Threads.cpp revision 9e8fcbcd8efa51d70d1207ff57bfbfe31324287a
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 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1434 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1435 mChannelCount = (uint16_t)popcount(mChannelMask); 1436 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1437 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1438 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1439 if (mFrameCount & 15) { 1440 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1441 mFrameCount); 1442 } 1443 1444 // Calculate size of normal mix buffer relative to the HAL output buffer size 1445 double multiplier = 1.0; 1446 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1447 kUseFastMixer == FastMixer_Dynamic)) { 1448 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1449 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1450 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1451 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1452 maxNormalFrameCount = maxNormalFrameCount & ~15; 1453 if (maxNormalFrameCount < minNormalFrameCount) { 1454 maxNormalFrameCount = minNormalFrameCount; 1455 } 1456 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1457 if (multiplier <= 1.0) { 1458 multiplier = 1.0; 1459 } else if (multiplier <= 2.0) { 1460 if (2 * mFrameCount <= maxNormalFrameCount) { 1461 multiplier = 2.0; 1462 } else { 1463 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1464 } 1465 } else { 1466 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1467 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1468 // track, but we sometimes have to do this to satisfy the maximum frame count 1469 // constraint) 1470 // FIXME this rounding up should not be done if no HAL SRC 1471 uint32_t truncMult = (uint32_t) multiplier; 1472 if ((truncMult & 1)) { 1473 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1474 ++truncMult; 1475 } 1476 } 1477 multiplier = (double) truncMult; 1478 } 1479 } 1480 mNormalFrameCount = multiplier * mFrameCount; 1481 // round up to nearest 16 frames to satisfy AudioMixer 1482 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1483 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1484 mNormalFrameCount); 1485 1486 delete[] mMixBuffer; 1487 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1488 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1489 1490 // force reconfiguration of effect chains and engines to take new buffer size and audio 1491 // parameters into account 1492 // Note that mLock is not held when readOutputParameters() is called from the constructor 1493 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1494 // matter. 1495 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1496 Vector< sp<EffectChain> > effectChains = mEffectChains; 1497 for (size_t i = 0; i < effectChains.size(); i ++) { 1498 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1499 } 1500} 1501 1502 1503status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1504{ 1505 if (halFrames == NULL || dspFrames == NULL) { 1506 return BAD_VALUE; 1507 } 1508 Mutex::Autolock _l(mLock); 1509 if (initCheck() != NO_ERROR) { 1510 return INVALID_OPERATION; 1511 } 1512 size_t framesWritten = mBytesWritten / mFrameSize; 1513 *halFrames = framesWritten; 1514 1515 if (isSuspended()) { 1516 // return an estimation of rendered frames when the output is suspended 1517 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1518 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1519 return NO_ERROR; 1520 } else { 1521 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1522 } 1523} 1524 1525uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1526{ 1527 Mutex::Autolock _l(mLock); 1528 uint32_t result = 0; 1529 if (getEffectChain_l(sessionId) != 0) { 1530 result = EFFECT_SESSION; 1531 } 1532 1533 for (size_t i = 0; i < mTracks.size(); ++i) { 1534 sp<Track> track = mTracks[i]; 1535 if (sessionId == track->sessionId() && !track->isInvalid()) { 1536 result |= TRACK_SESSION; 1537 break; 1538 } 1539 } 1540 1541 return result; 1542} 1543 1544uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1545{ 1546 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1547 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1548 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1549 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1550 } 1551 for (size_t i = 0; i < mTracks.size(); i++) { 1552 sp<Track> track = mTracks[i]; 1553 if (sessionId == track->sessionId() && !track->isInvalid()) { 1554 return AudioSystem::getStrategyForStream(track->streamType()); 1555 } 1556 } 1557 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1558} 1559 1560 1561AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1562{ 1563 Mutex::Autolock _l(mLock); 1564 return mOutput; 1565} 1566 1567AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1568{ 1569 Mutex::Autolock _l(mLock); 1570 AudioStreamOut *output = mOutput; 1571 mOutput = NULL; 1572 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1573 // must push a NULL and wait for ack 1574 mOutputSink.clear(); 1575 mPipeSink.clear(); 1576 mNormalSink.clear(); 1577 return output; 1578} 1579 1580// this method must always be called either with ThreadBase mLock held or inside the thread loop 1581audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1582{ 1583 if (mOutput == NULL) { 1584 return NULL; 1585 } 1586 return &mOutput->stream->common; 1587} 1588 1589uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1590{ 1591 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1592} 1593 1594status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1595{ 1596 if (!isValidSyncEvent(event)) { 1597 return BAD_VALUE; 1598 } 1599 1600 Mutex::Autolock _l(mLock); 1601 1602 for (size_t i = 0; i < mTracks.size(); ++i) { 1603 sp<Track> track = mTracks[i]; 1604 if (event->triggerSession() == track->sessionId()) { 1605 (void) track->setSyncEvent(event); 1606 return NO_ERROR; 1607 } 1608 } 1609 1610 return NAME_NOT_FOUND; 1611} 1612 1613bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1614{ 1615 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1616} 1617 1618void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1619 const Vector< sp<Track> >& tracksToRemove) 1620{ 1621 size_t count = tracksToRemove.size(); 1622 if (CC_UNLIKELY(count)) { 1623 for (size_t i = 0 ; i < count ; i++) { 1624 const sp<Track>& track = tracksToRemove.itemAt(i); 1625 if ((track->sharedBuffer() != 0) && 1626 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1627 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1628 } 1629 } 1630 } 1631 1632} 1633 1634void AudioFlinger::PlaybackThread::checkSilentMode_l() 1635{ 1636 if (!mMasterMute) { 1637 char value[PROPERTY_VALUE_MAX]; 1638 if (property_get("ro.audio.silent", value, "0") > 0) { 1639 char *endptr; 1640 unsigned long ul = strtoul(value, &endptr, 0); 1641 if (*endptr == '\0' && ul != 0) { 1642 ALOGD("Silence is golden"); 1643 // The setprop command will not allow a property to be changed after 1644 // the first time it is set, so we don't have to worry about un-muting. 1645 setMasterMute_l(true); 1646 } 1647 } 1648 } 1649} 1650 1651// shared by MIXER and DIRECT, overridden by DUPLICATING 1652void AudioFlinger::PlaybackThread::threadLoop_write() 1653{ 1654 // FIXME rewrite to reduce number of system calls 1655 mLastWriteTime = systemTime(); 1656 mInWrite = true; 1657 int bytesWritten; 1658 1659 // If an NBAIO sink is present, use it to write the normal mixer's submix 1660 if (mNormalSink != 0) { 1661#define mBitShift 2 // FIXME 1662 size_t count = mixBufferSize >> mBitShift; 1663 ATRACE_BEGIN("write"); 1664 // update the setpoint when AudioFlinger::mScreenState changes 1665 uint32_t screenState = AudioFlinger::mScreenState; 1666 if (screenState != mScreenState) { 1667 mScreenState = screenState; 1668 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1669 if (pipe != NULL) { 1670 pipe->setAvgFrames((mScreenState & 1) ? 1671 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1672 } 1673 } 1674 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1675 ATRACE_END(); 1676 if (framesWritten > 0) { 1677 bytesWritten = framesWritten << mBitShift; 1678 } else { 1679 bytesWritten = framesWritten; 1680 } 1681 // otherwise use the HAL / AudioStreamOut directly 1682 } else { 1683 // Direct output thread. 1684 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1685 } 1686 1687 if (bytesWritten > 0) { 1688 mBytesWritten += mixBufferSize; 1689 } 1690 mNumWrites++; 1691 mInWrite = false; 1692} 1693 1694/* 1695The derived values that are cached: 1696 - mixBufferSize from frame count * frame size 1697 - activeSleepTime from activeSleepTimeUs() 1698 - idleSleepTime from idleSleepTimeUs() 1699 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1700 - maxPeriod from frame count and sample rate (MIXER only) 1701 1702The parameters that affect these derived values are: 1703 - frame count 1704 - frame size 1705 - sample rate 1706 - device type: A2DP or not 1707 - device latency 1708 - format: PCM or not 1709 - active sleep time 1710 - idle sleep time 1711*/ 1712 1713void AudioFlinger::PlaybackThread::cacheParameters_l() 1714{ 1715 mixBufferSize = mNormalFrameCount * mFrameSize; 1716 activeSleepTime = activeSleepTimeUs(); 1717 idleSleepTime = idleSleepTimeUs(); 1718} 1719 1720void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1721{ 1722 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1723 this, streamType, mTracks.size()); 1724 Mutex::Autolock _l(mLock); 1725 1726 size_t size = mTracks.size(); 1727 for (size_t i = 0; i < size; i++) { 1728 sp<Track> t = mTracks[i]; 1729 if (t->streamType() == streamType) { 1730 t->invalidate(); 1731 } 1732 } 1733} 1734 1735status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1736{ 1737 int session = chain->sessionId(); 1738 int16_t *buffer = mMixBuffer; 1739 bool ownsBuffer = false; 1740 1741 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1742 if (session > 0) { 1743 // Only one effect chain can be present in direct output thread and it uses 1744 // the mix buffer as input 1745 if (mType != DIRECT) { 1746 size_t numSamples = mNormalFrameCount * mChannelCount; 1747 buffer = new int16_t[numSamples]; 1748 memset(buffer, 0, numSamples * sizeof(int16_t)); 1749 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1750 ownsBuffer = true; 1751 } 1752 1753 // Attach all tracks with same session ID to this chain. 