Threads.cpp revision d455cdfad40ca0558b8f4f800ec192027e272c14
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 273 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 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 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 285 for (size_t i = 0; i < mConfigEvents.size(); i++) { 286 delete mConfigEvents[i]; 287 } 288 mConfigEvents.clear(); 289 290 mParamCond.broadcast(); 291 // do not lock the mutex in destructor 292 releaseWakeLock_l(); 293 if (mPowerManager != 0) { 294 sp<IBinder> binder = mPowerManager->asBinder(); 295 binder->unlinkToDeath(mDeathRecipient); 296 } 297} 298 299status_t AudioFlinger::ThreadBase::readyToRun() 300{ 301 status_t status = initCheck(); 302 if (status == NO_ERROR) { 303 ALOGI("AudioFlinger's thread %p ready to run", this); 304 } else { 305 ALOGE("No working audio driver found."); 306 } 307 return status; 308} 309 310void AudioFlinger::ThreadBase::exit() 311{ 312 ALOGV("ThreadBase::exit"); 313 // do any cleanup required for exit to succeed 314 preExit(); 315 { 316 // This lock prevents the following race in thread (uniprocessor for illustration): 317 // if (!exitPending()) { 318 // // context switch from here to exit() 319 // // exit() calls requestExit(), what exitPending() observes 320 // // exit() calls signal(), which is dropped since no waiters 321 // // context switch back from exit() to here 322 // mWaitWorkCV.wait(...); 323 // // now thread is hung 324 // } 325 AutoMutex lock(mLock); 326 requestExit(); 327 mWaitWorkCV.broadcast(); 328 } 329 // When Thread::requestExitAndWait is made virtual and this method is renamed to 330 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 331 requestExitAndWait(); 332} 333 334status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 335{ 336 status_t status; 337 338 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 339 Mutex::Autolock _l(mLock); 340 341 mNewParameters.add(keyValuePairs); 342 mWaitWorkCV.signal(); 343 // wait condition with timeout in case the thread loop has exited 344 // before the request could be processed 345 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 346 status = mParamStatus; 347 mWaitWorkCV.signal(); 348 } else { 349 status = TIMED_OUT; 350 } 351 return status; 352} 353 354void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 355{ 356 Mutex::Autolock _l(mLock); 357 sendIoConfigEvent_l(event, param); 358} 359 360// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 361void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 362{ 363 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 364 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 365 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 366 param); 367 mWaitWorkCV.signal(); 368} 369 370// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 371void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 372{ 373 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 374 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 375 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 376 mConfigEvents.size(), pid, tid, prio); 377 mWaitWorkCV.signal(); 378} 379 380void AudioFlinger::ThreadBase::processConfigEvents() 381{ 382 Mutex::Autolock _l(mLock); 383 processConfigEvents_l(); 384} 385 386// post condition: mConfigEvents.isEmpty() 387void AudioFlinger::ThreadBase::processConfigEvents_l() 388{ 389 while (!mConfigEvents.isEmpty()) { 390 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 391 ConfigEvent *event = mConfigEvents[0]; 392 mConfigEvents.removeAt(0); 393 // release mLock before locking AudioFlinger mLock: lock order is always 394 // AudioFlinger then ThreadBase to avoid cross deadlock 395 mLock.unlock(); 396 switch (event->type()) { 397 case CFG_EVENT_PRIO: { 398 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 399 // FIXME Need to understand why this has be done asynchronously 400 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 401 true /*asynchronous*/); 402 if (err != 0) { 403 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 404 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 405 } 406 } break; 407 case CFG_EVENT_IO: { 408 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 409 { 410 Mutex::Autolock _l(mAudioFlinger->mLock); 411 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 412 } 413 } break; 414 default: 415 ALOGE("processConfigEvents() unknown event type %d", event->type()); 416 break; 417 } 418 delete event; 419 mLock.lock(); 420 } 421} 422 423void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 424{ 425 const size_t SIZE = 256; 426 char buffer[SIZE]; 427 String8 result; 428 429 bool locked = AudioFlinger::dumpTryLock(mLock); 430 if (!locked) { 431 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 432 write(fd, buffer, strlen(buffer)); 433 } 434 435 snprintf(buffer, SIZE, "io handle: %d\n", mId); 436 result.append(buffer); 437 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 438 result.append(buffer); 439 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 440 result.append(buffer); 441 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 442 result.append(buffer); 443 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 444 result.append(buffer); 445 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 446 result.append(buffer); 447 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 448 result.append(buffer); 449 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 450 result.append(buffer); 451 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 452 result.append(buffer); 453 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 454 result.append(buffer); 455 456 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 457 result.append(buffer); 458 result.append(" Index Command"); 459 for (size_t i = 0; i < mNewParameters.size(); ++i) { 460 snprintf(buffer, SIZE, "\n %02d ", i); 461 result.append(buffer); 462 result.append(mNewParameters[i]); 463 } 464 465 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 466 result.append(buffer); 467 for (size_t i = 0; i < mConfigEvents.size(); i++) { 468 mConfigEvents[i]->dump(buffer, SIZE); 469 result.append(buffer); 470 } 471 result.append("\n"); 472 473 write(fd, result.string(), result.size()); 474 475 if (locked) { 476 mLock.unlock(); 477 } 478} 479 480void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 481{ 482 const size_t SIZE = 256; 483 char buffer[SIZE]; 484 String8 result; 485 486 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 487 write(fd, buffer, strlen(buffer)); 488 489 for (size_t i = 0; i < mEffectChains.size(); ++i) { 490 sp<EffectChain> chain = mEffectChains[i]; 491 if (chain != 0) { 492 chain->dump(fd, args); 493 } 494 } 495} 496 497void AudioFlinger::ThreadBase::acquireWakeLock() 498{ 499 Mutex::Autolock _l(mLock); 500 acquireWakeLock_l(); 501} 502 503void AudioFlinger::ThreadBase::acquireWakeLock_l() 504{ 505 if (mPowerManager == 0) { 506 // use checkService() to avoid blocking if power service is not up yet 507 sp<IBinder> binder = 508 defaultServiceManager()->checkService(String16("power")); 509 if (binder == 0) { 510 ALOGW("Thread %s cannot connect to the power manager service", mName); 511 } else { 512 mPowerManager = interface_cast<IPowerManager>(binder); 513 binder->linkToDeath(mDeathRecipient); 514 } 515 } 516 if (mPowerManager != 0) { 517 sp<IBinder> binder = new BBinder(); 518 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 519 binder, 520 String16(mName), 521 String16("media")); 522 if (status == NO_ERROR) { 523 mWakeLockToken = binder; 524 } 525 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 526 } 527} 528 529void AudioFlinger::ThreadBase::releaseWakeLock() 530{ 531 Mutex::Autolock _l(mLock); 532 releaseWakeLock_l(); 533} 534 535void AudioFlinger::ThreadBase::releaseWakeLock_l() 536{ 537 if (mWakeLockToken != 0) { 538 ALOGV("releaseWakeLock_l() %s", mName); 539 if (mPowerManager != 0) { 540 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 541 } 542 mWakeLockToken.clear(); 543 } 544} 545 546void AudioFlinger::ThreadBase::clearPowerManager() 547{ 548 Mutex::Autolock _l(mLock); 549 releaseWakeLock_l(); 550 mPowerManager.clear(); 551} 552 553void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 554{ 555 sp<ThreadBase> thread = mThread.promote(); 556 if (thread != 0) { 557 thread->clearPowerManager(); 558 } 559 ALOGW("power manager service died !!!"); 560} 561 562void AudioFlinger::ThreadBase::setEffectSuspended( 563 const effect_uuid_t *type, bool suspend, int sessionId) 564{ 565 Mutex::Autolock _l(mLock); 566 setEffectSuspended_l(type, suspend, sessionId); 567} 568 569void AudioFlinger::ThreadBase::setEffectSuspended_l( 570 const effect_uuid_t *type, bool suspend, int sessionId) 571{ 572 sp<EffectChain> chain = getEffectChain_l(sessionId); 573 if (chain != 0) { 574 if (type != NULL) { 575 chain->setEffectSuspended_l(type, suspend); 576 } else { 577 chain->setEffectSuspendedAll_l(suspend); 578 } 579 } 580 581 updateSuspendedSessions_l(type, suspend, sessionId); 582} 583 584void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 585{ 586 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 587 if (index < 0) { 588 return; 589 } 590 591 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 592 mSuspendedSessions.valueAt(index); 593 594 for (size_t i = 0; i < sessionEffects.size(); i++) { 595 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 596 for (int j = 0; j < desc->mRefCount; j++) { 597 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 598 chain->setEffectSuspendedAll_l(true); 599 } else { 600 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 601 desc->mType.timeLow); 602 chain->setEffectSuspended_l(&desc->mType, true); 603 } 604 } 605 } 606} 607 608void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 609 bool suspend, 610 int sessionId) 611{ 612 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 613 614 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 615 616 if (suspend) { 617 if (index >= 0) { 618 sessionEffects = mSuspendedSessions.valueAt(index); 619 } else { 620 mSuspendedSessions.add(sessionId, sessionEffects); 621 } 622 } else { 623 if (index < 0) { 624 return; 625 } 626 sessionEffects = mSuspendedSessions.valueAt(index); 627 } 628 629 630 int key = EffectChain::kKeyForSuspendAll; 631 if (type != NULL) { 632 key = type->timeLow; 633 } 634 index = sessionEffects.indexOfKey(key); 635 636 sp<SuspendedSessionDesc> desc; 637 if (suspend) { 638 if (index >= 0) { 639 desc = sessionEffects.valueAt(index); 640 } else { 641 desc = new SuspendedSessionDesc(); 642 if (type != NULL) { 643 desc->mType = *type; 644 } 645 sessionEffects.add(key, desc); 646 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 647 } 648 desc->mRefCount++; 649 } else { 650 if (index < 0) { 651 return; 652 } 653 desc = sessionEffects.valueAt(index); 654 if (--desc->mRefCount == 0) { 655 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 656 sessionEffects.removeItemsAt(index); 657 if (sessionEffects.isEmpty()) { 658 ALOGV("updateSuspendedSessions_l() restore removing session %d", 659 sessionId); 660 mSuspendedSessions.removeItem(sessionId); 661 } 662 } 663 } 664 if (!sessionEffects.isEmpty()) { 665 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 666 } 667} 668 669void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 670 bool enabled, 671 int sessionId) 672{ 673 Mutex::Autolock _l(mLock); 674 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 675} 676 677void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 678 bool enabled, 679 int sessionId) 680{ 681 if (mType != RECORD) { 682 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 683 // another session. This gives the priority to well behaved effect control panels 684 // and applications not using global effects. 685 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 686 // global effects 687 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 688 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 689 } 690 } 691 692 sp<EffectChain> chain = getEffectChain_l(sessionId); 693 if (chain != 0) { 694 chain->checkSuspendOnEffectEnabled(effect, enabled); 695 } 696} 697 698// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 699sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 700 const sp<AudioFlinger::Client>& client, 701 const sp<IEffectClient>& effectClient, 702 int32_t priority, 703 int sessionId, 704 effect_descriptor_t *desc, 705 int *enabled, 706 status_t *status) 707{ 708 sp<EffectModule> effect; 709 sp<EffectHandle> handle; 710 status_t lStatus; 711 sp<EffectChain> chain; 712 bool chainCreated = false; 713 bool effectCreated = false; 714 bool effectRegistered = false; 715 716 lStatus = initCheck(); 717 if (lStatus != NO_ERROR) { 718 ALOGW("createEffect_l() Audio driver not initialized."); 719 goto Exit; 720 } 721 722 // Do not allow effects with session ID 0 on direct output or duplicating threads 723 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 724 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 725 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 726 desc->name, sessionId); 727 lStatus = BAD_VALUE; 728 goto Exit; 729 } 730 // Only Pre processor effects are allowed on input threads and only on input threads 731 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 732 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 733 desc->name, desc->flags, mType); 734 lStatus = BAD_VALUE; 735 goto Exit; 736 } 737 738 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 739 740 { // scope for mLock 741 Mutex::Autolock _l(mLock); 742 743 // check for existing effect chain with the requested audio session 744 chain = getEffectChain_l(sessionId); 745 if (chain == 0) { 746 // create a new chain for this session 747 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 748 chain = new EffectChain(this, sessionId); 749 addEffectChain_l(chain); 750 chain->setStrategy(getStrategyForSession_l(sessionId)); 751 chainCreated = true; 752 } else { 753 effect = chain->getEffectFromDesc_l(desc); 754 } 755 756 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 757 758 if (effect == 0) { 759 int id = mAudioFlinger->nextUniqueId(); 760 // Check CPU and memory usage 761 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 762 if (lStatus != NO_ERROR) { 763 goto Exit; 764 } 765 effectRegistered = true; 766 // create a new effect module if none present in the chain 767 effect = new EffectModule(this, chain, desc, id, sessionId); 768 lStatus = effect->status(); 769 if (lStatus != NO_ERROR) { 770 goto Exit; 771 } 772 lStatus = chain->addEffect_l(effect); 773 if (lStatus != NO_ERROR) { 774 goto Exit; 775 } 776 effectCreated = true; 777 778 effect->setDevice(mOutDevice); 779 effect->setDevice(mInDevice); 780 effect->setMode(mAudioFlinger->getMode()); 781 effect->setAudioSource(mAudioSource); 782 } 783 // create effect handle and connect it to effect module 784 handle = new EffectHandle(effect, client, effectClient, priority); 785 lStatus = effect->addHandle(handle.get()); 786 if (enabled != NULL) { 787 *enabled = (int)effect->isEnabled(); 788 } 789 } 790 791Exit: 792 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 793 Mutex::Autolock _l(mLock); 794 if (effectCreated) { 795 chain->removeEffect_l(effect); 796 } 797 if (effectRegistered) { 798 AudioSystem::unregisterEffect(effect->id()); 799 } 800 if (chainCreated) { 801 removeEffectChain_l(chain); 802 } 803 handle.clear(); 804 } 805 806 *status = lStatus; 807 return handle; 808} 809 810sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 811{ 812 Mutex::Autolock _l(mLock); 813 return getEffect_l(sessionId, effectId); 814} 815 816sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 817{ 818 sp<EffectChain> chain = getEffectChain_l(sessionId); 819 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 820} 821 822// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 823// PlaybackThread::mLock held 824status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 825{ 826 // check for existing effect chain with the requested audio session 827 int sessionId = effect->sessionId(); 828 sp<EffectChain> chain = getEffectChain_l(sessionId); 829 bool chainCreated = false; 830 831 if (chain == 0) { 832 // create a new chain for this session 833 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 834 chain = new EffectChain(this, sessionId); 835 addEffectChain_l(chain); 836 chain->setStrategy(getStrategyForSession_l(sessionId)); 837 chainCreated = true; 838 } 839 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 840 841 if (chain->getEffectFromId_l(effect->id()) != 0) { 842 ALOGW("addEffect_l() %p effect %s already present in chain %p", 843 this, effect->desc().name, chain.get()); 844 return BAD_VALUE; 845 } 846 847 status_t status = chain->addEffect_l(effect); 848 if (status != NO_ERROR) { 849 if (chainCreated) { 850 removeEffectChain_l(chain); 851 } 852 return status; 853 } 854 855 effect->setDevice(mOutDevice); 856 effect->setDevice(mInDevice); 857 effect->setMode(mAudioFlinger->getMode()); 858 effect->setAudioSource(mAudioSource); 859 return NO_ERROR; 860} 861 862void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 863 864 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 865 effect_descriptor_t desc = effect->desc(); 866 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 867 detachAuxEffect_l(effect->id()); 868 } 869 870 sp<EffectChain> chain = effect->chain().promote(); 871 if (chain != 0) { 872 // remove effect chain if removing last effect 873 if (chain->removeEffect_l(effect) == 0) { 874 removeEffectChain_l(chain); 875 } 876 } else { 877 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 878 } 879} 880 881void AudioFlinger::ThreadBase::lockEffectChains_l( 882 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 883{ 884 effectChains = mEffectChains; 885 for (size_t i = 0; i < mEffectChains.size(); i++) { 886 mEffectChains[i]->lock(); 887 } 888} 889 890void AudioFlinger::ThreadBase::unlockEffectChains( 891 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 892{ 893 for (size_t i = 0; i < effectChains.size(); i++) { 894 effectChains[i]->unlock(); 895 } 896} 897 898sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 899{ 900 Mutex::Autolock _l(mLock); 901 return getEffectChain_l(sessionId); 902} 903 904sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 905{ 906 size_t size = mEffectChains.size(); 907 for (size_t i = 0; i < size; i++) { 908 if (mEffectChains[i]->sessionId() == sessionId) { 909 return mEffectChains[i]; 910 } 911 } 912 return 0; 913} 914 915void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 916{ 917 Mutex::Autolock _l(mLock); 918 size_t size = mEffectChains.size(); 919 for (size_t i = 0; i < size; i++) { 920 mEffectChains[i]->setMode_l(mode); 921 } 922} 923 924void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 925 EffectHandle *handle, 926 bool unpinIfLast) { 927 928 Mutex::Autolock _l(mLock); 929 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 930 // delete the effect module if removing last handle on it 931 if (effect->removeHandle(handle) == 0) { 932 if (!effect->isPinned() || unpinIfLast) { 933 removeEffect_l(effect); 934 AudioSystem::unregisterEffect(effect->id()); 935 } 936 } 937} 938 939// ---------------------------------------------------------------------------- 940// Playback 941// ---------------------------------------------------------------------------- 942 943AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 944 AudioStreamOut* output, 945 audio_io_handle_t id, 946 audio_devices_t device, 947 type_t type) 948 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 949 mNormalFrameCount(0), mMixBuffer(NULL), 950 mSuspended(0), mBytesWritten(0), 951 // mStreamTypes[] initialized in constructor body 952 mOutput(output), 953 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 954 mMixerStatus(MIXER_IDLE), 955 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 956 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 957 mBytesRemaining(0), 958 mCurrentWriteLength(0), 959 mUseAsyncWrite(false), 960 mWriteAckSequence(0), 961 mDrainSequence(0), 962 mScreenState(AudioFlinger::mScreenState), 963 // index 0 is reserved for normal mixer's submix 964 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 965 // mLatchD, mLatchQ, 966 mLatchDValid(false), mLatchQValid(false) 967{ 968 snprintf(mName, kNameLength, "AudioOut_%X", id); 969 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 970 971 // Assumes constructor is called by AudioFlinger with it's mLock held, but 972 // it would be safer to explicitly pass initial masterVolume/masterMute as 973 // parameter. 