Threads.cpp revision 0fd582e3ce5243c3e5a429fee3330aafc69b69fa
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// Whether to use fast mixer 113static const enum { 114 FastMixer_Never, // never initialize or use: for debugging only 115 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 116 // normal mixer multiplier is 1 117 FastMixer_Static, // initialize if needed, then use all the time if initialized, 118 // multiplier is calculated based on min & max normal mixer buffer size 119 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 // FIXME for FastMixer_Dynamic: 122 // Supporting this option will require fixing HALs that can't handle large writes. 123 // For example, one HAL implementation returns an error from a large write, 124 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 125 // We could either fix the HAL implementations, or provide a wrapper that breaks 126 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 127} kUseFastMixer = FastMixer_Static; 128 129// Priorities for requestPriority 130static const int kPriorityAudioApp = 2; 131static const int kPriorityFastMixer = 3; 132 133// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 134// for the track. The client then sub-divides this into smaller buffers for its use. 135// Currently the client uses double-buffering by default, but doesn't tell us about that. 136// So for now we just assume that client is double-buffered. 137// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 138// N-buffering, so AudioFlinger could allocate the right amount of memory. 139// See the client's minBufCount and mNotificationFramesAct calculations for details. 140static const int kFastTrackMultiplier = 1; 141 142// ---------------------------------------------------------------------------- 143 144#ifdef ADD_BATTERY_DATA 145// To collect the amplifier usage 146static void addBatteryData(uint32_t params) { 147 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 148 if (service == NULL) { 149 // it already logged 150 return; 151 } 152 153 service->addBatteryData(params); 154} 155#endif 156 157 158// ---------------------------------------------------------------------------- 159// CPU Stats 160// ---------------------------------------------------------------------------- 161 162class CpuStats { 163public: 164 CpuStats(); 165 void sample(const String8 &title); 166#ifdef DEBUG_CPU_USAGE 167private: 168 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 169 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 170 171 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 172 173 int mCpuNum; // thread's current CPU number 174 int mCpukHz; // frequency of thread's current CPU in kHz 175#endif 176}; 177 178CpuStats::CpuStats() 179#ifdef DEBUG_CPU_USAGE 180 : mCpuNum(-1), mCpukHz(-1) 181#endif 182{ 183} 184 185void CpuStats::sample(const String8 &title) { 186#ifdef DEBUG_CPU_USAGE 187 // get current thread's delta CPU time in wall clock ns 188 double wcNs; 189 bool valid = mCpuUsage.sampleAndEnable(wcNs); 190 191 // record sample for wall clock statistics 192 if (valid) { 193 mWcStats.sample(wcNs); 194 } 195 196 // get the current CPU number 197 int cpuNum = sched_getcpu(); 198 199 // get the current CPU frequency in kHz 200 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 201 202 // check if either CPU number or frequency changed 203 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 204 mCpuNum = cpuNum; 205 mCpukHz = cpukHz; 206 // ignore sample for purposes of cycles 207 valid = false; 208 } 209 210 // if no change in CPU number or frequency, then record sample for cycle statistics 211 if (valid && mCpukHz > 0) { 212 double cycles = wcNs * cpukHz * 0.000001; 213 mHzStats.sample(cycles); 214 } 215 216 unsigned n = mWcStats.n(); 217 // mCpuUsage.elapsed() is expensive, so don't call it every loop 218 if ((n & 127) == 1) { 219 long long elapsed = mCpuUsage.elapsed(); 220 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 221 double perLoop = elapsed / (double) n; 222 double perLoop100 = perLoop * 0.01; 223 double perLoop1k = perLoop * 0.001; 224 double mean = mWcStats.mean(); 225 double stddev = mWcStats.stddev(); 226 double minimum = mWcStats.minimum(); 227 double maximum = mWcStats.maximum(); 228 double meanCycles = mHzStats.mean(); 229 double stddevCycles = mHzStats.stddev(); 230 double minCycles = mHzStats.minimum(); 231 double maxCycles = mHzStats.maximum(); 232 mCpuUsage.resetElapsed(); 233 mWcStats.reset(); 234 mHzStats.reset(); 235 ALOGD("CPU usage for %s over past %.1f secs\n" 236 " (%u mixer loops at %.1f mean ms per loop):\n" 237 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 238 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 239 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 240 title.string(), 241 elapsed * .000000001, n, perLoop * .000001, 242 mean * .001, 243 stddev * .001, 244 minimum * .001, 245 maximum * .001, 246 mean / perLoop100, 247 stddev / perLoop100, 248 minimum / perLoop100, 249 maximum / perLoop100, 250 meanCycles / perLoop1k, 251 stddevCycles / perLoop1k, 252 minCycles / perLoop1k, 253 maxCycles / perLoop1k); 254 255 } 256 } 257#endif 258}; 259 260// ---------------------------------------------------------------------------- 261// ThreadBase 262// ---------------------------------------------------------------------------- 263 264AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 265 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 266 : Thread(false /*canCallJava*/), 267 mType(type), 268 mAudioFlinger(audioFlinger), 269 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 270 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 271 mParamStatus(NO_ERROR), 272 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 273 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 274 // mName will be set by concrete (non-virtual) subclass 275 mDeathRecipient(new PMDeathRecipient(this)) 276{ 277} 278 279AudioFlinger::ThreadBase::~ThreadBase() 280{ 281 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 282 for (size_t i = 0; i < mConfigEvents.size(); i++) { 283 delete mConfigEvents[i]; 284 } 285 mConfigEvents.clear(); 286 287 mParamCond.broadcast(); 288 // do not lock the mutex in destructor 289 releaseWakeLock_l(); 290 if (mPowerManager != 0) { 291 sp<IBinder> binder = mPowerManager->asBinder(); 292 binder->unlinkToDeath(mDeathRecipient); 293 } 294} 295 296status_t AudioFlinger::ThreadBase::readyToRun() 297{ 298 status_t status = initCheck(); 299 if (status == NO_ERROR) { 300 ALOGI("AudioFlinger's thread %p ready to run", this); 301 } else { 302 ALOGE("No working audio driver found."); 303 } 304 return status; 305} 306 307void AudioFlinger::ThreadBase::exit() 308{ 309 ALOGV("ThreadBase::exit"); 310 // do any cleanup required for exit to succeed 311 preExit(); 312 { 313 // This lock prevents the following race in thread (uniprocessor for illustration): 314 // if (!exitPending()) { 315 // // context switch from here to exit() 316 // // exit() calls requestExit(), what exitPending() observes 317 // // exit() calls signal(), which is dropped since no waiters 318 // // context switch back from exit() to here 319 // mWaitWorkCV.wait(...); 320 // // now thread is hung 321 // } 322 AutoMutex lock(mLock); 323 requestExit(); 324 mWaitWorkCV.broadcast(); 325 } 326 // When Thread::requestExitAndWait is made virtual and this method is renamed to 327 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 328 requestExitAndWait(); 329} 330 331status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 332{ 333 status_t status; 334 335 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 336 Mutex::Autolock _l(mLock); 337 338 mNewParameters.add(keyValuePairs); 339 mWaitWorkCV.signal(); 340 // wait condition with timeout in case the thread loop has exited 341 // before the request could be processed 342 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 343 status = mParamStatus; 344 mWaitWorkCV.signal(); 345 } else { 346 status = TIMED_OUT; 347 } 348 return status; 349} 350 351void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 352{ 353 Mutex::Autolock _l(mLock); 354 sendIoConfigEvent_l(event, param); 355} 356 357// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 358void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 359{ 360 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 361 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 362 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 363 param); 364 mWaitWorkCV.signal(); 365} 366 367// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 368void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 369{ 370 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 371 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 372 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 373 mConfigEvents.size(), pid, tid, prio); 374 mWaitWorkCV.signal(); 375} 376 377void AudioFlinger::ThreadBase::processConfigEvents() 378{ 379 Mutex::Autolock _l(mLock); 380 processConfigEvents_l(); 381} 382 383// post condition: mConfigEvents.isEmpty() 384void AudioFlinger::ThreadBase::processConfigEvents_l() 385{ 386 while (!mConfigEvents.isEmpty()) { 387 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 388 ConfigEvent *event = mConfigEvents[0]; 389 mConfigEvents.removeAt(0); 390 // release mLock before locking AudioFlinger mLock: lock order is always 391 // AudioFlinger then ThreadBase to avoid cross deadlock 392 mLock.unlock(); 393 switch (event->type()) { 394 case CFG_EVENT_PRIO: { 395 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 396 // FIXME Need to understand why this has be done asynchronously 397 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 398 true /*asynchronous*/); 399 if (err != 0) { 400 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 401 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 402 } 403 } break; 404 case CFG_EVENT_IO: { 405 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 406 { 407 Mutex::Autolock _l(mAudioFlinger->mLock); 408 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 409 } 410 } break; 411 default: 412 ALOGE("processConfigEvents() unknown event type %d", event->type()); 413 break; 414 } 415 delete event; 416 mLock.lock(); 417 } 418} 419 420void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 421{ 422 const size_t SIZE = 256; 423 char buffer[SIZE]; 424 String8 result; 425 426 bool locked = AudioFlinger::dumpTryLock(mLock); 427 if (!locked) { 428 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 429 write(fd, buffer, strlen(buffer)); 430 } 431 432 snprintf(buffer, SIZE, "io handle: %d\n", mId); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 437 result.append(buffer); 438 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 439 result.append(buffer); 440 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 441 result.append(buffer); 442 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 443 result.append(buffer); 444 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 445 result.append(buffer); 446 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 447 result.append(buffer); 448 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 449 result.append(buffer); 450 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 451 result.append(buffer); 452 453 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 454 result.append(buffer); 455 result.append(" Index Command"); 456 for (size_t i = 0; i < mNewParameters.size(); ++i) { 457 snprintf(buffer, SIZE, "\n %02d ", i); 458 result.append(buffer); 459 result.append(mNewParameters[i]); 460 } 461 462 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 463 result.append(buffer); 464 for (size_t i = 0; i < mConfigEvents.size(); i++) { 465 mConfigEvents[i]->dump(buffer, SIZE); 466 result.append(buffer); 467 } 468 result.append("\n"); 469 470 write(fd, result.string(), result.size()); 471 472 if (locked) { 473 mLock.unlock(); 474 } 475} 476 477void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 478{ 479 const size_t SIZE = 256; 480 char buffer[SIZE]; 481 String8 result; 482 483 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 484 write(fd, buffer, strlen(buffer)); 485 486 for (size_t i = 0; i < mEffectChains.size(); ++i) { 487 sp<EffectChain> chain = mEffectChains[i]; 488 if (chain != 0) { 489 chain->dump(fd, args); 490 } 491 } 492} 493 494void AudioFlinger::ThreadBase::acquireWakeLock() 495{ 496 Mutex::Autolock _l(mLock); 497 acquireWakeLock_l(); 498} 499 500void AudioFlinger::ThreadBase::acquireWakeLock_l() 501{ 502 if (mPowerManager == 0) { 503 // use checkService() to avoid blocking if power service is not up yet 504 sp<IBinder> binder = 505 defaultServiceManager()->checkService(String16("power")); 506 if (binder == 0) { 507 ALOGW("Thread %s cannot connect to the power manager service", mName); 508 } else { 509 mPowerManager = interface_cast<IPowerManager>(binder); 510 binder->linkToDeath(mDeathRecipient); 511 } 512 } 513 if (mPowerManager != 0) { 514 sp<IBinder> binder = new BBinder(); 515 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 516 binder, 517 String16(mName), 518 String16("media")); 519 if (status == NO_ERROR) { 520 mWakeLockToken = binder; 521 } 522 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 523 } 524} 525 526void AudioFlinger::ThreadBase::releaseWakeLock() 527{ 528 Mutex::Autolock _l(mLock); 529 releaseWakeLock_l(); 530} 531 532void AudioFlinger::ThreadBase::releaseWakeLock_l() 533{ 534 if (mWakeLockToken != 0) { 535 ALOGV("releaseWakeLock_l() %s", mName); 536 if (mPowerManager != 0) { 537 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 538 } 539 mWakeLockToken.clear(); 540 } 541} 542 543void AudioFlinger::ThreadBase::clearPowerManager() 544{ 545 Mutex::Autolock _l(mLock); 546 releaseWakeLock_l(); 547 mPowerManager.clear(); 548} 549 550void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 551{ 552 sp<ThreadBase> thread = mThread.promote(); 553 if (thread != 0) { 554 thread->clearPowerManager(); 555 } 556 ALOGW("power manager service died !!!"); 557} 558 559void AudioFlinger::ThreadBase::setEffectSuspended( 560 const effect_uuid_t *type, bool suspend, int sessionId) 561{ 562 Mutex::Autolock _l(mLock); 563 setEffectSuspended_l(type, suspend, sessionId); 564} 565 566void AudioFlinger::ThreadBase::setEffectSuspended_l( 567 const effect_uuid_t *type, bool suspend, int sessionId) 568{ 569 sp<EffectChain> chain = getEffectChain_l(sessionId); 570 if (chain != 0) { 571 if (type != NULL) { 572 chain->setEffectSuspended_l(type, suspend); 573 } else { 574 chain->setEffectSuspendedAll_l(suspend); 575 } 576 } 577 578 updateSuspendedSessions_l(type, suspend, sessionId); 579} 580 581void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 582{ 583 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 584 if (index < 0) { 585 return; 586 } 587 588 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 589 mSuspendedSessions.valueAt(index); 590 591 for (size_t i = 0; i < sessionEffects.size(); i++) { 592 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 593 for (int j = 0; j < desc->mRefCount; j++) { 594 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 595 chain->setEffectSuspendedAll_l(true); 596 } else { 597 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 598 desc->mType.timeLow); 599 chain->setEffectSuspended_l(&desc->mType, true); 600 } 601 } 602 } 603} 604 605void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 606 bool suspend, 607 int sessionId) 608{ 609 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 610 611 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 612 613 if (suspend) { 614 if (index >= 0) { 615 sessionEffects = mSuspendedSessions.valueAt(index); 616 } else { 617 mSuspendedSessions.add(sessionId, sessionEffects); 618 } 619 } else { 620 if (index < 0) { 621 return; 622 } 623 sessionEffects = mSuspendedSessions.valueAt(index); 624 } 625 626 627 int key = EffectChain::kKeyForSuspendAll; 628 if (type != NULL) { 629 key = type->timeLow; 630 } 631 index = sessionEffects.indexOfKey(key); 632 633 sp<SuspendedSessionDesc> desc; 634 if (suspend) { 635 if (index >= 0) { 636 desc = sessionEffects.valueAt(index); 637 } else { 638 desc = new SuspendedSessionDesc(); 639 if (type != NULL) { 640 desc->mType = *type; 641 } 642 sessionEffects.add(key, desc); 643 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 644 } 645 desc->mRefCount++; 646 } else { 647 if (index < 0) { 648 return; 649 } 650 desc = sessionEffects.valueAt(index); 651 if (--desc->mRefCount == 0) { 652 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 653 sessionEffects.removeItemsAt(index); 654 if (sessionEffects.isEmpty()) { 655 ALOGV("updateSuspendedSessions_l() restore removing session %d", 656 sessionId); 657 mSuspendedSessions.removeItem(sessionId); 658 } 659 } 660 } 661 if (!sessionEffects.isEmpty()) { 662 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 663 } 664} 665 666void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 667 bool enabled, 668 int sessionId) 669{ 670 Mutex::Autolock _l(mLock); 671 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 672} 673 674void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 675 bool enabled, 676 int sessionId) 677{ 678 if (mType != RECORD) { 679 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 680 // another session. This gives the priority to well behaved effect control panels 681 // and applications not using global effects. 