Threads.cpp revision 5736c35b841de56ce394b4879389f669b61425e6
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include <math.h> 24#include <fcntl.h> 25#include <sys/stat.h> 26#include <cutils/properties.h> 27#include <cutils/compiler.h> 28#include <utils/Log.h> 29#include <utils/Trace.h> 30 31#include <private/media/AudioTrackShared.h> 32#include <hardware/audio.h> 33#include <audio_effects/effect_ns.h> 34#include <audio_effects/effect_aec.h> 35#include <audio_utils/primitives.h> 36 37// NBAIO implementations 38#include <media/nbaio/AudioStreamOutSink.h> 39#include <media/nbaio/MonoPipe.h> 40#include <media/nbaio/MonoPipeReader.h> 41#include <media/nbaio/Pipe.h> 42#include <media/nbaio/PipeReader.h> 43#include <media/nbaio/SourceAudioBufferProvider.h> 44 45#include <powermanager/PowerManager.h> 46 47#include <common_time/cc_helper.h> 48#include <common_time/local_clock.h> 49 50#include "AudioFlinger.h" 51#include "AudioMixer.h" 52#include "FastMixer.h" 53#include "ServiceUtilities.h" 54#include "SchedulingPolicyService.h" 55 56#undef ADD_BATTERY_DATA 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait for setParameters to complete 102static const nsecs_t kSetParametersTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal mix buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalMixBufferSizeMs = 20; 111// maximum normal mix buffer size 112static const uint32_t kMaxNormalMixBufferSizeMs = 24; 113 114// Whether to use fast mixer 115static const enum { 116 FastMixer_Never, // never initialize or use: for debugging only 117 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 118 // normal mixer multiplier is 1 119 FastMixer_Static, // initialize if needed, then use all the time if initialized, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 // FIXME for FastMixer_Dynamic: 124 // Supporting this option will require fixing HALs that can't handle large writes. 125 // For example, one HAL implementation returns an error from a large write, 126 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 127 // We could either fix the HAL implementations, or provide a wrapper that breaks 128 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 129} kUseFastMixer = FastMixer_Static; 130 131// Priorities for requestPriority 132static const int kPriorityAudioApp = 2; 133static const int kPriorityFastMixer = 3; 134 135// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 136// for the track. The client then sub-divides this into smaller buffers for its use. 137// Currently the client uses double-buffering by default, but doesn't tell us about that. 138// So for now we just assume that client is double-buffered. 139// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 140// N-buffering, so AudioFlinger could allocate the right amount of memory. 141// See the client's minBufCount and mNotificationFramesAct calculations for details. 142static const int kFastTrackMultiplier = 2; 143 144// ---------------------------------------------------------------------------- 145 146#ifdef ADD_BATTERY_DATA 147// To collect the amplifier usage 148static void addBatteryData(uint32_t params) { 149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150 if (service == NULL) { 151 // it already logged 152 return; 153 } 154 155 service->addBatteryData(params); 156} 157#endif 158 159 160// ---------------------------------------------------------------------------- 161// CPU Stats 162// ---------------------------------------------------------------------------- 163 164class CpuStats { 165public: 166 CpuStats(); 167 void sample(const String8 &title); 168#ifdef DEBUG_CPU_USAGE 169private: 170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172 173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174 175 int mCpuNum; // thread's current CPU number 176 int mCpukHz; // frequency of thread's current CPU in kHz 177#endif 178}; 179 180CpuStats::CpuStats() 181#ifdef DEBUG_CPU_USAGE 182 : mCpuNum(-1), mCpukHz(-1) 183#endif 184{ 185} 186 187void CpuStats::sample(const String8 &title) { 188#ifdef DEBUG_CPU_USAGE 189 // get current thread's delta CPU time in wall clock ns 190 double wcNs; 191 bool valid = mCpuUsage.sampleAndEnable(wcNs); 192 193 // record sample for wall clock statistics 194 if (valid) { 195 mWcStats.sample(wcNs); 196 } 197 198 // get the current CPU number 199 int cpuNum = sched_getcpu(); 200 201 // get the current CPU frequency in kHz 202 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203 204 // check if either CPU number or frequency changed 205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 206 mCpuNum = cpuNum; 207 mCpukHz = cpukHz; 208 // ignore sample for purposes of cycles 209 valid = false; 210 } 211 212 // if no change in CPU number or frequency, then record sample for cycle statistics 213 if (valid && mCpukHz > 0) { 214 double cycles = wcNs * cpukHz * 0.000001; 215 mHzStats.sample(cycles); 216 } 217 218 unsigned n = mWcStats.n(); 219 // mCpuUsage.elapsed() is expensive, so don't call it every loop 220 if ((n & 127) == 1) { 221 long long elapsed = mCpuUsage.elapsed(); 222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 223 double perLoop = elapsed / (double) n; 224 double perLoop100 = perLoop * 0.01; 225 double perLoop1k = perLoop * 0.001; 226 double mean = mWcStats.mean(); 227 double stddev = mWcStats.stddev(); 228 double minimum = mWcStats.minimum(); 229 double maximum = mWcStats.maximum(); 230 double meanCycles = mHzStats.mean(); 231 double stddevCycles = mHzStats.stddev(); 232 double minCycles = mHzStats.minimum(); 233 double maxCycles = mHzStats.maximum(); 234 mCpuUsage.resetElapsed(); 235 mWcStats.reset(); 236 mHzStats.reset(); 237 ALOGD("CPU usage for %s over past %.1f secs\n" 238 " (%u mixer loops at %.1f mean ms per loop):\n" 239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 242 title.string(), 243 elapsed * .000000001, n, perLoop * .000001, 244 mean * .001, 245 stddev * .001, 246 minimum * .001, 247 maximum * .001, 248 mean / perLoop100, 249 stddev / perLoop100, 250 minimum / perLoop100, 251 maximum / perLoop100, 252 meanCycles / perLoop1k, 253 stddevCycles / perLoop1k, 254 minCycles / perLoop1k, 255 maxCycles / perLoop1k); 256 257 } 258 } 259#endif 260}; 261 262// ---------------------------------------------------------------------------- 263// ThreadBase 264// ---------------------------------------------------------------------------- 265 266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 268 : Thread(false /*canCallJava*/), 269 mType(type), 270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 271 // mChannelMask 272 mChannelCount(0), 273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 mParamCond.broadcast(); 285 // do not lock the mutex in destructor 286 releaseWakeLock_l(); 287 if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 377 if (err != 0) { 378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 379 "error %d", 380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 381 } 382 } break; 383 case CFG_EVENT_IO: { 384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 385 mAudioFlinger->mLock.lock(); 386 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 387 mAudioFlinger->mLock.unlock(); 388 } break; 389 default: 390 ALOGE("processConfigEvents() unknown event type %d", event->type()); 391 break; 392 } 393 delete event; 394 mLock.lock(); 395 } 396 mLock.unlock(); 397} 398 399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 400{ 401 const size_t SIZE = 256; 402 char buffer[SIZE]; 403 String8 result; 404 405 bool locked = AudioFlinger::dumpTryLock(mLock); 406 if (!locked) { 407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 408 write(fd, buffer, strlen(buffer)); 409 } 410 411 snprintf(buffer, SIZE, "io handle: %d\n", mId); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 430 result.append(buffer); 431 432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 433 result.append(buffer); 434 result.append(" Index Command"); 435 for (size_t i = 0; i < mNewParameters.size(); ++i) { 436 snprintf(buffer, SIZE, "\n %02d ", i); 437 result.append(buffer); 438 result.append(mNewParameters[i]); 439 } 440 441 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 442 result.append(buffer); 443 for (size_t i = 0; i < mConfigEvents.size(); i++) { 444 mConfigEvents[i]->dump(buffer, SIZE); 445 result.append(buffer); 446 } 447 result.append("\n"); 448 449 write(fd, result.string(), result.size()); 450 451 if (locked) { 452 mLock.unlock(); 453 } 454} 455 456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 457{ 458 const size_t SIZE = 256; 459 char buffer[SIZE]; 460 String8 result; 461 462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 463 write(fd, buffer, strlen(buffer)); 464 465 for (size_t i = 0; i < mEffectChains.size(); ++i) { 466 sp<EffectChain> chain = mEffectChains[i]; 467 if (chain != 0) { 468 chain->dump(fd, args); 469 } 470 } 471} 472 473void AudioFlinger::ThreadBase::acquireWakeLock() 474{ 475 Mutex::Autolock _l(mLock); 476 acquireWakeLock_l(); 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock_l() 480{ 481 if (mPowerManager == 0) { 482 // use checkService() to avoid blocking if power service is not up yet 483 sp<IBinder> binder = 484 defaultServiceManager()->checkService(String16("power")); 485 if (binder == 0) { 486 ALOGW("Thread %s cannot connect to the power manager service", mName); 487 } else { 488 mPowerManager = interface_cast<IPowerManager>(binder); 489 binder->linkToDeath(mDeathRecipient); 490 } 491 } 492 if (mPowerManager != 0) { 493 sp<IBinder> binder = new BBinder(); 494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 495 binder, 496 String16(mName)); 497 if (status == NO_ERROR) { 498 mWakeLockToken = binder; 499 } 500 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 501 } 502} 503 504void AudioFlinger::ThreadBase::releaseWakeLock() 505{ 506 Mutex::Autolock _l(mLock); 507 releaseWakeLock_l(); 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock_l() 511{ 512 if (mWakeLockToken != 0) { 513 ALOGV("releaseWakeLock_l() %s", mName); 514 if (mPowerManager != 0) { 515 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 516 } 517 mWakeLockToken.clear(); 518 } 519} 520 521void AudioFlinger::ThreadBase::clearPowerManager() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525 mPowerManager.clear(); 526} 527 528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 529{ 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 thread->clearPowerManager(); 533 } 534 ALOGW("power manager service died !!!"); 535} 536 537void AudioFlinger::ThreadBase::setEffectSuspended( 538 const effect_uuid_t *type, bool suspend, int sessionId) 539{ 540 Mutex::Autolock _l(mLock); 541 setEffectSuspended_l(type, suspend, sessionId); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended_l( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 sp<EffectChain> chain = getEffectChain_l(sessionId); 548 if (chain != 0) { 549 if (type != NULL) { 550 chain->setEffectSuspended_l(type, suspend); 551 } else { 552 chain->setEffectSuspendedAll_l(suspend); 553 } 554 } 555 556 updateSuspendedSessions_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 560{ 561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 562 if (index < 0) { 563 return; 564 } 565 566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 567 mSuspendedSessions.valueAt(index); 568 569 for (size_t i = 0; i < sessionEffects.size(); i++) { 570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 571 for (int j = 0; j < desc->mRefCount; j++) { 572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 573 chain->setEffectSuspendedAll_l(true); 574 } else { 575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 576 desc->mType.timeLow); 577 chain->setEffectSuspended_l(&desc->mType, true); 578 } 579 } 580 } 581} 582 583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId) 586{ 587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 588 589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 590 591 if (suspend) { 592 if (index >= 0) { 593 sessionEffects = mSuspendedSessions.valueAt(index); 594 } else { 595 mSuspendedSessions.add(sessionId, sessionEffects); 596 } 597 } else { 598 if (index < 0) { 599 return; 600 } 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } 603 604 605 int key = EffectChain::kKeyForSuspendAll; 606 if (type != NULL) { 607 key = type->timeLow; 608 } 609 index = sessionEffects.indexOfKey(key); 610 611 sp<SuspendedSessionDesc> desc; 612 if (suspend) { 613 if (index >= 0) { 614 desc = sessionEffects.valueAt(index); 615 } else { 616 desc = new SuspendedSessionDesc(); 617 if (type != NULL) { 618 desc->mType = *type; 619 } 620 sessionEffects.add(key, desc); 621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 622 } 623 desc->mRefCount++; 624 } else { 625 if (index < 0) { 626 return; 627 } 628 desc = sessionEffects.valueAt(index); 629 if (--desc->mRefCount == 0) { 630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 631 sessionEffects.removeItemsAt(index); 632 if (sessionEffects.isEmpty()) { 633 ALOGV("updateSuspendedSessions_l() restore removing session %d", 634 sessionId); 635 mSuspendedSessions.removeItem(sessionId); 636 } 637 } 638 } 639 if (!sessionEffects.isEmpty()) { 640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 641 } 642} 643 644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 645 bool enabled, 646 int sessionId) 647{ 648 Mutex::Autolock _l(mLock); 649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 if (mType != RECORD) { 657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 658 // another session. This gives the priority to well behaved effect control panels 659 // and applications not using global effects. 660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 661 // global effects 662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 664 } 665 } 666 667 sp<EffectChain> chain = getEffectChain_l(sessionId); 668 if (chain != 0) { 669 chain->checkSuspendOnEffectEnabled(effect, enabled); 670 } 671} 672 673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 675 const sp<AudioFlinger::Client>& client, 676 const sp<IEffectClient>& effectClient, 677 int32_t priority, 678 int sessionId, 679 effect_descriptor_t *desc, 680 int *enabled, 681 status_t *status 682 ) 683{ 684 sp<EffectModule> effect; 685 sp<EffectHandle> handle; 686 status_t lStatus; 687 sp<EffectChain> chain; 688 bool chainCreated = false; 689 bool effectCreated = false; 690 bool effectRegistered = false; 691 692 lStatus = initCheck(); 693 if (lStatus != NO_ERROR) { 694 ALOGW("createEffect_l() Audio driver not initialized."); 695 goto Exit; 696 } 697 698 // Do not allow effects with session ID 0 on direct output or duplicating threads 699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 702 desc->name, sessionId); 703 lStatus = BAD_VALUE; 704 goto Exit; 705 } 706 // Only Pre processor effects are allowed on input threads and only on input threads 707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 709 desc->name, desc->flags, mType); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 715 716 { // scope for mLock 717 Mutex::Autolock _l(mLock); 718 719 // check for existing effect chain with the requested audio session 720 chain = getEffectChain_l(sessionId); 721 if (chain == 0) { 722 // create a new chain for this session 723 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 724 chain = new EffectChain(this, sessionId); 725 addEffectChain_l(chain); 726 chain->setStrategy(getStrategyForSession_l(sessionId)); 727 chainCreated = true; 728 } else { 729 effect = chain->getEffectFromDesc_l(desc); 730 } 731 732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 733 734 if (effect == 0) { 735 int id = mAudioFlinger->nextUniqueId(); 736 // Check CPU and memory usage 737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 738 if (lStatus != NO_ERROR) { 739 goto Exit; 740 } 741 effectRegistered = true; 742 // create a new effect module if none present in the chain 743 effect = new EffectModule(this, chain, desc, id, sessionId); 744 lStatus = effect->status(); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 lStatus = chain->addEffect_l(effect); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 effectCreated = true; 753 754 effect->setDevice(mOutDevice); 755 effect->setDevice(mInDevice); 756 effect->setMode(mAudioFlinger->getMode()); 757 