Threads.cpp revision 7f249fa9bcb64da324d19f551943fac7686d221c
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 <cutils/compiler.h> 29#include <media/AudioParameter.h> 30#include <utils/Log.h> 31#include <utils/Trace.h> 32 33#include <private/media/AudioTrackShared.h> 34#include <hardware/audio.h> 35#include <audio_effects/effect_ns.h> 36#include <audio_effects/effect_aec.h> 37#include <audio_utils/primitives.h> 38 39// NBAIO implementations 40#include <media/nbaio/AudioStreamOutSink.h> 41#include <media/nbaio/MonoPipe.h> 42#include <media/nbaio/MonoPipeReader.h> 43#include <media/nbaio/Pipe.h> 44#include <media/nbaio/PipeReader.h> 45#include <media/nbaio/SourceAudioBufferProvider.h> 46 47#include <powermanager/PowerManager.h> 48 49#include <common_time/cc_helper.h> 50#include <common_time/local_clock.h> 51 52#include "AudioFlinger.h" 53#include "AudioMixer.h" 54#include "FastMixer.h" 55#include "ServiceUtilities.h" 56#include "SchedulingPolicyService.h" 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85// retry counts for buffer fill timeout 86// 50 * ~20msecs = 1 second 87static const int8_t kMaxTrackRetries = 50; 88static const int8_t kMaxTrackStartupRetries = 50; 89// allow less retry attempts on direct output thread. 90// direct outputs can be a scarce resource in audio hardware and should 91// be released as quickly as possible. 92static const int8_t kMaxTrackRetriesDirect = 2; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait for setParameters to complete 101static const nsecs_t kSetParametersTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108// minimum normal mix buffer size, expressed in milliseconds rather than frames 109static const uint32_t kMinNormalMixBufferSizeMs = 20; 110// maximum normal mix buffer size 111static const uint32_t kMaxNormalMixBufferSizeMs = 24; 112 113// Whether to use fast mixer 114static const enum { 115 FastMixer_Never, // never initialize or use: for debugging only 116 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 117 // normal mixer multiplier is 1 118 FastMixer_Static, // initialize if needed, then use all the time if initialized, 119 // multiplier is calculated based on min & max normal mixer buffer size 120 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 // FIXME for FastMixer_Dynamic: 123 // Supporting this option will require fixing HALs that can't handle large writes. 124 // For example, one HAL implementation returns an error from a large write, 125 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 126 // We could either fix the HAL implementations, or provide a wrapper that breaks 127 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 128} kUseFastMixer = FastMixer_Static; 129 130// Priorities for requestPriority 131static const int kPriorityAudioApp = 2; 132static const int kPriorityFastMixer = 3; 133 134// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 135// for the track. The client then sub-divides this into smaller buffers for its use. 136// Currently the client uses double-buffering by default, but doesn't tell us about that. 137// So for now we just assume that client is double-buffered. 138// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 139// N-buffering, so AudioFlinger could allocate the right amount of memory. 140// See the client's minBufCount and mNotificationFramesAct calculations for details. 141static const int kFastTrackMultiplier = 1; 142 143// ---------------------------------------------------------------------------- 144 145#ifdef ADD_BATTERY_DATA 146// To collect the amplifier usage 147static void addBatteryData(uint32_t params) { 148 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 149 if (service == NULL) { 150 // it already logged 151 return; 152 } 153 154 service->addBatteryData(params); 155} 156#endif 157 158 159// ---------------------------------------------------------------------------- 160// CPU Stats 161// ---------------------------------------------------------------------------- 162 163class CpuStats { 164public: 165 CpuStats(); 166 void sample(const String8 &title); 167#ifdef DEBUG_CPU_USAGE 168private: 169 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 170 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 171 172 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 173 174 int mCpuNum; // thread's current CPU number 175 int mCpukHz; // frequency of thread's current CPU in kHz 176#endif 177}; 178 179CpuStats::CpuStats() 180#ifdef DEBUG_CPU_USAGE 181 : mCpuNum(-1), mCpukHz(-1) 182#endif 183{ 184} 185 186void CpuStats::sample(const String8 &title) { 187#ifdef DEBUG_CPU_USAGE 188 // get current thread's delta CPU time in wall clock ns 189 double wcNs; 190 bool valid = mCpuUsage.sampleAndEnable(wcNs); 191 192 // record sample for wall clock statistics 193 if (valid) { 194 mWcStats.sample(wcNs); 195 } 196 197 // get the current CPU number 198 int cpuNum = sched_getcpu(); 199 200 // get the current CPU frequency in kHz 201 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 202 203 // check if either CPU number or frequency changed 204 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 205 mCpuNum = cpuNum; 206 mCpukHz = cpukHz; 207 // ignore sample for purposes of cycles 208 valid = false; 209 } 210 211 // if no change in CPU number or frequency, then record sample for cycle statistics 212 if (valid && mCpukHz > 0) { 213 double cycles = wcNs * cpukHz * 0.000001; 214 mHzStats.sample(cycles); 215 } 216 217 unsigned n = mWcStats.n(); 218 // mCpuUsage.elapsed() is expensive, so don't call it every loop 219 if ((n & 127) == 1) { 220 long long elapsed = mCpuUsage.elapsed(); 221 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 222 double perLoop = elapsed / (double) n; 223 double perLoop100 = perLoop * 0.01; 224 double perLoop1k = perLoop * 0.001; 225 double mean = mWcStats.mean(); 226 double stddev = mWcStats.stddev(); 227 double minimum = mWcStats.minimum(); 228 double maximum = mWcStats.maximum(); 229 double meanCycles = mHzStats.mean(); 230 double stddevCycles = mHzStats.stddev(); 231 double minCycles = mHzStats.minimum(); 232 double maxCycles = mHzStats.maximum(); 233 mCpuUsage.resetElapsed(); 234 mWcStats.reset(); 235 mHzStats.reset(); 236 ALOGD("CPU usage for %s over past %.1f secs\n" 237 " (%u mixer loops at %.1f mean ms per loop):\n" 238 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 239 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 240 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 241 title.string(), 242 elapsed * .000000001, n, perLoop * .000001, 243 mean * .001, 244 stddev * .001, 245 minimum * .001, 246 maximum * .001, 247 mean / perLoop100, 248 stddev / perLoop100, 249 minimum / perLoop100, 250 maximum / perLoop100, 251 meanCycles / perLoop1k, 252 stddevCycles / perLoop1k, 253 minCycles / perLoop1k, 254 maxCycles / perLoop1k); 255 256 } 257 } 258#endif 259}; 260 261// ---------------------------------------------------------------------------- 262// ThreadBase 263// ---------------------------------------------------------------------------- 264 265AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 266 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 267 : Thread(false /*canCallJava*/), 268 mType(type), 269 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 270 // mChannelMask 271 mChannelCount(0), 272 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 273 mParamStatus(NO_ERROR), 274 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 275 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 276 // mName will be set by concrete (non-virtual) subclass 277 mDeathRecipient(new PMDeathRecipient(this)) 278{ 279} 280 281AudioFlinger::ThreadBase::~ThreadBase() 282{ 283 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 284 for (size_t i = 0; i < mConfigEvents.size(); i++) { 285 delete mConfigEvents[i]; 286 } 287 mConfigEvents.clear(); 288 289 mParamCond.broadcast(); 290 // do not lock the mutex in destructor 291 releaseWakeLock_l(); 292 if (mPowerManager != 0) { 293 sp<IBinder> binder = mPowerManager->asBinder(); 294 binder->unlinkToDeath(mDeathRecipient); 295 } 296} 297 298void AudioFlinger::ThreadBase::exit() 299{ 300 ALOGV("ThreadBase::exit"); 301 // do any cleanup required for exit to succeed 302 preExit(); 303 { 304 // This lock prevents the following race in thread (uniprocessor for illustration): 305 // if (!exitPending()) { 306 // // context switch from here to exit() 307 // // exit() calls requestExit(), what exitPending() observes 308 // // exit() calls signal(), which is dropped since no waiters 309 // // context switch back from exit() to here 310 // mWaitWorkCV.wait(...); 311 // // now thread is hung 312 // } 313 AutoMutex lock(mLock); 314 requestExit(); 315 mWaitWorkCV.broadcast(); 316 } 317 // When Thread::requestExitAndWait is made virtual and this method is renamed to 318 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 319 requestExitAndWait(); 320} 321 322status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 323{ 324 status_t status; 325 326 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 327 Mutex::Autolock _l(mLock); 328 329 mNewParameters.add(keyValuePairs); 330 mWaitWorkCV.signal(); 331 // wait condition with timeout in case the thread loop has exited 332 // before the request could be processed 333 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 334 status = mParamStatus; 335 mWaitWorkCV.signal(); 336 } else { 337 status = TIMED_OUT; 338 } 339 return status; 340} 341 342void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 343{ 344 Mutex::Autolock _l(mLock); 345 sendIoConfigEvent_l(event, param); 346} 347 348// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 349void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 350{ 351 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 352 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 353 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 354 param); 355 mWaitWorkCV.signal(); 356} 357 358// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 359void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 360{ 361 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 362 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 363 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 364 mConfigEvents.size(), pid, tid, prio); 365 mWaitWorkCV.signal(); 366} 367 368void AudioFlinger::ThreadBase::processConfigEvents() 369{ 370 mLock.lock(); 371 while (!mConfigEvents.isEmpty()) { 372 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 373 ConfigEvent *event = mConfigEvents[0]; 374 mConfigEvents.removeAt(0); 375 // release mLock before locking AudioFlinger mLock: lock order is always 376 // AudioFlinger then ThreadBase to avoid cross deadlock 377 mLock.unlock(); 378 switch(event->type()) { 379 case CFG_EVENT_PRIO: { 380 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 381 // FIXME Need to understand why this has be done asynchronously 382 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 383 true /*asynchronous*/); 384 if (err != 0) { 385 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 386 "error %d", 387 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 388 } 389 } break; 390 case CFG_EVENT_IO: { 391 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 392 mAudioFlinger->mLock.lock(); 393 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 394 mAudioFlinger->mLock.unlock(); 395 } break; 396 default: 397 ALOGE("processConfigEvents() unknown event type %d", event->type()); 398 break; 399 } 400 delete event; 401 mLock.lock(); 402 } 403 mLock.unlock(); 404} 405 406void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 407{ 408 const size_t SIZE = 256; 409 char buffer[SIZE]; 410 String8 result; 411 412 bool locked = AudioFlinger::dumpTryLock(mLock); 413 if (!locked) { 414 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 415 write(fd, buffer, strlen(buffer)); 416 } 417 418 snprintf(buffer, SIZE, "io handle: %d\n", mId); 419 result.append(buffer); 420 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 437 result.append(buffer); 438 439 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 440 result.append(buffer); 441 result.append(" Index Command"); 442 for (size_t i = 0; i < mNewParameters.size(); ++i) { 443 snprintf(buffer, SIZE, "\n %02d ", i); 444 result.append(buffer); 445 result.append(mNewParameters[i]); 446 } 447 448 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 449 result.append(buffer); 450 for (size_t i = 0; i < mConfigEvents.size(); i++) { 451 mConfigEvents[i]->dump(buffer, SIZE); 452 result.append(buffer); 453 } 454 result.append("\n"); 455 456 write(fd, result.string(), result.size()); 457 458 if (locked) { 459 mLock.unlock(); 460 } 461} 462 463void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 464{ 465 const size_t SIZE = 256; 466 char buffer[SIZE]; 467 String8 result; 468 469 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 470 write(fd, buffer, strlen(buffer)); 471 472 for (size_t i = 0; i < mEffectChains.size(); ++i) { 473 sp<EffectChain> chain = mEffectChains[i]; 474 if (chain != 0) { 475 chain->dump(fd, args); 476 } 477 } 478} 479 480void AudioFlinger::ThreadBase::acquireWakeLock() 481{ 482 Mutex::Autolock _l(mLock); 483 acquireWakeLock_l(); 484} 485 486void AudioFlinger::ThreadBase::acquireWakeLock_l() 487{ 488 if (mPowerManager == 0) { 489 // use checkService() to avoid blocking if power service is not up yet 490 sp<IBinder> binder = 491 defaultServiceManager()->checkService(String16("power")); 492 if (binder == 0) { 493 ALOGW("Thread %s cannot connect to the power manager service", mName); 494 } else { 495 mPowerManager = interface_cast<IPowerManager>(binder); 496 binder->linkToDeath(mDeathRecipient); 497 } 498 } 499 if (mPowerManager != 0) { 500 sp<IBinder> binder = new BBinder(); 501 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 502 binder, 503 String16(mName), 504 String16("media")); 505 if (status == NO_ERROR) { 506 mWakeLockToken = binder; 507 } 508 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 509 } 510} 511 512void AudioFlinger::ThreadBase::releaseWakeLock() 513{ 514 Mutex::Autolock _l(mLock); 515 releaseWakeLock_l(); 516} 517 518void AudioFlinger::ThreadBase::releaseWakeLock_l() 519{ 520 if (mWakeLockToken != 0) { 521 ALOGV("releaseWakeLock_l() %s", mName); 522 if (mPowerManager != 0) { 523 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 524 } 525 mWakeLockToken.clear(); 526 } 527} 528 529void AudioFlinger::ThreadBase::clearPowerManager() 530{ 531 Mutex::Autolock _l(mLock); 532 releaseWakeLock_l(); 533 mPowerManager.clear(); 534} 535 536void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 537{ 538 sp<ThreadBase> thread = mThread.promote(); 539 if (thread != 0) { 540 thread->clearPowerManager(); 541 } 542 ALOGW("power manager service died !!!"); 543} 544 545void AudioFlinger::ThreadBase::setEffectSuspended( 546 const effect_uuid_t *type, bool suspend, int sessionId) 547{ 548 Mutex::Autolock _l(mLock); 549 setEffectSuspended_l(type, suspend, sessionId); 550} 551 552void AudioFlinger::ThreadBase::setEffectSuspended_l( 553 const effect_uuid_t *type, bool suspend, int sessionId) 554{ 555 sp<EffectChain> chain = getEffectChain_l(sessionId); 556 if (chain != 0) { 557 if (type != NULL) { 558 chain->setEffectSuspended_l(type, suspend); 559 } else { 560 chain->setEffectSuspendedAll_l(suspend); 561 } 562 } 563 564 updateSuspendedSessions_l(type, suspend, sessionId); 565} 566 567void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 568{ 569 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 570 if (index < 0) { 571 return; 572 } 573 574 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 575 mSuspendedSessions.valueAt(index); 576 577 for (size_t i = 0; i < sessionEffects.size(); i++) { 578 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 579 for (int j = 0; j < desc->mRefCount; j++) { 580 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 581 chain->setEffectSuspendedAll_l(true); 582 } else { 583 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 584 desc->mType.timeLow); 585 chain->setEffectSuspended_l(&desc->mType, true); 586 } 587 } 588 } 589} 590 591void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 592 bool suspend, 593 int sessionId) 594{ 595 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 596 597 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 598 599 if (suspend) { 600 if (index >= 0) { 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } else { 603 mSuspendedSessions.add(sessionId, sessionEffects); 604 } 605 } else { 606 if (index < 0) { 607 return; 608 } 609 sessionEffects = mSuspendedSessions.valueAt(index); 610 } 611 612 613 int key = EffectChain::kKeyForSuspendAll; 614 if (type != NULL) { 615 key = type->timeLow; 616 } 617 index = sessionEffects.indexOfKey(key); 618 619 sp<SuspendedSessionDesc> desc; 620 if (suspend) { 621 if (index >= 0) { 622 desc = sessionEffects.valueAt(index); 623 } else { 624 desc = new SuspendedSessionDesc(); 625 if (type != NULL) { 626 desc->mType = *type; 627 } 628 sessionEffects.add(key, desc); 629 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 630 } 631 desc->mRefCount++; 632 } else { 633 if (index < 0) { 634 return; 635 } 636 desc = sessionEffects.valueAt(index); 637 if (--desc->mRefCount == 0) { 638 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 639 sessionEffects.removeItemsAt(index); 640 if (sessionEffects.isEmpty()) { 641 ALOGV("updateSuspendedSessions_l() restore removing session %d", 642 sessionId); 643 mSuspendedSessions.removeItem(sessionId); 644 } 645 } 646 } 647 if (!sessionEffects.isEmpty()) { 648 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 649 } 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 Mutex::Autolock _l(mLock); 657 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 658} 659 660void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 661 bool enabled, 662 int sessionId) 663{ 664 if (mType != RECORD) { 665 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 666 // another session. This gives the priority to well behaved effect control panels 667 // and applications not using global effects. 