682 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 683 // global effects 684 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 685 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 686 } 687 } 688 689 sp<EffectChain> chain = getEffectChain_l(sessionId); 690 if (chain != 0) { 691 chain->checkSuspendOnEffectEnabled(effect, enabled); 692 } 693} 694 695// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 696sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 697 const sp<AudioFlinger::Client>& client, 698 const sp<IEffectClient>& effectClient, 699 int32_t priority, 700 int sessionId, 701 effect_descriptor_t *desc, 702 int *enabled, 703 status_t *status) 704{ 705 sp<EffectModule> effect; 706 sp<EffectHandle> handle; 707 status_t lStatus; 708 sp<EffectChain> chain; 709 bool chainCreated = false; 710 bool effectCreated = false; 711 bool effectRegistered = false; 712 713 lStatus = initCheck(); 714 if (lStatus != NO_ERROR) { 715 ALOGW("createEffect_l() Audio driver not initialized."); 716 goto Exit; 717 } 718 719 // Do not allow effects with session ID 0 on direct output or duplicating threads 720 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 721 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 722 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 723 desc->name, sessionId); 724 lStatus = BAD_VALUE; 725 goto Exit; 726 } 727 // Only Pre processor effects are allowed on input threads and only on input threads 728 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 729 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 730 desc->name, desc->flags, mType); 731 lStatus = BAD_VALUE; 732 goto Exit; 733 } 734 735 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 736 737 { // scope for mLock 738 Mutex::Autolock _l(mLock); 739 740 // check for existing effect chain with the requested audio session 741 chain = getEffectChain_l(sessionId); 742 if (chain == 0) { 743 // create a new chain for this session 744 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 745 chain = new EffectChain(this, sessionId); 746 addEffectChain_l(chain); 747 chain->setStrategy(getStrategyForSession_l(sessionId)); 748 chainCreated = true; 749 } else { 750 effect = chain->getEffectFromDesc_l(desc); 751 } 752 753 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 754 755 if (effect == 0) { 756 int id = mAudioFlinger->nextUniqueId(); 757 // Check CPU and memory usage 758 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 759 if (lStatus != NO_ERROR) { 760 goto Exit; 761 } 762 effectRegistered = true; 763 // create a new effect module if none present in the chain 764 effect = new EffectModule(this, chain, desc, id, sessionId); 765 lStatus = effect->status(); 766 if (lStatus != NO_ERROR) { 767 goto Exit; 768 } 769 lStatus = chain->addEffect_l(effect); 770 if (lStatus != NO_ERROR) { 771 goto Exit; 772 } 773 effectCreated = true; 774 775 effect->setDevice(mOutDevice); 776 effect->setDevice(mInDevice); 777 effect->setMode(mAudioFlinger->getMode()); 778 effect->setAudioSource(mAudioSource); 779 } 780 // create effect handle and connect it to effect module 781 handle = new EffectHandle(effect, client, effectClient, priority); 782 lStatus = effect->addHandle(handle.get()); 783 if (enabled != NULL) { 784 *enabled = (int)effect->isEnabled(); 785 } 786 } 787 788Exit: 789 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 790 Mutex::Autolock _l(mLock); 791 if (effectCreated) { 792 chain->removeEffect_l(effect); 793 } 794 if (effectRegistered) { 795 AudioSystem::unregisterEffect(effect->id()); 796 } 797 if (chainCreated) { 798 removeEffectChain_l(chain); 799 } 800 handle.clear(); 801 } 802 803 *status = lStatus; 804 return handle; 805} 806 807sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 808{ 809 Mutex::Autolock _l(mLock); 810 return getEffect_l(sessionId, effectId); 811} 812 813sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 814{ 815 sp<EffectChain> chain = getEffectChain_l(sessionId); 816 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 817} 818 819// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 820// PlaybackThread::mLock held 821status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 822{ 823 // check for existing effect chain with the requested audio session 824 int sessionId = effect->sessionId(); 825 sp<EffectChain> chain = getEffectChain_l(sessionId); 826 bool chainCreated = false; 827 828 if (chain == 0) { 829 // create a new chain for this session 830 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 831 chain = new EffectChain(this, sessionId); 832 addEffectChain_l(chain); 833 chain->setStrategy(getStrategyForSession_l(sessionId)); 834 chainCreated = true; 835 } 836 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 837 838 if (chain->getEffectFromId_l(effect->id()) != 0) { 839 ALOGW("addEffect_l() %p effect %s already present in chain %p", 840 this, effect->desc().name, chain.get()); 841 return BAD_VALUE; 842 } 843 844 status_t status = chain->addEffect_l(effect); 845 if (status != NO_ERROR) { 846 if (chainCreated) { 847 removeEffectChain_l(chain); 848 } 849 return status; 850 } 851 852 effect->setDevice(mOutDevice); 853 effect->setDevice(mInDevice); 854 effect->setMode(mAudioFlinger->getMode()); 855 effect->setAudioSource(mAudioSource); 856 return NO_ERROR; 857} 858 859void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 860 861 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 862 effect_descriptor_t desc = effect->desc(); 863 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 864 detachAuxEffect_l(effect->id()); 865 } 866 867 sp<EffectChain> chain = effect->chain().promote(); 868 if (chain != 0) { 869 // remove effect chain if removing last effect 870 if (chain->removeEffect_l(effect) == 0) { 871 removeEffectChain_l(chain); 872 } 873 } else { 874 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 875 } 876} 877 878void AudioFlinger::ThreadBase::lockEffectChains_l( 879 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 880{ 881 effectChains = mEffectChains; 882 for (size_t i = 0; i < mEffectChains.size(); i++) { 883 mEffectChains[i]->lock(); 884 } 885} 886 887void AudioFlinger::ThreadBase::unlockEffectChains( 888 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 889{ 890 for (size_t i = 0; i < effectChains.size(); i++) { 891 effectChains[i]->unlock(); 892 } 893} 894 895sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 896{ 897 Mutex::Autolock _l(mLock); 898 return getEffectChain_l(sessionId); 899} 900 901sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 902{ 903 size_t size = mEffectChains.size(); 904 for (size_t i = 0; i < size; i++) { 905 if (mEffectChains[i]->sessionId() == sessionId) { 906 return mEffectChains[i]; 907 } 908 } 909 return 0; 910} 911 912void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 913{ 914 Mutex::Autolock _l(mLock); 915 size_t size = mEffectChains.size(); 916 for (size_t i = 0; i < size; i++) { 917 mEffectChains[i]->setMode_l(mode); 918 } 919} 920 921void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 922 EffectHandle *handle, 923 bool unpinIfLast) { 924 925 Mutex::Autolock _l(mLock); 926 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 927 // delete the effect module if removing last handle on it 928 if (effect->removeHandle(handle) == 0) { 929 if (!effect->isPinned() || unpinIfLast) { 930 removeEffect_l(effect); 931 AudioSystem::unregisterEffect(effect->id()); 932 } 933 } 934} 935 936// ---------------------------------------------------------------------------- 937// Playback 938// ---------------------------------------------------------------------------- 939 940AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 941 AudioStreamOut* output, 942 audio_io_handle_t id, 943 audio_devices_t device, 944 type_t type) 945 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 946 mNormalFrameCount(0), mMixBuffer(NULL), 947 mSuspended(0), mBytesWritten(0), 948 // mStreamTypes[] initialized in constructor body 949 mOutput(output), 950 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 951 mMixerStatus(MIXER_IDLE), 952 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 953 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 954 mBytesRemaining(0), 955 mCurrentWriteLength(0), 956 mUseAsyncWrite(false), 957 mWriteBlocked(false), 958 mDraining(false), 959 mScreenState(AudioFlinger::mScreenState), 960 // index 0 is reserved for normal mixer's submix 961 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 962 // mLatchD, mLatchQ, 963 mLatchDValid(false), mLatchQValid(false) 964{ 965 snprintf(mName, kNameLength, "AudioOut_%X", id); 966 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 967 968 // Assumes constructor is called by AudioFlinger with it's mLock held, but 969 // it would be safer to explicitly pass initial masterVolume/masterMute as 970 // parameter. 971 // 972 // If the HAL we are using has support for master volume or master mute, 973 // then do not attenuate or mute during mixing (just leave the volume at 1.0 974 // and the mute set to false). 975 mMasterVolume = audioFlinger->masterVolume_l(); 976 mMasterMute = audioFlinger->masterMute_l(); 977 if (mOutput && mOutput->audioHwDev) { 978 if (mOutput->audioHwDev->canSetMasterVolume()) { 979 mMasterVolume = 1.0; 980 } 981 982 if (mOutput->audioHwDev->canSetMasterMute()) { 983 mMasterMute = false; 984 } 985 } 986 987 readOutputParameters(); 988 989 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 990 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 991 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 992 stream = (audio_stream_type_t) (stream + 1)) { 993 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 994 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 995 } 996 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 997 // because mAudioFlinger doesn't have one to copy from 998} 999 1000AudioFlinger::PlaybackThread::~PlaybackThread() 1001{ 1002 mAudioFlinger->unregisterWriter(mNBLogWriter); 1003 delete[] mMixBuffer; 1004} 1005 1006void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1007{ 1008 dumpInternals(fd, args); 1009 dumpTracks(fd, args); 1010 dumpEffectChains(fd, args); 1011} 1012 1013void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1014{ 1015 const size_t SIZE = 256; 1016 char buffer[SIZE]; 1017 String8 result; 1018 1019 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1020 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1021 const stream_type_t *st = &mStreamTypes[i]; 1022 if (i > 0) { 1023 result.appendFormat(", "); 1024 } 1025 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1026 if (st->mute) { 1027 result.append("M"); 1028 } 1029 } 1030 result.append("\n"); 1031 write(fd, result.string(), result.length()); 1032 result.clear(); 1033 1034 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1035 result.append(buffer); 1036 Track::appendDumpHeader(result); 1037 for (size_t i = 0; i < mTracks.size(); ++i) { 1038 sp<Track> track = mTracks[i]; 1039 if (track != 0) { 1040 track->dump(buffer, SIZE); 1041 result.append(buffer); 1042 } 1043 } 1044 1045 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1046 result.append(buffer); 1047 Track::appendDumpHeader(result); 1048 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1049 sp<Track> track = mActiveTracks[i].promote(); 1050 if (track != 0) { 1051 track->dump(buffer, SIZE); 1052 result.append(buffer); 1053 } 1054 } 1055 write(fd, result.string(), result.size()); 1056 1057 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1058 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1059 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1060 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1061} 1062 1063void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1064{ 1065 const size_t SIZE = 256; 1066 char buffer[SIZE]; 1067 String8 result; 1068 1069 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1070 result.append(buffer); 1071 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1072 result.append(buffer); 1073 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1074 ns2ms(systemTime() - mLastWriteTime)); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1079 result.append(buffer); 1080 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1081 result.append(buffer); 1082 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1083 result.append(buffer); 1084 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1085 result.append(buffer); 1086 write(fd, result.string(), result.size()); 1087 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1088 1089 dumpBase(fd, args); 1090} 1091 1092// Thread virtuals 1093 1094void AudioFlinger::PlaybackThread::onFirstRef() 1095{ 1096 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1097} 1098 1099// ThreadBase virtuals 1100void AudioFlinger::PlaybackThread::preExit() 1101{ 1102 ALOGV(" preExit()"); 1103 // FIXME this is using hard-coded strings but in the future, this functionality will be 1104 // converted to use audio HAL extensions required to support tunneling 1105 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1106} 1107 1108// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1109sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1110 const sp<AudioFlinger::Client>& client, 1111 audio_stream_type_t streamType, 1112 uint32_t sampleRate, 1113 audio_format_t format, 1114 audio_channel_mask_t channelMask, 1115 size_t frameCount, 1116 const sp<IMemory>& sharedBuffer, 1117 int sessionId, 1118 IAudioFlinger::track_flags_t *flags, 1119 pid_t tid, 1120 status_t *status) 1121{ 1122 sp<Track> track; 1123 status_t lStatus; 1124 1125 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1126 1127 // client expresses a preference for FAST, but we get the final say 1128 if (*flags & IAudioFlinger::TRACK_FAST) { 1129 if ( 1130 // not timed 1131 (!isTimed) && 1132 // either of these use cases: 1133 ( 1134 // use case 1: shared buffer with any frame count 1135 ( 1136 (sharedBuffer != 0) 1137 ) || 1138 // use case 2: callback handler and frame count is default or at least as large as HAL 1139 ( 1140 (tid != -1) && 1141 ((frameCount == 0) || 1142 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1143 ) 1144 ) && 1145 // PCM data 1146 audio_is_linear_pcm(format) && 1147 // mono or stereo 1148 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1149 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1150#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1151 // hardware sample rate 1152 (sampleRate == mSampleRate) && 1153#endif 1154 // normal mixer has an associated fast mixer 1155 hasFastMixer() && 1156 // there are sufficient fast track slots available 1157 (mFastTrackAvailMask != 0) 1158 // FIXME test that MixerThread for this fast track has a capable output HAL 1159 // FIXME add a permission test also? 1160 ) { 1161 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1162 if (frameCount == 0) { 1163 frameCount = mFrameCount * kFastTrackMultiplier; 1164 } 1165 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1166 frameCount, mFrameCount); 1167 } else { 1168 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1169 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1170 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1171 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1172 audio_is_linear_pcm(format), 1173 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1174 *flags &= ~IAudioFlinger::TRACK_FAST; 1175 // For compatibility with AudioTrack calculation, buffer depth is forced 1176 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1177 // This is probably too conservative, but legacy application code may depend on it. 1178 // If you change this calculation, also review the start threshold which is related. 1179 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1180 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1181 if (minBufCount < 2) { 1182 minBufCount = 2; 1183 } 1184 size_t minFrameCount = mNormalFrameCount * minBufCount; 1185 if (frameCount < minFrameCount) { 1186 frameCount = minFrameCount; 1187 } 1188 } 1189 } 1190 1191 if (mType == DIRECT) { 1192 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1193 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1194 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1195 "for output %p with format %d", 1196 sampleRate, format, channelMask, mOutput, mFormat); 1197 lStatus = BAD_VALUE; 1198 goto Exit; 1199 } 1200 } 1201 } else if (mType == OFFLOAD) { 1202 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1203 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1204 "for output %p with format %d", 1205 sampleRate, format, channelMask, mOutput, mFormat); 1206 lStatus = BAD_VALUE; 1207 goto Exit; 1208 } 1209 } else { 1210 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1211 ALOGE("createTrack_l() Bad parameter: format %d \"" 1212 "for output %p with format %d", 1213 format, mOutput, mFormat); 1214 lStatus = BAD_VALUE; 1215 goto Exit; 1216 } 1217 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1218 if (sampleRate > mSampleRate*2) { 1219 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1220 lStatus = BAD_VALUE; 1221 goto Exit; 1222 } 1223 } 1224 1225 lStatus = initCheck(); 1226 if (lStatus != NO_ERROR) { 1227 ALOGE("Audio driver not initialized."); 1228 goto Exit; 1229 } 1230 1231 { // scope for mLock 1232 Mutex::Autolock _l(mLock); 1233 1234 // all tracks in same audio session must share the same routing strategy otherwise 1235 // conflicts will happen when tracks are moved from one output to another by audio policy 1236 // manager 1237 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1238 for (size_t i = 0; i < mTracks.size(); ++i) { 1239 sp<Track> t = mTracks[i]; 1240 if (t != 0 && !t->isOutputTrack()) { 1241 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1242 if (sessionId == t->sessionId() && strategy != actual) { 1243 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1244 strategy, actual); 1245 lStatus = BAD_VALUE; 1246 goto Exit; 1247 } 1248 } 1249 } 1250 1251 if (!isTimed) { 1252 track = new Track(this, client, streamType, sampleRate, format, 1253 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1254 } else { 1255 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1256 channelMask, frameCount, sharedBuffer, sessionId); 1257 } 1258 1259 // new Track always returns non-NULL, 1260 // but TimedTrack::create() is a factory that could fail by returning NULL 1261 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1262 if (lStatus != NO_ERROR) { 1263 track.clear(); 1264 goto Exit; 1265 } 1266 1267 mTracks.add(track); 1268 1269 sp<EffectChain> chain = getEffectChain_l(sessionId); 1270 if (chain != 0) { 1271 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1272 track->setMainBuffer(chain->inBuffer()); 1273 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1274 chain->incTrackCnt(); 1275 } 1276 1277 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1278 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1279 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1280 // so ask activity manager to do this on our behalf 1281 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1282 } 1283 } 1284 1285 lStatus = NO_ERROR; 1286 1287Exit: 1288 *status = lStatus; 1289 return track; 1290} 1291 1292uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1293{ 1294 return latency; 1295} 1296 1297uint32_t AudioFlinger::PlaybackThread::latency() const 1298{ 1299 Mutex::Autolock _l(mLock); 1300 return latency_l(); 1301} 1302uint32_t AudioFlinger::PlaybackThread::latency_l() const 1303{ 1304 if (initCheck() == NO_ERROR) { 1305 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1306 } else { 1307 return 0; 1308 } 1309} 1310 1311void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1312{ 1313 Mutex::Autolock _l(mLock); 1314 // Don't apply master volume in SW if our HAL can do it for us. 1315 if (mOutput && mOutput->audioHwDev && 1316 mOutput->audioHwDev->canSetMasterVolume()) { 1317 mMasterVolume = 1.0; 1318 } else { 1319 mMasterVolume = value; 1320 } 1321} 1322 1323void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1324{ 1325 Mutex::Autolock _l(mLock); 1326 // Don't apply master mute in SW if our HAL can do it for us. 1327 if (mOutput && mOutput->audioHwDev && 1328 mOutput->audioHwDev->canSetMasterMute()) { 1329 mMasterMute = false; 1330 } else { 1331 mMasterMute = muted; 1332 } 1333} 1334 1335void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1336{ 1337 Mutex::Autolock _l(mLock); 1338 mStreamTypes[stream].volume = value; 1339 signal_l(); 1340} 1341 1342void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1343{ 1344 Mutex::Autolock _l(mLock); 1345 mStreamTypes[stream].mute = muted; 1346 signal_l(); 1347} 1348 1349float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1350{ 1351 Mutex::Autolock _l(mLock); 1352 return mStreamTypes[stream].volume; 1353} 1354 1355// addTrack_l() must be called with ThreadBase::mLock held 1356status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1357{ 1358 status_t status = ALREADY_EXISTS; 1359 1360 // set retry count for buffer fill 1361 track->mRetryCount = kMaxTrackStartupRetries; 1362 if (mActiveTracks.indexOf(track) < 0) { 1363 // the track is newly added, make sure it fills up all its 1364 // buffers before playing. This is to ensure the client will 1365 // effectively get the latency it requested. 