Threads.cpp revision bf04b5860182d8f4130dcb5d6d88ee68a58c99cd
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 <math.h> 24#include <fcntl.h> 25#include <sys/stat.h> 26#include <cutils/properties.h> 27#include <cutils/compiler.h> 28#include <utils/Log.h> 29#include <utils/Trace.h> 30 31#include <private/media/AudioTrackShared.h> 32#include <hardware/audio.h> 33#include <audio_effects/effect_ns.h> 34#include <audio_effects/effect_aec.h> 35#include <audio_utils/primitives.h> 36 37// NBAIO implementations 38#include <media/nbaio/AudioStreamOutSink.h> 39#include <media/nbaio/MonoPipe.h> 40#include <media/nbaio/MonoPipeReader.h> 41#include <media/nbaio/Pipe.h> 42#include <media/nbaio/PipeReader.h> 43#include <media/nbaio/SourceAudioBufferProvider.h> 44 45#include <powermanager/PowerManager.h> 46 47#include <common_time/cc_helper.h> 48#include <common_time/local_clock.h> 49 50#include "AudioFlinger.h" 51#include "AudioMixer.h" 52#include "FastMixer.h" 53#include "ServiceUtilities.h" 54#include "SchedulingPolicyService.h" 55 56#undef ADD_BATTERY_DATA 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait for setParameters to complete 102static const nsecs_t kSetParametersTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal mix buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalMixBufferSizeMs = 20; 111// maximum normal mix buffer size 112static const uint32_t kMaxNormalMixBufferSizeMs = 24; 113 114// Whether to use fast mixer 115static const enum { 116 FastMixer_Never, // never initialize or use: for debugging only 117 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 118 // normal mixer multiplier is 1 119 FastMixer_Static, // initialize if needed, then use all the time if initialized, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 // FIXME for FastMixer_Dynamic: 124 // Supporting this option will require fixing HALs that can't handle large writes. 125 // For example, one HAL implementation returns an error from a large write, 126 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 127 // We could either fix the HAL implementations, or provide a wrapper that breaks 128 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 129} kUseFastMixer = FastMixer_Static; 130 131// Priorities for requestPriority 132static const int kPriorityAudioApp = 2; 133static const int kPriorityFastMixer = 3; 134 135// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 136// for the track. The client then sub-divides this into smaller buffers for its use. 137// Currently the client uses double-buffering by default, but doesn't tell us about that. 138// So for now we just assume that client is double-buffered. 139// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 140// N-buffering, so AudioFlinger could allocate the right amount of memory. 141// See the client's minBufCount and mNotificationFramesAct calculations for details. 142static const int kFastTrackMultiplier = 2; 143 144// ---------------------------------------------------------------------------- 145 146#ifdef ADD_BATTERY_DATA 147// To collect the amplifier usage 148static void addBatteryData(uint32_t params) { 149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150 if (service == NULL) { 151 // it already logged 152 return; 153 } 154 155 service->addBatteryData(params); 156} 157#endif 158 159 160// ---------------------------------------------------------------------------- 161// CPU Stats 162// ---------------------------------------------------------------------------- 163 164class CpuStats { 165public: 166 CpuStats(); 167 void sample(const String8 &title); 168#ifdef DEBUG_CPU_USAGE 169private: 170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172 173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174 175 int mCpuNum; // thread's current CPU number 176 int mCpukHz; // frequency of thread's current CPU in kHz 177#endif 178}; 179 180CpuStats::CpuStats() 181#ifdef DEBUG_CPU_USAGE 182 : mCpuNum(-1), mCpukHz(-1) 183#endif 184{ 185} 186 187void CpuStats::sample(const String8 &title) { 188#ifdef DEBUG_CPU_USAGE 189 // get current thread's delta CPU time in wall clock ns 190 double wcNs; 191 bool valid = mCpuUsage.sampleAndEnable(wcNs); 192 193 // record sample for wall clock statistics 194 if (valid) { 195 mWcStats.sample(wcNs); 196 } 197 198 // get the current CPU number 199 int cpuNum = sched_getcpu(); 200 201 // get the current CPU frequency in kHz 202 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203 204 // check if either CPU number or frequency changed 205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 206 mCpuNum = cpuNum; 207 mCpukHz = cpukHz; 208 // ignore sample for purposes of cycles 209 valid = false; 210 } 211 212 // if no change in CPU number or frequency, then record sample for cycle statistics 213 if (valid && mCpukHz > 0) { 214 double cycles = wcNs * cpukHz * 0.000001; 215 mHzStats.sample(cycles); 216 } 217 218 unsigned n = mWcStats.n(); 219 // mCpuUsage.elapsed() is expensive, so don't call it every loop 220 if ((n & 127) == 1) { 221 long long elapsed = mCpuUsage.elapsed(); 222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 223 double perLoop = elapsed / (double) n; 224 double perLoop100 = perLoop * 0.01; 225 double perLoop1k = perLoop * 0.001; 226 double mean = mWcStats.mean(); 227 double stddev = mWcStats.stddev(); 228 double minimum = mWcStats.minimum(); 229 double maximum = mWcStats.maximum(); 230 double meanCycles = mHzStats.mean(); 231 double stddevCycles = mHzStats.stddev(); 232 double minCycles = mHzStats.minimum(); 233 double maxCycles = mHzStats.maximum(); 234 mCpuUsage.resetElapsed(); 235 mWcStats.reset(); 236 mHzStats.reset(); 237 ALOGD("CPU usage for %s over past %.1f secs\n" 238 " (%u mixer loops at %.1f mean ms per loop):\n" 239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 242 title.string(), 243 elapsed * .000000001, n, perLoop * .000001, 244 mean * .001, 245 stddev * .001, 246 minimum * .001, 247 maximum * .001, 248 mean / perLoop100, 249 stddev / perLoop100, 250 minimum / perLoop100, 251 maximum / perLoop100, 252 meanCycles / perLoop1k, 253 stddevCycles / perLoop1k, 254 minCycles / perLoop1k, 255 maxCycles / perLoop1k); 256 257 } 258 } 259#endif 260}; 261 262// ---------------------------------------------------------------------------- 263// ThreadBase 264// ---------------------------------------------------------------------------- 265 266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 268 : Thread(false /*canCallJava*/), 269 mType(type), 270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 271 // mChannelMask 272 mChannelCount(0), 273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 mParamCond.broadcast(); 285 // do not lock the mutex in destructor 286 releaseWakeLock_l(); 287 if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 377 if (err != 0) { 378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 379 "error %d", 380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 381 } 382 } break; 383 case CFG_EVENT_IO: { 384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 385 mAudioFlinger->mLock.lock(); 386 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 387 mAudioFlinger->mLock.unlock(); 388 } break; 389 default: 390 ALOGE("processConfigEvents() unknown event type %d", event->type()); 391 break; 392 } 393 delete event; 394 mLock.lock(); 395 } 396 mLock.unlock(); 397} 398 399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 400{ 401 const size_t SIZE = 256; 402 char buffer[SIZE]; 403 String8 result; 404 405 bool locked = AudioFlinger::dumpTryLock(mLock); 406 if (!locked) { 407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 408 write(fd, buffer, strlen(buffer)); 409 } 410 411 snprintf(buffer, SIZE, "io handle: %d\n", mId); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 430 result.append(buffer); 431 432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 433 result.append(buffer); 434 result.append(" Index Command"); 435 for (size_t i = 0; i < mNewParameters.size(); ++i) { 436 snprintf(buffer, SIZE, "\n %02d ", i); 437 result.append(buffer); 438 result.append(mNewParameters[i]); 439 } 440 441 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 442 result.append(buffer); 443 for (size_t i = 0; i < mConfigEvents.size(); i++) { 444 mConfigEvents[i]->dump(buffer, SIZE); 445 result.append(buffer); 446 } 447 result.append("\n"); 448 449 write(fd, result.string(), result.size()); 450 451 if (locked) { 452 mLock.unlock(); 453 } 454} 455 456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 457{ 458 const size_t SIZE = 256; 459 char buffer[SIZE]; 460 String8 result; 461 462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 463 write(fd, buffer, strlen(buffer)); 464 465 for (size_t i = 0; i < mEffectChains.size(); ++i) { 466 sp<EffectChain> chain = mEffectChains[i]; 467 if (chain != 0) { 468 chain->dump(fd, args); 469 } 470 } 471} 472 473void AudioFlinger::ThreadBase::acquireWakeLock() 474{ 475 Mutex::Autolock _l(mLock); 476 acquireWakeLock_l(); 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock_l() 480{ 481 if (mPowerManager == 0) { 482 // use checkService() to avoid blocking if power service is not up yet 483 sp<IBinder> binder = 484 defaultServiceManager()->checkService(String16("power")); 485 if (binder == 0) { 486 ALOGW("Thread %s cannot connect to the power manager service", mName); 487 } else { 488 mPowerManager = interface_cast<IPowerManager>(binder); 489 binder->linkToDeath(mDeathRecipient); 490 } 491 } 492 if (mPowerManager != 0) { 493 sp<IBinder> binder = new BBinder(); 494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 495 binder, 496 String16(mName)); 497 if (status == NO_ERROR) { 498 mWakeLockToken = binder; 499 } 500 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 501 } 502} 503 504void AudioFlinger::ThreadBase::releaseWakeLock() 505{ 506 Mutex::Autolock _l(mLock); 507 releaseWakeLock_l(); 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock_l() 511{ 512 if (mWakeLockToken != 0) { 513 ALOGV("releaseWakeLock_l() %s", mName); 514 if (mPowerManager != 0) { 515 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 516 } 517 mWakeLockToken.clear(); 518 } 519} 520 521void AudioFlinger::ThreadBase::clearPowerManager() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525 mPowerManager.clear(); 526} 527 528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 529{ 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 thread->clearPowerManager(); 533 } 534 ALOGW("power manager service died !!!"); 535} 536 537void AudioFlinger::ThreadBase::setEffectSuspended( 538 const effect_uuid_t *type, bool suspend, int sessionId) 539{ 540 Mutex::Autolock _l(mLock); 541 