668 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 669 // global effects 670 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 671 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 672 } 673 } 674 675 sp<EffectChain> chain = getEffectChain_l(sessionId); 676 if (chain != 0) { 677 chain->checkSuspendOnEffectEnabled(effect, enabled); 678 } 679} 680 681// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 682sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 683 const sp<AudioFlinger::Client>& client, 684 const sp<IEffectClient>& effectClient, 685 int32_t priority, 686 int sessionId, 687 effect_descriptor_t *desc, 688 int *enabled, 689 status_t *status 690 ) 691{ 692 sp<EffectModule> effect; 693 sp<EffectHandle> handle; 694 status_t lStatus; 695 sp<EffectChain> chain; 696 bool chainCreated = false; 697 bool effectCreated = false; 698 bool effectRegistered = false; 699 700 lStatus = initCheck(); 701 if (lStatus != NO_ERROR) { 702 ALOGW("createEffect_l() Audio driver not initialized."); 703 goto Exit; 704 } 705 706 // Do not allow effects with session ID 0 on direct output or duplicating threads 707 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 708 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 709 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 710 desc->name, sessionId); 711 lStatus = BAD_VALUE; 712 goto Exit; 713 } 714 // Only Pre processor effects are allowed on input threads and only on input threads 715 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 716 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 717 desc->name, desc->flags, mType); 718 lStatus = BAD_VALUE; 719 goto Exit; 720 } 721 722 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 723 724 { // scope for mLock 725 Mutex::Autolock _l(mLock); 726 727 // check for existing effect chain with the requested audio session 728 chain = getEffectChain_l(sessionId); 729 if (chain == 0) { 730 // create a new chain for this session 731 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 732 chain = new EffectChain(this, sessionId); 733 addEffectChain_l(chain); 734 chain->setStrategy(getStrategyForSession_l(sessionId)); 735 chainCreated = true; 736 } else { 737 effect = chain->getEffectFromDesc_l(desc); 738 } 739 740 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 741 742 if (effect == 0) { 743 int id = mAudioFlinger->nextUniqueId(); 744 // Check CPU and memory usage 745 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 746 if (lStatus != NO_ERROR) { 747 goto Exit; 748 } 749 effectRegistered = true; 750 // create a new effect module if none present in the chain 751 effect = new EffectModule(this, chain, desc, id, sessionId); 752 lStatus = effect->status(); 753 if (lStatus != NO_ERROR) { 754 goto Exit; 755 } 756 lStatus = chain->addEffect_l(effect); 757 if (lStatus != NO_ERROR) { 758 goto Exit; 759 } 760 effectCreated = true; 761 762 effect->setDevice(mOutDevice); 763 effect->setDevice(mInDevice); 764 effect->setMode(mAudioFlinger->getMode()); 765 effect->setAudioSource(mAudioSource); 766 } 767 // create effect handle and connect it to effect module 768 handle = new EffectHandle(effect, client, effectClient, priority); 769 lStatus = effect->addHandle(handle.get()); 770 if (enabled != NULL) { 771 *enabled = (int)effect->isEnabled(); 772 } 773 } 774 775Exit: 776 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 777 Mutex::Autolock _l(mLock); 778 if (effectCreated) { 779 chain->removeEffect_l(effect); 780 } 781 if (effectRegistered) { 782 AudioSystem::unregisterEffect(effect->id()); 783 } 784 if (chainCreated) { 785 removeEffectChain_l(chain); 786 } 787 handle.clear(); 788 } 789 790 if (status != NULL) { 791 *status = lStatus; 792 } 793 return handle; 794} 795 796sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 797{ 798 Mutex::Autolock _l(mLock); 799 return getEffect_l(sessionId, effectId); 800} 801 802sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 803{ 804 sp<EffectChain> chain = getEffectChain_l(sessionId); 805 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 806} 807 808// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 809// PlaybackThread::mLock held 810status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 811{ 812 // check for existing effect chain with the requested audio session 813 int sessionId = effect->sessionId(); 814 sp<EffectChain> chain = getEffectChain_l(sessionId); 815 bool chainCreated = false; 816 817 if (chain == 0) { 818 // create a new chain for this session 819 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 820 chain = new EffectChain(this, sessionId); 821 addEffectChain_l(chain); 822 chain->setStrategy(getStrategyForSession_l(sessionId)); 823 chainCreated = true; 824 } 825 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 826 827 if (chain->getEffectFromId_l(effect->id()) != 0) { 828 ALOGW("addEffect_l() %p effect %s already present in chain %p", 829 this, effect->desc().name, chain.get()); 830 return BAD_VALUE; 831 } 832 833 status_t status = chain->addEffect_l(effect); 834 if (status != NO_ERROR) { 835 if (chainCreated) { 836 removeEffectChain_l(chain); 837 } 838 return status; 839 } 840 841 effect->setDevice(mOutDevice); 842 effect->setDevice(mInDevice); 843 effect->setMode(mAudioFlinger->getMode()); 844 effect->setAudioSource(mAudioSource); 845 return NO_ERROR; 846} 847 848void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 849 850 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 851 effect_descriptor_t desc = effect->desc(); 852 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 853 detachAuxEffect_l(effect->id()); 854 } 855 856 sp<EffectChain> chain = effect->chain().promote(); 857 if (chain != 0) { 858 // remove effect chain if removing last effect 859 if (chain->removeEffect_l(effect) == 0) { 860 removeEffectChain_l(chain); 861 } 862 } else { 863 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 864 } 865} 866 867void AudioFlinger::ThreadBase::lockEffectChains_l( 868 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 869{ 870 effectChains = mEffectChains; 871 for (size_t i = 0; i < mEffectChains.size(); i++) { 872 mEffectChains[i]->lock(); 873 } 874} 875 876void AudioFlinger::ThreadBase::unlockEffectChains( 877 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 878{ 879 for (size_t i = 0; i < effectChains.size(); i++) { 880 effectChains[i]->unlock(); 881 } 882} 883 884sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 885{ 886 Mutex::Autolock _l(mLock); 887 return getEffectChain_l(sessionId); 888} 889 890sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 891{ 892 size_t size = mEffectChains.size(); 893 for (size_t i = 0; i < size; i++) { 894 if (mEffectChains[i]->sessionId() == sessionId) { 895 return mEffectChains[i]; 896 } 897 } 898 return 0; 899} 900 901void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 902{ 903 Mutex::Autolock _l(mLock); 904 size_t size = mEffectChains.size(); 905 for (size_t i = 0; i < size; i++) { 906 mEffectChains[i]->setMode_l(mode); 907 } 908} 909 910void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 911 EffectHandle *handle, 912 bool unpinIfLast) { 913 914 Mutex::Autolock _l(mLock); 915 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 916 // delete the effect module if removing last handle on it 917 if (effect->removeHandle(handle) == 0) { 918 if (!effect->isPinned() || unpinIfLast) { 919 removeEffect_l(effect); 920 AudioSystem::unregisterEffect(effect->id()); 921 } 922 } 923} 924 925// ---------------------------------------------------------------------------- 926// Playback 927// ---------------------------------------------------------------------------- 928 929AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 930 AudioStreamOut* output, 931 audio_io_handle_t id, 932 audio_devices_t device, 933 type_t type) 934 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 935 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 936 // mStreamTypes[] initialized in constructor body 937 mOutput(output), 938 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 939 mMixerStatus(MIXER_IDLE), 940 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 941 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 942 mBytesRemaining(0), 943 mCurrentWriteLength(0), 944 mUseAsyncWrite(false), 945 mWriteBlocked(false), 946 mDraining(false), 947 mScreenState(AudioFlinger::mScreenState), 948 // index 0 is reserved for normal mixer's submix 949 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 950{ 951 snprintf(mName, kNameLength, "AudioOut_%X", id); 952 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 953 954 // Assumes constructor is called by AudioFlinger with it's mLock held, but 955 // it would be safer to explicitly pass initial masterVolume/masterMute as 956 // parameter. 957 // 958 // If the HAL we are using has support for master volume or master mute, 959 // then do not attenuate or mute during mixing (just leave the volume at 1.0 960 // and the mute set to false). 961 mMasterVolume = audioFlinger->masterVolume_l(); 962 mMasterMute = audioFlinger->masterMute_l(); 963 if (mOutput && mOutput->audioHwDev) { 964 if (mOutput->audioHwDev->canSetMasterVolume()) { 965 mMasterVolume = 1.0; 966 } 967 968 if (mOutput->audioHwDev->canSetMasterMute()) { 969 mMasterMute = false; 970 } 971 } 972 973 readOutputParameters(); 974 975 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 976 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 977 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 978 stream = (audio_stream_type_t) (stream + 1)) { 979 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 980 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 981 } 982 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 983 // because mAudioFlinger doesn't have one to copy from 984} 985 986AudioFlinger::PlaybackThread::~PlaybackThread() 987{ 988 mAudioFlinger->unregisterWriter(mNBLogWriter); 989 delete [] mAllocMixBuffer; 990} 991 992void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 993{ 994 dumpInternals(fd, args); 995 dumpTracks(fd, args); 996 dumpEffectChains(fd, args); 997} 998 999void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1000{ 1001 const size_t SIZE = 256; 1002 char buffer[SIZE]; 1003 String8 result; 1004 1005 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1006 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1007 const stream_type_t *st = &mStreamTypes[i]; 1008 if (i > 0) { 1009 result.appendFormat(", "); 1010 } 1011 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1012 if (st->mute) { 1013 result.append("M"); 1014 } 1015 } 1016 result.append("\n"); 1017 write(fd, result.string(), result.length()); 1018 result.clear(); 1019 1020 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1021 result.append(buffer); 1022 Track::appendDumpHeader(result); 1023 for (size_t i = 0; i < mTracks.size(); ++i) { 1024 sp<Track> track = mTracks[i]; 1025 if (track != 0) { 1026 track->dump(buffer, SIZE); 1027 result.append(buffer); 1028 } 1029 } 1030 1031 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1032 result.append(buffer); 1033 Track::appendDumpHeader(result); 1034 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1035 sp<Track> track = mActiveTracks[i].promote(); 1036 if (track != 0) { 1037 track->dump(buffer, SIZE); 1038 result.append(buffer); 1039 } 1040 } 1041 write(fd, result.string(), result.size()); 1042 1043 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1044 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1045 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1046 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1047} 1048 1049void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1050{ 1051 const size_t SIZE = 256; 1052 char buffer[SIZE]; 1053 String8 result; 1054 1055 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1056 result.append(buffer); 1057 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1058 ns2ms(systemTime() - mLastWriteTime)); 1059 result.append(buffer); 1060 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1061 result.append(buffer); 1062 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1063 result.append(buffer); 1064 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1065 result.append(buffer); 1066 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1067 result.append(buffer); 1068 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1069 result.append(buffer); 1070 write(fd, result.string(), result.size()); 1071 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1072 1073 dumpBase(fd, args); 1074} 1075 1076// Thread virtuals 1077status_t AudioFlinger::PlaybackThread::readyToRun() 1078{ 1079 status_t status = initCheck(); 1080 if (status == NO_ERROR) { 1081 ALOGI("AudioFlinger's thread %p ready to run", this); 1082 } else { 1083 ALOGE("No working audio driver found."); 1084 } 1085 return status; 1086} 1087 1088void AudioFlinger::PlaybackThread::onFirstRef() 1089{ 1090 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1091} 1092 1093// ThreadBase virtuals 1094void AudioFlinger::PlaybackThread::preExit() 1095{ 1096 ALOGV(" preExit()"); 1097 // FIXME this is using hard-coded strings but in the future, this functionality will be 1098 // converted to use audio HAL extensions required to support tunneling 1099 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1100} 1101 1102// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1103sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1104 const sp<AudioFlinger::Client>& client, 1105 audio_stream_type_t streamType, 1106 uint32_t sampleRate, 1107 audio_format_t format, 1108 audio_channel_mask_t channelMask, 1109 size_t frameCount, 1110 const sp<IMemory>& sharedBuffer, 1111 int sessionId, 1112 IAudioFlinger::track_flags_t *flags, 1113 pid_t tid, 1114 status_t *status) 1115{ 1116 sp<Track> track; 1117 status_t lStatus; 1118 1119 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1120 1121 // client expresses a preference for FAST, but we get the final say 1122 if (*flags & IAudioFlinger::TRACK_FAST) { 1123 if ( 1124 // not timed 1125 (!isTimed) && 1126 // either of these use cases: 1127 ( 1128 // use case 1: shared buffer with any frame count 1129 ( 1130 (sharedBuffer != 0) 1131 ) || 1132 // use case 2: callback handler and frame count is default or at least as large as HAL 1133 ( 1134 (tid != -1) && 1135 ((frameCount == 0) || 1136 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1137 ) 1138 ) && 1139 // PCM data 1140 audio_is_linear_pcm(format) && 1141 // mono or stereo 1142 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1143 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1144#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1145 // hardware sample rate 1146 (sampleRate == mSampleRate) && 1147#endif 1148 // normal mixer has an associated fast mixer 1149 hasFastMixer() && 1150 // there are sufficient fast track slots available 1151 (mFastTrackAvailMask != 0) 1152 // FIXME test that MixerThread for this fast track has a capable output HAL 1153 // FIXME add a permission test also? 1154 ) { 1155 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1156 if (frameCount == 0) { 1157 frameCount = mFrameCount * kFastTrackMultiplier; 1158 } 1159 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1160 frameCount, mFrameCount); 1161 } else { 1162 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1163 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1164 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1165 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1166 audio_is_linear_pcm(format), 1167 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1168 *flags &= ~IAudioFlinger::TRACK_FAST; 1169 // For compatibility with AudioTrack calculation, buffer depth is forced 1170 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1171 // This is probably too conservative, but legacy application code may depend on it. 1172 // If you change this calculation, also review the start threshold which is related. 1173 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1174 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1175 if (minBufCount < 2) { 1176 minBufCount = 2; 1177 } 1178 size_t minFrameCount = mNormalFrameCount * minBufCount; 1179 if (frameCount < minFrameCount) { 1180 frameCount = minFrameCount; 1181 } 1182 } 1183 } 1184 1185 if (mType == DIRECT) { 1186 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1187 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1188 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1189 "for output %p with format %d", 1190 sampleRate, format, channelMask, mOutput, mFormat); 1191 lStatus = BAD_VALUE; 1192 goto Exit; 1193 } 1194 } 1195 } else if (mType == OFFLOAD) { 1196 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1197 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1198 "for output %p with format %d", 1199 sampleRate, format, channelMask, mOutput, mFormat); 1200 lStatus = BAD_VALUE; 1201 goto Exit; 1202 } 1203 } else { 1204 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1205 ALOGE("createTrack_l() Bad parameter: format %d \"" 1206 "for output %p with format %d", 1207 format, mOutput, mFormat); 1208 lStatus = BAD_VALUE; 1209 goto Exit; 1210 } 1211 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1212 if (sampleRate > mSampleRate*2) { 1213 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1214 lStatus = BAD_VALUE; 1215 goto Exit; 1216 } 1217 } 1218 1219 lStatus = initCheck(); 1220 if (lStatus != NO_ERROR) { 1221 ALOGE("Audio driver not initialized."); 1222 goto Exit; 1223 } 1224 1225 { // scope for mLock 1226 Mutex::Autolock _l(mLock); 1227 1228 // all tracks in same audio session must share the same routing strategy otherwise 1229 // conflicts will happen when tracks are moved from one output to another by audio policy 1230 // manager 1231 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1232 for (size_t i = 0; i < mTracks.size(); ++i) { 1233 sp<Track> t = mTracks[i]; 1234 if (t != 0 && !t->isOutputTrack()) { 1235 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1236 if (sessionId == t->sessionId() && strategy != actual) { 1237 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1238 strategy, actual); 1239 lStatus = BAD_VALUE; 1240 goto Exit; 1241 } 1242 } 1243 } 1244 1245 if (!isTimed) { 1246 track = new Track(this, client, streamType, sampleRate, format, 1247 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1248 } else { 1249 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1250 channelMask, frameCount, sharedBuffer, sessionId); 1251 } 1252 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1253 lStatus = NO_MEMORY; 1254 goto Exit; 1255 } 1256 1257 mTracks.add(track); 1258 1259 sp<EffectChain> chain = getEffectChain_l(sessionId); 1260 if (chain != 0) { 1261 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1262 track->setMainBuffer(chain->inBuffer()); 1263 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1264 chain->incTrackCnt(); 1265 } 1266 1267 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1268 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1269 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1270 // so ask activity manager to do this on our behalf 1271 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1272 } 1273 } 1274 1275 lStatus = NO_ERROR; 1276 1277Exit: 1278 if (status) { 1279 *status = lStatus; 1280 } 1281 return track; 1282} 1283 1284uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1285{ 1286 return latency; 1287} 1288 1289uint32_t AudioFlinger::PlaybackThread::latency() const 1290{ 1291 Mutex::Autolock _l(mLock); 1292 return latency_l(); 1293} 1294uint32_t AudioFlinger::PlaybackThread::latency_l() const 1295{ 1296 if (initCheck() == NO_ERROR) { 1297 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1298 } else { 1299 return 0; 1300 } 1301} 1302 1303void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1304{ 1305 Mutex::Autolock _l(mLock); 1306 // Don't apply master volume in SW if our HAL can do it for us. 1307 if (mOutput && mOutput->audioHwDev && 1308 mOutput->audioHwDev->canSetMasterVolume()) { 1309 mMasterVolume = 1.0; 1310 } else { 1311 mMasterVolume = value; 1312 } 1313} 1314 1315void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 // Don't apply master mute in SW if our HAL can do it for us. 1319 if (mOutput && mOutput->audioHwDev && 1320 mOutput->audioHwDev->canSetMasterMute()) { 1321 mMasterMute = false; 1322 } else { 1323 mMasterMute = muted; 1324 } 1325} 1326 1327void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1328{ 1329 Mutex::Autolock _l(mLock); 1330 mStreamTypes[stream].volume = value; 1331 signal_l(); 1332} 1333 1334void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1335{ 1336 Mutex::Autolock _l(mLock); 1337 mStreamTypes[stream].mute = muted; 1338 signal_l(); 1339} 1340 1341float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1342{ 1343 Mutex::Autolock _l(mLock); 1344 return mStreamTypes[stream].volume; 1345} 1346 1347// addTrack_l() must be called with ThreadBase::mLock held 1348status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1349{ 1350 status_t status = ALREADY_EXISTS; 1351 1352 // set retry count for buffer fill 1353 track->mRetryCount = kMaxTrackStartupRetries; 1354 if (mActiveTracks.indexOf(track) < 0) { 1355 // the track is newly added, make sure it fills up all its 1356 // buffers before playing. This is to ensure the client will 1357 // effectively get the latency it requested. 