974 // 975 // If the HAL we are using has support for master volume or master mute, 976 // then do not attenuate or mute during mixing (just leave the volume at 1.0 977 // and the mute set to false). 978 mMasterVolume = audioFlinger->masterVolume_l(); 979 mMasterMute = audioFlinger->masterMute_l(); 980 if (mOutput && mOutput->audioHwDev) { 981 if (mOutput->audioHwDev->canSetMasterVolume()) { 982 mMasterVolume = 1.0; 983 } 984 985 if (mOutput->audioHwDev->canSetMasterMute()) { 986 mMasterMute = false; 987 } 988 } 989 990 readOutputParameters(); 991 992 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 993 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 994 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 995 stream = (audio_stream_type_t) (stream + 1)) { 996 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 997 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 998 } 999 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1000 // because mAudioFlinger doesn't have one to copy from 1001} 1002 1003AudioFlinger::PlaybackThread::~PlaybackThread() 1004{ 1005 mAudioFlinger->unregisterWriter(mNBLogWriter); 1006 delete[] mMixBuffer; 1007} 1008 1009void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1010{ 1011 dumpInternals(fd, args); 1012 dumpTracks(fd, args); 1013 dumpEffectChains(fd, args); 1014} 1015 1016void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1017{ 1018 const size_t SIZE = 256; 1019 char buffer[SIZE]; 1020 String8 result; 1021 1022 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1023 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1024 const stream_type_t *st = &mStreamTypes[i]; 1025 if (i > 0) { 1026 result.appendFormat(", "); 1027 } 1028 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1029 if (st->mute) { 1030 result.append("M"); 1031 } 1032 } 1033 result.append("\n"); 1034 write(fd, result.string(), result.length()); 1035 result.clear(); 1036 1037 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1038 result.append(buffer); 1039 Track::appendDumpHeader(result); 1040 for (size_t i = 0; i < mTracks.size(); ++i) { 1041 sp<Track> track = mTracks[i]; 1042 if (track != 0) { 1043 track->dump(buffer, SIZE); 1044 result.append(buffer); 1045 } 1046 } 1047 1048 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1049 result.append(buffer); 1050 Track::appendDumpHeader(result); 1051 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1052 sp<Track> track = mActiveTracks[i].promote(); 1053 if (track != 0) { 1054 track->dump(buffer, SIZE); 1055 result.append(buffer); 1056 } 1057 } 1058 write(fd, result.string(), result.size()); 1059 1060 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1061 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1062 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1063 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1064} 1065 1066void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1067{ 1068 const size_t SIZE = 256; 1069 char buffer[SIZE]; 1070 String8 result; 1071 1072 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1073 result.append(buffer); 1074 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1077 ns2ms(systemTime() - mLastWriteTime)); 1078 result.append(buffer); 1079 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1080 result.append(buffer); 1081 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1082 result.append(buffer); 1083 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1084 result.append(buffer); 1085 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1086 result.append(buffer); 1087 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1088 result.append(buffer); 1089 write(fd, result.string(), result.size()); 1090 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1091 1092 dumpBase(fd, args); 1093} 1094 1095// Thread virtuals 1096 1097void AudioFlinger::PlaybackThread::onFirstRef() 1098{ 1099 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1100} 1101 1102// ThreadBase virtuals 1103void AudioFlinger::PlaybackThread::preExit() 1104{ 1105 ALOGV(" preExit()"); 1106 // FIXME this is using hard-coded strings but in the future, this functionality will be 1107 // converted to use audio HAL extensions required to support tunneling 1108 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1109} 1110 1111// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1112sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1113 const sp<AudioFlinger::Client>& client, 1114 audio_stream_type_t streamType, 1115 uint32_t sampleRate, 1116 audio_format_t format, 1117 audio_channel_mask_t channelMask, 1118 size_t frameCount, 1119 const sp<IMemory>& sharedBuffer, 1120 int sessionId, 1121 IAudioFlinger::track_flags_t *flags, 1122 pid_t tid, 1123 status_t *status) 1124{ 1125 sp<Track> track; 1126 status_t lStatus; 1127 1128 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1129 1130 // client expresses a preference for FAST, but we get the final say 1131 if (*flags & IAudioFlinger::TRACK_FAST) { 1132 if ( 1133 // not timed 1134 (!isTimed) && 1135 // either of these use cases: 1136 ( 1137 // use case 1: shared buffer with any frame count 1138 ( 1139 (sharedBuffer != 0) 1140 ) || 1141 // use case 2: callback handler and frame count is default or at least as large as HAL 1142 ( 1143 (tid != -1) && 1144 ((frameCount == 0) || 1145 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1146 ) 1147 ) && 1148 // PCM data 1149 audio_is_linear_pcm(format) && 1150 // mono or stereo 1151 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1152 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1153#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1154 // hardware sample rate 1155 (sampleRate == mSampleRate) && 1156#endif 1157 // normal mixer has an associated fast mixer 1158 hasFastMixer() && 1159 // there are sufficient fast track slots available 1160 (mFastTrackAvailMask != 0) 1161 // FIXME test that MixerThread for this fast track has a capable output HAL 1162 // FIXME add a permission test also? 1163 ) { 1164 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1165 if (frameCount == 0) { 1166 frameCount = mFrameCount * kFastTrackMultiplier; 1167 } 1168 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1169 frameCount, mFrameCount); 1170 } else { 1171 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1172 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1173 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1174 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1175 audio_is_linear_pcm(format), 1176 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1177 *flags &= ~IAudioFlinger::TRACK_FAST; 1178 // For compatibility with AudioTrack calculation, buffer depth is forced 1179 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1180 // This is probably too conservative, but legacy application code may depend on it. 1181 // If you change this calculation, also review the start threshold which is related. 1182 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1183 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1184 if (minBufCount < 2) { 1185 minBufCount = 2; 1186 } 1187 size_t minFrameCount = mNormalFrameCount * minBufCount; 1188 if (frameCount < minFrameCount) { 1189 frameCount = minFrameCount; 1190 } 1191 } 1192 } 1193 1194 if (mType == DIRECT) { 1195 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1196 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1197 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1198 "for output %p with format %d", 1199 sampleRate, format, channelMask, mOutput, mFormat); 1200 lStatus = BAD_VALUE; 1201 goto Exit; 1202 } 1203 } 1204 } else if (mType == OFFLOAD) { 1205 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1206 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1207 "for output %p with format %d", 1208 sampleRate, format, channelMask, mOutput, mFormat); 1209 lStatus = BAD_VALUE; 1210 goto Exit; 1211 } 1212 } else { 1213 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1214 ALOGE("createTrack_l() Bad parameter: format %d \"" 1215 "for output %p with format %d", 1216 format, mOutput, mFormat); 1217 lStatus = BAD_VALUE; 1218 goto Exit; 1219 } 1220 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1221 if (sampleRate > mSampleRate*2) { 1222 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1223 lStatus = BAD_VALUE; 1224 goto Exit; 1225 } 1226 } 1227 1228 lStatus = initCheck(); 1229 if (lStatus != NO_ERROR) { 1230 ALOGE("Audio driver not initialized."); 1231 goto Exit; 1232 } 1233 1234 { // scope for mLock 1235 Mutex::Autolock _l(mLock); 1236 1237 // all tracks in same audio session must share the same routing strategy otherwise 1238 // conflicts will happen when tracks are moved from one output to another by audio policy 1239 // manager 1240 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1241 for (size_t i = 0; i < mTracks.size(); ++i) { 1242 sp<Track> t = mTracks[i]; 1243 if (t != 0 && !t->isOutputTrack()) { 1244 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1245 if (sessionId == t->sessionId() && strategy != actual) { 1246 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1247 strategy, actual); 1248 lStatus = BAD_VALUE; 1249 goto Exit; 1250 } 1251 } 1252 } 1253 1254 if (!isTimed) { 1255 track = new Track(this, client, streamType, sampleRate, format, 1256 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1257 } else { 1258 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1259 channelMask, frameCount, sharedBuffer, sessionId); 1260 } 1261 1262 // new Track always returns non-NULL, 1263 // but TimedTrack::create() is a factory that could fail by returning NULL 1264 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1265 if (lStatus != NO_ERROR) { 1266 track.clear(); 1267 goto Exit; 1268 } 1269 1270 mTracks.add(track); 1271 1272 sp<EffectChain> chain = getEffectChain_l(sessionId); 1273 if (chain != 0) { 1274 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1275 track->setMainBuffer(chain->inBuffer()); 1276 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1277 chain->incTrackCnt(); 1278 } 1279 1280 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1281 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1282 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1283 // so ask activity manager to do this on our behalf 1284 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1285 } 1286 } 1287 1288 lStatus = NO_ERROR; 1289 1290Exit: 1291 *status = lStatus; 1292 return track; 1293} 1294 1295uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1296{ 1297 return latency; 1298} 1299 1300uint32_t AudioFlinger::PlaybackThread::latency() const 1301{ 1302 Mutex::Autolock _l(mLock); 1303 return latency_l(); 1304} 1305uint32_t AudioFlinger::PlaybackThread::latency_l() const 1306{ 1307 if (initCheck() == NO_ERROR) { 1308 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1309 } else { 1310 return 0; 1311 } 1312} 1313 1314void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 // Don't apply master volume in SW if our HAL can do it for us. 1318 if (mOutput && mOutput->audioHwDev && 1319 mOutput->audioHwDev->canSetMasterVolume()) { 1320 mMasterVolume = 1.0; 1321 } else { 1322 mMasterVolume = value; 1323 } 1324} 1325 1326void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1327{ 1328 Mutex::Autolock _l(mLock); 1329 // Don't apply master mute in SW if our HAL can do it for us. 1330 if (mOutput && mOutput->audioHwDev && 1331 mOutput->audioHwDev->canSetMasterMute()) { 1332 mMasterMute = false; 1333 } else { 1334 mMasterMute = muted; 1335 } 1336} 1337 1338void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1339{ 1340 Mutex::Autolock _l(mLock); 1341 mStreamTypes[stream].volume = value; 1342 signal_l(); 1343} 1344 1345void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1346{ 1347 Mutex::Autolock _l(mLock); 1348 mStreamTypes[stream].mute = muted; 1349 signal_l(); 1350} 1351 1352float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1353{ 1354 Mutex::Autolock _l(mLock); 1355 return mStreamTypes[stream].volume; 1356} 1357 1358// addTrack_l() must be called with ThreadBase::mLock held 1359status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1360{ 1361 status_t status = ALREADY_EXISTS; 1362 1363 // set retry count for buffer fill 1364 track->mRetryCount = kMaxTrackStartupRetries; 1365 if (mActiveTracks.indexOf(track) < 0) { 1366 // the track is newly added, make sure it fills up all its 1367 // buffers before playing. This is to ensure the client will 1368 // effectively get the latency it requested. 1369 if (!track->isOutputTrack()) { 1370 TrackBase::track_state state = track->mState; 1371 mLock.unlock(); 1372 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1373 mLock.lock(); 1374 // abort track was stopped/paused while we released the lock 1375 if (state != track->mState) { 1376 if (status == NO_ERROR) { 1377 mLock.unlock(); 1378 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1379 mLock.lock(); 1380 } 1381 return INVALID_OPERATION; 1382 } 1383 // abort if start is rejected by audio policy manager 1384 if (status != NO_ERROR) { 1385 return PERMISSION_DENIED; 1386 } 1387#ifdef ADD_BATTERY_DATA 1388 // to track the speaker usage 1389 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1390#endif 1391 } 1392 1393 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1394 track->mResetDone = false; 1395 track->mPresentationCompleteFrames = 0; 1396 mActiveTracks.add(track); 1397 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1398 if (chain != 0) { 1399 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1400 track->sessionId()); 1401 chain->incActiveTrackCnt(); 1402 } 1403 1404 status = NO_ERROR; 1405 } 1406 1407 ALOGV("mWaitWorkCV.broadcast"); 1408 mWaitWorkCV.broadcast(); 1409 1410 return status; 1411} 1412 1413bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1414{ 1415 track->terminate(); 1416 // active tracks are removed by threadLoop() 1417 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1418 track->mState = TrackBase::STOPPED; 1419 if (!trackActive) { 1420 removeTrack_l(track); 1421 } else if (track->isFastTrack() || track->isOffloaded()) { 1422 track->mState = TrackBase::STOPPING_1; 1423 } 1424 1425 return trackActive; 1426} 1427 1428void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1429{ 1430 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1431 mTracks.remove(track); 1432 deleteTrackName_l(track->name()); 1433 // redundant as track is about to be destroyed, for dumpsys only 1434 track->mName = -1; 1435 if (track->isFastTrack()) { 1436 int index = track->mFastIndex; 1437 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1438 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1439 mFastTrackAvailMask |= 1 << index; 1440 // redundant as track is about to be destroyed, for dumpsys only 1441 track->mFastIndex = -1; 1442 } 1443 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1444 if (chain != 0) { 1445 chain->decTrackCnt(); 1446 } 1447} 1448 1449void AudioFlinger::PlaybackThread::signal_l() 1450{ 1451 // Thread could be blocked waiting for async 1452 // so signal it to handle state changes immediately 1453 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1454 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1455 mSignalPending = true; 1456 mWaitWorkCV.signal(); 1457} 1458 1459String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1460{ 1461 Mutex::Autolock _l(mLock); 1462 if (initCheck() != NO_ERROR) { 1463 return String8(); 1464 } 1465 1466 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1467 const String8 out_s8(s); 1468 free(s); 1469 return out_s8; 1470} 1471 1472// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1473void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1474 AudioSystem::OutputDescriptor desc; 1475 void *param2 = NULL; 1476 1477 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1478 param); 1479 1480 switch (event) { 1481 case AudioSystem::OUTPUT_OPENED: 1482 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1483 desc.channelMask = mChannelMask; 1484 desc.samplingRate = mSampleRate; 1485 desc.format = mFormat; 1486 desc.frameCount = mNormalFrameCount; // FIXME see 1487 // AudioFlinger::frameCount(audio_io_handle_t) 1488 desc.latency = latency(); 1489 param2 = &desc; 1490 break; 1491 1492 case AudioSystem::STREAM_CONFIG_CHANGED: 1493 param2 = ¶m; 1494 case AudioSystem::OUTPUT_CLOSED: 1495 default: 1496 break; 1497 } 1498 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1499} 1500 1501void AudioFlinger::PlaybackThread::writeCallback() 1502{ 1503 ALOG_ASSERT(mCallbackThread != 0); 1504 mCallbackThread->resetWriteBlocked(); 1505} 1506 1507void AudioFlinger::PlaybackThread::drainCallback() 1508{ 1509 ALOG_ASSERT(mCallbackThread != 0); 1510 mCallbackThread->resetDraining(); 1511} 1512 1513void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1514{ 1515 Mutex::Autolock _l(mLock); 1516 // reject out of sequence requests 1517 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1518 mWriteAckSequence &= ~1; 1519 mWaitWorkCV.signal(); 1520 } 1521} 1522 1523void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1524{ 1525 Mutex::Autolock _l(mLock); 1526 // reject out of sequence requests 1527 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1528 mDrainSequence &= ~1; 1529 mWaitWorkCV.signal(); 1530 } 1531} 1532 1533// static 1534int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1535 void *param, 1536 void *cookie) 1537{ 1538 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1539 ALOGV("asyncCallback() event %d", event); 1540 switch (event) { 1541 case STREAM_CBK_EVENT_WRITE_READY: 1542 me->writeCallback(); 1543 break; 1544 case STREAM_CBK_EVENT_DRAIN_READY: 1545 me->drainCallback(); 1546 break; 1547 default: 1548 ALOGW("asyncCallback() unknown event %d", event); 1549 break; 1550 } 1551 return 0; 1552} 1553 1554void AudioFlinger::PlaybackThread::readOutputParameters() 1555{ 1556 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1557 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1558 