Threads.cpp revision d8ea699dc8e7dac58bb32e9cdb31b0758da25817
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <cutils/compiler.h> 29#include <media/AudioParameter.h> 30#include <utils/Log.h> 31#include <utils/Trace.h> 32 33#include <private/media/AudioTrackShared.h> 34#include <hardware/audio.h> 35#include <audio_effects/effect_ns.h> 36#include <audio_effects/effect_aec.h> 37#include <audio_utils/primitives.h> 38 39// NBAIO implementations 40#include <media/nbaio/AudioStreamOutSink.h> 41#include <media/nbaio/MonoPipe.h> 42#include <media/nbaio/MonoPipeReader.h> 43#include <media/nbaio/Pipe.h> 44#include <media/nbaio/PipeReader.h> 45#include <media/nbaio/SourceAudioBufferProvider.h> 46 47#include <powermanager/PowerManager.h> 48 49#include <common_time/cc_helper.h> 50#include <common_time/local_clock.h> 51 52#include "AudioFlinger.h" 53#include "AudioMixer.h" 54#include "FastMixer.h" 55#include "ServiceUtilities.h" 56#include "SchedulingPolicyService.h" 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85// retry counts for buffer fill timeout 86// 50 * ~20msecs = 1 second 87static const int8_t kMaxTrackRetries = 50; 88static const int8_t kMaxTrackStartupRetries = 50; 89// allow less retry attempts on direct output thread. 90// direct outputs can be a scarce resource in audio hardware and should 91// be released as quickly as possible. 92static const int8_t kMaxTrackRetriesDirect = 2; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait for setParameters to complete 101static const nsecs_t kSetParametersTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108// minimum normal mix buffer size, expressed in milliseconds rather than frames 109static const uint32_t kMinNormalMixBufferSizeMs = 20; 110// maximum normal mix buffer size 111static const uint32_t kMaxNormalMixBufferSizeMs = 24; 112 113// Whether to use fast mixer 114static const enum { 115 FastMixer_Never, // never initialize or use: for debugging only 116 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 117 // normal mixer multiplier is 1 118 FastMixer_Static, // initialize if needed, then use all the time if initialized, 119 // multiplier is calculated based on min & max normal mixer buffer size 120 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 // FIXME for FastMixer_Dynamic: 123 // Supporting this option will require fixing HALs that can't handle large writes. 124 // For example, one HAL implementation returns an error from a large write, 125 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 126 // We could either fix the HAL implementations, or provide a wrapper that breaks 127 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 128} kUseFastMixer = FastMixer_Static; 129 130// Priorities for requestPriority 131static const int kPriorityAudioApp = 2; 132static const int kPriorityFastMixer = 3; 133 134// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 135// for the track. The client then sub-divides this into smaller buffers for its use. 136// Currently the client uses double-buffering by default, but doesn't tell us about that. 137// So for now we just assume that client is double-buffered. 138// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 139// N-buffering, so AudioFlinger could allocate the right amount of memory. 140// See the client's minBufCount and mNotificationFramesAct calculations for details. 141static const int kFastTrackMultiplier = 1; 142 143// ---------------------------------------------------------------------------- 144 145#ifdef ADD_BATTERY_DATA 146// To collect the amplifier usage 147static void addBatteryData(uint32_t params) { 148 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 149 if (service == NULL) { 150 // it already logged 151 return; 152 } 153 154 service->addBatteryData(params); 155} 156#endif 157 158 159// ---------------------------------------------------------------------------- 160// CPU Stats 161// ---------------------------------------------------------------------------- 162 163class CpuStats { 164public: 165 CpuStats(); 166 void sample(const String8 &title); 167#ifdef DEBUG_CPU_USAGE 168private: 169 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 170 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 171 172 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 173 174 int mCpuNum; // thread's current CPU number 175 int mCpukHz; // frequency of thread's current CPU in kHz 176#endif 177}; 178 179CpuStats::CpuStats() 180#ifdef DEBUG_CPU_USAGE 181 : mCpuNum(-1), mCpukHz(-1) 182#endif 183{ 184} 185 186void CpuStats::sample(const String8 &title) { 187#ifdef DEBUG_CPU_USAGE 188 // get current thread's delta CPU time in wall clock ns 189 double wcNs; 190 bool valid = mCpuUsage.sampleAndEnable(wcNs); 191 192 // record sample for wall clock statistics 193 if (valid) { 194 mWcStats.sample(wcNs); 195 } 196 197 // get the current CPU number 198 int cpuNum = sched_getcpu(); 199 200 // get the current CPU frequency in kHz 201 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 202 203 // check if either CPU number or frequency changed 204 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 205 mCpuNum = cpuNum; 206 mCpukHz = cpukHz; 207 // ignore sample for purposes of cycles 208 valid = false; 209 } 210 211 // if no change in CPU number or frequency, then record sample for cycle statistics 212 if (valid && mCpukHz > 0) { 213 double cycles = wcNs * cpukHz * 0.000001; 214 mHzStats.sample(cycles); 215 } 216 217 unsigned n = mWcStats.n(); 218 // mCpuUsage.elapsed() is expensive, so don't call it every loop 219 if ((n & 127) == 1) { 220 long long elapsed = mCpuUsage.elapsed(); 221 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 222 double perLoop = elapsed / (double) n; 223 double perLoop100 = perLoop * 0.01; 224 double perLoop1k = perLoop * 0.001; 225 double mean = mWcStats.mean(); 226 double stddev = mWcStats.stddev(); 227 double minimum = mWcStats.minimum(); 228 double maximum = mWcStats.maximum(); 229 double meanCycles = mHzStats.mean(); 230 double stddevCycles = mHzStats.stddev(); 231 double minCycles = mHzStats.minimum(); 232 double maxCycles = mHzStats.maximum(); 233 mCpuUsage.resetElapsed(); 234 mWcStats.reset(); 235 mHzStats.reset(); 236 ALOGD("CPU usage for %s over past %.1f secs\n" 237 " (%u mixer loops at %.1f mean ms per loop):\n" 238 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 239 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 240 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 241 title.string(), 242 elapsed * .000000001, n, perLoop * .000001, 243 mean * .001, 244 stddev * .001, 245 minimum * .001, 246 maximum * .001, 247 mean / perLoop100, 248 stddev / perLoop100, 249 minimum / perLoop100, 250 maximum / perLoop100, 251 meanCycles / perLoop1k, 252 stddevCycles / perLoop1k, 253 minCycles / perLoop1k, 254 maxCycles / perLoop1k); 255 256 } 257 } 258#endif 259}; 260 261// ---------------------------------------------------------------------------- 262// ThreadBase 263// ---------------------------------------------------------------------------- 264 265AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 266 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 267 : Thread(false /*canCallJava*/), 268 mType(type), 269 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 270 // mChannelMask 271 mChannelCount(0), 272 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 273 mParamStatus(NO_ERROR), 274 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 275 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 276 // mName will be set by concrete (non-virtual) subclass 277 mDeathRecipient(new PMDeathRecipient(this)) 278{ 279} 280 281AudioFlinger::ThreadBase::~ThreadBase() 282{ 283 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 284 for (size_t i = 0; i < mConfigEvents.size(); i++) { 285 delete mConfigEvents[i]; 286 } 287 mConfigEvents.clear(); 288 289 mParamCond.broadcast(); 290 // do not lock the mutex in destructor 291 releaseWakeLock_l(); 292 if (mPowerManager != 0) { 293 sp<IBinder> binder = mPowerManager->asBinder(); 294 binder->unlinkToDeath(mDeathRecipient); 295 } 296} 297 298void AudioFlinger::ThreadBase::exit() 299{ 300 ALOGV("ThreadBase::exit"); 301 // do any cleanup required for exit to succeed 302 preExit(); 303 { 304 // This lock prevents the following race in thread (uniprocessor for illustration): 305 // if (!exitPending()) { 306 // // context switch from here to exit() 307 // // exit() calls requestExit(), what exitPending() observes 308 // // exit() calls signal(), which is dropped since no waiters 309 // // context switch back from exit() to here 310 // mWaitWorkCV.wait(...); 311 // // now thread is hung 312 // } 313 AutoMutex lock(mLock); 314 requestExit(); 315 mWaitWorkCV.broadcast(); 316 } 317 // When Thread::requestExitAndWait is made virtual and this method is renamed to 318 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 319 requestExitAndWait(); 320} 321 322status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 323{ 324 status_t status; 325 326 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 327 Mutex::Autolock _l(mLock); 328 329 mNewParameters.add(keyValuePairs); 330 mWaitWorkCV.signal(); 331 // wait condition with timeout in case the thread loop has exited 332 // before the request could be processed 333 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 334 status = mParamStatus; 335 mWaitWorkCV.signal(); 336 } else { 337 status = TIMED_OUT; 338 } 339 return status; 340} 341 342void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 343{ 344 Mutex::Autolock _l(mLock); 345 sendIoConfigEvent_l(event, param); 346} 347 348// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 349void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 350{ 351 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 352 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 353 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 354 param); 355 mWaitWorkCV.signal(); 356} 357 358// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 359void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 360{ 361 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 362 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 363 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 364 mConfigEvents.size(), pid, tid, prio); 365 mWaitWorkCV.signal(); 366} 367 368void AudioFlinger::ThreadBase::processConfigEvents() 369{ 370 mLock.lock(); 371 while (!mConfigEvents.isEmpty()) { 372 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 373 ConfigEvent *event = mConfigEvents[0]; 374 mConfigEvents.removeAt(0); 375 // release mLock before locking AudioFlinger mLock: lock order is always 376 // AudioFlinger then ThreadBase to avoid cross deadlock 377 mLock.unlock(); 378 switch(event->type()) { 379 case CFG_EVENT_PRIO: { 380 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 381 // FIXME Need to understand why this has be done asynchronously 382 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 383 true /*asynchronous*/); 384 if (err != 0) { 385 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 386 "error %d", 387 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 388 } 389 } break; 390 case CFG_EVENT_IO: { 391 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 392 mAudioFlinger->mLock.lock(); 393 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 394 mAudioFlinger->mLock.unlock(); 395 } break; 396 default: 397 ALOGE("processConfigEvents() unknown event type %d", event->type()); 398 break; 399 } 400 delete event; 401 mLock.lock(); 402 } 403 mLock.unlock(); 404} 405 406void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 407{ 408 const size_t SIZE = 256; 409 char buffer[SIZE]; 410 String8 result; 411 412 bool locked = AudioFlinger::dumpTryLock(mLock); 413 if (!locked) { 414 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 415 write(fd, buffer, strlen(buffer)); 416 } 417 418 snprintf(buffer, SIZE, "io handle: %d\n", mId); 419 result.append(buffer); 420 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 437 result.append(buffer); 438 439 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 440 result.append(buffer); 441 result.append(" Index Command"); 442 for (size_t i = 0; i < mNewParameters.size(); ++i) { 443 snprintf(buffer, SIZE, "\n %02d ", i); 444 result.append(buffer); 445 result.append(mNewParameters[i]); 446 } 447 448 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 449 result.append(buffer); 450 for (size_t i = 0; i < mConfigEvents.size(); i++) { 451 mConfigEvents[i]->dump(buffer, SIZE); 452 result.append(buffer); 453 } 454 result.append("\n"); 455 456 write(fd, result.string(), result.size()); 457 458 if (locked) { 459 mLock.unlock(); 460 } 461} 462 463void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 464{ 465 const size_t SIZE = 256; 466 char buffer[SIZE]; 467 String8 result; 468 469 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 470 write(fd, buffer, strlen(buffer)); 471 472 for (size_t i = 0; i < mEffectChains.size(); ++i) { 473 sp<EffectChain> chain = mEffectChains[i]; 474 if (chain != 0) { 475 chain->dump(fd, args); 476 } 477 } 478} 479 480void AudioFlinger::ThreadBase::acquireWakeLock() 481{ 482 Mutex::Autolock _l(mLock); 483 acquireWakeLock_l(); 484} 485 486void AudioFlinger::ThreadBase::acquireWakeLock_l() 487{ 488 if (mPowerManager == 0) { 489 // use checkService() to avoid blocking if power service is not up yet 490 sp<IBinder> binder = 491 defaultServiceManager()->checkService(String16("power")); 492 if (binder == 0) { 493 ALOGW("Thread %s cannot connect to the power manager service", mName); 494 } else { 495 mPowerManager = interface_cast<IPowerManager>(binder); 496 binder->linkToDeath(mDeathRecipient); 497 } 498 } 499 if (mPowerManager != 0) { 500 sp<IBinder> binder = new BBinder(); 501 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 502 binder, 503 String16(mName), 504 String16("media")); 505 if (status == NO_ERROR) { 506 mWakeLockToken = binder; 507 } 508 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 509 } 510} 511 512void AudioFlinger::ThreadBase::releaseWakeLock() 513{ 514 Mutex::Autolock _l(mLock); 515 releaseWakeLock_l(); 516} 517 518void AudioFlinger::ThreadBase::releaseWakeLock_l() 519{ 520 if (mWakeLockToken != 0) { 521 ALOGV("releaseWakeLock_l() %s", mName); 522 if (mPowerManager != 0) { 523 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 524 } 525 mWakeLockToken.clear(); 526 } 527} 528 529void AudioFlinger::ThreadBase::clearPowerManager() 530{ 531 Mutex::Autolock _l(mLock); 532 releaseWakeLock_l(); 533 mPowerManager.clear(); 534} 535 536void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 537{ 538 sp<ThreadBase> thread = mThread.promote(); 539 if (thread != 0) { 540 thread->clearPowerManager(); 541 } 542 ALOGW("power manager service died !!!"); 543} 544 545void AudioFlinger::ThreadBase::setEffectSuspended( 546 const effect_uuid_t *type, bool suspend, int sessionId) 547{ 548 Mutex::Autolock _l(mLock); 549 setEffectSuspended_l(type, suspend, sessionId); 550} 551 552void AudioFlinger::ThreadBase::setEffectSuspended_l( 553 const effect_uuid_t *type, bool suspend, int sessionId) 554{ 555 sp<EffectChain> chain = getEffectChain_l(sessionId); 556 if (chain != 0) { 557 if (type != NULL) { 558 chain->setEffectSuspended_l(type, suspend); 559 } else { 560 chain->setEffectSuspendedAll_l(suspend); 561 } 562 } 563 564 updateSuspendedSessions_l(type, suspend, sessionId); 565} 566 567void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 568{ 569 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 570 if (index < 0) { 571 return; 572 } 573 574 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 575 mSuspendedSessions.valueAt(index); 576 577 for (size_t i = 0; i < sessionEffects.size(); i++) { 578 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 579 for (int j = 0; j < desc->mRefCount; j++) { 580 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 581 chain->setEffectSuspendedAll_l(true); 582 } else { 583 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 584 desc->mType.timeLow); 585 chain->setEffectSuspended_l(&desc->mType, true); 586 } 587 } 588 } 589} 590 591void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 592 bool suspend, 593 int sessionId) 594{ 595 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 596 597 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 598 599 if (suspend) { 600 if (index >= 0) { 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } else { 603 mSuspendedSessions.add(sessionId, sessionEffects); 604 } 605 } else { 606 if (index < 0) { 607 return; 608 } 609 sessionEffects = mSuspendedSessions.valueAt(index); 610 } 611 612 613 int key = EffectChain::kKeyForSuspendAll; 614 if (type != NULL) { 615 key = type->timeLow; 616 } 617 index = sessionEffects.indexOfKey(key); 618 619 sp<SuspendedSessionDesc> desc; 620 if (suspend) { 621 if (index >= 0) { 622 desc = sessionEffects.valueAt(index); 623 } else { 624 desc = new SuspendedSessionDesc(); 625 if (type != NULL) { 626 desc->mType = *type; 627 } 628 sessionEffects.add(key, desc); 629 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 630 } 631 desc->mRefCount++; 632 } else { 633 if (index < 0) { 634 return; 635 } 636 desc = sessionEffects.valueAt(index); 637 if (--desc->mRefCount == 0) { 638 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 639 sessionEffects.removeItemsAt(index); 640 if (sessionEffects.isEmpty()) { 641 ALOGV("updateSuspendedSessions_l() restore removing session %d", 642 sessionId); 643 mSuspendedSessions.removeItem(sessionId); 644 } 645 } 646 } 647 if (!sessionEffects.isEmpty()) { 648 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 649 } 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 Mutex::Autolock _l(mLock); 657 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 658} 659 660void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 661 bool enabled, 662 int sessionId) 663{ 664 if (mType != RECORD) { 665 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 666 // another session. This gives the priority to well behaved effect control panels 667 // and applications not using global effects. 