261794fd4445e4e152d1ada074a3f92f
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 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 286 for (size_t i = 0; i < mConfigEvents.size(); i++) { 287 delete mConfigEvents[i]; 288 } 289 mConfigEvents.clear(); 290 291 mParamCond.broadcast(); 292 // do not lock the mutex in destructor 293 releaseWakeLock_l(); 294 if (mPowerManager != 0) { 295 sp<IBinder> binder = mPowerManager->asBinder(); 296 binder->unlinkToDeath(mDeathRecipient); 297 } 298} 299 300void AudioFlinger::ThreadBase::exit() 301{ 302 ALOGV("ThreadBase::exit"); 303 // do any cleanup required for exit to succeed 304 preExit(); 305 { 306 // This lock prevents the following race in thread (uniprocessor for illustration): 307 // if (!exitPending()) { 308 // // context switch from here to exit() 309 // // exit() calls requestExit(), what exitPending() observes 310 // // exit() calls signal(), which is dropped since no waiters 311 // // context switch back from exit() to here 312 // mWaitWorkCV.wait(...); 313 // // now thread is hung 314 // } 315 AutoMutex lock(mLock); 316 requestExit(); 317 mWaitWorkCV.broadcast(); 318 } 319 // When Thread::requestExitAndWait is made virtual and this method is renamed to 320 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 321 requestExitAndWait(); 322} 323 324status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 325{ 326 status_t status; 327 328 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 329 Mutex::Autolock _l(mLock); 330 331 mNewParameters.add(keyValuePairs); 332 mWaitWorkCV.signal(); 333 // wait condition with timeout in case the thread loop has exited 334 // before the request could be processed 335 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 336 status = mParamStatus; 337 mWaitWorkCV.signal(); 338 } else { 339 status = TIMED_OUT; 340 } 341 return status; 342} 343 344void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 345{ 346 Mutex::Autolock _l(mLock); 347 sendIoConfigEvent_l(event, param); 348} 349 350// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 351void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 352{ 353 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 354 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 355 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 356 param); 357 mWaitWorkCV.signal(); 358} 359 360// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 361void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 362{ 363 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 364 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 365 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 366 mConfigEvents.size(), pid, tid, prio); 367 mWaitWorkCV.signal(); 368} 369 370void AudioFlinger::ThreadBase::processConfigEvents() 371{ 372 mLock.lock(); 373 while (!mConfigEvents.isEmpty()) { 374 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 375 ConfigEvent *event = mConfigEvents[0]; 376 mConfigEvents.removeAt(0); 377 // release mLock before locking AudioFlinger mLock: lock order is always 378 // AudioFlinger then ThreadBase to avoid cross deadlock 379 mLock.unlock(); 380 switch(event->type()) { 381 case CFG_EVENT_PRIO: { 382 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 383 // FIXME Need to understand why this has be done asynchronously 384 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 385 true /*asynchronous*/); 386 if (err != 0) { 387 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 388 "error %d", 389 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 390 } 391 } break; 392 case CFG_EVENT_IO: { 393 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 394 mAudioFlinger->mLock.lock(); 395 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 396 mAudioFlinger->mLock.unlock(); 397 } break; 398 default: 399 ALOGE("processConfigEvents() unknown event type %d", event->type()); 400 break; 401 } 402 delete event; 403 mLock.lock(); 404 } 405 mLock.unlock(); 406} 407 408void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 409{ 410 const size_t SIZE = 256; 411 char buffer[SIZE]; 412 String8 result; 413 414 bool locked = AudioFlinger::dumpTryLock(mLock); 415 if (!locked) { 416 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 417 write(fd, buffer, strlen(buffer)); 418 } 419 420 snprintf(buffer, SIZE, "io handle: %d\n", mId); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 437 result.append(buffer); 438 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 439 result.append(buffer); 440 441 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 442 result.append(buffer); 443 result.append(" Index Command"); 444 for (size_t i = 0; i < mNewParameters.size(); ++i) { 445 snprintf(buffer, SIZE, "\n %02d ", i); 446 result.append(buffer); 447 result.append(mNewParameters[i]); 448 } 449 450 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 451 result.append(buffer); 452 for (size_t i = 0; i < mConfigEvents.size(); i++) { 453 mConfigEvents[i]->dump(buffer, SIZE); 454 result.append(buffer); 455 } 456 result.append("\n"); 457 458 write(fd, result.string(), result.size()); 459 460 if (locked) { 461 mLock.unlock(); 462 } 463} 464 465void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 466{ 467 const size_t SIZE = 256; 468 char buffer[SIZE]; 469 String8 result; 470 471 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 472 write(fd, buffer, strlen(buffer)); 473 474 for (size_t i = 0; i < mEffectChains.size(); ++i) { 475 sp<EffectChain> chain = mEffectChains[i]; 476 if (chain != 0) { 477 chain->dump(fd, args); 478 } 479 } 480} 481 482void AudioFlinger::ThreadBase::acquireWakeLock() 483{ 484 Mutex::Autolock _l(mLock); 485 acquireWakeLock_l(); 486} 487 488void AudioFlinger::ThreadBase::acquireWakeLock_l() 489{ 490 if (mPowerManager == 0) { 491 // use checkService() to avoid blocking if power service is not up yet 492 sp<IBinder> binder = 493 defaultServiceManager()->checkService(String16("power")); 494 if (binder == 0) { 495 ALOGW("Thread %s cannot connect to the power manager service", mName); 496 } else { 497 mPowerManager = interface_cast<IPowerManager>(binder); 498 binder->linkToDeath(mDeathRecipient); 499 } 500 } 501 if (mPowerManager != 0) { 502 sp<IBinder> binder = new BBinder(); 503 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 504 binder, 505 String16(mName), 506 String16("media")); 507 if (status == NO_ERROR) { 508 mWakeLockToken = binder; 509 } 510 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 511 } 512} 513 514void AudioFlinger::ThreadBase::releaseWakeLock() 515{ 516 Mutex::Autolock _l(mLock); 517 releaseWakeLock_l(); 518} 519 520void AudioFlinger::ThreadBase::releaseWakeLock_l() 521{ 522 if (mWakeLockToken != 0) { 523 ALOGV("releaseWakeLock_l() %s", mName); 524 if (mPowerManager != 0) { 525 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 526 } 527 mWakeLockToken.clear(); 528 } 529} 530 531void AudioFlinger::ThreadBase::clearPowerManager() 532{ 533 Mutex::Autolock _l(mLock); 534 releaseWakeLock_l(); 535 mPowerManager.clear(); 536} 537 538void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 539{ 540 sp<ThreadBase> thread = mThread.promote(); 541 if (thread != 0) { 542 thread->clearPowerManager(); 543 } 544 ALOGW("power manager service died !!!"); 545} 546 547void AudioFlinger::ThreadBase::setEffectSuspended( 548 const effect_uuid_t *type, bool suspend, int sessionId) 549{ 550 Mutex::Autolock _l(mLock); 551 setEffectSuspended_l(type, suspend, sessionId); 552} 553 554void AudioFlinger::ThreadBase::setEffectSuspended_l( 555 const effect_uuid_t *type, bool suspend, int sessionId) 556{ 557 sp<EffectChain> chain = getEffectChain_l(sessionId); 558 if (chain != 0) { 559 if (type != NULL) { 560 chain->setEffectSuspended_l(type, suspend); 561 } else { 562 chain->setEffectSuspendedAll_l(suspend); 563 } 564 } 565 566 updateSuspendedSessions_l(type, suspend, sessionId); 567} 568 569void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 570{ 571 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 572 if (index < 0) { 573 return; 574 } 575 576 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 577 mSuspendedSessions.valueAt(index); 578 579 for (size_t i = 0; i < sessionEffects.size(); i++) { 580 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 581 for (int j = 0; j < desc->mRefCount; j++) { 582 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 583 chain->setEffectSuspendedAll_l(true); 584 } else { 585 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 586 desc->mType.timeLow); 587 chain->setEffectSuspended_l(&desc->mType, true); 588 } 589 } 590 } 591} 592 593void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 594 bool suspend, 595 int sessionId) 596{ 597 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 598 599 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 600 601 if (suspend) { 602 if (index >= 0) { 603 sessionEffects = mSuspendedSessions.valueAt(index); 604 } else { 605 mSuspendedSessions.add(sessionId, sessionEffects); 606 } 607 } else { 608 if (index < 0) { 609 return; 610 } 611 sessionEffects = mSuspendedSessions.valueAt(index); 612 } 613 614 615 int key = EffectChain::kKeyForSuspendAll; 616 if (type != NULL) { 617 key = type->timeLow; 618 } 619 index = sessionEffects.indexOfKey(key); 620 621 sp<SuspendedSessionDesc> desc; 622 if (suspend) { 623 if (index >= 0) { 624 desc = sessionEffects.valueAt(index); 625 } else { 626 desc = new SuspendedSessionDesc(); 627 if (type != NULL) { 628 desc->mType = *type; 629 } 630 sessionEffects.add(key, desc); 631 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 632 } 633 desc->mRefCount++; 634 } else { 635 if (index < 0) { 636 return; 637 } 638 desc = sessionEffects.valueAt(index); 639 if (--desc->mRefCount == 0) { 640 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 641 sessionEffects.removeItemsAt(index); 642 if (sessionEffects.isEmpty()) { 643 ALOGV("updateSuspendedSessions_l() restore removing session %d", 644 sessionId); 645 mSuspendedSessions.removeItem(sessionId); 646 } 647 } 648 } 649 if (!sessionEffects.isEmpty()) { 650 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 651 } 652} 653 654void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 655 bool enabled, 656 int sessionId) 657{ 658 Mutex::Autolock _l(mLock); 659 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 660} 661 662void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 663 bool enabled, 664 int sessionId) 665{ 666 if (mType != RECORD) { 667 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 668 // another session. This gives the priority to well behaved effect control panels 669 // and applications not using global effects. 670 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 671 // global effects 672 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 673 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 674 } 675 } 676 677 sp<EffectChain> chain = getEffectChain_l(sessionId); 678 if (chain != 0) { 679 chain->checkSuspendOnEffectEnabled(effect, enabled); 680 } 681} 682 683// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 684sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 685 const sp<AudioFlinger::Client>& client, 686 const sp<IEffectClient>& effectClient, 687 int32_t priority, 688 int sessionId, 689 effect_descriptor_t *desc, 690 int *enabled, 691 status_t *status 692 ) 693{ 694 sp<EffectModule> effect; 695 sp<EffectHandle> handle; 696 status_t lStatus; 697 sp<EffectChain> chain; 698 bool chainCreated = false; 699 bool effectCreated = false; 700 bool effectRegistered = false; 701 702 lStatus = initCheck(); 703 if (lStatus != NO_ERROR) { 704 ALOGW("createEffect_l() Audio driver not initialized."); 705 goto Exit; 706 } 707 708 // Do not allow effects with session ID 0 on direct output or duplicating threads 709 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 710 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 711 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 712 desc->name, sessionId); 713 lStatus = BAD_VALUE; 714 goto Exit; 715 } 716 // Only Pre processor effects are allowed on input threads and only on input threads 717 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 718 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 719 desc->name, desc->flags, mType); 720 lStatus = BAD_VALUE; 721 goto Exit; 722 } 723 724 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 725 726 { // scope for mLock 727 Mutex::Autolock _l(mLock); 728 729 // check for existing effect chain with the requested audio session 730 chain = getEffectChain_l(sessionId); 731 if (chain == 0) { 732 // create a new chain for this session 733 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 734 chain = new EffectChain(this, sessionId); 735 addEffectChain_l(chain); 736 chain->setStrategy(getStrategyForSession_l(sessionId)); 737 chainCreated = true; 738 } else { 739 effect = chain->getEffectFromDesc_l(desc); 740 } 741 742 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 743 744 if (effect == 0) { 745 int id = mAudioFlinger->nextUniqueId(); 746 // Check CPU and memory usage 747 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 748 if (lStatus != NO_ERROR) { 749 goto Exit; 750 } 751 effectRegistered = true; 752 // create a new effect module if none present in the chain 753 effect = new EffectModule(this, chain, desc, id, sessionId); 754 lStatus = effect->status(); 755 if (lStatus != NO_ERROR) { 756 goto Exit; 757 } 758 lStatus = chain->addEffect_l(effect); 759 if (lStatus != NO_ERROR) { 760 goto Exit; 761 } 762 effectCreated = true; 763 764 effect->setDevice(mOutDevice); 765 effect->setDevice(mInDevice); 766 effect->setMode(mAudioFlinger->getMode()); 767 effect->setAudioSource(mAudioSource); 768 } 769 // create effect handle and connect it to effect module 770 handle = new EffectHandle(effect, client, effectClient, priority); 771 lStatus = effect->addHandle(handle.get()); 772 if (enabled != NULL) { 773 *enabled = (int)effect->isEnabled(); 774 } 775 } 776 777Exit: 778 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 779 Mutex::Autolock _l(mLock); 780 if (effectCreated) { 781 chain->removeEffect_l(effect); 782 } 783 if (effectRegistered) { 784 AudioSystem::unregisterEffect(effect->id()); 785 } 786 if (chainCreated) { 787 removeEffectChain_l(chain); 788 } 789 handle.clear(); 790 } 791 792 if (status != NULL) { 793 *status = lStatus; 794 } 795 return handle; 796} 797 798sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 799{ 800 Mutex::Autolock _l(mLock); 801 return getEffect_l(sessionId, effectId); 802} 803 804sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 805{ 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 808} 809 810// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 811// PlaybackThread::mLock held 812status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 813{ 814 // check for existing effect chain with the requested audio session 815 int sessionId = effect->sessionId(); 816 sp<EffectChain> chain = getEffectChain_l(sessionId); 817 bool chainCreated = false; 818 819 if (chain == 0) { 820 // create a new chain for this session 821 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 822 chain = new EffectChain(this, sessionId); 823 addEffectChain_l(chain); 824 chain->setStrategy(getStrategyForSession_l(sessionId)); 825 chainCreated = true; 826 } 827 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 828 829 if (chain->getEffectFromId_l(effect->id()) != 0) { 830 ALOGW("addEffect_l() %p effect %s already present in chain %p", 831 this, effect->desc().name, chain.get()); 832 return BAD_VALUE; 833 } 834 835 status_t status = chain->addEffect_l(effect); 836 if (status != NO_ERROR) { 837 if (chainCreated) { 838 removeEffectChain_l(chain); 839 } 840 return status; 841 } 842 843 effect->setDevice(mOutDevice); 844 effect->setDevice(mInDevice); 845 effect->setMode(mAudioFlinger->getMode()); 846 effect->setAudioSource(mAudioSource); 847 return NO_ERROR; 848} 849 850void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 851 852 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 853 effect_descriptor_t desc = effect->desc(); 854 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 855 detachAuxEffect_l(effect->id()); 856 } 857 858 sp<EffectChain> chain = effect->chain().promote(); 859 if (chain != 0) { 860 // remove effect chain if removing