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1559 if (!audio_is_output_channel(mChannelMask)) { 1560 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1561 } 1562 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1563 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1564 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1565 } 1566 mChannelCount = popcount(mChannelMask); 1567 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1568 if (!audio_is_valid_format(mFormat)) { 1569 LOG_FATAL("HAL format %d not valid for output", mFormat); 1570 } 1571 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1572 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1573 mFormat); 1574 } 1575 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1576 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1577 mFrameCount = mBufferSize / mFrameSize; 1578 if (mFrameCount & 15) { 1579 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1580 mFrameCount); 1581 } 1582 1583 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1584 (mOutput->stream->set_callback != NULL)) { 1585 if (mOutput->stream->set_callback(mOutput->stream, 1586 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1587 mUseAsyncWrite = true; 1588 } 1589 } 1590 1591 // Calculate size of normal mix buffer relative to the HAL output buffer size 1592 double multiplier = 1.0; 1593 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1594 kUseFastMixer == FastMixer_Dynamic)) { 1595 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1596 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1597 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1598 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1599 maxNormalFrameCount = maxNormalFrameCount & ~15; 1600 if (maxNormalFrameCount < minNormalFrameCount) { 1601 maxNormalFrameCount = minNormalFrameCount; 1602 } 1603 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1604 if (multiplier <= 1.0) { 1605 multiplier = 1.0; 1606 } else if (multiplier <= 2.0) { 1607 if (2 * mFrameCount <= maxNormalFrameCount) { 1608 multiplier = 2.0; 1609 } else { 1610 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1611 } 1612 } else { 1613 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1614 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1615 // track, but we sometimes have to do this to satisfy the maximum frame count 1616 // constraint) 1617 // FIXME this rounding up should not be done if no HAL SRC 1618 uint32_t truncMult = (uint32_t) multiplier; 1619 if ((truncMult & 1)) { 1620 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1621 ++truncMult; 1622 } 1623 } 1624 multiplier = (double) truncMult; 1625 } 1626 } 1627 mNormalFrameCount = multiplier * mFrameCount; 1628 // round up to nearest 16 frames to satisfy AudioMixer 1629 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1630 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1631 mNormalFrameCount); 1632 1633 delete[] mMixBuffer; 1634 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1635 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1636 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1637 memset(mMixBuffer, 0, normalBufferSize); 1638 1639 // force reconfiguration of effect chains and engines to take new buffer size and audio 1640 // parameters into account 1641 // Note that mLock is not held when readOutputParameters() is called from the constructor 1642 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1643 // matter. 1644 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1645 Vector< sp<EffectChain> > effectChains = mEffectChains; 1646 for (size_t i = 0; i < effectChains.size(); i ++) { 1647 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1648 } 1649} 1650 1651 1652status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1653{ 1654 if (halFrames == NULL || dspFrames == NULL) { 1655 return BAD_VALUE; 1656 } 1657 Mutex::Autolock _l(mLock); 1658 if (initCheck() != NO_ERROR) { 1659 return INVALID_OPERATION; 1660 } 1661 size_t framesWritten = mBytesWritten / mFrameSize; 1662 *halFrames = framesWritten; 1663 1664 if (isSuspended()) { 1665 // return an estimation of rendered frames when the output is suspended 1666 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1667 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1668 return NO_ERROR; 1669 } else { 1670 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1671 } 1672} 1673 1674uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1675{ 1676 Mutex::Autolock _l(mLock); 1677 uint32_t result = 0; 1678 if (getEffectChain_l(sessionId) != 0) { 1679 result = EFFECT_SESSION; 1680 } 1681 1682 for (size_t i = 0; i < mTracks.size(); ++i) { 1683 sp<Track> track = mTracks[i]; 1684 if (sessionId == track->sessionId() && !track->isInvalid()) { 1685 result |= TRACK_SESSION; 1686 break; 1687 } 1688 } 1689 1690 return result; 1691} 1692 1693uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1694{ 1695 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1696 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1697 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1698 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1699 } 1700 for (size_t i = 0; i < mTracks.size(); i++) { 1701 sp<Track> track = mTracks[i]; 1702 if (sessionId == track->sessionId() && !track->isInvalid()) { 1703 return AudioSystem::getStrategyForStream(track->streamType()); 1704 } 1705 } 1706 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1707} 1708 1709 1710AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1711{ 1712 Mutex::Autolock _l(mLock); 1713 return mOutput; 1714} 1715 1716AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1717{ 1718 Mutex::Autolock _l(mLock); 1719 AudioStreamOut *output = mOutput; 1720 mOutput = NULL; 1721 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1722 // must push a NULL and wait for ack 1723 mOutputSink.clear(); 1724 mPipeSink.clear(); 1725 mNormalSink.clear(); 1726 return output; 1727} 1728 1729// this method must always be called either with ThreadBase mLock held or inside the thread loop 1730audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1731{ 1732 if (mOutput == NULL) { 1733 return NULL; 1734 } 1735 return &mOutput->stream->common; 1736} 1737 1738uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1739{ 1740 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1741} 1742 1743status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1744{ 1745 if (!isValidSyncEvent(event)) { 1746 return BAD_VALUE; 1747 } 1748 1749 Mutex::Autolock _l(mLock); 1750 1751 for (size_t i = 0; i < mTracks.size(); ++i) { 1752 sp<Track> track = mTracks[i]; 1753 if (event->triggerSession() == track->sessionId()) { 1754 (void) track->setSyncEvent(event); 1755 return NO_ERROR; 1756 } 1757 } 1758 1759 return NAME_NOT_FOUND; 1760} 1761 1762bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1763{ 1764 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1765} 1766 1767void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1768 const Vector< sp<Track> >& tracksToRemove) 1769{ 1770 size_t count = tracksToRemove.size(); 1771 if (count > 0) { 1772 for (size_t i = 0 ; i < count ; i++) { 1773 const sp<Track>& track = tracksToRemove.itemAt(i); 1774 if (!track->isOutputTrack()) { 1775 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1776#ifdef ADD_BATTERY_DATA 1777 // to track the speaker usage 1778 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1779#endif 1780 if (track->isTerminated()) { 1781 AudioSystem::releaseOutput(mId); 1782 } 1783 } 1784 } 1785 } 1786} 1787 1788void AudioFlinger::PlaybackThread::checkSilentMode_l() 1789{ 1790 if (!mMasterMute) { 1791 char value[PROPERTY_VALUE_MAX]; 1792 if (property_get("ro.audio.silent", value, "0") > 0) { 1793 char *endptr; 1794 unsigned long ul = strtoul(value, &endptr, 0); 1795 if (*endptr == '\0' && ul != 0) { 1796 ALOGD("Silence is golden"); 1797 // The setprop command will not allow a property to be changed after 1798 // the first time it is set, so we don't have to worry about un-muting. 1799 setMasterMute_l(true); 1800 } 1801 } 1802 } 1803} 1804 1805// shared by MIXER and DIRECT, overridden by DUPLICATING 1806ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1807{ 1808 // FIXME rewrite to reduce number of system calls 1809 mLastWriteTime = systemTime(); 1810 mInWrite = true; 1811 ssize_t bytesWritten; 1812 1813 // If an NBAIO sink is present, use it to write the normal mixer's submix 1814 if (mNormalSink != 0) { 1815#define mBitShift 2 // FIXME 1816 size_t count = mBytesRemaining >> mBitShift; 1817 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1818 ATRACE_BEGIN("write"); 1819 // update the setpoint when AudioFlinger::mScreenState changes 1820 uint32_t screenState = AudioFlinger::mScreenState; 1821 if (screenState != mScreenState) { 1822 mScreenState = screenState; 1823 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1824 if (pipe != NULL) { 1825 pipe->setAvgFrames((mScreenState & 1) ? 1826 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1827 } 1828 } 1829 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1830 ATRACE_END(); 1831 if (framesWritten > 0) { 1832 bytesWritten = framesWritten << mBitShift; 1833 } else { 1834 bytesWritten = framesWritten; 1835 } 1836 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1837 if (status == NO_ERROR) { 1838 size_t totalFramesWritten = mNormalSink->framesWritten(); 1839 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1840 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1841 mLatchDValid = true; 1842 } 1843 } 1844 // otherwise use the HAL / AudioStreamOut directly 1845 } else { 1846 // Direct output and offload threads 1847 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1848 if (mUseAsyncWrite) { 1849 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1850 mWriteAckSequence += 2; 1851 mWriteAckSequence |= 1; 1852 ALOG_ASSERT(mCallbackThread != 0); 1853 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1854 } 1855 // FIXME We should have an implementation of timestamps for direct output threads. 1856 // They are used e.g for multichannel PCM playback over HDMI. 1857 bytesWritten = mOutput->stream->write(mOutput->stream, 1858 mMixBuffer + offset, mBytesRemaining); 1859 if (mUseAsyncWrite && 1860 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1861 // do not wait for async callback in case of error of full write 1862 mWriteAckSequence &= ~1; 1863 ALOG_ASSERT(mCallbackThread != 0); 1864 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1865 } 1866 } 1867 1868 mNumWrites++; 1869 mInWrite = false; 1870 1871 return bytesWritten; 1872} 1873 1874void AudioFlinger::PlaybackThread::threadLoop_drain() 1875{ 1876 if (mOutput->stream->drain) { 1877 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1878 if (mUseAsyncWrite) { 1879 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1880 mDrainSequence |= 1; 1881 ALOG_ASSERT(mCallbackThread != 0); 1882 mCallbackThread->setDraining(mDrainSequence); 1883 } 1884 mOutput->stream->drain(mOutput->stream, 1885 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1886 : AUDIO_DRAIN_ALL); 1887 } 1888} 1889 1890void AudioFlinger::PlaybackThread::threadLoop_exit() 1891{ 1892 // Default implementation has nothing to do 1893} 1894 1895/* 1896The derived values that are cached: 1897 - mixBufferSize from frame count * frame size 1898 - activeSleepTime from activeSleepTimeUs() 1899 - idleSleepTime from idleSleepTimeUs() 1900 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1901 - maxPeriod from frame count and sample rate (MIXER only) 1902 1903The parameters that affect these derived values are: 1904 - frame count 1905 - frame size 1906 - sample rate 1907 - device type: A2DP or not 1908 - device latency 1909 - format: PCM or not 1910 - active sleep time 1911 - idle sleep time 1912*/ 1913 1914void AudioFlinger::PlaybackThread::cacheParameters_l() 1915{ 1916 mixBufferSize = mNormalFrameCount * mFrameSize; 1917 activeSleepTime = activeSleepTimeUs(); 1918 idleSleepTime = idleSleepTimeUs(); 1919} 1920 1921void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1922{ 1923 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1924 this, streamType, mTracks.size()); 1925 Mutex::Autolock _l(mLock); 1926 1927 size_t size = mTracks.size(); 1928 for (size_t i = 0; i < size; i++) { 1929 sp<Track> t = mTracks[i]; 1930 if (t->streamType() == streamType) { 1931 t->invalidate(); 1932 } 1933 } 1934} 1935 1936status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1937{ 1938 int session = chain->sessionId(); 1939 int16_t *buffer = mMixBuffer; 1940 bool ownsBuffer = false; 1941 1942 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1943 if (session > 0) { 1944 // Only one effect chain can be present in direct output thread and it uses 1945 // the mix buffer as input 1946 if (mType != DIRECT) { 1947 size_t numSamples = mNormalFrameCount * mChannelCount; 1948 buffer = new int16_t[numSamples]; 1949 memset(buffer, 0, numSamples * sizeof(int16_t)); 1950 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1951 ownsBuffer = true; 1952 } 1953 1954 // Attach all tracks with same session ID to this chain. 1955 for (size_t i = 0; i < mTracks.size(); ++i) { 1956 sp<Track> track = mTracks[i]; 1957 if (session == track->sessionId()) { 1958 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1959 buffer); 1960 track->setMainBuffer(buffer); 1961 chain->incTrackCnt(); 1962 } 1963 } 1964 1965 // indicate all active tracks in the chain 1966 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1967 sp<Track> track = mActiveTracks[i].promote(); 1968 if (track == 0) { 1969 continue; 1970 } 1971 if (session == track->sessionId()) { 1972 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1973 chain->incActiveTrackCnt(); 1974 } 1975 } 1976 } 1977 1978 chain->setInBuffer(buffer, ownsBuffer); 1979 chain->setOutBuffer(mMixBuffer); 1980 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1981 // chains list in order to be processed last as it contains output stage effects 1982 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1983 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1984 // after track specific effects and before output stage 1985 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1986 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1987 // Effect chain for other sessions are inserted at beginning of effect 1988 // chains list to be processed before output mix effects. Relative order between other 1989 // sessions is not important 1990 size_t size = mEffectChains.size(); 1991 size_t i = 0; 1992 for (i = 0; i < size; i++) { 1993 if (mEffectChains[i]->sessionId() < session) { 1994 break; 1995 } 1996 } 1997 mEffectChains.insertAt(chain, i); 1998 checkSuspendOnAddEffectChain_l(chain); 1999 2000 return NO_ERROR; 2001} 2002 2003size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2004{ 2005 int session = chain->sessionId(); 2006 2007 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2008 2009 for (size_t i = 0; i < mEffectChains.size(); i++) { 2010 if (chain == mEffectChains[i]) { 2011 mEffectChains.removeAt(i); 2012 // detach all active tracks from the chain 2013 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2014 sp<Track> track = mActiveTracks[i].promote(); 2015 if (track == 0) { 2016 continue; 2017 } 2018 if (session == track->sessionId()) { 2019 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2020 chain.get(), session); 2021 chain->decActiveTrackCnt(); 2022 } 2023 } 2024 2025 // detach all tracks with same session ID from this chain 2026 for (size_t i = 0; i < mTracks.size(); ++i) { 2027 sp<Track> track = mTracks[i]; 2028 if (session == track->sessionId()) { 2029 track->setMainBuffer(mMixBuffer); 2030 chain->decTrackCnt(); 2031 } 2032 } 2033 break; 2034 } 2035 } 2036 return mEffectChains.size(); 2037} 2038 2039status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2040 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2041{ 2042 Mutex::Autolock _l(mLock); 2043 return attachAuxEffect_l(track, EffectId); 2044} 2045 2046status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2047 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2048{ 2049 status_t status = NO_ERROR; 2050 2051 if (EffectId == 0) { 2052 track->setAuxBuffer(0, NULL); 2053 } else { 2054 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2055 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2056 if (effect != 0) { 2057 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2058 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2059 } else { 2060 status = INVALID_OPERATION; 2061 } 2062 } else { 2063 status = BAD_VALUE; 2064 } 2065 } 2066 return status; 2067} 2068 2069void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2070{ 2071 for (size_t i = 0; i < mTracks.size(); ++i) { 2072 sp<Track> track = mTracks[i]; 2073 if (track->auxEffectId() == effectId) { 2074 attachAuxEffect_l(track, 0); 2075 } 2076 } 2077} 2078 2079bool AudioFlinger::PlaybackThread::threadLoop() 2080{ 2081 Vector< sp<Track> > tracksToRemove; 2082 2083 standbyTime = systemTime(); 2084 2085 // MIXER 2086 nsecs_t lastWarning = 0; 2087 2088 // DUPLICATING 2089 // FIXME could this be made local to while loop? 2090 writeFrames = 0; 2091 2092 cacheParameters_l(); 2093 sleepTime = idleSleepTime; 2094 2095 if (mType == MIXER) { 2096 sleepTimeShift = 0; 2097 } 2098 2099 CpuStats cpuStats; 2100 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2101 2102 acquireWakeLock(); 2103 2104 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2105 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2106 // and then that string will be logged at the next convenient opportunity. 