668 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 669 // global effects 670 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 671 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 672 } 673 } 674 675 sp<EffectChain> chain = getEffectChain_l(sessionId); 676 if (chain != 0) { 677 chain->checkSuspendOnEffectEnabled(effect, enabled); 678 } 679} 680 681// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 682sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 683 const sp<AudioFlinger::Client>& client, 684 const sp<IEffectClient>& effectClient, 685 int32_t priority, 686 int sessionId, 687 effect_descriptor_t *desc, 688 int *enabled, 689 status_t *status 690 ) 691{ 692 sp<EffectModule> effect; 693 sp<EffectHandle> handle; 694 status_t lStatus; 695 sp<EffectChain> chain; 696 bool chainCreated = false; 697 bool effectCreated = false; 698 bool effectRegistered = false; 699 700 lStatus = initCheck(); 701 if (lStatus != NO_ERROR) { 702 ALOGW("createEffect_l() Audio driver not initialized."); 703 goto Exit; 704 } 705 706 // Do not allow effects with session ID 0 on direct output or duplicating threads 707 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 708 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 709 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 710 desc->name, sessionId); 711 lStatus = BAD_VALUE; 712 goto Exit; 713 } 714 // Only Pre processor effects are allowed on input threads and only on input threads 715 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 716 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 717 desc->name, desc->flags, mType); 718 lStatus = BAD_VALUE; 719 goto Exit; 720 } 721 722 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 723 724 { // scope for mLock 725 Mutex::Autolock _l(mLock); 726 727 // check for existing effect chain with the requested audio session 728 chain = getEffectChain_l(sessionId); 729 if (chain == 0) { 730 // create a new chain for this session 731 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 732 chain = new EffectChain(this, sessionId); 733 addEffectChain_l(chain); 734 chain->setStrategy(getStrategyForSession_l(sessionId)); 735 chainCreated = true; 736 } else { 737 effect = chain->getEffectFromDesc_l(desc); 738 } 739 740 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 741 742 if (effect == 0) { 743 int id = mAudioFlinger->nextUniqueId(); 744 // Check CPU and memory usage 745 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 746 if (lStatus != NO_ERROR) { 747 goto Exit; 748 } 749 effectRegistered = true; 750 // create a new effect module if none present in the chain 751 effect = new EffectModule(this, chain, desc, id, sessionId); 752 lStatus = effect->status(); 753 if (lStatus != NO_ERROR) { 754 goto Exit; 755 } 756 lStatus = chain->addEffect_l(effect); 757 if (lStatus != NO_ERROR) { 758 goto Exit; 759 } 760 effectCreated = true; 761 762 effect->setDevice(mOutDevice); 763 effect->setDevice(mInDevice); 764 effect->setMode(mAudioFlinger->getMode()); 765 effect->setAudioSource(mAudioSource); 766 } 767 // create effect handle and connect it to effect module 768 handle = new EffectHandle(effect, client, effectClient, priority); 769 lStatus = effect->addHandle(handle.get()); 770 if (enabled != NULL) { 771 *enabled = (int)effect->isEnabled(); 772 } 773 } 774 775Exit: 776 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 777 Mutex::Autolock _l(mLock); 778 if (effectCreated) { 779 chain->removeEffect_l(effect); 780 } 781 if (effectRegistered) { 782 AudioSystem::unregisterEffect(effect->id()); 783 } 784 if (chainCreated) { 785 removeEffectChain_l(chain); 786 } 787 handle.clear(); 788 } 789 790 if (status != NULL) { 791 *status = lStatus; 792 } 793 return handle; 794} 795 796sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 797{ 798 Mutex::Autolock _l(mLock); 799 return getEffect_l(sessionId, effectId); 800} 801 802sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 803{ 804 sp<EffectChain> chain = getEffectChain_l(sessionId); 805 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 806} 807 808// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 809// PlaybackThread::mLock held 810status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 811{ 812 // check for existing effect chain with the requested audio session 813 int sessionId = effect->sessionId(); 814 sp<EffectChain> chain = getEffectChain_l(sessionId); 815 bool chainCreated = false; 816 817 if (chain == 0) { 818 // create a new chain for this session 819 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 820 chain = new EffectChain(this, sessionId); 821 addEffectChain_l(chain); 822 chain->setStrategy(getStrategyForSession_l(sessionId)); 823 chainCreated = true; 824 } 825 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 826 827 if (chain->getEffectFromId_l(effect->id()) != 0) { 828 ALOGW("addEffect_l() %p effect %s already present in chain %p", 829 this, effect->desc().name, chain.get()); 830 return BAD_VALUE; 831 } 832 833 status_t status = chain->addEffect_l(effect); 834 if (status != NO_ERROR) { 835 if (chainCreated) { 836 removeEffectChain_l(chain); 837 } 838 return status; 839 } 840 841 effect->setDevice(mOutDevice); 842 effect->setDevice(mInDevice); 843 effect->setMode(mAudioFlinger->getMode()); 844 effect->setAudioSource(mAudioSource); 845 return NO_ERROR; 846} 847 848void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 849 850 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 851 effect_descriptor_t desc = effect->desc(); 852 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 853 detachAuxEffect_l(effect->id()); 854 } 855 856 sp<EffectChain> chain = effect->chain().promote(); 857 if (chain != 0) { 858 // remove effect chain if removing last effect 859 if (chain->removeEffect_l(effect) == 0) { 860 removeEffectChain_l(chain); 861 } 862 } else { 863 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 864 } 865} 866 867void AudioFlinger::ThreadBase::lockEffectChains_l( 868 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 869{ 870 effectChains = mEffectChains; 871 for (size_t i = 0; i < mEffectChains.size(); i++) { 872 mEffectChains[i]->lock(); 873 } 874} 875 876void AudioFlinger::ThreadBase::unlockEffectChains( 877 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 878{ 879 for (size_t i = 0; i < effectChains.size(); i++) { 880 effectChains[i]->unlock(); 881 } 882} 883 884sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 885{ 886 Mutex::Autolock _l(mLock); 887 return getEffectChain_l(sessionId); 888} 889 890sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 891{ 892 size_t size = mEffectChains.size(); 893 for (size_t i = 0; i < size; i++) { 894 if (mEffectChains[i]->sessionId() == sessionId) { 895 return mEffectChains[i]; 896 } 897 } 898 return 0; 899} 900 901void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 902{ 903 Mutex::Autolock _l(mLock); 904 size_t size = mEffectChains.size(); 905 for (size_t i = 0; i < size; i++) { 906 mEffectChains[i]->setMode_l(mode); 907 } 908} 909 910void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 911 EffectHandle *handle, 912 bool unpinIfLast) { 913 914 Mutex::Autolock _l(mLock); 915 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 916 // delete the effect module if removing last handle on it 917 if (effect->removeHandle(handle) == 0) { 918 if (!effect->isPinned() || unpinIfLast) { 919 removeEffect_l(effect); 920 AudioSystem::unregisterEffect(effect->id()); 921 } 922 } 923} 924 925// ---------------------------------------------------------------------------- 926// Playback 927// ---------------------------------------------------------------------------- 928 929AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 930 AudioStreamOut* output, 931 audio_io_handle_t id, 932 audio_devices_t device, 933 type_t type) 934 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 935 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 936 // mStreamTypes[] initialized in constructor body 937 mOutput(output), 938 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 939 mMixerStatus(MIXER_IDLE), 940 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 941 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 942 mScreenState(AudioFlinger::mScreenState), 943 // index 0 is reserved for normal mixer's submix 944 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 945{ 946 snprintf(mName, kNameLength, "AudioOut_%X", id); 947 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 948 949 // Assumes constructor is called by AudioFlinger with it's mLock held, but 950 // it would be safer to explicitly pass initial masterVolume/masterMute as 951 // parameter. 952 // 953 // If the HAL we are using has support for master volume or master mute, 954 // then do not attenuate or mute during mixing (just leave the volume at 1.0 955 // and the mute set to false). 956 mMasterVolume = audioFlinger->masterVolume_l(); 957 mMasterMute = audioFlinger->masterMute_l(); 958 if (mOutput && mOutput->audioHwDev) { 959 if (mOutput->audioHwDev->canSetMasterVolume()) { 960 mMasterVolume = 1.0; 961 } 962 963 if (mOutput->audioHwDev->canSetMasterMute()) { 964 mMasterMute = false; 965 } 966 } 967 968 readOutputParameters(); 969 970 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 971 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 972 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 973 stream = (audio_stream_type_t) (stream + 1)) { 974 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 975 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 976 } 977 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 978 // because mAudioFlinger doesn't have one to copy from 979} 980 981AudioFlinger::PlaybackThread::~PlaybackThread() 982{ 983 mAudioFlinger->unregisterWriter(mNBLogWriter); 984 delete [] mMixBuffer; 985} 986 987void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 988{ 989 dumpInternals(fd, args); 990 dumpTracks(fd, args); 991 dumpEffectChains(fd, args); 992} 993 994void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 995{ 996 const size_t SIZE = 256; 997 char buffer[SIZE]; 998 String8 result; 999 1000 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1001 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1002 const stream_type_t *st = &mStreamTypes[i]; 1003 if (i > 0) { 1004 result.appendFormat(", "); 1005 } 1006 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1007 if (st->mute) { 1008 result.append("M"); 1009 } 1010 } 1011 result.append("\n"); 1012 write(fd, result.string(), result.length()); 1013 result.clear(); 1014 1015 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1016 result.append(buffer); 1017 Track::appendDumpHeader(result); 1018 for (size_t i = 0; i < mTracks.size(); ++i) { 1019 sp<Track> track = mTracks[i]; 1020 if (track != 0) { 1021 track->dump(buffer, SIZE); 1022 result.append(buffer); 1023 } 1024 } 1025 1026 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1027 result.append(buffer); 1028 Track::appendDumpHeader(result); 1029 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1030 sp<Track> track = mActiveTracks[i].promote(); 1031 if (track != 0) { 1032 track->dump(buffer, SIZE); 1033 result.append(buffer); 1034 } 1035 } 1036 write(fd, result.string(), result.size()); 1037 1038 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1039 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1040 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1041 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1042} 1043 1044void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1045{ 1046 const size_t SIZE = 256; 1047 char buffer[SIZE]; 1048 String8 result; 1049 1050 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1051 result.append(buffer); 1052 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1053 ns2ms(systemTime() - mLastWriteTime)); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1056 result.append(buffer); 1057 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1058 result.append(buffer); 1059 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1060 result.append(buffer); 1061 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1062 result.append(buffer); 1063 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1064 result.append(buffer); 1065 write(fd, result.string(), result.size()); 1066 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1067 1068 dumpBase(fd, args); 1069} 1070 1071// Thread virtuals 1072status_t AudioFlinger::PlaybackThread::readyToRun() 1073{ 1074 status_t status = initCheck(); 1075 if (status == NO_ERROR) { 1076 ALOGI("AudioFlinger's thread %p ready to run", this); 1077 } else { 1078 ALOGE("No working audio driver found."); 1079 } 1080 return status; 1081} 1082 1083void AudioFlinger::PlaybackThread::onFirstRef() 1084{ 1085 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1086} 1087 1088// ThreadBase virtuals 1089void AudioFlinger::PlaybackThread::preExit() 1090{ 1091 ALOGV(" preExit()"); 1092 // FIXME this is using hard-coded strings but in the future, this functionality will be 1093 // converted to use audio HAL extensions required to support tunneling 1094 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1095} 1096 1097// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1098sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1099 const sp<AudioFlinger::Client>& client, 1100 audio_stream_type_t streamType, 1101 uint32_t sampleRate, 1102 audio_format_t format, 1103 audio_channel_mask_t channelMask, 1104 size_t frameCount, 1105 const sp<IMemory>& sharedBuffer, 1106 int sessionId, 1107 IAudioFlinger::track_flags_t *flags, 1108 pid_t tid, 1109 status_t *status) 1110{ 1111 sp<Track> track; 1112 status_t lStatus; 1113 1114 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1115 1116 // client expresses a preference for FAST, but we get the final say 1117 if (*flags & IAudioFlinger::TRACK_FAST) { 1118 if ( 1119 // not timed 1120 (!isTimed) && 1121 // either of these use cases: 1122 ( 1123 // use case 1: shared buffer with any frame count 1124 ( 1125 (sharedBuffer != 0) 1126 ) || 1127 // use case 2: callback handler and frame count is default or at least as large as HAL 1128 ( 1129 (tid != -1) && 1130 ((frameCount == 0) || 1131 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1132 ) 1133 ) && 1134 // PCM data 1135 audio_is_linear_pcm(format) && 1136 // mono or stereo 1137 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1138 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1139#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1140 // hardware sample rate 1141 (sampleRate == mSampleRate) && 1142#endif 1143 // normal mixer has an associated fast mixer 1144 hasFastMixer() && 1145 // there are sufficient fast track slots available 1146 (mFastTrackAvailMask != 0) 1147 // FIXME test that MixerThread for this fast track has a capable output HAL 1148 // FIXME add a permission test also? 1149 ) { 1150 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1151 if (frameCount == 0) { 1152 frameCount = mFrameCount * kFastTrackMultiplier; 1153 } 1154 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1155 frameCount, mFrameCount); 1156 } else { 1157 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1158 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1159 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1160 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1161 audio_is_linear_pcm(format), 1162 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1163 *flags &= ~IAudioFlinger::TRACK_FAST; 1164 // For compatibility with AudioTrack calculation, buffer depth is forced 1165 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1166 // This is probably too conservative, but legacy application code may depend on it. 1167 // If you change this calculation, also review the start threshold which is related. 