Threads.cpp revision a7fef85e7d419a4f5d6a3144f9ba70ceff2f122a
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 = mNormalSink->getTimestamp(mLatchD.mTimestamp); 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 // FIXME We should have an implementation of timestamps for direct output threads. 1849 // They are used e.g for multichannel PCM playback over HDMI. 1850 bytesWritten = mOutput->stream->write(mOutput->stream, 1851 mMixBuffer + offset, mBytesRemaining); 1852 if (mUseAsyncWrite && 1853 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1854 // do not wait for async callback in case of error of full write 1855 mWriteBlocked = false; 1856 ALOG_ASSERT(mCallbackThread != 0); 1857 mCallbackThread->setWriteBlocked(false); 1858 } 1859 } 1860 1861 mNumWrites++; 1862 mInWrite = false; 1863 1864 return bytesWritten; 1865} 1866 1867void AudioFlinger::PlaybackThread::threadLoop_drain() 1868{ 1869 if (mOutput->stream->drain) { 1870 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1871 if (mUseAsyncWrite) { 1872 mDraining = true; 1873 ALOG_ASSERT(mCallbackThread != 0); 1874 mCallbackThread->setDraining(true); 1875 } 1876 mOutput->stream->drain(mOutput->stream, 1877 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1878 : AUDIO_DRAIN_ALL); 1879 } 1880} 1881 1882void AudioFlinger::PlaybackThread::threadLoop_exit() 1883{ 1884 // Default implementation has nothing to do 1885} 1886 1887/* 1888The derived values that are cached: 1889 - mixBufferSize from frame count * frame size 1890 - activeSleepTime from activeSleepTimeUs() 1891 - idleSleepTime from idleSleepTimeUs() 1892 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1893 - maxPeriod from frame count and sample rate (MIXER only) 1894 1895The parameters that affect these derived values are: 1896 - frame count 1897 - frame size 1898 - sample rate 1899 - device type: A2DP or not 1900 - device latency 1901 - format: PCM or not 1902 - active sleep time 1903 - idle sleep time 1904*/ 1905 1906void AudioFlinger::PlaybackThread::cacheParameters_l() 1907{ 1908 mixBufferSize = mNormalFrameCount * mFrameSize; 1909 activeSleepTime = activeSleepTimeUs(); 1910 idleSleepTime = idleSleepTimeUs(); 1911} 1912 1913void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1914{ 1915 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1916 this, streamType, mTracks.size()); 1917 Mutex::Autolock _l(mLock); 1918 1919 size_t size = mTracks.size(); 1920 for (size_t i = 0; i < size; i++) { 1921 sp<Track> t = mTracks[i]; 1922 if (t->streamType() == streamType) { 1923 t->invalidate(); 1924 } 1925 } 1926} 1927 1928status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1929{ 1930 int session = chain->sessionId(); 1931 int16_t *buffer = mMixBuffer; 1932 bool ownsBuffer = false; 1933 1934 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1935 if (session > 0) { 1936 // Only one effect chain can be present in direct output thread and it uses 1937 // the mix buffer as input 1938 if (mType != DIRECT) { 1939 size_t numSamples = mNormalFrameCount * mChannelCount; 1940 buffer = new int16_t[numSamples]; 1941 memset(buffer, 0, numSamples * sizeof(int16_t)); 1942 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1943 ownsBuffer = true; 1944 } 1945 1946 // Attach all tracks with same session ID to this chain. 1947 for (size_t i = 0; i < mTracks.size(); ++i) { 1948 sp<Track> track = mTracks[i]; 1949 if (session == track->sessionId()) { 1950 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1951 buffer); 1952 track->setMainBuffer(buffer); 1953 chain->incTrackCnt(); 1954 } 1955 } 1956 1957 // indicate all active tracks in the chain 1958 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1959 sp<Track> track = mActiveTracks[i].promote(); 1960 if (track == 0) { 1961 continue; 1962 } 1963 if (session == track->sessionId()) { 1964 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1965 chain->incActiveTrackCnt(); 1966 } 1967 } 1968 } 1969 1970 chain->setInBuffer(buffer, ownsBuffer); 1971 chain->setOutBuffer(mMixBuffer); 1972 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1973 // chains list in order to be processed last as it contains output stage effects 1974 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1975 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1976 // after track specific effects and before output stage 1977 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1978 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1979 // Effect chain for other sessions are inserted at beginning of effect 1980 // chains list to be processed before output mix effects. Relative order between other 1981 // sessions is not important 1982 size_t size = mEffectChains.size(); 1983 size_t i = 0; 1984 for (i = 0; i < size; i++) { 1985 if (mEffectChains[i]->sessionId() < session) { 1986 break; 1987 } 1988 } 1989 mEffectChains.insertAt(chain, i); 1990 checkSuspendOnAddEffectChain_l(chain); 1991 1992 return NO_ERROR; 1993} 1994 1995size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1996{ 1997 int session = chain->sessionId(); 1998 1999 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2000 2001 for (size_t i = 0; i < mEffectChains.size(); i++) { 2002 if (chain == mEffectChains[i]) { 2003 mEffectChains.removeAt(i); 2004 // detach all active tracks from the chain 2005 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2006 sp<Track> track = mActiveTracks[i].promote(); 2007 if (track == 0) { 2008 continue; 2009 } 2010 if (session == track->sessionId()) { 2011 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2012 chain.get(), session); 2013 chain->decActiveTrackCnt(); 2014 } 2015 } 2016 2017 // detach all tracks with same session ID from this chain 2018 for (size_t i = 0; i < mTracks.size(); ++i) { 2019 sp<Track> track = mTracks[i]; 2020 if (session == track->sessionId()) { 2021 track->setMainBuffer(mMixBuffer); 2022 chain->decTrackCnt(); 2023 } 2024 } 2025 break; 2026 } 2027 } 2028 return mEffectChains.size(); 2029} 2030 2031status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2032 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2033{ 2034 Mutex::Autolock _l(mLock); 2035 return attachAuxEffect_l(track, EffectId); 2036} 2037 2038status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2039 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2040{ 2041 status_t status = NO_ERROR; 2042 2043 if (EffectId == 0) { 2044 track->setAuxBuffer(0, NULL); 2045 } else { 2046 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2047 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2048 if (effect != 0) { 2049 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2050 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2051 } else { 2052 status = INVALID_OPERATION; 2053 } 2054 } else { 2055 status = BAD_VALUE; 2056 } 2057 } 2058 return status; 2059} 2060 2061void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2062{ 2063 for (size_t i = 0; i < mTracks.size(); ++i) { 2064 sp<Track> track = mTracks[i]; 2065 if (track->auxEffectId() == effectId) { 2066 attachAuxEffect_l(track, 0); 2067 } 2068 } 2069} 2070 2071bool AudioFlinger::PlaybackThread::threadLoop() 2072{ 2073 Vector< sp<Track> > tracksToRemove; 2074 2075 standbyTime = systemTime(); 2076 2077 // MIXER 2078 nsecs_t lastWarning = 0; 2079 2080 // DUPLICATING 2081 // FIXME could this be made local to while loop? 2082 writeFrames = 0; 2083 2084 cacheParameters_l(); 2085 sleepTime = idleSleepTime; 2086 2087 if (mType == MIXER) { 2088 sleepTimeShift = 0; 2089 } 2090 2091 CpuStats cpuStats; 2092 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2093 2094 acquireWakeLock(); 2095 2096 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2097 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2098 // and then that string will be logged at the next convenient opportunity. 2099 const char *logString = NULL; 2100 2101 while (!exitPending()) 2102 { 2103 cpuStats.sample(myName); 2104 2105 Vector< sp<EffectChain> > effectChains; 2106 2107 processConfigEvents(); 2108 2109 { // scope for mLock 2110 2111 Mutex::Autolock _l(mLock); 2112 2113 if (logString != NULL) { 2114 mNBLogWriter->logTimestamp(); 2115 mNBLogWriter->log(logString); 2116 logString = NULL; 2117 } 2118 2119 if (mLatchDValid) { 2120 mLatchQ = mLatchD; 2121 mLatchDValid = false; 2122 mLatchQValid = true; 2123 } 2124 2125 if (checkForNewParameters_l()) { 2126 cacheParameters_l(); 2127 } 2128 2129 saveOutputTracks(); 2130 2131 if (mSignalPending) { 2132 // A signal was raised while we were unlocked 2133 mSignalPending = false; 2134 } else if (waitingAsyncCallback_l()) { 2135 if (exitPending()) { 2136 break; 2137 } 2138 releaseWakeLock_l(); 2139 ALOGV("wait async completion"); 2140 mWaitWorkCV.wait(mLock); 2141 ALOGV("async completion/wake"); 2142 acquireWakeLock_l(); 2143 if (exitPending()) { 2144 break; 2145 } 2146 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2147 continue; 2148 } 2149 sleepTime = 0; 2150 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2151 isSuspended()) { 2152 // put audio hardware into standby after short delay 2153 if (shouldStandby_l()) { 2154 2155 threadLoop_standby(); 2156 2157 mStandby = true; 2158 } 2159 2160 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2161 // we're about to wait, flush the binder command buffer 2162 IPCThreadState::self()->flushCommands(); 2163 2164 clearOutputTracks(); 2165 2166 if (exitPending()) { 2167 break; 2168 } 2169 2170 releaseWakeLock_l(); 2171 // wait until we have something to do... 2172 ALOGV("%s going to sleep", myName.string()); 2173 mWaitWorkCV.wait(mLock); 2174 ALOGV("%s waking up", myName.string()); 2175 acquireWakeLock_l(); 2176 2177 mMixerStatus = MIXER_IDLE; 2178 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2179 mBytesWritten = 0; 2180 mBytesRemaining = 0; 2181 checkSilentMode_l(); 2182 2183 standbyTime = systemTime() + standbyDelay; 2184 sleepTime = idleSleepTime; 2185 if (mType == MIXER) { 2186 sleepTimeShift = 0; 2187 } 2188 2189 continue; 2190 } 2191 } 2192 2193 // mMixerStatusIgnoringFastTracks is also updated internally 2194 mMixerStatus = prepareTracks_l(&tracksToRemove); 2195 2196 // prevent any changes in effect chain list and in each effect chain 2197 // during mixing and effect process as the audio buffers could be deleted 2198 // or modified if an effect is created or deleted 2199 lockEffectChains_l(effectChains); 2200 } 2201 2202 if (mBytesRemaining == 0) { 2203 mCurrentWriteLength = 0; 2204 if (mMixerStatus == MIXER_TRACKS_READY) { 2205 // threadLoop_mix() sets mCurrentWriteLength 2206 threadLoop_mix(); 2207 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2208 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2209 // threadLoop_sleepTime sets sleepTime to 0 if data 2210 // must be written to HAL 2211 threadLoop_sleepTime(); 2212 if (sleepTime == 0) { 2213 mCurrentWriteLength = mixBufferSize; 2214 } 2215 } 2216 mBytesRemaining = mCurrentWriteLength; 2217 if (isSuspended()) { 2218 sleepTime = suspendSleepTimeUs(); 2219 // simulate write to HAL when suspended 2220 mBytesWritten += mixBufferSize; 2221 mBytesRemaining = 0; 2222 } 2223 2224 // only process effects if we're going to write 2225 if (sleepTime == 0) { 2226 for (size_t i = 0; i < effectChains.size(); i ++) { 2227 effectChains[i]->process_l(); 2228 } 2229 } 2230 } 2231 2232 // enable changes in effect chain 2233 unlockEffectChains(effectChains); 2234 2235 if (!waitingAsyncCallback()) { 2236 // sleepTime == 0 means we must write to audio hardware 2237 if (sleepTime == 0) { 2238 if (mBytesRemaining) { 2239 ssize_t ret = threadLoop_write(); 2240 if (ret < 0) { 2241 mBytesRemaining = 0; 2242 } else { 2243 mBytesWritten += ret; 2244 mBytesRemaining -= ret; 2245 } 2246 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2247 (mMixerStatus == MIXER_DRAIN_ALL)) { 2248 threadLoop_drain(); 2249 } 2250if (mType == MIXER) { 2251 // write blocked detection 2252 nsecs_t now = systemTime(); 2253 nsecs_t delta = now - mLastWriteTime; 2254 if (!mStandby && delta > maxPeriod) { 2255 mNumDelayedWrites++; 2256 if ((now - lastWarning) > kWarningThrottleNs) { 2257 ATRACE_NAME("underrun"); 2258 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2259 ns2ms(delta), mNumDelayedWrites, this); 2260 lastWarning = now; 2261 } 2262 } 2263} 2264 2265 mStandby = false; 2266 } else { 2267 usleep(sleepTime); 2268 } 2269 } 2270 2271 // Finally let go of removed track(s), without the lock held 2272 // since we can't guarantee the destructors won't acquire that 2273 // same lock. This will also mutate and push a new fast mixer state. 2274 threadLoop_removeTracks(tracksToRemove); 2275 tracksToRemove.clear(); 2276 2277 // FIXME I don't understand the need for this here; 2278 // it was in the original code but maybe the 2279 // assignment in saveOutputTracks() makes this unnecessary? 2280 clearOutputTracks(); 2281 2282 // Effect chains will be actually deleted here if they were removed from 2283 // mEffectChains list during mixing or effects processing 2284 effectChains.clear(); 2285 2286 // FIXME Note that the above .clear() is no longer necessary since effectChains 2287 // is now local to this block, but will keep it for now (at least until merge done). 