setEffectSuspended_l(type, suspend, sessionId); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended_l( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 sp<EffectChain> chain = getEffectChain_l(sessionId); 548 if (chain != 0) { 549 if (type != NULL) { 550 chain->setEffectSuspended_l(type, suspend); 551 } else { 552 chain->setEffectSuspendedAll_l(suspend); 553 } 554 } 555 556 updateSuspendedSessions_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 560{ 561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 562 if (index < 0) { 563 return; 564 } 565 566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 567 mSuspendedSessions.valueAt(index); 568 569 for (size_t i = 0; i < sessionEffects.size(); i++) { 570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 571 for (int j = 0; j < desc->mRefCount; j++) { 572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 573 chain->setEffectSuspendedAll_l(true); 574 } else { 575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 576 desc->mType.timeLow); 577 chain->setEffectSuspended_l(&desc->mType, true); 578 } 579 } 580 } 581} 582 583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId) 586{ 587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 588 589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 590 591 if (suspend) { 592 if (index >= 0) { 593 sessionEffects = mSuspendedSessions.valueAt(index); 594 } else { 595 mSuspendedSessions.add(sessionId, sessionEffects); 596 } 597 } else { 598 if (index < 0) { 599 return; 600 } 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } 603 604 605 int key = EffectChain::kKeyForSuspendAll; 606 if (type != NULL) { 607 key = type->timeLow; 608 } 609 index = sessionEffects.indexOfKey(key); 610 611 sp<SuspendedSessionDesc> desc; 612 if (suspend) { 613 if (index >= 0) { 614 desc = sessionEffects.valueAt(index); 615 } else { 616 desc = new SuspendedSessionDesc(); 617 if (type != NULL) { 618 desc->mType = *type; 619 } 620 sessionEffects.add(key, desc); 621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 622 } 623 desc->mRefCount++; 624 } else { 625 if (index < 0) { 626 return; 627 } 628 desc = sessionEffects.valueAt(index); 629 if (--desc->mRefCount == 0) { 630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 631 sessionEffects.removeItemsAt(index); 632 if (sessionEffects.isEmpty()) { 633 ALOGV("updateSuspendedSessions_l() restore removing session %d", 634 sessionId); 635 mSuspendedSessions.removeItem(sessionId); 636 } 637 } 638 } 639 if (!sessionEffects.isEmpty()) { 640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 641 } 642} 643 644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 645 bool enabled, 646 int sessionId) 647{ 648 Mutex::Autolock _l(mLock); 649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 if (mType != RECORD) { 657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 658 // another session. This gives the priority to well behaved effect control panels 659 // and applications not using global effects. 660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 661 // global effects 662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 664 } 665 } 666 667 sp<EffectChain> chain = getEffectChain_l(sessionId); 668 if (chain != 0) { 669 chain->checkSuspendOnEffectEnabled(effect, enabled); 670 } 671} 672 673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 675 const sp<AudioFlinger::Client>& client, 676 const sp<IEffectClient>& effectClient, 677 int32_t priority, 678 int sessionId, 679 effect_descriptor_t *desc, 680 int *enabled, 681 status_t *status 682 ) 683{ 684 sp<EffectModule> effect; 685 sp<EffectHandle> handle; 686 status_t lStatus; 687 sp<EffectChain> chain; 688 bool chainCreated = false; 689 bool effectCreated = false; 690 bool effectRegistered = false; 691 692 lStatus = initCheck(); 693 if (lStatus != NO_ERROR) { 694 ALOGW("createEffect_l() Audio driver not initialized."); 695 goto Exit; 696 } 697 698 // Do not allow effects with session ID 0 on direct output or duplicating threads 699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 702 desc->name, sessionId); 703 lStatus = BAD_VALUE; 704 goto Exit; 705 } 706 // Only Pre processor effects are allowed on input threads and only on input threads 707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 709 desc->name, desc->flags, mType); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 715 716 { // scope for mLock 717 Mutex::Autolock _l(mLock); 718 719 // check for existing effect chain with the requested audio session 720 chain = getEffectChain_l(sessionId); 721 if (chain == 0) { 722 // create a new chain for this session 723 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 724 chain = new EffectChain(this, sessionId); 725 addEffectChain_l(chain); 726 chain->setStrategy(getStrategyForSession_l(sessionId)); 727 chainCreated = true; 728 } else { 729 effect = chain->getEffectFromDesc_l(desc); 730 } 731 732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 733 734 if (effect == 0) { 735 int id = mAudioFlinger->nextUniqueId(); 736 // Check CPU and memory usage 737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 738 if (lStatus != NO_ERROR) { 739 goto Exit; 740 } 741 effectRegistered = true; 742 // create a new effect module if none present in the chain 743 effect = new EffectModule(this, chain, desc, id, sessionId); 744 lStatus = effect->status(); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 lStatus = chain->addEffect_l(effect); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 effectCreated = true; 753 754 effect->setDevice(mOutDevice); 755 effect->setDevice(mInDevice); 756 effect->setMode(mAudioFlinger->getMode()); 757 effect->setAudioSource(mAudioSource); 758 } 759 // create effect handle and connect it to effect module 760 handle = new EffectHandle(effect, client, effectClient, priority); 761 lStatus = effect->addHandle(handle.get()); 762 if (enabled != NULL) { 763 *enabled = (int)effect->isEnabled(); 764 } 765 } 766 767Exit: 768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 769 Mutex::Autolock _l(mLock); 770 if (effectCreated) { 771 chain->removeEffect_l(effect); 772 } 773 if (effectRegistered) { 774 AudioSystem::unregisterEffect(effect->id()); 775 } 776 if (chainCreated) { 777 removeEffectChain_l(chain); 778 } 779 handle.clear(); 780 } 781 782 if (status != NULL) { 783 *status = lStatus; 784 } 785 return handle; 786} 787 788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 789{ 790 Mutex::Autolock _l(mLock); 791 return getEffect_l(sessionId, effectId); 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 795{ 796 sp<EffectChain> chain = getEffectChain_l(sessionId); 797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 798} 799 800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 801// PlaybackThread::mLock held 802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 803{ 804 // check for existing effect chain with the requested audio session 805 int sessionId = effect->sessionId(); 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 bool chainCreated = false; 808 809 if (chain == 0) { 810 // create a new chain for this session 811 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 812 chain = new EffectChain(this, sessionId); 813 addEffectChain_l(chain); 814 chain->setStrategy(getStrategyForSession_l(sessionId)); 815 chainCreated = true; 816 } 817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 818 819 if (chain->getEffectFromId_l(effect->id()) != 0) { 820 ALOGW("addEffect_l() %p effect %s already present in chain %p", 821 this, effect->desc().name, chain.get()); 822 return BAD_VALUE; 823 } 824 825 status_t status = chain->addEffect_l(effect); 826 if (status != NO_ERROR) { 827 if (chainCreated) { 828 removeEffectChain_l(chain); 829 } 830 return status; 831 } 832 833 effect->setDevice(mOutDevice); 834 effect->setDevice(mInDevice); 835 effect->setMode(mAudioFlinger->getMode()); 836 effect->setAudioSource(mAudioSource); 837 return NO_ERROR; 838} 839 840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 841 842 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 843 effect_descriptor_t desc = effect->desc(); 844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 845 detachAuxEffect_l(effect->id()); 846 } 847 848 sp<EffectChain> chain = effect->chain().promote(); 849 if (chain != 0) { 850 // remove effect chain if removing last effect 851 if (chain->removeEffect_l(effect) == 0) { 852 removeEffectChain_l(chain); 853 } 854 } else { 855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 856 } 857} 858 859void AudioFlinger::ThreadBase::lockEffectChains_l( 860 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 861{ 862 effectChains = mEffectChains; 863 for (size_t i = 0; i < mEffectChains.size(); i++) { 864 mEffectChains[i]->lock(); 865 } 866} 867 868void AudioFlinger::ThreadBase::unlockEffectChains( 869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 870{ 871 for (size_t i = 0; i < effectChains.size(); i++) { 872 effectChains[i]->unlock(); 873 } 874} 875 876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 877{ 878 Mutex::Autolock _l(mLock); 879 return getEffectChain_l(sessionId); 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 883{ 884 size_t size = mEffectChains.size(); 885 for (size_t i = 0; i < size; i++) { 886 if (mEffectChains[i]->sessionId() == sessionId) { 887 return mEffectChains[i]; 888 } 889 } 890 return 0; 891} 892 893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 894{ 895 Mutex::Autolock _l(mLock); 896 size_t size = mEffectChains.size(); 897 for (size_t i = 0; i < size; i++) { 898 mEffectChains[i]->setMode_l(mode); 899 } 900} 901 902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 903 EffectHandle *handle, 904 bool unpinIfLast) { 905 906 Mutex::Autolock _l(mLock); 907 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 908 // delete the effect module if removing last handle on it 909 if (effect->removeHandle(handle) == 0) { 910 if (!effect->isPinned() || unpinIfLast) { 911 removeEffect_l(effect); 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 } 915} 916 917// ---------------------------------------------------------------------------- 918// Playback 919// ---------------------------------------------------------------------------- 920 921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 922 AudioStreamOut* output, 923 audio_io_handle_t id, 924 audio_devices_t device, 925 type_t type) 926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 928 // mStreamTypes[] initialized in constructor body 929 mOutput(output), 930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 931 mMixerStatus(MIXER_IDLE), 932 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 934 mScreenState(AudioFlinger::mScreenState), 935 // index 0 is reserved for normal mixer's submix 936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 937{ 938 snprintf(mName, kNameLength, "AudioOut_%X", id); 939 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 940 941 // Assumes constructor is called by AudioFlinger with it's mLock held, but 942 // it would be safer to explicitly pass initial masterVolume/masterMute as 943 // parameter. 