1754 for (size_t i = 0; i < mTracks.size(); ++i) { 1755 sp<Track> track = mTracks[i]; 1756 if (session == track->sessionId()) { 1757 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1758 buffer); 1759 track->setMainBuffer(buffer); 1760 chain->incTrackCnt(); 1761 } 1762 } 1763 1764 // indicate all active tracks in the chain 1765 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1766 sp<Track> track = mActiveTracks[i].promote(); 1767 if (track == 0) { 1768 continue; 1769 } 1770 if (session == track->sessionId()) { 1771 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1772 chain->incActiveTrackCnt(); 1773 } 1774 } 1775 } 1776 1777 chain->setInBuffer(buffer, ownsBuffer); 1778 chain->setOutBuffer(mMixBuffer); 1779 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1780 // chains list in order to be processed last as it contains output stage effects 1781 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1782 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1783 // after track specific effects and before output stage 1784 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1785 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1786 // Effect chain for other sessions are inserted at beginning of effect 1787 // chains list to be processed before output mix effects. Relative order between other 1788 // sessions is not important 1789 size_t size = mEffectChains.size(); 1790 size_t i = 0; 1791 for (i = 0; i < size; i++) { 1792 if (mEffectChains[i]->sessionId() < session) { 1793 break; 1794 } 1795 } 1796 mEffectChains.insertAt(chain, i); 1797 checkSuspendOnAddEffectChain_l(chain); 1798 1799 return NO_ERROR; 1800} 1801 1802size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1803{ 1804 int session = chain->sessionId(); 1805 1806 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1807 1808 for (size_t i = 0; i < mEffectChains.size(); i++) { 1809 if (chain == mEffectChains[i]) { 1810 mEffectChains.removeAt(i); 1811 // detach all active tracks from the chain 1812 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1813 sp<Track> track = mActiveTracks[i].promote(); 1814 if (track == 0) { 1815 continue; 1816 } 1817 if (session == track->sessionId()) { 1818 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1819 chain.get(), session); 1820 chain->decActiveTrackCnt(); 1821 } 1822 } 1823 1824 // detach all tracks with same session ID from this chain 1825 for (size_t i = 0; i < mTracks.size(); ++i) { 1826 sp<Track> track = mTracks[i]; 1827 if (session == track->sessionId()) { 1828 track->setMainBuffer(mMixBuffer); 1829 chain->decTrackCnt(); 1830 } 1831 } 1832 break; 1833 } 1834 } 1835 return mEffectChains.size(); 1836} 1837 1838status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1839 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1840{ 1841 Mutex::Autolock _l(mLock); 1842 return attachAuxEffect_l(track, EffectId); 1843} 1844 1845status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1846 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1847{ 1848 status_t status = NO_ERROR; 1849 1850 if (EffectId == 0) { 1851 track->setAuxBuffer(0, NULL); 1852 } else { 1853 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1854 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1855 if (effect != 0) { 1856 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1857 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1858 } else { 1859 status = INVALID_OPERATION; 1860 } 1861 } else { 1862 status = BAD_VALUE; 1863 } 1864 } 1865 return status; 1866} 1867 1868void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1869{ 1870 for (size_t i = 0; i < mTracks.size(); ++i) { 1871 sp<Track> track = mTracks[i]; 1872 if (track->auxEffectId() == effectId) { 1873 attachAuxEffect_l(track, 0); 1874 } 1875 } 1876} 1877 1878bool AudioFlinger::PlaybackThread::threadLoop() 1879{ 1880 Vector< sp<Track> > tracksToRemove; 1881 1882 standbyTime = systemTime(); 1883 1884 // MIXER 1885 nsecs_t lastWarning = 0; 1886 1887 // DUPLICATING 1888 // FIXME could this be made local to while loop? 1889 writeFrames = 0; 1890 1891 cacheParameters_l(); 1892 sleepTime = idleSleepTime; 1893 1894 if (mType == MIXER) { 1895 sleepTimeShift = 0; 1896 } 1897 1898 CpuStats cpuStats; 1899 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1900 1901 acquireWakeLock(); 1902 1903 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1904 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1905 // and then that string will be logged at the next convenient opportunity. 1906 const char *logString = NULL; 1907 1908 while (!exitPending()) 1909 { 1910 cpuStats.sample(myName); 1911 1912 Vector< sp<EffectChain> > effectChains; 1913 1914 processConfigEvents(); 1915 1916 { // scope for mLock 1917 1918 Mutex::Autolock _l(mLock); 1919 1920 if (logString != NULL) { 1921 mNBLogWriter->logTimestamp(); 1922 mNBLogWriter->log(logString); 1923 logString = NULL; 1924 } 1925 1926 if (checkForNewParameters_l()) { 1927 cacheParameters_l(); 1928 } 1929 1930 saveOutputTracks(); 1931 1932 // put audio hardware into standby after short delay 1933 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1934 isSuspended())) { 1935 if (!mStandby) { 1936 1937 threadLoop_standby(); 1938 1939 mStandby = true; 1940 } 1941 1942 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1943 // we're about to wait, flush the binder command buffer 1944 IPCThreadState::self()->flushCommands(); 1945 1946 clearOutputTracks(); 1947 1948 if (exitPending()) { 1949 break; 1950 } 1951 1952 releaseWakeLock_l(); 1953 // wait until we have something to do... 1954 ALOGV("%s going to sleep", myName.string()); 1955 mWaitWorkCV.wait(mLock); 1956 ALOGV("%s waking up", myName.string()); 1957 acquireWakeLock_l(); 1958 1959 mMixerStatus = MIXER_IDLE; 1960 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1961 mBytesWritten = 0; 1962 1963 checkSilentMode_l(); 1964 1965 standbyTime = systemTime() + standbyDelay; 1966 sleepTime = idleSleepTime; 1967 if (mType == MIXER) { 1968 sleepTimeShift = 0; 1969 } 1970 1971 continue; 1972 } 1973 } 1974 1975 // mMixerStatusIgnoringFastTracks is also updated internally 1976 mMixerStatus = prepareTracks_l(&tracksToRemove); 1977 1978 // prevent any changes in effect chain list and in each effect chain 1979 // during mixing and effect process as the audio buffers could be deleted 1980 // or modified if an effect is created or deleted 1981 lockEffectChains_l(effectChains); 1982 } 1983 1984 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1985 threadLoop_mix(); 1986 } else { 1987 threadLoop_sleepTime(); 1988 } 1989 1990 if (isSuspended()) { 1991 sleepTime = suspendSleepTimeUs(); 1992 mBytesWritten += mixBufferSize; 1993 } 1994 1995 // only process effects if we're going to write 1996 if (sleepTime == 0) { 1997 for (size_t i = 0; i < effectChains.size(); i ++) { 1998 effectChains[i]->process_l(); 1999 } 2000 } 2001 2002 // enable changes in effect chain 2003 unlockEffectChains(effectChains); 2004 2005 // sleepTime == 0 means we must write to audio hardware 2006 if (sleepTime == 0) { 2007 2008 threadLoop_write(); 2009 2010if (mType == MIXER) { 2011 // write blocked detection 2012 nsecs_t now = systemTime(); 2013 nsecs_t delta = now - mLastWriteTime; 2014 if (!mStandby && delta > maxPeriod) { 2015 mNumDelayedWrites++; 2016 if ((now - lastWarning) > kWarningThrottleNs) { 2017 ATRACE_NAME("underrun"); 2018 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2019 ns2ms(delta), mNumDelayedWrites, this); 2020 lastWarning = now; 2021 } 2022 } 2023} 2024 2025 mStandby = false; 2026 } else { 2027 usleep(sleepTime); 2028 } 2029 2030 // Finally let go of removed track(s), without the lock held 2031 // since we can't guarantee the destructors won't acquire that 2032 // same lock. This will also mutate and push a new fast mixer state. 2033 threadLoop_removeTracks(tracksToRemove); 2034 tracksToRemove.clear(); 2035 2036 // FIXME I don't understand the need for this here; 2037 // it was in the original code but maybe the 2038 // assignment in saveOutputTracks() makes this unnecessary? 2039 clearOutputTracks(); 2040 2041 // Effect chains will be actually deleted here if they were removed from 2042 // mEffectChains list during mixing or effects processing 2043 effectChains.clear(); 2044 2045 // FIXME Note that the above .clear() is no longer necessary since effectChains 2046 // is now local to this block, but will keep it for now (at least until merge done). 2047 } 2048 2049 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2050 if (mType == MIXER || mType == DIRECT) { 2051 // put output stream into standby mode 2052 if (!mStandby) { 2053 mOutput->stream->common.standby(&mOutput->stream->common); 2054 } 2055 } 2056 2057 releaseWakeLock(); 2058 2059 ALOGV("Thread %p type %d exiting", this, mType); 2060 return false; 2061} 2062 2063 2064// ---------------------------------------------------------------------------- 2065 2066AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2067 audio_io_handle_t id, audio_devices_t device, type_t type) 2068 : PlaybackThread(audioFlinger, output, id, device, type), 2069 // mAudioMixer below 2070 // mFastMixer below 2071 mFastMixerFutex(0) 2072 // mOutputSink below 2073 // mPipeSink below 2074 // mNormalSink below 2075{ 2076 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2077 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2078 "mFrameCount=%d, mNormalFrameCount=%d", 2079 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2080 mNormalFrameCount); 2081 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2082 2083 // FIXME - Current mixer implementation only supports stereo output 2084 if (mChannelCount != FCC_2) { 2085 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2086 } 2087 2088 // create an NBAIO sink for the HAL output stream, and negotiate 2089 mOutputSink = new AudioStreamOutSink(output->stream); 2090 size_t numCounterOffers = 0; 2091 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2092 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2093 ALOG_ASSERT(index == 0); 2094 2095 // initialize fast mixer depending on configuration 2096 bool initFastMixer; 2097 switch (kUseFastMixer) { 2098 case FastMixer_Never: 2099 initFastMixer = false; 2100 break; 2101 case FastMixer_Always: 2102 initFastMixer = true; 2103 break; 2104 case FastMixer_Static: 2105 case FastMixer_Dynamic: 2106 initFastMixer = mFrameCount < mNormalFrameCount; 2107 break; 2108 } 2109 if (initFastMixer) { 2110 2111 // create a MonoPipe to connect our submix to FastMixer 2112 NBAIO_Format format = mOutputSink->format(); 2113 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2114 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2115 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2116 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2117 const NBAIO_Format offers[1] = {format}; 2118 size_t numCounterOffers = 0; 2119 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2120 ALOG_ASSERT(index == 0); 2121 monoPipe->setAvgFrames((mScreenState & 1) ? 2122 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2123 mPipeSink = monoPipe; 2124 2125#ifdef TEE_SINK 2126 if (mTeeSinkOutputEnabled) { 2127 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2128 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2129 numCounterOffers = 0; 2130 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2131 ALOG_ASSERT(index == 0); 2132 mTeeSink = teeSink; 2133 PipeReader *teeSource = new PipeReader(*teeSink); 2134 numCounterOffers = 0; 2135 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2136 ALOG_ASSERT(index == 0); 2137 mTeeSource = teeSource; 2138 } 2139#endif 2140 2141 // create fast mixer and configure it initially with just one fast track for our submix 2142 mFastMixer = new FastMixer(); 2143 FastMixerStateQueue *sq = mFastMixer->sq(); 2144#ifdef STATE_QUEUE_DUMP 2145 sq->setObserverDump(&mStateQueueObserverDump); 2146 sq->setMutatorDump(&mStateQueueMutatorDump); 2147#endif 2148 FastMixerState *state = sq->begin(); 2149 FastTrack *fastTrack = &state->mFastTracks[0]; 2150 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2151 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2152 fastTrack->mVolumeProvider = NULL; 2153 fastTrack->mGeneration++; 2154 state->mFastTracksGen++; 2155 state->mTrackMask = 1; 2156 // fast mixer will use the HAL output sink 2157 state->mOutputSink = mOutputSink.get(); 2158 state->mOutputSinkGen++; 2159 state->mFrameCount = mFrameCount; 2160 state->mCommand = FastMixerState::COLD_IDLE; 2161 // already done in constructor initialization list 2162 //mFastMixerFutex = 0; 2163 state->mColdFutexAddr = &mFastMixerFutex; 2164 state->mColdGen++; 2165 state->mDumpState = &mFastMixerDumpState; 2166#ifdef TEE_SINK 2167 state->mTeeSink = mTeeSink.get(); 2168#endif 2169 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2170 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2171 sq->end(); 2172 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2173 2174 // start the fast mixer 2175 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2176 pid_t tid = mFastMixer->getTid(); 2177 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2178 if (err != 0) { 2179 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2180 kPriorityFastMixer, getpid_cached, tid, err); 2181 } 2182 2183#ifdef AUDIO_WATCHDOG 2184 // create and start the watchdog 2185 mAudioWatchdog = new AudioWatchdog(); 2186 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2187 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2188 tid = mAudioWatchdog->getTid(); 2189 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2190 if (err != 0) { 2191 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2192 kPriorityFastMixer, getpid_cached, tid, err); 2193 } 2194#endif 2195 2196 } else { 2197 mFastMixer = NULL; 2198 } 2199 2200 switch (kUseFastMixer) { 2201 case FastMixer_Never: 2202 case FastMixer_Dynamic: 2203 mNormalSink = mOutputSink; 2204 break; 2205 case FastMixer_Always: 2206 mNormalSink = mPipeSink; 2207 break; 2208 case FastMixer_Static: 2209 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2210 break; 2211 } 2212} 2213 2214AudioFlinger::MixerThread::~MixerThread() 2215{ 2216 if (mFastMixer != NULL) { 2217 FastMixerStateQueue *sq = mFastMixer->sq(); 2218 FastMixerState *state = sq->begin(); 2219 if (state->mCommand == FastMixerState::COLD_IDLE) { 2220 int32_t old = android_atomic_inc(&mFastMixerFutex); 2221 if (old == -1) { 2222 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2223 } 2224 } 2225 state->mCommand = FastMixerState::EXIT; 2226 sq->end(); 2227 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2228 mFastMixer->join(); 2229 // Though the fast mixer thread has exited, it's state queue is still valid. 