2288 } 2289 2290 threadLoop_exit(); 2291 2292 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2293 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2294 // put output stream into standby mode 2295 if (!mStandby) { 2296 mOutput->stream->common.standby(&mOutput->stream->common); 2297 } 2298 } 2299 2300 releaseWakeLock(); 2301 2302 ALOGV("Thread %p type %d exiting", this, mType); 2303 return false; 2304} 2305 2306// removeTracks_l() must be called with ThreadBase::mLock held 2307void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2308{ 2309 size_t count = tracksToRemove.size(); 2310 if (count > 0) { 2311 for (size_t i=0 ; i<count ; i++) { 2312 const sp<Track>& track = tracksToRemove.itemAt(i); 2313 mActiveTracks.remove(track); 2314 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2315 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2316 if (chain != 0) { 2317 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2318 track->sessionId()); 2319 chain->decActiveTrackCnt(); 2320 } 2321 if (track->isTerminated()) { 2322 removeTrack_l(track); 2323 } 2324 } 2325 } 2326 2327} 2328 2329// ---------------------------------------------------------------------------- 2330 2331AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2332 audio_io_handle_t id, audio_devices_t device, type_t type) 2333 : PlaybackThread(audioFlinger, output, id, device, type), 2334 // mAudioMixer below 2335 // mFastMixer below 2336 mFastMixerFutex(0) 2337 // mOutputSink below 2338 // mPipeSink below 2339 // mNormalSink below 2340{ 2341 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2342 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2343 "mFrameCount=%d, mNormalFrameCount=%d", 2344 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2345 mNormalFrameCount); 2346 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2347 2348 // FIXME - Current mixer implementation only supports stereo output 2349 if (mChannelCount != FCC_2) { 2350 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2351 } 2352 2353 // create an NBAIO sink for the HAL output stream, and negotiate 2354 mOutputSink = new AudioStreamOutSink(output->stream); 2355 size_t numCounterOffers = 0; 2356 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2357 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2358 ALOG_ASSERT(index == 0); 2359 2360 // initialize fast mixer depending on configuration 2361 bool initFastMixer; 2362 switch (kUseFastMixer) { 2363 case FastMixer_Never: 2364 initFastMixer = false; 2365 break; 2366 case FastMixer_Always: 2367 initFastMixer = true; 2368 break; 2369 case FastMixer_Static: 2370 case FastMixer_Dynamic: 2371 initFastMixer = mFrameCount < mNormalFrameCount; 2372 break; 2373 } 2374 if (initFastMixer) { 2375 2376 // create a MonoPipe to connect our submix to FastMixer 2377 NBAIO_Format format = mOutputSink->format(); 2378 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2379 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2380 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2381 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2382 const NBAIO_Format offers[1] = {format}; 2383 size_t numCounterOffers = 0; 2384 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2385 ALOG_ASSERT(index == 0); 2386 monoPipe->setAvgFrames((mScreenState & 1) ? 2387 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2388 mPipeSink = monoPipe; 2389 2390#ifdef TEE_SINK 2391 if (mTeeSinkOutputEnabled) { 2392 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2393 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2394 numCounterOffers = 0; 2395 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2396 ALOG_ASSERT(index == 0); 2397 mTeeSink = teeSink; 2398 PipeReader *teeSource = new PipeReader(*teeSink); 2399 numCounterOffers = 0; 2400 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2401 ALOG_ASSERT(index == 0); 2402 mTeeSource = teeSource; 2403 } 2404#endif 2405 2406 // create fast mixer and configure it initially with just one fast track for our submix 2407 mFastMixer = new FastMixer(); 2408 FastMixerStateQueue *sq = mFastMixer->sq(); 2409#ifdef STATE_QUEUE_DUMP 2410 sq->setObserverDump(&mStateQueueObserverDump); 2411 sq->setMutatorDump(&mStateQueueMutatorDump); 2412#endif 2413 FastMixerState *state = sq->begin(); 2414 FastTrack *fastTrack = &state->mFastTracks[0]; 2415 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2416 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2417 fastTrack->mVolumeProvider = NULL; 2418 fastTrack->mGeneration++; 2419 state->mFastTracksGen++; 2420 state->mTrackMask = 1; 2421 // fast mixer will use the HAL output sink 2422 state->mOutputSink = mOutputSink.get(); 2423 state->mOutputSinkGen++; 2424 state->mFrameCount = mFrameCount; 2425 state->mCommand = FastMixerState::COLD_IDLE; 2426 // already done in constructor initialization list 2427 //mFastMixerFutex = 0; 2428 state->mColdFutexAddr = &mFastMixerFutex; 2429 state->mColdGen++; 2430 state->mDumpState = &mFastMixerDumpState; 2431#ifdef TEE_SINK 2432 state->mTeeSink = mTeeSink.get(); 2433#endif 2434 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2435 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2436 sq->end(); 2437 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2438 2439 // start the fast mixer 2440 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2441 pid_t tid = mFastMixer->getTid(); 2442 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2443 if (err != 0) { 2444 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2445 kPriorityFastMixer, getpid_cached, tid, err); 2446 } 2447 2448#ifdef AUDIO_WATCHDOG 2449 // create and start the watchdog 2450 mAudioWatchdog = new AudioWatchdog(); 2451 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2452 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2453 tid = mAudioWatchdog->getTid(); 2454 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2455 if (err != 0) { 2456 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2457 kPriorityFastMixer, getpid_cached, tid, err); 2458 } 2459#endif 2460 2461 } else { 2462 mFastMixer = NULL; 2463 } 2464 2465 switch (kUseFastMixer) { 2466 case FastMixer_Never: 2467 case FastMixer_Dynamic: 2468 mNormalSink = mOutputSink; 2469 break; 2470 case FastMixer_Always: 2471 mNormalSink = mPipeSink; 2472 break; 2473 case FastMixer_Static: 2474 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2475 break; 2476 } 2477} 2478 2479AudioFlinger::MixerThread::~MixerThread() 2480{ 2481 if (mFastMixer != NULL) { 2482 FastMixerStateQueue *sq = mFastMixer->sq(); 2483 FastMixerState *state = sq->begin(); 2484 if (state->mCommand == FastMixerState::COLD_IDLE) { 2485 int32_t old = android_atomic_inc(&mFastMixerFutex); 2486 if (old == -1) { 2487 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2488 } 2489 } 2490 state->mCommand = FastMixerState::EXIT; 2491 sq->end(); 2492 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2493 mFastMixer->join(); 2494 // Though the fast mixer thread has exited, it's state queue is still valid. 2495 // We'll use that extract the final state which contains one remaining fast track 2496 // corresponding to our sub-mix. 2497 state = sq->begin(); 2498 ALOG_ASSERT(state->mTrackMask == 1); 2499 FastTrack *fastTrack = &state->mFastTracks[0]; 2500 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2501 delete fastTrack->mBufferProvider; 2502 sq->end(false /*didModify*/); 2503 delete mFastMixer; 2504#ifdef AUDIO_WATCHDOG 2505 if (mAudioWatchdog != 0) { 2506 mAudioWatchdog->requestExit(); 2507 mAudioWatchdog->requestExitAndWait(); 2508 mAudioWatchdog.clear(); 2509 } 2510#endif 2511 } 2512 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2513 delete mAudioMixer; 2514} 2515 2516 2517uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2518{ 2519 if (mFastMixer != NULL) { 2520 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2521 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2522 } 2523 return latency; 2524} 2525 2526 2527void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2528{ 2529 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2530} 2531 2532ssize_t AudioFlinger::MixerThread::threadLoop_write() 2533{ 2534 // FIXME we should only do one push per cycle; confirm this is true 2535 // Start the fast mixer if it's not already running 2536 if (mFastMixer != NULL) { 2537 FastMixerStateQueue *sq = mFastMixer->sq(); 2538 FastMixerState *state = sq->begin(); 2539 if (state->mCommand != FastMixerState::MIX_WRITE && 2540 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2541 if (state->mCommand == FastMixerState::COLD_IDLE) { 2542 int32_t old = android_atomic_inc(&mFastMixerFutex); 2543 if (old == -1) { 2544 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2545 } 2546#ifdef AUDIO_WATCHDOG 2547 if (mAudioWatchdog != 0) { 2548 mAudioWatchdog->resume(); 2549 } 2550#endif 2551 } 2552 state->mCommand = FastMixerState::MIX_WRITE; 2553 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2554 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2555 sq->end(); 2556 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2557 if (kUseFastMixer == FastMixer_Dynamic) { 2558 mNormalSink = mPipeSink; 2559 } 2560 } else { 2561 sq->end(false /*didModify*/); 2562 } 2563 } 2564 return PlaybackThread::threadLoop_write(); 2565} 2566 2567void AudioFlinger::MixerThread::threadLoop_standby() 2568{ 2569 // Idle the fast mixer if it's currently running 2570 if (mFastMixer != NULL) { 2571 FastMixerStateQueue *sq = mFastMixer->sq(); 2572 FastMixerState *state = sq->begin(); 2573 if (!(state->mCommand & FastMixerState::IDLE)) { 2574 state->mCommand = FastMixerState::COLD_IDLE; 2575 state->mColdFutexAddr = &mFastMixerFutex; 2576 state->mColdGen++; 2577 mFastMixerFutex = 0; 2578 sq->end(); 2579 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2580 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2581 if (kUseFastMixer == FastMixer_Dynamic) { 2582 mNormalSink = mOutputSink; 2583 } 2584#ifdef AUDIO_WATCHDOG 2585 if (mAudioWatchdog != 0) { 2586 mAudioWatchdog->pause(); 2587 } 2588#endif 2589 } else { 2590 sq->end(false /*didModify*/); 2591 } 2592 } 2593 PlaybackThread::threadLoop_standby(); 2594} 2595 2596// Empty implementation for standard mixer 2597// Overridden for offloaded playback 2598void AudioFlinger::PlaybackThread::flushOutput_l() 2599{ 2600} 2601 2602bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2603{ 2604 return false; 2605} 2606 2607bool AudioFlinger::PlaybackThread::shouldStandby_l() 2608{ 2609 return !mStandby; 2610} 2611 2612bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2613{ 2614 Mutex::Autolock _l(mLock); 2615 return waitingAsyncCallback_l(); 2616} 2617 2618// shared by MIXER and DIRECT, overridden by DUPLICATING 2619void AudioFlinger::PlaybackThread::threadLoop_standby() 2620{ 2621 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2622 mOutput->stream->common.standby(&mOutput->stream->common); 2623 if (mUseAsyncWrite != 0) { 2624 mWriteBlocked = false; 2625 mDraining = false; 2626 ALOG_ASSERT(mCallbackThread != 0); 2627 mCallbackThread->setWriteBlocked(false); 2628 mCallbackThread->setDraining(false); 2629 } 2630} 2631 2632void AudioFlinger::MixerThread::threadLoop_mix() 2633{ 2634 // obtain the presentation timestamp of the next output buffer 2635 int64_t pts; 2636 status_t status = INVALID_OPERATION; 2637 2638 if (mNormalSink != 0) { 2639 status = mNormalSink->getNextWriteTimestamp(&pts); 2640 } else { 2641 status = mOutputSink->getNextWriteTimestamp(&pts); 2642 } 2643 2644 if (status != NO_ERROR) { 2645 pts = AudioBufferProvider::kInvalidPTS; 2646 } 2647 2648 // mix buffers... 2649 mAudioMixer->process(pts); 2650 mCurrentWriteLength = mixBufferSize; 2651 // increase sleep time progressively when application underrun condition clears. 2652 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2653 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2654 // such that we would underrun the audio HAL. 2655 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2656 sleepTimeShift--; 2657 } 2658 sleepTime = 0; 2659 standbyTime = systemTime() + standbyDelay; 2660 //TODO: delay standby when effects have a tail 2661} 2662 2663void AudioFlinger::MixerThread::threadLoop_sleepTime() 2664{ 2665 // If no tracks are ready, sleep once for the duration of an output 2666 // buffer size, then write 0s to the output 2667 if (sleepTime == 0) { 2668 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2669 sleepTime = activeSleepTime >> sleepTimeShift; 2670 if (sleepTime < kMinThreadSleepTimeUs) { 2671 sleepTime = kMinThreadSleepTimeUs; 2672 } 2673 // reduce sleep time in case of consecutive application underruns to avoid 2674 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2675 // duration we would end up writing less data than needed by the audio HAL if 2676 // the condition persists. 