Threads.cpp revision 7fc97ba08e2850f3f16db704b78ce78e3dbe1ff0
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: %d\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 mMixBuffer(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 mScreenState(AudioFlinger::mScreenState), 943 // index 0 is reserved for normal mixer's submix 944 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 945{ 946 snprintf(mName, kNameLength, "AudioOut_%X", id); 947 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 948 949 // Assumes constructor is called by AudioFlinger with it's mLock held, but 950 // it would be safer to explicitly pass initial masterVolume/masterMute as 951 // parameter. 952 // 953 // If the HAL we are using has support for master volume or master mute, 954 // then do not attenuate or mute during mixing (just leave the volume at 1.0 955 // and the mute set to false). 956 mMasterVolume = audioFlinger->masterVolume_l(); 957 mMasterMute = audioFlinger->masterMute_l(); 958 if (mOutput && mOutput->audioHwDev) { 959 if (mOutput->audioHwDev->canSetMasterVolume()) { 960 mMasterVolume = 1.0; 961 } 962 963 if (mOutput->audioHwDev->canSetMasterMute()) { 964 mMasterMute = false; 965 } 966 } 967 968 readOutputParameters(); 969 970 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 971 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 972 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 973 stream = (audio_stream_type_t) (stream + 1)) { 974 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 975 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 976 } 977 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 978 // because mAudioFlinger doesn't have one to copy from 979} 980 981AudioFlinger::PlaybackThread::~PlaybackThread() 982{ 983 mAudioFlinger->unregisterWriter(mNBLogWriter); 984 delete [] mMixBuffer; 985} 986 987void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 988{ 989 dumpInternals(fd, args); 990 dumpTracks(fd, args); 991 dumpEffectChains(fd, args); 992} 993 994void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 995{ 996 const size_t SIZE = 256; 997 char buffer[SIZE]; 998 String8 result; 999 1000 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1001 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1002 const stream_type_t *st = &mStreamTypes[i]; 1003 if (i > 0) { 1004 result.appendFormat(", "); 1005 } 1006 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1007 if (st->mute) { 1008 result.append("M"); 1009 } 1010 } 1011 result.append("\n"); 1012 write(fd, result.string(), result.length()); 1013 result.clear(); 1014 1015 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1016 result.append(buffer); 1017 Track::appendDumpHeader(result); 1018 for (size_t i = 0; i < mTracks.size(); ++i) { 1019 sp<Track> track = mTracks[i]; 1020 if (track != 0) { 1021 track->dump(buffer, SIZE); 1022 result.append(buffer); 1023 } 1024 } 1025 1026 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1027 result.append(buffer); 1028 Track::appendDumpHeader(result); 1029 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1030 sp<Track> track = mActiveTracks[i].promote(); 1031 if (track != 0) { 1032 track->dump(buffer, SIZE); 1033 result.append(buffer); 1034 } 1035 } 1036 write(fd, result.string(), result.size()); 1037 1038 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1039 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1040 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1041 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1042} 1043 1044void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1045{ 1046 const size_t SIZE = 256; 1047 char buffer[SIZE]; 1048 String8 result; 1049 1050 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1051 result.append(buffer); 1052 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1053 ns2ms(systemTime() - mLastWriteTime)); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1056 result.append(buffer); 1057 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1058 result.append(buffer); 1059 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1060 result.append(buffer); 1061 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1062 result.append(buffer); 1063 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1064 result.append(buffer); 1065 write(fd, result.string(), result.size()); 1066 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1067 1068 dumpBase(fd, args); 1069} 1070 1071// Thread virtuals 1072status_t AudioFlinger::PlaybackThread::readyToRun() 1073{ 1074 status_t status = initCheck(); 1075 if (status == NO_ERROR) { 1076 ALOGI("AudioFlinger's thread %p ready to run", this); 1077 } else { 1078 ALOGE("No working audio driver found."); 1079 } 1080 return status; 1081} 1082 1083void AudioFlinger::PlaybackThread::onFirstRef() 1084{ 1085 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1086} 1087 1088// ThreadBase virtuals 1089void AudioFlinger::PlaybackThread::preExit() 1090{ 1091 ALOGV(" preExit()"); 1092 // FIXME this is using hard-coded strings but in the future, this functionality will be 1093 // converted to use audio HAL extensions required to support tunneling 1094 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1095} 1096 1097// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1098sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1099 const sp<AudioFlinger::Client>& client, 1100 audio_stream_type_t streamType, 1101 uint32_t sampleRate, 1102 audio_format_t format, 1103 audio_channel_mask_t channelMask, 1104 size_t frameCount, 1105 const sp<IMemory>& sharedBuffer, 1106 int sessionId, 1107 IAudioFlinger::track_flags_t *flags, 1108 pid_t tid, 1109 status_t *status) 1110{ 1111 sp<Track> track; 1112 status_t lStatus; 1113 1114 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1115 1116 // client expresses a preference for FAST, but we get the final say 1117 if (*flags & IAudioFlinger::TRACK_FAST) { 1118 if ( 1119 // not timed 1120 (!isTimed) && 1121 // either of these use cases: 1122 ( 1123 // use case 1: shared buffer with any frame count 1124 ( 1125 (sharedBuffer != 0) 1126 ) || 1127 // use case 2: callback handler and frame count is default or at least as large as HAL 1128 ( 1129 (tid != -1) && 1130 ((frameCount == 0) || 1131 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1132 ) 1133 ) && 1134 // PCM data 1135 audio_is_linear_pcm(format) && 1136 // mono or stereo 1137 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1138 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1139#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1140 // hardware sample rate 1141 (sampleRate == mSampleRate) && 1142#endif 1143 // normal mixer has an associated fast mixer 1144 hasFastMixer() && 1145 // there are sufficient fast track slots available 1146 (mFastTrackAvailMask != 0) 1147 // FIXME test that MixerThread for this fast track has a capable output HAL 1148 // FIXME add a permission test also? 1149 ) { 1150 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1151 if (frameCount == 0) { 1152 frameCount = mFrameCount * kFastTrackMultiplier; 1153 } 1154 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1155 frameCount, mFrameCount); 1156 } else { 1157 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1158 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1159 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1160 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1161 audio_is_linear_pcm(format), 1162 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1163 *flags &= ~IAudioFlinger::TRACK_FAST; 1164 // For compatibility with AudioTrack calculation, buffer depth is forced 1165 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1166 // This is probably too conservative, but legacy application code may depend on it. 1167 // If you change this calculation, also review the start threshold which is related. 1168 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1169 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1170 if (minBufCount < 2) { 1171 minBufCount = 2; 1172 } 1173 size_t minFrameCount = mNormalFrameCount * minBufCount; 1174 if (frameCount < minFrameCount) { 1175 frameCount = minFrameCount; 1176 } 1177 } 1178 } 1179 1180 if (mType == DIRECT) { 1181 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1182 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1183 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1184 "for output %p with format %d", 1185 sampleRate, format, channelMask, mOutput, mFormat); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 } 1190 } else { 1191 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1192 if (sampleRate > mSampleRate*2) { 1193 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1194 lStatus = BAD_VALUE; 1195 goto Exit; 1196 } 1197 } 1198 1199 lStatus = initCheck(); 1200 if (lStatus != NO_ERROR) { 1201 ALOGE("Audio driver not initialized."); 1202 goto Exit; 1203 } 1204 1205 { // scope for mLock 1206 Mutex::Autolock _l(mLock); 1207 1208 // all tracks in same audio session must share the same routing strategy otherwise 1209 // conflicts will happen when tracks are moved from one output to another by audio policy 1210 // manager 1211 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1212 for (size_t i = 0; i < mTracks.size(); ++i) { 1213 sp<Track> t = mTracks[i]; 1214 if (t != 0 && !t->isOutputTrack()) { 1215 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1216 if (sessionId == t->sessionId() && strategy != actual) { 1217 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1218 strategy, actual); 1219 lStatus = BAD_VALUE; 1220 goto Exit; 1221 } 1222 } 1223 } 1224 1225 if (!isTimed) { 1226 track = new Track(this, client, streamType, sampleRate, format, 1227 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1228 } else { 1229 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1230 channelMask, frameCount, sharedBuffer, sessionId); 1231 } 1232 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1233 lStatus = NO_MEMORY; 1234 goto Exit; 1235 } 1236 mTracks.add(track); 1237 1238 sp<EffectChain> chain = getEffectChain_l(sessionId); 1239 if (chain != 0) { 1240 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1241 track->setMainBuffer(chain->inBuffer()); 1242 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1243 chain->incTrackCnt(); 1244 } 1245 1246 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1247 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1248 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1249 // so ask activity manager to do this on our behalf 1250 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1251 } 1252 } 1253 1254 lStatus = NO_ERROR; 1255 1256Exit: 1257 if (status) { 1258 *status = lStatus; 1259 } 1260 return track; 1261} 1262 1263uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1264{ 1265 return latency; 1266} 1267 1268uint32_t AudioFlinger::PlaybackThread::latency() const 1269{ 1270 Mutex::Autolock _l(mLock); 1271 return latency_l(); 1272} 1273uint32_t AudioFlinger::PlaybackThread::latency_l() const 1274{ 1275 if (initCheck() == NO_ERROR) { 1276 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1277 } else { 1278 return 0; 1279 } 1280} 1281 1282void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1283{ 1284 Mutex::Autolock _l(mLock); 1285 // Don't apply master volume in SW if our HAL can do it for us. 1286 if (mOutput && mOutput->audioHwDev && 1287 mOutput->audioHwDev->canSetMasterVolume()) { 1288 mMasterVolume = 1.0; 1289 } else { 1290 mMasterVolume = value; 1291 } 1292} 1293 1294void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1295{ 1296 Mutex::Autolock _l(mLock); 1297 // Don't apply master mute in SW if our HAL can do it for us. 1298 if (mOutput && mOutput->audioHwDev && 1299 mOutput->audioHwDev->canSetMasterMute()) { 1300 mMasterMute = false; 1301 } else { 1302 mMasterMute = muted; 1303 } 1304} 1305 1306void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1307{ 1308 Mutex::Autolock _l(mLock); 1309 mStreamTypes[stream].volume = value; 1310} 1311 1312void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1313{ 1314 Mutex::Autolock _l(mLock); 1315 mStreamTypes[stream].mute = muted; 1316} 1317 1318float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1319{ 1320 Mutex::Autolock _l(mLock); 1321 return mStreamTypes[stream].volume; 1322} 1323 1324// addTrack_l() must be called with ThreadBase::mLock held 1325status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1326{ 1327 status_t status = ALREADY_EXISTS; 1328 1329 // set retry count for buffer fill 1330 track->mRetryCount = kMaxTrackStartupRetries; 1331 if (mActiveTracks.indexOf(track) < 0) { 1332 // the track is newly added, make sure it fills up all its 1333 // buffers before playing. This is to ensure the client will 1334 // effectively get the latency it requested. 