2230 // We'll use that extract the final state which contains one remaining fast track 2231 // corresponding to our sub-mix. 2232 state = sq->begin(); 2233 ALOG_ASSERT(state->mTrackMask == 1); 2234 FastTrack *fastTrack = &state->mFastTracks[0]; 2235 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2236 delete fastTrack->mBufferProvider; 2237 sq->end(false /*didModify*/); 2238 delete mFastMixer; 2239#ifdef AUDIO_WATCHDOG 2240 if (mAudioWatchdog != 0) { 2241 mAudioWatchdog->requestExit(); 2242 mAudioWatchdog->requestExitAndWait(); 2243 mAudioWatchdog.clear(); 2244 } 2245#endif 2246 } 2247 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2248 delete mAudioMixer; 2249} 2250 2251 2252uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2253{ 2254 if (mFastMixer != NULL) { 2255 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2256 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2257 } 2258 return latency; 2259} 2260 2261 2262void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2263{ 2264 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2265} 2266 2267void AudioFlinger::MixerThread::threadLoop_write() 2268{ 2269 // FIXME we should only do one push per cycle; confirm this is true 2270 // Start the fast mixer if it's not already running 2271 if (mFastMixer != NULL) { 2272 FastMixerStateQueue *sq = mFastMixer->sq(); 2273 FastMixerState *state = sq->begin(); 2274 if (state->mCommand != FastMixerState::MIX_WRITE && 2275 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2276 if (state->mCommand == FastMixerState::COLD_IDLE) { 2277 int32_t old = android_atomic_inc(&mFastMixerFutex); 2278 if (old == -1) { 2279 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2280 } 2281#ifdef AUDIO_WATCHDOG 2282 if (mAudioWatchdog != 0) { 2283 mAudioWatchdog->resume(); 2284 } 2285#endif 2286 } 2287 state->mCommand = FastMixerState::MIX_WRITE; 2288 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2289 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2290 sq->end(); 2291 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2292 if (kUseFastMixer == FastMixer_Dynamic) { 2293 mNormalSink = mPipeSink; 2294 } 2295 } else { 2296 sq->end(false /*didModify*/); 2297 } 2298 } 2299 PlaybackThread::threadLoop_write(); 2300} 2301 2302void AudioFlinger::MixerThread::threadLoop_standby() 2303{ 2304 // Idle the fast mixer if it's currently running 2305 if (mFastMixer != NULL) { 2306 FastMixerStateQueue *sq = mFastMixer->sq(); 2307 FastMixerState *state = sq->begin(); 2308 if (!(state->mCommand & FastMixerState::IDLE)) { 2309 state->mCommand = FastMixerState::COLD_IDLE; 2310 state->mColdFutexAddr = &mFastMixerFutex; 2311 state->mColdGen++; 2312 mFastMixerFutex = 0; 2313 sq->end(); 2314 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2315 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2316 if (kUseFastMixer == FastMixer_Dynamic) { 2317 mNormalSink = mOutputSink; 2318 } 2319#ifdef AUDIO_WATCHDOG 2320 if (mAudioWatchdog != 0) { 2321 mAudioWatchdog->pause(); 2322 } 2323#endif 2324 } else { 2325 sq->end(false /*didModify*/); 2326 } 2327 } 2328 PlaybackThread::threadLoop_standby(); 2329} 2330 2331// shared by MIXER and DIRECT, overridden by DUPLICATING 2332void AudioFlinger::PlaybackThread::threadLoop_standby() 2333{ 2334 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2335 mOutput->stream->common.standby(&mOutput->stream->common); 2336} 2337 2338void AudioFlinger::MixerThread::threadLoop_mix() 2339{ 2340 // obtain the presentation timestamp of the next output buffer 2341 int64_t pts; 2342 status_t status = INVALID_OPERATION; 2343 2344 if (mNormalSink != 0) { 2345 status = mNormalSink->getNextWriteTimestamp(&pts); 2346 } else { 2347 status = mOutputSink->getNextWriteTimestamp(&pts); 2348 } 2349 2350 if (status != NO_ERROR) { 2351 pts = AudioBufferProvider::kInvalidPTS; 2352 } 2353 2354 // mix buffers... 2355 mAudioMixer->process(pts); 2356 // increase sleep time progressively when application underrun condition clears. 2357 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2358 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2359 // such that we would underrun the audio HAL. 2360 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2361 sleepTimeShift--; 2362 } 2363 sleepTime = 0; 2364 standbyTime = systemTime() + standbyDelay; 2365 //TODO: delay standby when effects have a tail 2366} 2367 2368void AudioFlinger::MixerThread::threadLoop_sleepTime() 2369{ 2370 // If no tracks are ready, sleep once for the duration of an output 2371 // buffer size, then write 0s to the output 2372 if (sleepTime == 0) { 2373 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2374 sleepTime = activeSleepTime >> sleepTimeShift; 2375 if (sleepTime < kMinThreadSleepTimeUs) { 2376 sleepTime = kMinThreadSleepTimeUs; 2377 } 2378 // reduce sleep time in case of consecutive application underruns to avoid 2379 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2380 // duration we would end up writing less data than needed by the audio HAL if 2381 // the condition persists. 2382 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2383 sleepTimeShift++; 2384 } 2385 } else { 2386 sleepTime = idleSleepTime; 2387 } 2388 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2389 memset (mMixBuffer, 0, mixBufferSize); 2390 sleepTime = 0; 2391 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2392 "anticipated start"); 2393 } 2394 // TODO add standby time extension fct of effect tail 2395} 2396 2397// prepareTracks_l() must be called with ThreadBase::mLock held 2398AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2399 Vector< sp<Track> > *tracksToRemove) 2400{ 2401 2402 mixer_state mixerStatus = MIXER_IDLE; 2403 // find out which tracks need to be processed 2404 size_t count = mActiveTracks.size(); 2405 size_t mixedTracks = 0; 2406 size_t tracksWithEffect = 0; 2407 // counts only _active_ fast tracks 2408 size_t fastTracks = 0; 2409 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2410 2411 float masterVolume = mMasterVolume; 2412 bool masterMute = mMasterMute; 2413 2414 if (masterMute) { 2415 masterVolume = 0; 2416 } 2417 // Delegate master volume control to effect in output mix effect chain if needed 2418 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2419 if (chain != 0) { 2420 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2421 chain->setVolume_l(&v, &v); 2422 masterVolume = (float)((v + (1 << 23)) >> 24); 2423 chain.clear(); 2424 } 2425 2426 // prepare a new state to push 2427 FastMixerStateQueue *sq = NULL; 2428 FastMixerState *state = NULL; 2429 bool didModify = false; 2430 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2431 if (mFastMixer != NULL) { 2432 sq = mFastMixer->sq(); 2433 state = sq->begin(); 2434 } 2435 2436 for (size_t i=0 ; i<count ; i++) { 2437 sp<Track> t = mActiveTracks[i].promote(); 2438 if (t == 0) { 2439 continue; 2440 } 2441 2442 // this const just means the local variable doesn't change 2443 Track* const track = t.get(); 2444 2445 // process fast tracks 2446 if (track->isFastTrack()) { 2447 2448 // It's theoretically possible (though unlikely) for a fast track to be created 2449 // and then removed within the same normal mix cycle. This is not a problem, as 2450 // the track never becomes active so it's fast mixer slot is never touched. 2451 // The converse, of removing an (active) track and then creating a new track 2452 // at the identical fast mixer slot within the same normal mix cycle, 2453 // is impossible because the slot isn't marked available until the end of each cycle. 2454 int j = track->mFastIndex; 2455 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2456 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2457 FastTrack *fastTrack = &state->mFastTracks[j]; 2458 2459 // Determine whether the track is currently in underrun condition, 2460 // and whether it had a recent underrun. 2461 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2462 FastTrackUnderruns underruns = ftDump->mUnderruns; 2463 uint32_t recentFull = (underruns.mBitFields.mFull - 2464 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2465 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2466 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2467 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2468 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2469 uint32_t recentUnderruns = recentPartial + recentEmpty; 2470 track->mObservedUnderruns = underruns; 2471 // don't count underruns that occur while stopping or pausing 2472 // or stopped which can occur when flush() is called while active 2473 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2474 track->mUnderrunCount += recentUnderruns; 2475 } 2476 2477 // This is similar to the state machine for normal tracks, 2478 // with a few modifications for fast tracks. 2479 bool isActive = true; 2480 switch (track->mState) { 2481 case TrackBase::STOPPING_1: 2482 // track stays active in STOPPING_1 state until first underrun 2483 if (recentUnderruns > 0) { 2484 track->mState = TrackBase::STOPPING_2; 2485 } 2486 break; 2487 case TrackBase::PAUSING: 2488 // ramp down is not yet implemented 2489 track->setPaused(); 2490 break; 2491 case TrackBase::RESUMING: 2492 // ramp up is not yet implemented 2493 track->mState = TrackBase::ACTIVE; 2494 break; 2495 case TrackBase::ACTIVE: 2496 if (recentFull > 0 || recentPartial > 0) { 2497 // track has provided at least some frames recently: reset retry count 2498 track->mRetryCount = kMaxTrackRetries; 2499 } 2500 if (recentUnderruns == 0) { 2501 // no recent underruns: stay active 2502 break; 2503 } 2504 // there has recently been an underrun of some kind 2505 if (track->sharedBuffer() == 0) { 2506 // were any of the recent underruns "empty" (no frames available)? 2507 if (recentEmpty == 0) { 2508 // no, then ignore the partial underruns as they are allowed indefinitely 2509 break; 2510 } 2511 // there has recently been an "empty" underrun: decrement the retry counter 2512 if (--(track->mRetryCount) > 0) { 2513 break; 2514 } 2515 // indicate to client process that the track was disabled because of underrun; 2516 // it will then automatically call start() when data is available 2517 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2518 // remove from active list, but state remains ACTIVE [confusing but true] 2519 isActive = false; 2520 break; 2521 } 2522 // fall through 2523 case TrackBase::STOPPING_2: 2524 case TrackBase::PAUSED: 2525 case TrackBase::TERMINATED: 2526 case TrackBase::STOPPED: 2527 case TrackBase::FLUSHED: // flush() while active 2528 // Check for presentation complete if track is inactive 2529 // We have consumed all the buffers of this track. 