2677 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2678 sleepTimeShift++; 2679 } 2680 } else { 2681 sleepTime = idleSleepTime; 2682 } 2683 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2684 memset(mMixBuffer, 0, mixBufferSize); 2685 sleepTime = 0; 2686 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2687 "anticipated start"); 2688 } 2689 // TODO add standby time extension fct of effect tail 2690} 2691 2692// prepareTracks_l() must be called with ThreadBase::mLock held 2693AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2694 Vector< sp<Track> > *tracksToRemove) 2695{ 2696 2697 mixer_state mixerStatus = MIXER_IDLE; 2698 // find out which tracks need to be processed 2699 size_t count = mActiveTracks.size(); 2700 size_t mixedTracks = 0; 2701 size_t tracksWithEffect = 0; 2702 // counts only _active_ fast tracks 2703 size_t fastTracks = 0; 2704 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2705 2706 float masterVolume = mMasterVolume; 2707 bool masterMute = mMasterMute; 2708 2709 if (masterMute) { 2710 masterVolume = 0; 2711 } 2712 // Delegate master volume control to effect in output mix effect chain if needed 2713 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2714 if (chain != 0) { 2715 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2716 chain->setVolume_l(&v, &v); 2717 masterVolume = (float)((v + (1 << 23)) >> 24); 2718 chain.clear(); 2719 } 2720 2721 // prepare a new state to push 2722 FastMixerStateQueue *sq = NULL; 2723 FastMixerState *state = NULL; 2724 bool didModify = false; 2725 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2726 if (mFastMixer != NULL) { 2727 sq = mFastMixer->sq(); 2728 state = sq->begin(); 2729 } 2730 2731 for (size_t i=0 ; i<count ; i++) { 2732 const sp<Track> t = mActiveTracks[i].promote(); 2733 if (t == 0) { 2734 continue; 2735 } 2736 2737 // this const just means the local variable doesn't change 2738 Track* const track = t.get(); 2739 2740 // process fast tracks 2741 if (track->isFastTrack()) { 2742 2743 // It's theoretically possible (though unlikely) for a fast track to be created 2744 // and then removed within the same normal mix cycle. This is not a problem, as 2745 // the track never becomes active so it's fast mixer slot is never touched. 2746 // The converse, of removing an (active) track and then creating a new track 2747 // at the identical fast mixer slot within the same normal mix cycle, 2748 // is impossible because the slot isn't marked available until the end of each cycle. 2749 int j = track->mFastIndex; 2750 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2751 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2752 FastTrack *fastTrack = &state->mFastTracks[j]; 2753 2754 // Determine whether the track is currently in underrun condition, 2755 // and whether it had a recent underrun. 2756 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2757 FastTrackUnderruns underruns = ftDump->mUnderruns; 2758 uint32_t recentFull = (underruns.mBitFields.mFull - 2759 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2760 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2761 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2762 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2763 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2764 uint32_t recentUnderruns = recentPartial + recentEmpty; 2765 track->mObservedUnderruns = underruns; 2766 // don't count underruns that occur while stopping or pausing 2767 // or stopped which can occur when flush() is called while active 2768 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2769 recentUnderruns > 0) { 2770 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2771 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2772 } 2773 2774 // This is similar to the state machine for normal tracks, 2775 // with a few modifications for fast tracks. 2776 bool isActive = true; 2777 switch (track->mState) { 2778 case TrackBase::STOPPING_1: 2779 // track stays active in STOPPING_1 state until first underrun 2780 if (recentUnderruns > 0 || track->isTerminated()) { 2781 track->mState = TrackBase::STOPPING_2; 2782 } 2783 break; 2784 case TrackBase::PAUSING: 2785 // ramp down is not yet implemented 2786 track->setPaused(); 2787 break; 2788 case TrackBase::RESUMING: 2789 // ramp up is not yet implemented 2790 track->mState = TrackBase::ACTIVE; 2791 break; 2792 case TrackBase::ACTIVE: 2793 if (recentFull > 0 || recentPartial > 0) { 2794 // track has provided at least some frames recently: reset retry count 2795 track->mRetryCount = kMaxTrackRetries; 2796 } 2797 if (recentUnderruns == 0) { 2798 // no recent underruns: stay active 2799 break; 2800 } 2801 // there has recently been an underrun of some kind 2802 if (track->sharedBuffer() == 0) { 2803 // were any of the recent underruns "empty" (no frames available)? 2804 if (recentEmpty == 0) { 2805 // no, then ignore the partial underruns as they are allowed indefinitely 2806 break; 2807 } 2808 // there has recently been an "empty" underrun: decrement the retry counter 2809 if (--(track->mRetryCount) > 0) { 2810 break; 2811 } 2812 // indicate to client process that the track was disabled because of underrun; 2813 // it will then automatically call start() when data is available 2814 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2815 // remove from active list, but state remains ACTIVE [confusing but true] 2816 isActive = false; 2817 break; 2818 } 2819 // fall through 2820 case TrackBase::STOPPING_2: 2821 case TrackBase::PAUSED: 2822 case TrackBase::STOPPED: 2823 case TrackBase::FLUSHED: // flush() while active 2824 // Check for presentation complete if track is inactive 2825 // We have consumed all the buffers of this track. 2826 // This would be incomplete if we auto-paused on underrun 2827 { 2828 size_t audioHALFrames = 2829 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2830 size_t framesWritten = mBytesWritten / mFrameSize; 2831 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2832 // track stays in active list until presentation is complete 2833 break; 2834 } 2835 } 2836 if (track->isStopping_2()) { 2837 track->mState = TrackBase::STOPPED; 2838 } 2839 if (track->isStopped()) { 2840 // Can't reset directly, as fast mixer is still polling this track 2841 // track->reset(); 2842 // So instead mark this track as needing to be reset after push with ack 2843 resetMask |= 1 << i; 2844 } 2845 isActive = false; 2846 break; 2847 case TrackBase::IDLE: 2848 default: 2849 LOG_FATAL("unexpected track state %d", track->mState); 2850 } 2851 2852 if (isActive) { 2853 // was it previously inactive? 2854 if (!(state->mTrackMask & (1 << j))) { 2855 ExtendedAudioBufferProvider *eabp = track; 2856 VolumeProvider *vp = track; 2857 fastTrack->mBufferProvider = eabp; 2858 fastTrack->mVolumeProvider = vp; 2859 fastTrack->mSampleRate = track->mSampleRate; 2860 fastTrack->mChannelMask = track->mChannelMask; 2861 fastTrack->mGeneration++; 2862 state->mTrackMask |= 1 << j; 2863 didModify = true; 2864 // no acknowledgement required for newly active tracks 2865 } 2866 // cache the combined master volume and stream type volume for fast mixer; this 2867 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2868 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2869 ++fastTracks; 2870 } else { 2871 // was it previously active? 2872 if (state->mTrackMask & (1 << j)) { 2873 fastTrack->mBufferProvider = NULL; 2874 fastTrack->mGeneration++; 2875 state->mTrackMask &= ~(1 << j); 2876 didModify = true; 2877 // If any fast tracks were removed, we must wait for acknowledgement 2878 // because we're about to decrement the last sp<> on those tracks. 2879 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2880 } else { 2881 LOG_FATAL("fast track %d should have been active", j); 2882 } 2883 tracksToRemove->add(track); 2884 // Avoids a misleading display in dumpsys 2885 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2886 } 2887 continue; 2888 } 2889 2890 { // local variable scope to avoid goto warning 2891 2892 audio_track_cblk_t* cblk = track->cblk(); 2893 2894 // The first time a track is added we wait 2895 // for all its buffers to be filled before processing it 2896 int name = track->name(); 2897 // make sure that we have enough frames to mix one full buffer. 2898 // enforce this condition only once to enable draining the buffer in case the client 2899 // app does not call stop() and relies on underrun to stop: 2900 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2901 // during last round 2902 size_t desiredFrames; 2903 uint32_t sr = track->sampleRate(); 2904 if (sr == mSampleRate) { 2905 desiredFrames = mNormalFrameCount; 2906 } else { 2907 // +1 for rounding and +1 for additional sample needed for interpolation 2908 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2909 // add frames already consumed but not yet released by the resampler 2910 // because mAudioTrackServerProxy->framesReady() will include these frames 2911 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2912 // the minimum track buffer size is normally twice the number of frames necessary 2913 // to fill one buffer and the resampler should not leave more than one buffer worth 2914 // of unreleased frames after each pass, but just in case... 2915 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2916 } 2917 uint32_t minFrames = 1; 2918 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2919 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2920 minFrames = desiredFrames; 2921 } 2922 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2923 size_t framesReady; 2924 if (track->sharedBuffer() == 0) { 2925 framesReady = track->framesReady(); 2926 } else if (track->isStopped()) { 2927 framesReady = 0; 2928 } else { 2929 framesReady = 1; 2930 } 2931 if ((framesReady >= minFrames) && track->isReady() && 2932 !track->isPaused() && !track->isTerminated()) 2933 { 2934 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2935 2936 mixedTracks++; 2937 2938 // track->mainBuffer() != mMixBuffer means there is an effect chain 2939 // connected to the track 2940 chain.clear(); 2941 if (track->mainBuffer() != mMixBuffer) { 2942 chain = getEffectChain_l(track->sessionId()); 2943 // Delegate volume control to effect in track effect chain if needed 2944 if (chain != 0) { 2945 tracksWithEffect++; 2946 } else { 2947 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2948 "session %d", 2949 name, track->sessionId()); 2950 } 2951 } 2952 2953 2954 int param = AudioMixer::VOLUME; 2955 if (track->mFillingUpStatus == Track::FS_FILLED) { 2956 // no ramp for the first volume setting 2957 track->mFillingUpStatus = Track::FS_ACTIVE; 2958 if (track->mState == TrackBase::RESUMING) { 2959 track->mState = TrackBase::ACTIVE; 2960 param = AudioMixer::RAMP_VOLUME; 2961 } 2962 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2963 // FIXME should not make a decision based on mServer 2964 } else if (cblk->mServer != 0) { 2965 // If the track is stopped before the first frame was mixed, 2966 // do not apply ramp 2967 param = AudioMixer::RAMP_VOLUME; 2968 } 2969 2970 // compute volume for this track 2971 uint32_t vl, vr, va; 2972 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2973 vl = vr = va = 0; 2974 if (track->isPausing()) { 2975 track->setPaused(); 2976 } 2977 } else { 2978 2979 // read original volumes with volume control 2980 float typeVolume = mStreamTypes[track->streamType()].volume; 2981 float v = masterVolume * typeVolume; 2982 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2983 uint32_t vlr = proxy->getVolumeLR(); 2984 vl = vlr & 0xFFFF; 2985 vr = vlr >> 16; 2986 // track volumes come from shared memory, so can't be trusted and must be clamped 2987 if (vl > MAX_GAIN_INT) { 2988 ALOGV("Track left volume out of range: %04X", vl); 2989 vl = MAX_GAIN_INT; 2990 } 2991 if (vr > MAX_GAIN_INT) { 2992 ALOGV("Track right volume out of range: %04X", vr); 2993 vr = MAX_GAIN_INT; 2994 } 2995 // now apply the master volume and stream type volume 2996 vl = (uint32_t)(v * vl) << 12; 2997 vr = (uint32_t)(v * vr) << 12; 2998 // assuming master volume and stream type volume each go up to 1.0, 2999 // vl and vr are now in 8.24 format 3000 3001 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3002 // send level comes from shared memory and so may be corrupt 3003 if (sendLevel > MAX_GAIN_INT) { 3004 ALOGV("Track send level out of range: %04X", sendLevel); 3005 sendLevel = MAX_GAIN_INT; 3006 } 3007 va = (uint32_t)(v * sendLevel); 3008 } 3009 3010 // Delegate volume control to effect in track effect chain if needed 3011 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3012 // Do not ramp volume if volume is controlled by effect 3013 param = AudioMixer::VOLUME; 3014 track->mHasVolumeController = true; 3015 } else { 3016 // force no volume ramp when volume controller was just disabled or removed 3017 // from effect chain to avoid volume spike 3018 if (track->mHasVolumeController) { 3019 param = AudioMixer::VOLUME; 3020 } 3021 track->mHasVolumeController = false; 3022 } 3023 3024 // Convert volumes from 8.24 to 4.12 format 3025 // This additional clamping is needed in case chain->setVolume_l() overshot 3026 vl = (vl + (1 << 11)) >> 12; 3027 if (vl > MAX_GAIN_INT) { 3028 vl = MAX_GAIN_INT; 3029 } 3030 vr = (vr + (1 << 11)) >> 12; 3031 if (vr > MAX_GAIN_INT) { 3032 vr = MAX_GAIN_INT; 3033 } 3034 3035 if (va > MAX_GAIN_INT) { 3036 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3037 } 3038 3039 // XXX: these things DON'T need to be done each time 3040 mAudioMixer->setBufferProvider(name, track); 3041 mAudioMixer->enable(name); 3042 3043 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3044 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3045 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3046 mAudioMixer->setParameter( 3047 name, 3048 AudioMixer::TRACK, 3049 AudioMixer::FORMAT, (void *)track->format()); 3050 mAudioMixer->setParameter( 3051 name, 3052 AudioMixer::TRACK, 3053 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3054 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3055 uint32_t maxSampleRate = mSampleRate * 2; 3056 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3057 if (reqSampleRate == 0) { 3058 reqSampleRate = mSampleRate; 3059 } else if (reqSampleRate > maxSampleRate) { 3060 reqSampleRate = maxSampleRate; 3061 } 3062 mAudioMixer->setParameter( 3063 name, 3064 AudioMixer::RESAMPLE, 3065 AudioMixer::SAMPLE_RATE, 3066 (void *)reqSampleRate); 3067 mAudioMixer->setParameter( 3068 name, 3069 AudioMixer::TRACK, 3070 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3071 mAudioMixer->setParameter( 3072 name, 3073 AudioMixer::TRACK, 3074 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3075 3076 // reset retry count 3077 track->mRetryCount = kMaxTrackRetries; 3078 3079 // If one track is ready, set the mixer ready if: 3080 // - the mixer was not ready during previous round OR 3081 // - no other track is not ready 3082 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3083 mixerStatus != MIXER_TRACKS_ENABLED) { 3084 mixerStatus = MIXER_TRACKS_READY; 3085 } 3086 } else { 3087 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3088 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3089 } 3090 // clear effect chain input buffer if an active track underruns to avoid sending 3091 // previous audio buffer again to effects 3092 chain = getEffectChain_l(track->sessionId()); 3093 if (chain != 0) { 3094 chain->clearInputBuffer(); 3095 } 3096 3097 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3098 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3099 track->isStopped() || track->isPaused()) { 3100 // We have consumed all the buffers of this track. 3101 // Remove it from the list of active tracks. 3102 // TODO: use actual buffer filling status instead of latency when available from 3103 // audio HAL 3104 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3105 size_t framesWritten = mBytesWritten / mFrameSize; 3106 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3107 if (track->isStopped()) { 3108 track->reset(); 3109 } 3110 tracksToRemove->add(track); 3111 } 3112 } else { 3113 // No buffers for this track. Give it a few chances to 3114 // fill a buffer, then remove it from active list. 3115 if (--(track->mRetryCount) <= 0) { 3116 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3117 tracksToRemove->add(track); 3118 // indicate to client process that the track was disabled because of underrun; 3119 // it will then automatically call start() when data is available 3120 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3121 // If one track is not ready, mark the mixer also not ready if: 3122 // - the mixer was ready during previous round OR 3123 // - no other track is ready 3124 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3125 mixerStatus != MIXER_TRACKS_READY) { 3126 mixerStatus = MIXER_TRACKS_ENABLED; 3127 } 3128 } 3129 mAudioMixer->disable(name); 3130 } 3131 3132 } // local variable scope to avoid goto warning 3133track_is_ready: ; 3134 3135 } 3136 3137 // Push the new FastMixer state if necessary 3138 bool pauseAudioWatchdog = false; 3139 if (didModify) { 3140 state->mFastTracksGen++; 3141 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3142 if (kUseFastMixer == FastMixer_Dynamic && 3143 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3144 state->mCommand = FastMixerState::COLD_IDLE; 3145 state->mColdFutexAddr = &mFastMixerFutex; 3146 state->mColdGen++; 3147 mFastMixerFutex = 0; 3148 if (kUseFastMixer == FastMixer_Dynamic) { 3149 mNormalSink = mOutputSink; 3150 } 3151 // If we go into cold idle, need to wait for acknowledgement 3152 // so that fast mixer stops doing I/O. 3153 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3154 pauseAudioWatchdog = true; 3155 } 3156 } 3157 if (sq != NULL) { 3158 sq->end(didModify); 3159 sq->push(block); 3160 } 3161#ifdef AUDIO_WATCHDOG 3162 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3163 mAudioWatchdog->pause(); 3164 } 3165#endif 3166 3167 // Now perform the deferred reset on fast tracks that have stopped 3168 while (resetMask != 0) { 3169 size_t i = __builtin_ctz(resetMask); 3170 ALOG_ASSERT(i < count); 3171 resetMask &= ~(1 << i); 3172 sp<Track> t = mActiveTracks[i].promote(); 3173 if (t == 0) { 3174 continue; 3175 } 3176 Track* track = t.get(); 3177 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3178 track->reset(); 3179 } 3180 3181 // remove all the tracks that need to be... 3182 removeTracks_l(*tracksToRemove); 3183 3184 // mix buffer must be cleared if all tracks are connected to an 3185 // effect chain as in this case the mixer will not write to 3186 // mix buffer and track effects will accumulate into it 3187 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3188 (mixedTracks == 0 && fastTracks > 0))) { 3189 // FIXME as a performance optimization, should remember previous zero status 3190 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3191 } 3192 3193 // if any fast tracks, then status is ready 3194 mMixerStatusIgnoringFastTracks = mixerStatus; 3195 if (fastTracks > 0) { 3196 mixerStatus = MIXER_TRACKS_READY; 3197 } 3198 return mixerStatus; 3199} 3200 3201// getTrackName_l() must be called with ThreadBase::mLock held 3202int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3203{ 3204 return mAudioMixer->getTrackName(channelMask, sessionId); 3205} 3206 3207// deleteTrackName_l() must be called with ThreadBase::mLock held 3208void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3209{ 3210 ALOGV("remove track (%d) and delete from mixer", name); 3211 mAudioMixer->deleteTrackName(name); 3212} 3213 3214// checkForNewParameters_l() must be called with ThreadBase::mLock held 3215bool AudioFlinger::MixerThread::checkForNewParameters_l() 3216{ 3217 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3218 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3219 bool reconfig = false; 3220 3221 while (!mNewParameters.isEmpty()) { 3222 3223 if (mFastMixer != NULL) { 3224 FastMixerStateQueue *sq = mFastMixer->sq(); 3225 FastMixerState *state = sq->begin(); 3226 if (!(state->mCommand & FastMixerState::IDLE)) { 3227 previousCommand = state->mCommand; 3228 state->mCommand = FastMixerState::HOT_IDLE; 3229 sq->end(); 3230 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3231 } else { 3232 sq->end(false /*didModify*/); 3233 } 3234 } 3235 3236 status_t status = NO_ERROR; 3237 String8 keyValuePair = mNewParameters[0]; 3238 AudioParameter param = AudioParameter(keyValuePair); 3239 int value; 3240 3241 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3242 reconfig = true; 3243 } 3244 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3245 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3246 status = BAD_VALUE; 3247 } else { 3248 // no need to save value, since it's constant 3249 reconfig = true; 3250 } 3251 } 3252 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3253 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3254 status = BAD_VALUE; 3255 } else { 3256 // no need to save value, since it's constant 3257 reconfig = true; 3258 } 3259 } 3260 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3261 // do not accept frame count changes if tracks are open as the track buffer 3262 // size depends on frame count and correct behavior would not be guaranteed 3263 // if frame count is changed after track creation 3264 if (!mTracks.isEmpty()) { 3265 status = INVALID_OPERATION; 3266 } else { 3267 reconfig = true; 3268 } 3269 } 3270 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3271#ifdef ADD_BATTERY_DATA 3272 // when changing the audio output device, call addBatteryData to notify 3273 // the change 3274 if (mOutDevice != value) { 3275 uint32_t params = 0; 3276 // check whether speaker is on 3277 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3278 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3279 } 3280 3281 audio_devices_t deviceWithoutSpeaker 3282 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3283 // check if any other device (except speaker) is on 3284 if (value & deviceWithoutSpeaker ) { 3285 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3286 } 3287 3288 if (params != 0) { 3289 addBatteryData(params); 3290 } 3291 } 3292#endif 3293 3294 // forward device change to effects that have requested to be 3295 // aware of attached audio device. 3296 if (value != AUDIO_DEVICE_NONE) { 3297 mOutDevice = value; 3298 for (size_t i = 0; i < mEffectChains.size(); i++) { 3299 mEffectChains[i]->setDevice_l(mOutDevice); 3300 } 3301 } 3302 } 3303 3304 if (status == NO_ERROR) { 3305 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3306 keyValuePair.string()); 3307 if (!mStandby && status == INVALID_OPERATION) { 3308 mOutput->stream->common.standby(&mOutput->stream->common); 3309 mStandby = true; 3310 mBytesWritten = 0; 3311 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3312 keyValuePair.string()); 3313 } 3314 if (status == NO_ERROR && reconfig) { 3315 readOutputParameters(); 3316 delete mAudioMixer; 3317 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3318 for (size_t i = 0; i < mTracks.size() ; i++) { 3319 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3320 if (name < 0) { 3321 break; 3322 } 3323 mTracks[i]->mName = name; 3324 } 3325 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3326 } 3327 } 3328 3329 mNewParameters.removeAt(0); 3330 3331 mParamStatus = status; 3332 mParamCond.signal(); 3333 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3334 // already timed out waiting for the status and will never signal the condition. 3335 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3336 } 3337 3338 if (!(previousCommand & FastMixerState::IDLE)) { 3339 ALOG_ASSERT(mFastMixer != NULL); 3340 FastMixerStateQueue *sq = mFastMixer->sq(); 3341 FastMixerState *state = sq->begin(); 3342 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3343 state->mCommand = previousCommand; 3344 sq->end(); 3345 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3346 } 3347 3348 return reconfig; 3349} 3350 3351 3352void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3353{ 3354 const size_t SIZE = 256; 3355 char buffer[SIZE]; 3356 String8 result; 3357 3358 PlaybackThread::dumpInternals(fd, args); 3359 3360 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3361 result.append(buffer); 3362 write(fd, result.string(), result.size()); 3363 3364 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3365 const FastMixerDumpState copy(mFastMixerDumpState); 3366 copy.dump(fd); 3367 3368#ifdef STATE_QUEUE_DUMP 3369 // Similar for state queue 3370 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3371 observerCopy.dump(fd); 3372 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3373 mutatorCopy.dump(fd); 3374#endif 3375 3376#ifdef TEE_SINK 3377 // Write the tee output to a .wav file 3378 dumpTee(fd, mTeeSource, mId); 3379#endif 3380 3381#ifdef AUDIO_WATCHDOG 3382 if (mAudioWatchdog != 0) { 3383 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3384 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3385 wdCopy.dump(fd); 3386 } 3387#endif 3388} 3389 3390uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3391{ 3392 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3393} 3394 3395uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3396{ 3397 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3398} 3399 3400void AudioFlinger::MixerThread::cacheParameters_l() 3401{ 3402 PlaybackThread::cacheParameters_l(); 3403 3404 // FIXME: Relaxed timing because of a certain device that can't meet latency 3405 // Should be reduced to 2x after the vendor fixes the driver issue 3406 // increase threshold again due to low power audio mode. The way this warning 3407 // threshold is calculated and its usefulness should be reconsidered anyway. 