Threads.cpp revision cf04c2cb8e031acc03c1c91cb1ccab15098c89b6
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, and mFormat are 270 // 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 mLock.lock(); 380 while (!mConfigEvents.isEmpty()) { 381 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 382 ConfigEvent *event = mConfigEvents[0]; 383 mConfigEvents.removeAt(0); 384 // release mLock before locking AudioFlinger mLock: lock order is always 385 // AudioFlinger then ThreadBase to avoid cross deadlock 386 mLock.unlock(); 387 switch(event->type()) { 388 case CFG_EVENT_PRIO: { 389 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 390 // FIXME Need to understand why this has be done asynchronously 391 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 392 true /*asynchronous*/); 393 if (err != 0) { 394 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 395 "error %d", 396 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 397 } 398 } break; 399 case CFG_EVENT_IO: { 400 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 401 mAudioFlinger->mLock.lock(); 402 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 403 mAudioFlinger->mLock.unlock(); 404 } break; 405 default: 406 ALOGE("processConfigEvents() unknown event type %d", event->type()); 407 break; 408 } 409 delete event; 410 mLock.lock(); 411 } 412 mLock.unlock(); 413} 414 415void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 416{ 417 const size_t SIZE = 256; 418 char buffer[SIZE]; 419 String8 result; 420 421 bool locked = AudioFlinger::dumpTryLock(mLock); 422 if (!locked) { 423 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 424 write(fd, buffer, strlen(buffer)); 425 } 426 427 snprintf(buffer, SIZE, "io handle: %d\n", mId); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 432 result.append(buffer); 433 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 434 result.append(buffer); 435 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 436 result.append(buffer); 437 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 438 result.append(buffer); 439 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 440 result.append(buffer); 441 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 442 result.append(buffer); 443 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 444 result.append(buffer); 445 446 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 447 result.append(buffer); 448 result.append(" Index Command"); 449 for (size_t i = 0; i < mNewParameters.size(); ++i) { 450 snprintf(buffer, SIZE, "\n %02d ", i); 451 result.append(buffer); 452 result.append(mNewParameters[i]); 453 } 454 455 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 456 result.append(buffer); 457 for (size_t i = 0; i < mConfigEvents.size(); i++) { 458 mConfigEvents[i]->dump(buffer, SIZE); 459 result.append(buffer); 460 } 461 result.append("\n"); 462 463 write(fd, result.string(), result.size()); 464 465 if (locked) { 466 mLock.unlock(); 467 } 468} 469 470void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 471{ 472 const size_t SIZE = 256; 473 char buffer[SIZE]; 474 String8 result; 475 476 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 477 write(fd, buffer, strlen(buffer)); 478 479 for (size_t i = 0; i < mEffectChains.size(); ++i) { 480 sp<EffectChain> chain = mEffectChains[i]; 481 if (chain != 0) { 482 chain->dump(fd, args); 483 } 484 } 485} 486 487void AudioFlinger::ThreadBase::acquireWakeLock() 488{ 489 Mutex::Autolock _l(mLock); 490 acquireWakeLock_l(); 491} 492 493void AudioFlinger::ThreadBase::acquireWakeLock_l() 494{ 495 if (mPowerManager == 0) { 496 // use checkService() to avoid blocking if power service is not up yet 497 sp<IBinder> binder = 498 defaultServiceManager()->checkService(String16("power")); 499 if (binder == 0) { 500 ALOGW("Thread %s cannot connect to the power manager service", mName); 501 } else { 502 mPowerManager = interface_cast<IPowerManager>(binder); 503 binder->linkToDeath(mDeathRecipient); 504 } 505 } 506 if (mPowerManager != 0) { 507 sp<IBinder> binder = new BBinder(); 508 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 509 binder, 510 String16(mName), 511 String16("media")); 512 if (status == NO_ERROR) { 513 mWakeLockToken = binder; 514 } 515 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 516 } 517} 518 519void AudioFlinger::ThreadBase::releaseWakeLock() 520{ 521 Mutex::Autolock _l(mLock); 522 releaseWakeLock_l(); 523} 524 525void AudioFlinger::ThreadBase::releaseWakeLock_l() 526{ 527 if (mWakeLockToken != 0) { 528 ALOGV("releaseWakeLock_l() %s", mName); 529 if (mPowerManager != 0) { 530 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 531 } 532 mWakeLockToken.clear(); 533 } 534} 535 536void AudioFlinger::ThreadBase::clearPowerManager() 537{ 538 Mutex::Autolock _l(mLock); 539 releaseWakeLock_l(); 540 mPowerManager.clear(); 541} 542 543void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 544{ 545 sp<ThreadBase> thread = mThread.promote(); 546 if (thread != 0) { 547 thread->clearPowerManager(); 548 } 549 ALOGW("power manager service died !!!"); 550} 551 552void AudioFlinger::ThreadBase::setEffectSuspended( 553 const effect_uuid_t *type, bool suspend, int sessionId) 554{ 555 Mutex::Autolock _l(mLock); 556 setEffectSuspended_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::setEffectSuspended_l( 560 const effect_uuid_t *type, bool suspend, int sessionId) 561{ 562 sp<EffectChain> chain = getEffectChain_l(sessionId); 563 if (chain != 0) { 564 if (type != NULL) { 565 chain->setEffectSuspended_l(type, suspend); 566 } else { 567 chain->setEffectSuspendedAll_l(suspend); 568 } 569 } 570 571 updateSuspendedSessions_l(type, suspend, sessionId); 572} 573 574void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 575{ 576 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 577 if (index < 0) { 578 return; 579 } 580 581 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 582 mSuspendedSessions.valueAt(index); 583 584 for (size_t i = 0; i < sessionEffects.size(); i++) { 585 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 586 for (int j = 0; j < desc->mRefCount; j++) { 587 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 588 chain->setEffectSuspendedAll_l(true); 589 } else { 590 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 591 desc->mType.timeLow); 592 chain->setEffectSuspended_l(&desc->mType, true); 593 } 594 } 595 } 596} 597 598void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 599 bool suspend, 600 int sessionId) 601{ 602 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 603 604 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 605 606 if (suspend) { 607 if (index >= 0) { 608 sessionEffects = mSuspendedSessions.valueAt(index); 609 } else { 610 mSuspendedSessions.add(sessionId, sessionEffects); 611 } 612 } else { 613 if (index < 0) { 614 return; 615 } 616 sessionEffects = mSuspendedSessions.valueAt(index); 617 } 618 619 620 int key = EffectChain::kKeyForSuspendAll; 621 if (type != NULL) { 622 key = type->timeLow; 623 } 624 index = sessionEffects.indexOfKey(key); 625 626 sp<SuspendedSessionDesc> desc; 627 if (suspend) { 628 if (index >= 0) { 629 desc = sessionEffects.valueAt(index); 630 } else { 631 desc = new SuspendedSessionDesc(); 632 if (type != NULL) { 633 desc->mType = *type; 634 } 635 sessionEffects.add(key, desc); 636 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 637 } 638 desc->mRefCount++; 639 } else { 640 if (index < 0) { 641 return; 642 } 643 desc = sessionEffects.valueAt(index); 644 if (--desc->mRefCount == 0) { 645 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 646 sessionEffects.removeItemsAt(index); 647 if (sessionEffects.isEmpty()) { 648 ALOGV("updateSuspendedSessions_l() restore removing session %d", 649 sessionId); 650 mSuspendedSessions.removeItem(sessionId); 651 } 652 } 653 } 654 if (!sessionEffects.isEmpty()) { 655 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 656 } 657} 658 659void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 660 bool enabled, 661 int sessionId) 662{ 663 Mutex::Autolock _l(mLock); 664 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 665} 666 667void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 668 bool enabled, 669 int sessionId) 670{ 671 if (mType != RECORD) { 672 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 673 // another session. This gives the priority to well behaved effect control panels 674 // and applications not using global effects. 675 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 676 // global effects 677 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 678 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 679 } 680 } 681 682 sp<EffectChain> chain = getEffectChain_l(sessionId); 683 if (chain != 0) { 684 chain->checkSuspendOnEffectEnabled(effect, enabled); 685 } 686} 687 688// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 689sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 690 const sp<AudioFlinger::Client>& client, 691 const sp<IEffectClient>& effectClient, 692 int32_t priority, 693 int sessionId, 694 effect_descriptor_t *desc, 695 int *enabled, 696 status_t *status 697 ) 698{ 699 sp<EffectModule> effect; 700 sp<EffectHandle> handle; 701 status_t lStatus; 702 sp<EffectChain> chain; 703 bool chainCreated = false; 704 bool effectCreated = false; 705 bool effectRegistered = false; 706 707 lStatus = initCheck(); 708 if (lStatus != NO_ERROR) { 709 ALOGW("createEffect_l() Audio driver not initialized."); 710 goto Exit; 711 } 712 713 // Do not allow effects with session ID 0 on direct output or duplicating threads 714 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 715 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 716 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 717 desc->name, sessionId); 718 lStatus = BAD_VALUE; 719 goto Exit; 720 } 721 // Only Pre processor effects are allowed on input threads and only on input threads 722 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 723 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 724 desc->name, desc->flags, mType); 725 lStatus = BAD_VALUE; 726 goto Exit; 727 } 728 729 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 730 731 { // scope for mLock 732 Mutex::Autolock _l(mLock); 733 734 // check for existing effect chain with the requested audio session 735 chain = getEffectChain_l(sessionId); 736 if (chain == 0) { 737 // create a new chain for this session 738 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 739 chain = new EffectChain(this, sessionId); 740 addEffectChain_l(chain); 741 chain->setStrategy(getStrategyForSession_l(sessionId)); 742 chainCreated = true; 743 } else { 744 effect = chain->getEffectFromDesc_l(desc); 745 } 746 747 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 748 749 if (effect == 0) { 750 int id = mAudioFlinger->nextUniqueId(); 751 // Check CPU and memory usage 752 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 753 if (lStatus != NO_ERROR) { 754 goto Exit; 755 } 756 effectRegistered = true; 757 // create a new effect module if none present in the chain 758 effect = new EffectModule(this, chain, desc, id, sessionId); 759 lStatus = effect->status(); 760 if (lStatus != NO_ERROR) { 761 goto Exit; 762 } 763 lStatus = chain->addEffect_l(effect); 764 if (lStatus != NO_ERROR) { 765 goto Exit; 766 } 767 effectCreated = true; 768 769 effect->setDevice(mOutDevice); 770 effect->setDevice(mInDevice); 771 effect->setMode(mAudioFlinger->getMode()); 772 effect->setAudioSource(mAudioSource); 773 } 774 // create effect handle and connect it to effect module 775 handle = new EffectHandle(effect, client, effectClient, priority); 776 lStatus = effect->addHandle(handle.get()); 777 if (enabled != NULL) { 778 *enabled = (int)effect->isEnabled(); 779 } 780 } 781 782Exit: 783 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 784 Mutex::Autolock _l(mLock); 785 if (effectCreated) { 786 chain->removeEffect_l(effect); 787 } 788 if (effectRegistered) { 789 AudioSystem::unregisterEffect(effect->id()); 790 } 791 if (chainCreated) { 792 removeEffectChain_l(chain); 793 } 794 handle.clear(); 795 } 796 797 if (status != NULL) { 798 *status = lStatus; 799 } 800 return handle; 801} 802 803sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 804{ 805 Mutex::Autolock _l(mLock); 806 return getEffect_l(sessionId, effectId); 807} 808 809sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 810{ 811 sp<EffectChain> chain = getEffectChain_l(sessionId); 812 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 813} 814 815// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 816// PlaybackThread::mLock held 817status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 818{ 819 // check for existing effect chain with the requested audio session 820 int sessionId = effect->sessionId(); 821 sp<EffectChain> chain = getEffectChain_l(sessionId); 822 bool chainCreated = false; 823 824 if (chain == 0) { 825 // create a new chain for this session 826 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 827 chain = new EffectChain(this, sessionId); 828 addEffectChain_l(chain); 829 chain->setStrategy(getStrategyForSession_l(sessionId)); 830 chainCreated = true; 831 } 832 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 833 834 if (chain->getEffectFromId_l(effect->id()) != 0) { 835 ALOGW("addEffect_l() %p effect %s already present in chain %p", 836 this, effect->desc().name, chain.get()); 837 return BAD_VALUE; 838 } 839 840 status_t status = chain->addEffect_l(effect); 841 if (status != NO_ERROR) { 842 if (chainCreated) { 843 removeEffectChain_l(chain); 844 } 845 return status; 846 } 847 848 effect->setDevice(mOutDevice); 849 effect->setDevice(mInDevice); 850 effect->setMode(mAudioFlinger->getMode()); 851 effect->setAudioSource(mAudioSource); 852 return NO_ERROR; 853} 854 855void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 856 857 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 858 effect_descriptor_t desc = effect->desc(); 859 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 860 detachAuxEffect_l(effect->id()); 861 } 862 863 sp<EffectChain> chain = effect->chain().promote(); 864 if (chain != 0) { 865 // remove effect chain if removing last effect 866 if (chain->removeEffect_l(effect) == 0) { 867 removeEffectChain_l(chain); 868 } 869 } else { 870 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 871 } 872} 873 874void AudioFlinger::ThreadBase::lockEffectChains_l( 875 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 876{ 877 effectChains = mEffectChains; 878 for (size_t i = 0; i < mEffectChains.size(); i++) { 879 mEffectChains[i]->lock(); 880 } 881} 882 883void AudioFlinger::ThreadBase::unlockEffectChains( 884 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 885{ 886 for (size_t i = 0; i < effectChains.size(); i++) { 887 effectChains[i]->unlock(); 888 } 889} 890 891sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 892{ 893 Mutex::Autolock _l(mLock); 894 return getEffectChain_l(sessionId); 895} 896 897sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 898{ 899 size_t size = mEffectChains.size(); 900 for (size_t i = 0; i < size; i++) { 901 if (mEffectChains[i]->sessionId() == sessionId) { 902 return mEffectChains[i]; 903 } 904 } 905 return 0; 906} 907 908void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 909{ 910 Mutex::Autolock _l(mLock); 911 size_t size = mEffectChains.size(); 912 for (size_t i = 0; i < size; i++) { 913 mEffectChains[i]->setMode_l(mode); 914 } 915} 916 917void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 918 EffectHandle *handle, 919 bool unpinIfLast) { 920 921 Mutex::Autolock _l(mLock); 922 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 923 // delete the effect module if removing last handle on it 924 if (effect->removeHandle(handle) == 0) { 925 if (!effect->isPinned() || unpinIfLast) { 926 removeEffect_l(effect); 927 AudioSystem::unregisterEffect(effect->id()); 928 } 929 } 930} 931 932// ---------------------------------------------------------------------------- 933// Playback 934// ---------------------------------------------------------------------------- 935 936AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 937 AudioStreamOut* output, 938 audio_io_handle_t id, 939 audio_devices_t device, 940 type_t type) 941 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 942 mNormalFrameCount(0), mMixBuffer(NULL), 943 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 944 // mStreamTypes[] initialized in constructor body 945 mOutput(output), 946 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 947 mMixerStatus(MIXER_IDLE), 948 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 949 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 950 mBytesRemaining(0), 951 mCurrentWriteLength(0), 952 mUseAsyncWrite(false), 953 mWriteBlocked(false), 954 mDraining(false), 955 mScreenState(AudioFlinger::mScreenState), 956 // index 0 is reserved for normal mixer's submix 957 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 958{ 959 snprintf(mName, kNameLength, "AudioOut_%X", id); 960 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 961 962 // Assumes constructor is called by AudioFlinger with it's mLock held, but 963 // it would be safer to explicitly pass initial masterVolume/masterMute as 964 // parameter. 965 // 966 // If the HAL we are using has support for master volume or master mute, 967 // then do not attenuate or mute during mixing (just leave the volume at 1.0 968 // and the mute set to false). 