last effect 861 if (chain->removeEffect_l(effect) == 0) { 862 removeEffectChain_l(chain); 863 } 864 } else { 865 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 866 } 867} 868 869void AudioFlinger::ThreadBase::lockEffectChains_l( 870 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 871{ 872 effectChains = mEffectChains; 873 for (size_t i = 0; i < mEffectChains.size(); i++) { 874 mEffectChains[i]->lock(); 875 } 876} 877 878void AudioFlinger::ThreadBase::unlockEffectChains( 879 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 880{ 881 for (size_t i = 0; i < effectChains.size(); i++) { 882 effectChains[i]->unlock(); 883 } 884} 885 886sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 887{ 888 Mutex::Autolock _l(mLock); 889 return getEffectChain_l(sessionId); 890} 891 892sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 893{ 894 size_t size = mEffectChains.size(); 895 for (size_t i = 0; i < size; i++) { 896 if (mEffectChains[i]->sessionId() == sessionId) { 897 return mEffectChains[i]; 898 } 899 } 900 return 0; 901} 902 903void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 904{ 905 Mutex::Autolock _l(mLock); 906 size_t size = mEffectChains.size(); 907 for (size_t i = 0; i < size; i++) { 908 mEffectChains[i]->setMode_l(mode); 909 } 910} 911 912void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 913 EffectHandle *handle, 914 bool unpinIfLast) { 915 916 Mutex::Autolock _l(mLock); 917 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 918 // delete the effect module if removing last handle on it 919 if (effect->removeHandle(handle) == 0) { 920 if (!effect->isPinned() || unpinIfLast) { 921 removeEffect_l(effect); 922 AudioSystem::unregisterEffect(effect->id()); 923 } 924 } 925} 926 927// ---------------------------------------------------------------------------- 928// Playback 929// ---------------------------------------------------------------------------- 930 931AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 932 AudioStreamOut* output, 933 audio_io_handle_t id, 934 audio_devices_t device, 935 type_t type) 936 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 937 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 938 // mStreamTypes[] initialized in constructor body 939 mOutput(output), 940 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 941 mMixerStatus(MIXER_IDLE), 942 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 943 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 944 mScreenState(AudioFlinger::mScreenState), 945 // index 0 is reserved for normal mixer's submix 946 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 947{ 948 snprintf(mName, kNameLength, "AudioOut_%X", id); 949 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 950 951 // Assumes constructor is called by AudioFlinger with it's mLock held, but 952 // it would be safer to explicitly pass initial masterVolume/masterMute as 953 // parameter. 954 // 955 // If the HAL we are using has support for master volume or master mute, 956 // then do not attenuate or mute during mixing (just leave the volume at 1.0 957 // and the mute set to false). 958 mMasterVolume = audioFlinger->masterVolume_l(); 959 mMasterMute = audioFlinger->masterMute_l(); 960 if (mOutput && mOutput->audioHwDev) { 961 if (mOutput->audioHwDev->canSetMasterVolume()) { 962 mMasterVolume = 1.0; 963 } 964 965 if (mOutput->audioHwDev->canSetMasterMute()) { 966 mMasterMute = false; 967 } 968 } 969 970 readOutputParameters(); 971 972 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 973 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 974 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 975 stream = (audio_stream_type_t) (stream + 1)) { 976 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 977 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 978 } 979 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 980 // because mAudioFlinger doesn't have one to copy from 981} 982 983AudioFlinger::PlaybackThread::~PlaybackThread() 984{ 985 mAudioFlinger->unregisterWriter(mNBLogWriter); 986 delete [] mMixBuffer; 987} 988 989void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 990{ 991 dumpInternals(fd, args); 992 dumpTracks(fd, args); 993 dumpEffectChains(fd, args); 994} 995 996void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 997{ 998 const size_t SIZE = 256; 999 char buffer[SIZE]; 1000 String8 result; 1001 1002 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1003 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1004 const stream_type_t *st = &mStreamTypes[i]; 1005 if (i > 0) { 1006 result.appendFormat(", "); 1007 } 1008 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1009 if (st->mute) { 1010 result.append("M"); 1011 } 1012 } 1013 result.append("\n"); 1014 write(fd, result.string(), result.length()); 1015 result.clear(); 1016 1017 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1018 result.append(buffer); 1019 Track::appendDumpHeader(result); 1020 for (size_t i = 0; i < mTracks.size(); ++i) { 1021 sp<Track> track = mTracks[i]; 1022 if (track != 0) { 1023 track->dump(buffer, SIZE); 1024 result.append(buffer); 1025 } 1026 } 1027 1028 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1029 result.append(buffer); 1030 Track::appendDumpHeader(result); 1031 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1032 sp<Track> track = mActiveTracks[i].promote(); 1033 if (track != 0) { 1034 track->dump(buffer, SIZE); 1035 result.append(buffer); 1036 } 1037 } 1038 write(fd, result.string(), result.size()); 1039 1040 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1041 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1042 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1043 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1044} 1045 1046void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1047{ 1048 const size_t SIZE = 256; 1049 char buffer[SIZE]; 1050 String8 result; 1051 1052 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1053 result.append(buffer); 1054 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1055 ns2ms(systemTime() - mLastWriteTime)); 1056 result.append(buffer); 1057 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1058 result.append(buffer); 1059 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1060 result.append(buffer); 1061 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1062 result.append(buffer); 1063 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1064 result.append(buffer); 1065 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1066 result.append(buffer); 1067 write(fd, result.string(), result.size()); 1068 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1069 1070 dumpBase(fd, args); 1071} 1072 1073// Thread virtuals 1074status_t AudioFlinger::PlaybackThread::readyToRun() 1075{ 1076 status_t status = initCheck(); 1077 if (status == NO_ERROR) { 1078 ALOGI("AudioFlinger's thread %p ready to run", this); 1079 } else { 1080 ALOGE("No working audio driver found."); 1081 } 1082 return status; 1083} 1084 1085void AudioFlinger::PlaybackThread::onFirstRef() 1086{ 1087 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1088} 1089 1090// ThreadBase virtuals 1091void AudioFlinger::PlaybackThread::preExit() 1092{ 1093 ALOGV(" preExit()"); 1094 // FIXME this is using hard-coded strings but in the future, this functionality will be 1095 // converted to use audio HAL extensions required to support tunneling 1096 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1097} 1098 1099// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1100sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1101 const sp<AudioFlinger::Client>& client, 1102 audio_stream_type_t streamType, 1103 uint32_t sampleRate, 1104 audio_format_t format, 1105 audio_channel_mask_t channelMask, 1106 size_t frameCount, 1107 const sp<IMemory>& sharedBuffer, 1108 int sessionId, 1109 IAudioFlinger::track_flags_t *flags, 1110 pid_t tid, 1111 status_t *status) 1112{ 1113 sp<Track> track; 1114 status_t lStatus; 1115 1116 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1117 1118 // client expresses a preference for FAST, but we get the final say 1119 if (*flags & IAudioFlinger::TRACK_FAST) { 1120 if ( 1121 // not timed 1122 (!isTimed) && 1123 // either of these use cases: 1124 ( 1125 // use case 1: shared buffer with any frame count 1126 ( 1127 (sharedBuffer != 0) 1128 ) || 1129 // use case 2: callback handler and frame count is default or at least as large as HAL 1130 ( 1131 (tid != -1) && 1132 ((frameCount == 0) || 1133 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1134 ) 1135 ) && 1136 // PCM data 1137 audio_is_linear_pcm(format) && 1138 // mono or stereo 1139 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1140 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1141#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1142 // hardware sample rate 1143 (sampleRate == mSampleRate) && 1144#endif 1145 // normal mixer has an associated fast mixer 1146 hasFastMixer() && 1147 // there are sufficient fast track slots available 1148 (mFastTrackAvailMask != 0) 1149 // FIXME test that MixerThread for this fast track has a capable output HAL 1150 // FIXME add a permission test also? 1151 ) { 1152 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1153 if (frameCount == 0) { 1154 frameCount = mFrameCount * kFastTrackMultiplier; 1155 } 1156 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1157 frameCount, mFrameCount); 1158 } else { 1159 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1160 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1161 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1162 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1163 audio_is_linear_pcm(format), 1164 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1165 *flags &= ~IAudioFlinger::TRACK_FAST; 1166 // For compatibility with AudioTrack calculation, buffer depth is forced 1167 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1168 // This is probably too conservative, but legacy application code may depend on it. 1169 // If you change this calculation, also review the start threshold which is related. 1170 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1171 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1172 if (minBufCount < 2) { 1173 minBufCount = 2; 1174 } 1175 size_t minFrameCount = mNormalFrameCount * minBufCount; 1176 if (frameCount < minFrameCount) { 1177 frameCount = minFrameCount; 1178 } 1179 } 1180 } 1181 1182 if (mType == DIRECT) { 1183 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1184 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1185 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1186 "for output %p with format %d", 1187 sampleRate, format, channelMask, mOutput, mFormat); 1188 lStatus = BAD_VALUE; 1189 goto Exit; 1190 } 1191 } 1192 } else { 1193 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1194 if (sampleRate > mSampleRate*2) { 1195 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1196 lStatus = BAD_VALUE; 1197 goto Exit; 1198 } 1199 } 1200 1201 lStatus = initCheck(); 1202 if (lStatus != NO_ERROR) { 1203 ALOGE("Audio driver not initialized."); 1204 goto Exit; 1205 } 1206 1207 { // scope for mLock 1208 Mutex::Autolock _l(mLock); 1209 1210 // all tracks in same audio session must share the same routing strategy otherwise 1211 // conflicts will happen when tracks are moved from one output to another by audio policy 1212 // manager 1213 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1214 for (size_t i = 0; i < mTracks.size(); ++i) { 1215 sp<Track> t = mTracks[i]; 1216 if (t != 0 && !t->isOutputTrack()) { 1217 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1218 if (sessionId == t->sessionId() && strategy != actual) { 1219 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1220 strategy, actual); 1221 lStatus = BAD_VALUE; 1222 goto Exit; 1223 } 1224 } 1225 } 1226 1227 if (!isTimed) { 1228 track = new Track(this, client, streamType, sampleRate, format, 1229 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1230 } else { 1231 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1232 channelMask, frameCount, sharedBuffer, sessionId); 1233 } 1234 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1235 lStatus = NO_MEMORY; 1236 goto Exit; 1237 } 1238 mTracks.add(track); 1239 1240 sp<EffectChain> chain = getEffectChain_l(sessionId); 1241 if (chain != 0) { 1242 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1243 track->setMainBuffer(chain->inBuffer()); 1244 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1245 chain->incTrackCnt(); 1246 } 1247 1248 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1249 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1250 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1251 // so ask activity manager to do this on our behalf 1252 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1253 } 1254 } 1255 1256 lStatus = NO_ERROR; 1257 1258Exit: 1259 if (status) { 1260 *status = lStatus; 1261 } 1262 return track; 1263} 1264 1265uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1266{ 1267 return latency; 1268} 1269 1270uint32_t AudioFlinger::PlaybackThread::latency() const 1271{ 1272 Mutex::Autolock _l(mLock); 1273 return latency_l(); 1274} 1275uint32_t AudioFlinger::PlaybackThread::latency_l() const 1276{ 1277 if (initCheck() == NO_ERROR) { 1278 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1279 } else { 1280 return 0; 1281 } 1282} 1283 1284void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1285{ 1286 Mutex::Autolock _l(mLock); 1287 // Don't apply master volume in SW if our HAL can do it for us. 1288 if (mOutput && mOutput->audioHwDev && 1289 mOutput->audioHwDev->canSetMasterVolume()) { 1290 mMasterVolume = 1.0; 1291 } else { 1292 mMasterVolume = value; 1293 } 1294} 1295 1296void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1297{ 1298 Mutex::Autolock _l(mLock); 1299 // Don't apply master mute in SW if our HAL can do it for us. 1300 if (mOutput && mOutput->audioHwDev && 1301 mOutput->audioHwDev->canSetMasterMute()) { 1302 mMasterMute = false; 1303 } else { 1304 mMasterMute = muted; 1305 } 1306} 1307 1308void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1309{ 1310 Mutex::Autolock _l(mLock); 1311 mStreamTypes[stream].volume = value; 1312} 1313 1314void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 mStreamTypes[stream].mute = muted; 1318} 1319 1320float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1321{ 1322 Mutex::Autolock _l(mLock); 1323 return mStreamTypes[stream].volume; 1324} 1325 1326// addTrack_l() must be called with ThreadBase::mLock held 1327status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1328{ 1329 status_t status = ALREADY_EXISTS; 1330 1331 // set retry count for buffer fill 1332 track->mRetryCount = kMaxTrackStartupRetries; 1333 if (mActiveTracks.indexOf(track) < 0) { 1334 // the track is newly added, make sure it fills up all its 1335 // buffers before playing. This is to ensure the client will 1336 // effectively get the latency it requested. 