1168 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1169 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1170 if (minBufCount < 2) { 1171 minBufCount = 2; 1172 } 1173 size_t minFrameCount = mNormalFrameCount * minBufCount; 1174 if (frameCount < minFrameCount) { 1175 frameCount = minFrameCount; 1176 } 1177 } 1178 } 1179 1180 if (mType == DIRECT) { 1181 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1182 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1183 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1184 "for output %p with format %d", 1185 sampleRate, format, channelMask, mOutput, mFormat); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 } 1190 } else { 1191 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1192 if (sampleRate > mSampleRate*2) { 1193 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1194 lStatus = BAD_VALUE; 1195 goto Exit; 1196 } 1197 } 1198 1199 lStatus = initCheck(); 1200 if (lStatus != NO_ERROR) { 1201 ALOGE("Audio driver not initialized."); 1202 goto Exit; 1203 } 1204 1205 { // scope for mLock 1206 Mutex::Autolock _l(mLock); 1207 1208 // all tracks in same audio session must share the same routing strategy otherwise 1209 // conflicts will happen when tracks are moved from one output to another by audio policy 1210 // manager 1211 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1212 for (size_t i = 0; i < mTracks.size(); ++i) { 1213 sp<Track> t = mTracks[i]; 1214 if (t != 0 && !t->isOutputTrack()) { 1215 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1216 if (sessionId == t->sessionId() && strategy != actual) { 1217 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1218 strategy, actual); 1219 lStatus = BAD_VALUE; 1220 goto Exit; 1221 } 1222 } 1223 } 1224 1225 if (!isTimed) { 1226 track = new Track(this, client, streamType, sampleRate, format, 1227 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1228 } else { 1229 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1230 channelMask, frameCount, sharedBuffer, sessionId); 1231 } 1232 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1233 lStatus = NO_MEMORY; 1234 goto Exit; 1235 } 1236 mTracks.add(track); 1237 1238 sp<EffectChain> chain = getEffectChain_l(sessionId); 1239 if (chain != 0) { 1240 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1241 track->setMainBuffer(chain->inBuffer()); 1242 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1243 chain->incTrackCnt(); 1244 } 1245 1246 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1247 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1248 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1249 // so ask activity manager to do this on our behalf 1250 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1251 } 1252 } 1253 1254 lStatus = NO_ERROR; 1255 1256Exit: 1257 if (status) { 1258 *status = lStatus; 1259 } 1260 return track; 1261} 1262 1263uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1264{ 1265 return latency; 1266} 1267 1268uint32_t AudioFlinger::PlaybackThread::latency() const 1269{ 1270 Mutex::Autolock _l(mLock); 1271 return latency_l(); 1272} 1273uint32_t AudioFlinger::PlaybackThread::latency_l() const 1274{ 1275 if (initCheck() == NO_ERROR) { 1276 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1277 } else { 1278 return 0; 1279 } 1280} 1281 1282void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1283{ 1284 Mutex::Autolock _l(mLock); 1285 // Don't apply master volume in SW if our HAL can do it for us. 1286 if (mOutput && mOutput->audioHwDev && 1287 mOutput->audioHwDev->canSetMasterVolume()) { 1288 mMasterVolume = 1.0; 1289 } else { 1290 mMasterVolume = value; 1291 } 1292} 1293 1294void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1295{ 1296 Mutex::Autolock _l(mLock); 1297 // Don't apply master mute in SW if our HAL can do it for us. 1298 if (mOutput && mOutput->audioHwDev && 1299 mOutput->audioHwDev->canSetMasterMute()) { 1300 mMasterMute = false; 1301 } else { 1302 mMasterMute = muted; 1303 } 1304} 1305 1306void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1307{ 1308 Mutex::Autolock _l(mLock); 1309 mStreamTypes[stream].volume = value; 1310} 1311 1312void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1313{ 1314 Mutex::Autolock _l(mLock); 1315 mStreamTypes[stream].mute = muted; 1316} 1317 1318float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1319{ 1320 Mutex::Autolock _l(mLock); 1321 return mStreamTypes[stream].volume; 1322} 1323 1324// addTrack_l() must be called with ThreadBase::mLock held 1325status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1326{ 1327 status_t status = ALREADY_EXISTS; 1328 1329 // set retry count for buffer fill 1330 track->mRetryCount = kMaxTrackStartupRetries; 1331 if (mActiveTracks.indexOf(track) < 0) { 1332 // the track is newly added, make sure it fills up all its 1333 // buffers before playing. This is to ensure the client will 1334 // effectively get the latency it requested. 1335 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1336 track->mResetDone = false; 1337 track->mPresentationCompleteFrames = 0; 1338 mActiveTracks.add(track); 1339 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1340 if (chain != 0) { 1341 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1342 track->sessionId()); 1343 chain->incActiveTrackCnt(); 1344 } 1345 1346 status = NO_ERROR; 1347 } 1348 1349 ALOGV("mWaitWorkCV.broadcast"); 1350 mWaitWorkCV.broadcast(); 1351 1352 return status; 1353} 1354 1355// destroyTrack_l() must be called with ThreadBase::mLock held 1356void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1357{ 1358 track->mState = TrackBase::TERMINATED; 1359 // active tracks are removed by threadLoop() 1360 if (mActiveTracks.indexOf(track) < 0) { 1361 removeTrack_l(track); 1362 } 1363} 1364 1365void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1366{ 1367 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1368 mTracks.remove(track); 1369 deleteTrackName_l(track->name()); 1370 // redundant as track is about to be destroyed, for dumpsys only 1371 track->mName = -1; 1372 if (track->isFastTrack()) { 1373 int index = track->mFastIndex; 1374 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1375 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1376 mFastTrackAvailMask |= 1 << index; 1377 // redundant as track is about to be destroyed, for dumpsys only 1378 track->mFastIndex = -1; 1379 } 1380 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1381 if (chain != 0) { 1382 chain->decTrackCnt(); 1383 } 1384} 1385 1386String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1387{ 1388 Mutex::Autolock _l(mLock); 1389 if (initCheck() != NO_ERROR) { 1390 return String8(); 1391 } 1392 1393 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1394 const String8 out_s8(s); 1395 free(s); 1396 return out_s8; 1397} 1398 1399// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1400void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1401 AudioSystem::OutputDescriptor desc; 1402 void *param2 = NULL; 1403 1404 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1405 param); 1406 1407 switch (event) { 1408 case AudioSystem::OUTPUT_OPENED: 1409 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1410 desc.channels = mChannelMask; 1411 desc.samplingRate = mSampleRate; 1412 desc.format = mFormat; 1413 desc.frameCount = mNormalFrameCount; // FIXME see 1414 // AudioFlinger::frameCount(audio_io_handle_t) 1415 desc.latency = latency(); 1416 param2 = &desc; 1417 break; 1418 1419 case AudioSystem::STREAM_CONFIG_CHANGED: 1420 param2 = ¶m; 1421 case AudioSystem::OUTPUT_CLOSED: 1422 default: 1423 break; 1424 } 1425 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1426} 1427 1428void AudioFlinger::PlaybackThread::readOutputParameters() 1429{ 1430 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1431 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1432 mChannelCount = (uint16_t)popcount(mChannelMask); 1433 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1434 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1435 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1436 if (mFrameCount & 15) { 1437 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1438 mFrameCount); 1439 } 1440 1441 // Calculate size of normal mix buffer relative to the HAL output buffer size 1442 double multiplier = 1.0; 1443 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1444 kUseFastMixer == FastMixer_Dynamic)) { 1445 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1446 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1447 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1448 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1449 maxNormalFrameCount = maxNormalFrameCount & ~15; 1450 if (maxNormalFrameCount < minNormalFrameCount) { 1451 maxNormalFrameCount = minNormalFrameCount; 1452 } 1453 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1454 if (multiplier <= 1.0) { 1455 multiplier = 1.0; 1456 } else if (multiplier <= 2.0) { 1457 if (2 * mFrameCount <= maxNormalFrameCount) { 1458 multiplier = 2.0; 1459 } else { 1460 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1461 } 1462 } else { 1463 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1464 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1465 // track, but we sometimes have to do this to satisfy the maximum frame count 1466 // constraint) 1467 // FIXME this rounding up should not be done if no HAL SRC 1468 uint32_t truncMult = (uint32_t) multiplier; 1469 if ((truncMult & 1)) { 1470 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1471 ++truncMult; 1472 } 1473 } 1474 multiplier = (double) truncMult; 1475 } 1476 } 1477 mNormalFrameCount = multiplier * mFrameCount; 1478 // round up to nearest 16 frames to satisfy AudioMixer 1479 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1480 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1481 mNormalFrameCount); 1482 1483 delete[] mMixBuffer; 1484 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1485 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1486 1487 // force reconfiguration of effect chains and engines to take new buffer size and audio 1488 // parameters into account 1489 // Note that mLock is not held when readOutputParameters() is called from the constructor 1490 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1491 // matter. 1492 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1493 Vector< sp<EffectChain> > effectChains = mEffectChains; 1494 for (size_t i = 0; i < effectChains.size(); i ++) { 1495 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1496 } 1497} 1498 1499 1500status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1501{ 1502 if (halFrames == NULL || dspFrames == NULL) { 1503 return BAD_VALUE; 1504 } 1505 Mutex::Autolock _l(mLock); 1506 if (initCheck() != NO_ERROR) { 1507 return INVALID_OPERATION; 1508 } 1509 size_t framesWritten = mBytesWritten / mFrameSize; 1510 *halFrames = framesWritten; 1511 1512 if (isSuspended()) { 1513 // return an estimation of rendered frames when the output is suspended 1514 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1515 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1516 return NO_ERROR; 1517 } else { 1518 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1519 } 1520} 1521 1522uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1523{ 1524 Mutex::Autolock _l(mLock); 1525 uint32_t result = 0; 1526 if (getEffectChain_l(sessionId) != 0) { 1527 result = EFFECT_SESSION; 1528 } 1529 1530 for (size_t i = 0; i < mTracks.size(); ++i) { 1531 sp<Track> track = mTracks[i]; 1532 if (sessionId == track->sessionId() && !track->isInvalid()) { 1533 result |= TRACK_SESSION; 1534 break; 1535 } 1536 } 1537 1538 return result; 1539} 1540 1541uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1542{ 1543 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1544 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1545 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1546 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1547 } 1548 for (size_t i = 0; i < mTracks.size(); i++) { 1549 sp<Track> track = mTracks[i]; 1550 if (sessionId == track->sessionId() && !track->isInvalid()) { 1551 return AudioSystem::getStrategyForStream(track->streamType()); 1552 } 1553 } 1554 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1555} 1556 1557 1558AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1559{ 1560 Mutex::Autolock _l(mLock); 1561 return mOutput; 1562} 1563 1564AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1565{ 1566 Mutex::Autolock _l(mLock); 1567 AudioStreamOut *output = mOutput; 1568 mOutput = NULL; 1569 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1570 // must push a NULL and wait for ack 1571 mOutputSink.clear(); 1572 mPipeSink.clear(); 1573 mNormalSink.clear(); 1574 return output; 1575} 1576 1577// this method must always be called either with ThreadBase mLock held or inside the thread loop 1578audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1579{ 1580 if (mOutput == NULL) { 1581 return NULL; 1582 } 1583 return &mOutput->stream->common; 1584} 1585 1586uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1587{ 1588 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1589} 1590 1591status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1592{ 1593 if (!isValidSyncEvent(event)) { 1594 return BAD_VALUE; 1595 } 1596 1597 Mutex::Autolock _l(mLock); 1598 1599 for (size_t i = 0; i < mTracks.size(); ++i) { 1600 sp<Track> track = mTracks[i]; 1601 if (event->triggerSession() == track->sessionId()) { 1602 (void) track->setSyncEvent(event); 1603 return NO_ERROR; 1604 } 1605 } 1606 1607 return NAME_NOT_FOUND; 1608} 1609 1610bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1611{ 1612 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1613} 1614 1615void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1616 const Vector< sp<Track> >& tracksToRemove) 1617{ 1618 size_t count = tracksToRemove.size(); 1619 if (CC_UNLIKELY(count)) { 1620 for (size_t i = 0 ; i < count ; i++) { 1621 const sp<Track>& track = tracksToRemove.itemAt(i); 1622 if ((track->sharedBuffer() != 0) && 1623 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1624 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1625 } 1626 } 1627 } 1628 1629} 1630 1631void AudioFlinger::PlaybackThread::checkSilentMode_l() 1632{ 1633 if (!mMasterMute) { 1634 char value[PROPERTY_VALUE_MAX]; 1635 if (property_get("ro.audio.silent", value, "0") > 0) { 1636 char *endptr; 1637 unsigned long ul = strtoul(value, &endptr, 0); 1638 if (*endptr == '\0' && ul != 0) { 1639 ALOGD("Silence is golden"); 1640 // The setprop command will not allow a property to be changed after 1641 // the first time it is set, so we don't have to worry about un-muting. 1642 setMasterMute_l(true); 1643 } 1644 } 1645 } 1646} 1647 1648// shared by MIXER and DIRECT, overridden by DUPLICATING 1649void AudioFlinger::PlaybackThread::threadLoop_write() 1650{ 1651 // FIXME rewrite to reduce number of system calls 1652 mLastWriteTime = systemTime(); 1653 mInWrite = true; 1654 int bytesWritten; 1655 1656 // If an NBAIO sink is present, use it to write the normal mixer's submix 1657 if (mNormalSink != 0) { 1658#define mBitShift 2 // FIXME 1659 size_t count = mixBufferSize >> mBitShift; 1660 ATRACE_BEGIN("write"); 1661 // update the setpoint when AudioFlinger::mScreenState changes 1662 uint32_t screenState = AudioFlinger::mScreenState; 1663 if (screenState != mScreenState) { 1664 mScreenState = screenState; 1665 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1666 if (pipe != NULL) { 1667 pipe->setAvgFrames((mScreenState & 1) ? 1668 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1669 } 1670 } 1671 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1672 ATRACE_END(); 1673 if (framesWritten > 0) { 1674 bytesWritten = framesWritten << mBitShift; 1675 } else { 1676 bytesWritten = framesWritten; 1677 } 1678 // otherwise use the HAL / AudioStreamOut directly 1679 } else { 1680 // Direct output thread. 1681 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1682 } 1683 1684 if (bytesWritten > 0) { 1685 mBytesWritten += mixBufferSize; 1686 } 1687 mNumWrites++; 1688 mInWrite = false; 1689} 1690 1691/* 1692The derived values that are cached: 1693 - mixBufferSize from frame count * frame size 1694 - activeSleepTime from activeSleepTimeUs() 1695 - idleSleepTime from idleSleepTimeUs() 1696 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1697 - maxPeriod from frame count and sample rate (MIXER only) 1698 1699The parameters that affect these derived values are: 1700 - frame count 1701 - frame size 1702 - sample rate 1703 - device type: A2DP or not 1704 - device latency 1705 - format: PCM or not 1706 - active sleep time 1707 - idle sleep time 1708*/ 1709 1710void AudioFlinger::PlaybackThread::cacheParameters_l() 1711{ 1712 mixBufferSize = mNormalFrameCount * mFrameSize; 1713 activeSleepTime = activeSleepTimeUs(); 1714 idleSleepTime = idleSleepTimeUs(); 1715} 1716 1717void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1718{ 1719 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1720 this, streamType, mTracks.size()); 1721 Mutex::Autolock _l(mLock); 1722 1723 size_t size = mTracks.size(); 1724 for (size_t i = 0; i < size; i++) { 1725 sp<Track> t = mTracks[i]; 1726 if (t->streamType() == streamType) { 1727 t->invalidate(); 1728 } 1729 } 1730} 1731 1732status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1733{ 1734 int session = chain->sessionId(); 1735 int16_t *buffer = mMixBuffer; 1736 bool ownsBuffer = false; 1737 1738 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1739 if (session > 0) { 1740 // Only one effect chain can be present in direct output thread and it uses 1741 // the mix buffer as input 1742 if (mType != DIRECT) { 1743 size_t numSamples = mNormalFrameCount * mChannelCount; 1744 buffer = new int16_t[numSamples]; 1745 memset(buffer, 0, numSamples * sizeof(int16_t)); 1746 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1747 ownsBuffer = true; 1748 } 1749 1750 // Attach all tracks with same session ID to this chain. 