944 // 945 // If the HAL we are using has support for master volume or master mute, 946 // then do not attenuate or mute during mixing (just leave the volume at 1.0 947 // and the mute set to false). 948 mMasterVolume = audioFlinger->masterVolume_l(); 949 mMasterMute = audioFlinger->masterMute_l(); 950 if (mOutput && mOutput->audioHwDev) { 951 if (mOutput->audioHwDev->canSetMasterVolume()) { 952 mMasterVolume = 1.0; 953 } 954 955 if (mOutput->audioHwDev->canSetMasterMute()) { 956 mMasterMute = false; 957 } 958 } 959 960 readOutputParameters(); 961 962 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 963 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 964 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 965 stream = (audio_stream_type_t) (stream + 1)) { 966 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 967 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 968 } 969 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 970 // because mAudioFlinger doesn't have one to copy from 971} 972 973AudioFlinger::PlaybackThread::~PlaybackThread() 974{ 975 mAudioFlinger->unregisterWriter(mNBLogWriter); 976 delete [] mMixBuffer; 977} 978 979void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 980{ 981 dumpInternals(fd, args); 982 dumpTracks(fd, args); 983 dumpEffectChains(fd, args); 984} 985 986void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 987{ 988 const size_t SIZE = 256; 989 char buffer[SIZE]; 990 String8 result; 991 992 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 993 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 994 const stream_type_t *st = &mStreamTypes[i]; 995 if (i > 0) { 996 result.appendFormat(", "); 997 } 998 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 999 if (st->mute) { 1000 result.append("M"); 1001 } 1002 } 1003 result.append("\n"); 1004 write(fd, result.string(), result.length()); 1005 result.clear(); 1006 1007 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1008 result.append(buffer); 1009 Track::appendDumpHeader(result); 1010 for (size_t i = 0; i < mTracks.size(); ++i) { 1011 sp<Track> track = mTracks[i]; 1012 if (track != 0) { 1013 track->dump(buffer, SIZE); 1014 result.append(buffer); 1015 } 1016 } 1017 1018 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1019 result.append(buffer); 1020 Track::appendDumpHeader(result); 1021 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1022 sp<Track> track = mActiveTracks[i].promote(); 1023 if (track != 0) { 1024 track->dump(buffer, SIZE); 1025 result.append(buffer); 1026 } 1027 } 1028 write(fd, result.string(), result.size()); 1029 1030 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1031 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1032 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1033 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1034} 1035 1036void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1037{ 1038 const size_t SIZE = 256; 1039 char buffer[SIZE]; 1040 String8 result; 1041 1042 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1043 result.append(buffer); 1044 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1045 ns2ms(systemTime() - mLastWriteTime)); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1056 result.append(buffer); 1057 write(fd, result.string(), result.size()); 1058 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1059 1060 dumpBase(fd, args); 1061} 1062 1063// Thread virtuals 1064status_t AudioFlinger::PlaybackThread::readyToRun() 1065{ 1066 status_t status = initCheck(); 1067 if (status == NO_ERROR) { 1068 ALOGI("AudioFlinger's thread %p ready to run", this); 1069 } else { 1070 ALOGE("No working audio driver found."); 1071 } 1072 return status; 1073} 1074 1075void AudioFlinger::PlaybackThread::onFirstRef() 1076{ 1077 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1078} 1079 1080// ThreadBase virtuals 1081void AudioFlinger::PlaybackThread::preExit() 1082{ 1083 ALOGV(" preExit()"); 1084 // FIXME this is using hard-coded strings but in the future, this functionality will be 1085 // converted to use audio HAL extensions required to support tunneling 1086 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1087} 1088 1089// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1090sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1091 const sp<AudioFlinger::Client>& client, 1092 audio_stream_type_t streamType, 1093 uint32_t sampleRate, 1094 audio_format_t format, 1095 audio_channel_mask_t channelMask, 1096 size_t frameCount, 1097 const sp<IMemory>& sharedBuffer, 1098 int sessionId, 1099 IAudioFlinger::track_flags_t *flags, 1100 pid_t tid, 1101 status_t *status) 1102{ 1103 sp<Track> track; 1104 status_t lStatus; 1105 1106 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1107 1108 // client expresses a preference for FAST, but we get the final say 1109 if (*flags & IAudioFlinger::TRACK_FAST) { 1110 if ( 1111 // not timed 1112 (!isTimed) && 1113 // either of these use cases: 1114 ( 1115 // use case 1: shared buffer with any frame count 1116 ( 1117 (sharedBuffer != 0) 1118 ) || 1119 // use case 2: callback handler and frame count is default or at least as large as HAL 1120 ( 1121 (tid != -1) && 1122 ((frameCount == 0) || 1123 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1124 ) 1125 ) && 1126 // PCM data 1127 audio_is_linear_pcm(format) && 1128 // mono or stereo 1129 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1130 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1131#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1132 // hardware sample rate 1133 (sampleRate == mSampleRate) && 1134#endif 1135 // normal mixer has an associated fast mixer 1136 hasFastMixer() && 1137 // there are sufficient fast track slots available 1138 (mFastTrackAvailMask != 0) 1139 // FIXME test that MixerThread for this fast track has a capable output HAL 1140 // FIXME add a permission test also? 1141 ) { 1142 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1143 if (frameCount == 0) { 1144 frameCount = mFrameCount * kFastTrackMultiplier; 1145 } 1146 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1147 frameCount, mFrameCount); 1148 } else { 1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1150 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1151 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1152 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1153 audio_is_linear_pcm(format), 1154 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1155 *flags &= ~IAudioFlinger::TRACK_FAST; 1156 // For compatibility with AudioTrack calculation, buffer depth is forced 1157 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1158 // This is probably too conservative, but legacy application code may depend on it. 1159 // If you change this calculation, also review the start threshold which is related. 1160 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1161 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1162 if (minBufCount < 2) { 1163 minBufCount = 2; 1164 } 1165 size_t minFrameCount = mNormalFrameCount * minBufCount; 1166 if (frameCount < minFrameCount) { 1167 frameCount = minFrameCount; 1168 } 1169 } 1170 } 1171 1172 if (mType == DIRECT) { 1173 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1174 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1175 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1176 "for output %p with format %d", 1177 sampleRate, format, channelMask, mOutput, mFormat); 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } else { 1183 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1184 if (sampleRate > mSampleRate*2) { 1185 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 } 1190 1191 lStatus = initCheck(); 1192 if (lStatus != NO_ERROR) { 1193 ALOGE("Audio driver not initialized."); 1194 goto Exit; 1195 } 1196 1197 { // scope for mLock 1198 Mutex::Autolock _l(mLock); 1199 1200 // all tracks in same audio session must share the same routing strategy otherwise 1201 // conflicts will happen when tracks are moved from one output to another by audio policy 1202 // manager 1203 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1204 for (size_t i = 0; i < mTracks.size(); ++i) { 1205 sp<Track> t = mTracks[i]; 1206 if (t != 0 && !t->isOutputTrack()) { 1207 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1208 if (sessionId == t->sessionId() && strategy != actual) { 1209 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1210 strategy, actual); 1211 lStatus = BAD_VALUE; 1212 goto Exit; 1213 } 1214 } 1215 } 1216 1217 if (!isTimed) { 1218 track = new Track(this, client, streamType, sampleRate, format, 1219 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1220 } else { 1221 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1222 channelMask, frameCount, sharedBuffer, sessionId); 1223 } 1224 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1225 lStatus = NO_MEMORY; 1226 goto Exit; 1227 } 1228 mTracks.add(track); 1229 1230 sp<EffectChain> chain = getEffectChain_l(sessionId); 1231 if (chain != 0) { 1232 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1233 track->setMainBuffer(chain->inBuffer()); 1234 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1235 chain->incTrackCnt(); 1236 } 1237 1238 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1239 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1240 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1241 // so ask activity manager to do this on our behalf 1242 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1243 } 1244 } 1245 1246 lStatus = NO_ERROR; 1247 1248Exit: 1249 if (status) { 1250 *status = lStatus; 1251 } 1252 return track; 1253} 1254 1255uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1256{ 1257 return latency; 1258} 1259 1260uint32_t AudioFlinger::PlaybackThread::latency() const 1261{ 1262 Mutex::Autolock _l(mLock); 1263 return latency_l(); 1264} 1265uint32_t AudioFlinger::PlaybackThread::latency_l() const 1266{ 1267 if (initCheck() == NO_ERROR) { 1268 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1269 } else { 1270 return 0; 1271 } 1272} 1273 1274void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1275{ 1276 Mutex::Autolock _l(mLock); 1277 // Don't apply master volume in SW if our HAL can do it for us. 1278 if (mOutput && mOutput->audioHwDev && 1279 mOutput->audioHwDev->canSetMasterVolume()) { 1280 mMasterVolume = 1.0; 1281 } else { 1282 mMasterVolume = value; 1283 } 1284} 1285 1286void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1287{ 1288 Mutex::Autolock _l(mLock); 1289 // Don't apply master mute in SW if our HAL can do it for us. 1290 if (mOutput && mOutput->audioHwDev && 1291 mOutput->audioHwDev->canSetMasterMute()) { 1292 mMasterMute = false; 1293 } else { 1294 mMasterMute = muted; 1295 } 1296} 1297 1298void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1299{ 1300 Mutex::Autolock _l(mLock); 1301 mStreamTypes[stream].volume = value; 1302} 1303 1304void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1305{ 1306 Mutex::Autolock _l(mLock); 1307 mStreamTypes[stream].mute = muted; 1308} 1309 1310float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1311{ 1312 Mutex::Autolock _l(mLock); 1313 return mStreamTypes[stream].volume; 1314} 1315 1316// addTrack_l() must be called with ThreadBase::mLock held 1317status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1318{ 1319 status_t status = ALREADY_EXISTS; 1320 1321 // set retry count for buffer fill 1322 track->mRetryCount = kMaxTrackStartupRetries; 1323 if (mActiveTracks.indexOf(track) < 0) { 1324 // the track is newly added, make sure it fills up all its 1325 // buffers before playing. This is to ensure the client will 1326 // effectively get the latency it requested. 