2107 const char *logString = NULL; 2108 2109 while (!exitPending()) 2110 { 2111 cpuStats.sample(myName); 2112 2113 Vector< sp<EffectChain> > effectChains; 2114 2115 processConfigEvents(); 2116 2117 { // scope for mLock 2118 2119 Mutex::Autolock _l(mLock); 2120 2121 if (logString != NULL) { 2122 mNBLogWriter->logTimestamp(); 2123 mNBLogWriter->log(logString); 2124 logString = NULL; 2125 } 2126 2127 if (mLatchDValid) { 2128 mLatchQ = mLatchD; 2129 mLatchDValid = false; 2130 mLatchQValid = true; 2131 } 2132 2133 if (checkForNewParameters_l()) { 2134 cacheParameters_l(); 2135 } 2136 2137 saveOutputTracks(); 2138 2139 if (mSignalPending) { 2140 // A signal was raised while we were unlocked 2141 mSignalPending = false; 2142 } else if (waitingAsyncCallback_l()) { 2143 if (exitPending()) { 2144 break; 2145 } 2146 releaseWakeLock_l(); 2147 ALOGV("wait async completion"); 2148 mWaitWorkCV.wait(mLock); 2149 ALOGV("async completion/wake"); 2150 acquireWakeLock_l(); 2151 standbyTime = systemTime() + standbyDelay; 2152 sleepTime = 0; 2153 if (exitPending()) { 2154 break; 2155 } 2156 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2157 isSuspended()) { 2158 // put audio hardware into standby after short delay 2159 if (shouldStandby_l()) { 2160 2161 threadLoop_standby(); 2162 2163 mStandby = true; 2164 } 2165 2166 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2167 // we're about to wait, flush the binder command buffer 2168 IPCThreadState::self()->flushCommands(); 2169 2170 clearOutputTracks(); 2171 2172 if (exitPending()) { 2173 break; 2174 } 2175 2176 releaseWakeLock_l(); 2177 // wait until we have something to do... 2178 ALOGV("%s going to sleep", myName.string()); 2179 mWaitWorkCV.wait(mLock); 2180 ALOGV("%s waking up", myName.string()); 2181 acquireWakeLock_l(); 2182 2183 mMixerStatus = MIXER_IDLE; 2184 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2185 mBytesWritten = 0; 2186 mBytesRemaining = 0; 2187 checkSilentMode_l(); 2188 2189 standbyTime = systemTime() + standbyDelay; 2190 sleepTime = idleSleepTime; 2191 if (mType == MIXER) { 2192 sleepTimeShift = 0; 2193 } 2194 2195 continue; 2196 } 2197 } 2198 2199 // mMixerStatusIgnoringFastTracks is also updated internally 2200 mMixerStatus = prepareTracks_l(&tracksToRemove); 2201 2202 // prevent any changes in effect chain list and in each effect chain 2203 // during mixing and effect process as the audio buffers could be deleted 2204 // or modified if an effect is created or deleted 2205 lockEffectChains_l(effectChains); 2206 } 2207 2208 if (mBytesRemaining == 0) { 2209 mCurrentWriteLength = 0; 2210 if (mMixerStatus == MIXER_TRACKS_READY) { 2211 // threadLoop_mix() sets mCurrentWriteLength 2212 threadLoop_mix(); 2213 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2214 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2215 // threadLoop_sleepTime sets sleepTime to 0 if data 2216 // must be written to HAL 2217 threadLoop_sleepTime(); 2218 if (sleepTime == 0) { 2219 mCurrentWriteLength = mixBufferSize; 2220 } 2221 } 2222 mBytesRemaining = mCurrentWriteLength; 2223 if (isSuspended()) { 2224 sleepTime = suspendSleepTimeUs(); 2225 // simulate write to HAL when suspended 2226 mBytesWritten += mixBufferSize; 2227 mBytesRemaining = 0; 2228 } 2229 2230 // only process effects if we're going to write 2231 if (sleepTime == 0) { 2232 for (size_t i = 0; i < effectChains.size(); i ++) { 2233 effectChains[i]->process_l(); 2234 } 2235 } 2236 } 2237 2238 // enable changes in effect chain 2239 unlockEffectChains(effectChains); 2240 2241 if (!waitingAsyncCallback()) { 2242 // sleepTime == 0 means we must write to audio hardware 2243 if (sleepTime == 0) { 2244 if (mBytesRemaining) { 2245 ssize_t ret = threadLoop_write(); 2246 if (ret < 0) { 2247 mBytesRemaining = 0; 2248 } else { 2249 mBytesWritten += ret; 2250 mBytesRemaining -= ret; 2251 } 2252 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2253 (mMixerStatus == MIXER_DRAIN_ALL)) { 2254 threadLoop_drain(); 2255 } 2256if (mType == MIXER) { 2257 // write blocked detection 2258 nsecs_t now = systemTime(); 2259 nsecs_t delta = now - mLastWriteTime; 2260 if (!mStandby && delta > maxPeriod) { 2261 mNumDelayedWrites++; 2262 if ((now - lastWarning) > kWarningThrottleNs) { 2263 ATRACE_NAME("underrun"); 2264 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2265 ns2ms(delta), mNumDelayedWrites, this); 2266 lastWarning = now; 2267 } 2268 } 2269} 2270 2271 mStandby = false; 2272 } else { 2273 usleep(sleepTime); 2274 } 2275 } 2276 2277 // Finally let go of removed track(s), without the lock held 2278 // since we can't guarantee the destructors won't acquire that 2279 // same lock. This will also mutate and push a new fast mixer state. 2280 threadLoop_removeTracks(tracksToRemove); 2281 tracksToRemove.clear(); 2282 2283 // FIXME I don't understand the need for this here; 2284 // it was in the original code but maybe the 2285 // assignment in saveOutputTracks() makes this unnecessary? 2286 clearOutputTracks(); 2287 2288 // Effect chains will be actually deleted here if they were removed from 2289 // mEffectChains list during mixing or effects processing 2290 effectChains.clear(); 2291 2292 // FIXME Note that the above .clear() is no longer necessary since effectChains 2293 // is now local to this block, but will keep it for now (at least until merge done). 2294 } 2295 2296 threadLoop_exit(); 2297 2298 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2299 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2300 // put output stream into standby mode 2301 if (!mStandby) { 2302 mOutput->stream->common.standby(&mOutput->stream->common); 2303 } 2304 } 2305 2306 releaseWakeLock(); 2307 2308 ALOGV("Thread %p type %d exiting", this, mType); 2309 return false; 2310} 2311 2312// removeTracks_l() must be called with ThreadBase::mLock held 2313void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2314{ 2315 size_t count = tracksToRemove.size(); 2316 if (count > 0) { 2317 for (size_t i=0 ; i<count ; i++) { 2318 const sp<Track>& track = tracksToRemove.itemAt(i); 2319 mActiveTracks.remove(track); 2320 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2321 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2322 if (chain != 0) { 2323 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2324 track->sessionId()); 2325 chain->decActiveTrackCnt(); 2326 } 2327 if (track->isTerminated()) { 2328 removeTrack_l(track); 2329 } 2330 } 2331 } 2332 2333} 2334 2335// ---------------------------------------------------------------------------- 2336 2337AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2338 audio_io_handle_t id, audio_devices_t device, type_t type) 2339 : PlaybackThread(audioFlinger, output, id, device, type), 2340 // mAudioMixer below 2341 // mFastMixer below 2342 mFastMixerFutex(0) 2343 // mOutputSink below 2344 // mPipeSink below 2345 // mNormalSink below 2346{ 2347 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2348 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2349 "mFrameCount=%d, mNormalFrameCount=%d", 2350 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2351 mNormalFrameCount); 2352 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2353 2354 // FIXME - Current mixer implementation only supports stereo output 2355 if (mChannelCount != FCC_2) { 2356 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2357 } 2358 2359 // create an NBAIO sink for the HAL output stream, and negotiate 2360 mOutputSink = new AudioStreamOutSink(output->stream); 2361 size_t numCounterOffers = 0; 2362 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2363 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2364 ALOG_ASSERT(index == 0); 2365 2366 // initialize fast mixer depending on configuration 2367 bool initFastMixer; 2368 switch (kUseFastMixer) { 2369 case FastMixer_Never: 2370 initFastMixer = false; 2371 break; 2372 case FastMixer_Always: 2373 initFastMixer = true; 2374 break; 2375 case FastMixer_Static: 2376 case FastMixer_Dynamic: 2377 initFastMixer = mFrameCount < mNormalFrameCount; 2378 break; 2379 } 2380 if (initFastMixer) { 2381 2382 // create a MonoPipe to connect our submix to FastMixer 2383 NBAIO_Format format = mOutputSink->format(); 2384 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2385 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2386 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2387 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2388 const NBAIO_Format offers[1] = {format}; 2389 size_t numCounterOffers = 0; 2390 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2391 ALOG_ASSERT(index == 0); 2392 monoPipe->setAvgFrames((mScreenState & 1) ? 2393 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2394 mPipeSink = monoPipe; 2395 2396#ifdef TEE_SINK 2397 if (mTeeSinkOutputEnabled) { 2398 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2399 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2400 numCounterOffers = 0; 2401 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2402 ALOG_ASSERT(index == 0); 2403 mTeeSink = teeSink; 2404 PipeReader *teeSource = new PipeReader(*teeSink); 2405 numCounterOffers = 0; 2406 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2407 ALOG_ASSERT(index == 0); 2408 mTeeSource = teeSource; 2409 } 2410#endif 2411 2412 // create fast mixer and configure it initially with just one fast track for our submix 2413 mFastMixer = new FastMixer(); 2414 FastMixerStateQueue *sq = mFastMixer->sq(); 2415#ifdef STATE_QUEUE_DUMP 2416 sq->setObserverDump(&mStateQueueObserverDump); 2417 sq->setMutatorDump(&mStateQueueMutatorDump); 2418#endif 2419 FastMixerState *state = sq->begin(); 2420 FastTrack *fastTrack = &state->mFastTracks[0]; 2421 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2422 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2423 fastTrack->mVolumeProvider = NULL; 2424 fastTrack->mGeneration++; 2425 state->mFastTracksGen++; 2426 state->mTrackMask = 1; 2427 // fast mixer will use the HAL output sink 2428 state->mOutputSink = mOutputSink.get(); 2429 state->mOutputSinkGen++; 2430 state->mFrameCount = mFrameCount; 2431 state->mCommand = FastMixerState::COLD_IDLE; 2432 // already done in constructor initialization list 2433 //mFastMixerFutex = 0; 2434 state->mColdFutexAddr = &mFastMixerFutex; 2435 state->mColdGen++; 2436 state->mDumpState = &mFastMixerDumpState; 2437#ifdef TEE_SINK 2438 state->mTeeSink = mTeeSink.get(); 2439#endif 2440 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2441 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2442 sq->end(); 2443 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2444 2445 // start the fast mixer 2446 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2447 pid_t tid = mFastMixer->getTid(); 2448 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2449 if (err != 0) { 2450 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2451 kPriorityFastMixer, getpid_cached, tid, err); 2452 } 2453 2454#ifdef AUDIO_WATCHDOG 2455 // create and start the watchdog 2456 mAudioWatchdog = new AudioWatchdog(); 2457 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2458 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2459 tid = mAudioWatchdog->getTid(); 2460 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2461 if (err != 0) { 2462 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2463 kPriorityFastMixer, getpid_cached, tid, err); 2464 } 2465#endif 2466 2467 } else { 2468 mFastMixer = NULL; 2469 } 2470 2471 switch (kUseFastMixer) { 2472 case FastMixer_Never: 2473 case FastMixer_Dynamic: 2474 mNormalSink = mOutputSink; 2475 break; 2476 case FastMixer_Always: 2477 mNormalSink = mPipeSink; 2478 break; 2479 case FastMixer_Static: 2480 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2481 break; 2482 } 2483} 2484 2485AudioFlinger::MixerThread::~MixerThread() 2486{ 2487 if (mFastMixer != NULL) { 2488 FastMixerStateQueue *sq = mFastMixer->sq(); 2489 FastMixerState *state = sq->begin(); 2490 if (state->mCommand == FastMixerState::COLD_IDLE) { 2491 int32_t old = android_atomic_inc(&mFastMixerFutex); 2492 if (old == -1) { 2493 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2494 } 2495 } 2496 state->mCommand = FastMixerState::EXIT; 2497 sq->end(); 2498 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2499 mFastMixer->join(); 2500 // Though the fast mixer thread has exited, it's state queue is still valid. 2501 // We'll use that extract the final state which contains one remaining fast track 2502 // corresponding to our sub-mix. 2503 state = sq->begin(); 2504 ALOG_ASSERT(state->mTrackMask == 1); 2505 FastTrack *fastTrack = &state->mFastTracks[0]; 2506 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2507 delete fastTrack->mBufferProvider; 2508 sq->end(false /*didModify*/); 2509 delete mFastMixer; 2510#ifdef AUDIO_WATCHDOG 2511 if (mAudioWatchdog != 0) { 2512 mAudioWatchdog->requestExit(); 2513 mAudioWatchdog->requestExitAndWait(); 2514 mAudioWatchdog.clear(); 2515 } 2516#endif 2517 } 2518 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2519 delete mAudioMixer; 2520} 2521 2522 2523uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2524{ 2525 if (mFastMixer != NULL) { 2526 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2527 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2528 } 2529 return latency; 2530} 2531 2532 2533void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2534{ 2535 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2536} 2537 2538ssize_t AudioFlinger::MixerThread::threadLoop_write() 2539{ 2540 // FIXME we should only do one push per cycle; confirm this is true 2541 // Start the fast mixer if it's not already running 2542 if (mFastMixer != NULL) { 2543 FastMixerStateQueue *sq = mFastMixer->sq(); 2544 FastMixerState *state = sq->begin(); 2545 if (state->mCommand != FastMixerState::MIX_WRITE && 2546 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2547 if (state->mCommand == FastMixerState::COLD_IDLE) { 2548 int32_t old = android_atomic_inc(&mFastMixerFutex); 2549 if (old == -1) { 2550 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2551 } 2552#ifdef AUDIO_WATCHDOG 2553 if (mAudioWatchdog != 0) { 2554 mAudioWatchdog->resume(); 2555 } 2556#endif 2557 } 2558 state->mCommand = FastMixerState::MIX_WRITE; 2559 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2560 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2561 sq->end(); 2562 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2563 if (kUseFastMixer == FastMixer_Dynamic) { 2564 mNormalSink = mPipeSink; 2565 } 2566 } else { 2567 sq->end(false /*didModify*/); 2568 } 2569 } 2570 return PlaybackThread::threadLoop_write(); 2571} 2572 2573void AudioFlinger::MixerThread::threadLoop_standby() 2574{ 2575 // Idle the fast mixer if it's currently running 2576 if (mFastMixer != NULL) { 2577 FastMixerStateQueue *sq = mFastMixer->sq(); 2578 FastMixerState *state = sq->begin(); 2579 if (!(state->mCommand & FastMixerState::IDLE)) { 2580 state->mCommand = FastMixerState::COLD_IDLE; 2581 state->mColdFutexAddr = &mFastMixerFutex; 2582 state->mColdGen++; 2583 mFastMixerFutex = 0; 2584 sq->end(); 2585 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2586 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2587 if (kUseFastMixer == FastMixer_Dynamic) { 2588 mNormalSink = mOutputSink; 2589 } 2590#ifdef AUDIO_WATCHDOG 2591 if (mAudioWatchdog != 0) { 2592 mAudioWatchdog->pause(); 2593 } 2594#endif 2595 } else { 2596 sq->end(false /*didModify*/); 2597 } 2598 } 2599 PlaybackThread::threadLoop_standby(); 2600} 2601 2602// Empty implementation for standard mixer 2603// Overridden for offloaded playback 2604void AudioFlinger::PlaybackThread::flushOutput_l() 2605{ 2606} 2607 2608bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2609{ 2610 return false; 2611} 2612 2613bool AudioFlinger::PlaybackThread::shouldStandby_l() 2614{ 2615 return !mStandby; 2616} 2617 2618bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2619{ 2620 Mutex::Autolock _l(mLock); 2621 return waitingAsyncCallback_l(); 2622} 2623 2624// shared by MIXER and DIRECT, overridden by DUPLICATING 2625void AudioFlinger::PlaybackThread::threadLoop_standby() 2626{ 2627 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2628 mOutput->stream->common.standby(&mOutput->stream->common); 2629 if (mUseAsyncWrite != 0) { 2630 // discard any pending drain or write ack by incrementing sequence 2631 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2632 mDrainSequence = (mDrainSequence + 2) & ~1; 2633 ALOG_ASSERT(mCallbackThread != 0); 2634 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2635 mCallbackThread->setDraining(mDrainSequence); 2636 } 2637} 2638 2639void AudioFlinger::MixerThread::threadLoop_mix() 2640{ 2641 // obtain the presentation timestamp of the next output buffer 2642 int64_t pts; 2643 status_t status = INVALID_OPERATION; 2644 2645 if (mNormalSink != 0) { 2646 status = mNormalSink->getNextWriteTimestamp(&pts); 2647 } else { 2648 status = mOutputSink->getNextWriteTimestamp(&pts); 2649 } 2650 2651 if (status != NO_ERROR) { 2652 pts = AudioBufferProvider::kInvalidPTS; 2653 } 2654 2655 // mix buffers... 2656 mAudioMixer->process(pts); 2657 mCurrentWriteLength = mixBufferSize; 2658 // increase sleep time progressively when application underrun condition clears. 2659 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2660 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2661 // such that we would underrun the audio HAL. 2662 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2663 sleepTimeShift--; 2664 } 2665 sleepTime = 0; 2666 standbyTime = systemTime() + standbyDelay; 2667 //TODO: delay standby when effects have a tail 2668} 2669 2670void AudioFlinger::MixerThread::threadLoop_sleepTime() 2671{ 2672 // If no tracks are ready, sleep once for the duration of an output 2673 // buffer size, then write 0s to the output 2674 if (sleepTime == 0) { 2675 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2676 sleepTime = activeSleepTime >> sleepTimeShift; 2677 if (sleepTime < kMinThreadSleepTimeUs) { 2678 sleepTime = kMinThreadSleepTimeUs; 2679 } 2680 // reduce sleep time in case of consecutive application underruns to avoid 2681 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2682 // duration we would end up writing less data than needed by the audio HAL if 2683 // the condition persists. 2684 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2685 sleepTimeShift++; 2686 } 2687 } else { 2688 sleepTime = idleSleepTime; 2689 } 2690 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2691 memset(mMixBuffer, 0, mixBufferSize); 2692 sleepTime = 0; 2693 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2694 "anticipated start"); 2695 } 2696 // TODO add standby time extension fct of effect tail 2697} 2698 2699// prepareTracks_l() must be called with ThreadBase::mLock held 2700AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2701 Vector< sp<Track> > *tracksToRemove) 2702{ 2703 2704 mixer_state mixerStatus = MIXER_IDLE; 2705 // find out which tracks need to be processed 2706 size_t count = mActiveTracks.size(); 2707 size_t mixedTracks = 0; 2708 size_t tracksWithEffect = 0; 2709 // counts only _active_ fast tracks 2710 size_t fastTracks = 0; 2711 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2712 2713 float masterVolume = mMasterVolume; 2714 bool masterMute = mMasterMute; 2715 2716 if (masterMute) { 2717 masterVolume = 0; 2718 } 2719 // Delegate master volume control to effect in output mix effect chain if needed 2720 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2721 if (chain != 0) { 2722 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2723 chain->setVolume_l(&v, &v); 2724 masterVolume = (float)((v + (1 << 23)) >> 24); 2725 chain.clear(); 2726 } 2727 2728 // prepare a new state to push 2729 FastMixerStateQueue *sq = NULL; 2730 FastMixerState *state = NULL; 2731 bool didModify = false; 2732 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2733 if (mFastMixer != NULL) { 2734 sq = mFastMixer->sq(); 2735 state = sq->begin(); 2736 } 2737 2738 for (size_t i=0 ; i<count ; i++) { 2739 const sp<Track> t = mActiveTracks[i].promote(); 2740 if (t == 0) { 2741 continue; 2742 } 2743 2744 // this const just means the local variable doesn't change 2745 Track* const track = t.get(); 2746 2747 // process fast tracks 2748 if (track->isFastTrack()) { 2749 2750 // It's theoretically possible (though unlikely) for a fast track to be created 2751 // and then removed within the same normal mix cycle. This is not a problem, as 2752 // the track never becomes active so it's fast mixer slot is never touched. 2753 // The converse, of removing an (active) track and then creating a new track 2754 // at the identical fast mixer slot within the same normal mix cycle, 2755 // is impossible because the slot isn't marked available until the end of each cycle. 2756 int j = track->mFastIndex; 2757 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2758 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2759 FastTrack *fastTrack = &state->mFastTracks[j]; 2760 2761 // Determine whether the track is currently in underrun condition, 2762 // and whether it had a recent underrun. 