Threads.cpp revision 4de95592980dba88a35b3dc8f3fd045588387a4f
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 285 for (size_t i = 0; i < mConfigEvents.size(); i++) { 286 delete mConfigEvents[i]; 287 } 288 mConfigEvents.clear(); 289 290 mParamCond.broadcast(); 291 // do not lock the mutex in destructor 292 releaseWakeLock_l(); 293 if (mPowerManager != 0) { 294 sp<IBinder> binder = mPowerManager->asBinder(); 295 binder->unlinkToDeath(mDeathRecipient); 296 } 297} 298 299void AudioFlinger::ThreadBase::exit() 300{ 301 ALOGV("ThreadBase::exit"); 302 // do any cleanup required for exit to succeed 303 preExit(); 304 { 305 // This lock prevents the following race in thread (uniprocessor for illustration): 306 // if (!exitPending()) { 307 // // context switch from here to exit() 308 // // exit() calls requestExit(), what exitPending() observes 309 // // exit() calls signal(), which is dropped since no waiters 310 // // context switch back from exit() to here 311 // mWaitWorkCV.wait(...); 312 // // now thread is hung 313 // } 314 AutoMutex lock(mLock); 315 requestExit(); 316 mWaitWorkCV.broadcast(); 317 } 318 // When Thread::requestExitAndWait is made virtual and this method is renamed to 319 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 320 requestExitAndWait(); 321} 322 323status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 324{ 325 status_t status; 326 327 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 328 Mutex::Autolock _l(mLock); 329 330 mNewParameters.add(keyValuePairs); 331 mWaitWorkCV.signal(); 332 // wait condition with timeout in case the thread loop has exited 333 // before the request could be processed 334 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 335 status = mParamStatus; 336 mWaitWorkCV.signal(); 337 } else { 338 status = TIMED_OUT; 339 } 340 return status; 341} 342 343void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 344{ 345 Mutex::Autolock _l(mLock); 346 sendIoConfigEvent_l(event, param); 347} 348 349// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 350void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 351{ 352 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 353 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 354 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 355 param); 356 mWaitWorkCV.signal(); 357} 358 359// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 360void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 361{ 362 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 363 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 364 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 365 mConfigEvents.size(), pid, tid, prio); 366 mWaitWorkCV.signal(); 367} 368 369void AudioFlinger::ThreadBase::processConfigEvents() 370{ 371 mLock.lock(); 372 while (!mConfigEvents.isEmpty()) { 373 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 374 ConfigEvent *event = mConfigEvents[0]; 375 mConfigEvents.removeAt(0); 376 // release mLock before locking AudioFlinger mLock: lock order is always 377 // AudioFlinger then ThreadBase to avoid cross deadlock 378 mLock.unlock(); 379 switch(event->type()) { 380 case CFG_EVENT_PRIO: { 381 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 382 // FIXME Need to understand why this has be done asynchronously 383 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 384 true /*asynchronous*/); 385 if (err != 0) { 386 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 387 "error %d", 388 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 389 } 390 } break; 391 case CFG_EVENT_IO: { 392 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 393 mAudioFlinger->mLock.lock(); 394 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 395 mAudioFlinger->mLock.unlock(); 396 } break; 397 default: 398 ALOGE("processConfigEvents() unknown event type %d", event->type()); 399 break; 400 } 401 delete event; 402 mLock.lock(); 403 } 404 mLock.unlock(); 405} 406 407void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 408{ 409 const size_t SIZE = 256; 410 char buffer[SIZE]; 411 String8 result; 412 413 bool locked = AudioFlinger::dumpTryLock(mLock); 414 if (!locked) { 415 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 416 write(fd, buffer, strlen(buffer)); 417 } 418 419 snprintf(buffer, SIZE, "io handle: %d\n", mId); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 432 result.append(buffer); 433 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 434 result.append(buffer); 435 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 436 result.append(buffer); 437 438 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 439 result.append(buffer); 440 result.append(" Index Command"); 441 for (size_t i = 0; i < mNewParameters.size(); ++i) { 442 snprintf(buffer, SIZE, "\n %02d ", i); 443 result.append(buffer); 444 result.append(mNewParameters[i]); 445 } 446 447 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 448 result.append(buffer); 449 for (size_t i = 0; i < mConfigEvents.size(); i++) { 450 mConfigEvents[i]->dump(buffer, SIZE); 451 result.append(buffer); 452 } 453 result.append("\n"); 454 455 write(fd, result.string(), result.size()); 456 457 if (locked) { 458 mLock.unlock(); 459 } 460} 461 462void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 463{ 464 const size_t SIZE = 256; 465 char buffer[SIZE]; 466 String8 result; 467 468 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 469 write(fd, buffer, strlen(buffer)); 470 471 for (size_t i = 0; i < mEffectChains.size(); ++i) { 472 sp<EffectChain> chain = mEffectChains[i]; 473 if (chain != 0) { 474 chain->dump(fd, args); 475 } 476 } 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock() 480{ 481 Mutex::Autolock _l(mLock); 482 acquireWakeLock_l(); 483} 484 485void AudioFlinger::ThreadBase::acquireWakeLock_l() 486{ 487 if (mPowerManager == 0) { 488 // use checkService() to avoid blocking if power service is not up yet 489 sp<IBinder> binder = 490 defaultServiceManager()->checkService(String16("power")); 491 if (binder == 0) { 492 ALOGW("Thread %s cannot connect to the power manager service", mName); 493 } else { 494 mPowerManager = interface_cast<IPowerManager>(binder); 495 binder->linkToDeath(mDeathRecipient); 496 } 497 } 498 if (mPowerManager != 0) { 499 sp<IBinder> binder = new BBinder(); 500 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 501 binder, 502 String16(mName), 503 String16("media")); 504 if (status == NO_ERROR) { 505 mWakeLockToken = binder; 506 } 507 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 508 } 509} 510 511void AudioFlinger::ThreadBase::releaseWakeLock() 512{ 513 Mutex::Autolock _l(mLock); 514 releaseWakeLock_l(); 515} 516 517void AudioFlinger::ThreadBase::releaseWakeLock_l() 518{ 519 if (mWakeLockToken != 0) { 520 ALOGV("releaseWakeLock_l() %s", mName); 521 if (mPowerManager != 0) { 522 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 523 } 524 mWakeLockToken.clear(); 525 } 526} 527 528void AudioFlinger::ThreadBase::clearPowerManager() 529{ 530 Mutex::Autolock _l(mLock); 531 releaseWakeLock_l(); 532 mPowerManager.clear(); 533} 534 535void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 536{ 537 sp<ThreadBase> thread = mThread.promote(); 538 if (thread != 0) { 539 thread->clearPowerManager(); 540 } 541 ALOGW("power manager service died !!!"); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 Mutex::Autolock _l(mLock); 548 setEffectSuspended_l(type, suspend, sessionId); 549} 550 551void AudioFlinger::ThreadBase::setEffectSuspended_l( 552 const effect_uuid_t *type, bool suspend, int sessionId) 553{ 554 sp<EffectChain> chain = getEffectChain_l(sessionId); 555 if (chain != 0) { 556 if (type != NULL) { 557 chain->setEffectSuspended_l(type, suspend); 558 } else { 559 chain->setEffectSuspendedAll_l(suspend); 560 } 561 } 562 563 updateSuspendedSessions_l(type, suspend, sessionId); 564} 565 566void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 567{ 568 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 569 if (index < 0) { 570 return; 571 } 572 573 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 574 mSuspendedSessions.valueAt(index); 575 576 for (size_t i = 0; i < sessionEffects.size(); i++) { 577 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 578 for (int j = 0; j < desc->mRefCount; j++) { 579 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 580 chain->setEffectSuspendedAll_l(true); 581 } else { 582 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 583 desc->mType.timeLow); 584 chain->setEffectSuspended_l(&desc->mType, true); 585 } 586 } 587 } 588} 589 590void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 591 bool suspend, 592 int sessionId) 593{ 594 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 595 596 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 597 598 if (suspend) { 599 if (index >= 0) { 600 sessionEffects = mSuspendedSessions.valueAt(index); 601 } else { 602 mSuspendedSessions.add(sessionId, sessionEffects); 603 } 604 } else { 605 if (index < 0) { 606 return; 607 } 608 sessionEffects = mSuspendedSessions.valueAt(index); 609 } 610 611 612 int key = EffectChain::kKeyForSuspendAll; 613 if (type != NULL) { 614 key = type->timeLow; 615 } 616 index = sessionEffects.indexOfKey(key); 617 618 sp<SuspendedSessionDesc> desc; 619 if (suspend) { 620 if (index >= 0) { 621 desc = sessionEffects.valueAt(index); 622 } else { 623 desc = new SuspendedSessionDesc(); 624 if (type != NULL) { 625 desc->mType = *type; 626 } 627 sessionEffects.add(key, desc); 628 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 629 } 630 desc->mRefCount++; 631 } else { 632 if (index < 0) { 633 return; 634 } 635 desc = sessionEffects.valueAt(index); 636 if (--desc->mRefCount == 0) { 637 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 638 sessionEffects.removeItemsAt(index); 639 if (sessionEffects.isEmpty()) { 640 ALOGV("updateSuspendedSessions_l() restore removing session %d", 641 sessionId); 642 mSuspendedSessions.removeItem(sessionId); 643 } 644 } 645 } 646 if (!sessionEffects.isEmpty()) { 647 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 648 } 649} 650 651void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 652 bool enabled, 653 int sessionId) 654{ 655 Mutex::Autolock _l(mLock); 656 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 657} 658 659void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 660 bool enabled, 661 int sessionId) 662{ 663 if (mType != RECORD) { 664 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 665 // another session. This gives the priority to well behaved effect control panels 666 // and applications not using global effects. 667 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 668 // global effects 669 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 670 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 671 } 672 } 673 674 sp<EffectChain> chain = getEffectChain_l(sessionId); 675 if (chain != 0) { 676 chain->checkSuspendOnEffectEnabled(effect, enabled); 677 } 678} 679 680// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 681sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 682 const sp<AudioFlinger::Client>& client, 683 const sp<IEffectClient>& effectClient, 684 int32_t priority, 685 int sessionId, 686 effect_descriptor_t *desc, 687 int *enabled, 688 status_t *status 689 ) 690{ 691 sp<EffectModule> effect; 692 sp<EffectHandle> handle; 693 status_t lStatus; 694 sp<EffectChain> chain; 695 bool chainCreated = false; 696 bool effectCreated = false; 697 bool effectRegistered = false; 698 699 lStatus = initCheck(); 700 if (lStatus != NO_ERROR) { 701 ALOGW("createEffect_l() Audio driver not initialized."); 702 goto Exit; 703 } 704 705 // Allow global effects only on offloaded and mixer threads 706 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 707 switch (mType) { 708 case MIXER: 709 case OFFLOAD: 710 break; 711 case DIRECT: 712 case DUPLICATING: 713 case RECORD: 714 default: 715 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 716 lStatus = BAD_VALUE; 717 goto Exit; 718 } 719 } 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 effect->setOffloaded(mType == OFFLOAD, mId); 764 765 lStatus = chain->addEffect_l(effect); 766 if (lStatus != NO_ERROR) { 767 goto Exit; 768 } 769 effectCreated = true; 770 771 effect->setDevice(mOutDevice); 772 effect->setDevice(mInDevice); 773 effect->setMode(mAudioFlinger->getMode()); 774 effect->setAudioSource(mAudioSource); 775 } 776 // create effect handle and connect it to effect module 777 handle = new EffectHandle(effect, client, effectClient, priority); 778 lStatus = effect->addHandle(handle.get()); 779 if (enabled != NULL) { 780 *enabled = (int)effect->isEnabled(); 781 } 782 } 783 784Exit: 785 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 786 Mutex::Autolock _l(mLock); 787 if (effectCreated) { 788 chain->removeEffect_l(effect); 789 } 790 if (effectRegistered) { 791 AudioSystem::unregisterEffect(effect->id()); 792 } 793 if (chainCreated) { 794 removeEffectChain_l(chain); 795 } 796 handle.clear(); 797 } 798 799 if (status != NULL) { 800 *status = lStatus; 801 } 802 return handle; 803} 804 805sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 806{ 807 Mutex::Autolock _l(mLock); 808 return getEffect_l(sessionId, effectId); 809} 810 811sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 812{ 813 sp<EffectChain> chain = getEffectChain_l(sessionId); 814 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 815} 816 817// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 818// PlaybackThread::mLock held 819status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 820{ 821 // check for existing effect chain with the requested audio session 822 int sessionId = effect->sessionId(); 823 sp<EffectChain> chain = getEffectChain_l(sessionId); 824 bool chainCreated = false; 825 826 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 827 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 828 this, effect->desc().name, effect->desc().flags); 829 830 if (chain == 0) { 831 // create a new chain for this session 832 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 833 chain = new EffectChain(this, sessionId); 834 addEffectChain_l(chain); 835 chain->setStrategy(getStrategyForSession_l(sessionId)); 836 chainCreated = true; 837 } 838 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 839 840 if (chain->getEffectFromId_l(effect->id()) != 0) { 841 ALOGW("addEffect_l() %p effect %s already present in chain %p", 842 this, effect->desc().name, chain.get()); 843 return BAD_VALUE; 844 } 845 846 effect->setOffloaded(mType == OFFLOAD, mId); 847 848 status_t status = chain->addEffect_l(effect); 849 if (status != NO_ERROR) { 850 if (chainCreated) { 851 removeEffectChain_l(chain); 852 } 853 return status; 854 } 855 856 effect->setDevice(mOutDevice); 857 effect->setDevice(mInDevice); 858 effect->setMode(mAudioFlinger->getMode()); 859 effect->setAudioSource(mAudioSource); 860 return NO_ERROR; 861} 862 863void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 864 865 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 866 effect_descriptor_t desc = effect->desc(); 867 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 868 detachAuxEffect_l(effect->id()); 869 } 870 871 sp<EffectChain> chain = effect->chain().promote(); 872 if (chain != 0) { 873 // remove effect chain if removing last effect 874 if (chain->removeEffect_l(effect) == 0) { 875 removeEffectChain_l(chain); 876 } 877 } else { 878 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 879 } 880} 881 882void AudioFlinger::ThreadBase::lockEffectChains_l( 883 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 884{ 885 effectChains = mEffectChains; 886 for (size_t i = 0; i < mEffectChains.size(); i++) { 887 mEffectChains[i]->lock(); 888 } 889} 890 891void AudioFlinger::ThreadBase::unlockEffectChains( 892 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 893{ 894 for (size_t i = 0; i < effectChains.size(); i++) { 895 effectChains[i]->unlock(); 896 } 897} 898 899sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 900{ 901 Mutex::Autolock _l(mLock); 902 return getEffectChain_l(sessionId); 903} 904 905sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 906{ 907 size_t size = mEffectChains.size(); 908 for (size_t i = 0; i < size; i++) { 909 if (mEffectChains[i]->sessionId() == sessionId) { 910 return mEffectChains[i]; 911 } 912 } 913 return 0; 914} 915 916void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 917{ 918 Mutex::Autolock _l(mLock); 919 size_t size = mEffectChains.size(); 920 for (size_t i = 0; i < size; i++) { 921 mEffectChains[i]->setMode_l(mode); 922 } 923} 924 925void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 926 EffectHandle *handle, 927 bool unpinIfLast) { 928 929 Mutex::Autolock _l(mLock); 930 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 931 // delete the effect module if removing last handle on it 932 if (effect->removeHandle(handle) == 0) { 933 if (!effect->isPinned() || unpinIfLast) { 934 removeEffect_l(effect); 935 AudioSystem::unregisterEffect(effect->id()); 936 } 937 } 938} 939 940// ---------------------------------------------------------------------------- 941// Playback 942// ---------------------------------------------------------------------------- 943 944AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 945 AudioStreamOut* output, 946 audio_io_handle_t id, 947 audio_devices_t device, 948 type_t type) 949 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 950 mNormalFrameCount(0), mMixBuffer(NULL), 951 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 952 // mStreamTypes[] initialized in constructor body 953 mOutput(output), 954 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 955 mMixerStatus(MIXER_IDLE), 956 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 957 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 958 mBytesRemaining(0), 959 mCurrentWriteLength(0), 960 mUseAsyncWrite(false), 961 mWriteAckSequence(0), 962 mDrainSequence(0), 963 mSignalPending(false), 964 mScreenState(AudioFlinger::mScreenState), 965 // index 0 is reserved for normal mixer's submix 966 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 967 // mLatchD, mLatchQ, 968 mLatchDValid(false), mLatchQValid(false) 969{ 970 snprintf(mName, kNameLength, "AudioOut_%X", id); 971 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 972 973 // Assumes constructor is called by AudioFlinger with it's mLock held, but 974 // it would be safer to explicitly pass initial masterVolume/masterMute as 975 // parameter. 976 // 977 // If the HAL we are using has support for master volume or master mute, 978 // then do not attenuate or mute during mixing (just leave the volume at 1.0 979 // and the mute set to false). 