2530 // This would be incomplete if we auto-paused on underrun 2531 { 2532 size_t audioHALFrames = 2533 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2534 size_t framesWritten = mBytesWritten / mFrameSize; 2535 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2536 // track stays in active list until presentation is complete 2537 break; 2538 } 2539 } 2540 if (track->isStopping_2()) { 2541 track->mState = TrackBase::STOPPED; 2542 } 2543 if (track->isStopped()) { 2544 // Can't reset directly, as fast mixer is still polling this track 2545 // track->reset(); 2546 // So instead mark this track as needing to be reset after push with ack 2547 resetMask |= 1 << i; 2548 } 2549 isActive = false; 2550 break; 2551 case TrackBase::IDLE: 2552 default: 2553 LOG_FATAL("unexpected track state %d", track->mState); 2554 } 2555 2556 if (isActive) { 2557 // was it previously inactive? 2558 if (!(state->mTrackMask & (1 << j))) { 2559 ExtendedAudioBufferProvider *eabp = track; 2560 VolumeProvider *vp = track; 2561 fastTrack->mBufferProvider = eabp; 2562 fastTrack->mVolumeProvider = vp; 2563 fastTrack->mSampleRate = track->mSampleRate; 2564 fastTrack->mChannelMask = track->mChannelMask; 2565 fastTrack->mGeneration++; 2566 state->mTrackMask |= 1 << j; 2567 didModify = true; 2568 // no acknowledgement required for newly active tracks 2569 } 2570 // cache the combined master volume and stream type volume for fast mixer; this 2571 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2572 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2573 ++fastTracks; 2574 } else { 2575 // was it previously active? 2576 if (state->mTrackMask & (1 << j)) { 2577 fastTrack->mBufferProvider = NULL; 2578 fastTrack->mGeneration++; 2579 state->mTrackMask &= ~(1 << j); 2580 didModify = true; 2581 // If any fast tracks were removed, we must wait for acknowledgement 2582 // because we're about to decrement the last sp<> on those tracks. 2583 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2584 } else { 2585 LOG_FATAL("fast track %d should have been active", j); 2586 } 2587 tracksToRemove->add(track); 2588 // Avoids a misleading display in dumpsys 2589 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2590 } 2591 continue; 2592 } 2593 2594 { // local variable scope to avoid goto warning 2595 2596 audio_track_cblk_t* cblk = track->cblk(); 2597 2598 // The first time a track is added we wait 2599 // for all its buffers to be filled before processing it 2600 int name = track->name(); 2601 // make sure that we have enough frames to mix one full buffer. 2602 // enforce this condition only once to enable draining the buffer in case the client 2603 // app does not call stop() and relies on underrun to stop: 2604 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2605 // during last round 2606 size_t desiredFrames; 2607 if (t->sampleRate() == mSampleRate) { 2608 desiredFrames = mNormalFrameCount; 2609 } else { 2610 // +1 for rounding and +1 for additional sample needed for interpolation 2611 desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2612 // add frames already consumed but not yet released by the resampler 2613 // because cblk->framesReady() will include these frames 2614 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2615 // the minimum track buffer size is normally twice the number of frames necessary 2616 // to fill one buffer and the resampler should not leave more than one buffer worth 2617 // of unreleased frames after each pass, but just in case... 2618 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2619 } 2620 uint32_t minFrames = 1; 2621 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2622 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2623 minFrames = desiredFrames; 2624 } 2625 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2626 size_t framesReady; 2627 if (track->sharedBuffer() == 0) { 2628 framesReady = track->framesReady(); 2629 } else if (track->isStopped()) { 2630 framesReady = 0; 2631 } else { 2632 framesReady = 1; 2633 } 2634 if ((framesReady >= minFrames) && track->isReady() && 2635 !track->isPaused() && !track->isTerminated()) 2636 { 2637 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2638 this); 2639 2640 mixedTracks++; 2641 2642 // track->mainBuffer() != mMixBuffer means there is an effect chain 2643 // connected to the track 2644 chain.clear(); 2645 if (track->mainBuffer() != mMixBuffer) { 2646 chain = getEffectChain_l(track->sessionId()); 2647 // Delegate volume control to effect in track effect chain if needed 2648 if (chain != 0) { 2649 tracksWithEffect++; 2650 } else { 2651 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2652 "session %d", 2653 name, track->sessionId()); 2654 } 2655 } 2656 2657 2658 int param = AudioMixer::VOLUME; 2659 if (track->mFillingUpStatus == Track::FS_FILLED) { 2660 // no ramp for the first volume setting 2661 track->mFillingUpStatus = Track::FS_ACTIVE; 2662 if (track->mState == TrackBase::RESUMING) { 2663 track->mState = TrackBase::ACTIVE; 2664 param = AudioMixer::RAMP_VOLUME; 2665 } 2666 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2667 } else if (cblk->server != 0) { 2668 // If the track is stopped before the first frame was mixed, 2669 // do not apply ramp 2670 param = AudioMixer::RAMP_VOLUME; 2671 } 2672 2673 // compute volume for this track 2674 uint32_t vl, vr, va; 2675 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2676 vl = vr = va = 0; 2677 if (track->isPausing()) { 2678 track->setPaused(); 2679 } 2680 } else { 2681 2682 // read original volumes with volume control 2683 float typeVolume = mStreamTypes[track->streamType()].volume; 2684 float v = masterVolume * typeVolume; 2685 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2686 uint32_t vlr = proxy->getVolumeLR(); 2687 vl = vlr & 0xFFFF; 2688 vr = vlr >> 16; 2689 // track volumes come from shared memory, so can't be trusted and must be clamped 2690 if (vl > MAX_GAIN_INT) { 2691 ALOGV("Track left volume out of range: %04X", vl); 2692 vl = MAX_GAIN_INT; 2693 } 2694 if (vr > MAX_GAIN_INT) { 2695 ALOGV("Track right volume out of range: %04X", vr); 2696 vr = MAX_GAIN_INT; 2697 } 2698 // now apply the master volume and stream type volume 2699 vl = (uint32_t)(v * vl) << 12; 2700 vr = (uint32_t)(v * vr) << 12; 2701 // assuming master volume and stream type volume each go up to 1.0, 2702 // vl and vr are now in 8.24 format 2703 2704 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2705 // send level comes from shared memory and so may be corrupt 2706 if (sendLevel > MAX_GAIN_INT) { 2707 ALOGV("Track send level out of range: %04X", sendLevel); 2708 sendLevel = MAX_GAIN_INT; 2709 } 2710 va = (uint32_t)(v * sendLevel); 2711 } 2712 // Delegate volume control to effect in track effect chain if needed 2713 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2714 // Do not ramp volume if volume is controlled by effect 2715 param = AudioMixer::VOLUME; 2716 track->mHasVolumeController = true; 2717 } else { 2718 // force no volume ramp when volume controller was just disabled or removed 2719 // from effect chain to avoid volume spike 2720 if (track->mHasVolumeController) { 2721 param = AudioMixer::VOLUME; 2722 } 2723 track->mHasVolumeController = false; 2724 } 2725 2726 // Convert volumes from 8.24 to 4.12 format 2727 // This additional clamping is needed in case chain->setVolume_l() overshot 2728 vl = (vl + (1 << 11)) >> 12; 2729 if (vl > MAX_GAIN_INT) { 2730 vl = MAX_GAIN_INT; 2731 } 2732 vr = (vr + (1 << 11)) >> 12; 2733 if (vr > MAX_GAIN_INT) { 2734 vr = MAX_GAIN_INT; 2735 } 2736 2737 if (va > MAX_GAIN_INT) { 2738 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2739 } 2740 2741 // XXX: these things DON'T need to be done each time 2742 mAudioMixer->setBufferProvider(name, track); 2743 mAudioMixer->enable(name); 2744 2745 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2746 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2747 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2748 mAudioMixer->setParameter( 2749 name, 2750 AudioMixer::TRACK, 2751 AudioMixer::FORMAT, (void *)track->format()); 2752 mAudioMixer->setParameter( 2753 name, 2754 AudioMixer::TRACK, 2755 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2756 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2757 uint32_t maxSampleRate = mSampleRate * 2; 2758 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 2759 if (reqSampleRate == 0) { 2760 reqSampleRate = mSampleRate; 2761 } else if (reqSampleRate > maxSampleRate) { 2762 reqSampleRate = maxSampleRate; 2763 } 2764 mAudioMixer->setParameter( 2765 name, 2766 AudioMixer::RESAMPLE, 2767 AudioMixer::SAMPLE_RATE, 2768 (void *)reqSampleRate); 2769 mAudioMixer->setParameter( 2770 name, 2771 AudioMixer::TRACK, 2772 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2773 mAudioMixer->setParameter( 2774 name, 2775 AudioMixer::TRACK, 2776 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2777 2778 // reset retry count 2779 track->mRetryCount = kMaxTrackRetries; 2780 2781 // If one track is ready, set the mixer ready if: 2782 // - the mixer was not ready during previous round OR 2783 // - no other track is not ready 2784 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2785 mixerStatus != MIXER_TRACKS_ENABLED) { 2786 mixerStatus = MIXER_TRACKS_READY; 2787 } 2788 } else { 2789 // only implemented for normal tracks, not fast tracks 2790 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 2791 // we missed desiredFrames whatever the actual number of frames missing was 2792 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 2793 // FIXME also wake futex so that underrun is noticed more quickly 2794 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 2795 } 2796 // clear effect chain input buffer if an active track underruns to avoid sending 2797 // previous audio buffer again to effects 2798 chain = getEffectChain_l(track->sessionId()); 2799 if (chain != 0) { 2800 chain->clearInputBuffer(); 2801 } 2802 2803 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2804 cblk->server, this); 2805 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2806 track->isStopped() || track->isPaused()) { 2807 // We have consumed all the buffers of this track. 2808 // Remove it from the list of active tracks. 2809 // TODO: use actual buffer filling status instead of latency when available from 2810 // audio HAL 2811 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2812 size_t framesWritten = mBytesWritten / mFrameSize; 2813 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2814 if (track->isStopped()) { 2815 track->reset(); 2816 } 2817 tracksToRemove->add(track); 2818 } 2819 } else { 2820 track->mUnderrunCount++; 2821 // No buffers for this track. Give it a few chances to 2822 // fill a buffer, then remove it from active list. 2823 if (--(track->mRetryCount) <= 0) { 2824 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2825 tracksToRemove->add(track); 2826 // indicate to client process that the track was disabled because of underrun; 2827 // it will then automatically call start() when data is available 2828 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2829 // If one track is not ready, mark the mixer also not ready if: 2830 // - the mixer was ready during previous round OR 2831 // - no other track is ready 2832 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2833 mixerStatus != MIXER_TRACKS_READY) { 2834 mixerStatus = MIXER_TRACKS_ENABLED; 2835 } 2836 } 2837 mAudioMixer->disable(name); 2838 } 2839 2840 } // local variable scope to avoid goto warning 2841track_is_ready: ; 2842 2843 } 2844 2845 // Push the new FastMixer state if necessary 2846 bool pauseAudioWatchdog = false; 2847 if (didModify) { 2848 state->mFastTracksGen++; 2849 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2850 if (kUseFastMixer == FastMixer_Dynamic && 2851 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2852 state->mCommand = FastMixerState::COLD_IDLE; 2853 state->mColdFutexAddr = &mFastMixerFutex; 2854 state->mColdGen++; 2855 mFastMixerFutex = 0; 2856 if (kUseFastMixer == FastMixer_Dynamic) { 2857 mNormalSink = mOutputSink; 2858 } 2859 // If we go into cold idle, need to wait for acknowledgement 2860 // so that fast mixer stops doing I/O. 2861 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2862 pauseAudioWatchdog = true; 2863 } 2864 } 2865 if (sq != NULL) { 2866 sq->end(didModify); 2867 sq->push(block); 2868 } 2869#ifdef