2763 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2764 FastTrackUnderruns underruns = ftDump->mUnderruns; 2765 uint32_t recentFull = (underruns.mBitFields.mFull - 2766 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2767 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2768 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2769 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2770 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2771 uint32_t recentUnderruns = recentPartial + recentEmpty; 2772 track->mObservedUnderruns = underruns; 2773 // don't count underruns that occur while stopping or pausing 2774 // or stopped which can occur when flush() is called while active 2775 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2776 recentUnderruns > 0) { 2777 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2778 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2779 } 2780 2781 // This is similar to the state machine for normal tracks, 2782 // with a few modifications for fast tracks. 2783 bool isActive = true; 2784 switch (track->mState) { 2785 case TrackBase::STOPPING_1: 2786 // track stays active in STOPPING_1 state until first underrun 2787 if (recentUnderruns > 0 || track->isTerminated()) { 2788 track->mState = TrackBase::STOPPING_2; 2789 } 2790 break; 2791 case TrackBase::PAUSING: 2792 // ramp down is not yet implemented 2793 track->setPaused(); 2794 break; 2795 case TrackBase::RESUMING: 2796 // ramp up is not yet implemented 2797 track->mState = TrackBase::ACTIVE; 2798 break; 2799 case TrackBase::ACTIVE: 2800 if (recentFull > 0 || recentPartial > 0) { 2801 // track has provided at least some frames recently: reset retry count 2802 track->mRetryCount = kMaxTrackRetries; 2803 } 2804 if (recentUnderruns == 0) { 2805 // no recent underruns: stay active 2806 break; 2807 } 2808 // there has recently been an underrun of some kind 2809 if (track->sharedBuffer() == 0) { 2810 // were any of the recent underruns "empty" (no frames available)? 2811 if (recentEmpty == 0) { 2812 // no, then ignore the partial underruns as they are allowed indefinitely 2813 break; 2814 } 2815 // there has recently been an "empty" underrun: decrement the retry counter 2816 if (--(track->mRetryCount) > 0) { 2817 break; 2818 } 2819 // indicate to client process that the track was disabled because of underrun; 2820 // it will then automatically call start() when data is available 2821 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2822 // remove from active list, but state remains ACTIVE [confusing but true] 2823 isActive = false; 2824 break; 2825 } 2826 // fall through 2827 case TrackBase::STOPPING_2: 2828 case TrackBase::PAUSED: 2829 case TrackBase::STOPPED: 2830 case TrackBase::FLUSHED: // flush() while active 2831 // Check for presentation complete if track is inactive 2832 // We have consumed all the buffers of this track. 2833 // This would be incomplete if we auto-paused on underrun 2834 { 2835 size_t audioHALFrames = 2836 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2837 size_t framesWritten = mBytesWritten / mFrameSize; 2838 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2839 // track stays in active list until presentation is complete 2840 break; 2841 } 2842 } 2843 if (track->isStopping_2()) { 2844 track->mState = TrackBase::STOPPED; 2845 } 2846 if (track->isStopped()) { 2847 // Can't reset directly, as fast mixer is still polling this track 2848 // track->reset(); 2849 // So instead mark this track as needing to be reset after push with ack 2850 resetMask |= 1 << i; 2851 } 2852 isActive = false; 2853 break; 2854 case TrackBase::IDLE: 2855 default: 2856 LOG_FATAL("unexpected track state %d", track->mState); 2857 } 2858 2859 if (isActive) { 2860 // was it previously inactive? 2861 if (!(state->mTrackMask & (1 << j))) { 2862 ExtendedAudioBufferProvider *eabp = track; 2863 VolumeProvider *vp = track; 2864 fastTrack->mBufferProvider = eabp; 2865 fastTrack->mVolumeProvider = vp; 2866 fastTrack->mSampleRate = track->mSampleRate; 2867 fastTrack->mChannelMask = track->mChannelMask; 2868 fastTrack->mGeneration++; 2869 state->mTrackMask |= 1 << j; 2870 didModify = true; 2871 // no acknowledgement required for newly active tracks 2872 } 2873 // cache the combined master volume and stream type volume for fast mixer; this 2874 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2875 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2876 ++fastTracks; 2877 } else { 2878 // was it previously active? 2879 if (state->mTrackMask & (1 << j)) { 2880 fastTrack->mBufferProvider = NULL; 2881 fastTrack->mGeneration++; 2882 state->mTrackMask &= ~(1 << j); 2883 didModify = true; 2884 // If any fast tracks were removed, we must wait for acknowledgement 2885 // because we're about to decrement the last sp<> on those tracks. 2886 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2887 } else { 2888 LOG_FATAL("fast track %d should have been active", j); 2889 } 2890 tracksToRemove->add(track); 2891 // Avoids a misleading display in dumpsys 2892 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2893 } 2894 continue; 2895 } 2896 2897 { // local variable scope to avoid goto warning 2898 2899 audio_track_cblk_t* cblk = track->cblk(); 2900 2901 // The first time a track is added we wait 2902 // for all its buffers to be filled before processing it 2903 int name = track->name(); 2904 // make sure that we have enough frames to mix one full buffer. 2905 // enforce this condition only once to enable draining the buffer in case the client 2906 // app does not call stop() and relies on underrun to stop: 2907 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2908 // during last round 2909 size_t desiredFrames; 2910 uint32_t sr = track->sampleRate(); 2911 if (sr == mSampleRate) { 2912 desiredFrames = mNormalFrameCount; 2913 } else { 2914 // +1 for rounding and +1 for additional sample needed for interpolation 2915 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2916 // add frames already consumed but not yet released by the resampler 2917 // because mAudioTrackServerProxy->framesReady() will include these frames 2918 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2919 // the minimum track buffer size is normally twice the number of frames necessary 2920 // to fill one buffer and the resampler should not leave more than one buffer worth 2921 // of unreleased frames after each pass, but just in case... 2922 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2923 } 2924 uint32_t minFrames = 1; 2925 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2926 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2927 minFrames = desiredFrames; 2928 } 2929 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2930 size_t framesReady; 2931 if (track->sharedBuffer() == 0) { 2932 framesReady = track->framesReady(); 2933 } else if (track->isStopped()) { 2934 framesReady = 0; 2935 } else { 2936 framesReady = 1; 2937 } 2938 if ((framesReady >= minFrames) && track->isReady() && 2939 !track->isPaused() && !track->isTerminated()) 2940 { 2941 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2942 2943 mixedTracks++; 2944 2945 // track->mainBuffer() != mMixBuffer means there is an effect chain 2946 // connected to the track 2947 chain.clear(); 2948 if (track->mainBuffer() != mMixBuffer) { 2949 chain = getEffectChain_l(track->sessionId()); 2950 // Delegate volume control to effect in track effect chain if needed 2951 if (chain != 0) { 2952 tracksWithEffect++; 2953 } else { 2954 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2955 "session %d", 2956 name, track->sessionId()); 2957 } 2958 } 2959 2960 2961 int param = AudioMixer::VOLUME; 2962 if (track->mFillingUpStatus == Track::FS_FILLED) { 2963 // no ramp for the first volume setting 2964 track->mFillingUpStatus = Track::FS_ACTIVE; 2965 if (track->mState == TrackBase::RESUMING) { 2966 track->mState = TrackBase::ACTIVE; 2967 param = AudioMixer::RAMP_VOLUME; 2968 } 2969 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2970 // FIXME should not make a decision based on mServer 2971 } else if (cblk->mServer != 0) { 2972 // If the track is stopped before the first frame was mixed, 2973 // do not apply ramp 2974 param = AudioMixer::RAMP_VOLUME; 2975 } 2976 2977 // compute volume for this track 2978 uint32_t vl, vr, va; 2979 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2980 vl = vr = va = 0; 2981 if (track->isPausing()) { 2982 track->setPaused(); 2983 } 2984 } else { 2985 2986 // read original volumes with volume control 2987 float typeVolume = mStreamTypes[track->streamType()].volume; 2988 float v = masterVolume * typeVolume; 2989 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2990 uint32_t vlr = proxy->getVolumeLR(); 2991 vl = vlr & 0xFFFF; 2992 vr = vlr >> 16; 2993 // track volumes come from shared memory, so can't be trusted and must be clamped 2994 if (vl > MAX_GAIN_INT) { 2995 ALOGV("Track left volume out of range: %04X", vl); 2996 vl = MAX_GAIN_INT; 2997 } 2998 if (vr > MAX_GAIN_INT) { 2999 ALOGV("Track right volume out of range: %04X", vr); 3000 vr = MAX_GAIN_INT; 3001 } 3002 // now apply the master volume and stream type volume 3003 vl = (uint32_t)(v * vl) << 12; 3004 vr = (uint32_t)(v * vr) << 12; 3005 // assuming master volume and stream type volume each go up to 1.0, 3006 // vl and vr are now in 8.24 format 3007 3008 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3009 // send level comes from shared memory and so may be corrupt 3010 if (sendLevel > MAX_GAIN_INT) { 3011 ALOGV("Track send level out of range: %04X", sendLevel); 3012 sendLevel = MAX_GAIN_INT; 3013 } 3014 va = (uint32_t)(v * sendLevel); 3015 } 3016 3017 // Delegate volume control to effect in track effect chain if needed 3018 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3019 // Do not ramp volume if volume is controlled by effect 3020 param = AudioMixer::VOLUME; 3021 track->mHasVolumeController = true; 3022 } else { 3023 // force no volume ramp when volume controller was just disabled or removed 3024 // from effect chain to avoid volume spike 3025 if (track->mHasVolumeController) { 3026 param = AudioMixer::VOLUME; 3027 } 3028 track->mHasVolumeController = false; 3029 } 3030 3031 // Convert volumes from 8.24 to 4.12 format 3032 // This additional clamping is needed in case chain->setVolume_l() overshot 3033 vl = (vl + (1 << 11)) >> 12; 3034 if (vl > MAX_GAIN_INT) { 3035 vl = MAX_GAIN_INT; 3036 } 3037 vr = (vr + (1 << 11)) >> 12; 3038 if (vr > MAX_GAIN_INT) { 3039 vr = MAX_GAIN_INT; 3040 } 3041 3042 if (va > MAX_GAIN_INT) { 3043 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3044 } 3045 3046 // XXX: these things DON'T need to be done each time 3047 mAudioMixer->setBufferProvider(name, track); 3048 mAudioMixer->enable(name); 3049 3050 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3051 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3052 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3053 mAudioMixer->setParameter( 3054 name, 3055 AudioMixer::TRACK, 3056 AudioMixer::FORMAT, (void *)track->format()); 3057 mAudioMixer->setParameter( 3058 name, 3059 AudioMixer::TRACK, 3060 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3061 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3062 uint32_t maxSampleRate = mSampleRate * 2; 3063 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3064 if (reqSampleRate == 0) { 3065 reqSampleRate = mSampleRate; 3066 } else if (reqSampleRate > maxSampleRate) { 3067 reqSampleRate = maxSampleRate; 3068 } 3069 mAudioMixer->setParameter( 3070 name, 3071 AudioMixer::RESAMPLE, 3072 AudioMixer::SAMPLE_RATE, 3073 (void *)reqSampleRate); 3074 mAudioMixer->setParameter( 3075 name, 3076 AudioMixer::TRACK, 3077 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3078 mAudioMixer->setParameter( 3079 name, 3080 AudioMixer::TRACK, 3081 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3082 3083 // reset retry count 3084 track->mRetryCount = kMaxTrackRetries; 3085 3086 // If one track is ready, set the mixer ready if: 3087 // - the mixer was not ready during previous round OR 3088 // - no other track is not ready 3089 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3090 mixerStatus != MIXER_TRACKS_ENABLED) { 3091 mixerStatus = MIXER_TRACKS_READY; 3092 } 3093 } else { 3094 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3095 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3096 } 3097 // clear effect chain input buffer if an active track underruns to avoid sending 3098 // previous audio buffer again to effects 3099 chain = getEffectChain_l(track->sessionId()); 3100 if (chain != 0) { 3101 chain->clearInputBuffer(); 3102 } 3103 3104 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3105 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3106 track->isStopped() || track->isPaused()) { 3107 // We have consumed all the buffers of this track. 3108 // Remove it from the list of active tracks. 3109 // TODO: use actual buffer filling status instead of latency when available from 3110 // audio HAL 3111 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3112 size_t framesWritten = mBytesWritten / mFrameSize; 3113 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3114 if (track->isStopped()) { 3115 track->reset(); 3116 } 3117 tracksToRemove->add(track); 3118 } 3119 } else { 3120 // No buffers for this track. Give it a few chances to 3121 // fill a buffer, then remove it from active list. 3122 if (--(track->mRetryCount) <= 0) { 3123 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3124 tracksToRemove->add(track); 3125 // indicate to client process that the track was disabled because of underrun; 3126 // it will then automatically call start() when data is available 3127 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3128 // If one track is not ready, mark the mixer also not ready if: 3129 // - the mixer was ready during previous round OR 3130 // - no other track is ready 3131 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3132 mixerStatus != MIXER_TRACKS_READY) { 3133 mixerStatus = MIXER_TRACKS_ENABLED; 3134 } 3135 } 3136 mAudioMixer->disable(name); 3137 } 3138 3139 } // local variable scope to avoid goto warning 3140track_is_ready: ; 3141 3142 } 3143 3144 // Push the new FastMixer state if necessary 3145 bool pauseAudioWatchdog = false; 3146 if (didModify) { 3147 state->mFastTracksGen++; 3148 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3149 if (kUseFastMixer == FastMixer_Dynamic && 3150 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3151 state->mCommand = FastMixerState::COLD_IDLE; 3152 state->mColdFutexAddr = &mFastMixerFutex; 3153 state->mColdGen++; 3154 mFastMixerFutex = 0; 3155 if (kUseFastMixer == FastMixer_Dynamic) { 3156 mNormalSink = mOutputSink; 3157 } 3158 // If we go into cold idle, need to wait for acknowledgement 3159 // so that fast mixer stops doing I/O. 3160 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3161 pauseAudioWatchdog = true; 3162 } 3163 } 3164 if (sq != NULL) { 3165 sq->end(didModify); 3166 sq->push(block); 3167 } 3168#ifdef AUDIO_WATCHDOG 3169 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3170 mAudioWatchdog->pause(); 3171 } 3172#endif 3173 3174 // Now perform the deferred reset on fast tracks that have stopped 3175 while (resetMask != 0) { 3176 size_t i = __builtin_ctz(resetMask); 3177 ALOG_ASSERT(i < count); 3178 resetMask &= ~(1 << i); 3179 sp<Track> t = mActiveTracks[i].promote(); 3180 if (t == 0) { 3181 continue; 3182 } 3183 Track* track = t.get(); 3184 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3185 track->reset(); 3186 } 3187 3188 // remove all the tracks that need to be... 3189 removeTracks_l(*tracksToRemove); 3190 3191 // mix buffer must be cleared if all tracks are connected to an 3192 // effect chain as in this case the mixer will not write to 3193 // mix buffer and track effects will accumulate into it 3194 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3195 (mixedTracks == 0 && fastTracks > 0))) { 3196 // FIXME as a performance optimization, should remember previous zero status 3197 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3198 } 3199 3200 // if any fast tracks, then status is ready 3201 mMixerStatusIgnoringFastTracks = mixerStatus; 3202 if (fastTracks > 0) { 3203 mixerStatus = MIXER_TRACKS_READY; 3204 } 3205 return mixerStatus; 3206} 3207 3208// getTrackName_l() must be called with ThreadBase::mLock held 3209int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3210{ 3211 return mAudioMixer->getTrackName(channelMask, sessionId); 3212} 3213 3214// deleteTrackName_l() must be called with ThreadBase::mLock held 3215void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3216{ 3217 ALOGV("remove track (%d) and delete from mixer", name); 3218 mAudioMixer->deleteTrackName(name); 3219} 3220 3221// checkForNewParameters_l() must be called with ThreadBase::mLock held 3222bool AudioFlinger::MixerThread::checkForNewParameters_l() 3223{ 3224 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3225 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3226 bool reconfig = false; 3227 3228 while (!mNewParameters.isEmpty()) { 3229 3230 if (mFastMixer != NULL) { 3231 FastMixerStateQueue *sq = mFastMixer->sq(); 3232 FastMixerState *state = sq->begin(); 3233 if (!(state->mCommand & FastMixerState::IDLE)) { 3234 previousCommand = state->mCommand; 3235 state->mCommand = FastMixerState::HOT_IDLE; 3236 sq->end(); 3237 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3238 } else { 3239 sq->end(false /*didModify*/); 3240 } 3241 } 3242 3243 status_t status = NO_ERROR; 3244 String8 keyValuePair = mNewParameters[0]; 3245 AudioParameter param = AudioParameter(keyValuePair); 3246 int value; 3247 3248 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3249 reconfig = true; 3250 } 3251 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3252 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3253 status = BAD_VALUE; 3254 } else { 3255 // no need to save value, since it's constant 3256 reconfig = true; 3257 } 3258 } 3259 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3260 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3261 status = BAD_VALUE; 3262 } else { 3263 // no need to save value, since it's constant 3264 reconfig = true; 3265 } 3266 } 3267 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3268 // do not accept frame count changes if tracks are open as the track buffer 3269 // size depends on frame count and correct behavior would not be guaranteed 3270 // if frame count is changed after track creation 3271 if (!mTracks.isEmpty()) { 3272 status = INVALID_OPERATION; 3273 } else { 3274 reconfig = true; 3275 } 3276 } 3277 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3278#ifdef ADD_BATTERY_DATA 3279 // when changing the audio output device, call addBatteryData to notify 3280 // the change 3281 if (mOutDevice != value) { 3282 uint32_t params = 0; 3283 // check whether speaker is on 3284 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3285 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3286 } 3287 3288 audio_devices_t deviceWithoutSpeaker 3289 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3290 // check if any other device (except speaker) is on 3291 if (value & deviceWithoutSpeaker ) { 3292 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3293 } 3294 3295 if (params != 0) { 3296 addBatteryData(params); 3297 } 3298 } 3299#endif 3300 3301 // forward device change to effects that have requested to be 3302 // aware of attached audio device. 