1358 if (!track->isOutputTrack()) { 1359 TrackBase::track_state state = track->mState; 1360 mLock.unlock(); 1361 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1362 mLock.lock(); 1363 // abort track was stopped/paused while we released the lock 1364 if (state != track->mState) { 1365 if (status == NO_ERROR) { 1366 mLock.unlock(); 1367 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1368 mLock.lock(); 1369 } 1370 return INVALID_OPERATION; 1371 } 1372 // abort if start is rejected by audio policy manager 1373 if (status != NO_ERROR) { 1374 return PERMISSION_DENIED; 1375 } 1376#ifdef ADD_BATTERY_DATA 1377 // to track the speaker usage 1378 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1379#endif 1380 } 1381 1382 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1383 track->mResetDone = false; 1384 track->mPresentationCompleteFrames = 0; 1385 mActiveTracks.add(track); 1386 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1387 if (chain != 0) { 1388 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1389 track->sessionId()); 1390 chain->incActiveTrackCnt(); 1391 } 1392 1393 status = NO_ERROR; 1394 } 1395 1396 ALOGV("mWaitWorkCV.broadcast"); 1397 mWaitWorkCV.broadcast(); 1398 1399 return status; 1400} 1401 1402bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1403{ 1404 track->terminate(); 1405 // active tracks are removed by threadLoop() 1406 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1407 track->mState = TrackBase::STOPPED; 1408 if (!trackActive) { 1409 removeTrack_l(track); 1410 } else if (track->isFastTrack() || track->isOffloaded()) { 1411 track->mState = TrackBase::STOPPING_1; 1412 } 1413 1414 return trackActive; 1415} 1416 1417void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1418{ 1419 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1420 mTracks.remove(track); 1421 deleteTrackName_l(track->name()); 1422 // redundant as track is about to be destroyed, for dumpsys only 1423 track->mName = -1; 1424 if (track->isFastTrack()) { 1425 int index = track->mFastIndex; 1426 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1427 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1428 mFastTrackAvailMask |= 1 << index; 1429 // redundant as track is about to be destroyed, for dumpsys only 1430 track->mFastIndex = -1; 1431 } 1432 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1433 if (chain != 0) { 1434 chain->decTrackCnt(); 1435 } 1436} 1437 1438void AudioFlinger::PlaybackThread::signal_l() 1439{ 1440 // Thread could be blocked waiting for async 1441 // so signal it to handle state changes immediately 1442 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1443 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1444 mSignalPending = true; 1445 mWaitWorkCV.signal(); 1446} 1447 1448String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1449{ 1450 Mutex::Autolock _l(mLock); 1451 if (initCheck() != NO_ERROR) { 1452 return String8(); 1453 } 1454 1455 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1456 const String8 out_s8(s); 1457 free(s); 1458 return out_s8; 1459} 1460 1461// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1462void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1463 AudioSystem::OutputDescriptor desc; 1464 void *param2 = NULL; 1465 1466 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1467 param); 1468 1469 switch (event) { 1470 case AudioSystem::OUTPUT_OPENED: 1471 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1472 desc.channelMask = mChannelMask; 1473 desc.samplingRate = mSampleRate; 1474 desc.format = mFormat; 1475 desc.frameCount = mNormalFrameCount; // FIXME see 1476 // AudioFlinger::frameCount(audio_io_handle_t) 1477 desc.latency = latency(); 1478 param2 = &desc; 1479 break; 1480 1481 case AudioSystem::STREAM_CONFIG_CHANGED: 1482 param2 = ¶m; 1483 case AudioSystem::OUTPUT_CLOSED: 1484 default: 1485 break; 1486 } 1487 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1488} 1489 1490void AudioFlinger::PlaybackThread::writeCallback() 1491{ 1492 ALOG_ASSERT(mCallbackThread != 0); 1493 mCallbackThread->setWriteBlocked(false); 1494} 1495 1496void AudioFlinger::PlaybackThread::drainCallback() 1497{ 1498 ALOG_ASSERT(mCallbackThread != 0); 1499 mCallbackThread->setDraining(false); 1500} 1501 1502void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1503{ 1504 Mutex::Autolock _l(mLock); 1505 mWriteBlocked = value; 1506 if (!value) { 1507 mWaitWorkCV.signal(); 1508 } 1509} 1510 1511void AudioFlinger::PlaybackThread::setDraining(bool value) 1512{ 1513 Mutex::Autolock _l(mLock); 1514 mDraining = value; 1515 if (!value) { 1516 mWaitWorkCV.signal(); 1517 } 1518} 1519 1520// static 1521int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1522 void *param, 1523 void *cookie) 1524{ 1525 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1526 ALOGV("asyncCallback() event %d", event); 1527 switch (event) { 1528 case STREAM_CBK_EVENT_WRITE_READY: 1529 me->writeCallback(); 1530 break; 1531 case STREAM_CBK_EVENT_DRAIN_READY: 1532 me->drainCallback(); 1533 break; 1534 default: 1535 ALOGW("asyncCallback() unknown event %d", event); 1536 break; 1537 } 1538 return 0; 1539} 1540 1541void AudioFlinger::PlaybackThread::readOutputParameters() 1542{ 1543 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1544 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1545 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1546 if (!audio_is_output_channel(mChannelMask)) { 1547 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1548 } 1549 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1550 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1551 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1552 } 1553 mChannelCount = popcount(mChannelMask); 1554 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1555 if (!audio_is_valid_format(mFormat)) { 1556 LOG_FATAL("HAL format %d not valid for output", mFormat); 1557 } 1558 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1559 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1560 mFormat); 1561 } 1562 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1563 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1564 if (mFrameCount & 15) { 1565 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1566 mFrameCount); 1567 } 1568 1569 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1570 (mOutput->stream->set_callback != NULL)) { 1571 if (mOutput->stream->set_callback(mOutput->stream, 1572 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1573 mUseAsyncWrite = true; 1574 } 1575 } 1576 1577 // Calculate size of normal mix buffer relative to the HAL output buffer size 1578 double multiplier = 1.0; 1579 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1580 kUseFastMixer == FastMixer_Dynamic)) { 1581 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1582 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1583 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1584 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1585 maxNormalFrameCount = maxNormalFrameCount & ~15; 1586 if (maxNormalFrameCount < minNormalFrameCount) { 1587 maxNormalFrameCount = minNormalFrameCount; 1588 } 1589 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1590 if (multiplier <= 1.0) { 1591 multiplier = 1.0; 1592 } else if (multiplier <= 2.0) { 1593 if (2 * mFrameCount <= maxNormalFrameCount) { 1594 multiplier = 2.0; 1595 } else { 1596 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1597 } 1598 } else { 1599 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1600 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1601 // track, but we sometimes have to do this to satisfy the maximum frame count 1602 // constraint) 1603 // FIXME this rounding up should not be done if no HAL SRC 1604 uint32_t truncMult = (uint32_t) multiplier; 1605 if ((truncMult & 1)) { 1606 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1607 ++truncMult; 1608 } 1609 } 1610 multiplier = (double) truncMult; 1611 } 1612 } 1613 mNormalFrameCount = multiplier * mFrameCount; 1614 // round up to nearest 16 frames to satisfy AudioMixer 1615 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1616 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1617 mNormalFrameCount); 1618 1619 delete[] mAllocMixBuffer; 1620 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1621 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1622 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1623 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1624 1625 // force reconfiguration of effect chains and engines to take new buffer size and audio 1626 // parameters into account 1627 // Note that mLock is not held when readOutputParameters() is called from the constructor 1628 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1629 // matter. 1630 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1631 Vector< sp<EffectChain> > effectChains = mEffectChains; 1632 for (size_t i = 0; i < effectChains.size(); i ++) { 1633 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1634 } 1635} 1636 1637 1638status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1639{ 1640 if (halFrames == NULL || dspFrames == NULL) { 1641 return BAD_VALUE; 1642 } 1643 Mutex::Autolock _l(mLock); 1644 if (initCheck() != NO_ERROR) { 1645 return INVALID_OPERATION; 1646 } 1647 size_t framesWritten = mBytesWritten / mFrameSize; 1648 *halFrames = framesWritten; 1649 1650 if (isSuspended()) { 1651 // return an estimation of rendered frames when the output is suspended 1652 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1653 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1654 return NO_ERROR; 1655 } else { 1656 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1657 } 1658} 1659 1660uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1661{ 1662 Mutex::Autolock _l(mLock); 1663 uint32_t result = 0; 1664 if (getEffectChain_l(sessionId) != 0) { 1665 result = EFFECT_SESSION; 1666 } 1667 1668 for (size_t i = 0; i < mTracks.size(); ++i) { 1669 sp<Track> track = mTracks[i]; 1670 if (sessionId == track->sessionId() && !track->isInvalid()) { 1671 result |= TRACK_SESSION; 1672 break; 1673 } 1674 } 1675 1676 return result; 1677} 1678 1679uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1680{ 1681 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1682 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1683 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1684 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1685 } 1686 for (size_t i = 0; i < mTracks.size(); i++) { 1687 sp<Track> track = mTracks[i]; 1688 if (sessionId == track->sessionId() && !track->isInvalid()) { 1689 return AudioSystem::getStrategyForStream(track->streamType()); 1690 } 1691 } 1692 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1693} 1694 1695 1696AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1697{ 1698 Mutex::Autolock _l(mLock); 1699 return mOutput; 1700} 1701 1702AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1703{ 1704 Mutex::Autolock _l(mLock); 1705 AudioStreamOut *output = mOutput; 1706 mOutput = NULL; 1707 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1708 // must push a NULL and wait for ack 1709 mOutputSink.clear(); 1710 mPipeSink.clear(); 1711 mNormalSink.clear(); 1712 return output; 1713} 1714 1715// this method must always be called either with ThreadBase mLock held or inside the thread loop 1716audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1717{ 1718 if (mOutput == NULL) { 1719 return NULL; 1720 } 1721 return &mOutput->stream->common; 1722} 1723 1724uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1725{ 1726 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1727} 1728 1729status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1730{ 1731 if (!isValidSyncEvent(event)) { 1732 return BAD_VALUE; 1733 } 1734 1735 Mutex::Autolock _l(mLock); 1736 1737 for (size_t i = 0; i < mTracks.size(); ++i) { 1738 sp<Track> track = mTracks[i]; 1739 if (event->triggerSession() == track->sessionId()) { 1740 (void) track->setSyncEvent(event); 1741 return NO_ERROR; 1742 } 1743 } 1744 1745 return NAME_NOT_FOUND; 1746} 1747 1748bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1749{ 1750 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1751} 1752 1753void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1754 const Vector< sp<Track> >& tracksToRemove) 1755{ 1756 size_t count = tracksToRemove.size(); 1757 if (CC_UNLIKELY(count)) { 1758 for (size_t i = 0 ; i < count ; i++) { 1759 const sp<Track>& track = tracksToRemove.itemAt(i); 1760 if (!track->isOutputTrack()) { 1761 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1762#ifdef ADD_BATTERY_DATA 1763 // to track the speaker usage 1764 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1765#endif 1766 if (track->isTerminated()) { 1767 AudioSystem::releaseOutput(mId); 1768 } 1769 } 1770 } 1771 } 1772} 1773 1774void AudioFlinger::PlaybackThread::checkSilentMode_l() 1775{ 1776 if (!mMasterMute) { 1777 char value[PROPERTY_VALUE_MAX]; 1778 if (property_get("ro.audio.silent", value, "0") > 0) { 1779 char *endptr; 1780 unsigned long ul = strtoul(value, &endptr, 0); 1781 if (*endptr == '\0' && ul != 0) { 1782 ALOGD("Silence is golden"); 1783 // The setprop command will not allow a property to be changed after 1784 // the first time it is set, so we don't have to worry about un-muting. 1785 setMasterMute_l(true); 1786 } 1787 } 1788 } 1789} 1790 1791// shared by MIXER and DIRECT, overridden by DUPLICATING 1792ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1793{ 1794 // FIXME rewrite to reduce number of system calls 1795 mLastWriteTime = systemTime(); 1796 mInWrite = true; 1797 ssize_t bytesWritten; 1798 1799 // If an NBAIO sink is present, use it to write the normal mixer's submix 1800 if (mNormalSink != 0) { 1801#define mBitShift 2 // FIXME 1802 size_t count = mBytesRemaining >> mBitShift; 1803 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1804 ATRACE_BEGIN("write"); 1805 // update the setpoint when AudioFlinger::mScreenState changes 1806 uint32_t screenState = AudioFlinger::mScreenState; 1807 if (screenState != mScreenState) { 1808 mScreenState = screenState; 1809 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1810 if (pipe != NULL) { 1811 pipe->setAvgFrames((mScreenState & 1) ? 1812 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1813 } 1814 } 1815 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1816 ATRACE_END(); 1817 if (framesWritten > 0) { 1818 bytesWritten = framesWritten << mBitShift; 1819 } else { 1820 bytesWritten = framesWritten; 1821 } 1822 // otherwise use the HAL / AudioStreamOut directly 1823 } else { 1824 // Direct output and offload threads 1825 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1826 if (mUseAsyncWrite) { 1827 mWriteBlocked = true; 1828 ALOG_ASSERT(mCallbackThread != 0); 1829 mCallbackThread->setWriteBlocked(true); 1830 } 1831 bytesWritten = mOutput->stream->write(mOutput->stream, 1832 mMixBuffer + offset, mBytesRemaining); 1833 if (mUseAsyncWrite && 1834 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1835 // do not wait for async callback in case of error of full write 1836 mWriteBlocked = false; 1837 ALOG_ASSERT(mCallbackThread != 0); 1838 mCallbackThread->setWriteBlocked(false); 1839 } 1840 } 1841 1842 mNumWrites++; 1843 mInWrite = false; 1844 1845 return bytesWritten; 1846} 1847 1848void AudioFlinger::PlaybackThread::threadLoop_drain() 1849{ 1850 if (mOutput->stream->drain) { 1851 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1852 if (mUseAsyncWrite) { 1853 mDraining = true; 1854 ALOG_ASSERT(mCallbackThread != 0); 1855 mCallbackThread->setDraining(true); 1856 } 1857 mOutput->stream->drain(mOutput->stream, 1858 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1859 : AUDIO_DRAIN_ALL); 1860 } 1861} 1862 1863void AudioFlinger::PlaybackThread::threadLoop_exit() 1864{ 1865 // Default implementation has nothing to do 1866} 1867 1868/* 1869The derived values that are cached: 1870 - mixBufferSize from frame count * frame size 1871 - activeSleepTime from activeSleepTimeUs() 1872 - idleSleepTime from idleSleepTimeUs() 1873 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1874 - maxPeriod from frame count and sample rate (MIXER only) 1875 1876The parameters that affect these derived values are: 1877 - frame count 1878 - frame size 1879 - sample rate 1880 - device type: A2DP or not 1881 - device latency 1882 - format: PCM or not 1883 - active sleep time 1884 - idle sleep time 1885*/ 1886 1887void AudioFlinger::PlaybackThread::cacheParameters_l() 1888{ 1889 mixBufferSize = mNormalFrameCount * mFrameSize; 1890 activeSleepTime = activeSleepTimeUs(); 1891 idleSleepTime = idleSleepTimeUs(); 1892} 1893 1894void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1895{ 1896 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1897 this, streamType, mTracks.size()); 1898 Mutex::Autolock _l(mLock); 1899 1900 size_t size = mTracks.size(); 1901 for (size_t i = 0; i < size; i++) { 1902 sp<Track> t = mTracks[i]; 1903 if (t->streamType() == streamType) { 1904 t->invalidate(); 1905 } 1906 } 1907} 1908 1909status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1910{ 1911 int session = chain->sessionId(); 1912 int16_t *buffer = mMixBuffer; 1913 bool ownsBuffer = false; 1914 1915 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1916 if (session > 0) { 1917 // Only one effect chain can be present in direct output thread and it uses 1918 // the mix buffer as input 1919 if (mType != DIRECT) { 1920 size_t numSamples = mNormalFrameCount * mChannelCount; 1921 buffer = new int16_t[numSamples]; 1922 memset(buffer, 0, numSamples * sizeof(int16_t)); 1923 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1924 ownsBuffer = true; 1925 } 1926 1927 // Attach all tracks with same session ID to this chain. 