AUDIO_WATCHDOG 2870 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2871 mAudioWatchdog->pause(); 2872 } 2873#endif 2874 2875 // Now perform the deferred reset on fast tracks that have stopped 2876 while (resetMask != 0) { 2877 size_t i = __builtin_ctz(resetMask); 2878 ALOG_ASSERT(i < count); 2879 resetMask &= ~(1 << i); 2880 sp<Track> t = mActiveTracks[i].promote(); 2881 if (t == 0) { 2882 continue; 2883 } 2884 Track* track = t.get(); 2885 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2886 track->reset(); 2887 } 2888 2889 // remove all the tracks that need to be... 2890 count = tracksToRemove->size(); 2891 if (CC_UNLIKELY(count)) { 2892 for (size_t i=0 ; i<count ; i++) { 2893 const sp<Track>& track = tracksToRemove->itemAt(i); 2894 mActiveTracks.remove(track); 2895 if (track->mainBuffer() != mMixBuffer) { 2896 chain = getEffectChain_l(track->sessionId()); 2897 if (chain != 0) { 2898 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2899 track->sessionId()); 2900 chain->decActiveTrackCnt(); 2901 } 2902 } 2903 if (track->isTerminated()) { 2904 removeTrack_l(track); 2905 } 2906 } 2907 } 2908 2909 // mix buffer must be cleared if all tracks are connected to an 2910 // effect chain as in this case the mixer will not write to 2911 // mix buffer and track effects will accumulate into it 2912 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2913 (mixedTracks == 0 && fastTracks > 0)) { 2914 // FIXME as a performance optimization, should remember previous zero status 2915 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2916 } 2917 2918 // if any fast tracks, then status is ready 2919 mMixerStatusIgnoringFastTracks = mixerStatus; 2920 if (fastTracks > 0) { 2921 mixerStatus = MIXER_TRACKS_READY; 2922 } 2923 return mixerStatus; 2924} 2925 2926// getTrackName_l() must be called with ThreadBase::mLock held 2927int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2928{ 2929 return mAudioMixer->getTrackName(channelMask, sessionId); 2930} 2931 2932// deleteTrackName_l() must be called with ThreadBase::mLock held 2933void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2934{ 2935 ALOGV("remove track (%d) and delete from mixer", name); 2936 mAudioMixer->deleteTrackName(name); 2937} 2938 2939// checkForNewParameters_l() must be called with ThreadBase::mLock held 2940bool AudioFlinger::MixerThread::checkForNewParameters_l() 2941{ 2942 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2943 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2944 bool reconfig = false; 2945 2946 while (!mNewParameters.isEmpty()) { 2947 2948 if (mFastMixer != NULL) { 2949 FastMixerStateQueue *sq = mFastMixer->sq(); 2950 FastMixerState *state = sq->begin(); 2951 if (!(state->mCommand & FastMixerState::IDLE)) { 2952 previousCommand = state->mCommand; 2953 state->mCommand = FastMixerState::HOT_IDLE; 2954 sq->end(); 2955 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2956 } else { 2957 sq->end(false /*didModify*/); 2958 } 2959 } 2960 2961 status_t status = NO_ERROR; 2962 String8 keyValuePair = mNewParameters[0]; 2963 AudioParameter param = AudioParameter(keyValuePair); 2964 int value; 2965 2966 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2967 reconfig = true; 2968 } 2969 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2970 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2971 status = BAD_VALUE; 2972 } else { 2973 reconfig = true; 2974 } 2975 } 2976 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2977 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2978 status = BAD_VALUE; 2979 } else { 2980 reconfig = true; 2981 } 2982 } 2983 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2984 // do not accept frame count changes if tracks are open as the track buffer 2985 // size depends on frame count and correct behavior would not be guaranteed 2986 // if frame count is changed after track creation 2987 if (!mTracks.isEmpty()) { 2988 status = INVALID_OPERATION; 2989 } else { 2990 reconfig = true; 2991 } 2992 } 2993 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2994#ifdef ADD_BATTERY_DATA 2995 // when changing the audio output device, call addBatteryData to notify 2996 // the change 2997 if (mOutDevice != value) { 2998 uint32_t params = 0; 2999 // check whether speaker is on 3000 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3001 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3002 } 3003 3004 audio_devices_t deviceWithoutSpeaker 3005 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3006 // check if any other device (except speaker) is on 3007 if (value & deviceWithoutSpeaker ) { 3008 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3009 } 3010 3011 if (params != 0) { 3012 addBatteryData(params); 3013 } 3014 } 3015#endif 3016 3017 // forward device change to effects that have requested to be 3018 // aware of attached audio device. 3019 if (value != AUDIO_DEVICE_NONE) { 3020 mOutDevice = value; 3021 for (size_t i = 0; i < mEffectChains.size(); i++) { 3022 mEffectChains[i]->setDevice_l(mOutDevice); 3023 } 3024 } 3025 } 3026 3027 if (status == NO_ERROR) { 3028 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3029 keyValuePair.string()); 3030 if (!mStandby && status == INVALID_OPERATION) { 3031 mOutput->stream->common.standby(&mOutput->stream->common); 3032 mStandby = true; 3033 mBytesWritten = 0; 3034 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3035 keyValuePair.string()); 3036 } 3037 if (status == NO_ERROR && reconfig) { 3038 readOutputParameters(); 3039 delete mAudioMixer; 3040 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3041 for (size_t i = 0; i < mTracks.size() ; i++) { 3042 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3043 if (name < 0) { 3044 break; 3045 } 3046 mTracks[i]->mName = name; 3047 } 3048 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3049 } 3050 } 3051 3052 mNewParameters.removeAt(0); 3053 3054 mParamStatus = status; 3055 mParamCond.signal(); 3056 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3057 // already timed out waiting for the status and will never signal the condition. 3058 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3059 } 3060 3061 if (!(previousCommand & FastMixerState::IDLE)) { 3062 ALOG_ASSERT(mFastMixer != NULL); 3063 FastMixerStateQueue *sq = mFastMixer->sq(); 3064 FastMixerState *state = sq->begin(); 3065 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3066 state->mCommand = previousCommand; 3067 sq->end(); 3068 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3069 } 3070 3071 return reconfig; 3072} 3073 3074 3075void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3076{ 3077 const size_t SIZE = 256; 3078 char buffer[SIZE]; 3079 String8 result; 3080 3081 PlaybackThread::dumpInternals(fd, args); 3082 3083 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3084 result.append(buffer); 3085 write(fd, result.string(), result.size()); 3086 3087 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3088 const FastMixerDumpState copy(mFastMixerDumpState); 3089 copy.dump(fd); 3090 3091#ifdef STATE_QUEUE_DUMP 3092 // Similar for state queue 3093 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3094 observerCopy.dump(fd); 3095 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3096 mutatorCopy.dump(fd); 3097#endif 3098 3099#ifdef TEE_SINK 3100 // Write the tee output to a .wav file 3101 dumpTee(fd, mTeeSource, mId); 3102#endif 3103 3104#ifdef AUDIO_WATCHDOG 3105 if (mAudioWatchdog != 0) { 3106 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3107 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3108 wdCopy.dump(fd); 3109 } 3110#endif 3111} 3112 3113uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3114{ 3115 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3116} 3117 3118uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3119{ 3120 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3121} 3122 3123void AudioFlinger::MixerThread::cacheParameters_l() 3124{ 3125 PlaybackThread::cacheParameters_l(); 3126 3127 // FIXME: Relaxed timing because of a certain device that can't meet latency 3128 // Should be reduced to 2x after the vendor fixes the driver issue 3129 // increase threshold again due to low power audio mode. The way this warning 3130 // threshold is calculated and its usefulness should be reconsidered anyway. 3131 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3132} 3133 3134// ---------------------------------------------------------------------------- 3135 3136AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3137 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3138 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3139 // mLeftVolFloat, mRightVolFloat 3140{ 3141} 3142 3143AudioFlinger::DirectOutputThread::~DirectOutputThread() 3144{ 3145} 3146 3147AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3148 Vector< sp<Track> > *tracksToRemove 3149) 3150{ 3151 size_t count = mActiveTracks.size(); 3152 mixer_state mixerStatus = MIXER_IDLE; 3153 3154 // find out which tracks need to be processed 3155 for (size_t i = 0; i < count; i++) { 3156 sp<Track> t = mActiveTracks[i].promote(); 3157 // The track died recently 3158 if (t == 0) { 3159 continue; 3160 } 3161 3162 Track* const track = t.get(); 3163 audio_track_cblk_t* cblk = track->cblk(); 3164 3165 // The first time a track is added we wait 3166 // for all its buffers to be filled before processing it 3167 uint32_t minFrames; 3168 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3169 minFrames = mNormalFrameCount; 3170 } else { 3171 minFrames = 1; 3172 } 3173 if ((track->framesReady() >= minFrames) && track->isReady() && 3174 !track->isPaused() && !track->isTerminated()) 3175 { 3176 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3177 3178 if (track->mFillingUpStatus == Track::FS_FILLED) { 3179 track->mFillingUpStatus = Track::FS_ACTIVE; 3180 mLeftVolFloat = mRightVolFloat = 0; 3181 if (track->mState == TrackBase::RESUMING) { 3182 track->mState = TrackBase::ACTIVE; 3183 } 3184 } 3185 3186 // compute volume for this track 3187 float left, right; 3188 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3189 left = right = 0; 3190 if (track->isPausing()) { 3191 track->setPaused(); 3192 } 3193 } else { 3194 float typeVolume = mStreamTypes[track->streamType()].volume; 3195 float v = mMasterVolume * typeVolume; 3196 uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); 3197 float v_clamped = v * (vlr & 0xFFFF); 3198 if (v_clamped > MAX_GAIN) { 3199 v_clamped = MAX_GAIN; 3200 } 3201 left = v_clamped/MAX_GAIN; 3202 v_clamped = v * (vlr >> 16); 3203 if (v_clamped > MAX_GAIN) { 3204 v_clamped = MAX_GAIN; 3205 } 3206 right = v_clamped/MAX_GAIN; 3207 } 3208 // Only consider last track started for volume and mixer state control. 3209 // This is the last entry in mActiveTracks unless a track underruns. 3210 // As we only care about the transition phase between two tracks on a 3211 // direct output, it is not a problem to ignore the underrun case. 3212 if (i == (count - 1)) { 3213 if (left != mLeftVolFloat || right != mRightVolFloat) { 3214 mLeftVolFloat = left; 3215 mRightVolFloat = right; 3216 3217 // Convert volumes from float to 8.24 3218 uint32_t vl = (uint32_t)(left * (1 << 24)); 3219 uint32_t vr = (uint32_t)(right * (1 << 24)); 3220 3221 // Delegate volume control to effect in track effect chain if needed 3222 // only one effect chain can be present on DirectOutputThread, so if 3223 // there is one, the track is connected to it 3224 if (!mEffectChains.isEmpty()) { 3225 // Do not ramp volume if volume is controlled by effect 3226 mEffectChains[0]->setVolume_l(&vl, &vr); 3227 left = (float)vl / (1 << 24); 3228 right = (float)vr / (1 << 24); 3229 } 3230 mOutput->stream->set_volume(mOutput->stream, left, right); 3231 } 3232 3233 // reset retry count 3234 track->mRetryCount = kMaxTrackRetriesDirect; 3235 mActiveTrack = t; 3236 mixerStatus = MIXER_TRACKS_READY; 3237 } 3238 } else { 3239 // clear effect chain input buffer if the last active track started underruns 3240 // to avoid sending previous audio buffer again to effects 3241 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3242 mEffectChains[0]->clearInputBuffer(); 3243 } 3244 3245 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3246 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3247 track->isStopped() || track->isPaused()) { 3248 // We have consumed all the buffers of this track. 