effect->setAudioSource(mAudioSource); 758 } 759 // create effect handle and connect it to effect module 760 handle = new EffectHandle(effect, client, effectClient, priority); 761 lStatus = effect->addHandle(handle.get()); 762 if (enabled != NULL) { 763 *enabled = (int)effect->isEnabled(); 764 } 765 } 766 767Exit: 768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 769 Mutex::Autolock _l(mLock); 770 if (effectCreated) { 771 chain->removeEffect_l(effect); 772 } 773 if (effectRegistered) { 774 AudioSystem::unregisterEffect(effect->id()); 775 } 776 if (chainCreated) { 777 removeEffectChain_l(chain); 778 } 779 handle.clear(); 780 } 781 782 if (status != NULL) { 783 *status = lStatus; 784 } 785 return handle; 786} 787 788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 789{ 790 Mutex::Autolock _l(mLock); 791 return getEffect_l(sessionId, effectId); 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 795{ 796 sp<EffectChain> chain = getEffectChain_l(sessionId); 797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 798} 799 800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 801// PlaybackThread::mLock held 802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 803{ 804 // check for existing effect chain with the requested audio session 805 int sessionId = effect->sessionId(); 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 bool chainCreated = false; 808 809 if (chain == 0) { 810 // create a new chain for this session 811 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 812 chain = new EffectChain(this, sessionId); 813 addEffectChain_l(chain); 814 chain->setStrategy(getStrategyForSession_l(sessionId)); 815 chainCreated = true; 816 } 817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 818 819 if (chain->getEffectFromId_l(effect->id()) != 0) { 820 ALOGW("addEffect_l() %p effect %s already present in chain %p", 821 this, effect->desc().name, chain.get()); 822 return BAD_VALUE; 823 } 824 825 status_t status = chain->addEffect_l(effect); 826 if (status != NO_ERROR) { 827 if (chainCreated) { 828 removeEffectChain_l(chain); 829 } 830 return status; 831 } 832 833 effect->setDevice(mOutDevice); 834 effect->setDevice(mInDevice); 835 effect->setMode(mAudioFlinger->getMode()); 836 effect->setAudioSource(mAudioSource); 837 return NO_ERROR; 838} 839 840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 841 842 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 843 effect_descriptor_t desc = effect->desc(); 844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 845 detachAuxEffect_l(effect->id()); 846 } 847 848 sp<EffectChain> chain = effect->chain().promote(); 849 if (chain != 0) { 850 // remove effect chain if removing last effect 851 if (chain->removeEffect_l(effect) == 0) { 852 removeEffectChain_l(chain); 853 } 854 } else { 855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 856 } 857} 858 859void AudioFlinger::ThreadBase::lockEffectChains_l( 860 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 861{ 862 effectChains = mEffectChains; 863 for (size_t i = 0; i < mEffectChains.size(); i++) { 864 mEffectChains[i]->lock(); 865 } 866} 867 868void AudioFlinger::ThreadBase::unlockEffectChains( 869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 870{ 871 for (size_t i = 0; i < effectChains.size(); i++) { 872 effectChains[i]->unlock(); 873 } 874} 875 876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 877{ 878 Mutex::Autolock _l(mLock); 879 return getEffectChain_l(sessionId); 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 883{ 884 size_t size = mEffectChains.size(); 885 for (size_t i = 0; i < size; i++) { 886 if (mEffectChains[i]->sessionId() == sessionId) { 887 return mEffectChains[i]; 888 } 889 } 890 return 0; 891} 892 893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 894{ 895 Mutex::Autolock _l(mLock); 896 size_t size = mEffectChains.size(); 897 for (size_t i = 0; i < size; i++) { 898 mEffectChains[i]->setMode_l(mode); 899 } 900} 901 902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 903 EffectHandle *handle, 904 bool unpinIfLast) { 905 906 Mutex::Autolock _l(mLock); 907 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 908 // delete the effect module if removing last handle on it 909 if (effect->removeHandle(handle) == 0) { 910 if (!effect->isPinned() || unpinIfLast) { 911 removeEffect_l(effect); 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 } 915} 916 917// ---------------------------------------------------------------------------- 918// Playback 919// ---------------------------------------------------------------------------- 920 921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 922 AudioStreamOut* output, 923 audio_io_handle_t id, 924 audio_devices_t device, 925 type_t type) 926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 928 // mStreamTypes[] initialized in constructor body 929 mOutput(output), 930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 931 mMixerStatus(MIXER_IDLE), 932 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 934 mScreenState(AudioFlinger::mScreenState), 935 // index 0 is reserved for normal mixer's submix 936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 937{ 938 snprintf(mName, kNameLength, "AudioOut_%X", id); 939 940 // Assumes constructor is called by AudioFlinger with it's mLock held, but 941 // it would be safer to explicitly pass initial masterVolume/masterMute as 942 // parameter. 943 // 944 // If the HAL we are using has support for master volume or master mute, 945 // then do not attenuate or mute during mixing (just leave the volume at 1.0 946 // and the mute set to false). 947 mMasterVolume = audioFlinger->masterVolume_l(); 948 mMasterMute = audioFlinger->masterMute_l(); 949 if (mOutput && mOutput->audioHwDev) { 950 if (mOutput->audioHwDev->canSetMasterVolume()) { 951 mMasterVolume = 1.0; 952 } 953 954 if (mOutput->audioHwDev->canSetMasterMute()) { 955 mMasterMute = false; 956 } 957 } 958 959 readOutputParameters(); 960 961 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 962 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 963 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 964 stream = (audio_stream_type_t) (stream + 1)) { 965 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 966 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 967 } 968 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 969 // because mAudioFlinger doesn't have one to copy from 970} 971 972AudioFlinger::PlaybackThread::~PlaybackThread() 973{ 974 delete [] mMixBuffer; 975} 976 977void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 978{ 979 dumpInternals(fd, args); 980 dumpTracks(fd, args); 981 dumpEffectChains(fd, args); 982} 983 984void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 985{ 986 const size_t SIZE = 256; 987 char buffer[SIZE]; 988 String8 result; 989 990 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 991 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 992 const stream_type_t *st = &mStreamTypes[i]; 993 if (i > 0) { 994 result.appendFormat(", "); 995 } 996 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 997 if (st->mute) { 998 result.append("M"); 999 } 1000 } 1001 result.append("\n"); 1002 write(fd, result.string(), result.length()); 1003 result.clear(); 1004 1005 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1006 result.append(buffer); 1007 Track::appendDumpHeader(result); 1008 for (size_t i = 0; i < mTracks.size(); ++i) { 1009 sp<Track> track = mTracks[i]; 1010 if (track != 0) { 1011 track->dump(buffer, SIZE); 1012 result.append(buffer); 1013 } 1014 } 1015 1016 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1017 result.append(buffer); 1018 Track::appendDumpHeader(result); 1019 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1020 sp<Track> track = mActiveTracks[i].promote(); 1021 if (track != 0) { 1022 track->dump(buffer, SIZE); 1023 result.append(buffer); 1024 } 1025 } 1026 write(fd, result.string(), result.size()); 1027 1028 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1029 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1030 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1031 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1032} 1033 1034void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1035{ 1036 const size_t SIZE = 256; 1037 char buffer[SIZE]; 1038 String8 result; 1039 1040 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1041 result.append(buffer); 1042 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1043 ns2ms(systemTime() - mLastWriteTime)); 1044 result.append(buffer); 1045 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1054 result.append(buffer); 1055 write(fd, result.string(), result.size()); 1056 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1057 1058 dumpBase(fd, args); 1059} 1060 1061// Thread virtuals 1062status_t AudioFlinger::PlaybackThread::readyToRun() 1063{ 1064 status_t status = initCheck(); 1065 if (status == NO_ERROR) { 1066 ALOGI("AudioFlinger's thread %p ready to run", this); 1067 } else { 1068 ALOGE("No working audio driver found."); 1069 } 1070 return status; 1071} 1072 1073void AudioFlinger::PlaybackThread::onFirstRef() 1074{ 1075 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1076} 1077 1078// ThreadBase virtuals 1079void AudioFlinger::PlaybackThread::preExit() 1080{ 1081 ALOGV(" preExit()"); 1082 // FIXME this is using hard-coded strings but in the future, this functionality will be 1083 // converted to use audio HAL extensions required to support tunneling 1084 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1085} 1086 1087// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1088sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1089 const sp<AudioFlinger::Client>& client, 1090 audio_stream_type_t streamType, 1091 uint32_t sampleRate, 1092 audio_format_t format, 1093 audio_channel_mask_t channelMask, 1094 size_t frameCount, 1095 const sp<IMemory>& sharedBuffer, 1096 int sessionId, 1097 IAudioFlinger::track_flags_t *flags, 1098 pid_t tid, 1099 status_t *status) 1100{ 1101 sp<Track> track; 1102 status_t lStatus; 1103 1104 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1105 1106 // client expresses a preference for FAST, but we get the final say 1107 if (*flags & IAudioFlinger::TRACK_FAST) { 1108 if ( 1109 // not timed 1110 (!isTimed) && 1111 // either of these use cases: 1112 ( 1113 // use case 1: shared buffer with any frame count 1114 ( 1115 (sharedBuffer != 0) 1116 ) || 1117 // use case 2: callback handler and frame count is default or at least as large as HAL 1118 ( 1119 (tid != -1) && 1120 ((frameCount == 0) || 1121 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1122 ) 1123 ) && 1124 // PCM data 1125 audio_is_linear_pcm(format) && 1126 // mono or stereo 1127 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1128 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1129#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1130 // hardware sample rate 1131 (sampleRate == mSampleRate) && 1132#endif 1133 // normal mixer has an associated fast mixer 1134 hasFastMixer() && 1135 // there are sufficient fast track slots available 1136 (mFastTrackAvailMask != 0) 1137 // FIXME test that MixerThread for this fast track has a capable output HAL 1138 // FIXME add a permission test also? 1139 ) { 1140 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1141 if (frameCount == 0) { 1142 frameCount = mFrameCount * kFastTrackMultiplier; 1143 } 1144 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1145 frameCount, mFrameCount); 1146 } else { 1147 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1148 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1149 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1150 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1151 audio_is_linear_pcm(format), 1152 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1153 *flags &= ~IAudioFlinger::TRACK_FAST; 1154 // For compatibility with AudioTrack calculation, buffer depth is forced 1155 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1156 // This is probably too conservative, but legacy application code may depend on it. 1157 // If you change this calculation, also review the start threshold which is related. 1158 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1159 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1160 if (minBufCount < 2) { 1161 minBufCount = 2; 1162 } 1163 size_t minFrameCount = mNormalFrameCount * minBufCount; 1164 if (frameCount < minFrameCount) { 1165 frameCount = minFrameCount; 1166 } 1167 } 1168 } 1169 1170 if (mType == DIRECT) { 1171 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1172 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1173 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1174 "for output %p with format %d", 1175 sampleRate, format, channelMask, mOutput, mFormat); 1176 lStatus = BAD_VALUE; 1177 goto Exit; 1178 } 1179 } 1180 } else { 1181 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1182 if (sampleRate > mSampleRate*2) { 1183 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1184 lStatus = BAD_VALUE; 1185 goto Exit; 1186 } 1187 } 1188 1189 lStatus = initCheck(); 1190 if (lStatus != NO_ERROR) { 1191 ALOGE("Audio driver not initialized."); 1192 goto Exit; 1193 } 1194 1195 { // scope for mLock 1196 Mutex::Autolock _l(mLock); 1197 1198 // all tracks in same audio session must share the same routing strategy otherwise 1199 // conflicts will happen when tracks are moved from one output to another by audio policy 1200 // manager 1201 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1202 for (size_t i = 0; i < mTracks.size(); ++i) { 1203 sp<Track> t = mTracks[i]; 1204 if (t != 0 && !t->isOutputTrack()) { 1205 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1206 if (sessionId == t->sessionId() && strategy != actual) { 1207 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1208 strategy, actual); 1209 lStatus = BAD_VALUE; 1210 goto Exit; 1211 } 1212 } 1213 } 1214 1215 if (!isTimed) { 1216 track = new Track(this, client, streamType, sampleRate, format, 1217 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1218 } else { 1219 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1220 channelMask, frameCount, sharedBuffer, sessionId); 1221 } 1222 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1223 lStatus = NO_MEMORY; 1224 goto Exit; 1225 } 1226 mTracks.add(track); 1227 1228 sp<EffectChain> chain = getEffectChain_l(sessionId); 1229 if (chain != 0) { 1230 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1231 track->setMainBuffer(chain->inBuffer()); 1232 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1233 chain->incTrackCnt(); 1234 } 1235 1236 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1237 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1238 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1239 // so ask activity manager to do this on our behalf 1240 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1241 } 1242 } 1243 1244 lStatus = NO_ERROR; 1245 1246Exit: 1247 if (status) { 1248 *status = lStatus; 1249 } 1250 return track; 1251} 1252 1253uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1254{ 1255 return latency; 1256} 1257 1258uint32_t AudioFlinger::PlaybackThread::latency() const 1259{ 1260 Mutex::Autolock _l(mLock); 1261 return latency_l(); 1262} 1263uint32_t AudioFlinger::PlaybackThread::latency_l() const 1264{ 1265 if (initCheck() == NO_ERROR) { 1266 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1267 } else { 1268 return 0; 1269 } 1270} 1271 1272void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1273{ 1274 Mutex::Autolock _l(mLock); 1275 // Don't apply master volume in SW if our HAL can do it for us. 1276 if (mOutput && mOutput->audioHwDev && 1277 mOutput->audioHwDev->canSetMasterVolume()) { 1278 mMasterVolume = 1.0; 1279 } else { 1280 mMasterVolume = value; 1281 } 1282} 1283 1284void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1285{ 1286 Mutex::Autolock _l(mLock); 1287 // Don't apply master mute in SW if our HAL can do it for us. 1288 if (mOutput && mOutput->audioHwDev && 1289 mOutput->audioHwDev->canSetMasterMute()) { 1290 mMasterMute = false; 1291 } else { 1292 mMasterMute = muted; 1293 } 1294} 1295 1296void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1297{ 1298 Mutex::Autolock _l(mLock); 1299 mStreamTypes[stream].volume = value; 1300} 1301 1302void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1303{ 1304 Mutex::Autolock _l(mLock); 1305 mStreamTypes[stream].mute = muted; 1306} 1307 1308float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1309{ 1310 Mutex::Autolock _l(mLock); 1311 return mStreamTypes[stream].volume; 1312} 1313 1314// addTrack_l() must be called with ThreadBase::mLock held 1315status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1316{ 1317 status_t status = ALREADY_EXISTS; 1318 1319 // set retry count for buffer fill 1320 track->mRetryCount = kMaxTrackStartupRetries; 1321 if (mActiveTracks.indexOf(track) < 0) { 1322 // the track is newly added, make sure it fills up all its 1323 // buffers before playing. This is to ensure the client will 1324 // effectively get the latency it requested. 