980 mMasterVolume = audioFlinger->masterVolume_l(); 981 mMasterMute = audioFlinger->masterMute_l(); 982 if (mOutput && mOutput->audioHwDev) { 983 if (mOutput->audioHwDev->canSetMasterVolume()) { 984 mMasterVolume = 1.0; 985 } 986 987 if (mOutput->audioHwDev->canSetMasterMute()) { 988 mMasterMute = false; 989 } 990 } 991 992 readOutputParameters(); 993 994 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 995 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 996 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 997 stream = (audio_stream_type_t) (stream + 1)) { 998 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 999 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1000 } 1001 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1002 // because mAudioFlinger doesn't have one to copy from 1003} 1004 1005AudioFlinger::PlaybackThread::~PlaybackThread() 1006{ 1007 mAudioFlinger->unregisterWriter(mNBLogWriter); 1008 delete [] mAllocMixBuffer; 1009} 1010 1011void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1012{ 1013 dumpInternals(fd, args); 1014 dumpTracks(fd, args); 1015 dumpEffectChains(fd, args); 1016} 1017 1018void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1019{ 1020 const size_t SIZE = 256; 1021 char buffer[SIZE]; 1022 String8 result; 1023 1024 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1025 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1026 const stream_type_t *st = &mStreamTypes[i]; 1027 if (i > 0) { 1028 result.appendFormat(", "); 1029 } 1030 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1031 if (st->mute) { 1032 result.append("M"); 1033 } 1034 } 1035 result.append("\n"); 1036 write(fd, result.string(), result.length()); 1037 result.clear(); 1038 1039 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1040 result.append(buffer); 1041 Track::appendDumpHeader(result); 1042 for (size_t i = 0; i < mTracks.size(); ++i) { 1043 sp<Track> track = mTracks[i]; 1044 if (track != 0) { 1045 track->dump(buffer, SIZE); 1046 result.append(buffer); 1047 } 1048 } 1049 1050 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1051 result.append(buffer); 1052 Track::appendDumpHeader(result); 1053 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1054 sp<Track> track = mActiveTracks[i].promote(); 1055 if (track != 0) { 1056 track->dump(buffer, SIZE); 1057 result.append(buffer); 1058 } 1059 } 1060 write(fd, result.string(), result.size()); 1061 1062 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1063 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1064 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1065 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1066} 1067 1068void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1069{ 1070 const size_t SIZE = 256; 1071 char buffer[SIZE]; 1072 String8 result; 1073 1074 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1079 ns2ms(systemTime() - mLastWriteTime)); 1080 result.append(buffer); 1081 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1082 result.append(buffer); 1083 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1084 result.append(buffer); 1085 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1086 result.append(buffer); 1087 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1088 result.append(buffer); 1089 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1090 result.append(buffer); 1091 write(fd, result.string(), result.size()); 1092 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1093 1094 dumpBase(fd, args); 1095} 1096 1097// Thread virtuals 1098status_t AudioFlinger::PlaybackThread::readyToRun() 1099{ 1100 status_t status = initCheck(); 1101 if (status == NO_ERROR) { 1102 ALOGI("AudioFlinger's thread %p ready to run", this); 1103 } else { 1104 ALOGE("No working audio driver found."); 1105 } 1106 return status; 1107} 1108 1109void AudioFlinger::PlaybackThread::onFirstRef() 1110{ 1111 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1112} 1113 1114// ThreadBase virtuals 1115void AudioFlinger::PlaybackThread::preExit() 1116{ 1117 ALOGV(" preExit()"); 1118 // FIXME this is using hard-coded strings but in the future, this functionality will be 1119 // converted to use audio HAL extensions required to support tunneling 1120 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1121} 1122 1123// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1124sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1125 const sp<AudioFlinger::Client>& client, 1126 audio_stream_type_t streamType, 1127 uint32_t sampleRate, 1128 audio_format_t format, 1129 audio_channel_mask_t channelMask, 1130 size_t frameCount, 1131 const sp<IMemory>& sharedBuffer, 1132 int sessionId, 1133 IAudioFlinger::track_flags_t *flags, 1134 pid_t tid, 1135 status_t *status) 1136{ 1137 sp<Track> track; 1138 status_t lStatus; 1139 1140 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1141 1142 // client expresses a preference for FAST, but we get the final say 1143 if (*flags & IAudioFlinger::TRACK_FAST) { 1144 if ( 1145 // not timed 1146 (!isTimed) && 1147 // either of these use cases: 1148 ( 1149 // use case 1: shared buffer with any frame count 1150 ( 1151 (sharedBuffer != 0) 1152 ) || 1153 // use case 2: callback handler and frame count is default or at least as large as HAL 1154 ( 1155 (tid != -1) && 1156 ((frameCount == 0) || 1157 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1158 ) 1159 ) && 1160 // PCM data 1161 audio_is_linear_pcm(format) && 1162 // mono or stereo 1163 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1164 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1165#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1166 // hardware sample rate 1167 (sampleRate == mSampleRate) && 1168#endif 1169 // normal mixer has an associated fast mixer 1170 hasFastMixer() && 1171 // there are sufficient fast track slots available 1172 (mFastTrackAvailMask != 0) 1173 // FIXME test that MixerThread for this fast track has a capable output HAL 1174 // FIXME add a permission test also? 1175 ) { 1176 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1177 if (frameCount == 0) { 1178 frameCount = mFrameCount * kFastTrackMultiplier; 1179 } 1180 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1181 frameCount, mFrameCount); 1182 } else { 1183 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1184 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1185 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1186 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1187 audio_is_linear_pcm(format), 1188 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1189 *flags &= ~IAudioFlinger::TRACK_FAST; 1190 // For compatibility with AudioTrack calculation, buffer depth is forced 1191 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1192 // This is probably too conservative, but legacy application code may depend on it. 1193 // If you change this calculation, also review the start threshold which is related. 1194 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1195 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1196 if (minBufCount < 2) { 1197 minBufCount = 2; 1198 } 1199 size_t minFrameCount = mNormalFrameCount * minBufCount; 1200 if (frameCount < minFrameCount) { 1201 frameCount = minFrameCount; 1202 } 1203 } 1204 } 1205 1206 if (mType == DIRECT) { 1207 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1208 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1209 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1210 "for output %p with format %d", 1211 sampleRate, format, channelMask, mOutput, mFormat); 1212 lStatus = BAD_VALUE; 1213 goto Exit; 1214 } 1215 } 1216 } else if (mType == OFFLOAD) { 1217 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1218 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1219 "for output %p with format %d", 1220 sampleRate, format, channelMask, mOutput, mFormat); 1221 lStatus = BAD_VALUE; 1222 goto Exit; 1223 } 1224 } else { 1225 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1226 ALOGE("createTrack_l() Bad parameter: format %d \"" 1227 "for output %p with format %d", 1228 format, mOutput, mFormat); 1229 lStatus = BAD_VALUE; 1230 goto Exit; 1231 } 1232 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1233 if (sampleRate > mSampleRate*2) { 1234 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1235 lStatus = BAD_VALUE; 1236 goto Exit; 1237 } 1238 } 1239 1240 lStatus = initCheck(); 1241 if (lStatus != NO_ERROR) { 1242 ALOGE("Audio driver not initialized."); 1243 goto Exit; 1244 } 1245 1246 { // scope for mLock 1247 Mutex::Autolock _l(mLock); 1248 1249 // all tracks in same audio session must share the same routing strategy otherwise 1250 // conflicts will happen when tracks are moved from one output to another by audio policy 1251 // manager 1252 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1253 for (size_t i = 0; i < mTracks.size(); ++i) { 1254 sp<Track> t = mTracks[i]; 1255 if (t != 0 && !t->isOutputTrack()) { 1256 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1257 if (sessionId == t->sessionId() && strategy != actual) { 1258 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1259 strategy, actual); 1260 lStatus = BAD_VALUE; 1261 goto Exit; 1262 } 1263 } 1264 } 1265 1266 if (!isTimed) { 1267 track = new Track(this, client, streamType, sampleRate, format, 1268 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1269 } else { 1270 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1271 channelMask, frameCount, sharedBuffer, sessionId); 1272 } 1273 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1274 lStatus = NO_MEMORY; 1275 goto Exit; 1276 } 1277 1278 mTracks.add(track); 1279 1280 sp<EffectChain> chain = getEffectChain_l(sessionId); 1281 if (chain != 0) { 1282 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1283 track->setMainBuffer(chain->inBuffer()); 1284 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1285 chain->incTrackCnt(); 1286 } 1287 1288 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1289 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1290 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1291 // so ask activity manager to do this on our behalf 1292 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1293 } 1294 } 1295 1296 lStatus = NO_ERROR; 1297 1298Exit: 1299 if (status) { 1300 *status = lStatus; 1301 } 1302 return track; 1303} 1304 1305uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1306{ 1307 return latency; 1308} 1309 1310uint32_t AudioFlinger::PlaybackThread::latency() const 1311{ 1312 Mutex::Autolock _l(mLock); 1313 return latency_l(); 1314} 1315uint32_t AudioFlinger::PlaybackThread::latency_l() const 1316{ 1317 if (initCheck() == NO_ERROR) { 1318 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1319 } else { 1320 return 0; 1321 } 1322} 1323 1324void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1325{ 1326 Mutex::Autolock _l(mLock); 1327 // Don't apply master volume in SW if our HAL can do it for us. 1328 if (mOutput && mOutput->audioHwDev && 1329 mOutput->audioHwDev->canSetMasterVolume()) { 1330 mMasterVolume = 1.0; 1331 } else { 1332 mMasterVolume = value; 1333 } 1334} 1335 1336void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1337{ 1338 Mutex::Autolock _l(mLock); 1339 // Don't apply master mute in SW if our HAL can do it for us. 1340 if (mOutput && mOutput->audioHwDev && 1341 mOutput->audioHwDev->canSetMasterMute()) { 1342 mMasterMute = false; 1343 } else { 1344 mMasterMute = muted; 1345 } 1346} 1347 1348void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1349{ 1350 Mutex::Autolock _l(mLock); 1351 mStreamTypes[stream].volume = value; 1352 broadcast_l(); 1353} 1354 1355void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1356{ 1357 Mutex::Autolock _l(mLock); 1358 mStreamTypes[stream].mute = muted; 1359 broadcast_l(); 1360} 1361 1362float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1363{ 1364 Mutex::Autolock _l(mLock); 1365 return mStreamTypes[stream].volume; 1366} 1367 1368// addTrack_l() must be called with ThreadBase::mLock held 1369status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1370{ 1371 status_t status = ALREADY_EXISTS; 1372 1373 // set retry count for buffer fill 1374 track->mRetryCount = kMaxTrackStartupRetries; 1375 if (mActiveTracks.indexOf(track) < 0) { 1376 // the track is newly added, make sure it fills up all its 1377 // buffers before playing. This is to ensure the client will 1378 // effectively get the latency it requested. 1379 if (!track->isOutputTrack()) { 1380 TrackBase::track_state state = track->mState; 1381 mLock.unlock(); 1382 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1383 mLock.lock(); 1384 // abort track was stopped/paused while we released the lock 1385 if (state != track->mState) { 1386 if (status == NO_ERROR) { 1387 mLock.unlock(); 1388 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1389 mLock.lock(); 1390 } 1391 return INVALID_OPERATION; 1392 } 1393 // abort if start is rejected by audio policy manager 1394 if (status != NO_ERROR) { 1395 return PERMISSION_DENIED; 1396 } 1397#ifdef ADD_BATTERY_DATA 1398 // to track the speaker usage 1399 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1400#endif 1401 } 1402 1403 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1404 track->mResetDone = false; 1405 track->mPresentationCompleteFrames = 0; 1406 mActiveTracks.add(track); 1407 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1408 if (chain != 0) { 1409 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1410 track->sessionId()); 1411 chain->incActiveTrackCnt(); 1412 } 1413 1414 status = NO_ERROR; 1415 } 1416 1417 ALOGV("signal playback thread"); 1418 broadcast_l(); 1419 1420 return status; 1421} 1422 1423bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1424{ 1425 track->terminate(); 1426 // active tracks are removed by threadLoop() 1427 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1428 track->mState = TrackBase::STOPPED; 1429 if (!trackActive) { 1430 removeTrack_l(track); 1431 } else if (track->isFastTrack() || track->isOffloaded()) { 1432 track->mState = TrackBase::STOPPING_1; 1433 } 1434 1435 return trackActive; 1436} 1437 1438void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1439{ 1440 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1441 mTracks.remove(track); 1442 deleteTrackName_l(track->name()); 1443 // redundant as track is about to be destroyed, for dumpsys only 1444 track->mName = -1; 1445 if (track->isFastTrack()) { 1446 int index = track->mFastIndex; 1447 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1448 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1449 mFastTrackAvailMask |= 1 << index; 1450 // redundant as track is about to be destroyed, for dumpsys only 1451 track->mFastIndex = -1; 1452 } 1453 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1454 if (chain != 0) { 1455 chain->decTrackCnt(); 1456 } 1457} 1458 1459void AudioFlinger::PlaybackThread::broadcast_l() 1460{ 1461 // Thread could be blocked waiting for async 1462 // so signal it to handle state changes immediately 1463 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1464 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1465 mSignalPending = true; 1466 mWaitWorkCV.broadcast(); 1467} 1468 1469String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1470{ 1471 Mutex::Autolock _l(mLock); 1472 if (initCheck() != NO_ERROR) { 1473 return String8(); 1474 } 1475 1476 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1477 const String8 out_s8(s); 1478 free(s); 1479 return out_s8; 1480} 1481 1482// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1483void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1484 AudioSystem::OutputDescriptor desc; 1485 void *param2 = NULL; 1486 1487 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1488 param); 1489 1490 switch (event) { 1491 case AudioSystem::OUTPUT_OPENED: 1492 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1493 desc.channelMask = mChannelMask; 1494 desc.samplingRate = mSampleRate; 1495 desc.format = mFormat; 1496 desc.frameCount = mNormalFrameCount; // FIXME see 1497 // AudioFlinger::frameCount(audio_io_handle_t) 1498 desc.latency = latency(); 1499 param2 = &desc; 1500 break; 1501 1502 case AudioSystem::STREAM_CONFIG_CHANGED: 1503 param2 = ¶m; 1504 case AudioSystem::OUTPUT_CLOSED: 1505 default: 1506 break; 1507 } 1508 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1509} 1510 1511void AudioFlinger::PlaybackThread::writeCallback() 1512{ 1513 ALOG_ASSERT(mCallbackThread != 0); 1514 mCallbackThread->resetWriteBlocked(); 1515} 1516 1517void AudioFlinger::PlaybackThread::drainCallback() 1518{ 1519 ALOG_ASSERT(mCallbackThread != 0); 1520 mCallbackThread->resetDraining(); 1521} 1522 1523void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1524{ 1525 Mutex::Autolock _l(mLock); 1526 // reject out of sequence requests 1527 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1528 mWriteAckSequence &= ~1; 1529 mWaitWorkCV.signal(); 1530 } 1531} 1532 1533void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1534{ 1535 Mutex::Autolock _l(mLock); 1536 // reject out of sequence requests 1537 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1538 mDrainSequence &= ~1; 1539 mWaitWorkCV.signal(); 1540 } 1541} 1542 1543// static 1544int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1545 void *param, 1546 void *cookie) 1547{ 1548 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1549 ALOGV("asyncCallback() event %d", event); 1550 switch (event) { 1551 case STREAM_CBK_EVENT_WRITE_READY: 1552 me->writeCallback(); 1553 break; 1554 case STREAM_CBK_EVENT_DRAIN_READY: 1555 me->drainCallback(); 1556 break; 1557 default: 1558 ALOGW("asyncCallback() unknown event %d", event); 1559 break; 1560 } 1561 return 0; 1562} 1563 1564void AudioFlinger::PlaybackThread::readOutputParameters() 1565{ 1566 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1567 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1568 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1569 if (!audio_is_output_channel(mChannelMask)) { 1570 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1571 } 1572 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1573 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1574 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1575 } 1576 mChannelCount = popcount(mChannelMask); 1577 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1578 if (!audio_is_valid_format(mFormat)) { 1579 LOG_FATAL("HAL format %d not valid for output", mFormat); 1580 } 1581 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1582 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1583 mFormat); 1584 } 1585 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1586 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1587 if (mFrameCount & 15) { 1588 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1589 mFrameCount); 1590 } 1591 1592 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1593 (mOutput->stream->set_callback != NULL)) { 1594 if (mOutput->stream->set_callback(mOutput->stream, 1595 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1596 mUseAsyncWrite = true; 1597 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1598 } 1599 } 1600 1601 // Calculate size of normal mix buffer relative to the HAL output buffer size 1602 double multiplier = 1.0; 1603 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1604 kUseFastMixer == FastMixer_Dynamic)) { 1605 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1606 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1607 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1608 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1609 maxNormalFrameCount = maxNormalFrameCount & ~15; 1610 if (maxNormalFrameCount < minNormalFrameCount) { 1611 maxNormalFrameCount = minNormalFrameCount; 1612 } 1613 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1614 if (multiplier <= 1.0) { 1615 multiplier = 1.0; 1616 } else if (multiplier <= 2.0) { 1617 if (2 * mFrameCount <= maxNormalFrameCount) { 1618 multiplier = 2.0; 1619 } else { 1620 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1621 } 1622 } else { 1623 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1624 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1625 // track, but we sometimes have to do this to satisfy the maximum frame count 1626 // constraint) 1627 // FIXME this rounding up should not be done if no HAL SRC 1628 uint32_t truncMult = (uint32_t) multiplier; 1629 if ((truncMult & 1)) { 1630 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1631 ++truncMult; 1632 } 1633 } 1634 multiplier = (double) truncMult; 1635 } 1636 } 1637 mNormalFrameCount = multiplier * mFrameCount; 1638 // round up to nearest 16 frames to satisfy AudioMixer 1639 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1640 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1641 mNormalFrameCount); 1642 1643 delete[] mAllocMixBuffer; 1644 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1645 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1646 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1647 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1648 1649 // force reconfiguration of effect chains and engines to take new buffer size and audio 1650 // parameters into account 1651 // Note that mLock is not held when readOutputParameters() is called from the constructor 1652 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1653 // matter. 