1366 if (!track->isOutputTrack()) { 1367 TrackBase::track_state state = track->mState; 1368 mLock.unlock(); 1369 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1370 mLock.lock(); 1371 // abort track was stopped/paused while we released the lock 1372 if (state != track->mState) { 1373 if (status == NO_ERROR) { 1374 mLock.unlock(); 1375 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1376 mLock.lock(); 1377 } 1378 return INVALID_OPERATION; 1379 } 1380 // abort if start is rejected by audio policy manager 1381 if (status != NO_ERROR) { 1382 return PERMISSION_DENIED; 1383 } 1384#ifdef ADD_BATTERY_DATA 1385 // to track the speaker usage 1386 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1387#endif 1388 } 1389 1390 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1391 track->mResetDone = false; 1392 track->mPresentationCompleteFrames = 0; 1393 mActiveTracks.add(track); 1394 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1395 if (chain != 0) { 1396 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1397 track->sessionId()); 1398 chain->incActiveTrackCnt(); 1399 } 1400 1401 status = NO_ERROR; 1402 } 1403 1404 ALOGV("mWaitWorkCV.broadcast"); 1405 mWaitWorkCV.broadcast(); 1406 1407 return status; 1408} 1409 1410bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1411{ 1412 track->terminate(); 1413 // active tracks are removed by threadLoop() 1414 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1415 track->mState = TrackBase::STOPPED; 1416 if (!trackActive) { 1417 removeTrack_l(track); 1418 } else if (track->isFastTrack() || track->isOffloaded()) { 1419 track->mState = TrackBase::STOPPING_1; 1420 } 1421 1422 return trackActive; 1423} 1424 1425void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1426{ 1427 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1428 mTracks.remove(track); 1429 deleteTrackName_l(track->name()); 1430 // redundant as track is about to be destroyed, for dumpsys only 1431 track->mName = -1; 1432 if (track->isFastTrack()) { 1433 int index = track->mFastIndex; 1434 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1435 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1436 mFastTrackAvailMask |= 1 << index; 1437 // redundant as track is about to be destroyed, for dumpsys only 1438 track->mFastIndex = -1; 1439 } 1440 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1441 if (chain != 0) { 1442 chain->decTrackCnt(); 1443 } 1444} 1445 1446void AudioFlinger::PlaybackThread::signal_l() 1447{ 1448 // Thread could be blocked waiting for async 1449 // so signal it to handle state changes immediately 1450 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1451 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1452 mSignalPending = true; 1453 mWaitWorkCV.signal(); 1454} 1455 1456String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1457{ 1458 Mutex::Autolock _l(mLock); 1459 if (initCheck() != NO_ERROR) { 1460 return String8(); 1461 } 1462 1463 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1464 const String8 out_s8(s); 1465 free(s); 1466 return out_s8; 1467} 1468 1469// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1470void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1471 AudioSystem::OutputDescriptor desc; 1472 void *param2 = NULL; 1473 1474 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1475 param); 1476 1477 switch (event) { 1478 case AudioSystem::OUTPUT_OPENED: 1479 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1480 desc.channelMask = mChannelMask; 1481 desc.samplingRate = mSampleRate; 1482 desc.format = mFormat; 1483 desc.frameCount = mNormalFrameCount; // FIXME see 1484 // AudioFlinger::frameCount(audio_io_handle_t) 1485 desc.latency = latency(); 1486 param2 = &desc; 1487 break; 1488 1489 case AudioSystem::STREAM_CONFIG_CHANGED: 1490 param2 = ¶m; 1491 case AudioSystem::OUTPUT_CLOSED: 1492 default: 1493 break; 1494 } 1495 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1496} 1497 1498void AudioFlinger::PlaybackThread::writeCallback() 1499{ 1500 ALOG_ASSERT(mCallbackThread != 0); 1501 mCallbackThread->setWriteBlocked(false); 1502} 1503 1504void AudioFlinger::PlaybackThread::drainCallback() 1505{ 1506 ALOG_ASSERT(mCallbackThread != 0); 1507 mCallbackThread->setDraining(false); 1508} 1509 1510void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1511{ 1512 Mutex::Autolock _l(mLock); 1513 mWriteBlocked = value; 1514 if (!value) { 1515 mWaitWorkCV.signal(); 1516 } 1517} 1518 1519void AudioFlinger::PlaybackThread::setDraining(bool value) 1520{ 1521 Mutex::Autolock _l(mLock); 1522 mDraining = value; 1523 if (!value) { 1524 mWaitWorkCV.signal(); 1525 } 1526} 1527 1528// static 1529int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1530 void *param, 1531 void *cookie) 1532{ 1533 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1534 ALOGV("asyncCallback() event %d", event); 1535 switch (event) { 1536 case STREAM_CBK_EVENT_WRITE_READY: 1537 me->writeCallback(); 1538 break; 1539 case STREAM_CBK_EVENT_DRAIN_READY: 1540 me->drainCallback(); 1541 break; 1542 default: 1543 ALOGW("asyncCallback() unknown event %d", event); 1544 break; 1545 } 1546 return 0; 1547} 1548 1549void AudioFlinger::PlaybackThread::readOutputParameters() 1550{ 1551 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1552 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1553 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1554 if (!audio_is_output_channel(mChannelMask)) { 1555 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1556 } 1557 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1558 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1559 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1560 } 1561 mChannelCount = popcount(mChannelMask); 1562 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1563 if (!audio_is_valid_format(mFormat)) { 1564 LOG_FATAL("HAL format %d not valid for output", mFormat); 1565 } 1566 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1567 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1568 mFormat); 1569 } 1570 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1571 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1572 mFrameCount = mBufferSize / mFrameSize; 1573 if (mFrameCount & 15) { 1574 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1575 mFrameCount); 1576 } 1577 1578 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1579 (mOutput->stream->set_callback != NULL)) { 1580 if (mOutput->stream->set_callback(mOutput->stream, 1581 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1582 mUseAsyncWrite = true; 1583 } 1584 } 1585 1586 // Calculate size of normal mix buffer relative to the HAL output buffer size 1587 double multiplier = 1.0; 1588 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1589 kUseFastMixer == FastMixer_Dynamic)) { 1590 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1591 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1592 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1593 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1594 maxNormalFrameCount = maxNormalFrameCount & ~15; 1595 if (maxNormalFrameCount < minNormalFrameCount) { 1596 maxNormalFrameCount = minNormalFrameCount; 1597 } 1598 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1599 if (multiplier <= 1.0) { 1600 multiplier = 1.0; 1601 } else if (multiplier <= 2.0) { 1602 if (2 * mFrameCount <= maxNormalFrameCount) { 1603 multiplier = 2.0; 1604 } else { 1605 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1606 } 1607 } else { 1608 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1609 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1610 // track, but we sometimes have to do this to satisfy the maximum frame count 1611 // constraint) 1612 // FIXME this rounding up should not be done if no HAL SRC 1613 uint32_t truncMult = (uint32_t) multiplier; 1614 if ((truncMult & 1)) { 1615 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1616 ++truncMult; 1617 } 1618 } 1619 multiplier = (double) truncMult; 1620 } 1621 } 1622 mNormalFrameCount = multiplier * mFrameCount; 1623 // round up to nearest 16 frames to satisfy AudioMixer 1624 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1625 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1626 mNormalFrameCount); 1627 1628 delete[] mMixBuffer; 1629 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1630 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1631 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1632 memset(mMixBuffer, 0, normalBufferSize); 1633 1634 // force reconfiguration of effect chains and engines to take new buffer size and audio 1635 // parameters into account 1636 // Note that mLock is not held when readOutputParameters() is called from the constructor 1637 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1638 // matter. 1639 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1640 Vector< sp<EffectChain> > effectChains = mEffectChains; 1641 for (size_t i = 0; i < effectChains.size(); i ++) { 1642 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1643 } 1644} 1645 1646 1647status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1648{ 1649 if (halFrames == NULL || dspFrames == NULL) { 1650 return BAD_VALUE; 1651 } 1652 Mutex::Autolock _l(mLock); 1653 if (initCheck() != NO_ERROR) { 1654 return INVALID_OPERATION; 1655 } 1656 size_t framesWritten = mBytesWritten / mFrameSize; 1657 *halFrames = framesWritten; 1658 1659 if (isSuspended()) { 1660 // return an estimation of rendered frames when the output is suspended 1661 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1662 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1663 return NO_ERROR; 1664 } else { 1665 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1666 } 1667} 1668 1669uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1670{ 1671 Mutex::Autolock _l(mLock); 1672 uint32_t result = 0; 1673 if (getEffectChain_l(sessionId) != 0) { 1674 result = EFFECT_SESSION; 1675 } 1676 1677 for (size_t i = 0; i < mTracks.size(); ++i) { 1678 sp<Track> track = mTracks[i]; 1679 if (sessionId == track->sessionId() && !track->isInvalid()) { 1680 result |= TRACK_SESSION; 1681 break; 1682 } 1683 } 1684 1685 return result; 1686} 1687 1688uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1689{ 1690 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1691 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1692 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1693 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1694 } 1695 for (size_t i = 0; i < mTracks.size(); i++) { 1696 sp<Track> track = mTracks[i]; 1697 if (sessionId == track->sessionId() && !track->isInvalid()) { 1698 return AudioSystem::getStrategyForStream(track->streamType()); 1699 } 1700 } 1701 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1702} 1703 1704 1705AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1706{ 1707 Mutex::Autolock _l(mLock); 1708 return mOutput; 1709} 1710 1711AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1712{ 1713 Mutex::Autolock _l(mLock); 1714 AudioStreamOut *output = mOutput; 1715 mOutput = NULL; 1716 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1717 // must push a NULL and wait for ack 1718 mOutputSink.clear(); 1719 mPipeSink.clear(); 1720 mNormalSink.clear(); 1721 return output; 1722} 1723 1724// this method must always be called either with ThreadBase mLock held or inside the thread loop 1725audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1726{ 1727 if (mOutput == NULL) { 1728 return NULL; 1729 } 1730 return &mOutput->stream->common; 1731} 1732 1733uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1734{ 1735 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1736} 1737 1738status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1739{ 1740 if (!isValidSyncEvent(event)) { 1741 return BAD_VALUE; 1742 } 1743 1744 Mutex::Autolock _l(mLock); 1745 1746 for (size_t i = 0; i < mTracks.size(); ++i) { 1747 sp<Track> track = mTracks[i]; 1748 if (event->triggerSession() == track->sessionId()) { 1749 (void) track->setSyncEvent(event); 1750 return NO_ERROR; 1751 } 1752 } 1753 1754 return NAME_NOT_FOUND; 1755} 1756 1757bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1758{ 1759 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1760} 1761 1762void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1763 const Vector< sp<Track> >& tracksToRemove) 1764{ 1765 size_t count = tracksToRemove.size(); 1766 if (count > 0) { 1767 for (size_t i = 0 ; i < count ; i++) { 1768 const sp<Track>& track = tracksToRemove.itemAt(i); 1769 if (!track->isOutputTrack()) { 1770 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1771#ifdef ADD_BATTERY_DATA 1772 // to track the speaker usage 1773 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1774#endif 1775 if (track->isTerminated()) { 1776 AudioSystem::releaseOutput(mId); 1777 } 1778 } 1779 } 1780 } 1781} 1782 1783void AudioFlinger::PlaybackThread::checkSilentMode_l() 1784{ 1785 if (!mMasterMute) { 1786 char value[PROPERTY_VALUE_MAX]; 1787 if (property_get("ro.audio.silent", value, "0") > 0) { 1788 char *endptr; 1789 unsigned long ul = strtoul(value, &endptr, 0); 1790 if (*endptr == '\0' && ul != 0) { 1791 ALOGD("Silence is golden"); 1792 // The setprop command will not allow a property to be changed after 1793 // the first time it is set, so we don't have to worry about un-muting. 1794 setMasterMute_l(true); 1795 } 1796 } 1797 } 1798} 1799 1800// shared by MIXER and DIRECT, overridden by DUPLICATING 1801ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1802{ 1803 // FIXME rewrite to reduce number of system calls 1804 mLastWriteTime = systemTime(); 1805 mInWrite = true; 1806 ssize_t bytesWritten; 1807 1808 // If an NBAIO sink is present, use it to write the normal mixer's submix 1809 if (mNormalSink != 0) { 1810#define mBitShift 2 // FIXME 1811 size_t count = mBytesRemaining >> mBitShift; 1812 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1813 ATRACE_BEGIN("write"); 1814 // update the setpoint when AudioFlinger::mScreenState changes 1815 uint32_t screenState = AudioFlinger::mScreenState; 1816 if (screenState != mScreenState) { 1817 mScreenState = screenState; 1818 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1819 if (pipe != NULL) { 1820 pipe->setAvgFrames((mScreenState & 1) ? 1821 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1822 } 1823 } 1824 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1825 ATRACE_END(); 1826 if (framesWritten > 0) { 1827 bytesWritten = framesWritten << mBitShift; 1828 } else { 1829 bytesWritten = framesWritten; 1830 } 1831 status_t status = INVALID_OPERATION; // mLatchD.mTimestamp is invalid 1832 if (status == NO_ERROR) { 1833 size_t totalFramesWritten = mNormalSink->framesWritten(); 1834 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1835 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1836 mLatchDValid = true; 1837 } 1838 } 1839 // otherwise use the HAL / AudioStreamOut directly 1840 } else { 1841 // Direct output and offload threads 1842 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1843 if (mUseAsyncWrite) { 1844 mWriteBlocked = true; 1845 ALOG_ASSERT(mCallbackThread != 0); 1846 mCallbackThread->setWriteBlocked(true); 1847 } 1848 bytesWritten = mOutput->stream->write(mOutput->stream, 1849 mMixBuffer + offset, mBytesRemaining); 1850 if (mUseAsyncWrite && 1851 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1852 // do not wait for async callback in case of error of full write 1853 mWriteBlocked = false; 1854 ALOG_ASSERT(mCallbackThread != 0); 1855 mCallbackThread->setWriteBlocked(false); 1856 } 1857 } 1858 1859 mNumWrites++; 1860 mInWrite = false; 1861 1862 return bytesWritten; 1863} 1864 1865void AudioFlinger::PlaybackThread::threadLoop_drain() 1866{ 1867 if (mOutput->stream->drain) { 1868 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1869 if (mUseAsyncWrite) { 1870 mDraining = true; 1871 ALOG_ASSERT(mCallbackThread != 0); 1872 mCallbackThread->setDraining(true); 1873 } 1874 mOutput->stream->drain(mOutput->stream, 1875 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1876 : AUDIO_DRAIN_ALL); 1877 } 1878} 1879 1880void AudioFlinger::PlaybackThread::threadLoop_exit() 1881{ 1882 // Default implementation has nothing to do 1883} 1884 1885/* 1886The derived values that are cached: 1887 - mixBufferSize from frame count * frame size 1888 - activeSleepTime from activeSleepTimeUs() 1889 - idleSleepTime from idleSleepTimeUs() 1890 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1891 - maxPeriod from frame count and sample rate (MIXER only) 1892 1893The parameters that affect these derived values are: 1894 - frame count 1895 - frame size 1896 - sample rate 1897 - device type: A2DP or not 1898 - device latency 1899 - format: PCM or not 1900 - active sleep time 1901 - idle sleep time 1902*/ 1903 1904void AudioFlinger::PlaybackThread::cacheParameters_l() 1905{ 1906 mixBufferSize = mNormalFrameCount * mFrameSize; 1907 activeSleepTime = activeSleepTimeUs(); 1908 idleSleepTime = idleSleepTimeUs(); 1909} 1910 1911void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1912{ 1913 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1914 this, streamType, mTracks.size()); 1915 Mutex::Autolock _l(mLock); 1916 1917 size_t size = mTracks.size(); 1918 for (size_t i = 0; i < size; i++) { 1919 sp<Track> t = mTracks[i]; 1920 if (t->streamType() == streamType) { 1921 t->invalidate(); 1922 } 1923 } 1924} 1925 1926status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1927{ 1928 int session = chain->sessionId(); 1929 int16_t *buffer = mMixBuffer; 1930 bool ownsBuffer = false; 1931 1932 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1933 if (session > 0) { 1934 // Only one effect chain can be present in direct output thread and it uses 1935 // the mix buffer as input 1936 if (mType != DIRECT) { 1937 size_t numSamples = mNormalFrameCount * mChannelCount; 1938 buffer = new int16_t[numSamples]; 1939 memset(buffer, 0, numSamples * sizeof(int16_t)); 1940 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1941 ownsBuffer = true; 1942 } 1943 1944 // Attach all tracks with same session ID to this chain. 1945 for (size_t i = 0; i < mTracks.size(); ++i) { 1946 sp<Track> track = mTracks[i]; 1947 if (session == track->sessionId()) { 1948 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1949 buffer); 1950 track->setMainBuffer(buffer); 1951 chain->incTrackCnt(); 1952 } 1953 } 1954 1955 // indicate all active tracks in the chain 1956 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1957 sp<Track> track = mActiveTracks[i].promote(); 1958 if (track == 0) { 1959 continue; 1960 } 1961 if (session == track->sessionId()) { 1962 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1963 chain->incActiveTrackCnt(); 1964 } 1965 } 1966 } 1967 1968 chain->setInBuffer(buffer, ownsBuffer); 1969 chain->setOutBuffer(mMixBuffer); 1970 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1971 // chains list in order to be processed last as it contains output stage effects 1972 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1973 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1974 // after track specific effects and before output stage 1975 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1976 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1977 // Effect chain for other sessions are inserted at beginning of effect 1978 // chains list to be processed before output mix effects. Relative order between other 1979 // sessions is not important 1980 size_t size = mEffectChains.size(); 1981 size_t i = 0; 1982 for (i = 0; i < size; i++) { 1983 if (mEffectChains[i]->sessionId() < session) { 1984 break; 1985 } 1986 } 1987 mEffectChains.insertAt(chain, i); 1988 checkSuspendOnAddEffectChain_l(chain); 1989 1990 return NO_ERROR; 1991} 1992 1993size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1994{ 1995 int session = chain->sessionId(); 1996 1997 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1998 1999 for (size_t i = 0; i < mEffectChains.size(); i++) { 2000 if (chain == mEffectChains[i]) { 2001 mEffectChains.removeAt(i); 2002 // detach all active tracks from the chain 2003 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2004 sp<Track> track = mActiveTracks[i].promote(); 2005 if (track == 0) { 2006 continue; 2007 } 2008 if (session == track->sessionId()) { 2009 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2010 chain.get(), session); 2011 chain->decActiveTrackCnt(); 2012 } 2013 } 2014 2015 // detach all tracks with same session ID from this chain 2016 for (size_t i = 0; i < mTracks.size(); ++i) { 2017 sp<Track> track = mTracks[i]; 2018 if (session == track->sessionId()) { 2019 track->setMainBuffer(mMixBuffer); 2020 chain->decTrackCnt(); 2021 } 2022 } 2023 break; 2024 } 2025 } 2026 return mEffectChains.size(); 2027} 2028 2029status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2030 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2031{ 2032 Mutex::Autolock _l(mLock); 2033 return attachAuxEffect_l(track, EffectId); 2034} 2035 2036status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2037 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2038{ 2039 status_t status = NO_ERROR; 2040 2041 if (EffectId == 0) { 2042 track->setAuxBuffer(0, NULL); 2043 } else { 2044 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2045 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2046 if (effect != 0) { 2047 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2048 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2049 } else { 2050 status = INVALID_OPERATION; 2051 } 2052 } else { 2053 status = BAD_VALUE; 2054 } 2055 } 2056 return status; 2057} 2058 2059void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2060{ 2061 for (size_t i = 0; i < mTracks.size(); ++i) { 2062 sp<Track> track = mTracks[i]; 2063 if (track->auxEffectId() == effectId) { 2064 attachAuxEffect_l(track, 0); 2065 } 2066 } 2067} 2068 2069bool AudioFlinger::PlaybackThread::threadLoop() 2070{ 2071 Vector< sp<Track> > tracksToRemove; 2072 2073 standbyTime = systemTime(); 2074 2075 // MIXER 2076 nsecs_t lastWarning = 0; 2077 2078 // DUPLICATING 2079 // FIXME could this be made local to while loop? 2080 writeFrames = 0; 2081 2082 cacheParameters_l(); 2083 sleepTime = idleSleepTime; 2084 2085 if (mType == MIXER) { 2086 sleepTimeShift = 0; 2087 } 2088 2089 CpuStats cpuStats; 2090 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2091 2092 acquireWakeLock(); 2093 2094 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2095 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2096 // and then that string will be logged at the next convenient opportunity. 