969 mMasterVolume = audioFlinger->masterVolume_l(); 970 mMasterMute = audioFlinger->masterMute_l(); 971 if (mOutput && mOutput->audioHwDev) { 972 if (mOutput->audioHwDev->canSetMasterVolume()) { 973 mMasterVolume = 1.0; 974 } 975 976 if (mOutput->audioHwDev->canSetMasterMute()) { 977 mMasterMute = false; 978 } 979 } 980 981 readOutputParameters(); 982 983 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 984 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 985 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 986 stream = (audio_stream_type_t) (stream + 1)) { 987 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 988 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 989 } 990 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 991 // because mAudioFlinger doesn't have one to copy from 992} 993 994AudioFlinger::PlaybackThread::~PlaybackThread() 995{ 996 mAudioFlinger->unregisterWriter(mNBLogWriter); 997 delete [] mAllocMixBuffer; 998} 999 1000void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1001{ 1002 dumpInternals(fd, args); 1003 dumpTracks(fd, args); 1004 dumpEffectChains(fd, args); 1005} 1006 1007void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1008{ 1009 const size_t SIZE = 256; 1010 char buffer[SIZE]; 1011 String8 result; 1012 1013 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1014 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1015 const stream_type_t *st = &mStreamTypes[i]; 1016 if (i > 0) { 1017 result.appendFormat(", "); 1018 } 1019 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1020 if (st->mute) { 1021 result.append("M"); 1022 } 1023 } 1024 result.append("\n"); 1025 write(fd, result.string(), result.length()); 1026 result.clear(); 1027 1028 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1029 result.append(buffer); 1030 Track::appendDumpHeader(result); 1031 for (size_t i = 0; i < mTracks.size(); ++i) { 1032 sp<Track> track = mTracks[i]; 1033 if (track != 0) { 1034 track->dump(buffer, SIZE); 1035 result.append(buffer); 1036 } 1037 } 1038 1039 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1040 result.append(buffer); 1041 Track::appendDumpHeader(result); 1042 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1043 sp<Track> track = mActiveTracks[i].promote(); 1044 if (track != 0) { 1045 track->dump(buffer, SIZE); 1046 result.append(buffer); 1047 } 1048 } 1049 write(fd, result.string(), result.size()); 1050 1051 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1052 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1053 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1054 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1055} 1056 1057void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1058{ 1059 const size_t SIZE = 256; 1060 char buffer[SIZE]; 1061 String8 result; 1062 1063 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1064 result.append(buffer); 1065 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1066 result.append(buffer); 1067 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1068 ns2ms(systemTime() - mLastWriteTime)); 1069 result.append(buffer); 1070 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1071 result.append(buffer); 1072 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1073 result.append(buffer); 1074 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1079 result.append(buffer); 1080 write(fd, result.string(), result.size()); 1081 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1082 1083 dumpBase(fd, args); 1084} 1085 1086// Thread virtuals 1087 1088void AudioFlinger::PlaybackThread::onFirstRef() 1089{ 1090 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1091} 1092 1093// ThreadBase virtuals 1094void AudioFlinger::PlaybackThread::preExit() 1095{ 1096 ALOGV(" preExit()"); 1097 // FIXME this is using hard-coded strings but in the future, this functionality will be 1098 // converted to use audio HAL extensions required to support tunneling 1099 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1100} 1101 1102// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1103sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1104 const sp<AudioFlinger::Client>& client, 1105 audio_stream_type_t streamType, 1106 uint32_t sampleRate, 1107 audio_format_t format, 1108 audio_channel_mask_t channelMask, 1109 size_t frameCount, 1110 const sp<IMemory>& sharedBuffer, 1111 int sessionId, 1112 IAudioFlinger::track_flags_t *flags, 1113 pid_t tid, 1114 status_t *status) 1115{ 1116 sp<Track> track; 1117 status_t lStatus; 1118 1119 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1120 1121 // client expresses a preference for FAST, but we get the final say 1122 if (*flags & IAudioFlinger::TRACK_FAST) { 1123 if ( 1124 // not timed 1125 (!isTimed) && 1126 // either of these use cases: 1127 ( 1128 // use case 1: shared buffer with any frame count 1129 ( 1130 (sharedBuffer != 0) 1131 ) || 1132 // use case 2: callback handler and frame count is default or at least as large as HAL 1133 ( 1134 (tid != -1) && 1135 ((frameCount == 0) || 1136 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1137 ) 1138 ) && 1139 // PCM data 1140 audio_is_linear_pcm(format) && 1141 // mono or stereo 1142 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1143 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1144#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1145 // hardware sample rate 1146 (sampleRate == mSampleRate) && 1147#endif 1148 // normal mixer has an associated fast mixer 1149 hasFastMixer() && 1150 // there are sufficient fast track slots available 1151 (mFastTrackAvailMask != 0) 1152 // FIXME test that MixerThread for this fast track has a capable output HAL 1153 // FIXME add a permission test also? 1154 ) { 1155 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1156 if (frameCount == 0) { 1157 frameCount = mFrameCount * kFastTrackMultiplier; 1158 } 1159 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1160 frameCount, mFrameCount); 1161 } else { 1162 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1163 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1164 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1165 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1166 audio_is_linear_pcm(format), 1167 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1168 *flags &= ~IAudioFlinger::TRACK_FAST; 1169 // For compatibility with AudioTrack calculation, buffer depth is forced 1170 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1171 // This is probably too conservative, but legacy application code may depend on it. 1172 // If you change this calculation, also review the start threshold which is related. 1173 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1174 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1175 if (minBufCount < 2) { 1176 minBufCount = 2; 1177 } 1178 size_t minFrameCount = mNormalFrameCount * minBufCount; 1179 if (frameCount < minFrameCount) { 1180 frameCount = minFrameCount; 1181 } 1182 } 1183 } 1184 1185 if (mType == DIRECT) { 1186 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1187 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1188 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1189 "for output %p with format %d", 1190 sampleRate, format, channelMask, mOutput, mFormat); 1191 lStatus = BAD_VALUE; 1192 goto Exit; 1193 } 1194 } 1195 } else if (mType == OFFLOAD) { 1196 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1197 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1198 "for output %p with format %d", 1199 sampleRate, format, channelMask, mOutput, mFormat); 1200 lStatus = BAD_VALUE; 1201 goto Exit; 1202 } 1203 } else { 1204 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1205 ALOGE("createTrack_l() Bad parameter: format %d \"" 1206 "for output %p with format %d", 1207 format, mOutput, mFormat); 1208 lStatus = BAD_VALUE; 1209 goto Exit; 1210 } 1211 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1212 if (sampleRate > mSampleRate*2) { 1213 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1214 lStatus = BAD_VALUE; 1215 goto Exit; 1216 } 1217 } 1218 1219 lStatus = initCheck(); 1220 if (lStatus != NO_ERROR) { 1221 ALOGE("Audio driver not initialized."); 1222 goto Exit; 1223 } 1224 1225 { // scope for mLock 1226 Mutex::Autolock _l(mLock); 1227 1228 // all tracks in same audio session must share the same routing strategy otherwise 1229 // conflicts will happen when tracks are moved from one output to another by audio policy 1230 // manager 1231 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1232 for (size_t i = 0; i < mTracks.size(); ++i) { 1233 sp<Track> t = mTracks[i]; 1234 if (t != 0 && !t->isOutputTrack()) { 1235 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1236 if (sessionId == t->sessionId() && strategy != actual) { 1237 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1238 strategy, actual); 1239 lStatus = BAD_VALUE; 1240 goto Exit; 1241 } 1242 } 1243 } 1244 1245 if (!isTimed) { 1246 track = new Track(this, client, streamType, sampleRate, format, 1247 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1248 } else { 1249 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1250 channelMask, frameCount, sharedBuffer, sessionId); 1251 } 1252 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1253 lStatus = NO_MEMORY; 1254 goto Exit; 1255 } 1256 1257 mTracks.add(track); 1258 1259 sp<EffectChain> chain = getEffectChain_l(sessionId); 1260 if (chain != 0) { 1261 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1262 track->setMainBuffer(chain->inBuffer()); 1263 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1264 chain->incTrackCnt(); 1265 } 1266 1267 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1268 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1269 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1270 // so ask activity manager to do this on our behalf 1271 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1272 } 1273 } 1274 1275 lStatus = NO_ERROR; 1276 1277Exit: 1278 if (status) { 1279 *status = lStatus; 1280 } 1281 return track; 1282} 1283 1284uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1285{ 1286 return latency; 1287} 1288 1289uint32_t AudioFlinger::PlaybackThread::latency() const 1290{ 1291 Mutex::Autolock _l(mLock); 1292 return latency_l(); 1293} 1294uint32_t AudioFlinger::PlaybackThread::latency_l() const 1295{ 1296 if (initCheck() == NO_ERROR) { 1297 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1298 } else { 1299 return 0; 1300 } 1301} 1302 1303void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1304{ 1305 Mutex::Autolock _l(mLock); 1306 // Don't apply master volume in SW if our HAL can do it for us. 1307 if (mOutput && mOutput->audioHwDev && 1308 mOutput->audioHwDev->canSetMasterVolume()) { 1309 mMasterVolume = 1.0; 1310 } else { 1311 mMasterVolume = value; 1312 } 1313} 1314 1315void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 // Don't apply master mute in SW if our HAL can do it for us. 1319 if (mOutput && mOutput->audioHwDev && 1320 mOutput->audioHwDev->canSetMasterMute()) { 1321 mMasterMute = false; 1322 } else { 1323 mMasterMute = muted; 1324 } 1325} 1326 1327void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1328{ 1329 Mutex::Autolock _l(mLock); 1330 mStreamTypes[stream].volume = value; 1331 signal_l(); 1332} 1333 1334void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1335{ 1336 Mutex::Autolock _l(mLock); 1337 mStreamTypes[stream].mute = muted; 1338 signal_l(); 1339} 1340 1341float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1342{ 1343 Mutex::Autolock _l(mLock); 1344 return mStreamTypes[stream].volume; 1345} 1346 1347// addTrack_l() must be called with ThreadBase::mLock held 1348status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1349{ 1350 status_t status = ALREADY_EXISTS; 1351 1352 // set retry count for buffer fill 1353 track->mRetryCount = kMaxTrackStartupRetries; 1354 if (mActiveTracks.indexOf(track) < 0) { 1355 // the track is newly added, make sure it fills up all its 1356 // buffers before playing. This is to ensure the client will 1357 // effectively get the latency it requested. 1358 if (!track->isOutputTrack()) { 1359 TrackBase::track_state state = track->mState; 1360 mLock.unlock(); 1361 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1362 mLock.lock(); 1363 // abort track was stopped/paused while we released the lock 1364 if (state != track->mState) { 1365 if (status == NO_ERROR) { 1366 mLock.unlock(); 1367 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1368 mLock.lock(); 1369 } 1370 return INVALID_OPERATION; 1371 } 1372 // abort if start is rejected by audio policy manager 1373 if (status != NO_ERROR) { 1374 return PERMISSION_DENIED; 1375 } 1376#ifdef ADD_BATTERY_DATA 1377 // to track the speaker usage 1378 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1379#endif 1380 } 1381 1382 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1383 track->mResetDone = false; 1384 track->mPresentationCompleteFrames = 0; 1385 mActiveTracks.add(track); 1386 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1387 if (chain != 0) { 1388 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1389 track->sessionId()); 1390 chain->incActiveTrackCnt(); 1391 } 1392 1393 status = NO_ERROR; 1394 } 1395 1396 ALOGV("mWaitWorkCV.broadcast"); 1397 mWaitWorkCV.broadcast(); 1398 1399 return status; 1400} 1401 1402bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1403{ 1404 track->terminate(); 1405 // active tracks are removed by threadLoop() 1406 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1407 track->mState = TrackBase::STOPPED; 1408 if (!trackActive) { 1409 removeTrack_l(track); 1410 } else if (track->isFastTrack() || track->isOffloaded()) { 1411 track->mState = TrackBase::STOPPING_1; 1412 } 1413 1414 return trackActive; 1415} 1416 1417void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1418{ 1419 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1420 mTracks.remove(track); 1421 deleteTrackName_l(track->name()); 1422 // redundant as track is about to be destroyed, for dumpsys only 1423 track->mName = -1; 1424 if (track->isFastTrack()) { 1425 int index = track->mFastIndex; 1426 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1427 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1428 mFastTrackAvailMask |= 1 << index; 1429 // redundant as track is about to be destroyed, for dumpsys only 1430 track->mFastIndex = -1; 1431 } 1432 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1433 if (chain != 0) { 1434 chain->decTrackCnt(); 1435 } 1436} 1437 1438void AudioFlinger::PlaybackThread::signal_l() 1439{ 1440 // Thread could be blocked waiting for async 1441 // so signal it to handle state changes immediately 1442 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1443 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1444 mSignalPending = true; 1445 mWaitWorkCV.signal(); 1446} 1447 1448String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1449{ 1450 Mutex::Autolock _l(mLock); 1451 if (initCheck() != NO_ERROR) { 1452 return String8(); 1453 } 1454 1455 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1456 const String8 out_s8(s); 1457 free(s); 1458 return out_s8; 1459} 1460 1461// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1462void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1463 AudioSystem::OutputDescriptor desc; 1464 void *param2 = NULL; 1465 1466 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1467 param); 1468 1469 switch (event) { 1470 case AudioSystem::OUTPUT_OPENED: 1471 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1472 desc.channelMask = mChannelMask; 1473 desc.samplingRate = mSampleRate; 1474 desc.format = mFormat; 1475 desc.frameCount = mNormalFrameCount; // FIXME see 1476 // AudioFlinger::frameCount(audio_io_handle_t) 1477 desc.latency = latency(); 1478 param2 = &desc; 1479 break; 1480 1481 case AudioSystem::STREAM_CONFIG_CHANGED: 1482 param2 = ¶m; 1483 case AudioSystem::OUTPUT_CLOSED: 1484 default: 1485 break; 1486 } 1487 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1488} 1489 1490void AudioFlinger::PlaybackThread::writeCallback() 1491{ 1492 ALOG_ASSERT(mCallbackThread != 0); 1493 mCallbackThread->setWriteBlocked(false); 1494} 1495 1496void AudioFlinger::PlaybackThread::drainCallback() 1497{ 1498 ALOG_ASSERT(mCallbackThread != 0); 1499 mCallbackThread->setDraining(false); 1500} 1501 1502void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1503{ 1504 Mutex::Autolock _l(mLock); 1505 mWriteBlocked = value; 1506 if (!value) { 1507 mWaitWorkCV.signal(); 1508 } 1509} 1510 1511void AudioFlinger::PlaybackThread::setDraining(bool value) 1512{ 1513 Mutex::Autolock _l(mLock); 1514 mDraining = value; 1515 if (!value) { 1516 mWaitWorkCV.signal(); 1517 } 1518} 1519 1520// static 1521int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1522 void *param, 1523 void *cookie) 1524{ 1525 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1526 ALOGV("asyncCallback() event %d", event); 1527 switch (event) { 1528 case STREAM_CBK_EVENT_WRITE_READY: 1529 me->writeCallback(); 1530 break; 1531 case STREAM_CBK_EVENT_DRAIN_READY: 1532 me->drainCallback(); 1533 break; 1534 default: 1535 ALOGW("asyncCallback() unknown event %d", event); 1536 break; 1537 } 1538 return 0; 1539} 1540 1541void AudioFlinger::PlaybackThread::readOutputParameters() 1542{ 1543 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1544 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1545 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1546 if (!audio_is_output_channel(mChannelMask)) { 1547 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1548 } 1549 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1550 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1551 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1552 } 1553 mChannelCount = popcount(mChannelMask); 1554 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1555 if (!audio_is_valid_format(mFormat)) { 1556 LOG_FATAL("HAL format %d not valid for output", mFormat); 1557 } 1558 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1559 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1560 mFormat); 1561 } 1562 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1563 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1564 if (mFrameCount & 15) { 1565 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1566 mFrameCount); 1567 } 1568 1569 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1570 (mOutput->stream->set_callback != NULL)) { 1571 if (mOutput->stream->set_callback(mOutput->stream, 1572 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1573 mUseAsyncWrite = true; 1574 } 1575 } 1576 1577 // Calculate size of normal mix buffer relative to the HAL output buffer size 1578 double multiplier = 1.0; 1579 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1580 kUseFastMixer == FastMixer_Dynamic)) { 1581 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1582 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1583 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1584 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1585 maxNormalFrameCount = maxNormalFrameCount & ~15; 1586 if (maxNormalFrameCount < minNormalFrameCount) { 1587 maxNormalFrameCount = minNormalFrameCount; 1588 } 1589 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1590 if (multiplier <= 1.0) { 1591 multiplier = 1.0; 1592 } else if (multiplier <= 2.0) { 1593 if (2 * mFrameCount <= maxNormalFrameCount) { 1594 multiplier = 2.0; 1595 } else { 1596 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1597 } 1598 } else { 1599 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1600 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1601 // track, but we sometimes have to do this to satisfy the maximum frame count 1602 // constraint) 1603 // FIXME this rounding up should not be done if no HAL SRC 1604 uint32_t truncMult = (uint32_t) multiplier; 1605 if ((truncMult & 1)) { 1606 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1607 ++truncMult; 1608 } 1609 } 1610 multiplier = (double) truncMult; 1611 } 1612 } 1613 mNormalFrameCount = multiplier * mFrameCount; 1614 // round up to nearest 16 frames to satisfy AudioMixer 1615 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1616 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1617 mNormalFrameCount); 1618 1619 delete[] mAllocMixBuffer; 1620 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1621 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1622 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1623 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1624 1625 // force reconfiguration of effect chains and engines to take new buffer size and audio 1626 // parameters into account 1627 // Note that mLock is not held when readOutputParameters() is called from the constructor 1628 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1629 // matter. 