1654 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1655 Vector< sp<EffectChain> > effectChains = mEffectChains; 1656 for (size_t i = 0; i < effectChains.size(); i ++) { 1657 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1658 } 1659} 1660 1661 1662status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1663{ 1664 if (halFrames == NULL || dspFrames == NULL) { 1665 return BAD_VALUE; 1666 } 1667 Mutex::Autolock _l(mLock); 1668 if (initCheck() != NO_ERROR) { 1669 return INVALID_OPERATION; 1670 } 1671 size_t framesWritten = mBytesWritten / mFrameSize; 1672 *halFrames = framesWritten; 1673 1674 if (isSuspended()) { 1675 // return an estimation of rendered frames when the output is suspended 1676 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1677 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1678 return NO_ERROR; 1679 } else { 1680 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1681 } 1682} 1683 1684uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1685{ 1686 Mutex::Autolock _l(mLock); 1687 uint32_t result = 0; 1688 if (getEffectChain_l(sessionId) != 0) { 1689 result = EFFECT_SESSION; 1690 } 1691 1692 for (size_t i = 0; i < mTracks.size(); ++i) { 1693 sp<Track> track = mTracks[i]; 1694 if (sessionId == track->sessionId() && !track->isInvalid()) { 1695 result |= TRACK_SESSION; 1696 break; 1697 } 1698 } 1699 1700 return result; 1701} 1702 1703uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1704{ 1705 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1706 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1707 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1708 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1709 } 1710 for (size_t i = 0; i < mTracks.size(); i++) { 1711 sp<Track> track = mTracks[i]; 1712 if (sessionId == track->sessionId() && !track->isInvalid()) { 1713 return AudioSystem::getStrategyForStream(track->streamType()); 1714 } 1715 } 1716 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1717} 1718 1719 1720AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1721{ 1722 Mutex::Autolock _l(mLock); 1723 return mOutput; 1724} 1725 1726AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1727{ 1728 Mutex::Autolock _l(mLock); 1729 AudioStreamOut *output = mOutput; 1730 mOutput = NULL; 1731 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1732 // must push a NULL and wait for ack 1733 mOutputSink.clear(); 1734 mPipeSink.clear(); 1735 mNormalSink.clear(); 1736 return output; 1737} 1738 1739// this method must always be called either with ThreadBase mLock held or inside the thread loop 1740audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1741{ 1742 if (mOutput == NULL) { 1743 return NULL; 1744 } 1745 return &mOutput->stream->common; 1746} 1747 1748uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1749{ 1750 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1751} 1752 1753status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1754{ 1755 if (!isValidSyncEvent(event)) { 1756 return BAD_VALUE; 1757 } 1758 1759 Mutex::Autolock _l(mLock); 1760 1761 for (size_t i = 0; i < mTracks.size(); ++i) { 1762 sp<Track> track = mTracks[i]; 1763 if (event->triggerSession() == track->sessionId()) { 1764 (void) track->setSyncEvent(event); 1765 return NO_ERROR; 1766 } 1767 } 1768 1769 return NAME_NOT_FOUND; 1770} 1771 1772bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1773{ 1774 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1775} 1776 1777void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1778 const Vector< sp<Track> >& tracksToRemove) 1779{ 1780 size_t count = tracksToRemove.size(); 1781 if (count) { 1782 for (size_t i = 0 ; i < count ; i++) { 1783 const sp<Track>& track = tracksToRemove.itemAt(i); 1784 if (!track->isOutputTrack()) { 1785 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1786#ifdef ADD_BATTERY_DATA 1787 // to track the speaker usage 1788 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1789#endif 1790 if (track->isTerminated()) { 1791 AudioSystem::releaseOutput(mId); 1792 } 1793 } 1794 } 1795 } 1796} 1797 1798void AudioFlinger::PlaybackThread::checkSilentMode_l() 1799{ 1800 if (!mMasterMute) { 1801 char value[PROPERTY_VALUE_MAX]; 1802 if (property_get("ro.audio.silent", value, "0") > 0) { 1803 char *endptr; 1804 unsigned long ul = strtoul(value, &endptr, 0); 1805 if (*endptr == '\0' && ul != 0) { 1806 ALOGD("Silence is golden"); 1807 // The setprop command will not allow a property to be changed after 1808 // the first time it is set, so we don't have to worry about un-muting. 1809 setMasterMute_l(true); 1810 } 1811 } 1812 } 1813} 1814 1815// shared by MIXER and DIRECT, overridden by DUPLICATING 1816ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1817{ 1818 // FIXME rewrite to reduce number of system calls 1819 mLastWriteTime = systemTime(); 1820 mInWrite = true; 1821 ssize_t bytesWritten; 1822 1823 // If an NBAIO sink is present, use it to write the normal mixer's submix 1824 if (mNormalSink != 0) { 1825#define mBitShift 2 // FIXME 1826 size_t count = mBytesRemaining >> mBitShift; 1827 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1828 ATRACE_BEGIN("write"); 1829 // update the setpoint when AudioFlinger::mScreenState changes 1830 uint32_t screenState = AudioFlinger::mScreenState; 1831 if (screenState != mScreenState) { 1832 mScreenState = screenState; 1833 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1834 if (pipe != NULL) { 1835 pipe->setAvgFrames((mScreenState & 1) ? 1836 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1837 } 1838 } 1839 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1840 ATRACE_END(); 1841 if (framesWritten > 0) { 1842 bytesWritten = framesWritten << mBitShift; 1843 } else { 1844 bytesWritten = framesWritten; 1845 } 1846 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1847 if (status == NO_ERROR) { 1848 size_t totalFramesWritten = mNormalSink->framesWritten(); 1849 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1850 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1851 mLatchDValid = true; 1852 } 1853 } 1854 // otherwise use the HAL / AudioStreamOut directly 1855 } else { 1856 // Direct output and offload threads 1857 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1858 if (mUseAsyncWrite) { 1859 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1860 mWriteAckSequence += 2; 1861 mWriteAckSequence |= 1; 1862 ALOG_ASSERT(mCallbackThread != 0); 1863 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1864 } 1865 // FIXME We should have an implementation of timestamps for direct output threads. 1866 // They are used e.g for multichannel PCM playback over HDMI. 1867 bytesWritten = mOutput->stream->write(mOutput->stream, 1868 mMixBuffer + offset, mBytesRemaining); 1869 if (mUseAsyncWrite && 1870 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1871 // do not wait for async callback in case of error of full write 1872 mWriteAckSequence &= ~1; 1873 ALOG_ASSERT(mCallbackThread != 0); 1874 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1875 } 1876 } 1877 1878 mNumWrites++; 1879 mInWrite = false; 1880 1881 return bytesWritten; 1882} 1883 1884void AudioFlinger::PlaybackThread::threadLoop_drain() 1885{ 1886 if (mOutput->stream->drain) { 1887 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1888 if (mUseAsyncWrite) { 1889 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1890 mDrainSequence |= 1; 1891 ALOG_ASSERT(mCallbackThread != 0); 1892 mCallbackThread->setDraining(mDrainSequence); 1893 } 1894 mOutput->stream->drain(mOutput->stream, 1895 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1896 : AUDIO_DRAIN_ALL); 1897 } 1898} 1899 1900void AudioFlinger::PlaybackThread::threadLoop_exit() 1901{ 1902 // Default implementation has nothing to do 1903} 1904 1905/* 1906The derived values that are cached: 1907 - mixBufferSize from frame count * frame size 1908 - activeSleepTime from activeSleepTimeUs() 1909 - idleSleepTime from idleSleepTimeUs() 1910 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1911 - maxPeriod from frame count and sample rate (MIXER only) 1912 1913The parameters that affect these derived values are: 1914 - frame count 1915 - frame size 1916 - sample rate 1917 - device type: A2DP or not 1918 - device latency 1919 - format: PCM or not 1920 - active sleep time 1921 - idle sleep time 1922*/ 1923 1924void AudioFlinger::PlaybackThread::cacheParameters_l() 1925{ 1926 mixBufferSize = mNormalFrameCount * mFrameSize; 1927 activeSleepTime = activeSleepTimeUs(); 1928 idleSleepTime = idleSleepTimeUs(); 1929} 1930 1931void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1932{ 1933 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1934 this, streamType, mTracks.size()); 1935 Mutex::Autolock _l(mLock); 1936 1937 size_t size = mTracks.size(); 1938 for (size_t i = 0; i < size; i++) { 1939 sp<Track> t = mTracks[i]; 1940 if (t->streamType() == streamType) { 1941 t->invalidate(); 1942 } 1943 } 1944} 1945 1946status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1947{ 1948 int session = chain->sessionId(); 1949 int16_t *buffer = mMixBuffer; 1950 bool ownsBuffer = false; 1951 1952 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1953 if (session > 0) { 1954 // Only one effect chain can be present in direct output thread and it uses 1955 // the mix buffer as input 1956 if (mType != DIRECT) { 1957 size_t numSamples = mNormalFrameCount * mChannelCount; 1958 buffer = new int16_t[numSamples]; 1959 memset(buffer, 0, numSamples * sizeof(int16_t)); 1960 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1961 ownsBuffer = true; 1962 } 1963 1964 // Attach all tracks with same session ID to this chain. 1965 for (size_t i = 0; i < mTracks.size(); ++i) { 1966 sp<Track> track = mTracks[i]; 1967 if (session == track->sessionId()) { 1968 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1969 buffer); 1970 track->setMainBuffer(buffer); 1971 chain->incTrackCnt(); 1972 } 1973 } 1974 1975 // indicate all active tracks in the chain 1976 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1977 sp<Track> track = mActiveTracks[i].promote(); 1978 if (track == 0) { 1979 continue; 1980 } 1981 if (session == track->sessionId()) { 1982 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1983 chain->incActiveTrackCnt(); 1984 } 1985 } 1986 } 1987 1988 chain->setInBuffer(buffer, ownsBuffer); 1989 chain->setOutBuffer(mMixBuffer); 1990 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1991 // chains list in order to be processed last as it contains output stage effects 1992 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1993 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1994 // after track specific effects and before output stage 1995 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1996 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1997 // Effect chain for other sessions are inserted at beginning of effect 1998 // chains list to be processed before output mix effects. Relative order between other 1999 // sessions is not important 2000 size_t size = mEffectChains.size(); 2001 size_t i = 0; 2002 for (i = 0; i < size; i++) { 2003 if (mEffectChains[i]->sessionId() < session) { 2004 break; 2005 } 2006 } 2007 mEffectChains.insertAt(chain, i); 2008 checkSuspendOnAddEffectChain_l(chain); 2009 2010 return NO_ERROR; 2011} 2012 2013size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2014{ 2015 int session = chain->sessionId(); 2016 2017 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2018 2019 for (size_t i = 0; i < mEffectChains.size(); i++) { 2020 if (chain == mEffectChains[i]) { 2021 mEffectChains.removeAt(i); 2022 // detach all active tracks from the chain 2023 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2024 sp<Track> track = mActiveTracks[i].promote(); 2025 if (track == 0) { 2026 continue; 2027 } 2028 if (session == track->sessionId()) { 2029 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2030 chain.get(), session); 2031 chain->decActiveTrackCnt(); 2032 } 2033 } 2034 2035 // detach all tracks with same session ID from this chain 2036 for (size_t i = 0; i < mTracks.size(); ++i) { 2037 sp<Track> track = mTracks[i]; 2038 if (session == track->sessionId()) { 2039 track->setMainBuffer(mMixBuffer); 2040 chain->decTrackCnt(); 2041 } 2042 } 2043 break; 2044 } 2045 } 2046 return mEffectChains.size(); 2047} 2048 2049status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2050 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2051{ 2052 Mutex::Autolock _l(mLock); 2053 return attachAuxEffect_l(track, EffectId); 2054} 2055 2056status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2057 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2058{ 2059 status_t status = NO_ERROR; 2060 2061 if (EffectId == 0) { 2062 track->setAuxBuffer(0, NULL); 2063 } else { 2064 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2065 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2066 if (effect != 0) { 2067 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2068 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2069 } else { 2070 status = INVALID_OPERATION; 2071 } 2072 } else { 2073 status = BAD_VALUE; 2074 } 2075 } 2076 return status; 2077} 2078 2079void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2080{ 2081 for (size_t i = 0; i < mTracks.size(); ++i) { 2082 sp<Track> track = mTracks[i]; 2083 if (track->auxEffectId() == effectId) { 2084 attachAuxEffect_l(track, 0); 2085 } 2086 } 2087} 2088 2089bool AudioFlinger::PlaybackThread::threadLoop() 2090{ 2091 Vector< sp<Track> > tracksToRemove; 2092 2093 standbyTime = systemTime(); 2094 2095 // MIXER 2096 nsecs_t lastWarning = 0; 2097 2098 // DUPLICATING 2099 // FIXME could this be made local to while loop? 2100 writeFrames = 0; 2101 2102 cacheParameters_l(); 2103 sleepTime = idleSleepTime; 2104 2105 if (mType == MIXER) { 2106 sleepTimeShift = 0; 2107 } 2108 2109 CpuStats cpuStats; 2110 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2111 2112 acquireWakeLock(); 2113 2114 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2115 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2116 // and then that string will be logged at the next convenient opportunity. 2117 const char *logString = NULL; 2118 2119 checkSilentMode_l(); 2120 2121 while (!exitPending()) 2122 { 2123 cpuStats.sample(myName); 2124 2125 Vector< sp<EffectChain> > effectChains; 2126 2127 processConfigEvents(); 2128 2129 { // scope for mLock 2130 2131 Mutex::Autolock _l(mLock); 2132 2133 if (logString != NULL) { 2134 mNBLogWriter->logTimestamp(); 2135 mNBLogWriter->log(logString); 2136 logString = NULL; 2137 } 2138 2139 if (mLatchDValid) { 2140 mLatchQ = mLatchD; 2141 mLatchDValid = false; 2142 mLatchQValid = true; 2143 } 2144 2145 if (checkForNewParameters_l()) { 2146 cacheParameters_l(); 2147 } 2148 2149 saveOutputTracks(); 2150 if (mSignalPending) { 2151 // A signal was raised while we were unlocked 2152 mSignalPending = false; 2153 } else if (waitingAsyncCallback_l()) { 2154 if (exitPending()) { 2155 break; 2156 } 2157 releaseWakeLock_l(); 2158 ALOGV("wait async completion"); 2159 mWaitWorkCV.wait(mLock); 2160 ALOGV("async completion/wake"); 2161 acquireWakeLock_l(); 2162 standbyTime = systemTime() + standbyDelay; 2163 sleepTime = 0; 2164 2165 continue; 2166 } 2167 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2168 isSuspended()) { 2169 // put audio hardware into standby after short delay 2170 if (shouldStandby_l()) { 2171 2172 threadLoop_standby(); 2173 2174 mStandby = true; 2175 } 2176 2177 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2178 // we're about to wait, flush the binder command buffer 2179 IPCThreadState::self()->flushCommands(); 2180 2181 clearOutputTracks(); 2182 2183 if (exitPending()) { 2184 break; 2185 } 2186 2187 releaseWakeLock_l(); 2188 // wait until we have something to do... 2189 ALOGV("%s going to sleep", myName.string()); 2190 mWaitWorkCV.wait(mLock); 2191 ALOGV("%s waking up", myName.string()); 2192 acquireWakeLock_l(); 2193 2194 mMixerStatus = MIXER_IDLE; 2195 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2196 mBytesWritten = 0; 2197 mBytesRemaining = 0; 2198 checkSilentMode_l(); 2199 2200 standbyTime = systemTime() + standbyDelay; 2201 sleepTime = idleSleepTime; 2202 if (mType == MIXER) { 2203 sleepTimeShift = 0; 2204 } 2205 2206 continue; 2207 } 2208 } 2209 // mMixerStatusIgnoringFastTracks is also updated internally 2210 mMixerStatus = prepareTracks_l(&tracksToRemove); 2211 2212 // prevent any changes in effect chain list and in each effect chain 2213 // during mixing and effect process as the audio buffers could be deleted 2214 // or modified if an effect is created or deleted 2215 lockEffectChains_l(effectChains); 2216 } 2217 2218 if (mBytesRemaining == 0) { 2219 mCurrentWriteLength = 0; 2220 if (mMixerStatus == MIXER_TRACKS_READY) { 2221 // threadLoop_mix() sets mCurrentWriteLength 2222 threadLoop_mix(); 2223 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2224 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2225 // threadLoop_sleepTime sets sleepTime to 0 if data 2226 // must be written to HAL 2227 threadLoop_sleepTime(); 2228 if (sleepTime == 0) { 2229 mCurrentWriteLength = mixBufferSize; 2230 } 2231 } 2232 mBytesRemaining = mCurrentWriteLength; 2233 if (isSuspended()) { 2234 sleepTime = suspendSleepTimeUs(); 2235 // simulate write to HAL when suspended 2236 mBytesWritten += mixBufferSize; 2237 mBytesRemaining = 0; 2238 } 2239 2240 // only process effects if we're going to write 2241 if (sleepTime == 0) { 2242 for (size_t i = 0; i < effectChains.size(); i ++) { 2243 effectChains[i]->process_l(); 2244 } 2245 } 2246 } 2247 2248 // enable changes in effect chain 2249 unlockEffectChains(effectChains); 2250 2251 if (!waitingAsyncCallback()) { 2252 // sleepTime == 0 means we must write to audio hardware 2253 if (sleepTime == 0) { 2254 if (mBytesRemaining) { 2255 ssize_t ret = threadLoop_write(); 2256 if (ret < 0) { 2257 mBytesRemaining = 0; 2258 } else { 2259 mBytesWritten += ret; 2260 mBytesRemaining -= ret; 2261 } 2262 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2263 (mMixerStatus == MIXER_DRAIN_ALL)) { 2264 threadLoop_drain(); 2265 } 2266if (mType == MIXER) { 2267 // write blocked detection 2268 nsecs_t now = systemTime(); 2269 nsecs_t delta = now - mLastWriteTime; 2270 if (!mStandby && delta > maxPeriod) { 2271 mNumDelayedWrites++; 2272 if ((now - lastWarning) > kWarningThrottleNs) { 2273 ATRACE_NAME("underrun"); 2274 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2275 ns2ms(delta), mNumDelayedWrites, this); 2276 lastWarning = now; 2277 } 2278 } 2279} 2280 2281 mStandby = false; 2282 } else { 2283 usleep(sleepTime); 2284 } 2285 } 2286 2287 // Finally let go of removed track(s), without the lock held 2288 // since we can't guarantee the destructors won't acquire that 2289 // same lock. This will also mutate and push a new fast mixer state. 