3303 if (value != AUDIO_DEVICE_NONE) { 3304 mOutDevice = value; 3305 for (size_t i = 0; i < mEffectChains.size(); i++) { 3306 mEffectChains[i]->setDevice_l(mOutDevice); 3307 } 3308 } 3309 } 3310 3311 if (status == NO_ERROR) { 3312 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3313 keyValuePair.string()); 3314 if (!mStandby && status == INVALID_OPERATION) { 3315 mOutput->stream->common.standby(&mOutput->stream->common); 3316 mStandby = true; 3317 mBytesWritten = 0; 3318 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3319 keyValuePair.string()); 3320 } 3321 if (status == NO_ERROR && reconfig) { 3322 readOutputParameters(); 3323 delete mAudioMixer; 3324 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3325 for (size_t i = 0; i < mTracks.size() ; i++) { 3326 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3327 if (name < 0) { 3328 break; 3329 } 3330 mTracks[i]->mName = name; 3331 } 3332 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3333 } 3334 } 3335 3336 mNewParameters.removeAt(0); 3337 3338 mParamStatus = status; 3339 mParamCond.signal(); 3340 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3341 // already timed out waiting for the status and will never signal the condition. 3342 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3343 } 3344 3345 if (!(previousCommand & FastMixerState::IDLE)) { 3346 ALOG_ASSERT(mFastMixer != NULL); 3347 FastMixerStateQueue *sq = mFastMixer->sq(); 3348 FastMixerState *state = sq->begin(); 3349 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3350 state->mCommand = previousCommand; 3351 sq->end(); 3352 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3353 } 3354 3355 return reconfig; 3356} 3357 3358 3359void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3360{ 3361 const size_t SIZE = 256; 3362 char buffer[SIZE]; 3363 String8 result; 3364 3365 PlaybackThread::dumpInternals(fd, args); 3366 3367 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3368 result.append(buffer); 3369 write(fd, result.string(), result.size()); 3370 3371 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3372 const FastMixerDumpState copy(mFastMixerDumpState); 3373 copy.dump(fd); 3374 3375#ifdef STATE_QUEUE_DUMP 3376 // Similar for state queue 3377 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3378 observerCopy.dump(fd); 3379 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3380 mutatorCopy.dump(fd); 3381#endif 3382 3383#ifdef TEE_SINK 3384 // Write the tee output to a .wav file 3385 dumpTee(fd, mTeeSource, mId); 3386#endif 3387 3388#ifdef AUDIO_WATCHDOG 3389 if (mAudioWatchdog != 0) { 3390 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3391 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3392 wdCopy.dump(fd); 3393 } 3394#endif 3395} 3396 3397uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3398{ 3399 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3400} 3401 3402uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3403{ 3404 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3405} 3406 3407void AudioFlinger::MixerThread::cacheParameters_l() 3408{ 3409 PlaybackThread::cacheParameters_l(); 3410 3411 // FIXME: Relaxed timing because of a certain device that can't meet latency 3412 // Should be reduced to 2x after the vendor fixes the driver issue 3413 // increase threshold again due to low power audio mode. The way this warning 3414 // threshold is calculated and its usefulness should be reconsidered anyway. 3415 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3416} 3417 3418// ---------------------------------------------------------------------------- 3419 3420AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3421 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3422 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3423 // mLeftVolFloat, mRightVolFloat 3424{ 3425} 3426 3427AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3428 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3429 ThreadBase::type_t type) 3430 : PlaybackThread(audioFlinger, output, id, device, type) 3431 // mLeftVolFloat, mRightVolFloat 3432{ 3433} 3434 3435AudioFlinger::DirectOutputThread::~DirectOutputThread() 3436{ 3437} 3438 3439void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3440{ 3441 audio_track_cblk_t* cblk = track->cblk(); 3442 float left, right; 3443 3444 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3445 left = right = 0; 3446 } else { 3447 float typeVolume = mStreamTypes[track->streamType()].volume; 3448 float v = mMasterVolume * typeVolume; 3449 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3450 uint32_t vlr = proxy->getVolumeLR(); 3451 float v_clamped = v * (vlr & 0xFFFF); 3452 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3453 left = v_clamped/MAX_GAIN; 3454 v_clamped = v * (vlr >> 16); 3455 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3456 right = v_clamped/MAX_GAIN; 3457 } 3458 3459 if (lastTrack) { 3460 if (left != mLeftVolFloat || right != mRightVolFloat) { 3461 mLeftVolFloat = left; 3462 mRightVolFloat = right; 3463 3464 // Convert volumes from float to 8.24 3465 uint32_t vl = (uint32_t)(left * (1 << 24)); 3466 uint32_t vr = (uint32_t)(right * (1 << 24)); 3467 3468 // Delegate volume control to effect in track effect chain if needed 3469 // only one effect chain can be present on DirectOutputThread, so if 3470 // there is one, the track is connected to it 3471 if (!mEffectChains.isEmpty()) { 3472 mEffectChains[0]->setVolume_l(&vl, &vr); 3473 left = (float)vl / (1 << 24); 3474 right = (float)vr / (1 << 24); 3475 } 3476 if (mOutput->stream->set_volume) { 3477 mOutput->stream->set_volume(mOutput->stream, left, right); 3478 } 3479 } 3480 } 3481} 3482 3483 3484AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3485 Vector< sp<Track> > *tracksToRemove 3486) 3487{ 3488 size_t count = mActiveTracks.size(); 3489 mixer_state mixerStatus = MIXER_IDLE; 3490 3491 // find out which tracks need to be processed 3492 for (size_t i = 0; i < count; i++) { 3493 sp<Track> t = mActiveTracks[i].promote(); 3494 // The track died recently 3495 if (t == 0) { 3496 continue; 3497 } 3498 3499 Track* const track = t.get(); 3500 audio_track_cblk_t* cblk = track->cblk(); 3501 3502 // The first time a track is added we wait 3503 // for all its buffers to be filled before processing it 3504 uint32_t minFrames; 3505 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3506 minFrames = mNormalFrameCount; 3507 } else { 3508 minFrames = 1; 3509 } 3510 // Only consider last track started for volume and mixer state control. 3511 // This is the last entry in mActiveTracks unless a track underruns. 3512 // As we only care about the transition phase between two tracks on a 3513 // direct output, it is not a problem to ignore the underrun case. 3514 bool last = (i == (count - 1)); 3515 3516 if ((track->framesReady() >= minFrames) && track->isReady() && 3517 !track->isPaused() && !track->isTerminated()) 3518 { 3519 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3520 3521 if (track->mFillingUpStatus == Track::FS_FILLED) { 3522 track->mFillingUpStatus = Track::FS_ACTIVE; 3523 mLeftVolFloat = mRightVolFloat = 0; 3524 if (track->mState == TrackBase::RESUMING) { 3525 track->mState = TrackBase::ACTIVE; 3526 } 3527 } 3528 3529 // compute volume for this track 3530 processVolume_l(track, last); 3531 if (last) { 3532 // reset retry count 3533 track->mRetryCount = kMaxTrackRetriesDirect; 3534 mActiveTrack = t; 3535 mixerStatus = MIXER_TRACKS_READY; 3536 } 3537 } else { 3538 // clear effect chain input buffer if the last active track started underruns 3539 // to avoid sending previous audio buffer again to effects 3540 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3541 mEffectChains[0]->clearInputBuffer(); 3542 } 3543 3544 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3545 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3546 track->isStopped() || track->isPaused()) { 3547 // We have consumed all the buffers of this track. 3548 // Remove it from the list of active tracks. 3549 // TODO: implement behavior for compressed audio 3550 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3551 size_t framesWritten = mBytesWritten / mFrameSize; 3552 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3553 if (track->isStopped()) { 3554 track->reset(); 3555 } 3556 tracksToRemove->add(track); 3557 } 3558 } else { 3559 // No buffers for this track. Give it a few chances to 3560 // fill a buffer, then remove it from active list. 3561 // Only consider last track started for mixer state control 3562 if (--(track->mRetryCount) <= 0) { 3563 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3564 tracksToRemove->add(track); 3565 } else if (last) { 3566 mixerStatus = MIXER_TRACKS_ENABLED; 3567 } 3568 } 3569 } 3570 } 3571 3572 // remove all the tracks that need to be... 3573 removeTracks_l(*tracksToRemove); 3574 3575 return mixerStatus; 3576} 3577 3578void AudioFlinger::DirectOutputThread::threadLoop_mix() 3579{ 3580 size_t frameCount = mFrameCount; 3581 int8_t *curBuf = (int8_t *)mMixBuffer; 3582 // output audio to hardware 3583 while (frameCount) { 3584 AudioBufferProvider::Buffer buffer; 3585 buffer.frameCount = frameCount; 3586 mActiveTrack->getNextBuffer(&buffer); 3587 if (buffer.raw == NULL) { 3588 memset(curBuf, 0, frameCount * mFrameSize); 3589 break; 3590 } 3591 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3592 frameCount -= buffer.frameCount; 3593 curBuf += buffer.frameCount * mFrameSize; 3594 mActiveTrack->releaseBuffer(&buffer); 3595 } 3596 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3597 sleepTime = 0; 3598 standbyTime = systemTime() + standbyDelay; 3599 mActiveTrack.clear(); 3600} 3601 3602void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3603{ 3604 if (sleepTime == 0) { 3605 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3606 sleepTime = activeSleepTime; 3607 } else { 3608 sleepTime = idleSleepTime; 3609 } 3610 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3611 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3612 sleepTime = 0; 3613 } 3614} 3615 3616// getTrackName_l() must be called with ThreadBase::mLock held 3617int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3618 int sessionId) 3619{ 3620 return 0; 3621} 3622 3623// deleteTrackName_l() must be called with ThreadBase::mLock held 3624void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3625{ 3626} 3627 3628// checkForNewParameters_l() must be called with ThreadBase::mLock held 3629bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3630{ 3631 bool reconfig = false; 3632 3633 while (!mNewParameters.isEmpty()) { 3634 status_t status = NO_ERROR; 3635 String8 keyValuePair = mNewParameters[0]; 3636 AudioParameter param = AudioParameter(keyValuePair); 3637 int value; 3638 3639 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3640 // do not accept frame count changes if tracks are open as the track buffer 3641 // size depends on frame count and correct behavior would not be garantied 3642 // if frame count is changed after track creation 3643 if (!mTracks.isEmpty()) { 3644 status = INVALID_OPERATION; 3645 } else { 3646 reconfig = true; 3647 } 3648 } 3649 if (status == NO_ERROR) { 3650 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3651 keyValuePair.string()); 3652 if (!mStandby && status == INVALID_OPERATION) { 3653 mOutput->stream->common.standby(&mOutput->stream->common); 3654 mStandby = true; 3655 mBytesWritten = 0; 3656 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3657 keyValuePair.string()); 3658 } 3659 if (status == NO_ERROR && reconfig) { 3660 readOutputParameters(); 3661 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3662 } 3663 } 3664 3665 mNewParameters.removeAt(0); 3666 3667 mParamStatus = status; 3668 mParamCond.signal(); 3669 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3670 // already timed out waiting for the status and will never signal the condition. 3671 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3672 } 3673 return reconfig; 3674} 3675 3676uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3677{ 3678 uint32_t time; 3679 if (audio_is_linear_pcm(mFormat)) { 3680 time = PlaybackThread::activeSleepTimeUs(); 3681 } else { 3682 time = 10000; 3683 } 3684 return time; 3685} 3686 3687uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3688{ 3689 uint32_t time; 3690 if (audio_is_linear_pcm(mFormat)) { 3691 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3692 } else { 3693 time = 10000; 3694 } 3695 return time; 3696} 3697 3698uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3699{ 3700 uint32_t time; 3701 if (audio_is_linear_pcm(mFormat)) { 3702 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3703 } else { 3704 time = 10000; 3705 } 3706 return time; 3707} 3708 3709void AudioFlinger::DirectOutputThread::cacheParameters_l() 3710{ 3711 PlaybackThread::cacheParameters_l(); 3712 3713 // use shorter standby delay as on normal output to release 3714 // hardware resources as soon as possible 3715 if (audio_is_linear_pcm(mFormat)) { 3716 standbyDelay = microseconds(activeSleepTime*2); 3717 } else { 3718 standbyDelay = kOffloadStandbyDelayNs; 3719 } 3720} 3721 3722// ---------------------------------------------------------------------------- 3723 3724AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3725 const sp<AudioFlinger::OffloadThread>& offloadThread) 3726 : Thread(false /*canCallJava*/), 3727 mOffloadThread(offloadThread), 3728 mWriteAckSequence(0), 3729 mDrainSequence(0) 3730{ 3731} 3732 3733AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3734{ 3735} 3736 3737void AudioFlinger::AsyncCallbackThread::onFirstRef() 3738{ 3739 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3740} 3741 3742bool AudioFlinger::AsyncCallbackThread::threadLoop() 3743{ 3744 while (!exitPending()) { 3745 uint32_t writeAckSequence; 3746 uint32_t drainSequence; 3747 3748 { 3749 Mutex::Autolock _l(mLock); 3750 mWaitWorkCV.wait(mLock); 3751 if (exitPending()) { 3752 break; 3753 } 3754 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3755 mWriteAckSequence, mDrainSequence); 3756 writeAckSequence = mWriteAckSequence; 3757 mWriteAckSequence &= ~1; 3758 drainSequence = mDrainSequence; 3759 mDrainSequence &= ~1; 3760 } 3761 { 3762 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3763 if (offloadThread != 0) { 3764 if (writeAckSequence & 1) { 3765 offloadThread->resetWriteBlocked(writeAckSequence >> 1); 3766 } 3767 if (drainSequence & 1) { 3768 offloadThread->resetDraining(drainSequence >> 1); 3769 } 3770 } 3771 } 3772 } 3773 return false; 3774} 3775 3776void AudioFlinger::AsyncCallbackThread::exit() 3777{ 3778 ALOGV("AsyncCallbackThread::exit"); 3779 Mutex::Autolock _l(mLock); 3780 requestExit(); 3781 mWaitWorkCV.broadcast(); 3782} 3783 3784void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3785{ 3786 Mutex::Autolock _l(mLock); 3787 // bit 0 is cleared 3788 mWriteAckSequence = sequence << 1; 3789} 3790 3791void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3792{ 3793 Mutex::Autolock _l(mLock); 3794 // ignore unexpected callbacks 3795 if (mWriteAckSequence & 2) { 3796 mWriteAckSequence |= 1; 3797 mWaitWorkCV.signal(); 3798 } 3799} 3800 3801void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3802{ 3803 Mutex::Autolock _l(mLock); 3804 // bit 0 is cleared 3805 mDrainSequence = sequence << 1; 3806} 3807 3808void AudioFlinger::AsyncCallbackThread::resetDraining() 3809{ 3810 Mutex::Autolock _l(mLock); 3811 // ignore unexpected callbacks 3812 if (mDrainSequence & 2) { 3813 mDrainSequence |= 1; 3814 mWaitWorkCV.signal(); 3815 } 3816} 3817 3818 3819// ---------------------------------------------------------------------------- 3820AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3821 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3822 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3823 mHwPaused(false), 3824 mPausedBytesRemaining(0) 3825{ 3826 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3827} 3828 3829AudioFlinger::OffloadThread::~OffloadThread() 3830{ 3831 mPreviousTrack.clear(); 3832} 3833 3834void AudioFlinger::OffloadThread::threadLoop_exit() 3835{ 3836 if (mFlushPending || mHwPaused) { 3837 // If a flush is pending or track was paused, just discard buffered data 3838 flushHw_l(); 3839 } else { 3840 mMixerStatus = MIXER_DRAIN_ALL; 3841 threadLoop_drain(); 3842 } 3843 mCallbackThread->exit(); 3844 PlaybackThread::threadLoop_exit(); 3845} 3846 3847AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3848 Vector< sp<Track> > *tracksToRemove 3849) 3850{ 3851 ALOGV("OffloadThread::prepareTracks_l"); 3852 size_t count = mActiveTracks.size(); 3853 3854 mixer_state mixerStatus = MIXER_IDLE; 3855 bool doHwPause = false; 3856 bool doHwResume = false; 3857 3858 // find out which tracks need to be processed 3859 for (size_t i = 0; i < count; i++) { 3860 sp<Track> t = mActiveTracks[i].promote(); 3861 // The track died recently 3862 if (t == 0) { 3863 continue; 3864 } 3865 Track* const track = t.get(); 3866 audio_track_cblk_t* cblk = track->cblk(); 3867 if (mPreviousTrack != NULL) { 3868 if (t != mPreviousTrack) { 3869 // Flush any data still being written from last track 3870 mBytesRemaining = 0; 3871 if (mPausedBytesRemaining) { 3872 // Last track was paused so we also need to flush saved 3873 // mixbuffer state and invalidate track so that it will 3874 // re-submit that unwritten data when it is next resumed 3875 mPausedBytesRemaining = 0; 3876 // Invalidate is a bit drastic - would be more efficient 3877 // to have a flag to tell client that some of the 3878 // previously written data was lost 3879 mPreviousTrack->invalidate(); 3880 } 3881 } 3882 } 3883 mPreviousTrack = t; 3884 bool last = (i == (count - 1)); 3885 if (track->isPausing()) { 3886 track->setPaused(); 3887 if (last) { 3888 if (!mHwPaused) { 3889 doHwPause = true; 3890 mHwPaused = true; 3891 } 3892 // If we were part way through writing the mixbuffer to 3893 // the HAL we must save this until we resume 3894 // BUG - this will be wrong if a different track is made active, 3895 // in that case we want to discard the pending data in the 3896 // mixbuffer and tell the client to present it again when the 3897 // track is resumed 3898 mPausedWriteLength = mCurrentWriteLength; 3899 mPausedBytesRemaining = mBytesRemaining; 3900 mBytesRemaining = 0; // stop writing 3901 } 3902 tracksToRemove->add(track); 3903 } else if (track->framesReady() && track->isReady() && 3904 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3905 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3906 if (track->mFillingUpStatus == Track::FS_FILLED) { 3907 track->mFillingUpStatus = Track::FS_ACTIVE; 3908 mLeftVolFloat = mRightVolFloat = 0; 3909 if (track->mState == TrackBase::RESUMING) { 3910 if (mPausedBytesRemaining) { 3911 // Need to continue write that was interrupted 3912 mCurrentWriteLength = mPausedWriteLength; 3913 mBytesRemaining = mPausedBytesRemaining; 3914 mPausedBytesRemaining = 0; 3915 } 3916 track->mState = TrackBase::ACTIVE; 3917 } 3918 } 3919 3920 if (last) { 3921 if (mHwPaused) { 3922 doHwResume = true; 3923 mHwPaused = false; 3924 // threadLoop_mix() will handle the case that we need to 3925 // resume an interrupted write 3926 } 3927 // reset retry count 3928 track->mRetryCount = kMaxTrackRetriesOffload; 3929 mActiveTrack = t; 3930 mixerStatus = MIXER_TRACKS_READY; 3931 } 3932 } else { 3933 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3934 if (track->isStopping_1()) { 3935 // Hardware buffer can hold a large amount of audio so we must 3936 // wait for all current track's data to drain before we say 3937 // that the track is stopped. 