1630 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1631 Vector< sp<EffectChain> > effectChains = mEffectChains; 1632 for (size_t i = 0; i < effectChains.size(); i ++) { 1633 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1634 } 1635} 1636 1637 1638status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1639{ 1640 if (halFrames == NULL || dspFrames == NULL) { 1641 return BAD_VALUE; 1642 } 1643 Mutex::Autolock _l(mLock); 1644 if (initCheck() != NO_ERROR) { 1645 return INVALID_OPERATION; 1646 } 1647 size_t framesWritten = mBytesWritten / mFrameSize; 1648 *halFrames = framesWritten; 1649 1650 if (isSuspended()) { 1651 // return an estimation of rendered frames when the output is suspended 1652 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1653 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1654 return NO_ERROR; 1655 } else { 1656 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1657 } 1658} 1659 1660uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1661{ 1662 Mutex::Autolock _l(mLock); 1663 uint32_t result = 0; 1664 if (getEffectChain_l(sessionId) != 0) { 1665 result = EFFECT_SESSION; 1666 } 1667 1668 for (size_t i = 0; i < mTracks.size(); ++i) { 1669 sp<Track> track = mTracks[i]; 1670 if (sessionId == track->sessionId() && !track->isInvalid()) { 1671 result |= TRACK_SESSION; 1672 break; 1673 } 1674 } 1675 1676 return result; 1677} 1678 1679uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1680{ 1681 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1682 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1683 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1684 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1685 } 1686 for (size_t i = 0; i < mTracks.size(); i++) { 1687 sp<Track> track = mTracks[i]; 1688 if (sessionId == track->sessionId() && !track->isInvalid()) { 1689 return AudioSystem::getStrategyForStream(track->streamType()); 1690 } 1691 } 1692 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1693} 1694 1695 1696AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1697{ 1698 Mutex::Autolock _l(mLock); 1699 return mOutput; 1700} 1701 1702AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1703{ 1704 Mutex::Autolock _l(mLock); 1705 AudioStreamOut *output = mOutput; 1706 mOutput = NULL; 1707 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1708 // must push a NULL and wait for ack 1709 mOutputSink.clear(); 1710 mPipeSink.clear(); 1711 mNormalSink.clear(); 1712 return output; 1713} 1714 1715// this method must always be called either with ThreadBase mLock held or inside the thread loop 1716audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1717{ 1718 if (mOutput == NULL) { 1719 return NULL; 1720 } 1721 return &mOutput->stream->common; 1722} 1723 1724uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1725{ 1726 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1727} 1728 1729status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1730{ 1731 if (!isValidSyncEvent(event)) { 1732 return BAD_VALUE; 1733 } 1734 1735 Mutex::Autolock _l(mLock); 1736 1737 for (size_t i = 0; i < mTracks.size(); ++i) { 1738 sp<Track> track = mTracks[i]; 1739 if (event->triggerSession() == track->sessionId()) { 1740 (void) track->setSyncEvent(event); 1741 return NO_ERROR; 1742 } 1743 } 1744 1745 return NAME_NOT_FOUND; 1746} 1747 1748bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1749{ 1750 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1751} 1752 1753void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1754 const Vector< sp<Track> >& tracksToRemove) 1755{ 1756 size_t count = tracksToRemove.size(); 1757 if (count) { 1758 for (size_t i = 0 ; i < count ; i++) { 1759 const sp<Track>& track = tracksToRemove.itemAt(i); 1760 if (!track->isOutputTrack()) { 1761 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1762#ifdef ADD_BATTERY_DATA 1763 // to track the speaker usage 1764 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1765#endif 1766 if (track->isTerminated()) { 1767 AudioSystem::releaseOutput(mId); 1768 } 1769 } 1770 } 1771 } 1772} 1773 1774void AudioFlinger::PlaybackThread::checkSilentMode_l() 1775{ 1776 if (!mMasterMute) { 1777 char value[PROPERTY_VALUE_MAX]; 1778 if (property_get("ro.audio.silent", value, "0") > 0) { 1779 char *endptr; 1780 unsigned long ul = strtoul(value, &endptr, 0); 1781 if (*endptr == '\0' && ul != 0) { 1782 ALOGD("Silence is golden"); 1783 // The setprop command will not allow a property to be changed after 1784 // the first time it is set, so we don't have to worry about un-muting. 1785 setMasterMute_l(true); 1786 } 1787 } 1788 } 1789} 1790 1791// shared by MIXER and DIRECT, overridden by DUPLICATING 1792ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1793{ 1794 // FIXME rewrite to reduce number of system calls 1795 mLastWriteTime = systemTime(); 1796 mInWrite = true; 1797 ssize_t bytesWritten; 1798 1799 // If an NBAIO sink is present, use it to write the normal mixer's submix 1800 if (mNormalSink != 0) { 1801#define mBitShift 2 // FIXME 1802 size_t count = mBytesRemaining >> mBitShift; 1803 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1804 ATRACE_BEGIN("write"); 1805 // update the setpoint when AudioFlinger::mScreenState changes 1806 uint32_t screenState = AudioFlinger::mScreenState; 1807 if (screenState != mScreenState) { 1808 mScreenState = screenState; 1809 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1810 if (pipe != NULL) { 1811 pipe->setAvgFrames((mScreenState & 1) ? 1812 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1813 } 1814 } 1815 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1816 ATRACE_END(); 1817 if (framesWritten > 0) { 1818 bytesWritten = framesWritten << mBitShift; 1819 } else { 1820 bytesWritten = framesWritten; 1821 } 1822 // otherwise use the HAL / AudioStreamOut directly 1823 } else { 1824 // Direct output and offload threads 1825 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1826 if (mUseAsyncWrite) { 1827 mWriteBlocked = true; 1828 ALOG_ASSERT(mCallbackThread != 0); 1829 mCallbackThread->setWriteBlocked(true); 1830 } 1831 bytesWritten = mOutput->stream->write(mOutput->stream, 1832 mMixBuffer + offset, mBytesRemaining); 1833 if (mUseAsyncWrite && 1834 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1835 // do not wait for async callback in case of error of full write 1836 mWriteBlocked = false; 1837 ALOG_ASSERT(mCallbackThread != 0); 1838 mCallbackThread->setWriteBlocked(false); 1839 } 1840 } 1841 1842 mNumWrites++; 1843 mInWrite = false; 1844 1845 return bytesWritten; 1846} 1847 1848void AudioFlinger::PlaybackThread::threadLoop_drain() 1849{ 1850 if (mOutput->stream->drain) { 1851 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1852 if (mUseAsyncWrite) { 1853 mDraining = true; 1854 ALOG_ASSERT(mCallbackThread != 0); 1855 mCallbackThread->setDraining(true); 1856 } 1857 mOutput->stream->drain(mOutput->stream, 1858 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1859 : AUDIO_DRAIN_ALL); 1860 } 1861} 1862 1863void AudioFlinger::PlaybackThread::threadLoop_exit() 1864{ 1865 // Default implementation has nothing to do 1866} 1867 1868/* 1869The derived values that are cached: 1870 - mixBufferSize from frame count * frame size 1871 - activeSleepTime from activeSleepTimeUs() 1872 - idleSleepTime from idleSleepTimeUs() 1873 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1874 - maxPeriod from frame count and sample rate (MIXER only) 1875 1876The parameters that affect these derived values are: 1877 - frame count 1878 - frame size 1879 - sample rate 1880 - device type: A2DP or not 1881 - device latency 1882 - format: PCM or not 1883 - active sleep time 1884 - idle sleep time 1885*/ 1886 1887void AudioFlinger::PlaybackThread::cacheParameters_l() 1888{ 1889 mixBufferSize = mNormalFrameCount * mFrameSize; 1890 activeSleepTime = activeSleepTimeUs(); 1891 idleSleepTime = idleSleepTimeUs(); 1892} 1893 1894void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1895{ 1896 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1897 this, streamType, mTracks.size()); 1898 Mutex::Autolock _l(mLock); 1899 1900 size_t size = mTracks.size(); 1901 for (size_t i = 0; i < size; i++) { 1902 sp<Track> t = mTracks[i]; 1903 if (t->streamType() == streamType) { 1904 t->invalidate(); 1905 } 1906 } 1907} 1908 1909status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1910{ 1911 int session = chain->sessionId(); 1912 int16_t *buffer = mMixBuffer; 1913 bool ownsBuffer = false; 1914 1915 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1916 if (session > 0) { 1917 // Only one effect chain can be present in direct output thread and it uses 1918 // the mix buffer as input 1919 if (mType != DIRECT) { 1920 size_t numSamples = mNormalFrameCount * mChannelCount; 1921 buffer = new int16_t[numSamples]; 1922 memset(buffer, 0, numSamples * sizeof(int16_t)); 1923 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1924 ownsBuffer = true; 1925 } 1926 1927 // Attach all tracks with same session ID to this chain. 1928 for (size_t i = 0; i < mTracks.size(); ++i) { 1929 sp<Track> track = mTracks[i]; 1930 if (session == track->sessionId()) { 1931 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1932 buffer); 1933 track->setMainBuffer(buffer); 1934 chain->incTrackCnt(); 1935 } 1936 } 1937 1938 // indicate all active tracks in the chain 1939 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1940 sp<Track> track = mActiveTracks[i].promote(); 1941 if (track == 0) { 1942 continue; 1943 } 1944 if (session == track->sessionId()) { 1945 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1946 chain->incActiveTrackCnt(); 1947 } 1948 } 1949 } 1950 1951 chain->setInBuffer(buffer, ownsBuffer); 1952 chain->setOutBuffer(mMixBuffer); 1953 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1954 // chains list in order to be processed last as it contains output stage effects 1955 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1956 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1957 // after track specific effects and before output stage 1958 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1959 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1960 // Effect chain for other sessions are inserted at beginning of effect 1961 // chains list to be processed before output mix effects. Relative order between other 1962 // sessions is not important 1963 size_t size = mEffectChains.size(); 1964 size_t i = 0; 1965 for (i = 0; i < size; i++) { 1966 if (mEffectChains[i]->sessionId() < session) { 1967 break; 1968 } 1969 } 1970 mEffectChains.insertAt(chain, i); 1971 checkSuspendOnAddEffectChain_l(chain); 1972 1973 return NO_ERROR; 1974} 1975 1976size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1977{ 1978 int session = chain->sessionId(); 1979 1980 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1981 1982 for (size_t i = 0; i < mEffectChains.size(); i++) { 1983 if (chain == mEffectChains[i]) { 1984 mEffectChains.removeAt(i); 1985 // detach all active tracks from the chain 1986 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1987 sp<Track> track = mActiveTracks[i].promote(); 1988 if (track == 0) { 1989 continue; 1990 } 1991 if (session == track->sessionId()) { 1992 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1993 chain.get(), session); 1994 chain->decActiveTrackCnt(); 1995 } 1996 } 1997 1998 // detach all tracks with same session ID from this chain 1999 for (size_t i = 0; i < mTracks.size(); ++i) { 2000 sp<Track> track = mTracks[i]; 2001 if (session == track->sessionId()) { 2002 track->setMainBuffer(mMixBuffer); 2003 chain->decTrackCnt(); 2004 } 2005 } 2006 break; 2007 } 2008 } 2009 return mEffectChains.size(); 2010} 2011 2012status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2013 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2014{ 2015 Mutex::Autolock _l(mLock); 2016 return attachAuxEffect_l(track, EffectId); 2017} 2018 2019status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2020 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2021{ 2022 status_t status = NO_ERROR; 2023 2024 if (EffectId == 0) { 2025 track->setAuxBuffer(0, NULL); 2026 } else { 2027 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2028 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2029 if (effect != 0) { 2030 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2031 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2032 } else { 2033 status = INVALID_OPERATION; 2034 } 2035 } else { 2036 status = BAD_VALUE; 2037 } 2038 } 2039 return status; 2040} 2041 2042void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2043{ 2044 for (size_t i = 0; i < mTracks.size(); ++i) { 2045 sp<Track> track = mTracks[i]; 2046 if (track->auxEffectId() == effectId) { 2047 attachAuxEffect_l(track, 0); 2048 } 2049 } 2050} 2051 2052bool AudioFlinger::PlaybackThread::threadLoop() 2053{ 2054 Vector< sp<Track> > tracksToRemove; 2055 2056 standbyTime = systemTime(); 2057 2058 // MIXER 2059 nsecs_t lastWarning = 0; 2060 2061 // DUPLICATING 2062 // FIXME could this be made local to while loop? 2063 writeFrames = 0; 2064 2065 cacheParameters_l(); 2066 sleepTime = idleSleepTime; 2067 2068 if (mType == MIXER) { 2069 sleepTimeShift = 0; 2070 } 2071 2072 CpuStats cpuStats; 2073 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2074 2075 acquireWakeLock(); 2076 2077 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2078 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2079 // and then that string will be logged at the next convenient opportunity. 2080 const char *logString = NULL; 2081 2082 while (!exitPending()) 2083 { 2084 cpuStats.sample(myName); 2085 2086 Vector< sp<EffectChain> > effectChains; 2087 2088 processConfigEvents(); 2089 2090 { // scope for mLock 2091 2092 Mutex::Autolock _l(mLock); 2093 2094 if (logString != NULL) { 2095 mNBLogWriter->logTimestamp(); 2096 mNBLogWriter->log(logString); 2097 logString = NULL; 2098 } 2099 2100 if (checkForNewParameters_l()) { 2101 cacheParameters_l(); 2102 } 2103 2104 saveOutputTracks(); 2105 2106 if (mSignalPending) { 2107 // A signal was raised while we were unlocked 2108 mSignalPending = false; 2109 } else if (waitingAsyncCallback_l()) { 2110 if (exitPending()) { 2111 break; 2112 } 2113 releaseWakeLock_l(); 2114 ALOGV("wait async completion"); 2115 mWaitWorkCV.wait(mLock); 2116 ALOGV("async completion/wake"); 2117 acquireWakeLock_l(); 2118 if (exitPending()) { 2119 break; 2120 } 2121 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2122 continue; 2123 } 2124 sleepTime = 0; 2125 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2126 isSuspended()) { 2127 // put audio hardware into standby after short delay 2128 if (shouldStandby_l()) { 2129 2130 threadLoop_standby(); 2131 2132 mStandby = true; 2133 } 2134 2135 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2136 // we're about to wait, flush the binder command buffer 2137 IPCThreadState::self()->flushCommands(); 2138 2139 clearOutputTracks(); 2140 2141 if (exitPending()) { 2142 break; 2143 } 2144 2145 releaseWakeLock_l(); 2146 // wait until we have something to do... 2147 ALOGV("%s going to sleep", myName.string()); 2148 mWaitWorkCV.wait(mLock); 2149 ALOGV("%s waking up", myName.string()); 2150 acquireWakeLock_l(); 2151 2152 mMixerStatus = MIXER_IDLE; 2153 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2154 mBytesWritten = 0; 2155 mBytesRemaining = 0; 2156 checkSilentMode_l(); 2157 2158 standbyTime = systemTime() + standbyDelay; 2159 sleepTime = idleSleepTime; 2160 if (mType == MIXER) { 2161 sleepTimeShift = 0; 2162 } 2163 2164 continue; 2165 } 2166 } 2167 2168 // mMixerStatusIgnoringFastTracks is also updated internally 2169 mMixerStatus = prepareTracks_l(&tracksToRemove); 2170 2171 // prevent any changes in effect chain list and in each effect chain 2172 // during mixing and effect process as the audio buffers could be deleted 2173 // or modified if an effect is created or deleted 2174 lockEffectChains_l(effectChains); 2175 } 2176 2177 if (mBytesRemaining == 0) { 2178 mCurrentWriteLength = 0; 2179 if (mMixerStatus == MIXER_TRACKS_READY) { 2180 // threadLoop_mix() sets mCurrentWriteLength 2181 threadLoop_mix(); 2182 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2183 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2184 // threadLoop_sleepTime sets sleepTime to 0 if data 2185 // must be written to HAL 2186 threadLoop_sleepTime(); 2187 if (sleepTime == 0) { 2188 mCurrentWriteLength = mixBufferSize; 2189 } 2190 } 2191 mBytesRemaining = mCurrentWriteLength; 2192 if (isSuspended()) { 2193 sleepTime = suspendSleepTimeUs(); 2194 // simulate write to HAL when suspended 2195 mBytesWritten += mixBufferSize; 2196 mBytesRemaining = 0; 2197 } 2198 2199 // only process effects if we're going to write 2200 if (sleepTime == 0) { 2201 for (size_t i = 0; i < effectChains.size(); i ++) { 2202 effectChains[i]->process_l(); 2203 } 2204 } 2205 } 2206 2207 // enable changes in effect chain 2208 unlockEffectChains(effectChains); 2209 2210 if (!waitingAsyncCallback()) { 2211 // sleepTime == 0 means we must write to audio hardware 2212 if (sleepTime == 0) { 2213 if (mBytesRemaining) { 2214 ssize_t ret = threadLoop_write(); 2215 if (ret < 0) { 2216 mBytesRemaining = 0; 2217 } else { 2218 mBytesWritten += ret; 2219 mBytesRemaining -= ret; 2220 } 2221 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2222 (mMixerStatus == MIXER_DRAIN_ALL)) { 2223 threadLoop_drain(); 2224 } 2225if (mType == MIXER) { 2226 // write blocked detection 2227 nsecs_t now = systemTime(); 2228 nsecs_t delta = now - mLastWriteTime; 2229 if (!mStandby && delta > maxPeriod) { 2230 mNumDelayedWrites++; 2231 if ((now - lastWarning) > kWarningThrottleNs) { 2232 ATRACE_NAME("underrun"); 2233 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2234 ns2ms(delta), mNumDelayedWrites, this); 2235 lastWarning = now; 2236 } 2237 } 2238} 2239 2240 mStandby = false; 2241 } else { 2242 usleep(sleepTime); 2243 } 2244 } 2245 2246 // Finally let go of removed track(s), without the lock held 2247 // since we can't guarantee the destructors won't acquire that 2248 // same lock. This will also mutate and push a new fast mixer state. 