1335 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1336 track->mResetDone = false; 1337 track->mPresentationCompleteFrames = 0; 1338 mActiveTracks.add(track); 1339 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1340 if (chain != 0) { 1341 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1342 track->sessionId()); 1343 chain->incActiveTrackCnt(); 1344 } 1345 1346 status = NO_ERROR; 1347 } 1348 1349 ALOGV("mWaitWorkCV.broadcast"); 1350 mWaitWorkCV.broadcast(); 1351 1352 return status; 1353} 1354 1355// destroyTrack_l() must be called with ThreadBase::mLock held 1356void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1357{ 1358 track->mState = TrackBase::TERMINATED; 1359 // active tracks are removed by threadLoop() 1360 if (mActiveTracks.indexOf(track) < 0) { 1361 removeTrack_l(track); 1362 } 1363} 1364 1365void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1366{ 1367 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1368 mTracks.remove(track); 1369 deleteTrackName_l(track->name()); 1370 // redundant as track is about to be destroyed, for dumpsys only 1371 track->mName = -1; 1372 if (track->isFastTrack()) { 1373 int index = track->mFastIndex; 1374 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1375 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1376 mFastTrackAvailMask |= 1 << index; 1377 // redundant as track is about to be destroyed, for dumpsys only 1378 track->mFastIndex = -1; 1379 } 1380 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1381 if (chain != 0) { 1382 chain->decTrackCnt(); 1383 } 1384} 1385 1386String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1387{ 1388 String8 out_s8 = String8(""); 1389 char *s; 1390 1391 Mutex::Autolock _l(mLock); 1392 if (initCheck() != NO_ERROR) { 1393 return out_s8; 1394 } 1395 1396 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1397 out_s8 = String8(s); 1398 free(s); 1399 return out_s8; 1400} 1401 1402// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1403void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1404 AudioSystem::OutputDescriptor desc; 1405 void *param2 = NULL; 1406 1407 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1408 param); 1409 1410 switch (event) { 1411 case AudioSystem::OUTPUT_OPENED: 1412 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1413 desc.channels = mChannelMask; 1414 desc.samplingRate = mSampleRate; 1415 desc.format = mFormat; 1416 desc.frameCount = mNormalFrameCount; // FIXME see 1417 // AudioFlinger::frameCount(audio_io_handle_t) 1418 desc.latency = latency(); 1419 param2 = &desc; 1420 break; 1421 1422 case AudioSystem::STREAM_CONFIG_CHANGED: 1423 param2 = ¶m; 1424 case AudioSystem::OUTPUT_CLOSED: 1425 default: 1426 break; 1427 } 1428 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1429} 1430 1431void AudioFlinger::PlaybackThread::readOutputParameters() 1432{ 1433 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1434 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1435 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1436 if (!audio_is_output_channel(mChannelMask)) { 1437 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1438 } 1439 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1440 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1441 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1442 } 1443 mChannelCount = (uint16_t)popcount(mChannelMask); 1444 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1445 if (!audio_is_valid_format(mFormat)) { 1446 LOG_FATAL("HAL format %d not valid for output", mFormat); 1447 } 1448 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1449 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1450 mFormat); 1451 } 1452 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1453 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1454 if (mFrameCount & 15) { 1455 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1456 mFrameCount); 1457 } 1458 1459 // Calculate size of normal mix buffer relative to the HAL output buffer size 1460 double multiplier = 1.0; 1461 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1462 kUseFastMixer == FastMixer_Dynamic)) { 1463 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1464 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1465 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1466 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1467 maxNormalFrameCount = maxNormalFrameCount & ~15; 1468 if (maxNormalFrameCount < minNormalFrameCount) { 1469 maxNormalFrameCount = minNormalFrameCount; 1470 } 1471 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1472 if (multiplier <= 1.0) { 1473 multiplier = 1.0; 1474 } else if (multiplier <= 2.0) { 1475 if (2 * mFrameCount <= maxNormalFrameCount) { 1476 multiplier = 2.0; 1477 } else { 1478 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1479 } 1480 } else { 1481 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1482 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1483 // track, but we sometimes have to do this to satisfy the maximum frame count 1484 // constraint) 1485 // FIXME this rounding up should not be done if no HAL SRC 1486 uint32_t truncMult = (uint32_t) multiplier; 1487 if ((truncMult & 1)) { 1488 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1489 ++truncMult; 1490 } 1491 } 1492 multiplier = (double) truncMult; 1493 } 1494 } 1495 mNormalFrameCount = multiplier * mFrameCount; 1496 // round up to nearest 16 frames to satisfy AudioMixer 1497 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1498 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1499 mNormalFrameCount); 1500 1501 delete[] mMixBuffer; 1502 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1503 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1504 1505 // force reconfiguration of effect chains and engines to take new buffer size and audio 1506 // parameters into account 1507 // Note that mLock is not held when readOutputParameters() is called from the constructor 1508 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1509 // matter. 1510 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1511 Vector< sp<EffectChain> > effectChains = mEffectChains; 1512 for (size_t i = 0; i < effectChains.size(); i ++) { 1513 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1514 } 1515} 1516 1517 1518status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1519{ 1520 if (halFrames == NULL || dspFrames == NULL) { 1521 return BAD_VALUE; 1522 } 1523 Mutex::Autolock _l(mLock); 1524 if (initCheck() != NO_ERROR) { 1525 return INVALID_OPERATION; 1526 } 1527 size_t framesWritten = mBytesWritten / mFrameSize; 1528 *halFrames = framesWritten; 1529 1530 if (isSuspended()) { 1531 // return an estimation of rendered frames when the output is suspended 1532 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1533 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1534 return NO_ERROR; 1535 } else { 1536 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1537 } 1538} 1539 1540uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1541{ 1542 Mutex::Autolock _l(mLock); 1543 uint32_t result = 0; 1544 if (getEffectChain_l(sessionId) != 0) { 1545 result = EFFECT_SESSION; 1546 } 1547 1548 for (size_t i = 0; i < mTracks.size(); ++i) { 1549 sp<Track> track = mTracks[i]; 1550 if (sessionId == track->sessionId() && !track->isInvalid()) { 1551 result |= TRACK_SESSION; 1552 break; 1553 } 1554 } 1555 1556 return result; 1557} 1558 1559uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1560{ 1561 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1562 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1563 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1564 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1565 } 1566 for (size_t i = 0; i < mTracks.size(); i++) { 1567 sp<Track> track = mTracks[i]; 1568 if (sessionId == track->sessionId() && !track->isInvalid()) { 1569 return AudioSystem::getStrategyForStream(track->streamType()); 1570 } 1571 } 1572 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1573} 1574 1575 1576AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1577{ 1578 Mutex::Autolock _l(mLock); 1579 return mOutput; 1580} 1581 1582AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1583{ 1584 Mutex::Autolock _l(mLock); 1585 AudioStreamOut *output = mOutput; 1586 mOutput = NULL; 1587 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1588 // must push a NULL and wait for ack 1589 mOutputSink.clear(); 1590 mPipeSink.clear(); 1591 mNormalSink.clear(); 1592 return output; 1593} 1594 1595// this method must always be called either with ThreadBase mLock held or inside the thread loop 1596audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1597{ 1598 if (mOutput == NULL) { 1599 return NULL; 1600 } 1601 return &mOutput->stream->common; 1602} 1603 1604uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1605{ 1606 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1607} 1608 1609status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1610{ 1611 if (!isValidSyncEvent(event)) { 1612 return BAD_VALUE; 1613 } 1614 1615 Mutex::Autolock _l(mLock); 1616 1617 for (size_t i = 0; i < mTracks.size(); ++i) { 1618 sp<Track> track = mTracks[i]; 1619 if (event->triggerSession() == track->sessionId()) { 1620 (void) track->setSyncEvent(event); 1621 return NO_ERROR; 1622 } 1623 } 1624 1625 return NAME_NOT_FOUND; 1626} 1627 1628bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1629{ 1630 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1631} 1632 1633void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1634 const Vector< sp<Track> >& tracksToRemove) 1635{ 1636 size_t count = tracksToRemove.size(); 1637 if (CC_UNLIKELY(count)) { 1638 for (size_t i = 0 ; i < count ; i++) { 1639 const sp<Track>& track = tracksToRemove.itemAt(i); 1640 if ((track->sharedBuffer() != 0) && 1641 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1642 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1643 } 1644 } 1645 } 1646 1647} 1648 1649void AudioFlinger::PlaybackThread::checkSilentMode_l() 1650{ 1651 if (!mMasterMute) { 1652 char value[PROPERTY_VALUE_MAX]; 1653 if (property_get("ro.audio.silent", value, "0") > 0) { 1654 char *endptr; 1655 unsigned long ul = strtoul(value, &endptr, 0); 1656 if (*endptr == '\0' && ul != 0) { 1657 ALOGD("Silence is golden"); 1658 // The setprop command will not allow a property to be changed after 1659 // the first time it is set, so we don't have to worry about un-muting. 1660 setMasterMute_l(true); 1661 } 1662 } 1663 } 1664} 1665 1666// shared by MIXER and DIRECT, overridden by DUPLICATING 1667void AudioFlinger::PlaybackThread::threadLoop_write() 1668{ 1669 // FIXME rewrite to reduce number of system calls 1670 mLastWriteTime = systemTime(); 1671 mInWrite = true; 1672 int bytesWritten; 1673 1674 // If an NBAIO sink is present, use it to write the normal mixer's submix 1675 if (mNormalSink != 0) { 1676#define mBitShift 2 // FIXME 1677 size_t count = mixBufferSize >> mBitShift; 1678 ATRACE_BEGIN("write"); 1679 // update the setpoint when AudioFlinger::mScreenState changes 1680 uint32_t screenState = AudioFlinger::mScreenState; 1681 if (screenState != mScreenState) { 1682 mScreenState = screenState; 1683 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1684 if (pipe != NULL) { 1685 pipe->setAvgFrames((mScreenState & 1) ? 1686 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1687 } 1688 } 1689 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1690 ATRACE_END(); 1691 if (framesWritten > 0) { 1692 bytesWritten = framesWritten << mBitShift; 1693 } else { 1694 bytesWritten = framesWritten; 1695 } 1696 // otherwise use the HAL / AudioStreamOut directly 1697 } else { 1698 // Direct output thread. 1699 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1700 } 1701 1702 if (bytesWritten > 0) { 1703 mBytesWritten += mixBufferSize; 1704 } 1705 mNumWrites++; 1706 mInWrite = false; 1707} 1708 1709/* 1710The derived values that are cached: 1711 - mixBufferSize from frame count * frame size 1712 - activeSleepTime from activeSleepTimeUs() 1713 - idleSleepTime from idleSleepTimeUs() 1714 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1715 - maxPeriod from frame count and sample rate (MIXER only) 1716 1717The parameters that affect these derived values are: 1718 - frame count 1719 - frame size 1720 - sample rate 1721 - device type: A2DP or not 1722 - device latency 1723 - format: PCM or not 1724 - active sleep time 1725 - idle sleep time 1726*/ 1727 1728void AudioFlinger::PlaybackThread::cacheParameters_l() 1729{ 1730 mixBufferSize = mNormalFrameCount * mFrameSize; 1731 activeSleepTime = activeSleepTimeUs(); 1732 idleSleepTime = idleSleepTimeUs(); 1733} 1734 1735void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1736{ 1737 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1738 this, streamType, mTracks.size()); 1739 Mutex::Autolock _l(mLock); 1740 1741 size_t size = mTracks.size(); 1742 for (size_t i = 0; i < size; i++) { 1743 sp<Track> t = mTracks[i]; 1744 if (t->streamType() == streamType) { 1745 t->invalidate(); 1746 } 1747 } 1748} 1749 1750status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1751{ 1752 int session = chain->sessionId(); 1753 int16_t *buffer = mMixBuffer; 1754 bool ownsBuffer = false; 1755 1756 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1757 if (session > 0) { 1758 // Only one effect chain can be present in direct output thread and it uses 1759 // the mix buffer as input 1760 if (mType != DIRECT) { 1761 size_t numSamples = mNormalFrameCount * mChannelCount; 1762 buffer = new int16_t[numSamples]; 1763 memset(buffer, 0, numSamples * sizeof(int16_t)); 1764 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1765 ownsBuffer = true; 1766 } 1767 1768 // Attach all tracks with same session ID to this chain. 