2097 const char *logString = NULL; 2098 2099 while (!exitPending()) 2100 { 2101 cpuStats.sample(myName); 2102 2103 Vector< sp<EffectChain> > effectChains; 2104 2105 processConfigEvents(); 2106 2107 { // scope for mLock 2108 2109 Mutex::Autolock _l(mLock); 2110 2111 if (logString != NULL) { 2112 mNBLogWriter->logTimestamp(); 2113 mNBLogWriter->log(logString); 2114 logString = NULL; 2115 } 2116 2117 if (mLatchDValid) { 2118 mLatchQ = mLatchD; 2119 mLatchDValid = false; 2120 mLatchQValid = true; 2121 } 2122 2123 if (checkForNewParameters_l()) { 2124 cacheParameters_l(); 2125 } 2126 2127 saveOutputTracks(); 2128 2129 if (mSignalPending) { 2130 // A signal was raised while we were unlocked 2131 mSignalPending = false; 2132 } else if (waitingAsyncCallback_l()) { 2133 if (exitPending()) { 2134 break; 2135 } 2136 releaseWakeLock_l(); 2137 ALOGV("wait async completion"); 2138 mWaitWorkCV.wait(mLock); 2139 ALOGV("async completion/wake"); 2140 acquireWakeLock_l(); 2141 if (exitPending()) { 2142 break; 2143 } 2144 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2145 continue; 2146 } 2147 sleepTime = 0; 2148 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2149 isSuspended()) { 2150 // put audio hardware into standby after short delay 2151 if (shouldStandby_l()) { 2152 2153 threadLoop_standby(); 2154 2155 mStandby = true; 2156 } 2157 2158 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2159 // we're about to wait, flush the binder command buffer 2160 IPCThreadState::self()->flushCommands(); 2161 2162 clearOutputTracks(); 2163 2164 if (exitPending()) { 2165 break; 2166 } 2167 2168 releaseWakeLock_l(); 2169 // wait until we have something to do... 2170 ALOGV("%s going to sleep", myName.string()); 2171 mWaitWorkCV.wait(mLock); 2172 ALOGV("%s waking up", myName.string()); 2173 acquireWakeLock_l(); 2174 2175 mMixerStatus = MIXER_IDLE; 2176 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2177 mBytesWritten = 0; 2178 mBytesRemaining = 0; 2179 checkSilentMode_l(); 2180 2181 standbyTime = systemTime() + standbyDelay; 2182 sleepTime = idleSleepTime; 2183 if (mType == MIXER) { 2184 sleepTimeShift = 0; 2185 } 2186 2187 continue; 2188 } 2189 } 2190 2191 // mMixerStatusIgnoringFastTracks is also updated internally 2192 mMixerStatus = prepareTracks_l(&tracksToRemove); 2193 2194 // prevent any changes in effect chain list and in each effect chain 2195 // during mixing and effect process as the audio buffers could be deleted 2196 // or modified if an effect is created or deleted 2197 lockEffectChains_l(effectChains); 2198 } 2199 2200 if (mBytesRemaining == 0) { 2201 mCurrentWriteLength = 0; 2202 if (mMixerStatus == MIXER_TRACKS_READY) { 2203 // threadLoop_mix() sets mCurrentWriteLength 2204 threadLoop_mix(); 2205 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2206 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2207 // threadLoop_sleepTime sets sleepTime to 0 if data 2208 // must be written to HAL 2209 threadLoop_sleepTime(); 2210 if (sleepTime == 0) { 2211 mCurrentWriteLength = mixBufferSize; 2212 } 2213 } 2214 mBytesRemaining = mCurrentWriteLength; 2215 if (isSuspended()) { 2216 sleepTime = suspendSleepTimeUs(); 2217 // simulate write to HAL when suspended 2218 mBytesWritten += mixBufferSize; 2219 mBytesRemaining = 0; 2220 } 2221 2222 // only process effects if we're going to write 2223 if (sleepTime == 0) { 2224 for (size_t i = 0; i < effectChains.size(); i ++) { 2225 effectChains[i]->process_l(); 2226 } 2227 } 2228 } 2229 2230 // enable changes in effect chain 2231 unlockEffectChains(effectChains); 2232 2233 if (!waitingAsyncCallback()) { 2234 // sleepTime == 0 means we must write to audio hardware 2235 if (sleepTime == 0) { 2236 if (mBytesRemaining) { 2237 ssize_t ret = threadLoop_write(); 2238 if (ret < 0) { 2239 mBytesRemaining = 0; 2240 } else { 2241 mBytesWritten += ret; 2242 mBytesRemaining -= ret; 2243 } 2244 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2245 (mMixerStatus == MIXER_DRAIN_ALL)) { 2246 threadLoop_drain(); 2247 } 2248if (mType == MIXER) { 2249 // write blocked detection 2250 nsecs_t now = systemTime(); 2251 nsecs_t delta = now - mLastWriteTime; 2252 if (!mStandby && delta > maxPeriod) { 2253 mNumDelayedWrites++; 2254 if ((now - lastWarning) > kWarningThrottleNs) { 2255 ATRACE_NAME("underrun"); 2256 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2257 ns2ms(delta), mNumDelayedWrites, this); 2258 lastWarning = now; 2259 } 2260 } 2261} 2262 2263 mStandby = false; 2264 } else { 2265 usleep(sleepTime); 2266 } 2267 } 2268 2269 // Finally let go of removed track(s), without the lock held 2270 // since we can't guarantee the destructors won't acquire that 2271 // same lock. This will also mutate and push a new fast mixer state. 2272 threadLoop_removeTracks(tracksToRemove); 2273 tracksToRemove.clear(); 2274 2275 // FIXME I don't understand the need for this here; 2276 // it was in the original code but maybe the 2277 // assignment in saveOutputTracks() makes this unnecessary? 2278 clearOutputTracks(); 2279 2280 // Effect chains will be actually deleted here if they were removed from 2281 // mEffectChains list during mixing or effects processing 2282 effectChains.clear(); 2283 2284 // FIXME Note that the above .clear() is no longer necessary since effectChains 2285 // is now local to this block, but will keep it for now (at least until merge done). 2286 } 2287 2288 threadLoop_exit(); 2289 2290 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2291 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2292 // put output stream into standby mode 2293 if (!mStandby) { 2294 mOutput->stream->common.standby(&mOutput->stream->common); 2295 } 2296 } 2297 2298 releaseWakeLock(); 2299 2300 ALOGV("Thread %p type %d exiting", this, mType); 2301 return false; 2302} 2303 2304// removeTracks_l() must be called with ThreadBase::mLock held 2305void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2306{ 2307 size_t count = tracksToRemove.size(); 2308 if (count > 0) { 2309 for (size_t i=0 ; i<count ; i++) { 2310 const sp<Track>& track = tracksToRemove.itemAt(i); 2311 mActiveTracks.remove(track); 2312 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2313 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2314 if (chain != 0) { 2315 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2316 track->sessionId()); 2317 chain->decActiveTrackCnt(); 2318 } 2319 if (track->isTerminated()) { 2320 removeTrack_l(track); 2321 } 2322 } 2323 } 2324 2325} 2326 2327// ---------------------------------------------------------------------------- 2328 2329AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2330 audio_io_handle_t id, audio_devices_t device, type_t type) 2331 : PlaybackThread(audioFlinger, output, id, device, type), 2332 // mAudioMixer below 2333 // mFastMixer below 2334 mFastMixerFutex(0) 2335 // mOutputSink below 2336 // mPipeSink below 2337 // mNormalSink below 2338{ 2339 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2340 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2341 "mFrameCount=%d, mNormalFrameCount=%d", 2342 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2343 mNormalFrameCount); 2344 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2345 2346 // FIXME - Current mixer implementation only supports stereo output 2347 if (mChannelCount != FCC_2) { 2348 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2349 } 2350 2351 // create an NBAIO sink for the HAL output stream, and negotiate 2352 mOutputSink = new AudioStreamOutSink(output->stream); 2353 size_t numCounterOffers = 0; 2354 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2355 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2356 ALOG_ASSERT(index == 0); 2357 2358 // initialize fast mixer depending on configuration 2359 bool initFastMixer; 2360 switch (kUseFastMixer) { 2361 case FastMixer_Never: 2362 initFastMixer = false; 2363 break; 2364 case FastMixer_Always: 2365 initFastMixer = true; 2366 break; 2367 case FastMixer_Static: 2368 case FastMixer_Dynamic: 2369 initFastMixer = mFrameCount < mNormalFrameCount; 2370 break; 2371 } 2372 if (initFastMixer) { 2373 2374 // create a MonoPipe to connect our submix to FastMixer 2375 NBAIO_Format format = mOutputSink->format(); 2376 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2377 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2378 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2379 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2380 const NBAIO_Format offers[1] = {format}; 2381 size_t numCounterOffers = 0; 2382 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2383 ALOG_ASSERT(index == 0); 2384 monoPipe->setAvgFrames((mScreenState & 1) ? 2385 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2386 mPipeSink = monoPipe; 2387 2388#ifdef TEE_SINK 2389 if (mTeeSinkOutputEnabled) { 2390 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2391 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2392 numCounterOffers = 0; 2393 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2394 ALOG_ASSERT(index == 0); 2395 mTeeSink = teeSink; 2396 PipeReader *teeSource = new PipeReader(*teeSink); 2397 numCounterOffers = 0; 2398 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2399 ALOG_ASSERT(index == 0); 2400 mTeeSource = teeSource; 2401 } 2402#endif 2403 2404 // create fast mixer and configure it initially with just one fast track for our submix 2405 mFastMixer = new FastMixer(); 2406 FastMixerStateQueue *sq = mFastMixer->sq(); 2407#ifdef STATE_QUEUE_DUMP 2408 sq->setObserverDump(&mStateQueueObserverDump); 2409 sq->setMutatorDump(&mStateQueueMutatorDump); 2410#endif 2411 FastMixerState *state = sq->begin(); 2412 FastTrack *fastTrack = &state->mFastTracks[0]; 2413 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2414 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2415 fastTrack->mVolumeProvider = NULL; 2416 fastTrack->mGeneration++; 2417 state->mFastTracksGen++; 2418 state->mTrackMask = 1; 2419 // fast mixer will use the HAL output sink 2420 state->mOutputSink = mOutputSink.get(); 2421 state->mOutputSinkGen++; 2422 state->mFrameCount = mFrameCount; 2423 state->mCommand = FastMixerState::COLD_IDLE; 2424 // already done in constructor initialization list 2425 //mFastMixerFutex = 0; 2426 state->mColdFutexAddr = &mFastMixerFutex; 2427 state->mColdGen++; 2428 state->mDumpState = &mFastMixerDumpState; 2429#ifdef TEE_SINK 2430 state->mTeeSink = mTeeSink.get(); 2431#endif 2432 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2433 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2434 sq->end(); 2435 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2436 2437 // start the fast mixer 2438 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2439 pid_t tid = mFastMixer->getTid(); 2440 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2441 if (err != 0) { 2442 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2443 kPriorityFastMixer, getpid_cached, tid, err); 2444 } 2445 2446#ifdef AUDIO_WATCHDOG 2447 // create and start the watchdog 2448 mAudioWatchdog = new AudioWatchdog(); 2449 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2450 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2451 tid = mAudioWatchdog->getTid(); 2452 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2453 if (err != 0) { 2454 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2455 kPriorityFastMixer, getpid_cached, tid, err); 2456 } 2457#endif 2458 2459 } else { 2460 mFastMixer = NULL; 2461 } 2462 2463 switch (kUseFastMixer) { 2464 case FastMixer_Never: 2465 case FastMixer_Dynamic: 2466 mNormalSink = mOutputSink; 2467 break; 2468 case FastMixer_Always: 2469 mNormalSink = mPipeSink; 2470 break; 2471 case FastMixer_Static: 2472 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2473 break; 2474 } 2475} 2476 2477AudioFlinger::MixerThread::~MixerThread() 2478{ 2479 if (mFastMixer != NULL) { 2480 FastMixerStateQueue *sq = mFastMixer->sq(); 2481 FastMixerState *state = sq->begin(); 2482 if (state->mCommand == FastMixerState::COLD_IDLE) { 2483 int32_t old = android_atomic_inc(&mFastMixerFutex); 2484 if (old == -1) { 2485 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2486 } 2487 } 2488 state->mCommand = FastMixerState::EXIT; 2489 sq->end(); 2490 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2491 mFastMixer->join(); 2492 // Though the fast mixer thread has exited, it's state queue is still valid. 2493 // We'll use that extract the final state which contains one remaining fast track 2494 // corresponding to our sub-mix. 2495 state = sq->begin(); 2496 ALOG_ASSERT(state->mTrackMask == 1); 2497 FastTrack *fastTrack = &state->mFastTracks[0]; 2498 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2499 delete fastTrack->mBufferProvider; 2500 sq->end(false /*didModify*/); 2501 delete mFastMixer; 2502#ifdef AUDIO_WATCHDOG 2503 if (mAudioWatchdog != 0) { 2504 mAudioWatchdog->requestExit(); 2505 mAudioWatchdog->requestExitAndWait(); 2506 mAudioWatchdog.clear(); 2507 } 2508#endif 2509 } 2510 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2511 delete mAudioMixer; 2512} 2513 2514 2515uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2516{ 2517 if (mFastMixer != NULL) { 2518 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2519 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2520 } 2521 return latency; 2522} 2523 2524 2525void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2526{ 2527 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2528} 2529 2530ssize_t AudioFlinger::MixerThread::threadLoop_write() 2531{ 2532 // FIXME we should only do one push per cycle; confirm this is true 2533 // Start the fast mixer if it's not already running 2534 if (mFastMixer != NULL) { 2535 FastMixerStateQueue *sq = mFastMixer->sq(); 2536 FastMixerState *state = sq->begin(); 2537 if (state->mCommand != FastMixerState::MIX_WRITE && 2538 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2539 if (state->mCommand == FastMixerState::COLD_IDLE) { 2540 int32_t old = android_atomic_inc(&mFastMixerFutex); 2541 if (old == -1) { 2542 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2543 } 2544#ifdef AUDIO_WATCHDOG 2545 if (mAudioWatchdog != 0) { 2546 mAudioWatchdog->resume(); 2547 } 2548#endif 2549 } 2550 state->mCommand = FastMixerState::MIX_WRITE; 2551 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2552 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2553 sq->end(); 2554 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2555 if (kUseFastMixer == FastMixer_Dynamic) { 2556 mNormalSink = mPipeSink; 2557 } 2558 } else { 2559 sq->end(false /*didModify*/); 2560 } 2561 } 2562 return PlaybackThread::threadLoop_write(); 2563} 2564 2565void AudioFlinger::MixerThread::threadLoop_standby() 2566{ 2567 // Idle the fast mixer if it's currently running 2568 if (mFastMixer != NULL) { 2569 FastMixerStateQueue *sq = mFastMixer->sq(); 2570 FastMixerState *state = sq->begin(); 2571 if (!(state->mCommand & FastMixerState::IDLE)) { 2572 state->mCommand = FastMixerState::COLD_IDLE; 2573 state->mColdFutexAddr = &mFastMixerFutex; 2574 state->mColdGen++; 2575 mFastMixerFutex = 0; 2576 sq->end(); 2577 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2578 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2579 if (kUseFastMixer == FastMixer_Dynamic) { 2580 mNormalSink = mOutputSink; 2581 } 2582#ifdef AUDIO_WATCHDOG 2583 if (mAudioWatchdog != 0) { 2584 mAudioWatchdog->pause(); 2585 } 2586#endif 2587 } else { 2588 sq->end(false /*didModify*/); 2589 } 2590 } 2591 PlaybackThread::threadLoop_standby(); 2592} 2593 2594// Empty implementation for standard mixer 2595// Overridden for offloaded playback 2596void AudioFlinger::PlaybackThread::flushOutput_l() 2597{ 2598} 2599 2600bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2601{ 2602 return false; 2603} 2604 2605bool AudioFlinger::PlaybackThread::shouldStandby_l() 2606{ 2607 return !mStandby; 2608} 2609 2610bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2611{ 2612 Mutex::Autolock _l(mLock); 2613 return waitingAsyncCallback_l(); 2614} 2615 2616// shared by MIXER and DIRECT, overridden by DUPLICATING 2617void AudioFlinger::PlaybackThread::threadLoop_standby() 2618{ 2619 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2620 mOutput->stream->common.standby(&mOutput->stream->common); 2621 if (mUseAsyncWrite != 0) { 2622 mWriteBlocked = false; 2623 mDraining = false; 2624 ALOG_ASSERT(mCallbackThread != 0); 2625 mCallbackThread->setWriteBlocked(false); 2626 mCallbackThread->setDraining(false); 2627 } 2628} 2629 2630void AudioFlinger::MixerThread::threadLoop_mix() 2631{ 2632 // obtain the presentation timestamp of the next output buffer 2633 int64_t pts; 2634 status_t status = INVALID_OPERATION; 2635 2636 if (mNormalSink != 0) { 2637 status = mNormalSink->getNextWriteTimestamp(&pts); 2638 } else { 2639 status = mOutputSink->getNextWriteTimestamp(&pts); 2640 } 2641 2642 if (status != NO_ERROR) { 2643 pts = AudioBufferProvider::kInvalidPTS; 2644 } 2645 2646 // mix buffers... 2647 mAudioMixer->process(pts); 2648 mCurrentWriteLength = mixBufferSize; 2649 // increase sleep time progressively when application underrun condition clears. 2650 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2651 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2652 // such that we would underrun the audio HAL. 2653 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2654 sleepTimeShift--; 2655 } 2656 sleepTime = 0; 2657 standbyTime = systemTime() + standbyDelay; 2658 //TODO: delay standby when effects have a tail 2659} 2660 2661void AudioFlinger::MixerThread::threadLoop_sleepTime() 2662{ 2663 // If no tracks are ready, sleep once for the duration of an output 2664 // buffer size, then write 0s to the output 2665 if (sleepTime == 0) { 2666 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2667 sleepTime = activeSleepTime >> sleepTimeShift; 2668 if (sleepTime < kMinThreadSleepTimeUs) { 2669 sleepTime = kMinThreadSleepTimeUs; 2670 } 2671 // reduce sleep time in case of consecutive application underruns to avoid 2672 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2673 // duration we would end up writing less data than needed by the audio HAL if 2674 // the condition persists. 