1337 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1338 track->mResetDone = false; 1339 track->mPresentationCompleteFrames = 0; 1340 mActiveTracks.add(track); 1341 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1342 if (chain != 0) { 1343 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1344 track->sessionId()); 1345 chain->incActiveTrackCnt(); 1346 } 1347 1348 status = NO_ERROR; 1349 } 1350 1351 ALOGV("mWaitWorkCV.broadcast"); 1352 mWaitWorkCV.broadcast(); 1353 1354 return status; 1355} 1356 1357// destroyTrack_l() must be called with ThreadBase::mLock held 1358void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1359{ 1360 track->mState = TrackBase::TERMINATED; 1361 // active tracks are removed by threadLoop() 1362 if (mActiveTracks.indexOf(track) < 0) { 1363 removeTrack_l(track); 1364 } 1365} 1366 1367void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1368{ 1369 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1370 mTracks.remove(track); 1371 deleteTrackName_l(track->name()); 1372 // redundant as track is about to be destroyed, for dumpsys only 1373 track->mName = -1; 1374 if (track->isFastTrack()) { 1375 int index = track->mFastIndex; 1376 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1377 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1378 mFastTrackAvailMask |= 1 << index; 1379 // redundant as track is about to be destroyed, for dumpsys only 1380 track->mFastIndex = -1; 1381 } 1382 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1383 if (chain != 0) { 1384 chain->decTrackCnt(); 1385 } 1386} 1387 1388String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1389{ 1390 String8 out_s8 = String8(""); 1391 char *s; 1392 1393 Mutex::Autolock _l(mLock); 1394 if (initCheck() != NO_ERROR) { 1395 return out_s8; 1396 } 1397 1398 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1399 out_s8 = String8(s); 1400 free(s); 1401 return out_s8; 1402} 1403 1404// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1405void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1406 AudioSystem::OutputDescriptor desc; 1407 void *param2 = NULL; 1408 1409 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1410 param); 1411 1412 switch (event) { 1413 case AudioSystem::OUTPUT_OPENED: 1414 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1415 desc.channels = mChannelMask; 1416 desc.samplingRate = mSampleRate; 1417 desc.format = mFormat; 1418 desc.frameCount = mNormalFrameCount; // FIXME see 1419 // AudioFlinger::frameCount(audio_io_handle_t) 1420 desc.latency = latency(); 1421 param2 = &desc; 1422 break; 1423 1424 case AudioSystem::STREAM_CONFIG_CHANGED: 1425 param2 = ¶m; 1426 case AudioSystem::OUTPUT_CLOSED: 1427 default: 1428 break; 1429 } 1430 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1431} 1432 1433void AudioFlinger::PlaybackThread::readOutputParameters() 1434{ 1435 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1436 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1437 mChannelCount = (uint16_t)popcount(mChannelMask); 1438 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1439 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1440 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1441 if (mFrameCount & 15) { 1442 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1443 mFrameCount); 1444 } 1445 1446 // Calculate size of normal mix buffer relative to the HAL output buffer size 1447 double multiplier = 1.0; 1448 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1449 kUseFastMixer == FastMixer_Dynamic)) { 1450 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1451 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1452 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1453 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1454 maxNormalFrameCount = maxNormalFrameCount & ~15; 1455 if (maxNormalFrameCount < minNormalFrameCount) { 1456 maxNormalFrameCount = minNormalFrameCount; 1457 } 1458 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1459 if (multiplier <= 1.0) { 1460 multiplier = 1.0; 1461 } else if (multiplier <= 2.0) { 1462 if (2 * mFrameCount <= maxNormalFrameCount) { 1463 multiplier = 2.0; 1464 } else { 1465 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1466 } 1467 } else { 1468 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1469 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1470 // track, but we sometimes have to do this to satisfy the maximum frame count 1471 // constraint) 1472 // FIXME this rounding up should not be done if no HAL SRC 1473 uint32_t truncMult = (uint32_t) multiplier; 1474 if ((truncMult & 1)) { 1475 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1476 ++truncMult; 1477 } 1478 } 1479 multiplier = (double) truncMult; 1480 } 1481 } 1482 mNormalFrameCount = multiplier * mFrameCount; 1483 // round up to nearest 16 frames to satisfy AudioMixer 1484 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1485 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1486 mNormalFrameCount); 1487 1488 delete[] mMixBuffer; 1489 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1490 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1491 1492 // force reconfiguration of effect chains and engines to take new buffer size and audio 1493 // parameters into account 1494 // Note that mLock is not held when readOutputParameters() is called from the constructor 1495 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1496 // matter. 1497 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1498 Vector< sp<EffectChain> > effectChains = mEffectChains; 1499 for (size_t i = 0; i < effectChains.size(); i ++) { 1500 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1501 } 1502} 1503 1504 1505status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1506{ 1507 if (halFrames == NULL || dspFrames == NULL) { 1508 return BAD_VALUE; 1509 } 1510 Mutex::Autolock _l(mLock); 1511 if (initCheck() != NO_ERROR) { 1512 return INVALID_OPERATION; 1513 } 1514 size_t framesWritten = mBytesWritten / mFrameSize; 1515 *halFrames = framesWritten; 1516 1517 if (isSuspended()) { 1518 // return an estimation of rendered frames when the output is suspended 1519 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1520 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1521 return NO_ERROR; 1522 } else { 1523 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1524 } 1525} 1526 1527uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1528{ 1529 Mutex::Autolock _l(mLock); 1530 uint32_t result = 0; 1531 if (getEffectChain_l(sessionId) != 0) { 1532 result = EFFECT_SESSION; 1533 } 1534 1535 for (size_t i = 0; i < mTracks.size(); ++i) { 1536 sp<Track> track = mTracks[i]; 1537 if (sessionId == track->sessionId() && !track->isInvalid()) { 1538 result |= TRACK_SESSION; 1539 break; 1540 } 1541 } 1542 1543 return result; 1544} 1545 1546uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1547{ 1548 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1549 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1550 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1551 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1552 } 1553 for (size_t i = 0; i < mTracks.size(); i++) { 1554 sp<Track> track = mTracks[i]; 1555 if (sessionId == track->sessionId() && !track->isInvalid()) { 1556 return AudioSystem::getStrategyForStream(track->streamType()); 1557 } 1558 } 1559 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1560} 1561 1562 1563AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1564{ 1565 Mutex::Autolock _l(mLock); 1566 return mOutput; 1567} 1568 1569AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1570{ 1571 Mutex::Autolock _l(mLock); 1572 AudioStreamOut *output = mOutput; 1573 mOutput = NULL; 1574 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1575 // must push a NULL and wait for ack 1576 mOutputSink.clear(); 1577 mPipeSink.clear(); 1578 mNormalSink.clear(); 1579 return output; 1580} 1581 1582// this method must always be called either with ThreadBase mLock held or inside the thread loop 1583audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1584{ 1585 if (mOutput == NULL) { 1586 return NULL; 1587 } 1588 return &mOutput->stream->common; 1589} 1590 1591uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1592{ 1593 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1594} 1595 1596status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1597{ 1598 if (!isValidSyncEvent(event)) { 1599 return BAD_VALUE; 1600 } 1601 1602 Mutex::Autolock _l(mLock); 1603 1604 for (size_t i = 0; i < mTracks.size(); ++i) { 1605 sp<Track> track = mTracks[i]; 1606 if (event->triggerSession() == track->sessionId()) { 1607 (void) track->setSyncEvent(event); 1608 return NO_ERROR; 1609 } 1610 } 1611 1612 return NAME_NOT_FOUND; 1613} 1614 1615bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1616{ 1617 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1618} 1619 1620void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1621 const Vector< sp<Track> >& tracksToRemove) 1622{ 1623 size_t count = tracksToRemove.size(); 1624 if (CC_UNLIKELY(count)) { 1625 for (size_t i = 0 ; i < count ; i++) { 1626 const sp<Track>& track = tracksToRemove.itemAt(i); 1627 if ((track->sharedBuffer() != 0) && 1628 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1629 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1630 } 1631 } 1632 } 1633 1634} 1635 1636void AudioFlinger::PlaybackThread::checkSilentMode_l() 1637{ 1638 if (!mMasterMute) { 1639 char value[PROPERTY_VALUE_MAX]; 1640 if (property_get("ro.audio.silent", value, "0") > 0) { 1641 char *endptr; 1642 unsigned long ul = strtoul(value, &endptr, 0); 1643 if (*endptr == '\0' && ul != 0) { 1644 ALOGD("Silence is golden"); 1645 // The setprop command will not allow a property to be changed after 1646 // the first time it is set, so we don't have to worry about un-muting. 1647 setMasterMute_l(true); 1648 } 1649 } 1650 } 1651} 1652 1653// shared by MIXER and DIRECT, overridden by DUPLICATING 1654void AudioFlinger::PlaybackThread::threadLoop_write() 1655{ 1656 // FIXME rewrite to reduce number of system calls 1657 mLastWriteTime = systemTime(); 1658 mInWrite = true; 1659 int bytesWritten; 1660 1661 // If an NBAIO sink is present, use it to write the normal mixer's submix 1662 if (mNormalSink != 0) { 1663#define mBitShift 2 // FIXME 1664 size_t count = mixBufferSize >> mBitShift; 1665 ATRACE_BEGIN("write"); 1666 // update the setpoint when AudioFlinger::mScreenState changes 1667 uint32_t screenState = AudioFlinger::mScreenState; 1668 if (screenState != mScreenState) { 1669 mScreenState = screenState; 1670 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1671 if (pipe != NULL) { 1672 pipe->setAvgFrames((mScreenState & 1) ? 1673 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1674 } 1675 } 1676 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1677 ATRACE_END(); 1678 if (framesWritten > 0) { 1679 bytesWritten = framesWritten << mBitShift; 1680 } else { 1681 bytesWritten = framesWritten; 1682 } 1683 // otherwise use the HAL / AudioStreamOut directly 1684 } else { 1685 // Direct output thread. 1686 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1687 } 1688 1689 if (bytesWritten > 0) { 1690 mBytesWritten += mixBufferSize; 1691 } 1692 mNumWrites++; 1693 mInWrite = false; 1694} 1695 1696/* 1697The derived values that are cached: 1698 - mixBufferSize from frame count * frame size 1699 - activeSleepTime from activeSleepTimeUs() 1700 - idleSleepTime from idleSleepTimeUs() 1701 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1702 - maxPeriod from frame count and sample rate (MIXER only) 1703 1704The parameters that affect these derived values are: 1705 - frame count 1706 - frame size 1707 - sample rate 1708 - device type: A2DP or not 1709 - device latency 1710 - format: PCM or not 1711 - active sleep time 1712 - idle sleep time 1713*/ 1714 1715void AudioFlinger::PlaybackThread::cacheParameters_l() 1716{ 1717 mixBufferSize = mNormalFrameCount * mFrameSize; 1718 activeSleepTime = activeSleepTimeUs(); 1719 idleSleepTime = idleSleepTimeUs(); 1720} 1721 1722void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1723{ 1724 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1725 this, streamType, mTracks.size()); 1726 Mutex::Autolock _l(mLock); 1727 1728 size_t size = mTracks.size(); 1729 for (size_t i = 0; i < size; i++) { 1730 sp<Track> t = mTracks[i]; 1731 if (t->streamType() == streamType) { 1732 t->invalidate(); 1733 } 1734 } 1735} 1736 1737status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1738{ 1739 int session = chain->sessionId(); 1740 int16_t *buffer = mMixBuffer; 1741 bool ownsBuffer = false; 1742 1743 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1744 if (session > 0) { 1745 // Only one effect chain can be present in direct output thread and it uses 1746 // the mix buffer as input 1747 if (mType != DIRECT) { 1748 size_t numSamples = mNormalFrameCount * mChannelCount; 1749 buffer = new int16_t[numSamples]; 1750 memset(buffer, 0, numSamples * sizeof(int16_t)); 1751 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1752 ownsBuffer = true; 1753 } 1754 1755 // Attach all tracks with same session ID to this chain. 1756 for (size_t i = 0; i < mTracks.size(); ++i) { 1757 sp<Track> track = mTracks[i]; 1758 if (session == track->sessionId()) { 1759 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1760 buffer); 1761 track->setMainBuffer(buffer); 1762 chain->incTrackCnt(); 1763 } 1764 } 1765 1766 // indicate all active tracks in the chain 1767 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1768 sp<Track> track = mActiveTracks[i].promote(); 1769 if (track == 0) { 1770 continue; 1771 } 1772 if (session == track->sessionId()) { 1773 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1774 chain->incActiveTrackCnt(); 1775 } 1776 } 1777 } 1778 1779 chain->setInBuffer(buffer, ownsBuffer); 1780 chain->setOutBuffer(mMixBuffer); 1781 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1782 // chains list in order to be processed last as it contains output stage effects 1783 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1784 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1785 // after track specific effects and before output stage 1786 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1787 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1788 // Effect chain for other sessions are inserted at beginning of effect 1789 // chains list to be processed before output mix effects. Relative order between other 1790 // sessions is not important 1791 size_t size = mEffectChains.size(); 1792 size_t i = 0; 1793 for (i = 0; i < size; i++) { 1794 if (mEffectChains[i]->sessionId() < session) { 1795 break; 1796 } 1797 } 1798 mEffectChains.insertAt(chain, i); 1799 checkSuspendOnAddEffectChain_l(chain); 1800 1801 return NO_ERROR; 1802} 1803 1804size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1805{ 1806 int session = chain->sessionId(); 1807 1808 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1809 1810 for (size_t i = 0; i < mEffectChains.size(); i++) { 1811 if (chain == mEffectChains[i]) { 1812 mEffectChains.removeAt(i); 1813 // detach all active tracks from the chain 1814 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1815 sp<Track> track = mActiveTracks[i].promote(); 1816 if (track == 0) { 1817 continue; 1818 } 1819 if (session == track->sessionId()) { 1820 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1821 chain.get(), session); 1822 chain->decActiveTrackCnt(); 1823 } 1824 } 1825 1826 // detach all tracks with same session ID from this chain 1827 for (size_t i = 0; i < mTracks.size(); ++i) { 1828 sp<Track> track = mTracks[i]; 1829 if (session == track->sessionId()) { 1830 track->setMainBuffer(mMixBuffer); 1831 chain->decTrackCnt(); 1832 } 1833 } 1834 break; 1835 } 1836 } 1837 return mEffectChains.size(); 1838} 1839 1840status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1841 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1842{ 1843 Mutex::Autolock _l(mLock); 1844 return attachAuxEffect_l(track, EffectId); 1845} 1846 1847status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1848 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1849{ 1850 status_t status = NO_ERROR; 1851 1852 if (EffectId == 0) { 1853 track->setAuxBuffer(0, NULL); 1854 } else { 1855 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1856 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1857 if (effect != 0) { 1858 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1859 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1860 } else { 1861 status = INVALID_OPERATION; 1862 } 1863 } else { 1864 status = BAD_VALUE; 1865 } 1866 } 1867 return status; 1868} 1869 1870void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1871{ 1872 for (size_t i = 0; i < mTracks.size(); ++i) { 1873 sp<Track> track = mTracks[i]; 1874 if (track->auxEffectId() == effectId) { 1875 attachAuxEffect_l(track, 0); 1876 } 1877 } 1878} 1879 1880bool AudioFlinger::PlaybackThread::threadLoop() 1881{ 1882 Vector< sp<Track> > tracksToRemove; 1883 1884 standbyTime = systemTime(); 1885 1886 // MIXER 1887 nsecs_t lastWarning = 0; 1888 1889 // DUPLICATING 1890 // FIXME could this be made local to while loop? 1891 writeFrames = 0; 1892 1893 cacheParameters_l(); 1894 sleepTime = idleSleepTime; 1895 1896 if (mType == MIXER) { 1897 sleepTimeShift = 0; 1898 } 1899 1900 CpuStats cpuStats; 1901 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1902 1903 acquireWakeLock(); 1904 1905 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1906 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1907 // and then that string will be logged at the next convenient opportunity. 