2290 threadLoop_removeTracks(tracksToRemove); 2291 tracksToRemove.clear(); 2292 2293 // FIXME I don't understand the need for this here; 2294 // it was in the original code but maybe the 2295 // assignment in saveOutputTracks() makes this unnecessary? 2296 clearOutputTracks(); 2297 2298 // Effect chains will be actually deleted here if they were removed from 2299 // mEffectChains list during mixing or effects processing 2300 effectChains.clear(); 2301 2302 // FIXME Note that the above .clear() is no longer necessary since effectChains 2303 // is now local to this block, but will keep it for now (at least until merge done). 2304 } 2305 2306 threadLoop_exit(); 2307 2308 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2309 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2310 // put output stream into standby mode 2311 if (!mStandby) { 2312 mOutput->stream->common.standby(&mOutput->stream->common); 2313 } 2314 } 2315 2316 releaseWakeLock(); 2317 2318 ALOGV("Thread %p type %d exiting", this, mType); 2319 return false; 2320} 2321 2322// removeTracks_l() must be called with ThreadBase::mLock held 2323void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2324{ 2325 size_t count = tracksToRemove.size(); 2326 if (count) { 2327 for (size_t i=0 ; i<count ; i++) { 2328 const sp<Track>& track = tracksToRemove.itemAt(i); 2329 mActiveTracks.remove(track); 2330 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2331 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2332 if (chain != 0) { 2333 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2334 track->sessionId()); 2335 chain->decActiveTrackCnt(); 2336 } 2337 if (track->isTerminated()) { 2338 removeTrack_l(track); 2339 } 2340 } 2341 } 2342 2343} 2344 2345status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2346{ 2347 if (mNormalSink != 0) { 2348 return mNormalSink->getTimestamp(timestamp); 2349 } 2350 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2351 uint64_t position64; 2352 int ret = mOutput->stream->get_presentation_position( 2353 mOutput->stream, &position64, ×tamp.mTime); 2354 if (ret == 0) { 2355 timestamp.mPosition = (uint32_t)position64; 2356 return NO_ERROR; 2357 } 2358 } 2359 return INVALID_OPERATION; 2360} 2361// ---------------------------------------------------------------------------- 2362 2363AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2364 audio_io_handle_t id, audio_devices_t device, type_t type) 2365 : PlaybackThread(audioFlinger, output, id, device, type), 2366 // mAudioMixer below 2367 // mFastMixer below 2368 mFastMixerFutex(0) 2369 // mOutputSink below 2370 // mPipeSink below 2371 // mNormalSink below 2372{ 2373 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2374 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2375 "mFrameCount=%d, mNormalFrameCount=%d", 2376 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2377 mNormalFrameCount); 2378 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2379 2380 // FIXME - Current mixer implementation only supports stereo output 2381 if (mChannelCount != FCC_2) { 2382 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2383 } 2384 2385 // create an NBAIO sink for the HAL output stream, and negotiate 2386 mOutputSink = new AudioStreamOutSink(output->stream); 2387 size_t numCounterOffers = 0; 2388 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2389 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2390 ALOG_ASSERT(index == 0); 2391 2392 // initialize fast mixer depending on configuration 2393 bool initFastMixer; 2394 switch (kUseFastMixer) { 2395 case FastMixer_Never: 2396 initFastMixer = false; 2397 break; 2398 case FastMixer_Always: 2399 initFastMixer = true; 2400 break; 2401 case FastMixer_Static: 2402 case FastMixer_Dynamic: 2403 initFastMixer = mFrameCount < mNormalFrameCount; 2404 break; 2405 } 2406 if (initFastMixer) { 2407 2408 // create a MonoPipe to connect our submix to FastMixer 2409 NBAIO_Format format = mOutputSink->format(); 2410 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2411 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2412 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2413 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2414 const NBAIO_Format offers[1] = {format}; 2415 size_t numCounterOffers = 0; 2416 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2417 ALOG_ASSERT(index == 0); 2418 monoPipe->setAvgFrames((mScreenState & 1) ? 2419 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2420 mPipeSink = monoPipe; 2421 2422#ifdef TEE_SINK 2423 if (mTeeSinkOutputEnabled) { 2424 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2425 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2426 numCounterOffers = 0; 2427 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2428 ALOG_ASSERT(index == 0); 2429 mTeeSink = teeSink; 2430 PipeReader *teeSource = new PipeReader(*teeSink); 2431 numCounterOffers = 0; 2432 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2433 ALOG_ASSERT(index == 0); 2434 mTeeSource = teeSource; 2435 } 2436#endif 2437 2438 // create fast mixer and configure it initially with just one fast track for our submix 2439 mFastMixer = new FastMixer(); 2440 FastMixerStateQueue *sq = mFastMixer->sq(); 2441#ifdef STATE_QUEUE_DUMP 2442 sq->setObserverDump(&mStateQueueObserverDump); 2443 sq->setMutatorDump(&mStateQueueMutatorDump); 2444#endif 2445 FastMixerState *state = sq->begin(); 2446 FastTrack *fastTrack = &state->mFastTracks[0]; 2447 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2448 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2449 fastTrack->mVolumeProvider = NULL; 2450 fastTrack->mGeneration++; 2451 state->mFastTracksGen++; 2452 state->mTrackMask = 1; 2453 // fast mixer will use the HAL output sink 2454 state->mOutputSink = mOutputSink.get(); 2455 state->mOutputSinkGen++; 2456 state->mFrameCount = mFrameCount; 2457 state->mCommand = FastMixerState::COLD_IDLE; 2458 // already done in constructor initialization list 2459 //mFastMixerFutex = 0; 2460 state->mColdFutexAddr = &mFastMixerFutex; 2461 state->mColdGen++; 2462 state->mDumpState = &mFastMixerDumpState; 2463#ifdef TEE_SINK 2464 state->mTeeSink = mTeeSink.get(); 2465#endif 2466 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2467 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2468 sq->end(); 2469 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2470 2471 // start the fast mixer 2472 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2473 pid_t tid = mFastMixer->getTid(); 2474 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2475 if (err != 0) { 2476 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2477 kPriorityFastMixer, getpid_cached, tid, err); 2478 } 2479 2480#ifdef AUDIO_WATCHDOG 2481 // create and start the watchdog 2482 mAudioWatchdog = new AudioWatchdog(); 2483 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2484 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2485 tid = mAudioWatchdog->getTid(); 2486 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2487 if (err != 0) { 2488 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2489 kPriorityFastMixer, getpid_cached, tid, err); 2490 } 2491#endif 2492 2493 } else { 2494 mFastMixer = NULL; 2495 } 2496 2497 switch (kUseFastMixer) { 2498 case FastMixer_Never: 2499 case FastMixer_Dynamic: 2500 mNormalSink = mOutputSink; 2501 break; 2502 case FastMixer_Always: 2503 mNormalSink = mPipeSink; 2504 break; 2505 case FastMixer_Static: 2506 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2507 break; 2508 } 2509} 2510 2511AudioFlinger::MixerThread::~MixerThread() 2512{ 2513 if (mFastMixer != NULL) { 2514 FastMixerStateQueue *sq = mFastMixer->sq(); 2515 FastMixerState *state = sq->begin(); 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 } 2522 state->mCommand = FastMixerState::EXIT; 2523 sq->end(); 2524 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2525 mFastMixer->join(); 2526 // Though the fast mixer thread has exited, it's state queue is still valid. 2527 // We'll use that extract the final state which contains one remaining fast track 2528 // corresponding to our sub-mix. 2529 state = sq->begin(); 2530 ALOG_ASSERT(state->mTrackMask == 1); 2531 FastTrack *fastTrack = &state->mFastTracks[0]; 2532 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2533 delete fastTrack->mBufferProvider; 2534 sq->end(false /*didModify*/); 2535 delete mFastMixer; 2536#ifdef AUDIO_WATCHDOG 2537 if (mAudioWatchdog != 0) { 2538 mAudioWatchdog->requestExit(); 2539 mAudioWatchdog->requestExitAndWait(); 2540 mAudioWatchdog.clear(); 2541 } 2542#endif 2543 } 2544 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2545 delete mAudioMixer; 2546} 2547 2548 2549uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2550{ 2551 if (mFastMixer != NULL) { 2552 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2553 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2554 } 2555 return latency; 2556} 2557 2558 2559void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2560{ 2561 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2562} 2563 2564ssize_t AudioFlinger::MixerThread::threadLoop_write() 2565{ 2566 // FIXME we should only do one push per cycle; confirm this is true 2567 // Start the fast mixer if it's not already running 2568 if (mFastMixer != NULL) { 2569 FastMixerStateQueue *sq = mFastMixer->sq(); 2570 FastMixerState *state = sq->begin(); 2571 if (state->mCommand != FastMixerState::MIX_WRITE && 2572 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2573 if (state->mCommand == FastMixerState::COLD_IDLE) { 2574 int32_t old = android_atomic_inc(&mFastMixerFutex); 2575 if (old == -1) { 2576 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2577 } 2578#ifdef AUDIO_WATCHDOG 2579 if (mAudioWatchdog != 0) { 2580 mAudioWatchdog->resume(); 2581 } 2582#endif 2583 } 2584 state->mCommand = FastMixerState::MIX_WRITE; 2585 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2586 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2587 sq->end(); 2588 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2589 if (kUseFastMixer == FastMixer_Dynamic) { 2590 mNormalSink = mPipeSink; 2591 } 2592 } else { 2593 sq->end(false /*didModify*/); 2594 } 2595 } 2596 return PlaybackThread::threadLoop_write(); 2597} 2598 2599void AudioFlinger::MixerThread::threadLoop_standby() 2600{ 2601 // Idle the fast mixer if it's currently running 2602 if (mFastMixer != NULL) { 2603 FastMixerStateQueue *sq = mFastMixer->sq(); 2604 FastMixerState *state = sq->begin(); 2605 if (!(state->mCommand & FastMixerState::IDLE)) { 2606 state->mCommand = FastMixerState::COLD_IDLE; 2607 state->mColdFutexAddr = &mFastMixerFutex; 2608 state->mColdGen++; 2609 mFastMixerFutex = 0; 2610 sq->end(); 2611 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2612 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2613 if (kUseFastMixer == FastMixer_Dynamic) { 2614 mNormalSink = mOutputSink; 2615 } 2616#ifdef AUDIO_WATCHDOG 2617 if (mAudioWatchdog != 0) { 2618 mAudioWatchdog->pause(); 2619 } 2620#endif 2621 } else { 2622 sq->end(false /*didModify*/); 2623 } 2624 } 2625 PlaybackThread::threadLoop_standby(); 2626} 2627 2628// Empty implementation for standard mixer 2629// Overridden for offloaded playback 2630void AudioFlinger::PlaybackThread::flushOutput_l() 2631{ 2632} 2633 2634bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2635{ 2636 return false; 2637} 2638 2639bool AudioFlinger::PlaybackThread::shouldStandby_l() 2640{ 2641 return !mStandby; 2642} 2643 2644bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2645{ 2646 Mutex::Autolock _l(mLock); 2647 return waitingAsyncCallback_l(); 2648} 2649 2650// shared by MIXER and DIRECT, overridden by DUPLICATING 2651void AudioFlinger::PlaybackThread::threadLoop_standby() 2652{ 2653 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2654 mOutput->stream->common.standby(&mOutput->stream->common); 2655 if (mUseAsyncWrite != 0) { 2656 // discard any pending drain or write ack by incrementing sequence 2657 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2658 mDrainSequence = (mDrainSequence + 2) & ~1; 2659 ALOG_ASSERT(mCallbackThread != 0); 2660 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2661 mCallbackThread->setDraining(mDrainSequence); 2662 } 2663} 2664 2665void AudioFlinger::MixerThread::threadLoop_mix() 2666{ 2667 // obtain the presentation timestamp of the next output buffer 2668 int64_t pts; 2669 status_t status = INVALID_OPERATION; 2670 2671 if (mNormalSink != 0) { 2672 status = mNormalSink->getNextWriteTimestamp(&pts); 2673 } else { 2674 status = mOutputSink->getNextWriteTimestamp(&pts); 2675 } 2676 2677 if (status != NO_ERROR) { 2678 pts = AudioBufferProvider::kInvalidPTS; 2679 } 2680 2681 // mix buffers... 2682 mAudioMixer->process(pts); 2683 mCurrentWriteLength = mixBufferSize; 2684 // increase sleep time progressively when application underrun condition clears. 2685 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2686 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2687 // such that we would underrun the audio HAL. 2688 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2689 sleepTimeShift--; 2690 } 2691 sleepTime = 0; 2692 standbyTime = systemTime() + standbyDelay; 2693 //TODO: delay standby when effects have a tail 2694} 2695 2696void AudioFlinger::MixerThread::threadLoop_sleepTime() 2697{ 2698 // If no tracks are ready, sleep once for the duration of an output 2699 // buffer size, then write 0s to the output 2700 if (sleepTime == 0) { 2701 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2702 sleepTime = activeSleepTime >> sleepTimeShift; 2703 if (sleepTime < kMinThreadSleepTimeUs) { 2704 sleepTime = kMinThreadSleepTimeUs; 2705 } 2706 // reduce sleep time in case of consecutive application underruns to avoid 2707 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2708 // duration we would end up writing less data than needed by the audio HAL if 2709 // the condition persists. 2710 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2711 sleepTimeShift++; 2712 } 2713 } else { 2714 sleepTime = idleSleepTime; 2715 } 2716 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2717 memset (mMixBuffer, 0, mixBufferSize); 2718 sleepTime = 0; 2719 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2720 "anticipated start"); 2721 } 2722 // TODO add standby time extension fct of effect tail 2723} 2724 2725// prepareTracks_l() must be called with ThreadBase::mLock held 2726AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2727 Vector< sp<Track> > *tracksToRemove) 2728{ 2729 2730 mixer_state mixerStatus = MIXER_IDLE; 2731 // find out which tracks need to be processed 2732 size_t count = mActiveTracks.size(); 2733 size_t mixedTracks = 0; 2734 size_t tracksWithEffect = 0; 2735 // counts only _active_ fast tracks 2736 size_t fastTracks = 0; 2737 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2738 2739 float masterVolume = mMasterVolume; 2740 bool masterMute = mMasterMute; 2741 2742 if (masterMute) { 2743 masterVolume = 0; 2744 } 2745 // Delegate master volume control to effect in output mix effect chain if needed 2746 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2747 if (chain != 0) { 2748 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2749 chain->setVolume_l(&v, &v); 2750 masterVolume = (float)((v + (1 << 23)) >> 24); 2751 chain.clear(); 2752 } 2753 2754 // prepare a new state to push 2755 FastMixerStateQueue *sq = NULL; 2756 FastMixerState *state = NULL; 2757 bool didModify = false; 2758 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2759 if (mFastMixer != NULL) { 2760 sq = mFastMixer->sq(); 2761 state = sq->begin(); 2762 } 2763 2764 for (size_t i=0 ; i<count ; i++) { 2765 const sp<Track> t = mActiveTracks[i].promote(); 2766 if (t == 0) { 2767 continue; 2768 } 2769 2770 // this const just means the local variable doesn't change 2771 Track* const track = t.get(); 2772 2773 // process fast tracks 2774 if (track->isFastTrack()) { 2775 2776 // It's theoretically possible (though unlikely) for a fast track to be created 2777 // and then removed within the same normal mix cycle. This is not a problem, as 2778 // the track never becomes active so it's fast mixer slot is never touched. 2779 // The converse, of removing an (active) track and then creating a new track 2780 // at the identical fast mixer slot within the same normal mix cycle, 2781 // is impossible because the slot isn't marked available until the end of each cycle. 2782 int j = track->mFastIndex; 2783 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2784 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2785 FastTrack *fastTrack = &state->mFastTracks[j]; 2786 2787 // Determine whether the track is currently in underrun condition, 2788 // and whether it had a recent underrun. 