1928 for (size_t i = 0; i < mTracks.size(); ++i) { 1929 sp<Track> track = mTracks[i]; 1930 if (session == track->sessionId()) { 1931 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1932 buffer); 1933 track->setMainBuffer(buffer); 1934 chain->incTrackCnt(); 1935 } 1936 } 1937 1938 // indicate all active tracks in the chain 1939 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1940 sp<Track> track = mActiveTracks[i].promote(); 1941 if (track == 0) { 1942 continue; 1943 } 1944 if (session == track->sessionId()) { 1945 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1946 chain->incActiveTrackCnt(); 1947 } 1948 } 1949 } 1950 1951 chain->setInBuffer(buffer, ownsBuffer); 1952 chain->setOutBuffer(mMixBuffer); 1953 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1954 // chains list in order to be processed last as it contains output stage effects 1955 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1956 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1957 // after track specific effects and before output stage 1958 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1959 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1960 // Effect chain for other sessions are inserted at beginning of effect 1961 // chains list to be processed before output mix effects. Relative order between other 1962 // sessions is not important 1963 size_t size = mEffectChains.size(); 1964 size_t i = 0; 1965 for (i = 0; i < size; i++) { 1966 if (mEffectChains[i]->sessionId() < session) { 1967 break; 1968 } 1969 } 1970 mEffectChains.insertAt(chain, i); 1971 checkSuspendOnAddEffectChain_l(chain); 1972 1973 return NO_ERROR; 1974} 1975 1976size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1977{ 1978 int session = chain->sessionId(); 1979 1980 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1981 1982 for (size_t i = 0; i < mEffectChains.size(); i++) { 1983 if (chain == mEffectChains[i]) { 1984 mEffectChains.removeAt(i); 1985 // detach all active tracks from the chain 1986 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1987 sp<Track> track = mActiveTracks[i].promote(); 1988 if (track == 0) { 1989 continue; 1990 } 1991 if (session == track->sessionId()) { 1992 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1993 chain.get(), session); 1994 chain->decActiveTrackCnt(); 1995 } 1996 } 1997 1998 // detach all tracks with same session ID from this chain 1999 for (size_t i = 0; i < mTracks.size(); ++i) { 2000 sp<Track> track = mTracks[i]; 2001 if (session == track->sessionId()) { 2002 track->setMainBuffer(mMixBuffer); 2003 chain->decTrackCnt(); 2004 } 2005 } 2006 break; 2007 } 2008 } 2009 return mEffectChains.size(); 2010} 2011 2012status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2013 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2014{ 2015 Mutex::Autolock _l(mLock); 2016 return attachAuxEffect_l(track, EffectId); 2017} 2018 2019status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2020 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2021{ 2022 status_t status = NO_ERROR; 2023 2024 if (EffectId == 0) { 2025 track->setAuxBuffer(0, NULL); 2026 } else { 2027 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2028 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2029 if (effect != 0) { 2030 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2031 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2032 } else { 2033 status = INVALID_OPERATION; 2034 } 2035 } else { 2036 status = BAD_VALUE; 2037 } 2038 } 2039 return status; 2040} 2041 2042void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2043{ 2044 for (size_t i = 0; i < mTracks.size(); ++i) { 2045 sp<Track> track = mTracks[i]; 2046 if (track->auxEffectId() == effectId) { 2047 attachAuxEffect_l(track, 0); 2048 } 2049 } 2050} 2051 2052bool AudioFlinger::PlaybackThread::threadLoop() 2053{ 2054 Vector< sp<Track> > tracksToRemove; 2055 2056 standbyTime = systemTime(); 2057 2058 // MIXER 2059 nsecs_t lastWarning = 0; 2060 2061 // DUPLICATING 2062 // FIXME could this be made local to while loop? 2063 writeFrames = 0; 2064 2065 cacheParameters_l(); 2066 sleepTime = idleSleepTime; 2067 2068 if (mType == MIXER) { 2069 sleepTimeShift = 0; 2070 } 2071 2072 CpuStats cpuStats; 2073 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2074 2075 acquireWakeLock(); 2076 2077 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2078 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2079 // and then that string will be logged at the next convenient opportunity. 2080 const char *logString = NULL; 2081 2082 while (!exitPending()) 2083 { 2084 cpuStats.sample(myName); 2085 2086 Vector< sp<EffectChain> > effectChains; 2087 2088 processConfigEvents(); 2089 2090 { // scope for mLock 2091 2092 Mutex::Autolock _l(mLock); 2093 2094 if (logString != NULL) { 2095 mNBLogWriter->logTimestamp(); 2096 mNBLogWriter->log(logString); 2097 logString = NULL; 2098 } 2099 2100 if (checkForNewParameters_l()) { 2101 cacheParameters_l(); 2102 } 2103 2104 saveOutputTracks(); 2105 2106 if (mSignalPending) { 2107 // A signal was raised while we were unlocked 2108 mSignalPending = false; 2109 } else if (waitingAsyncCallback_l()) { 2110 if (exitPending()) { 2111 break; 2112 } 2113 releaseWakeLock_l(); 2114 ALOGV("wait async completion"); 2115 mWaitWorkCV.wait(mLock); 2116 ALOGV("async completion/wake"); 2117 acquireWakeLock_l(); 2118 if (exitPending()) { 2119 break; 2120 } 2121 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2122 continue; 2123 } 2124 sleepTime = 0; 2125 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2126 isSuspended()) { 2127 // put audio hardware into standby after short delay 2128 if (shouldStandby_l()) { 2129 2130 threadLoop_standby(); 2131 2132 mStandby = true; 2133 } 2134 2135 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2136 // we're about to wait, flush the binder command buffer 2137 IPCThreadState::self()->flushCommands(); 2138 2139 clearOutputTracks(); 2140 2141 if (exitPending()) { 2142 break; 2143 } 2144 2145 releaseWakeLock_l(); 2146 // wait until we have something to do... 2147 ALOGV("%s going to sleep", myName.string()); 2148 mWaitWorkCV.wait(mLock); 2149 ALOGV("%s waking up", myName.string()); 2150 acquireWakeLock_l(); 2151 2152 mMixerStatus = MIXER_IDLE; 2153 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2154 mBytesWritten = 0; 2155 mBytesRemaining = 0; 2156 checkSilentMode_l(); 2157 2158 standbyTime = systemTime() + standbyDelay; 2159 sleepTime = idleSleepTime; 2160 if (mType == MIXER) { 2161 sleepTimeShift = 0; 2162 } 2163 2164 continue; 2165 } 2166 } 2167 2168 // mMixerStatusIgnoringFastTracks is also updated internally 2169 mMixerStatus = prepareTracks_l(&tracksToRemove); 2170 2171 // prevent any changes in effect chain list and in each effect chain 2172 // during mixing and effect process as the audio buffers could be deleted 2173 // or modified if an effect is created or deleted 2174 lockEffectChains_l(effectChains); 2175 } 2176 2177 if (mBytesRemaining == 0) { 2178 mCurrentWriteLength = 0; 2179 if (mMixerStatus == MIXER_TRACKS_READY) { 2180 // threadLoop_mix() sets mCurrentWriteLength 2181 threadLoop_mix(); 2182 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2183 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2184 // threadLoop_sleepTime sets sleepTime to 0 if data 2185 // must be written to HAL 2186 threadLoop_sleepTime(); 2187 if (sleepTime == 0) { 2188 mCurrentWriteLength = mixBufferSize; 2189 } 2190 } 2191 mBytesRemaining = mCurrentWriteLength; 2192 if (isSuspended()) { 2193 sleepTime = suspendSleepTimeUs(); 2194 // simulate write to HAL when suspended 2195 mBytesWritten += mixBufferSize; 2196 mBytesRemaining = 0; 2197 } 2198 2199 // only process effects if we're going to write 2200 if (sleepTime == 0) { 2201 for (size_t i = 0; i < effectChains.size(); i ++) { 2202 effectChains[i]->process_l(); 2203 } 2204 } 2205 } 2206 2207 // enable changes in effect chain 2208 unlockEffectChains(effectChains); 2209 2210 if (!waitingAsyncCallback()) { 2211 // sleepTime == 0 means we must write to audio hardware 2212 if (sleepTime == 0) { 2213 if (mBytesRemaining) { 2214 ssize_t ret = threadLoop_write(); 2215 if (ret < 0) { 2216 mBytesRemaining = 0; 2217 } else { 2218 mBytesWritten += ret; 2219 mBytesRemaining -= ret; 2220 } 2221 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2222 (mMixerStatus == MIXER_DRAIN_ALL)) { 2223 threadLoop_drain(); 2224 } 2225if (mType == MIXER) { 2226 // write blocked detection 2227 nsecs_t now = systemTime(); 2228 nsecs_t delta = now - mLastWriteTime; 2229 if (!mStandby && delta > maxPeriod) { 2230 mNumDelayedWrites++; 2231 if ((now - lastWarning) > kWarningThrottleNs) { 2232 ATRACE_NAME("underrun"); 2233 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2234 ns2ms(delta), mNumDelayedWrites, this); 2235 lastWarning = now; 2236 } 2237 } 2238} 2239 2240 mStandby = false; 2241 } else { 2242 usleep(sleepTime); 2243 } 2244 } 2245 2246 // Finally let go of removed track(s), without the lock held 2247 // since we can't guarantee the destructors won't acquire that 2248 // same lock. This will also mutate and push a new fast mixer state. 2249 threadLoop_removeTracks(tracksToRemove); 2250 tracksToRemove.clear(); 2251 2252 // FIXME I don't understand the need for this here; 2253 // it was in the original code but maybe the 2254 // assignment in saveOutputTracks() makes this unnecessary? 2255 clearOutputTracks(); 2256 2257 // Effect chains will be actually deleted here if they were removed from 2258 // mEffectChains list during mixing or effects processing 2259 effectChains.clear(); 2260 2261 // FIXME Note that the above .clear() is no longer necessary since effectChains 2262 // is now local to this block, but will keep it for now (at least until merge done). 2263 } 2264 2265 threadLoop_exit(); 2266 2267 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2268 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2269 // put output stream into standby mode 2270 if (!mStandby) { 2271 mOutput->stream->common.standby(&mOutput->stream->common); 2272 } 2273 } 2274 2275 releaseWakeLock(); 2276 2277 ALOGV("Thread %p type %d exiting", this, mType); 2278 return false; 2279} 2280 2281// removeTracks_l() must be called with ThreadBase::mLock held 2282void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2283{ 2284 size_t count = tracksToRemove.size(); 2285 if (CC_UNLIKELY(count)) { 2286 for (size_t i=0 ; i<count ; i++) { 2287 const sp<Track>& track = tracksToRemove.itemAt(i); 2288 mActiveTracks.remove(track); 2289 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2290 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2291 if (chain != 0) { 2292 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2293 track->sessionId()); 2294 chain->decActiveTrackCnt(); 2295 } 2296 if (track->isTerminated()) { 2297 removeTrack_l(track); 2298 } 2299 } 2300 } 2301 2302} 2303 2304// ---------------------------------------------------------------------------- 2305 2306AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2307 audio_io_handle_t id, audio_devices_t device, type_t type) 2308 : PlaybackThread(audioFlinger, output, id, device, type), 2309 // mAudioMixer below 2310 // mFastMixer below 2311 mFastMixerFutex(0) 2312 // mOutputSink below 2313 // mPipeSink below 2314 // mNormalSink below 2315{ 2316 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2317 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2318 "mFrameCount=%d, mNormalFrameCount=%d", 2319 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2320 mNormalFrameCount); 2321 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2322 2323 // FIXME - Current mixer implementation only supports stereo output 2324 if (mChannelCount != FCC_2) { 2325 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2326 } 2327 2328 // create an NBAIO sink for the HAL output stream, and negotiate 2329 mOutputSink = new AudioStreamOutSink(output->stream); 2330 size_t numCounterOffers = 0; 2331 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2332 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2333 ALOG_ASSERT(index == 0); 2334 2335 // initialize fast mixer depending on configuration 2336 bool initFastMixer; 2337 switch (kUseFastMixer) { 2338 case FastMixer_Never: 2339 initFastMixer = false; 2340 break; 2341 case FastMixer_Always: 2342 initFastMixer = true; 2343 break; 2344 case FastMixer_Static: 2345 case FastMixer_Dynamic: 2346 initFastMixer = mFrameCount < mNormalFrameCount; 2347 break; 2348 } 2349 if (initFastMixer) { 2350 2351 // create a MonoPipe to connect our submix to FastMixer 2352 NBAIO_Format format = mOutputSink->format(); 2353 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2354 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2355 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2356 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2357 const NBAIO_Format offers[1] = {format}; 2358 size_t numCounterOffers = 0; 2359 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2360 ALOG_ASSERT(index == 0); 2361 monoPipe->setAvgFrames((mScreenState & 1) ? 2362 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2363 mPipeSink = monoPipe; 2364 2365#ifdef TEE_SINK 2366 if (mTeeSinkOutputEnabled) { 2367 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2368 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2369 numCounterOffers = 0; 2370 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2371 ALOG_ASSERT(index == 0); 2372 mTeeSink = teeSink; 2373 PipeReader *teeSource = new PipeReader(*teeSink); 2374 numCounterOffers = 0; 2375 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2376 ALOG_ASSERT(index == 0); 2377 mTeeSource = teeSource; 2378 } 2379#endif 2380 2381 // create fast mixer and configure it initially with just one fast track for our submix 2382 mFastMixer = new FastMixer(); 2383 FastMixerStateQueue *sq = mFastMixer->sq(); 2384#ifdef STATE_QUEUE_DUMP 2385 sq->setObserverDump(&mStateQueueObserverDump); 2386 sq->setMutatorDump(&mStateQueueMutatorDump); 2387#endif 2388 FastMixerState *state = sq->begin(); 2389 FastTrack *fastTrack = &state->mFastTracks[0]; 2390 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2391 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2392 fastTrack->mVolumeProvider = NULL; 2393 fastTrack->mGeneration++; 2394 state->mFastTracksGen++; 2395 state->mTrackMask = 1; 2396 // fast mixer will use the HAL output sink 2397 state->mOutputSink = mOutputSink.get(); 2398 state->mOutputSinkGen++; 2399 state->mFrameCount = mFrameCount; 2400 state->mCommand = FastMixerState::COLD_IDLE; 2401 // already done in constructor initialization list 2402 //mFastMixerFutex = 0; 2403 state->mColdFutexAddr = &mFastMixerFutex; 2404 state->mColdGen++; 2405 state->mDumpState = &mFastMixerDumpState; 2406#ifdef TEE_SINK 2407 state->mTeeSink = mTeeSink.get(); 2408#endif 2409 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2410 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2411 sq->end(); 2412 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2413 2414 // start the fast mixer 2415 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2416 pid_t tid = mFastMixer->getTid(); 2417 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2418 if (err != 0) { 2419 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2420 kPriorityFastMixer, getpid_cached, tid, err); 2421 } 2422 2423#ifdef AUDIO_WATCHDOG 2424 // create and start the watchdog 2425 mAudioWatchdog = new AudioWatchdog(); 2426 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2427 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2428 tid = mAudioWatchdog->getTid(); 2429 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2430 if (err != 0) { 2431 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2432 kPriorityFastMixer, getpid_cached, tid, err); 2433 } 2434#endif 2435 2436 } else { 2437 mFastMixer = NULL; 2438 } 2439 2440 switch (kUseFastMixer) { 2441 case FastMixer_Never: 2442 case FastMixer_Dynamic: 2443 mNormalSink = mOutputSink; 2444 break; 2445 case FastMixer_Always: 2446 mNormalSink = mPipeSink; 2447 break; 2448 case FastMixer_Static: 2449 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2450 break; 2451 } 2452} 2453 2454AudioFlinger::MixerThread::~MixerThread() 2455{ 2456 if (mFastMixer != NULL) { 2457 FastMixerStateQueue *sq = mFastMixer->sq(); 2458 FastMixerState *state = sq->begin(); 2459 if (state->mCommand == FastMixerState::COLD_IDLE) { 2460 int32_t old = android_atomic_inc(&mFastMixerFutex); 2461 if (old == -1) { 2462 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2463 } 2464 } 2465 state->mCommand = FastMixerState::EXIT; 2466 sq->end(); 2467 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2468 mFastMixer->join(); 2469 // Though the fast mixer thread has exited, it's state queue is still valid. 2470 // We'll use that extract the final state which contains one remaining fast track 2471 // corresponding to our sub-mix. 2472 state = sq->begin(); 2473 ALOG_ASSERT(state->mTrackMask == 1); 2474 FastTrack *fastTrack = &state->mFastTracks[0]; 2475 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2476 delete fastTrack->mBufferProvider; 2477 sq->end(false /*didModify*/); 2478 delete mFastMixer; 2479#ifdef AUDIO_WATCHDOG 2480 if (mAudioWatchdog != 0) { 2481 mAudioWatchdog->requestExit(); 2482 mAudioWatchdog->requestExitAndWait(); 2483 mAudioWatchdog.clear(); 2484 } 2485#endif 2486 } 2487 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2488 delete mAudioMixer; 2489} 2490 2491 2492uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2493{ 2494 if (mFastMixer != NULL) { 2495 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2496 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2497 } 2498 return latency; 2499} 2500 2501 2502void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2503{ 2504 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2505} 2506 2507ssize_t AudioFlinger::MixerThread::threadLoop_write() 2508{ 2509 // FIXME we should only do one push per cycle; confirm this is true 2510 // Start the fast mixer if it's not already running 2511 if (mFastMixer != NULL) { 2512 FastMixerStateQueue *sq = mFastMixer->sq(); 2513 FastMixerState *state = sq->begin(); 2514 if (state->mCommand != FastMixerState::MIX_WRITE && 2515 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2516 if (state->mCommand == FastMixerState::COLD_IDLE) { 2517 int32_t old = android_atomic_inc(&mFastMixerFutex); 2518 if (old == -1) { 2519 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2520 } 2521#ifdef AUDIO_WATCHDOG 2522 if (mAudioWatchdog != 0) { 2523 mAudioWatchdog->resume(); 2524 } 2525#endif 2526 } 2527 state->mCommand = FastMixerState::MIX_WRITE; 2528 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2529 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2530 sq->end(); 2531 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2532 if (kUseFastMixer == FastMixer_Dynamic) { 2533 mNormalSink = mPipeSink; 2534 } 2535 } else { 2536 sq->end(false /*didModify*/); 2537 } 2538 } 2539 return PlaybackThread::threadLoop_write(); 2540} 2541 2542void AudioFlinger::MixerThread::threadLoop_standby() 2543{ 2544 // Idle the fast mixer if it's currently running 2545 if (mFastMixer != NULL) { 2546 FastMixerStateQueue *sq = mFastMixer->sq(); 2547 FastMixerState *state = sq->begin(); 2548 if (!