1769 for (size_t i = 0; i < mTracks.size(); ++i) { 1770 sp<Track> track = mTracks[i]; 1771 if (session == track->sessionId()) { 1772 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1773 buffer); 1774 track->setMainBuffer(buffer); 1775 chain->incTrackCnt(); 1776 } 1777 } 1778 1779 // indicate all active tracks in the chain 1780 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1781 sp<Track> track = mActiveTracks[i].promote(); 1782 if (track == 0) { 1783 continue; 1784 } 1785 if (session == track->sessionId()) { 1786 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1787 chain->incActiveTrackCnt(); 1788 } 1789 } 1790 } 1791 1792 chain->setInBuffer(buffer, ownsBuffer); 1793 chain->setOutBuffer(mMixBuffer); 1794 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1795 // chains list in order to be processed last as it contains output stage effects 1796 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1797 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1798 // after track specific effects and before output stage 1799 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1800 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1801 // Effect chain for other sessions are inserted at beginning of effect 1802 // chains list to be processed before output mix effects. Relative order between other 1803 // sessions is not important 1804 size_t size = mEffectChains.size(); 1805 size_t i = 0; 1806 for (i = 0; i < size; i++) { 1807 if (mEffectChains[i]->sessionId() < session) { 1808 break; 1809 } 1810 } 1811 mEffectChains.insertAt(chain, i); 1812 checkSuspendOnAddEffectChain_l(chain); 1813 1814 return NO_ERROR; 1815} 1816 1817size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1818{ 1819 int session = chain->sessionId(); 1820 1821 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1822 1823 for (size_t i = 0; i < mEffectChains.size(); i++) { 1824 if (chain == mEffectChains[i]) { 1825 mEffectChains.removeAt(i); 1826 // detach all active tracks from the chain 1827 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1828 sp<Track> track = mActiveTracks[i].promote(); 1829 if (track == 0) { 1830 continue; 1831 } 1832 if (session == track->sessionId()) { 1833 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1834 chain.get(), session); 1835 chain->decActiveTrackCnt(); 1836 } 1837 } 1838 1839 // detach all tracks with same session ID from this chain 1840 for (size_t i = 0; i < mTracks.size(); ++i) { 1841 sp<Track> track = mTracks[i]; 1842 if (session == track->sessionId()) { 1843 track->setMainBuffer(mMixBuffer); 1844 chain->decTrackCnt(); 1845 } 1846 } 1847 break; 1848 } 1849 } 1850 return mEffectChains.size(); 1851} 1852 1853status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1854 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1855{ 1856 Mutex::Autolock _l(mLock); 1857 return attachAuxEffect_l(track, EffectId); 1858} 1859 1860status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1861 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1862{ 1863 status_t status = NO_ERROR; 1864 1865 if (EffectId == 0) { 1866 track->setAuxBuffer(0, NULL); 1867 } else { 1868 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1869 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1870 if (effect != 0) { 1871 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1872 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1873 } else { 1874 status = INVALID_OPERATION; 1875 } 1876 } else { 1877 status = BAD_VALUE; 1878 } 1879 } 1880 return status; 1881} 1882 1883void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1884{ 1885 for (size_t i = 0; i < mTracks.size(); ++i) { 1886 sp<Track> track = mTracks[i]; 1887 if (track->auxEffectId() == effectId) { 1888 attachAuxEffect_l(track, 0); 1889 } 1890 } 1891} 1892 1893bool AudioFlinger::PlaybackThread::threadLoop() 1894{ 1895 Vector< sp<Track> > tracksToRemove; 1896 1897 standbyTime = systemTime(); 1898 1899 // MIXER 1900 nsecs_t lastWarning = 0; 1901 1902 // DUPLICATING 1903 // FIXME could this be made local to while loop? 1904 writeFrames = 0; 1905 1906 cacheParameters_l(); 1907 sleepTime = idleSleepTime; 1908 1909 if (mType == MIXER) { 1910 sleepTimeShift = 0; 1911 } 1912 1913 CpuStats cpuStats; 1914 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1915 1916 acquireWakeLock(); 1917 1918 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1919 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1920 // and then that string will be logged at the next convenient opportunity. 1921 const char *logString = NULL; 1922 1923 while (!exitPending()) 1924 { 1925 cpuStats.sample(myName); 1926 1927 Vector< sp<EffectChain> > effectChains; 1928 1929 processConfigEvents(); 1930 1931 { // scope for mLock 1932 1933 Mutex::Autolock _l(mLock); 1934 1935 if (logString != NULL) { 1936 mNBLogWriter->logTimestamp(); 1937 mNBLogWriter->log(logString); 1938 logString = NULL; 1939 } 1940 1941 if (checkForNewParameters_l()) { 1942 cacheParameters_l(); 1943 } 1944 1945 saveOutputTracks(); 1946 1947 // put audio hardware into standby after short delay 1948 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1949 isSuspended())) { 1950 if (!mStandby) { 1951 1952 threadLoop_standby(); 1953 1954 mStandby = true; 1955 } 1956 1957 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1958 // we're about to wait, flush the binder command buffer 1959 IPCThreadState::self()->flushCommands(); 1960 1961 clearOutputTracks(); 1962 1963 if (exitPending()) { 1964 break; 1965 } 1966 1967 releaseWakeLock_l(); 1968 // wait until we have something to do... 1969 ALOGV("%s going to sleep", myName.string()); 1970 mWaitWorkCV.wait(mLock); 1971 ALOGV("%s waking up", myName.string()); 1972 acquireWakeLock_l(); 1973 1974 mMixerStatus = MIXER_IDLE; 1975 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1976 mBytesWritten = 0; 1977 1978 checkSilentMode_l(); 1979 1980 standbyTime = systemTime() + standbyDelay; 1981 sleepTime = idleSleepTime; 1982 if (mType == MIXER) { 1983 sleepTimeShift = 0; 1984 } 1985 1986 continue; 1987 } 1988 } 1989 1990 // mMixerStatusIgnoringFastTracks is also updated internally 1991 mMixerStatus = prepareTracks_l(&tracksToRemove); 1992 1993 // prevent any changes in effect chain list and in each effect chain 1994 // during mixing and effect process as the audio buffers could be deleted 1995 // or modified if an effect is created or deleted 1996 lockEffectChains_l(effectChains); 1997 } 1998 1999 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2000 threadLoop_mix(); 2001 } else { 2002 threadLoop_sleepTime(); 2003 } 2004 2005 if (isSuspended()) { 2006 sleepTime = suspendSleepTimeUs(); 2007 mBytesWritten += mixBufferSize; 2008 } 2009 2010 // only process effects if we're going to write 2011 if (sleepTime == 0) { 2012 for (size_t i = 0; i < effectChains.size(); i ++) { 2013 effectChains[i]->process_l(); 2014 } 2015 } 2016 2017 // enable changes in effect chain 2018 unlockEffectChains(effectChains); 2019 2020 // sleepTime == 0 means we must write to audio hardware 2021 if (sleepTime == 0) { 2022 2023 threadLoop_write(); 2024 2025if (mType == MIXER) { 2026 // write blocked detection 2027 nsecs_t now = systemTime(); 2028 nsecs_t delta = now - mLastWriteTime; 2029 if (!mStandby && delta > maxPeriod) { 2030 mNumDelayedWrites++; 2031 if ((now - lastWarning) > kWarningThrottleNs) { 2032 ATRACE_NAME("underrun"); 2033 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2034 ns2ms(delta), mNumDelayedWrites, this); 2035 lastWarning = now; 2036 } 2037 } 2038} 2039 2040 mStandby = false; 2041 } else { 2042 usleep(sleepTime); 2043 } 2044 2045 // Finally let go of removed track(s), without the lock held 2046 // since we can't guarantee the destructors won't acquire that 2047 // same lock. This will also mutate and push a new fast mixer state. 2048 threadLoop_removeTracks(tracksToRemove); 2049 tracksToRemove.clear(); 2050 2051 // FIXME I don't understand the need for this here; 2052 // it was in the original code but maybe the 2053 // assignment in saveOutputTracks() makes this unnecessary? 2054 clearOutputTracks(); 2055 2056 // Effect chains will be actually deleted here if they were removed from 2057 // mEffectChains list during mixing or effects processing 2058 effectChains.clear(); 2059 2060 // FIXME Note that the above .clear() is no longer necessary since effectChains 2061 // is now local to this block, but will keep it for now (at least until merge done). 2062 } 2063 2064 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2065 if (mType == MIXER || mType == DIRECT) { 2066 // put output stream into standby mode 2067 if (!mStandby) { 2068 mOutput->stream->common.standby(&mOutput->stream->common); 2069 } 2070 } 2071 2072 releaseWakeLock(); 2073 2074 ALOGV("Thread %p type %d exiting", this, mType); 2075 return false; 2076} 2077 2078 2079// ---------------------------------------------------------------------------- 2080 2081AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2082 audio_io_handle_t id, audio_devices_t device, type_t type) 2083 : PlaybackThread(audioFlinger, output, id, device, type), 2084 // mAudioMixer below 2085 // mFastMixer below 2086 mFastMixerFutex(0) 2087 // mOutputSink below 2088 // mPipeSink below 2089 // mNormalSink below 2090{ 2091 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2092 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2093 "mFrameCount=%d, mNormalFrameCount=%d", 2094 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2095 mNormalFrameCount); 2096 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2097 2098 // FIXME - Current mixer implementation only supports stereo output 2099 if (mChannelCount != FCC_2) { 2100 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2101 } 2102 2103 // create an NBAIO sink for the HAL output stream, and negotiate 2104 mOutputSink = new AudioStreamOutSink(output->stream); 2105 size_t numCounterOffers = 0; 2106 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2107 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2108 ALOG_ASSERT(index == 0); 2109 2110 // initialize fast mixer depending on configuration 2111 bool initFastMixer; 2112 switch (kUseFastMixer) { 2113 case FastMixer_Never: 2114 initFastMixer = false; 2115 break; 2116 case FastMixer_Always: 2117 initFastMixer = true; 2118 break; 2119 case FastMixer_Static: 2120 case FastMixer_Dynamic: 2121 initFastMixer = mFrameCount < mNormalFrameCount; 2122 break; 2123 } 2124 if (initFastMixer) { 2125 2126 // create a MonoPipe to connect our submix to FastMixer 2127 NBAIO_Format format = mOutputSink->format(); 2128 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2129 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2130 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2131 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2132 const NBAIO_Format offers[1] = {format}; 2133 size_t numCounterOffers = 0; 2134 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2135 ALOG_ASSERT(index == 0); 2136 monoPipe->setAvgFrames((mScreenState & 1) ? 2137 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2138 mPipeSink = monoPipe; 2139 2140#ifdef TEE_SINK 2141 if (mTeeSinkOutputEnabled) { 2142 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2143 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2144 numCounterOffers = 0; 2145 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2146 ALOG_ASSERT(index == 0); 2147 mTeeSink = teeSink; 2148 PipeReader *teeSource = new PipeReader(*teeSink); 2149 numCounterOffers = 0; 2150 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2151 ALOG_ASSERT(index == 0); 2152 mTeeSource = teeSource; 2153 } 2154#endif 2155 2156 // create fast mixer and configure it initially with just one fast track for our submix 2157 mFastMixer = new FastMixer(); 2158 FastMixerStateQueue *sq = mFastMixer->sq(); 2159#ifdef STATE_QUEUE_DUMP 2160 sq->setObserverDump(&mStateQueueObserverDump); 2161 sq->setMutatorDump(&mStateQueueMutatorDump); 2162#endif 2163 FastMixerState *state = sq->begin(); 2164 FastTrack *fastTrack = &state->mFastTracks[0]; 2165 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2166 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2167 fastTrack->mVolumeProvider = NULL; 2168 fastTrack->mGeneration++; 2169 state->mFastTracksGen++; 2170 state->mTrackMask = 1; 2171 // fast mixer will use the HAL output sink 2172 state->mOutputSink = mOutputSink.get(); 2173 state->mOutputSinkGen++; 2174 state->mFrameCount = mFrameCount; 2175 state->mCommand = FastMixerState::COLD_IDLE; 2176 // already done in constructor initialization list 2177 //mFastMixerFutex = 0; 2178 state->mColdFutexAddr = &mFastMixerFutex; 2179 state->mColdGen++; 2180 state->mDumpState = &mFastMixerDumpState; 2181#ifdef TEE_SINK 2182 state->mTeeSink = mTeeSink.get(); 2183#endif 2184 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2185 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2186 sq->end(); 2187 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2188 2189 // start the fast mixer 2190 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2191 pid_t tid = mFastMixer->getTid(); 2192 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2193 if (err != 0) { 2194 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2195 kPriorityFastMixer, getpid_cached, tid, err); 2196 } 2197 2198#ifdef AUDIO_WATCHDOG 2199 // create and start the watchdog 2200 mAudioWatchdog = new AudioWatchdog(); 2201 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2202 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2203 tid = mAudioWatchdog->getTid(); 2204 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2205 if (err != 0) { 2206 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2207 kPriorityFastMixer, getpid_cached, tid, err); 2208 } 2209#endif 2210 2211 } else { 2212 mFastMixer = NULL; 2213 } 2214 2215 switch (kUseFastMixer) { 2216 case FastMixer_Never: 2217 case FastMixer_Dynamic: 2218 mNormalSink = mOutputSink; 2219 break; 2220 case FastMixer_Always: 2221 mNormalSink = mPipeSink; 2222 break; 2223 case FastMixer_Static: 2224 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2225 break; 2226 } 2227} 2228 2229AudioFlinger::MixerThread::~MixerThread() 2230{ 2231 if (mFastMixer != NULL) { 2232 FastMixerStateQueue *sq = mFastMixer->sq(); 2233 FastMixerState *state = sq->begin(); 2234 if (state->mCommand == FastMixerState::COLD_IDLE) { 2235 int32_t old = android_atomic_inc(&mFastMixerFutex); 2236 if (old == -1) { 2237 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2238 } 2239 } 2240 state->mCommand = FastMixerState::EXIT; 2241 sq->end(); 2242 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2243 mFastMixer->join(); 2244 // Though the fast mixer thread has exited, it's state queue is still valid. 