2249 threadLoop_removeTracks(tracksToRemove); 2250 tracksToRemove.clear(); 2251 2252 // FIXME I don't understand the need for this here; 2253 // it was in the original code but maybe the 2254 // assignment in saveOutputTracks() makes this unnecessary? 2255 clearOutputTracks(); 2256 2257 // Effect chains will be actually deleted here if they were removed from 2258 // mEffectChains list during mixing or effects processing 2259 effectChains.clear(); 2260 2261 // FIXME Note that the above .clear() is no longer necessary since effectChains 2262 // is now local to this block, but will keep it for now (at least until merge done). 2263 } 2264 2265 threadLoop_exit(); 2266 2267 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2268 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2269 // put output stream into standby mode 2270 if (!mStandby) { 2271 mOutput->stream->common.standby(&mOutput->stream->common); 2272 } 2273 } 2274 2275 releaseWakeLock(); 2276 2277 ALOGV("Thread %p type %d exiting", this, mType); 2278 return false; 2279} 2280 2281// removeTracks_l() must be called with ThreadBase::mLock held 2282void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2283{ 2284 size_t count = tracksToRemove.size(); 2285 if (count) { 2286 for (size_t i=0 ; i<count ; i++) { 2287 const sp<Track>& track = tracksToRemove.itemAt(i); 2288 mActiveTracks.remove(track); 2289 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2290 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2291 if (chain != 0) { 2292 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2293 track->sessionId()); 2294 chain->decActiveTrackCnt(); 2295 } 2296 if (track->isTerminated()) { 2297 removeTrack_l(track); 2298 } 2299 } 2300 } 2301 2302} 2303 2304// ---------------------------------------------------------------------------- 2305 2306AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2307 audio_io_handle_t id, audio_devices_t device, type_t type) 2308 : PlaybackThread(audioFlinger, output, id, device, type), 2309 // mAudioMixer below 2310 // mFastMixer below 2311 mFastMixerFutex(0) 2312 // mOutputSink below 2313 // mPipeSink below 2314 // mNormalSink below 2315{ 2316 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2317 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2318 "mFrameCount=%d, mNormalFrameCount=%d", 2319 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2320 mNormalFrameCount); 2321 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2322 2323 // FIXME - Current mixer implementation only supports stereo output 2324 if (mChannelCount != FCC_2) { 2325 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2326 } 2327 2328 // create an NBAIO sink for the HAL output stream, and negotiate 2329 mOutputSink = new AudioStreamOutSink(output->stream); 2330 size_t numCounterOffers = 0; 2331 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2332 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2333 ALOG_ASSERT(index == 0); 2334 2335 // initialize fast mixer depending on configuration 2336 bool initFastMixer; 2337 switch (kUseFastMixer) { 2338 case FastMixer_Never: 2339 initFastMixer = false; 2340 break; 2341 case FastMixer_Always: 2342 initFastMixer = true; 2343 break; 2344 case FastMixer_Static: 2345 case FastMixer_Dynamic: 2346 initFastMixer = mFrameCount < mNormalFrameCount; 2347 break; 2348 } 2349 if (initFastMixer) { 2350 2351 // create a MonoPipe to connect our submix to FastMixer 2352 NBAIO_Format format = mOutputSink->format(); 2353 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2354 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2355 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2356 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2357 const NBAIO_Format offers[1] = {format}; 2358 size_t numCounterOffers = 0; 2359 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2360 ALOG_ASSERT(index == 0); 2361 monoPipe->setAvgFrames((mScreenState & 1) ? 2362 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2363 mPipeSink = monoPipe; 2364 2365#ifdef TEE_SINK 2366 if (mTeeSinkOutputEnabled) { 2367 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2368 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2369 numCounterOffers = 0; 2370 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2371 ALOG_ASSERT(index == 0); 2372 mTeeSink = teeSink; 2373 PipeReader *teeSource = new PipeReader(*teeSink); 2374 numCounterOffers = 0; 2375 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2376 ALOG_ASSERT(index == 0); 2377 mTeeSource = teeSource; 2378 } 2379#endif 2380 2381 // create fast mixer and configure it initially with just one fast track for our submix 2382 mFastMixer = new FastMixer(); 2383 FastMixerStateQueue *sq = mFastMixer->sq(); 2384#ifdef STATE_QUEUE_DUMP 2385 sq->setObserverDump(&mStateQueueObserverDump); 2386 sq->setMutatorDump(&mStateQueueMutatorDump); 2387#endif 2388 FastMixerState *state = sq->begin(); 2389 FastTrack *fastTrack = &state->mFastTracks[0]; 2390 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2391 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2392 fastTrack->mVolumeProvider = NULL; 2393 fastTrack->mGeneration++; 2394 state->mFastTracksGen++; 2395 state->mTrackMask = 1; 2396 // fast mixer will use the HAL output sink 2397 state->mOutputSink = mOutputSink.get(); 2398 state->mOutputSinkGen++; 2399 state->mFrameCount = mFrameCount; 2400 state->mCommand = FastMixerState::COLD_IDLE; 2401 // already done in constructor initialization list 2402 //mFastMixerFutex = 0; 2403 state->mColdFutexAddr = &mFastMixerFutex; 2404 state->mColdGen++; 2405 state->mDumpState = &mFastMixerDumpState; 2406#ifdef TEE_SINK 2407 state->mTeeSink = mTeeSink.get(); 2408#endif 2409 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2410 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2411 sq->end(); 2412 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2413 2414 // start the fast mixer 2415 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2416 pid_t tid = mFastMixer->getTid(); 2417 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2418 if (err != 0) { 2419 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2420 kPriorityFastMixer, getpid_cached, tid, err); 2421 } 2422 2423#ifdef AUDIO_WATCHDOG 2424 // create and start the watchdog 2425 mAudioWatchdog = new AudioWatchdog(); 2426 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2427 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2428 tid = mAudioWatchdog->getTid(); 2429 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2430 if (err != 0) { 2431 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2432 kPriorityFastMixer, getpid_cached, tid, err); 2433 } 2434#endif 2435 2436 } else { 2437 mFastMixer = NULL; 2438 } 2439 2440 switch (kUseFastMixer) { 2441 case FastMixer_Never: 2442 case FastMixer_Dynamic: 2443 mNormalSink = mOutputSink; 2444 break; 2445 case FastMixer_Always: 2446 mNormalSink = mPipeSink; 2447 break; 2448 case FastMixer_Static: 2449 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2450 break; 2451 } 2452} 2453 2454AudioFlinger::MixerThread::~MixerThread() 2455{ 2456 if (mFastMixer != NULL) { 2457 FastMixerStateQueue *sq = mFastMixer->sq(); 2458 FastMixerState *state = sq->begin(); 2459 if (state->mCommand == FastMixerState::COLD_IDLE) { 2460 int32_t old = android_atomic_inc(&mFastMixerFutex); 2461 if (old == -1) { 2462 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2463 } 2464 } 2465 state->mCommand = FastMixerState::EXIT; 2466 sq->end(); 2467 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2468 mFastMixer->join(); 2469 // Though the fast mixer thread has exited, it's state queue is still valid. 2470 // We'll use that extract the final state which contains one remaining fast track 2471 // corresponding to our sub-mix. 2472 state = sq->begin(); 2473 ALOG_ASSERT(state->mTrackMask == 1); 2474 FastTrack *fastTrack = &state->mFastTracks[0]; 2475 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2476 delete fastTrack->mBufferProvider; 2477 sq->end(false /*didModify*/); 2478 delete mFastMixer; 2479#ifdef AUDIO_WATCHDOG 2480 if (mAudioWatchdog != 0) { 2481 mAudioWatchdog->requestExit(); 2482 mAudioWatchdog->requestExitAndWait(); 2483 mAudioWatchdog.clear(); 2484 } 2485#endif 2486 } 2487 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2488 delete mAudioMixer; 2489} 2490 2491 2492uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2493{ 2494 if (mFastMixer != NULL) { 2495 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2496 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2497 } 2498 return latency; 2499} 2500 2501 2502void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2503{ 2504 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2505} 2506 2507ssize_t AudioFlinger::MixerThread::threadLoop_write() 2508{ 2509 // FIXME we should only do one push per cycle; confirm this is true 2510 // Start the fast mixer if it's not already running 2511 if (mFastMixer != NULL) { 2512 FastMixerStateQueue *sq = mFastMixer->sq(); 2513 FastMixerState *state = sq->begin(); 2514 if (state->mCommand != FastMixerState::MIX_WRITE && 2515 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2516 if (state->mCommand == FastMixerState::COLD_IDLE) { 2517 int32_t old = android_atomic_inc(&mFastMixerFutex); 2518 if (old == -1) { 2519 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2520 } 2521#ifdef AUDIO_WATCHDOG 2522 if (mAudioWatchdog != 0) { 2523 mAudioWatchdog->resume(); 2524 } 2525#endif 2526 } 2527 state->mCommand = FastMixerState::MIX_WRITE; 2528 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2529 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2530 sq->end(); 2531 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2532 if (kUseFastMixer == FastMixer_Dynamic) { 2533 mNormalSink = mPipeSink; 2534 } 2535 } else { 2536 sq->end(false /*didModify*/); 2537 } 2538 } 2539 return PlaybackThread::threadLoop_write(); 2540} 2541 2542void AudioFlinger::MixerThread::threadLoop_standby() 2543{ 2544 // Idle the fast mixer if it's currently running 2545 if (mFastMixer != NULL) { 2546 FastMixerStateQueue *sq = mFastMixer->sq(); 2547 FastMixerState *state = sq->begin(); 2548 if (!(state->mCommand & FastMixerState::IDLE)) { 2549 state->mCommand = FastMixerState::COLD_IDLE; 2550 state->mColdFutexAddr = &mFastMixerFutex; 2551 state->mColdGen++; 2552 mFastMixerFutex = 0; 2553 sq->end(); 2554 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2555 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2556 if (kUseFastMixer == FastMixer_Dynamic) { 2557 mNormalSink = mOutputSink; 2558 } 2559#ifdef AUDIO_WATCHDOG 2560 if (mAudioWatchdog != 0) { 2561 mAudioWatchdog->pause(); 2562 } 2563#endif 2564 } else { 2565 sq->end(false /*didModify*/); 2566 } 2567 } 2568 PlaybackThread::threadLoop_standby(); 2569} 2570 2571// Empty implementation for standard mixer 2572// Overridden for offloaded playback 2573void AudioFlinger::PlaybackThread::flushOutput_l() 2574{ 2575} 2576 2577bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2578{ 2579 return false; 2580} 2581 2582bool AudioFlinger::PlaybackThread::shouldStandby_l() 2583{ 2584 return !mStandby; 2585} 2586 2587bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2588{ 2589 Mutex::Autolock _l(mLock); 2590 return waitingAsyncCallback_l(); 2591} 2592 2593// shared by MIXER and DIRECT, overridden by DUPLICATING 2594void AudioFlinger::PlaybackThread::threadLoop_standby() 2595{ 2596 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2597 mOutput->stream->common.standby(&mOutput->stream->common); 2598 if (mUseAsyncWrite != 0) { 2599 mWriteBlocked = false; 2600 mDraining = false; 2601 ALOG_ASSERT(mCallbackThread != 0); 2602 mCallbackThread->setWriteBlocked(false); 2603 mCallbackThread->setDraining(false); 2604 } 2605} 2606 2607void AudioFlinger::MixerThread::threadLoop_mix() 2608{ 2609 // obtain the presentation timestamp of the next output buffer 2610 int64_t pts; 2611 status_t status = INVALID_OPERATION; 2612 2613 if (mNormalSink != 0) { 2614 status = mNormalSink->getNextWriteTimestamp(&pts); 2615 } else { 2616 status = mOutputSink->getNextWriteTimestamp(&pts); 2617 } 2618 2619 if (status != NO_ERROR) { 2620 pts = AudioBufferProvider::kInvalidPTS; 2621 } 2622 2623 // mix buffers... 2624 mAudioMixer->process(pts); 2625 mCurrentWriteLength = mixBufferSize; 2626 // increase sleep time progressively when application underrun condition clears. 2627 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2628 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2629 // such that we would underrun the audio HAL. 2630 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2631 sleepTimeShift--; 2632 } 2633 sleepTime = 0; 2634 standbyTime = systemTime() + standbyDelay; 2635 //TODO: delay standby when effects have a tail 2636} 2637 2638void AudioFlinger::MixerThread::threadLoop_sleepTime() 2639{ 2640 // If no tracks are ready, sleep once for the duration of an output 2641 // buffer size, then write 0s to the output 2642 if (sleepTime == 0) { 2643 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2644 sleepTime = activeSleepTime >> sleepTimeShift; 2645 if (sleepTime < kMinThreadSleepTimeUs) { 2646 sleepTime = kMinThreadSleepTimeUs; 2647 } 2648 // reduce sleep time in case of consecutive application underruns to avoid 2649 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2650 // duration we would end up writing less data than needed by the audio HAL if 2651 // the condition persists. 2652 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2653 sleepTimeShift++; 2654 } 2655 } else { 2656 sleepTime = idleSleepTime; 2657 } 2658 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2659 memset (mMixBuffer, 0, mixBufferSize); 2660 sleepTime = 0; 2661 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2662 "anticipated start"); 2663 } 2664 // TODO add standby time extension fct of effect tail 2665} 2666 2667// prepareTracks_l() must be called with ThreadBase::mLock held 2668AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2669 Vector< sp<Track> > *tracksToRemove) 2670{ 2671 2672 mixer_state mixerStatus = MIXER_IDLE; 2673 // find out which tracks need to be processed 2674 size_t count = mActiveTracks.size(); 2675 size_t mixedTracks = 0; 2676 size_t tracksWithEffect = 0; 2677 // counts only _active_ fast tracks 2678 size_t fastTracks = 0; 2679 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2680 2681 float masterVolume = mMasterVolume; 2682 bool masterMute = mMasterMute; 2683 2684 if (masterMute) { 2685 masterVolume = 0; 2686 } 2687 // Delegate master volume control to effect in output mix effect chain if needed 2688 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2689 if (chain != 0) { 2690 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2691 chain->setVolume_l(&v, &v); 2692 masterVolume = (float)((v + (1 << 23)) >> 24); 2693 chain.clear(); 2694 } 2695 2696 // prepare a new state to push 2697 FastMixerStateQueue *sq = NULL; 2698 FastMixerState *state = NULL; 2699 bool didModify = false; 2700 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2701 if (mFastMixer != NULL) { 2702 sq = mFastMixer->sq(); 2703 state = sq->begin(); 2704 } 2705 2706 for (size_t i=0 ; i<count ; i++) { 2707 const sp<Track> t = mActiveTracks[i].promote(); 2708 if (t == 0) { 2709 continue; 2710 } 2711 2712 // this const just means the local variable doesn't change 2713 Track* const track = t.get(); 2714 2715 // process fast tracks 2716 if (track->isFastTrack()) { 2717 2718 // It's theoretically possible (though unlikely) for a fast track to be created 2719 // and then removed within the same normal mix cycle. This is not a problem, as 2720 // the track never becomes active so it's fast mixer slot is never touched. 