3249 // Remove it from the list of active tracks. 3250 // TODO: implement behavior for compressed audio 3251 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3252 size_t framesWritten = mBytesWritten / mFrameSize; 3253 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3254 if (track->isStopped()) { 3255 track->reset(); 3256 } 3257 tracksToRemove->add(track); 3258 } 3259 } else { 3260 // No buffers for this track. Give it a few chances to 3261 // fill a buffer, then remove it from active list. 3262 // Only consider last track started for mixer state control 3263 if (--(track->mRetryCount) <= 0) { 3264 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3265 tracksToRemove->add(track); 3266 } else if (i == (count -1)){ 3267 mixerStatus = MIXER_TRACKS_ENABLED; 3268 } 3269 } 3270 } 3271 } 3272 3273 // remove all the tracks that need to be... 3274 count = tracksToRemove->size(); 3275 if (CC_UNLIKELY(count)) { 3276 for (size_t i = 0 ; i < count ; i++) { 3277 const sp<Track>& track = tracksToRemove->itemAt(i); 3278 mActiveTracks.remove(track); 3279 if (!mEffectChains.isEmpty()) { 3280 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3281 track->sessionId()); 3282 mEffectChains[0]->decActiveTrackCnt(); 3283 } 3284 if (track->isTerminated()) { 3285 removeTrack_l(track); 3286 } 3287 } 3288 } 3289 3290 return mixerStatus; 3291} 3292 3293void AudioFlinger::DirectOutputThread::threadLoop_mix() 3294{ 3295 AudioBufferProvider::Buffer buffer; 3296 size_t frameCount = mFrameCount; 3297 int8_t *curBuf = (int8_t *)mMixBuffer; 3298 // output audio to hardware 3299 while (frameCount) { 3300 buffer.frameCount = frameCount; 3301 mActiveTrack->getNextBuffer(&buffer); 3302 if (CC_UNLIKELY(buffer.raw == NULL)) { 3303 memset(curBuf, 0, frameCount * mFrameSize); 3304 break; 3305 } 3306 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3307 frameCount -= buffer.frameCount; 3308 curBuf += buffer.frameCount * mFrameSize; 3309 mActiveTrack->releaseBuffer(&buffer); 3310 } 3311 sleepTime = 0; 3312 standbyTime = systemTime() + standbyDelay; 3313 mActiveTrack.clear(); 3314 3315} 3316 3317void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3318{ 3319 if (sleepTime == 0) { 3320 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3321 sleepTime = activeSleepTime; 3322 } else { 3323 sleepTime = idleSleepTime; 3324 } 3325 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3326 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3327 sleepTime = 0; 3328 } 3329} 3330 3331// getTrackName_l() must be called with ThreadBase::mLock held 3332int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3333 int sessionId) 3334{ 3335 return 0; 3336} 3337 3338// deleteTrackName_l() must be called with ThreadBase::mLock held 3339void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3340{ 3341} 3342 3343// checkForNewParameters_l() must be called with ThreadBase::mLock held 3344bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3345{ 3346 bool reconfig = false; 3347 3348 while (!mNewParameters.isEmpty()) { 3349 status_t status = NO_ERROR; 3350 String8 keyValuePair = mNewParameters[0]; 3351 AudioParameter param = AudioParameter(keyValuePair); 3352 int value; 3353 3354 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3355 // do not accept frame count changes if tracks are open as the track buffer 3356 // size depends on frame count and correct behavior would not be garantied 3357 // if frame count is changed after track creation 3358 if (!mTracks.isEmpty()) { 3359 status = INVALID_OPERATION; 3360 } else { 3361 reconfig = true; 3362 } 3363 } 3364 if (status == NO_ERROR) { 3365 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3366 keyValuePair.string()); 3367 if (!mStandby && status == INVALID_OPERATION) { 3368 mOutput->stream->common.standby(&mOutput->stream->common); 3369 mStandby = true; 3370 mBytesWritten = 0; 3371 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3372 keyValuePair.string()); 3373 } 3374 if (status == NO_ERROR && reconfig) { 3375 readOutputParameters(); 3376 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3377 } 3378 } 3379 3380 mNewParameters.removeAt(0); 3381 3382 mParamStatus = status; 3383 mParamCond.signal(); 3384 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3385 // already timed out waiting for the status and will never signal the condition. 3386 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3387 } 3388 return reconfig; 3389} 3390 3391uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3392{ 3393 uint32_t time; 3394 if (audio_is_linear_pcm(mFormat)) { 3395 time = PlaybackThread::activeSleepTimeUs(); 3396 } else { 3397 time = 10000; 3398 } 3399 return time; 3400} 3401 3402uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3403{ 3404 uint32_t time; 3405 if (audio_is_linear_pcm(mFormat)) { 3406 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3407 } else { 3408 time = 10000; 3409 } 3410 return time; 3411} 3412 3413uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3414{ 3415 uint32_t time; 3416 if (audio_is_linear_pcm(mFormat)) { 3417 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3418 } else { 3419 time = 10000; 3420 } 3421 return time; 3422} 3423 3424void AudioFlinger::DirectOutputThread::cacheParameters_l() 3425{ 3426 PlaybackThread::cacheParameters_l(); 3427 3428 // use shorter standby delay as on normal output to release 3429 // hardware resources as soon as possible 3430 standbyDelay = microseconds(activeSleepTime*2); 3431} 3432 3433// ---------------------------------------------------------------------------- 3434 3435AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3436 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3437 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3438 DUPLICATING), 3439 mWaitTimeMs(UINT_MAX) 3440{ 3441 addOutputTrack(mainThread); 3442} 3443 3444AudioFlinger::DuplicatingThread::~DuplicatingThread() 3445{ 3446 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3447 mOutputTracks[i]->destroy(); 3448 } 3449} 3450 3451void AudioFlinger::DuplicatingThread::threadLoop_mix() 3452{ 3453 // mix buffers... 3454 if (outputsReady(outputTracks)) { 3455 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3456 } else { 3457 memset(mMixBuffer, 0, mixBufferSize); 3458 } 3459 sleepTime = 0; 3460 writeFrames = mNormalFrameCount; 3461 standbyTime = systemTime() + standbyDelay; 3462} 3463 3464void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3465{ 3466 if (sleepTime == 0) { 3467 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3468 sleepTime = activeSleepTime; 3469 } else { 3470 sleepTime = idleSleepTime; 3471 } 3472 } else if (mBytesWritten != 0) { 3473 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3474 writeFrames = mNormalFrameCount; 3475 memset(mMixBuffer, 0, mixBufferSize); 3476 } else { 3477 // flush remaining overflow buffers in output tracks 3478 writeFrames = 0; 3479 } 3480 sleepTime = 0; 3481 } 3482} 3483 3484void AudioFlinger::DuplicatingThread::threadLoop_write() 3485{ 3486 for (size_t i = 0; i < outputTracks.size(); i++) { 3487 outputTracks[i]->write(mMixBuffer, writeFrames); 3488 } 3489 mBytesWritten += mixBufferSize; 3490} 3491 3492void AudioFlinger::DuplicatingThread::threadLoop_standby() 3493{ 3494 // DuplicatingThread implements standby by stopping all tracks 3495 for (size_t i = 0; i < outputTracks.size(); i++) { 3496 outputTracks[i]->stop(); 3497 } 3498} 3499 3500void AudioFlinger::DuplicatingThread::saveOutputTracks() 3501{ 3502 outputTracks = mOutputTracks; 3503} 3504 3505void AudioFlinger::DuplicatingThread::clearOutputTracks() 3506{ 3507 outputTracks.clear(); 3508} 3509 3510void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3511{ 3512 Mutex::Autolock _l(mLock); 3513 // FIXME explain this formula 3514 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3515 OutputTrack *outputTrack = new OutputTrack(thread, 3516 this, 3517 mSampleRate, 3518 mFormat, 3519 mChannelMask, 3520 frameCount); 3521 if (outputTrack->cblk() != NULL) { 3522 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3523 mOutputTracks.add(outputTrack); 3524 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3525 updateWaitTime_l(); 3526 } 3527} 3528 3529void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3530{ 3531 Mutex::Autolock _l(mLock); 3532 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3533 if (mOutputTracks[i]->thread() == thread) { 3534 mOutputTracks[i]->destroy(); 3535 mOutputTracks.removeAt(i); 3536 updateWaitTime_l(); 3537 return; 3538 } 3539 } 3540 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3541} 3542 3543// caller must hold mLock 3544void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3545{ 3546 mWaitTimeMs = UINT_MAX; 3547 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3548 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3549 if (strong != 0) { 3550 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3551 if (waitTimeMs < mWaitTimeMs) { 3552 mWaitTimeMs = waitTimeMs; 3553 } 3554 } 3555 } 3556} 3557 3558 3559bool AudioFlinger::DuplicatingThread::outputsReady( 3560 const SortedVector< sp<OutputTrack> > &outputTracks) 3561{ 3562 for (size_t i = 0; i < outputTracks.size(); i++) { 3563 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3564 if (thread == 0) { 3565 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3566 outputTracks[i].get()); 3567 return false; 3568 } 3569 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3570 // see note at standby() declaration 3571 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3572 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3573 thread.get()); 3574 return false; 3575 } 3576 } 3577 return true; 3578} 3579 3580uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3581{ 3582 return (mWaitTimeMs * 1000) / 2; 3583} 3584 3585void AudioFlinger::DuplicatingThread::cacheParameters_l() 3586{ 3587 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3588 updateWaitTime_l(); 3589 3590 MixerThread::cacheParameters_l(); 3591} 3592 3593// ---------------------------------------------------------------------------- 3594// Record 3595// ---------------------------------------------------------------------------- 3596 3597AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3598 AudioStreamIn *input, 3599 uint32_t sampleRate, 3600 audio_channel_mask_t channelMask, 3601 audio_io_handle_t id, 3602 audio_devices_t outDevice, 3603 audio_devices_t inDevice 3604#ifdef TEE_SINK 3605 , const sp<NBAIO_Sink>& teeSink 3606#endif 3607 ) : 3608 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3609 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3610 // mRsmpInIndex and mInputBytes set by readInputParameters() 3611 mReqChannelCount(popcount(channelMask)), 3612 mReqSampleRate(sampleRate) 3613 // mBytesRead is only meaningful while active, and so is cleared in start() 3614 // (but might be better to also clear here for dump?) 3615#ifdef TEE_SINK 3616 , mTeeSink(teeSink) 3617#endif 3618{ 3619 snprintf(mName, kNameLength, "AudioIn_%X", id); 3620 3621 readInputParameters(); 3622 3623} 3624 3625 3626AudioFlinger::RecordThread::~RecordThread() 3627{ 3628 delete[] mRsmpInBuffer; 3629 delete mResampler; 3630 delete[] mRsmpOutBuffer; 3631} 3632 3633void AudioFlinger::RecordThread::onFirstRef() 3634{ 3635 run(mName, PRIORITY_URGENT_AUDIO); 3636} 3637 3638status_t AudioFlinger::RecordThread::readyToRun() 3639{ 3640 status_t status = initCheck(); 3641 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3642 return status; 3643} 3644 3645bool AudioFlinger::RecordThread::threadLoop() 3646{ 3647 AudioBufferProvider::Buffer buffer; 3648 sp<RecordTrack> activeTrack; 3649 Vector< sp<EffectChain> > effectChains; 3650 3651 nsecs_t lastWarning = 0; 3652 3653 inputStandBy(); 