1327 track->mFillingUpStatus = Track::FS_FILLING; 1328 track->mResetDone = false; 1329 track->mPresentationCompleteFrames = 0; 1330 mActiveTracks.add(track); 1331 if (track->mainBuffer() != mMixBuffer) { 1332 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1333 if (chain != 0) { 1334 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1335 track->sessionId()); 1336 chain->incActiveTrackCnt(); 1337 } 1338 } 1339 1340 status = NO_ERROR; 1341 } 1342 1343 ALOGV("mWaitWorkCV.broadcast"); 1344 mWaitWorkCV.broadcast(); 1345 1346 return status; 1347} 1348 1349// destroyTrack_l() must be called with ThreadBase::mLock held 1350void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1351{ 1352 track->mState = TrackBase::TERMINATED; 1353 // active tracks are removed by threadLoop() 1354 if (mActiveTracks.indexOf(track) < 0) { 1355 removeTrack_l(track); 1356 } 1357} 1358 1359void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1360{ 1361 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1362 mTracks.remove(track); 1363 deleteTrackName_l(track->name()); 1364 // redundant as track is about to be destroyed, for dumpsys only 1365 track->mName = -1; 1366 if (track->isFastTrack()) { 1367 int index = track->mFastIndex; 1368 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1369 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1370 mFastTrackAvailMask |= 1 << index; 1371 // redundant as track is about to be destroyed, for dumpsys only 1372 track->mFastIndex = -1; 1373 } 1374 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1375 if (chain != 0) { 1376 chain->decTrackCnt(); 1377 } 1378} 1379 1380String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1381{ 1382 String8 out_s8 = String8(""); 1383 char *s; 1384 1385 Mutex::Autolock _l(mLock); 1386 if (initCheck() != NO_ERROR) { 1387 return out_s8; 1388 } 1389 1390 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1391 out_s8 = String8(s); 1392 free(s); 1393 return out_s8; 1394} 1395 1396// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1397void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1398 AudioSystem::OutputDescriptor desc; 1399 void *param2 = NULL; 1400 1401 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1402 param); 1403 1404 switch (event) { 1405 case AudioSystem::OUTPUT_OPENED: 1406 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1407 desc.channels = mChannelMask; 1408 desc.samplingRate = mSampleRate; 1409 desc.format = mFormat; 1410 desc.frameCount = mNormalFrameCount; // FIXME see 1411 // AudioFlinger::frameCount(audio_io_handle_t) 1412 desc.latency = latency(); 1413 param2 = &desc; 1414 break; 1415 1416 case AudioSystem::STREAM_CONFIG_CHANGED: 1417 param2 = ¶m; 1418 case AudioSystem::OUTPUT_CLOSED: 1419 default: 1420 break; 1421 } 1422 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1423} 1424 1425void AudioFlinger::PlaybackThread::readOutputParameters() 1426{ 1427 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1428 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1429 mChannelCount = (uint16_t)popcount(mChannelMask); 1430 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1431 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1432 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1433 if (mFrameCount & 15) { 1434 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1435 mFrameCount); 1436 } 1437 1438 // Calculate size of normal mix buffer relative to the HAL output buffer size 1439 double multiplier = 1.0; 1440 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1441 kUseFastMixer == FastMixer_Dynamic)) { 1442 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1443 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1444 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1445 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1446 maxNormalFrameCount = maxNormalFrameCount & ~15; 1447 if (maxNormalFrameCount < minNormalFrameCount) { 1448 maxNormalFrameCount = minNormalFrameCount; 1449 } 1450 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1451 if (multiplier <= 1.0) { 1452 multiplier = 1.0; 1453 } else if (multiplier <= 2.0) { 1454 if (2 * mFrameCount <= maxNormalFrameCount) { 1455 multiplier = 2.0; 1456 } else { 1457 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1458 } 1459 } else { 1460 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1461 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1462 // track, but we sometimes have to do this to satisfy the maximum frame count 1463 // constraint) 1464 // FIXME this rounding up should not be done if no HAL SRC 1465 uint32_t truncMult = (uint32_t) multiplier; 1466 if ((truncMult & 1)) { 1467 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1468 ++truncMult; 1469 } 1470 } 1471 multiplier = (double) truncMult; 1472 } 1473 } 1474 mNormalFrameCount = multiplier * mFrameCount; 1475 // round up to nearest 16 frames to satisfy AudioMixer 1476 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1477 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1478 mNormalFrameCount); 1479 1480 delete[] mMixBuffer; 1481 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1482 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1483 1484 // force reconfiguration of effect chains and engines to take new buffer size and audio 1485 // parameters into account 1486 // Note that mLock is not held when readOutputParameters() is called from the constructor 1487 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1488 // matter. 1489 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1490 Vector< sp<EffectChain> > effectChains = mEffectChains; 1491 for (size_t i = 0; i < effectChains.size(); i ++) { 1492 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1493 } 1494} 1495 1496 1497status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1498{ 1499 if (halFrames == NULL || dspFrames == NULL) { 1500 return BAD_VALUE; 1501 } 1502 Mutex::Autolock _l(mLock); 1503 if (initCheck() != NO_ERROR) { 1504 return INVALID_OPERATION; 1505 } 1506 size_t framesWritten = mBytesWritten / mFrameSize; 1507 *halFrames = framesWritten; 1508 1509 if (isSuspended()) { 1510 // return an estimation of rendered frames when the output is suspended 1511 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1512 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1513 return NO_ERROR; 1514 } else { 1515 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1516 } 1517} 1518 1519uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1520{ 1521 Mutex::Autolock _l(mLock); 1522 uint32_t result = 0; 1523 if (getEffectChain_l(sessionId) != 0) { 1524 result = EFFECT_SESSION; 1525 } 1526 1527 for (size_t i = 0; i < mTracks.size(); ++i) { 1528 sp<Track> track = mTracks[i]; 1529 if (sessionId == track->sessionId() && !track->isInvalid()) { 1530 result |= TRACK_SESSION; 1531 break; 1532 } 1533 } 1534 1535 return result; 1536} 1537 1538uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1539{ 1540 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1541 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1542 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1543 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1544 } 1545 for (size_t i = 0; i < mTracks.size(); i++) { 1546 sp<Track> track = mTracks[i]; 1547 if (sessionId == track->sessionId() && !track->isInvalid()) { 1548 return AudioSystem::getStrategyForStream(track->streamType()); 1549 } 1550 } 1551 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1552} 1553 1554 1555AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1556{ 1557 Mutex::Autolock _l(mLock); 1558 return mOutput; 1559} 1560 1561AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1562{ 1563 Mutex::Autolock _l(mLock); 1564 AudioStreamOut *output = mOutput; 1565 mOutput = NULL; 1566 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1567 // must push a NULL and wait for ack 1568 mOutputSink.clear(); 1569 mPipeSink.clear(); 1570 mNormalSink.clear(); 1571 return output; 1572} 1573 1574// this method must always be called either with ThreadBase mLock held or inside the thread loop 1575audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1576{ 1577 if (mOutput == NULL) { 1578 return NULL; 1579 } 1580 return &mOutput->stream->common; 1581} 1582 1583uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1584{ 1585 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1586} 1587 1588status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1589{ 1590 if (!isValidSyncEvent(event)) { 1591 return BAD_VALUE; 1592 } 1593 1594 Mutex::Autolock _l(mLock); 1595 1596 for (size_t i = 0; i < mTracks.size(); ++i) { 1597 sp<Track> track = mTracks[i]; 1598 if (event->triggerSession() == track->sessionId()) { 1599 (void) track->setSyncEvent(event); 1600 return NO_ERROR; 1601 } 1602 } 1603 1604 return NAME_NOT_FOUND; 1605} 1606 1607bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1608{ 1609 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1610} 1611 1612void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1613 const Vector< sp<Track> >& tracksToRemove) 1614{ 1615 size_t count = tracksToRemove.size(); 1616 if (CC_UNLIKELY(count)) { 1617 for (size_t i = 0 ; i < count ; i++) { 1618 const sp<Track>& track = tracksToRemove.itemAt(i); 1619 if ((track->sharedBuffer() != 0) && 1620 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1621 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1622 } 1623 } 1624 } 1625 1626} 1627 1628void AudioFlinger::PlaybackThread::checkSilentMode_l() 1629{ 1630 if (!mMasterMute) { 1631 char value[PROPERTY_VALUE_MAX]; 1632 if (property_get("ro.audio.silent", value, "0") > 0) { 1633 char *endptr; 1634 unsigned long ul = strtoul(value, &endptr, 0); 1635 if (*endptr == '\0' && ul != 0) { 1636 ALOGD("Silence is golden"); 1637 // The setprop command will not allow a property to be changed after 1638 // the first time it is set, so we don't have to worry about un-muting. 