1908 const char *logString = NULL; 1909 1910 while (!exitPending()) 1911 { 1912 cpuStats.sample(myName); 1913 1914 Vector< sp<EffectChain> > effectChains; 1915 1916 processConfigEvents(); 1917 1918 { // scope for mLock 1919 1920 Mutex::Autolock _l(mLock); 1921 1922 if (logString != NULL) { 1923 mNBLogWriter->logTimestamp(); 1924 mNBLogWriter->log(logString); 1925 logString = NULL; 1926 } 1927 1928 if (checkForNewParameters_l()) { 1929 cacheParameters_l(); 1930 } 1931 1932 saveOutputTracks(); 1933 1934 // put audio hardware into standby after short delay 1935 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1936 isSuspended())) { 1937 if (!mStandby) { 1938 1939 threadLoop_standby(); 1940 1941 mStandby = true; 1942 } 1943 1944 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1945 // we're about to wait, flush the binder command buffer 1946 IPCThreadState::self()->flushCommands(); 1947 1948 clearOutputTracks(); 1949 1950 if (exitPending()) { 1951 break; 1952 } 1953 1954 releaseWakeLock_l(); 1955 // wait until we have something to do... 1956 ALOGV("%s going to sleep", myName.string()); 1957 mWaitWorkCV.wait(mLock); 1958 ALOGV("%s waking up", myName.string()); 1959 acquireWakeLock_l(); 1960 1961 mMixerStatus = MIXER_IDLE; 1962 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1963 mBytesWritten = 0; 1964 1965 checkSilentMode_l(); 1966 1967 standbyTime = systemTime() + standbyDelay; 1968 sleepTime = idleSleepTime; 1969 if (mType == MIXER) { 1970 sleepTimeShift = 0; 1971 } 1972 1973 continue; 1974 } 1975 } 1976 1977 // mMixerStatusIgnoringFastTracks is also updated internally 1978 mMixerStatus = prepareTracks_l(&tracksToRemove); 1979 1980 // prevent any changes in effect chain list and in each effect chain 1981 // during mixing and effect process as the audio buffers could be deleted 1982 // or modified if an effect is created or deleted 1983 lockEffectChains_l(effectChains); 1984 } 1985 1986 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1987 threadLoop_mix(); 1988 } else { 1989 threadLoop_sleepTime(); 1990 } 1991 1992 if (isSuspended()) { 1993 sleepTime = suspendSleepTimeUs(); 1994 mBytesWritten += mixBufferSize; 1995 } 1996 1997 // only process effects if we're going to write 1998 if (sleepTime == 0) { 1999 for (size_t i = 0; i < effectChains.size(); i ++) { 2000 effectChains[i]->process_l(); 2001 } 2002 } 2003 2004 // enable changes in effect chain 2005 unlockEffectChains(effectChains); 2006 2007 // sleepTime == 0 means we must write to audio hardware 2008 if (sleepTime == 0) { 2009 2010 threadLoop_write(); 2011 2012if (mType == MIXER) { 2013 // write blocked detection 2014 nsecs_t now = systemTime(); 2015 nsecs_t delta = now - mLastWriteTime; 2016 if (!mStandby && delta > maxPeriod) { 2017 mNumDelayedWrites++; 2018 if ((now - lastWarning) > kWarningThrottleNs) { 2019 ATRACE_NAME("underrun"); 2020 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2021 ns2ms(delta), mNumDelayedWrites, this); 2022 lastWarning = now; 2023 } 2024 } 2025} 2026 2027 mStandby = false; 2028 } else { 2029 usleep(sleepTime); 2030 } 2031 2032 // Finally let go of removed track(s), without the lock held 2033 // since we can't guarantee the destructors won't acquire that 2034 // same lock. This will also mutate and push a new fast mixer state. 2035 threadLoop_removeTracks(tracksToRemove); 2036 tracksToRemove.clear(); 2037 2038 // FIXME I don't understand the need for this here; 2039 // it was in the original code but maybe the 2040 // assignment in saveOutputTracks() makes this unnecessary? 2041 clearOutputTracks(); 2042 2043 // Effect chains will be actually deleted here if they were removed from 2044 // mEffectChains list during mixing or effects processing 2045 effectChains.clear(); 2046 2047 // FIXME Note that the above .clear() is no longer necessary since effectChains 2048 // is now local to this block, but will keep it for now (at least until merge done). 2049 } 2050 2051 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2052 if (mType == MIXER || mType == DIRECT) { 2053 // put output stream into standby mode 2054 if (!mStandby) { 2055 mOutput->stream->common.standby(&mOutput->stream->common); 2056 } 2057 } 2058 2059 releaseWakeLock(); 2060 2061 ALOGV("Thread %p type %d exiting", this, mType); 2062 return false; 2063} 2064 2065 2066// ---------------------------------------------------------------------------- 2067 2068AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2069 audio_io_handle_t id, audio_devices_t device, type_t type) 2070 : PlaybackThread(audioFlinger, output, id, device, type), 2071 // mAudioMixer below 2072 // mFastMixer below 2073 mFastMixerFutex(0) 2074 // mOutputSink below 2075 // mPipeSink below 2076 // mNormalSink below 2077{ 2078 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2079 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2080 "mFrameCount=%d, mNormalFrameCount=%d", 2081 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2082 mNormalFrameCount); 2083 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2084 2085 // FIXME - Current mixer implementation only supports stereo output 2086 if (mChannelCount != FCC_2) { 2087 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2088 } 2089 2090 // create an NBAIO sink for the HAL output stream, and negotiate 2091 mOutputSink = new AudioStreamOutSink(output->stream); 2092 size_t numCounterOffers = 0; 2093 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2094 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2095 ALOG_ASSERT(index == 0); 2096 2097 // initialize fast mixer depending on configuration 2098 bool initFastMixer; 2099 switch (kUseFastMixer) { 2100 case FastMixer_Never: 2101 initFastMixer = false; 2102 break; 2103 case FastMixer_Always: 2104 initFastMixer = true; 2105 break; 2106 case FastMixer_Static: 2107 case FastMixer_Dynamic: 2108 initFastMixer = mFrameCount < mNormalFrameCount; 2109 break; 2110 } 2111 if (initFastMixer) { 2112 2113 // create a MonoPipe to connect our submix to FastMixer 2114 NBAIO_Format format = mOutputSink->format(); 2115 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2116 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2117 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2118 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2119 const NBAIO_Format offers[1] = {format}; 2120 size_t numCounterOffers = 0; 2121 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2122 ALOG_ASSERT(index == 0); 2123 monoPipe->setAvgFrames((mScreenState & 1) ? 2124 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2125 mPipeSink = monoPipe; 2126 2127#ifdef TEE_SINK 2128 if (mTeeSinkOutputEnabled) { 2129 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2130 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2131 numCounterOffers = 0; 2132 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2133 ALOG_ASSERT(index == 0); 2134 mTeeSink = teeSink; 2135 PipeReader *teeSource = new PipeReader(*teeSink); 2136 numCounterOffers = 0; 2137 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2138 ALOG_ASSERT(index == 0); 2139 mTeeSource = teeSource; 2140 } 2141#endif 2142 2143 // create fast mixer and configure it initially with just one fast track for our submix 2144 mFastMixer = new FastMixer(); 2145 FastMixerStateQueue *sq = mFastMixer->sq(); 2146#ifdef STATE_QUEUE_DUMP 2147 sq->setObserverDump(&mStateQueueObserverDump); 2148 sq->setMutatorDump(&mStateQueueMutatorDump); 2149#endif 2150 FastMixerState *state = sq->begin(); 2151 FastTrack *fastTrack = &state->mFastTracks[0]; 2152 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2153 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2154 fastTrack->mVolumeProvider = NULL; 2155 fastTrack->mGeneration++; 2156 state->mFastTracksGen++; 2157 state->mTrackMask = 1; 2158 // fast mixer will use the HAL output sink 2159 state->mOutputSink = mOutputSink.get(); 2160 state->mOutputSinkGen++; 2161 state->mFrameCount = mFrameCount; 2162 state->mCommand = FastMixerState::COLD_IDLE; 2163 // already done in constructor initialization list 2164 //mFastMixerFutex = 0; 2165 state->mColdFutexAddr = &mFastMixerFutex; 2166 state->mColdGen++; 2167 state->mDumpState = &mFastMixerDumpState; 2168#ifdef TEE_SINK 2169 state->mTeeSink = mTeeSink.get(); 2170#endif 2171 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2172 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2173 sq->end(); 2174 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2175 2176 // start the fast mixer 2177 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2178 pid_t tid = mFastMixer->getTid(); 2179 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2180 if (err != 0) { 2181 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2182 kPriorityFastMixer, getpid_cached, tid, err); 2183 } 2184 2185#ifdef AUDIO_WATCHDOG 2186 // create and start the watchdog 2187 mAudioWatchdog = new AudioWatchdog(); 2188 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2189 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2190 tid = mAudioWatchdog->getTid(); 2191 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2192 if (err != 0) { 2193 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2194 kPriorityFastMixer, getpid_cached, tid, err); 2195 } 2196#endif 2197 2198 } else { 2199 mFastMixer = NULL; 2200 } 2201 2202 switch (kUseFastMixer) { 2203 case FastMixer_Never: 2204 case FastMixer_Dynamic: 2205 mNormalSink = mOutputSink; 2206 break; 2207 case FastMixer_Always: 2208 mNormalSink = mPipeSink; 2209 break; 2210 case FastMixer_Static: 2211 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2212 break; 2213 } 2214} 2215 2216AudioFlinger::MixerThread::~MixerThread() 2217{ 2218 if (mFastMixer != NULL) { 2219 FastMixerStateQueue *sq = mFastMixer->sq(); 2220 FastMixerState *state = sq->begin(); 2221 if (state->mCommand == FastMixerState::COLD_IDLE) { 2222 int32_t old = android_atomic_inc(&mFastMixerFutex); 2223 if (old == -1) { 2224 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2225 } 2226 } 2227 state->mCommand = FastMixerState::EXIT; 2228 sq->end(); 2229 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2230 mFastMixer->join(); 2231 // Though the fast mixer thread has exited, it's state queue is still valid. 2232 // We'll use that extract the final state which contains one remaining fast track 2233 // corresponding to our sub-mix. 2234 state = sq->begin(); 2235 ALOG_ASSERT(state->mTrackMask == 1); 2236 FastTrack *fastTrack = &state->mFastTracks[0]; 2237 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2238 delete fastTrack->mBufferProvider; 2239 sq->end(false /*didModify*/); 2240 delete mFastMixer; 2241#ifdef AUDIO_WATCHDOG 2242 if (mAudioWatchdog != 0) { 2243 mAudioWatchdog->requestExit(); 2244 mAudioWatchdog->requestExitAndWait(); 2245 mAudioWatchdog.clear(); 2246 } 2247#endif 2248 } 2249 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2250 delete mAudioMixer; 2251} 2252 2253 2254uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2255{ 2256 if (mFastMixer != NULL) { 2257 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2258 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2259 } 2260 return latency; 2261} 2262 2263 2264void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2265{ 2266 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2267} 2268 2269void AudioFlinger::MixerThread::threadLoop_write() 2270{ 2271 // FIXME we should only do one push per cycle; confirm this is true 2272 // Start the fast mixer if it's not already running 2273 if (mFastMixer != NULL) { 2274 FastMixerStateQueue *sq = mFastMixer->sq(); 2275 FastMixerState *state = sq->begin(); 2276 if (state->mCommand != FastMixerState::MIX_WRITE && 2277 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2278 if (state->mCommand == FastMixerState::COLD_IDLE) { 2279 int32_t old = android_atomic_inc(&mFastMixerFutex); 2280 if (old == -1) { 2281 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2282 } 2283#ifdef AUDIO_WATCHDOG 2284 if (mAudioWatchdog != 0) { 2285 mAudioWatchdog->resume(); 2286 } 2287#endif 2288 } 2289 state->mCommand = FastMixerState::MIX_WRITE; 2290 sq->end(); 2291 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2292 if (kUseFastMixer == FastMixer_Dynamic) { 2293 mNormalSink = mPipeSink; 2294 } 2295 } else { 2296 sq->end(false /*didModify*/); 2297 } 2298 } 2299 PlaybackThread::threadLoop_write(); 2300} 2301 2302void AudioFlinger::MixerThread::threadLoop_standby() 2303{ 2304 // Idle the fast mixer if it's currently running 2305 if (mFastMixer != NULL) { 2306 FastMixerStateQueue *sq = mFastMixer->sq(); 2307 FastMixerState *state = sq->begin(); 2308 if (!(state->mCommand & FastMixerState::IDLE)) { 2309 state->mCommand = FastMixerState::COLD_IDLE; 2310 state->mColdFutexAddr = &mFastMixerFutex; 2311 state->mColdGen++; 2312 mFastMixerFutex = 0; 2313 sq->end(); 2314 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2315 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2316 if (kUseFastMixer == FastMixer_Dynamic) { 2317 mNormalSink = mOutputSink; 2318 } 2319#ifdef AUDIO_WATCHDOG 2320 if (mAudioWatchdog != 0) { 2321 mAudioWatchdog->pause(); 2322 } 2323#endif 2324 } else { 2325 sq->end(false /*didModify*/); 2326 } 2327 } 2328 PlaybackThread::threadLoop_standby(); 2329} 2330 2331// shared by MIXER and DIRECT, overridden by DUPLICATING 2332void AudioFlinger::PlaybackThread::threadLoop_standby() 2333{ 2334 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2335 mOutput->stream->common.standby(&mOutput->stream->common); 2336} 2337 2338void AudioFlinger::MixerThread::threadLoop_mix() 2339{ 2340 // obtain the presentation timestamp of the next output buffer 2341 int64_t pts; 2342 status_t status = INVALID_OPERATION; 2343 2344 if (mNormalSink != 0) { 2345 status = mNormalSink->getNextWriteTimestamp(&pts); 2346 } else { 2347 status = mOutputSink->getNextWriteTimestamp(&pts); 2348 } 2349 2350 if (status != NO_ERROR) { 2351 pts = AudioBufferProvider::kInvalidPTS; 2352 } 2353 2354 // mix buffers... 2355 mAudioMixer->process(pts); 2356 // increase sleep time progressively when application underrun condition clears. 2357 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2358 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2359 // such that we would underrun the audio HAL. 2360 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2361 sleepTimeShift--; 2362 } 2363 sleepTime = 0; 2364 standbyTime = systemTime() + standbyDelay; 2365 //TODO: delay standby when effects have a tail 2366} 2367 2368void AudioFlinger::MixerThread::threadLoop_sleepTime() 2369{ 2370 // If no tracks are ready, sleep once for the duration of an output 2371 // buffer size, then write 0s to the output 2372 if (sleepTime == 0) { 2373 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2374 sleepTime = activeSleepTime >> sleepTimeShift; 2375 if (sleepTime < kMinThreadSleepTimeUs) { 2376 sleepTime = kMinThreadSleepTimeUs; 2377 } 2378 // reduce sleep time in case of consecutive application underruns to avoid 2379 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2380 // duration we would end up writing less data than needed by the audio HAL if 2381 // the condition persists. 