2721 // The converse, of removing an (active) track and then creating a new track 2722 // at the identical fast mixer slot within the same normal mix cycle, 2723 // is impossible because the slot isn't marked available until the end of each cycle. 2724 int j = track->mFastIndex; 2725 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2726 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2727 FastTrack *fastTrack = &state->mFastTracks[j]; 2728 2729 // Determine whether the track is currently in underrun condition, 2730 // and whether it had a recent underrun. 2731 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2732 FastTrackUnderruns underruns = ftDump->mUnderruns; 2733 uint32_t recentFull = (underruns.mBitFields.mFull - 2734 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2735 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2736 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2737 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2738 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2739 uint32_t recentUnderruns = recentPartial + recentEmpty; 2740 track->mObservedUnderruns = underruns; 2741 // don't count underruns that occur while stopping or pausing 2742 // or stopped which can occur when flush() is called while active 2743 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2744 recentUnderruns > 0) { 2745 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2746 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2747 } 2748 2749 // This is similar to the state machine for normal tracks, 2750 // with a few modifications for fast tracks. 2751 bool isActive = true; 2752 switch (track->mState) { 2753 case TrackBase::STOPPING_1: 2754 // track stays active in STOPPING_1 state until first underrun 2755 if (recentUnderruns > 0 || track->isTerminated()) { 2756 track->mState = TrackBase::STOPPING_2; 2757 } 2758 break; 2759 case TrackBase::PAUSING: 2760 // ramp down is not yet implemented 2761 track->setPaused(); 2762 break; 2763 case TrackBase::RESUMING: 2764 // ramp up is not yet implemented 2765 track->mState = TrackBase::ACTIVE; 2766 break; 2767 case TrackBase::ACTIVE: 2768 if (recentFull > 0 || recentPartial > 0) { 2769 // track has provided at least some frames recently: reset retry count 2770 track->mRetryCount = kMaxTrackRetries; 2771 } 2772 if (recentUnderruns == 0) { 2773 // no recent underruns: stay active 2774 break; 2775 } 2776 // there has recently been an underrun of some kind 2777 if (track->sharedBuffer() == 0) { 2778 // were any of the recent underruns "empty" (no frames available)? 2779 if (recentEmpty == 0) { 2780 // no, then ignore the partial underruns as they are allowed indefinitely 2781 break; 2782 } 2783 // there has recently been an "empty" underrun: decrement the retry counter 2784 if (--(track->mRetryCount) > 0) { 2785 break; 2786 } 2787 // indicate to client process that the track was disabled because of underrun; 2788 // it will then automatically call start() when data is available 2789 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2790 // remove from active list, but state remains ACTIVE [confusing but true] 2791 isActive = false; 2792 break; 2793 } 2794 // fall through 2795 case TrackBase::STOPPING_2: 2796 case TrackBase::PAUSED: 2797 case TrackBase::STOPPED: 2798 case TrackBase::FLUSHED: // flush() while active 2799 // Check for presentation complete if track is inactive 2800 // We have consumed all the buffers of this track. 2801 // This would be incomplete if we auto-paused on underrun 2802 { 2803 size_t audioHALFrames = 2804 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2805 size_t framesWritten = mBytesWritten / mFrameSize; 2806 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2807 // track stays in active list until presentation is complete 2808 break; 2809 } 2810 } 2811 if (track->isStopping_2()) { 2812 track->mState = TrackBase::STOPPED; 2813 } 2814 if (track->isStopped()) { 2815 // Can't reset directly, as fast mixer is still polling this track 2816 // track->reset(); 2817 // So instead mark this track as needing to be reset after push with ack 2818 resetMask |= 1 << i; 2819 } 2820 isActive = false; 2821 break; 2822 case TrackBase::IDLE: 2823 default: 2824 LOG_FATAL("unexpected track state %d", track->mState); 2825 } 2826 2827 if (isActive) { 2828 // was it previously inactive? 2829 if (!(state->mTrackMask & (1 << j))) { 2830 ExtendedAudioBufferProvider *eabp = track; 2831 VolumeProvider *vp = track; 2832 fastTrack->mBufferProvider = eabp; 2833 fastTrack->mVolumeProvider = vp; 2834 fastTrack->mSampleRate = track->mSampleRate; 2835 fastTrack->mChannelMask = track->mChannelMask; 2836 fastTrack->mGeneration++; 2837 state->mTrackMask |= 1 << j; 2838 didModify = true; 2839 // no acknowledgement required for newly active tracks 2840 } 2841 // cache the combined master volume and stream type volume for fast mixer; this 2842 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2843 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2844 ++fastTracks; 2845 } else { 2846 // was it previously active? 2847 if (state->mTrackMask & (1 << j)) { 2848 fastTrack->mBufferProvider = NULL; 2849 fastTrack->mGeneration++; 2850 state->mTrackMask &= ~(1 << j); 2851 didModify = true; 2852 // If any fast tracks were removed, we must wait for acknowledgement 2853 // because we're about to decrement the last sp<> on those tracks. 2854 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2855 } else { 2856 LOG_FATAL("fast track %d should have been active", j); 2857 } 2858 tracksToRemove->add(track); 2859 // Avoids a misleading display in dumpsys 2860 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2861 } 2862 continue; 2863 } 2864 2865 { // local variable scope to avoid goto warning 2866 2867 audio_track_cblk_t* cblk = track->cblk(); 2868 2869 // The first time a track is added we wait 2870 // for all its buffers to be filled before processing it 2871 int name = track->name(); 2872 // make sure that we have enough frames to mix one full buffer. 2873 // enforce this condition only once to enable draining the buffer in case the client 2874 // app does not call stop() and relies on underrun to stop: 2875 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2876 // during last round 2877 size_t desiredFrames; 2878 uint32_t sr = track->sampleRate(); 2879 if (sr == mSampleRate) { 2880 desiredFrames = mNormalFrameCount; 2881 } else { 2882 // +1 for rounding and +1 for additional sample needed for interpolation 2883 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2884 // add frames already consumed but not yet released by the resampler 2885 // because cblk->framesReady() will include these frames 2886 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2887 // the minimum track buffer size is normally twice the number of frames necessary 2888 // to fill one buffer and the resampler should not leave more than one buffer worth 2889 // of unreleased frames after each pass, but just in case... 2890 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2891 } 2892 uint32_t minFrames = 1; 2893 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2894 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2895 minFrames = desiredFrames; 2896 } 2897 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2898 size_t framesReady; 2899 if (track->sharedBuffer() == 0) { 2900 framesReady = track->framesReady(); 2901 } else if (track->isStopped()) { 2902 framesReady = 0; 2903 } else { 2904 framesReady = 1; 2905 } 2906 if ((framesReady >= minFrames) && track->isReady() && 2907 !track->isPaused() && !track->isTerminated()) 2908 { 2909 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2910 2911 mixedTracks++; 2912 2913 // track->mainBuffer() != mMixBuffer means there is an effect chain 2914 // connected to the track 2915 chain.clear(); 2916 if (track->mainBuffer() != mMixBuffer) { 2917 chain = getEffectChain_l(track->sessionId()); 2918 // Delegate volume control to effect in track effect chain if needed 2919 if (chain != 0) { 2920 tracksWithEffect++; 2921 } else { 2922 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2923 "session %d", 2924 name, track->sessionId()); 2925 } 2926 } 2927 2928 2929 int param = AudioMixer::VOLUME; 2930 if (track->mFillingUpStatus == Track::FS_FILLED) { 2931 // no ramp for the first volume setting 2932 track->mFillingUpStatus = Track::FS_ACTIVE; 2933 if (track->mState == TrackBase::RESUMING) { 2934 track->mState = TrackBase::ACTIVE; 2935 param = AudioMixer::RAMP_VOLUME; 2936 } 2937 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2938 // FIXME should not make a decision based on mServer 2939 } else if (cblk->mServer != 0) { 2940 // If the track is stopped before the first frame was mixed, 2941 // do not apply ramp 2942 param = AudioMixer::RAMP_VOLUME; 2943 } 2944 2945 // compute volume for this track 2946 uint32_t vl, vr, va; 2947 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2948 vl = vr = va = 0; 2949 if (track->isPausing()) { 2950 track->setPaused(); 2951 } 2952 } else { 2953 2954 // read original volumes with volume control 2955 float typeVolume = mStreamTypes[track->streamType()].volume; 2956 float v = masterVolume * typeVolume; 2957 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2958 uint32_t vlr = proxy->getVolumeLR(); 2959 vl = vlr & 0xFFFF; 2960 vr = vlr >> 16; 2961 // track volumes come from shared memory, so can't be trusted and must be clamped 2962 if (vl > MAX_GAIN_INT) { 2963 ALOGV("Track left volume out of range: %04X", vl); 2964 vl = MAX_GAIN_INT; 2965 } 2966 if (vr > MAX_GAIN_INT) { 2967 ALOGV("Track right volume out of range: %04X", vr); 2968 vr = MAX_GAIN_INT; 2969 } 2970 // now apply the master volume and stream type volume 2971 vl = (uint32_t)(v * vl) << 12; 2972 vr = (uint32_t)(v * vr) << 12; 2973 // assuming master volume and stream type volume each go up to 1.0, 2974 // vl and vr are now in 8.24 format 2975 2976 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2977 // send level comes from shared memory and so may be corrupt 2978 if (sendLevel > MAX_GAIN_INT) { 2979 ALOGV("Track send level out of range: %04X", sendLevel); 2980 sendLevel = MAX_GAIN_INT; 2981 } 2982 va = (uint32_t)(v * sendLevel); 2983 } 2984 2985 // Delegate volume control to effect in track effect chain if needed 2986 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2987 // Do not ramp volume if volume is controlled by effect 2988 param = AudioMixer::VOLUME; 2989 track->mHasVolumeController = true; 2990 } else { 2991 // force no volume ramp when volume controller was just disabled or removed 2992 // from effect chain to avoid volume spike 2993 if (track->mHasVolumeController) { 2994 param = AudioMixer::VOLUME; 2995 } 2996 track->mHasVolumeController = false; 2997 } 2998 2999 // Convert volumes from 8.24 to 4.12 format 3000 // This additional clamping is needed in case chain->setVolume_l() overshot 3001 vl = (vl + (1 << 11)) >> 12; 3002 if (vl > MAX_GAIN_INT) { 3003 vl = MAX_GAIN_INT; 3004 } 3005 vr = (vr + (1 << 11)) >> 12; 3006 if (vr > MAX_GAIN_INT) { 3007 vr = MAX_GAIN_INT; 3008 } 3009 3010 if (va > MAX_GAIN_INT) { 3011 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3012 } 3013 3014 // XXX: these things DON'T need to be done each time 3015 mAudioMixer->setBufferProvider(name, track); 3016 mAudioMixer->enable(name); 3017 3018 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3019 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3020 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3021 mAudioMixer->setParameter( 3022 name, 3023 AudioMixer::TRACK, 3024 AudioMixer::FORMAT, (void *)track->format()); 3025 mAudioMixer->setParameter( 3026 name, 3027 AudioMixer::TRACK, 3028 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3029 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3030 uint32_t maxSampleRate = mSampleRate * 2; 3031 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3032 if (reqSampleRate == 0) { 3033 reqSampleRate = mSampleRate; 3034 } else if (reqSampleRate > maxSampleRate) { 3035 reqSampleRate = maxSampleRate; 3036 } 3037 mAudioMixer->setParameter( 3038 name, 3039 AudioMixer::RESAMPLE, 3040 AudioMixer::SAMPLE_RATE, 3041 (void *)reqSampleRate); 3042 mAudioMixer->setParameter( 3043 name, 3044 AudioMixer::TRACK, 3045 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3046 mAudioMixer->setParameter( 3047 name, 3048 AudioMixer::TRACK, 3049 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3050 3051 // reset retry count 3052 track->mRetryCount = kMaxTrackRetries; 3053 3054 // If one track is ready, set the mixer ready if: 3055 // - the mixer was not ready during previous round OR 3056 // - no other track is not ready 3057 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3058 mixerStatus != MIXER_TRACKS_ENABLED) { 3059 mixerStatus = MIXER_TRACKS_READY; 3060 } 3061 } else { 3062 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3063 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3064 } 3065 // clear effect chain input buffer if an active track underruns to avoid sending 3066 // previous audio buffer again to effects 3067 chain = getEffectChain_l(track->sessionId()); 3068 if (chain != 0) { 3069 chain->clearInputBuffer(); 3070 } 3071 3072 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3073 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3074 track->isStopped() || track->isPaused()) { 3075 // We have consumed all the buffers of this track. 3076 // Remove it from the list of active tracks. 3077 // TODO: use actual buffer filling status instead of latency when available from 3078 // audio HAL 3079 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3080 size_t framesWritten = mBytesWritten / mFrameSize; 3081 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3082 if (track->isStopped()) { 3083 track->reset(); 3084 } 3085 tracksToRemove->add(track); 3086 } 3087 } else { 3088 // No buffers for this track. Give it a few chances to 3089 // fill a buffer, then remove it from active list. 3090 if (--(track->mRetryCount) <= 0) { 3091 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3092 tracksToRemove->add(track); 3093 // indicate to client process that the track was disabled because of underrun; 3094 // it will then automatically call start() when data is available 3095 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3096 // If one track is not ready, mark the mixer also not ready if: 3097 // - the mixer was ready during previous round OR 3098 // - no other track is ready 3099 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3100 mixerStatus != MIXER_TRACKS_READY) { 3101 mixerStatus = MIXER_TRACKS_ENABLED; 3102 } 3103 } 3104 mAudioMixer->disable(name); 3105 } 3106 3107 } // local variable scope to avoid goto warning 3108track_is_ready: ; 3109 3110 } 3111 3112 // Push the new FastMixer state if necessary 3113 bool pauseAudioWatchdog = false; 3114 if (didModify) { 3115 state->mFastTracksGen++; 3116 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3117 if (kUseFastMixer == FastMixer_Dynamic && 3118 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3119 state->mCommand = FastMixerState::COLD_IDLE; 3120 state->mColdFutexAddr = &mFastMixerFutex; 3121 state->mColdGen++; 3122 mFastMixerFutex = 0; 3123 if (kUseFastMixer == FastMixer_Dynamic) { 3124 mNormalSink = mOutputSink; 3125 } 3126 // If we go into cold idle, need to wait for acknowledgement 3127 // so that fast mixer stops doing I/O. 3128 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3129 pauseAudioWatchdog = true; 3130 } 3131 } 3132 if (sq != NULL) { 3133 sq->end(didModify); 3134 sq->push(block); 3135 } 3136#ifdef AUDIO_WATCHDOG 3137 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3138 mAudioWatchdog->pause(); 3139 } 3140#endif 3141 3142 // Now perform the deferred reset on fast tracks that have stopped 3143 while (resetMask != 0) { 3144 size_t i = __builtin_ctz(resetMask); 3145 ALOG_ASSERT(i < count); 3146 resetMask &= ~(1 << i); 3147 sp<Track> t = mActiveTracks[i].promote(); 3148 if (t == 0) { 3149 continue; 3150 } 3151 Track* track = t.get(); 3152 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3153 track->reset(); 3154 } 3155 3156 // remove all the tracks that need to be... 3157 removeTracks_l(*tracksToRemove); 3158 3159 // mix buffer must be cleared if all tracks are connected to an 3160 // effect chain as in this case the mixer will not write to 3161 // mix buffer and track effects will accumulate into it 3162 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3163 (mixedTracks == 0 && fastTracks > 0))) { 3164 // FIXME as a performance optimization, should remember previous zero status 3165 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3166 } 3167 3168 // if any fast tracks, then status is ready 3169 mMixerStatusIgnoringFastTracks = mixerStatus; 3170 if (fastTracks > 0) { 3171 mixerStatus = MIXER_TRACKS_READY; 3172 } 3173 return mixerStatus; 3174} 3175 3176// getTrackName_l() must be called with ThreadBase::mLock held 3177int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3178{ 3179 return mAudioMixer->getTrackName(channelMask, sessionId); 3180} 3181 3182// deleteTrackName_l() must be called with ThreadBase::mLock held 3183void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3184{ 3185 ALOGV("remove track (%d) and delete from mixer", name); 3186 mAudioMixer->deleteTrackName(name); 3187} 3188 3189// checkForNewParameters_l() must be called with ThreadBase::mLock held 3190bool AudioFlinger::MixerThread::checkForNewParameters_l() 3191{ 3192 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3193 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3194 bool reconfig = false; 3195 3196 while (!mNewParameters.isEmpty()) { 3197 3198 if (mFastMixer != NULL) { 3199 FastMixerStateQueue *sq = mFastMixer->sq(); 3200 FastMixerState *state = sq->begin(); 3201 if (!