1639 setMasterMute_l(true); 1640 } 1641 } 1642 } 1643} 1644 1645// shared by MIXER and DIRECT, overridden by DUPLICATING 1646void AudioFlinger::PlaybackThread::threadLoop_write() 1647{ 1648 // FIXME rewrite to reduce number of system calls 1649 mLastWriteTime = systemTime(); 1650 mInWrite = true; 1651 int bytesWritten; 1652 1653 // If an NBAIO sink is present, use it to write the normal mixer's submix 1654 if (mNormalSink != 0) { 1655#define mBitShift 2 // FIXME 1656 size_t count = mixBufferSize >> mBitShift; 1657 ATRACE_BEGIN("write"); 1658 // update the setpoint when AudioFlinger::mScreenState changes 1659 uint32_t screenState = AudioFlinger::mScreenState; 1660 if (screenState != mScreenState) { 1661 mScreenState = screenState; 1662 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1663 if (pipe != NULL) { 1664 pipe->setAvgFrames((mScreenState & 1) ? 1665 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1666 } 1667 } 1668 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1669 ATRACE_END(); 1670 if (framesWritten > 0) { 1671 bytesWritten = framesWritten << mBitShift; 1672 } else { 1673 bytesWritten = framesWritten; 1674 } 1675 // otherwise use the HAL / AudioStreamOut directly 1676 } else { 1677 // Direct output thread. 1678 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1679 } 1680 1681 if (bytesWritten > 0) { 1682 mBytesWritten += mixBufferSize; 1683 } 1684 mNumWrites++; 1685 mInWrite = false; 1686} 1687 1688/* 1689The derived values that are cached: 1690 - mixBufferSize from frame count * frame size 1691 - activeSleepTime from activeSleepTimeUs() 1692 - idleSleepTime from idleSleepTimeUs() 1693 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1694 - maxPeriod from frame count and sample rate (MIXER only) 1695 1696The parameters that affect these derived values are: 1697 - frame count 1698 - frame size 1699 - sample rate 1700 - device type: A2DP or not 1701 - device latency 1702 - format: PCM or not 1703 - active sleep time 1704 - idle sleep time 1705*/ 1706 1707void AudioFlinger::PlaybackThread::cacheParameters_l() 1708{ 1709 mixBufferSize = mNormalFrameCount * mFrameSize; 1710 activeSleepTime = activeSleepTimeUs(); 1711 idleSleepTime = idleSleepTimeUs(); 1712} 1713 1714void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1715{ 1716 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1717 this, streamType, mTracks.size()); 1718 Mutex::Autolock _l(mLock); 1719 1720 size_t size = mTracks.size(); 1721 for (size_t i = 0; i < size; i++) { 1722 sp<Track> t = mTracks[i]; 1723 if (t->streamType() == streamType) { 1724 t->invalidate(); 1725 } 1726 } 1727} 1728 1729status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1730{ 1731 int session = chain->sessionId(); 1732 int16_t *buffer = mMixBuffer; 1733 bool ownsBuffer = false; 1734 1735 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1736 if (session > 0) { 1737 // Only one effect chain can be present in direct output thread and it uses 1738 // the mix buffer as input 1739 if (mType != DIRECT) { 1740 size_t numSamples = mNormalFrameCount * mChannelCount; 1741 buffer = new int16_t[numSamples]; 1742 memset(buffer, 0, numSamples * sizeof(int16_t)); 1743 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1744 ownsBuffer = true; 1745 } 1746 1747 // Attach all tracks with same session ID to this chain. 1748 for (size_t i = 0; i < mTracks.size(); ++i) { 1749 sp<Track> track = mTracks[i]; 1750 if (session == track->sessionId()) { 1751 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1752 buffer); 1753 track->setMainBuffer(buffer); 1754 chain->incTrackCnt(); 1755 } 1756 } 1757 1758 // indicate all active tracks in the chain 1759 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1760 sp<Track> track = mActiveTracks[i].promote(); 1761 if (track == 0) { 1762 continue; 1763 } 1764 if (session == track->sessionId()) { 1765 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1766 chain->incActiveTrackCnt(); 1767 } 1768 } 1769 } 1770 1771 chain->setInBuffer(buffer, ownsBuffer); 1772 chain->setOutBuffer(mMixBuffer); 1773 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1774 // chains list in order to be processed last as it contains output stage effects 1775 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1776 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1777 // after track specific effects and before output stage 1778 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1779 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1780 // Effect chain for other sessions are inserted at beginning of effect 1781 // chains list to be processed before output mix effects. Relative order between other 1782 // sessions is not important 1783 size_t size = mEffectChains.size(); 1784 size_t i = 0; 1785 for (i = 0; i < size; i++) { 1786 if (mEffectChains[i]->sessionId() < session) { 1787 break; 1788 } 1789 } 1790 mEffectChains.insertAt(chain, i); 1791 checkSuspendOnAddEffectChain_l(chain); 1792 1793 return NO_ERROR; 1794} 1795 1796size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1797{ 1798 int session = chain->sessionId(); 1799 1800 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1801 1802 for (size_t i = 0; i < mEffectChains.size(); i++) { 1803 if (chain == mEffectChains[i]) { 1804 mEffectChains.removeAt(i); 1805 // detach all active tracks from the chain 1806 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1807 sp<Track> track = mActiveTracks[i].promote(); 1808 if (track == 0) { 1809 continue; 1810 } 1811 if (session == track->sessionId()) { 1812 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1813 chain.get(), session); 1814 chain->decActiveTrackCnt(); 1815 } 1816 } 1817 1818 // detach all tracks with same session ID from this chain 1819 for (size_t i = 0; i < mTracks.size(); ++i) { 1820 sp<Track> track = mTracks[i]; 1821 if (session == track->sessionId()) { 1822 track->setMainBuffer(mMixBuffer); 1823 chain->decTrackCnt(); 1824 } 1825 } 1826 break; 1827 } 1828 } 1829 return mEffectChains.size(); 1830} 1831 1832status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1833 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1834{ 1835 Mutex::Autolock _l(mLock); 1836 return attachAuxEffect_l(track, EffectId); 1837} 1838 1839status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1840 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1841{ 1842 status_t status = NO_ERROR; 1843 1844 if (EffectId == 0) { 1845 track->setAuxBuffer(0, NULL); 1846 } else { 1847 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1848 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1849 if (effect != 0) { 1850 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1851 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1852 } else { 1853 status = INVALID_OPERATION; 1854 } 1855 } else { 1856 status = BAD_VALUE; 1857 } 1858 } 1859 return status; 1860} 1861 1862void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1863{ 1864 for (size_t i = 0; i < mTracks.size(); ++i) { 1865 sp<Track> track = mTracks[i]; 1866 if (track->auxEffectId() == effectId) { 1867 attachAuxEffect_l(track, 0); 1868 } 1869 } 1870} 1871 1872bool AudioFlinger::PlaybackThread::threadLoop() 1873{ 1874 Vector< sp<Track> > tracksToRemove; 1875 1876 standbyTime = systemTime(); 1877 1878 // MIXER 1879 nsecs_t lastWarning = 0; 1880 1881 // DUPLICATING 1882 // FIXME could this be made local to while loop? 1883 writeFrames = 0; 1884 1885 cacheParameters_l(); 1886 sleepTime = idleSleepTime; 1887 1888 if (mType == MIXER) { 1889 sleepTimeShift = 0; 1890 } 1891 1892 CpuStats cpuStats; 1893 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1894 1895 acquireWakeLock(); 1896 1897 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1898 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1899 // and then that string will be logged at the next convenient opportunity. 1900 const char *logString = NULL; 1901 1902 while (!exitPending()) 1903 { 1904 cpuStats.sample(myName); 1905 1906 Vector< sp<EffectChain> > effectChains; 1907 1908 processConfigEvents(); 1909 1910 { // scope for mLock 1911 1912 Mutex::Autolock _l(mLock); 1913 1914 if (logString != NULL) { 1915 mNBLogWriter->logTimestamp(); 1916 mNBLogWriter->log(logString); 1917 logString = NULL; 1918 } 1919 1920 if (checkForNewParameters_l()) { 1921 cacheParameters_l(); 1922 } 1923 1924 saveOutputTracks(); 1925 1926 // put audio hardware into standby after short delay 1927 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1928 isSuspended())) { 1929 if (!mStandby) { 1930 1931 threadLoop_standby(); 1932 1933 mStandby = true; 1934 } 1935 1936 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1937 // we're about to wait, flush the binder command buffer 1938 IPCThreadState::self()->flushCommands(); 1939 1940 clearOutputTracks(); 1941 1942 if (exitPending()) { 1943 break; 1944 } 1945 1946 releaseWakeLock_l(); 1947 // wait until we have something to do... 1948 ALOGV("%s going to sleep", myName.string()); 1949 mWaitWorkCV.wait(mLock); 1950 ALOGV("%s waking up", myName.string()); 1951 acquireWakeLock_l(); 1952 1953 mMixerStatus = MIXER_IDLE; 1954 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1955 mBytesWritten = 0; 1956 1957 checkSilentMode_l(); 1958 1959 standbyTime = systemTime() + standbyDelay; 1960 sleepTime = idleSleepTime; 1961 if (mType == MIXER) { 1962 sleepTimeShift = 0; 1963 } 1964 1965 continue; 1966 } 1967 } 1968 1969 // mMixerStatusIgnoringFastTracks is also updated internally 1970 mMixerStatus = prepareTracks_l(&tracksToRemove); 1971 1972 // prevent any changes in effect chain list and in each effect chain 1973 // during mixing and effect process as the audio buffers could be deleted 1974 // or modified if an effect is created or deleted 1975 lockEffectChains_l(effectChains); 1976 } 1977 1978 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1979 threadLoop_mix(); 1980 } else { 1981 threadLoop_sleepTime(); 1982 } 1983 1984 if (isSuspended()) { 1985 sleepTime = suspendSleepTimeUs(); 1986 mBytesWritten += mixBufferSize; 1987 } 1988 1989 // only process effects if we're going to write 1990 if (sleepTime == 0) { 1991 for (size_t i = 0; i < effectChains.size(); i ++) { 1992 effectChains[i]->process_l(); 1993 } 1994 } 1995 1996 // enable changes in effect chain 1997 unlockEffectChains(effectChains); 1998 1999 // sleepTime == 0 means we must write to audio hardware 2000 if (sleepTime == 0) { 2001 2002 threadLoop_write(); 2003 2004if (mType == MIXER) { 2005 // write blocked detection 2006 nsecs_t now = systemTime(); 2007 nsecs_t delta = now - mLastWriteTime; 2008 if (!mStandby && delta > maxPeriod) { 2009 mNumDelayedWrites++; 2010 if ((now - lastWarning) > kWarningThrottleNs) { 2011 ATRACE_NAME("underrun"); 2012 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2013 ns2ms(delta), mNumDelayedWrites, this); 2014 lastWarning = now; 2015 } 2016 } 2017} 2018 2019 mStandby = false; 2020 } else { 2021 usleep(sleepTime); 2022 } 2023 2024 // Finally let go of removed track(s), without the lock held 2025 // since we can't guarantee the destructors won't acquire that 2026 // same lock. This will also mutate and push a new fast mixer state. 