2245 // We'll use that extract the final state which contains one remaining fast track 2246 // corresponding to our sub-mix. 2247 state = sq->begin(); 2248 ALOG_ASSERT(state->mTrackMask == 1); 2249 FastTrack *fastTrack = &state->mFastTracks[0]; 2250 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2251 delete fastTrack->mBufferProvider; 2252 sq->end(false /*didModify*/); 2253 delete mFastMixer; 2254#ifdef AUDIO_WATCHDOG 2255 if (mAudioWatchdog != 0) { 2256 mAudioWatchdog->requestExit(); 2257 mAudioWatchdog->requestExitAndWait(); 2258 mAudioWatchdog.clear(); 2259 } 2260#endif 2261 } 2262 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2263 delete mAudioMixer; 2264} 2265 2266 2267uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2268{ 2269 if (mFastMixer != NULL) { 2270 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2271 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2272 } 2273 return latency; 2274} 2275 2276 2277void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2278{ 2279 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2280} 2281 2282void AudioFlinger::MixerThread::threadLoop_write() 2283{ 2284 // FIXME we should only do one push per cycle; confirm this is true 2285 // Start the fast mixer if it's not already running 2286 if (mFastMixer != NULL) { 2287 FastMixerStateQueue *sq = mFastMixer->sq(); 2288 FastMixerState *state = sq->begin(); 2289 if (state->mCommand != FastMixerState::MIX_WRITE && 2290 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2291 if (state->mCommand == FastMixerState::COLD_IDLE) { 2292 int32_t old = android_atomic_inc(&mFastMixerFutex); 2293 if (old == -1) { 2294 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2295 } 2296#ifdef AUDIO_WATCHDOG 2297 if (mAudioWatchdog != 0) { 2298 mAudioWatchdog->resume(); 2299 } 2300#endif 2301 } 2302 state->mCommand = FastMixerState::MIX_WRITE; 2303 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2304 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2305 sq->end(); 2306 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2307 if (kUseFastMixer == FastMixer_Dynamic) { 2308 mNormalSink = mPipeSink; 2309 } 2310 } else { 2311 sq->end(false /*didModify*/); 2312 } 2313 } 2314 PlaybackThread::threadLoop_write(); 2315} 2316 2317void AudioFlinger::MixerThread::threadLoop_standby() 2318{ 2319 // Idle the fast mixer if it's currently running 2320 if (mFastMixer != NULL) { 2321 FastMixerStateQueue *sq = mFastMixer->sq(); 2322 FastMixerState *state = sq->begin(); 2323 if (!(state->mCommand & FastMixerState::IDLE)) { 2324 state->mCommand = FastMixerState::COLD_IDLE; 2325 state->mColdFutexAddr = &mFastMixerFutex; 2326 state->mColdGen++; 2327 mFastMixerFutex = 0; 2328 sq->end(); 2329 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2330 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2331 if (kUseFastMixer == FastMixer_Dynamic) { 2332 mNormalSink = mOutputSink; 2333 } 2334#ifdef AUDIO_WATCHDOG 2335 if (mAudioWatchdog != 0) { 2336 mAudioWatchdog->pause(); 2337 } 2338#endif 2339 } else { 2340 sq->end(false /*didModify*/); 2341 } 2342 } 2343 PlaybackThread::threadLoop_standby(); 2344} 2345 2346// shared by MIXER and DIRECT, overridden by DUPLICATING 2347void AudioFlinger::PlaybackThread::threadLoop_standby() 2348{ 2349 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2350 mOutput->stream->common.standby(&mOutput->stream->common); 2351} 2352 2353void AudioFlinger::MixerThread::threadLoop_mix() 2354{ 2355 // obtain the presentation timestamp of the next output buffer 2356 int64_t pts; 2357 status_t status = INVALID_OPERATION; 2358 2359 if (mNormalSink != 0) { 2360 status = mNormalSink->getNextWriteTimestamp(&pts); 2361 } else { 2362 status = mOutputSink->getNextWriteTimestamp(&pts); 2363 } 2364 2365 if (status != NO_ERROR) { 2366 pts = AudioBufferProvider::kInvalidPTS; 2367 } 2368 2369 // mix buffers... 2370 mAudioMixer->process(pts); 2371 // increase sleep time progressively when application underrun condition clears. 2372 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2373 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2374 // such that we would underrun the audio HAL. 2375 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2376 sleepTimeShift--; 2377 } 2378 sleepTime = 0; 2379 standbyTime = systemTime() + standbyDelay; 2380 //TODO: delay standby when effects have a tail 2381} 2382 2383void AudioFlinger::MixerThread::threadLoop_sleepTime() 2384{ 2385 // If no tracks are ready, sleep once for the duration of an output 2386 // buffer size, then write 0s to the output 2387 if (sleepTime == 0) { 2388 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2389 sleepTime = activeSleepTime >> sleepTimeShift; 2390 if (sleepTime < kMinThreadSleepTimeUs) { 2391 sleepTime = kMinThreadSleepTimeUs; 2392 } 2393 // reduce sleep time in case of consecutive application underruns to avoid 2394 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2395 // duration we would end up writing less data than needed by the audio HAL if 2396 // the condition persists. 2397 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2398 sleepTimeShift++; 2399 } 2400 } else { 2401 sleepTime = idleSleepTime; 2402 } 2403 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2404 memset (mMixBuffer, 0, mixBufferSize); 2405 sleepTime = 0; 2406 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2407 "anticipated start"); 2408 } 2409 // TODO add standby time extension fct of effect tail 2410} 2411 2412// prepareTracks_l() must be called with ThreadBase::mLock held 2413AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2414 Vector< sp<Track> > *tracksToRemove) 2415{ 2416 2417 mixer_state mixerStatus = MIXER_IDLE; 2418 // find out which tracks need to be processed 2419 size_t count = mActiveTracks.size(); 2420 size_t mixedTracks = 0; 2421 size_t tracksWithEffect = 0; 2422 // counts only _active_ fast tracks 2423 size_t fastTracks = 0; 2424 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2425 2426 float masterVolume = mMasterVolume; 2427 bool masterMute = mMasterMute; 2428 2429 if (masterMute) { 2430 masterVolume = 0; 2431 } 2432 // Delegate master volume control to effect in output mix effect chain if needed 2433 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2434 if (chain != 0) { 2435 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2436 chain->setVolume_l(&v, &v); 2437 masterVolume = (float)((v + (1 << 23)) >> 24); 2438 chain.clear(); 2439 } 2440 2441 // prepare a new state to push 2442 FastMixerStateQueue *sq = NULL; 2443 FastMixerState *state = NULL; 2444 bool didModify = false; 2445 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2446 if (mFastMixer != NULL) { 2447 sq = mFastMixer->sq(); 2448 state = sq->begin(); 2449 } 2450 2451 for (size_t i=0 ; i<count ; i++) { 2452 sp<Track> t = mActiveTracks[i].promote(); 2453 if (t == 0) { 2454 continue; 2455 } 2456 2457 // this const just means the local variable doesn't change 2458 Track* const track = t.get(); 2459 2460 // process fast tracks 2461 if (track->isFastTrack()) { 2462 2463 // It's theoretically possible (though unlikely) for a fast track to be created 2464 // and then removed within the same normal mix cycle. This is not a problem, as 2465 // the track never becomes active so it's fast mixer slot is never touched. 2466 // The converse, of removing an (active) track and then creating a new track 2467 // at the identical fast mixer slot within the same normal mix cycle, 2468 // is impossible because the slot isn't marked available until the end of each cycle. 2469 int j = track->mFastIndex; 2470 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2471 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2472 FastTrack *fastTrack = &state->mFastTracks[j]; 2473 2474 // Determine whether the track is currently in underrun condition, 2475 // and whether it had a recent underrun. 2476 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2477 FastTrackUnderruns underruns = ftDump->mUnderruns; 2478 uint32_t recentFull = (underruns.mBitFields.mFull - 2479 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2480 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2481 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2482 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2483 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2484 uint32_t recentUnderruns = recentPartial + recentEmpty; 2485 track->mObservedUnderruns = underruns; 2486 // don't count underruns that occur while stopping or pausing 2487 // or stopped which can occur when flush() is called while active 2488 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2489 track->mUnderrunCount += recentUnderruns; 2490 } 2491 2492 // This is similar to the state machine for normal tracks, 2493 // with a few modifications for fast tracks. 2494 bool isActive = true; 2495 switch (track->mState) { 2496 case TrackBase::STOPPING_1: 2497 // track stays active in STOPPING_1 state until first underrun 2498 if (recentUnderruns > 0) { 2499 track->mState = TrackBase::STOPPING_2; 2500 } 2501 break; 2502 case TrackBase::PAUSING: 2503 // ramp down is not yet implemented 2504 track->setPaused(); 2505 break; 2506 case TrackBase::RESUMING: 2507 // ramp up is not yet implemented 2508 track->mState = TrackBase::ACTIVE; 2509 break; 2510 case TrackBase::ACTIVE: 2511 if (recentFull > 0 || recentPartial > 0) { 2512 // track has provided at least some frames recently: reset retry count 2513 track->mRetryCount = kMaxTrackRetries; 2514 } 2515 if (recentUnderruns == 0) { 2516 // no recent underruns: stay active 2517 break; 2518 } 2519 // there has recently been an underrun of some kind 2520 if (track->sharedBuffer() == 0) { 2521 // were any of the recent underruns "empty" (no frames available)? 2522 if (recentEmpty == 0) { 2523 // no, then ignore the partial underruns as they are allowed indefinitely 2524 break; 2525 } 2526 // there has recently been an "empty" underrun: decrement the retry counter 2527 if (--(track->mRetryCount) > 0) { 2528 break; 2529 } 2530 // indicate to client process that the track was disabled because of underrun; 2531 // it will then automatically call start() when data is available 2532 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2533 // remove from active list, but state remains ACTIVE [confusing but true] 2534 isActive = false; 2535 break; 2536 } 2537 // fall through 2538 case TrackBase::STOPPING_2: 2539 case TrackBase::PAUSED: 2540 case TrackBase::TERMINATED: 2541 case TrackBase::STOPPED: 2542 case TrackBase::FLUSHED: // flush() while active 2543 // Check for presentation complete if track is inactive 2544 // We have consumed all the buffers of this track. 2545 // This would be incomplete if we auto-paused on underrun 2546 { 2547 size_t audioHALFrames = 2548 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2549 size_t framesWritten = mBytesWritten / mFrameSize; 2550 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2551 // track stays in active list until presentation is complete 2552 break; 2553 } 2554 } 2555 if (track->isStopping_2()) { 2556 track->mState = TrackBase::STOPPED; 2557 } 2558 if (track->isStopped()) { 2559 // Can't reset directly, as fast mixer is still polling this track 2560 // track->reset(); 2561 // So instead mark this track as needing to be reset after push with ack 2562 resetMask |= 1 << i; 2563 } 2564 isActive = false; 2565 break; 2566 case TrackBase::IDLE: 2567 default: 2568 LOG_FATAL("unexpected track state %d", track->mState); 2569 } 2570 2571 if (isActive) { 2572 // was it previously inactive? 2573 if (!(state->mTrackMask & (1 << j))) { 2574 ExtendedAudioBufferProvider *eabp = track; 2575 VolumeProvider *vp = track; 2576 fastTrack->mBufferProvider = eabp; 2577 fastTrack->mVolumeProvider = vp; 2578 fastTrack->mSampleRate = track->mSampleRate; 2579 fastTrack->mChannelMask = track->mChannelMask; 2580 fastTrack->mGeneration++; 2581 state->mTrackMask |= 1 << j; 2582 didModify = true; 2583 // no acknowledgement required for newly active tracks 2584 } 2585 // cache the combined master volume and stream type volume for fast mixer; this 2586 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2587 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2588 ++fastTracks; 2589 } else { 2590 // was it previously active? 2591 if (state->mTrackMask & (1 << j)) { 2592 fastTrack->mBufferProvider = NULL; 2593 fastTrack->mGeneration++; 2594 state->mTrackMask &= ~(1 << j); 2595 didModify = true; 2596 // If any fast tracks were removed, we must wait for acknowledgement 2597 // because we're about to decrement the last sp<> on those tracks. 2598 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2599 } else { 2600 LOG_FATAL("fast track %d should have been active", j); 2601 } 2602 tracksToRemove->add(track); 2603 // Avoids a misleading display in dumpsys 2604 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2605 } 2606 continue; 2607 } 2608 2609 { // local variable scope to avoid goto warning 2610 2611 audio_track_cblk_t* cblk = track->cblk(); 2612 2613 // The first time a track is added we wait 2614 // for all its buffers to be filled before processing it 2615 int name = track->name(); 2616 // make sure that we have enough frames to mix one full buffer. 