(state->mCommand & FastMixerState::IDLE)) { 3202 previousCommand = state->mCommand; 3203 state->mCommand = FastMixerState::HOT_IDLE; 3204 sq->end(); 3205 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3206 } else { 3207 sq->end(false /*didModify*/); 3208 } 3209 } 3210 3211 status_t status = NO_ERROR; 3212 String8 keyValuePair = mNewParameters[0]; 3213 AudioParameter param = AudioParameter(keyValuePair); 3214 int value; 3215 3216 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3217 reconfig = true; 3218 } 3219 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3220 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3221 status = BAD_VALUE; 3222 } else { 3223 reconfig = true; 3224 } 3225 } 3226 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3227 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3228 status = BAD_VALUE; 3229 } else { 3230 reconfig = true; 3231 } 3232 } 3233 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3234 // do not accept frame count changes if tracks are open as the track buffer 3235 // size depends on frame count and correct behavior would not be guaranteed 3236 // if frame count is changed after track creation 3237 if (!mTracks.isEmpty()) { 3238 status = INVALID_OPERATION; 3239 } else { 3240 reconfig = true; 3241 } 3242 } 3243 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3244#ifdef ADD_BATTERY_DATA 3245 // when changing the audio output device, call addBatteryData to notify 3246 // the change 3247 if (mOutDevice != value) { 3248 uint32_t params = 0; 3249 // check whether speaker is on 3250 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3251 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3252 } 3253 3254 audio_devices_t deviceWithoutSpeaker 3255 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3256 // check if any other device (except speaker) is on 3257 if (value & deviceWithoutSpeaker ) { 3258 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3259 } 3260 3261 if (params != 0) { 3262 addBatteryData(params); 3263 } 3264 } 3265#endif 3266 3267 // forward device change to effects that have requested to be 3268 // aware of attached audio device. 3269 if (value != AUDIO_DEVICE_NONE) { 3270 mOutDevice = value; 3271 for (size_t i = 0; i < mEffectChains.size(); i++) { 3272 mEffectChains[i]->setDevice_l(mOutDevice); 3273 } 3274 } 3275 } 3276 3277 if (status == NO_ERROR) { 3278 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3279 keyValuePair.string()); 3280 if (!mStandby && status == INVALID_OPERATION) { 3281 mOutput->stream->common.standby(&mOutput->stream->common); 3282 mStandby = true; 3283 mBytesWritten = 0; 3284 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3285 keyValuePair.string()); 3286 } 3287 if (status == NO_ERROR && reconfig) { 3288 readOutputParameters(); 3289 delete mAudioMixer; 3290 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3291 for (size_t i = 0; i < mTracks.size() ; i++) { 3292 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3293 if (name < 0) { 3294 break; 3295 } 3296 mTracks[i]->mName = name; 3297 } 3298 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3299 } 3300 } 3301 3302 mNewParameters.removeAt(0); 3303 3304 mParamStatus = status; 3305 mParamCond.signal(); 3306 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3307 // already timed out waiting for the status and will never signal the condition. 3308 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3309 } 3310 3311 if (!(previousCommand & FastMixerState::IDLE)) { 3312 ALOG_ASSERT(mFastMixer != NULL); 3313 FastMixerStateQueue *sq = mFastMixer->sq(); 3314 FastMixerState *state = sq->begin(); 3315 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3316 state->mCommand = previousCommand; 3317 sq->end(); 3318 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3319 } 3320 3321 return reconfig; 3322} 3323 3324 3325void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3326{ 3327 const size_t SIZE = 256; 3328 char buffer[SIZE]; 3329 String8 result; 3330 3331 PlaybackThread::dumpInternals(fd, args); 3332 3333 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3334 result.append(buffer); 3335 write(fd, result.string(), result.size()); 3336 3337 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3338 const FastMixerDumpState copy(mFastMixerDumpState); 3339 copy.dump(fd); 3340 3341#ifdef STATE_QUEUE_DUMP 3342 // Similar for state queue 3343 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3344 observerCopy.dump(fd); 3345 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3346 mutatorCopy.dump(fd); 3347#endif 3348 3349#ifdef TEE_SINK 3350 // Write the tee output to a .wav file 3351 dumpTee(fd, mTeeSource, mId); 3352#endif 3353 3354#ifdef AUDIO_WATCHDOG 3355 if (mAudioWatchdog != 0) { 3356 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3357 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3358 wdCopy.dump(fd); 3359 } 3360#endif 3361} 3362 3363uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3364{ 3365 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3366} 3367 3368uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3369{ 3370 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3371} 3372 3373void AudioFlinger::MixerThread::cacheParameters_l() 3374{ 3375 PlaybackThread::cacheParameters_l(); 3376 3377 // FIXME: Relaxed timing because of a certain device that can't meet latency 3378 // Should be reduced to 2x after the vendor fixes the driver issue 3379 // increase threshold again due to low power audio mode. The way this warning 3380 // threshold is calculated and its usefulness should be reconsidered anyway. 3381 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3382} 3383 3384// ---------------------------------------------------------------------------- 3385 3386AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3387 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3388 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3389 // mLeftVolFloat, mRightVolFloat 3390{ 3391} 3392 3393AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3394 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3395 ThreadBase::type_t type) 3396 : PlaybackThread(audioFlinger, output, id, device, type) 3397 // mLeftVolFloat, mRightVolFloat 3398{ 3399} 3400 3401AudioFlinger::DirectOutputThread::~DirectOutputThread() 3402{ 3403} 3404 3405void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3406{ 3407 audio_track_cblk_t* cblk = track->cblk(); 3408 float left, right; 3409 3410 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3411 left = right = 0; 3412 } else { 3413 float typeVolume = mStreamTypes[track->streamType()].volume; 3414 float v = mMasterVolume * typeVolume; 3415 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3416 uint32_t vlr = proxy->getVolumeLR(); 3417 float v_clamped = v * (vlr & 0xFFFF); 3418 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3419 left = v_clamped/MAX_GAIN; 3420 v_clamped = v * (vlr >> 16); 3421 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3422 right = v_clamped/MAX_GAIN; 3423 } 3424 3425 if (lastTrack) { 3426 if (left != mLeftVolFloat || right != mRightVolFloat) { 3427 mLeftVolFloat = left; 3428 mRightVolFloat = right; 3429 3430 // Convert volumes from float to 8.24 3431 uint32_t vl = (uint32_t)(left * (1 << 24)); 3432 uint32_t vr = (uint32_t)(right * (1 << 24)); 3433 3434 // Delegate volume control to effect in track effect chain if needed 3435 // only one effect chain can be present on DirectOutputThread, so if 3436 // there is one, the track is connected to it 3437 if (!mEffectChains.isEmpty()) { 3438 mEffectChains[0]->setVolume_l(&vl, &vr); 3439 left = (float)vl / (1 << 24); 3440 right = (float)vr / (1 << 24); 3441 } 3442 if (mOutput->stream->set_volume) { 3443 mOutput->stream->set_volume(mOutput->stream, left, right); 3444 } 3445 } 3446 } 3447} 3448 3449 3450AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3451 Vector< sp<Track> > *tracksToRemove 3452) 3453{ 3454 size_t count = mActiveTracks.size(); 3455 mixer_state mixerStatus = MIXER_IDLE; 3456 3457 // find out which tracks need to be processed 3458 for (size_t i = 0; i < count; i++) { 3459 sp<Track> t = mActiveTracks[i].promote(); 3460 // The track died recently 3461 if (t == 0) { 3462 continue; 3463 } 3464 3465 Track* const track = t.get(); 3466 audio_track_cblk_t* cblk = track->cblk(); 3467 3468 // The first time a track is added we wait 3469 // for all its buffers to be filled before processing it 3470 uint32_t minFrames; 3471 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3472 minFrames = mNormalFrameCount; 3473 } else { 3474 minFrames = 1; 3475 } 3476 // Only consider last track started for volume and mixer state control. 3477 // This is the last entry in mActiveTracks unless a track underruns. 3478 // As we only care about the transition phase between two tracks on a 3479 // direct output, it is not a problem to ignore the underrun case. 3480 bool last = (i == (count - 1)); 3481 3482 if ((track->framesReady() >= minFrames) && track->isReady() && 3483 !track->isPaused() && !track->isTerminated()) 3484 { 3485 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3486 3487 if (track->mFillingUpStatus == Track::FS_FILLED) { 3488 track->mFillingUpStatus = Track::FS_ACTIVE; 3489 mLeftVolFloat = mRightVolFloat = 0; 3490 if (track->mState == TrackBase::RESUMING) { 3491 track->mState = TrackBase::ACTIVE; 3492 } 3493 } 3494 3495 // compute volume for this track 3496 processVolume_l(track, last); 3497 if (last) { 3498 // reset retry count 3499 track->mRetryCount = kMaxTrackRetriesDirect; 3500 mActiveTrack = t; 3501 mixerStatus = MIXER_TRACKS_READY; 3502 } 3503 } else { 3504 // clear effect chain input buffer if the last active track started underruns 3505 // to avoid sending previous audio buffer again to effects 3506 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3507 mEffectChains[0]->clearInputBuffer(); 3508 } 3509 3510 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3511 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3512 track->isStopped() || track->isPaused()) { 3513 // We have consumed all the buffers of this track. 3514 // Remove it from the list of active tracks. 3515 // TODO: implement behavior for compressed audio 3516 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3517 size_t framesWritten = mBytesWritten / mFrameSize; 3518 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3519 if (track->isStopped()) { 3520 track->reset(); 3521 } 3522 tracksToRemove->add(track); 3523 } 3524 } else { 3525 // No buffers for this track. Give it a few chances to 3526 // fill a buffer, then remove it from active list. 3527 // Only consider last track started for mixer state control 3528 if (--(track->mRetryCount) <= 0) { 3529 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3530 tracksToRemove->add(track); 3531 } else if (last) { 3532 mixerStatus = MIXER_TRACKS_ENABLED; 3533 } 3534 } 3535 } 3536 } 3537 3538 // remove all the tracks that need to be... 3539 removeTracks_l(*tracksToRemove); 3540 3541 return mixerStatus; 3542} 3543 3544void AudioFlinger::DirectOutputThread::threadLoop_mix() 3545{ 3546 size_t frameCount = mFrameCount; 3547 int8_t *curBuf = (int8_t *)mMixBuffer; 3548 // output audio to hardware 3549 while (frameCount) { 3550 AudioBufferProvider::Buffer buffer; 3551 buffer.frameCount = frameCount; 3552 mActiveTrack->getNextBuffer(&buffer); 3553 if (buffer.raw == NULL) { 3554 memset(curBuf, 0, frameCount * mFrameSize); 3555 break; 3556 } 3557 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3558 frameCount -= buffer.frameCount; 3559 curBuf += buffer.frameCount * mFrameSize; 3560 mActiveTrack->releaseBuffer(&buffer); 3561 } 3562 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3563 sleepTime = 0; 3564 standbyTime = systemTime() + standbyDelay; 3565 mActiveTrack.clear(); 3566} 3567 3568void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3569{ 3570 if (sleepTime == 0) { 3571 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3572 sleepTime = activeSleepTime; 3573 } else { 3574 sleepTime = idleSleepTime; 3575 } 3576 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3577 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3578 sleepTime = 0; 3579 } 3580} 3581 3582// getTrackName_l() must be called with ThreadBase::mLock held 3583int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3584 int sessionId) 3585{ 3586 return 0; 3587} 3588 3589// deleteTrackName_l() must be called with ThreadBase::mLock held 3590void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3591{ 3592} 3593 3594// checkForNewParameters_l() must be called with ThreadBase::mLock held 3595bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3596{ 3597 bool reconfig = false; 3598 3599 while (!mNewParameters.isEmpty()) { 3600 status_t status = NO_ERROR; 3601 String8 keyValuePair = mNewParameters[0]; 3602 AudioParameter param = AudioParameter(keyValuePair); 3603 int value; 3604 3605 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3606 // do not accept frame count changes if tracks are open as the track buffer 3607 // size depends on frame count and correct behavior would not be garantied 3608 // if frame count is changed after track creation 3609 if (!mTracks.isEmpty()) { 3610 status = INVALID_OPERATION; 3611 } else { 3612 reconfig = true; 3613 } 3614 } 3615 if (status == NO_ERROR) { 3616 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3617 keyValuePair.string()); 3618 if (!mStandby && status == INVALID_OPERATION) { 3619 mOutput->stream->common.standby(&mOutput->stream->common); 3620 mStandby = true; 3621 mBytesWritten = 0; 3622 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3623 keyValuePair.string()); 3624 } 3625 if (status == NO_ERROR && reconfig) { 3626 readOutputParameters(); 3627 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3628 } 3629 } 3630 3631 mNewParameters.removeAt(0); 3632 3633 mParamStatus = status; 3634 mParamCond.signal(); 3635 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3636 // already timed out waiting for the status and will never signal the condition. 3637 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3638 } 3639 return reconfig; 3640} 3641 3642uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3643{ 3644 uint32_t time; 3645 if (audio_is_linear_pcm(mFormat)) { 3646 time = PlaybackThread::activeSleepTimeUs(); 3647 } else { 3648 time = 10000; 3649 } 3650 return time; 3651} 3652 3653uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3654{ 3655 uint32_t time; 3656 if (audio_is_linear_pcm(mFormat)) { 3657 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3658 } else { 3659 time = 10000; 3660 } 3661 return time; 3662} 3663 3664uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3665{ 3666 uint32_t time; 3667 if (audio_is_linear_pcm(mFormat)) { 3668 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3669 } else { 3670 time = 10000; 3671 } 3672 return time; 3673} 3674 3675void AudioFlinger::DirectOutputThread::cacheParameters_l() 3676{ 3677 PlaybackThread::cacheParameters_l(); 3678 3679 // use shorter standby delay as on normal output to release 3680 // hardware resources as soon as possible 3681 standbyDelay = microseconds(activeSleepTime*2); 3682} 3683 3684// ---------------------------------------------------------------------------- 3685 3686AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3687 const sp<AudioFlinger::OffloadThread>& offloadThread) 3688 : Thread(false /*canCallJava*/), 3689 mOffloadThread(offloadThread), 3690 mWriteBlocked(false), 3691 mDraining(false) 3692{ 3693} 3694 3695AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3696{ 3697} 3698 3699void AudioFlinger::AsyncCallbackThread::onFirstRef() 3700{ 3701 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3702} 3703 3704bool AudioFlinger::AsyncCallbackThread::threadLoop() 3705{ 3706 while (!exitPending()) { 3707 bool writeBlocked; 3708 bool draining; 3709 3710 { 3711 Mutex::Autolock _l(mLock); 3712 mWaitWorkCV.wait(mLock); 3713 if (exitPending()) { 3714 break; 3715 } 3716 writeBlocked = mWriteBlocked; 3717 draining = mDraining; 3718 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3719 } 3720 { 3721 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3722 if (offloadThread != 0) { 3723 if (writeBlocked == false) { 3724 offloadThread->setWriteBlocked(false); 3725 } 3726 if (draining == false) { 3727 offloadThread->setDraining(false); 3728 } 3729 } 3730 } 3731 } 3732 return false; 3733} 3734 3735void AudioFlinger::AsyncCallbackThread::exit() 3736{ 3737 ALOGV("AsyncCallbackThread::exit"); 3738 Mutex::Autolock _l(mLock); 3739 requestExit(); 3740 mWaitWorkCV.broadcast(); 3741} 3742 3743void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3744{ 3745 Mutex::Autolock _l(mLock); 3746 mWriteBlocked = value; 3747 if (!value) { 3748 mWaitWorkCV.signal(); 3749 } 3750} 3751 3752void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3753{ 3754 Mutex::Autolock _l(mLock); 3755 mDraining = value; 3756 if (!value) { 3757 mWaitWorkCV.signal(); 3758 } 3759} 3760 3761 3762// ---------------------------------------------------------------------------- 3763AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3764 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3765 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3766 mHwPaused(false), 3767 mPausedBytesRemaining(0) 3768{ 3769 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3770} 3771 3772AudioFlinger::OffloadThread::~OffloadThread() 3773{ 3774 mPreviousTrack.clear(); 3775} 3776 3777void AudioFlinger::OffloadThread::threadLoop_exit() 3778{ 3779 if (mFlushPending || mHwPaused) { 3780 // If a flush is pending or track was paused, just discard buffered data 3781 flushHw_l(); 3782 } else { 3783 mMixerStatus = MIXER_DRAIN_ALL; 3784 threadLoop_drain(); 3785 } 3786 mCallbackThread->exit(); 3787 PlaybackThread::threadLoop_exit(); 3788} 3789 3790AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3791 Vector< sp<Track> > *tracksToRemove 3792) 3793{ 3794 ALOGV("OffloadThread::prepareTracks_l"); 3795 size_t count = mActiveTracks.size(); 3796 3797 mixer_state mixerStatus = MIXER_IDLE; 3798 if (mFlushPending) { 3799 flushHw_l(); 3800 mFlushPending = false; 3801 } 3802 // find out which tracks need to be processed 3803 for (size_t i = 0; i < count; i++) { 3804 sp<Track> t = mActiveTracks[i].promote(); 3805 // The track died recently 3806 if (t == 0) { 3807 continue; 3808 } 3809 Track* const track = t.get(); 3810 audio_track_cblk_t* cblk = track->cblk(); 3811 if (mPreviousTrack != NULL) { 3812 if (t != mPreviousTrack) { 3813 // Flush any data still being written from last track 3814 mBytesRemaining = 0; 3815 if (mPausedBytesRemaining) { 3816 // Last track was paused so we also need to flush saved 3817 // mixbuffer state and invalidate track so that it will 3818 // re-submit that unwritten data when it is next resumed 3819 mPausedBytesRemaining = 0; 3820 // Invalidate is a bit drastic - would be more efficient 3821 // to have a flag to tell client that some of the 3822 // previously written data was lost 3823 mPreviousTrack->invalidate(); 3824 } 3825 } 3826 } 3827 mPreviousTrack = t; 3828 bool last = (i == (count - 1)); 3829 if (track->isPausing()) { 3830 track->setPaused(); 3831 if (last) { 3832 if (!mHwPaused) { 3833 mOutput->stream->pause(mOutput->stream); 3834 mHwPaused = true; 3835 } 3836 // If we were part way through writing the mixbuffer to 3837 // the HAL we must save this until we resume 3838 // BUG - this will be wrong if a different track is made active, 3839 // in that case we want to discard the pending data in the 3840 // mixbuffer and tell the client to present it again when the 3841 // track is resumed 3842 mPausedWriteLength = mCurrentWriteLength; 3843 mPausedBytesRemaining = mBytesRemaining; 3844 mBytesRemaining = 0; // stop writing 3845 } 3846 tracksToRemove->add(track); 3847 } else if (track->framesReady() && track->isReady() && 3848 !track->isPaused() && !track->isTerminated()) { 3849 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3850 if (track->mFillingUpStatus == Track::FS_FILLED) { 3851 track->mFillingUpStatus = Track::FS_ACTIVE; 3852 mLeftVolFloat = mRightVolFloat = 0; 3853 if (track->mState == TrackBase::RESUMING) { 3854 if (mPausedBytesRemaining) { 3855 // Need to continue write that was interrupted 3856 mCurrentWriteLength = mPausedWriteLength; 3857 mBytesRemaining = mPausedBytesRemaining; 3858 mPausedBytesRemaining = 0; 3859 } 3860 track->mState = TrackBase::ACTIVE; 3861 } 3862 } 3863 3864 if (last) { 3865 if (mHwPaused) { 3866 mOutput->stream->resume(mOutput->stream); 3867 mHwPaused = false; 3868 // threadLoop_mix() will handle the case that we need to 3869 // resume an interrupted write 3870 } 3871 // reset retry count 3872 track->mRetryCount = kMaxTrackRetriesOffload; 3873 mActiveTrack = t; 3874 mixerStatus = MIXER_TRACKS_READY; 3875 } 3876 } else { 3877 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3878 if (track->isStopping_1()) { 3879 // Hardware buffer can hold a large amount of audio so we must 3880 // wait for all current track's data to drain before we say 3881 // that the track is stopped. 