2675 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2676 sleepTimeShift++; 2677 } 2678 } else { 2679 sleepTime = idleSleepTime; 2680 } 2681 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2682 memset(mMixBuffer, 0, mixBufferSize); 2683 sleepTime = 0; 2684 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2685 "anticipated start"); 2686 } 2687 // TODO add standby time extension fct of effect tail 2688} 2689 2690// prepareTracks_l() must be called with ThreadBase::mLock held 2691AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2692 Vector< sp<Track> > *tracksToRemove) 2693{ 2694 2695 mixer_state mixerStatus = MIXER_IDLE; 2696 // find out which tracks need to be processed 2697 size_t count = mActiveTracks.size(); 2698 size_t mixedTracks = 0; 2699 size_t tracksWithEffect = 0; 2700 // counts only _active_ fast tracks 2701 size_t fastTracks = 0; 2702 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2703 2704 float masterVolume = mMasterVolume; 2705 bool masterMute = mMasterMute; 2706 2707 if (masterMute) { 2708 masterVolume = 0; 2709 } 2710 // Delegate master volume control to effect in output mix effect chain if needed 2711 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2712 if (chain != 0) { 2713 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2714 chain->setVolume_l(&v, &v); 2715 masterVolume = (float)((v + (1 << 23)) >> 24); 2716 chain.clear(); 2717 } 2718 2719 // prepare a new state to push 2720 FastMixerStateQueue *sq = NULL; 2721 FastMixerState *state = NULL; 2722 bool didModify = false; 2723 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2724 if (mFastMixer != NULL) { 2725 sq = mFastMixer->sq(); 2726 state = sq->begin(); 2727 } 2728 2729 for (size_t i=0 ; i<count ; i++) { 2730 const sp<Track> t = mActiveTracks[i].promote(); 2731 if (t == 0) { 2732 continue; 2733 } 2734 2735 // this const just means the local variable doesn't change 2736 Track* const track = t.get(); 2737 2738 // process fast tracks 2739 if (track->isFastTrack()) { 2740 2741 // It's theoretically possible (though unlikely) for a fast track to be created 2742 // and then removed within the same normal mix cycle. This is not a problem, as 2743 // the track never becomes active so it's fast mixer slot is never touched. 2744 // The converse, of removing an (active) track and then creating a new track 2745 // at the identical fast mixer slot within the same normal mix cycle, 2746 // is impossible because the slot isn't marked available until the end of each cycle. 2747 int j = track->mFastIndex; 2748 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2749 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2750 FastTrack *fastTrack = &state->mFastTracks[j]; 2751 2752 // Determine whether the track is currently in underrun condition, 2753 // and whether it had a recent underrun. 2754 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2755 FastTrackUnderruns underruns = ftDump->mUnderruns; 2756 uint32_t recentFull = (underruns.mBitFields.mFull - 2757 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2758 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2759 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2760 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2761 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2762 uint32_t recentUnderruns = recentPartial + recentEmpty; 2763 track->mObservedUnderruns = underruns; 2764 // don't count underruns that occur while stopping or pausing 2765 // or stopped which can occur when flush() is called while active 2766 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2767 recentUnderruns > 0) { 2768 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2769 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2770 } 2771 2772 // This is similar to the state machine for normal tracks, 2773 // with a few modifications for fast tracks. 2774 bool isActive = true; 2775 switch (track->mState) { 2776 case TrackBase::STOPPING_1: 2777 // track stays active in STOPPING_1 state until first underrun 2778 if (recentUnderruns > 0 || track->isTerminated()) { 2779 track->mState = TrackBase::STOPPING_2; 2780 } 2781 break; 2782 case TrackBase::PAUSING: 2783 // ramp down is not yet implemented 2784 track->setPaused(); 2785 break; 2786 case TrackBase::RESUMING: 2787 // ramp up is not yet implemented 2788 track->mState = TrackBase::ACTIVE; 2789 break; 2790 case TrackBase::ACTIVE: 2791 if (recentFull > 0 || recentPartial > 0) { 2792 // track has provided at least some frames recently: reset retry count 2793 track->mRetryCount = kMaxTrackRetries; 2794 } 2795 if (recentUnderruns == 0) { 2796 // no recent underruns: stay active 2797 break; 2798 } 2799 // there has recently been an underrun of some kind 2800 if (track->sharedBuffer() == 0) { 2801 // were any of the recent underruns "empty" (no frames available)? 2802 if (recentEmpty == 0) { 2803 // no, then ignore the partial underruns as they are allowed indefinitely 2804 break; 2805 } 2806 // there has recently been an "empty" underrun: decrement the retry counter 2807 if (--(track->mRetryCount) > 0) { 2808 break; 2809 } 2810 // indicate to client process that the track was disabled because of underrun; 2811 // it will then automatically call start() when data is available 2812 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2813 // remove from active list, but state remains ACTIVE [confusing but true] 2814 isActive = false; 2815 break; 2816 } 2817 // fall through 2818 case TrackBase::STOPPING_2: 2819 case TrackBase::PAUSED: 2820 case TrackBase::STOPPED: 2821 case TrackBase::FLUSHED: // flush() while active 2822 // Check for presentation complete if track is inactive 2823 // We have consumed all the buffers of this track. 2824 // This would be incomplete if we auto-paused on underrun 2825 { 2826 size_t audioHALFrames = 2827 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2828 size_t framesWritten = mBytesWritten / mFrameSize; 2829 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2830 // track stays in active list until presentation is complete 2831 break; 2832 } 2833 } 2834 if (track->isStopping_2()) { 2835 track->mState = TrackBase::STOPPED; 2836 } 2837 if (track->isStopped()) { 2838 // Can't reset directly, as fast mixer is still polling this track 2839 // track->reset(); 2840 // So instead mark this track as needing to be reset after push with ack 2841 resetMask |= 1 << i; 2842 } 2843 isActive = false; 2844 break; 2845 case TrackBase::IDLE: 2846 default: 2847 LOG_FATAL("unexpected track state %d", track->mState); 2848 } 2849 2850 if (isActive) { 2851 // was it previously inactive? 2852 if (!(state->mTrackMask & (1 << j))) { 2853 ExtendedAudioBufferProvider *eabp = track; 2854 VolumeProvider *vp = track; 2855 fastTrack->mBufferProvider = eabp; 2856 fastTrack->mVolumeProvider = vp; 2857 fastTrack->mSampleRate = track->mSampleRate; 2858 fastTrack->mChannelMask = track->mChannelMask; 2859 fastTrack->mGeneration++; 2860 state->mTrackMask |= 1 << j; 2861 didModify = true; 2862 // no acknowledgement required for newly active tracks 2863 } 2864 // cache the combined master volume and stream type volume for fast mixer; this 2865 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2866 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2867 ++fastTracks; 2868 } else { 2869 // was it previously active? 2870 if (state->mTrackMask & (1 << j)) { 2871 fastTrack->mBufferProvider = NULL; 2872 fastTrack->mGeneration++; 2873 state->mTrackMask &= ~(1 << j); 2874 didModify = true; 2875 // If any fast tracks were removed, we must wait for acknowledgement 2876 // because we're about to decrement the last sp<> on those tracks. 2877 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2878 } else { 2879 LOG_FATAL("fast track %d should have been active", j); 2880 } 2881 tracksToRemove->add(track); 2882 // Avoids a misleading display in dumpsys 2883 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2884 } 2885 continue; 2886 } 2887 2888 { // local variable scope to avoid goto warning 2889 2890 audio_track_cblk_t* cblk = track->cblk(); 2891 2892 // The first time a track is added we wait 2893 // for all its buffers to be filled before processing it 2894 int name = track->name(); 2895 // make sure that we have enough frames to mix one full buffer. 2896 // enforce this condition only once to enable draining the buffer in case the client 2897 // app does not call stop() and relies on underrun to stop: 2898 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2899 // during last round 2900 size_t desiredFrames; 2901 uint32_t sr = track->sampleRate(); 2902 if (sr == mSampleRate) { 2903 desiredFrames = mNormalFrameCount; 2904 } else { 2905 // +1 for rounding and +1 for additional sample needed for interpolation 2906 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2907 // add frames already consumed but not yet released by the resampler 2908 // because mAudioTrackServerProxy->framesReady() will include these frames 2909 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2910 // the minimum track buffer size is normally twice the number of frames necessary 2911 // to fill one buffer and the resampler should not leave more than one buffer worth 2912 // of unreleased frames after each pass, but just in case... 2913 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2914 } 2915 uint32_t minFrames = 1; 2916 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2917 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2918 minFrames = desiredFrames; 2919 } 2920 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2921 size_t framesReady; 2922 if (track->sharedBuffer() == 0) { 2923 framesReady = track->framesReady(); 2924 } else if (track->isStopped()) { 2925 framesReady = 0; 2926 } else { 2927 framesReady = 1; 2928 } 2929 if ((framesReady >= minFrames) && track->isReady() && 2930 !track->isPaused() && !track->isTerminated()) 2931 { 2932 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2933 2934 mixedTracks++; 2935 2936 // track->mainBuffer() != mMixBuffer means there is an effect chain 2937 // connected to the track 2938 chain.clear(); 2939 if (track->mainBuffer() != mMixBuffer) { 2940 chain = getEffectChain_l(track->sessionId()); 2941 // Delegate volume control to effect in track effect chain if needed 2942 if (chain != 0) { 2943 tracksWithEffect++; 2944 } else { 2945 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2946 "session %d", 2947 name, track->sessionId()); 2948 } 2949 } 2950 2951 2952 int param = AudioMixer::VOLUME; 2953 if (track->mFillingUpStatus == Track::FS_FILLED) { 2954 // no ramp for the first volume setting 2955 track->mFillingUpStatus = Track::FS_ACTIVE; 2956 if (track->mState == TrackBase::RESUMING) { 2957 track->mState = TrackBase::ACTIVE; 2958 param = AudioMixer::RAMP_VOLUME; 2959 } 2960 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2961 // FIXME should not make a decision based on mServer 2962 } else if (cblk->mServer != 0) { 2963 // If the track is stopped before the first frame was mixed, 2964 // do not apply ramp 2965 param = AudioMixer::RAMP_VOLUME; 2966 } 2967 2968 // compute volume for this track 2969 uint32_t vl, vr, va; 2970 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2971 vl = vr = va = 0; 2972 if (track->isPausing()) { 2973 track->setPaused(); 2974 } 2975 } else { 2976 2977 // read original volumes with volume control 2978 float typeVolume = mStreamTypes[track->streamType()].volume; 2979 float v = masterVolume * typeVolume; 2980 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2981 uint32_t vlr = proxy->getVolumeLR(); 2982 vl = vlr & 0xFFFF; 2983 vr = vlr >> 16; 2984 // track volumes come from shared memory, so can't be trusted and must be clamped 2985 if (vl > MAX_GAIN_INT) { 2986 ALOGV("Track left volume out of range: %04X", vl); 2987 vl = MAX_GAIN_INT; 2988 } 2989 if (vr > MAX_GAIN_INT) { 2990 ALOGV("Track right volume out of range: %04X", vr); 2991 vr = MAX_GAIN_INT; 2992 } 2993 // now apply the master volume and stream type volume 2994 vl = (uint32_t)(v * vl) << 12; 2995 vr = (uint32_t)(v * vr) << 12; 2996 // assuming master volume and stream type volume each go up to 1.0, 2997 // vl and vr are now in 8.24 format 2998 2999 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3000 // send level comes from shared memory and so may be corrupt 3001 if (sendLevel > MAX_GAIN_INT) { 3002 ALOGV("Track send level out of range: %04X", sendLevel); 3003 sendLevel = MAX_GAIN_INT; 3004 } 3005 va = (uint32_t)(v * sendLevel); 3006 } 3007 3008 // Delegate volume control to effect in track effect chain if needed 3009 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3010 // Do not ramp volume if volume is controlled by effect 3011 param = AudioMixer::VOLUME; 3012 track->mHasVolumeController = true; 3013 } else { 3014 // force no volume ramp when volume controller was just disabled or removed 3015 // from effect chain to avoid volume spike 3016 if (track->mHasVolumeController) { 3017 param = AudioMixer::VOLUME; 3018 } 3019 track->mHasVolumeController = false; 3020 } 3021 3022 // Convert volumes from 8.24 to 4.12 format 3023 // This additional clamping is needed in case chain->setVolume_l() overshot 3024 vl = (vl + (1 << 11)) >> 12; 3025 if (vl > MAX_GAIN_INT) { 3026 vl = MAX_GAIN_INT; 3027 } 3028 vr = (vr + (1 << 11)) >> 12; 3029 if (vr > MAX_GAIN_INT) { 3030 vr = MAX_GAIN_INT; 3031 } 3032 3033 if (va > MAX_GAIN_INT) { 3034 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3035 } 3036 3037 // XXX: these things DON'T need to be done each time 3038 mAudioMixer->setBufferProvider(name, track); 3039 mAudioMixer->enable(name); 3040 3041 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3042 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3043 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3044 mAudioMixer->setParameter( 3045 name, 3046 AudioMixer::TRACK, 3047 AudioMixer::FORMAT, (void *)track->format()); 3048 mAudioMixer->setParameter( 3049 name, 3050 AudioMixer::TRACK, 3051 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3052 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3053 uint32_t maxSampleRate = mSampleRate * 2; 3054 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3055 if (reqSampleRate == 0) { 3056 reqSampleRate = mSampleRate; 3057 } else if (reqSampleRate > maxSampleRate) { 3058 reqSampleRate = maxSampleRate; 3059 } 3060 mAudioMixer->setParameter( 3061 name, 3062 AudioMixer::RESAMPLE, 3063 AudioMixer::SAMPLE_RATE, 3064 (void *)reqSampleRate); 3065 mAudioMixer->setParameter( 3066 name, 3067 AudioMixer::TRACK, 3068 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3069 mAudioMixer->setParameter( 3070 name, 3071 AudioMixer::TRACK, 3072 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3073 3074 // reset retry count 3075 track->mRetryCount = kMaxTrackRetries; 3076 3077 // If one track is ready, set the mixer ready if: 3078 // - the mixer was not ready during previous round OR 3079 // - no other track is not ready 3080 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3081 mixerStatus != MIXER_TRACKS_ENABLED) { 3082 mixerStatus = MIXER_TRACKS_READY; 3083 } 3084 } else { 3085 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3086 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3087 } 3088 // clear effect chain input buffer if an active track underruns to avoid sending 3089 // previous audio buffer again to effects 3090 chain = getEffectChain_l(track->sessionId()); 3091 if (chain != 0) { 3092 chain->clearInputBuffer(); 3093 } 3094 3095 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3096 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3097 track->isStopped() || track->isPaused()) { 3098 // We have consumed all the buffers of this track. 3099 // Remove it from the list of active tracks. 3100 // TODO: use actual buffer filling status instead of latency when available from 3101 // audio HAL 3102 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3103 size_t framesWritten = mBytesWritten / mFrameSize; 3104 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3105 if (track->isStopped()) { 3106 track->reset(); 3107 } 3108 tracksToRemove->add(track); 3109 } 3110 } else { 3111 // No buffers for this track. Give it a few chances to 3112 // fill a buffer, then remove it from active list. 3113 if (--(track->mRetryCount) <= 0) { 3114 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3115 tracksToRemove->add(track); 3116 // indicate to client process that the track was disabled because of underrun; 3117 // it will then automatically call start() when data is available 3118 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3119 // If one track is not ready, mark the mixer also not ready if: 3120 // - the mixer was ready during previous round OR 3121 // - no other track is ready 3122 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3123 mixerStatus != MIXER_TRACKS_READY) { 3124 mixerStatus = MIXER_TRACKS_ENABLED; 3125 } 3126 } 3127 mAudioMixer->disable(name); 3128 } 3129 3130 } // local variable scope to avoid goto warning 3131track_is_ready: ; 3132 3133 } 3134 3135 // Push the new FastMixer state if necessary 3136 bool pauseAudioWatchdog = false; 3137 if (didModify) { 3138 state->mFastTracksGen++; 3139 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3140 if (kUseFastMixer == FastMixer_Dynamic && 3141 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3142 state->mCommand = FastMixerState::COLD_IDLE; 3143 state->mColdFutexAddr = &mFastMixerFutex; 3144 state->mColdGen++; 3145 mFastMixerFutex = 0; 3146 if (kUseFastMixer == FastMixer_Dynamic) { 3147 mNormalSink = mOutputSink; 3148 } 3149 // If we go into cold idle, need to wait for acknowledgement 3150 // so that fast mixer stops doing I/O. 3151 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3152 pauseAudioWatchdog = true; 3153 } 3154 } 3155 if (sq != NULL) { 3156 sq->end(didModify); 3157 sq->push(block); 3158 } 3159#ifdef AUDIO_WATCHDOG 3160 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3161 mAudioWatchdog->pause(); 3162 } 3163#endif 3164 3165 // Now perform the deferred reset on fast tracks that have stopped 3166 while (resetMask != 0) { 3167 size_t i = __builtin_ctz(resetMask); 3168 ALOG_ASSERT(i < count); 3169 resetMask &= ~(1 << i); 3170 sp<Track> t = mActiveTracks[i].promote(); 3171 if (t == 0) { 3172 continue; 3173 } 3174 Track* track = t.get(); 3175 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3176 track->reset(); 3177 } 3178 3179 // remove all the tracks that need to be... 3180 removeTracks_l(*tracksToRemove); 3181 3182 // mix buffer must be cleared if all tracks are connected to an 3183 // effect chain as in this case the mixer will not write to 3184 // mix buffer and track effects will accumulate into it 3185 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3186 (mixedTracks == 0 && fastTracks > 0))) { 3187 // FIXME as a performance optimization, should remember previous zero status 3188 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3189 } 3190 3191 // if any fast tracks, then status is ready 3192 mMixerStatusIgnoringFastTracks = mixerStatus; 3193 if (fastTracks > 0) { 3194 mixerStatus = MIXER_TRACKS_READY; 3195 } 3196 return mixerStatus; 3197} 3198 3199// getTrackName_l() must be called with ThreadBase::mLock held 3200int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3201{ 3202 return mAudioMixer->getTrackName(channelMask, sessionId); 3203} 3204 3205// deleteTrackName_l() must be called with ThreadBase::mLock held 3206void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3207{ 3208 ALOGV("remove track (%d) and delete from mixer", name); 3209 mAudioMixer->deleteTrackName(name); 3210} 3211 3212// checkForNewParameters_l() must be called with ThreadBase::mLock held 3213bool AudioFlinger::MixerThread::checkForNewParameters_l() 3214{ 3215 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3216 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3217 bool reconfig = false; 3218 3219 while (!mNewParameters.isEmpty()) { 3220 3221 if (mFastMixer != NULL) { 3222 FastMixerStateQueue *sq = mFastMixer->sq(); 3223 FastMixerState *state = sq->begin(); 3224 if (!