3938 if (mBytesRemaining == 0) { 3939 // Only start draining when all data in mixbuffer 3940 // has been written 3941 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3942 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3943 sleepTime = 0; 3944 standbyTime = systemTime() + standbyDelay; 3945 if (last) { 3946 mixerStatus = MIXER_DRAIN_TRACK; 3947 mDrainSequence += 2; 3948 if (mHwPaused) { 3949 // It is possible to move from PAUSED to STOPPING_1 without 3950 // a resume so we must ensure hardware is running 3951 mOutput->stream->resume(mOutput->stream); 3952 mHwPaused = false; 3953 } 3954 } 3955 } 3956 } else if (track->isStopping_2()) { 3957 // Drain has completed, signal presentation complete 3958 if (!(mDrainSequence & 1) || !last) { 3959 track->mState = TrackBase::STOPPED; 3960 size_t audioHALFrames = 3961 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3962 size_t framesWritten = 3963 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3964 track->presentationComplete(framesWritten, audioHALFrames); 3965 track->reset(); 3966 tracksToRemove->add(track); 3967 } 3968 } else { 3969 // No buffers for this track. Give it a few chances to 3970 // fill a buffer, then remove it from active list. 3971 if (--(track->mRetryCount) <= 0) { 3972 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3973 track->name()); 3974 tracksToRemove->add(track); 3975 } else if (last){ 3976 mixerStatus = MIXER_TRACKS_ENABLED; 3977 } 3978 } 3979 } 3980 // compute volume for this track 3981 processVolume_l(track, last); 3982 } 3983 3984 // make sure the pause/flush/resume sequence is executed in the right order 3985 if (doHwPause) { 3986 mOutput->stream->pause(mOutput->stream); 3987 } 3988 if (mFlushPending) { 3989 flushHw_l(); 3990 mFlushPending = false; 3991 } 3992 if (doHwResume) { 3993 mOutput->stream->resume(mOutput->stream); 3994 } 3995 3996 // remove all the tracks that need to be... 3997 removeTracks_l(*tracksToRemove); 3998 3999 return mixerStatus; 4000} 4001 4002void AudioFlinger::OffloadThread::flushOutput_l() 4003{ 4004 mFlushPending = true; 4005} 4006 4007// must be called with thread mutex locked 4008bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4009{ 4010 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4011 mWriteAckSequence, mDrainSequence); 4012 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4013 return true; 4014 } 4015 return false; 4016} 4017 4018// must be called with thread mutex locked 4019bool AudioFlinger::OffloadThread::shouldStandby_l() 4020{ 4021 bool TrackPaused = false; 4022 4023 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4024 // after a timeout and we will enter standby then. 4025 if (mTracks.size() > 0) { 4026 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4027 } 4028 4029 return !mStandby && !TrackPaused; 4030} 4031 4032 4033bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4034{ 4035 Mutex::Autolock _l(mLock); 4036 return waitingAsyncCallback_l(); 4037} 4038 4039void AudioFlinger::OffloadThread::flushHw_l() 4040{ 4041 mOutput->stream->flush(mOutput->stream); 4042 // Flush anything still waiting in the mixbuffer 4043 mCurrentWriteLength = 0; 4044 mBytesRemaining = 0; 4045 mPausedWriteLength = 0; 4046 mPausedBytesRemaining = 0; 4047 if (mUseAsyncWrite) { 4048 // discard any pending drain or write ack by incrementing sequence 4049 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4050 mDrainSequence = (mDrainSequence + 2) & ~1; 4051 ALOG_ASSERT(mCallbackThread != 0); 4052 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4053 mCallbackThread->setDraining(mDrainSequence); 4054 } 4055} 4056 4057// ---------------------------------------------------------------------------- 4058 4059AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4060 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4061 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4062 DUPLICATING), 4063 mWaitTimeMs(UINT_MAX) 4064{ 4065 addOutputTrack(mainThread); 4066} 4067 4068AudioFlinger::DuplicatingThread::~DuplicatingThread() 4069{ 4070 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4071 mOutputTracks[i]->destroy(); 4072 } 4073} 4074 4075void AudioFlinger::DuplicatingThread::threadLoop_mix() 4076{ 4077 // mix buffers... 4078 if (outputsReady(outputTracks)) { 4079 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4080 } else { 4081 memset(mMixBuffer, 0, mixBufferSize); 4082 } 4083 sleepTime = 0; 4084 writeFrames = mNormalFrameCount; 4085 mCurrentWriteLength = mixBufferSize; 4086 standbyTime = systemTime() + standbyDelay; 4087} 4088 4089void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4090{ 4091 if (sleepTime == 0) { 4092 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4093 sleepTime = activeSleepTime; 4094 } else { 4095 sleepTime = idleSleepTime; 4096 } 4097 } else if (mBytesWritten != 0) { 4098 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4099 writeFrames = mNormalFrameCount; 4100 memset(mMixBuffer, 0, mixBufferSize); 4101 } else { 4102 // flush remaining overflow buffers in output tracks 4103 writeFrames = 0; 4104 } 4105 sleepTime = 0; 4106 } 4107} 4108 4109ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4110{ 4111 for (size_t i = 0; i < outputTracks.size(); i++) { 4112 outputTracks[i]->write(mMixBuffer, writeFrames); 4113 } 4114 return (ssize_t)mixBufferSize; 4115} 4116 4117void AudioFlinger::DuplicatingThread::threadLoop_standby() 4118{ 4119 // DuplicatingThread implements standby by stopping all tracks 4120 for (size_t i = 0; i < outputTracks.size(); i++) { 4121 outputTracks[i]->stop(); 4122 } 4123} 4124 4125void AudioFlinger::DuplicatingThread::saveOutputTracks() 4126{ 4127 outputTracks = mOutputTracks; 4128} 4129 4130void AudioFlinger::DuplicatingThread::clearOutputTracks() 4131{ 4132 outputTracks.clear(); 4133} 4134 4135void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4136{ 4137 Mutex::Autolock _l(mLock); 4138 // FIXME explain this formula 4139 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4140 OutputTrack *outputTrack = new OutputTrack(thread, 4141 this, 4142 mSampleRate, 4143 mFormat, 4144 mChannelMask, 4145 frameCount); 4146 if (outputTrack->cblk() != NULL) { 4147 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4148 mOutputTracks.add(outputTrack); 4149 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4150 updateWaitTime_l(); 4151 } 4152} 4153 4154void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4155{ 4156 Mutex::Autolock _l(mLock); 4157 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4158 if (mOutputTracks[i]->thread() == thread) { 4159 mOutputTracks[i]->destroy(); 4160 mOutputTracks.removeAt(i); 4161 updateWaitTime_l(); 4162 return; 4163 } 4164 } 4165 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4166} 4167 4168// caller must hold mLock 4169void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4170{ 4171 mWaitTimeMs = UINT_MAX; 4172 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4173 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4174 if (strong != 0) { 4175 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4176 if (waitTimeMs < mWaitTimeMs) { 4177 mWaitTimeMs = waitTimeMs; 4178 } 4179 } 4180 } 4181} 4182 4183 4184bool AudioFlinger::DuplicatingThread::outputsReady( 4185 const SortedVector< sp<OutputTrack> > &outputTracks) 4186{ 4187 for (size_t i = 0; i < outputTracks.size(); i++) { 4188 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4189 if (thread == 0) { 4190 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4191 outputTracks[i].get()); 4192 return false; 4193 } 4194 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4195 // see note at standby() declaration 4196 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4197 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4198 thread.get()); 4199 return false; 4200 } 4201 } 4202 return true; 4203} 4204 4205uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4206{ 4207 return (mWaitTimeMs * 1000) / 2; 4208} 4209 4210void AudioFlinger::DuplicatingThread::cacheParameters_l() 4211{ 4212 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4213 updateWaitTime_l(); 4214 4215 MixerThread::cacheParameters_l(); 4216} 4217 4218// ---------------------------------------------------------------------------- 4219// Record 4220// ---------------------------------------------------------------------------- 4221 4222AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4223 AudioStreamIn *input, 4224 uint32_t sampleRate, 4225 audio_channel_mask_t channelMask, 4226 audio_io_handle_t id, 4227 audio_devices_t outDevice, 4228 audio_devices_t inDevice 4229#ifdef TEE_SINK 4230 , const sp<NBAIO_Sink>& teeSink 4231#endif 4232 ) : 4233 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4234 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4235 // mRsmpInIndex set by readInputParameters() 4236 mReqChannelCount(popcount(channelMask)), 4237 mReqSampleRate(sampleRate) 4238 // mBytesRead is only meaningful while active, and so is cleared in start() 4239 // (but might be better to also clear here for dump?) 4240#ifdef TEE_SINK 4241 , mTeeSink(teeSink) 4242#endif 4243{ 4244 snprintf(mName, kNameLength, "AudioIn_%X", id); 4245 4246 readInputParameters(); 4247 4248} 4249 4250 4251AudioFlinger::RecordThread::~RecordThread() 4252{ 4253 delete[] mRsmpInBuffer; 4254 delete mResampler; 4255 delete[] mRsmpOutBuffer; 4256} 4257 4258void AudioFlinger::RecordThread::onFirstRef() 4259{ 4260 run(mName, PRIORITY_URGENT_AUDIO); 4261} 4262 4263bool AudioFlinger::RecordThread::threadLoop() 4264{ 4265 AudioBufferProvider::Buffer buffer; 4266 4267 nsecs_t lastWarning = 0; 4268 4269 inputStandBy(); 4270 acquireWakeLock(); 4271 4272 // used to verify we've read at least once before evaluating how many bytes were read 4273 bool readOnce = false; 4274 4275 // used to request a deferred sleep, to be executed later while mutex is unlocked 4276 bool doSleep = false; 4277 4278 // start recording 4279 for (;;) { 4280 sp<RecordTrack> activeTrack; 4281 TrackBase::track_state activeTrackState; 4282 Vector< sp<EffectChain> > effectChains; 4283 4284 // sleep with mutex unlocked 4285 if (doSleep) { 4286 doSleep = false; 4287 usleep(kRecordThreadSleepUs); 4288 } 4289 4290 { // scope for mLock 4291 Mutex::Autolock _l(mLock); 4292 if (exitPending()) { 4293 break; 4294 } 4295 processConfigEvents_l(); 4296 // return value 'reconfig' is currently unused 4297 bool reconfig = checkForNewParameters_l(); 4298 // make a stable copy of mActiveTrack 4299 activeTrack = mActiveTrack; 4300 if (activeTrack == 0) { 4301 standby(); 4302 // exitPending() can't become true here 4303 releaseWakeLock_l(); 4304 ALOGV("RecordThread: loop stopping"); 4305 // go to sleep 4306 mWaitWorkCV.wait(mLock); 4307 ALOGV("RecordThread: loop starting"); 4308 acquireWakeLock_l(); 4309 continue; 4310 } 4311 4312 if (activeTrack->isTerminated()) { 4313 removeTrack_l(activeTrack); 4314 mActiveTrack.clear(); 4315 continue; 4316 } 4317 4318 activeTrackState = activeTrack->mState; 4319 switch (activeTrackState) { 4320 case TrackBase::PAUSING: 4321 standby(); 4322 mActiveTrack.clear(); 4323 mStartStopCond.broadcast(); 4324 doSleep = true; 4325 continue; 4326 4327 case TrackBase::RESUMING: 4328 mStandby = false; 4329 if (mReqChannelCount != activeTrack->channelCount()) { 4330 mActiveTrack.clear(); 4331 mStartStopCond.broadcast(); 4332 continue; 4333 } 4334 if (readOnce) { 4335 mStartStopCond.broadcast(); 4336 // record start succeeds only if first read from audio input succeeds 4337 if (mBytesRead < 0) { 4338 mActiveTrack.clear(); 4339 continue; 4340 } 4341 activeTrack->mState = TrackBase::ACTIVE; 4342 } 4343 break; 4344 4345 case TrackBase::ACTIVE: 4346 break; 4347 4348 case TrackBase::IDLE: 4349 doSleep = true; 4350 continue; 4351 4352 default: 4353 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4354 } 4355 4356 lockEffectChains_l(effectChains); 4357 } 4358 4359 // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable 4360 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4361 4362 for (size_t i = 0; i < effectChains.size(); i ++) { 4363 // thread mutex is not locked, but effect chain is locked 4364 effectChains[i]->process_l(); 4365 } 4366 4367 buffer.frameCount = mFrameCount; 4368 status_t status = activeTrack->getNextBuffer(&buffer); 4369 if (status == NO_ERROR) { 4370 readOnce = true; 4371 size_t framesOut = buffer.frameCount; 4372 if (mResampler == NULL) { 4373 // no resampling 4374 while (framesOut) { 4375 size_t framesIn = mFrameCount - mRsmpInIndex; 4376 if (framesIn > 0) { 4377 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4378 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4379 activeTrack->mFrameSize; 4380 if (framesIn > framesOut) { 4381 framesIn = framesOut; 4382 } 4383 mRsmpInIndex += framesIn; 4384 framesOut -= framesIn; 4385 if (mChannelCount == mReqChannelCount) { 4386 memcpy(dst, src, framesIn * mFrameSize); 4387 } else { 4388 if (mChannelCount == 1) { 4389 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4390 (int16_t *)src, framesIn); 4391 } else { 4392 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4393 (int16_t *)src, framesIn); 4394 } 4395 } 4396 } 4397 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4398 void *readInto; 4399 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4400 readInto = buffer.raw; 4401 framesOut = 0; 4402 } else { 4403 readInto = mRsmpInBuffer; 4404 mRsmpInIndex = 0; 4405 } 4406 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4407 mBufferSize); 4408 if (mBytesRead <= 0) { 4409 // TODO: verify that it's benign to use a stale track state 4410 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4411 { 4412 ALOGE("Error reading audio input"); 4413 // Force input into standby so that it tries to 4414 // recover at next read attempt 4415 inputStandBy(); 4416 doSleep = true; 4417 } 4418 mRsmpInIndex = mFrameCount; 4419 framesOut = 0; 4420 buffer.frameCount = 0; 4421 } 4422#ifdef TEE_SINK 4423 else if (mTeeSink != 0) { 4424 (void) mTeeSink->write(readInto, 4425 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4426 } 4427#endif 4428 } 4429 } 4430 } else { 4431 // resampling 4432 4433 // resampler accumulates, but we only have one source track 4434 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4435 // alter output frame count as if we were expecting stereo samples 4436 if (mChannelCount == 1 && mReqChannelCount == 1) { 4437 framesOut >>= 1; 4438 } 4439 mResampler->resample(mRsmpOutBuffer, framesOut, 4440 this /* AudioBufferProvider* */); 4441 // ditherAndClamp() works as long as all buffers returned by 4442 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4443 if (mChannelCount == 2 && mReqChannelCount == 1) { 4444 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4445 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4446 // the resampler always outputs stereo samples: 4447 // do post stereo to mono conversion 4448 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4449 framesOut); 4450 } else { 4451 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4452 } 4453 // now done with mRsmpOutBuffer 4454 4455 } 4456 if (mFramestoDrop == 0) { 4457 activeTrack->releaseBuffer(&buffer); 4458 } else { 4459 if (mFramestoDrop > 0) { 4460 mFramestoDrop -= buffer.frameCount; 4461 if (mFramestoDrop <= 0) { 4462 clearSyncStartEvent(); 4463 } 4464 } else { 4465 mFramestoDrop += buffer.frameCount; 4466 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4467 mSyncStartEvent->isCancelled()) { 4468 ALOGW("Synced record %s, session %d, trigger session %d", 4469 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4470 activeTrack->sessionId(), 4471 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4472 clearSyncStartEvent(); 4473 } 4474 } 4475 } 4476 activeTrack->clearOverflow(); 4477 } 4478 // client isn't retrieving buffers fast enough 4479 else { 4480 if (!activeTrack->setOverflow()) { 4481 nsecs_t now = systemTime(); 4482 if ((now - lastWarning) > kWarningThrottleNs) { 4483 ALOGW("RecordThread: buffer overflow"); 4484 lastWarning = now; 4485 } 4486 } 4487 // Release the processor for a while before asking for a new buffer. 4488 // This will give the application more chance to read from the buffer and 4489 // clear the overflow. 4490 doSleep = true; 4491 } 4492 4493 // enable changes in effect chain 4494 unlockEffectChains(effectChains); 4495 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4496 } 4497 4498 standby(); 4499 4500 { 4501 Mutex::Autolock _l(mLock); 4502 for (size_t i = 0; i < mTracks.size(); i++) { 4503 sp<RecordTrack> track = mTracks[i]; 4504 track->invalidate(); 4505 } 4506 mActiveTrack.clear(); 4507 mStartStopCond.broadcast(); 4508 } 4509 4510 releaseWakeLock(); 4511 4512 ALOGV("RecordThread %p exiting", this); 4513 return false; 4514} 4515 4516void AudioFlinger::RecordThread::standby() 4517{ 4518 if (!mStandby) { 4519 inputStandBy(); 4520 mStandby = true; 4521 } 4522} 4523 4524void AudioFlinger::RecordThread::inputStandBy() 4525{ 4526 mInput->stream->common.standby(&mInput->stream->common); 4527} 4528 4529sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4530 const sp<AudioFlinger::Client>& client, 4531 uint32_t sampleRate, 4532 audio_format_t format, 4533 audio_channel_mask_t channelMask, 4534 size_t frameCount, 4535 int sessionId, 4536 IAudioFlinger::track_flags_t *flags, 4537 pid_t tid, 4538 status_t *status) 4539{ 4540 sp<RecordTrack> track; 4541 status_t lStatus; 4542 4543 lStatus = initCheck(); 4544 if (lStatus != NO_ERROR) { 4545 ALOGE("Audio driver not initialized."); 4546 goto Exit; 4547 } 4548 4549 // client expresses a preference for FAST, but we get the final say 4550 if (*flags & IAudioFlinger::TRACK_FAST) { 4551 if ( 4552 // use case: callback handler and frame count is default or at least as large as HAL 4553 ( 4554 (tid != -1) && 4555 ((frameCount == 0) || 4556 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4557 ) && 4558 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4559 // mono or stereo 4560 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4561 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4562 // hardware sample rate 4563 (sampleRate == mSampleRate) && 4564 // record thread has an associated fast recorder 4565 hasFastRecorder() 4566 // FIXME test that RecordThread for this fast track has a capable output HAL 4567 // FIXME add a permission test also? 4568 ) { 4569 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4570 if (frameCount == 0) { 4571 frameCount = mFrameCount * kFastTrackMultiplier; 4572 } 4573 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4574 frameCount, mFrameCount); 4575 } else { 4576 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4577 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4578 "hasFastRecorder=%d tid=%d", 4579 frameCount, mFrameCount, format, 4580 audio_is_linear_pcm(format), 4581 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4582 *flags &= ~IAudioFlinger::TRACK_FAST; 4583 // For compatibility with AudioRecord calculation, buffer depth is forced 4584 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4585 // This is probably too conservative, but legacy application code may depend on it. 4586 // If you change this calculation, also review the start threshold which is related. 