1751 for (size_t i = 0; i < mTracks.size(); ++i) { 1752 sp<Track> track = mTracks[i]; 1753 if (session == track->sessionId()) { 1754 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1755 buffer); 1756 track->setMainBuffer(buffer); 1757 chain->incTrackCnt(); 1758 } 1759 } 1760 1761 // indicate all active tracks in the chain 1762 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1763 sp<Track> track = mActiveTracks[i].promote(); 1764 if (track == 0) { 1765 continue; 1766 } 1767 if (session == track->sessionId()) { 1768 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1769 chain->incActiveTrackCnt(); 1770 } 1771 } 1772 } 1773 1774 chain->setInBuffer(buffer, ownsBuffer); 1775 chain->setOutBuffer(mMixBuffer); 1776 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1777 // chains list in order to be processed last as it contains output stage effects 1778 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1779 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1780 // after track specific effects and before output stage 1781 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1782 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1783 // Effect chain for other sessions are inserted at beginning of effect 1784 // chains list to be processed before output mix effects. Relative order between other 1785 // sessions is not important 1786 size_t size = mEffectChains.size(); 1787 size_t i = 0; 1788 for (i = 0; i < size; i++) { 1789 if (mEffectChains[i]->sessionId() < session) { 1790 break; 1791 } 1792 } 1793 mEffectChains.insertAt(chain, i); 1794 checkSuspendOnAddEffectChain_l(chain); 1795 1796 return NO_ERROR; 1797} 1798 1799size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1800{ 1801 int session = chain->sessionId(); 1802 1803 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1804 1805 for (size_t i = 0; i < mEffectChains.size(); i++) { 1806 if (chain == mEffectChains[i]) { 1807 mEffectChains.removeAt(i); 1808 // detach all active tracks from the chain 1809 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1810 sp<Track> track = mActiveTracks[i].promote(); 1811 if (track == 0) { 1812 continue; 1813 } 1814 if (session == track->sessionId()) { 1815 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1816 chain.get(), session); 1817 chain->decActiveTrackCnt(); 1818 } 1819 } 1820 1821 // detach all tracks with same session ID from this chain 1822 for (size_t i = 0; i < mTracks.size(); ++i) { 1823 sp<Track> track = mTracks[i]; 1824 if (session == track->sessionId()) { 1825 track->setMainBuffer(mMixBuffer); 1826 chain->decTrackCnt(); 1827 } 1828 } 1829 break; 1830 } 1831 } 1832 return mEffectChains.size(); 1833} 1834 1835status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1836 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1837{ 1838 Mutex::Autolock _l(mLock); 1839 return attachAuxEffect_l(track, EffectId); 1840} 1841 1842status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1843 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1844{ 1845 status_t status = NO_ERROR; 1846 1847 if (EffectId == 0) { 1848 track->setAuxBuffer(0, NULL); 1849 } else { 1850 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1851 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1852 if (effect != 0) { 1853 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1854 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1855 } else { 1856 status = INVALID_OPERATION; 1857 } 1858 } else { 1859 status = BAD_VALUE; 1860 } 1861 } 1862 return status; 1863} 1864 1865void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1866{ 1867 for (size_t i = 0; i < mTracks.size(); ++i) { 1868 sp<Track> track = mTracks[i]; 1869 if (track->auxEffectId() == effectId) { 1870 attachAuxEffect_l(track, 0); 1871 } 1872 } 1873} 1874 1875bool AudioFlinger::PlaybackThread::threadLoop() 1876{ 1877 Vector< sp<Track> > tracksToRemove; 1878 1879 standbyTime = systemTime(); 1880 1881 // MIXER 1882 nsecs_t lastWarning = 0; 1883 1884 // DUPLICATING 1885 // FIXME could this be made local to while loop? 1886 writeFrames = 0; 1887 1888 cacheParameters_l(); 1889 sleepTime = idleSleepTime; 1890 1891 if (mType == MIXER) { 1892 sleepTimeShift = 0; 1893 } 1894 1895 CpuStats cpuStats; 1896 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1897 1898 acquireWakeLock(); 1899 1900 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1901 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1902 // and then that string will be logged at the next convenient opportunity. 1903 const char *logString = NULL; 1904 1905 while (!exitPending()) 1906 { 1907 cpuStats.sample(myName); 1908 1909 Vector< sp<EffectChain> > effectChains; 1910 1911 processConfigEvents(); 1912 1913 { // scope for mLock 1914 1915 Mutex::Autolock _l(mLock); 1916 1917 if (logString != NULL) { 1918 mNBLogWriter->logTimestamp(); 1919 mNBLogWriter->log(logString); 1920 logString = NULL; 1921 } 1922 1923 if (checkForNewParameters_l()) { 1924 cacheParameters_l(); 1925 } 1926 1927 saveOutputTracks(); 1928 1929 // put audio hardware into standby after short delay 1930 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1931 isSuspended())) { 1932 if (!mStandby) { 1933 1934 threadLoop_standby(); 1935 1936 mStandby = true; 1937 } 1938 1939 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1940 // we're about to wait, flush the binder command buffer 1941 IPCThreadState::self()->flushCommands(); 1942 1943 clearOutputTracks(); 1944 1945 if (exitPending()) { 1946 break; 1947 } 1948 1949 releaseWakeLock_l(); 1950 // wait until we have something to do... 1951 ALOGV("%s going to sleep", myName.string()); 1952 mWaitWorkCV.wait(mLock); 1953 ALOGV("%s waking up", myName.string()); 1954 acquireWakeLock_l(); 1955 1956 mMixerStatus = MIXER_IDLE; 1957 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1958 mBytesWritten = 0; 1959 1960 checkSilentMode_l(); 1961 1962 standbyTime = systemTime() + standbyDelay; 1963 sleepTime = idleSleepTime; 1964 if (mType == MIXER) { 1965 sleepTimeShift = 0; 1966 } 1967 1968 continue; 1969 } 1970 } 1971 1972 // mMixerStatusIgnoringFastTracks is also updated internally 1973 mMixerStatus = prepareTracks_l(&tracksToRemove); 1974 1975 // prevent any changes in effect chain list and in each effect chain 1976 // during mixing and effect process as the audio buffers could be deleted 1977 // or modified if an effect is created or deleted 1978 lockEffectChains_l(effectChains); 1979 } 1980 1981 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1982 threadLoop_mix(); 1983 } else { 1984 threadLoop_sleepTime(); 1985 } 1986 1987 if (isSuspended()) { 1988 sleepTime = suspendSleepTimeUs(); 1989 mBytesWritten += mixBufferSize; 1990 } 1991 1992 // only process effects if we're going to write 1993 if (sleepTime == 0) { 1994 for (size_t i = 0; i < effectChains.size(); i ++) { 1995 effectChains[i]->process_l(); 1996 } 1997 } 1998 1999 // enable changes in effect chain 2000 unlockEffectChains(effectChains); 2001 2002 // sleepTime == 0 means we must write to audio hardware 2003 if (sleepTime == 0) { 2004 2005 threadLoop_write(); 2006 2007if (mType == MIXER) { 2008 // write blocked detection 2009 nsecs_t now = systemTime(); 2010 nsecs_t delta = now - mLastWriteTime; 2011 if (!mStandby && delta > maxPeriod) { 2012 mNumDelayedWrites++; 2013 if ((now - lastWarning) > kWarningThrottleNs) { 2014 ATRACE_NAME("underrun"); 2015 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2016 ns2ms(delta), mNumDelayedWrites, this); 2017 lastWarning = now; 2018 } 2019 } 2020} 2021 2022 mStandby = false; 2023 } else { 2024 usleep(sleepTime); 2025 } 2026 2027 // Finally let go of removed track(s), without the lock held 2028 // since we can't guarantee the destructors won't acquire that 2029 // same lock. This will also mutate and push a new fast mixer state. 2030 threadLoop_removeTracks(tracksToRemove); 2031 tracksToRemove.clear(); 2032 2033 // FIXME I don't understand the need for this here; 2034 // it was in the original code but maybe the 2035 // assignment in saveOutputTracks() makes this unnecessary? 2036 clearOutputTracks(); 2037 2038 // Effect chains will be actually deleted here if they were removed from 2039 // mEffectChains list during mixing or effects processing 2040 effectChains.clear(); 2041 2042 // FIXME Note that the above .clear() is no longer necessary since effectChains 2043 // is now local to this block, but will keep it for now (at least until merge done). 2044 } 2045 2046 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2047 if (mType == MIXER || mType == DIRECT) { 2048 // put output stream into standby mode 2049 if (!mStandby) { 2050 mOutput->stream->common.standby(&mOutput->stream->common); 2051 } 2052 } 2053 2054 releaseWakeLock(); 2055 2056 ALOGV("Thread %p type %d exiting", this, mType); 2057 return false; 2058} 2059 2060 2061// ---------------------------------------------------------------------------- 2062 2063AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2064 audio_io_handle_t id, audio_devices_t device, type_t type) 2065 : PlaybackThread(audioFlinger, output, id, device, type), 2066 // mAudioMixer below 2067 // mFastMixer below 2068 mFastMixerFutex(0) 2069 // mOutputSink below 2070 // mPipeSink below 2071 // mNormalSink below 2072{ 2073 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2074 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2075 "mFrameCount=%d, mNormalFrameCount=%d", 2076 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2077 mNormalFrameCount); 2078 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2079 2080 // FIXME - Current mixer implementation only supports stereo output 2081 if (mChannelCount != FCC_2) { 2082 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2083 } 2084 2085 // create an NBAIO sink for the HAL output stream, and negotiate 2086 mOutputSink = new AudioStreamOutSink(output->stream); 2087 size_t numCounterOffers = 0; 2088 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2089 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2090 ALOG_ASSERT(index == 0); 2091 2092 // initialize fast mixer depending on configuration 2093 bool initFastMixer; 2094 switch (kUseFastMixer) { 2095 case FastMixer_Never: 2096 initFastMixer = false; 2097 break; 2098 case FastMixer_Always: 2099 initFastMixer = true; 2100 break; 2101 case FastMixer_Static: 2102 case FastMixer_Dynamic: 2103 initFastMixer = mFrameCount < mNormalFrameCount; 2104 break; 2105 } 2106 if (initFastMixer) { 2107 2108 // create a MonoPipe to connect our submix to FastMixer 2109 NBAIO_Format format = mOutputSink->format(); 2110 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2111 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2112 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2113 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2114 const NBAIO_Format offers[1] = {format}; 2115 size_t numCounterOffers = 0; 2116 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2117 ALOG_ASSERT(index == 0); 2118 monoPipe->setAvgFrames((mScreenState & 1) ? 2119 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2120 mPipeSink = monoPipe; 2121 2122#ifdef TEE_SINK 2123 if (mTeeSinkOutputEnabled) { 2124 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2125 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2126 numCounterOffers = 0; 2127 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2128 ALOG_ASSERT(index == 0); 2129 mTeeSink = teeSink; 2130 PipeReader *teeSource = new PipeReader(*teeSink); 2131 numCounterOffers = 0; 2132 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2133 ALOG_ASSERT(index == 0); 2134 mTeeSource = teeSource; 2135 } 2136#endif 2137 2138 // create fast mixer and configure it initially with just one fast track for our submix 2139 mFastMixer = new FastMixer(); 2140 FastMixerStateQueue *sq = mFastMixer->sq(); 2141#ifdef STATE_QUEUE_DUMP 2142 sq->setObserverDump(&mStateQueueObserverDump); 2143 sq->setMutatorDump(&mStateQueueMutatorDump); 2144#endif 2145 FastMixerState *state = sq->begin(); 2146 FastTrack *fastTrack = &state->mFastTracks[0]; 2147 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2148 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2149 fastTrack->mVolumeProvider = NULL; 2150 fastTrack->mGeneration++; 2151 state->mFastTracksGen++; 2152 state->mTrackMask = 1; 2153 // fast mixer will use the HAL output sink 2154 state->mOutputSink = mOutputSink.get(); 2155 state->mOutputSinkGen++; 2156 state->mFrameCount = mFrameCount; 2157 state->mCommand = FastMixerState::COLD_IDLE; 2158 // already done in constructor initialization list 2159 //mFastMixerFutex = 0; 2160 state->mColdFutexAddr = &mFastMixerFutex; 2161 state->mColdGen++; 2162 state->mDumpState = &mFastMixerDumpState; 2163#ifdef TEE_SINK 2164 state->mTeeSink = mTeeSink.get(); 2165#endif 2166 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2167 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2168 sq->end(); 2169 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2170 2171 // start the fast mixer 2172 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2173 pid_t tid = mFastMixer->getTid(); 2174 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2175 if (err != 0) { 2176 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2177 kPriorityFastMixer, getpid_cached, tid, err); 2178 } 2179 2180#ifdef AUDIO_WATCHDOG 2181 // create and start the watchdog 2182 mAudioWatchdog = new AudioWatchdog(); 2183 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2184 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2185 tid = mAudioWatchdog->getTid(); 2186 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2187 if (err != 0) { 2188 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2189 kPriorityFastMixer, getpid_cached, tid, err); 2190 } 2191#endif 2192 2193 } else { 2194 mFastMixer = NULL; 2195 } 2196 2197 switch (kUseFastMixer) { 2198 case FastMixer_Never: 2199 case FastMixer_Dynamic: 2200 mNormalSink = mOutputSink; 2201 break; 2202 case FastMixer_Always: 2203 mNormalSink = mPipeSink; 2204 break; 2205 case FastMixer_Static: 2206 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2207 break; 2208 } 2209} 2210 2211AudioFlinger::MixerThread::~MixerThread() 2212{ 2213 if (mFastMixer != NULL) { 2214 FastMixerStateQueue *sq = mFastMixer->sq(); 2215 FastMixerState *state = sq->begin(); 2216 if (state->mCommand == FastMixerState::COLD_IDLE) { 2217 int32_t old = android_atomic_inc(&mFastMixerFutex); 2218 if (old == -1) { 2219 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2220 } 2221 } 2222 state->mCommand = FastMixerState::EXIT; 2223 sq->end(); 2224 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2225 mFastMixer->join(); 2226 // Though the fast mixer thread has exited, it's state queue is still valid. 2227 // We'll use that extract the final state which contains one remaining fast track 2228 // corresponding to our sub-mix. 2229 state = sq->begin(); 2230 ALOG_ASSERT(state->mTrackMask == 1); 2231 FastTrack *fastTrack = &state->mFastTracks[0]; 2232 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2233 delete fastTrack->mBufferProvider; 2234 sq->end(false /*didModify*/); 2235 delete mFastMixer; 2236#ifdef AUDIO_WATCHDOG 2237 if (mAudioWatchdog != 0) { 2238 mAudioWatchdog->requestExit(); 2239 mAudioWatchdog->requestExitAndWait(); 2240 mAudioWatchdog.clear(); 2241 } 2242#endif 2243 } 2244 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2245 delete mAudioMixer; 2246} 2247 2248 2249uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2250{ 2251 if (mFastMixer != NULL) { 2252 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2253 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2254 } 2255 return latency; 2256} 2257 2258 2259void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2260{ 2261 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2262} 2263 2264void AudioFlinger::MixerThread::threadLoop_write() 2265{ 2266 // FIXME we should only do one push per cycle; confirm this is true 2267 // Start the fast mixer if it's not already running 2268 if (mFastMixer != NULL) { 2269 FastMixerStateQueue *sq = mFastMixer->sq(); 2270 FastMixerState *state = sq->begin(); 2271 if (state->mCommand != FastMixerState::MIX_WRITE && 2272 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2273 if (state->mCommand == FastMixerState::COLD_IDLE) { 2274 int32_t old = android_atomic_inc(&mFastMixerFutex); 2275 if (old == -1) { 2276 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2277 } 2278#ifdef AUDIO_WATCHDOG 2279 if (mAudioWatchdog != 0) { 2280 mAudioWatchdog->resume(); 2281 } 2282#endif 2283 } 2284 state->mCommand = FastMixerState::MIX_WRITE; 2285 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2286 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2287 sq->end(); 2288 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2289 if (kUseFastMixer == FastMixer_Dynamic) { 2290 mNormalSink = mPipeSink; 2291 } 2292 } else { 2293 sq->end(false /*didModify*/); 2294 } 2295 } 2296 PlaybackThread::threadLoop_write(); 2297} 2298 2299void AudioFlinger::MixerThread::threadLoop_standby() 2300{ 2301 // Idle the fast mixer if it's currently running 2302 if (mFastMixer != NULL) { 2303 FastMixerStateQueue *sq = mFastMixer->sq(); 2304 FastMixerState *state = sq->begin(); 2305 if (!