(state->mCommand & FastMixerState::IDLE)) { 2549 state->mCommand = FastMixerState::COLD_IDLE; 2550 state->mColdFutexAddr = &mFastMixerFutex; 2551 state->mColdGen++; 2552 mFastMixerFutex = 0; 2553 sq->end(); 2554 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2555 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2556 if (kUseFastMixer == FastMixer_Dynamic) { 2557 mNormalSink = mOutputSink; 2558 } 2559#ifdef AUDIO_WATCHDOG 2560 if (mAudioWatchdog != 0) { 2561 mAudioWatchdog->pause(); 2562 } 2563#endif 2564 } else { 2565 sq->end(false /*didModify*/); 2566 } 2567 } 2568 PlaybackThread::threadLoop_standby(); 2569} 2570 2571// Empty implementation for standard mixer 2572// Overridden for offloaded playback 2573void AudioFlinger::PlaybackThread::flushOutput_l() 2574{ 2575} 2576 2577bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2578{ 2579 return false; 2580} 2581 2582bool AudioFlinger::PlaybackThread::shouldStandby_l() 2583{ 2584 return !mStandby; 2585} 2586 2587bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2588{ 2589 Mutex::Autolock _l(mLock); 2590 return waitingAsyncCallback_l(); 2591} 2592 2593// shared by MIXER and DIRECT, overridden by DUPLICATING 2594void AudioFlinger::PlaybackThread::threadLoop_standby() 2595{ 2596 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2597 mOutput->stream->common.standby(&mOutput->stream->common); 2598 if (mUseAsyncWrite != 0) { 2599 mWriteBlocked = false; 2600 mDraining = false; 2601 ALOG_ASSERT(mCallbackThread != 0); 2602 mCallbackThread->setWriteBlocked(false); 2603 mCallbackThread->setDraining(false); 2604 } 2605} 2606 2607void AudioFlinger::MixerThread::threadLoop_mix() 2608{ 2609 // obtain the presentation timestamp of the next output buffer 2610 int64_t pts; 2611 status_t status = INVALID_OPERATION; 2612 2613 if (mNormalSink != 0) { 2614 status = mNormalSink->getNextWriteTimestamp(&pts); 2615 } else { 2616 status = mOutputSink->getNextWriteTimestamp(&pts); 2617 } 2618 2619 if (status != NO_ERROR) { 2620 pts = AudioBufferProvider::kInvalidPTS; 2621 } 2622 2623 // mix buffers... 2624 mAudioMixer->process(pts); 2625 mCurrentWriteLength = mixBufferSize; 2626 // increase sleep time progressively when application underrun condition clears. 2627 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2628 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2629 // such that we would underrun the audio HAL. 2630 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2631 sleepTimeShift--; 2632 } 2633 sleepTime = 0; 2634 standbyTime = systemTime() + standbyDelay; 2635 //TODO: delay standby when effects have a tail 2636} 2637 2638void AudioFlinger::MixerThread::threadLoop_sleepTime() 2639{ 2640 // If no tracks are ready, sleep once for the duration of an output 2641 // buffer size, then write 0s to the output 2642 if (sleepTime == 0) { 2643 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2644 sleepTime = activeSleepTime >> sleepTimeShift; 2645 if (sleepTime < kMinThreadSleepTimeUs) { 2646 sleepTime = kMinThreadSleepTimeUs; 2647 } 2648 // reduce sleep time in case of consecutive application underruns to avoid 2649 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2650 // duration we would end up writing less data than needed by the audio HAL if 2651 // the condition persists. 2652 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2653 sleepTimeShift++; 2654 } 2655 } else { 2656 sleepTime = idleSleepTime; 2657 } 2658 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2659 memset (mMixBuffer, 0, mixBufferSize); 2660 sleepTime = 0; 2661 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2662 "anticipated start"); 2663 } 2664 // TODO add standby time extension fct of effect tail 2665} 2666 2667// prepareTracks_l() must be called with ThreadBase::mLock held 2668AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2669 Vector< sp<Track> > *tracksToRemove) 2670{ 2671 2672 mixer_state mixerStatus = MIXER_IDLE; 2673 // find out which tracks need to be processed 2674 size_t count = mActiveTracks.size(); 2675 size_t mixedTracks = 0; 2676 size_t tracksWithEffect = 0; 2677 // counts only _active_ fast tracks 2678 size_t fastTracks = 0; 2679 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2680 2681 float masterVolume = mMasterVolume; 2682 bool masterMute = mMasterMute; 2683 2684 if (masterMute) { 2685 masterVolume = 0; 2686 } 2687 // Delegate master volume control to effect in output mix effect chain if needed 2688 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2689 if (chain != 0) { 2690 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2691 chain->setVolume_l(&v, &v); 2692 masterVolume = (float)((v + (1 << 23)) >> 24); 2693 chain.clear(); 2694 } 2695 2696 // prepare a new state to push 2697 FastMixerStateQueue *sq = NULL; 2698 FastMixerState *state = NULL; 2699 bool didModify = false; 2700 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2701 if (mFastMixer != NULL) { 2702 sq = mFastMixer->sq(); 2703 state = sq->begin(); 2704 } 2705 2706 for (size_t i=0 ; i<count ; i++) { 2707 sp<Track> t = mActiveTracks[i].promote(); 2708 if (t == 0) { 2709 continue; 2710 } 2711 2712 // this const just means the local variable doesn't change 2713 Track* const track = t.get(); 2714 2715 // process fast tracks 2716 if (track->isFastTrack()) { 2717 2718 // It's theoretically possible (though unlikely) for a fast track to be created 2719 // and then removed within the same normal mix cycle. This is not a problem, as 2720 // the track never becomes active so it's fast mixer slot is never touched. 2721 // The converse, of removing an (active) track and then creating a new track 2722 // at the identical fast mixer slot within the same normal mix cycle, 2723 // is impossible because the slot isn't marked available until the end of each cycle. 2724 int j = track->mFastIndex; 2725 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2726 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2727 FastTrack *fastTrack = &state->mFastTracks[j]; 2728 2729 // Determine whether the track is currently in underrun condition, 2730 // and whether it had a recent underrun. 2731 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2732 FastTrackUnderruns underruns = ftDump->mUnderruns; 2733 uint32_t recentFull = (underruns.mBitFields.mFull - 2734 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2735 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2736 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2737 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2738 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2739 uint32_t recentUnderruns = recentPartial + recentEmpty; 2740 track->mObservedUnderruns = underruns; 2741 // don't count underruns that occur while stopping or pausing 2742 // or stopped which can occur when flush() is called while active 2743 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2744 track->mUnderrunCount += recentUnderruns; 2745 } 2746 2747 // This is similar to the state machine for normal tracks, 2748 // with a few modifications for fast tracks. 2749 bool isActive = true; 2750 switch (track->mState) { 2751 case TrackBase::STOPPING_1: 2752 // track stays active in STOPPING_1 state until first underrun 2753 if (recentUnderruns > 0 || track->isTerminated()) { 2754 track->mState = TrackBase::STOPPING_2; 2755 } 2756 break; 2757 case TrackBase::PAUSING: 2758 // ramp down is not yet implemented 2759 track->setPaused(); 2760 break; 2761 case TrackBase::RESUMING: 2762 // ramp up is not yet implemented 2763 track->mState = TrackBase::ACTIVE; 2764 break; 2765 case TrackBase::ACTIVE: 2766 if (recentFull > 0 || recentPartial > 0) { 2767 // track has provided at least some frames recently: reset retry count 2768 track->mRetryCount = kMaxTrackRetries; 2769 } 2770 if (recentUnderruns == 0) { 2771 // no recent underruns: stay active 2772 break; 2773 } 2774 // there has recently been an underrun of some kind 2775 if (track->sharedBuffer() == 0) { 2776 // were any of the recent underruns "empty" (no frames available)? 2777 if (recentEmpty == 0) { 2778 // no, then ignore the partial underruns as they are allowed indefinitely 2779 break; 2780 } 2781 // there has recently been an "empty" underrun: decrement the retry counter 2782 if (--(track->mRetryCount) > 0) { 2783 break; 2784 } 2785 // indicate to client process that the track was disabled because of underrun; 2786 // it will then automatically call start() when data is available 2787 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2788 // remove from active list, but state remains ACTIVE [confusing but true] 2789 isActive = false; 2790 break; 2791 } 2792 // fall through 2793 case TrackBase::STOPPING_2: 2794 case TrackBase::PAUSED: 2795 case TrackBase::STOPPED: 2796 case TrackBase::FLUSHED: // flush() while active 2797 // Check for presentation complete if track is inactive 2798 // We have consumed all the buffers of this track. 2799 // This would be incomplete if we auto-paused on underrun 2800 { 2801 size_t audioHALFrames = 2802 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2803 size_t framesWritten = mBytesWritten / mFrameSize; 2804 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2805 // track stays in active list until presentation is complete 2806 break; 2807 } 2808 } 2809 if (track->isStopping_2()) { 2810 track->mState = TrackBase::STOPPED; 2811 } 2812 if (track->isStopped()) { 2813 // Can't reset directly, as fast mixer is still polling this track 2814 // track->reset(); 2815 // So instead mark this track as needing to be reset after push with ack 2816 resetMask |= 1 << i; 2817 } 2818 isActive = false; 2819 break; 2820 case TrackBase::IDLE: 2821 default: 2822 LOG_FATAL("unexpected track state %d", track->mState); 2823 } 2824 2825 if (isActive) { 2826 // was it previously inactive? 2827 if (!(state->mTrackMask & (1 << j))) { 2828 ExtendedAudioBufferProvider *eabp = track; 2829 VolumeProvider *vp = track; 2830 fastTrack->mBufferProvider = eabp; 2831 fastTrack->mVolumeProvider = vp; 2832 fastTrack->mSampleRate = track->mSampleRate; 2833 fastTrack->mChannelMask = track->mChannelMask; 2834 fastTrack->mGeneration++; 2835 state->mTrackMask |= 1 << j; 2836 didModify = true; 2837 // no acknowledgement required for newly active tracks 2838 } 2839 // cache the combined master volume and stream type volume for fast mixer; this 2840 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2841 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2842 ++fastTracks; 2843 } else { 2844 // was it previously active? 2845 if (state->mTrackMask & (1 << j)) { 2846 fastTrack->mBufferProvider = NULL; 2847 fastTrack->mGeneration++; 2848 state->mTrackMask &= ~(1 << j); 2849 didModify = true; 2850 // If any fast tracks were removed, we must wait for acknowledgement 2851 // because we're about to decrement the last sp<> on those tracks. 2852 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2853 } else { 2854 LOG_FATAL("fast track %d should have been active", j); 2855 } 2856 tracksToRemove->add(track); 2857 // Avoids a misleading display in dumpsys 2858 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2859 } 2860 continue; 2861 } 2862 2863 { // local variable scope to avoid goto warning 2864 2865 audio_track_cblk_t* cblk = track->cblk(); 2866 2867 // The first time a track is added we wait 2868 // for all its buffers to be filled before processing it 2869 int name = track->name(); 2870 // make sure that we have enough frames to mix one full buffer. 2871 // enforce this condition only once to enable draining the buffer in case the client 2872 // app does not call stop() and relies on underrun to stop: 2873 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2874 // during last round 2875 size_t desiredFrames; 2876 if (t->sampleRate() == mSampleRate) { 2877 desiredFrames = mNormalFrameCount; 2878 } else { 2879 // +1 for rounding and +1 for additional sample needed for interpolation 2880 desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2881 // add frames already consumed but not yet released by the resampler 2882 // because cblk->framesReady() will include these frames 2883 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2884 // the minimum track buffer size is normally twice the number of frames necessary 2885 // to fill one buffer and the resampler should not leave more than one buffer worth 2886 // of unreleased frames after each pass, but just in case... 2887 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2888 } 2889 uint32_t minFrames = 1; 2890 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2891 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2892 minFrames = desiredFrames; 2893 } 2894 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2895 size_t framesReady; 2896 if (track->sharedBuffer() == 0) { 2897 framesReady = track->framesReady(); 2898 } else if (track->isStopped()) { 2899 framesReady = 0; 2900 } else { 2901 framesReady = 1; 2902 } 2903 if ((framesReady >= minFrames) && track->isReady() && 2904 !track->isPaused() && !track->isTerminated()) 2905 { 2906 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->server, this); 2907 2908 mixedTracks++; 2909 2910 // track->mainBuffer() != mMixBuffer means there is an effect chain 2911 // connected to the track 2912 chain.clear(); 2913 if (track->mainBuffer() != mMixBuffer) { 2914 chain = getEffectChain_l(track->sessionId()); 2915 // Delegate volume control to effect in track effect chain if needed 2916 if (chain != 0) { 2917 tracksWithEffect++; 2918 } else { 2919 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2920 "session %d", 2921 name, track->sessionId()); 2922 } 2923 } 2924 2925 2926 int param = AudioMixer::VOLUME; 2927 if (track->mFillingUpStatus == Track::FS_FILLED) { 2928 // no ramp for the first volume setting 2929 track->mFillingUpStatus = Track::FS_ACTIVE; 2930 if (track->mState == TrackBase::RESUMING) { 2931 track->mState = TrackBase::ACTIVE; 2932 param = AudioMixer::RAMP_VOLUME; 2933 } 2934 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2935 } else if (cblk->server != 0) { 2936 // If the track is stopped before the first frame was mixed, 2937 // do not apply ramp 2938 param = AudioMixer::RAMP_VOLUME; 2939 } 2940 2941 // compute volume for this track 2942 uint32_t vl, vr, va; 2943 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2944 vl = vr = va = 0; 2945 if (track->isPausing()) { 2946 track->setPaused(); 2947 } 2948 } else { 2949 2950 // read original volumes with volume control 2951 float typeVolume = mStreamTypes[track->streamType()].volume; 2952 float v = masterVolume * typeVolume; 2953 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2954 uint32_t vlr = proxy->getVolumeLR(); 2955 vl = vlr & 0xFFFF; 2956 vr = vlr >> 16; 2957 // track volumes come from shared memory, so can't be trusted and must be clamped 2958 if (vl > MAX_GAIN_INT) { 2959 ALOGV("Track left volume out of range: %04X", vl); 2960 vl = MAX_GAIN_INT; 2961 } 2962 if (vr > MAX_GAIN_INT) { 2963 ALOGV("Track right volume out of range: %04X", vr); 2964 vr = MAX_GAIN_INT; 2965 } 2966 // now apply the master volume and stream type volume 2967 vl = (uint32_t)(v * vl) << 12; 2968 vr = (uint32_t)(v * vr) << 12; 2969 // assuming master volume and stream type volume each go up to 1.0, 2970 // vl and vr are now in 8.24 format 2971 2972 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2973 // send level comes from shared memory and so may be corrupt 2974 if (sendLevel > MAX_GAIN_INT) { 2975 ALOGV("Track send level out of range: %04X", sendLevel); 2976 sendLevel = MAX_GAIN_INT; 2977 } 2978 va = (uint32_t)(v * sendLevel); 2979 } 2980 2981 // Delegate volume control to effect in track effect chain if needed 2982 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2983 // Do not ramp volume if volume is controlled by effect 2984 param = AudioMixer::VOLUME; 2985 track->mHasVolumeController = true; 2986 } else { 2987 // force no volume ramp when volume controller was just disabled or removed 2988 // from effect chain to avoid volume spike 2989 if (track->mHasVolumeController) { 2990 param = AudioMixer::VOLUME; 2991 } 2992 track->mHasVolumeController = false; 2993 } 2994 2995 // Convert volumes from 8.24 to 4.12 format 2996 // This additional clamping is needed in case chain->setVolume_l() overshot 2997 vl = (vl + (1 << 11)) >> 12; 2998 if (vl > MAX_GAIN_INT) { 2999 vl = MAX_GAIN_INT; 3000 } 3001 vr = (vr + (1 << 11)) >> 12; 3002 if (vr > MAX_GAIN_INT) { 3003 vr = MAX_GAIN_INT; 3004 } 3005 3006 if (va > MAX_GAIN_INT) { 3007 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3008 } 3009 3010 // XXX: these things DON'T need to be done each time 3011 mAudioMixer->setBufferProvider(name, track); 3012 mAudioMixer->enable(name); 3013 3014 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3015 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3016 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3017 mAudioMixer->setParameter( 3018 name, 3019 AudioMixer::TRACK, 3020 AudioMixer::FORMAT, (void *)track->format()); 3021 mAudioMixer->setParameter( 3022 name, 3023 AudioMixer::TRACK, 3024 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3025 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3026 uint32_t maxSampleRate = mSampleRate * 2; 3027 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3028 if (reqSampleRate == 0) { 3029 reqSampleRate = mSampleRate; 3030 } else if (reqSampleRate > maxSampleRate) { 3031 reqSampleRate = maxSampleRate; 3032 } 3033 mAudioMixer->setParameter( 3034 name, 3035 AudioMixer::RESAMPLE, 3036 AudioMixer::SAMPLE_RATE, 3037 (void *)reqSampleRate); 3038 mAudioMixer->setParameter( 3039 name, 3040 AudioMixer::TRACK, 3041 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3042 mAudioMixer->setParameter( 3043 name, 3044 AudioMixer::TRACK, 3045 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3046 3047 // reset retry count 3048 track->mRetryCount = kMaxTrackRetries; 3049 3050 // If one track is ready, set the mixer ready if: 3051 // - the mixer was not ready during previous round OR 3052 // - no other track is not ready 3053 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3054 mixerStatus != MIXER_TRACKS_ENABLED) { 3055 mixerStatus = MIXER_TRACKS_READY; 3056 } 3057 } else { 3058 // only implemented for normal tracks, not fast tracks 3059 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3060 // we missed desiredFrames whatever the actual number of frames missing was 3061 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 3062 // FIXME also wake futex so that underrun is noticed more quickly 3063 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 3064 } 3065 // clear effect chain input buffer if an active track underruns to avoid sending 3066 // previous audio buffer again to effects 3067 chain = getEffectChain_l(track->sessionId()); 3068 if (chain != 0) { 3069 chain->clearInputBuffer(); 3070 } 3071 3072 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->server, this); 3073 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3074 track->isStopped() || track->isPaused()) { 3075 // We have consumed all the buffers of this track. 3076 // Remove it from the list of active tracks. 