1325 track->mFillingUpStatus = Track::FS_FILLING; 1326 track->mResetDone = false; 1327 track->mPresentationCompleteFrames = 0; 1328 mActiveTracks.add(track); 1329 if (track->mainBuffer() != mMixBuffer) { 1330 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1331 if (chain != 0) { 1332 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1333 track->sessionId()); 1334 chain->incActiveTrackCnt(); 1335 } 1336 } 1337 1338 status = NO_ERROR; 1339 } 1340 1341 ALOGV("mWaitWorkCV.broadcast"); 1342 mWaitWorkCV.broadcast(); 1343 1344 return status; 1345} 1346 1347// destroyTrack_l() must be called with ThreadBase::mLock held 1348void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1349{ 1350 track->mState = TrackBase::TERMINATED; 1351 // active tracks are removed by threadLoop() 1352 if (mActiveTracks.indexOf(track) < 0) { 1353 removeTrack_l(track); 1354 } 1355} 1356 1357void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1358{ 1359 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1360 mTracks.remove(track); 1361 deleteTrackName_l(track->name()); 1362 // redundant as track is about to be destroyed, for dumpsys only 1363 track->mName = -1; 1364 if (track->isFastTrack()) { 1365 int index = track->mFastIndex; 1366 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1367 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1368 mFastTrackAvailMask |= 1 << index; 1369 // redundant as track is about to be destroyed, for dumpsys only 1370 track->mFastIndex = -1; 1371 } 1372 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1373 if (chain != 0) { 1374 chain->decTrackCnt(); 1375 } 1376} 1377 1378String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1379{ 1380 String8 out_s8 = String8(""); 1381 char *s; 1382 1383 Mutex::Autolock _l(mLock); 1384 if (initCheck() != NO_ERROR) { 1385 return out_s8; 1386 } 1387 1388 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1389 out_s8 = String8(s); 1390 free(s); 1391 return out_s8; 1392} 1393 1394// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1395void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1396 AudioSystem::OutputDescriptor desc; 1397 void *param2 = NULL; 1398 1399 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1400 param); 1401 1402 switch (event) { 1403 case AudioSystem::OUTPUT_OPENED: 1404 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1405 desc.channels = mChannelMask; 1406 desc.samplingRate = mSampleRate; 1407 desc.format = mFormat; 1408 desc.frameCount = mNormalFrameCount; // FIXME see 1409 // AudioFlinger::frameCount(audio_io_handle_t) 1410 desc.latency = latency(); 1411 param2 = &desc; 1412 break; 1413 1414 case AudioSystem::STREAM_CONFIG_CHANGED: 1415 param2 = ¶m; 1416 case AudioSystem::OUTPUT_CLOSED: 1417 default: 1418 break; 1419 } 1420 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1421} 1422 1423void AudioFlinger::PlaybackThread::readOutputParameters() 1424{ 1425 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1426 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1427 mChannelCount = (uint16_t)popcount(mChannelMask); 1428 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1429 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1430 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1431 if (mFrameCount & 15) { 1432 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1433 mFrameCount); 1434 } 1435 1436 // Calculate size of normal mix buffer relative to the HAL output buffer size 1437 double multiplier = 1.0; 1438 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1439 kUseFastMixer == FastMixer_Dynamic)) { 1440 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1441 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1442 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1443 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1444 maxNormalFrameCount = maxNormalFrameCount & ~15; 1445 if (maxNormalFrameCount < minNormalFrameCount) { 1446 maxNormalFrameCount = minNormalFrameCount; 1447 } 1448 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1449 if (multiplier <= 1.0) { 1450 multiplier = 1.0; 1451 } else if (multiplier <= 2.0) { 1452 if (2 * mFrameCount <= maxNormalFrameCount) { 1453 multiplier = 2.0; 1454 } else { 1455 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1456 } 1457 } else { 1458 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1459 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1460 // track, but we sometimes have to do this to satisfy the maximum frame count 1461 // constraint) 1462 // FIXME this rounding up should not be done if no HAL SRC 1463 uint32_t truncMult = (uint32_t) multiplier; 1464 if ((truncMult & 1)) { 1465 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1466 ++truncMult; 1467 } 1468 } 1469 multiplier = (double) truncMult; 1470 } 1471 } 1472 mNormalFrameCount = multiplier * mFrameCount; 1473 // round up to nearest 16 frames to satisfy AudioMixer 1474 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1475 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1476 mNormalFrameCount); 1477 1478 delete[] mMixBuffer; 1479 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1480 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1481 1482 // force reconfiguration of effect chains and engines to take new buffer size and audio 1483 // parameters into account 1484 // Note that mLock is not held when readOutputParameters() is called from the constructor 1485 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1486 // matter. 1487 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1488 Vector< sp<EffectChain> > effectChains = mEffectChains; 1489 for (size_t i = 0; i < effectChains.size(); i ++) { 1490 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1491 } 1492} 1493 1494 1495status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1496{ 1497 if (halFrames == NULL || dspFrames == NULL) { 1498 return BAD_VALUE; 1499 } 1500 Mutex::Autolock _l(mLock); 1501 if (initCheck() != NO_ERROR) { 1502 return INVALID_OPERATION; 1503 } 1504 size_t framesWritten = mBytesWritten / mFrameSize; 1505 *halFrames = framesWritten; 1506 1507 if (isSuspended()) { 1508 // return an estimation of rendered frames when the output is suspended 1509 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1510 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1511 return NO_ERROR; 1512 } else { 1513 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1514 } 1515} 1516 1517uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1518{ 1519 Mutex::Autolock _l(mLock); 1520 uint32_t result = 0; 1521 if (getEffectChain_l(sessionId) != 0) { 1522 result = EFFECT_SESSION; 1523 } 1524 1525 for (size_t i = 0; i < mTracks.size(); ++i) { 1526 sp<Track> track = mTracks[i]; 1527 if (sessionId == track->sessionId() && !track->isInvalid()) { 1528 result |= TRACK_SESSION; 1529 break; 1530 } 1531 } 1532 1533 return result; 1534} 1535 1536uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1537{ 1538 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1539 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1540 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1541 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1542 } 1543 for (size_t i = 0; i < mTracks.size(); i++) { 1544 sp<Track> track = mTracks[i]; 1545 if (sessionId == track->sessionId() && !track->isInvalid()) { 1546 return AudioSystem::getStrategyForStream(track->streamType()); 1547 } 1548 } 1549 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1550} 1551 1552 1553AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1554{ 1555 Mutex::Autolock _l(mLock); 1556 return mOutput; 1557} 1558 1559AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1560{ 1561 Mutex::Autolock _l(mLock); 1562 AudioStreamOut *output = mOutput; 1563 mOutput = NULL; 1564 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1565 // must push a NULL and wait for ack 1566 mOutputSink.clear(); 1567 mPipeSink.clear(); 1568 mNormalSink.clear(); 1569 return output; 1570} 1571 1572// this method must always be called either with ThreadBase mLock held or inside the thread loop 1573audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1574{ 1575 if (mOutput == NULL) { 1576 return NULL; 1577 } 1578 return &mOutput->stream->common; 1579} 1580 1581uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1582{ 1583 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1584} 1585 1586status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1587{ 1588 if (!isValidSyncEvent(event)) { 1589 return BAD_VALUE; 1590 } 1591 1592 Mutex::Autolock _l(mLock); 1593 1594 for (size_t i = 0; i < mTracks.size(); ++i) { 1595 sp<Track> track = mTracks[i]; 1596 if (event->triggerSession() == track->sessionId()) { 1597 (void) track->setSyncEvent(event); 1598 return NO_ERROR; 1599 } 1600 } 1601 1602 return NAME_NOT_FOUND; 1603} 1604 1605bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1606{ 1607 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1608} 1609 1610void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1611 const Vector< sp<Track> >& tracksToRemove) 1612{ 1613 size_t count = tracksToRemove.size(); 1614 if (CC_UNLIKELY(count)) { 1615 for (size_t i = 0 ; i < count ; i++) { 1616 const sp<Track>& track = tracksToRemove.itemAt(i); 1617 if ((track->sharedBuffer() != 0) && 1618 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1619 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1620 } 1621 } 1622 } 1623 1624} 1625 1626void AudioFlinger::PlaybackThread::checkSilentMode_l() 1627{ 1628 if (!mMasterMute) { 1629 char value[PROPERTY_VALUE_MAX]; 1630 if (property_get("ro.audio.silent", value, "0") > 0) { 1631 char *endptr; 1632 unsigned long ul = strtoul(value, &endptr, 0); 1633 if (*endptr == '\0' && ul != 0) { 1634 ALOGD("Silence is golden"); 1635 // The setprop command will not allow a property to be changed after 1636 // the first time it is set, so we don't have to worry about un-muting. 1637 setMasterMute_l(true); 1638 } 1639 } 1640 } 1641} 1642 1643// shared by MIXER and DIRECT, overridden by DUPLICATING 1644void AudioFlinger::PlaybackThread::threadLoop_write() 1645{ 1646 // FIXME rewrite to reduce number of system calls 1647 mLastWriteTime = systemTime(); 1648 mInWrite = true; 1649 int bytesWritten; 1650 1651 // If an NBAIO sink is present, use it to write the normal mixer's submix 1652 if (mNormalSink != 0) { 1653#define mBitShift 2 // FIXME 1654 size_t count = mixBufferSize >> mBitShift; 1655 ATRACE_BEGIN("write"); 1656 // update the setpoint when AudioFlinger::mScreenState changes 1657 uint32_t screenState = AudioFlinger::mScreenState; 1658 if (screenState != mScreenState) { 1659 mScreenState = screenState; 1660 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1661 if (pipe != NULL) { 1662 pipe->setAvgFrames((mScreenState & 1) ? 1663 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1664 } 1665 } 1666 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1667 ATRACE_END(); 1668 if (framesWritten > 0) { 1669 bytesWritten = framesWritten << mBitShift; 1670 } else { 1671 bytesWritten = framesWritten; 1672 } 1673 // otherwise use the HAL / AudioStreamOut directly 1674 } else { 1675 // Direct output thread. 1676 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1677 } 1678 1679 if (bytesWritten > 0) { 1680 mBytesWritten += mixBufferSize; 1681 } 1682 mNumWrites++; 1683 mInWrite = false; 1684} 1685 1686/* 1687The derived values that are cached: 1688 - mixBufferSize from frame count * frame size 1689 - activeSleepTime from activeSleepTimeUs() 1690 - idleSleepTime from idleSleepTimeUs() 1691 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1692 - maxPeriod from frame count and sample rate (MIXER only) 1693 1694The parameters that affect these derived values are: 1695 - frame count 1696 - frame size 1697 - sample rate 1698 - device type: A2DP or not 1699 - device latency 1700 - format: PCM or not 1701 - active sleep time 1702 - idle sleep time 1703*/ 1704 1705void AudioFlinger::PlaybackThread::cacheParameters_l() 1706{ 1707 mixBufferSize = mNormalFrameCount * mFrameSize; 1708 activeSleepTime = activeSleepTimeUs(); 1709 idleSleepTime = idleSleepTimeUs(); 1710} 1711 1712void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1713{ 1714 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1715 this, streamType, mTracks.size()); 1716 Mutex::Autolock _l(mLock); 1717 1718 size_t size = mTracks.size(); 1719 for (size_t i = 0; i < size; i++) { 1720 sp<Track> t = mTracks[i]; 1721 if (t->streamType() == streamType) { 1722 t->invalidate(); 1723 } 1724 } 1725} 1726 1727status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1728{ 1729 int session = chain->sessionId(); 1730 int16_t *buffer = mMixBuffer; 1731 bool ownsBuffer = false; 1732 1733 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1734 if (session > 0) { 1735 // Only one effect chain can be present in direct output thread and it uses 1736 // the mix buffer as input 1737 if (mType != DIRECT) { 1738 size_t numSamples = mNormalFrameCount * mChannelCount; 1739 buffer = new int16_t[numSamples]; 1740 memset(buffer, 0, numSamples * sizeof(int16_t)); 1741 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1742 ownsBuffer = true; 1743 } 1744 1745 // Attach all tracks with same session ID to this chain. 