3882 if (mBytesRemaining == 0) { 3883 // Only start draining when all data in mixbuffer 3884 // has been written 3885 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3886 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3887 sleepTime = 0; 3888 standbyTime = systemTime() + standbyDelay; 3889 if (last) { 3890 mixerStatus = MIXER_DRAIN_TRACK; 3891 if (mHwPaused) { 3892 // It is possible to move from PAUSED to STOPPING_1 without 3893 // a resume so we must ensure hardware is running 3894 mOutput->stream->resume(mOutput->stream); 3895 mHwPaused = false; 3896 } 3897 } 3898 } 3899 } else if (track->isStopping_2()) { 3900 // Drain has completed, signal presentation complete 3901 if (!mDraining || !last) { 3902 track->mState = TrackBase::STOPPED; 3903 size_t audioHALFrames = 3904 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3905 size_t framesWritten = 3906 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3907 track->presentationComplete(framesWritten, audioHALFrames); 3908 track->reset(); 3909 tracksToRemove->add(track); 3910 } 3911 } else { 3912 // No buffers for this track. Give it a few chances to 3913 // fill a buffer, then remove it from active list. 3914 if (--(track->mRetryCount) <= 0) { 3915 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3916 track->name()); 3917 tracksToRemove->add(track); 3918 } else if (last){ 3919 mixerStatus = MIXER_TRACKS_ENABLED; 3920 } 3921 } 3922 } 3923 // compute volume for this track 3924 processVolume_l(track, last); 3925 } 3926 // remove all the tracks that need to be... 3927 removeTracks_l(*tracksToRemove); 3928 3929 return mixerStatus; 3930} 3931 3932void AudioFlinger::OffloadThread::flushOutput_l() 3933{ 3934 mFlushPending = true; 3935} 3936 3937// must be called with thread mutex locked 3938bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3939{ 3940 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3941 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3942 return true; 3943 } 3944 return false; 3945} 3946 3947// must be called with thread mutex locked 3948bool AudioFlinger::OffloadThread::shouldStandby_l() 3949{ 3950 bool TrackPaused = false; 3951 3952 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3953 // after a timeout and we will enter standby then. 3954 if (mTracks.size() > 0) { 3955 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3956 } 3957 3958 return !mStandby && !TrackPaused; 3959} 3960 3961 3962bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3963{ 3964 Mutex::Autolock _l(mLock); 3965 return waitingAsyncCallback_l(); 3966} 3967 3968void AudioFlinger::OffloadThread::flushHw_l() 3969{ 3970 mOutput->stream->flush(mOutput->stream); 3971 // Flush anything still waiting in the mixbuffer 3972 mCurrentWriteLength = 0; 3973 mBytesRemaining = 0; 3974 mPausedWriteLength = 0; 3975 mPausedBytesRemaining = 0; 3976 if (mUseAsyncWrite) { 3977 mWriteBlocked = false; 3978 mDraining = false; 3979 ALOG_ASSERT(mCallbackThread != 0); 3980 mCallbackThread->setWriteBlocked(false); 3981 mCallbackThread->setDraining(false); 3982 } 3983} 3984 3985// ---------------------------------------------------------------------------- 3986 3987AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3988 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3989 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3990 DUPLICATING), 3991 mWaitTimeMs(UINT_MAX) 3992{ 3993 addOutputTrack(mainThread); 3994} 3995 3996AudioFlinger::DuplicatingThread::~DuplicatingThread() 3997{ 3998 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3999 mOutputTracks[i]->destroy(); 4000 } 4001} 4002 4003void AudioFlinger::DuplicatingThread::threadLoop_mix() 4004{ 4005 // mix buffers... 4006 if (outputsReady(outputTracks)) { 4007 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4008 } else { 4009 memset(mMixBuffer, 0, mixBufferSize); 4010 } 4011 sleepTime = 0; 4012 writeFrames = mNormalFrameCount; 4013 mCurrentWriteLength = mixBufferSize; 4014 standbyTime = systemTime() + standbyDelay; 4015} 4016 4017void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4018{ 4019 if (sleepTime == 0) { 4020 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4021 sleepTime = activeSleepTime; 4022 } else { 4023 sleepTime = idleSleepTime; 4024 } 4025 } else if (mBytesWritten != 0) { 4026 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4027 writeFrames = mNormalFrameCount; 4028 memset(mMixBuffer, 0, mixBufferSize); 4029 } else { 4030 // flush remaining overflow buffers in output tracks 4031 writeFrames = 0; 4032 } 4033 sleepTime = 0; 4034 } 4035} 4036 4037ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4038{ 4039 for (size_t i = 0; i < outputTracks.size(); i++) { 4040 outputTracks[i]->write(mMixBuffer, writeFrames); 4041 } 4042 return (ssize_t)mixBufferSize; 4043} 4044 4045void AudioFlinger::DuplicatingThread::threadLoop_standby() 4046{ 4047 // DuplicatingThread implements standby by stopping all tracks 4048 for (size_t i = 0; i < outputTracks.size(); i++) { 4049 outputTracks[i]->stop(); 4050 } 4051} 4052 4053void AudioFlinger::DuplicatingThread::saveOutputTracks() 4054{ 4055 outputTracks = mOutputTracks; 4056} 4057 4058void AudioFlinger::DuplicatingThread::clearOutputTracks() 4059{ 4060 outputTracks.clear(); 4061} 4062 4063void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4064{ 4065 Mutex::Autolock _l(mLock); 4066 // FIXME explain this formula 4067 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4068 OutputTrack *outputTrack = new OutputTrack(thread, 4069 this, 4070 mSampleRate, 4071 mFormat, 4072 mChannelMask, 4073 frameCount); 4074 if (outputTrack->cblk() != NULL) { 4075 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4076 mOutputTracks.add(outputTrack); 4077 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4078 updateWaitTime_l(); 4079 } 4080} 4081 4082void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4083{ 4084 Mutex::Autolock _l(mLock); 4085 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4086 if (mOutputTracks[i]->thread() == thread) { 4087 mOutputTracks[i]->destroy(); 4088 mOutputTracks.removeAt(i); 4089 updateWaitTime_l(); 4090 return; 4091 } 4092 } 4093 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4094} 4095 4096// caller must hold mLock 4097void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4098{ 4099 mWaitTimeMs = UINT_MAX; 4100 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4101 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4102 if (strong != 0) { 4103 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4104 if (waitTimeMs < mWaitTimeMs) { 4105 mWaitTimeMs = waitTimeMs; 4106 } 4107 } 4108 } 4109} 4110 4111 4112bool AudioFlinger::DuplicatingThread::outputsReady( 4113 const SortedVector< sp<OutputTrack> > &outputTracks) 4114{ 4115 for (size_t i = 0; i < outputTracks.size(); i++) { 4116 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4117 if (thread == 0) { 4118 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4119 outputTracks[i].get()); 4120 return false; 4121 } 4122 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4123 // see note at standby() declaration 4124 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4125 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4126 thread.get()); 4127 return false; 4128 } 4129 } 4130 return true; 4131} 4132 4133uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4134{ 4135 return (mWaitTimeMs * 1000) / 2; 4136} 4137 4138void AudioFlinger::DuplicatingThread::cacheParameters_l() 4139{ 4140 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4141 updateWaitTime_l(); 4142 4143 MixerThread::cacheParameters_l(); 4144} 4145 4146// ---------------------------------------------------------------------------- 4147// Record 4148// ---------------------------------------------------------------------------- 4149 4150AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4151 AudioStreamIn *input, 4152 uint32_t sampleRate, 4153 audio_channel_mask_t channelMask, 4154 audio_io_handle_t id, 4155 audio_devices_t outDevice, 4156 audio_devices_t inDevice 4157#ifdef TEE_SINK 4158 , const sp<NBAIO_Sink>& teeSink 4159#endif 4160 ) : 4161 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4162 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4163 // mRsmpInIndex and mBufferSize set by readInputParameters() 4164 mReqChannelCount(popcount(channelMask)), 4165 mReqSampleRate(sampleRate) 4166 // mBytesRead is only meaningful while active, and so is cleared in start() 4167 // (but might be better to also clear here for dump?) 4168#ifdef TEE_SINK 4169 , mTeeSink(teeSink) 4170#endif 4171{ 4172 snprintf(mName, kNameLength, "AudioIn_%X", id); 4173 4174 readInputParameters(); 4175 4176} 4177 4178 4179AudioFlinger::RecordThread::~RecordThread() 4180{ 4181 delete[] mRsmpInBuffer; 4182 delete mResampler; 4183 delete[] mRsmpOutBuffer; 4184} 4185 4186void AudioFlinger::RecordThread::onFirstRef() 4187{ 4188 run(mName, PRIORITY_URGENT_AUDIO); 4189} 4190 4191bool AudioFlinger::RecordThread::threadLoop() 4192{ 4193 AudioBufferProvider::Buffer buffer; 4194 sp<RecordTrack> activeTrack; 4195 Vector< sp<EffectChain> > effectChains; 4196 4197 nsecs_t lastWarning = 0; 4198 4199 inputStandBy(); 4200 acquireWakeLock(); 4201 4202 // used to verify we've read at least once before evaluating how many bytes were read 4203 bool readOnce = false; 4204 4205 // start recording 4206 while (!exitPending()) { 4207 4208 processConfigEvents(); 4209 4210 { // scope for mLock 4211 Mutex::Autolock _l(mLock); 4212 checkForNewParameters_l(); 4213 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4214 standby(); 4215 4216 if (exitPending()) { 4217 break; 4218 } 4219 4220 releaseWakeLock_l(); 4221 ALOGV("RecordThread: loop stopping"); 4222 // go to sleep 4223 mWaitWorkCV.wait(mLock); 4224 ALOGV("RecordThread: loop starting"); 4225 acquireWakeLock_l(); 4226 continue; 4227 } 4228 if (mActiveTrack != 0) { 4229 if (mActiveTrack->isTerminated()) { 4230 removeTrack_l(mActiveTrack); 4231 mActiveTrack.clear(); 4232 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4233 standby(); 4234 mActiveTrack.clear(); 4235 mStartStopCond.broadcast(); 4236 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4237 if (mReqChannelCount != mActiveTrack->channelCount()) { 4238 mActiveTrack.clear(); 4239 mStartStopCond.broadcast(); 4240 } else if (readOnce) { 4241 // record start succeeds only if first read from audio input 4242 // succeeds 4243 if (mBytesRead >= 0) { 4244 mActiveTrack->mState = TrackBase::ACTIVE; 4245 } else { 4246 mActiveTrack.clear(); 4247 } 4248 mStartStopCond.broadcast(); 4249 } 4250 mStandby = false; 4251 } 4252 } 4253 lockEffectChains_l(effectChains); 4254 } 4255 4256 if (mActiveTrack != 0) { 4257 if (mActiveTrack->mState != TrackBase::ACTIVE && 4258 mActiveTrack->mState != TrackBase::RESUMING) { 4259 unlockEffectChains(effectChains); 4260 usleep(kRecordThreadSleepUs); 4261 continue; 4262 } 4263 for (size_t i = 0; i < effectChains.size(); i ++) { 4264 effectChains[i]->process_l(); 4265 } 4266 4267 buffer.frameCount = mFrameCount; 4268 status_t status = mActiveTrack->getNextBuffer(&buffer); 4269 if (status == NO_ERROR) { 4270 readOnce = true; 4271 size_t framesOut = buffer.frameCount; 4272 if (mResampler == NULL) { 4273 // no resampling 4274 while (framesOut) { 4275 size_t framesIn = mFrameCount - mRsmpInIndex; 4276 if (framesIn) { 4277 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4278 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4279 mActiveTrack->mFrameSize; 4280 if (framesIn > framesOut) 4281 framesIn = framesOut; 4282 mRsmpInIndex += framesIn; 4283 framesOut -= framesIn; 4284 if (mChannelCount == mReqChannelCount) { 4285 memcpy(dst, src, framesIn * mFrameSize); 4286 } else { 4287 if (mChannelCount == 1) { 4288 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4289 (int16_t *)src, framesIn); 4290 } else { 4291 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4292 (int16_t *)src, framesIn); 4293 } 4294 } 4295 } 4296 if (framesOut && mFrameCount == mRsmpInIndex) { 4297 void *readInto; 4298 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4299 readInto = buffer.raw; 4300 framesOut = 0; 4301 } else { 4302 readInto = mRsmpInBuffer; 4303 mRsmpInIndex = 0; 4304 } 4305 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4306 mBufferSize); 4307 if (mBytesRead <= 0) { 4308 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4309 { 4310 ALOGE("Error reading audio input"); 4311 // Force input into standby so that it tries to 4312 // recover at next read attempt 4313 inputStandBy(); 4314 usleep(kRecordThreadSleepUs); 4315 } 4316 mRsmpInIndex = mFrameCount; 4317 framesOut = 0; 4318 buffer.frameCount = 0; 4319 } 4320#ifdef TEE_SINK 4321 else if (mTeeSink != 0) { 4322 (void) mTeeSink->write(readInto, 4323 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4324 } 4325#endif 4326 } 4327 } 4328 } else { 4329 // resampling 4330 4331 // resampler accumulates, but we only have one source track 4332 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4333 // alter output frame count as if we were expecting stereo samples 4334 if (mChannelCount == 1 && mReqChannelCount == 1) { 4335 framesOut >>= 1; 4336 } 4337 mResampler->resample(mRsmpOutBuffer, framesOut, 4338 this /* AudioBufferProvider* */); 4339 // ditherAndClamp() works as long as all buffers returned by 4340 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4341 if (mChannelCount == 2 && mReqChannelCount == 1) { 4342 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4343 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4344 // the resampler always outputs stereo samples: 4345 // do post stereo to mono conversion 4346 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4347 framesOut); 4348 } else { 4349 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4350 } 4351 // now done with mRsmpOutBuffer 4352 4353 } 4354 if (mFramestoDrop == 0) { 4355 mActiveTrack->releaseBuffer(&buffer); 4356 } else { 4357 if (mFramestoDrop > 0) { 4358 mFramestoDrop -= buffer.frameCount; 4359 if (mFramestoDrop <= 0) { 4360 clearSyncStartEvent(); 4361 } 4362 } else { 4363 mFramestoDrop += buffer.frameCount; 4364 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4365 mSyncStartEvent->isCancelled()) { 4366 ALOGW("Synced record %s, session %d, trigger session %d", 4367 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4368 mActiveTrack->sessionId(), 4369 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4370 clearSyncStartEvent(); 4371 } 4372 } 4373 } 4374 mActiveTrack->clearOverflow(); 4375 } 4376 // client isn't retrieving buffers fast enough 4377 else { 4378 if (!mActiveTrack->setOverflow()) { 4379 nsecs_t now = systemTime(); 4380 if ((now - lastWarning) > kWarningThrottleNs) { 4381 ALOGW("RecordThread: buffer overflow"); 4382 lastWarning = now; 4383 } 4384 } 4385 // Release the processor for a while before asking for a new buffer. 4386 // This will give the application more chance to read from the buffer and 4387 // clear the overflow. 