2027 threadLoop_removeTracks(tracksToRemove); 2028 tracksToRemove.clear(); 2029 2030 // FIXME I don't understand the need for this here; 2031 // it was in the original code but maybe the 2032 // assignment in saveOutputTracks() makes this unnecessary? 2033 clearOutputTracks(); 2034 2035 // Effect chains will be actually deleted here if they were removed from 2036 // mEffectChains list during mixing or effects processing 2037 effectChains.clear(); 2038 2039 // FIXME Note that the above .clear() is no longer necessary since effectChains 2040 // is now local to this block, but will keep it for now (at least until merge done). 2041 } 2042 2043 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2044 if (mType == MIXER || mType == DIRECT) { 2045 // put output stream into standby mode 2046 if (!mStandby) { 2047 mOutput->stream->common.standby(&mOutput->stream->common); 2048 } 2049 } 2050 2051 releaseWakeLock(); 2052 2053 ALOGV("Thread %p type %d exiting", this, mType); 2054 return false; 2055} 2056 2057 2058// ---------------------------------------------------------------------------- 2059 2060AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2061 audio_io_handle_t id, audio_devices_t device, type_t type) 2062 : PlaybackThread(audioFlinger, output, id, device, type), 2063 // mAudioMixer below 2064 // mFastMixer below 2065 mFastMixerFutex(0) 2066 // mOutputSink below 2067 // mPipeSink below 2068 // mNormalSink below 2069{ 2070 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2071 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2072 "mFrameCount=%d, mNormalFrameCount=%d", 2073 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2074 mNormalFrameCount); 2075 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2076 2077 // FIXME - Current mixer implementation only supports stereo output 2078 if (mChannelCount != FCC_2) { 2079 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2080 } 2081 2082 // create an NBAIO sink for the HAL output stream, and negotiate 2083 mOutputSink = new AudioStreamOutSink(output->stream); 2084 size_t numCounterOffers = 0; 2085 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2086 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2087 ALOG_ASSERT(index == 0); 2088 2089 // initialize fast mixer depending on configuration 2090 bool initFastMixer; 2091 switch (kUseFastMixer) { 2092 case FastMixer_Never: 2093 initFastMixer = false; 2094 break; 2095 case FastMixer_Always: 2096 initFastMixer = true; 2097 break; 2098 case FastMixer_Static: 2099 case FastMixer_Dynamic: 2100 initFastMixer = mFrameCount < mNormalFrameCount; 2101 break; 2102 } 2103 if (initFastMixer) { 2104 2105 // create a MonoPipe to connect our submix to FastMixer 2106 NBAIO_Format format = mOutputSink->format(); 2107 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2108 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2109 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2110 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2111 const NBAIO_Format offers[1] = {format}; 2112 size_t numCounterOffers = 0; 2113 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2114 ALOG_ASSERT(index == 0); 2115 monoPipe->setAvgFrames((mScreenState & 1) ? 2116 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2117 mPipeSink = monoPipe; 2118 2119#ifdef TEE_SINK 2120 if (mTeeSinkOutputEnabled) { 2121 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2122 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2123 numCounterOffers = 0; 2124 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2125 ALOG_ASSERT(index == 0); 2126 mTeeSink = teeSink; 2127 PipeReader *teeSource = new PipeReader(*teeSink); 2128 numCounterOffers = 0; 2129 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2130 ALOG_ASSERT(index == 0); 2131 mTeeSource = teeSource; 2132 } 2133#endif 2134 2135 // create fast mixer and configure it initially with just one fast track for our submix 2136 mFastMixer = new FastMixer(); 2137 FastMixerStateQueue *sq = mFastMixer->sq(); 2138#ifdef STATE_QUEUE_DUMP 2139 sq->setObserverDump(&mStateQueueObserverDump); 2140 sq->setMutatorDump(&mStateQueueMutatorDump); 2141#endif 2142 FastMixerState *state = sq->begin(); 2143 FastTrack *fastTrack = &state->mFastTracks[0]; 2144 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2145 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2146 fastTrack->mVolumeProvider = NULL; 2147 fastTrack->mGeneration++; 2148 state->mFastTracksGen++; 2149 state->mTrackMask = 1; 2150 // fast mixer will use the HAL output sink 2151 state->mOutputSink = mOutputSink.get(); 2152 state->mOutputSinkGen++; 2153 state->mFrameCount = mFrameCount; 2154 state->mCommand = FastMixerState::COLD_IDLE; 2155 // already done in constructor initialization list 2156 //mFastMixerFutex = 0; 2157 state->mColdFutexAddr = &mFastMixerFutex; 2158 state->mColdGen++; 2159 state->mDumpState = &mFastMixerDumpState; 2160#ifdef TEE_SINK 2161 state->mTeeSink = mTeeSink.get(); 2162#endif 2163 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2164 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2165 sq->end(); 2166 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2167 2168 // start the fast mixer 2169 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2170 pid_t tid = mFastMixer->getTid(); 2171 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2172 if (err != 0) { 2173 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2174 kPriorityFastMixer, getpid_cached, tid, err); 2175 } 2176 2177#ifdef AUDIO_WATCHDOG 2178 // create and start the watchdog 2179 mAudioWatchdog = new AudioWatchdog(); 2180 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2181 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2182 tid = mAudioWatchdog->getTid(); 2183 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2184 if (err != 0) { 2185 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2186 kPriorityFastMixer, getpid_cached, tid, err); 2187 } 2188#endif 2189 2190 } else { 2191 mFastMixer = NULL; 2192 } 2193 2194 switch (kUseFastMixer) { 2195 case FastMixer_Never: 2196 case FastMixer_Dynamic: 2197 mNormalSink = mOutputSink; 2198 break; 2199 case FastMixer_Always: 2200 mNormalSink = mPipeSink; 2201 break; 2202 case FastMixer_Static: 2203 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2204 break; 2205 } 2206} 2207 2208AudioFlinger::MixerThread::~MixerThread() 2209{ 2210 if (mFastMixer != NULL) { 2211 FastMixerStateQueue *sq = mFastMixer->sq(); 2212 FastMixerState *state = sq->begin(); 2213 if (state->mCommand == FastMixerState::COLD_IDLE) { 2214 int32_t old = android_atomic_inc(&mFastMixerFutex); 2215 if (old == -1) { 2216 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2217 } 2218 } 2219 state->mCommand = FastMixerState::EXIT; 2220 sq->end(); 2221 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2222 mFastMixer->join(); 2223 // Though the fast mixer thread has exited, it's state queue is still valid. 2224 // We'll use that extract the final state which contains one remaining fast track 2225 // corresponding to our sub-mix. 2226 state = sq->begin(); 2227 ALOG_ASSERT(state->mTrackMask == 1); 2228 FastTrack *fastTrack = &state->mFastTracks[0]; 2229 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2230 delete fastTrack->mBufferProvider; 2231 sq->end(false /*didModify*/); 2232 delete mFastMixer; 2233#ifdef AUDIO_WATCHDOG 2234 if (mAudioWatchdog != 0) { 2235 mAudioWatchdog->requestExit(); 2236 mAudioWatchdog->requestExitAndWait(); 2237 mAudioWatchdog.clear(); 2238 } 2239#endif 2240 } 2241 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2242 delete mAudioMixer; 2243} 2244 2245 2246uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2247{ 2248 if (mFastMixer != NULL) { 2249 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2250 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2251 } 2252 return latency; 2253} 2254 2255 2256void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2257{ 2258 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2259} 2260 2261void AudioFlinger::MixerThread::threadLoop_write() 2262{ 2263 // FIXME we should only do one push per cycle; confirm this is true 2264 // Start the fast mixer if it's not already running 2265 if (mFastMixer != NULL) { 2266 FastMixerStateQueue *sq = mFastMixer->sq(); 2267 FastMixerState *state = sq->begin(); 2268 if (state->mCommand != FastMixerState::MIX_WRITE && 2269 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2270 if (state->mCommand == FastMixerState::COLD_IDLE) { 2271 int32_t old = android_atomic_inc(&mFastMixerFutex); 2272 if (old == -1) { 2273 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2274 } 2275#ifdef AUDIO_WATCHDOG 2276 if (mAudioWatchdog != 0) { 2277 mAudioWatchdog->resume(); 2278 } 2279#endif 2280 } 2281 state->mCommand = FastMixerState::MIX_WRITE; 2282 sq->end(); 2283 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2284 if (kUseFastMixer == FastMixer_Dynamic) { 2285 mNormalSink = mPipeSink; 2286 } 2287 } else { 2288 sq->end(false /*didModify*/); 2289 } 2290 } 2291 PlaybackThread::threadLoop_write(); 2292} 2293 2294void AudioFlinger::MixerThread::threadLoop_standby() 2295{ 2296 // Idle the fast mixer if it's currently running 2297 if (mFastMixer != NULL) { 2298 FastMixerStateQueue *sq = mFastMixer->sq(); 2299 FastMixerState *state = sq->begin(); 2300 if (!