3654 acquireWakeLock(); 3655 3656 // used to verify we've read at least once before evaluating how many bytes were read 3657 bool readOnce = false; 3658 3659 // start recording 3660 while (!exitPending()) { 3661 3662 processConfigEvents(); 3663 3664 { // scope for mLock 3665 Mutex::Autolock _l(mLock); 3666 checkForNewParameters_l(); 3667 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3668 standby(); 3669 3670 if (exitPending()) { 3671 break; 3672 } 3673 3674 releaseWakeLock_l(); 3675 ALOGV("RecordThread: loop stopping"); 3676 // go to sleep 3677 mWaitWorkCV.wait(mLock); 3678 ALOGV("RecordThread: loop starting"); 3679 acquireWakeLock_l(); 3680 continue; 3681 } 3682 if (mActiveTrack != 0) { 3683 if (mActiveTrack->mState == TrackBase::PAUSING) { 3684 standby(); 3685 mActiveTrack.clear(); 3686 mStartStopCond.broadcast(); 3687 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3688 if (mReqChannelCount != mActiveTrack->channelCount()) { 3689 mActiveTrack.clear(); 3690 mStartStopCond.broadcast(); 3691 } else if (readOnce) { 3692 // record start succeeds only if first read from audio input 3693 // succeeds 3694 if (mBytesRead >= 0) { 3695 mActiveTrack->mState = TrackBase::ACTIVE; 3696 } else { 3697 mActiveTrack.clear(); 3698 } 3699 mStartStopCond.broadcast(); 3700 } 3701 mStandby = false; 3702 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3703 removeTrack_l(mActiveTrack); 3704 mActiveTrack.clear(); 3705 } 3706 } 3707 lockEffectChains_l(effectChains); 3708 } 3709 3710 if (mActiveTrack != 0) { 3711 if (mActiveTrack->mState != TrackBase::ACTIVE && 3712 mActiveTrack->mState != TrackBase::RESUMING) { 3713 unlockEffectChains(effectChains); 3714 usleep(kRecordThreadSleepUs); 3715 continue; 3716 } 3717 for (size_t i = 0; i < effectChains.size(); i ++) { 3718 effectChains[i]->process_l(); 3719 } 3720 3721 buffer.frameCount = mFrameCount; 3722 status_t status = mActiveTrack->getNextBuffer(&buffer); 3723 if (CC_LIKELY(status == NO_ERROR)) { 3724 readOnce = true; 3725 size_t framesOut = buffer.frameCount; 3726 if (mResampler == NULL) { 3727 // no resampling 3728 while (framesOut) { 3729 size_t framesIn = mFrameCount - mRsmpInIndex; 3730 if (framesIn) { 3731 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3732 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3733 mActiveTrack->mFrameSize; 3734 if (framesIn > framesOut) 3735 framesIn = framesOut; 3736 mRsmpInIndex += framesIn; 3737 framesOut -= framesIn; 3738 if (mChannelCount == mReqChannelCount || 3739 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3740 memcpy(dst, src, framesIn * mFrameSize); 3741 } else { 3742 if (mChannelCount == 1) { 3743 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3744 (int16_t *)src, framesIn); 3745 } else { 3746 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3747 (int16_t *)src, framesIn); 3748 } 3749 } 3750 } 3751 if (framesOut && mFrameCount == mRsmpInIndex) { 3752 void *readInto; 3753 if (framesOut == mFrameCount && 3754 (mChannelCount == mReqChannelCount || 3755 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3756 readInto = buffer.raw; 3757 framesOut = 0; 3758 } else { 3759 readInto = mRsmpInBuffer; 3760 mRsmpInIndex = 0; 3761 } 3762 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3763 mInputBytes); 3764 if (mBytesRead <= 0) { 3765 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3766 { 3767 ALOGE("Error reading audio input"); 3768 // Force input into standby so that it tries to 3769 // recover at next read attempt 3770 inputStandBy(); 3771 usleep(kRecordThreadSleepUs); 3772 } 3773 mRsmpInIndex = mFrameCount; 3774 framesOut = 0; 3775 buffer.frameCount = 0; 3776 } 3777#ifdef TEE_SINK 3778 else if (mTeeSink != 0) { 3779 (void) mTeeSink->write(readInto, 3780 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3781 } 3782#endif 3783 } 3784 } 3785 } else { 3786 // resampling 3787 3788 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3789 // alter output frame count as if we were expecting stereo samples 3790 if (mChannelCount == 1 && mReqChannelCount == 1) { 3791 framesOut >>= 1; 3792 } 3793 mResampler->resample(mRsmpOutBuffer, framesOut, 3794 this /* AudioBufferProvider* */); 3795 // ditherAndClamp() works as long as all buffers returned by 3796 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3797 if (mChannelCount == 2 && mReqChannelCount == 1) { 3798 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3799 // the resampler always outputs stereo samples: 3800 // do post stereo to mono conversion 3801 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3802 framesOut); 3803 } else { 3804 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3805 } 3806 3807 } 3808 if (mFramestoDrop == 0) { 3809 mActiveTrack->releaseBuffer(&buffer); 3810 } else { 3811 if (mFramestoDrop > 0) { 3812 mFramestoDrop -= buffer.frameCount; 3813 if (mFramestoDrop <= 0) { 3814 clearSyncStartEvent(); 3815 } 3816 } else { 3817 mFramestoDrop += buffer.frameCount; 3818 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3819 mSyncStartEvent->isCancelled()) { 3820 ALOGW("Synced record %s, session %d, trigger session %d", 3821 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3822 mActiveTrack->sessionId(), 3823 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3824 clearSyncStartEvent(); 3825 } 3826 } 3827 } 3828 mActiveTrack->clearOverflow(); 3829 } 3830 // client isn't retrieving buffers fast enough 3831 else { 3832 if (!mActiveTrack->setOverflow()) { 3833 nsecs_t now = systemTime(); 3834 if ((now - lastWarning) > kWarningThrottleNs) { 3835 ALOGW("RecordThread: buffer overflow"); 3836 lastWarning = now; 3837 } 3838 } 3839 // Release the processor for a while before asking for a new buffer. 3840 // This will give the application more chance to read from the buffer and 3841 // clear the overflow. 3842 usleep(kRecordThreadSleepUs); 3843 } 3844 } 3845 // enable changes in effect chain 3846 unlockEffectChains(effectChains); 3847 effectChains.clear(); 3848 } 3849 3850 standby(); 3851 3852 { 3853 Mutex::Autolock _l(mLock); 3854 mActiveTrack.clear(); 3855 mStartStopCond.broadcast(); 3856 } 3857 3858 releaseWakeLock(); 3859 3860 ALOGV("RecordThread %p exiting", this); 3861 return false; 3862} 3863 3864void AudioFlinger::RecordThread::standby() 3865{ 3866 if (!mStandby) { 3867 inputStandBy(); 3868 mStandby = true; 3869 } 3870} 3871 3872void AudioFlinger::RecordThread::inputStandBy() 3873{ 3874 mInput->stream->common.standby(&mInput->stream->common); 3875} 3876 3877sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3878 const sp<AudioFlinger::Client>& client, 3879 uint32_t sampleRate, 3880 audio_format_t format, 3881 audio_channel_mask_t channelMask, 3882 size_t frameCount, 3883 int sessionId, 3884 IAudioFlinger::track_flags_t flags, 3885 pid_t tid, 3886 status_t *status) 3887{ 3888 sp<RecordTrack> track; 3889 status_t lStatus; 3890 3891 lStatus = initCheck(); 3892 if (lStatus != NO_ERROR) { 3893 ALOGE("Audio driver not initialized."); 3894 goto Exit; 3895 } 3896 3897 // FIXME use flags and tid similar to createTrack_l() 3898 3899 { // scope for mLock 3900 Mutex::Autolock _l(mLock); 3901 3902 track = new RecordTrack(this, client, sampleRate, 3903 format, channelMask, frameCount, sessionId); 3904 3905 if (track->getCblk() == 0) { 3906 lStatus = NO_MEMORY; 3907 goto Exit; 3908 } 3909 mTracks.add(track); 3910 3911 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3912 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3913 mAudioFlinger->btNrecIsOff(); 3914 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3915 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3916 } 3917 lStatus = NO_ERROR; 3918 3919Exit: 3920 if (status) { 3921 *status = lStatus; 3922 } 3923 return track; 3924} 3925 3926status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3927 AudioSystem::sync_event_t event, 3928 int triggerSession) 3929{ 3930 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3931 sp<ThreadBase> strongMe = this; 3932 status_t status = NO_ERROR; 3933 3934 if (event == AudioSystem::SYNC_EVENT_NONE) { 3935 clearSyncStartEvent(); 3936 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3937 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3938 triggerSession, 3939 recordTrack->sessionId(), 3940 syncStartEventCallback, 3941 this); 3942 // Sync event can be cancelled by the trigger session if the track is not in a 3943 // compatible state in which case we start record immediately 3944 if (mSyncStartEvent->isCancelled()) { 3945 clearSyncStartEvent(); 3946 } else { 3947 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3948 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3949 } 3950 } 3951 3952 { 3953 AutoMutex lock(mLock); 3954 if (mActiveTrack != 0) { 3955 if (recordTrack != mActiveTrack.get()) { 3956 status = -EBUSY; 3957 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3958 mActiveTrack->mState = TrackBase::ACTIVE; 3959 } 3960 return status; 3961 } 3962 3963 recordTrack->mState = TrackBase::IDLE; 3964 mActiveTrack = recordTrack; 3965 mLock.unlock(); 3966 status_t status = AudioSystem::startInput(mId); 3967 mLock.lock(); 3968 if (status != NO_ERROR) { 3969 mActiveTrack.clear(); 3970 clearSyncStartEvent(); 3971 return status; 3972 } 3973 mRsmpInIndex = mFrameCount; 3974 mBytesRead = 0; 3975 if (mResampler != NULL) { 3976 mResampler->reset(); 3977 } 3978 mActiveTrack->mState = TrackBase::RESUMING; 3979 // signal thread to start 3980 ALOGV("Signal record thread"); 3981 mWaitWorkCV.broadcast(); 3982 // do not wait for mStartStopCond if exiting 3983 if (exitPending()) { 3984 mActiveTrack.clear(); 3985 status = INVALID_OPERATION; 3986 goto startError; 3987 } 3988 mStartStopCond.wait(mLock); 3989 if (mActiveTrack == 0) { 3990 ALOGV("Record failed to start"); 3991 status = BAD_VALUE; 3992 goto startError; 3993 } 3994 ALOGV("Record started OK"); 3995 return status; 3996 } 3997 3998startError: 3999 AudioSystem::stopInput(mId); 4000 clearSyncStartEvent(); 4001 return status; 4002} 4003 4004void AudioFlinger::RecordThread::clearSyncStartEvent() 4005{ 4006 if (mSyncStartEvent != 0) { 4007 mSyncStartEvent->cancel(); 4008 } 4009 mSyncStartEvent.clear(); 4010 mFramestoDrop = 0; 4011} 4012 4013void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4014{ 4015 sp<SyncEvent> strongEvent = event.promote(); 4016 4017 if (strongEvent != 0) { 4018 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4019 me->handleSyncStartEvent(strongEvent); 4020 } 4021} 4022 4023void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4024{ 4025 if (event == mSyncStartEvent) { 4026 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4027 // from audio HAL 4028 mFramestoDrop = mFrameCount * 2; 4029 } 4030} 4031 4032bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4033 ALOGV("RecordThread::stop"); 4034 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4035 return false; 4036 } 4037 recordTrack->mState = TrackBase::PAUSING; 4038 // do not wait for mStartStopCond if exiting 4039 if (exitPending()) { 4040 return true; 4041 } 4042 mStartStopCond.wait(mLock); 4043 // if we have been restarted, recordTrack == mActiveTrack.get() here 4044 if (exitPending() || recordTrack != mActiveTrack.get()) { 4045 ALOGV("Record stopped OK"); 4046 return true; 4047 } 4048 return false; 4049} 4050 4051bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4052{ 4053 return false; 4054} 4055 4056status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4057{ 4058#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4059 if (!isValidSyncEvent(event)) { 4060 return BAD_VALUE; 4061 } 4062 4063 int eventSession = event->triggerSession(); 4064 status_t ret = NAME_NOT_FOUND; 4065 4066 Mutex::Autolock _l(mLock); 4067 4068 for (size_t i = 0; i < mTracks.size(); i++) { 4069 sp<RecordTrack> track = mTracks[i]; 4070 if (eventSession == track->sessionId()) { 4071 (void) track->setSyncEvent(event); 4072 ret = NO_ERROR; 4073 } 4074 } 4075 return ret; 4076#else 4077 return BAD_VALUE; 4078#endif 4079} 4080 4081// destroyTrack_l() must be called with ThreadBase::mLock held 4082void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4083{ 