3408 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3409} 3410 3411// ---------------------------------------------------------------------------- 3412 3413AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3414 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3415 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3416 // mLeftVolFloat, mRightVolFloat 3417{ 3418} 3419 3420AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3421 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3422 ThreadBase::type_t type) 3423 : PlaybackThread(audioFlinger, output, id, device, type) 3424 // mLeftVolFloat, mRightVolFloat 3425{ 3426} 3427 3428AudioFlinger::DirectOutputThread::~DirectOutputThread() 3429{ 3430} 3431 3432void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3433{ 3434 audio_track_cblk_t* cblk = track->cblk(); 3435 float left, right; 3436 3437 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3438 left = right = 0; 3439 } else { 3440 float typeVolume = mStreamTypes[track->streamType()].volume; 3441 float v = mMasterVolume * typeVolume; 3442 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3443 uint32_t vlr = proxy->getVolumeLR(); 3444 float v_clamped = v * (vlr & 0xFFFF); 3445 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3446 left = v_clamped/MAX_GAIN; 3447 v_clamped = v * (vlr >> 16); 3448 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3449 right = v_clamped/MAX_GAIN; 3450 } 3451 3452 if (lastTrack) { 3453 if (left != mLeftVolFloat || right != mRightVolFloat) { 3454 mLeftVolFloat = left; 3455 mRightVolFloat = right; 3456 3457 // Convert volumes from float to 8.24 3458 uint32_t vl = (uint32_t)(left * (1 << 24)); 3459 uint32_t vr = (uint32_t)(right * (1 << 24)); 3460 3461 // Delegate volume control to effect in track effect chain if needed 3462 // only one effect chain can be present on DirectOutputThread, so if 3463 // there is one, the track is connected to it 3464 if (!mEffectChains.isEmpty()) { 3465 mEffectChains[0]->setVolume_l(&vl, &vr); 3466 left = (float)vl / (1 << 24); 3467 right = (float)vr / (1 << 24); 3468 } 3469 if (mOutput->stream->set_volume) { 3470 mOutput->stream->set_volume(mOutput->stream, left, right); 3471 } 3472 } 3473 } 3474} 3475 3476 3477AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3478 Vector< sp<Track> > *tracksToRemove 3479) 3480{ 3481 size_t count = mActiveTracks.size(); 3482 mixer_state mixerStatus = MIXER_IDLE; 3483 3484 // find out which tracks need to be processed 3485 for (size_t i = 0; i < count; i++) { 3486 sp<Track> t = mActiveTracks[i].promote(); 3487 // The track died recently 3488 if (t == 0) { 3489 continue; 3490 } 3491 3492 Track* const track = t.get(); 3493 audio_track_cblk_t* cblk = track->cblk(); 3494 3495 // The first time a track is added we wait 3496 // for all its buffers to be filled before processing it 3497 uint32_t minFrames; 3498 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3499 minFrames = mNormalFrameCount; 3500 } else { 3501 minFrames = 1; 3502 } 3503 // Only consider last track started for volume and mixer state control. 3504 // This is the last entry in mActiveTracks unless a track underruns. 3505 // As we only care about the transition phase between two tracks on a 3506 // direct output, it is not a problem to ignore the underrun case. 3507 bool last = (i == (count - 1)); 3508 3509 if ((track->framesReady() >= minFrames) && track->isReady() && 3510 !track->isPaused() && !track->isTerminated()) 3511 { 3512 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3513 3514 if (track->mFillingUpStatus == Track::FS_FILLED) { 3515 track->mFillingUpStatus = Track::FS_ACTIVE; 3516 mLeftVolFloat = mRightVolFloat = 0; 3517 if (track->mState == TrackBase::RESUMING) { 3518 track->mState = TrackBase::ACTIVE; 3519 } 3520 } 3521 3522 // compute volume for this track 3523 processVolume_l(track, last); 3524 if (last) { 3525 // reset retry count 3526 track->mRetryCount = kMaxTrackRetriesDirect; 3527 mActiveTrack = t; 3528 mixerStatus = MIXER_TRACKS_READY; 3529 } 3530 } else { 3531 // clear effect chain input buffer if the last active track started underruns 3532 // to avoid sending previous audio buffer again to effects 3533 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3534 mEffectChains[0]->clearInputBuffer(); 3535 } 3536 3537 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3538 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3539 track->isStopped() || track->isPaused()) { 3540 // We have consumed all the buffers of this track. 3541 // Remove it from the list of active tracks. 3542 // TODO: implement behavior for compressed audio 3543 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3544 size_t framesWritten = mBytesWritten / mFrameSize; 3545 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3546 if (track->isStopped()) { 3547 track->reset(); 3548 } 3549 tracksToRemove->add(track); 3550 } 3551 } else { 3552 // No buffers for this track. Give it a few chances to 3553 // fill a buffer, then remove it from active list. 3554 // Only consider last track started for mixer state control 3555 if (--(track->mRetryCount) <= 0) { 3556 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3557 tracksToRemove->add(track); 3558 } else if (last) { 3559 mixerStatus = MIXER_TRACKS_ENABLED; 3560 } 3561 } 3562 } 3563 } 3564 3565 // remove all the tracks that need to be... 3566 removeTracks_l(*tracksToRemove); 3567 3568 return mixerStatus; 3569} 3570 3571void AudioFlinger::DirectOutputThread::threadLoop_mix() 3572{ 3573 size_t frameCount = mFrameCount; 3574 int8_t *curBuf = (int8_t *)mMixBuffer; 3575 // output audio to hardware 3576 while (frameCount) { 3577 AudioBufferProvider::Buffer buffer; 3578 buffer.frameCount = frameCount; 3579 mActiveTrack->getNextBuffer(&buffer); 3580 if (buffer.raw == NULL) { 3581 memset(curBuf, 0, frameCount * mFrameSize); 3582 break; 3583 } 3584 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3585 frameCount -= buffer.frameCount; 3586 curBuf += buffer.frameCount * mFrameSize; 3587 mActiveTrack->releaseBuffer(&buffer); 3588 } 3589 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3590 sleepTime = 0; 3591 standbyTime = systemTime() + standbyDelay; 3592 mActiveTrack.clear(); 3593} 3594 3595void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3596{ 3597 if (sleepTime == 0) { 3598 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3599 sleepTime = activeSleepTime; 3600 } else { 3601 sleepTime = idleSleepTime; 3602 } 3603 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3604 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3605 sleepTime = 0; 3606 } 3607} 3608 3609// getTrackName_l() must be called with ThreadBase::mLock held 3610int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3611 int sessionId) 3612{ 3613 return 0; 3614} 3615 3616// deleteTrackName_l() must be called with ThreadBase::mLock held 3617void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3618{ 3619} 3620 3621// checkForNewParameters_l() must be called with ThreadBase::mLock held 3622bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3623{ 3624 bool reconfig = false; 3625 3626 while (!mNewParameters.isEmpty()) { 3627 status_t status = NO_ERROR; 3628 String8 keyValuePair = mNewParameters[0]; 3629 AudioParameter param = AudioParameter(keyValuePair); 3630 int value; 3631 3632 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3633 // do not accept frame count changes if tracks are open as the track buffer 3634 // size depends on frame count and correct behavior would not be garantied 3635 // if frame count is changed after track creation 3636 if (!mTracks.isEmpty()) { 3637 status = INVALID_OPERATION; 3638 } else { 3639 reconfig = true; 3640 } 3641 } 3642 if (status == NO_ERROR) { 3643 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3644 keyValuePair.string()); 3645 if (!mStandby && status == INVALID_OPERATION) { 3646 mOutput->stream->common.standby(&mOutput->stream->common); 3647 mStandby = true; 3648 mBytesWritten = 0; 3649 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3650 keyValuePair.string()); 3651 } 3652 if (status == NO_ERROR && reconfig) { 3653 readOutputParameters(); 3654 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3655 } 3656 } 3657 3658 mNewParameters.removeAt(0); 3659 3660 mParamStatus = status; 3661 mParamCond.signal(); 3662 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3663 // already timed out waiting for the status and will never signal the condition. 3664 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3665 } 3666 return reconfig; 3667} 3668 3669uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3670{ 3671 uint32_t time; 3672 if (audio_is_linear_pcm(mFormat)) { 3673 time = PlaybackThread::activeSleepTimeUs(); 3674 } else { 3675 time = 10000; 3676 } 3677 return time; 3678} 3679 3680uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3681{ 3682 uint32_t time; 3683 if (audio_is_linear_pcm(mFormat)) { 3684 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3685 } else { 3686 time = 10000; 3687 } 3688 return time; 3689} 3690 3691uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3692{ 3693 uint32_t time; 3694 if (audio_is_linear_pcm(mFormat)) { 3695 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3696 } else { 3697 time = 10000; 3698 } 3699 return time; 3700} 3701 3702void AudioFlinger::DirectOutputThread::cacheParameters_l() 3703{ 3704 PlaybackThread::cacheParameters_l(); 3705 3706 // use shorter standby delay as on normal output to release 3707 // hardware resources as soon as possible 3708 standbyDelay = microseconds(activeSleepTime*2); 3709} 3710 3711// ---------------------------------------------------------------------------- 3712 3713AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3714 const sp<AudioFlinger::OffloadThread>& offloadThread) 3715 : Thread(false /*canCallJava*/), 3716 mOffloadThread(offloadThread), 3717 mWriteBlocked(false), 3718 mDraining(false) 3719{ 3720} 3721 3722AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3723{ 3724} 3725 3726void AudioFlinger::AsyncCallbackThread::onFirstRef() 3727{ 3728 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3729} 3730 3731bool AudioFlinger::AsyncCallbackThread::threadLoop() 3732{ 3733 while (!exitPending()) { 3734 bool writeBlocked; 3735 bool draining; 3736 3737 { 3738 Mutex::Autolock _l(mLock); 3739 mWaitWorkCV.wait(mLock); 3740 if (exitPending()) { 3741 break; 3742 } 3743 writeBlocked = mWriteBlocked; 3744 draining = mDraining; 3745 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3746 } 3747 { 3748 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3749 if (offloadThread != 0) { 3750 if (writeBlocked == false) { 3751 offloadThread->setWriteBlocked(false); 3752 } 3753 if (draining == false) { 3754 offloadThread->setDraining(false); 3755 } 3756 } 3757 } 3758 } 3759 return false; 3760} 3761 3762void AudioFlinger::AsyncCallbackThread::exit() 3763{ 3764 ALOGV("AsyncCallbackThread::exit"); 3765 Mutex::Autolock _l(mLock); 3766 requestExit(); 3767 mWaitWorkCV.broadcast(); 3768} 3769 3770void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3771{ 3772 Mutex::Autolock _l(mLock); 3773 mWriteBlocked = value; 3774 if (!value) { 3775 mWaitWorkCV.signal(); 3776 } 3777} 3778 3779void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3780{ 3781 Mutex::Autolock _l(mLock); 3782 mDraining = value; 3783 if (!value) { 3784 mWaitWorkCV.signal(); 3785 } 3786} 3787 3788 3789// ---------------------------------------------------------------------------- 3790AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3791 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3792 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3793 mHwPaused(false), 3794 mPausedBytesRemaining(0) 3795{ 3796 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3797} 3798 3799AudioFlinger::OffloadThread::~OffloadThread() 3800{ 3801 mPreviousTrack.clear(); 3802} 3803 3804void AudioFlinger::OffloadThread::threadLoop_exit() 3805{ 3806 if (mFlushPending || mHwPaused) { 3807 // If a flush is pending or track was paused, just discard buffered data 3808 flushHw_l(); 3809 } else { 3810 mMixerStatus = MIXER_DRAIN_ALL; 3811 threadLoop_drain(); 3812 } 3813 mCallbackThread->exit(); 3814 PlaybackThread::threadLoop_exit(); 3815} 3816 3817AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3818 Vector< sp<Track> > *tracksToRemove 3819) 3820{ 3821 ALOGV("OffloadThread::prepareTracks_l"); 3822 size_t count = mActiveTracks.size(); 3823 3824 mixer_state mixerStatus = MIXER_IDLE; 3825 // find out which tracks need to be processed 3826 for (size_t i = 0; i < count; i++) { 3827 sp<Track> t = mActiveTracks[i].promote(); 3828 // The track died recently 3829 if (t == 0) { 3830 continue; 3831 } 3832 Track* const track = t.get(); 3833 audio_track_cblk_t* cblk = track->cblk(); 3834 if (mPreviousTrack != NULL) { 3835 if (t != mPreviousTrack) { 3836 // Flush any data still being written from last track 3837 mBytesRemaining = 0; 3838 if (mPausedBytesRemaining) { 3839 // Last track was paused so we also need to flush saved 3840 // mixbuffer state and invalidate track so that it will 3841 // re-submit that unwritten data when it is next resumed 3842 mPausedBytesRemaining = 0; 3843 // Invalidate is a bit drastic - would be more efficient 3844 // to have a flag to tell client that some of the 3845 // previously written data was lost 3846 mPreviousTrack->invalidate(); 3847 } 3848 } 3849 } 3850 mPreviousTrack = t; 3851 bool last = (i == (count - 1)); 3852 if (track->isPausing()) { 3853 track->setPaused(); 3854 if (last) { 3855 if (!mHwPaused) { 3856 mOutput->stream->pause(mOutput->stream); 3857 mHwPaused = true; 3858 } 3859 // If we were part way through writing the mixbuffer to 3860 // the HAL we must save this until we resume 3861 // BUG - this will be wrong if a different track is made active, 3862 // in that case we want to discard the pending data in the 3863 // mixbuffer and tell the client to present it again when the 3864 // track is resumed 3865 mPausedWriteLength = mCurrentWriteLength; 3866 mPausedBytesRemaining = mBytesRemaining; 3867 mBytesRemaining = 0; // stop writing 3868 } 3869 tracksToRemove->add(track); 3870 } else if (track->framesReady() && track->isReady() && 3871 !track->isPaused() && !track->isTerminated()) { 3872 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3873 if (track->mFillingUpStatus == Track::FS_FILLED) { 3874 track->mFillingUpStatus = Track::FS_ACTIVE; 3875 mLeftVolFloat = mRightVolFloat = 0; 3876 if (track->mState == TrackBase::RESUMING) { 3877 if (mPausedBytesRemaining) { 3878 // Need to continue write that was interrupted 3879 mCurrentWriteLength = mPausedWriteLength; 3880 mBytesRemaining = mPausedBytesRemaining; 3881 mPausedBytesRemaining = 0; 3882 } 3883 track->mState = TrackBase::ACTIVE; 3884 } 3885 } 3886 3887 if (last) { 3888 if (mHwPaused) { 3889 mOutput->stream->resume(mOutput->stream); 3890 mHwPaused = false; 3891 // threadLoop_mix() will handle the case that we need to 3892 // resume an interrupted write 3893 } 3894 // reset retry count 3895 track->mRetryCount = kMaxTrackRetriesOffload; 3896 mActiveTrack = t; 3897 mixerStatus = MIXER_TRACKS_READY; 3898 } 3899 } else { 3900 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3901 if (track->isStopping_1()) { 3902 // Hardware buffer can hold a large amount of audio so we must 3903 // wait for all current track's data to drain before we say 3904 // that the track is stopped. 