2382 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2383 sleepTimeShift++; 2384 } 2385 } else { 2386 sleepTime = idleSleepTime; 2387 } 2388 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2389 memset (mMixBuffer, 0, mixBufferSize); 2390 sleepTime = 0; 2391 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2392 "anticipated start"); 2393 } 2394 // TODO add standby time extension fct of effect tail 2395} 2396 2397// prepareTracks_l() must be called with ThreadBase::mLock held 2398AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2399 Vector< sp<Track> > *tracksToRemove) 2400{ 2401 2402 mixer_state mixerStatus = MIXER_IDLE; 2403 // find out which tracks need to be processed 2404 size_t count = mActiveTracks.size(); 2405 size_t mixedTracks = 0; 2406 size_t tracksWithEffect = 0; 2407 // counts only _active_ fast tracks 2408 size_t fastTracks = 0; 2409 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2410 2411 float masterVolume = mMasterVolume; 2412 bool masterMute = mMasterMute; 2413 2414 if (masterMute) { 2415 masterVolume = 0; 2416 } 2417 // Delegate master volume control to effect in output mix effect chain if needed 2418 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2419 if (chain != 0) { 2420 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2421 chain->setVolume_l(&v, &v); 2422 masterVolume = (float)((v + (1 << 23)) >> 24); 2423 chain.clear(); 2424 } 2425 2426 // prepare a new state to push 2427 FastMixerStateQueue *sq = NULL; 2428 FastMixerState *state = NULL; 2429 bool didModify = false; 2430 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2431 if (mFastMixer != NULL) { 2432 sq = mFastMixer->sq(); 2433 state = sq->begin(); 2434 } 2435 2436 for (size_t i=0 ; i<count ; i++) { 2437 sp<Track> t = mActiveTracks[i].promote(); 2438 if (t == 0) { 2439 continue; 2440 } 2441 2442 // this const just means the local variable doesn't change 2443 Track* const track = t.get(); 2444 2445 // process fast tracks 2446 if (track->isFastTrack()) { 2447 2448 // It's theoretically possible (though unlikely) for a fast track to be created 2449 // and then removed within the same normal mix cycle. This is not a problem, as 2450 // the track never becomes active so it's fast mixer slot is never touched. 2451 // The converse, of removing an (active) track and then creating a new track 2452 // at the identical fast mixer slot within the same normal mix cycle, 2453 // is impossible because the slot isn't marked available until the end of each cycle. 2454 int j = track->mFastIndex; 2455 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2456 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2457 FastTrack *fastTrack = &state->mFastTracks[j]; 2458 2459 // Determine whether the track is currently in underrun condition, 2460 // and whether it had a recent underrun. 2461 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2462 FastTrackUnderruns underruns = ftDump->mUnderruns; 2463 uint32_t recentFull = (underruns.mBitFields.mFull - 2464 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2465 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2466 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2467 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2468 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2469 uint32_t recentUnderruns = recentPartial + recentEmpty; 2470 track->mObservedUnderruns = underruns; 2471 // don't count underruns that occur while stopping or pausing 2472 // or stopped which can occur when flush() is called while active 2473 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2474 track->mUnderrunCount += recentUnderruns; 2475 } 2476 2477 // This is similar to the state machine for normal tracks, 2478 // with a few modifications for fast tracks. 2479 bool isActive = true; 2480 switch (track->mState) { 2481 case TrackBase::STOPPING_1: 2482 // track stays active in STOPPING_1 state until first underrun 2483 if (recentUnderruns > 0) { 2484 track->mState = TrackBase::STOPPING_2; 2485 } 2486 break; 2487 case TrackBase::PAUSING: 2488 // ramp down is not yet implemented 2489 track->setPaused(); 2490 break; 2491 case TrackBase::RESUMING: 2492 // ramp up is not yet implemented 2493 track->mState = TrackBase::ACTIVE; 2494 break; 2495 case TrackBase::ACTIVE: 2496 if (recentFull > 0 || recentPartial > 0) { 2497 // track has provided at least some frames recently: reset retry count 2498 track->mRetryCount = kMaxTrackRetries; 2499 } 2500 if (recentUnderruns == 0) { 2501 // no recent underruns: stay active 2502 break; 2503 } 2504 // there has recently been an underrun of some kind 2505 if (track->sharedBuffer() == 0) { 2506 // were any of the recent underruns "empty" (no frames available)? 2507 if (recentEmpty == 0) { 2508 // no, then ignore the partial underruns as they are allowed indefinitely 2509 break; 2510 } 2511 // there has recently been an "empty" underrun: decrement the retry counter 2512 if (--(track->mRetryCount) > 0) { 2513 break; 2514 } 2515 // indicate to client process that the track was disabled because of underrun; 2516 // it will then automatically call start() when data is available 2517 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2518 // remove from active list, but state remains ACTIVE [confusing but true] 2519 isActive = false; 2520 break; 2521 } 2522 // fall through 2523 case TrackBase::STOPPING_2: 2524 case TrackBase::PAUSED: 2525 case TrackBase::TERMINATED: 2526 case TrackBase::STOPPED: 2527 case TrackBase::FLUSHED: // flush() while active 2528 // Check for presentation complete if track is inactive 2529 // We have consumed all the buffers of this track. 2530 // This would be incomplete if we auto-paused on underrun 2531 { 2532 size_t audioHALFrames = 2533 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2534 size_t framesWritten = mBytesWritten / mFrameSize; 2535 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2536 // track stays in active list until presentation is complete 2537 break; 2538 } 2539 } 2540 if (track->isStopping_2()) { 2541 track->mState = TrackBase::STOPPED; 2542 } 2543 if (track->isStopped()) { 2544 // Can't reset directly, as fast mixer is still polling this track 2545 // track->reset(); 2546 // So instead mark this track as needing to be reset after push with ack 2547 resetMask |= 1 << i; 2548 } 2549 isActive = false; 2550 break; 2551 case TrackBase::IDLE: 2552 default: 2553 LOG_FATAL("unexpected track state %d", track->mState); 2554 } 2555 2556 if (isActive) { 2557 // was it previously inactive? 2558 if (!(state->mTrackMask & (1 << j))) { 2559 ExtendedAudioBufferProvider *eabp = track; 2560 VolumeProvider *vp = track; 2561 fastTrack->mBufferProvider = eabp; 2562 fastTrack->mVolumeProvider = vp; 2563 fastTrack->mSampleRate = track->mSampleRate; 2564 fastTrack->mChannelMask = track->mChannelMask; 2565 fastTrack->mGeneration++; 2566 state->mTrackMask |= 1 << j; 2567 didModify = true; 2568 // no acknowledgement required for newly active tracks 2569 } 2570 // cache the combined master volume and stream type volume for fast mixer; this 2571 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2572 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2573 ++fastTracks; 2574 } else { 2575 // was it previously active? 2576 if (state->mTrackMask & (1 << j)) { 2577 fastTrack->mBufferProvider = NULL; 2578 fastTrack->mGeneration++; 2579 state->mTrackMask &= ~(1 << j); 2580 didModify = true; 2581 // If any fast tracks were removed, we must wait for acknowledgement 2582 // because we're about to decrement the last sp<> on those tracks. 2583 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2584 } else { 2585 LOG_FATAL("fast track %d should have been active", j); 2586 } 2587 tracksToRemove->add(track); 2588 // Avoids a misleading display in dumpsys 2589 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2590 } 2591 continue; 2592 } 2593 2594 { // local variable scope to avoid goto warning 2595 2596 audio_track_cblk_t* cblk = track->cblk(); 2597 2598 // The first time a track is added we wait 2599 // for all its buffers to be filled before processing it 2600 int name = track->name(); 2601 // make sure that we have enough frames to mix one full buffer. 2602 // enforce this condition only once to enable draining the buffer in case the client 2603 // app does not call stop() and relies on underrun to stop: 2604 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2605 // during last round 2606 size_t desiredFrames; 2607 if (t->sampleRate() == mSampleRate) { 2608 desiredFrames = mNormalFrameCount; 2609 } else { 2610 // +1 for rounding and +1 for additional sample needed for interpolation 2611 desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2612 // add frames already consumed but not yet released by the resampler 2613 // because cblk->framesReady() will include these frames 2614 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2615 // the minimum track buffer size is normally twice the number of frames necessary 2616 // to fill one buffer and the resampler should not leave more than one buffer worth 2617 // of unreleased frames after each pass, but just in case... 2618 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2619 } 2620 uint32_t minFrames = 1; 2621 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2622 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2623 minFrames = desiredFrames; 2624 } 2625 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2626 size_t framesReady; 2627 if (track->sharedBuffer() == 0) { 2628 framesReady = track->framesReady(); 2629 } else if (track->isStopped()) { 2630 framesReady = 0; 2631 } else { 2632 framesReady = 1; 2633 } 2634 if ((framesReady >= minFrames) && track->isReady() && 2635 !track->isPaused() && !track->isTerminated()) 2636 { 2637 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2638 this); 2639 2640 mixedTracks++; 2641 2642 // track->mainBuffer() != mMixBuffer means there is an effect chain 2643 // connected to the track 2644 chain.clear(); 2645 if (track->mainBuffer() != mMixBuffer) { 2646 chain = getEffectChain_l(track->sessionId()); 2647 // Delegate volume control to effect in track effect chain if needed 2648 if (chain != 0) { 2649 tracksWithEffect++; 2650 } else { 2651 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2652 "session %d", 2653 name, track->sessionId()); 2654 } 2655 } 2656 2657 2658 int param = AudioMixer::VOLUME; 2659 if (track->mFillingUpStatus == Track::FS_FILLED) { 2660 // no ramp for the first volume setting 2661 track->mFillingUpStatus = Track::FS_ACTIVE; 2662 if (track->mState == TrackBase::RESUMING) { 2663 track->mState = TrackBase::ACTIVE; 2664 param = AudioMixer::RAMP_VOLUME; 2665 } 2666 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2667 } else if (cblk->server != 0) { 2668 // If the track is stopped before the first frame was mixed, 2669 // do not apply ramp 2670 param = AudioMixer::RAMP_VOLUME; 2671 } 2672 2673 // compute volume for this track 2674 uint32_t vl, vr, va; 2675 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2676 vl = vr = va = 0; 2677 if (track->isPausing()) { 2678 track->setPaused(); 2679 } 2680 } else { 2681 2682 // read original volumes with volume control 2683 float typeVolume = mStreamTypes[track->streamType()].volume; 2684 float v = masterVolume * typeVolume; 2685 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2686 uint32_t vlr = proxy->getVolumeLR(); 2687 vl = vlr & 0xFFFF; 2688 vr = vlr >> 16; 2689 // track volumes come from shared memory, so can't be trusted and must be clamped 2690 if (vl > MAX_GAIN_INT) { 2691 ALOGV("Track left volume out of range: %04X", vl); 2692 vl = MAX_GAIN_INT; 2693 } 2694 if (vr > MAX_GAIN_INT) { 2695 ALOGV("Track right volume out of range: %04X", vr); 2696 vr = MAX_GAIN_INT; 2697 } 2698 // now apply the master volume and stream type volume 2699 vl = (uint32_t)(v * vl) << 12; 2700 vr = (uint32_t)(v * vr) << 12; 2701 // assuming master volume and stream type volume each go up to 1.0, 2702 // vl and vr are now in 8.24 format 2703 2704 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2705 // send level comes from shared memory and so may be corrupt 2706 if (sendLevel > MAX_GAIN_INT) { 2707 ALOGV("Track send level out of range: %04X", sendLevel); 2708 sendLevel = MAX_GAIN_INT; 2709 } 2710 va = (uint32_t)(v * sendLevel); 2711 } 2712 // Delegate volume control to effect in track effect chain if needed 2713 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2714 // Do not ramp volume if volume is controlled by effect 2715 param = AudioMixer::VOLUME; 2716 track->mHasVolumeController = true; 2717 } else { 2718 // force no volume ramp when volume controller was just disabled or removed 2719 // from effect chain to avoid volume spike 2720 if (track->mHasVolumeController) { 2721 param = AudioMixer::VOLUME; 2722 } 2723 track->mHasVolumeController = false; 2724 } 2725 2726 // Convert volumes from 8.24 to 4.12 format 2727 // This additional clamping is needed in case chain->setVolume_l() overshot 2728 vl = (vl + (1 << 11)) >> 12; 2729 if (vl > MAX_GAIN_INT) { 2730 vl = MAX_GAIN_INT; 2731 } 2732 vr = (vr + (1 << 11)) >> 12; 2733 if (vr > MAX_GAIN_INT) { 2734 vr = MAX_GAIN_INT; 2735 } 2736 2737 if (va > MAX_GAIN_INT) { 2738 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2739 } 2740 2741 // XXX: these things DON'T need to be done each time 2742 mAudioMixer->setBufferProvider(name, track); 2743 mAudioMixer->enable(name); 2744 2745 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2746 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2747 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2748 mAudioMixer->setParameter( 2749 name, 2750 AudioMixer::TRACK, 2751 AudioMixer::FORMAT, (void *)track->format()); 2752 mAudioMixer->setParameter( 2753 name, 2754 AudioMixer::TRACK, 2755 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2756 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2757 uint32_t maxSampleRate = mSampleRate * 2; 2758 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 2759 if (reqSampleRate == 0) { 2760 reqSampleRate = mSampleRate; 2761 } else if (reqSampleRate > maxSampleRate) { 2762 reqSampleRate = maxSampleRate; 2763 } 2764 mAudioMixer->setParameter( 2765 name, 2766 AudioMixer::RESAMPLE, 2767 AudioMixer::SAMPLE_RATE, 2768 (void *)reqSampleRate); 2769 mAudioMixer->setParameter( 2770 name, 2771 AudioMixer::TRACK, 2772 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2773 mAudioMixer->setParameter( 2774 name, 2775 AudioMixer::TRACK, 2776 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2777 2778 // reset retry count 2779 track->mRetryCount = kMaxTrackRetries; 2780 2781 // If one track is ready, set the mixer ready if: 2782 // - the mixer was not ready during previous round OR 2783 // - no other track is not ready 2784 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2785 mixerStatus != MIXER_TRACKS_ENABLED) { 2786 mixerStatus = MIXER_TRACKS_READY; 2787 } 2788 } else { 2789 // only implemented for normal tracks, not fast tracks 2790 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 2791 // we missed desiredFrames whatever the actual number of frames missing was 2792 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 2793 // FIXME also wake futex so that underrun is noticed more quickly 2794 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 2795 } 2796 // clear effect chain input buffer if an active track underruns to avoid sending 2797 // previous audio buffer again to effects 2798 chain = getEffectChain_l(track->sessionId()); 2799 if (chain != 0) { 2800 chain->clearInputBuffer(); 2801 } 2802 2803 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2804 cblk->server, this); 2805 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2806 track->isStopped() || track->isPaused()) { 2807 // We have consumed all the buffers of this track. 2808 // Remove it from the list of active tracks. 