1746 for (size_t i = 0; i < mTracks.size(); ++i) { 1747 sp<Track> track = mTracks[i]; 1748 if (session == track->sessionId()) { 1749 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1750 buffer); 1751 track->setMainBuffer(buffer); 1752 chain->incTrackCnt(); 1753 } 1754 } 1755 1756 // indicate all active tracks in the chain 1757 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1758 sp<Track> track = mActiveTracks[i].promote(); 1759 if (track == 0) { 1760 continue; 1761 } 1762 if (session == track->sessionId()) { 1763 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1764 chain->incActiveTrackCnt(); 1765 } 1766 } 1767 } 1768 1769 chain->setInBuffer(buffer, ownsBuffer); 1770 chain->setOutBuffer(mMixBuffer); 1771 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1772 // chains list in order to be processed last as it contains output stage effects 1773 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1774 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1775 // after track specific effects and before output stage 1776 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1777 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1778 // Effect chain for other sessions are inserted at beginning of effect 1779 // chains list to be processed before output mix effects. Relative order between other 1780 // sessions is not important 1781 size_t size = mEffectChains.size(); 1782 size_t i = 0; 1783 for (i = 0; i < size; i++) { 1784 if (mEffectChains[i]->sessionId() < session) { 1785 break; 1786 } 1787 } 1788 mEffectChains.insertAt(chain, i); 1789 checkSuspendOnAddEffectChain_l(chain); 1790 1791 return NO_ERROR; 1792} 1793 1794size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1795{ 1796 int session = chain->sessionId(); 1797 1798 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1799 1800 for (size_t i = 0; i < mEffectChains.size(); i++) { 1801 if (chain == mEffectChains[i]) { 1802 mEffectChains.removeAt(i); 1803 // detach all active tracks from the chain 1804 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1805 sp<Track> track = mActiveTracks[i].promote(); 1806 if (track == 0) { 1807 continue; 1808 } 1809 if (session == track->sessionId()) { 1810 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1811 chain.get(), session); 1812 chain->decActiveTrackCnt(); 1813 } 1814 } 1815 1816 // detach all tracks with same session ID from this chain 1817 for (size_t i = 0; i < mTracks.size(); ++i) { 1818 sp<Track> track = mTracks[i]; 1819 if (session == track->sessionId()) { 1820 track->setMainBuffer(mMixBuffer); 1821 chain->decTrackCnt(); 1822 } 1823 } 1824 break; 1825 } 1826 } 1827 return mEffectChains.size(); 1828} 1829 1830status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1831 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1832{ 1833 Mutex::Autolock _l(mLock); 1834 return attachAuxEffect_l(track, EffectId); 1835} 1836 1837status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1838 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1839{ 1840 status_t status = NO_ERROR; 1841 1842 if (EffectId == 0) { 1843 track->setAuxBuffer(0, NULL); 1844 } else { 1845 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1846 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1847 if (effect != 0) { 1848 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1849 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1850 } else { 1851 status = INVALID_OPERATION; 1852 } 1853 } else { 1854 status = BAD_VALUE; 1855 } 1856 } 1857 return status; 1858} 1859 1860void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1861{ 1862 for (size_t i = 0; i < mTracks.size(); ++i) { 1863 sp<Track> track = mTracks[i]; 1864 if (track->auxEffectId() == effectId) { 1865 attachAuxEffect_l(track, 0); 1866 } 1867 } 1868} 1869 1870bool AudioFlinger::PlaybackThread::threadLoop() 1871{ 1872 Vector< sp<Track> > tracksToRemove; 1873 1874 standbyTime = systemTime(); 1875 1876 // MIXER 1877 nsecs_t lastWarning = 0; 1878 1879 // DUPLICATING 1880 // FIXME could this be made local to while loop? 1881 writeFrames = 0; 1882 1883 cacheParameters_l(); 1884 sleepTime = idleSleepTime; 1885 1886 if (mType == MIXER) { 1887 sleepTimeShift = 0; 1888 } 1889 1890 CpuStats cpuStats; 1891 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1892 1893 acquireWakeLock(); 1894 1895 while (!exitPending()) 1896 { 1897 cpuStats.sample(myName); 1898 1899 Vector< sp<EffectChain> > effectChains; 1900 1901 processConfigEvents(); 1902 1903 { // scope for mLock 1904 1905 Mutex::Autolock _l(mLock); 1906 1907 if (checkForNewParameters_l()) { 1908 cacheParameters_l(); 1909 } 1910 1911 saveOutputTracks(); 1912 1913 // put audio hardware into standby after short delay 1914 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1915 isSuspended())) { 1916 if (!mStandby) { 1917 1918 threadLoop_standby(); 1919 1920 mStandby = true; 1921 } 1922 1923 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1924 // we're about to wait, flush the binder command buffer 1925 IPCThreadState::self()->flushCommands(); 1926 1927 clearOutputTracks(); 1928 1929 if (exitPending()) { 1930 break; 1931 } 1932 1933 releaseWakeLock_l(); 1934 // wait until we have something to do... 1935 ALOGV("%s going to sleep", myName.string()); 1936 mWaitWorkCV.wait(mLock); 1937 ALOGV("%s waking up", myName.string()); 1938 acquireWakeLock_l(); 1939 1940 mMixerStatus = MIXER_IDLE; 1941 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1942 mBytesWritten = 0; 1943 1944 checkSilentMode_l(); 1945 1946 standbyTime = systemTime() + standbyDelay; 1947 sleepTime = idleSleepTime; 1948 if (mType == MIXER) { 1949 sleepTimeShift = 0; 1950 } 1951 1952 continue; 1953 } 1954 } 1955 1956 // mMixerStatusIgnoringFastTracks is also updated internally 1957 mMixerStatus = prepareTracks_l(&tracksToRemove); 1958 1959 // prevent any changes in effect chain list and in each effect chain 1960 // during mixing and effect process as the audio buffers could be deleted 1961 // or modified if an effect is created or deleted 1962 lockEffectChains_l(effectChains); 1963 } 1964 1965 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1966 threadLoop_mix(); 1967 } else { 1968 threadLoop_sleepTime(); 1969 } 1970 1971 if (isSuspended()) { 1972 sleepTime = suspendSleepTimeUs(); 1973 mBytesWritten += mixBufferSize; 1974 } 1975 1976 // only process effects if we're going to write 1977 if (sleepTime == 0) { 1978 for (size_t i = 0; i < effectChains.size(); i ++) { 1979 effectChains[i]->process_l(); 1980 } 1981 } 1982 1983 // enable changes in effect chain 1984 unlockEffectChains(effectChains); 1985 1986 // sleepTime == 0 means we must write to audio hardware 1987 if (sleepTime == 0) { 1988 1989 threadLoop_write(); 1990 1991if (mType == MIXER) { 1992 // write blocked detection 1993 nsecs_t now = systemTime(); 1994 nsecs_t delta = now - mLastWriteTime; 1995 if (!mStandby && delta > maxPeriod) { 1996 mNumDelayedWrites++; 1997 if ((now - lastWarning) > kWarningThrottleNs) { 1998 ATRACE_NAME("underrun"); 1999 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2000 ns2ms(delta), mNumDelayedWrites, this); 2001 lastWarning = now; 2002 } 2003 } 2004} 2005 2006 mStandby = false; 2007 } else { 2008 usleep(sleepTime); 2009 } 2010 2011 // Finally let go of removed track(s), without the lock held 2012 // since we can't guarantee the destructors won't acquire that 2013 // same lock. This will also mutate and push a new fast mixer state. 2014 threadLoop_removeTracks(tracksToRemove); 2015 tracksToRemove.clear(); 2016 2017 // FIXME I don't understand the need for this here; 2018 // it was in the original code but maybe the 2019 // assignment in saveOutputTracks() makes this unnecessary? 2020 clearOutputTracks(); 2021 2022 // Effect chains will be actually deleted here if they were removed from 2023 // mEffectChains list during mixing or effects processing 2024 effectChains.clear(); 2025 2026 // FIXME Note that the above .clear() is no longer necessary since effectChains 2027 // is now local to this block, but will keep it for now (at least until merge done). 2028 } 2029 2030 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2031 if (mType == MIXER || mType == DIRECT) { 2032 // put output stream into standby mode 2033 if (!mStandby) { 2034 mOutput->stream->common.standby(&mOutput->stream->common); 2035 } 2036 } 2037 2038 releaseWakeLock(); 2039 2040 ALOGV("Thread %p type %d exiting", this, mType); 2041 return false; 2042} 2043 2044 2045// ---------------------------------------------------------------------------- 2046 2047AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2048 audio_io_handle_t id, audio_devices_t device, type_t type) 2049 : PlaybackThread(audioFlinger, output, id, device, type), 2050 // mAudioMixer below 2051 // mFastMixer below 2052 mFastMixerFutex(0) 2053 // mOutputSink below 2054 // mPipeSink below 2055 // mNormalSink below 2056{ 2057 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2058 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2059 "mFrameCount=%d, mNormalFrameCount=%d", 2060 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2061 mNormalFrameCount); 2062 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2063 2064 // FIXME - Current mixer implementation only supports stereo output 2065 if (mChannelCount != FCC_2) { 2066 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2067 } 2068 2069 // create an NBAIO sink for the HAL output stream, and negotiate 2070 mOutputSink = new AudioStreamOutSink(output->stream); 2071 size_t numCounterOffers = 0; 2072 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2073 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2074 ALOG_ASSERT(index == 0); 2075 2076 // initialize fast mixer depending on configuration 2077 bool initFastMixer; 2078 switch (kUseFastMixer) { 2079 case FastMixer_Never: 2080 initFastMixer = false; 2081 break; 2082 case FastMixer_Always: 2083 initFastMixer = true; 2084 break; 2085 case FastMixer_Static: 2086 case FastMixer_Dynamic: 2087 initFastMixer = mFrameCount < mNormalFrameCount; 2088 break; 2089 } 2090 if (initFastMixer) { 2091 2092 // create a MonoPipe to connect our submix to FastMixer 2093 NBAIO_Format format = mOutputSink->format(); 2094 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2095 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2096 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2097 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2098 const NBAIO_Format offers[1] = {format}; 2099 size_t numCounterOffers = 0; 2100 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2101 ALOG_ASSERT(index == 0); 2102 monoPipe->setAvgFrames((mScreenState & 1) ? 2103 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2104 mPipeSink = monoPipe; 2105 2106#ifdef TEE_SINK_FRAMES 2107 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2108 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2109 numCounterOffers = 0; 2110 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2111 ALOG_ASSERT(index == 0); 2112 mTeeSink = teeSink; 2113 PipeReader *teeSource = new PipeReader(*teeSink); 2114 numCounterOffers = 0; 2115 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2116 ALOG_ASSERT(index == 0); 2117 mTeeSource = teeSource; 2118#endif 2119 2120 // create fast mixer and configure it initially with just one fast track for our submix 2121 mFastMixer = new FastMixer(); 2122 FastMixerStateQueue *sq = mFastMixer->sq(); 2123#ifdef STATE_QUEUE_DUMP 2124 sq->setObserverDump(&mStateQueueObserverDump); 2125 sq->setMutatorDump(&mStateQueueMutatorDump); 2126#endif 2127 FastMixerState *state = sq->begin(); 2128 FastTrack *fastTrack = &state->mFastTracks[0]; 2129 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2130 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2131 fastTrack->mVolumeProvider = NULL; 2132 fastTrack->mGeneration++; 2133 state->mFastTracksGen++; 2134 state->mTrackMask = 1; 2135 // fast mixer will use the HAL output sink 2136 state->mOutputSink = mOutputSink.get(); 2137 state->mOutputSinkGen++; 2138 state->mFrameCount = mFrameCount; 2139 state->mCommand = FastMixerState::COLD_IDLE; 2140 // already done in constructor initialization list 2141 //mFastMixerFutex = 0; 2142 state->mColdFutexAddr = &mFastMixerFutex; 2143 state->mColdGen++; 2144 state->mDumpState = &mFastMixerDumpState; 2145 state->mTeeSink = mTeeSink.get(); 2146 sq->end(); 2147 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2148 2149 // start the fast mixer 2150 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2151 pid_t tid = mFastMixer->getTid(); 2152 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2153 if (err != 0) { 2154 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2155 kPriorityFastMixer, getpid_cached, tid, err); 2156 } 2157 2158#ifdef AUDIO_WATCHDOG 2159 // create and start the watchdog 2160 mAudioWatchdog = new AudioWatchdog(); 2161 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2162 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2163 tid = mAudioWatchdog->getTid(); 2164 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2165 if (err != 0) { 2166 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2167 kPriorityFastMixer, getpid_cached, tid, err); 2168 } 2169#endif 2170 2171 } else { 2172 mFastMixer = NULL; 2173 } 2174 2175 switch (kUseFastMixer) { 2176 case FastMixer_Never: 2177 case FastMixer_Dynamic: 2178 mNormalSink = mOutputSink; 2179 break; 2180 case FastMixer_Always: 2181 mNormalSink = mPipeSink; 2182 break; 2183 case FastMixer_Static: 2184 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2185 break; 2186 } 2187} 2188 2189AudioFlinger::MixerThread::~MixerThread() 2190{ 2191 if (mFastMixer != NULL) { 2192 FastMixerStateQueue *sq = mFastMixer->sq(); 2193 FastMixerState *state = sq->begin(); 2194 if (state->mCommand == FastMixerState::COLD_IDLE) { 2195 int32_t old = android_atomic_inc(&mFastMixerFutex); 2196 if (old == -1) { 2197 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2198 } 2199 } 2200 state->mCommand = FastMixerState::EXIT; 2201 sq->end(); 2202 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2203 mFastMixer->join(); 2204 // Though the fast mixer thread has exited, it's state queue is still valid. 2205 // We'll use that extract the final state which contains one remaining fast track 2206 // corresponding to our sub-mix. 2207 state = sq->begin(); 2208 ALOG_ASSERT(state->mTrackMask == 1); 2209 FastTrack *fastTrack = &state->mFastTracks[0]; 2210 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2211 delete fastTrack->mBufferProvider; 2212 sq->end(false /*didModify*/); 2213 delete mFastMixer; 2214#ifdef AUDIO_WATCHDOG 2215 if (mAudioWatchdog != 0) { 2216 mAudioWatchdog->requestExit(); 2217 mAudioWatchdog->requestExitAndWait(); 2218 mAudioWatchdog.clear(); 2219 } 2220#endif 2221 } 2222 delete mAudioMixer; 2223} 2224 2225 2226uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2227{ 2228 if (mFastMixer != NULL) { 2229 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2230 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2231 } 2232 return latency; 2233} 2234 2235 2236void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2237{ 2238 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2239} 2240 2241void AudioFlinger::MixerThread::threadLoop_write() 2242{ 2243 // FIXME we should only do one push per cycle; confirm this is true 2244 // Start the fast mixer if it's not already running 2245 if (mFastMixer != NULL) { 2246 FastMixerStateQueue *sq = mFastMixer->sq(); 2247 FastMixerState *state = sq->begin(); 2248 if (state->mCommand != FastMixerState::MIX_WRITE && 2249 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2250 if (state->mCommand == FastMixerState::COLD_IDLE) { 2251 int32_t old = android_atomic_inc(&mFastMixerFutex); 2252 if (old == -1) { 2253 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2254 } 2255#ifdef AUDIO_WATCHDOG 2256 if (mAudioWatchdog != 0) { 2257 mAudioWatchdog->resume(); 2258 } 2259#endif 2260 } 2261 state->mCommand = FastMixerState::MIX_WRITE; 2262 sq->end(); 2263 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2264 if (kUseFastMixer == FastMixer_Dynamic) { 2265 mNormalSink = mPipeSink; 2266 } 2267 } else { 2268 sq->end(false /*didModify*/); 2269 } 2270 } 2271 PlaybackThread::threadLoop_write(); 2272} 2273 2274void AudioFlinger::MixerThread::threadLoop_standby() 2275{ 2276 // Idle the fast mixer if it's currently running 2277 if (mFastMixer != NULL) { 2278 FastMixerStateQueue *sq = mFastMixer->sq(); 2279 FastMixerState *state = sq->begin(); 2280 if (!(state->mCommand & FastMixerState::IDLE)) { 2281 state->mCommand = FastMixerState::COLD_IDLE; 2282 state->mColdFutexAddr = &mFastMixerFutex; 2283 state->mColdGen++; 2284 mFastMixerFutex = 0; 2285 sq->end(); 2286 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2287 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2288 if (kUseFastMixer == FastMixer_Dynamic) { 2289 mNormalSink = mOutputSink; 2290 } 2291#ifdef AUDIO_WATCHDOG 2292 if (mAudioWatchdog != 0) { 2293 mAudioWatchdog->pause(); 2294 } 2295#endif 2296 } else { 2297 sq->end(false /*didModify*/); 2298 } 2299 } 2300 PlaybackThread::threadLoop_standby(); 2301} 2302 2303// shared by MIXER and DIRECT, overridden by DUPLICATING 2304void AudioFlinger::PlaybackThread::threadLoop_standby() 2305{ 2306 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2307 mOutput->stream->common.standby(&mOutput->stream->common); 2308} 2309 2310void AudioFlinger::MixerThread::threadLoop_mix() 2311{ 2312 // obtain the presentation timestamp of the next output buffer 2313 int64_t pts; 2314 status_t status = INVALID_OPERATION; 2315 2316 if (mNormalSink != 0) { 2317 status = mNormalSink->getNextWriteTimestamp(&pts); 2318 } else { 2319 status = mOutputSink->getNextWriteTimestamp(&pts); 2320 } 2321 2322 if (status != NO_ERROR) { 2323 pts = AudioBufferProvider::kInvalidPTS; 2324 } 2325 2326 // mix buffers... 2327 mAudioMixer->process(pts); 2328 // increase sleep time progressively when application underrun condition clears. 2329 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2330 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2331 // such that we would underrun the audio HAL. 2332 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2333 sleepTimeShift--; 2334 } 2335 sleepTime = 0; 2336 standbyTime = systemTime() + standbyDelay; 2337 //TODO: delay standby when effects have a tail 2338} 2339 2340void AudioFlinger::MixerThread::threadLoop_sleepTime() 2341{ 2342 // If no tracks are ready, sleep once for the duration of an output 2343 // buffer size, then write 0s to the output 2344 if (sleepTime == 0) { 2345 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2346 sleepTime = activeSleepTime >> sleepTimeShift; 2347 if (sleepTime < kMinThreadSleepTimeUs) { 2348 sleepTime = kMinThreadSleepTimeUs; 2349 } 2350 // reduce sleep time in case of consecutive application underruns to avoid 2351 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2352 // duration we would end up writing less data than needed by the audio HAL if 2353 // the condition persists. 