3905 if (mBytesRemaining == 0) { 3906 // Only start draining when all data in mixbuffer 3907 // has been written 3908 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3909 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3910 sleepTime = 0; 3911 standbyTime = systemTime() + standbyDelay; 3912 if (last) { 3913 mixerStatus = MIXER_DRAIN_TRACK; 3914 if (mHwPaused) { 3915 // It is possible to move from PAUSED to STOPPING_1 without 3916 // a resume so we must ensure hardware is running 3917 mOutput->stream->resume(mOutput->stream); 3918 mHwPaused = false; 3919 } 3920 } 3921 } 3922 } else if (track->isStopping_2()) { 3923 // Drain has completed, signal presentation complete 3924 if (!mDraining || !last) { 3925 track->mState = TrackBase::STOPPED; 3926 size_t audioHALFrames = 3927 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3928 size_t framesWritten = 3929 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3930 track->presentationComplete(framesWritten, audioHALFrames); 3931 track->reset(); 3932 tracksToRemove->add(track); 3933 } 3934 } else { 3935 // No buffers for this track. Give it a few chances to 3936 // fill a buffer, then remove it from active list. 3937 if (--(track->mRetryCount) <= 0) { 3938 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3939 track->name()); 3940 tracksToRemove->add(track); 3941 } else if (last){ 3942 mixerStatus = MIXER_TRACKS_ENABLED; 3943 } 3944 } 3945 } 3946 // compute volume for this track 3947 processVolume_l(track, last); 3948 } 3949 3950 if (mFlushPending) { 3951 flushHw_l(); 3952 mFlushPending = false; 3953 } 3954 3955 // remove all the tracks that need to be... 3956 removeTracks_l(*tracksToRemove); 3957 3958 return mixerStatus; 3959} 3960 3961void AudioFlinger::OffloadThread::flushOutput_l() 3962{ 3963 mFlushPending = true; 3964} 3965 3966// must be called with thread mutex locked 3967bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3968{ 3969 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3970 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3971 return true; 3972 } 3973 return false; 3974} 3975 3976// must be called with thread mutex locked 3977bool AudioFlinger::OffloadThread::shouldStandby_l() 3978{ 3979 bool TrackPaused = false; 3980 3981 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3982 // after a timeout and we will enter standby then. 3983 if (mTracks.size() > 0) { 3984 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3985 } 3986 3987 return !mStandby && !TrackPaused; 3988} 3989 3990 3991bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3992{ 3993 Mutex::Autolock _l(mLock); 3994 return waitingAsyncCallback_l(); 3995} 3996 3997void AudioFlinger::OffloadThread::flushHw_l() 3998{ 3999 mOutput->stream->flush(mOutput->stream); 4000 // Flush anything still waiting in the mixbuffer 4001 mCurrentWriteLength = 0; 4002 mBytesRemaining = 0; 4003 mPausedWriteLength = 0; 4004 mPausedBytesRemaining = 0; 4005 if (mUseAsyncWrite) { 4006 mWriteBlocked = false; 4007 mDraining = false; 4008 ALOG_ASSERT(mCallbackThread != 0); 4009 mCallbackThread->setWriteBlocked(false); 4010 mCallbackThread->setDraining(false); 4011 } 4012} 4013 4014// ---------------------------------------------------------------------------- 4015 4016AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4017 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4018 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4019 DUPLICATING), 4020 mWaitTimeMs(UINT_MAX) 4021{ 4022 addOutputTrack(mainThread); 4023} 4024 4025AudioFlinger::DuplicatingThread::~DuplicatingThread() 4026{ 4027 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4028 mOutputTracks[i]->destroy(); 4029 } 4030} 4031 4032void AudioFlinger::DuplicatingThread::threadLoop_mix() 4033{ 4034 // mix buffers... 4035 if (outputsReady(outputTracks)) { 4036 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4037 } else { 4038 memset(mMixBuffer, 0, mixBufferSize); 4039 } 4040 sleepTime = 0; 4041 writeFrames = mNormalFrameCount; 4042 mCurrentWriteLength = mixBufferSize; 4043 standbyTime = systemTime() + standbyDelay; 4044} 4045 4046void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4047{ 4048 if (sleepTime == 0) { 4049 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4050 sleepTime = activeSleepTime; 4051 } else { 4052 sleepTime = idleSleepTime; 4053 } 4054 } else if (mBytesWritten != 0) { 4055 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4056 writeFrames = mNormalFrameCount; 4057 memset(mMixBuffer, 0, mixBufferSize); 4058 } else { 4059 // flush remaining overflow buffers in output tracks 4060 writeFrames = 0; 4061 } 4062 sleepTime = 0; 4063 } 4064} 4065 4066ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4067{ 4068 for (size_t i = 0; i < outputTracks.size(); i++) { 4069 outputTracks[i]->write(mMixBuffer, writeFrames); 4070 } 4071 return (ssize_t)mixBufferSize; 4072} 4073 4074void AudioFlinger::DuplicatingThread::threadLoop_standby() 4075{ 4076 // DuplicatingThread implements standby by stopping all tracks 4077 for (size_t i = 0; i < outputTracks.size(); i++) { 4078 outputTracks[i]->stop(); 4079 } 4080} 4081 4082void AudioFlinger::DuplicatingThread::saveOutputTracks() 4083{ 4084 outputTracks = mOutputTracks; 4085} 4086 4087void AudioFlinger::DuplicatingThread::clearOutputTracks() 4088{ 4089 outputTracks.clear(); 4090} 4091 4092void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4093{ 4094 Mutex::Autolock _l(mLock); 4095 // FIXME explain this formula 4096 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4097 OutputTrack *outputTrack = new OutputTrack(thread, 4098 this, 4099 mSampleRate, 4100 mFormat, 4101 mChannelMask, 4102 frameCount); 4103 if (outputTrack->cblk() != NULL) { 4104 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4105 mOutputTracks.add(outputTrack); 4106 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4107 updateWaitTime_l(); 4108 } 4109} 4110 4111void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4112{ 4113 Mutex::Autolock _l(mLock); 4114 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4115 if (mOutputTracks[i]->thread() == thread) { 4116 mOutputTracks[i]->destroy(); 4117 mOutputTracks.removeAt(i); 4118 updateWaitTime_l(); 4119 return; 4120 } 4121 } 4122 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4123} 4124 4125// caller must hold mLock 4126void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4127{ 4128 mWaitTimeMs = UINT_MAX; 4129 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4130 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4131 if (strong != 0) { 4132 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4133 if (waitTimeMs < mWaitTimeMs) { 4134 mWaitTimeMs = waitTimeMs; 4135 } 4136 } 4137 } 4138} 4139 4140 4141bool AudioFlinger::DuplicatingThread::outputsReady( 4142 const SortedVector< sp<OutputTrack> > &outputTracks) 4143{ 4144 for (size_t i = 0; i < outputTracks.size(); i++) { 4145 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4146 if (thread == 0) { 4147 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4148 outputTracks[i].get()); 4149 return false; 4150 } 4151 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4152 // see note at standby() declaration 4153 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4154 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4155 thread.get()); 4156 return false; 4157 } 4158 } 4159 return true; 4160} 4161 4162uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4163{ 4164 return (mWaitTimeMs * 1000) / 2; 4165} 4166 4167void AudioFlinger::DuplicatingThread::cacheParameters_l() 4168{ 4169 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4170 updateWaitTime_l(); 4171 4172 MixerThread::cacheParameters_l(); 4173} 4174 4175// ---------------------------------------------------------------------------- 4176// Record 4177// ---------------------------------------------------------------------------- 4178 4179AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4180 AudioStreamIn *input, 4181 uint32_t sampleRate, 4182 audio_channel_mask_t channelMask, 4183 audio_io_handle_t id, 4184 audio_devices_t outDevice, 4185 audio_devices_t inDevice 4186#ifdef TEE_SINK 4187 , const sp<NBAIO_Sink>& teeSink 4188#endif 4189 ) : 4190 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4191 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4192 // mRsmpInIndex set by readInputParameters() 4193 mReqChannelCount(popcount(channelMask)), 4194 mReqSampleRate(sampleRate) 4195 // mBytesRead is only meaningful while active, and so is cleared in start() 4196 // (but might be better to also clear here for dump?) 4197#ifdef TEE_SINK 4198 , mTeeSink(teeSink) 4199#endif 4200{ 4201 snprintf(mName, kNameLength, "AudioIn_%X", id); 4202 4203 readInputParameters(); 4204 4205} 4206 4207 4208AudioFlinger::RecordThread::~RecordThread() 4209{ 4210 delete[] mRsmpInBuffer; 4211 delete mResampler; 4212 delete[] mRsmpOutBuffer; 4213} 4214 4215void AudioFlinger::RecordThread::onFirstRef() 4216{ 4217 run(mName, PRIORITY_URGENT_AUDIO); 4218} 4219 4220bool AudioFlinger::RecordThread::threadLoop() 4221{ 4222 AudioBufferProvider::Buffer buffer; 4223 4224 nsecs_t lastWarning = 0; 4225 4226 inputStandBy(); 4227 acquireWakeLock(); 4228 4229 // used to verify we've read at least once before evaluating how many bytes were read 4230 bool readOnce = false; 4231 4232 // used to request a deferred sleep, to be executed later while mutex is unlocked 4233 bool doSleep = false; 4234 4235 // start recording 4236 for (;;) { 4237 sp<RecordTrack> activeTrack; 4238 TrackBase::track_state activeTrackState; 4239 Vector< sp<EffectChain> > effectChains; 4240 4241 // sleep with mutex unlocked 4242 if (doSleep) { 4243 doSleep = false; 4244 usleep(kRecordThreadSleepUs); 4245 } 4246 4247 { // scope for mLock 4248 Mutex::Autolock _l(mLock); 4249 if (exitPending()) { 4250 break; 4251 } 4252 processConfigEvents_l(); 4253 // return value 'reconfig' is currently unused 4254 bool reconfig = checkForNewParameters_l(); 4255 // make a stable copy of mActiveTrack 4256 activeTrack = mActiveTrack; 4257 if (activeTrack == 0) { 4258 standby(); 4259 // exitPending() can't become true here 4260 releaseWakeLock_l(); 4261 ALOGV("RecordThread: loop stopping"); 4262 // go to sleep 4263 mWaitWorkCV.wait(mLock); 4264 ALOGV("RecordThread: loop starting"); 4265 acquireWakeLock_l(); 4266 continue; 4267 } 4268 4269 if (activeTrack->isTerminated()) { 4270 removeTrack_l(activeTrack); 4271 mActiveTrack.clear(); 4272 continue; 4273 } 4274 4275 activeTrackState = activeTrack->mState; 4276 switch (activeTrackState) { 4277 case TrackBase::PAUSING: 4278 standby(); 4279 mActiveTrack.clear(); 4280 mStartStopCond.broadcast(); 4281 doSleep = true; 4282 continue; 4283 4284 case TrackBase::RESUMING: 4285 mStandby = false; 4286 if (mReqChannelCount != activeTrack->channelCount()) { 4287 mActiveTrack.clear(); 4288 mStartStopCond.broadcast(); 4289 continue; 4290 } 4291 if (readOnce) { 4292 mStartStopCond.broadcast(); 4293 // record start succeeds only if first read from audio input succeeds 4294 if (mBytesRead < 0) { 4295 mActiveTrack.clear(); 4296 continue; 4297 } 4298 activeTrack->mState = TrackBase::ACTIVE; 4299 } 4300 break; 4301 4302 case TrackBase::ACTIVE: 4303 break; 4304 4305 case TrackBase::IDLE: 4306 doSleep = true; 4307 continue; 4308 4309 default: 4310 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4311 } 4312 4313 lockEffectChains_l(effectChains); 4314 } 4315 4316 // thread mutex is now unlocked, mActiveTrack unknown, activeTrack != 0, kept, immutable 4317 // activeTrack->mState unknown, activeTrackState immutable and is ACTIVE or RESUMING 4318 4319 for (size_t i = 0; i < effectChains.size(); i ++) { 4320 // thread mutex is not locked, but effect chain is locked 4321 effectChains[i]->process_l(); 4322 } 4323 4324 buffer.frameCount = mFrameCount; 4325 status_t status = activeTrack->getNextBuffer(&buffer); 4326 if (status == NO_ERROR) { 4327 readOnce = true; 4328 size_t framesOut = buffer.frameCount; 4329 if (mResampler == NULL) { 4330 // no resampling 4331 while (framesOut) { 4332 size_t framesIn = mFrameCount - mRsmpInIndex; 4333 if (framesIn > 0) { 4334 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4335 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4336 activeTrack->mFrameSize; 4337 if (framesIn > framesOut) { 4338 framesIn = framesOut; 4339 } 4340 mRsmpInIndex += framesIn; 4341 framesOut -= framesIn; 4342 if (mChannelCount == mReqChannelCount) { 4343 memcpy(dst, src, framesIn * mFrameSize); 4344 } else { 4345 if (mChannelCount == 1) { 4346 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4347 (int16_t *)src, framesIn); 4348 } else { 4349 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4350 (int16_t *)src, framesIn); 4351 } 4352 } 4353 } 4354 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4355 void *readInto; 4356 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4357 readInto = buffer.raw; 4358 framesOut = 0; 4359 } else { 4360 readInto = mRsmpInBuffer; 4361 mRsmpInIndex = 0; 4362 } 4363 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4364 mBufferSize); 4365 if (mBytesRead <= 0) { 4366 // TODO: verify that it's benign to use a stale track state 4367 if ((mBytesRead < 0) && (activeTrackState == TrackBase::ACTIVE)) 4368 { 4369 ALOGE("Error reading audio input"); 4370 // Force input into standby so that it tries to 4371 // recover at next read attempt 4372 inputStandBy(); 4373 doSleep = true; 4374 } 4375 mRsmpInIndex = mFrameCount; 4376 framesOut = 0; 4377 buffer.frameCount = 0; 4378 } 4379#ifdef TEE_SINK 4380 else if (mTeeSink != 0) { 4381 (void) mTeeSink->write(readInto, 4382 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4383 } 4384#endif 4385 } 4386 } 4387 } else { 4388 // resampling 4389 4390 // resampler accumulates, but we only have one source track 4391 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4392 // alter output frame count as if we were expecting stereo samples 4393 if (mChannelCount == 1 && mReqChannelCount == 1) { 4394 framesOut >>= 1; 4395 } 4396 mResampler->resample(mRsmpOutBuffer, framesOut, 4397 this /* AudioBufferProvider* */); 4398 // ditherAndClamp() works as long as all buffers returned by 4399 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4400 if (mChannelCount == 2 && mReqChannelCount == 1) { 4401 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4402 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4403 // the resampler always outputs stereo samples: 4404 // do post stereo to mono conversion 4405 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4406 framesOut); 4407 } else { 4408 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4409 } 4410 // now done with mRsmpOutBuffer 4411 4412 } 4413 if (mFramestoDrop == 0) { 4414 activeTrack->releaseBuffer(&buffer); 4415 } else { 4416 if (mFramestoDrop > 0) { 4417 mFramestoDrop -= buffer.frameCount; 4418 if (mFramestoDrop <= 0) { 4419 clearSyncStartEvent(); 4420 } 4421 } else { 4422 mFramestoDrop += buffer.frameCount; 4423 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4424 mSyncStartEvent->isCancelled()) { 4425 ALOGW("Synced record %s, session %d, trigger session %d", 4426 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4427 activeTrack->sessionId(), 4428 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4429 clearSyncStartEvent(); 4430 } 4431 } 4432 } 4433 activeTrack->clearOverflow(); 4434 } 4435 // client isn't retrieving buffers fast enough 4436 else { 4437 if (!activeTrack->setOverflow()) { 4438 nsecs_t now = systemTime(); 4439 if ((now - lastWarning) > kWarningThrottleNs) { 4440 ALOGW("RecordThread: buffer overflow"); 4441 lastWarning = now; 4442 } 4443 } 4444 // Release the processor for a while before asking for a new buffer. 4445 // This will give the application more chance to read from the buffer and 4446 // clear the overflow. 4447 doSleep = true; 4448 } 4449 4450 // enable changes in effect chain 4451 unlockEffectChains(effectChains); 4452 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4453 } 4454 4455 standby(); 4456 4457 { 4458 Mutex::Autolock _l(mLock); 4459 mActiveTrack.clear(); 4460 mStartStopCond.broadcast(); 4461 } 4462 4463 releaseWakeLock(); 4464 4465 ALOGV("RecordThread %p exiting", this); 4466 return false; 4467} 4468 4469void AudioFlinger::RecordThread::standby() 4470{ 4471 if (!mStandby) { 4472 inputStandBy(); 4473 mStandby = true; 4474 } 4475} 4476 4477void AudioFlinger::RecordThread::inputStandBy() 4478{ 4479 mInput->stream->common.standby(&mInput->stream->common); 4480} 4481 4482sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4483 const sp<AudioFlinger::Client>& client, 4484 uint32_t sampleRate, 4485 audio_format_t format, 4486 audio_channel_mask_t channelMask, 4487 size_t frameCount, 4488 int sessionId, 4489 IAudioFlinger::track_flags_t *flags, 4490 pid_t tid, 4491 status_t *status) 4492{ 4493 sp<RecordTrack> track; 4494 status_t lStatus; 4495 4496 lStatus = initCheck(); 4497 if (lStatus != NO_ERROR) { 4498 ALOGE("Audio driver not initialized."); 4499 goto Exit; 4500 } 4501 4502 // client expresses a preference for FAST, but we get the final say 4503 if (*flags & IAudioFlinger::TRACK_FAST) { 4504 if ( 4505 // use case: callback handler and frame count is default or at least as large as HAL 4506 ( 4507 (tid != -1) && 4508 ((frameCount == 0) || 4509 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4510 ) && 4511 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4512 // mono or stereo 4513 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4514 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4515 // hardware sample rate 4516 (sampleRate == mSampleRate) && 4517 // record thread has an associated fast recorder 4518 hasFastRecorder() 4519 // FIXME test that RecordThread for this fast track has a capable output HAL 4520 // FIXME add a permission test also? 4521 ) { 4522 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4523 if (frameCount == 0) { 4524 frameCount = mFrameCount * kFastTrackMultiplier; 4525 } 4526 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4527 frameCount, mFrameCount); 4528 } else { 4529 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4530 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4531 "hasFastRecorder=%d tid=%d", 4532 frameCount, mFrameCount, format, 4533 audio_is_linear_pcm(format), 4534 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4535 *flags &= ~IAudioFlinger::TRACK_FAST; 4536 // For compatibility with AudioRecord calculation, buffer depth is forced 4537 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4538 // This is probably too conservative, but legacy application code may depend on it. 4539 // If you change this calculation, also review the start threshold which is related. 4540 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4541 size_t mNormalFrameCount = 2048; // FIXME 4542 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4543 if (minBufCount < 2) { 4544 minBufCount = 2; 4545 } 4546 size_t minFrameCount = mNormalFrameCount * minBufCount; 4547 if (frameCount < minFrameCount) { 4548 frameCount = minFrameCount; 4549 } 4550 } 4551 } 4552 4553 // FIXME use flags and tid similar to createTrack_l() 4554 4555 { // scope for mLock 4556 Mutex::Autolock _l(mLock); 4557 4558 track = new RecordTrack(this, client, sampleRate, 4559 format, channelMask, frameCount, sessionId); 4560 4561 lStatus = track->initCheck(); 4562 if (lStatus != NO_ERROR) { 4563 track.clear(); 4564 goto Exit; 4565 } 4566 mTracks.add(track); 4567 4568 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4569 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4570 mAudioFlinger->btNrecIsOff(); 4571 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4572 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4573 4574 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4575 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4576 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4577 // so ask activity manager to do this on our behalf 4578 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4579 } 4580 } 4581 lStatus = NO_ERROR; 4582 4583Exit: 4584 *status = lStatus; 4585 return track; 4586} 4587 4588status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4589 AudioSystem::sync_event_t event, 4590 int triggerSession) 4591{ 4592 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4593 sp<ThreadBase> strongMe = this; 4594 status_t status = NO_ERROR; 4595 4596 if (event == AudioSystem::SYNC_EVENT_NONE) { 4597 clearSyncStartEvent(); 4598 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4599 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4600 triggerSession, 4601 recordTrack->sessionId(), 4602 syncStartEventCallback, 4603 this); 4604 // Sync event can be cancelled by the trigger session if the track is not in a 4605 // compatible state in which case we start record immediately 4606 if (mSyncStartEvent->isCancelled()) { 4607 clearSyncStartEvent(); 4608 } else { 4609 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4610 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4611 } 4612 } 4613 4614 { 4615 // This section is a rendezvous between binder thread executing start() and RecordThread 4616 AutoMutex lock(mLock); 4617 if (mActiveTrack != 0) { 4618 if (recordTrack != mActiveTrack.get()) { 4619 status = -EBUSY; 4620 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4621 mActiveTrack->mState = TrackBase::ACTIVE; 4622 } 4623 return status; 4624 } 4625 4626 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4627 recordTrack->mState = TrackBase::IDLE; 4628 mActiveTrack = recordTrack; 4629 mLock.unlock(); 4630 status_t status = AudioSystem::startInput(mId); 4631 mLock.lock(); 4632 // FIXME should verify that mActiveTrack is still == recordTrack 4633 if (status != NO_ERROR) { 4634 mActiveTrack.clear(); 4635 clearSyncStartEvent(); 4636 return status; 4637 } 4638 mRsmpInIndex = mFrameCount; 4639 mBytesRead = 0; 4640 if (mResampler != NULL) { 4641 mResampler->reset(); 4642 } 4643 // FIXME hijacking a playback track state name which was intended for start after pause; 4644 // here 'STARTING_2' would be more accurate 4645 mActiveTrack->mState = TrackBase::RESUMING; 4646 // signal thread to start 4647 ALOGV("Signal record thread"); 4648 mWaitWorkCV.broadcast(); 4649 // do not wait for mStartStopCond if exiting 4650 if (exitPending()) { 4651 mActiveTrack.clear(); 4652 status = INVALID_OPERATION; 4653 goto startError; 4654 } 4655 // FIXME incorrect usage of wait: no explicit predicate or loop 4656 mStartStopCond.wait(mLock); 4657 if (mActiveTrack == 0) { 4658 ALOGV("Record failed to start"); 4659 status = BAD_VALUE; 4660 goto startError; 4661 } 4662 ALOGV("Record started OK"); 4663 return status; 4664 } 4665 4666startError: 4667 AudioSystem::stopInput(mId); 4668 clearSyncStartEvent(); 4669 return status; 4670} 4671 4672void AudioFlinger::RecordThread::clearSyncStartEvent() 4673{ 4674 if (mSyncStartEvent != 0) { 4675 mSyncStartEvent->cancel(); 4676 } 4677 mSyncStartEvent.clear(); 4678 mFramestoDrop = 0; 4679} 4680 4681void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4682{ 4683 sp<SyncEvent> strongEvent = event.promote(); 4684 4685 if (strongEvent != 0) { 4686 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4687 me->handleSyncStartEvent(strongEvent); 4688 } 4689} 4690 4691void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4692{ 4693 if (event == mSyncStartEvent) { 4694 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4695 // from audio HAL 4696 mFramestoDrop = mFrameCount * 2; 4697 } 4698} 4699 4700bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4701 ALOGV("RecordThread::stop"); 4702 AutoMutex _l(mLock); 4703 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4704 return false; 4705 } 4706 // note that threadLoop may still be processing the track at this point [without lock] 4707 recordTrack->mState = TrackBase::PAUSING; 4708 // do not wait for mStartStopCond if exiting 4709 if (exitPending()) { 4710 return true; 4711 } 4712 // FIXME incorrect usage of wait: no explicit predicate or loop 4713 mStartStopCond.wait(mLock); 4714 // if we have been restarted, recordTrack == mActiveTrack.get() here 4715 if (exitPending() || recordTrack != mActiveTrack.get()) { 4716 ALOGV("Record stopped OK"); 4717 return true; 4718 } 4719 return false; 4720} 4721 4722bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4723{ 4724 return false; 4725} 4726 4727status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4728{ 4729#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4730 if (!isValidSyncEvent(event)) { 4731 return BAD_VALUE; 4732 } 4733 4734 int eventSession = event->triggerSession(); 4735 status_t ret = NAME_NOT_FOUND; 4736 4737 Mutex::Autolock _l(mLock); 4738 4739 for (size_t i = 0; i < mTracks.size(); i++) { 4740 sp<RecordTrack> track = mTracks[i]; 4741 if (eventSession == track->sessionId()) { 4742 (void) track->setSyncEvent(event); 4743 ret = NO_ERROR; 4744 } 4745 } 4746 return ret; 4747#else 4748 return BAD_VALUE; 4749#endif 4750} 4751 4752// destroyTrack_l() must be called with ThreadBase::mLock held 4753void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4754{ 4755 track->terminate(); 4756 track->mState = TrackBase::STOPPED; 4757 // active tracks are removed by threadLoop() 4758 if (mActiveTrack != track) { 4759 removeTrack_l(track); 4760 } 4761} 4762 4763void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4764{ 4765 mTracks.remove(track); 4766 // need anything related to effects here? 4767} 4768 4769void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4770{ 4771 dumpInternals(fd, args); 4772 dumpTracks(fd, args); 4773 dumpEffectChains(fd, args); 4774} 4775 4776void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4777{ 4778 const size_t SIZE = 256; 4779 char buffer[SIZE]; 4780 String8 result; 4781 4782 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4783 result.append(buffer); 4784 4785 if (mActiveTrack != 0) { 4786 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4787 result.append(buffer); 4788 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4789 result.append(buffer); 4790 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4791 result.append(buffer); 4792 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4793 result.append(buffer); 4794 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4795 result.append(buffer); 4796 } else { 4797 result.append("No active record client\n"); 4798 } 4799 4800 write(fd, result.string(), result.size()); 4801 4802 dumpBase(fd, args); 4803} 4804 4805void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4806{ 4807 const size_t SIZE = 256; 4808 char buffer[SIZE]; 4809 String8 result; 4810 4811 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4812 result.append(buffer); 4813 RecordTrack::appendDumpHeader(result); 4814 for (size_t i = 0; i < mTracks.size(); ++i) { 4815 sp<RecordTrack> track = mTracks[i]; 4816 if (track != 0) { 4817 track->dump(buffer, SIZE); 4818 result.append(buffer); 4819 } 4820 } 4821 4822 if (mActiveTrack != 0) { 4823 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4824 result.append(buffer); 4825 RecordTrack::appendDumpHeader(result); 4826 mActiveTrack->dump(buffer, SIZE); 4827 result.append(buffer); 4828 4829 } 4830 write(fd, result.string(), result.size()); 4831} 4832 4833// AudioBufferProvider interface 4834status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4835{ 4836 size_t framesReq = buffer->frameCount; 4837 size_t framesReady = mFrameCount - mRsmpInIndex; 4838 int channelCount; 4839 4840 if (framesReady == 0) { 4841 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4842 if (mBytesRead <= 0) { 4843 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4844 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4845 // Force input into standby so that it tries to 4846 // recover at next read attempt 4847 inputStandBy(); 4848 // FIXME an awkward place to sleep, consider using doSleep when this is pulled up 4849 usleep(kRecordThreadSleepUs); 4850 } 4851 buffer->raw = NULL; 4852 buffer->frameCount = 0; 4853 return NOT_ENOUGH_DATA; 4854 } 4855 mRsmpInIndex = 0; 4856 framesReady = mFrameCount; 4857 } 4858 4859 if (framesReq > framesReady) { 4860 framesReq = framesReady; 4861 } 4862 4863 if (mChannelCount == 1 && mReqChannelCount == 2) { 4864 channelCount = 1; 4865 } else { 4866 channelCount = 2; 4867 } 4868 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4869 buffer->frameCount = framesReq; 4870 return NO_ERROR; 4871} 4872 4873// AudioBufferProvider interface 4874void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4875{ 4876 mRsmpInIndex += buffer->frameCount; 4877 buffer->frameCount = 0; 4878} 4879 4880bool AudioFlinger::RecordThread::checkForNewParameters_l() 4881{ 4882 bool reconfig = false; 4883 4884 while (!mNewParameters.isEmpty()) { 4885 status_t status = NO_ERROR; 4886 String8 keyValuePair = mNewParameters[0]; 4887 AudioParameter param = AudioParameter(keyValuePair); 4888 int value; 4889 audio_format_t reqFormat = mFormat; 4890 uint32_t reqSamplingRate = mReqSampleRate; 4891 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 4892 4893 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4894 reqSamplingRate = value; 4895 reconfig = true; 4896 } 4897 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4898 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4899 status = BAD_VALUE; 4900 } else { 4901 reqFormat = (audio_format_t) value; 4902 reconfig = true; 4903 } 4904 } 4905 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4906 audio_channel_mask_t mask = (audio_channel_mask_t) value; 4907 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 4908 status = BAD_VALUE; 4909 } else { 4910 reqChannelMask = mask; 4911 reconfig = true; 4912 } 4913 } 4914 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4915 // do not accept frame count changes if tracks are open as the track buffer 4916 // size depends on frame count and correct behavior would not be guaranteed 4917 // if frame count is changed after track creation 4918 if (mActiveTrack != 0) { 4919 status = INVALID_OPERATION; 4920 } else { 4921 reconfig = true; 4922 } 4923 } 4924 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4925 // forward device change to effects that have requested to be 4926 // aware of attached audio device. 4927 for (size_t i = 0; i < mEffectChains.size(); i++) { 4928 mEffectChains[i]->setDevice_l(value); 4929 } 4930 4931 // store input device and output device but do not forward output device to audio HAL. 4932 // Note that status is ignored by the caller for output device 4933 // (see AudioFlinger::setParameters() 4934 if (audio_is_output_devices(value)) { 4935 mOutDevice = value; 4936 status = BAD_VALUE; 4937 } else { 4938 mInDevice = value; 4939 // disable AEC and NS if the device is a BT SCO headset supporting those 4940 // pre processings 4941 if (mTracks.size() > 0) { 4942 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4943 mAudioFlinger->btNrecIsOff(); 4944 for (size_t i = 0; i < mTracks.size(); i++) { 4945 sp<RecordTrack> track = mTracks[i]; 4946 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4947 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4948 } 4949 } 4950 } 4951 } 4952 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4953 mAudioSource != (audio_source_t)value) { 4954 // forward device change to effects that have requested to be 4955 // aware of attached audio device. 4956 for (size_t i = 0; i < mEffectChains.size(); i++) { 4957 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4958 } 4959 mAudioSource = (audio_source_t)value; 4960 } 4961 4962 if (status == NO_ERROR) { 4963 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4964 keyValuePair.string()); 4965 if (status == INVALID_OPERATION) { 4966 inputStandBy(); 4967 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4968 keyValuePair.string()); 4969 } 4970 if (reconfig) { 4971 if (status == BAD_VALUE && 4972 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4973 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4974 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4975 <= (2 * reqSamplingRate)) && 4976 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4977 <= FCC_2 && 4978 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 4979 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 4980 status = NO_ERROR; 4981 } 4982 if (status == NO_ERROR) { 4983 readInputParameters(); 4984 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4985 } 4986 } 4987 } 4988 4989 mNewParameters.removeAt(0); 4990 4991 mParamStatus = status; 4992 mParamCond.signal(); 4993 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4994 // already timed out waiting for the status and will never signal the condition. 4995 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4996 } 4997 return reconfig; 4998} 4999 5000String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5001{ 5002 Mutex::Autolock _l(mLock); 5003 if (initCheck() != NO_ERROR) { 5004 return String8(); 5005 } 5006 5007 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5008 const String8 out_s8(s); 5009 free(s); 5010 return out_s8; 5011} 5012 5013void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5014 AudioSystem::OutputDescriptor desc; 5015 void *param2 = NULL; 5016 5017 switch (event) { 5018 case AudioSystem::INPUT_OPENED: 5019 case AudioSystem::INPUT_CONFIG_CHANGED: 5020 desc.channelMask = mChannelMask; 5021 desc.samplingRate = mSampleRate; 5022 desc.format = mFormat; 5023 desc.frameCount = mFrameCount; 5024 desc.latency = 0; 5025 param2 = &desc; 5026 break; 5027 5028 case AudioSystem::INPUT_CLOSED: 5029 default: 5030 break; 5031 } 5032 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5033} 5034 5035void AudioFlinger::RecordThread::readInputParameters() 5036{ 5037 delete[] mRsmpInBuffer; 5038 // mRsmpInBuffer is always assigned a new[] below 5039 delete[] mRsmpOutBuffer; 5040 mRsmpOutBuffer = NULL; 5041 delete mResampler; 5042 mResampler = NULL; 5043 5044 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5045 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5046 mChannelCount = popcount(mChannelMask); 5047 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5048 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5049 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5050 } 5051 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5052 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5053 mFrameCount = mBufferSize / mFrameSize; 5054 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5055 5056 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5057 int channelCount; 5058 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5059 // stereo to mono post process as the resampler always outputs stereo. 5060 if (mChannelCount == 1 && mReqChannelCount == 2) { 5061 channelCount = 1; 5062 } else { 5063 channelCount = 2; 5064 } 5065 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5066 mResampler->setSampleRate(mSampleRate); 5067 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5068 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5069 5070 // optmization: if mono to mono, alter input frame count as if we were inputing 5071 // stereo samples 5072 if (mChannelCount == 1 && mReqChannelCount == 1) { 5073 mFrameCount >>= 1; 5074 } 5075 5076 } 5077 mRsmpInIndex = mFrameCount; 5078} 5079 5080unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5081{ 5082 Mutex::Autolock _l(mLock); 5083 if (initCheck() != NO_ERROR) { 5084 return 0; 5085 } 5086 5087 return mInput->stream->get_input_frames_lost(mInput->stream); 5088} 5089 5090uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5091{ 5092 Mutex::Autolock _l(mLock); 5093 uint32_t result = 0; 5094 if (getEffectChain_l(sessionId) != 0) { 5095 result = EFFECT_SESSION; 5096 } 5097 5098 for (size_t i = 0; i < mTracks.size(); ++i) { 5099 if (sessionId == mTracks[i]->sessionId()) { 5100 result |= TRACK_SESSION; 5101 break; 5102 } 5103 } 5104 5105 return result; 5106} 5107 5108KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5109{ 5110 KeyedVector<int, bool> ids; 5111 Mutex::Autolock _l(mLock); 5112 for (size_t j = 0; j < mTracks.size(); ++j) { 5113 sp<RecordThread::RecordTrack> track = mTracks[j]; 5114 int sessionId = track->sessionId(); 5115 if (ids.indexOfKey(sessionId) < 0) { 5116 ids.add(sessionId, true); 5117 } 5118 } 5119 return ids; 5120} 5121 5122AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5123{ 5124 Mutex::Autolock _l(mLock); 5125 AudioStreamIn *input = mInput; 5126 mInput = NULL; 5127 return input; 5128} 5129 5130// this method must always be called either with ThreadBase mLock held or inside the thread loop 5131audio_stream_t* AudioFlinger::RecordThread::stream() const 5132{ 5133 if (mInput == NULL) { 5134 return NULL; 5135 } 5136 return &mInput->stream->common; 5137} 5138 5139status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5140{ 5141 // only one chain per input thread 5142 if (mEffectChains.size() != 0) { 5143 return INVALID_OPERATION; 5144 } 5145 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5146 5147 chain->setInBuffer(NULL); 5148 chain->setOutBuffer(NULL); 5149 5150 checkSuspendOnAddEffectChain_l(chain); 5151 5152 mEffectChains.add(chain); 5153 5154 return NO_ERROR; 5155} 5156 5157size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5158{ 5159 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5160 ALOGW_IF(mEffectChains.size() != 1, 5161 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5162 chain.get(), mEffectChains.size(), this); 5163 if (mEffectChains.size() == 1) { 5164 mEffectChains.removeAt(0); 5165 } 5166 return 0; 5167} 5168 5169}; // namespace android 5170