(state->mCommand & FastMixerState::IDLE)) { 3225 previousCommand = state->mCommand; 3226 state->mCommand = FastMixerState::HOT_IDLE; 3227 sq->end(); 3228 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3229 } else { 3230 sq->end(false /*didModify*/); 3231 } 3232 } 3233 3234 status_t status = NO_ERROR; 3235 String8 keyValuePair = mNewParameters[0]; 3236 AudioParameter param = AudioParameter(keyValuePair); 3237 int value; 3238 3239 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3240 reconfig = true; 3241 } 3242 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3243 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3244 status = BAD_VALUE; 3245 } else { 3246 // no need to save value, since it's constant 3247 reconfig = true; 3248 } 3249 } 3250 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3251 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3252 status = BAD_VALUE; 3253 } else { 3254 // no need to save value, since it's constant 3255 reconfig = true; 3256 } 3257 } 3258 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3259 // do not accept frame count changes if tracks are open as the track buffer 3260 // size depends on frame count and correct behavior would not be guaranteed 3261 // if frame count is changed after track creation 3262 if (!mTracks.isEmpty()) { 3263 status = INVALID_OPERATION; 3264 } else { 3265 reconfig = true; 3266 } 3267 } 3268 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3269#ifdef ADD_BATTERY_DATA 3270 // when changing the audio output device, call addBatteryData to notify 3271 // the change 3272 if (mOutDevice != value) { 3273 uint32_t params = 0; 3274 // check whether speaker is on 3275 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3276 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3277 } 3278 3279 audio_devices_t deviceWithoutSpeaker 3280 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3281 // check if any other device (except speaker) is on 3282 if (value & deviceWithoutSpeaker ) { 3283 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3284 } 3285 3286 if (params != 0) { 3287 addBatteryData(params); 3288 } 3289 } 3290#endif 3291 3292 // forward device change to effects that have requested to be 3293 // aware of attached audio device. 3294 if (value != AUDIO_DEVICE_NONE) { 3295 mOutDevice = value; 3296 for (size_t i = 0; i < mEffectChains.size(); i++) { 3297 mEffectChains[i]->setDevice_l(mOutDevice); 3298 } 3299 } 3300 } 3301 3302 if (status == NO_ERROR) { 3303 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3304 keyValuePair.string()); 3305 if (!mStandby && status == INVALID_OPERATION) { 3306 mOutput->stream->common.standby(&mOutput->stream->common); 3307 mStandby = true; 3308 mBytesWritten = 0; 3309 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3310 keyValuePair.string()); 3311 } 3312 if (status == NO_ERROR && reconfig) { 3313 readOutputParameters(); 3314 delete mAudioMixer; 3315 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3316 for (size_t i = 0; i < mTracks.size() ; i++) { 3317 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3318 if (name < 0) { 3319 break; 3320 } 3321 mTracks[i]->mName = name; 3322 } 3323 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3324 } 3325 } 3326 3327 mNewParameters.removeAt(0); 3328 3329 mParamStatus = status; 3330 mParamCond.signal(); 3331 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3332 // already timed out waiting for the status and will never signal the condition. 3333 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3334 } 3335 3336 if (!(previousCommand & FastMixerState::IDLE)) { 3337 ALOG_ASSERT(mFastMixer != NULL); 3338 FastMixerStateQueue *sq = mFastMixer->sq(); 3339 FastMixerState *state = sq->begin(); 3340 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3341 state->mCommand = previousCommand; 3342 sq->end(); 3343 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3344 } 3345 3346 return reconfig; 3347} 3348 3349 3350void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3351{ 3352 const size_t SIZE = 256; 3353 char buffer[SIZE]; 3354 String8 result; 3355 3356 PlaybackThread::dumpInternals(fd, args); 3357 3358 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3359 result.append(buffer); 3360 write(fd, result.string(), result.size()); 3361 3362 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3363 const FastMixerDumpState copy(mFastMixerDumpState); 3364 copy.dump(fd); 3365 3366#ifdef STATE_QUEUE_DUMP 3367 // Similar for state queue 3368 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3369 observerCopy.dump(fd); 3370 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3371 mutatorCopy.dump(fd); 3372#endif 3373 3374#ifdef TEE_SINK 3375 // Write the tee output to a .wav file 3376 dumpTee(fd, mTeeSource, mId); 3377#endif 3378 3379#ifdef AUDIO_WATCHDOG 3380 if (mAudioWatchdog != 0) { 3381 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3382 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3383 wdCopy.dump(fd); 3384 } 3385#endif 3386} 3387 3388uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3389{ 3390 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3391} 3392 3393uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3394{ 3395 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3396} 3397 3398void AudioFlinger::MixerThread::cacheParameters_l() 3399{ 3400 PlaybackThread::cacheParameters_l(); 3401 3402 // FIXME: Relaxed timing because of a certain device that can't meet latency 3403 // Should be reduced to 2x after the vendor fixes the driver issue 3404 // increase threshold again due to low power audio mode. The way this warning 3405 // threshold is calculated and its usefulness should be reconsidered anyway. 3406 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3407} 3408 3409// ---------------------------------------------------------------------------- 3410 3411AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3412 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3413 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3414 // mLeftVolFloat, mRightVolFloat 3415{ 3416} 3417 3418AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3419 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3420 ThreadBase::type_t type) 3421 : PlaybackThread(audioFlinger, output, id, device, type) 3422 // mLeftVolFloat, mRightVolFloat 3423{ 3424} 3425 3426AudioFlinger::DirectOutputThread::~DirectOutputThread() 3427{ 3428} 3429 3430void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3431{ 3432 audio_track_cblk_t* cblk = track->cblk(); 3433 float left, right; 3434 3435 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3436 left = right = 0; 3437 } else { 3438 float typeVolume = mStreamTypes[track->streamType()].volume; 3439 float v = mMasterVolume * typeVolume; 3440 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3441 uint32_t vlr = proxy->getVolumeLR(); 3442 float v_clamped = v * (vlr & 0xFFFF); 3443 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3444 left = v_clamped/MAX_GAIN; 3445 v_clamped = v * (vlr >> 16); 3446 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3447 right = v_clamped/MAX_GAIN; 3448 } 3449 3450 if (lastTrack) { 3451 if (left != mLeftVolFloat || right != mRightVolFloat) { 3452 mLeftVolFloat = left; 3453 mRightVolFloat = right; 3454 3455 // Convert volumes from float to 8.24 3456 uint32_t vl = (uint32_t)(left * (1 << 24)); 3457 uint32_t vr = (uint32_t)(right * (1 << 24)); 3458 3459 // Delegate volume control to effect in track effect chain if needed 3460 // only one effect chain can be present on DirectOutputThread, so if 3461 // there is one, the track is connected to it 3462 if (!mEffectChains.isEmpty()) { 3463 mEffectChains[0]->setVolume_l(&vl, &vr); 3464 left = (float)vl / (1 << 24); 3465 right = (float)vr / (1 << 24); 3466 } 3467 if (mOutput->stream->set_volume) { 3468 mOutput->stream->set_volume(mOutput->stream, left, right); 3469 } 3470 } 3471 } 3472} 3473 3474 3475AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3476 Vector< sp<Track> > *tracksToRemove 3477) 3478{ 3479 size_t count = mActiveTracks.size(); 3480 mixer_state mixerStatus = MIXER_IDLE; 3481 3482 // find out which tracks need to be processed 3483 for (size_t i = 0; i < count; i++) { 3484 sp<Track> t = mActiveTracks[i].promote(); 3485 // The track died recently 3486 if (t == 0) { 3487 continue; 3488 } 3489 3490 Track* const track = t.get(); 3491 audio_track_cblk_t* cblk = track->cblk(); 3492 3493 // The first time a track is added we wait 3494 // for all its buffers to be filled before processing it 3495 uint32_t minFrames; 3496 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3497 minFrames = mNormalFrameCount; 3498 } else { 3499 minFrames = 1; 3500 } 3501 // Only consider last track started for volume and mixer state control. 3502 // This is the last entry in mActiveTracks unless a track underruns. 3503 // As we only care about the transition phase between two tracks on a 3504 // direct output, it is not a problem to ignore the underrun case. 3505 bool last = (i == (count - 1)); 3506 3507 if ((track->framesReady() >= minFrames) && track->isReady() && 3508 !track->isPaused() && !track->isTerminated()) 3509 { 3510 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3511 3512 if (track->mFillingUpStatus == Track::FS_FILLED) { 3513 track->mFillingUpStatus = Track::FS_ACTIVE; 3514 mLeftVolFloat = mRightVolFloat = 0; 3515 if (track->mState == TrackBase::RESUMING) { 3516 track->mState = TrackBase::ACTIVE; 3517 } 3518 } 3519 3520 // compute volume for this track 3521 processVolume_l(track, last); 3522 if (last) { 3523 // reset retry count 3524 track->mRetryCount = kMaxTrackRetriesDirect; 3525 mActiveTrack = t; 3526 mixerStatus = MIXER_TRACKS_READY; 3527 } 3528 } else { 3529 // clear effect chain input buffer if the last active track started underruns 3530 // to avoid sending previous audio buffer again to effects 3531 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3532 mEffectChains[0]->clearInputBuffer(); 3533 } 3534 3535 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3536 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3537 track->isStopped() || track->isPaused()) { 3538 // We have consumed all the buffers of this track. 3539 // Remove it from the list of active tracks. 3540 // TODO: implement behavior for compressed audio 3541 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3542 size_t framesWritten = mBytesWritten / mFrameSize; 3543 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3544 if (track->isStopped()) { 3545 track->reset(); 3546 } 3547 tracksToRemove->add(track); 3548 } 3549 } else { 3550 // No buffers for this track. Give it a few chances to 3551 // fill a buffer, then remove it from active list. 3552 // Only consider last track started for mixer state control 3553 if (--(track->mRetryCount) <= 0) { 3554 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3555 tracksToRemove->add(track); 3556 } else if (last) { 3557 mixerStatus = MIXER_TRACKS_ENABLED; 3558 } 3559 } 3560 } 3561 } 3562 3563 // remove all the tracks that need to be... 3564 removeTracks_l(*tracksToRemove); 3565 3566 return mixerStatus; 3567} 3568 3569void AudioFlinger::DirectOutputThread::threadLoop_mix() 3570{ 3571 size_t frameCount = mFrameCount; 3572 int8_t *curBuf = (int8_t *)mMixBuffer; 3573 // output audio to hardware 3574 while (frameCount) { 3575 AudioBufferProvider::Buffer buffer; 3576 buffer.frameCount = frameCount; 3577 mActiveTrack->getNextBuffer(&buffer); 3578 if (buffer.raw == NULL) { 3579 memset(curBuf, 0, frameCount * mFrameSize); 3580 break; 3581 } 3582 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3583 frameCount -= buffer.frameCount; 3584 curBuf += buffer.frameCount * mFrameSize; 3585 mActiveTrack->releaseBuffer(&buffer); 3586 } 3587 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3588 sleepTime = 0; 3589 standbyTime = systemTime() + standbyDelay; 3590 mActiveTrack.clear(); 3591} 3592 3593void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3594{ 3595 if (sleepTime == 0) { 3596 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3597 sleepTime = activeSleepTime; 3598 } else { 3599 sleepTime = idleSleepTime; 3600 } 3601 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3602 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3603 sleepTime = 0; 3604 } 3605} 3606 3607// getTrackName_l() must be called with ThreadBase::mLock held 3608int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3609 int sessionId) 3610{ 3611 return 0; 3612} 3613 3614// deleteTrackName_l() must be called with ThreadBase::mLock held 3615void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3616{ 3617} 3618 3619// checkForNewParameters_l() must be called with ThreadBase::mLock held 3620bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3621{ 3622 bool reconfig = false; 3623 3624 while (!mNewParameters.isEmpty()) { 3625 status_t status = NO_ERROR; 3626 String8 keyValuePair = mNewParameters[0]; 3627 AudioParameter param = AudioParameter(keyValuePair); 3628 int value; 3629 3630 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3631 // do not accept frame count changes if tracks are open as the track buffer 3632 // size depends on frame count and correct behavior would not be garantied 3633 // if frame count is changed after track creation 3634 if (!mTracks.isEmpty()) { 3635 status = INVALID_OPERATION; 3636 } else { 3637 reconfig = true; 3638 } 3639 } 3640 if (status == NO_ERROR) { 3641 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3642 keyValuePair.string()); 3643 if (!mStandby && status == INVALID_OPERATION) { 3644 mOutput->stream->common.standby(&mOutput->stream->common); 3645 mStandby = true; 3646 mBytesWritten = 0; 3647 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3648 keyValuePair.string()); 3649 } 3650 if (status == NO_ERROR && reconfig) { 3651 readOutputParameters(); 3652 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3653 } 3654 } 3655 3656 mNewParameters.removeAt(0); 3657 3658 mParamStatus = status; 3659 mParamCond.signal(); 3660 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3661 // already timed out waiting for the status and will never signal the condition. 3662 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3663 } 3664 return reconfig; 3665} 3666 3667uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3668{ 3669 uint32_t time; 3670 if (audio_is_linear_pcm(mFormat)) { 3671 time = PlaybackThread::activeSleepTimeUs(); 3672 } else { 3673 time = 10000; 3674 } 3675 return time; 3676} 3677 3678uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3679{ 3680 uint32_t time; 3681 if (audio_is_linear_pcm(mFormat)) { 3682 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3683 } else { 3684 time = 10000; 3685 } 3686 return time; 3687} 3688 3689uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3690{ 3691 uint32_t time; 3692 if (audio_is_linear_pcm(mFormat)) { 3693 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3694 } else { 3695 time = 10000; 3696 } 3697 return time; 3698} 3699 3700void AudioFlinger::DirectOutputThread::cacheParameters_l() 3701{ 3702 PlaybackThread::cacheParameters_l(); 3703 3704 // use shorter standby delay as on normal output to release 3705 // hardware resources as soon as possible 3706 standbyDelay = microseconds(activeSleepTime*2); 3707} 3708 3709// ---------------------------------------------------------------------------- 3710 3711AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3712 const sp<AudioFlinger::OffloadThread>& offloadThread) 3713 : Thread(false /*canCallJava*/), 3714 mOffloadThread(offloadThread), 3715 mWriteBlocked(false), 3716 mDraining(false) 3717{ 3718} 3719 3720AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3721{ 3722} 3723 3724void AudioFlinger::AsyncCallbackThread::onFirstRef() 3725{ 3726 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3727} 3728 3729bool AudioFlinger::AsyncCallbackThread::threadLoop() 3730{ 3731 while (!exitPending()) { 3732 bool writeBlocked; 3733 bool draining; 3734 3735 { 3736 Mutex::Autolock _l(mLock); 3737 mWaitWorkCV.wait(mLock); 3738 if (exitPending()) { 3739 break; 3740 } 3741 writeBlocked = mWriteBlocked; 3742 draining = mDraining; 3743 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3744 } 3745 { 3746 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3747 if (offloadThread != 0) { 3748 if (writeBlocked == false) { 3749 offloadThread->setWriteBlocked(false); 3750 } 3751 if (draining == false) { 3752 offloadThread->setDraining(false); 3753 } 3754 } 3755 } 3756 } 3757 return false; 3758} 3759 3760void AudioFlinger::AsyncCallbackThread::exit() 3761{ 3762 ALOGV("AsyncCallbackThread::exit"); 3763 Mutex::Autolock _l(mLock); 3764 requestExit(); 3765 mWaitWorkCV.broadcast(); 3766} 3767 3768void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3769{ 3770 Mutex::Autolock _l(mLock); 3771 mWriteBlocked = value; 3772 if (!value) { 3773 mWaitWorkCV.signal(); 3774 } 3775} 3776 3777void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3778{ 3779 Mutex::Autolock _l(mLock); 3780 mDraining = value; 3781 if (!value) { 3782 mWaitWorkCV.signal(); 3783 } 3784} 3785 3786 3787// ---------------------------------------------------------------------------- 3788AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3789 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3790 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3791 mHwPaused(false), 3792 mPausedBytesRemaining(0) 3793{ 3794 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3795} 3796 3797AudioFlinger::OffloadThread::~OffloadThread() 3798{ 3799 mPreviousTrack.clear(); 3800} 3801 3802void AudioFlinger::OffloadThread::threadLoop_exit() 3803{ 3804 if (mFlushPending || mHwPaused) { 3805 // If a flush is pending or track was paused, just discard buffered data 3806 flushHw_l(); 3807 } else { 3808 mMixerStatus = MIXER_DRAIN_ALL; 3809 threadLoop_drain(); 3810 } 3811 mCallbackThread->exit(); 3812 PlaybackThread::threadLoop_exit(); 3813} 3814 3815AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3816 Vector< sp<Track> > *tracksToRemove 3817) 3818{ 3819 ALOGV("OffloadThread::prepareTracks_l"); 3820 size_t count = mActiveTracks.size(); 3821 3822 mixer_state mixerStatus = MIXER_IDLE; 3823 // find out which tracks need to be processed 3824 for (size_t i = 0; i < count; i++) { 3825 sp<Track> t = mActiveTracks[i].promote(); 3826 // The track died recently 3827 if (t == 0) { 3828 continue; 3829 } 3830 Track* const track = t.get(); 3831 audio_track_cblk_t* cblk = track->cblk(); 3832 if (mPreviousTrack != NULL) { 3833 if (t != mPreviousTrack) { 3834 // Flush any data still being written from last track 3835 mBytesRemaining = 0; 3836 if (mPausedBytesRemaining) { 3837 // Last track was paused so we also need to flush saved 3838 // mixbuffer state and invalidate track so that it will 3839 // re-submit that unwritten data when it is next resumed 3840 mPausedBytesRemaining = 0; 3841 // Invalidate is a bit drastic - would be more efficient 3842 // to have a flag to tell client that some of the 3843 // previously written data was lost 3844 mPreviousTrack->invalidate(); 3845 } 3846 } 3847 } 3848 mPreviousTrack = t; 3849 bool last = (i == (count - 1)); 3850 if (track->isPausing()) { 3851 track->setPaused(); 3852 if (last) { 3853 if (!mHwPaused) { 3854 mOutput->stream->pause(mOutput->stream); 3855 mHwPaused = true; 3856 } 3857 // If we were part way through writing the mixbuffer to 3858 // the HAL we must save this until we resume 3859 // BUG - this will be wrong if a different track is made active, 3860 // in that case we want to discard the pending data in the 3861 // mixbuffer and tell the client to present it again when the 3862 // track is resumed 3863 mPausedWriteLength = mCurrentWriteLength; 3864 mPausedBytesRemaining = mBytesRemaining; 3865 mBytesRemaining = 0; // stop writing 3866 } 3867 tracksToRemove->add(track); 3868 } else if (track->framesReady() && track->isReady() && 3869 !track->isPaused() && !track->isTerminated()) { 3870 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3871 if (track->mFillingUpStatus == Track::FS_FILLED) { 3872 track->mFillingUpStatus = Track::FS_ACTIVE; 3873 mLeftVolFloat = mRightVolFloat = 0; 3874 if (track->mState == TrackBase::RESUMING) { 3875 if (mPausedBytesRemaining) { 3876 // Need to continue write that was interrupted 3877 mCurrentWriteLength = mPausedWriteLength; 3878 mBytesRemaining = mPausedBytesRemaining; 3879 mPausedBytesRemaining = 0; 3880 } 3881 track->mState = TrackBase::ACTIVE; 3882 } 3883 } 3884 3885 if (last) { 3886 if (mHwPaused) { 3887 mOutput->stream->resume(mOutput->stream); 3888 mHwPaused = false; 3889 // threadLoop_mix() will handle the case that we need to 3890 // resume an interrupted write 3891 } 3892 // reset retry count 3893 track->mRetryCount = kMaxTrackRetriesOffload; 3894 mActiveTrack = t; 3895 mixerStatus = MIXER_TRACKS_READY; 3896 } 3897 } else { 3898 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3899 if (track->isStopping_1()) { 3900 // Hardware buffer can hold a large amount of audio so we must 3901 // wait for all current track's data to drain before we say 3902 // that the track is stopped. 