2809 // TODO: use actual buffer filling status instead of latency when available from 2810 // audio HAL 2811 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2812 size_t framesWritten = mBytesWritten / mFrameSize; 2813 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2814 if (track->isStopped()) { 2815 track->reset(); 2816 } 2817 tracksToRemove->add(track); 2818 } 2819 } else { 2820 track->mUnderrunCount++; 2821 // No buffers for this track. Give it a few chances to 2822 // fill a buffer, then remove it from active list. 2823 if (--(track->mRetryCount) <= 0) { 2824 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2825 tracksToRemove->add(track); 2826 // indicate to client process that the track was disabled because of underrun; 2827 // it will then automatically call start() when data is available 2828 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2829 // If one track is not ready, mark the mixer also not ready if: 2830 // - the mixer was ready during previous round OR 2831 // - no other track is ready 2832 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2833 mixerStatus != MIXER_TRACKS_READY) { 2834 mixerStatus = MIXER_TRACKS_ENABLED; 2835 } 2836 } 2837 mAudioMixer->disable(name); 2838 } 2839 2840 } // local variable scope to avoid goto warning 2841track_is_ready: ; 2842 2843 } 2844 2845 // Push the new FastMixer state if necessary 2846 bool pauseAudioWatchdog = false; 2847 if (didModify) { 2848 state->mFastTracksGen++; 2849 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2850 if (kUseFastMixer == FastMixer_Dynamic && 2851 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2852 state->mCommand = FastMixerState::COLD_IDLE; 2853 state->mColdFutexAddr = &mFastMixerFutex; 2854 state->mColdGen++; 2855 mFastMixerFutex = 0; 2856 if (kUseFastMixer == FastMixer_Dynamic) { 2857 mNormalSink = mOutputSink; 2858 } 2859 // If we go into cold idle, need to wait for acknowledgement 2860 // so that fast mixer stops doing I/O. 2861 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2862 pauseAudioWatchdog = true; 2863 } 2864 } 2865 if (sq != NULL) { 2866 sq->end(didModify); 2867 sq->push(block); 2868 } 2869#ifdef AUDIO_WATCHDOG 2870 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2871 mAudioWatchdog->pause(); 2872 } 2873#endif 2874 2875 // Now perform the deferred reset on fast tracks that have stopped 2876 while (resetMask != 0) { 2877 size_t i = __builtin_ctz(resetMask); 2878 ALOG_ASSERT(i < count); 2879 resetMask &= ~(1 << i); 2880 sp<Track> t = mActiveTracks[i].promote(); 2881 if (t == 0) { 2882 continue; 2883 } 2884 Track* track = t.get(); 2885 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2886 track->reset(); 2887 } 2888 2889 // remove all the tracks that need to be... 2890 count = tracksToRemove->size(); 2891 if (CC_UNLIKELY(count)) { 2892 for (size_t i=0 ; i<count ; i++) { 2893 const sp<Track>& track = tracksToRemove->itemAt(i); 2894 mActiveTracks.remove(track); 2895 if (track->mainBuffer() != mMixBuffer) { 2896 chain = getEffectChain_l(track->sessionId()); 2897 if (chain != 0) { 2898 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2899 track->sessionId()); 2900 chain->decActiveTrackCnt(); 2901 } 2902 } 2903 if (track->isTerminated()) { 2904 removeTrack_l(track); 2905 } 2906 } 2907 } 2908 2909 // mix buffer must be cleared if all tracks are connected to an 2910 // effect chain as in this case the mixer will not write to 2911 // mix buffer and track effects will accumulate into it 2912 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2913 (mixedTracks == 0 && fastTracks > 0)) { 2914 // FIXME as a performance optimization, should remember previous zero status 2915 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2916 } 2917 2918 // if any fast tracks, then status is ready 2919 mMixerStatusIgnoringFastTracks = mixerStatus; 2920 if (fastTracks > 0) { 2921 mixerStatus = MIXER_TRACKS_READY; 2922 } 2923 return mixerStatus; 2924} 2925 2926// getTrackName_l() must be called with ThreadBase::mLock held 2927int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2928{ 2929 return mAudioMixer->getTrackName(channelMask, sessionId); 2930} 2931 2932// deleteTrackName_l() must be called with ThreadBase::mLock held 2933void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2934{ 2935 ALOGV("remove track (%d) and delete from mixer", name); 2936 mAudioMixer->deleteTrackName(name); 2937} 2938 2939// checkForNewParameters_l() must be called with ThreadBase::mLock held 2940bool AudioFlinger::MixerThread::checkForNewParameters_l() 2941{ 2942 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2943 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2944 bool reconfig = false; 2945 2946 while (!mNewParameters.isEmpty()) { 2947 2948 if (mFastMixer != NULL) { 2949 FastMixerStateQueue *sq = mFastMixer->sq(); 2950 FastMixerState *state = sq->begin(); 2951 if (!(state->mCommand & FastMixerState::IDLE)) { 2952 previousCommand = state->mCommand; 2953 state->mCommand = FastMixerState::HOT_IDLE; 2954 sq->end(); 2955 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2956 } else { 2957 sq->end(false /*didModify*/); 2958 } 2959 } 2960 2961 status_t status = NO_ERROR; 2962 String8 keyValuePair = mNewParameters[0]; 2963 AudioParameter param = AudioParameter(keyValuePair); 2964 int value; 2965 2966 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2967 reconfig = true; 2968 } 2969 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2970 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2971 status = BAD_VALUE; 2972 } else { 2973 reconfig = true; 2974 } 2975 } 2976 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2977 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2978 status = BAD_VALUE; 2979 } else { 2980 reconfig = true; 2981 } 2982 } 2983 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2984 // do not accept frame count changes if tracks are open as the track buffer 2985 // size depends on frame count and correct behavior would not be guaranteed 2986 // if frame count is changed after track creation 2987 if (!mTracks.isEmpty()) { 2988 status = INVALID_OPERATION; 2989 } else { 2990 reconfig = true; 2991 } 2992 } 2993 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2994#ifdef ADD_BATTERY_DATA 2995 // when changing the audio output device, call addBatteryData to notify 2996 // the change 2997 if (mOutDevice != value) { 2998 uint32_t params = 0; 2999 // check whether speaker is on 3000 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3001 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3002 } 3003 3004 audio_devices_t deviceWithoutSpeaker 3005 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3006 // check if any other device (except speaker) is on 3007 if (value & deviceWithoutSpeaker ) { 3008 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3009 } 3010 3011 if (params != 0) { 3012 addBatteryData(params); 3013 } 3014 } 3015#endif 3016 3017 // forward device change to effects that have requested to be 3018 // aware of attached audio device. 3019 if (value != AUDIO_DEVICE_NONE) { 3020 mOutDevice = value; 3021 for (size_t i = 0; i < mEffectChains.size(); i++) { 3022 mEffectChains[i]->setDevice_l(mOutDevice); 3023 } 3024 } 3025 } 3026 3027 if (status == NO_ERROR) { 3028 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3029 keyValuePair.string()); 3030 if (!mStandby && status == INVALID_OPERATION) { 3031 mOutput->stream->common.standby(&mOutput->stream->common); 3032 mStandby = true; 3033 mBytesWritten = 0; 3034 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3035 keyValuePair.string()); 3036 } 3037 if (status == NO_ERROR && reconfig) { 3038 delete mAudioMixer; 3039 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3040 mAudioMixer = NULL; 3041 readOutputParameters(); 3042 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3043 for (size_t i = 0; i < mTracks.size() ; i++) { 3044 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3045 if (name < 0) { 3046 break; 3047 } 3048 mTracks[i]->mName = name; 3049 } 3050 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3051 } 3052 } 3053 3054 mNewParameters.removeAt(0); 3055 3056 mParamStatus = status; 3057 mParamCond.signal(); 3058 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3059 // already timed out waiting for the status and will never signal the condition. 3060 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3061 } 3062 3063 if (!(previousCommand & FastMixerState::IDLE)) { 3064 ALOG_ASSERT(mFastMixer != NULL); 3065 FastMixerStateQueue *sq = mFastMixer->sq(); 3066 FastMixerState *state = sq->begin(); 3067 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3068 state->mCommand = previousCommand; 3069 sq->end(); 3070 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3071 } 3072 3073 return reconfig; 3074} 3075 3076 3077void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3078{ 3079 const size_t SIZE = 256; 3080 char buffer[SIZE]; 3081 String8 result; 3082 3083 PlaybackThread::dumpInternals(fd, args); 3084 3085 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3086 result.append(buffer); 3087 write(fd, result.string(), result.size()); 3088 3089 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3090 FastMixerDumpState copy = mFastMixerDumpState; 3091 copy.dump(fd); 3092 3093#ifdef STATE_QUEUE_DUMP 3094 // Similar for state queue 3095 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3096 observerCopy.dump(fd); 3097 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3098 mutatorCopy.dump(fd); 3099#endif 3100 3101#ifdef TEE_SINK 3102 // Write the tee output to a .wav file 3103 dumpTee(fd, mTeeSource, mId); 3104#endif 3105 3106#ifdef AUDIO_WATCHDOG 3107 if (mAudioWatchdog != 0) { 3108 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3109 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3110 wdCopy.dump(fd); 3111 } 3112#endif 3113} 3114 3115uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3116{ 3117 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3118} 3119 3120uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3121{ 3122 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3123} 3124 3125void AudioFlinger::MixerThread::cacheParameters_l() 3126{ 3127 PlaybackThread::cacheParameters_l(); 3128 3129 // FIXME: Relaxed timing because of a certain device that can't meet latency 3130 // Should be reduced to 2x after the vendor fixes the driver issue 3131 // increase threshold again due to low power audio mode. The way this warning 3132 // threshold is calculated and its usefulness should be reconsidered anyway. 3133 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3134} 3135 3136// ---------------------------------------------------------------------------- 3137 3138AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3139 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3140 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3141 // mLeftVolFloat, mRightVolFloat 3142{ 3143} 3144 3145AudioFlinger::DirectOutputThread::~DirectOutputThread() 3146{ 3147} 3148 3149AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3150 Vector< sp<Track> > *tracksToRemove 3151) 3152{ 3153 size_t count = mActiveTracks.size(); 3154 mixer_state mixerStatus = MIXER_IDLE; 3155 3156 // find out which tracks need to be processed 3157 for (size_t i = 0; i < count; i++) { 3158 sp<Track> t = mActiveTracks[i].promote(); 3159 // The track died recently 3160 if (t == 0) { 3161 continue; 3162 } 3163 3164 Track* const track = t.get(); 3165 audio_track_cblk_t* cblk = track->cblk(); 3166 3167 // The first time a track is added we wait 3168 // for all its buffers to be filled before processing it 3169 uint32_t minFrames; 3170 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3171 minFrames = mNormalFrameCount; 3172 } else { 3173 minFrames = 1; 3174 } 3175 if ((track->framesReady() >= minFrames) && track->isReady() && 3176 !track->isPaused() && !track->isTerminated()) 3177 { 3178 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3179 3180 if (track->mFillingUpStatus == Track::FS_FILLED) { 3181 track->mFillingUpStatus = Track::FS_ACTIVE; 3182 mLeftVolFloat = mRightVolFloat = 0; 3183 if (track->mState == TrackBase::RESUMING) { 3184 track->mState = TrackBase::ACTIVE; 3185 } 3186 } 3187 3188 // compute volume for this track 3189 float left, right; 3190 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3191 left = right = 0; 3192 if (track->isPausing()) { 3193 track->setPaused(); 3194 } 3195 } else { 3196 float typeVolume = mStreamTypes[track->streamType()].volume; 3197 float v = mMasterVolume * typeVolume; 3198 uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); 3199 float v_clamped = v * (vlr & 0xFFFF); 3200 if (v_clamped > MAX_GAIN) { 3201 v_clamped = MAX_GAIN; 3202 } 3203 left = v_clamped/MAX_GAIN; 3204 v_clamped = v * (vlr >> 16); 3205 if (v_clamped > MAX_GAIN) { 3206 v_clamped = MAX_GAIN; 3207 } 3208 right = v_clamped/MAX_GAIN; 3209 } 3210 // Only consider last track started for volume and mixer state control. 3211 // This is the last entry in mActiveTracks unless a track underruns. 3212 // As we only care about the transition phase between two tracks on a 3213 // direct output, it is not a problem to ignore the underrun case. 3214 if (i == (count - 1)) { 3215 if (left != mLeftVolFloat || right != mRightVolFloat) { 3216 mLeftVolFloat = left; 3217 mRightVolFloat = right; 3218 3219 // Convert volumes from float to 8.24 3220 uint32_t vl = (uint32_t)(left * (1 << 24)); 3221 uint32_t vr = (uint32_t)(right * (1 << 24)); 3222 3223 // Delegate volume control to effect in track effect chain if needed 3224 // only one effect chain can be present on DirectOutputThread, so if 3225 // there is one, the track is connected to it 3226 if (!mEffectChains.isEmpty()) { 3227 // Do not ramp volume if volume is controlled by effect 3228 mEffectChains[0]->setVolume_l(&vl, &vr); 3229 left = (float)vl / (1 << 24); 3230 right = (float)vr / (1 << 24); 3231 } 3232 mOutput->stream->set_volume(mOutput->stream, left, right); 3233 } 3234 3235 // reset retry count 3236 track->mRetryCount = kMaxTrackRetriesDirect; 3237 mActiveTrack = t; 3238 mixerStatus = MIXER_TRACKS_READY; 3239 } 3240 } else { 3241 // clear effect chain input buffer if the last active track started underruns 3242 // to avoid sending previous audio buffer again to effects 3243 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3244 mEffectChains[0]->clearInputBuffer(); 3245 } 3246 3247 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3248 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3249 track->isStopped() || track->isPaused()) { 3250 // We have consumed all the buffers of this track. 3251 // Remove it from the list of active tracks. 3252 // TODO: implement behavior for compressed audio 3253 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3254 size_t framesWritten = mBytesWritten / mFrameSize; 3255 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3256 if (track->isStopped()) { 3257 track->reset(); 3258 } 3259 tracksToRemove->add(track); 3260 } 3261 } else { 3262 // No buffers for this track. Give it a few chances to 3263 // fill a buffer, then remove it from active list. 3264 // Only consider last track started for mixer state control 3265 if (--(track->mRetryCount) <= 0) { 3266 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3267 tracksToRemove->add(track); 3268 } else if (i == (count -1)){ 3269 mixerStatus = MIXER_TRACKS_ENABLED; 3270 } 3271 } 3272 } 3273 } 3274 3275 // remove all the tracks that need to be... 3276 count = tracksToRemove->size(); 3277 if (CC_UNLIKELY(count)) { 3278 for (size_t i = 0 ; i < count ; i++) { 3279 const sp<Track>& track = tracksToRemove->itemAt(i); 3280 mActiveTracks.remove(track); 3281 if (!mEffectChains.isEmpty()) { 3282 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3283 track->sessionId()); 3284 mEffectChains[0]->decActiveTrackCnt(); 3285 } 3286 if (track->isTerminated()) { 3287 removeTrack_l(track); 3288 } 3289 } 3290 } 3291 3292 return mixerStatus; 3293} 3294 3295void AudioFlinger::DirectOutputThread::threadLoop_mix() 3296{ 3297 AudioBufferProvider::Buffer buffer; 3298 size_t frameCount = mFrameCount; 3299 int8_t *curBuf = (int8_t *)mMixBuffer; 3300 // output audio to hardware 3301 while (frameCount) { 3302 buffer.frameCount = frameCount; 3303 mActiveTrack->getNextBuffer(&buffer); 3304 if (CC_UNLIKELY(buffer.raw == NULL)) { 3305 memset(curBuf, 0, frameCount * mFrameSize); 3306 break; 3307 } 3308 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3309 frameCount -= buffer.frameCount; 3310 curBuf += buffer.frameCount * mFrameSize; 3311 mActiveTrack->releaseBuffer(&buffer); 3312 } 3313 sleepTime = 0; 3314 standbyTime = systemTime() + standbyDelay; 3315 mActiveTrack.clear(); 3316 3317} 3318 3319void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3320{ 3321 if (sleepTime == 0) { 3322 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3323 sleepTime = activeSleepTime; 3324 } else { 3325 sleepTime = idleSleepTime; 3326 } 3327 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3328 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3329 sleepTime = 0; 3330 } 3331} 3332 3333// getTrackName_l() must be called with ThreadBase::mLock held 3334int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3335 int sessionId) 3336{ 3337 return 0; 3338} 3339 3340// deleteTrackName_l() must be called with ThreadBase::mLock held 3341void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3342{ 3343} 3344 3345// checkForNewParameters_l() must be called with ThreadBase::mLock held 3346bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3347{ 3348 bool reconfig = false; 3349 3350 while (!mNewParameters.isEmpty()) { 3351 status_t status = NO_ERROR; 3352 String8 keyValuePair = mNewParameters[0]; 3353 AudioParameter param = AudioParameter(keyValuePair); 3354 int value; 3355 3356 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3357 // do not accept frame count changes if tracks are open as the track buffer 3358 // size depends on frame count and correct behavior would not be garantied 3359 // if frame count is changed after track creation 3360 if (!mTracks.isEmpty()) { 3361 status = INVALID_OPERATION; 3362 } else { 3363 reconfig = true; 3364 } 3365 } 3366 if (status == NO_ERROR) { 3367 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3368 keyValuePair.string()); 3369 if (!mStandby && status == INVALID_OPERATION) { 3370 mOutput->stream->common.standby(&mOutput->stream->common); 3371 mStandby = true; 3372 mBytesWritten = 0; 3373 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3374 keyValuePair.string()); 3375 } 3376 if (status == NO_ERROR && reconfig) { 3377 readOutputParameters(); 3378 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3379 } 3380 } 3381 3382 mNewParameters.removeAt(0); 3383 3384 mParamStatus = status; 3385 mParamCond.signal(); 3386 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3387 // already timed out waiting for the status and will never signal the condition. 