2617 // enforce this condition only once to enable draining the buffer in case the client 2618 // app does not call stop() and relies on underrun to stop: 2619 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2620 // during last round 2621 size_t desiredFrames; 2622 if (t->sampleRate() == mSampleRate) { 2623 desiredFrames = mNormalFrameCount; 2624 } else { 2625 // +1 for rounding and +1 for additional sample needed for interpolation 2626 desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2627 // add frames already consumed but not yet released by the resampler 2628 // because cblk->framesReady() will include these frames 2629 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2630 // the minimum track buffer size is normally twice the number of frames necessary 2631 // to fill one buffer and the resampler should not leave more than one buffer worth 2632 // of unreleased frames after each pass, but just in case... 2633 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2634 } 2635 uint32_t minFrames = 1; 2636 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2637 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2638 minFrames = desiredFrames; 2639 } 2640 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2641 size_t framesReady; 2642 if (track->sharedBuffer() == 0) { 2643 framesReady = track->framesReady(); 2644 } else if (track->isStopped()) { 2645 framesReady = 0; 2646 } else { 2647 framesReady = 1; 2648 } 2649 if ((framesReady >= minFrames) && track->isReady() && 2650 !track->isPaused() && !track->isTerminated()) 2651 { 2652 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2653 this); 2654 2655 mixedTracks++; 2656 2657 // track->mainBuffer() != mMixBuffer means there is an effect chain 2658 // connected to the track 2659 chain.clear(); 2660 if (track->mainBuffer() != mMixBuffer) { 2661 chain = getEffectChain_l(track->sessionId()); 2662 // Delegate volume control to effect in track effect chain if needed 2663 if (chain != 0) { 2664 tracksWithEffect++; 2665 } else { 2666 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2667 "session %d", 2668 name, track->sessionId()); 2669 } 2670 } 2671 2672 2673 int param = AudioMixer::VOLUME; 2674 if (track->mFillingUpStatus == Track::FS_FILLED) { 2675 // no ramp for the first volume setting 2676 track->mFillingUpStatus = Track::FS_ACTIVE; 2677 if (track->mState == TrackBase::RESUMING) { 2678 track->mState = TrackBase::ACTIVE; 2679 param = AudioMixer::RAMP_VOLUME; 2680 } 2681 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2682 } else if (cblk->server != 0) { 2683 // If the track is stopped before the first frame was mixed, 2684 // do not apply ramp 2685 param = AudioMixer::RAMP_VOLUME; 2686 } 2687 2688 // compute volume for this track 2689 uint32_t vl, vr, va; 2690 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2691 vl = vr = va = 0; 2692 if (track->isPausing()) { 2693 track->setPaused(); 2694 } 2695 } else { 2696 2697 // read original volumes with volume control 2698 float typeVolume = mStreamTypes[track->streamType()].volume; 2699 float v = masterVolume * typeVolume; 2700 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2701 uint32_t vlr = proxy->getVolumeLR(); 2702 vl = vlr & 0xFFFF; 2703 vr = vlr >> 16; 2704 // track volumes come from shared memory, so can't be trusted and must be clamped 2705 if (vl > MAX_GAIN_INT) { 2706 ALOGV("Track left volume out of range: %04X", vl); 2707 vl = MAX_GAIN_INT; 2708 } 2709 if (vr > MAX_GAIN_INT) { 2710 ALOGV("Track right volume out of range: %04X", vr); 2711 vr = MAX_GAIN_INT; 2712 } 2713 // now apply the master volume and stream type volume 2714 vl = (uint32_t)(v * vl) << 12; 2715 vr = (uint32_t)(v * vr) << 12; 2716 // assuming master volume and stream type volume each go up to 1.0, 2717 // vl and vr are now in 8.24 format 2718 2719 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2720 // send level comes from shared memory and so may be corrupt 2721 if (sendLevel > MAX_GAIN_INT) { 2722 ALOGV("Track send level out of range: %04X", sendLevel); 2723 sendLevel = MAX_GAIN_INT; 2724 } 2725 va = (uint32_t)(v * sendLevel); 2726 } 2727 // Delegate volume control to effect in track effect chain if needed 2728 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2729 // Do not ramp volume if volume is controlled by effect 2730 param = AudioMixer::VOLUME; 2731 track->mHasVolumeController = true; 2732 } else { 2733 // force no volume ramp when volume controller was just disabled or removed 2734 // from effect chain to avoid volume spike 2735 if (track->mHasVolumeController) { 2736 param = AudioMixer::VOLUME; 2737 } 2738 track->mHasVolumeController = false; 2739 } 2740 2741 // Convert volumes from 8.24 to 4.12 format 2742 // This additional clamping is needed in case chain->setVolume_l() overshot 2743 vl = (vl + (1 << 11)) >> 12; 2744 if (vl > MAX_GAIN_INT) { 2745 vl = MAX_GAIN_INT; 2746 } 2747 vr = (vr + (1 << 11)) >> 12; 2748 if (vr > MAX_GAIN_INT) { 2749 vr = MAX_GAIN_INT; 2750 } 2751 2752 if (va > MAX_GAIN_INT) { 2753 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2754 } 2755 2756 // XXX: these things DON'T need to be done each time 2757 mAudioMixer->setBufferProvider(name, track); 2758 mAudioMixer->enable(name); 2759 2760 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2761 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2762 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2763 mAudioMixer->setParameter( 2764 name, 2765 AudioMixer::TRACK, 2766 AudioMixer::FORMAT, (void *)track->format()); 2767 mAudioMixer->setParameter( 2768 name, 2769 AudioMixer::TRACK, 2770 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2771 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2772 uint32_t maxSampleRate = mSampleRate * 2; 2773 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 2774 if (reqSampleRate == 0) { 2775 reqSampleRate = mSampleRate; 2776 } else if (reqSampleRate > maxSampleRate) { 2777 reqSampleRate = maxSampleRate; 2778 } 2779 mAudioMixer->setParameter( 2780 name, 2781 AudioMixer::RESAMPLE, 2782 AudioMixer::SAMPLE_RATE, 2783 (void *)reqSampleRate); 2784 mAudioMixer->setParameter( 2785 name, 2786 AudioMixer::TRACK, 2787 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2788 mAudioMixer->setParameter( 2789 name, 2790 AudioMixer::TRACK, 2791 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2792 2793 // reset retry count 2794 track->mRetryCount = kMaxTrackRetries; 2795 2796 // If one track is ready, set the mixer ready if: 2797 // - the mixer was not ready during previous round OR 2798 // - no other track is not ready 2799 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2800 mixerStatus != MIXER_TRACKS_ENABLED) { 2801 mixerStatus = MIXER_TRACKS_READY; 2802 } 2803 } else { 2804 // only implemented for normal tracks, not fast tracks 2805 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 2806 // we missed desiredFrames whatever the actual number of frames missing was 2807 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 2808 // FIXME also wake futex so that underrun is noticed more quickly 2809 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 2810 } 2811 // clear effect chain input buffer if an active track underruns to avoid sending 2812 // previous audio buffer again to effects 2813 chain = getEffectChain_l(track->sessionId()); 2814 if (chain != 0) { 2815 chain->clearInputBuffer(); 2816 } 2817 2818 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2819 cblk->server, this); 2820 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2821 track->isStopped() || track->isPaused()) { 2822 // We have consumed all the buffers of this track. 2823 // Remove it from the list of active tracks. 2824 // TODO: use actual buffer filling status instead of latency when available from 2825 // audio HAL 2826 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2827 size_t framesWritten = mBytesWritten / mFrameSize; 2828 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2829 if (track->isStopped()) { 2830 track->reset(); 2831 } 2832 tracksToRemove->add(track); 2833 } 2834 } else { 2835 track->mUnderrunCount++; 2836 // No buffers for this track. Give it a few chances to 2837 // fill a buffer, then remove it from active list. 2838 if (--(track->mRetryCount) <= 0) { 2839 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2840 tracksToRemove->add(track); 2841 // indicate to client process that the track was disabled because of underrun; 2842 // it will then automatically call start() when data is available 2843 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2844 // If one track is not ready, mark the mixer also not ready if: 2845 // - the mixer was ready during previous round OR 2846 // - no other track is ready 2847 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2848 mixerStatus != MIXER_TRACKS_READY) { 2849 mixerStatus = MIXER_TRACKS_ENABLED; 2850 } 2851 } 2852 mAudioMixer->disable(name); 2853 } 2854 2855 } // local variable scope to avoid goto warning 2856track_is_ready: ; 2857 2858 } 2859 2860 // Push the new FastMixer state if necessary 2861 bool pauseAudioWatchdog = false; 2862 if (didModify) { 2863 state->mFastTracksGen++; 2864 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2865 if (kUseFastMixer == FastMixer_Dynamic && 2866 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2867 state->mCommand = FastMixerState::COLD_IDLE; 2868 state->mColdFutexAddr = &mFastMixerFutex; 2869 state->mColdGen++; 2870 mFastMixerFutex = 0; 2871 if (kUseFastMixer == FastMixer_Dynamic) { 2872 mNormalSink = mOutputSink; 2873 } 2874 // If we go into cold idle, need to wait for acknowledgement 2875 // so that fast mixer stops doing I/O. 2876 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2877 pauseAudioWatchdog = true; 2878 } 2879 } 2880 if (sq != NULL) { 2881 sq->end(didModify); 2882 sq->push(block); 2883 } 2884#ifdef AUDIO_WATCHDOG 2885 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2886 mAudioWatchdog->pause(); 2887 } 2888#endif 2889 2890 // Now perform the deferred reset on fast tracks that have stopped 2891 while (resetMask != 0) { 2892 size_t i = __builtin_ctz(resetMask); 2893 ALOG_ASSERT(i < count); 2894 resetMask &= ~(1 << i); 2895 sp<Track> t = mActiveTracks[i].promote(); 2896 if (t == 0) { 2897 continue; 2898 } 2899 Track* track = t.get(); 2900 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2901 track->reset(); 2902 } 2903 2904 // remove all the tracks that need to be... 2905 count = tracksToRemove->size(); 2906 if (CC_UNLIKELY(count)) { 2907 for (size_t i=0 ; i<count ; i++) { 2908 const sp<Track>& track = tracksToRemove->itemAt(i); 2909 mActiveTracks.remove(track); 2910 if (track->mainBuffer() != mMixBuffer) { 2911 chain = getEffectChain_l(track->sessionId()); 2912 if (chain != 0) { 2913 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2914 track->sessionId()); 2915 chain->decActiveTrackCnt(); 2916 } 2917 } 2918 if (track->isTerminated()) { 2919 removeTrack_l(track); 2920 } 2921 } 2922 } 2923 2924 // mix buffer must be cleared if all tracks are connected to an 2925 // effect chain as in this case the mixer will not write to 2926 // mix buffer and track effects will accumulate into it 2927 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2928 (mixedTracks == 0 && fastTracks > 0)) { 2929 // FIXME as a performance optimization, should remember previous zero status 2930 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2931 } 2932 2933 // if any fast tracks, then status is ready 2934 mMixerStatusIgnoringFastTracks = mixerStatus; 2935 if (fastTracks > 0) { 2936 mixerStatus = MIXER_TRACKS_READY; 2937 } 2938 return mixerStatus; 2939} 2940 2941// getTrackName_l() must be called with ThreadBase::mLock held 2942int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2943{ 2944 return mAudioMixer->getTrackName(channelMask, sessionId); 2945} 2946 2947// deleteTrackName_l() must be called with ThreadBase::mLock held 2948void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2949{ 2950 ALOGV("remove track (%d) and delete from mixer", name); 2951 mAudioMixer->deleteTrackName(name); 2952} 2953 2954// checkForNewParameters_l() must be called with ThreadBase::mLock held 2955bool AudioFlinger::MixerThread::checkForNewParameters_l() 2956{ 2957 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2958 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2959 bool reconfig = false; 2960 2961 while (!mNewParameters.isEmpty()) { 2962 2963 if (mFastMixer != NULL) { 2964 FastMixerStateQueue *sq = mFastMixer->sq(); 2965 FastMixerState *state = sq->begin(); 2966 if (!(state->mCommand & FastMixerState::IDLE)) { 2967 previousCommand = state->mCommand; 2968 state->mCommand = FastMixerState::HOT_IDLE; 2969 sq->end(); 2970 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2971 } else { 2972 sq->end(false /*didModify*/); 2973 } 2974 } 2975 2976 status_t status = NO_ERROR; 2977 String8 keyValuePair = mNewParameters[0]; 2978 AudioParameter param = AudioParameter(keyValuePair); 2979 int value; 2980 2981 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2982 reconfig = true; 2983 } 2984 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2985 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2986 status = BAD_VALUE; 2987 } else { 2988 reconfig = true; 2989 } 2990 } 2991 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2992 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2993 status = BAD_VALUE; 2994 } else { 2995 reconfig = true; 2996 } 2997 } 2998 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2999 // do not accept frame count changes if tracks are open as the track buffer 3000 // size depends on frame count and correct behavior would not be guaranteed 3001 // if frame count is changed after track creation 3002 if (!mTracks.isEmpty()) { 3003 status = INVALID_OPERATION; 3004 } else { 3005 reconfig = true; 3006 } 3007 } 3008 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3009#ifdef ADD_BATTERY_DATA 3010 // when changing the audio output device, call addBatteryData to notify 3011 // the change 3012 if (mOutDevice != value) { 3013 uint32_t params = 0; 3014 // check whether speaker is on 3015 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3016 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3017 } 3018 3019 audio_devices_t deviceWithoutSpeaker 3020 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3021 // check if any other device (except speaker) is on 3022 if (value & deviceWithoutSpeaker ) { 3023 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3024 } 3025 3026 if (params != 0) { 3027 addBatteryData(params); 3028 } 3029 } 3030#endif 3031 3032 // forward device change to effects that have requested to be 3033 // aware of attached audio device. 3034 if (value != AUDIO_DEVICE_NONE) { 3035 mOutDevice = value; 3036 for (size_t i = 0; i < mEffectChains.size(); i++) { 3037 mEffectChains[i]->setDevice_l(mOutDevice); 3038 } 3039 } 3040 } 3041 3042 if (status == NO_ERROR) { 3043 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3044 keyValuePair.string()); 3045 if (!mStandby && status == INVALID_OPERATION) { 3046 mOutput->stream->common.standby(&mOutput->stream->common); 3047 mStandby = true; 3048 mBytesWritten = 0; 3049 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3050 keyValuePair.string()); 3051 } 3052 if (status == NO_ERROR && reconfig) { 3053 delete mAudioMixer; 3054 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3055 mAudioMixer = NULL; 3056 readOutputParameters(); 3057 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3058 for (size_t i = 0; i < mTracks.size() ; i++) { 3059 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3060 if (name < 0) { 3061 break; 3062 } 3063 mTracks[i]->mName = name; 3064 } 3065 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3066 } 3067 } 3068 3069 mNewParameters.removeAt(0); 3070 3071 mParamStatus = status; 3072 mParamCond.signal(); 3073 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3074 // already timed out waiting for the status and will never signal the condition. 