2354 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2355 sleepTimeShift++; 2356 } 2357 } else { 2358 sleepTime = idleSleepTime; 2359 } 2360 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2361 memset (mMixBuffer, 0, mixBufferSize); 2362 sleepTime = 0; 2363 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2364 "anticipated start"); 2365 } 2366 // TODO add standby time extension fct of effect tail 2367} 2368 2369// prepareTracks_l() must be called with ThreadBase::mLock held 2370AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2371 Vector< sp<Track> > *tracksToRemove) 2372{ 2373 2374 mixer_state mixerStatus = MIXER_IDLE; 2375 // find out which tracks need to be processed 2376 size_t count = mActiveTracks.size(); 2377 size_t mixedTracks = 0; 2378 size_t tracksWithEffect = 0; 2379 // counts only _active_ fast tracks 2380 size_t fastTracks = 0; 2381 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2382 2383 float masterVolume = mMasterVolume; 2384 bool masterMute = mMasterMute; 2385 2386 if (masterMute) { 2387 masterVolume = 0; 2388 } 2389 // Delegate master volume control to effect in output mix effect chain if needed 2390 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2391 if (chain != 0) { 2392 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2393 chain->setVolume_l(&v, &v); 2394 masterVolume = (float)((v + (1 << 23)) >> 24); 2395 chain.clear(); 2396 } 2397 2398 // prepare a new state to push 2399 FastMixerStateQueue *sq = NULL; 2400 FastMixerState *state = NULL; 2401 bool didModify = false; 2402 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2403 if (mFastMixer != NULL) { 2404 sq = mFastMixer->sq(); 2405 state = sq->begin(); 2406 } 2407 2408 for (size_t i=0 ; i<count ; i++) { 2409 sp<Track> t = mActiveTracks[i].promote(); 2410 if (t == 0) { 2411 continue; 2412 } 2413 2414 // this const just means the local variable doesn't change 2415 Track* const track = t.get(); 2416 2417 // process fast tracks 2418 if (track->isFastTrack()) { 2419 2420 // It's theoretically possible (though unlikely) for a fast track to be created 2421 // and then removed within the same normal mix cycle. This is not a problem, as 2422 // the track never becomes active so it's fast mixer slot is never touched. 2423 // The converse, of removing an (active) track and then creating a new track 2424 // at the identical fast mixer slot within the same normal mix cycle, 2425 // is impossible because the slot isn't marked available until the end of each cycle. 2426 int j = track->mFastIndex; 2427 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2428 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2429 FastTrack *fastTrack = &state->mFastTracks[j]; 2430 2431 // Determine whether the track is currently in underrun condition, 2432 // and whether it had a recent underrun. 2433 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2434 FastTrackUnderruns underruns = ftDump->mUnderruns; 2435 uint32_t recentFull = (underruns.mBitFields.mFull - 2436 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2437 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2438 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2439 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2440 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2441 uint32_t recentUnderruns = recentPartial + recentEmpty; 2442 track->mObservedUnderruns = underruns; 2443 // don't count underruns that occur while stopping or pausing 2444 // or stopped which can occur when flush() is called while active 2445 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2446 track->mUnderrunCount += recentUnderruns; 2447 } 2448 2449 // This is similar to the state machine for normal tracks, 2450 // with a few modifications for fast tracks. 2451 bool isActive = true; 2452 switch (track->mState) { 2453 case TrackBase::STOPPING_1: 2454 // track stays active in STOPPING_1 state until first underrun 2455 if (recentUnderruns > 0) { 2456 track->mState = TrackBase::STOPPING_2; 2457 } 2458 break; 2459 case TrackBase::PAUSING: 2460 // ramp down is not yet implemented 2461 track->setPaused(); 2462 break; 2463 case TrackBase::RESUMING: 2464 // ramp up is not yet implemented 2465 track->mState = TrackBase::ACTIVE; 2466 break; 2467 case TrackBase::ACTIVE: 2468 if (recentFull > 0 || recentPartial > 0) { 2469 // track has provided at least some frames recently: reset retry count 2470 track->mRetryCount = kMaxTrackRetries; 2471 } 2472 if (recentUnderruns == 0) { 2473 // no recent underruns: stay active 2474 break; 2475 } 2476 // there has recently been an underrun of some kind 2477 if (track->sharedBuffer() == 0) { 2478 // were any of the recent underruns "empty" (no frames available)? 2479 if (recentEmpty == 0) { 2480 // no, then ignore the partial underruns as they are allowed indefinitely 2481 break; 2482 } 2483 // there has recently been an "empty" underrun: decrement the retry counter 2484 if (--(track->mRetryCount) > 0) { 2485 break; 2486 } 2487 // indicate to client process that the track was disabled because of underrun; 2488 // it will then automatically call start() when data is available 2489 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2490 // remove from active list, but state remains ACTIVE [confusing but true] 2491 isActive = false; 2492 break; 2493 } 2494 // fall through 2495 case TrackBase::STOPPING_2: 2496 case TrackBase::PAUSED: 2497 case TrackBase::TERMINATED: 2498 case TrackBase::STOPPED: 2499 case TrackBase::FLUSHED: // flush() while active 2500 // Check for presentation complete if track is inactive 2501 // We have consumed all the buffers of this track. 2502 // This would be incomplete if we auto-paused on underrun 2503 { 2504 size_t audioHALFrames = 2505 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2506 size_t framesWritten = mBytesWritten / mFrameSize; 2507 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2508 // track stays in active list until presentation is complete 2509 break; 2510 } 2511 } 2512 if (track->isStopping_2()) { 2513 track->mState = TrackBase::STOPPED; 2514 } 2515 if (track->isStopped()) { 2516 // Can't reset directly, as fast mixer is still polling this track 2517 // track->reset(); 2518 // So instead mark this track as needing to be reset after push with ack 2519 resetMask |= 1 << i; 2520 } 2521 isActive = false; 2522 break; 2523 case TrackBase::IDLE: 2524 default: 2525 LOG_FATAL("unexpected track state %d", track->mState); 2526 } 2527 2528 if (isActive) { 2529 // was it previously inactive? 2530 if (!(state->mTrackMask & (1 << j))) { 2531 ExtendedAudioBufferProvider *eabp = track; 2532 VolumeProvider *vp = track; 2533 fastTrack->mBufferProvider = eabp; 2534 fastTrack->mVolumeProvider = vp; 2535 fastTrack->mSampleRate = track->mSampleRate; 2536 fastTrack->mChannelMask = track->mChannelMask; 2537 fastTrack->mGeneration++; 2538 state->mTrackMask |= 1 << j; 2539 didModify = true; 2540 // no acknowledgement required for newly active tracks 2541 } 2542 // cache the combined master volume and stream type volume for fast mixer; this 2543 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2544 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2545 ++fastTracks; 2546 } else { 2547 // was it previously active? 2548 if (state->mTrackMask & (1 << j)) { 2549 fastTrack->mBufferProvider = NULL; 2550 fastTrack->mGeneration++; 2551 state->mTrackMask &= ~(1 << j); 2552 didModify = true; 2553 // If any fast tracks were removed, we must wait for acknowledgement 2554 // because we're about to decrement the last sp<> on those tracks. 2555 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2556 } else { 2557 LOG_FATAL("fast track %d should have been active", j); 2558 } 2559 tracksToRemove->add(track); 2560 // Avoids a misleading display in dumpsys 2561 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2562 } 2563 continue; 2564 } 2565 2566 { // local variable scope to avoid goto warning 2567 2568 audio_track_cblk_t* cblk = track->cblk(); 2569 2570 // The first time a track is added we wait 2571 // for all its buffers to be filled before processing it 2572 int name = track->name(); 2573 // make sure that we have enough frames to mix one full buffer. 2574 // enforce this condition only once to enable draining the buffer in case the client 2575 // app does not call stop() and relies on underrun to stop: 2576 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2577 // during last round 2578 uint32_t minFrames = 1; 2579 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2580 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2581 if (t->sampleRate() == mSampleRate) { 2582 minFrames = mNormalFrameCount; 2583 } else { 2584 // +1 for rounding and +1 for additional sample needed for interpolation 2585 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2586 // add frames already consumed but not yet released by the resampler 2587 // because cblk->framesReady() will include these frames 2588 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2589 // the minimum track buffer size is normally twice the number of frames necessary 2590 // to fill one buffer and the resampler should not leave more than one buffer worth 2591 // of unreleased frames after each pass, but just in case... 2592 ALOG_ASSERT(minFrames <= cblk->frameCount); 2593 } 2594 } 2595 if ((track->framesReady() >= minFrames) && track->isReady() && 2596 !track->isPaused() && !track->isTerminated()) 2597 { 2598 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2599 this); 2600 2601 mixedTracks++; 2602 2603 // track->mainBuffer() != mMixBuffer means there is an effect chain 2604 // connected to the track 2605 chain.clear(); 2606 if (track->mainBuffer() != mMixBuffer) { 2607 chain = getEffectChain_l(track->sessionId()); 2608 // Delegate volume control to effect in track effect chain if needed 2609 if (chain != 0) { 2610 tracksWithEffect++; 2611 } else { 2612 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2613 "session %d", 2614 name, track->sessionId()); 2615 } 2616 } 2617 2618 2619 int param = AudioMixer::VOLUME; 2620 if (track->mFillingUpStatus == Track::FS_FILLED) { 2621 // no ramp for the first volume setting 2622 track->mFillingUpStatus = Track::FS_ACTIVE; 2623 if (track->mState == TrackBase::RESUMING) { 2624 track->mState = TrackBase::ACTIVE; 2625 param = AudioMixer::RAMP_VOLUME; 2626 } 2627 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2628 } else if (cblk->server != 0) { 2629 // If the track is stopped before the first frame was mixed, 2630 // do not apply ramp 2631 param = AudioMixer::RAMP_VOLUME; 2632 } 2633 2634 // compute volume for this track 2635 uint32_t vl, vr, va; 2636 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2637 vl = vr = va = 0; 2638 if (track->isPausing()) { 2639 track->setPaused(); 2640 } 2641 } else { 2642 2643 // read original volumes with volume control 2644 float typeVolume = mStreamTypes[track->streamType()].volume; 2645 float v = masterVolume * typeVolume; 2646 uint32_t vlr = cblk->getVolumeLR(); 2647 vl = vlr & 0xFFFF; 2648 vr = vlr >> 16; 2649 // track volumes come from shared memory, so can't be trusted and must be clamped 2650 if (vl > MAX_GAIN_INT) { 2651 ALOGV("Track left volume out of range: %04X", vl); 2652 vl = MAX_GAIN_INT; 2653 } 2654 if (vr > MAX_GAIN_INT) { 2655 ALOGV("Track right volume out of range: %04X", vr); 2656 vr = MAX_GAIN_INT; 2657 } 2658 // now apply the master volume and stream type volume 2659 vl = (uint32_t)(v * vl) << 12; 2660 vr = (uint32_t)(v * vr) << 12; 2661 // assuming master volume and stream type volume each go up to 1.0, 2662 // vl and vr are now in 8.24 format 2663 2664 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2665 // send level comes from shared memory and so may be corrupt 2666 if (sendLevel > MAX_GAIN_INT) { 2667 ALOGV("Track send level out of range: %04X", sendLevel); 2668 sendLevel = MAX_GAIN_INT; 2669 } 2670 va = (uint32_t)(v * sendLevel); 2671 } 2672 // Delegate volume control to effect in track effect chain if needed 2673 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2674 // Do not ramp volume if volume is controlled by effect 2675 param = AudioMixer::VOLUME; 2676 track->mHasVolumeController = true; 2677 } else { 2678 // force no volume ramp when volume controller was just disabled or removed 2679 // from effect chain to avoid volume spike 2680 if (track->mHasVolumeController) { 2681 param = AudioMixer::VOLUME; 2682 } 2683 track->mHasVolumeController = false; 2684 } 2685 2686 // Convert volumes from 8.24 to 4.12 format 2687 // This additional clamping is needed in case chain->setVolume_l() overshot 2688 vl = (vl + (1 << 11)) >> 12; 2689 if (vl > MAX_GAIN_INT) { 2690 vl = MAX_GAIN_INT; 2691 } 2692 vr = (vr + (1 << 11)) >> 12; 2693 if (vr > MAX_GAIN_INT) { 2694 vr = MAX_GAIN_INT; 2695 } 2696 2697 if (va > MAX_GAIN_INT) { 2698 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2699 } 2700 2701 // XXX: these things DON'T need to be done each time 2702 mAudioMixer->setBufferProvider(name, track); 2703 mAudioMixer->enable(name); 2704 2705 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2706 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2707 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2708 mAudioMixer->setParameter( 2709 name, 2710 AudioMixer::TRACK, 2711 AudioMixer::FORMAT, (void *)track->format()); 2712 mAudioMixer->setParameter( 2713 name, 2714 AudioMixer::TRACK, 2715 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2716 mAudioMixer->setParameter( 2717 name, 2718 AudioMixer::RESAMPLE, 2719 AudioMixer::SAMPLE_RATE, 2720 (void *)(cblk->sampleRate)); 2721 mAudioMixer->setParameter( 2722 name, 2723 AudioMixer::TRACK, 2724 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2725 mAudioMixer->setParameter( 2726 name, 2727 AudioMixer::TRACK, 2728 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2729 2730 // reset retry count 2731 track->mRetryCount = kMaxTrackRetries; 2732 2733 // If one track is ready, set the mixer ready if: 2734 // - the mixer was not ready during previous round OR 2735 // - no other track is not ready 2736 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2737 mixerStatus != MIXER_TRACKS_ENABLED) { 2738 mixerStatus = MIXER_TRACKS_READY; 2739 } 2740 } else { 2741 // clear effect chain input buffer if an active track underruns to avoid sending 2742 // previous audio buffer again to effects 2743 chain = getEffectChain_l(track->sessionId()); 2744 if (chain != 0) { 2745 chain->clearInputBuffer(); 2746 } 2747 2748 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2749 cblk->server, this); 2750 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2751 track->isStopped() || track->isPaused()) { 2752 // We have consumed all the buffers of this track. 2753 // Remove it from the list of active tracks. 