3077 // TODO: use actual buffer filling status instead of latency when available from 3078 // audio HAL 3079 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3080 size_t framesWritten = mBytesWritten / mFrameSize; 3081 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3082 if (track->isStopped()) { 3083 track->reset(); 3084 } 3085 tracksToRemove->add(track); 3086 } 3087 } else { 3088 track->mUnderrunCount++; 3089 // No buffers for this track. Give it a few chances to 3090 // fill a buffer, then remove it from active list. 3091 if (--(track->mRetryCount) <= 0) { 3092 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3093 tracksToRemove->add(track); 3094 // indicate to client process that the track was disabled because of underrun; 3095 // it will then automatically call start() when data is available 3096 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3097 // If one track is not ready, mark the mixer also not ready if: 3098 // - the mixer was ready during previous round OR 3099 // - no other track is ready 3100 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3101 mixerStatus != MIXER_TRACKS_READY) { 3102 mixerStatus = MIXER_TRACKS_ENABLED; 3103 } 3104 } 3105 mAudioMixer->disable(name); 3106 } 3107 3108 } // local variable scope to avoid goto warning 3109track_is_ready: ; 3110 3111 } 3112 3113 // Push the new FastMixer state if necessary 3114 bool pauseAudioWatchdog = false; 3115 if (didModify) { 3116 state->mFastTracksGen++; 3117 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3118 if (kUseFastMixer == FastMixer_Dynamic && 3119 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3120 state->mCommand = FastMixerState::COLD_IDLE; 3121 state->mColdFutexAddr = &mFastMixerFutex; 3122 state->mColdGen++; 3123 mFastMixerFutex = 0; 3124 if (kUseFastMixer == FastMixer_Dynamic) { 3125 mNormalSink = mOutputSink; 3126 } 3127 // If we go into cold idle, need to wait for acknowledgement 3128 // so that fast mixer stops doing I/O. 3129 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3130 pauseAudioWatchdog = true; 3131 } 3132 } 3133 if (sq != NULL) { 3134 sq->end(didModify); 3135 sq->push(block); 3136 } 3137#ifdef AUDIO_WATCHDOG 3138 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3139 mAudioWatchdog->pause(); 3140 } 3141#endif 3142 3143 // Now perform the deferred reset on fast tracks that have stopped 3144 while (resetMask != 0) { 3145 size_t i = __builtin_ctz(resetMask); 3146 ALOG_ASSERT(i < count); 3147 resetMask &= ~(1 << i); 3148 sp<Track> t = mActiveTracks[i].promote(); 3149 if (t == 0) { 3150 continue; 3151 } 3152 Track* track = t.get(); 3153 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3154 track->reset(); 3155 } 3156 3157 // remove all the tracks that need to be... 3158 removeTracks_l(*tracksToRemove); 3159 3160 // mix buffer must be cleared if all tracks are connected to an 3161 // effect chain as in this case the mixer will not write to 3162 // mix buffer and track effects will accumulate into it 3163 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3164 (mixedTracks == 0 && fastTracks > 0))) { 3165 // FIXME as a performance optimization, should remember previous zero status 3166 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3167 } 3168 3169 // if any fast tracks, then status is ready 3170 mMixerStatusIgnoringFastTracks = mixerStatus; 3171 if (fastTracks > 0) { 3172 mixerStatus = MIXER_TRACKS_READY; 3173 } 3174 return mixerStatus; 3175} 3176 3177// getTrackName_l() must be called with ThreadBase::mLock held 3178int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3179{ 3180 return mAudioMixer->getTrackName(channelMask, sessionId); 3181} 3182 3183// deleteTrackName_l() must be called with ThreadBase::mLock held 3184void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3185{ 3186 ALOGV("remove track (%d) and delete from mixer", name); 3187 mAudioMixer->deleteTrackName(name); 3188} 3189 3190// checkForNewParameters_l() must be called with ThreadBase::mLock held 3191bool AudioFlinger::MixerThread::checkForNewParameters_l() 3192{ 3193 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3194 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3195 bool reconfig = false; 3196 3197 while (!mNewParameters.isEmpty()) { 3198 3199 if (mFastMixer != NULL) { 3200 FastMixerStateQueue *sq = mFastMixer->sq(); 3201 FastMixerState *state = sq->begin(); 3202 if (!(state->mCommand & FastMixerState::IDLE)) { 3203 previousCommand = state->mCommand; 3204 state->mCommand = FastMixerState::HOT_IDLE; 3205 sq->end(); 3206 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3207 } else { 3208 sq->end(false /*didModify*/); 3209 } 3210 } 3211 3212 status_t status = NO_ERROR; 3213 String8 keyValuePair = mNewParameters[0]; 3214 AudioParameter param = AudioParameter(keyValuePair); 3215 int value; 3216 3217 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3218 reconfig = true; 3219 } 3220 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3221 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3222 status = BAD_VALUE; 3223 } else { 3224 reconfig = true; 3225 } 3226 } 3227 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3228 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3229 status = BAD_VALUE; 3230 } else { 3231 reconfig = true; 3232 } 3233 } 3234 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3235 // do not accept frame count changes if tracks are open as the track buffer 3236 // size depends on frame count and correct behavior would not be guaranteed 3237 // if frame count is changed after track creation 3238 if (!mTracks.isEmpty()) { 3239 status = INVALID_OPERATION; 3240 } else { 3241 reconfig = true; 3242 } 3243 } 3244 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3245#ifdef ADD_BATTERY_DATA 3246 // when changing the audio output device, call addBatteryData to notify 3247 // the change 3248 if (mOutDevice != value) { 3249 uint32_t params = 0; 3250 // check whether speaker is on 3251 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3252 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3253 } 3254 3255 audio_devices_t deviceWithoutSpeaker 3256 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3257 // check if any other device (except speaker) is on 3258 if (value & deviceWithoutSpeaker ) { 3259 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3260 } 3261 3262 if (params != 0) { 3263 addBatteryData(params); 3264 } 3265 } 3266#endif 3267 3268 // forward device change to effects that have requested to be 3269 // aware of attached audio device. 3270 if (value != AUDIO_DEVICE_NONE) { 3271 mOutDevice = value; 3272 for (size_t i = 0; i < mEffectChains.size(); i++) { 3273 mEffectChains[i]->setDevice_l(mOutDevice); 3274 } 3275 } 3276 } 3277 3278 if (status == NO_ERROR) { 3279 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3280 keyValuePair.string()); 3281 if (!mStandby && status == INVALID_OPERATION) { 3282 mOutput->stream->common.standby(&mOutput->stream->common); 3283 mStandby = true; 3284 mBytesWritten = 0; 3285 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3286 keyValuePair.string()); 3287 } 3288 if (status == NO_ERROR && reconfig) { 3289 readOutputParameters(); 3290 delete mAudioMixer; 3291 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3292 for (size_t i = 0; i < mTracks.size() ; i++) { 3293 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3294 if (name < 0) { 3295 break; 3296 } 3297 mTracks[i]->mName = name; 3298 } 3299 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3300 } 3301 } 3302 3303 mNewParameters.removeAt(0); 3304 3305 mParamStatus = status; 3306 mParamCond.signal(); 3307 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3308 // already timed out waiting for the status and will never signal the condition. 3309 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3310 } 3311 3312 if (!(previousCommand & FastMixerState::IDLE)) { 3313 ALOG_ASSERT(mFastMixer != NULL); 3314 FastMixerStateQueue *sq = mFastMixer->sq(); 3315 FastMixerState *state = sq->begin(); 3316 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3317 state->mCommand = previousCommand; 3318 sq->end(); 3319 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3320 } 3321 3322 return reconfig; 3323} 3324 3325 3326void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3327{ 3328 const size_t SIZE = 256; 3329 char buffer[SIZE]; 3330 String8 result; 3331 3332 PlaybackThread::dumpInternals(fd, args); 3333 3334 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3335 result.append(buffer); 3336 write(fd, result.string(), result.size()); 3337 3338 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3339 const FastMixerDumpState copy(mFastMixerDumpState); 3340 copy.dump(fd); 3341 3342#ifdef STATE_QUEUE_DUMP 3343 // Similar for state queue 3344 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3345 observerCopy.dump(fd); 3346 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3347 mutatorCopy.dump(fd); 3348#endif 3349 3350#ifdef TEE_SINK 3351 // Write the tee output to a .wav file 3352 dumpTee(fd, mTeeSource, mId); 3353#endif 3354 3355#ifdef AUDIO_WATCHDOG 3356 if (mAudioWatchdog != 0) { 3357 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3358 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3359 wdCopy.dump(fd); 3360 } 3361#endif 3362} 3363 3364uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3365{ 3366 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3367} 3368 3369uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3370{ 3371 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3372} 3373 3374void AudioFlinger::MixerThread::cacheParameters_l() 3375{ 3376 PlaybackThread::cacheParameters_l(); 3377 3378 // FIXME: Relaxed timing because of a certain device that can't meet latency 3379 // Should be reduced to 2x after the vendor fixes the driver issue 3380 // increase threshold again due to low power audio mode. The way this warning 3381 // threshold is calculated and its usefulness should be reconsidered anyway. 3382 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3383} 3384 3385// ---------------------------------------------------------------------------- 3386 3387AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3388 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3389 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3390 // mLeftVolFloat, mRightVolFloat 3391{ 3392} 3393 3394AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3395 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3396 ThreadBase::type_t type) 3397 : PlaybackThread(audioFlinger, output, id, device, type) 3398 // mLeftVolFloat, mRightVolFloat 3399{ 3400} 3401 3402AudioFlinger::DirectOutputThread::~DirectOutputThread() 3403{ 3404} 3405 3406void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3407{ 3408 audio_track_cblk_t* cblk = track->cblk(); 3409 float left, right; 3410 3411 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3412 left = right = 0; 3413 } else { 3414 float typeVolume = mStreamTypes[track->streamType()].volume; 3415 float v = mMasterVolume * typeVolume; 3416 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3417 uint32_t vlr = proxy->getVolumeLR(); 3418 float v_clamped = v * (vlr & 0xFFFF); 3419 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3420 left = v_clamped/MAX_GAIN; 3421 v_clamped = v * (vlr >> 16); 3422 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3423 right = v_clamped/MAX_GAIN; 3424 } 3425 3426 if (lastTrack) { 3427 if (left != mLeftVolFloat || right != mRightVolFloat) { 3428 mLeftVolFloat = left; 3429 mRightVolFloat = right; 3430 3431 // Convert volumes from float to 8.24 3432 uint32_t vl = (uint32_t)(left * (1 << 24)); 3433 uint32_t vr = (uint32_t)(right * (1 << 24)); 3434 3435 // Delegate volume control to effect in track effect chain if needed 3436 // only one effect chain can be present on DirectOutputThread, so if 3437 // there is one, the track is connected to it 3438 if (!mEffectChains.isEmpty()) { 3439 mEffectChains[0]->setVolume_l(&vl, &vr); 3440 left = (float)vl / (1 << 24); 3441 right = (float)vr / (1 << 24); 3442 } 3443 if (mOutput->stream->set_volume) { 3444 mOutput->stream->set_volume(mOutput->stream, left, right); 3445 } 3446 } 3447 } 3448} 3449 3450 3451AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3452 Vector< sp<Track> > *tracksToRemove 3453) 3454{ 3455 size_t count = mActiveTracks.size(); 3456 mixer_state mixerStatus = MIXER_IDLE; 3457 3458 // find out which tracks need to be processed 3459 for (size_t i = 0; i < count; i++) { 3460 sp<Track> t = mActiveTracks[i].promote(); 3461 // The track died recently 3462 if (t == 0) { 3463 continue; 3464 } 3465 3466 Track* const track = t.get(); 3467 audio_track_cblk_t* cblk = track->cblk(); 3468 3469 // The first time a track is added we wait 3470 // for all its buffers to be filled before processing it 3471 uint32_t minFrames; 3472 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3473 minFrames = mNormalFrameCount; 3474 } else { 3475 minFrames = 1; 3476 } 3477 // Only consider last track started for volume and mixer state control. 3478 // This is the last entry in mActiveTracks unless a track underruns. 3479 // As we only care about the transition phase between two tracks on a 3480 // direct output, it is not a problem to ignore the underrun case. 3481 bool last = (i == (count - 1)); 3482 3483 if ((track->framesReady() >= minFrames) && track->isReady() && 3484 !track->isPaused() && !track->isTerminated()) 3485 { 3486 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3487 3488 if (track->mFillingUpStatus == Track::FS_FILLED) { 3489 track->mFillingUpStatus = Track::FS_ACTIVE; 3490 mLeftVolFloat = mRightVolFloat = 0; 3491 if (track->mState == TrackBase::RESUMING) { 3492 track->mState = TrackBase::ACTIVE; 3493 } 3494 } 3495 3496 // compute volume for this track 3497 processVolume_l(track, last); 3498 if (last) { 3499 // reset retry count 3500 track->mRetryCount = kMaxTrackRetriesDirect; 3501 mActiveTrack = t; 3502 mixerStatus = MIXER_TRACKS_READY; 3503 } 3504 } else { 3505 // clear effect chain input buffer if the last active track started underruns 3506 // to avoid sending previous audio buffer again to effects 3507 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3508 mEffectChains[0]->clearInputBuffer(); 3509 } 3510 3511 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3512 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3513 track->isStopped() || track->isPaused()) { 3514 // We have consumed all the buffers of this track. 3515 // Remove it from the list of active tracks. 3516 // TODO: implement behavior for compressed audio 3517 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3518 size_t framesWritten = mBytesWritten / mFrameSize; 3519 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3520 if (track->isStopped()) { 3521 track->reset(); 3522 } 3523 tracksToRemove->add(track); 3524 } 3525 } else { 3526 // No buffers for this track. Give it a few chances to 3527 // fill a buffer, then remove it from active list. 3528 // Only consider last track started for mixer state control 3529 if (--(track->mRetryCount) <= 0) { 3530 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3531 tracksToRemove->add(track); 3532 } else if (last) { 3533 mixerStatus = MIXER_TRACKS_ENABLED; 3534 } 3535 } 3536 } 3537 } 3538 3539 // remove all the tracks that need to be... 3540 removeTracks_l(*tracksToRemove); 3541 3542 return mixerStatus; 3543} 3544 3545void AudioFlinger::DirectOutputThread::threadLoop_mix() 3546{ 3547 size_t frameCount = mFrameCount; 3548 int8_t *curBuf = (int8_t *)mMixBuffer; 3549 // output audio to hardware 3550 while (frameCount) { 3551 AudioBufferProvider::Buffer buffer; 3552 buffer.frameCount = frameCount; 3553 mActiveTrack->getNextBuffer(&buffer); 3554 if (CC_UNLIKELY(buffer.raw == NULL)) { 3555 memset(curBuf, 0, frameCount * mFrameSize); 3556 break; 3557 } 3558 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3559 frameCount -= buffer.frameCount; 3560 curBuf += buffer.frameCount * mFrameSize; 3561 mActiveTrack->releaseBuffer(&buffer); 3562 } 3563 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3564 sleepTime = 0; 3565 standbyTime = systemTime() + standbyDelay; 3566 mActiveTrack.clear(); 3567} 3568 3569void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3570{ 3571 if (sleepTime == 0) { 3572 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3573 sleepTime = activeSleepTime; 3574 } else { 3575 sleepTime = idleSleepTime; 3576 } 3577 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3578 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3579 sleepTime = 0; 3580 } 3581} 3582 3583// getTrackName_l() must be called with ThreadBase::mLock held 3584int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3585 int sessionId) 3586{ 3587 return 0; 3588} 3589 3590// deleteTrackName_l() must be called with ThreadBase::mLock held 3591void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3592{ 3593} 3594 3595// checkForNewParameters_l() must be called with ThreadBase::mLock held 3596bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3597{ 3598 bool reconfig = false; 3599 3600 while (!mNewParameters.isEmpty()) { 3601 status_t status = NO_ERROR; 3602 String8 keyValuePair = mNewParameters[0]; 3603 AudioParameter param = AudioParameter(keyValuePair); 3604 int value; 3605 3606 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3607 // do not accept frame count changes if tracks are open as the track buffer 3608 // size depends on frame count and correct behavior would not be garantied 3609 // if frame count is changed after track creation 3610 if (!mTracks.isEmpty()) { 3611 status = INVALID_OPERATION; 3612 } else { 3613 reconfig = true; 3614 } 3615 } 3616 if (status == NO_ERROR) { 3617 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3618 keyValuePair.string()); 3619 if (!mStandby && status == INVALID_OPERATION) { 3620 mOutput->stream->common.standby(&mOutput->stream->common); 3621 mStandby = true; 3622 mBytesWritten = 0; 3623 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3624 keyValuePair.string()); 3625 } 3626 if (status == NO_ERROR && reconfig) { 3627 readOutputParameters(); 3628 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3629 } 3630 } 3631 3632 mNewParameters.removeAt(0); 3633 3634 mParamStatus = status; 3635 mParamCond.signal(); 3636 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3637 // already timed out waiting for the status and will never signal the condition. 