(state->mCommand & FastMixerState::IDLE)) { 2306 state->mCommand = FastMixerState::COLD_IDLE; 2307 state->mColdFutexAddr = &mFastMixerFutex; 2308 state->mColdGen++; 2309 mFastMixerFutex = 0; 2310 sq->end(); 2311 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2312 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2313 if (kUseFastMixer == FastMixer_Dynamic) { 2314 mNormalSink = mOutputSink; 2315 } 2316#ifdef AUDIO_WATCHDOG 2317 if (mAudioWatchdog != 0) { 2318 mAudioWatchdog->pause(); 2319 } 2320#endif 2321 } else { 2322 sq->end(false /*didModify*/); 2323 } 2324 } 2325 PlaybackThread::threadLoop_standby(); 2326} 2327 2328// shared by MIXER and DIRECT, overridden by DUPLICATING 2329void AudioFlinger::PlaybackThread::threadLoop_standby() 2330{ 2331 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2332 mOutput->stream->common.standby(&mOutput->stream->common); 2333} 2334 2335void AudioFlinger::MixerThread::threadLoop_mix() 2336{ 2337 // obtain the presentation timestamp of the next output buffer 2338 int64_t pts; 2339 status_t status = INVALID_OPERATION; 2340 2341 if (mNormalSink != 0) { 2342 status = mNormalSink->getNextWriteTimestamp(&pts); 2343 } else { 2344 status = mOutputSink->getNextWriteTimestamp(&pts); 2345 } 2346 2347 if (status != NO_ERROR) { 2348 pts = AudioBufferProvider::kInvalidPTS; 2349 } 2350 2351 // mix buffers... 2352 mAudioMixer->process(pts); 2353 // increase sleep time progressively when application underrun condition clears. 2354 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2355 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2356 // such that we would underrun the audio HAL. 2357 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2358 sleepTimeShift--; 2359 } 2360 sleepTime = 0; 2361 standbyTime = systemTime() + standbyDelay; 2362 //TODO: delay standby when effects have a tail 2363} 2364 2365void AudioFlinger::MixerThread::threadLoop_sleepTime() 2366{ 2367 // If no tracks are ready, sleep once for the duration of an output 2368 // buffer size, then write 0s to the output 2369 if (sleepTime == 0) { 2370 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2371 sleepTime = activeSleepTime >> sleepTimeShift; 2372 if (sleepTime < kMinThreadSleepTimeUs) { 2373 sleepTime = kMinThreadSleepTimeUs; 2374 } 2375 // reduce sleep time in case of consecutive application underruns to avoid 2376 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2377 // duration we would end up writing less data than needed by the audio HAL if 2378 // the condition persists. 2379 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2380 sleepTimeShift++; 2381 } 2382 } else { 2383 sleepTime = idleSleepTime; 2384 } 2385 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2386 memset (mMixBuffer, 0, mixBufferSize); 2387 sleepTime = 0; 2388 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2389 "anticipated start"); 2390 } 2391 // TODO add standby time extension fct of effect tail 2392} 2393 2394// prepareTracks_l() must be called with ThreadBase::mLock held 2395AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2396 Vector< sp<Track> > *tracksToRemove) 2397{ 2398 2399 mixer_state mixerStatus = MIXER_IDLE; 2400 // find out which tracks need to be processed 2401 size_t count = mActiveTracks.size(); 2402 size_t mixedTracks = 0; 2403 size_t tracksWithEffect = 0; 2404 // counts only _active_ fast tracks 2405 size_t fastTracks = 0; 2406 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2407 2408 float masterVolume = mMasterVolume; 2409 bool masterMute = mMasterMute; 2410 2411 if (masterMute) { 2412 masterVolume = 0; 2413 } 2414 // Delegate master volume control to effect in output mix effect chain if needed 2415 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2416 if (chain != 0) { 2417 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2418 chain->setVolume_l(&v, &v); 2419 masterVolume = (float)((v + (1 << 23)) >> 24); 2420 chain.clear(); 2421 } 2422 2423 // prepare a new state to push 2424 FastMixerStateQueue *sq = NULL; 2425 FastMixerState *state = NULL; 2426 bool didModify = false; 2427 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2428 if (mFastMixer != NULL) { 2429 sq = mFastMixer->sq(); 2430 state = sq->begin(); 2431 } 2432 2433 for (size_t i=0 ; i<count ; i++) { 2434 sp<Track> t = mActiveTracks[i].promote(); 2435 if (t == 0) { 2436 continue; 2437 } 2438 2439 // this const just means the local variable doesn't change 2440 Track* const track = t.get(); 2441 2442 // process fast tracks 2443 if (track->isFastTrack()) { 2444 2445 // It's theoretically possible (though unlikely) for a fast track to be created 2446 // and then removed within the same normal mix cycle. This is not a problem, as 2447 // the track never becomes active so it's fast mixer slot is never touched. 2448 // The converse, of removing an (active) track and then creating a new track 2449 // at the identical fast mixer slot within the same normal mix cycle, 2450 // is impossible because the slot isn't marked available until the end of each cycle. 2451 int j = track->mFastIndex; 2452 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2453 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2454 FastTrack *fastTrack = &state->mFastTracks[j]; 2455 2456 // Determine whether the track is currently in underrun condition, 2457 // and whether it had a recent underrun. 2458 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2459 FastTrackUnderruns underruns = ftDump->mUnderruns; 2460 uint32_t recentFull = (underruns.mBitFields.mFull - 2461 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2462 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2463 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2464 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2465 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2466 uint32_t recentUnderruns = recentPartial + recentEmpty; 2467 track->mObservedUnderruns = underruns; 2468 // don't count underruns that occur while stopping or pausing 2469 // or stopped which can occur when flush() is called while active 2470 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2471 track->mUnderrunCount += recentUnderruns; 2472 } 2473 2474 // This is similar to the state machine for normal tracks, 2475 // with a few modifications for fast tracks. 2476 bool isActive = true; 2477 switch (track->mState) { 2478 case TrackBase::STOPPING_1: 2479 // track stays active in STOPPING_1 state until first underrun 2480 if (recentUnderruns > 0) { 2481 track->mState = TrackBase::STOPPING_2; 2482 } 2483 break; 2484 case TrackBase::PAUSING: 2485 // ramp down is not yet implemented 2486 track->setPaused(); 2487 break; 2488 case TrackBase::RESUMING: 2489 // ramp up is not yet implemented 2490 track->mState = TrackBase::ACTIVE; 2491 break; 2492 case TrackBase::ACTIVE: 2493 if (recentFull > 0 || recentPartial > 0) { 2494 // track has provided at least some frames recently: reset retry count 2495 track->mRetryCount = kMaxTrackRetries; 2496 } 2497 if (recentUnderruns == 0) { 2498 // no recent underruns: stay active 2499 break; 2500 } 2501 // there has recently been an underrun of some kind 2502 if (track->sharedBuffer() == 0) { 2503 // were any of the recent underruns "empty" (no frames available)? 2504 if (recentEmpty == 0) { 2505 // no, then ignore the partial underruns as they are allowed indefinitely 2506 break; 2507 } 2508 // there has recently been an "empty" underrun: decrement the retry counter 2509 if (--(track->mRetryCount) > 0) { 2510 break; 2511 } 2512 // indicate to client process that the track was disabled because of underrun; 2513 // it will then automatically call start() when data is available 2514 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2515 // remove from active list, but state remains ACTIVE [confusing but true] 2516 isActive = false; 2517 break; 2518 } 2519 // fall through 2520 case TrackBase::STOPPING_2: 2521 case TrackBase::PAUSED: 2522 case TrackBase::TERMINATED: 2523 case TrackBase::STOPPED: 2524 case TrackBase::FLUSHED: // flush() while active 2525 // Check for presentation complete if track is inactive 2526 // We have consumed all the buffers of this track. 2527 // This would be incomplete if we auto-paused on underrun 2528 { 2529 size_t audioHALFrames = 2530 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2531 size_t framesWritten = mBytesWritten / mFrameSize; 2532 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2533 // track stays in active list until presentation is complete 2534 break; 2535 } 2536 } 2537 if (track->isStopping_2()) { 2538 track->mState = TrackBase::STOPPED; 2539 } 2540 if (track->isStopped()) { 2541 // Can't reset directly, as fast mixer is still polling this track 2542 // track->reset(); 2543 // So instead mark this track as needing to be reset after push with ack 2544 resetMask |= 1 << i; 2545 } 2546 isActive = false; 2547 break; 2548 case TrackBase::IDLE: 2549 default: 2550 LOG_FATAL("unexpected track state %d", track->mState); 2551 } 2552 2553 if (isActive) { 2554 // was it previously inactive? 2555 if (!(state->mTrackMask & (1 << j))) { 2556 ExtendedAudioBufferProvider *eabp = track; 2557 VolumeProvider *vp = track; 2558 fastTrack->mBufferProvider = eabp; 2559 fastTrack->mVolumeProvider = vp; 2560 fastTrack->mSampleRate = track->mSampleRate; 2561 fastTrack->mChannelMask = track->mChannelMask; 2562 fastTrack->mGeneration++; 2563 state->mTrackMask |= 1 << j; 2564 didModify = true; 2565 // no acknowledgement required for newly active tracks 2566 } 2567 // cache the combined master volume and stream type volume for fast mixer; this 2568 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2569 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2570 ++fastTracks; 2571 } else { 2572 // was it previously active? 2573 if (state->mTrackMask & (1 << j)) { 2574 fastTrack->mBufferProvider = NULL; 2575 fastTrack->mGeneration++; 2576 state->mTrackMask &= ~(1 << j); 2577 didModify = true; 2578 // If any fast tracks were removed, we must wait for acknowledgement 2579 // because we're about to decrement the last sp<> on those tracks. 2580 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2581 } else { 2582 LOG_FATAL("fast track %d should have been active", j); 2583 } 2584 tracksToRemove->add(track); 2585 // Avoids a misleading display in dumpsys 2586 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2587 } 2588 continue; 2589 } 2590 2591 { // local variable scope to avoid goto warning 2592 2593 audio_track_cblk_t* cblk = track->cblk(); 2594 2595 // The first time a track is added we wait 2596 // for all its buffers to be filled before processing it 2597 int name = track->name(); 2598 // make sure that we have enough frames to mix one full buffer. 2599 // enforce this condition only once to enable draining the buffer in case the client 2600 // app does not call stop() and relies on underrun to stop: 2601 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2602 // during last round 2603 size_t desiredFrames; 2604 if (t->sampleRate() == mSampleRate) { 2605 desiredFrames = mNormalFrameCount; 2606 } else { 2607 // +1 for rounding and +1 for additional sample needed for interpolation 2608 desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2609 // add frames already consumed but not yet released by the resampler 2610 // because cblk->framesReady() will include these frames 2611 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2612 // the minimum track buffer size is normally twice the number of frames necessary 2613 // to fill one buffer and the resampler should not leave more than one buffer worth 2614 // of unreleased frames after each pass, but just in case... 2615 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2616 } 2617 uint32_t minFrames = 1; 2618 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2619 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2620 minFrames = desiredFrames; 2621 } 2622 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2623 size_t framesReady; 2624 if (track->sharedBuffer() == 0) { 2625 framesReady = track->framesReady(); 2626 } else if (track->isStopped()) { 2627 framesReady = 0; 2628 } else { 2629 framesReady = 1; 2630 } 2631 if ((framesReady >= minFrames) && track->isReady() && 2632 !track->isPaused() && !track->isTerminated()) 2633 { 2634 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2635 this); 2636 2637 mixedTracks++; 2638 2639 // track->mainBuffer() != mMixBuffer means there is an effect chain 2640 // connected to the track 2641 chain.clear(); 2642 if (track->mainBuffer() != mMixBuffer) { 2643 chain = getEffectChain_l(track->sessionId()); 2644 // Delegate volume control to effect in track effect chain if needed 2645 if (chain != 0) { 2646 tracksWithEffect++; 2647 } else { 2648 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2649 "session %d", 2650 name, track->sessionId()); 2651 } 2652 } 2653 2654 2655 int param = AudioMixer::VOLUME; 2656 if (track->mFillingUpStatus == Track::FS_FILLED) { 2657 // no ramp for the first volume setting 2658 track->mFillingUpStatus = Track::FS_ACTIVE; 2659 if (track->mState == TrackBase::RESUMING) { 2660 track->mState = TrackBase::ACTIVE; 2661 param = AudioMixer::RAMP_VOLUME; 2662 } 2663 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2664 } else if (cblk->server != 0) { 2665 // If the track is stopped before the first frame was mixed, 2666 // do not apply ramp 2667 param = AudioMixer::RAMP_VOLUME; 2668 } 2669 2670 // compute volume for this track 2671 uint32_t vl, vr, va; 2672 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2673 vl = vr = va = 0; 2674 if (track->isPausing()) { 2675 track->setPaused(); 2676 } 2677 } else { 2678 2679 // read original volumes with volume control 2680 float typeVolume = mStreamTypes[track->streamType()].volume; 2681 float v = masterVolume * typeVolume; 2682 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2683 uint32_t vlr = proxy->getVolumeLR(); 2684 vl = vlr & 0xFFFF; 2685 vr = vlr >> 16; 2686 // track volumes come from shared memory, so can't be trusted and must be clamped 2687 if (vl > MAX_GAIN_INT) { 2688 ALOGV("Track left volume out of range: %04X", vl); 2689 vl = MAX_GAIN_INT; 2690 } 2691 if (vr > MAX_GAIN_INT) { 2692 ALOGV("Track right volume out of range: %04X", vr); 2693 vr = MAX_GAIN_INT; 2694 } 2695 // now apply the master volume and stream type volume 2696 vl = (uint32_t)(v * vl) << 12; 2697 vr = (uint32_t)(v * vr) << 12; 2698 // assuming master volume and stream type volume each go up to 1.0, 2699 // vl and vr are now in 8.24 format 2700 2701 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2702 // send level comes from shared memory and so may be corrupt 2703 if (sendLevel > MAX_GAIN_INT) { 2704 ALOGV("Track send level out of range: %04X", sendLevel); 2705 sendLevel = MAX_GAIN_INT; 2706 } 2707 va = (uint32_t)(v * sendLevel); 2708 } 2709 // Delegate volume control to effect in track effect chain if needed 2710 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2711 // Do not ramp volume if volume is controlled by effect 2712 param = AudioMixer::VOLUME; 2713 track->mHasVolumeController = true; 2714 } else { 2715 // force no volume ramp when volume controller was just disabled or removed 2716 // from effect chain to avoid volume spike 2717 if (track->mHasVolumeController) { 2718 param = AudioMixer::VOLUME; 2719 } 2720 track->mHasVolumeController = false; 2721 } 2722 2723 // Convert volumes from 8.24 to 4.12 format 2724 // This additional clamping is needed in case chain->setVolume_l() overshot 2725 vl = (vl + (1 << 11)) >> 12; 2726 if (vl > MAX_GAIN_INT) { 2727 vl = MAX_GAIN_INT; 2728 } 2729 vr = (vr + (1 << 11)) >> 12; 2730 if (vr > MAX_GAIN_INT) { 2731 vr = MAX_GAIN_INT; 2732 } 2733 2734 if (va > MAX_GAIN_INT) { 2735 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2736 } 2737 2738 // XXX: these things DON'T need to be done each time 2739 mAudioMixer->setBufferProvider(name, track); 2740 mAudioMixer->enable(name); 2741 2742 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2743 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2744 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2745 mAudioMixer->setParameter( 2746 name, 2747 AudioMixer::TRACK, 2748 AudioMixer::FORMAT, (void *)track->format()); 2749 mAudioMixer->setParameter( 2750 name, 2751 AudioMixer::TRACK, 2752 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2753 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2754 uint32_t maxSampleRate = mSampleRate * 2; 2755 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 2756 if (reqSampleRate == 0) { 2757 reqSampleRate = mSampleRate; 2758 } else if (reqSampleRate > maxSampleRate) { 2759 reqSampleRate = maxSampleRate; 2760 } 2761 mAudioMixer->setParameter( 2762 name, 2763 AudioMixer::RESAMPLE, 2764 AudioMixer::SAMPLE_RATE, 2765 (void *)reqSampleRate); 2766 mAudioMixer->setParameter( 2767 name, 2768 AudioMixer::TRACK, 2769 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2770 mAudioMixer->setParameter( 2771 name, 2772 AudioMixer::TRACK, 2773 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2774 2775 // reset retry count 2776 track->mRetryCount = kMaxTrackRetries; 2777 2778 // If one track is ready, set the mixer ready if: 2779 // - the mixer was not ready during previous round OR 2780 // - no other track is not ready 2781 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2782 mixerStatus != MIXER_TRACKS_ENABLED) { 2783 mixerStatus = MIXER_TRACKS_READY; 2784 } 2785 } else { 2786 // only implemented for normal tracks, not fast tracks 2787 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 2788 // we missed desiredFrames whatever the actual number of frames missing was 2789 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 2790 // FIXME also wake futex so that underrun is noticed more quickly 2791 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 2792 } 2793 // clear effect chain input buffer if an active track underruns to avoid sending 2794 // previous audio buffer again to effects 2795 chain = getEffectChain_l(track->sessionId()); 2796 if (chain != 0) { 2797 chain->clearInputBuffer(); 2798 } 2799 2800 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2801 cblk->server, this); 2802 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2803 track->isStopped() || track->isPaused()) { 2804 // We have consumed all the buffers of this track. 