2789 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2790 FastTrackUnderruns underruns = ftDump->mUnderruns; 2791 uint32_t recentFull = (underruns.mBitFields.mFull - 2792 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2793 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2794 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2795 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2796 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2797 uint32_t recentUnderruns = recentPartial + recentEmpty; 2798 track->mObservedUnderruns = underruns; 2799 // don't count underruns that occur while stopping or pausing 2800 // or stopped which can occur when flush() is called while active 2801 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2802 recentUnderruns > 0) { 2803 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2804 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2805 } 2806 2807 // This is similar to the state machine for normal tracks, 2808 // with a few modifications for fast tracks. 2809 bool isActive = true; 2810 switch (track->mState) { 2811 case TrackBase::STOPPING_1: 2812 // track stays active in STOPPING_1 state until first underrun 2813 if (recentUnderruns > 0 || track->isTerminated()) { 2814 track->mState = TrackBase::STOPPING_2; 2815 } 2816 break; 2817 case TrackBase::PAUSING: 2818 // ramp down is not yet implemented 2819 track->setPaused(); 2820 break; 2821 case TrackBase::RESUMING: 2822 // ramp up is not yet implemented 2823 track->mState = TrackBase::ACTIVE; 2824 break; 2825 case TrackBase::ACTIVE: 2826 if (recentFull > 0 || recentPartial > 0) { 2827 // track has provided at least some frames recently: reset retry count 2828 track->mRetryCount = kMaxTrackRetries; 2829 } 2830 if (recentUnderruns == 0) { 2831 // no recent underruns: stay active 2832 break; 2833 } 2834 // there has recently been an underrun of some kind 2835 if (track->sharedBuffer() == 0) { 2836 // were any of the recent underruns "empty" (no frames available)? 2837 if (recentEmpty == 0) { 2838 // no, then ignore the partial underruns as they are allowed indefinitely 2839 break; 2840 } 2841 // there has recently been an "empty" underrun: decrement the retry counter 2842 if (--(track->mRetryCount) > 0) { 2843 break; 2844 } 2845 // indicate to client process that the track was disabled because of underrun; 2846 // it will then automatically call start() when data is available 2847 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2848 // remove from active list, but state remains ACTIVE [confusing but true] 2849 isActive = false; 2850 break; 2851 } 2852 // fall through 2853 case TrackBase::STOPPING_2: 2854 case TrackBase::PAUSED: 2855 case TrackBase::STOPPED: 2856 case TrackBase::FLUSHED: // flush() while active 2857 // Check for presentation complete if track is inactive 2858 // We have consumed all the buffers of this track. 2859 // This would be incomplete if we auto-paused on underrun 2860 { 2861 size_t audioHALFrames = 2862 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2863 size_t framesWritten = mBytesWritten / mFrameSize; 2864 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2865 // track stays in active list until presentation is complete 2866 break; 2867 } 2868 } 2869 if (track->isStopping_2()) { 2870 track->mState = TrackBase::STOPPED; 2871 } 2872 if (track->isStopped()) { 2873 // Can't reset directly, as fast mixer is still polling this track 2874 // track->reset(); 2875 // So instead mark this track as needing to be reset after push with ack 2876 resetMask |= 1 << i; 2877 } 2878 isActive = false; 2879 break; 2880 case TrackBase::IDLE: 2881 default: 2882 LOG_FATAL("unexpected track state %d", track->mState); 2883 } 2884 2885 if (isActive) { 2886 // was it previously inactive? 2887 if (!(state->mTrackMask & (1 << j))) { 2888 ExtendedAudioBufferProvider *eabp = track; 2889 VolumeProvider *vp = track; 2890 fastTrack->mBufferProvider = eabp; 2891 fastTrack->mVolumeProvider = vp; 2892 fastTrack->mSampleRate = track->mSampleRate; 2893 fastTrack->mChannelMask = track->mChannelMask; 2894 fastTrack->mGeneration++; 2895 state->mTrackMask |= 1 << j; 2896 didModify = true; 2897 // no acknowledgement required for newly active tracks 2898 } 2899 // cache the combined master volume and stream type volume for fast mixer; this 2900 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2901 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2902 ++fastTracks; 2903 } else { 2904 // was it previously active? 2905 if (state->mTrackMask & (1 << j)) { 2906 fastTrack->mBufferProvider = NULL; 2907 fastTrack->mGeneration++; 2908 state->mTrackMask &= ~(1 << j); 2909 didModify = true; 2910 // If any fast tracks were removed, we must wait for acknowledgement 2911 // because we're about to decrement the last sp<> on those tracks. 2912 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2913 } else { 2914 LOG_FATAL("fast track %d should have been active", j); 2915 } 2916 tracksToRemove->add(track); 2917 // Avoids a misleading display in dumpsys 2918 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2919 } 2920 continue; 2921 } 2922 2923 { // local variable scope to avoid goto warning 2924 2925 audio_track_cblk_t* cblk = track->cblk(); 2926 2927 // The first time a track is added we wait 2928 // for all its buffers to be filled before processing it 2929 int name = track->name(); 2930 // make sure that we have enough frames to mix one full buffer. 2931 // enforce this condition only once to enable draining the buffer in case the client 2932 // app does not call stop() and relies on underrun to stop: 2933 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2934 // during last round 2935 size_t desiredFrames; 2936 uint32_t sr = track->sampleRate(); 2937 if (sr == mSampleRate) { 2938 desiredFrames = mNormalFrameCount; 2939 } else { 2940 // +1 for rounding and +1 for additional sample needed for interpolation 2941 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2942 // add frames already consumed but not yet released by the resampler 2943 // because cblk->framesReady() will include these frames 2944 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2945 // the minimum track buffer size is normally twice the number of frames necessary 2946 // to fill one buffer and the resampler should not leave more than one buffer worth 2947 // of unreleased frames after each pass, but just in case... 2948 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2949 } 2950 uint32_t minFrames = 1; 2951 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2952 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2953 minFrames = desiredFrames; 2954 } 2955 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2956 size_t framesReady; 2957 if (track->sharedBuffer() == 0) { 2958 framesReady = track->framesReady(); 2959 } else if (track->isStopped()) { 2960 framesReady = 0; 2961 } else { 2962 framesReady = 1; 2963 } 2964 if ((framesReady >= minFrames) && track->isReady() && 2965 !track->isPaused() && !track->isTerminated()) 2966 { 2967 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2968 2969 mixedTracks++; 2970 2971 // track->mainBuffer() != mMixBuffer means there is an effect chain 2972 // connected to the track 2973 chain.clear(); 2974 if (track->mainBuffer() != mMixBuffer) { 2975 chain = getEffectChain_l(track->sessionId()); 2976 // Delegate volume control to effect in track effect chain if needed 2977 if (chain != 0) { 2978 tracksWithEffect++; 2979 } else { 2980 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2981 "session %d", 2982 name, track->sessionId()); 2983 } 2984 } 2985 2986 2987 int param = AudioMixer::VOLUME; 2988 if (track->mFillingUpStatus == Track::FS_FILLED) { 2989 // no ramp for the first volume setting 2990 track->mFillingUpStatus = Track::FS_ACTIVE; 2991 if (track->mState == TrackBase::RESUMING) { 2992 track->mState = TrackBase::ACTIVE; 2993 param = AudioMixer::RAMP_VOLUME; 2994 } 2995 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2996 // FIXME should not make a decision based on mServer 2997 } else if (cblk->mServer != 0) { 2998 // If the track is stopped before the first frame was mixed, 2999 // do not apply ramp 3000 param = AudioMixer::RAMP_VOLUME; 3001 } 3002 3003 // compute volume for this track 3004 uint32_t vl, vr, va; 3005 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3006 vl = vr = va = 0; 3007 if (track->isPausing()) { 3008 track->setPaused(); 3009 } 3010 } else { 3011 3012 // read original volumes with volume control 3013 float typeVolume = mStreamTypes[track->streamType()].volume; 3014 float v = masterVolume * typeVolume; 3015 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3016 uint32_t vlr = proxy->getVolumeLR(); 3017 vl = vlr & 0xFFFF; 3018 vr = vlr >> 16; 3019 // track volumes come from shared memory, so can't be trusted and must be clamped 3020 if (vl > MAX_GAIN_INT) { 3021 ALOGV("Track left volume out of range: %04X", vl); 3022 vl = MAX_GAIN_INT; 3023 } 3024 if (vr > MAX_GAIN_INT) { 3025 ALOGV("Track right volume out of range: %04X", vr); 3026 vr = MAX_GAIN_INT; 3027 } 3028 // now apply the master volume and stream type volume 3029 vl = (uint32_t)(v * vl) << 12; 3030 vr = (uint32_t)(v * vr) << 12; 3031 // assuming master volume and stream type volume each go up to 1.0, 3032 // vl and vr are now in 8.24 format 3033 3034 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3035 // send level comes from shared memory and so may be corrupt 3036 if (sendLevel > MAX_GAIN_INT) { 3037 ALOGV("Track send level out of range: %04X", sendLevel); 3038 sendLevel = MAX_GAIN_INT; 3039 } 3040 va = (uint32_t)(v * sendLevel); 3041 } 3042 3043 // Delegate volume control to effect in track effect chain if needed 3044 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3045 // Do not ramp volume if volume is controlled by effect 3046 param = AudioMixer::VOLUME; 3047 track->mHasVolumeController = true; 3048 } else { 3049 // force no volume ramp when volume controller was just disabled or removed 3050 // from effect chain to avoid volume spike 3051 if (track->mHasVolumeController) { 3052 param = AudioMixer::VOLUME; 3053 } 3054 track->mHasVolumeController = false; 3055 } 3056 3057 // Convert volumes from 8.24 to 4.12 format 3058 // This additional clamping is needed in case chain->setVolume_l() overshot 3059 vl = (vl + (1 << 11)) >> 12; 3060 if (vl > MAX_GAIN_INT) { 3061 vl = MAX_GAIN_INT; 3062 } 3063 vr = (vr + (1 << 11)) >> 12; 3064 if (vr > MAX_GAIN_INT) { 3065 vr = MAX_GAIN_INT; 3066 } 3067 3068 if (va > MAX_GAIN_INT) { 3069 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3070 } 3071 3072 // XXX: these things DON'T need to be done each time 3073 mAudioMixer->setBufferProvider(name, track); 3074 mAudioMixer->enable(name); 3075 3076 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3077 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3078 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3079 mAudioMixer->setParameter( 3080 name, 3081 AudioMixer::TRACK, 3082 AudioMixer::FORMAT, (void *)track->format()); 3083 mAudioMixer->setParameter( 3084 name, 3085 AudioMixer::TRACK, 3086 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3087 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3088 uint32_t maxSampleRate = mSampleRate * 2; 3089 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3090 if (reqSampleRate == 0) { 3091 reqSampleRate = mSampleRate; 3092 } else if (reqSampleRate > maxSampleRate) { 3093 reqSampleRate = maxSampleRate; 3094 } 3095 mAudioMixer->setParameter( 3096 name, 3097 AudioMixer::RESAMPLE, 3098 AudioMixer::SAMPLE_RATE, 3099 (void *)reqSampleRate); 3100 mAudioMixer->setParameter( 3101 name, 3102 AudioMixer::TRACK, 3103 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3104 mAudioMixer->setParameter( 3105 name, 3106 AudioMixer::TRACK, 3107 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3108 3109 // reset retry count 3110 track->mRetryCount = kMaxTrackRetries; 3111 3112 // If one track is ready, set the mixer ready if: 3113 // - the mixer was not ready during previous round OR 3114 // - no other track is not ready 3115 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3116 mixerStatus != MIXER_TRACKS_ENABLED) { 3117 mixerStatus = MIXER_TRACKS_READY; 3118 } 3119 } else { 3120 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3121 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3122 } 3123 // clear effect chain input buffer if an active track underruns to avoid sending 3124 // previous audio buffer again to effects 3125 chain = getEffectChain_l(track->sessionId()); 3126 if (chain != 0) { 3127 chain->clearInputBuffer(); 3128 } 3129 3130 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3131 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3132 track->isStopped() || track->isPaused()) { 3133 // We have consumed all the buffers of this track. 3134 // Remove it from the list of active tracks. 3135 // TODO: use actual buffer filling status instead of latency when available from 3136 // audio HAL 3137 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3138 size_t framesWritten = mBytesWritten / mFrameSize; 3139 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3140 if (track->isStopped()) { 3141 track->reset(); 3142 } 3143 tracksToRemove->add(track); 3144 } 3145 } else { 3146 // No buffers for this track. Give it a few chances to 3147 // fill a buffer, then remove it from active list. 3148 if (--(track->mRetryCount) <= 0) { 3149 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3150 tracksToRemove->add(track); 3151 // indicate to client process that the track was disabled because of underrun; 3152 // it will then automatically call start() when data is available 3153 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3154 // If one track is not ready, mark the mixer also not ready if: 3155 // - the mixer was ready during previous round OR 3156 // - no other track is ready 3157 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3158 mixerStatus != MIXER_TRACKS_READY) { 3159 mixerStatus = MIXER_TRACKS_ENABLED; 3160 } 3161 } 3162 mAudioMixer->disable(name); 3163 } 3164 3165 } // local variable scope to avoid goto warning 3166track_is_ready: ; 3167 3168 } 3169 3170 // Push the new FastMixer state if necessary 3171 bool pauseAudioWatchdog = false; 3172 if (didModify) { 3173 state->mFastTracksGen++; 3174 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3175 if (kUseFastMixer == FastMixer_Dynamic && 3176 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3177 state->mCommand = FastMixerState::COLD_IDLE; 3178 state->mColdFutexAddr = &mFastMixerFutex; 3179 state->mColdGen++; 3180 mFastMixerFutex = 0; 3181 if (kUseFastMixer == FastMixer_Dynamic) { 3182 mNormalSink = mOutputSink; 3183 } 3184 // If we go into cold idle, need to wait for acknowledgement 3185 // so that fast mixer stops doing I/O. 3186 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3187 pauseAudioWatchdog = true; 3188 } 3189 } 3190 if (sq != NULL) { 3191 sq->end(didModify); 3192 sq->push(block); 3193 } 3194#ifdef AUDIO_WATCHDOG 3195 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3196 mAudioWatchdog->pause(); 3197 } 3198#endif 3199 3200 // Now perform the deferred reset on fast tracks that have stopped 3201 while (resetMask != 0) { 3202 size_t i = __builtin_ctz(resetMask); 3203 ALOG_ASSERT(i < count); 3204 resetMask &= ~(1 << i); 3205 sp<Track> t = mActiveTracks[i].promote(); 3206 if (t == 0) { 3207 continue; 3208 } 3209 Track* track = t.get(); 3210 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3211 track->reset(); 3212 } 3213 3214 // remove all the tracks that need to be... 3215 removeTracks_l(*tracksToRemove); 3216 3217 // mix buffer must be cleared if all tracks are connected to an 3218 // effect chain as in this case the mixer will not write to 3219 // mix buffer and track effects will accumulate into it 3220 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3221 (mixedTracks == 0 && fastTracks > 0))) { 3222 // FIXME as a performance optimization, should remember previous zero status 3223 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3224 } 3225 3226 // if any fast tracks, then status is ready 3227 mMixerStatusIgnoringFastTracks = mixerStatus; 3228 if (fastTracks > 0) { 3229 mixerStatus = MIXER_TRACKS_READY; 3230 } 3231 return mixerStatus; 3232} 3233 3234// getTrackName_l() must be called with ThreadBase::mLock held 3235int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3236{ 3237 return mAudioMixer->getTrackName(channelMask, sessionId); 3238} 3239 3240// deleteTrackName_l() must be called with ThreadBase::mLock held 3241void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3242{ 3243 ALOGV("remove track (%d) and delete from mixer", name); 3244 mAudioMixer->deleteTrackName(name); 3245} 3246 3247// checkForNewParameters_l() must be called with ThreadBase::mLock held 3248bool AudioFlinger::MixerThread::checkForNewParameters_l() 3249{ 3250 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3251 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3252 bool reconfig = false; 3253 3254 while (!mNewParameters.isEmpty()) { 3255 3256 if (mFastMixer != NULL) { 3257 FastMixerStateQueue *sq = mFastMixer->sq(); 3258 FastMixerState *state = sq->begin(); 3259 if (!(state->mCommand & FastMixerState::IDLE)) { 3260 previousCommand = state->mCommand; 3261 state->mCommand = FastMixerState::HOT_IDLE; 3262 sq->end(); 3263 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3264 } else { 3265 sq->end(false /*didModify*/); 3266 } 3267 } 3268 3269 status_t status = NO_ERROR; 3270 String8 keyValuePair = mNewParameters[0]; 3271 AudioParameter param = AudioParameter(keyValuePair); 3272 int value; 3273 3274 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3275 reconfig = true; 3276 } 3277 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3278 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3279 status = BAD_VALUE; 3280 } else { 3281 reconfig = true; 3282 } 3283 } 3284 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3285 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3286 status = BAD_VALUE; 3287 } else { 3288 reconfig = true; 3289 } 3290 } 3291 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3292 // do not accept frame count changes if tracks are open as the track buffer 3293 // size depends on frame count and correct behavior would not be guaranteed 3294 // if frame count is changed after track creation 3295 if (!mTracks.isEmpty()) { 3296 status = INVALID_OPERATION; 3297 } else { 3298 reconfig = true; 3299 } 3300 } 3301 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3302#ifdef ADD_BATTERY_DATA 3303 // when changing the audio output device, call addBatteryData to notify 3304 // the change 3305 if (mOutDevice != value) { 3306 uint32_t params = 0; 3307 // check whether speaker is on 3308 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3309 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3310 } 3311 3312 audio_devices_t deviceWithoutSpeaker 3313 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3314 // check if any other device (except speaker) is on 3315 if (value & deviceWithoutSpeaker ) { 3316 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3317 } 3318 3319 if (params != 0) { 3320 addBatteryData(params); 3321 } 3322 } 3323#endif 3324 3325 // forward device change to effects that have requested to be 3326 // aware of attached audio device. 