3388 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3389 } 3390 return reconfig; 3391} 3392 3393uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3394{ 3395 uint32_t time; 3396 if (audio_is_linear_pcm(mFormat)) { 3397 time = PlaybackThread::activeSleepTimeUs(); 3398 } else { 3399 time = 10000; 3400 } 3401 return time; 3402} 3403 3404uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3405{ 3406 uint32_t time; 3407 if (audio_is_linear_pcm(mFormat)) { 3408 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3409 } else { 3410 time = 10000; 3411 } 3412 return time; 3413} 3414 3415uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3416{ 3417 uint32_t time; 3418 if (audio_is_linear_pcm(mFormat)) { 3419 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3420 } else { 3421 time = 10000; 3422 } 3423 return time; 3424} 3425 3426void AudioFlinger::DirectOutputThread::cacheParameters_l() 3427{ 3428 PlaybackThread::cacheParameters_l(); 3429 3430 // use shorter standby delay as on normal output to release 3431 // hardware resources as soon as possible 3432 standbyDelay = microseconds(activeSleepTime*2); 3433} 3434 3435// ---------------------------------------------------------------------------- 3436 3437AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3438 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3439 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3440 DUPLICATING), 3441 mWaitTimeMs(UINT_MAX) 3442{ 3443 addOutputTrack(mainThread); 3444} 3445 3446AudioFlinger::DuplicatingThread::~DuplicatingThread() 3447{ 3448 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3449 mOutputTracks[i]->destroy(); 3450 } 3451} 3452 3453void AudioFlinger::DuplicatingThread::threadLoop_mix() 3454{ 3455 // mix buffers... 3456 if (outputsReady(outputTracks)) { 3457 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3458 } else { 3459 memset(mMixBuffer, 0, mixBufferSize); 3460 } 3461 sleepTime = 0; 3462 writeFrames = mNormalFrameCount; 3463 standbyTime = systemTime() + standbyDelay; 3464} 3465 3466void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3467{ 3468 if (sleepTime == 0) { 3469 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3470 sleepTime = activeSleepTime; 3471 } else { 3472 sleepTime = idleSleepTime; 3473 } 3474 } else if (mBytesWritten != 0) { 3475 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3476 writeFrames = mNormalFrameCount; 3477 memset(mMixBuffer, 0, mixBufferSize); 3478 } else { 3479 // flush remaining overflow buffers in output tracks 3480 writeFrames = 0; 3481 } 3482 sleepTime = 0; 3483 } 3484} 3485 3486void AudioFlinger::DuplicatingThread::threadLoop_write() 3487{ 3488 for (size_t i = 0; i < outputTracks.size(); i++) { 3489 outputTracks[i]->write(mMixBuffer, writeFrames); 3490 } 3491 mBytesWritten += mixBufferSize; 3492} 3493 3494void AudioFlinger::DuplicatingThread::threadLoop_standby() 3495{ 3496 // DuplicatingThread implements standby by stopping all tracks 3497 for (size_t i = 0; i < outputTracks.size(); i++) { 3498 outputTracks[i]->stop(); 3499 } 3500} 3501 3502void AudioFlinger::DuplicatingThread::saveOutputTracks() 3503{ 3504 outputTracks = mOutputTracks; 3505} 3506 3507void AudioFlinger::DuplicatingThread::clearOutputTracks() 3508{ 3509 outputTracks.clear(); 3510} 3511 3512void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3513{ 3514 Mutex::Autolock _l(mLock); 3515 // FIXME explain this formula 3516 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3517 OutputTrack *outputTrack = new OutputTrack(thread, 3518 this, 3519 mSampleRate, 3520 mFormat, 3521 mChannelMask, 3522 frameCount); 3523 if (outputTrack->cblk() != NULL) { 3524 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3525 mOutputTracks.add(outputTrack); 3526 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3527 updateWaitTime_l(); 3528 } 3529} 3530 3531void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3532{ 3533 Mutex::Autolock _l(mLock); 3534 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3535 if (mOutputTracks[i]->thread() == thread) { 3536 mOutputTracks[i]->destroy(); 3537 mOutputTracks.removeAt(i); 3538 updateWaitTime_l(); 3539 return; 3540 } 3541 } 3542 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3543} 3544 3545// caller must hold mLock 3546void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3547{ 3548 mWaitTimeMs = UINT_MAX; 3549 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3550 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3551 if (strong != 0) { 3552 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3553 if (waitTimeMs < mWaitTimeMs) { 3554 mWaitTimeMs = waitTimeMs; 3555 } 3556 } 3557 } 3558} 3559 3560 3561bool AudioFlinger::DuplicatingThread::outputsReady( 3562 const SortedVector< sp<OutputTrack> > &outputTracks) 3563{ 3564 for (size_t i = 0; i < outputTracks.size(); i++) { 3565 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3566 if (thread == 0) { 3567 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3568 outputTracks[i].get()); 3569 return false; 3570 } 3571 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3572 // see note at standby() declaration 3573 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3574 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3575 thread.get()); 3576 return false; 3577 } 3578 } 3579 return true; 3580} 3581 3582uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3583{ 3584 return (mWaitTimeMs * 1000) / 2; 3585} 3586 3587void AudioFlinger::DuplicatingThread::cacheParameters_l() 3588{ 3589 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3590 updateWaitTime_l(); 3591 3592 MixerThread::cacheParameters_l(); 3593} 3594 3595// ---------------------------------------------------------------------------- 3596// Record 3597// ---------------------------------------------------------------------------- 3598 3599AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3600 AudioStreamIn *input, 3601 uint32_t sampleRate, 3602 audio_channel_mask_t channelMask, 3603 audio_io_handle_t id, 3604 audio_devices_t outDevice, 3605 audio_devices_t inDevice 3606#ifdef TEE_SINK 3607 , const sp<NBAIO_Sink>& teeSink 3608#endif 3609 ) : 3610 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3611 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3612 // mRsmpInIndex and mInputBytes set by readInputParameters() 3613 mReqChannelCount(popcount(channelMask)), 3614 mReqSampleRate(sampleRate) 3615 // mBytesRead is only meaningful while active, and so is cleared in start() 3616 // (but might be better to also clear here for dump?) 3617#ifdef TEE_SINK 3618 , mTeeSink(teeSink) 3619#endif 3620{ 3621 snprintf(mName, kNameLength, "AudioIn_%X", id); 3622 3623 readInputParameters(); 3624 3625} 3626 3627 3628AudioFlinger::RecordThread::~RecordThread() 3629{ 3630 delete[] mRsmpInBuffer; 3631 delete mResampler; 3632 delete[] mRsmpOutBuffer; 3633} 3634 3635void AudioFlinger::RecordThread::onFirstRef() 3636{ 3637 run(mName, PRIORITY_URGENT_AUDIO); 3638} 3639 3640status_t AudioFlinger::RecordThread::readyToRun() 3641{ 3642 status_t status = initCheck(); 3643 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3644 return status; 3645} 3646 3647bool AudioFlinger::RecordThread::threadLoop() 3648{ 3649 AudioBufferProvider::Buffer buffer; 3650 sp<RecordTrack> activeTrack; 3651 Vector< sp<EffectChain> > effectChains; 3652 3653 nsecs_t lastWarning = 0; 3654 3655 inputStandBy(); 3656 acquireWakeLock(); 3657 3658 // used to verify we've read at least once before evaluating how many bytes were read 3659 bool readOnce = false; 3660 3661 // start recording 3662 while (!exitPending()) { 3663 3664 processConfigEvents(); 3665 3666 { // scope for mLock 3667 Mutex::Autolock _l(mLock); 3668 checkForNewParameters_l(); 3669 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3670 standby(); 3671 3672 if (exitPending()) { 3673 break; 3674 } 3675 3676 releaseWakeLock_l(); 3677 ALOGV("RecordThread: loop stopping"); 3678 // go to sleep 3679 mWaitWorkCV.wait(mLock); 3680 ALOGV("RecordThread: loop starting"); 3681 acquireWakeLock_l(); 3682 continue; 3683 } 3684 if (mActiveTrack != 0) { 3685 if (mActiveTrack->mState == TrackBase::PAUSING) { 3686 standby(); 3687 mActiveTrack.clear(); 3688 mStartStopCond.broadcast(); 3689 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3690 if (mReqChannelCount != mActiveTrack->channelCount()) { 3691 mActiveTrack.clear(); 3692 mStartStopCond.broadcast(); 3693 } else if (readOnce) { 3694 // record start succeeds only if first read from audio input 3695 // succeeds 3696 if (mBytesRead >= 0) { 3697 mActiveTrack->mState = TrackBase::ACTIVE; 3698 } else { 3699 mActiveTrack.clear(); 3700 } 3701 mStartStopCond.broadcast(); 3702 } 3703 mStandby = false; 3704 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3705 removeTrack_l(mActiveTrack); 3706 mActiveTrack.clear(); 3707 } 3708 } 3709 lockEffectChains_l(effectChains); 3710 } 3711 3712 if (mActiveTrack != 0) { 3713 if (mActiveTrack->mState != TrackBase::ACTIVE && 3714 mActiveTrack->mState != TrackBase::RESUMING) { 3715 unlockEffectChains(effectChains); 3716 usleep(kRecordThreadSleepUs); 3717 continue; 3718 } 3719 for (size_t i = 0; i < effectChains.size(); i ++) { 3720 effectChains[i]->process_l(); 3721 } 3722 3723 buffer.frameCount = mFrameCount; 3724 status_t status = mActiveTrack->getNextBuffer(&buffer); 3725 if (CC_LIKELY(status == NO_ERROR)) { 3726 readOnce = true; 3727 size_t framesOut = buffer.frameCount; 3728 if (mResampler == NULL) { 3729 // no resampling 3730 while (framesOut) { 3731 size_t framesIn = mFrameCount - mRsmpInIndex; 3732 if (framesIn) { 3733 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3734 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3735 mActiveTrack->mFrameSize; 3736 if (framesIn > framesOut) 3737 framesIn = framesOut; 3738 mRsmpInIndex += framesIn; 3739 framesOut -= framesIn; 3740 if (mChannelCount == mReqChannelCount || 3741 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3742 memcpy(dst, src, framesIn * mFrameSize); 3743 } else { 3744 if (mChannelCount == 1) { 3745 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3746 (int16_t *)src, framesIn); 3747 } else { 3748 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3749 (int16_t *)src, framesIn); 3750 } 3751 } 3752 } 3753 if (framesOut && mFrameCount == mRsmpInIndex) { 3754 void *readInto; 3755 if (framesOut == mFrameCount && 3756 (mChannelCount == mReqChannelCount || 3757 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3758 readInto = buffer.raw; 3759 framesOut = 0; 3760 } else { 3761 readInto = mRsmpInBuffer; 3762 mRsmpInIndex = 0; 3763 } 3764 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3765 mInputBytes); 3766 if (mBytesRead <= 0) { 3767 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3768 { 3769 ALOGE("Error reading audio input"); 3770 // Force input into standby so that it tries to 3771 // recover at next read attempt 3772 inputStandBy(); 3773 usleep(kRecordThreadSleepUs); 3774 } 3775 mRsmpInIndex = mFrameCount; 3776 framesOut = 0; 3777 buffer.frameCount = 0; 3778 } 3779#ifdef TEE_SINK 3780 else if (mTeeSink != 0) { 3781 (void) mTeeSink->write(readInto, 3782 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3783 } 3784#endif 3785 } 3786 } 3787 } else { 3788 // resampling 3789 3790 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3791 // alter output frame count as if we were expecting stereo samples 3792 if (mChannelCount == 1 && mReqChannelCount == 1) { 3793 framesOut >>= 1; 3794 } 3795 mResampler->resample(mRsmpOutBuffer, framesOut, 3796 this /* AudioBufferProvider* */); 3797 // ditherAndClamp() works as long as all buffers returned by 3798 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3799 if (mChannelCount == 2 && mReqChannelCount == 1) { 3800 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3801 // the resampler always outputs stereo samples: 3802 // do post stereo to mono conversion 3803 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3804 framesOut); 3805 } else { 3806 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3807 } 3808 3809 } 3810 if (mFramestoDrop == 0) { 3811 mActiveTrack->releaseBuffer(&buffer); 3812 } else { 3813 if (mFramestoDrop > 0) { 3814 mFramestoDrop -= buffer.frameCount; 3815 if (mFramestoDrop <= 0) { 3816 clearSyncStartEvent(); 3817 } 3818 } else { 3819 mFramestoDrop += buffer.frameCount; 3820 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3821 mSyncStartEvent->isCancelled()) { 3822 ALOGW("Synced record %s, session %d, trigger session %d", 3823 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3824 mActiveTrack->sessionId(), 3825 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3826 clearSyncStartEvent(); 3827 } 3828 } 3829 } 3830 mActiveTrack->clearOverflow(); 3831 } 3832 // client isn't retrieving buffers fast enough 3833 else { 3834 if (!mActiveTrack->setOverflow()) { 3835 nsecs_t now = systemTime(); 3836 if ((now - lastWarning) > kWarningThrottleNs) { 3837 ALOGW("RecordThread: buffer overflow"); 3838 lastWarning = now; 3839 } 3840 } 3841 // Release the processor for a while before asking for a new buffer. 3842 // This will give the application more chance to read from the buffer and 3843 // clear the overflow. 