2805 // Remove it from the list of active tracks. 2806 // TODO: use actual buffer filling status instead of latency when available from 2807 // audio HAL 2808 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2809 size_t framesWritten = mBytesWritten / mFrameSize; 2810 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2811 if (track->isStopped()) { 2812 track->reset(); 2813 } 2814 tracksToRemove->add(track); 2815 } 2816 } else { 2817 track->mUnderrunCount++; 2818 // No buffers for this track. Give it a few chances to 2819 // fill a buffer, then remove it from active list. 2820 if (--(track->mRetryCount) <= 0) { 2821 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2822 tracksToRemove->add(track); 2823 // indicate to client process that the track was disabled because of underrun; 2824 // it will then automatically call start() when data is available 2825 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2826 // If one track is not ready, mark the mixer also not ready if: 2827 // - the mixer was ready during previous round OR 2828 // - no other track is ready 2829 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2830 mixerStatus != MIXER_TRACKS_READY) { 2831 mixerStatus = MIXER_TRACKS_ENABLED; 2832 } 2833 } 2834 mAudioMixer->disable(name); 2835 } 2836 2837 } // local variable scope to avoid goto warning 2838track_is_ready: ; 2839 2840 } 2841 2842 // Push the new FastMixer state if necessary 2843 bool pauseAudioWatchdog = false; 2844 if (didModify) { 2845 state->mFastTracksGen++; 2846 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2847 if (kUseFastMixer == FastMixer_Dynamic && 2848 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2849 state->mCommand = FastMixerState::COLD_IDLE; 2850 state->mColdFutexAddr = &mFastMixerFutex; 2851 state->mColdGen++; 2852 mFastMixerFutex = 0; 2853 if (kUseFastMixer == FastMixer_Dynamic) { 2854 mNormalSink = mOutputSink; 2855 } 2856 // If we go into cold idle, need to wait for acknowledgement 2857 // so that fast mixer stops doing I/O. 2858 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2859 pauseAudioWatchdog = true; 2860 } 2861 } 2862 if (sq != NULL) { 2863 sq->end(didModify); 2864 sq->push(block); 2865 } 2866#ifdef AUDIO_WATCHDOG 2867 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2868 mAudioWatchdog->pause(); 2869 } 2870#endif 2871 2872 // Now perform the deferred reset on fast tracks that have stopped 2873 while (resetMask != 0) { 2874 size_t i = __builtin_ctz(resetMask); 2875 ALOG_ASSERT(i < count); 2876 resetMask &= ~(1 << i); 2877 sp<Track> t = mActiveTracks[i].promote(); 2878 if (t == 0) { 2879 continue; 2880 } 2881 Track* track = t.get(); 2882 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2883 track->reset(); 2884 } 2885 2886 // remove all the tracks that need to be... 2887 count = tracksToRemove->size(); 2888 if (CC_UNLIKELY(count)) { 2889 for (size_t i=0 ; i<count ; i++) { 2890 const sp<Track>& track = tracksToRemove->itemAt(i); 2891 mActiveTracks.remove(track); 2892 if (track->mainBuffer() != mMixBuffer) { 2893 chain = getEffectChain_l(track->sessionId()); 2894 if (chain != 0) { 2895 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2896 track->sessionId()); 2897 chain->decActiveTrackCnt(); 2898 } 2899 } 2900 if (track->isTerminated()) { 2901 removeTrack_l(track); 2902 } 2903 } 2904 } 2905 2906 // mix buffer must be cleared if all tracks are connected to an 2907 // effect chain as in this case the mixer will not write to 2908 // mix buffer and track effects will accumulate into it 2909 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2910 (mixedTracks == 0 && fastTracks > 0)) { 2911 // FIXME as a performance optimization, should remember previous zero status 2912 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2913 } 2914 2915 // if any fast tracks, then status is ready 2916 mMixerStatusIgnoringFastTracks = mixerStatus; 2917 if (fastTracks > 0) { 2918 mixerStatus = MIXER_TRACKS_READY; 2919 } 2920 return mixerStatus; 2921} 2922 2923// getTrackName_l() must be called with ThreadBase::mLock held 2924int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2925{ 2926 return mAudioMixer->getTrackName(channelMask, sessionId); 2927} 2928 2929// deleteTrackName_l() must be called with ThreadBase::mLock held 2930void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2931{ 2932 ALOGV("remove track (%d) and delete from mixer", name); 2933 mAudioMixer->deleteTrackName(name); 2934} 2935 2936// checkForNewParameters_l() must be called with ThreadBase::mLock held 2937bool AudioFlinger::MixerThread::checkForNewParameters_l() 2938{ 2939 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2940 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2941 bool reconfig = false; 2942 2943 while (!mNewParameters.isEmpty()) { 2944 2945 if (mFastMixer != NULL) { 2946 FastMixerStateQueue *sq = mFastMixer->sq(); 2947 FastMixerState *state = sq->begin(); 2948 if (!(state->mCommand & FastMixerState::IDLE)) { 2949 previousCommand = state->mCommand; 2950 state->mCommand = FastMixerState::HOT_IDLE; 2951 sq->end(); 2952 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2953 } else { 2954 sq->end(false /*didModify*/); 2955 } 2956 } 2957 2958 status_t status = NO_ERROR; 2959 String8 keyValuePair = mNewParameters[0]; 2960 AudioParameter param = AudioParameter(keyValuePair); 2961 int value; 2962 2963 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2964 reconfig = true; 2965 } 2966 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2967 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2968 status = BAD_VALUE; 2969 } else { 2970 reconfig = true; 2971 } 2972 } 2973 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2974 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2975 status = BAD_VALUE; 2976 } else { 2977 reconfig = true; 2978 } 2979 } 2980 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2981 // do not accept frame count changes if tracks are open as the track buffer 2982 // size depends on frame count and correct behavior would not be guaranteed 2983 // if frame count is changed after track creation 2984 if (!mTracks.isEmpty()) { 2985 status = INVALID_OPERATION; 2986 } else { 2987 reconfig = true; 2988 } 2989 } 2990 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2991#ifdef ADD_BATTERY_DATA 2992 // when changing the audio output device, call addBatteryData to notify 2993 // the change 2994 if (mOutDevice != value) { 2995 uint32_t params = 0; 2996 // check whether speaker is on 2997 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2998 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2999 } 3000 3001 audio_devices_t deviceWithoutSpeaker 3002 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3003 // check if any other device (except speaker) is on 3004 if (value & deviceWithoutSpeaker ) { 3005 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3006 } 3007 3008 if (params != 0) { 3009 addBatteryData(params); 3010 } 3011 } 3012#endif 3013 3014 // forward device change to effects that have requested to be 3015 // aware of attached audio device. 3016 if (value != AUDIO_DEVICE_NONE) { 3017 mOutDevice = value; 3018 for (size_t i = 0; i < mEffectChains.size(); i++) { 3019 mEffectChains[i]->setDevice_l(mOutDevice); 3020 } 3021 } 3022 } 3023 3024 if (status == NO_ERROR) { 3025 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3026 keyValuePair.string()); 3027 if (!mStandby && status == INVALID_OPERATION) { 3028 mOutput->stream->common.standby(&mOutput->stream->common); 3029 mStandby = true; 3030 mBytesWritten = 0; 3031 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3032 keyValuePair.string()); 3033 } 3034 if (status == NO_ERROR && reconfig) { 3035 delete mAudioMixer; 3036 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3037 mAudioMixer = NULL; 3038 readOutputParameters(); 3039 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3040 for (size_t i = 0; i < mTracks.size() ; i++) { 3041 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3042 if (name < 0) { 3043 break; 3044 } 3045 mTracks[i]->mName = name; 3046 } 3047 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3048 } 3049 } 3050 3051 mNewParameters.removeAt(0); 3052 3053 mParamStatus = status; 3054 mParamCond.signal(); 3055 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3056 // already timed out waiting for the status and will never signal the condition. 3057 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3058 } 3059 3060 if (!(previousCommand & FastMixerState::IDLE)) { 3061 ALOG_ASSERT(mFastMixer != NULL); 3062 FastMixerStateQueue *sq = mFastMixer->sq(); 3063 FastMixerState *state = sq->begin(); 3064 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3065 state->mCommand = previousCommand; 3066 sq->end(); 3067 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3068 } 3069 3070 return reconfig; 3071} 3072 3073 3074void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3075{ 3076 const size_t SIZE = 256; 3077 char buffer[SIZE]; 3078 String8 result; 3079 3080 PlaybackThread::dumpInternals(fd, args); 3081 3082 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3083 result.append(buffer); 3084 write(fd, result.string(), result.size()); 3085 3086 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3087 const FastMixerDumpState copy(mFastMixerDumpState); 3088 copy.dump(fd); 3089 3090#ifdef STATE_QUEUE_DUMP 3091 // Similar for state queue 3092 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3093 observerCopy.dump(fd); 3094 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3095 mutatorCopy.dump(fd); 3096#endif 3097 3098#ifdef TEE_SINK 3099 // Write the tee output to a .wav file 3100 dumpTee(fd, mTeeSource, mId); 3101#endif 3102 3103#ifdef AUDIO_WATCHDOG 3104 if (mAudioWatchdog != 0) { 3105 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3106 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3107 wdCopy.dump(fd); 3108 } 3109#endif 3110} 3111 3112uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3113{ 3114 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3115} 3116 3117uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3118{ 3119 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3120} 3121 3122void AudioFlinger::MixerThread::cacheParameters_l() 3123{ 3124 PlaybackThread::cacheParameters_l(); 3125 3126 // FIXME: Relaxed timing because of a certain device that can't meet latency 3127 // Should be reduced to 2x after the vendor fixes the driver issue 3128 // increase threshold again due to low power audio mode. The way this warning 3129 // threshold is calculated and its usefulness should be reconsidered anyway. 3130 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3131} 3132 3133// ---------------------------------------------------------------------------- 3134 3135AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3136 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3137 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3138 // mLeftVolFloat, mRightVolFloat 3139{ 3140} 3141 3142AudioFlinger::DirectOutputThread::~DirectOutputThread() 3143{ 3144} 3145 3146AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3147 Vector< sp<Track> > *tracksToRemove 3148) 3149{ 3150 size_t count = mActiveTracks.size(); 3151 mixer_state mixerStatus = MIXER_IDLE; 3152 3153 // find out which tracks need to be processed 3154 for (size_t i = 0; i < count; i++) { 3155 sp<Track> t = mActiveTracks[i].promote(); 3156 // The track died recently 3157 if (t == 0) { 3158 continue; 3159 } 3160 3161 Track* const track = t.get(); 3162 audio_track_cblk_t* cblk = track->cblk(); 3163 3164 // The first time a track is added we wait 3165 // for all its buffers to be filled before processing it 3166 uint32_t minFrames; 3167 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3168 minFrames = mNormalFrameCount; 3169 } else { 3170 minFrames = 1; 3171 } 3172 if ((track->framesReady() >= minFrames) && track->isReady() && 3173 !track->isPaused() && !track->isTerminated()) 3174 { 3175 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3176 3177 if (track->mFillingUpStatus == Track::FS_FILLED) { 3178 track->mFillingUpStatus = Track::FS_ACTIVE; 3179 mLeftVolFloat = mRightVolFloat = 0; 3180 if (track->mState == TrackBase::RESUMING) { 3181 track->mState = TrackBase::ACTIVE; 3182 } 3183 } 3184 3185 // compute volume for this track 3186 float left, right; 3187 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3188 left = right = 0; 3189 if (track->isPausing()) { 3190 track->setPaused(); 3191 } 3192 } else { 3193 float typeVolume = mStreamTypes[track->streamType()].volume; 3194 float v = mMasterVolume * typeVolume; 3195 uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); 3196 float v_clamped = v * (vlr & 0xFFFF); 3197 if (v_clamped > MAX_GAIN) { 3198 v_clamped = MAX_GAIN; 3199 } 3200 left = v_clamped/MAX_GAIN; 3201 v_clamped = v * (vlr >> 16); 3202 if (v_clamped > MAX_GAIN) { 3203 v_clamped = MAX_GAIN; 3204 } 3205 right = v_clamped/MAX_GAIN; 3206 } 3207 // Only consider last track started for volume and mixer state control. 3208 // This is the last entry in mActiveTracks unless a track underruns. 3209 // As we only care about the transition phase between two tracks on a 3210 // direct output, it is not a problem to ignore the underrun case. 3211 if (i == (count - 1)) { 3212 if (left != mLeftVolFloat || right != mRightVolFloat) { 3213 mLeftVolFloat = left; 3214 mRightVolFloat = right; 3215 3216 // Convert volumes from float to 8.24 3217 uint32_t vl = (uint32_t)(left * (1 << 24)); 3218 uint32_t vr = (uint32_t)(right * (1 << 24)); 3219 3220 // Delegate volume control to effect in track effect chain if needed 3221 // only one effect chain can be present on DirectOutputThread, so if 3222 // there is one, the track is connected to it 3223 if (!mEffectChains.isEmpty()) { 3224 // Do not ramp volume if volume is controlled by effect 3225 mEffectChains[0]->setVolume_l(&vl, &vr); 3226 left = (float)vl / (1 << 24); 3227 right = (float)vr / (1 << 24); 3228 } 3229 mOutput->stream->set_volume(mOutput->stream, left, right); 3230 } 3231 3232 // reset retry count 3233 track->mRetryCount = kMaxTrackRetriesDirect; 3234 mActiveTrack = t; 3235 mixerStatus = MIXER_TRACKS_READY; 3236 } 3237 } else { 3238 // clear effect chain input buffer if the last active track started underruns 3239 // to avoid sending previous audio buffer again to effects 3240 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3241 mEffectChains[0]->clearInputBuffer(); 3242 } 3243 3244 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3245 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3246 track->isStopped() || track->isPaused()) { 3247 // We have consumed all the buffers of this track. 3248 // Remove it from the list of active tracks. 