3903 if (mBytesRemaining == 0) { 3904 // Only start draining when all data in mixbuffer 3905 // has been written 3906 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3907 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3908 sleepTime = 0; 3909 standbyTime = systemTime() + standbyDelay; 3910 if (last) { 3911 mixerStatus = MIXER_DRAIN_TRACK; 3912 if (mHwPaused) { 3913 // It is possible to move from PAUSED to STOPPING_1 without 3914 // a resume so we must ensure hardware is running 3915 mOutput->stream->resume(mOutput->stream); 3916 mHwPaused = false; 3917 } 3918 } 3919 } 3920 } else if (track->isStopping_2()) { 3921 // Drain has completed, signal presentation complete 3922 if (!mDraining || !last) { 3923 track->mState = TrackBase::STOPPED; 3924 size_t audioHALFrames = 3925 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3926 size_t framesWritten = 3927 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3928 track->presentationComplete(framesWritten, audioHALFrames); 3929 track->reset(); 3930 tracksToRemove->add(track); 3931 } 3932 } else { 3933 // No buffers for this track. Give it a few chances to 3934 // fill a buffer, then remove it from active list. 3935 if (--(track->mRetryCount) <= 0) { 3936 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3937 track->name()); 3938 tracksToRemove->add(track); 3939 } else if (last){ 3940 mixerStatus = MIXER_TRACKS_ENABLED; 3941 } 3942 } 3943 } 3944 // compute volume for this track 3945 processVolume_l(track, last); 3946 } 3947 3948 if (mFlushPending) { 3949 flushHw_l(); 3950 mFlushPending = false; 3951 } 3952 3953 // remove all the tracks that need to be... 3954 removeTracks_l(*tracksToRemove); 3955 3956 return mixerStatus; 3957} 3958 3959void AudioFlinger::OffloadThread::flushOutput_l() 3960{ 3961 mFlushPending = true; 3962} 3963 3964// must be called with thread mutex locked 3965bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3966{ 3967 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3968 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3969 return true; 3970 } 3971 return false; 3972} 3973 3974// must be called with thread mutex locked 3975bool AudioFlinger::OffloadThread::shouldStandby_l() 3976{ 3977 bool TrackPaused = false; 3978 3979 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3980 // after a timeout and we will enter standby then. 3981 if (mTracks.size() > 0) { 3982 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3983 } 3984 3985 return !mStandby && !TrackPaused; 3986} 3987 3988 3989bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3990{ 3991 Mutex::Autolock _l(mLock); 3992 return waitingAsyncCallback_l(); 3993} 3994 3995void AudioFlinger::OffloadThread::flushHw_l() 3996{ 3997 mOutput->stream->flush(mOutput->stream); 3998 // Flush anything still waiting in the mixbuffer 3999 mCurrentWriteLength = 0; 4000 mBytesRemaining = 0; 4001 mPausedWriteLength = 0; 4002 mPausedBytesRemaining = 0; 4003 if (mUseAsyncWrite) { 4004 mWriteBlocked = false; 4005 mDraining = false; 4006 ALOG_ASSERT(mCallbackThread != 0); 4007 mCallbackThread->setWriteBlocked(false); 4008 mCallbackThread->setDraining(false); 4009 } 4010} 4011 4012// ---------------------------------------------------------------------------- 4013 4014AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4015 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4016 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4017 DUPLICATING), 4018 mWaitTimeMs(UINT_MAX) 4019{ 4020 addOutputTrack(mainThread); 4021} 4022 4023AudioFlinger::DuplicatingThread::~DuplicatingThread() 4024{ 4025 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4026 mOutputTracks[i]->destroy(); 4027 } 4028} 4029 4030void AudioFlinger::DuplicatingThread::threadLoop_mix() 4031{ 4032 // mix buffers... 4033 if (outputsReady(outputTracks)) { 4034 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4035 } else { 4036 memset(mMixBuffer, 0, mixBufferSize); 4037 } 4038 sleepTime = 0; 4039 writeFrames = mNormalFrameCount; 4040 mCurrentWriteLength = mixBufferSize; 4041 standbyTime = systemTime() + standbyDelay; 4042} 4043 4044void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4045{ 4046 if (sleepTime == 0) { 4047 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4048 sleepTime = activeSleepTime; 4049 } else { 4050 sleepTime = idleSleepTime; 4051 } 4052 } else if (mBytesWritten != 0) { 4053 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4054 writeFrames = mNormalFrameCount; 4055 memset(mMixBuffer, 0, mixBufferSize); 4056 } else { 4057 // flush remaining overflow buffers in output tracks 4058 writeFrames = 0; 4059 } 4060 sleepTime = 0; 4061 } 4062} 4063 4064ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4065{ 4066 for (size_t i = 0; i < outputTracks.size(); i++) { 4067 outputTracks[i]->write(mMixBuffer, writeFrames); 4068 } 4069 return (ssize_t)mixBufferSize; 4070} 4071 4072void AudioFlinger::DuplicatingThread::threadLoop_standby() 4073{ 4074 // DuplicatingThread implements standby by stopping all tracks 4075 for (size_t i = 0; i < outputTracks.size(); i++) { 4076 outputTracks[i]->stop(); 4077 } 4078} 4079 4080void AudioFlinger::DuplicatingThread::saveOutputTracks() 4081{ 4082 outputTracks = mOutputTracks; 4083} 4084 4085void AudioFlinger::DuplicatingThread::clearOutputTracks() 4086{ 4087 outputTracks.clear(); 4088} 4089 4090void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4091{ 4092 Mutex::Autolock _l(mLock); 4093 // FIXME explain this formula 4094 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4095 OutputTrack *outputTrack = new OutputTrack(thread, 4096 this, 4097 mSampleRate, 4098 mFormat, 4099 mChannelMask, 4100 frameCount); 4101 if (outputTrack->cblk() != NULL) { 4102 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4103 mOutputTracks.add(outputTrack); 4104 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4105 updateWaitTime_l(); 4106 } 4107} 4108 4109void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4110{ 4111 Mutex::Autolock _l(mLock); 4112 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4113 if (mOutputTracks[i]->thread() == thread) { 4114 mOutputTracks[i]->destroy(); 4115 mOutputTracks.removeAt(i); 4116 updateWaitTime_l(); 4117 return; 4118 } 4119 } 4120 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4121} 4122 4123// caller must hold mLock 4124void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4125{ 4126 mWaitTimeMs = UINT_MAX; 4127 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4128 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4129 if (strong != 0) { 4130 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4131 if (waitTimeMs < mWaitTimeMs) { 4132 mWaitTimeMs = waitTimeMs; 4133 } 4134 } 4135 } 4136} 4137 4138 4139bool AudioFlinger::DuplicatingThread::outputsReady( 4140 const SortedVector< sp<OutputTrack> > &outputTracks) 4141{ 4142 for (size_t i = 0; i < outputTracks.size(); i++) { 4143 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4144 if (thread == 0) { 4145 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4146 outputTracks[i].get()); 4147 return false; 4148 } 4149 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4150 // see note at standby() declaration 4151 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4152 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4153 thread.get()); 4154 return false; 4155 } 4156 } 4157 return true; 4158} 4159 4160uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4161{ 4162 return (mWaitTimeMs * 1000) / 2; 4163} 4164 4165void AudioFlinger::DuplicatingThread::cacheParameters_l() 4166{ 4167 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4168 updateWaitTime_l(); 4169 4170 MixerThread::cacheParameters_l(); 4171} 4172 4173// ---------------------------------------------------------------------------- 4174// Record 4175// ---------------------------------------------------------------------------- 4176 4177AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4178 AudioStreamIn *input, 4179 uint32_t sampleRate, 4180 audio_channel_mask_t channelMask, 4181 audio_io_handle_t id, 4182 audio_devices_t outDevice, 4183 audio_devices_t inDevice 4184#ifdef TEE_SINK 4185 , const sp<NBAIO_Sink>& teeSink 4186#endif 4187 ) : 4188 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4189 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4190 // mRsmpInIndex set by readInputParameters() 4191 mReqChannelCount(popcount(channelMask)), 4192 mReqSampleRate(sampleRate) 4193 // mBytesRead is only meaningful while active, and so is cleared in start() 4194 // (but might be better to also clear here for dump?) 4195#ifdef TEE_SINK 4196 , mTeeSink(teeSink) 4197#endif 4198{ 4199 snprintf(mName, kNameLength, "AudioIn_%X", id); 4200 4201 readInputParameters(); 4202 4203} 4204 4205 4206AudioFlinger::RecordThread::~RecordThread() 4207{ 4208 delete[] mRsmpInBuffer; 4209 delete mResampler; 4210 delete[] mRsmpOutBuffer; 4211} 4212 4213void AudioFlinger::RecordThread::onFirstRef() 4214{ 4215 run(mName, PRIORITY_URGENT_AUDIO); 4216} 4217 4218bool AudioFlinger::RecordThread::threadLoop() 4219{ 4220 AudioBufferProvider::Buffer buffer; 4221 4222 nsecs_t lastWarning = 0; 4223 4224 inputStandBy(); 4225 acquireWakeLock(); 4226 4227 // used to verify we've read at least once before evaluating how many bytes were read 4228 bool readOnce = false; 4229 4230 // used to request a deferred sleep, to be executed later while mutex is unlocked 4231 bool doSleep = false; 4232 4233 // start recording 4234 for (;;) { 4235 sp<RecordTrack> activeTrack; 4236 TrackBase::track_state activeTrackState; 4237 Vector< sp<EffectChain> > effectChains; 4238 4239 // sleep with mutex unlocked 4240 if (doSleep) { 4241 doSleep = false; 4242 usleep(kRecordThreadSleepUs); 4243 } 4244 4245 { // scope for mLock 4246 Mutex::Autolock _l(mLock); 4247 if (exitPending()) { 4248 break; 4249 } 4250 processConfigEvents_l(); 4251 // return value 'reconfig' is currently unused 4252 bool reconfig = checkForNewParameters_l(); 4253 // make a stable copy of mActiveTrack 4254 activeTrack = mActiveTrack; 4255 if (activeTrack == 0) { 4256 standby(); 4257 // exitPending() can't become true here 4258 releaseWakeLock_l(); 4259 ALOGV("RecordThread: loop stopping"); 4260 // go to sleep 4261 mWaitWorkCV.wait(mLock); 4262 ALOGV("RecordThread: loop starting"); 4263 acquireWakeLock_l(); 4264 continue; 4265 } 4266 4267 if (activeTrack->isTerminated()) { 4268 removeTrack_l(activeTrack); 4269 mActiveTrack.clear(); 4270 continue; 4271 } 4272 4273 activeTrackState = activeTrack->mState; 4274 switch (activeTrackState) { 4275 case TrackBase::PAUSING: 4276 standby(); 4277 mActiveTrack.clear(); 4278 mStartStopCond.broadcast(); 4279 doSleep = true; 4280 continue; 4281 4282 case TrackBase::RESUMING: 4283 mStandby = false; 4284 if (mReqChannelCount != activeTrack->channelCount()) { 4285 mActiveTrack.clear(); 4286 mStartStopCond.broadcast(); 4287 continue; 4288 } 4289 if (readOnce) { 4290 mStartStopCond.broadcast(); 4291 // record start succeeds only if first read from audio input succeeds 4292 if (mBytesRead < 0) { 4293 mActiveTrack.clear(); 4294 continue; 4295 } 4296 activeTrack->mState = TrackBase::ACTIVE; 4297 } 4298 break; 4299 4300 case TrackBase::ACTIVE: 4301 break; 4302 4303 case TrackBase::IDLE: 4304 doSleep = true; 4305 continue; 4306 4307 default: 4308 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4309 } 4310 4311 lockEffectChains_l(effectChains); 4312 } 4313 4314 // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable 4315 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4316 4317 for (size_t i = 0; i < effectChains.size(); i ++) { 4318 // thread mutex is not locked, but effect chain is locked 4319 effectChains[i]->process_l(); 4320 } 4321 4322 buffer.frameCount = mFrameCount; 4323 status_t status = activeTrack->getNextBuffer(&buffer); 4324 if (status == NO_ERROR) { 4325 readOnce = true; 4326 size_t framesOut = buffer.frameCount; 4327 if (mResampler == NULL) { 4328 // no resampling 4329 while (framesOut) { 4330 size_t framesIn = mFrameCount - mRsmpInIndex; 4331 if (framesIn > 0) { 4332 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4333 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4334 activeTrack->mFrameSize; 4335 if (framesIn > framesOut) { 4336 framesIn = framesOut; 4337 } 4338 mRsmpInIndex += framesIn; 4339 framesOut -= framesIn; 4340 if (mChannelCount == mReqChannelCount) { 4341 memcpy(dst, src, framesIn * mFrameSize); 4342 } else { 4343 if (mChannelCount == 1) { 4344 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4345 (int16_t *)src, framesIn); 4346 } else { 4347 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4348 (int16_t *)src, framesIn); 4349 } 4350 } 4351 } 4352 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4353 void *readInto; 4354 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4355 readInto = buffer.raw; 4356 framesOut = 0; 4357 } else { 4358 readInto = mRsmpInBuffer; 4359 mRsmpInIndex = 0; 4360 } 4361 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4362 mBufferSize); 4363 if (mBytesRead <= 0) { 4364 // TODO: verify that it's benign to use a stale track state 4365 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4366 { 4367 ALOGE("Error reading audio input"); 4368 // Force input into standby so that it tries to 4369 // recover at next read attempt 4370 inputStandBy(); 4371 doSleep = true; 4372 } 4373 mRsmpInIndex = mFrameCount; 4374 framesOut = 0; 4375 buffer.frameCount = 0; 4376 } 4377#ifdef TEE_SINK 4378 else if (mTeeSink != 0) { 4379 (void) mTeeSink->write(readInto, 4380 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4381 } 4382#endif 4383 } 4384 } 4385 } else { 4386 // resampling 4387 4388 // resampler accumulates, but we only have one source track 4389 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4390 // alter output frame count as if we were expecting stereo samples 4391 if (mChannelCount == 1 && mReqChannelCount == 1) { 4392 framesOut >>= 1; 4393 } 4394 mResampler->resample(mRsmpOutBuffer, framesOut, 4395 this /* AudioBufferProvider* */); 4396 // ditherAndClamp() works as long as all buffers returned by 4397 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4398 if (mChannelCount == 2 && mReqChannelCount == 1) { 4399 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4400 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4401 // the resampler always outputs stereo samples: 4402 // do post stereo to mono conversion 4403 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4404 framesOut); 4405 } else { 4406 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4407 } 4408 // now done with mRsmpOutBuffer 4409 4410 } 4411 if (mFramestoDrop == 0) { 4412 activeTrack->releaseBuffer(&buffer); 4413 } else { 4414 if (mFramestoDrop > 0) { 4415 mFramestoDrop -= buffer.frameCount; 4416 if (mFramestoDrop <= 0) { 4417 clearSyncStartEvent(); 4418 } 4419 } else { 4420 mFramestoDrop += buffer.frameCount; 4421 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4422 mSyncStartEvent->isCancelled()) { 4423 ALOGW("Synced record %s, session %d, trigger session %d", 4424 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4425 activeTrack->sessionId(), 4426 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4427 clearSyncStartEvent(); 4428 } 4429 } 4430 } 4431 activeTrack->clearOverflow(); 4432 } 4433 // client isn't retrieving buffers fast enough 4434 else { 4435 if (!activeTrack->setOverflow()) { 4436 nsecs_t now = systemTime(); 4437 if ((now - lastWarning) > kWarningThrottleNs) { 4438 ALOGW("RecordThread: buffer overflow"); 4439 lastWarning = now; 4440 } 4441 } 4442 // Release the processor for a while before asking for a new buffer. 4443 // This will give the application more chance to read from the buffer and 4444 // clear the overflow. 4445 doSleep = true; 4446 } 4447 4448 // enable changes in effect chain 4449 unlockEffectChains(effectChains); 4450 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4451 } 4452 4453 standby(); 4454 4455 { 4456 Mutex::Autolock _l(mLock); 4457 mActiveTrack.clear(); 4458 mStartStopCond.broadcast(); 4459 } 4460 4461 releaseWakeLock(); 4462 4463 ALOGV("RecordThread %p exiting", this); 4464 return false; 4465} 4466 4467void AudioFlinger::RecordThread::standby() 4468{ 4469 if (!mStandby) { 4470 inputStandBy(); 4471 mStandby = true; 4472 } 4473} 4474 4475void AudioFlinger::RecordThread::inputStandBy() 4476{ 4477 mInput->stream->common.standby(&mInput->stream->common); 4478} 4479 4480sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4481 const sp<AudioFlinger::Client>& client, 4482 uint32_t sampleRate, 4483 audio_format_t format, 4484 audio_channel_mask_t channelMask, 4485 size_t frameCount, 4486 int sessionId, 4487 IAudioFlinger::track_flags_t *flags, 4488 pid_t tid, 4489 status_t *status) 4490{ 4491 sp<RecordTrack> track; 4492 status_t lStatus; 4493 4494 lStatus = initCheck(); 4495 if (lStatus != NO_ERROR) { 4496 ALOGE("Audio driver not initialized."); 4497 goto Exit; 4498 } 4499 4500 // client expresses a preference for FAST, but we get the final say 4501 if (*flags & IAudioFlinger::TRACK_FAST) { 4502 if ( 4503 // use case: callback handler and frame count is default or at least as large as HAL 4504 ( 4505 (tid != -1) && 4506 ((frameCount == 0) || 4507 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4508 ) && 4509 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4510 // mono or stereo 4511 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4512 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4513 // hardware sample rate 4514 (sampleRate == mSampleRate) && 4515 // record thread has an associated fast recorder 4516 hasFastRecorder() 4517 // FIXME test that RecordThread for this fast track has a capable output HAL 4518 // FIXME add a permission test also? 4519 ) { 4520 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4521 if (frameCount == 0) { 4522 frameCount = mFrameCount * kFastTrackMultiplier; 4523 } 4524 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4525 frameCount, mFrameCount); 4526 } else { 4527 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4528 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4529 "hasFastRecorder=%d tid=%d", 4530 frameCount, mFrameCount, format, 4531 audio_is_linear_pcm(format), 4532 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4533 *flags &= ~IAudioFlinger::TRACK_FAST; 4534 // For compatibility with AudioRecord calculation, buffer depth is forced 4535 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4536 // This is probably too conservative, but legacy application code may depend on it. 4537 // If you change this calculation, also review the start threshold which is related. 