4388 usleep(kRecordThreadSleepUs); 4389 } 4390 } 4391 // enable changes in effect chain 4392 unlockEffectChains(effectChains); 4393 effectChains.clear(); 4394 } 4395 4396 standby(); 4397 4398 { 4399 Mutex::Autolock _l(mLock); 4400 mActiveTrack.clear(); 4401 mStartStopCond.broadcast(); 4402 } 4403 4404 releaseWakeLock(); 4405 4406 ALOGV("RecordThread %p exiting", this); 4407 return false; 4408} 4409 4410void AudioFlinger::RecordThread::standby() 4411{ 4412 if (!mStandby) { 4413 inputStandBy(); 4414 mStandby = true; 4415 } 4416} 4417 4418void AudioFlinger::RecordThread::inputStandBy() 4419{ 4420 mInput->stream->common.standby(&mInput->stream->common); 4421} 4422 4423sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4424 const sp<AudioFlinger::Client>& client, 4425 uint32_t sampleRate, 4426 audio_format_t format, 4427 audio_channel_mask_t channelMask, 4428 size_t frameCount, 4429 int sessionId, 4430 IAudioFlinger::track_flags_t flags, 4431 pid_t tid, 4432 status_t *status) 4433{ 4434 sp<RecordTrack> track; 4435 status_t lStatus; 4436 4437 lStatus = initCheck(); 4438 if (lStatus != NO_ERROR) { 4439 ALOGE("Audio driver not initialized."); 4440 goto Exit; 4441 } 4442 4443 // FIXME use flags and tid similar to createTrack_l() 4444 4445 { // scope for mLock 4446 Mutex::Autolock _l(mLock); 4447 4448 track = new RecordTrack(this, client, sampleRate, 4449 format, channelMask, frameCount, sessionId); 4450 4451 if (track->getCblk() == 0) { 4452 lStatus = NO_MEMORY; 4453 goto Exit; 4454 } 4455 mTracks.add(track); 4456 4457 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4458 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4459 mAudioFlinger->btNrecIsOff(); 4460 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4461 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4462 } 4463 lStatus = NO_ERROR; 4464 4465Exit: 4466 if (status) { 4467 *status = lStatus; 4468 } 4469 return track; 4470} 4471 4472status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4473 AudioSystem::sync_event_t event, 4474 int triggerSession) 4475{ 4476 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4477 sp<ThreadBase> strongMe = this; 4478 status_t status = NO_ERROR; 4479 4480 if (event == AudioSystem::SYNC_EVENT_NONE) { 4481 clearSyncStartEvent(); 4482 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4483 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4484 triggerSession, 4485 recordTrack->sessionId(), 4486 syncStartEventCallback, 4487 this); 4488 // Sync event can be cancelled by the trigger session if the track is not in a 4489 // compatible state in which case we start record immediately 4490 if (mSyncStartEvent->isCancelled()) { 4491 clearSyncStartEvent(); 4492 } else { 4493 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4494 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4495 } 4496 } 4497 4498 { 4499 AutoMutex lock(mLock); 4500 if (mActiveTrack != 0) { 4501 if (recordTrack != mActiveTrack.get()) { 4502 status = -EBUSY; 4503 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4504 mActiveTrack->mState = TrackBase::ACTIVE; 4505 } 4506 return status; 4507 } 4508 4509 recordTrack->mState = TrackBase::IDLE; 4510 mActiveTrack = recordTrack; 4511 mLock.unlock(); 4512 status_t status = AudioSystem::startInput(mId); 4513 mLock.lock(); 4514 if (status != NO_ERROR) { 4515 mActiveTrack.clear(); 4516 clearSyncStartEvent(); 4517 return status; 4518 } 4519 mRsmpInIndex = mFrameCount; 4520 mBytesRead = 0; 4521 if (mResampler != NULL) { 4522 mResampler->reset(); 4523 } 4524 mActiveTrack->mState = TrackBase::RESUMING; 4525 // signal thread to start 4526 ALOGV("Signal record thread"); 4527 mWaitWorkCV.broadcast(); 4528 // do not wait for mStartStopCond if exiting 4529 if (exitPending()) { 4530 mActiveTrack.clear(); 4531 status = INVALID_OPERATION; 4532 goto startError; 4533 } 4534 mStartStopCond.wait(mLock); 4535 if (mActiveTrack == 0) { 4536 ALOGV("Record failed to start"); 4537 status = BAD_VALUE; 4538 goto startError; 4539 } 4540 ALOGV("Record started OK"); 4541 return status; 4542 } 4543 4544startError: 4545 AudioSystem::stopInput(mId); 4546 clearSyncStartEvent(); 4547 return status; 4548} 4549 4550void AudioFlinger::RecordThread::clearSyncStartEvent() 4551{ 4552 if (mSyncStartEvent != 0) { 4553 mSyncStartEvent->cancel(); 4554 } 4555 mSyncStartEvent.clear(); 4556 mFramestoDrop = 0; 4557} 4558 4559void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4560{ 4561 sp<SyncEvent> strongEvent = event.promote(); 4562 4563 if (strongEvent != 0) { 4564 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4565 me->handleSyncStartEvent(strongEvent); 4566 } 4567} 4568 4569void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4570{ 4571 if (event == mSyncStartEvent) { 4572 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4573 // from audio HAL 4574 mFramestoDrop = mFrameCount * 2; 4575 } 4576} 4577 4578bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4579 ALOGV("RecordThread::stop"); 4580 AutoMutex _l(mLock); 4581 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4582 return false; 4583 } 4584 recordTrack->mState = TrackBase::PAUSING; 4585 // do not wait for mStartStopCond if exiting 4586 if (exitPending()) { 4587 return true; 4588 } 4589 mStartStopCond.wait(mLock); 4590 // if we have been restarted, recordTrack == mActiveTrack.get() here 4591 if (exitPending() || recordTrack != mActiveTrack.get()) { 4592 ALOGV("Record stopped OK"); 4593 return true; 4594 } 4595 return false; 4596} 4597 4598bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4599{ 4600 return false; 4601} 4602 4603status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4604{ 4605#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4606 if (!isValidSyncEvent(event)) { 4607 return BAD_VALUE; 4608 } 4609 4610 int eventSession = event->triggerSession(); 4611 status_t ret = NAME_NOT_FOUND; 4612 4613 Mutex::Autolock _l(mLock); 4614 4615 for (size_t i = 0; i < mTracks.size(); i++) { 4616 sp<RecordTrack> track = mTracks[i]; 4617 if (eventSession == track->sessionId()) { 4618 (void) track->setSyncEvent(event); 4619 ret = NO_ERROR; 4620 } 4621 } 4622 return ret; 4623#else 4624 return BAD_VALUE; 4625#endif 4626} 4627 4628// destroyTrack_l() must be called with ThreadBase::mLock held 4629void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4630{ 4631 track->terminate(); 4632 track->mState = TrackBase::STOPPED; 4633 // active tracks are removed by threadLoop() 4634 if (mActiveTrack != track) { 4635 removeTrack_l(track); 4636 } 4637} 4638 4639void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4640{ 4641 mTracks.remove(track); 4642 // need anything related to effects here? 4643} 4644 4645void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4646{ 4647 dumpInternals(fd, args); 4648 dumpTracks(fd, args); 4649 dumpEffectChains(fd, args); 4650} 4651 4652void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4653{ 4654 const size_t SIZE = 256; 4655 char buffer[SIZE]; 4656 String8 result; 4657 4658 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4659 result.append(buffer); 4660 4661 if (mActiveTrack != 0) { 4662 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4663 result.append(buffer); 4664 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4665 result.append(buffer); 4666 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4667 result.append(buffer); 4668 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4669 result.append(buffer); 4670 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4671 result.append(buffer); 4672 } else { 4673 result.append("No active record client\n"); 4674 } 4675 4676 write(fd, result.string(), result.size()); 4677 4678 dumpBase(fd, args); 4679} 4680 4681void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4682{ 4683 const size_t SIZE = 256; 4684 char buffer[SIZE]; 4685 String8 result; 4686 4687 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4688 result.append(buffer); 4689 RecordTrack::appendDumpHeader(result); 4690 for (size_t i = 0; i < mTracks.size(); ++i) { 4691 sp<RecordTrack> track = mTracks[i]; 4692 if (track != 0) { 4693 track->dump(buffer, SIZE); 4694 result.append(buffer); 4695 } 4696 } 4697 4698 if (mActiveTrack != 0) { 4699 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4700 result.append(buffer); 4701 RecordTrack::appendDumpHeader(result); 4702 mActiveTrack->dump(buffer, SIZE); 4703 result.append(buffer); 4704 4705 } 4706 write(fd, result.string(), result.size()); 4707} 4708 4709// AudioBufferProvider interface 4710status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4711{ 4712 size_t framesReq = buffer->frameCount; 4713 size_t framesReady = mFrameCount - mRsmpInIndex; 4714 int channelCount; 4715 4716 if (framesReady == 0) { 4717 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4718 if (mBytesRead <= 0) { 4719 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4720 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4721 // Force input into standby so that it tries to 4722 // recover at next read attempt 4723 inputStandBy(); 4724 usleep(kRecordThreadSleepUs); 4725 } 4726 buffer->raw = NULL; 4727 buffer->frameCount = 0; 4728 return NOT_ENOUGH_DATA; 4729 } 4730 mRsmpInIndex = 0; 4731 framesReady = mFrameCount; 4732 } 4733 4734 if (framesReq > framesReady) { 4735 framesReq = framesReady; 4736 } 4737 4738 if (mChannelCount == 1 && mReqChannelCount == 2) { 4739 channelCount = 1; 4740 } else { 4741 channelCount = 2; 4742 } 4743 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4744 buffer->frameCount = framesReq; 4745 return NO_ERROR; 4746} 4747 4748// AudioBufferProvider interface 4749void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4750{ 4751 mRsmpInIndex += buffer->frameCount; 4752 buffer->frameCount = 0; 4753} 4754 4755bool AudioFlinger::RecordThread::checkForNewParameters_l() 4756{ 4757 bool reconfig = false; 4758 4759 while (!mNewParameters.isEmpty()) { 4760 status_t status = NO_ERROR; 4761 String8 keyValuePair = mNewParameters[0]; 4762 AudioParameter param = AudioParameter(keyValuePair); 4763 int value; 4764 audio_format_t reqFormat = mFormat; 4765 uint32_t reqSamplingRate = mReqSampleRate; 4766 uint32_t reqChannelCount = mReqChannelCount; 4767 4768 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4769 reqSamplingRate = value; 4770 reconfig = true; 4771 } 4772 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4773 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4774 status = BAD_VALUE; 4775 } else { 4776 reqFormat = (audio_format_t) value; 4777 reconfig = true; 4778 } 4779 } 4780 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4781 reqChannelCount = popcount(value); 4782 reconfig = true; 4783 } 4784 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4785 // do not accept frame count changes if tracks are open as the track buffer 4786 // size depends on frame count and correct behavior would not be guaranteed 4787 // if frame count is changed after track creation 4788 if (mActiveTrack != 0) { 4789 status = INVALID_OPERATION; 4790 } else { 4791 reconfig = true; 4792 } 4793 } 4794 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4795 // forward device change to effects that have requested to be 4796 // aware of attached audio device. 4797 for (size_t i = 0; i < mEffectChains.size(); i++) { 4798 mEffectChains[i]->setDevice_l(value); 4799 } 4800 4801 // store input device and output device but do not forward output device to audio HAL. 4802 // Note that status is ignored by the caller for output device 4803 // (see AudioFlinger::setParameters() 4804 if (audio_is_output_devices(value)) { 4805 mOutDevice = value; 4806 status = BAD_VALUE; 4807 } else { 4808 mInDevice = value; 4809 // disable AEC and NS if the device is a BT SCO headset supporting those 4810 // pre processings 4811 if (mTracks.size() > 0) { 4812 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4813 mAudioFlinger->btNrecIsOff(); 4814 for (size_t i = 0; i < mTracks.size(); i++) { 4815 sp<RecordTrack> track = mTracks[i]; 4816 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4817 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4818 } 4819 } 4820 } 4821 } 4822 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4823 mAudioSource != (audio_source_t)value) { 4824 // forward device change to effects that have requested to be 4825 // aware of attached audio device. 4826 for (size_t i = 0; i < mEffectChains.size(); i++) { 4827 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4828 } 4829 mAudioSource = (audio_source_t)value; 4830 } 4831 if (status == NO_ERROR) { 4832 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4833 keyValuePair.string()); 4834 if (status == INVALID_OPERATION) { 4835 inputStandBy(); 4836 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4837 keyValuePair.string()); 4838 } 4839 if (reconfig) { 4840 if (status == BAD_VALUE && 4841 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4842 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4843 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4844 <= (2 * reqSamplingRate)) && 4845 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4846 <= FCC_2 && 4847 (reqChannelCount <= FCC_2)) { 4848 status = NO_ERROR; 4849 } 4850 if (status == NO_ERROR) { 4851 readInputParameters(); 4852 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4853 } 4854 } 4855 } 4856 4857 mNewParameters.removeAt(0); 4858 4859 mParamStatus = status; 4860 mParamCond.signal(); 4861 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4862 // already timed out waiting for the status and will never signal the condition. 4863 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4864 } 4865 return reconfig; 4866} 4867 4868String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4869{ 4870 Mutex::Autolock _l(mLock); 4871 if (initCheck() != NO_ERROR) { 4872 return String8(); 4873 } 4874 4875 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4876 const String8 out_s8(s); 4877 free(s); 4878 return out_s8; 4879} 4880 4881void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4882 AudioSystem::OutputDescriptor desc; 4883 void *param2 = NULL; 4884 4885 switch (event) { 4886 case AudioSystem::INPUT_OPENED: 4887 case AudioSystem::INPUT_CONFIG_CHANGED: 4888 desc.channelMask = mChannelMask; 4889 desc.samplingRate = mSampleRate; 4890 desc.format = mFormat; 4891 desc.frameCount = mFrameCount; 4892 desc.latency = 0; 4893 param2 = &desc; 4894 break; 4895 4896 case AudioSystem::INPUT_CLOSED: 4897 default: 4898 break; 4899 } 4900 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4901} 4902 4903void AudioFlinger::RecordThread::readInputParameters() 4904{ 4905 delete[] mRsmpInBuffer; 4906 // mRsmpInBuffer is always assigned a new[] below 4907 delete[] mRsmpOutBuffer; 4908 mRsmpOutBuffer = NULL; 4909 delete mResampler; 4910 mResampler = NULL; 4911 4912 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4913 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4914 mChannelCount = popcount(mChannelMask); 4915 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4916 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4917 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 4918 } 4919 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4920 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4921 mFrameCount = mBufferSize / mFrameSize; 4922 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4923 4924 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4925 { 4926 int channelCount; 4927 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4928 // stereo to mono post process as the resampler always outputs stereo. 4929 if (mChannelCount == 1 && mReqChannelCount == 2) { 4930 channelCount = 1; 4931 } else { 4932 channelCount = 2; 4933 } 4934 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4935 mResampler->setSampleRate(mSampleRate); 4936 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4937 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 4938 4939 // optmization: if mono to mono, alter input frame count as if we were inputing 4940 // stereo samples 4941 if (mChannelCount == 1 && mReqChannelCount == 1) { 4942 mFrameCount >>= 1; 4943 } 4944 4945 } 4946 mRsmpInIndex = mFrameCount; 4947} 4948 4949unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4950{ 4951 Mutex::Autolock _l(mLock); 4952 if (initCheck() != NO_ERROR) { 4953 return 0; 4954 } 4955 4956 return mInput->stream->get_input_frames_lost(mInput->stream); 4957} 4958 4959uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4960{ 4961 Mutex::Autolock _l(mLock); 4962 uint32_t result = 0; 4963 if (getEffectChain_l(sessionId) != 0) { 4964 result = EFFECT_SESSION; 4965 } 4966 4967 for (size_t i = 0; i < mTracks.size(); ++i) { 4968 if (sessionId == mTracks[i]->sessionId()) { 4969 result |= TRACK_SESSION; 4970 break; 4971 } 4972 } 4973 4974 return result; 4975} 4976 4977KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4978{ 4979 KeyedVector<int, bool> ids; 4980 Mutex::Autolock _l(mLock); 4981 for (size_t j = 0; j < mTracks.size(); ++j) { 4982 sp<RecordThread::RecordTrack> track = mTracks[j]; 4983 int sessionId = track->sessionId(); 4984 if (ids.indexOfKey(sessionId) < 0) { 4985 ids.add(sessionId, true); 4986 } 4987 } 4988 return ids; 4989} 4990 4991AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4992{ 4993 Mutex::Autolock _l(mLock); 4994 AudioStreamIn *input = mInput; 4995 mInput = NULL; 4996 return input; 4997} 4998 4999// this method must always be called either with ThreadBase mLock held or inside the thread loop 5000audio_stream_t* AudioFlinger::RecordThread::stream() const 5001{ 5002 if (mInput == NULL) { 5003 return NULL; 5004 } 5005 return &mInput->stream->common; 5006} 5007 5008status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5009{ 5010 // only one chain per input thread 5011 if (mEffectChains.size() != 0) { 5012 return INVALID_OPERATION; 5013 } 5014 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5015 5016 chain->setInBuffer(NULL); 5017 chain->setOutBuffer(NULL); 5018 5019 checkSuspendOnAddEffectChain_l(chain); 5020 5021 mEffectChains.add(chain); 5022 5023 return NO_ERROR; 5024} 5025 5026size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5027{ 5028 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5029 ALOGW_IF(mEffectChains.size() != 1, 5030 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5031 chain.get(), mEffectChains.size(), this); 5032 if (mEffectChains.size() == 1) { 5033 mEffectChains.removeAt(0); 5034 } 5035 return 0; 5036} 5037 5038}; // namespace android 5039