3638 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3639 } 3640 return reconfig; 3641} 3642 3643uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3644{ 3645 uint32_t time; 3646 if (audio_is_linear_pcm(mFormat)) { 3647 time = PlaybackThread::activeSleepTimeUs(); 3648 } else { 3649 time = 10000; 3650 } 3651 return time; 3652} 3653 3654uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3655{ 3656 uint32_t time; 3657 if (audio_is_linear_pcm(mFormat)) { 3658 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3659 } else { 3660 time = 10000; 3661 } 3662 return time; 3663} 3664 3665uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3666{ 3667 uint32_t time; 3668 if (audio_is_linear_pcm(mFormat)) { 3669 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3670 } else { 3671 time = 10000; 3672 } 3673 return time; 3674} 3675 3676void AudioFlinger::DirectOutputThread::cacheParameters_l() 3677{ 3678 PlaybackThread::cacheParameters_l(); 3679 3680 // use shorter standby delay as on normal output to release 3681 // hardware resources as soon as possible 3682 standbyDelay = microseconds(activeSleepTime*2); 3683} 3684 3685// ---------------------------------------------------------------------------- 3686 3687AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3688 const sp<AudioFlinger::OffloadThread>& offloadThread) 3689 : Thread(false /*canCallJava*/), 3690 mOffloadThread(offloadThread), 3691 mWriteBlocked(false), 3692 mDraining(false) 3693{ 3694} 3695 3696AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3697{ 3698} 3699 3700void AudioFlinger::AsyncCallbackThread::onFirstRef() 3701{ 3702 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3703} 3704 3705bool AudioFlinger::AsyncCallbackThread::threadLoop() 3706{ 3707 while (!exitPending()) { 3708 bool writeBlocked; 3709 bool draining; 3710 3711 { 3712 Mutex::Autolock _l(mLock); 3713 mWaitWorkCV.wait(mLock); 3714 if (exitPending()) { 3715 break; 3716 } 3717 writeBlocked = mWriteBlocked; 3718 draining = mDraining; 3719 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3720 } 3721 { 3722 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3723 if (offloadThread != 0) { 3724 if (writeBlocked == false) { 3725 offloadThread->setWriteBlocked(false); 3726 } 3727 if (draining == false) { 3728 offloadThread->setDraining(false); 3729 } 3730 } 3731 } 3732 } 3733 return false; 3734} 3735 3736void AudioFlinger::AsyncCallbackThread::exit() 3737{ 3738 ALOGV("AsyncCallbackThread::exit"); 3739 Mutex::Autolock _l(mLock); 3740 requestExit(); 3741 mWaitWorkCV.broadcast(); 3742} 3743 3744void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3745{ 3746 Mutex::Autolock _l(mLock); 3747 mWriteBlocked = value; 3748 if (!value) { 3749 mWaitWorkCV.signal(); 3750 } 3751} 3752 3753void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3754{ 3755 Mutex::Autolock _l(mLock); 3756 mDraining = value; 3757 if (!value) { 3758 mWaitWorkCV.signal(); 3759 } 3760} 3761 3762 3763// ---------------------------------------------------------------------------- 3764AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3765 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3766 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3767 mHwPaused(false), 3768 mPausedBytesRemaining(0) 3769{ 3770 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3771} 3772 3773AudioFlinger::OffloadThread::~OffloadThread() 3774{ 3775 mPreviousTrack.clear(); 3776} 3777 3778void AudioFlinger::OffloadThread::threadLoop_exit() 3779{ 3780 if (mFlushPending || mHwPaused) { 3781 // If a flush is pending or track was paused, just discard buffered data 3782 flushHw_l(); 3783 } else { 3784 mMixerStatus = MIXER_DRAIN_ALL; 3785 threadLoop_drain(); 3786 } 3787 mCallbackThread->exit(); 3788 PlaybackThread::threadLoop_exit(); 3789} 3790 3791AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3792 Vector< sp<Track> > *tracksToRemove 3793) 3794{ 3795 ALOGV("OffloadThread::prepareTracks_l"); 3796 size_t count = mActiveTracks.size(); 3797 3798 mixer_state mixerStatus = MIXER_IDLE; 3799 if (mFlushPending) { 3800 flushHw_l(); 3801 mFlushPending = false; 3802 } 3803 // find out which tracks need to be processed 3804 for (size_t i = 0; i < count; i++) { 3805 sp<Track> t = mActiveTracks[i].promote(); 3806 // The track died recently 3807 if (t == 0) { 3808 continue; 3809 } 3810 Track* const track = t.get(); 3811 audio_track_cblk_t* cblk = track->cblk(); 3812 if (mPreviousTrack != NULL) { 3813 if (t != mPreviousTrack) { 3814 // Flush any data still being written from last track 3815 mBytesRemaining = 0; 3816 if (mPausedBytesRemaining) { 3817 // Last track was paused so we also need to flush saved 3818 // mixbuffer state and invalidate track so that it will 3819 // re-submit that unwritten data when it is next resumed 3820 mPausedBytesRemaining = 0; 3821 // Invalidate is a bit drastic - would be more efficient 3822 // to have a flag to tell client that some of the 3823 // previously written data was lost 3824 mPreviousTrack->invalidate(); 3825 } 3826 } 3827 } 3828 mPreviousTrack = t; 3829 bool last = (i == (count - 1)); 3830 if (track->isPausing()) { 3831 track->setPaused(); 3832 if (last) { 3833 if (!mHwPaused) { 3834 mOutput->stream->pause(mOutput->stream); 3835 mHwPaused = true; 3836 } 3837 // If we were part way through writing the mixbuffer to 3838 // the HAL we must save this until we resume 3839 // BUG - this will be wrong if a different track is made active, 3840 // in that case we want to discard the pending data in the 3841 // mixbuffer and tell the client to present it again when the 3842 // track is resumed 3843 mPausedWriteLength = mCurrentWriteLength; 3844 mPausedBytesRemaining = mBytesRemaining; 3845 mBytesRemaining = 0; // stop writing 3846 } 3847 tracksToRemove->add(track); 3848 } else if (track->framesReady() && track->isReady() && 3849 !track->isPaused() && !track->isTerminated()) { 3850 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->server); 3851 if (track->mFillingUpStatus == Track::FS_FILLED) { 3852 track->mFillingUpStatus = Track::FS_ACTIVE; 3853 mLeftVolFloat = mRightVolFloat = 0; 3854 if (track->mState == TrackBase::RESUMING) { 3855 if (CC_UNLIKELY(mPausedBytesRemaining)) { 3856 // Need to continue write that was interrupted 3857 mCurrentWriteLength = mPausedWriteLength; 3858 mBytesRemaining = mPausedBytesRemaining; 3859 mPausedBytesRemaining = 0; 3860 } 3861 track->mState = TrackBase::ACTIVE; 3862 } 3863 } 3864 3865 if (last) { 3866 if (mHwPaused) { 3867 mOutput->stream->resume(mOutput->stream); 3868 mHwPaused = false; 3869 // threadLoop_mix() will handle the case that we need to 3870 // resume an interrupted write 3871 } 3872 // reset retry count 3873 track->mRetryCount = kMaxTrackRetriesOffload; 3874 mActiveTrack = t; 3875 mixerStatus = MIXER_TRACKS_READY; 3876 } 3877 } else { 3878 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->server); 3879 if (track->isStopping_1()) { 3880 // Hardware buffer can hold a large amount of audio so we must 3881 // wait for all current track's data to drain before we say 3882 // that the track is stopped. 3883 if (mBytesRemaining == 0) { 3884 // Only start draining when all data in mixbuffer 3885 // has been written 3886 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3887 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3888 sleepTime = 0; 3889 standbyTime = systemTime() + standbyDelay; 3890 if (last) { 3891 mixerStatus = MIXER_DRAIN_TRACK; 3892 if (mHwPaused) { 3893 // It is possible to move from PAUSED to STOPPING_1 without 3894 // a resume so we must ensure hardware is running 3895 mOutput->stream->resume(mOutput->stream); 3896 mHwPaused = false; 3897 } 3898 } 3899 } 3900 } else if (track->isStopping_2()) { 3901 // Drain has completed, signal presentation complete 3902 if (!mDraining || !last) { 3903 track->mState = TrackBase::STOPPED; 3904 size_t audioHALFrames = 3905 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3906 size_t framesWritten = 3907 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3908 track->presentationComplete(framesWritten, audioHALFrames); 3909 track->reset(); 3910 tracksToRemove->add(track); 3911 } 3912 } else { 3913 // No buffers for this track. Give it a few chances to 3914 // fill a buffer, then remove it from active list. 3915 if (--(track->mRetryCount) <= 0) { 3916 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3917 track->name()); 3918 tracksToRemove->add(track); 3919 } else if (last){ 3920 mixerStatus = MIXER_TRACKS_ENABLED; 3921 } 3922 } 3923 } 3924 // compute volume for this track 3925 processVolume_l(track, last); 3926 } 3927 // remove all the tracks that need to be... 3928 removeTracks_l(*tracksToRemove); 3929 3930 return mixerStatus; 3931} 3932 3933void AudioFlinger::OffloadThread::flushOutput_l() 3934{ 3935 mFlushPending = true; 3936} 3937 3938// must be called with thread mutex locked 3939bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3940{ 3941 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3942 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3943 return true; 3944 } 3945 return false; 3946} 3947 3948// must be called with thread mutex locked 3949bool AudioFlinger::OffloadThread::shouldStandby_l() 3950{ 3951 bool TrackPaused = false; 3952 3953 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3954 // after a timeout and we will enter standby then. 3955 if (mTracks.size() > 0) { 3956 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3957 } 3958 3959 return !mStandby && !TrackPaused; 3960} 3961 3962 3963bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3964{ 3965 Mutex::Autolock _l(mLock); 3966 return waitingAsyncCallback_l(); 3967} 3968 3969void AudioFlinger::OffloadThread::flushHw_l() 3970{ 3971 mOutput->stream->flush(mOutput->stream); 3972 // Flush anything still waiting in the mixbuffer 3973 mCurrentWriteLength = 0; 3974 mBytesRemaining = 0; 3975 mPausedWriteLength = 0; 3976 mPausedBytesRemaining = 0; 3977 if (mUseAsyncWrite) { 3978 mWriteBlocked = false; 3979 mDraining = false; 3980 ALOG_ASSERT(mCallbackThread != 0); 3981 mCallbackThread->setWriteBlocked(false); 3982 mCallbackThread->setDraining(false); 3983 } 3984} 3985 3986// ---------------------------------------------------------------------------- 3987 3988AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3989 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3990 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3991 DUPLICATING), 3992 mWaitTimeMs(UINT_MAX) 3993{ 3994 addOutputTrack(mainThread); 3995} 3996 3997AudioFlinger::DuplicatingThread::~DuplicatingThread() 3998{ 3999 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4000 mOutputTracks[i]->destroy(); 4001 } 4002} 4003 4004void AudioFlinger::DuplicatingThread::threadLoop_mix() 4005{ 4006 // mix buffers... 4007 if (outputsReady(outputTracks)) { 4008 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4009 } else { 4010 memset(mMixBuffer, 0, mixBufferSize); 4011 } 4012 sleepTime = 0; 4013 writeFrames = mNormalFrameCount; 4014 mCurrentWriteLength = mixBufferSize; 4015 standbyTime = systemTime() + standbyDelay; 4016} 4017 4018void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4019{ 4020 if (sleepTime == 0) { 4021 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4022 sleepTime = activeSleepTime; 4023 } else { 4024 sleepTime = idleSleepTime; 4025 } 4026 } else if (mBytesWritten != 0) { 4027 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4028 writeFrames = mNormalFrameCount; 4029 memset(mMixBuffer, 0, mixBufferSize); 4030 } else { 4031 // flush remaining overflow buffers in output tracks 4032 writeFrames = 0; 4033 } 4034 sleepTime = 0; 4035 } 4036} 4037 4038ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4039{ 4040 for (size_t i = 0; i < outputTracks.size(); i++) { 4041 outputTracks[i]->write(mMixBuffer, writeFrames); 4042 } 4043 return (ssize_t)mixBufferSize; 4044} 4045 4046void AudioFlinger::DuplicatingThread::threadLoop_standby() 4047{ 4048 // DuplicatingThread implements standby by stopping all tracks 4049 for (size_t i = 0; i < outputTracks.size(); i++) { 4050 outputTracks[i]->stop(); 4051 } 4052} 4053 4054void AudioFlinger::DuplicatingThread::saveOutputTracks() 4055{ 4056 outputTracks = mOutputTracks; 4057} 4058 4059void AudioFlinger::DuplicatingThread::clearOutputTracks() 4060{ 4061 outputTracks.clear(); 4062} 4063 4064void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4065{ 4066 Mutex::Autolock _l(mLock); 4067 // FIXME explain this formula 4068 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4069 OutputTrack *outputTrack = new OutputTrack(thread, 4070 this, 4071 mSampleRate, 4072 mFormat, 4073 mChannelMask, 4074 frameCount); 4075 if (outputTrack->cblk() != NULL) { 4076 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4077 mOutputTracks.add(outputTrack); 4078 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4079 updateWaitTime_l(); 4080 } 4081} 4082 4083void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4084{ 4085 Mutex::Autolock _l(mLock); 4086 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4087 if (mOutputTracks[i]->thread() == thread) { 4088 mOutputTracks[i]->destroy(); 4089 mOutputTracks.removeAt(i); 4090 updateWaitTime_l(); 4091 return; 4092 } 4093 } 4094 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4095} 4096 4097// caller must hold mLock 4098void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4099{ 4100 mWaitTimeMs = UINT_MAX; 4101 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4102 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4103 if (strong != 0) { 4104 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4105 if (waitTimeMs < mWaitTimeMs) { 4106 mWaitTimeMs = waitTimeMs; 4107 } 4108 } 4109 } 4110} 4111 4112 4113bool AudioFlinger::DuplicatingThread::outputsReady( 4114 const SortedVector< sp<OutputTrack> > &outputTracks) 4115{ 4116 for (size_t i = 0; i < outputTracks.size(); i++) { 4117 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4118 if (thread == 0) { 4119 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4120 outputTracks[i].get()); 4121 return false; 4122 } 4123 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4124 // see note at standby() declaration 4125 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4126 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4127 thread.get()); 4128 return false; 4129 } 4130 } 4131 return true; 4132} 4133 4134uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4135{ 4136 return (mWaitTimeMs * 1000) / 2; 4137} 4138 4139void AudioFlinger::DuplicatingThread::cacheParameters_l() 4140{ 4141 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4142 updateWaitTime_l(); 4143 4144 MixerThread::cacheParameters_l(); 4145} 4146 4147// ---------------------------------------------------------------------------- 4148// Record 4149// ---------------------------------------------------------------------------- 4150 4151AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4152 AudioStreamIn *input, 4153 uint32_t sampleRate, 4154 audio_channel_mask_t channelMask, 4155 audio_io_handle_t id, 4156 audio_devices_t outDevice, 4157 audio_devices_t inDevice 4158#ifdef TEE_SINK 4159 , const sp<NBAIO_Sink>& teeSink 4160#endif 4161 ) : 4162 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4163 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4164 // mRsmpInIndex and mInputBytes set by readInputParameters() 4165 mReqChannelCount(popcount(channelMask)), 4166 mReqSampleRate(sampleRate) 4167 // mBytesRead is only meaningful while active, and so is cleared in start() 4168 // (but might be better to also clear here for dump?) 4169#ifdef TEE_SINK 4170 , mTeeSink(teeSink) 4171#endif 4172{ 4173 snprintf(mName, kNameLength, "AudioIn_%X", id); 4174 4175 readInputParameters(); 4176 4177} 4178 4179 4180AudioFlinger::RecordThread::~RecordThread() 4181{ 4182 delete[] mRsmpInBuffer; 4183 delete mResampler; 4184 delete[] mRsmpOutBuffer; 4185} 4186 4187void AudioFlinger::RecordThread::onFirstRef() 4188{ 4189 run(mName, PRIORITY_URGENT_AUDIO); 4190} 4191 4192status_t AudioFlinger::RecordThread::readyToRun() 4193{ 4194 status_t status = initCheck(); 4195 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4196 return status; 4197} 4198 4199bool AudioFlinger::RecordThread::threadLoop() 4200{ 4201 AudioBufferProvider::Buffer buffer; 4202 sp<RecordTrack> activeTrack; 4203 Vector< sp<EffectChain> > effectChains; 4204 4205 nsecs_t lastWarning = 0; 4206 4207 inputStandBy(); 4208 acquireWakeLock(); 4209 4210 // used to verify we've read at least once before evaluating how many bytes were read 4211 bool readOnce = false; 4212 4213 // start recording 4214 while (!exitPending()) { 4215 4216 processConfigEvents(); 4217 4218 { // scope for mLock 4219 Mutex::Autolock _l(mLock); 4220 checkForNewParameters_l(); 4221 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4222 standby(); 4223 4224 if (exitPending()) { 4225 break; 4226 } 4227 4228 releaseWakeLock_l(); 4229 ALOGV("RecordThread: loop stopping"); 4230 // go to sleep 4231 mWaitWorkCV.wait(mLock); 4232 ALOGV("RecordThread: loop starting"); 4233 acquireWakeLock_l(); 4234 continue; 4235 } 4236 if (mActiveTrack != 0) { 4237 if (mActiveTrack->isTerminated()) { 4238 removeTrack_l(mActiveTrack); 4239 mActiveTrack.clear(); 4240 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4241 standby(); 4242 mActiveTrack.clear(); 4243 mStartStopCond.broadcast(); 4244 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4245 if (mReqChannelCount != mActiveTrack->channelCount()) { 4246 mActiveTrack.clear(); 4247 mStartStopCond.broadcast(); 4248 } else if (readOnce) { 4249 // record start succeeds only if first read from audio input 4250 // succeeds 4251 if (mBytesRead >= 0) { 4252 mActiveTrack->mState = TrackBase::ACTIVE; 4253 } else { 4254 mActiveTrack.clear(); 4255 } 4256 mStartStopCond.broadcast(); 4257 } 4258 mStandby = false; 4259 } 4260 } 4261 lockEffectChains_l(effectChains); 4262 } 4263 4264 if (mActiveTrack != 0) { 4265 if (mActiveTrack->mState != TrackBase::ACTIVE && 4266 mActiveTrack->mState != TrackBase::RESUMING) { 4267 unlockEffectChains(effectChains); 4268 usleep(kRecordThreadSleepUs); 4269 continue; 4270 } 4271 for (size_t i = 0; i < effectChains.size(); i ++) { 4272 effectChains[i]->process_l(); 4273 } 4274 4275 buffer.frameCount = mFrameCount; 4276 status_t status = mActiveTrack->getNextBuffer(&buffer); 4277 if (CC_LIKELY(status == NO_ERROR)) { 4278 readOnce = true; 4279 size_t framesOut = buffer.frameCount; 4280 if (mResampler == NULL) { 4281 // no resampling 4282 while (framesOut) { 4283 size_t framesIn = mFrameCount - mRsmpInIndex; 4284 if (framesIn) { 4285 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4286 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4287 mActiveTrack->mFrameSize; 4288 if (framesIn > framesOut) 4289 framesIn = framesOut; 4290 mRsmpInIndex += framesIn; 4291 framesOut -= framesIn; 4292 if (mChannelCount == mReqChannelCount || 4293 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4294 memcpy(dst, src, framesIn * mFrameSize); 4295 } else { 4296 if (mChannelCount == 1) { 4297 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4298 (int16_t *)src, framesIn); 4299 } else { 4300 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4301 (int16_t *)src, framesIn); 4302 } 4303 } 4304 } 4305 if (framesOut && mFrameCount == mRsmpInIndex) { 4306 void *readInto; 4307 if (framesOut == mFrameCount && 4308 (mChannelCount == mReqChannelCount || 4309 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4310 readInto = buffer.raw; 4311 framesOut = 0; 4312 } else { 4313 readInto = mRsmpInBuffer; 4314 mRsmpInIndex = 0; 4315 } 4316 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4317 mInputBytes); 4318 if (mBytesRead <= 0) { 4319 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4320 { 4321 ALOGE("Error reading audio input"); 4322 // Force input into standby so that it tries to 4323 // recover at next read attempt 4324 inputStandBy(); 4325 usleep(kRecordThreadSleepUs); 4326 } 4327 mRsmpInIndex = mFrameCount; 4328 framesOut = 0; 4329 buffer.frameCount = 0; 4330 } 4331#ifdef TEE_SINK 4332 else if (mTeeSink != 0) { 4333 (void) mTeeSink->write(readInto, 4334 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4335 } 4336#endif 4337 } 4338 } 4339 } else { 4340 // resampling 4341 4342 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4343 // alter output frame count as if we were expecting stereo samples 4344 if (mChannelCount == 1 && mReqChannelCount == 1) { 4345 framesOut >>= 1; 4346 } 4347 mResampler->resample(mRsmpOutBuffer, framesOut, 4348 this /* AudioBufferProvider* */); 4349 // ditherAndClamp() works as long as all buffers returned by 4350 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4351 if (mChannelCount == 2 && mReqChannelCount == 1) { 4352 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4353 // the resampler always outputs stereo samples: 4354 // do post stereo to mono conversion 4355 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4356 framesOut); 4357 } else { 4358 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4359 } 4360 4361 } 4362 if (mFramestoDrop == 0) { 4363 mActiveTrack->releaseBuffer(&buffer); 4364 } else { 4365 if (mFramestoDrop > 0) { 4366 mFramestoDrop -= buffer.frameCount; 4367 if (mFramestoDrop <= 0) { 4368 clearSyncStartEvent(); 4369 } 4370 } else { 4371 mFramestoDrop += buffer.frameCount; 4372 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4373 mSyncStartEvent->isCancelled()) { 4374 ALOGW("Synced record %s, session %d, trigger session %d", 4375 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4376 mActiveTrack->sessionId(), 4377 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4378 clearSyncStartEvent(); 4379 } 4380 } 4381 } 4382 mActiveTrack->clearOverflow(); 4383 } 4384 // client isn't retrieving buffers fast enough 4385 else { 4386 if (!mActiveTrack->setOverflow()) { 4387 nsecs_t now = systemTime(); 4388 if ((now - lastWarning) > kWarningThrottleNs) { 4389 ALOGW("RecordThread: buffer overflow"); 4390 lastWarning = now; 4391 } 4392 } 4393 // Release the processor for a while before asking for a new buffer. 