4538 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4539 size_t mNormalFrameCount = 2048; // FIXME 4540 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4541 if (minBufCount < 2) { 4542 minBufCount = 2; 4543 } 4544 size_t minFrameCount = mNormalFrameCount * minBufCount; 4545 if (frameCount < minFrameCount) { 4546 frameCount = minFrameCount; 4547 } 4548 } 4549 } 4550 4551 // FIXME use flags and tid similar to createTrack_l() 4552 4553 { // scope for mLock 4554 Mutex::Autolock _l(mLock); 4555 4556 track = new RecordTrack(this, client, sampleRate, 4557 format, channelMask, frameCount, sessionId); 4558 4559 lStatus = track->initCheck(); 4560 if (lStatus != NO_ERROR) { 4561 track.clear(); 4562 goto Exit; 4563 } 4564 mTracks.add(track); 4565 4566 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4567 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4568 mAudioFlinger->btNrecIsOff(); 4569 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4570 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4571 4572 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4573 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4574 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4575 // so ask activity manager to do this on our behalf 4576 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4577 } 4578 } 4579 lStatus = NO_ERROR; 4580 4581Exit: 4582 *status = lStatus; 4583 return track; 4584} 4585 4586status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4587 AudioSystem::sync_event_t event, 4588 int triggerSession) 4589{ 4590 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4591 sp<ThreadBase> strongMe = this; 4592 status_t status = NO_ERROR; 4593 4594 if (event == AudioSystem::SYNC_EVENT_NONE) { 4595 clearSyncStartEvent(); 4596 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4597 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4598 triggerSession, 4599 recordTrack->sessionId(), 4600 syncStartEventCallback, 4601 this); 4602 // Sync event can be cancelled by the trigger session if the track is not in a 4603 // compatible state in which case we start record immediately 4604 if (mSyncStartEvent->isCancelled()) { 4605 clearSyncStartEvent(); 4606 } else { 4607 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4608 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4609 } 4610 } 4611 4612 { 4613 // This section is a rendezvous between binder thread executing start() and RecordThread 4614 AutoMutex lock(mLock); 4615 if (mActiveTrack != 0) { 4616 if (recordTrack != mActiveTrack.get()) { 4617 status = -EBUSY; 4618 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4619 mActiveTrack->mState = TrackBase::ACTIVE; 4620 } 4621 return status; 4622 } 4623 4624 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4625 recordTrack->mState = TrackBase::IDLE; 4626 mActiveTrack = recordTrack; 4627 mLock.unlock(); 4628 status_t status = AudioSystem::startInput(mId); 4629 mLock.lock(); 4630 // FIXME should verify that mActiveTrack is still == recordTrack 4631 if (status != NO_ERROR) { 4632 mActiveTrack.clear(); 4633 clearSyncStartEvent(); 4634 return status; 4635 } 4636 mRsmpInIndex = mFrameCount; 4637 mBytesRead = 0; 4638 if (mResampler != NULL) { 4639 mResampler->reset(); 4640 } 4641 // FIXME hijacking a playback track state name which was intended for start after pause; 4642 // here 'STARTING_2' would be more accurate 4643 mActiveTrack->mState = TrackBase::RESUMING; 4644 // signal thread to start 4645 ALOGV("Signal record thread"); 4646 mWaitWorkCV.broadcast(); 4647 // do not wait for mStartStopCond if exiting 4648 if (exitPending()) { 4649 mActiveTrack.clear(); 4650 status = INVALID_OPERATION; 4651 goto startError; 4652 } 4653 // FIXME incorrect usage of wait: no explicit predicate or loop 4654 mStartStopCond.wait(mLock); 4655 if (mActiveTrack == 0) { 4656 ALOGV("Record failed to start"); 4657 status = BAD_VALUE; 4658 goto startError; 4659 } 4660 ALOGV("Record started OK"); 4661 return status; 4662 } 4663 4664startError: 4665 AudioSystem::stopInput(mId); 4666 clearSyncStartEvent(); 4667 return status; 4668} 4669 4670void AudioFlinger::RecordThread::clearSyncStartEvent() 4671{ 4672 if (mSyncStartEvent != 0) { 4673 mSyncStartEvent->cancel(); 4674 } 4675 mSyncStartEvent.clear(); 4676 mFramestoDrop = 0; 4677} 4678 4679void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4680{ 4681 sp<SyncEvent> strongEvent = event.promote(); 4682 4683 if (strongEvent != 0) { 4684 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4685 me->handleSyncStartEvent(strongEvent); 4686 } 4687} 4688 4689void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4690{ 4691 if (event == mSyncStartEvent) { 4692 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4693 // from audio HAL 4694 mFramestoDrop = mFrameCount * 2; 4695 } 4696} 4697 4698bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4699 ALOGV("RecordThread::stop"); 4700 AutoMutex _l(mLock); 4701 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4702 return false; 4703 } 4704 // note that threadLoop may still be processing the track at this point [without lock] 4705 recordTrack->mState = TrackBase::PAUSING; 4706 // do not wait for mStartStopCond if exiting 4707 if (exitPending()) { 4708 return true; 4709 } 4710 // FIXME incorrect usage of wait: no explicit predicate or loop 4711 mStartStopCond.wait(mLock); 4712 // if we have been restarted, recordTrack == mActiveTrack.get() here 4713 if (exitPending() || recordTrack != mActiveTrack.get()) { 4714 ALOGV("Record stopped OK"); 4715 return true; 4716 } 4717 return false; 4718} 4719 4720bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4721{ 4722 return false; 4723} 4724 4725status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4726{ 4727#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4728 if (!isValidSyncEvent(event)) { 4729 return BAD_VALUE; 4730 } 4731 4732 int eventSession = event->triggerSession(); 4733 status_t ret = NAME_NOT_FOUND; 4734 4735 Mutex::Autolock _l(mLock); 4736 4737 for (size_t i = 0; i < mTracks.size(); i++) { 4738 sp<RecordTrack> track = mTracks[i]; 4739 if (eventSession == track->sessionId()) { 4740 (void) track->setSyncEvent(event); 4741 ret = NO_ERROR; 4742 } 4743 } 4744 return ret; 4745#else 4746 return BAD_VALUE; 4747#endif 4748} 4749 4750// destroyTrack_l() must be called with ThreadBase::mLock held 4751void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4752{ 4753 track->terminate(); 4754 track->mState = TrackBase::STOPPED; 4755 // active tracks are removed by threadLoop() 4756 if (mActiveTrack != track) { 4757 removeTrack_l(track); 4758 } 4759} 4760 4761void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4762{ 4763 mTracks.remove(track); 4764 // need anything related to effects here? 4765} 4766 4767void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4768{ 4769 dumpInternals(fd, args); 4770 dumpTracks(fd, args); 4771 dumpEffectChains(fd, args); 4772} 4773 4774void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4775{ 4776 const size_t SIZE = 256; 4777 char buffer[SIZE]; 4778 String8 result; 4779 4780 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4781 result.append(buffer); 4782 4783 if (mActiveTrack != 0) { 4784 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4785 result.append(buffer); 4786 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4787 result.append(buffer); 4788 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4789 result.append(buffer); 4790 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4791 result.append(buffer); 4792 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4793 result.append(buffer); 4794 } else { 4795 result.append("No active record client\n"); 4796 } 4797 4798 write(fd, result.string(), result.size()); 4799 4800 dumpBase(fd, args); 4801} 4802 4803void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4804{ 4805 const size_t SIZE = 256; 4806 char buffer[SIZE]; 4807 String8 result; 4808 4809 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4810 result.append(buffer); 4811 RecordTrack::appendDumpHeader(result); 4812 for (size_t i = 0; i < mTracks.size(); ++i) { 4813 sp<RecordTrack> track = mTracks[i]; 4814 if (track != 0) { 4815 track->dump(buffer, SIZE); 4816 result.append(buffer); 4817 } 4818 } 4819 4820 if (mActiveTrack != 0) { 4821 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4822 result.append(buffer); 4823 RecordTrack::appendDumpHeader(result); 4824 mActiveTrack->dump(buffer, SIZE); 4825 result.append(buffer); 4826 4827 } 4828 write(fd, result.string(), result.size()); 4829} 4830 4831// AudioBufferProvider interface 4832status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4833{ 4834 size_t framesReq = buffer->frameCount; 4835 size_t framesReady = mFrameCount - mRsmpInIndex; 4836 int channelCount; 4837 4838 if (framesReady == 0) { 4839 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4840 if (mBytesRead <= 0) { 4841 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4842 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4843 // Force input into standby so that it tries to 4844 // recover at next read attempt 4845 inputStandBy(); 4846 // FIXME an awkward place to sleep, consider using doSleep when this is pulled up 4847 usleep(kRecordThreadSleepUs); 4848 } 4849 buffer->raw = NULL; 4850 buffer->frameCount = 0; 4851 return NOT_ENOUGH_DATA; 4852 } 4853 mRsmpInIndex = 0; 4854 framesReady = mFrameCount; 4855 } 4856 4857 if (framesReq > framesReady) { 4858 framesReq = framesReady; 4859 } 4860 4861 if (mChannelCount == 1 && mReqChannelCount == 2) { 4862 channelCount = 1; 4863 } else { 4864 channelCount = 2; 4865 } 4866 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4867 buffer->frameCount = framesReq; 4868 return NO_ERROR; 4869} 4870 4871// AudioBufferProvider interface 4872void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4873{ 4874 mRsmpInIndex += buffer->frameCount; 4875 buffer->frameCount = 0; 4876} 4877 4878bool AudioFlinger::RecordThread::checkForNewParameters_l() 4879{ 4880 bool reconfig = false; 4881 4882 while (!mNewParameters.isEmpty()) { 4883 status_t status = NO_ERROR; 4884 String8 keyValuePair = mNewParameters[0]; 4885 AudioParameter param = AudioParameter(keyValuePair); 4886 int value; 4887 audio_format_t reqFormat = mFormat; 4888 uint32_t reqSamplingRate = mReqSampleRate; 4889 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 4890 4891 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4892 reqSamplingRate = value; 4893 reconfig = true; 4894 } 4895 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4896 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4897 status = BAD_VALUE; 4898 } else { 4899 reqFormat = (audio_format_t) value; 4900 reconfig = true; 4901 } 4902 } 4903 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4904 audio_channel_mask_t mask = (audio_channel_mask_t) value; 4905 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 4906 status = BAD_VALUE; 4907 } else { 4908 reqChannelMask = mask; 4909 reconfig = true; 4910 } 4911 } 4912 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4913 // do not accept frame count changes if tracks are open as the track buffer 4914 // size depends on frame count and correct behavior would not be guaranteed 4915 // if frame count is changed after track creation 4916 if (mActiveTrack != 0) { 4917 status = INVALID_OPERATION; 4918 } else { 4919 reconfig = true; 4920 } 4921 } 4922 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4923 // forward device change to effects that have requested to be 4924 // aware of attached audio device. 4925 for (size_t i = 0; i < mEffectChains.size(); i++) { 4926 mEffectChains[i]->setDevice_l(value); 4927 } 4928 4929 // store input device and output device but do not forward output device to audio HAL. 4930 // Note that status is ignored by the caller for output device 4931 // (see AudioFlinger::setParameters() 4932 if (audio_is_output_devices(value)) { 4933 mOutDevice = value; 4934 status = BAD_VALUE; 4935 } else { 4936 mInDevice = value; 4937 // disable AEC and NS if the device is a BT SCO headset supporting those 4938 // pre processings 4939 if (mTracks.size() > 0) { 4940 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4941 mAudioFlinger->btNrecIsOff(); 4942 for (size_t i = 0; i < mTracks.size(); i++) { 4943 sp<RecordTrack> track = mTracks[i]; 4944 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4945 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4946 } 4947 } 4948 } 4949 } 4950 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4951 mAudioSource != (audio_source_t)value) { 4952 // forward device change to effects that have requested to be 4953 // aware of attached audio device. 4954 for (size_t i = 0; i < mEffectChains.size(); i++) { 4955 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4956 } 4957 mAudioSource = (audio_source_t)value; 4958 } 4959 4960 if (status == NO_ERROR) { 4961 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4962 keyValuePair.string()); 4963 if (status == INVALID_OPERATION) { 4964 inputStandBy(); 4965 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4966 keyValuePair.string()); 4967 } 4968 if (reconfig) { 4969 if (status == BAD_VALUE && 4970 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4971 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4972 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4973 <= (2 * reqSamplingRate)) && 4974 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4975 <= FCC_2 && 4976 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 4977 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 4978 status = NO_ERROR; 4979 } 4980 if (status == NO_ERROR) { 4981 readInputParameters(); 4982 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4983 } 4984 } 4985 } 4986 4987 mNewParameters.removeAt(0); 4988 4989 mParamStatus = status; 4990 mParamCond.signal(); 4991 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4992 // already timed out waiting for the status and will never signal the condition. 4993 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4994 } 4995 return reconfig; 4996} 4997 4998String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4999{ 5000 Mutex::Autolock _l(mLock); 5001 if (initCheck() != NO_ERROR) { 5002 return String8(); 5003 } 5004 5005 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5006 const String8 out_s8(s); 5007 free(s); 5008 return out_s8; 5009} 5010 5011void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5012 AudioSystem::OutputDescriptor desc; 5013 void *param2 = NULL; 5014 5015 switch (event) { 5016 case AudioSystem::INPUT_OPENED: 5017 case AudioSystem::INPUT_CONFIG_CHANGED: 5018 desc.channelMask = mChannelMask; 5019 desc.samplingRate = mSampleRate; 5020 desc.format = mFormat; 5021 desc.frameCount = mFrameCount; 5022 desc.latency = 0; 5023 param2 = &desc; 5024 break; 5025 5026 case AudioSystem::INPUT_CLOSED: 5027 default: 5028 break; 5029 } 5030 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5031} 5032 5033void AudioFlinger::RecordThread::readInputParameters() 5034{ 5035 delete[] mRsmpInBuffer; 5036 // mRsmpInBuffer is always assigned a new[] below 5037 delete[] mRsmpOutBuffer; 5038 mRsmpOutBuffer = NULL; 5039 delete mResampler; 5040 mResampler = NULL; 5041 5042 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5043 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5044 mChannelCount = popcount(mChannelMask); 5045 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5046 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5047 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5048 } 5049 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5050 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5051 mFrameCount = mBufferSize / mFrameSize; 5052 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5053 5054 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5055 int channelCount; 5056 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5057 // stereo to mono post process as the resampler always outputs stereo. 5058 if (mChannelCount == 1 && mReqChannelCount == 2) { 5059 channelCount = 1; 5060 } else { 5061 channelCount = 2; 5062 } 5063 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5064 mResampler->setSampleRate(mSampleRate); 5065 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5066 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5067 5068 // optmization: if mono to mono, alter input frame count as if we were inputing 5069 // stereo samples 5070 if (mChannelCount == 1 && mReqChannelCount == 1) { 5071 mFrameCount >>= 1; 5072 } 5073 5074 } 5075 mRsmpInIndex = mFrameCount; 5076} 5077 5078unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5079{ 5080 Mutex::Autolock _l(mLock); 5081 if (initCheck() != NO_ERROR) { 5082 return 0; 5083 } 5084 5085 return mInput->stream->get_input_frames_lost(mInput->stream); 5086} 5087 5088uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5089{ 5090 Mutex::Autolock _l(mLock); 5091 uint32_t result = 0; 5092 if (getEffectChain_l(sessionId) != 0) { 5093 result = EFFECT_SESSION; 5094 } 5095 5096 for (size_t i = 0; i < mTracks.size(); ++i) { 5097 if (sessionId == mTracks[i]->sessionId()) { 5098 result |= TRACK_SESSION; 5099 break; 5100 } 5101 } 5102 5103 return result; 5104} 5105 5106KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5107{ 5108 KeyedVector<int, bool> ids; 5109 Mutex::Autolock _l(mLock); 5110 for (size_t j = 0; j < mTracks.size(); ++j) { 5111 sp<RecordThread::RecordTrack> track = mTracks[j]; 5112 int sessionId = track->sessionId(); 5113 if (ids.indexOfKey(sessionId) < 0) { 5114 ids.add(sessionId, true); 5115 } 5116 } 5117 return ids; 5118} 5119 5120AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5121{ 5122 Mutex::Autolock _l(mLock); 5123 AudioStreamIn *input = mInput; 5124 mInput = NULL; 5125 return input; 5126} 5127 5128// this method must always be called either with ThreadBase mLock held or inside the thread loop 5129audio_stream_t* AudioFlinger::RecordThread::stream() const 5130{ 5131 if (mInput == NULL) { 5132 return NULL; 5133 } 5134 return &mInput->stream->common; 5135} 5136 5137status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5138{ 5139 // only one chain per input thread 5140 if (mEffectChains.size() != 0) { 5141 return INVALID_OPERATION; 5142 } 5143 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5144 5145 chain->setInBuffer(NULL); 5146 chain->setOutBuffer(NULL); 5147 5148 checkSuspendOnAddEffectChain_l(chain); 5149 5150 mEffectChains.add(chain); 5151 5152 return NO_ERROR; 5153} 5154 5155size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5156{ 5157 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5158 ALOGW_IF(mEffectChains.size() != 1, 5159 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5160 chain.get(), mEffectChains.size(), this); 5161 if (mEffectChains.size() == 1) { 5162 mEffectChains.removeAt(0); 5163 } 5164 return 0; 5165} 5166 5167}; // namespace android 5168