3249 // TODO: implement behavior for compressed audio 3250 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3251 size_t framesWritten = mBytesWritten / mFrameSize; 3252 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3253 if (track->isStopped()) { 3254 track->reset(); 3255 } 3256 tracksToRemove->add(track); 3257 } 3258 } else { 3259 // No buffers for this track. Give it a few chances to 3260 // fill a buffer, then remove it from active list. 3261 // Only consider last track started for mixer state control 3262 if (--(track->mRetryCount) <= 0) { 3263 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3264 tracksToRemove->add(track); 3265 } else if (i == (count -1)){ 3266 mixerStatus = MIXER_TRACKS_ENABLED; 3267 } 3268 } 3269 } 3270 } 3271 3272 // remove all the tracks that need to be... 3273 count = tracksToRemove->size(); 3274 if (CC_UNLIKELY(count)) { 3275 for (size_t i = 0 ; i < count ; i++) { 3276 const sp<Track>& track = tracksToRemove->itemAt(i); 3277 mActiveTracks.remove(track); 3278 if (!mEffectChains.isEmpty()) { 3279 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3280 track->sessionId()); 3281 mEffectChains[0]->decActiveTrackCnt(); 3282 } 3283 if (track->isTerminated()) { 3284 removeTrack_l(track); 3285 } 3286 } 3287 } 3288 3289 return mixerStatus; 3290} 3291 3292void AudioFlinger::DirectOutputThread::threadLoop_mix() 3293{ 3294 AudioBufferProvider::Buffer buffer; 3295 size_t frameCount = mFrameCount; 3296 int8_t *curBuf = (int8_t *)mMixBuffer; 3297 // output audio to hardware 3298 while (frameCount) { 3299 buffer.frameCount = frameCount; 3300 mActiveTrack->getNextBuffer(&buffer); 3301 if (CC_UNLIKELY(buffer.raw == NULL)) { 3302 memset(curBuf, 0, frameCount * mFrameSize); 3303 break; 3304 } 3305 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3306 frameCount -= buffer.frameCount; 3307 curBuf += buffer.frameCount * mFrameSize; 3308 mActiveTrack->releaseBuffer(&buffer); 3309 } 3310 sleepTime = 0; 3311 standbyTime = systemTime() + standbyDelay; 3312 mActiveTrack.clear(); 3313 3314} 3315 3316void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3317{ 3318 if (sleepTime == 0) { 3319 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3320 sleepTime = activeSleepTime; 3321 } else { 3322 sleepTime = idleSleepTime; 3323 } 3324 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3325 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3326 sleepTime = 0; 3327 } 3328} 3329 3330// getTrackName_l() must be called with ThreadBase::mLock held 3331int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3332 int sessionId) 3333{ 3334 return 0; 3335} 3336 3337// deleteTrackName_l() must be called with ThreadBase::mLock held 3338void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3339{ 3340} 3341 3342// checkForNewParameters_l() must be called with ThreadBase::mLock held 3343bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3344{ 3345 bool reconfig = false; 3346 3347 while (!mNewParameters.isEmpty()) { 3348 status_t status = NO_ERROR; 3349 String8 keyValuePair = mNewParameters[0]; 3350 AudioParameter param = AudioParameter(keyValuePair); 3351 int value; 3352 3353 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3354 // do not accept frame count changes if tracks are open as the track buffer 3355 // size depends on frame count and correct behavior would not be garantied 3356 // if frame count is changed after track creation 3357 if (!mTracks.isEmpty()) { 3358 status = INVALID_OPERATION; 3359 } else { 3360 reconfig = true; 3361 } 3362 } 3363 if (status == NO_ERROR) { 3364 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3365 keyValuePair.string()); 3366 if (!mStandby && status == INVALID_OPERATION) { 3367 mOutput->stream->common.standby(&mOutput->stream->common); 3368 mStandby = true; 3369 mBytesWritten = 0; 3370 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3371 keyValuePair.string()); 3372 } 3373 if (status == NO_ERROR && reconfig) { 3374 readOutputParameters(); 3375 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3376 } 3377 } 3378 3379 mNewParameters.removeAt(0); 3380 3381 mParamStatus = status; 3382 mParamCond.signal(); 3383 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3384 // already timed out waiting for the status and will never signal the condition. 3385 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3386 } 3387 return reconfig; 3388} 3389 3390uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3391{ 3392 uint32_t time; 3393 if (audio_is_linear_pcm(mFormat)) { 3394 time = PlaybackThread::activeSleepTimeUs(); 3395 } else { 3396 time = 10000; 3397 } 3398 return time; 3399} 3400 3401uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3402{ 3403 uint32_t time; 3404 if (audio_is_linear_pcm(mFormat)) { 3405 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3406 } else { 3407 time = 10000; 3408 } 3409 return time; 3410} 3411 3412uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3413{ 3414 uint32_t time; 3415 if (audio_is_linear_pcm(mFormat)) { 3416 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3417 } else { 3418 time = 10000; 3419 } 3420 return time; 3421} 3422 3423void AudioFlinger::DirectOutputThread::cacheParameters_l() 3424{ 3425 PlaybackThread::cacheParameters_l(); 3426 3427 // use shorter standby delay as on normal output to release 3428 // hardware resources as soon as possible 3429 standbyDelay = microseconds(activeSleepTime*2); 3430} 3431 3432// ---------------------------------------------------------------------------- 3433 3434AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3435 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3436 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3437 DUPLICATING), 3438 mWaitTimeMs(UINT_MAX) 3439{ 3440 addOutputTrack(mainThread); 3441} 3442 3443AudioFlinger::DuplicatingThread::~DuplicatingThread() 3444{ 3445 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3446 mOutputTracks[i]->destroy(); 3447 } 3448} 3449 3450void AudioFlinger::DuplicatingThread::threadLoop_mix() 3451{ 3452 // mix buffers... 3453 if (outputsReady(outputTracks)) { 3454 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3455 } else { 3456 memset(mMixBuffer, 0, mixBufferSize); 3457 } 3458 sleepTime = 0; 3459 writeFrames = mNormalFrameCount; 3460 standbyTime = systemTime() + standbyDelay; 3461} 3462 3463void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3464{ 3465 if (sleepTime == 0) { 3466 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3467 sleepTime = activeSleepTime; 3468 } else { 3469 sleepTime = idleSleepTime; 3470 } 3471 } else if (mBytesWritten != 0) { 3472 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3473 writeFrames = mNormalFrameCount; 3474 memset(mMixBuffer, 0, mixBufferSize); 3475 } else { 3476 // flush remaining overflow buffers in output tracks 3477 writeFrames = 0; 3478 } 3479 sleepTime = 0; 3480 } 3481} 3482 3483void AudioFlinger::DuplicatingThread::threadLoop_write() 3484{ 3485 for (size_t i = 0; i < outputTracks.size(); i++) { 3486 outputTracks[i]->write(mMixBuffer, writeFrames); 3487 } 3488 mBytesWritten += mixBufferSize; 3489} 3490 3491void AudioFlinger::DuplicatingThread::threadLoop_standby() 3492{ 3493 // DuplicatingThread implements standby by stopping all tracks 3494 for (size_t i = 0; i < outputTracks.size(); i++) { 3495 outputTracks[i]->stop(); 3496 } 3497} 3498 3499void AudioFlinger::DuplicatingThread::saveOutputTracks() 3500{ 3501 outputTracks = mOutputTracks; 3502} 3503 3504void AudioFlinger::DuplicatingThread::clearOutputTracks() 3505{ 3506 outputTracks.clear(); 3507} 3508 3509void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3510{ 3511 Mutex::Autolock _l(mLock); 3512 // FIXME explain this formula 3513 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3514 OutputTrack *outputTrack = new OutputTrack(thread, 3515 this, 3516 mSampleRate, 3517 mFormat, 3518 mChannelMask, 3519 frameCount); 3520 if (outputTrack->cblk() != NULL) { 3521 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3522 mOutputTracks.add(outputTrack); 3523 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3524 updateWaitTime_l(); 3525 } 3526} 3527 3528void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3529{ 3530 Mutex::Autolock _l(mLock); 3531 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3532 if (mOutputTracks[i]->thread() == thread) { 3533 mOutputTracks[i]->destroy(); 3534 mOutputTracks.removeAt(i); 3535 updateWaitTime_l(); 3536 return; 3537 } 3538 } 3539 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3540} 3541 3542// caller must hold mLock 3543void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3544{ 3545 mWaitTimeMs = UINT_MAX; 3546 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3547 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3548 if (strong != 0) { 3549 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3550 if (waitTimeMs < mWaitTimeMs) { 3551 mWaitTimeMs = waitTimeMs; 3552 } 3553 } 3554 } 3555} 3556 3557 3558bool AudioFlinger::DuplicatingThread::outputsReady( 3559 const SortedVector< sp<OutputTrack> > &outputTracks) 3560{ 3561 for (size_t i = 0; i < outputTracks.size(); i++) { 3562 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3563 if (thread == 0) { 3564 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3565 outputTracks[i].get()); 3566 return false; 3567 } 3568 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3569 // see note at standby() declaration 3570 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3571 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3572 thread.get()); 3573 return false; 3574 } 3575 } 3576 return true; 3577} 3578 3579uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3580{ 3581 return (mWaitTimeMs * 1000) / 2; 3582} 3583 3584void AudioFlinger::DuplicatingThread::cacheParameters_l() 3585{ 3586 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3587 updateWaitTime_l(); 3588 3589 MixerThread::cacheParameters_l(); 3590} 3591 3592// ---------------------------------------------------------------------------- 3593// Record 3594// ---------------------------------------------------------------------------- 3595 3596AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3597 AudioStreamIn *input, 3598 uint32_t sampleRate, 3599 audio_channel_mask_t channelMask, 3600 audio_io_handle_t id, 3601 audio_devices_t outDevice, 3602 audio_devices_t inDevice 3603#ifdef TEE_SINK 3604 , const sp<NBAIO_Sink>& teeSink 3605#endif 3606 ) : 3607 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3608 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3609 // mRsmpInIndex and mInputBytes set by readInputParameters() 3610 mReqChannelCount(popcount(channelMask)), 3611 mReqSampleRate(sampleRate) 3612 // mBytesRead is only meaningful while active, and so is cleared in start() 3613 // (but might be better to also clear here for dump?) 3614#ifdef TEE_SINK 3615 , mTeeSink(teeSink) 3616#endif 3617{ 3618 snprintf(mName, kNameLength, "AudioIn_%X", id); 3619 3620 readInputParameters(); 3621 3622} 3623 3624 3625AudioFlinger::RecordThread::~RecordThread() 3626{ 3627 delete[] mRsmpInBuffer; 3628 delete mResampler; 3629 delete[] mRsmpOutBuffer; 3630} 3631 3632void AudioFlinger::RecordThread::onFirstRef() 3633{ 3634 run(mName, PRIORITY_URGENT_AUDIO); 3635} 3636 3637status_t AudioFlinger::RecordThread::readyToRun() 3638{ 3639 status_t status = initCheck(); 3640 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3641 return status; 3642} 3643 3644bool AudioFlinger::RecordThread::threadLoop() 3645{ 3646 AudioBufferProvider::Buffer buffer; 3647 sp<RecordTrack> activeTrack; 3648 Vector< sp<EffectChain> > effectChains; 3649 3650 nsecs_t lastWarning = 0; 3651 3652 inputStandBy(); 3653 acquireWakeLock(); 3654 3655 // used to verify we've read at least once before evaluating how many bytes were read 3656 bool readOnce = false; 3657 3658 // start recording 3659 while (!exitPending()) { 3660 3661 processConfigEvents(); 3662 3663 { // scope for mLock 3664 Mutex::Autolock _l(mLock); 3665 checkForNewParameters_l(); 3666 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3667 standby(); 3668 3669 if (exitPending()) { 3670 break; 3671 } 3672 3673 releaseWakeLock_l(); 3674 ALOGV("RecordThread: loop stopping"); 3675 // go to sleep 3676 mWaitWorkCV.wait(mLock); 3677 ALOGV("RecordThread: loop starting"); 3678 acquireWakeLock_l(); 3679 continue; 3680 } 3681 if (mActiveTrack != 0) { 3682 if (mActiveTrack->mState == TrackBase::PAUSING) { 3683 standby(); 3684 mActiveTrack.clear(); 3685 mStartStopCond.broadcast(); 3686 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3687 if (mReqChannelCount != mActiveTrack->channelCount()) { 3688 mActiveTrack.clear(); 3689 mStartStopCond.broadcast(); 3690 } else if (readOnce) { 3691 // record start succeeds only if first read from audio input 3692 // succeeds 3693 if (mBytesRead >= 0) { 3694 mActiveTrack->mState = TrackBase::ACTIVE; 3695 } else { 3696 mActiveTrack.clear(); 3697 } 3698 mStartStopCond.broadcast(); 3699 } 3700 mStandby = false; 3701 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3702 removeTrack_l(mActiveTrack); 3703 mActiveTrack.clear(); 3704 } 3705 } 3706 lockEffectChains_l(effectChains); 3707 } 3708 3709 if (mActiveTrack != 0) { 3710 if (mActiveTrack->mState != TrackBase::ACTIVE && 3711 mActiveTrack->mState != TrackBase::RESUMING) { 3712 unlockEffectChains(effectChains); 3713 usleep(kRecordThreadSleepUs); 3714 continue; 3715 } 3716 for (size_t i = 0; i < effectChains.size(); i ++) { 3717 effectChains[i]->process_l(); 3718 } 3719 3720 buffer.frameCount = mFrameCount; 3721 status_t status = mActiveTrack->getNextBuffer(&buffer); 3722 if (CC_LIKELY(status == NO_ERROR)) { 3723 readOnce = true; 3724 size_t framesOut = buffer.frameCount; 3725 if (mResampler == NULL) { 3726 // no resampling 3727 while (framesOut) { 3728 size_t framesIn = mFrameCount - mRsmpInIndex; 3729 if (framesIn) { 3730 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3731 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3732 mActiveTrack->mFrameSize; 3733 if (framesIn > framesOut) 3734 framesIn = framesOut; 3735 mRsmpInIndex += framesIn; 3736 framesOut -= framesIn; 3737 if (mChannelCount == mReqChannelCount || 3738 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3739 memcpy(dst, src, framesIn * mFrameSize); 3740 } else { 3741 if (mChannelCount == 1) { 3742 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3743 (int16_t *)src, framesIn); 3744 } else { 3745 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3746 (int16_t *)src, framesIn); 3747 } 3748 } 3749 } 3750 if (framesOut && mFrameCount == mRsmpInIndex) { 3751 void *readInto; 3752 if (framesOut == mFrameCount && 3753 (mChannelCount == mReqChannelCount || 3754 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3755 readInto = buffer.raw; 3756 framesOut = 0; 3757 } else { 3758 readInto = mRsmpInBuffer; 3759 mRsmpInIndex = 0; 3760 } 3761 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3762 mInputBytes); 3763 if (mBytesRead <= 0) { 3764 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3765 { 3766 ALOGE("Error reading audio input"); 3767 // Force input into standby so that it tries to 3768 // recover at next read attempt 3769 inputStandBy(); 3770 usleep(kRecordThreadSleepUs); 3771 } 3772 mRsmpInIndex = mFrameCount; 3773 framesOut = 0; 3774 buffer.frameCount = 0; 3775 } 3776#ifdef TEE_SINK 3777 else if (mTeeSink != 0) { 3778 (void) mTeeSink->write(readInto, 3779 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3780 } 3781#endif 3782 } 3783 } 3784 } else { 3785 // resampling 3786 3787 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3788 // alter output frame count as if we were expecting stereo samples 3789 if (mChannelCount == 1 && mReqChannelCount == 1) { 3790 framesOut >>= 1; 3791 } 3792 mResampler->resample(mRsmpOutBuffer, framesOut, 3793 this /* AudioBufferProvider* */); 3794 // ditherAndClamp() works as long as all buffers returned by 3795 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3796 if (mChannelCount == 2 && mReqChannelCount == 1) { 3797 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3798 // the resampler always outputs stereo samples: 3799 // do post stereo to mono conversion 3800 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3801 framesOut); 3802 } else { 3803 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3804 } 3805 3806 } 3807 if (mFramestoDrop == 0) { 3808 mActiveTrack->releaseBuffer(&buffer); 3809 } else { 3810 if (mFramestoDrop > 0) { 3811 mFramestoDrop -= buffer.frameCount; 3812 if (mFramestoDrop <= 0) { 3813 clearSyncStartEvent(); 3814 } 3815 } else { 3816 mFramestoDrop += buffer.frameCount; 3817 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3818 mSyncStartEvent->isCancelled()) { 3819 ALOGW("Synced record %s, session %d, trigger session %d", 3820 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3821 mActiveTrack->sessionId(), 3822 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3823 clearSyncStartEvent(); 3824 } 3825 } 3826 } 3827 mActiveTrack->clearOverflow(); 3828 } 3829 // client isn't retrieving buffers fast enough 3830 else { 3831 if (!mActiveTrack->setOverflow()) { 3832 nsecs_t now = systemTime(); 3833 if ((now - lastWarning) > kWarningThrottleNs) { 3834 ALOGW("RecordThread: buffer overflow"); 3835 lastWarning = now; 3836 } 3837 } 3838 // Release the processor for a while before asking for a new buffer. 3839 // This will give the application more chance to read from the buffer and 3840 // clear the overflow. 