3075 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3076 } 3077 3078 if (!(previousCommand & FastMixerState::IDLE)) { 3079 ALOG_ASSERT(mFastMixer != NULL); 3080 FastMixerStateQueue *sq = mFastMixer->sq(); 3081 FastMixerState *state = sq->begin(); 3082 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3083 state->mCommand = previousCommand; 3084 sq->end(); 3085 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3086 } 3087 3088 return reconfig; 3089} 3090 3091 3092void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3093{ 3094 const size_t SIZE = 256; 3095 char buffer[SIZE]; 3096 String8 result; 3097 3098 PlaybackThread::dumpInternals(fd, args); 3099 3100 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3101 result.append(buffer); 3102 write(fd, result.string(), result.size()); 3103 3104 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3105 const FastMixerDumpState copy(mFastMixerDumpState); 3106 copy.dump(fd); 3107 3108#ifdef STATE_QUEUE_DUMP 3109 // Similar for state queue 3110 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3111 observerCopy.dump(fd); 3112 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3113 mutatorCopy.dump(fd); 3114#endif 3115 3116#ifdef TEE_SINK 3117 // Write the tee output to a .wav file 3118 dumpTee(fd, mTeeSource, mId); 3119#endif 3120 3121#ifdef AUDIO_WATCHDOG 3122 if (mAudioWatchdog != 0) { 3123 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3124 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3125 wdCopy.dump(fd); 3126 } 3127#endif 3128} 3129 3130uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3131{ 3132 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3133} 3134 3135uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3136{ 3137 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3138} 3139 3140void AudioFlinger::MixerThread::cacheParameters_l() 3141{ 3142 PlaybackThread::cacheParameters_l(); 3143 3144 // FIXME: Relaxed timing because of a certain device that can't meet latency 3145 // Should be reduced to 2x after the vendor fixes the driver issue 3146 // increase threshold again due to low power audio mode. The way this warning 3147 // threshold is calculated and its usefulness should be reconsidered anyway. 3148 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3149} 3150 3151// ---------------------------------------------------------------------------- 3152 3153AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3154 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3155 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3156 // mLeftVolFloat, mRightVolFloat 3157{ 3158} 3159 3160AudioFlinger::DirectOutputThread::~DirectOutputThread() 3161{ 3162} 3163 3164AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3165 Vector< sp<Track> > *tracksToRemove 3166) 3167{ 3168 size_t count = mActiveTracks.size(); 3169 mixer_state mixerStatus = MIXER_IDLE; 3170 3171 // find out which tracks need to be processed 3172 for (size_t i = 0; i < count; i++) { 3173 sp<Track> t = mActiveTracks[i].promote(); 3174 // The track died recently 3175 if (t == 0) { 3176 continue; 3177 } 3178 3179 Track* const track = t.get(); 3180 audio_track_cblk_t* cblk = track->cblk(); 3181 3182 // The first time a track is added we wait 3183 // for all its buffers to be filled before processing it 3184 uint32_t minFrames; 3185 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3186 minFrames = mNormalFrameCount; 3187 } else { 3188 minFrames = 1; 3189 } 3190 if ((track->framesReady() >= minFrames) && track->isReady() && 3191 !track->isPaused() && !track->isTerminated()) 3192 { 3193 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3194 3195 if (track->mFillingUpStatus == Track::FS_FILLED) { 3196 track->mFillingUpStatus = Track::FS_ACTIVE; 3197 mLeftVolFloat = mRightVolFloat = 0; 3198 if (track->mState == TrackBase::RESUMING) { 3199 track->mState = TrackBase::ACTIVE; 3200 } 3201 } 3202 3203 // compute volume for this track 3204 float left, right; 3205 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3206 left = right = 0; 3207 if (track->isPausing()) { 3208 track->setPaused(); 3209 } 3210 } else { 3211 float typeVolume = mStreamTypes[track->streamType()].volume; 3212 float v = mMasterVolume * typeVolume; 3213 uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); 3214 float v_clamped = v * (vlr & 0xFFFF); 3215 if (v_clamped > MAX_GAIN) { 3216 v_clamped = MAX_GAIN; 3217 } 3218 left = v_clamped/MAX_GAIN; 3219 v_clamped = v * (vlr >> 16); 3220 if (v_clamped > MAX_GAIN) { 3221 v_clamped = MAX_GAIN; 3222 } 3223 right = v_clamped/MAX_GAIN; 3224 } 3225 // Only consider last track started for volume and mixer state control. 3226 // This is the last entry in mActiveTracks unless a track underruns. 3227 // As we only care about the transition phase between two tracks on a 3228 // direct output, it is not a problem to ignore the underrun case. 3229 if (i == (count - 1)) { 3230 if (left != mLeftVolFloat || right != mRightVolFloat) { 3231 mLeftVolFloat = left; 3232 mRightVolFloat = right; 3233 3234 // Convert volumes from float to 8.24 3235 uint32_t vl = (uint32_t)(left * (1 << 24)); 3236 uint32_t vr = (uint32_t)(right * (1 << 24)); 3237 3238 // Delegate volume control to effect in track effect chain if needed 3239 // only one effect chain can be present on DirectOutputThread, so if 3240 // there is one, the track is connected to it 3241 if (!mEffectChains.isEmpty()) { 3242 // Do not ramp volume if volume is controlled by effect 3243 mEffectChains[0]->setVolume_l(&vl, &vr); 3244 left = (float)vl / (1 << 24); 3245 right = (float)vr / (1 << 24); 3246 } 3247 mOutput->stream->set_volume(mOutput->stream, left, right); 3248 } 3249 3250 // reset retry count 3251 track->mRetryCount = kMaxTrackRetriesDirect; 3252 mActiveTrack = t; 3253 mixerStatus = MIXER_TRACKS_READY; 3254 } 3255 } else { 3256 // clear effect chain input buffer if the last active track started underruns 3257 // to avoid sending previous audio buffer again to effects 3258 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3259 mEffectChains[0]->clearInputBuffer(); 3260 } 3261 3262 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3263 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3264 track->isStopped() || track->isPaused()) { 3265 // We have consumed all the buffers of this track. 3266 // Remove it from the list of active tracks. 3267 // TODO: implement behavior for compressed audio 3268 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3269 size_t framesWritten = mBytesWritten / mFrameSize; 3270 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3271 if (track->isStopped()) { 3272 track->reset(); 3273 } 3274 tracksToRemove->add(track); 3275 } 3276 } else { 3277 // No buffers for this track. Give it a few chances to 3278 // fill a buffer, then remove it from active list. 3279 // Only consider last track started for mixer state control 3280 if (--(track->mRetryCount) <= 0) { 3281 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3282 tracksToRemove->add(track); 3283 } else if (i == (count -1)){ 3284 mixerStatus = MIXER_TRACKS_ENABLED; 3285 } 3286 } 3287 } 3288 } 3289 3290 // remove all the tracks that need to be... 3291 count = tracksToRemove->size(); 3292 if (CC_UNLIKELY(count)) { 3293 for (size_t i = 0 ; i < count ; i++) { 3294 const sp<Track>& track = tracksToRemove->itemAt(i); 3295 mActiveTracks.remove(track); 3296 if (!mEffectChains.isEmpty()) { 3297 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3298 track->sessionId()); 3299 mEffectChains[0]->decActiveTrackCnt(); 3300 } 3301 if (track->isTerminated()) { 3302 removeTrack_l(track); 3303 } 3304 } 3305 } 3306 3307 return mixerStatus; 3308} 3309 3310void AudioFlinger::DirectOutputThread::threadLoop_mix() 3311{ 3312 AudioBufferProvider::Buffer buffer; 3313 size_t frameCount = mFrameCount; 3314 int8_t *curBuf = (int8_t *)mMixBuffer; 3315 // output audio to hardware 3316 while (frameCount) { 3317 buffer.frameCount = frameCount; 3318 mActiveTrack->getNextBuffer(&buffer); 3319 if (CC_UNLIKELY(buffer.raw == NULL)) { 3320 memset(curBuf, 0, frameCount * mFrameSize); 3321 break; 3322 } 3323 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3324 frameCount -= buffer.frameCount; 3325 curBuf += buffer.frameCount * mFrameSize; 3326 mActiveTrack->releaseBuffer(&buffer); 3327 } 3328 sleepTime = 0; 3329 standbyTime = systemTime() + standbyDelay; 3330 mActiveTrack.clear(); 3331 3332} 3333 3334void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3335{ 3336 if (sleepTime == 0) { 3337 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3338 sleepTime = activeSleepTime; 3339 } else { 3340 sleepTime = idleSleepTime; 3341 } 3342 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3343 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3344 sleepTime = 0; 3345 } 3346} 3347 3348// getTrackName_l() must be called with ThreadBase::mLock held 3349int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3350 int sessionId) 3351{ 3352 return 0; 3353} 3354 3355// deleteTrackName_l() must be called with ThreadBase::mLock held 3356void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3357{ 3358} 3359 3360// checkForNewParameters_l() must be called with ThreadBase::mLock held 3361bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3362{ 3363 bool reconfig = false; 3364 3365 while (!mNewParameters.isEmpty()) { 3366 status_t status = NO_ERROR; 3367 String8 keyValuePair = mNewParameters[0]; 3368 AudioParameter param = AudioParameter(keyValuePair); 3369 int value; 3370 3371 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3372 // do not accept frame count changes if tracks are open as the track buffer 3373 // size depends on frame count and correct behavior would not be garantied 3374 // if frame count is changed after track creation 3375 if (!mTracks.isEmpty()) { 3376 status = INVALID_OPERATION; 3377 } else { 3378 reconfig = true; 3379 } 3380 } 3381 if (status == NO_ERROR) { 3382 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3383 keyValuePair.string()); 3384 if (!mStandby && status == INVALID_OPERATION) { 3385 mOutput->stream->common.standby(&mOutput->stream->common); 3386 mStandby = true; 3387 mBytesWritten = 0; 3388 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3389 keyValuePair.string()); 3390 } 3391 if (status == NO_ERROR && reconfig) { 3392 readOutputParameters(); 3393 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3394 } 3395 } 3396 3397 mNewParameters.removeAt(0); 3398 3399 mParamStatus = status; 3400 mParamCond.signal(); 3401 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3402 // already timed out waiting for the status and will never signal the condition. 3403 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3404 } 3405 return reconfig; 3406} 3407 3408uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3409{ 3410 uint32_t time; 3411 if (audio_is_linear_pcm(mFormat)) { 3412 time = PlaybackThread::activeSleepTimeUs(); 3413 } else { 3414 time = 10000; 3415 } 3416 return time; 3417} 3418 3419uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3420{ 3421 uint32_t time; 3422 if (audio_is_linear_pcm(mFormat)) { 3423 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3424 } else { 3425 time = 10000; 3426 } 3427 return time; 3428} 3429 3430uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3431{ 3432 uint32_t time; 3433 if (audio_is_linear_pcm(mFormat)) { 3434 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3435 } else { 3436 time = 10000; 3437 } 3438 return time; 3439} 3440 3441void AudioFlinger::DirectOutputThread::cacheParameters_l() 3442{ 3443 PlaybackThread::cacheParameters_l(); 3444 3445 // use shorter standby delay as on normal output to release 3446 // hardware resources as soon as possible 3447 standbyDelay = microseconds(activeSleepTime*2); 3448} 3449 3450// ---------------------------------------------------------------------------- 3451 3452AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3453 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3454 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3455 DUPLICATING), 3456 mWaitTimeMs(UINT_MAX) 3457{ 3458 addOutputTrack(mainThread); 3459} 3460 3461AudioFlinger::DuplicatingThread::~DuplicatingThread() 3462{ 3463 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3464 mOutputTracks[i]->destroy(); 3465 } 3466} 3467 3468void AudioFlinger::DuplicatingThread::threadLoop_mix() 3469{ 3470 // mix buffers... 