4587 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4588 size_t mNormalFrameCount = 2048; // FIXME 4589 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4590 if (minBufCount < 2) { 4591 minBufCount = 2; 4592 } 4593 size_t minFrameCount = mNormalFrameCount * minBufCount; 4594 if (frameCount < minFrameCount) { 4595 frameCount = minFrameCount; 4596 } 4597 } 4598 } 4599 4600 // FIXME use flags and tid similar to createTrack_l() 4601 4602 { // scope for mLock 4603 Mutex::Autolock _l(mLock); 4604 4605 track = new RecordTrack(this, client, sampleRate, 4606 format, channelMask, frameCount, sessionId); 4607 4608 lStatus = track->initCheck(); 4609 if (lStatus != NO_ERROR) { 4610 track.clear(); 4611 goto Exit; 4612 } 4613 mTracks.add(track); 4614 4615 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4616 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4617 mAudioFlinger->btNrecIsOff(); 4618 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4619 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4620 4621 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4622 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4623 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4624 // so ask activity manager to do this on our behalf 4625 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4626 } 4627 } 4628 lStatus = NO_ERROR; 4629 4630Exit: 4631 *status = lStatus; 4632 return track; 4633} 4634 4635status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4636 AudioSystem::sync_event_t event, 4637 int triggerSession) 4638{ 4639 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4640 sp<ThreadBase> strongMe = this; 4641 status_t status = NO_ERROR; 4642 4643 if (event == AudioSystem::SYNC_EVENT_NONE) { 4644 clearSyncStartEvent(); 4645 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4646 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4647 triggerSession, 4648 recordTrack->sessionId(), 4649 syncStartEventCallback, 4650 this); 4651 // Sync event can be cancelled by the trigger session if the track is not in a 4652 // compatible state in which case we start record immediately 4653 if (mSyncStartEvent->isCancelled()) { 4654 clearSyncStartEvent(); 4655 } else { 4656 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4657 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4658 } 4659 } 4660 4661 { 4662 // This section is a rendezvous between binder thread executing start() and RecordThread 4663 AutoMutex lock(mLock); 4664 if (mActiveTrack != 0) { 4665 if (recordTrack != mActiveTrack.get()) { 4666 status = -EBUSY; 4667 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4668 mActiveTrack->mState = TrackBase::ACTIVE; 4669 } 4670 return status; 4671 } 4672 4673 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4674 recordTrack->mState = TrackBase::IDLE; 4675 mActiveTrack = recordTrack; 4676 mLock.unlock(); 4677 status_t status = AudioSystem::startInput(mId); 4678 mLock.lock(); 4679 // FIXME should verify that mActiveTrack is still == recordTrack 4680 if (status != NO_ERROR) { 4681 mActiveTrack.clear(); 4682 clearSyncStartEvent(); 4683 return status; 4684 } 4685 mRsmpInIndex = mFrameCount; 4686 mBytesRead = 0; 4687 if (mResampler != NULL) { 4688 mResampler->reset(); 4689 } 4690 // FIXME hijacking a playback track state name which was intended for start after pause; 4691 // here 'STARTING_2' would be more accurate 4692 mActiveTrack->mState = TrackBase::RESUMING; 4693 // signal thread to start 4694 ALOGV("Signal record thread"); 4695 mWaitWorkCV.broadcast(); 4696 // do not wait for mStartStopCond if exiting 4697 if (exitPending()) { 4698 mActiveTrack.clear(); 4699 status = INVALID_OPERATION; 4700 goto startError; 4701 } 4702 // FIXME incorrect usage of wait: no explicit predicate or loop 4703 mStartStopCond.wait(mLock); 4704 if (mActiveTrack == 0) { 4705 ALOGV("Record failed to start"); 4706 status = BAD_VALUE; 4707 goto startError; 4708 } 4709 ALOGV("Record started OK"); 4710 return status; 4711 } 4712 4713startError: 4714 AudioSystem::stopInput(mId); 4715 clearSyncStartEvent(); 4716 return status; 4717} 4718 4719void AudioFlinger::RecordThread::clearSyncStartEvent() 4720{ 4721 if (mSyncStartEvent != 0) { 4722 mSyncStartEvent->cancel(); 4723 } 4724 mSyncStartEvent.clear(); 4725 mFramestoDrop = 0; 4726} 4727 4728void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4729{ 4730 sp<SyncEvent> strongEvent = event.promote(); 4731 4732 if (strongEvent != 0) { 4733 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4734 me->handleSyncStartEvent(strongEvent); 4735 } 4736} 4737 4738void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4739{ 4740 if (event == mSyncStartEvent) { 4741 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4742 // from audio HAL 4743 mFramestoDrop = mFrameCount * 2; 4744 } 4745} 4746 4747bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4748 ALOGV("RecordThread::stop"); 4749 AutoMutex _l(mLock); 4750 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4751 return false; 4752 } 4753 // note that threadLoop may still be processing the track at this point [without lock] 4754 recordTrack->mState = TrackBase::PAUSING; 4755 // do not wait for mStartStopCond if exiting 4756 if (exitPending()) { 4757 return true; 4758 } 4759 // FIXME incorrect usage of wait: no explicit predicate or loop 4760 mStartStopCond.wait(mLock); 4761 // if we have been restarted, recordTrack == mActiveTrack.get() here 4762 if (exitPending() || recordTrack != mActiveTrack.get()) { 4763 ALOGV("Record stopped OK"); 4764 return true; 4765 } 4766 return false; 4767} 4768 4769bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4770{ 4771 return false; 4772} 4773 4774status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4775{ 4776#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4777 if (!isValidSyncEvent(event)) { 4778 return BAD_VALUE; 4779 } 4780 4781 int eventSession = event->triggerSession(); 4782 status_t ret = NAME_NOT_FOUND; 4783 4784 Mutex::Autolock _l(mLock); 4785 4786 for (size_t i = 0; i < mTracks.size(); i++) { 4787 sp<RecordTrack> track = mTracks[i]; 4788 if (eventSession == track->sessionId()) { 4789 (void) track->setSyncEvent(event); 4790 ret = NO_ERROR; 4791 } 4792 } 4793 return ret; 4794#else 4795 return BAD_VALUE; 4796#endif 4797} 4798 4799// destroyTrack_l() must be called with ThreadBase::mLock held 4800void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4801{ 4802 track->terminate(); 4803 track->mState = TrackBase::STOPPED; 4804 // active tracks are removed by threadLoop() 4805 if (mActiveTrack != track) { 4806 removeTrack_l(track); 4807 } 4808} 4809 4810void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4811{ 4812 mTracks.remove(track); 4813 // need anything related to effects here? 4814} 4815 4816void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4817{ 4818 dumpInternals(fd, args); 4819 dumpTracks(fd, args); 4820 dumpEffectChains(fd, args); 4821} 4822 4823void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4824{ 4825 const size_t SIZE = 256; 4826 char buffer[SIZE]; 4827 String8 result; 4828 4829 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4830 result.append(buffer); 4831 4832 if (mActiveTrack != 0) { 4833 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4834 result.append(buffer); 4835 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4836 result.append(buffer); 4837 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4838 result.append(buffer); 4839 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4840 result.append(buffer); 4841 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4842 result.append(buffer); 4843 } else { 4844 result.append("No active record client\n"); 4845 } 4846 4847 write(fd, result.string(), result.size()); 4848 4849 dumpBase(fd, args); 4850} 4851 4852void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4853{ 4854 const size_t SIZE = 256; 4855 char buffer[SIZE]; 4856 String8 result; 4857 4858 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4859 result.append(buffer); 4860 RecordTrack::appendDumpHeader(result); 4861 for (size_t i = 0; i < mTracks.size(); ++i) { 4862 sp<RecordTrack> track = mTracks[i]; 4863 if (track != 0) { 4864 track->dump(buffer, SIZE); 4865 result.append(buffer); 4866 } 4867 } 4868 4869 if (mActiveTrack != 0) { 4870 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4871 result.append(buffer); 4872 RecordTrack::appendDumpHeader(result); 4873 mActiveTrack->dump(buffer, SIZE); 4874 result.append(buffer); 4875 4876 } 4877 write(fd, result.string(), result.size()); 4878} 4879 4880// AudioBufferProvider interface 4881status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4882{ 4883 size_t framesReq = buffer->frameCount; 4884 size_t framesReady = mFrameCount - mRsmpInIndex; 4885 int channelCount; 4886 4887 if (framesReady == 0) { 4888 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4889 if (mBytesRead <= 0) { 4890 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4891 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4892 // Force input into standby so that it tries to 4893 // recover at next read attempt 4894 inputStandBy(); 4895 // FIXME an awkward place to sleep, consider using doSleep when this is pulled up 4896 usleep(kRecordThreadSleepUs); 4897 } 4898 buffer->raw = NULL; 4899 buffer->frameCount = 0; 4900 return NOT_ENOUGH_DATA; 4901 } 4902 mRsmpInIndex = 0; 4903 framesReady = mFrameCount; 4904 } 4905 4906 if (framesReq > framesReady) { 4907 framesReq = framesReady; 4908 } 4909 4910 if (mChannelCount == 1 && mReqChannelCount == 2) { 4911 channelCount = 1; 4912 } else { 4913 channelCount = 2; 4914 } 4915 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4916 buffer->frameCount = framesReq; 4917 return NO_ERROR; 4918} 4919 4920// AudioBufferProvider interface 4921void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4922{ 4923 mRsmpInIndex += buffer->frameCount; 4924 buffer->frameCount = 0; 4925} 4926 4927bool AudioFlinger::RecordThread::checkForNewParameters_l() 4928{ 4929 bool reconfig = false; 4930 4931 while (!mNewParameters.isEmpty()) { 4932 status_t status = NO_ERROR; 4933 String8 keyValuePair = mNewParameters[0]; 4934 AudioParameter param = AudioParameter(keyValuePair); 4935 int value; 4936 audio_format_t reqFormat = mFormat; 4937 uint32_t reqSamplingRate = mReqSampleRate; 4938 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 4939 4940 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4941 reqSamplingRate = value; 4942 reconfig = true; 4943 } 4944 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4945 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4946 status = BAD_VALUE; 4947 } else { 4948 reqFormat = (audio_format_t) value; 4949 reconfig = true; 4950 } 4951 } 4952 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4953 audio_channel_mask_t mask = (audio_channel_mask_t) value; 4954 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 4955 status = BAD_VALUE; 4956 } else { 4957 reqChannelMask = mask; 4958 reconfig = true; 4959 } 4960 } 4961 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4962 // do not accept frame count changes if tracks are open as the track buffer 4963 // size depends on frame count and correct behavior would not be guaranteed 4964 // if frame count is changed after track creation 4965 if (mActiveTrack != 0) { 4966 status = INVALID_OPERATION; 4967 } else { 4968 reconfig = true; 4969 } 4970 } 4971 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4972 // forward device change to effects that have requested to be 4973 // aware of attached audio device. 4974 for (size_t i = 0; i < mEffectChains.size(); i++) { 4975 mEffectChains[i]->setDevice_l(value); 4976 } 4977 4978 // store input device and output device but do not forward output device to audio HAL. 4979 // Note that status is ignored by the caller for output device 4980 // (see AudioFlinger::setParameters() 4981 if (audio_is_output_devices(value)) { 4982 mOutDevice = value; 4983 status = BAD_VALUE; 4984 } else { 4985 mInDevice = value; 4986 // disable AEC and NS if the device is a BT SCO headset supporting those 4987 // pre processings 4988 if (mTracks.size() > 0) { 4989 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4990 mAudioFlinger->btNrecIsOff(); 4991 for (size_t i = 0; i < mTracks.size(); i++) { 4992 sp<RecordTrack> track = mTracks[i]; 4993 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4994 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4995 } 4996 } 4997 } 4998 } 4999 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5000 mAudioSource != (audio_source_t)value) { 5001 // forward device change to effects that have requested to be 5002 // aware of attached audio device. 5003 for (size_t i = 0; i < mEffectChains.size(); i++) { 5004 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5005 } 5006 mAudioSource = (audio_source_t)value; 5007 } 5008 5009 if (status == NO_ERROR) { 5010 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5011 keyValuePair.string()); 5012 if (status == INVALID_OPERATION) { 5013 inputStandBy(); 5014 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5015 keyValuePair.string()); 5016 } 5017 if (reconfig) { 5018 if (status == BAD_VALUE && 5019 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5020 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5021 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5022 <= (2 * reqSamplingRate)) && 5023 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5024 <= FCC_2 && 5025 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 5026 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 5027 status = NO_ERROR; 5028 } 5029 if (status == NO_ERROR) { 5030 readInputParameters(); 5031 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5032 } 5033 } 5034 } 5035 5036 mNewParameters.removeAt(0); 5037 5038 mParamStatus = status; 5039 mParamCond.signal(); 5040 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5041 // already timed out waiting for the status and will never signal the condition. 5042 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5043 } 5044 return reconfig; 5045} 5046 5047String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5048{ 5049 Mutex::Autolock _l(mLock); 5050 if (initCheck() != NO_ERROR) { 5051 return String8(); 5052 } 5053 5054 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5055 const String8 out_s8(s); 5056 free(s); 5057 return out_s8; 5058} 5059 5060void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5061 AudioSystem::OutputDescriptor desc; 5062 void *param2 = NULL; 5063 5064 switch (event) { 5065 case AudioSystem::INPUT_OPENED: 5066 case AudioSystem::INPUT_CONFIG_CHANGED: 5067 desc.channelMask = mChannelMask; 5068 desc.samplingRate = mSampleRate; 5069 desc.format = mFormat; 5070 desc.frameCount = mFrameCount; 5071 desc.latency = 0; 5072 param2 = &desc; 5073 break; 5074 5075 case AudioSystem::INPUT_CLOSED: 5076 default: 5077 break; 5078 } 5079 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5080} 5081 5082void AudioFlinger::RecordThread::readInputParameters() 5083{ 5084 delete[] mRsmpInBuffer; 5085 // mRsmpInBuffer is always assigned a new[] below 5086 delete[] mRsmpOutBuffer; 5087 mRsmpOutBuffer = NULL; 5088 delete mResampler; 5089 mResampler = NULL; 5090 5091 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5092 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5093 mChannelCount = popcount(mChannelMask); 5094 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5095 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5096 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5097 } 5098 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5099 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5100 mFrameCount = mBufferSize / mFrameSize; 5101 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5102 5103 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5104 int channelCount; 5105 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5106 // stereo to mono post process as the resampler always outputs stereo. 5107 if (mChannelCount == 1 && mReqChannelCount == 2) { 5108 channelCount = 1; 5109 } else { 5110 channelCount = 2; 5111 } 5112 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5113 mResampler->setSampleRate(mSampleRate); 5114 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5115 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5116 5117 // optmization: if mono to mono, alter input frame count as if we were inputing 5118 // stereo samples 5119 if (mChannelCount == 1 && mReqChannelCount == 1) { 5120 mFrameCount >>= 1; 5121 } 5122 5123 } 5124 mRsmpInIndex = mFrameCount; 5125} 5126 5127unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5128{ 5129 Mutex::Autolock _l(mLock); 5130 if (initCheck() != NO_ERROR) { 5131 return 0; 5132 } 5133 5134 return mInput->stream->get_input_frames_lost(mInput->stream); 5135} 5136 5137uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5138{ 5139 Mutex::Autolock _l(mLock); 5140 uint32_t result = 0; 5141 if (getEffectChain_l(sessionId) != 0) { 5142 result = EFFECT_SESSION; 5143 } 5144 5145 for (size_t i = 0; i < mTracks.size(); ++i) { 5146 if (sessionId == mTracks[i]->sessionId()) { 5147 result |= TRACK_SESSION; 5148 break; 5149 } 5150 } 5151 5152 return result; 5153} 5154 5155KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5156{ 5157 KeyedVector<int, bool> ids; 5158 Mutex::Autolock _l(mLock); 5159 for (size_t j = 0; j < mTracks.size(); ++j) { 5160 sp<RecordThread::RecordTrack> track = mTracks[j]; 5161 int sessionId = track->sessionId(); 5162 if (ids.indexOfKey(sessionId) < 0) { 5163 ids.add(sessionId, true); 5164 } 5165 } 5166 return ids; 5167} 5168 5169AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5170{ 5171 Mutex::Autolock _l(mLock); 5172 AudioStreamIn *input = mInput; 5173 mInput = NULL; 5174 return input; 5175} 5176 5177// this method must always be called either with ThreadBase mLock held or inside the thread loop 5178audio_stream_t* AudioFlinger::RecordThread::stream() const 5179{ 5180 if (mInput == NULL) { 5181 return NULL; 5182 } 5183 return &mInput->stream->common; 5184} 5185 5186status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5187{ 5188 // only one chain per input thread 5189 if (mEffectChains.size() != 0) { 5190 return INVALID_OPERATION; 5191 } 5192 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5193 5194 chain->setInBuffer(NULL); 5195 chain->setOutBuffer(NULL); 5196 5197 checkSuspendOnAddEffectChain_l(chain); 5198 5199 mEffectChains.add(chain); 5200 5201 return NO_ERROR; 5202} 5203 5204size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5205{ 5206 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5207 ALOGW_IF(mEffectChains.size() != 1, 5208 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5209 chain.get(), mEffectChains.size(), this); 5210 if (mEffectChains.size() == 1) { 5211 mEffectChains.removeAt(0); 5212 } 5213 return 0; 5214} 5215 5216}; // namespace android 5217