(state->mCommand & FastMixerState::IDLE)) { 2301 state->mCommand = FastMixerState::COLD_IDLE; 2302 state->mColdFutexAddr = &mFastMixerFutex; 2303 state->mColdGen++; 2304 mFastMixerFutex = 0; 2305 sq->end(); 2306 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2307 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2308 if (kUseFastMixer == FastMixer_Dynamic) { 2309 mNormalSink = mOutputSink; 2310 } 2311#ifdef AUDIO_WATCHDOG 2312 if (mAudioWatchdog != 0) { 2313 mAudioWatchdog->pause(); 2314 } 2315#endif 2316 } else { 2317 sq->end(false /*didModify*/); 2318 } 2319 } 2320 PlaybackThread::threadLoop_standby(); 2321} 2322 2323// shared by MIXER and DIRECT, overridden by DUPLICATING 2324void AudioFlinger::PlaybackThread::threadLoop_standby() 2325{ 2326 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2327 mOutput->stream->common.standby(&mOutput->stream->common); 2328} 2329 2330void AudioFlinger::MixerThread::threadLoop_mix() 2331{ 2332 // obtain the presentation timestamp of the next output buffer 2333 int64_t pts; 2334 status_t status = INVALID_OPERATION; 2335 2336 if (mNormalSink != 0) { 2337 status = mNormalSink->getNextWriteTimestamp(&pts); 2338 } else { 2339 status = mOutputSink->getNextWriteTimestamp(&pts); 2340 } 2341 2342 if (status != NO_ERROR) { 2343 pts = AudioBufferProvider::kInvalidPTS; 2344 } 2345 2346 // mix buffers... 2347 mAudioMixer->process(pts); 2348 // increase sleep time progressively when application underrun condition clears. 2349 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2350 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2351 // such that we would underrun the audio HAL. 2352 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2353 sleepTimeShift--; 2354 } 2355 sleepTime = 0; 2356 standbyTime = systemTime() + standbyDelay; 2357 //TODO: delay standby when effects have a tail 2358} 2359 2360void AudioFlinger::MixerThread::threadLoop_sleepTime() 2361{ 2362 // If no tracks are ready, sleep once for the duration of an output 2363 // buffer size, then write 0s to the output 2364 if (sleepTime == 0) { 2365 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2366 sleepTime = activeSleepTime >> sleepTimeShift; 2367 if (sleepTime < kMinThreadSleepTimeUs) { 2368 sleepTime = kMinThreadSleepTimeUs; 2369 } 2370 // reduce sleep time in case of consecutive application underruns to avoid 2371 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2372 // duration we would end up writing less data than needed by the audio HAL if 2373 // the condition persists. 2374 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2375 sleepTimeShift++; 2376 } 2377 } else { 2378 sleepTime = idleSleepTime; 2379 } 2380 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2381 memset (mMixBuffer, 0, mixBufferSize); 2382 sleepTime = 0; 2383 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2384 "anticipated start"); 2385 } 2386 // TODO add standby time extension fct of effect tail 2387} 2388 2389// prepareTracks_l() must be called with ThreadBase::mLock held 2390AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2391 Vector< sp<Track> > *tracksToRemove) 2392{ 2393 2394 mixer_state mixerStatus = MIXER_IDLE; 2395 // find out which tracks need to be processed 2396 size_t count = mActiveTracks.size(); 2397 size_t mixedTracks = 0; 2398 size_t tracksWithEffect = 0; 2399 // counts only _active_ fast tracks 2400 size_t fastTracks = 0; 2401 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2402 2403 float masterVolume = mMasterVolume; 2404 bool masterMute = mMasterMute; 2405 2406 if (masterMute) { 2407 masterVolume = 0; 2408 } 2409 // Delegate master volume control to effect in output mix effect chain if needed 2410 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2411 if (chain != 0) { 2412 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2413 chain->setVolume_l(&v, &v); 2414 masterVolume = (float)((v + (1 << 23)) >> 24); 2415 chain.clear(); 2416 } 2417 2418 // prepare a new state to push 2419 FastMixerStateQueue *sq = NULL; 2420 FastMixerState *state = NULL; 2421 bool didModify = false; 2422 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2423 if (mFastMixer != NULL) { 2424 sq = mFastMixer->sq(); 2425 state = sq->begin(); 2426 } 2427 2428 for (size_t i=0 ; i<count ; i++) { 2429 sp<Track> t = mActiveTracks[i].promote(); 2430 if (t == 0) { 2431 continue; 2432 } 2433 2434 // this const just means the local variable doesn't change 2435 Track* const track = t.get(); 2436 2437 // process fast tracks 2438 if (track->isFastTrack()) { 2439 2440 // It's theoretically possible (though unlikely) for a fast track to be created 2441 // and then removed within the same normal mix cycle. This is not a problem, as 2442 // the track never becomes active so it's fast mixer slot is never touched. 2443 // The converse, of removing an (active) track and then creating a new track 2444 // at the identical fast mixer slot within the same normal mix cycle, 2445 // is impossible because the slot isn't marked available until the end of each cycle. 2446 int j = track->mFastIndex; 2447 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2448 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2449 FastTrack *fastTrack = &state->mFastTracks[j]; 2450 2451 // Determine whether the track is currently in underrun condition, 2452 // and whether it had a recent underrun. 2453 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2454 FastTrackUnderruns underruns = ftDump->mUnderruns; 2455 uint32_t recentFull = (underruns.mBitFields.mFull - 2456 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2457 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2458 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2459 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2460 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2461 uint32_t recentUnderruns = recentPartial + recentEmpty; 2462 track->mObservedUnderruns = underruns; 2463 // don't count underruns that occur while stopping or pausing 2464 // or stopped which can occur when flush() is called while active 2465 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2466 track->mUnderrunCount += recentUnderruns; 2467 } 2468 2469 // This is similar to the state machine for normal tracks, 2470 // with a few modifications for fast tracks. 2471 bool isActive = true; 2472 switch (track->mState) { 2473 case TrackBase::STOPPING_1: 2474 // track stays active in STOPPING_1 state until first underrun 2475 if (recentUnderruns > 0) { 2476 track->mState = TrackBase::STOPPING_2; 2477 } 2478 break; 2479 case TrackBase::PAUSING: 2480 // ramp down is not yet implemented 2481 track->setPaused(); 2482 break; 2483 case TrackBase::RESUMING: 2484 // ramp up is not yet implemented 2485 track->mState = TrackBase::ACTIVE; 2486 break; 2487 case TrackBase::ACTIVE: 2488 if (recentFull > 0 || recentPartial > 0) { 2489 // track has provided at least some frames recently: reset retry count 2490 track->mRetryCount = kMaxTrackRetries; 2491 } 2492 if (recentUnderruns == 0) { 2493 // no recent underruns: stay active 2494 break; 2495 } 2496 // there has recently been an underrun of some kind 2497 if (track->sharedBuffer() == 0) { 2498 // were any of the recent underruns "empty" (no frames available)? 2499 if (recentEmpty == 0) { 2500 // no, then ignore the partial underruns as they are allowed indefinitely 2501 break; 2502 } 2503 // there has recently been an "empty" underrun: decrement the retry counter 2504 if (--(track->mRetryCount) > 0) { 2505 break; 2506 } 2507 // indicate to client process that the track was disabled because of underrun; 2508 // it will then automatically call start() when data is available 2509 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2510 // remove from active list, but state remains ACTIVE [confusing but true] 2511 isActive = false; 2512 break; 2513 } 2514 // fall through 2515 case TrackBase::STOPPING_2: 2516 case TrackBase::PAUSED: 2517 case TrackBase::TERMINATED: 2518 case TrackBase::STOPPED: 2519 case TrackBase::FLUSHED: // flush() while active 2520 // Check for presentation complete if track is inactive 2521 // We have consumed all the buffers of this track. 2522 // This would be incomplete if we auto-paused on underrun 2523 { 2524 size_t audioHALFrames = 2525 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2526 size_t framesWritten = mBytesWritten / mFrameSize; 2527 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2528 // track stays in active list until presentation is complete 2529 break; 2530 } 2531 } 2532 if (track->isStopping_2()) { 2533 track->mState = TrackBase::STOPPED; 2534 } 2535 if (track->isStopped()) { 2536 // Can't reset directly, as fast mixer is still polling this track 2537 // track->reset(); 2538 // So instead mark this track as needing to be reset after push with ack 2539 resetMask |= 1 << i; 2540 } 2541 isActive = false; 2542 break; 2543 case TrackBase::IDLE: 2544 default: 2545 LOG_FATAL("unexpected track state %d", track->mState); 2546 } 2547 2548 if (isActive) { 2549 // was it previously inactive? 2550 if (!(state->mTrackMask & (1 << j))) { 2551 ExtendedAudioBufferProvider *eabp = track; 2552 VolumeProvider *vp = track; 2553 fastTrack->mBufferProvider = eabp; 2554 fastTrack->mVolumeProvider = vp; 2555 fastTrack->mSampleRate = track->mSampleRate; 2556 fastTrack->mChannelMask = track->mChannelMask; 2557 fastTrack->mGeneration++; 2558 state->mTrackMask |= 1 << j; 2559 didModify = true; 2560 // no acknowledgement required for newly active tracks 2561 } 2562 // cache the combined master volume and stream type volume for fast mixer; this 2563 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2564 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2565 ++fastTracks; 2566 } else { 2567 // was it previously active? 2568 if (state->mTrackMask & (1 << j)) { 2569 fastTrack->mBufferProvider = NULL; 2570 fastTrack->mGeneration++; 2571 state->mTrackMask &= ~(1 << j); 2572 didModify = true; 2573 // If any fast tracks were removed, we must wait for acknowledgement 2574 // because we're about to decrement the last sp<> on those tracks. 2575 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2576 } else { 2577 LOG_FATAL("fast track %d should have been active", j); 2578 } 2579 tracksToRemove->add(track); 2580 // Avoids a misleading display in dumpsys 2581 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2582 } 2583 continue; 2584 } 2585 2586 { // local variable scope to avoid goto warning 2587 2588 audio_track_cblk_t* cblk = track->cblk(); 2589 2590 // The first time a track is added we wait 2591 // for all its buffers to be filled before processing it 2592 int name = track->name(); 2593 // make sure that we have enough frames to mix one full buffer. 