3471 if (outputsReady(outputTracks)) { 3472 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3473 } else { 3474 memset(mMixBuffer, 0, mixBufferSize); 3475 } 3476 sleepTime = 0; 3477 writeFrames = mNormalFrameCount; 3478 standbyTime = systemTime() + standbyDelay; 3479} 3480 3481void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3482{ 3483 if (sleepTime == 0) { 3484 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3485 sleepTime = activeSleepTime; 3486 } else { 3487 sleepTime = idleSleepTime; 3488 } 3489 } else if (mBytesWritten != 0) { 3490 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3491 writeFrames = mNormalFrameCount; 3492 memset(mMixBuffer, 0, mixBufferSize); 3493 } else { 3494 // flush remaining overflow buffers in output tracks 3495 writeFrames = 0; 3496 } 3497 sleepTime = 0; 3498 } 3499} 3500 3501void AudioFlinger::DuplicatingThread::threadLoop_write() 3502{ 3503 for (size_t i = 0; i < outputTracks.size(); i++) { 3504 outputTracks[i]->write(mMixBuffer, writeFrames); 3505 } 3506 mBytesWritten += mixBufferSize; 3507} 3508 3509void AudioFlinger::DuplicatingThread::threadLoop_standby() 3510{ 3511 // DuplicatingThread implements standby by stopping all tracks 3512 for (size_t i = 0; i < outputTracks.size(); i++) { 3513 outputTracks[i]->stop(); 3514 } 3515} 3516 3517void AudioFlinger::DuplicatingThread::saveOutputTracks() 3518{ 3519 outputTracks = mOutputTracks; 3520} 3521 3522void AudioFlinger::DuplicatingThread::clearOutputTracks() 3523{ 3524 outputTracks.clear(); 3525} 3526 3527void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3528{ 3529 Mutex::Autolock _l(mLock); 3530 // FIXME explain this formula 3531 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3532 OutputTrack *outputTrack = new OutputTrack(thread, 3533 this, 3534 mSampleRate, 3535 mFormat, 3536 mChannelMask, 3537 frameCount); 3538 if (outputTrack->cblk() != NULL) { 3539 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3540 mOutputTracks.add(outputTrack); 3541 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3542 updateWaitTime_l(); 3543 } 3544} 3545 3546void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3547{ 3548 Mutex::Autolock _l(mLock); 3549 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3550 if (mOutputTracks[i]->thread() == thread) { 3551 mOutputTracks[i]->destroy(); 3552 mOutputTracks.removeAt(i); 3553 updateWaitTime_l(); 3554 return; 3555 } 3556 } 3557 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3558} 3559 3560// caller must hold mLock 3561void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3562{ 3563 mWaitTimeMs = UINT_MAX; 3564 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3565 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3566 if (strong != 0) { 3567 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3568 if (waitTimeMs < mWaitTimeMs) { 3569 mWaitTimeMs = waitTimeMs; 3570 } 3571 } 3572 } 3573} 3574 3575 3576bool AudioFlinger::DuplicatingThread::outputsReady( 3577 const SortedVector< sp<OutputTrack> > &outputTracks) 3578{ 3579 for (size_t i = 0; i < outputTracks.size(); i++) { 3580 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3581 if (thread == 0) { 3582 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3583 outputTracks[i].get()); 3584 return false; 3585 } 3586 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3587 // see note at standby() declaration 3588 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3589 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3590 thread.get()); 3591 return false; 3592 } 3593 } 3594 return true; 3595} 3596 3597uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3598{ 3599 return (mWaitTimeMs * 1000) / 2; 3600} 3601 3602void AudioFlinger::DuplicatingThread::cacheParameters_l() 3603{ 3604 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3605 updateWaitTime_l(); 3606 3607 MixerThread::cacheParameters_l(); 3608} 3609 3610// ---------------------------------------------------------------------------- 3611// Record 3612// ---------------------------------------------------------------------------- 3613 3614AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3615 AudioStreamIn *input, 3616 uint32_t sampleRate, 3617 audio_channel_mask_t channelMask, 3618 audio_io_handle_t id, 3619 audio_devices_t outDevice, 3620 audio_devices_t inDevice 3621#ifdef TEE_SINK 3622 , const sp<NBAIO_Sink>& teeSink 3623#endif 3624 ) : 3625 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3626 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3627 // mRsmpInIndex and mInputBytes set by readInputParameters() 3628 mReqChannelCount(popcount(channelMask)), 3629 mReqSampleRate(sampleRate) 3630 // mBytesRead is only meaningful while active, and so is cleared in start() 3631 // (but might be better to also clear here for dump?) 3632#ifdef TEE_SINK 3633 , mTeeSink(teeSink) 3634#endif 3635{ 3636 snprintf(mName, kNameLength, "AudioIn_%X", id); 3637 3638 readInputParameters(); 3639 3640} 3641 3642 3643AudioFlinger::RecordThread::~RecordThread() 3644{ 3645 delete[] mRsmpInBuffer; 3646 delete mResampler; 3647 delete[] mRsmpOutBuffer; 3648} 3649 3650void AudioFlinger::RecordThread::onFirstRef() 3651{ 3652 run(mName, PRIORITY_URGENT_AUDIO); 3653} 3654 3655status_t AudioFlinger::RecordThread::readyToRun() 3656{ 3657 status_t status = initCheck(); 3658 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3659 return status; 3660} 3661 3662bool AudioFlinger::RecordThread::threadLoop() 3663{ 3664 AudioBufferProvider::Buffer buffer; 3665 sp<RecordTrack> activeTrack; 3666 Vector< sp<EffectChain> > effectChains; 3667 3668 nsecs_t lastWarning = 0; 3669 3670 inputStandBy(); 3671 acquireWakeLock(); 3672 3673 // used to verify we've read at least once before evaluating how many bytes were read 3674 bool readOnce = false; 3675 3676 // start recording 3677 while (!exitPending()) { 3678 3679 processConfigEvents(); 3680 3681 { // scope for mLock 3682 Mutex::Autolock _l(mLock); 3683 checkForNewParameters_l(); 3684 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3685 standby(); 3686 3687 if (exitPending()) { 3688 break; 3689 } 3690 3691 releaseWakeLock_l(); 3692 ALOGV("RecordThread: loop stopping"); 3693 // go to sleep 3694 mWaitWorkCV.wait(mLock); 3695 ALOGV("RecordThread: loop starting"); 3696 acquireWakeLock_l(); 3697 continue; 3698 } 3699 if (mActiveTrack != 0) { 3700 if (mActiveTrack->mState == TrackBase::PAUSING) { 3701 standby(); 3702 mActiveTrack.clear(); 3703 mStartStopCond.broadcast(); 3704 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3705 if (mReqChannelCount != mActiveTrack->channelCount()) { 3706 mActiveTrack.clear(); 3707 mStartStopCond.broadcast(); 3708 } else if (readOnce) { 3709 // record start succeeds only if first read from audio input 3710 // succeeds 3711 if (mBytesRead >= 0) { 3712 mActiveTrack->mState = TrackBase::ACTIVE; 3713 } else { 3714 mActiveTrack.clear(); 3715 } 3716 mStartStopCond.broadcast(); 3717 } 3718 mStandby = false; 3719 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3720 removeTrack_l(mActiveTrack); 3721 mActiveTrack.clear(); 3722 } 3723 } 3724 lockEffectChains_l(effectChains); 3725 } 3726 3727 if (mActiveTrack != 0) { 3728 if (mActiveTrack->mState != TrackBase::ACTIVE && 3729 mActiveTrack->mState != TrackBase::RESUMING) { 3730 unlockEffectChains(effectChains); 3731 usleep(kRecordThreadSleepUs); 3732 continue; 3733 } 3734 for (size_t i = 0; i < effectChains.size(); i ++) { 3735 effectChains[i]->process_l(); 3736 } 3737 3738 buffer.frameCount = mFrameCount; 3739 status_t status = mActiveTrack->getNextBuffer(&buffer); 3740 if (CC_LIKELY(status == NO_ERROR)) { 3741 readOnce = true; 3742 size_t framesOut = buffer.frameCount; 3743 if (mResampler == NULL) { 3744 // no resampling 3745 while (framesOut) { 3746 size_t framesIn = mFrameCount - mRsmpInIndex; 3747 if (framesIn) { 3748 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3749 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3750 mActiveTrack->mFrameSize; 3751 if (framesIn > framesOut) 3752 framesIn = framesOut; 3753 mRsmpInIndex += framesIn; 3754 framesOut -= framesIn; 3755 if (mChannelCount == mReqChannelCount || 3756 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3757 memcpy(dst, src, framesIn * mFrameSize); 3758 } else { 3759 if (mChannelCount == 1) { 3760 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3761 (int16_t *)src, framesIn); 3762 } else { 3763 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3764 (int16_t *)src, framesIn); 3765 } 3766 } 3767 } 3768 if (framesOut && mFrameCount == mRsmpInIndex) { 3769 void *readInto; 3770 if (framesOut == mFrameCount && 3771 (mChannelCount == mReqChannelCount || 3772 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3773 readInto = buffer.raw; 3774 framesOut = 0; 3775 } else { 3776 readInto = mRsmpInBuffer; 3777 mRsmpInIndex = 0; 3778 } 3779 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3780 mInputBytes); 3781 if (mBytesRead <= 0) { 3782 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3783 { 3784 ALOGE("Error reading audio input"); 3785 // Force input into standby so that it tries to 3786 // recover at next read attempt 3787 inputStandBy(); 3788 usleep(kRecordThreadSleepUs); 3789 } 3790 mRsmpInIndex = mFrameCount; 3791 framesOut = 0; 3792 buffer.frameCount = 0; 3793 } 3794#ifdef TEE_SINK 3795 else if (mTeeSink != 0) { 3796 (void) mTeeSink->write(readInto, 3797 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3798 } 3799#endif 3800 } 3801 } 3802 } else { 3803 // resampling 3804 3805 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3806 // alter output frame count as if we were expecting stereo samples 3807 if (mChannelCount == 1 && mReqChannelCount == 1) { 3808 framesOut >>= 1; 3809 } 3810 mResampler->resample(mRsmpOutBuffer, framesOut, 3811 this /* AudioBufferProvider* */); 3812 // ditherAndClamp() works as long as all buffers returned by 3813 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3814 if (mChannelCount == 2 && mReqChannelCount == 1) { 3815 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3816 // the resampler always outputs stereo samples: 3817 // do post stereo to mono conversion 3818 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3819 framesOut); 3820 } else { 3821 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3822 } 3823 3824 } 3825 if (mFramestoDrop == 0) { 3826 mActiveTrack->releaseBuffer(&buffer); 3827 } else { 3828 if (mFramestoDrop > 0) { 3829 mFramestoDrop -= buffer.frameCount; 3830 if (mFramestoDrop <= 0) { 3831 clearSyncStartEvent(); 3832 } 3833 } else { 3834 mFramestoDrop += buffer.frameCount; 3835 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3836 mSyncStartEvent->isCancelled()) { 3837 ALOGW("Synced record %s, session %d, trigger session %d", 3838 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3839 mActiveTrack->sessionId(), 3840 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3841 clearSyncStartEvent(); 3842 } 3843 } 3844 } 3845 mActiveTrack->clearOverflow(); 3846 } 3847 // client isn't retrieving buffers fast enough 3848 else { 3849 if (!mActiveTrack->setOverflow()) { 3850 nsecs_t now = systemTime(); 3851 if ((now - lastWarning) > kWarningThrottleNs) { 3852 ALOGW("RecordThread: buffer overflow"); 3853 lastWarning = now; 3854 } 3855 } 3856 // Release the processor for a while before asking for a new buffer. 3857 // This will give the application more chance to read from the buffer and 3858 // clear the overflow. 3859 usleep(kRecordThreadSleepUs); 3860 } 3861 } 3862 // enable changes in effect chain 3863 unlockEffectChains(effectChains); 3864 effectChains.clear(); 3865 } 3866 3867 standby(); 3868 3869 { 3870 Mutex::Autolock _l(mLock); 3871 mActiveTrack.clear(); 3872 mStartStopCond.broadcast(); 3873 } 3874 3875 releaseWakeLock(); 3876 3877 ALOGV("RecordThread %p exiting", this); 3878 return false; 3879} 3880 3881void AudioFlinger::RecordThread::standby() 3882{ 3883 if (!mStandby) { 3884 inputStandBy(); 3885 mStandby = true; 3886 } 3887} 3888 3889void AudioFlinger::RecordThread::inputStandBy() 3890{ 3891 mInput->stream->common.standby(&mInput->stream->common); 3892} 3893 3894sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3895 const sp<AudioFlinger::Client>& client, 3896 uint32_t sampleRate, 3897 audio_format_t format, 3898 audio_channel_mask_t channelMask, 3899 size_t frameCount, 3900 int sessionId, 3901 IAudioFlinger::track_flags_t flags, 3902 pid_t tid, 3903 status_t *status) 3904{ 3905 sp<RecordTrack> track; 3906 status_t lStatus; 3907 3908 lStatus = initCheck(); 3909 if (lStatus != NO_ERROR) { 3910 ALOGE("Audio driver not initialized."); 3911 goto Exit; 3912 } 3913 3914 // FIXME use flags and tid similar to createTrack_l() 3915 3916 { // scope for mLock 3917 Mutex::Autolock _l(mLock); 3918 3919 track = new RecordTrack(this, client, sampleRate, 3920 format, channelMask, frameCount, sessionId); 3921 3922 if (track->getCblk() == 0) { 3923 lStatus = NO_MEMORY; 3924 goto Exit; 3925 } 3926 mTracks.add(track); 3927 3928 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3929 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3930 mAudioFlinger->btNrecIsOff(); 3931 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3932 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3933 } 3934 lStatus = NO_ERROR; 3935 3936Exit: 3937 if (status) { 3938 *status = lStatus; 3939 } 3940 return track; 3941} 3942 3943status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3944 AudioSystem::sync_event_t event, 3945 int triggerSession) 3946{ 3947 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3948 sp<ThreadBase> strongMe = this; 3949 status_t status = NO_ERROR; 3950 3951 if (event == AudioSystem::SYNC_EVENT_NONE) { 3952 clearSyncStartEvent(); 3953 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3954 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3955 triggerSession, 3956 recordTrack->sessionId(), 3957 syncStartEventCallback, 3958 this); 3959 // Sync event can be cancelled by the trigger session if the track is not in a 3960 // compatible state in which case we start record immediately 3961 if (mSyncStartEvent->isCancelled()) { 3962 clearSyncStartEvent(); 3963 } else { 3964 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3965 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3966 } 3967 } 3968 3969 { 3970 AutoMutex lock(mLock); 3971 if (mActiveTrack != 0) { 3972 if (recordTrack != mActiveTrack.get()) { 3973 status = -EBUSY; 3974 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3975 mActiveTrack->mState = TrackBase::ACTIVE; 3976 } 3977 return status; 3978 } 3979 3980 recordTrack->mState = TrackBase::IDLE; 3981 mActiveTrack = recordTrack; 3982 mLock.unlock(); 3983 status_t status = AudioSystem::startInput(mId); 3984 mLock.lock(); 3985 if (status != NO_ERROR) { 3986 mActiveTrack.clear(); 3987 clearSyncStartEvent(); 3988 return status; 3989 } 3990 mRsmpInIndex = mFrameCount; 3991 mBytesRead = 0; 3992 if (mResampler != NULL) { 3993 mResampler->reset(); 3994 } 3995 mActiveTrack->mState = TrackBase::RESUMING; 3996 // signal thread to start 3997 ALOGV("Signal record thread"); 3998 mWaitWorkCV.broadcast(); 3999 // do not wait for mStartStopCond if exiting 4000 if (exitPending()) { 4001 mActiveTrack.clear(); 4002 status = INVALID_OPERATION; 4003 goto startError; 4004 } 4005 mStartStopCond.wait(mLock); 4006 if (mActiveTrack == 0) { 4007 ALOGV("Record failed to start"); 4008 status = BAD_VALUE; 4009 goto startError; 4010 } 4011 ALOGV("Record started OK"); 4012 return status; 4013 } 4014 4015startError: 4016 AudioSystem::stopInput(mId); 4017 clearSyncStartEvent(); 4018 return status; 4019} 4020 4021void AudioFlinger::RecordThread::clearSyncStartEvent() 4022{ 4023 if (mSyncStartEvent != 0) { 4024 mSyncStartEvent->cancel(); 4025 } 4026 mSyncStartEvent.clear(); 4027 mFramestoDrop = 0; 4028} 4029 4030void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4031{ 4032 sp<SyncEvent> strongEvent = event.promote(); 4033 4034 if (strongEvent != 0) { 4035 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4036 me->handleSyncStartEvent(strongEvent); 4037 } 4038} 4039 4040void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4041{ 4042 if (event == mSyncStartEvent) { 4043 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4044 // from audio HAL 4045 mFramestoDrop = mFrameCount * 2; 4046 } 4047} 4048 4049bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4050 ALOGV("RecordThread::stop"); 4051 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4052 return false; 4053 } 4054 recordTrack->mState = TrackBase::PAUSING; 4055 // do not wait for mStartStopCond if exiting 4056 if (exitPending()) { 4057 return true; 4058 } 4059 mStartStopCond.wait(mLock); 4060 // if we have been restarted, recordTrack == mActiveTrack.get() here 4061 if (exitPending() || recordTrack != mActiveTrack.get()) { 4062 ALOGV("Record stopped OK"); 4063 return true; 4064 } 4065 return false; 4066} 4067 4068bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4069{ 4070 return false; 4071} 4072 4073status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4074{ 4075#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4076 if (!isValidSyncEvent(event)) { 4077 return BAD_VALUE; 4078 } 4079 4080 int eventSession = event->triggerSession(); 4081 status_t ret = NAME_NOT_FOUND; 4082 4083 Mutex::Autolock _l(mLock); 4084 4085 for (size_t i = 0; i < mTracks.size(); i++) { 4086 sp<RecordTrack> track = mTracks[i]; 4087 if (eventSession == track->sessionId()) { 4088 (void) track->setSyncEvent(event); 4089 ret = NO_ERROR; 4090 } 4091 } 4092 return ret; 4093#else 4094 return BAD_VALUE; 4095#endif 4096} 4097 4098// destroyTrack_l() must be called with ThreadBase::mLock held 4099void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4100{ 4101 track->mState = TrackBase::TERMINATED; 4102 // active tracks are removed by threadLoop() 4103 if (mActiveTrack != track) { 4104 removeTrack_l(track); 4105 } 4106} 4107 4108void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4109{ 4110 mTracks.remove(track); 4111 // need anything related to effects here? 4112} 4113 4114void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4115{ 4116 dumpInternals(fd, args); 4117 dumpTracks(fd, args); 4118 dumpEffectChains(fd, args); 4119} 4120 4121void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4122{ 4123 const size_t SIZE = 256; 4124 char buffer[SIZE]; 4125 String8 result; 4126 4127 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4128 result.append(buffer); 4129 4130 if (mActiveTrack != 0) { 4131 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4132 result.append(buffer); 4133 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4134 result.append(buffer); 4135 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4136 result.append(buffer); 4137 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4138 result.append(buffer); 4139 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4140 result.append(buffer); 4141 } else { 4142 result.append("No active record client\n"); 4143 } 4144 4145 write(fd, result.string(), result.size()); 4146 4147 dumpBase(fd, args); 4148} 4149 4150void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4151{ 4152 const size_t SIZE = 256; 4153 char buffer[SIZE]; 4154 String8 result; 4155 4156 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4157 result.append(buffer); 4158 RecordTrack::appendDumpHeader(result); 4159 for (size_t i = 0; i < mTracks.size(); ++i) { 4160 sp<RecordTrack> track = mTracks[i]; 4161 if (track != 0) { 4162 track->dump(buffer, SIZE); 4163 result.append(buffer); 4164 } 4165 } 4166 4167 if (mActiveTrack != 0) { 4168 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4169 result.append(buffer); 4170 RecordTrack::appendDumpHeader(result); 4171 mActiveTrack->dump(buffer, SIZE); 4172 result.append(buffer); 4173 4174 } 4175 write(fd, result.string(), result.size()); 4176} 4177 4178// AudioBufferProvider interface 4179status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4180{ 4181 size_t framesReq = buffer->frameCount; 4182 size_t framesReady = mFrameCount - mRsmpInIndex; 4183 int channelCount; 4184 4185 if (framesReady == 0) { 4186 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4187 if (mBytesRead <= 0) { 4188 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4189 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4190 // Force input into standby so that it tries to 4191 // recover at next read attempt 4192 inputStandBy(); 4193 usleep(kRecordThreadSleepUs); 4194 } 4195 buffer->raw = NULL; 4196 buffer->frameCount = 0; 4197 return NOT_ENOUGH_DATA; 4198 } 4199 mRsmpInIndex = 0; 4200 framesReady = mFrameCount; 4201 } 4202 4203 if (framesReq > framesReady) { 4204 framesReq = framesReady; 4205 } 4206 4207 if (mChannelCount == 1 && mReqChannelCount == 2) { 4208 channelCount = 1; 4209 } else { 4210 channelCount = 2; 4211 } 4212 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4213 buffer->frameCount = framesReq; 4214 return NO_ERROR; 4215} 4216 4217// AudioBufferProvider interface 4218void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4219{ 4220 mRsmpInIndex += buffer->frameCount; 4221 buffer->frameCount = 0; 4222} 4223 4224bool AudioFlinger::RecordThread::checkForNewParameters_l() 4225{ 4226 bool reconfig = false; 4227 4228 while (!mNewParameters.isEmpty()) { 4229 status_t status = NO_ERROR; 4230 String8 keyValuePair = mNewParameters[0]; 4231 AudioParameter param = AudioParameter(keyValuePair); 4232 int value; 4233 audio_format_t reqFormat = mFormat; 4234 uint32_t reqSamplingRate = mReqSampleRate; 4235 uint32_t reqChannelCount = mReqChannelCount; 4236 4237 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4238 reqSamplingRate = value; 4239 reconfig = true; 4240 } 4241 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4242 reqFormat = (audio_format_t) value; 4243 reconfig = true; 4244 } 4245 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4246 reqChannelCount = popcount(value); 4247 reconfig = true; 4248 } 4249 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4250 // do not accept frame count changes if tracks are open as the track buffer 4251 // size depends on frame count and correct behavior would not be guaranteed 4252 // if frame count is changed after track creation 4253 if (mActiveTrack != 0) { 4254 status = INVALID_OPERATION; 4255 } else { 4256 reconfig = true; 4257 } 4258 } 4259 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4260 // forward device change to effects that have requested to be 4261 // aware of attached audio device. 4262 for (size_t i = 0; i < mEffectChains.size(); i++) { 4263 mEffectChains[i]->setDevice_l(value); 4264 } 4265 4266 // store input device and output device but do not forward output device to audio HAL. 4267 // Note that status is ignored by the caller for output device 4268 // (see AudioFlinger::setParameters() 4269 if (audio_is_output_devices(value)) { 4270 mOutDevice = value; 4271 status = BAD_VALUE; 4272 } else { 4273 mInDevice = value; 4274 // disable AEC and NS if the device is a BT SCO headset supporting those 4275 // pre processings 4276 if (mTracks.size() > 0) { 4277 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4278 mAudioFlinger->btNrecIsOff(); 4279 for (size_t i = 0; i < mTracks.size(); i++) { 4280 sp<RecordTrack> track = mTracks[i]; 4281 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4282 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4283 } 4284 } 4285 } 4286 } 4287 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4288 mAudioSource != (audio_source_t)value) { 4289 // forward device change to effects that have requested to be 4290 // aware of attached audio device. 4291 for (size_t i = 0; i < mEffectChains.size(); i++) { 4292 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4293 } 4294 mAudioSource = (audio_source_t)value; 4295 } 4296 if (status == NO_ERROR) { 4297 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4298 keyValuePair.string()); 4299 if (status == INVALID_OPERATION) { 4300 inputStandBy(); 4301 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4302 keyValuePair.string()); 4303 } 4304 if (reconfig) { 4305 if (status == BAD_VALUE && 4306 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4307 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4308 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4309 <= (2 * reqSamplingRate)) && 4310 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4311 <= FCC_2 && 4312 (reqChannelCount <= FCC_2)) { 4313 status = NO_ERROR; 4314 } 4315 if (status == NO_ERROR) { 4316 readInputParameters(); 4317 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4318 } 4319 } 4320 } 4321 4322 mNewParameters.removeAt(0); 4323 4324 mParamStatus = status; 4325 mParamCond.signal(); 4326 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4327 // already timed out waiting for the status and will never signal the condition. 4328 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4329 } 4330 return reconfig; 4331} 4332 4333String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4334{ 4335 char *s; 4336 String8 out_s8 = String8(); 4337 4338 Mutex::Autolock _l(mLock); 4339 if (initCheck() != NO_ERROR) { 4340 return out_s8; 4341 } 4342 4343 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4344 out_s8 = String8(s); 4345 free(s); 4346 return out_s8; 4347} 4348 4349void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4350 AudioSystem::OutputDescriptor desc; 4351 void *param2 = NULL; 4352 4353 switch (event) { 4354 case AudioSystem::INPUT_OPENED: 4355 case AudioSystem::INPUT_CONFIG_CHANGED: 4356 desc.channels = mChannelMask; 4357 desc.samplingRate = mSampleRate; 4358 desc.format = mFormat; 4359 desc.frameCount = mFrameCount; 4360 desc.latency = 0; 4361 param2 = &desc; 4362 break; 4363 4364 case AudioSystem::INPUT_CLOSED: 4365 default: 4366 break; 4367 } 4368 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4369} 4370 4371void AudioFlinger::RecordThread::readInputParameters() 4372{ 4373 delete mRsmpInBuffer; 4374 // mRsmpInBuffer is always assigned a new[] below 4375 delete mRsmpOutBuffer; 4376 mRsmpOutBuffer = NULL; 4377 delete mResampler; 4378 mResampler = NULL; 4379 4380 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4381 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4382 mChannelCount = (uint16_t)popcount(mChannelMask); 4383 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4384 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4385 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4386 mFrameCount = mInputBytes / mFrameSize; 4387 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4388 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4389 4390 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4391 { 4392 int channelCount; 4393 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4394 // stereo to mono post process as the resampler always outputs stereo. 4395 if (mChannelCount == 1 && mReqChannelCount == 2) { 4396 channelCount = 1; 4397 } else { 4398 channelCount = 2; 4399 } 4400 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4401 mResampler->setSampleRate(mSampleRate); 4402 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4403 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4404 4405 // optmization: if mono to mono, alter input frame count as if we were inputing 4406 // stereo samples 4407 if (mChannelCount == 1 && mReqChannelCount == 1) { 4408 mFrameCount >>= 1; 4409 } 4410 4411 } 4412 mRsmpInIndex = mFrameCount; 4413} 4414 4415unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4416{ 4417 Mutex::Autolock _l(mLock); 4418 if (initCheck() != NO_ERROR) { 4419 return 0; 4420 } 4421 4422 return mInput->stream->get_input_frames_lost(mInput->stream); 4423} 4424 4425uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4426{ 4427 Mutex::Autolock _l(mLock); 4428 uint32_t result = 0; 4429 if (getEffectChain_l(sessionId) != 0) { 4430 result = EFFECT_SESSION; 4431 } 4432 4433 for (size_t i = 0; i < mTracks.size(); ++i) { 4434 if (sessionId == mTracks[i]->sessionId()) { 4435 result |= TRACK_SESSION; 4436 break; 4437 } 4438 } 4439 4440 return result; 4441} 4442 4443KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4444{ 4445 KeyedVector<int, bool> ids; 4446 Mutex::Autolock _l(mLock); 4447 for (size_t j = 0; j < mTracks.size(); ++j) { 4448 sp<RecordThread::RecordTrack> track = mTracks[j]; 4449 int sessionId = track->sessionId(); 4450 if (ids.indexOfKey(sessionId) < 0) { 4451 ids.add(sessionId, true); 4452 } 4453 } 4454 return ids; 4455} 4456 4457AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4458{ 4459 Mutex::Autolock _l(mLock); 4460 AudioStreamIn *input = mInput; 4461 mInput = NULL; 4462 return input; 4463} 4464 4465// this method must always be called either with ThreadBase mLock held or inside the thread loop 4466audio_stream_t* AudioFlinger::RecordThread::stream() const 4467{ 4468 if (mInput == NULL) { 4469 return NULL; 4470 } 4471 return &mInput->stream->common; 4472} 4473 4474status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4475{ 4476 // only one chain per input thread 4477 if (mEffectChains.size() != 0) { 4478 return INVALID_OPERATION; 4479 } 4480 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4481 4482 chain->setInBuffer(NULL); 4483 chain->setOutBuffer(NULL); 4484 4485 checkSuspendOnAddEffectChain_l(chain); 4486 4487 mEffectChains.add(chain); 4488 4489 return NO_ERROR; 4490} 4491 4492size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4493{ 4494 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4495 ALOGW_IF(mEffectChains.size() != 1, 4496 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4497 chain.get(), mEffectChains.size(), this); 4498 if (mEffectChains.size() == 1) { 4499 mEffectChains.removeAt(0); 4500 } 4501 return 0; 4502} 4503 4504}; // namespace android 4505