3327 if (value != AUDIO_DEVICE_NONE) { 3328 mOutDevice = value; 3329 for (size_t i = 0; i < mEffectChains.size(); i++) { 3330 mEffectChains[i]->setDevice_l(mOutDevice); 3331 } 3332 } 3333 } 3334 3335 if (status == NO_ERROR) { 3336 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3337 keyValuePair.string()); 3338 if (!mStandby && status == INVALID_OPERATION) { 3339 mOutput->stream->common.standby(&mOutput->stream->common); 3340 mStandby = true; 3341 mBytesWritten = 0; 3342 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3343 keyValuePair.string()); 3344 } 3345 if (status == NO_ERROR && reconfig) { 3346 readOutputParameters(); 3347 delete mAudioMixer; 3348 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3349 for (size_t i = 0; i < mTracks.size() ; i++) { 3350 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3351 if (name < 0) { 3352 break; 3353 } 3354 mTracks[i]->mName = name; 3355 } 3356 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3357 } 3358 } 3359 3360 mNewParameters.removeAt(0); 3361 3362 mParamStatus = status; 3363 mParamCond.signal(); 3364 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3365 // already timed out waiting for the status and will never signal the condition. 3366 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3367 } 3368 3369 if (!(previousCommand & FastMixerState::IDLE)) { 3370 ALOG_ASSERT(mFastMixer != NULL); 3371 FastMixerStateQueue *sq = mFastMixer->sq(); 3372 FastMixerState *state = sq->begin(); 3373 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3374 state->mCommand = previousCommand; 3375 sq->end(); 3376 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3377 } 3378 3379 return reconfig; 3380} 3381 3382 3383void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3384{ 3385 const size_t SIZE = 256; 3386 char buffer[SIZE]; 3387 String8 result; 3388 3389 PlaybackThread::dumpInternals(fd, args); 3390 3391 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3392 result.append(buffer); 3393 write(fd, result.string(), result.size()); 3394 3395 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3396 const FastMixerDumpState copy(mFastMixerDumpState); 3397 copy.dump(fd); 3398 3399#ifdef STATE_QUEUE_DUMP 3400 // Similar for state queue 3401 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3402 observerCopy.dump(fd); 3403 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3404 mutatorCopy.dump(fd); 3405#endif 3406 3407#ifdef TEE_SINK 3408 // Write the tee output to a .wav file 3409 dumpTee(fd, mTeeSource, mId); 3410#endif 3411 3412#ifdef AUDIO_WATCHDOG 3413 if (mAudioWatchdog != 0) { 3414 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3415 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3416 wdCopy.dump(fd); 3417 } 3418#endif 3419} 3420 3421uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3422{ 3423 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3424} 3425 3426uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3427{ 3428 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3429} 3430 3431void AudioFlinger::MixerThread::cacheParameters_l() 3432{ 3433 PlaybackThread::cacheParameters_l(); 3434 3435 // FIXME: Relaxed timing because of a certain device that can't meet latency 3436 // Should be reduced to 2x after the vendor fixes the driver issue 3437 // increase threshold again due to low power audio mode. The way this warning 3438 // threshold is calculated and its usefulness should be reconsidered anyway. 3439 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3440} 3441 3442// ---------------------------------------------------------------------------- 3443 3444AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3445 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3446 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3447 // mLeftVolFloat, mRightVolFloat 3448{ 3449} 3450 3451AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3452 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3453 ThreadBase::type_t type) 3454 : PlaybackThread(audioFlinger, output, id, device, type) 3455 // mLeftVolFloat, mRightVolFloat 3456{ 3457} 3458 3459AudioFlinger::DirectOutputThread::~DirectOutputThread() 3460{ 3461} 3462 3463void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3464{ 3465 audio_track_cblk_t* cblk = track->cblk(); 3466 float left, right; 3467 3468 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3469 left = right = 0; 3470 } else { 3471 float typeVolume = mStreamTypes[track->streamType()].volume; 3472 float v = mMasterVolume * typeVolume; 3473 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3474 uint32_t vlr = proxy->getVolumeLR(); 3475 float v_clamped = v * (vlr & 0xFFFF); 3476 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3477 left = v_clamped/MAX_GAIN; 3478 v_clamped = v * (vlr >> 16); 3479 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3480 right = v_clamped/MAX_GAIN; 3481 } 3482 3483 if (lastTrack) { 3484 if (left != mLeftVolFloat || right != mRightVolFloat) { 3485 mLeftVolFloat = left; 3486 mRightVolFloat = right; 3487 3488 // Convert volumes from float to 8.24 3489 uint32_t vl = (uint32_t)(left * (1 << 24)); 3490 uint32_t vr = (uint32_t)(right * (1 << 24)); 3491 3492 // Delegate volume control to effect in track effect chain if needed 3493 // only one effect chain can be present on DirectOutputThread, so if 3494 // there is one, the track is connected to it 3495 if (!mEffectChains.isEmpty()) { 3496 mEffectChains[0]->setVolume_l(&vl, &vr); 3497 left = (float)vl / (1 << 24); 3498 right = (float)vr / (1 << 24); 3499 } 3500 if (mOutput->stream->set_volume) { 3501 mOutput->stream->set_volume(mOutput->stream, left, right); 3502 } 3503 } 3504 } 3505} 3506 3507 3508AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3509 Vector< sp<Track> > *tracksToRemove 3510) 3511{ 3512 size_t count = mActiveTracks.size(); 3513 mixer_state mixerStatus = MIXER_IDLE; 3514 3515 // find out which tracks need to be processed 3516 for (size_t i = 0; i < count; i++) { 3517 sp<Track> t = mActiveTracks[i].promote(); 3518 // The track died recently 3519 if (t == 0) { 3520 continue; 3521 } 3522 3523 Track* const track = t.get(); 3524 audio_track_cblk_t* cblk = track->cblk(); 3525 3526 // The first time a track is added we wait 3527 // for all its buffers to be filled before processing it 3528 uint32_t minFrames; 3529 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3530 minFrames = mNormalFrameCount; 3531 } else { 3532 minFrames = 1; 3533 } 3534 // Only consider last track started for volume and mixer state control. 3535 // This is the last entry in mActiveTracks unless a track underruns. 3536 // As we only care about the transition phase between two tracks on a 3537 // direct output, it is not a problem to ignore the underrun case. 3538 bool last = (i == (count - 1)); 3539 3540 if ((track->framesReady() >= minFrames) && track->isReady() && 3541 !track->isPaused() && !track->isTerminated()) 3542 { 3543 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3544 3545 if (track->mFillingUpStatus == Track::FS_FILLED) { 3546 track->mFillingUpStatus = Track::FS_ACTIVE; 3547 // make sure processVolume_l() will apply new volume even if 0 3548 mLeftVolFloat = mRightVolFloat = -1.0; 3549 if (track->mState == TrackBase::RESUMING) { 3550 track->mState = TrackBase::ACTIVE; 3551 } 3552 } 3553 3554 // compute volume for this track 3555 processVolume_l(track, last); 3556 if (last) { 3557 // reset retry count 3558 track->mRetryCount = kMaxTrackRetriesDirect; 3559 mActiveTrack = t; 3560 mixerStatus = MIXER_TRACKS_READY; 3561 } 3562 } else { 3563 // clear effect chain input buffer if the last active track started underruns 3564 // to avoid sending previous audio buffer again to effects 3565 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3566 mEffectChains[0]->clearInputBuffer(); 3567 } 3568 3569 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3570 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3571 track->isStopped() || track->isPaused()) { 3572 // We have consumed all the buffers of this track. 3573 // Remove it from the list of active tracks. 3574 // TODO: implement behavior for compressed audio 3575 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3576 size_t framesWritten = mBytesWritten / mFrameSize; 3577 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3578 if (track->isStopped()) { 3579 track->reset(); 3580 } 3581 tracksToRemove->add(track); 3582 } 3583 } else { 3584 // No buffers for this track. Give it a few chances to 3585 // fill a buffer, then remove it from active list. 3586 // Only consider last track started for mixer state control 3587 if (--(track->mRetryCount) <= 0) { 3588 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3589 tracksToRemove->add(track); 3590 } else if (last) { 3591 mixerStatus = MIXER_TRACKS_ENABLED; 3592 } 3593 } 3594 } 3595 } 3596 3597 // remove all the tracks that need to be... 3598 removeTracks_l(*tracksToRemove); 3599 3600 return mixerStatus; 3601} 3602 3603void AudioFlinger::DirectOutputThread::threadLoop_mix() 3604{ 3605 size_t frameCount = mFrameCount; 3606 int8_t *curBuf = (int8_t *)mMixBuffer; 3607 // output audio to hardware 3608 while (frameCount) { 3609 AudioBufferProvider::Buffer buffer; 3610 buffer.frameCount = frameCount; 3611 mActiveTrack->getNextBuffer(&buffer); 3612 if (buffer.raw == NULL) { 3613 memset(curBuf, 0, frameCount * mFrameSize); 3614 break; 3615 } 3616 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3617 frameCount -= buffer.frameCount; 3618 curBuf += buffer.frameCount * mFrameSize; 3619 mActiveTrack->releaseBuffer(&buffer); 3620 } 3621 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3622 sleepTime = 0; 3623 standbyTime = systemTime() + standbyDelay; 3624 mActiveTrack.clear(); 3625} 3626 3627void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3628{ 3629 if (sleepTime == 0) { 3630 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3631 sleepTime = activeSleepTime; 3632 } else { 3633 sleepTime = idleSleepTime; 3634 } 3635 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3636 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3637 sleepTime = 0; 3638 } 3639} 3640 3641// getTrackName_l() must be called with ThreadBase::mLock held 3642int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3643 int sessionId) 3644{ 3645 return 0; 3646} 3647 3648// deleteTrackName_l() must be called with ThreadBase::mLock held 3649void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3650{ 3651} 3652 3653// checkForNewParameters_l() must be called with ThreadBase::mLock held 3654bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3655{ 3656 bool reconfig = false; 3657 3658 while (!mNewParameters.isEmpty()) { 3659 status_t status = NO_ERROR; 3660 String8 keyValuePair = mNewParameters[0]; 3661 AudioParameter param = AudioParameter(keyValuePair); 3662 int value; 3663 3664 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3665 // do not accept frame count changes if tracks are open as the track buffer 3666 // size depends on frame count and correct behavior would not be garantied 3667 // if frame count is changed after track creation 3668 if (!mTracks.isEmpty()) { 3669 status = INVALID_OPERATION; 3670 } else { 3671 reconfig = true; 3672 } 3673 } 3674 if (status == NO_ERROR) { 3675 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3676 keyValuePair.string()); 3677 if (!mStandby && status == INVALID_OPERATION) { 3678 mOutput->stream->common.standby(&mOutput->stream->common); 3679 mStandby = true; 3680 mBytesWritten = 0; 3681 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3682 keyValuePair.string()); 3683 } 3684 if (status == NO_ERROR && reconfig) { 3685 readOutputParameters(); 3686 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3687 } 3688 } 3689 3690 mNewParameters.removeAt(0); 3691 3692 mParamStatus = status; 3693 mParamCond.signal(); 3694 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3695 // already timed out waiting for the status and will never signal the condition. 3696 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3697 } 3698 return reconfig; 3699} 3700 3701uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3702{ 3703 uint32_t time; 3704 if (audio_is_linear_pcm(mFormat)) { 3705 time = PlaybackThread::activeSleepTimeUs(); 3706 } else { 3707 time = 10000; 3708 } 3709 return time; 3710} 3711 3712uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3713{ 3714 uint32_t time; 3715 if (audio_is_linear_pcm(mFormat)) { 3716 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3717 } else { 3718 time = 10000; 3719 } 3720 return time; 3721} 3722 3723uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3724{ 3725 uint32_t time; 3726 if (audio_is_linear_pcm(mFormat)) { 3727 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3728 } else { 3729 time = 10000; 3730 } 3731 return time; 3732} 3733 3734void AudioFlinger::DirectOutputThread::cacheParameters_l() 3735{ 3736 PlaybackThread::cacheParameters_l(); 3737 3738 // use shorter standby delay as on normal output to release 3739 // hardware resources as soon as possible 3740 if (audio_is_linear_pcm(mFormat)) { 3741 standbyDelay = microseconds(activeSleepTime*2); 3742 } else { 3743 standbyDelay = kOffloadStandbyDelayNs; 3744 } 3745} 3746 3747// ---------------------------------------------------------------------------- 3748 3749AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3750 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3751 : Thread(false /*canCallJava*/), 3752 mPlaybackThread(playbackThread), 3753 mWriteAckSequence(0), 3754 mDrainSequence(0) 3755{ 3756} 3757 3758AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3759{ 3760} 3761 3762void AudioFlinger::AsyncCallbackThread::onFirstRef() 3763{ 3764 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3765} 3766 3767bool AudioFlinger::AsyncCallbackThread::threadLoop() 3768{ 3769 while (!exitPending()) { 3770 uint32_t writeAckSequence; 3771 uint32_t drainSequence; 3772 3773 { 3774 Mutex::Autolock _l(mLock); 3775 mWaitWorkCV.wait(mLock); 3776 if (exitPending()) { 3777 break; 3778 } 3779 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3780 mWriteAckSequence, mDrainSequence); 3781 writeAckSequence = mWriteAckSequence; 3782 mWriteAckSequence &= ~1; 3783 drainSequence = mDrainSequence; 3784 mDrainSequence &= ~1; 3785 } 3786 { 3787 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3788 if (playbackThread != 0) { 3789 if (writeAckSequence & 1) { 3790 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3791 } 3792 if (drainSequence & 1) { 3793 playbackThread->resetDraining(drainSequence >> 1); 3794 } 3795 } 3796 } 3797 } 3798 return false; 3799} 3800 3801void AudioFlinger::AsyncCallbackThread::exit() 3802{ 3803 ALOGV("AsyncCallbackThread::exit"); 3804 Mutex::Autolock _l(mLock); 3805 requestExit(); 3806 mWaitWorkCV.broadcast(); 3807} 3808 3809void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3810{ 3811 Mutex::Autolock _l(mLock); 3812 // bit 0 is cleared 3813 mWriteAckSequence = sequence << 1; 3814} 3815 3816void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3817{ 3818 Mutex::Autolock _l(mLock); 3819 // ignore unexpected callbacks 3820 if (mWriteAckSequence & 2) { 3821 mWriteAckSequence |= 1; 3822 mWaitWorkCV.signal(); 3823 } 3824} 3825 3826void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3827{ 3828 Mutex::Autolock _l(mLock); 3829 // bit 0 is cleared 3830 mDrainSequence = sequence << 1; 3831} 3832 3833void AudioFlinger::AsyncCallbackThread::resetDraining() 3834{ 3835 Mutex::Autolock _l(mLock); 3836 // ignore unexpected callbacks 3837 if (mDrainSequence & 2) { 3838 mDrainSequence |= 1; 3839 mWaitWorkCV.signal(); 3840 } 3841} 3842 3843 3844// ---------------------------------------------------------------------------- 3845AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3846 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3847 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3848 mHwPaused(false), 3849 mPausedBytesRemaining(0) 3850{ 3851} 3852 3853AudioFlinger::OffloadThread::~OffloadThread() 3854{ 3855 mPreviousTrack.clear(); 3856} 3857 3858void AudioFlinger::OffloadThread::threadLoop_exit() 3859{ 3860 if (mFlushPending || mHwPaused) { 3861 // If a flush is pending or track was paused, just discard buffered data 3862 flushHw_l(); 3863 } else { 3864 mMixerStatus = MIXER_DRAIN_ALL; 3865 threadLoop_drain(); 3866 } 3867 mCallbackThread->exit(); 3868 PlaybackThread::threadLoop_exit(); 3869} 3870 3871AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3872 Vector< sp<Track> > *tracksToRemove 3873) 3874{ 3875 size_t count = mActiveTracks.size(); 3876 3877 mixer_state mixerStatus = MIXER_IDLE; 3878 bool doHwPause = false; 3879 bool doHwResume = false; 3880 3881 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3882 3883 // find out which tracks need to be processed 3884 for (size_t i = 0; i < count; i++) { 3885 sp<Track> t = mActiveTracks[i].promote(); 3886 // The track died recently 3887 if (t == 0) { 3888 continue; 3889 } 3890 Track* const track = t.get(); 3891 audio_track_cblk_t* cblk = track->cblk(); 3892 if (mPreviousTrack != NULL) { 3893 if (t != mPreviousTrack) { 3894 // Flush any data still being written from last track 3895 mBytesRemaining = 0; 3896 if (mPausedBytesRemaining) { 3897 // Last track was paused so we also need to flush saved 3898 // mixbuffer state and invalidate track so that it will 3899 // re-submit that unwritten data when it is next resumed 3900 mPausedBytesRemaining = 0; 3901 // Invalidate is a bit drastic - would be more efficient 3902 // to have a flag to tell client that some of the 3903 // previously written data was lost 3904 mPreviousTrack->invalidate(); 3905 } 3906 } 3907 } 3908 mPreviousTrack = t; 3909 bool last = (i == (count - 1)); 3910 if (track->isPausing()) { 3911 track->setPaused(); 3912 if (last) { 3913 if (!mHwPaused) { 3914 doHwPause = true; 3915 mHwPaused = true; 3916 } 3917 // If we were part way through writing the mixbuffer to 3918 // the HAL we must save this until we resume 3919 // BUG - this will be wrong if a different track is made active, 3920 // in that case we want to discard the pending data in the 3921 // mixbuffer and tell the client to present it again when the 3922 // track is resumed 3923 mPausedWriteLength = mCurrentWriteLength; 3924 mPausedBytesRemaining = mBytesRemaining; 3925 mBytesRemaining = 0; // stop writing 3926 } 3927 tracksToRemove->add(track); 3928 } else if (track->framesReady() && track->isReady() && 3929 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3930 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3931 if (track->mFillingUpStatus == Track::FS_FILLED) { 3932 track->mFillingUpStatus = Track::FS_ACTIVE; 3933 // make sure processVolume_l() will apply new volume even if 0 3934 mLeftVolFloat = mRightVolFloat = -1.0; 3935 if (track->mState == TrackBase::RESUMING) { 3936 track->mState = TrackBase::ACTIVE; 3937 if (last) { 3938 if (mPausedBytesRemaining) { 3939 // Need to continue write that was interrupted 3940 mCurrentWriteLength = mPausedWriteLength; 3941 mBytesRemaining = mPausedBytesRemaining; 3942 mPausedBytesRemaining = 0; 3943 } 3944 if (mHwPaused) { 3945 doHwResume = true; 3946 mHwPaused = false; 3947 // threadLoop_mix() will handle the case that we need to 3948 // resume an interrupted write 3949 } 3950 // enable write to audio HAL 3951 sleepTime = 0; 3952 } 3953 } 3954 } 3955 3956 if (last) { 3957 // reset retry count 3958 track->mRetryCount = kMaxTrackRetriesOffload; 3959 mActiveTrack = t; 3960 mixerStatus = MIXER_TRACKS_READY; 3961 } 3962 } else { 3963 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3964 if (track->isStopping_1()) { 3965 // Hardware buffer can hold a large amount of audio so we must 3966 // wait for all current track's data to drain before we say 3967 // that the track is stopped. 