2754 // TODO: use actual buffer filling status instead of latency when available from 2755 // audio HAL 2756 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2757 size_t framesWritten = mBytesWritten / mFrameSize; 2758 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2759 if (track->isStopped()) { 2760 track->reset(); 2761 } 2762 tracksToRemove->add(track); 2763 } 2764 } else { 2765 track->mUnderrunCount++; 2766 // No buffers for this track. Give it a few chances to 2767 // fill a buffer, then remove it from active list. 2768 if (--(track->mRetryCount) <= 0) { 2769 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2770 tracksToRemove->add(track); 2771 // indicate to client process that the track was disabled because of underrun; 2772 // it will then automatically call start() when data is available 2773 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2774 // If one track is not ready, mark the mixer also not ready if: 2775 // - the mixer was ready during previous round OR 2776 // - no other track is ready 2777 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2778 mixerStatus != MIXER_TRACKS_READY) { 2779 mixerStatus = MIXER_TRACKS_ENABLED; 2780 } 2781 } 2782 mAudioMixer->disable(name); 2783 } 2784 2785 } // local variable scope to avoid goto warning 2786track_is_ready: ; 2787 2788 } 2789 2790 // Push the new FastMixer state if necessary 2791 bool pauseAudioWatchdog = false; 2792 if (didModify) { 2793 state->mFastTracksGen++; 2794 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2795 if (kUseFastMixer == FastMixer_Dynamic && 2796 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2797 state->mCommand = FastMixerState::COLD_IDLE; 2798 state->mColdFutexAddr = &mFastMixerFutex; 2799 state->mColdGen++; 2800 mFastMixerFutex = 0; 2801 if (kUseFastMixer == FastMixer_Dynamic) { 2802 mNormalSink = mOutputSink; 2803 } 2804 // If we go into cold idle, need to wait for acknowledgement 2805 // so that fast mixer stops doing I/O. 2806 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2807 pauseAudioWatchdog = true; 2808 } 2809 sq->end(); 2810 } 2811 if (sq != NULL) { 2812 sq->end(didModify); 2813 sq->push(block); 2814 } 2815#ifdef AUDIO_WATCHDOG 2816 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2817 mAudioWatchdog->pause(); 2818 } 2819#endif 2820 2821 // Now perform the deferred reset on fast tracks that have stopped 2822 while (resetMask != 0) { 2823 size_t i = __builtin_ctz(resetMask); 2824 ALOG_ASSERT(i < count); 2825 resetMask &= ~(1 << i); 2826 sp<Track> t = mActiveTracks[i].promote(); 2827 if (t == 0) { 2828 continue; 2829 } 2830 Track* track = t.get(); 2831 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2832 track->reset(); 2833 } 2834 2835 // remove all the tracks that need to be... 2836 count = tracksToRemove->size(); 2837 if (CC_UNLIKELY(count)) { 2838 for (size_t i=0 ; i<count ; i++) { 2839 const sp<Track>& track = tracksToRemove->itemAt(i); 2840 mActiveTracks.remove(track); 2841 if (track->mainBuffer() != mMixBuffer) { 2842 chain = getEffectChain_l(track->sessionId()); 2843 if (chain != 0) { 2844 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2845 track->sessionId()); 2846 chain->decActiveTrackCnt(); 2847 } 2848 } 2849 if (track->isTerminated()) { 2850 removeTrack_l(track); 2851 } 2852 } 2853 } 2854 2855 // mix buffer must be cleared if all tracks are connected to an 2856 // effect chain as in this case the mixer will not write to 2857 // mix buffer and track effects will accumulate into it 2858 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2859 (mixedTracks == 0 && fastTracks > 0)) { 2860 // FIXME as a performance optimization, should remember previous zero status 2861 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2862 } 2863 2864 // if any fast tracks, then status is ready 2865 mMixerStatusIgnoringFastTracks = mixerStatus; 2866 if (fastTracks > 0) { 2867 mixerStatus = MIXER_TRACKS_READY; 2868 } 2869 return mixerStatus; 2870} 2871 2872// getTrackName_l() must be called with ThreadBase::mLock held 2873int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2874{ 2875 return mAudioMixer->getTrackName(channelMask, sessionId); 2876} 2877 2878// deleteTrackName_l() must be called with ThreadBase::mLock held 2879void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2880{ 2881 ALOGV("remove track (%d) and delete from mixer", name); 2882 mAudioMixer->deleteTrackName(name); 2883} 2884 2885// checkForNewParameters_l() must be called with ThreadBase::mLock held 2886bool AudioFlinger::MixerThread::checkForNewParameters_l() 2887{ 2888 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2889 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2890 bool reconfig = false; 2891 2892 while (!mNewParameters.isEmpty()) { 2893 2894 if (mFastMixer != NULL) { 2895 FastMixerStateQueue *sq = mFastMixer->sq(); 2896 FastMixerState *state = sq->begin(); 2897 if (!(state->mCommand & FastMixerState::IDLE)) { 2898 previousCommand = state->mCommand; 2899 state->mCommand = FastMixerState::HOT_IDLE; 2900 sq->end(); 2901 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2902 } else { 2903 sq->end(false /*didModify*/); 2904 } 2905 } 2906 2907 status_t status = NO_ERROR; 2908 String8 keyValuePair = mNewParameters[0]; 2909 AudioParameter param = AudioParameter(keyValuePair); 2910 int value; 2911 2912 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2913 reconfig = true; 2914 } 2915 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2916 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2917 status = BAD_VALUE; 2918 } else { 2919 reconfig = true; 2920 } 2921 } 2922 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2923 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2924 status = BAD_VALUE; 2925 } else { 2926 reconfig = true; 2927 } 2928 } 2929 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2930 // do not accept frame count changes if tracks are open as the track buffer 2931 // size depends on frame count and correct behavior would not be guaranteed 2932 // if frame count is changed after track creation 2933 if (!mTracks.isEmpty()) { 2934 status = INVALID_OPERATION; 2935 } else { 2936 reconfig = true; 2937 } 2938 } 2939 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2940#ifdef ADD_BATTERY_DATA 2941 // when changing the audio output device, call addBatteryData to notify 2942 // the change 2943 if (mOutDevice != value) { 2944 uint32_t params = 0; 2945 // check whether speaker is on 2946 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2947 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2948 } 2949 2950 audio_devices_t deviceWithoutSpeaker 2951 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2952 // check if any other device (except speaker) is on 2953 if (value & deviceWithoutSpeaker ) { 2954 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2955 } 2956 2957 if (params != 0) { 2958 addBatteryData(params); 2959 } 2960 } 2961#endif 2962 2963 // forward device change to effects that have requested to be 2964 // aware of attached audio device. 2965 mOutDevice = value; 2966 for (size_t i = 0; i < mEffectChains.size(); i++) { 2967 mEffectChains[i]->setDevice_l(mOutDevice); 2968 } 2969 } 2970 2971 if (status == NO_ERROR) { 2972 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2973 keyValuePair.string()); 2974 if (!mStandby && status == INVALID_OPERATION) { 2975 mOutput->stream->common.standby(&mOutput->stream->common); 2976 mStandby = true; 2977 mBytesWritten = 0; 2978 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2979 keyValuePair.string()); 2980 } 2981 if (status == NO_ERROR && reconfig) { 2982 delete mAudioMixer; 2983 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2984 mAudioMixer = NULL; 2985 readOutputParameters(); 2986 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2987 for (size_t i = 0; i < mTracks.size() ; i++) { 2988 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 2989 if (name < 0) { 2990 break; 2991 } 2992 mTracks[i]->mName = name; 2993 // limit track sample rate to 2 x new output sample rate 2994 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2995 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2996 } 2997 } 2998 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2999 } 3000 } 3001 3002 mNewParameters.removeAt(0); 3003 3004 mParamStatus = status; 3005 mParamCond.signal(); 3006 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3007 // already timed out waiting for the status and will never signal the condition. 3008 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3009 } 3010 3011 if (!(previousCommand & FastMixerState::IDLE)) { 3012 ALOG_ASSERT(mFastMixer != NULL); 3013 FastMixerStateQueue *sq = mFastMixer->sq(); 3014 FastMixerState *state = sq->begin(); 3015 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3016 state->mCommand = previousCommand; 3017 sq->end(); 3018 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3019 } 3020 3021 return reconfig; 3022} 3023 3024 3025void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3026{ 3027 const size_t SIZE = 256; 3028 char buffer[SIZE]; 3029 String8 result; 3030 3031 PlaybackThread::dumpInternals(fd, args); 3032 3033 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3034 result.append(buffer); 3035 write(fd, result.string(), result.size()); 3036 3037 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3038 FastMixerDumpState copy = mFastMixerDumpState; 3039 copy.dump(fd); 3040 3041#ifdef STATE_QUEUE_DUMP 3042 // Similar for state queue 3043 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3044 observerCopy.dump(fd); 3045 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3046 mutatorCopy.dump(fd); 3047#endif 3048 3049 // Write the tee output to a .wav file 3050 dumpTee(fd, mTeeSource, mId); 3051 3052#ifdef AUDIO_WATCHDOG 3053 if (mAudioWatchdog != 0) { 3054 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3055 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3056 wdCopy.dump(fd); 3057 } 3058#endif 3059} 3060 3061uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3062{ 3063 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3064} 3065 3066uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3067{ 3068 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3069} 3070 3071void AudioFlinger::MixerThread::cacheParameters_l() 3072{ 3073 PlaybackThread::cacheParameters_l(); 3074 3075 // FIXME: Relaxed timing because of a certain device that can't meet latency 3076 // Should be reduced to 2x after the vendor fixes the driver issue 3077 // increase threshold again due to low power audio mode. The way this warning 3078 // threshold is calculated and its usefulness should be reconsidered anyway. 3079 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3080} 3081 3082// ---------------------------------------------------------------------------- 3083 3084AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3085 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3086 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3087 // mLeftVolFloat, mRightVolFloat 3088{ 3089} 3090 3091AudioFlinger::DirectOutputThread::~DirectOutputThread() 3092{ 3093} 3094 3095AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3096 Vector< sp<Track> > *tracksToRemove 3097) 3098{ 3099 sp<Track> trackToRemove; 3100 3101 mixer_state mixerStatus = MIXER_IDLE; 3102 3103 // find out which tracks need to be processed 3104 if (mActiveTracks.size() != 0) { 3105 sp<Track> t = mActiveTracks[0].promote(); 3106 // The track died recently 3107 if (t == 0) { 3108 return MIXER_IDLE; 3109 } 3110 3111 Track* const track = t.get(); 3112 audio_track_cblk_t* cblk = track->cblk(); 3113 3114 // The first time a track is added we wait 3115 // for all its buffers to be filled before processing it 3116 uint32_t minFrames; 3117 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3118 minFrames = mNormalFrameCount; 3119 } else { 3120 minFrames = 1; 3121 } 3122 if ((track->framesReady() >= minFrames) && track->isReady() && 3123 !track->isPaused() && !track->isTerminated()) 3124 { 3125 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3126 3127 if (track->mFillingUpStatus == Track::FS_FILLED) { 3128 track->mFillingUpStatus = Track::FS_ACTIVE; 3129 mLeftVolFloat = mRightVolFloat = 0; 3130 if (track->mState == TrackBase::RESUMING) { 3131 track->mState = TrackBase::ACTIVE; 3132 } 3133 } 3134 3135 // compute volume for this track 3136 float left, right; 3137 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3138 left = right = 0; 3139 if (track->isPausing()) { 3140 track->setPaused(); 3141 } 3142 } else { 3143 float typeVolume = mStreamTypes[track->streamType()].volume; 3144 float v = mMasterVolume * typeVolume; 3145 uint32_t vlr = cblk->getVolumeLR(); 3146 float v_clamped = v * (vlr & 0xFFFF); 3147 if (v_clamped > MAX_GAIN) { 3148 v_clamped = MAX_GAIN; 3149 } 3150 left = v_clamped/MAX_GAIN; 3151 v_clamped = v * (vlr >> 16); 3152 if (v_clamped > MAX_GAIN) { 3153 v_clamped = MAX_GAIN; 3154 } 3155 right = v_clamped/MAX_GAIN; 3156 } 3157 3158 if (left != mLeftVolFloat || right != mRightVolFloat) { 3159 mLeftVolFloat = left; 3160 mRightVolFloat = right; 3161 3162 // Convert volumes from float to 8.24 3163 uint32_t vl = (uint32_t)(left * (1 << 24)); 3164 uint32_t vr = (uint32_t)(right * (1 << 24)); 3165 3166 // Delegate volume control to effect in track effect chain if needed 3167 // only one effect chain can be present on DirectOutputThread, so if 3168 // there is one, the track is connected to it 3169 if (!mEffectChains.isEmpty()) { 3170 // Do not ramp volume if volume is controlled by effect 3171 mEffectChains[0]->setVolume_l(&vl, &vr); 3172 left = (float)vl / (1 << 24); 3173 right = (float)vr / (1 << 24); 3174 } 3175 mOutput->stream->set_volume(mOutput->stream, left, right); 3176 } 3177 3178 // reset retry count 3179 track->mRetryCount = kMaxTrackRetriesDirect; 3180 mActiveTrack = t; 3181 mixerStatus = MIXER_TRACKS_READY; 3182 } else { 3183 // clear effect chain input buffer if an active track underruns to avoid sending 3184 // previous audio buffer again to effects 3185 if (!mEffectChains.isEmpty()) { 3186 mEffectChains[0]->clearInputBuffer(); 3187 } 3188 3189 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3190 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3191 track->isStopped() || track->isPaused()) { 3192 // We have consumed all the buffers of this track. 3193 // Remove it from the list of active tracks. 3194 // TODO: implement behavior for compressed audio 3195 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3196 size_t framesWritten = mBytesWritten / mFrameSize; 3197 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3198 if (track->isStopped()) { 3199 track->reset(); 3200 } 3201 trackToRemove = track; 3202 } 3203 } else { 3204 // No buffers for this track. Give it a few chances to 3205 // fill a buffer, then remove it from active list. 3206 if (--(track->mRetryCount) <= 0) { 3207 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3208 trackToRemove = track; 3209 } else { 3210 mixerStatus = MIXER_TRACKS_ENABLED; 3211 } 3212 } 3213 } 3214 } 3215 3216 // FIXME merge this with similar code for removing multiple tracks 3217 // remove all the tracks that need to be... 3218 if (CC_UNLIKELY(trackToRemove != 0)) { 3219 tracksToRemove->add(trackToRemove); 3220 mActiveTracks.remove(trackToRemove); 3221 if (!mEffectChains.isEmpty()) { 3222 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3223 trackToRemove->sessionId()); 3224 mEffectChains[0]->decActiveTrackCnt(); 3225 } 3226 if (trackToRemove->isTerminated()) { 3227 removeTrack_l(trackToRemove); 3228 } 3229 } 3230 3231 return mixerStatus; 3232} 3233 3234void AudioFlinger::DirectOutputThread::threadLoop_mix() 3235{ 3236 AudioBufferProvider::Buffer buffer; 3237 size_t frameCount = mFrameCount; 3238 int8_t *curBuf = (int8_t *)mMixBuffer; 3239 // output audio to hardware 3240 while (frameCount) { 3241 buffer.frameCount = frameCount; 3242 mActiveTrack->getNextBuffer(&buffer); 3243 if (CC_UNLIKELY(buffer.raw == NULL)) { 3244 memset(curBuf, 0, frameCount * mFrameSize); 3245 break; 3246 } 3247 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3248 frameCount -= buffer.frameCount; 3249 curBuf += buffer.frameCount * mFrameSize; 3250 mActiveTrack->releaseBuffer(&buffer); 3251 } 3252 sleepTime = 0; 3253 standbyTime = systemTime() + standbyDelay; 3254 mActiveTrack.clear(); 3255 3256} 3257 3258void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3259{ 3260 if (sleepTime == 0) { 3261 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3262 sleepTime = activeSleepTime; 3263 } else { 3264 sleepTime = idleSleepTime; 3265 } 3266 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3267 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3268 sleepTime = 0; 3269 } 3270} 3271 3272// getTrackName_l() must be called with ThreadBase::mLock held 3273int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3274 int sessionId) 3275{ 3276 return 0; 3277} 3278 3279// deleteTrackName_l() must be called with ThreadBase::mLock held 3280void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3281{ 3282} 3283 3284// checkForNewParameters_l() must be called with ThreadBase::mLock held 3285bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3286{ 3287 bool reconfig = false; 3288 3289 while (!mNewParameters.isEmpty()) { 3290 status_t status = NO_ERROR; 3291 String8 keyValuePair = mNewParameters[0]; 3292 AudioParameter param = AudioParameter(keyValuePair); 3293 int value; 3294 3295 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3296 // do not accept frame count changes if tracks are open as the track buffer 3297 // size depends on frame count and correct behavior would not be garantied 3298 // if frame count is changed after track creation 3299 if (!mTracks.isEmpty()) { 3300 status = INVALID_OPERATION; 3301 } else { 3302 reconfig = true; 3303 } 3304 } 3305 if (status == NO_ERROR) { 3306 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3307 keyValuePair.string()); 3308 if (!mStandby && status == INVALID_OPERATION) { 3309 mOutput->stream->common.standby(&mOutput->stream->common); 3310 mStandby = true; 3311 mBytesWritten = 0; 3312 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3313 keyValuePair.string()); 3314 } 3315 if (status == NO_ERROR && reconfig) { 3316 readOutputParameters(); 3317 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3318 } 3319 } 3320 3321 mNewParameters.removeAt(0); 3322 3323 mParamStatus = status; 3324 mParamCond.signal(); 3325 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3326 // already timed out waiting for the status and will never signal the condition. 