4394 // This will give the application more chance to read from the buffer and 4395 // clear the overflow. 4396 usleep(kRecordThreadSleepUs); 4397 } 4398 } 4399 // enable changes in effect chain 4400 unlockEffectChains(effectChains); 4401 effectChains.clear(); 4402 } 4403 4404 standby(); 4405 4406 { 4407 Mutex::Autolock _l(mLock); 4408 mActiveTrack.clear(); 4409 mStartStopCond.broadcast(); 4410 } 4411 4412 releaseWakeLock(); 4413 4414 ALOGV("RecordThread %p exiting", this); 4415 return false; 4416} 4417 4418void AudioFlinger::RecordThread::standby() 4419{ 4420 if (!mStandby) { 4421 inputStandBy(); 4422 mStandby = true; 4423 } 4424} 4425 4426void AudioFlinger::RecordThread::inputStandBy() 4427{ 4428 mInput->stream->common.standby(&mInput->stream->common); 4429} 4430 4431sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4432 const sp<AudioFlinger::Client>& client, 4433 uint32_t sampleRate, 4434 audio_format_t format, 4435 audio_channel_mask_t channelMask, 4436 size_t frameCount, 4437 int sessionId, 4438 IAudioFlinger::track_flags_t flags, 4439 pid_t tid, 4440 status_t *status) 4441{ 4442 sp<RecordTrack> track; 4443 status_t lStatus; 4444 4445 lStatus = initCheck(); 4446 if (lStatus != NO_ERROR) { 4447 ALOGE("Audio driver not initialized."); 4448 goto Exit; 4449 } 4450 4451 // FIXME use flags and tid similar to createTrack_l() 4452 4453 { // scope for mLock 4454 Mutex::Autolock _l(mLock); 4455 4456 track = new RecordTrack(this, client, sampleRate, 4457 format, channelMask, frameCount, sessionId); 4458 4459 if (track->getCblk() == 0) { 4460 lStatus = NO_MEMORY; 4461 goto Exit; 4462 } 4463 mTracks.add(track); 4464 4465 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4466 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4467 mAudioFlinger->btNrecIsOff(); 4468 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4469 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4470 } 4471 lStatus = NO_ERROR; 4472 4473Exit: 4474 if (status) { 4475 *status = lStatus; 4476 } 4477 return track; 4478} 4479 4480status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4481 AudioSystem::sync_event_t event, 4482 int triggerSession) 4483{ 4484 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4485 sp<ThreadBase> strongMe = this; 4486 status_t status = NO_ERROR; 4487 4488 if (event == AudioSystem::SYNC_EVENT_NONE) { 4489 clearSyncStartEvent(); 4490 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4491 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4492 triggerSession, 4493 recordTrack->sessionId(), 4494 syncStartEventCallback, 4495 this); 4496 // Sync event can be cancelled by the trigger session if the track is not in a 4497 // compatible state in which case we start record immediately 4498 if (mSyncStartEvent->isCancelled()) { 4499 clearSyncStartEvent(); 4500 } else { 4501 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4502 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4503 } 4504 } 4505 4506 { 4507 AutoMutex lock(mLock); 4508 if (mActiveTrack != 0) { 4509 if (recordTrack != mActiveTrack.get()) { 4510 status = -EBUSY; 4511 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4512 mActiveTrack->mState = TrackBase::ACTIVE; 4513 } 4514 return status; 4515 } 4516 4517 recordTrack->mState = TrackBase::IDLE; 4518 mActiveTrack = recordTrack; 4519 mLock.unlock(); 4520 status_t status = AudioSystem::startInput(mId); 4521 mLock.lock(); 4522 if (status != NO_ERROR) { 4523 mActiveTrack.clear(); 4524 clearSyncStartEvent(); 4525 return status; 4526 } 4527 mRsmpInIndex = mFrameCount; 4528 mBytesRead = 0; 4529 if (mResampler != NULL) { 4530 mResampler->reset(); 4531 } 4532 mActiveTrack->mState = TrackBase::RESUMING; 4533 // signal thread to start 4534 ALOGV("Signal record thread"); 4535 mWaitWorkCV.broadcast(); 4536 // do not wait for mStartStopCond if exiting 4537 if (exitPending()) { 4538 mActiveTrack.clear(); 4539 status = INVALID_OPERATION; 4540 goto startError; 4541 } 4542 mStartStopCond.wait(mLock); 4543 if (mActiveTrack == 0) { 4544 ALOGV("Record failed to start"); 4545 status = BAD_VALUE; 4546 goto startError; 4547 } 4548 ALOGV("Record started OK"); 4549 return status; 4550 } 4551 4552startError: 4553 AudioSystem::stopInput(mId); 4554 clearSyncStartEvent(); 4555 return status; 4556} 4557 4558void AudioFlinger::RecordThread::clearSyncStartEvent() 4559{ 4560 if (mSyncStartEvent != 0) { 4561 mSyncStartEvent->cancel(); 4562 } 4563 mSyncStartEvent.clear(); 4564 mFramestoDrop = 0; 4565} 4566 4567void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4568{ 4569 sp<SyncEvent> strongEvent = event.promote(); 4570 4571 if (strongEvent != 0) { 4572 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4573 me->handleSyncStartEvent(strongEvent); 4574 } 4575} 4576 4577void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4578{ 4579 if (event == mSyncStartEvent) { 4580 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4581 // from audio HAL 4582 mFramestoDrop = mFrameCount * 2; 4583 } 4584} 4585 4586bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4587 ALOGV("RecordThread::stop"); 4588 AutoMutex _l(mLock); 4589 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4590 return false; 4591 } 4592 recordTrack->mState = TrackBase::PAUSING; 4593 // do not wait for mStartStopCond if exiting 4594 if (exitPending()) { 4595 return true; 4596 } 4597 mStartStopCond.wait(mLock); 4598 // if we have been restarted, recordTrack == mActiveTrack.get() here 4599 if (exitPending() || recordTrack != mActiveTrack.get()) { 4600 ALOGV("Record stopped OK"); 4601 return true; 4602 } 4603 return false; 4604} 4605 4606bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4607{ 4608 return false; 4609} 4610 4611status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4612{ 4613#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4614 if (!isValidSyncEvent(event)) { 4615 return BAD_VALUE; 4616 } 4617 4618 int eventSession = event->triggerSession(); 4619 status_t ret = NAME_NOT_FOUND; 4620 4621 Mutex::Autolock _l(mLock); 4622 4623 for (size_t i = 0; i < mTracks.size(); i++) { 4624 sp<RecordTrack> track = mTracks[i]; 4625 if (eventSession == track->sessionId()) { 4626 (void) track->setSyncEvent(event); 4627 ret = NO_ERROR; 4628 } 4629 } 4630 return ret; 4631#else 4632 return BAD_VALUE; 4633#endif 4634} 4635 4636// destroyTrack_l() must be called with ThreadBase::mLock held 4637void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4638{ 4639 track->terminate(); 4640 track->mState = TrackBase::STOPPED; 4641 // active tracks are removed by threadLoop() 4642 if (mActiveTrack != track) { 4643 removeTrack_l(track); 4644 } 4645} 4646 4647void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4648{ 4649 mTracks.remove(track); 4650 // need anything related to effects here? 4651} 4652 4653void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4654{ 4655 dumpInternals(fd, args); 4656 dumpTracks(fd, args); 4657 dumpEffectChains(fd, args); 4658} 4659 4660void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4661{ 4662 const size_t SIZE = 256; 4663 char buffer[SIZE]; 4664 String8 result; 4665 4666 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4667 result.append(buffer); 4668 4669 if (mActiveTrack != 0) { 4670 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4671 result.append(buffer); 4672 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4673 result.append(buffer); 4674 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4675 result.append(buffer); 4676 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4677 result.append(buffer); 4678 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4679 result.append(buffer); 4680 } else { 4681 result.append("No active record client\n"); 4682 } 4683 4684 write(fd, result.string(), result.size()); 4685 4686 dumpBase(fd, args); 4687} 4688 4689void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4690{ 4691 const size_t SIZE = 256; 4692 char buffer[SIZE]; 4693 String8 result; 4694 4695 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4696 result.append(buffer); 4697 RecordTrack::appendDumpHeader(result); 4698 for (size_t i = 0; i < mTracks.size(); ++i) { 4699 sp<RecordTrack> track = mTracks[i]; 4700 if (track != 0) { 4701 track->dump(buffer, SIZE); 4702 result.append(buffer); 4703 } 4704 } 4705 4706 if (mActiveTrack != 0) { 4707 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4708 result.append(buffer); 4709 RecordTrack::appendDumpHeader(result); 4710 mActiveTrack->dump(buffer, SIZE); 4711 result.append(buffer); 4712 4713 } 4714 write(fd, result.string(), result.size()); 4715} 4716 4717// AudioBufferProvider interface 4718status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4719{ 4720 size_t framesReq = buffer->frameCount; 4721 size_t framesReady = mFrameCount - mRsmpInIndex; 4722 int channelCount; 4723 4724 if (framesReady == 0) { 4725 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4726 if (mBytesRead <= 0) { 4727 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4728 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4729 // Force input into standby so that it tries to 4730 // recover at next read attempt 4731 inputStandBy(); 4732 usleep(kRecordThreadSleepUs); 4733 } 4734 buffer->raw = NULL; 4735 buffer->frameCount = 0; 4736 return NOT_ENOUGH_DATA; 4737 } 4738 mRsmpInIndex = 0; 4739 framesReady = mFrameCount; 4740 } 4741 4742 if (framesReq > framesReady) { 4743 framesReq = framesReady; 4744 } 4745 4746 if (mChannelCount == 1 && mReqChannelCount == 2) { 4747 channelCount = 1; 4748 } else { 4749 channelCount = 2; 4750 } 4751 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4752 buffer->frameCount = framesReq; 4753 return NO_ERROR; 4754} 4755 4756// AudioBufferProvider interface 4757void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4758{ 4759 mRsmpInIndex += buffer->frameCount; 4760 buffer->frameCount = 0; 4761} 4762 4763bool AudioFlinger::RecordThread::checkForNewParameters_l() 4764{ 4765 bool reconfig = false; 4766 4767 while (!mNewParameters.isEmpty()) { 4768 status_t status = NO_ERROR; 4769 String8 keyValuePair = mNewParameters[0]; 4770 AudioParameter param = AudioParameter(keyValuePair); 4771 int value; 4772 audio_format_t reqFormat = mFormat; 4773 uint32_t reqSamplingRate = mReqSampleRate; 4774 uint32_t reqChannelCount = mReqChannelCount; 4775 4776 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4777 reqSamplingRate = value; 4778 reconfig = true; 4779 } 4780 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4781 reqFormat = (audio_format_t) value; 4782 reconfig = true; 4783 } 4784 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4785 reqChannelCount = popcount(value); 4786 reconfig = true; 4787 } 4788 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4789 // do not accept frame count changes if tracks are open as the track buffer 4790 // size depends on frame count and correct behavior would not be guaranteed 4791 // if frame count is changed after track creation 4792 if (mActiveTrack != 0) { 4793 status = INVALID_OPERATION; 4794 } else { 4795 reconfig = true; 4796 } 4797 } 4798 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4799 // forward device change to effects that have requested to be 4800 // aware of attached audio device. 4801 for (size_t i = 0; i < mEffectChains.size(); i++) { 4802 mEffectChains[i]->setDevice_l(value); 4803 } 4804 4805 // store input device and output device but do not forward output device to audio HAL. 4806 // Note that status is ignored by the caller for output device 4807 // (see AudioFlinger::setParameters() 4808 if (audio_is_output_devices(value)) { 4809 mOutDevice = value; 4810 status = BAD_VALUE; 4811 } else { 4812 mInDevice = value; 4813 // disable AEC and NS if the device is a BT SCO headset supporting those 4814 // pre processings 4815 if (mTracks.size() > 0) { 4816 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4817 mAudioFlinger->btNrecIsOff(); 4818 for (size_t i = 0; i < mTracks.size(); i++) { 4819 sp<RecordTrack> track = mTracks[i]; 4820 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4821 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4822 } 4823 } 4824 } 4825 } 4826 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4827 mAudioSource != (audio_source_t)value) { 4828 // forward device change to effects that have requested to be 4829 // aware of attached audio device. 4830 for (size_t i = 0; i < mEffectChains.size(); i++) { 4831 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4832 } 4833 mAudioSource = (audio_source_t)value; 4834 } 4835 if (status == NO_ERROR) { 4836 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4837 keyValuePair.string()); 4838 if (status == INVALID_OPERATION) { 4839 inputStandBy(); 4840 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4841 keyValuePair.string()); 4842 } 4843 if (reconfig) { 4844 if (status == BAD_VALUE && 4845 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4846 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4847 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4848 <= (2 * reqSamplingRate)) && 4849 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4850 <= FCC_2 && 4851 (reqChannelCount <= FCC_2)) { 4852 status = NO_ERROR; 4853 } 4854 if (status == NO_ERROR) { 4855 readInputParameters(); 4856 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4857 } 4858 } 4859 } 4860 4861 mNewParameters.removeAt(0); 4862 4863 mParamStatus = status; 4864 mParamCond.signal(); 4865 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4866 // already timed out waiting for the status and will never signal the condition. 4867 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4868 } 4869 return reconfig; 4870} 4871 4872String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4873{ 4874 Mutex::Autolock _l(mLock); 4875 if (initCheck() != NO_ERROR) { 4876 return String8(); 4877 } 4878 4879 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4880 const String8 out_s8(s); 4881 free(s); 4882 return out_s8; 4883} 4884 4885void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4886 AudioSystem::OutputDescriptor desc; 4887 void *param2 = NULL; 4888 4889 switch (event) { 4890 case AudioSystem::INPUT_OPENED: 4891 case AudioSystem::INPUT_CONFIG_CHANGED: 4892 desc.channelMask = mChannelMask; 4893 desc.samplingRate = mSampleRate; 4894 desc.format = mFormat; 4895 desc.frameCount = mFrameCount; 4896 desc.latency = 0; 4897 param2 = &desc; 4898 break; 4899 4900 case AudioSystem::INPUT_CLOSED: 4901 default: 4902 break; 4903 } 4904 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4905} 4906 4907void AudioFlinger::RecordThread::readInputParameters() 4908{ 4909 delete mRsmpInBuffer; 4910 // mRsmpInBuffer is always assigned a new[] below 4911 delete mRsmpOutBuffer; 4912 mRsmpOutBuffer = NULL; 4913 delete mResampler; 4914 mResampler = NULL; 4915 4916 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4917 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4918 mChannelCount = popcount(mChannelMask); 4919 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4920 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4921 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4922 mFrameCount = mInputBytes / mFrameSize; 4923 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4924 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4925 4926 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4927 { 4928 int channelCount; 4929 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4930 // stereo to mono post process as the resampler always outputs stereo. 4931 if (mChannelCount == 1 && mReqChannelCount == 2) { 4932 channelCount = 1; 4933 } else { 4934 channelCount = 2; 4935 } 4936 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4937 mResampler->setSampleRate(mSampleRate); 4938 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4939 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4940 4941 // optmization: if mono to mono, alter input frame count as if we were inputing 4942 // stereo samples 4943 if (mChannelCount == 1 && mReqChannelCount == 1) { 4944 mFrameCount >>= 1; 4945 } 4946 4947 } 4948 mRsmpInIndex = mFrameCount; 4949} 4950 4951unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4952{ 4953 Mutex::Autolock _l(mLock); 4954 if (initCheck() != NO_ERROR) { 4955 return 0; 4956 } 4957 4958 return mInput->stream->get_input_frames_lost(mInput->stream); 4959} 4960 4961uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4962{ 4963 Mutex::Autolock _l(mLock); 4964 uint32_t result = 0; 4965 if (getEffectChain_l(sessionId) != 0) { 4966 result = EFFECT_SESSION; 4967 } 4968 4969 for (size_t i = 0; i < mTracks.size(); ++i) { 4970 if (sessionId == mTracks[i]->sessionId()) { 4971 result |= TRACK_SESSION; 4972 break; 4973 } 4974 } 4975 4976 return result; 4977} 4978 4979KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4980{ 4981 KeyedVector<int, bool> ids; 4982 Mutex::Autolock _l(mLock); 4983 for (size_t j = 0; j < mTracks.size(); ++j) { 4984 sp<RecordThread::RecordTrack> track = mTracks[j]; 4985 int sessionId = track->sessionId(); 4986 if (ids.indexOfKey(sessionId) < 0) { 4987 ids.add(sessionId, true); 4988 } 4989 } 4990 return ids; 4991} 4992 4993AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4994{ 4995 Mutex::Autolock _l(mLock); 4996 AudioStreamIn *input = mInput; 4997 mInput = NULL; 4998 return input; 4999} 5000 5001// this method must always be called either with ThreadBase mLock held or inside the thread loop 5002audio_stream_t* AudioFlinger::RecordThread::stream() const 5003{ 5004 if (mInput == NULL) { 5005 return NULL; 5006 } 5007 return &mInput->stream->common; 5008} 5009 5010status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5011{ 5012 // only one chain per input thread 5013 if (mEffectChains.size() != 0) { 5014 return INVALID_OPERATION; 5015 } 5016 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5017 5018 chain->setInBuffer(NULL); 5019 chain->setOutBuffer(NULL); 5020 5021 checkSuspendOnAddEffectChain_l(chain); 5022 5023 mEffectChains.add(chain); 5024 5025 return NO_ERROR; 5026} 5027 5028size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5029{ 5030 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5031 ALOGW_IF(mEffectChains.size() != 1, 5032 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5033 chain.get(), mEffectChains.size(), this); 5034 if (mEffectChains.size() == 1) { 5035 mEffectChains.removeAt(0); 5036 } 5037 return 0; 5038} 5039 5040}; // namespace android 5041