3968 if (mBytesRemaining == 0) { 3969 // Only start draining when all data in mixbuffer 3970 // has been written 3971 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3972 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3973 if (last) { 3974 sleepTime = 0; 3975 standbyTime = systemTime() + standbyDelay; 3976 mixerStatus = MIXER_DRAIN_TRACK; 3977 mDrainSequence += 2; 3978 if (mHwPaused) { 3979 // It is possible to move from PAUSED to STOPPING_1 without 3980 // a resume so we must ensure hardware is running 3981 mOutput->stream->resume(mOutput->stream); 3982 mHwPaused = false; 3983 } 3984 } 3985 } 3986 } else if (track->isStopping_2()) { 3987 // Drain has completed, signal presentation complete 3988 if (!(mDrainSequence & 1) || !last) { 3989 track->mState = TrackBase::STOPPED; 3990 size_t audioHALFrames = 3991 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3992 size_t framesWritten = 3993 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3994 track->presentationComplete(framesWritten, audioHALFrames); 3995 track->reset(); 3996 tracksToRemove->add(track); 3997 } 3998 } else { 3999 // No buffers for this track. Give it a few chances to 4000 // fill a buffer, then remove it from active list. 4001 if (--(track->mRetryCount) <= 0) { 4002 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4003 track->name()); 4004 tracksToRemove->add(track); 4005 } else if (last){ 4006 mixerStatus = MIXER_TRACKS_ENABLED; 4007 } 4008 } 4009 } 4010 // compute volume for this track 4011 processVolume_l(track, last); 4012 } 4013 4014 // make sure the pause/flush/resume sequence is executed in the right order 4015 if (doHwPause) { 4016 mOutput->stream->pause(mOutput->stream); 4017 } 4018 if (mFlushPending) { 4019 flushHw_l(); 4020 mFlushPending = false; 4021 } 4022 if (doHwResume) { 4023 mOutput->stream->resume(mOutput->stream); 4024 } 4025 4026 // remove all the tracks that need to be... 4027 removeTracks_l(*tracksToRemove); 4028 4029 return mixerStatus; 4030} 4031 4032void AudioFlinger::OffloadThread::flushOutput_l() 4033{ 4034 mFlushPending = true; 4035} 4036 4037// must be called with thread mutex locked 4038bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4039{ 4040 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4041 mWriteAckSequence, mDrainSequence); 4042 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4043 return true; 4044 } 4045 return false; 4046} 4047 4048// must be called with thread mutex locked 4049bool AudioFlinger::OffloadThread::shouldStandby_l() 4050{ 4051 bool TrackPaused = false; 4052 4053 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4054 // after a timeout and we will enter standby then. 4055 if (mTracks.size() > 0) { 4056 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4057 } 4058 4059 return !mStandby && !TrackPaused; 4060} 4061 4062 4063bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4064{ 4065 Mutex::Autolock _l(mLock); 4066 return waitingAsyncCallback_l(); 4067} 4068 4069void AudioFlinger::OffloadThread::flushHw_l() 4070{ 4071 mOutput->stream->flush(mOutput->stream); 4072 // Flush anything still waiting in the mixbuffer 4073 mCurrentWriteLength = 0; 4074 mBytesRemaining = 0; 4075 mPausedWriteLength = 0; 4076 mPausedBytesRemaining = 0; 4077 if (mUseAsyncWrite) { 4078 // discard any pending drain or write ack by incrementing sequence 4079 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4080 mDrainSequence = (mDrainSequence + 2) & ~1; 4081 ALOG_ASSERT(mCallbackThread != 0); 4082 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4083 mCallbackThread->setDraining(mDrainSequence); 4084 } 4085} 4086 4087// ---------------------------------------------------------------------------- 4088 4089AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4090 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4091 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4092 DUPLICATING), 4093 mWaitTimeMs(UINT_MAX) 4094{ 4095 addOutputTrack(mainThread); 4096} 4097 4098AudioFlinger::DuplicatingThread::~DuplicatingThread() 4099{ 4100 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4101 mOutputTracks[i]->destroy(); 4102 } 4103} 4104 4105void AudioFlinger::DuplicatingThread::threadLoop_mix() 4106{ 4107 // mix buffers... 4108 if (outputsReady(outputTracks)) { 4109 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4110 } else { 4111 memset(mMixBuffer, 0, mixBufferSize); 4112 } 4113 sleepTime = 0; 4114 writeFrames = mNormalFrameCount; 4115 mCurrentWriteLength = mixBufferSize; 4116 standbyTime = systemTime() + standbyDelay; 4117} 4118 4119void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4120{ 4121 if (sleepTime == 0) { 4122 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4123 sleepTime = activeSleepTime; 4124 } else { 4125 sleepTime = idleSleepTime; 4126 } 4127 } else if (mBytesWritten != 0) { 4128 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4129 writeFrames = mNormalFrameCount; 4130 memset(mMixBuffer, 0, mixBufferSize); 4131 } else { 4132 // flush remaining overflow buffers in output tracks 4133 writeFrames = 0; 4134 } 4135 sleepTime = 0; 4136 } 4137} 4138 4139ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4140{ 4141 for (size_t i = 0; i < outputTracks.size(); i++) { 4142 outputTracks[i]->write(mMixBuffer, writeFrames); 4143 } 4144 return (ssize_t)mixBufferSize; 4145} 4146 4147void AudioFlinger::DuplicatingThread::threadLoop_standby() 4148{ 4149 // DuplicatingThread implements standby by stopping all tracks 4150 for (size_t i = 0; i < outputTracks.size(); i++) { 4151 outputTracks[i]->stop(); 4152 } 4153} 4154 4155void AudioFlinger::DuplicatingThread::saveOutputTracks() 4156{ 4157 outputTracks = mOutputTracks; 4158} 4159 4160void AudioFlinger::DuplicatingThread::clearOutputTracks() 4161{ 4162 outputTracks.clear(); 4163} 4164 4165void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4166{ 4167 Mutex::Autolock _l(mLock); 4168 // FIXME explain this formula 4169 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4170 OutputTrack *outputTrack = new OutputTrack(thread, 4171 this, 4172 mSampleRate, 4173 mFormat, 4174 mChannelMask, 4175 frameCount); 4176 if (outputTrack->cblk() != NULL) { 4177 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4178 mOutputTracks.add(outputTrack); 4179 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4180 updateWaitTime_l(); 4181 } 4182} 4183 4184void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4185{ 4186 Mutex::Autolock _l(mLock); 4187 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4188 if (mOutputTracks[i]->thread() == thread) { 4189 mOutputTracks[i]->destroy(); 4190 mOutputTracks.removeAt(i); 4191 updateWaitTime_l(); 4192 return; 4193 } 4194 } 4195 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4196} 4197 4198// caller must hold mLock 4199void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4200{ 4201 mWaitTimeMs = UINT_MAX; 4202 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4203 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4204 if (strong != 0) { 4205 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4206 if (waitTimeMs < mWaitTimeMs) { 4207 mWaitTimeMs = waitTimeMs; 4208 } 4209 } 4210 } 4211} 4212 4213 4214bool AudioFlinger::DuplicatingThread::outputsReady( 4215 const SortedVector< sp<OutputTrack> > &outputTracks) 4216{ 4217 for (size_t i = 0; i < outputTracks.size(); i++) { 4218 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4219 if (thread == 0) { 4220 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4221 outputTracks[i].get()); 4222 return false; 4223 } 4224 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4225 // see note at standby() declaration 4226 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4227 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4228 thread.get()); 4229 return false; 4230 } 4231 } 4232 return true; 4233} 4234 4235uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4236{ 4237 return (mWaitTimeMs * 1000) / 2; 4238} 4239 4240void AudioFlinger::DuplicatingThread::cacheParameters_l() 4241{ 4242 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4243 updateWaitTime_l(); 4244 4245 MixerThread::cacheParameters_l(); 4246} 4247 4248// ---------------------------------------------------------------------------- 4249// Record 4250// ---------------------------------------------------------------------------- 4251 4252AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4253 AudioStreamIn *input, 4254 uint32_t sampleRate, 4255 audio_channel_mask_t channelMask, 4256 audio_io_handle_t id, 4257 audio_devices_t outDevice, 4258 audio_devices_t inDevice 4259#ifdef TEE_SINK 4260 , const sp<NBAIO_Sink>& teeSink 4261#endif 4262 ) : 4263 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4264 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4265 // mRsmpInIndex and mBufferSize set by readInputParameters() 4266 mReqChannelCount(popcount(channelMask)), 4267 mReqSampleRate(sampleRate) 4268 // mBytesRead is only meaningful while active, and so is cleared in start() 4269 // (but might be better to also clear here for dump?) 4270#ifdef TEE_SINK 4271 , mTeeSink(teeSink) 4272#endif 4273{ 4274 snprintf(mName, kNameLength, "AudioIn_%X", id); 4275 4276 readInputParameters(); 4277 4278} 4279 4280 4281AudioFlinger::RecordThread::~RecordThread() 4282{ 4283 delete[] mRsmpInBuffer; 4284 delete mResampler; 4285 delete[] mRsmpOutBuffer; 4286} 4287 4288void AudioFlinger::RecordThread::onFirstRef() 4289{ 4290 run(mName, PRIORITY_URGENT_AUDIO); 4291} 4292 4293status_t AudioFlinger::RecordThread::readyToRun() 4294{ 4295 status_t status = initCheck(); 4296 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4297 return status; 4298} 4299 4300bool AudioFlinger::RecordThread::threadLoop() 4301{ 4302 AudioBufferProvider::Buffer buffer; 4303 sp<RecordTrack> activeTrack; 4304 Vector< sp<EffectChain> > effectChains; 4305 4306 nsecs_t lastWarning = 0; 4307 4308 inputStandBy(); 4309 acquireWakeLock(); 4310 4311 // used to verify we've read at least once before evaluating how many bytes were read 4312 bool readOnce = false; 4313 4314 // start recording 4315 while (!exitPending()) { 4316 4317 processConfigEvents(); 4318 4319 { // scope for mLock 4320 Mutex::Autolock _l(mLock); 4321 checkForNewParameters_l(); 4322 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4323 standby(); 4324 4325 if (exitPending()) { 4326 break; 4327 } 4328 4329 releaseWakeLock_l(); 4330 ALOGV("RecordThread: loop stopping"); 4331 // go to sleep 4332 mWaitWorkCV.wait(mLock); 4333 ALOGV("RecordThread: loop starting"); 4334 acquireWakeLock_l(); 4335 continue; 4336 } 4337 if (mActiveTrack != 0) { 4338 if (mActiveTrack->isTerminated()) { 4339 removeTrack_l(mActiveTrack); 4340 mActiveTrack.clear(); 4341 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4342 standby(); 4343 mActiveTrack.clear(); 4344 mStartStopCond.broadcast(); 4345 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4346 if (mReqChannelCount != mActiveTrack->channelCount()) { 4347 mActiveTrack.clear(); 4348 mStartStopCond.broadcast(); 4349 } else if (readOnce) { 4350 // record start succeeds only if first read from audio input 4351 // succeeds 4352 if (mBytesRead >= 0) { 4353 mActiveTrack->mState = TrackBase::ACTIVE; 4354 } else { 4355 mActiveTrack.clear(); 4356 } 4357 mStartStopCond.broadcast(); 4358 } 4359 mStandby = false; 4360 } 4361 } 4362 4363 lockEffectChains_l(effectChains); 4364 } 4365 4366 if (mActiveTrack != 0) { 4367 if (mActiveTrack->mState != TrackBase::ACTIVE && 4368 mActiveTrack->mState != TrackBase::RESUMING) { 4369 unlockEffectChains(effectChains); 4370 usleep(kRecordThreadSleepUs); 4371 continue; 4372 } 4373 for (size_t i = 0; i < effectChains.size(); i ++) { 4374 effectChains[i]->process_l(); 4375 } 4376 4377 buffer.frameCount = mFrameCount; 4378 status_t status = mActiveTrack->getNextBuffer(&buffer); 4379 if (status == NO_ERROR) { 4380 readOnce = true; 4381 size_t framesOut = buffer.frameCount; 4382 if (mResampler == NULL) { 4383 // no resampling 4384 while (framesOut) { 4385 size_t framesIn = mFrameCount - mRsmpInIndex; 4386 if (framesIn) { 4387 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4388 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4389 mActiveTrack->mFrameSize; 4390 if (framesIn > framesOut) 4391 framesIn = framesOut; 4392 mRsmpInIndex += framesIn; 4393 framesOut -= framesIn; 4394 if (mChannelCount == mReqChannelCount) { 4395 memcpy(dst, src, framesIn * mFrameSize); 4396 } else { 4397 if (mChannelCount == 1) { 4398 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4399 (int16_t *)src, framesIn); 4400 } else { 4401 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4402 (int16_t *)src, framesIn); 4403 } 4404 } 4405 } 4406 if (framesOut && mFrameCount == mRsmpInIndex) { 4407 void *readInto; 4408 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4409 readInto = buffer.raw; 4410 framesOut = 0; 4411 } else { 4412 readInto = mRsmpInBuffer; 4413 mRsmpInIndex = 0; 4414 } 4415 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4416 mBufferSize); 4417 if (mBytesRead <= 0) { 4418 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4419 { 4420 ALOGE("Error reading audio input"); 4421 // Force input into standby so that it tries to 4422 // recover at next read attempt 4423 inputStandBy(); 4424 usleep(kRecordThreadSleepUs); 4425 } 4426 mRsmpInIndex = mFrameCount; 4427 framesOut = 0; 4428 buffer.frameCount = 0; 4429 } 4430#ifdef TEE_SINK 4431 else if (mTeeSink != 0) { 4432 (void) mTeeSink->write(readInto, 4433 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4434 } 4435#endif 4436 } 4437 } 4438 } else { 4439 // resampling 4440 4441 // resampler accumulates, but we only have one source track 4442 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4443 // alter output frame count as if we were expecting stereo samples 4444 if (mChannelCount == 1 && mReqChannelCount == 1) { 4445 framesOut >>= 1; 4446 } 4447 mResampler->resample(mRsmpOutBuffer, framesOut, 4448 this /* AudioBufferProvider* */); 4449 // ditherAndClamp() works as long as all buffers returned by 4450 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4451 if (mChannelCount == 2 && mReqChannelCount == 1) { 4452 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4453 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4454 // the resampler always outputs stereo samples: 4455 // do post stereo to mono conversion 4456 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4457 framesOut); 4458 } else { 4459 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4460 } 4461 // now done with mRsmpOutBuffer 4462 4463 } 4464 if (mFramestoDrop == 0) { 4465 mActiveTrack->releaseBuffer(&buffer); 4466 } else { 4467 if (mFramestoDrop > 0) { 4468 mFramestoDrop -= buffer.frameCount; 4469 if (mFramestoDrop <= 0) { 4470 clearSyncStartEvent(); 4471 } 4472 } else { 4473 mFramestoDrop += buffer.frameCount; 4474 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4475 mSyncStartEvent->isCancelled()) { 4476 ALOGW("Synced record %s, session %d, trigger session %d", 4477 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4478 mActiveTrack->sessionId(), 4479 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4480 clearSyncStartEvent(); 4481 } 4482 } 4483 } 4484 mActiveTrack->clearOverflow(); 4485 } 4486 // client isn't retrieving buffers fast enough 4487 else { 4488 if (!mActiveTrack->setOverflow()) { 4489 nsecs_t now = systemTime(); 4490 if ((now - lastWarning) > kWarningThrottleNs) { 4491 ALOGW("RecordThread: buffer overflow"); 4492 lastWarning = now; 4493 } 4494 } 4495 // Release the processor for a while before asking for a new buffer. 4496 // This will give the application more chance to read from the buffer and 4497 // clear the overflow. 4498 usleep(kRecordThreadSleepUs); 4499 } 4500 } 4501 // enable changes in effect chain 4502 unlockEffectChains(effectChains); 4503 effectChains.clear(); 4504 } 4505 4506 standby(); 4507 4508 { 4509 Mutex::Autolock _l(mLock); 4510 for (size_t i = 0; i < mTracks.size(); i++) { 4511 sp<RecordTrack> track = mTracks[i]; 4512 track->invalidate(); 4513 } 4514 mActiveTrack.clear(); 4515 mStartStopCond.broadcast(); 4516 } 4517 4518 releaseWakeLock(); 4519 4520 ALOGV("RecordThread %p exiting", this); 4521 return false; 4522} 4523 4524void AudioFlinger::RecordThread::standby() 4525{ 4526 if (!mStandby) { 4527 inputStandBy(); 4528 mStandby = true; 4529 } 4530} 4531 4532void AudioFlinger::RecordThread::inputStandBy() 4533{ 4534 mInput->stream->common.standby(&mInput->stream->common); 4535} 4536 4537sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4538 const sp<AudioFlinger::Client>& client, 4539 uint32_t sampleRate, 4540 audio_format_t format, 4541 audio_channel_mask_t channelMask, 4542 size_t frameCount, 4543 int sessionId, 4544 IAudioFlinger::track_flags_t *flags, 4545 pid_t tid, 4546 status_t *status) 4547{ 4548 sp<RecordTrack> track; 4549 status_t lStatus; 4550 4551 lStatus = initCheck(); 4552 if (lStatus != NO_ERROR) { 4553 ALOGE("Audio driver not initialized."); 4554 goto Exit; 4555 } 4556 4557 // client expresses a preference for FAST, but we get the final say 4558 if (*flags & IAudioFlinger::TRACK_FAST) { 4559 if ( 4560 // use case: callback handler and frame count is default or at least as large as HAL 4561 ( 4562 (tid != -1) && 4563 ((frameCount == 0) || 4564 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4565 ) && 4566 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4567 // mono or stereo 4568 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4569 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4570 // hardware sample rate 4571 (sampleRate == mSampleRate) && 4572 // record thread has an associated fast recorder 4573 hasFastRecorder() 4574 // FIXME test that RecordThread for this fast track has a capable output HAL 4575 // FIXME add a permission test also? 4576 ) { 4577 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4578 if (frameCount == 0) { 4579 frameCount = mFrameCount * kFastTrackMultiplier; 4580 } 4581 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4582 frameCount, mFrameCount); 4583 } else { 4584 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4585 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4586 "hasFastRecorder=%d tid=%d", 4587 frameCount, mFrameCount, format, 4588 audio_is_linear_pcm(format), 4589 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4590 *flags &= ~IAudioFlinger::TRACK_FAST; 4591 // For compatibility with AudioRecord calculation, buffer depth is forced 4592 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4593 // This is probably too conservative, but legacy application code may depend on it. 4594 // If you change this calculation, also review the start threshold which is related. 