3841 usleep(kRecordThreadSleepUs); 3842 } 3843 } 3844 // enable changes in effect chain 3845 unlockEffectChains(effectChains); 3846 effectChains.clear(); 3847 } 3848 3849 standby(); 3850 3851 { 3852 Mutex::Autolock _l(mLock); 3853 mActiveTrack.clear(); 3854 mStartStopCond.broadcast(); 3855 } 3856 3857 releaseWakeLock(); 3858 3859 ALOGV("RecordThread %p exiting", this); 3860 return false; 3861} 3862 3863void AudioFlinger::RecordThread::standby() 3864{ 3865 if (!mStandby) { 3866 inputStandBy(); 3867 mStandby = true; 3868 } 3869} 3870 3871void AudioFlinger::RecordThread::inputStandBy() 3872{ 3873 mInput->stream->common.standby(&mInput->stream->common); 3874} 3875 3876sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3877 const sp<AudioFlinger::Client>& client, 3878 uint32_t sampleRate, 3879 audio_format_t format, 3880 audio_channel_mask_t channelMask, 3881 size_t frameCount, 3882 int sessionId, 3883 IAudioFlinger::track_flags_t flags, 3884 pid_t tid, 3885 status_t *status) 3886{ 3887 sp<RecordTrack> track; 3888 status_t lStatus; 3889 3890 lStatus = initCheck(); 3891 if (lStatus != NO_ERROR) { 3892 ALOGE("Audio driver not initialized."); 3893 goto Exit; 3894 } 3895 3896 // FIXME use flags and tid similar to createTrack_l() 3897 3898 { // scope for mLock 3899 Mutex::Autolock _l(mLock); 3900 3901 track = new RecordTrack(this, client, sampleRate, 3902 format, channelMask, frameCount, sessionId); 3903 3904 if (track->getCblk() == 0) { 3905 lStatus = NO_MEMORY; 3906 goto Exit; 3907 } 3908 mTracks.add(track); 3909 3910 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3911 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3912 mAudioFlinger->btNrecIsOff(); 3913 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3914 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3915 } 3916 lStatus = NO_ERROR; 3917 3918Exit: 3919 if (status) { 3920 *status = lStatus; 3921 } 3922 return track; 3923} 3924 3925status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3926 AudioSystem::sync_event_t event, 3927 int triggerSession) 3928{ 3929 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3930 sp<ThreadBase> strongMe = this; 3931 status_t status = NO_ERROR; 3932 3933 if (event == AudioSystem::SYNC_EVENT_NONE) { 3934 clearSyncStartEvent(); 3935 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3936 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3937 triggerSession, 3938 recordTrack->sessionId(), 3939 syncStartEventCallback, 3940 this); 3941 // Sync event can be cancelled by the trigger session if the track is not in a 3942 // compatible state in which case we start record immediately 3943 if (mSyncStartEvent->isCancelled()) { 3944 clearSyncStartEvent(); 3945 } else { 3946 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3947 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3948 } 3949 } 3950 3951 { 3952 AutoMutex lock(mLock); 3953 if (mActiveTrack != 0) { 3954 if (recordTrack != mActiveTrack.get()) { 3955 status = -EBUSY; 3956 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3957 mActiveTrack->mState = TrackBase::ACTIVE; 3958 } 3959 return status; 3960 } 3961 3962 recordTrack->mState = TrackBase::IDLE; 3963 mActiveTrack = recordTrack; 3964 mLock.unlock(); 3965 status_t status = AudioSystem::startInput(mId); 3966 mLock.lock(); 3967 if (status != NO_ERROR) { 3968 mActiveTrack.clear(); 3969 clearSyncStartEvent(); 3970 return status; 3971 } 3972 mRsmpInIndex = mFrameCount; 3973 mBytesRead = 0; 3974 if (mResampler != NULL) { 3975 mResampler->reset(); 3976 } 3977 mActiveTrack->mState = TrackBase::RESUMING; 3978 // signal thread to start 3979 ALOGV("Signal record thread"); 3980 mWaitWorkCV.broadcast(); 3981 // do not wait for mStartStopCond if exiting 3982 if (exitPending()) { 3983 mActiveTrack.clear(); 3984 status = INVALID_OPERATION; 3985 goto startError; 3986 } 3987 mStartStopCond.wait(mLock); 3988 if (mActiveTrack == 0) { 3989 ALOGV("Record failed to start"); 3990 status = BAD_VALUE; 3991 goto startError; 3992 } 3993 ALOGV("Record started OK"); 3994 return status; 3995 } 3996 3997startError: 3998 AudioSystem::stopInput(mId); 3999 clearSyncStartEvent(); 4000 return status; 4001} 4002 4003void AudioFlinger::RecordThread::clearSyncStartEvent() 4004{ 4005 if (mSyncStartEvent != 0) { 4006 mSyncStartEvent->cancel(); 4007 } 4008 mSyncStartEvent.clear(); 4009 mFramestoDrop = 0; 4010} 4011 4012void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4013{ 4014 sp<SyncEvent> strongEvent = event.promote(); 4015 4016 if (strongEvent != 0) { 4017 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4018 me->handleSyncStartEvent(strongEvent); 4019 } 4020} 4021 4022void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4023{ 4024 if (event == mSyncStartEvent) { 4025 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4026 // from audio HAL 4027 mFramestoDrop = mFrameCount * 2; 4028 } 4029} 4030 4031bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4032 ALOGV("RecordThread::stop"); 4033 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4034 return false; 4035 } 4036 recordTrack->mState = TrackBase::PAUSING; 4037 // do not wait for mStartStopCond if exiting 4038 if (exitPending()) { 4039 return true; 4040 } 4041 mStartStopCond.wait(mLock); 4042 // if we have been restarted, recordTrack == mActiveTrack.get() here 4043 if (exitPending() || recordTrack != mActiveTrack.get()) { 4044 ALOGV("Record stopped OK"); 4045 return true; 4046 } 4047 return false; 4048} 4049 4050bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4051{ 4052 return false; 4053} 4054 4055status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4056{ 4057#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4058 if (!isValidSyncEvent(event)) { 4059 return BAD_VALUE; 4060 } 4061 4062 int eventSession = event->triggerSession(); 4063 status_t ret = NAME_NOT_FOUND; 4064 4065 Mutex::Autolock _l(mLock); 4066 4067 for (size_t i = 0; i < mTracks.size(); i++) { 4068 sp<RecordTrack> track = mTracks[i]; 4069 if (eventSession == track->sessionId()) { 4070 (void) track->setSyncEvent(event); 4071 ret = NO_ERROR; 4072 } 4073 } 4074 return ret; 4075#else 4076 return BAD_VALUE; 4077#endif 4078} 4079 4080// destroyTrack_l() must be called with ThreadBase::mLock held 4081void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4082{ 4083 track->mState = TrackBase::TERMINATED; 4084 // active tracks are removed by threadLoop() 4085 if (mActiveTrack != track) { 4086 removeTrack_l(track); 4087 } 4088} 4089 4090void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4091{ 4092 mTracks.remove(track); 4093 // need anything related to effects here? 4094} 4095 4096void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4097{ 4098 dumpInternals(fd, args); 4099 dumpTracks(fd, args); 4100 dumpEffectChains(fd, args); 4101} 4102 4103void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4104{ 4105 const size_t SIZE = 256; 4106 char buffer[SIZE]; 4107 String8 result; 4108 4109 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4110 result.append(buffer); 4111 4112 if (mActiveTrack != 0) { 4113 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4114 result.append(buffer); 4115 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4116 result.append(buffer); 4117 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4118 result.append(buffer); 4119 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4120 result.append(buffer); 4121 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4122 result.append(buffer); 4123 } else { 4124 result.append("No active record client\n"); 4125 } 4126 4127 write(fd, result.string(), result.size()); 4128 4129 dumpBase(fd, args); 4130} 4131 4132void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4133{ 4134 const size_t SIZE = 256; 4135 char buffer[SIZE]; 4136 String8 result; 4137 4138 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4139 result.append(buffer); 4140 RecordTrack::appendDumpHeader(result); 4141 for (size_t i = 0; i < mTracks.size(); ++i) { 4142 sp<RecordTrack> track = mTracks[i]; 4143 if (track != 0) { 4144 track->dump(buffer, SIZE); 4145 result.append(buffer); 4146 } 4147 } 4148 4149 if (mActiveTrack != 0) { 4150 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4151 result.append(buffer); 4152 RecordTrack::appendDumpHeader(result); 4153 mActiveTrack->dump(buffer, SIZE); 4154 result.append(buffer); 4155 4156 } 4157 write(fd, result.string(), result.size()); 4158} 4159 4160// AudioBufferProvider interface 4161status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4162{ 4163 size_t framesReq = buffer->frameCount; 4164 size_t framesReady = mFrameCount - mRsmpInIndex; 4165 int channelCount; 4166 4167 if (framesReady == 0) { 4168 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4169 if (mBytesRead <= 0) { 4170 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4171 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4172 // Force input into standby so that it tries to 4173 // recover at next read attempt 4174 inputStandBy(); 4175 usleep(kRecordThreadSleepUs); 4176 } 4177 buffer->raw = NULL; 4178 buffer->frameCount = 0; 4179 return NOT_ENOUGH_DATA; 4180 } 4181 mRsmpInIndex = 0; 4182 framesReady = mFrameCount; 4183 } 4184 4185 if (framesReq > framesReady) { 4186 framesReq = framesReady; 4187 } 4188 4189 if (mChannelCount == 1 && mReqChannelCount == 2) { 4190 channelCount = 1; 4191 } else { 4192 channelCount = 2; 4193 } 4194 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4195 buffer->frameCount = framesReq; 4196 return NO_ERROR; 4197} 4198 4199// AudioBufferProvider interface 4200void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4201{ 4202 mRsmpInIndex += buffer->frameCount; 4203 buffer->frameCount = 0; 4204} 4205 4206bool AudioFlinger::RecordThread::checkForNewParameters_l() 4207{ 4208 bool reconfig = false; 4209 4210 while (!mNewParameters.isEmpty()) { 4211 status_t status = NO_ERROR; 4212 String8 keyValuePair = mNewParameters[0]; 4213 AudioParameter param = AudioParameter(keyValuePair); 4214 int value; 4215 audio_format_t reqFormat = mFormat; 4216 uint32_t reqSamplingRate = mReqSampleRate; 4217 uint32_t reqChannelCount = mReqChannelCount; 4218 4219 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4220 reqSamplingRate = value; 4221 reconfig = true; 4222 } 4223 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4224 reqFormat = (audio_format_t) value; 4225 reconfig = true; 4226 } 4227 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4228 reqChannelCount = popcount(value); 4229 reconfig = true; 4230 } 4231 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4232 // do not accept frame count changes if tracks are open as the track buffer 4233 // size depends on frame count and correct behavior would not be guaranteed 4234 // if frame count is changed after track creation 4235 if (mActiveTrack != 0) { 4236 status = INVALID_OPERATION; 4237 } else { 4238 reconfig = true; 4239 } 4240 } 4241 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4242 // forward device change to effects that have requested to be 4243 // aware of attached audio device. 4244 for (size_t i = 0; i < mEffectChains.size(); i++) { 4245 mEffectChains[i]->setDevice_l(value); 4246 } 4247 4248 // store input device and output device but do not forward output device to audio HAL. 4249 // Note that status is ignored by the caller for output device 4250 // (see AudioFlinger::setParameters() 4251 if (audio_is_output_devices(value)) { 4252 mOutDevice = value; 4253 status = BAD_VALUE; 4254 } else { 4255 mInDevice = value; 4256 // disable AEC and NS if the device is a BT SCO headset supporting those 4257 // pre processings 4258 if (mTracks.size() > 0) { 4259 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4260 mAudioFlinger->btNrecIsOff(); 4261 for (size_t i = 0; i < mTracks.size(); i++) { 4262 sp<RecordTrack> track = mTracks[i]; 4263 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4264 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4265 } 4266 } 4267 } 4268 } 4269 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4270 mAudioSource != (audio_source_t)value) { 4271 // forward device change to effects that have requested to be 4272 // aware of attached audio device. 4273 for (size_t i = 0; i < mEffectChains.size(); i++) { 4274 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4275 } 4276 mAudioSource = (audio_source_t)value; 4277 } 4278 if (status == NO_ERROR) { 4279 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4280 keyValuePair.string()); 4281 if (status == INVALID_OPERATION) { 4282 inputStandBy(); 4283 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4284 keyValuePair.string()); 4285 } 4286 if (reconfig) { 4287 if (status == BAD_VALUE && 4288 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4289 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4290 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4291 <= (2 * reqSamplingRate)) && 4292 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4293 <= FCC_2 && 4294 (reqChannelCount <= FCC_2)) { 4295 status = NO_ERROR; 4296 } 4297 if (status == NO_ERROR) { 4298 readInputParameters(); 4299 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4300 } 4301 } 4302 } 4303 4304 mNewParameters.removeAt(0); 4305 4306 mParamStatus = status; 4307 mParamCond.signal(); 4308 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4309 // already timed out waiting for the status and will never signal the condition. 4310 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4311 } 4312 return reconfig; 4313} 4314 4315String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4316{ 4317 Mutex::Autolock _l(mLock); 4318 if (initCheck() != NO_ERROR) { 4319 return String8(); 4320 } 4321 4322 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4323 const String8 out_s8(s); 4324 free(s); 4325 return out_s8; 4326} 4327 4328void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4329 AudioSystem::OutputDescriptor desc; 4330 void *param2 = NULL; 4331 4332 switch (event) { 4333 case AudioSystem::INPUT_OPENED: 4334 case AudioSystem::INPUT_CONFIG_CHANGED: 4335 desc.channels = mChannelMask; 4336 desc.samplingRate = mSampleRate; 4337 desc.format = mFormat; 4338 desc.frameCount = mFrameCount; 4339 desc.latency = 0; 4340 param2 = &desc; 4341 break; 4342 4343 case AudioSystem::INPUT_CLOSED: 4344 default: 4345 break; 4346 } 4347 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4348} 4349 4350void AudioFlinger::RecordThread::readInputParameters() 4351{ 4352 delete mRsmpInBuffer; 4353 // mRsmpInBuffer is always assigned a new[] below 4354 delete mRsmpOutBuffer; 4355 mRsmpOutBuffer = NULL; 4356 delete mResampler; 4357 mResampler = NULL; 4358 4359 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4360 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4361 mChannelCount = (uint16_t)popcount(mChannelMask); 4362 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4363 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4364 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4365 mFrameCount = mInputBytes / mFrameSize; 4366 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4367 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4368 4369 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4370 { 4371 int channelCount; 4372 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4373 // stereo to mono post process as the resampler always outputs stereo. 4374 if (mChannelCount == 1 && mReqChannelCount == 2) { 4375 channelCount = 1; 4376 } else { 4377 channelCount = 2; 4378 } 4379 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4380 mResampler->setSampleRate(mSampleRate); 4381 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4382 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4383 4384 // optmization: if mono to mono, alter input frame count as if we were inputing 4385 // stereo samples 4386 if (mChannelCount == 1 && mReqChannelCount == 1) { 4387 mFrameCount >>= 1; 4388 } 4389 4390 } 4391 mRsmpInIndex = mFrameCount; 4392} 4393 4394unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4395{ 4396 Mutex::Autolock _l(mLock); 4397 if (initCheck() != NO_ERROR) { 4398 return 0; 4399 } 4400 4401 return mInput->stream->get_input_frames_lost(mInput->stream); 4402} 4403 4404uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4405{ 4406 Mutex::Autolock _l(mLock); 4407 uint32_t result = 0; 4408 if (getEffectChain_l(sessionId) != 0) { 4409 result = EFFECT_SESSION; 4410 } 4411 4412 for (size_t i = 0; i < mTracks.size(); ++i) { 4413 if (sessionId == mTracks[i]->sessionId()) { 4414 result |= TRACK_SESSION; 4415 break; 4416 } 4417 } 4418 4419 return result; 4420} 4421 4422KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4423{ 4424 KeyedVector<int, bool> ids; 4425 Mutex::Autolock _l(mLock); 4426 for (size_t j = 0; j < mTracks.size(); ++j) { 4427 sp<RecordThread::RecordTrack> track = mTracks[j]; 4428 int sessionId = track->sessionId(); 4429 if (ids.indexOfKey(sessionId) < 0) { 4430 ids.add(sessionId, true); 4431 } 4432 } 4433 return ids; 4434} 4435 4436AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4437{ 4438 Mutex::Autolock _l(mLock); 4439 AudioStreamIn *input = mInput; 4440 mInput = NULL; 4441 return input; 4442} 4443 4444// this method must always be called either with ThreadBase mLock held or inside the thread loop 4445audio_stream_t* AudioFlinger::RecordThread::stream() const 4446{ 4447 if (mInput == NULL) { 4448 return NULL; 4449 } 4450 return &mInput->stream->common; 4451} 4452 4453status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4454{ 4455 // only one chain per input thread 4456 if (mEffectChains.size() != 0) { 4457 return INVALID_OPERATION; 4458 } 4459 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4460 4461 chain->setInBuffer(NULL); 4462 chain->setOutBuffer(NULL); 4463 4464 checkSuspendOnAddEffectChain_l(chain); 4465 4466 mEffectChains.add(chain); 4467 4468 return NO_ERROR; 4469} 4470 4471size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4472{ 4473 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4474 ALOGW_IF(mEffectChains.size() != 1, 4475 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4476 chain.get(), mEffectChains.size(), this); 4477 if (mEffectChains.size() == 1) { 4478 mEffectChains.removeAt(0); 4479 } 4480 return 0; 4481} 4482 4483}; // namespace android 4484