3327 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3328 } 3329 return reconfig; 3330} 3331 3332uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3333{ 3334 uint32_t time; 3335 if (audio_is_linear_pcm(mFormat)) { 3336 time = PlaybackThread::activeSleepTimeUs(); 3337 } else { 3338 time = 10000; 3339 } 3340 return time; 3341} 3342 3343uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3344{ 3345 uint32_t time; 3346 if (audio_is_linear_pcm(mFormat)) { 3347 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3348 } else { 3349 time = 10000; 3350 } 3351 return time; 3352} 3353 3354uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3355{ 3356 uint32_t time; 3357 if (audio_is_linear_pcm(mFormat)) { 3358 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3359 } else { 3360 time = 10000; 3361 } 3362 return time; 3363} 3364 3365void AudioFlinger::DirectOutputThread::cacheParameters_l() 3366{ 3367 PlaybackThread::cacheParameters_l(); 3368 3369 // use shorter standby delay as on normal output to release 3370 // hardware resources as soon as possible 3371 standbyDelay = microseconds(activeSleepTime*2); 3372} 3373 3374// ---------------------------------------------------------------------------- 3375 3376AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3377 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3378 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3379 DUPLICATING), 3380 mWaitTimeMs(UINT_MAX) 3381{ 3382 addOutputTrack(mainThread); 3383} 3384 3385AudioFlinger::DuplicatingThread::~DuplicatingThread() 3386{ 3387 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3388 mOutputTracks[i]->destroy(); 3389 } 3390} 3391 3392void AudioFlinger::DuplicatingThread::threadLoop_mix() 3393{ 3394 // mix buffers... 3395 if (outputsReady(outputTracks)) { 3396 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3397 } else { 3398 memset(mMixBuffer, 0, mixBufferSize); 3399 } 3400 sleepTime = 0; 3401 writeFrames = mNormalFrameCount; 3402 standbyTime = systemTime() + standbyDelay; 3403} 3404 3405void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3406{ 3407 if (sleepTime == 0) { 3408 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3409 sleepTime = activeSleepTime; 3410 } else { 3411 sleepTime = idleSleepTime; 3412 } 3413 } else if (mBytesWritten != 0) { 3414 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3415 writeFrames = mNormalFrameCount; 3416 memset(mMixBuffer, 0, mixBufferSize); 3417 } else { 3418 // flush remaining overflow buffers in output tracks 3419 writeFrames = 0; 3420 } 3421 sleepTime = 0; 3422 } 3423} 3424 3425void AudioFlinger::DuplicatingThread::threadLoop_write() 3426{ 3427 for (size_t i = 0; i < outputTracks.size(); i++) { 3428 outputTracks[i]->write(mMixBuffer, writeFrames); 3429 } 3430 mBytesWritten += mixBufferSize; 3431} 3432 3433void AudioFlinger::DuplicatingThread::threadLoop_standby() 3434{ 3435 // DuplicatingThread implements standby by stopping all tracks 3436 for (size_t i = 0; i < outputTracks.size(); i++) { 3437 outputTracks[i]->stop(); 3438 } 3439} 3440 3441void AudioFlinger::DuplicatingThread::saveOutputTracks() 3442{ 3443 outputTracks = mOutputTracks; 3444} 3445 3446void AudioFlinger::DuplicatingThread::clearOutputTracks() 3447{ 3448 outputTracks.clear(); 3449} 3450 3451void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3452{ 3453 Mutex::Autolock _l(mLock); 3454 // FIXME explain this formula 3455 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3456 OutputTrack *outputTrack = new OutputTrack(thread, 3457 this, 3458 mSampleRate, 3459 mFormat, 3460 mChannelMask, 3461 frameCount); 3462 if (outputTrack->cblk() != NULL) { 3463 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3464 mOutputTracks.add(outputTrack); 3465 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3466 updateWaitTime_l(); 3467 } 3468} 3469 3470void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3471{ 3472 Mutex::Autolock _l(mLock); 3473 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3474 if (mOutputTracks[i]->thread() == thread) { 3475 mOutputTracks[i]->destroy(); 3476 mOutputTracks.removeAt(i); 3477 updateWaitTime_l(); 3478 return; 3479 } 3480 } 3481 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3482} 3483 3484// caller must hold mLock 3485void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3486{ 3487 mWaitTimeMs = UINT_MAX; 3488 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3489 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3490 if (strong != 0) { 3491 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3492 if (waitTimeMs < mWaitTimeMs) { 3493 mWaitTimeMs = waitTimeMs; 3494 } 3495 } 3496 } 3497} 3498 3499 3500bool AudioFlinger::DuplicatingThread::outputsReady( 3501 const SortedVector< sp<OutputTrack> > &outputTracks) 3502{ 3503 for (size_t i = 0; i < outputTracks.size(); i++) { 3504 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3505 if (thread == 0) { 3506 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3507 outputTracks[i].get()); 3508 return false; 3509 } 3510 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3511 // see note at standby() declaration 3512 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3513 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3514 thread.get()); 3515 return false; 3516 } 3517 } 3518 return true; 3519} 3520 3521uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3522{ 3523 return (mWaitTimeMs * 1000) / 2; 3524} 3525 3526void AudioFlinger::DuplicatingThread::cacheParameters_l() 3527{ 3528 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3529 updateWaitTime_l(); 3530 3531 MixerThread::cacheParameters_l(); 3532} 3533 3534// ---------------------------------------------------------------------------- 3535// Record 3536// ---------------------------------------------------------------------------- 3537 3538AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3539 AudioStreamIn *input, 3540 uint32_t sampleRate, 3541 audio_channel_mask_t channelMask, 3542 audio_io_handle_t id, 3543 audio_devices_t device, 3544 const sp<NBAIO_Sink>& teeSink) : 3545 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 3546 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3547 // mRsmpInIndex and mInputBytes set by readInputParameters() 3548 mReqChannelCount(popcount(channelMask)), 3549 mReqSampleRate(sampleRate), 3550 // mBytesRead is only meaningful while active, and so is cleared in start() 3551 // (but might be better to also clear here for dump?) 3552 mTeeSink(teeSink) 3553{ 3554 snprintf(mName, kNameLength, "AudioIn_%X", id); 3555 3556 readInputParameters(); 3557 3558} 3559 3560 3561AudioFlinger::RecordThread::~RecordThread() 3562{ 3563 delete[] mRsmpInBuffer; 3564 delete mResampler; 3565 delete[] mRsmpOutBuffer; 3566} 3567 3568void AudioFlinger::RecordThread::onFirstRef() 3569{ 3570 run(mName, PRIORITY_URGENT_AUDIO); 3571} 3572 3573status_t AudioFlinger::RecordThread::readyToRun() 3574{ 3575 status_t status = initCheck(); 3576 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3577 return status; 3578} 3579 3580bool AudioFlinger::RecordThread::threadLoop() 3581{ 3582 AudioBufferProvider::Buffer buffer; 3583 sp<RecordTrack> activeTrack; 3584 Vector< sp<EffectChain> > effectChains; 3585 3586 nsecs_t lastWarning = 0; 3587 3588 inputStandBy(); 3589 acquireWakeLock(); 3590 3591 // used to verify we've read at least once before evaluating how many bytes were read 3592 bool readOnce = false; 3593 3594 // start recording 3595 while (!exitPending()) { 3596 3597 processConfigEvents(); 3598 3599 { // scope for mLock 3600 Mutex::Autolock _l(mLock); 3601 checkForNewParameters_l(); 3602 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3603 standby(); 3604 3605 if (exitPending()) { 3606 break; 3607 } 3608 3609 releaseWakeLock_l(); 3610 ALOGV("RecordThread: loop stopping"); 3611 // go to sleep 3612 mWaitWorkCV.wait(mLock); 3613 ALOGV("RecordThread: loop starting"); 3614 acquireWakeLock_l(); 3615 continue; 3616 } 3617 if (mActiveTrack != 0) { 3618 if (mActiveTrack->mState == TrackBase::PAUSING) { 3619 standby(); 3620 mActiveTrack.clear(); 3621 mStartStopCond.broadcast(); 3622 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3623 if (mReqChannelCount != mActiveTrack->channelCount()) { 3624 mActiveTrack.clear(); 3625 mStartStopCond.broadcast(); 3626 } else if (readOnce) { 3627 // record start succeeds only if first read from audio input 3628 // succeeds 3629 if (mBytesRead >= 0) { 3630 mActiveTrack->mState = TrackBase::ACTIVE; 3631 } else { 3632 mActiveTrack.clear(); 3633 } 3634 mStartStopCond.broadcast(); 3635 } 3636 mStandby = false; 3637 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3638 removeTrack_l(mActiveTrack); 3639 mActiveTrack.clear(); 3640 } 3641 } 3642 lockEffectChains_l(effectChains); 3643 } 3644 3645 if (mActiveTrack != 0) { 3646 if (mActiveTrack->mState != TrackBase::ACTIVE && 3647 mActiveTrack->mState != TrackBase::RESUMING) { 3648 unlockEffectChains(effectChains); 3649 usleep(kRecordThreadSleepUs); 3650 continue; 3651 } 3652 for (size_t i = 0; i < effectChains.size(); i ++) { 3653 effectChains[i]->process_l(); 3654 } 3655 3656 buffer.frameCount = mFrameCount; 3657 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3658 readOnce = true; 3659 size_t framesOut = buffer.frameCount; 3660 if (mResampler == NULL) { 3661 // no resampling 3662 while (framesOut) { 3663 size_t framesIn = mFrameCount - mRsmpInIndex; 3664 if (framesIn) { 3665 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3666 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3667 mActiveTrack->mFrameSize; 3668 if (framesIn > framesOut) 3669 framesIn = framesOut; 3670 mRsmpInIndex += framesIn; 3671 framesOut -= framesIn; 3672 if (mChannelCount == mReqChannelCount || 3673 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3674 memcpy(dst, src, framesIn * mFrameSize); 3675 } else { 3676 if (mChannelCount == 1) { 3677 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3678 (int16_t *)src, framesIn); 3679 } else { 3680 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3681 (int16_t *)src, framesIn); 3682 } 3683 } 3684 } 3685 if (framesOut && mFrameCount == mRsmpInIndex) { 3686 void *readInto; 3687 if (framesOut == mFrameCount && 3688 (mChannelCount == mReqChannelCount || 3689 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3690 readInto = buffer.raw; 3691 framesOut = 0; 3692 } else { 3693 readInto = mRsmpInBuffer; 3694 mRsmpInIndex = 0; 3695 } 3696 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3697 if (mBytesRead <= 0) { 3698 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3699 { 3700 ALOGE("Error reading audio input"); 3701 // Force input into standby so that it tries to 3702 // recover at next read attempt 3703 inputStandBy(); 3704 usleep(kRecordThreadSleepUs); 3705 } 3706 mRsmpInIndex = mFrameCount; 3707 framesOut = 0; 3708 buffer.frameCount = 0; 3709 } else if (mTeeSink != 0) { 3710 (void) mTeeSink->write(readInto, 3711 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3712 } 3713 } 3714 } 3715 } else { 3716 // resampling 3717 3718 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3719 // alter output frame count as if we were expecting stereo samples 3720 if (mChannelCount == 1 && mReqChannelCount == 1) { 3721 framesOut >>= 1; 3722 } 3723 mResampler->resample(mRsmpOutBuffer, framesOut, 3724 this /* AudioBufferProvider* */); 3725 // ditherAndClamp() works as long as all buffers returned by 3726 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3727 if (mChannelCount == 2 && mReqChannelCount == 1) { 3728 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3729 // the resampler always outputs stereo samples: 3730 // do post stereo to mono conversion 3731 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3732 framesOut); 3733 } else { 3734 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3735 } 3736 3737 } 3738 if (mFramestoDrop == 0) { 3739 mActiveTrack->releaseBuffer(&buffer); 3740 } else { 3741 if (mFramestoDrop > 0) { 3742 mFramestoDrop -= buffer.frameCount; 3743 if (mFramestoDrop <= 0) { 3744 clearSyncStartEvent(); 3745 } 3746 } else { 3747 mFramestoDrop += buffer.frameCount; 3748 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3749 mSyncStartEvent->isCancelled()) { 3750 ALOGW("Synced record %s, session %d, trigger session %d", 3751 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3752 mActiveTrack->sessionId(), 3753 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3754 clearSyncStartEvent(); 3755 } 3756 } 3757 } 3758 mActiveTrack->clearOverflow(); 3759 } 3760 // client isn't retrieving buffers fast enough 3761 else { 3762 if (!mActiveTrack->setOverflow()) { 3763 nsecs_t now = systemTime(); 3764 if ((now - lastWarning) > kWarningThrottleNs) { 3765 ALOGW("RecordThread: buffer overflow"); 3766 lastWarning = now; 3767 } 3768 } 3769 // Release the processor for a while before asking for a new buffer. 3770 // This will give the application more chance to read from the buffer and 3771 // clear the overflow. 