4595 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4596 size_t mNormalFrameCount = 2048; // FIXME 4597 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4598 if (minBufCount < 2) { 4599 minBufCount = 2; 4600 } 4601 size_t minFrameCount = mNormalFrameCount * minBufCount; 4602 if (frameCount < minFrameCount) { 4603 frameCount = minFrameCount; 4604 } 4605 } 4606 } 4607 4608 // FIXME use flags and tid similar to createTrack_l() 4609 4610 { // scope for mLock 4611 Mutex::Autolock _l(mLock); 4612 4613 track = new RecordTrack(this, client, sampleRate, 4614 format, channelMask, frameCount, sessionId); 4615 4616 if (track->getCblk() == 0) { 4617 lStatus = NO_MEMORY; 4618 goto Exit; 4619 } 4620 mTracks.add(track); 4621 4622 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4623 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4624 mAudioFlinger->btNrecIsOff(); 4625 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4626 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4627 4628 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4629 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4630 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4631 // so ask activity manager to do this on our behalf 4632 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4633 } 4634 } 4635 lStatus = NO_ERROR; 4636 4637Exit: 4638 if (status) { 4639 *status = lStatus; 4640 } 4641 return track; 4642} 4643 4644status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4645 AudioSystem::sync_event_t event, 4646 int triggerSession) 4647{ 4648 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4649 sp<ThreadBase> strongMe = this; 4650 status_t status = NO_ERROR; 4651 4652 if (event == AudioSystem::SYNC_EVENT_NONE) { 4653 clearSyncStartEvent(); 4654 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4655 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4656 triggerSession, 4657 recordTrack->sessionId(), 4658 syncStartEventCallback, 4659 this); 4660 // Sync event can be cancelled by the trigger session if the track is not in a 4661 // compatible state in which case we start record immediately 4662 if (mSyncStartEvent->isCancelled()) { 4663 clearSyncStartEvent(); 4664 } else { 4665 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4666 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4667 } 4668 } 4669 4670 { 4671 AutoMutex lock(mLock); 4672 if (mActiveTrack != 0) { 4673 if (recordTrack != mActiveTrack.get()) { 4674 status = -EBUSY; 4675 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4676 mActiveTrack->mState = TrackBase::ACTIVE; 4677 } 4678 return status; 4679 } 4680 4681 recordTrack->mState = TrackBase::IDLE; 4682 mActiveTrack = recordTrack; 4683 mLock.unlock(); 4684 status_t status = AudioSystem::startInput(mId); 4685 mLock.lock(); 4686 if (status != NO_ERROR) { 4687 mActiveTrack.clear(); 4688 clearSyncStartEvent(); 4689 return status; 4690 } 4691 mRsmpInIndex = mFrameCount; 4692 mBytesRead = 0; 4693 if (mResampler != NULL) { 4694 mResampler->reset(); 4695 } 4696 mActiveTrack->mState = TrackBase::RESUMING; 4697 // signal thread to start 4698 ALOGV("Signal record thread"); 4699 mWaitWorkCV.broadcast(); 4700 // do not wait for mStartStopCond if exiting 4701 if (exitPending()) { 4702 mActiveTrack.clear(); 4703 status = INVALID_OPERATION; 4704 goto startError; 4705 } 4706 mStartStopCond.wait(mLock); 4707 if (mActiveTrack == 0) { 4708 ALOGV("Record failed to start"); 4709 status = BAD_VALUE; 4710 goto startError; 4711 } 4712 ALOGV("Record started OK"); 4713 return status; 4714 } 4715 4716startError: 4717 AudioSystem::stopInput(mId); 4718 clearSyncStartEvent(); 4719 return status; 4720} 4721 4722void AudioFlinger::RecordThread::clearSyncStartEvent() 4723{ 4724 if (mSyncStartEvent != 0) { 4725 mSyncStartEvent->cancel(); 4726 } 4727 mSyncStartEvent.clear(); 4728 mFramestoDrop = 0; 4729} 4730 4731void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4732{ 4733 sp<SyncEvent> strongEvent = event.promote(); 4734 4735 if (strongEvent != 0) { 4736 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4737 me->handleSyncStartEvent(strongEvent); 4738 } 4739} 4740 4741void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4742{ 4743 if (event == mSyncStartEvent) { 4744 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4745 // from audio HAL 4746 mFramestoDrop = mFrameCount * 2; 4747 } 4748} 4749 4750bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4751 ALOGV("RecordThread::stop"); 4752 AutoMutex _l(mLock); 4753 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4754 return false; 4755 } 4756 recordTrack->mState = TrackBase::PAUSING; 4757 // do not wait for mStartStopCond if exiting 4758 if (exitPending()) { 4759 return true; 4760 } 4761 mStartStopCond.wait(mLock); 4762 // if we have been restarted, recordTrack == mActiveTrack.get() here 4763 if (exitPending() || recordTrack != mActiveTrack.get()) { 4764 ALOGV("Record stopped OK"); 4765 return true; 4766 } 4767 return false; 4768} 4769 4770bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4771{ 4772 return false; 4773} 4774 4775status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4776{ 4777#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4778 if (!isValidSyncEvent(event)) { 4779 return BAD_VALUE; 4780 } 4781 4782 int eventSession = event->triggerSession(); 4783 status_t ret = NAME_NOT_FOUND; 4784 4785 Mutex::Autolock _l(mLock); 4786 4787 for (size_t i = 0; i < mTracks.size(); i++) { 4788 sp<RecordTrack> track = mTracks[i]; 4789 if (eventSession == track->sessionId()) { 4790 (void) track->setSyncEvent(event); 4791 ret = NO_ERROR; 4792 } 4793 } 4794 return ret; 4795#else 4796 return BAD_VALUE; 4797#endif 4798} 4799 4800// destroyTrack_l() must be called with ThreadBase::mLock held 4801void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4802{ 4803 track->terminate(); 4804 track->mState = TrackBase::STOPPED; 4805 // active tracks are removed by threadLoop() 4806 if (mActiveTrack != track) { 4807 removeTrack_l(track); 4808 } 4809} 4810 4811void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4812{ 4813 mTracks.remove(track); 4814 // need anything related to effects here? 4815} 4816 4817void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4818{ 4819 dumpInternals(fd, args); 4820 dumpTracks(fd, args); 4821 dumpEffectChains(fd, args); 4822} 4823 4824void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4825{ 4826 const size_t SIZE = 256; 4827 char buffer[SIZE]; 4828 String8 result; 4829 4830 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4831 result.append(buffer); 4832 4833 if (mActiveTrack != 0) { 4834 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4835 result.append(buffer); 4836 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4837 result.append(buffer); 4838 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4839 result.append(buffer); 4840 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4841 result.append(buffer); 4842 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4843 result.append(buffer); 4844 } else { 4845 result.append("No active record client\n"); 4846 } 4847 4848 write(fd, result.string(), result.size()); 4849 4850 dumpBase(fd, args); 4851} 4852 4853void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4854{ 4855 const size_t SIZE = 256; 4856 char buffer[SIZE]; 4857 String8 result; 4858 4859 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4860 result.append(buffer); 4861 RecordTrack::appendDumpHeader(result); 4862 for (size_t i = 0; i < mTracks.size(); ++i) { 4863 sp<RecordTrack> track = mTracks[i]; 4864 if (track != 0) { 4865 track->dump(buffer, SIZE); 4866 result.append(buffer); 4867 } 4868 } 4869 4870 if (mActiveTrack != 0) { 4871 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4872 result.append(buffer); 4873 RecordTrack::appendDumpHeader(result); 4874 mActiveTrack->dump(buffer, SIZE); 4875 result.append(buffer); 4876 4877 } 4878 write(fd, result.string(), result.size()); 4879} 4880 4881// AudioBufferProvider interface 4882status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4883{ 4884 size_t framesReq = buffer->frameCount; 4885 size_t framesReady = mFrameCount - mRsmpInIndex; 4886 int channelCount; 4887 4888 if (framesReady == 0) { 4889 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4890 if (mBytesRead <= 0) { 4891 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4892 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4893 // Force input into standby so that it tries to 4894 // recover at next read attempt 4895 inputStandBy(); 4896 usleep(kRecordThreadSleepUs); 4897 } 4898 buffer->raw = NULL; 4899 buffer->frameCount = 0; 4900 return NOT_ENOUGH_DATA; 4901 } 4902 mRsmpInIndex = 0; 4903 framesReady = mFrameCount; 4904 } 4905 4906 if (framesReq > framesReady) { 4907 framesReq = framesReady; 4908 } 4909 4910 if (mChannelCount == 1 && mReqChannelCount == 2) { 4911 channelCount = 1; 4912 } else { 4913 channelCount = 2; 4914 } 4915 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4916 buffer->frameCount = framesReq; 4917 return NO_ERROR; 4918} 4919 4920// AudioBufferProvider interface 4921void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4922{ 4923 mRsmpInIndex += buffer->frameCount; 4924 buffer->frameCount = 0; 4925} 4926 4927bool AudioFlinger::RecordThread::checkForNewParameters_l() 4928{ 4929 bool reconfig = false; 4930 4931 while (!mNewParameters.isEmpty()) { 4932 status_t status = NO_ERROR; 4933 String8 keyValuePair = mNewParameters[0]; 4934 AudioParameter param = AudioParameter(keyValuePair); 4935 int value; 4936 audio_format_t reqFormat = mFormat; 4937 uint32_t reqSamplingRate = mReqSampleRate; 4938 uint32_t reqChannelCount = mReqChannelCount; 4939 4940 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4941 reqSamplingRate = value; 4942 reconfig = true; 4943 } 4944 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4945 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4946 status = BAD_VALUE; 4947 } else { 4948 reqFormat = (audio_format_t) value; 4949 reconfig = true; 4950 } 4951 } 4952 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4953 reqChannelCount = popcount(value); 4954 reconfig = true; 4955 } 4956 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4957 // do not accept frame count changes if tracks are open as the track buffer 4958 // size depends on frame count and correct behavior would not be guaranteed 4959 // if frame count is changed after track creation 4960 if (mActiveTrack != 0) { 4961 status = INVALID_OPERATION; 4962 } else { 4963 reconfig = true; 4964 } 4965 } 4966 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4967 // forward device change to effects that have requested to be 4968 // aware of attached audio device. 4969 for (size_t i = 0; i < mEffectChains.size(); i++) { 4970 mEffectChains[i]->setDevice_l(value); 4971 } 4972 4973 // store input device and output device but do not forward output device to audio HAL. 4974 // Note that status is ignored by the caller for output device 4975 // (see AudioFlinger::setParameters() 4976 if (audio_is_output_devices(value)) { 4977 mOutDevice = value; 4978 status = BAD_VALUE; 4979 } else { 4980 mInDevice = value; 4981 // disable AEC and NS if the device is a BT SCO headset supporting those 4982 // pre processings 4983 if (mTracks.size() > 0) { 4984 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4985 mAudioFlinger->btNrecIsOff(); 4986 for (size_t i = 0; i < mTracks.size(); i++) { 4987 sp<RecordTrack> track = mTracks[i]; 4988 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4989 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4990 } 4991 } 4992 } 4993 } 4994 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4995 mAudioSource != (audio_source_t)value) { 4996 // forward device change to effects that have requested to be 4997 // aware of attached audio device. 4998 for (size_t i = 0; i < mEffectChains.size(); i++) { 4999 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5000 } 5001 mAudioSource = (audio_source_t)value; 5002 } 5003 if (status == NO_ERROR) { 5004 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5005 keyValuePair.string()); 5006 if (status == INVALID_OPERATION) { 5007 inputStandBy(); 5008 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5009 keyValuePair.string()); 5010 } 5011 if (reconfig) { 5012 if (status == BAD_VALUE && 5013 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5014 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5015 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5016 <= (2 * reqSamplingRate)) && 5017 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5018 <= FCC_2 && 5019 (reqChannelCount <= FCC_2)) { 5020 status = NO_ERROR; 5021 } 5022 if (status == NO_ERROR) { 5023 readInputParameters(); 5024 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5025 } 5026 } 5027 } 5028 5029 mNewParameters.removeAt(0); 5030 5031 mParamStatus = status; 5032 mParamCond.signal(); 5033 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5034 // already timed out waiting for the status and will never signal the condition. 5035 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5036 } 5037 return reconfig; 5038} 5039 5040String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5041{ 5042 Mutex::Autolock _l(mLock); 5043 if (initCheck() != NO_ERROR) { 5044 return String8(); 5045 } 5046 5047 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5048 const String8 out_s8(s); 5049 free(s); 5050 return out_s8; 5051} 5052 5053void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5054 AudioSystem::OutputDescriptor desc; 5055 void *param2 = NULL; 5056 5057 switch (event) { 5058 case AudioSystem::INPUT_OPENED: 5059 case AudioSystem::INPUT_CONFIG_CHANGED: 5060 desc.channelMask = mChannelMask; 5061 desc.samplingRate = mSampleRate; 5062 desc.format = mFormat; 5063 desc.frameCount = mFrameCount; 5064 desc.latency = 0; 5065 param2 = &desc; 5066 break; 5067 5068 case AudioSystem::INPUT_CLOSED: 5069 default: 5070 break; 5071 } 5072 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5073} 5074 5075void AudioFlinger::RecordThread::readInputParameters() 5076{ 5077 delete[] mRsmpInBuffer; 5078 // mRsmpInBuffer is always assigned a new[] below 5079 delete[] mRsmpOutBuffer; 5080 mRsmpOutBuffer = NULL; 5081 delete mResampler; 5082 mResampler = NULL; 5083 5084 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5085 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5086 mChannelCount = popcount(mChannelMask); 5087 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5088 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5089 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5090 } 5091 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5092 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5093 mFrameCount = mBufferSize / mFrameSize; 5094 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5095 5096 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5097 { 5098 int channelCount; 5099 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5100 // stereo to mono post process as the resampler always outputs stereo. 5101 if (mChannelCount == 1 && mReqChannelCount == 2) { 5102 channelCount = 1; 5103 } else { 5104 channelCount = 2; 5105 } 5106 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5107 mResampler->setSampleRate(mSampleRate); 5108 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5109 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5110 5111 // optmization: if mono to mono, alter input frame count as if we were inputing 5112 // stereo samples 5113 if (mChannelCount == 1 && mReqChannelCount == 1) { 5114 mFrameCount >>= 1; 5115 } 5116 5117 } 5118 mRsmpInIndex = mFrameCount; 5119} 5120 5121unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5122{ 5123 Mutex::Autolock _l(mLock); 5124 if (initCheck() != NO_ERROR) { 5125 return 0; 5126 } 5127 5128 return mInput->stream->get_input_frames_lost(mInput->stream); 5129} 5130 5131uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5132{ 5133 Mutex::Autolock _l(mLock); 5134 uint32_t result = 0; 5135 if (getEffectChain_l(sessionId) != 0) { 5136 result = EFFECT_SESSION; 5137 } 5138 5139 for (size_t i = 0; i < mTracks.size(); ++i) { 5140 if (sessionId == mTracks[i]->sessionId()) { 5141 result |= TRACK_SESSION; 5142 break; 5143 } 5144 } 5145 5146 return result; 5147} 5148 5149KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5150{ 5151 KeyedVector<int, bool> ids; 5152 Mutex::Autolock _l(mLock); 5153 for (size_t j = 0; j < mTracks.size(); ++j) { 5154 sp<RecordThread::RecordTrack> track = mTracks[j]; 5155 int sessionId = track->sessionId(); 5156 if (ids.indexOfKey(sessionId) < 0) { 5157 ids.add(sessionId, true); 5158 } 5159 } 5160 return ids; 5161} 5162 5163AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5164{ 5165 Mutex::Autolock _l(mLock); 5166 AudioStreamIn *input = mInput; 5167 mInput = NULL; 5168 return input; 5169} 5170 5171// this method must always be called either with ThreadBase mLock held or inside the thread loop 5172audio_stream_t* AudioFlinger::RecordThread::stream() const 5173{ 5174 if (mInput == NULL) { 5175 return NULL; 5176 } 5177 return &mInput->stream->common; 5178} 5179 5180status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5181{ 5182 // only one chain per input thread 5183 if (mEffectChains.size() != 0) { 5184 return INVALID_OPERATION; 5185 } 5186 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5187 5188 chain->setInBuffer(NULL); 5189 chain->setOutBuffer(NULL); 5190 5191 checkSuspendOnAddEffectChain_l(chain); 5192 5193 mEffectChains.add(chain); 5194 5195 return NO_ERROR; 5196} 5197 5198size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5199{ 5200 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5201 ALOGW_IF(mEffectChains.size() != 1, 5202 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5203 chain.get(), mEffectChains.size(), this); 5204 if (mEffectChains.size() == 1) { 5205 mEffectChains.removeAt(0); 5206 } 5207 return 0; 5208} 5209 5210}; // namespace android 5211