3844 usleep(kRecordThreadSleepUs); 3845 } 3846 } 3847 // enable changes in effect chain 3848 unlockEffectChains(effectChains); 3849 effectChains.clear(); 3850 } 3851 3852 standby(); 3853 3854 { 3855 Mutex::Autolock _l(mLock); 3856 mActiveTrack.clear(); 3857 mStartStopCond.broadcast(); 3858 } 3859 3860 releaseWakeLock(); 3861 3862 ALOGV("RecordThread %p exiting", this); 3863 return false; 3864} 3865 3866void AudioFlinger::RecordThread::standby() 3867{ 3868 if (!mStandby) { 3869 inputStandBy(); 3870 mStandby = true; 3871 } 3872} 3873 3874void AudioFlinger::RecordThread::inputStandBy() 3875{ 3876 mInput->stream->common.standby(&mInput->stream->common); 3877} 3878 3879sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3880 const sp<AudioFlinger::Client>& client, 3881 uint32_t sampleRate, 3882 audio_format_t format, 3883 audio_channel_mask_t channelMask, 3884 size_t frameCount, 3885 int sessionId, 3886 IAudioFlinger::track_flags_t flags, 3887 pid_t tid, 3888 status_t *status) 3889{ 3890 sp<RecordTrack> track; 3891 status_t lStatus; 3892 3893 lStatus = initCheck(); 3894 if (lStatus != NO_ERROR) { 3895 ALOGE("Audio driver not initialized."); 3896 goto Exit; 3897 } 3898 3899 // FIXME use flags and tid similar to createTrack_l() 3900 3901 { // scope for mLock 3902 Mutex::Autolock _l(mLock); 3903 3904 track = new RecordTrack(this, client, sampleRate, 3905 format, channelMask, frameCount, sessionId); 3906 3907 if (track->getCblk() == 0) { 3908 lStatus = NO_MEMORY; 3909 goto Exit; 3910 } 3911 mTracks.add(track); 3912 3913 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3914 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3915 mAudioFlinger->btNrecIsOff(); 3916 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3917 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3918 } 3919 lStatus = NO_ERROR; 3920 3921Exit: 3922 if (status) { 3923 *status = lStatus; 3924 } 3925 return track; 3926} 3927 3928status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3929 AudioSystem::sync_event_t event, 3930 int triggerSession) 3931{ 3932 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3933 sp<ThreadBase> strongMe = this; 3934 status_t status = NO_ERROR; 3935 3936 if (event == AudioSystem::SYNC_EVENT_NONE) { 3937 clearSyncStartEvent(); 3938 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3939 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3940 triggerSession, 3941 recordTrack->sessionId(), 3942 syncStartEventCallback, 3943 this); 3944 // Sync event can be cancelled by the trigger session if the track is not in a 3945 // compatible state in which case we start record immediately 3946 if (mSyncStartEvent->isCancelled()) { 3947 clearSyncStartEvent(); 3948 } else { 3949 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3950 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3951 } 3952 } 3953 3954 { 3955 AutoMutex lock(mLock); 3956 if (mActiveTrack != 0) { 3957 if (recordTrack != mActiveTrack.get()) { 3958 status = -EBUSY; 3959 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3960 mActiveTrack->mState = TrackBase::ACTIVE; 3961 } 3962 return status; 3963 } 3964 3965 recordTrack->mState = TrackBase::IDLE; 3966 mActiveTrack = recordTrack; 3967 mLock.unlock(); 3968 status_t status = AudioSystem::startInput(mId); 3969 mLock.lock(); 3970 if (status != NO_ERROR) { 3971 mActiveTrack.clear(); 3972 clearSyncStartEvent(); 3973 return status; 3974 } 3975 mRsmpInIndex = mFrameCount; 3976 mBytesRead = 0; 3977 if (mResampler != NULL) { 3978 mResampler->reset(); 3979 } 3980 mActiveTrack->mState = TrackBase::RESUMING; 3981 // signal thread to start 3982 ALOGV("Signal record thread"); 3983 mWaitWorkCV.broadcast(); 3984 // do not wait for mStartStopCond if exiting 3985 if (exitPending()) { 3986 mActiveTrack.clear(); 3987 status = INVALID_OPERATION; 3988 goto startError; 3989 } 3990 mStartStopCond.wait(mLock); 3991 if (mActiveTrack == 0) { 3992 ALOGV("Record failed to start"); 3993 status = BAD_VALUE; 3994 goto startError; 3995 } 3996 ALOGV("Record started OK"); 3997 return status; 3998 } 3999 4000startError: 4001 AudioSystem::stopInput(mId); 4002 clearSyncStartEvent(); 4003 return status; 4004} 4005 4006void AudioFlinger::RecordThread::clearSyncStartEvent() 4007{ 4008 if (mSyncStartEvent != 0) { 4009 mSyncStartEvent->cancel(); 4010 } 4011 mSyncStartEvent.clear(); 4012 mFramestoDrop = 0; 4013} 4014 4015void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4016{ 4017 sp<SyncEvent> strongEvent = event.promote(); 4018 4019 if (strongEvent != 0) { 4020 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4021 me->handleSyncStartEvent(strongEvent); 4022 } 4023} 4024 4025void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4026{ 4027 if (event == mSyncStartEvent) { 4028 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4029 // from audio HAL 4030 mFramestoDrop = mFrameCount * 2; 4031 } 4032} 4033 4034bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4035 ALOGV("RecordThread::stop"); 4036 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4037 return false; 4038 } 4039 recordTrack->mState = TrackBase::PAUSING; 4040 // do not wait for mStartStopCond if exiting 4041 if (exitPending()) { 4042 return true; 4043 } 4044 mStartStopCond.wait(mLock); 4045 // if we have been restarted, recordTrack == mActiveTrack.get() here 4046 if (exitPending() || recordTrack != mActiveTrack.get()) { 4047 ALOGV("Record stopped OK"); 4048 return true; 4049 } 4050 return false; 4051} 4052 4053bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4054{ 4055 return false; 4056} 4057 4058status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4059{ 4060#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4061 if (!isValidSyncEvent(event)) { 4062 return BAD_VALUE; 4063 } 4064 4065 int eventSession = event->triggerSession(); 4066 status_t ret = NAME_NOT_FOUND; 4067 4068 Mutex::Autolock _l(mLock); 4069 4070 for (size_t i = 0; i < mTracks.size(); i++) { 4071 sp<RecordTrack> track = mTracks[i]; 4072 if (eventSession == track->sessionId()) { 4073 (void) track->setSyncEvent(event); 4074 ret = NO_ERROR; 4075 } 4076 } 4077 return ret; 4078#else 4079 return BAD_VALUE; 4080#endif 4081} 4082 4083// destroyTrack_l() must be called with ThreadBase::mLock held 4084void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4085{ 4086 track->mState = TrackBase::TERMINATED; 4087 // active tracks are removed by threadLoop() 4088 if (mActiveTrack != track) { 4089 removeTrack_l(track); 4090 } 4091} 4092 4093void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4094{ 4095 mTracks.remove(track); 4096 // need anything related to effects here? 4097} 4098 4099void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4100{ 4101 dumpInternals(fd, args); 4102 dumpTracks(fd, args); 4103 dumpEffectChains(fd, args); 4104} 4105 4106void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4107{ 4108 const size_t SIZE = 256; 4109 char buffer[SIZE]; 4110 String8 result; 4111 4112 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4113 result.append(buffer); 4114 4115 if (mActiveTrack != 0) { 4116 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4117 result.append(buffer); 4118 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4119 result.append(buffer); 4120 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4121 result.append(buffer); 4122 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4123 result.append(buffer); 4124 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4125 result.append(buffer); 4126 } else { 4127 result.append("No active record client\n"); 4128 } 4129 4130 write(fd, result.string(), result.size()); 4131 4132 dumpBase(fd, args); 4133} 4134 4135void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4136{ 4137 const size_t SIZE = 256; 4138 char buffer[SIZE]; 4139 String8 result; 4140 4141 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4142 result.append(buffer); 4143 RecordTrack::appendDumpHeader(result); 4144 for (size_t i = 0; i < mTracks.size(); ++i) { 4145 sp<RecordTrack> track = mTracks[i]; 4146 if (track != 0) { 4147 track->dump(buffer, SIZE); 4148 result.append(buffer); 4149 } 4150 } 4151 4152 if (mActiveTrack != 0) { 4153 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4154 result.append(buffer); 4155 RecordTrack::appendDumpHeader(result); 4156 mActiveTrack->dump(buffer, SIZE); 4157 result.append(buffer); 4158 4159 } 4160 write(fd, result.string(), result.size()); 4161} 4162 4163// AudioBufferProvider interface 4164status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4165{ 4166 size_t framesReq = buffer->frameCount; 4167 size_t framesReady = mFrameCount - mRsmpInIndex; 4168 int channelCount; 4169 4170 if (framesReady == 0) { 4171 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4172 if (mBytesRead <= 0) { 4173 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4174 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4175 // Force input into standby so that it tries to 4176 // recover at next read attempt 4177 inputStandBy(); 4178 usleep(kRecordThreadSleepUs); 4179 } 4180 buffer->raw = NULL; 4181 buffer->frameCount = 0; 4182 return NOT_ENOUGH_DATA; 4183 } 4184 mRsmpInIndex = 0; 4185 framesReady = mFrameCount; 4186 } 4187 4188 if (framesReq > framesReady) { 4189 framesReq = framesReady; 4190 } 4191 4192 if (mChannelCount == 1 && mReqChannelCount == 2) { 4193 channelCount = 1; 4194 } else { 4195 channelCount = 2; 4196 } 4197 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4198 buffer->frameCount = framesReq; 4199 return NO_ERROR; 4200} 4201 4202// AudioBufferProvider interface 4203void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4204{ 4205 mRsmpInIndex += buffer->frameCount; 4206 buffer->frameCount = 0; 4207} 4208 4209bool AudioFlinger::RecordThread::checkForNewParameters_l() 4210{ 4211 bool reconfig = false; 4212 4213 while (!mNewParameters.isEmpty()) { 4214 status_t status = NO_ERROR; 4215 String8 keyValuePair = mNewParameters[0]; 4216 AudioParameter param = AudioParameter(keyValuePair); 4217 int value; 4218 audio_format_t reqFormat = mFormat; 4219 uint32_t reqSamplingRate = mReqSampleRate; 4220 uint32_t reqChannelCount = mReqChannelCount; 4221 4222 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4223 reqSamplingRate = value; 4224 reconfig = true; 4225 } 4226 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4227 reqFormat = (audio_format_t) value; 4228 reconfig = true; 4229 } 4230 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4231 reqChannelCount = popcount(value); 4232 reconfig = true; 4233 } 4234 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4235 // do not accept frame count changes if tracks are open as the track buffer 4236 // size depends on frame count and correct behavior would not be guaranteed 4237 // if frame count is changed after track creation 4238 if (mActiveTrack != 0) { 4239 status = INVALID_OPERATION; 4240 } else { 4241 reconfig = true; 4242 } 4243 } 4244 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4245 // forward device change to effects that have requested to be 4246 // aware of attached audio device. 4247 for (size_t i = 0; i < mEffectChains.size(); i++) { 4248 mEffectChains[i]->setDevice_l(value); 4249 } 4250 4251 // store input device and output device but do not forward output device to audio HAL. 4252 // Note that status is ignored by the caller for output device 4253 // (see AudioFlinger::setParameters() 4254 if (audio_is_output_devices(value)) { 4255 mOutDevice = value; 4256 status = BAD_VALUE; 4257 } else { 4258 mInDevice = value; 4259 // disable AEC and NS if the device is a BT SCO headset supporting those 4260 // pre processings 4261 if (mTracks.size() > 0) { 4262 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4263 mAudioFlinger->btNrecIsOff(); 4264 for (size_t i = 0; i < mTracks.size(); i++) { 4265 sp<RecordTrack> track = mTracks[i]; 4266 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4267 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4268 } 4269 } 4270 } 4271 } 4272 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4273 mAudioSource != (audio_source_t)value) { 4274 // forward device change to effects that have requested to be 4275 // aware of attached audio device. 4276 for (size_t i = 0; i < mEffectChains.size(); i++) { 4277 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4278 } 4279 mAudioSource = (audio_source_t)value; 4280 } 4281 if (status == NO_ERROR) { 4282 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4283 keyValuePair.string()); 4284 if (status == INVALID_OPERATION) { 4285 inputStandBy(); 4286 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4287 keyValuePair.string()); 4288 } 4289 if (reconfig) { 4290 if (status == BAD_VALUE && 4291 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4292 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4293 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4294 <= (2 * reqSamplingRate)) && 4295 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4296 <= FCC_2 && 4297 (reqChannelCount <= FCC_2)) { 4298 status = NO_ERROR; 4299 } 4300 if (status == NO_ERROR) { 4301 readInputParameters(); 4302 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4303 } 4304 } 4305 } 4306 4307 mNewParameters.removeAt(0); 4308 4309 mParamStatus = status; 4310 mParamCond.signal(); 4311 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4312 // already timed out waiting for the status and will never signal the condition. 4313 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4314 } 4315 return reconfig; 4316} 4317 4318String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4319{ 4320 char *s; 4321 String8 out_s8 = String8(); 4322 4323 Mutex::Autolock _l(mLock); 4324 if (initCheck() != NO_ERROR) { 4325 return out_s8; 4326 } 4327 4328 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4329 out_s8 = String8(s); 4330 free(s); 4331 return out_s8; 4332} 4333 4334void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4335 AudioSystem::OutputDescriptor desc; 4336 void *param2 = NULL; 4337 4338 switch (event) { 4339 case AudioSystem::INPUT_OPENED: 4340 case AudioSystem::INPUT_CONFIG_CHANGED: 4341 desc.channels = mChannelMask; 4342 desc.samplingRate = mSampleRate; 4343 desc.format = mFormat; 4344 desc.frameCount = mFrameCount; 4345 desc.latency = 0; 4346 param2 = &desc; 4347 break; 4348 4349 case AudioSystem::INPUT_CLOSED: 4350 default: 4351 break; 4352 } 4353 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4354} 4355 4356void AudioFlinger::RecordThread::readInputParameters() 4357{ 4358 delete mRsmpInBuffer; 4359 // mRsmpInBuffer is always assigned a new[] below 4360 delete mRsmpOutBuffer; 4361 mRsmpOutBuffer = NULL; 4362 delete mResampler; 4363 mResampler = NULL; 4364 4365 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4366 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4367 mChannelCount = (uint16_t)popcount(mChannelMask); 4368 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4369 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4370 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4371 mFrameCount = mInputBytes / mFrameSize; 4372 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4373 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4374 4375 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4376 { 4377 int channelCount; 4378 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4379 // stereo to mono post process as the resampler always outputs stereo. 4380 if (mChannelCount == 1 && mReqChannelCount == 2) { 4381 channelCount = 1; 4382 } else { 4383 channelCount = 2; 4384 } 4385 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4386 mResampler->setSampleRate(mSampleRate); 4387 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4388 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4389 4390 // optmization: if mono to mono, alter input frame count as if we were inputing 4391 // stereo samples 4392 if (mChannelCount == 1 && mReqChannelCount == 1) { 4393 mFrameCount >>= 1; 4394 } 4395 4396 } 4397 mRsmpInIndex = mFrameCount; 4398} 4399 4400unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4401{ 4402 Mutex::Autolock _l(mLock); 4403 if (initCheck() != NO_ERROR) { 4404 return 0; 4405 } 4406 4407 return mInput->stream->get_input_frames_lost(mInput->stream); 4408} 4409 4410uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4411{ 4412 Mutex::Autolock _l(mLock); 4413 uint32_t result = 0; 4414 if (getEffectChain_l(sessionId) != 0) { 4415 result = EFFECT_SESSION; 4416 } 4417 4418 for (size_t i = 0; i < mTracks.size(); ++i) { 4419 if (sessionId == mTracks[i]->sessionId()) { 4420 result |= TRACK_SESSION; 4421 break; 4422 } 4423 } 4424 4425 return result; 4426} 4427 4428KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4429{ 4430 KeyedVector<int, bool> ids; 4431 Mutex::Autolock _l(mLock); 4432 for (size_t j = 0; j < mTracks.size(); ++j) { 4433 sp<RecordThread::RecordTrack> track = mTracks[j]; 4434 int sessionId = track->sessionId(); 4435 if (ids.indexOfKey(sessionId) < 0) { 4436 ids.add(sessionId, true); 4437 } 4438 } 4439 return ids; 4440} 4441 4442AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4443{ 4444 Mutex::Autolock _l(mLock); 4445 AudioStreamIn *input = mInput; 4446 mInput = NULL; 4447 return input; 4448} 4449 4450// this method must always be called either with ThreadBase mLock held or inside the thread loop 4451audio_stream_t* AudioFlinger::RecordThread::stream() const 4452{ 4453 if (mInput == NULL) { 4454 return NULL; 4455 } 4456 return &mInput->stream->common; 4457} 4458 4459status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4460{ 4461 // only one chain per input thread 4462 if (mEffectChains.size() != 0) { 4463 return INVALID_OPERATION; 4464 } 4465 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4466 4467 chain->setInBuffer(NULL); 4468 chain->setOutBuffer(NULL); 4469 4470 checkSuspendOnAddEffectChain_l(chain); 4471 4472 mEffectChains.add(chain); 4473 4474 return NO_ERROR; 4475} 4476 4477size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4478{ 4479 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4480 ALOGW_IF(mEffectChains.size() != 1, 4481 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4482 chain.get(), mEffectChains.size(), this); 4483 if (mEffectChains.size() == 1) { 4484 mEffectChains.removeAt(0); 4485 } 4486 return 0; 4487} 4488 4489}; // namespace android 4490