4084 track->mState = TrackBase::TERMINATED; 4085 // active tracks are removed by threadLoop() 4086 if (mActiveTrack != track) { 4087 removeTrack_l(track); 4088 } 4089} 4090 4091void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4092{ 4093 mTracks.remove(track); 4094 // need anything related to effects here? 4095} 4096 4097void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4098{ 4099 dumpInternals(fd, args); 4100 dumpTracks(fd, args); 4101 dumpEffectChains(fd, args); 4102} 4103 4104void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4105{ 4106 const size_t SIZE = 256; 4107 char buffer[SIZE]; 4108 String8 result; 4109 4110 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4111 result.append(buffer); 4112 4113 if (mActiveTrack != 0) { 4114 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4115 result.append(buffer); 4116 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4117 result.append(buffer); 4118 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4119 result.append(buffer); 4120 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4121 result.append(buffer); 4122 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4123 result.append(buffer); 4124 } else { 4125 result.append("No active record client\n"); 4126 } 4127 4128 write(fd, result.string(), result.size()); 4129 4130 dumpBase(fd, args); 4131} 4132 4133void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4134{ 4135 const size_t SIZE = 256; 4136 char buffer[SIZE]; 4137 String8 result; 4138 4139 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4140 result.append(buffer); 4141 RecordTrack::appendDumpHeader(result); 4142 for (size_t i = 0; i < mTracks.size(); ++i) { 4143 sp<RecordTrack> track = mTracks[i]; 4144 if (track != 0) { 4145 track->dump(buffer, SIZE); 4146 result.append(buffer); 4147 } 4148 } 4149 4150 if (mActiveTrack != 0) { 4151 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4152 result.append(buffer); 4153 RecordTrack::appendDumpHeader(result); 4154 mActiveTrack->dump(buffer, SIZE); 4155 result.append(buffer); 4156 4157 } 4158 write(fd, result.string(), result.size()); 4159} 4160 4161// AudioBufferProvider interface 4162status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4163{ 4164 size_t framesReq = buffer->frameCount; 4165 size_t framesReady = mFrameCount - mRsmpInIndex; 4166 int channelCount; 4167 4168 if (framesReady == 0) { 4169 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4170 if (mBytesRead <= 0) { 4171 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4172 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4173 // Force input into standby so that it tries to 4174 // recover at next read attempt 4175 inputStandBy(); 4176 usleep(kRecordThreadSleepUs); 4177 } 4178 buffer->raw = NULL; 4179 buffer->frameCount = 0; 4180 return NOT_ENOUGH_DATA; 4181 } 4182 mRsmpInIndex = 0; 4183 framesReady = mFrameCount; 4184 } 4185 4186 if (framesReq > framesReady) { 4187 framesReq = framesReady; 4188 } 4189 4190 if (mChannelCount == 1 && mReqChannelCount == 2) { 4191 channelCount = 1; 4192 } else { 4193 channelCount = 2; 4194 } 4195 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4196 buffer->frameCount = framesReq; 4197 return NO_ERROR; 4198} 4199 4200// AudioBufferProvider interface 4201void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4202{ 4203 mRsmpInIndex += buffer->frameCount; 4204 buffer->frameCount = 0; 4205} 4206 4207bool AudioFlinger::RecordThread::checkForNewParameters_l() 4208{ 4209 bool reconfig = false; 4210 4211 while (!mNewParameters.isEmpty()) { 4212 status_t status = NO_ERROR; 4213 String8 keyValuePair = mNewParameters[0]; 4214 AudioParameter param = AudioParameter(keyValuePair); 4215 int value; 4216 audio_format_t reqFormat = mFormat; 4217 uint32_t reqSamplingRate = mReqSampleRate; 4218 uint32_t reqChannelCount = mReqChannelCount; 4219 4220 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4221 reqSamplingRate = value; 4222 reconfig = true; 4223 } 4224 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4225 reqFormat = (audio_format_t) value; 4226 reconfig = true; 4227 } 4228 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4229 reqChannelCount = popcount(value); 4230 reconfig = true; 4231 } 4232 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4233 // do not accept frame count changes if tracks are open as the track buffer 4234 // size depends on frame count and correct behavior would not be guaranteed 4235 // if frame count is changed after track creation 4236 if (mActiveTrack != 0) { 4237 status = INVALID_OPERATION; 4238 } else { 4239 reconfig = true; 4240 } 4241 } 4242 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4243 // forward device change to effects that have requested to be 4244 // aware of attached audio device. 4245 for (size_t i = 0; i < mEffectChains.size(); i++) { 4246 mEffectChains[i]->setDevice_l(value); 4247 } 4248 4249 // store input device and output device but do not forward output device to audio HAL. 4250 // Note that status is ignored by the caller for output device 4251 // (see AudioFlinger::setParameters() 4252 if (audio_is_output_devices(value)) { 4253 mOutDevice = value; 4254 status = BAD_VALUE; 4255 } else { 4256 mInDevice = value; 4257 // disable AEC and NS if the device is a BT SCO headset supporting those 4258 // pre processings 4259 if (mTracks.size() > 0) { 4260 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4261 mAudioFlinger->btNrecIsOff(); 4262 for (size_t i = 0; i < mTracks.size(); i++) { 4263 sp<RecordTrack> track = mTracks[i]; 4264 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4265 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4266 } 4267 } 4268 } 4269 } 4270 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4271 mAudioSource != (audio_source_t)value) { 4272 // forward device change to effects that have requested to be 4273 // aware of attached audio device. 4274 for (size_t i = 0; i < mEffectChains.size(); i++) { 4275 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4276 } 4277 mAudioSource = (audio_source_t)value; 4278 } 4279 if (status == NO_ERROR) { 4280 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4281 keyValuePair.string()); 4282 if (status == INVALID_OPERATION) { 4283 inputStandBy(); 4284 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4285 keyValuePair.string()); 4286 } 4287 if (reconfig) { 4288 if (status == BAD_VALUE && 4289 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4290 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4291 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4292 <= (2 * reqSamplingRate)) && 4293 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4294 <= FCC_2 && 4295 (reqChannelCount <= FCC_2)) { 4296 status = NO_ERROR; 4297 } 4298 if (status == NO_ERROR) { 4299 readInputParameters(); 4300 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4301 } 4302 } 4303 } 4304 4305 mNewParameters.removeAt(0); 4306 4307 mParamStatus = status; 4308 mParamCond.signal(); 4309 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4310 // already timed out waiting for the status and will never signal the condition. 4311 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4312 } 4313 return reconfig; 4314} 4315 4316String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4317{ 4318 char *s; 4319 String8 out_s8 = String8(); 4320 4321 Mutex::Autolock _l(mLock); 4322 if (initCheck() != NO_ERROR) { 4323 return out_s8; 4324 } 4325 4326 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4327 out_s8 = String8(s); 4328 free(s); 4329 return out_s8; 4330} 4331 4332void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4333 AudioSystem::OutputDescriptor desc; 4334 void *param2 = NULL; 4335 4336 switch (event) { 4337 case AudioSystem::INPUT_OPENED: 4338 case AudioSystem::INPUT_CONFIG_CHANGED: 4339 desc.channels = mChannelMask; 4340 desc.samplingRate = mSampleRate; 4341 desc.format = mFormat; 4342 desc.frameCount = mFrameCount; 4343 desc.latency = 0; 4344 param2 = &desc; 4345 break; 4346 4347 case AudioSystem::INPUT_CLOSED: 4348 default: 4349 break; 4350 } 4351 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4352} 4353 4354void AudioFlinger::RecordThread::readInputParameters() 4355{ 4356 delete mRsmpInBuffer; 4357 // mRsmpInBuffer is always assigned a new[] below 4358 delete mRsmpOutBuffer; 4359 mRsmpOutBuffer = NULL; 4360 delete mResampler; 4361 mResampler = NULL; 4362 4363 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4364 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4365 mChannelCount = (uint16_t)popcount(mChannelMask); 4366 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4367 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4368 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4369 mFrameCount = mInputBytes / mFrameSize; 4370 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4371 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4372 4373 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4374 { 4375 int channelCount; 4376 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4377 // stereo to mono post process as the resampler always outputs stereo. 4378 if (mChannelCount == 1 && mReqChannelCount == 2) { 4379 channelCount = 1; 4380 } else { 4381 channelCount = 2; 4382 } 4383 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4384 mResampler->setSampleRate(mSampleRate); 4385 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4386 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4387 4388 // optmization: if mono to mono, alter input frame count as if we were inputing 4389 // stereo samples 4390 if (mChannelCount == 1 && mReqChannelCount == 1) { 4391 mFrameCount >>= 1; 4392 } 4393 4394 } 4395 mRsmpInIndex = mFrameCount; 4396} 4397 4398unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4399{ 4400 Mutex::Autolock _l(mLock); 4401 if (initCheck() != NO_ERROR) { 4402 return 0; 4403 } 4404 4405 return mInput->stream->get_input_frames_lost(mInput->stream); 4406} 4407 4408uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4409{ 4410 Mutex::Autolock _l(mLock); 4411 uint32_t result = 0; 4412 if (getEffectChain_l(sessionId) != 0) { 4413 result = EFFECT_SESSION; 4414 } 4415 4416 for (size_t i = 0; i < mTracks.size(); ++i) { 4417 if (sessionId == mTracks[i]->sessionId()) { 4418 result |= TRACK_SESSION; 4419 break; 4420 } 4421 } 4422 4423 return result; 4424} 4425 4426KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4427{ 4428 KeyedVector<int, bool> ids; 4429 Mutex::Autolock _l(mLock); 4430 for (size_t j = 0; j < mTracks.size(); ++j) { 4431 sp<RecordThread::RecordTrack> track = mTracks[j]; 4432 int sessionId = track->sessionId(); 4433 if (ids.indexOfKey(sessionId) < 0) { 4434 ids.add(sessionId, true); 4435 } 4436 } 4437 return ids; 4438} 4439 4440AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4441{ 4442 Mutex::Autolock _l(mLock); 4443 AudioStreamIn *input = mInput; 4444 mInput = NULL; 4445 return input; 4446} 4447 4448// this method must always be called either with ThreadBase mLock held or inside the thread loop 4449audio_stream_t* AudioFlinger::RecordThread::stream() const 4450{ 4451 if (mInput == NULL) { 4452 return NULL; 4453 } 4454 return &mInput->stream->common; 4455} 4456 4457status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4458{ 4459 // only one chain per input thread 4460 if (mEffectChains.size() != 0) { 4461 return INVALID_OPERATION; 4462 } 4463 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4464 4465 chain->setInBuffer(NULL); 4466 chain->setOutBuffer(NULL); 4467 4468 checkSuspendOnAddEffectChain_l(chain); 4469 4470 mEffectChains.add(chain); 4471 4472 return NO_ERROR; 4473} 4474 4475size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4476{ 4477 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4478 ALOGW_IF(mEffectChains.size() != 1, 4479 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4480 chain.get(), mEffectChains.size(), this); 4481 if (mEffectChains.size() == 1) { 4482 mEffectChains.removeAt(0); 4483 } 4484 return 0; 4485} 4486 4487}; // namespace android 4488