3772 usleep(kRecordThreadSleepUs); 3773 } 3774 } 3775 // enable changes in effect chain 3776 unlockEffectChains(effectChains); 3777 effectChains.clear(); 3778 } 3779 3780 standby(); 3781 3782 { 3783 Mutex::Autolock _l(mLock); 3784 mActiveTrack.clear(); 3785 mStartStopCond.broadcast(); 3786 } 3787 3788 releaseWakeLock(); 3789 3790 ALOGV("RecordThread %p exiting", this); 3791 return false; 3792} 3793 3794void AudioFlinger::RecordThread::standby() 3795{ 3796 if (!mStandby) { 3797 inputStandBy(); 3798 mStandby = true; 3799 } 3800} 3801 3802void AudioFlinger::RecordThread::inputStandBy() 3803{ 3804 mInput->stream->common.standby(&mInput->stream->common); 3805} 3806 3807sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3808 const sp<AudioFlinger::Client>& client, 3809 uint32_t sampleRate, 3810 audio_format_t format, 3811 audio_channel_mask_t channelMask, 3812 size_t frameCount, 3813 int sessionId, 3814 IAudioFlinger::track_flags_t flags, 3815 pid_t tid, 3816 status_t *status) 3817{ 3818 sp<RecordTrack> track; 3819 status_t lStatus; 3820 3821 lStatus = initCheck(); 3822 if (lStatus != NO_ERROR) { 3823 ALOGE("Audio driver not initialized."); 3824 goto Exit; 3825 } 3826 3827 // FIXME use flags and tid similar to createTrack_l() 3828 3829 { // scope for mLock 3830 Mutex::Autolock _l(mLock); 3831 3832 track = new RecordTrack(this, client, sampleRate, 3833 format, channelMask, frameCount, sessionId); 3834 3835 if (track->getCblk() == 0) { 3836 lStatus = NO_MEMORY; 3837 goto Exit; 3838 } 3839 mTracks.add(track); 3840 3841 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3842 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3843 mAudioFlinger->btNrecIsOff(); 3844 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3845 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3846 } 3847 lStatus = NO_ERROR; 3848 3849Exit: 3850 if (status) { 3851 *status = lStatus; 3852 } 3853 return track; 3854} 3855 3856status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3857 AudioSystem::sync_event_t event, 3858 int triggerSession) 3859{ 3860 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3861 sp<ThreadBase> strongMe = this; 3862 status_t status = NO_ERROR; 3863 3864 if (event == AudioSystem::SYNC_EVENT_NONE) { 3865 clearSyncStartEvent(); 3866 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3867 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3868 triggerSession, 3869 recordTrack->sessionId(), 3870 syncStartEventCallback, 3871 this); 3872 // Sync event can be cancelled by the trigger session if the track is not in a 3873 // compatible state in which case we start record immediately 3874 if (mSyncStartEvent->isCancelled()) { 3875 clearSyncStartEvent(); 3876 } else { 3877 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3878 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3879 } 3880 } 3881 3882 { 3883 AutoMutex lock(mLock); 3884 if (mActiveTrack != 0) { 3885 if (recordTrack != mActiveTrack.get()) { 3886 status = -EBUSY; 3887 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3888 mActiveTrack->mState = TrackBase::ACTIVE; 3889 } 3890 return status; 3891 } 3892 3893 recordTrack->mState = TrackBase::IDLE; 3894 mActiveTrack = recordTrack; 3895 mLock.unlock(); 3896 status_t status = AudioSystem::startInput(mId); 3897 mLock.lock(); 3898 if (status != NO_ERROR) { 3899 mActiveTrack.clear(); 3900 clearSyncStartEvent(); 3901 return status; 3902 } 3903 mRsmpInIndex = mFrameCount; 3904 mBytesRead = 0; 3905 if (mResampler != NULL) { 3906 mResampler->reset(); 3907 } 3908 mActiveTrack->mState = TrackBase::RESUMING; 3909 // signal thread to start 3910 ALOGV("Signal record thread"); 3911 mWaitWorkCV.broadcast(); 3912 // do not wait for mStartStopCond if exiting 3913 if (exitPending()) { 3914 mActiveTrack.clear(); 3915 status = INVALID_OPERATION; 3916 goto startError; 3917 } 3918 mStartStopCond.wait(mLock); 3919 if (mActiveTrack == 0) { 3920 ALOGV("Record failed to start"); 3921 status = BAD_VALUE; 3922 goto startError; 3923 } 3924 ALOGV("Record started OK"); 3925 return status; 3926 } 3927startError: 3928 AudioSystem::stopInput(mId); 3929 clearSyncStartEvent(); 3930 return status; 3931} 3932 3933void AudioFlinger::RecordThread::clearSyncStartEvent() 3934{ 3935 if (mSyncStartEvent != 0) { 3936 mSyncStartEvent->cancel(); 3937 } 3938 mSyncStartEvent.clear(); 3939 mFramestoDrop = 0; 3940} 3941 3942void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3943{ 3944 sp<SyncEvent> strongEvent = event.promote(); 3945 3946 if (strongEvent != 0) { 3947 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3948 me->handleSyncStartEvent(strongEvent); 3949 } 3950} 3951 3952void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 3953{ 3954 if (event == mSyncStartEvent) { 3955 // TODO: use actual buffer filling status instead of 2 buffers when info is available 3956 // from audio HAL 3957 mFramestoDrop = mFrameCount * 2; 3958 } 3959} 3960 3961bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 3962 ALOGV("RecordThread::stop"); 3963 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 3964 return false; 3965 } 3966 recordTrack->mState = TrackBase::PAUSING; 3967 // do not wait for mStartStopCond if exiting 3968 if (exitPending()) { 3969 return true; 3970 } 3971 mStartStopCond.wait(mLock); 3972 // if we have been restarted, recordTrack == mActiveTrack.get() here 3973 if (exitPending() || recordTrack != mActiveTrack.get()) { 3974 ALOGV("Record stopped OK"); 3975 return true; 3976 } 3977 return false; 3978} 3979 3980bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 3981{ 3982 return false; 3983} 3984 3985status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 3986{ 3987#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 3988 if (!isValidSyncEvent(event)) { 3989 return BAD_VALUE; 3990 } 3991 3992 int eventSession = event->triggerSession(); 3993 status_t ret = NAME_NOT_FOUND; 3994 3995 Mutex::Autolock _l(mLock); 3996 3997 for (size_t i = 0; i < mTracks.size(); i++) { 3998 sp<RecordTrack> track = mTracks[i]; 3999 if (eventSession == track->sessionId()) { 4000 (void) track->setSyncEvent(event); 4001 ret = NO_ERROR; 4002 } 4003 } 4004 return ret; 4005#else 4006 return BAD_VALUE; 4007#endif 4008} 4009 4010// destroyTrack_l() must be called with ThreadBase::mLock held 4011void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4012{ 4013 track->mState = TrackBase::TERMINATED; 4014 // active tracks are removed by threadLoop() 4015 if (mActiveTrack != track) { 4016 removeTrack_l(track); 4017 } 4018} 4019 4020void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4021{ 4022 mTracks.remove(track); 4023 // need anything related to effects here? 4024} 4025 4026void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4027{ 4028 dumpInternals(fd, args); 4029 dumpTracks(fd, args); 4030 dumpEffectChains(fd, args); 4031} 4032 4033void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4034{ 4035 const size_t SIZE = 256; 4036 char buffer[SIZE]; 4037 String8 result; 4038 4039 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4040 result.append(buffer); 4041 4042 if (mActiveTrack != 0) { 4043 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4044 result.append(buffer); 4045 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4046 result.append(buffer); 4047 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4048 result.append(buffer); 4049 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4050 result.append(buffer); 4051 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4052 result.append(buffer); 4053 } else { 4054 result.append("No active record client\n"); 4055 } 4056 4057 write(fd, result.string(), result.size()); 4058 4059 dumpBase(fd, args); 4060} 4061 4062void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4063{ 4064 const size_t SIZE = 256; 4065 char buffer[SIZE]; 4066 String8 result; 4067 4068 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4069 result.append(buffer); 4070 RecordTrack::appendDumpHeader(result); 4071 for (size_t i = 0; i < mTracks.size(); ++i) { 4072 sp<RecordTrack> track = mTracks[i]; 4073 if (track != 0) { 4074 track->dump(buffer, SIZE); 4075 result.append(buffer); 4076 } 4077 } 4078 4079 if (mActiveTrack != 0) { 4080 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4081 result.append(buffer); 4082 RecordTrack::appendDumpHeader(result); 4083 mActiveTrack->dump(buffer, SIZE); 4084 result.append(buffer); 4085 4086 } 4087 write(fd, result.string(), result.size()); 4088} 4089 4090// AudioBufferProvider interface 4091status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4092{ 4093 size_t framesReq = buffer->frameCount; 4094 size_t framesReady = mFrameCount - mRsmpInIndex; 4095 int channelCount; 4096 4097 if (framesReady == 0) { 4098 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4099 if (mBytesRead <= 0) { 4100 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4101 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4102 // Force input into standby so that it tries to 4103 // recover at next read attempt 4104 inputStandBy(); 4105 usleep(kRecordThreadSleepUs); 4106 } 4107 buffer->raw = NULL; 4108 buffer->frameCount = 0; 4109 return NOT_ENOUGH_DATA; 4110 } 4111 mRsmpInIndex = 0; 4112 framesReady = mFrameCount; 4113 } 4114 4115 if (framesReq > framesReady) { 4116 framesReq = framesReady; 4117 } 4118 4119 if (mChannelCount == 1 && mReqChannelCount == 2) { 4120 channelCount = 1; 4121 } else { 4122 channelCount = 2; 4123 } 4124 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4125 buffer->frameCount = framesReq; 4126 return NO_ERROR; 4127} 4128 4129// AudioBufferProvider interface 4130void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4131{ 4132 mRsmpInIndex += buffer->frameCount; 4133 buffer->frameCount = 0; 4134} 4135 4136bool AudioFlinger::RecordThread::checkForNewParameters_l() 4137{ 4138 bool reconfig = false; 4139 4140 while (!mNewParameters.isEmpty()) { 4141 status_t status = NO_ERROR; 4142 String8 keyValuePair = mNewParameters[0]; 4143 AudioParameter param = AudioParameter(keyValuePair); 4144 int value; 4145 audio_format_t reqFormat = mFormat; 4146 uint32_t reqSamplingRate = mReqSampleRate; 4147 uint32_t reqChannelCount = mReqChannelCount; 4148 4149 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4150 reqSamplingRate = value; 4151 reconfig = true; 4152 } 4153 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4154 reqFormat = (audio_format_t) value; 4155 reconfig = true; 4156 } 4157 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4158 reqChannelCount = popcount(value); 4159 reconfig = true; 4160 } 4161 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4162 // do not accept frame count changes if tracks are open as the track buffer 4163 // size depends on frame count and correct behavior would not be guaranteed 4164 // if frame count is changed after track creation 4165 if (mActiveTrack != 0) { 4166 status = INVALID_OPERATION; 4167 } else { 4168 reconfig = true; 4169 } 4170 } 4171 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4172 // forward device change to effects that have requested to be 4173 // aware of attached audio device. 4174 for (size_t i = 0; i < mEffectChains.size(); i++) { 4175 mEffectChains[i]->setDevice_l(value); 4176 } 4177 4178 // store input device and output device but do not forward output device to audio HAL. 4179 // Note that status is ignored by the caller for output device 4180 // (see AudioFlinger::setParameters() 4181 if (audio_is_output_devices(value)) { 4182 mOutDevice = value; 4183 status = BAD_VALUE; 4184 } else { 4185 mInDevice = value; 4186 // disable AEC and NS if the device is a BT SCO headset supporting those 4187 // pre processings 4188 if (mTracks.size() > 0) { 4189 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4190 mAudioFlinger->btNrecIsOff(); 4191 for (size_t i = 0; i < mTracks.size(); i++) { 4192 sp<RecordTrack> track = mTracks[i]; 4193 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4194 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4195 } 4196 } 4197 } 4198 } 4199 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4200 mAudioSource != (audio_source_t)value) { 4201 // forward device change to effects that have requested to be 4202 // aware of attached audio device. 4203 for (size_t i = 0; i < mEffectChains.size(); i++) { 4204 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4205 } 4206 mAudioSource = (audio_source_t)value; 4207 } 4208 if (status == NO_ERROR) { 4209 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4210 keyValuePair.string()); 4211 if (status == INVALID_OPERATION) { 4212 inputStandBy(); 4213 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4214 keyValuePair.string()); 4215 } 4216 if (reconfig) { 4217 if (status == BAD_VALUE && 4218 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4219 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4220 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 4221 <= (2 * reqSamplingRate)) && 4222 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4223 <= FCC_2 && 4224 (reqChannelCount <= FCC_2)) { 4225 status = NO_ERROR; 4226 } 4227 if (status == NO_ERROR) { 4228 readInputParameters(); 4229 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4230 } 4231 } 4232 } 4233 4234 mNewParameters.removeAt(0); 4235 4236 mParamStatus = status; 4237 mParamCond.signal(); 4238 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4239 // already timed out waiting for the status and will never signal the condition. 4240 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4241 } 4242 return reconfig; 4243} 4244 4245String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4246{ 4247 char *s; 4248 String8 out_s8 = String8(); 4249 4250 Mutex::Autolock _l(mLock); 4251 if (initCheck() != NO_ERROR) { 4252 return out_s8; 4253 } 4254 4255 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4256 out_s8 = String8(s); 4257 free(s); 4258 return out_s8; 4259} 4260 4261void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4262 AudioSystem::OutputDescriptor desc; 4263 void *param2 = NULL; 4264 4265 switch (event) { 4266 case AudioSystem::INPUT_OPENED: 4267 case AudioSystem::INPUT_CONFIG_CHANGED: 4268 desc.channels = mChannelMask; 4269 desc.samplingRate = mSampleRate; 4270 desc.format = mFormat; 4271 desc.frameCount = mFrameCount; 4272 desc.latency = 0; 4273 param2 = &desc; 4274 break; 4275 4276 case AudioSystem::INPUT_CLOSED: 4277 default: 4278 break; 4279 } 4280 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4281} 4282 4283void AudioFlinger::RecordThread::readInputParameters() 4284{ 4285 delete mRsmpInBuffer; 4286 // mRsmpInBuffer is always assigned a new[] below 4287 delete mRsmpOutBuffer; 4288 mRsmpOutBuffer = NULL; 4289 delete mResampler; 4290 mResampler = NULL; 4291 4292 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4293 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4294 mChannelCount = (uint16_t)popcount(mChannelMask); 4295 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4296 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4297 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4298 mFrameCount = mInputBytes / mFrameSize; 4299 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4300 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4301 4302 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4303 { 4304 int channelCount; 4305 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4306 // stereo to mono post process as the resampler always outputs stereo. 4307 if (mChannelCount == 1 && mReqChannelCount == 2) { 4308 channelCount = 1; 4309 } else { 4310 channelCount = 2; 4311 } 4312 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4313 mResampler->setSampleRate(mSampleRate); 4314 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4315 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4316 4317 // optmization: if mono to mono, alter input frame count as if we were inputing 4318 // stereo samples 4319 if (mChannelCount == 1 && mReqChannelCount == 1) { 4320 mFrameCount >>= 1; 4321 } 4322 4323 } 4324 mRsmpInIndex = mFrameCount; 4325} 4326 4327unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4328{ 4329 Mutex::Autolock _l(mLock); 4330 if (initCheck() != NO_ERROR) { 4331 return 0; 4332 } 4333 4334 return mInput->stream->get_input_frames_lost(mInput->stream); 4335} 4336 4337uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4338{ 4339 Mutex::Autolock _l(mLock); 4340 uint32_t result = 0; 4341 if (getEffectChain_l(sessionId) != 0) { 4342 result = EFFECT_SESSION; 4343 } 4344 4345 for (size_t i = 0; i < mTracks.size(); ++i) { 4346 if (sessionId == mTracks[i]->sessionId()) { 4347 result |= TRACK_SESSION; 4348 break; 4349 } 4350 } 4351 4352 return result; 4353} 4354 4355KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4356{ 4357 KeyedVector<int, bool> ids; 4358 Mutex::Autolock _l(mLock); 4359 for (size_t j = 0; j < mTracks.size(); ++j) { 4360 sp<RecordThread::RecordTrack> track = mTracks[j]; 4361 int sessionId = track->sessionId(); 4362 if (ids.indexOfKey(sessionId) < 0) { 4363 ids.add(sessionId, true); 4364 } 4365 } 4366 return ids; 4367} 4368 4369AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4370{ 4371 Mutex::Autolock _l(mLock); 4372 AudioStreamIn *input = mInput; 4373 mInput = NULL; 4374 return input; 4375} 4376 4377// this method must always be called either with ThreadBase mLock held or inside the thread loop 4378audio_stream_t* AudioFlinger::RecordThread::stream() const 4379{ 4380 if (mInput == NULL) { 4381 return NULL; 4382 } 4383 return &mInput->stream->common; 4384} 4385 4386status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4387{ 4388 // only one chain per input thread 4389 if (mEffectChains.size() != 0) { 4390 return INVALID_OPERATION; 4391 } 4392 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4393 4394 chain->setInBuffer(NULL); 4395 chain->setOutBuffer(NULL); 4396 4397 checkSuspendOnAddEffectChain_l(chain); 4398 4399 mEffectChains.add(chain); 4400 4401 return NO_ERROR; 4402} 4403 4404size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4405{ 4406 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4407 ALOGW_IF(mEffectChains.size() != 1, 4408 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4409 chain.get(), mEffectChains.size(), this); 4410 if (mEffectChains.size() == 1) { 4411 mEffectChains.removeAt(0); 4412 } 4413 return 0; 4414} 4415 4416}; // namespace android 4417