2594 // enforce this condition only once to enable draining the buffer in case the client 2595 // app does not call stop() and relies on underrun to stop: 2596 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2597 // during last round 2598 uint32_t minFrames = 1; 2599 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2600 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2601 if (t->sampleRate() == mSampleRate) { 2602 minFrames = mNormalFrameCount; 2603 } else { 2604 // +1 for rounding and +1 for additional sample needed for interpolation 2605 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2606 // add frames already consumed but not yet released by the resampler 2607 // because cblk->framesReady() will include these frames 2608 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2609 // the minimum track buffer size is normally twice the number of frames necessary 2610 // to fill one buffer and the resampler should not leave more than one buffer worth 2611 // of unreleased frames after each pass, but just in case... 2612 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2613 } 2614 } 2615 if ((track->framesReady() >= minFrames) && track->isReady() && 2616 !track->isPaused() && !track->isTerminated()) 2617 { 2618 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2619 this); 2620 2621 mixedTracks++; 2622 2623 // track->mainBuffer() != mMixBuffer means there is an effect chain 2624 // connected to the track 2625 chain.clear(); 2626 if (track->mainBuffer() != mMixBuffer) { 2627 chain = getEffectChain_l(track->sessionId()); 2628 // Delegate volume control to effect in track effect chain if needed 2629 if (chain != 0) { 2630 tracksWithEffect++; 2631 } else { 2632 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2633 "session %d", 2634 name, track->sessionId()); 2635 } 2636 } 2637 2638 2639 int param = AudioMixer::VOLUME; 2640 if (track->mFillingUpStatus == Track::FS_FILLED) { 2641 // no ramp for the first volume setting 2642 track->mFillingUpStatus = Track::FS_ACTIVE; 2643 if (track->mState == TrackBase::RESUMING) { 2644 track->mState = TrackBase::ACTIVE; 2645 param = AudioMixer::RAMP_VOLUME; 2646 } 2647 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2648 } else if (cblk->server != 0) { 2649 // If the track is stopped before the first frame was mixed, 2650 // do not apply ramp 2651 param = AudioMixer::RAMP_VOLUME; 2652 } 2653 2654 // compute volume for this track 2655 uint32_t vl, vr, va; 2656 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2657 vl = vr = va = 0; 2658 if (track->isPausing()) { 2659 track->setPaused(); 2660 } 2661 } else { 2662 2663 // read original volumes with volume control 2664 float typeVolume = mStreamTypes[track->streamType()].volume; 2665 float v = masterVolume * typeVolume; 2666 ServerProxy *proxy = track->mServerProxy; 2667 uint32_t vlr = proxy->getVolumeLR(); 2668 vl = vlr & 0xFFFF; 2669 vr = vlr >> 16; 2670 // track volumes come from shared memory, so can't be trusted and must be clamped 2671 if (vl > MAX_GAIN_INT) { 2672 ALOGV("Track left volume out of range: %04X", vl); 2673 vl = MAX_GAIN_INT; 2674 } 2675 if (vr > MAX_GAIN_INT) { 2676 ALOGV("Track right volume out of range: %04X", vr); 2677 vr = MAX_GAIN_INT; 2678 } 2679 // now apply the master volume and stream type volume 2680 vl = (uint32_t)(v * vl) << 12; 2681 vr = (uint32_t)(v * vr) << 12; 2682 // assuming master volume and stream type volume each go up to 1.0, 2683 // vl and vr are now in 8.24 format 2684 2685 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2686 // send level comes from shared memory and so may be corrupt 2687 if (sendLevel > MAX_GAIN_INT) { 2688 ALOGV("Track send level out of range: %04X", sendLevel); 2689 sendLevel = MAX_GAIN_INT; 2690 } 2691 va = (uint32_t)(v * sendLevel); 2692 } 2693 // Delegate volume control to effect in track effect chain if needed 2694 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2695 // Do not ramp volume if volume is controlled by effect 2696 param = AudioMixer::VOLUME; 2697 track->mHasVolumeController = true; 2698 } else { 2699 // force no volume ramp when volume controller was just disabled or removed 2700 // from effect chain to avoid volume spike 2701 if (track->mHasVolumeController) { 2702 param = AudioMixer::VOLUME; 2703 } 2704 track->mHasVolumeController = false; 2705 } 2706 2707 // Convert volumes from 8.24 to 4.12 format 2708 // This additional clamping is needed in case chain->setVolume_l() overshot 2709 vl = (vl + (1 << 11)) >> 12; 2710 if (vl > MAX_GAIN_INT) { 2711 vl = MAX_GAIN_INT; 2712 } 2713 vr = (vr + (1 << 11)) >> 12; 2714 if (vr > MAX_GAIN_INT) { 2715 vr = MAX_GAIN_INT; 2716 } 2717 2718 if (va > MAX_GAIN_INT) { 2719 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2720 } 2721 2722 // XXX: these things DON'T need to be done each time 2723 mAudioMixer->setBufferProvider(name, track); 2724 mAudioMixer->enable(name); 2725 2726 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2727 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2728 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2729 mAudioMixer->setParameter( 2730 name, 2731 AudioMixer::TRACK, 2732 AudioMixer::FORMAT, (void *)track->format()); 2733 mAudioMixer->setParameter( 2734 name, 2735 AudioMixer::TRACK, 2736 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2737 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2738 uint32_t maxSampleRate = mSampleRate * 2; 2739 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2740 if (reqSampleRate == 0) { 2741 reqSampleRate = mSampleRate; 2742 } else if (reqSampleRate > maxSampleRate) { 2743 reqSampleRate = maxSampleRate; 2744 } 2745 mAudioMixer->setParameter( 2746 name, 2747 AudioMixer::RESAMPLE, 2748 AudioMixer::SAMPLE_RATE, 2749 (void *)reqSampleRate); 2750 mAudioMixer->setParameter( 2751 name, 2752 AudioMixer::TRACK, 2753 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2754 mAudioMixer->setParameter( 2755 name, 2756 AudioMixer::TRACK, 2757 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2758 2759 // reset retry count 2760 track->mRetryCount = kMaxTrackRetries; 2761 2762 // If one track is ready, set the mixer ready if: 2763 // - the mixer was not ready during previous round OR 2764 // - no other track is not ready 2765 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2766 mixerStatus != MIXER_TRACKS_ENABLED) { 2767 mixerStatus = MIXER_TRACKS_READY; 2768 } 2769 } else { 2770 // clear effect chain input buffer if an active track underruns to avoid sending 2771 // previous audio buffer again to effects 2772 chain = getEffectChain_l(track->sessionId()); 2773 if (chain != 0) { 2774 chain->clearInputBuffer(); 2775 } 2776 2777 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2778 cblk->server, this); 2779 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2780 track->isStopped() || track->isPaused()) { 2781 // We have consumed all the buffers of this track. 2782 // Remove it from the list of active tracks. 2783 // TODO: use actual buffer filling status instead of latency when available from 2784 // audio HAL 2785 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2786 size_t framesWritten = mBytesWritten / mFrameSize; 2787 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2788 if (track->isStopped()) { 2789 track->reset(); 2790 } 2791 tracksToRemove->add(track); 2792 } 2793 } else { 2794 track->mUnderrunCount++; 2795 // No buffers for this track. Give it a few chances to 2796 // fill a buffer, then remove it from active list. 2797 if (--(track->mRetryCount) <= 0) { 2798 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2799 tracksToRemove->add(track); 2800 // indicate to client process that the track was disabled because of underrun; 2801 // it will then automatically call start() when data is available 2802 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2803 // If one track is not ready, mark the mixer also not ready if: 2804 // - the mixer was ready during previous round OR 2805 // - no other track is ready 2806 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2807 mixerStatus != MIXER_TRACKS_READY) { 2808 mixerStatus = MIXER_TRACKS_ENABLED; 2809 } 2810 } 2811 mAudioMixer->disable(name); 2812 } 2813 2814 } // local variable scope to avoid goto warning 2815track_is_ready: ; 2816 2817 } 2818 2819 // Push the new FastMixer state if necessary 2820 bool pauseAudioWatchdog = false; 2821 if (didModify) { 2822 state->mFastTracksGen++; 2823 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2824 if (kUseFastMixer == FastMixer_Dynamic && 2825 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2826 state->mCommand = FastMixerState::COLD_IDLE; 2827 state->mColdFutexAddr = &mFastMixerFutex; 2828 state->mColdGen++; 2829 mFastMixerFutex = 0; 2830 if (kUseFastMixer == FastMixer_Dynamic) { 2831 mNormalSink = mOutputSink; 2832 } 2833 // If we go into cold idle, need to wait for acknowledgement 2834 // so that fast mixer stops doing I/O. 2835 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2836 pauseAudioWatchdog = true; 2837 } 2838 } 2839 if (sq != NULL) { 2840 sq->end(didModify); 2841 sq->push(block); 2842 } 2843#ifdef AUDIO_WATCHDOG 2844 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2845 mAudioWatchdog->pause(); 2846 } 2847#endif 2848 2849 // Now perform the deferred reset on fast tracks that have stopped 2850 while (resetMask != 0) { 2851 size_t i = __builtin_ctz(resetMask); 2852 ALOG_ASSERT(i < count); 2853 resetMask &= ~(1 << i); 2854 sp<Track> t = mActiveTracks[i].promote(); 2855 if (t == 0) { 2856 continue; 2857 } 2858 Track* track = t.get(); 2859 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2860 track->reset(); 2861 } 2862 2863 // remove all the tracks that need to be... 2864 count = tracksToRemove->size(); 2865 if (CC_UNLIKELY(count)) { 2866 for (size_t i=0 ; i<count ; i++) { 2867 const sp<Track>& track = tracksToRemove->itemAt(i); 2868 mActiveTracks.remove(track); 2869 if (track->mainBuffer() != mMixBuffer) { 2870 chain = getEffectChain_l(track->sessionId()); 2871 if (chain != 0) { 2872 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2873 track->sessionId()); 2874 chain->decActiveTrackCnt(); 2875 } 2876 } 2877 if (track->isTerminated()) { 2878 removeTrack_l(track); 2879 } 2880 } 2881 } 2882 2883 // mix buffer must be cleared if all tracks are connected to an 2884 // effect chain as in this case the mixer will not write to 2885 // mix buffer and track effects will accumulate into it 2886 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2887 (mixedTracks == 0 && fastTracks > 0)) { 2888 // FIXME as a performance optimization, should remember previous zero status 2889 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2890 } 2891 2892 // if any fast tracks, then status is ready 2893 mMixerStatusIgnoringFastTracks = mixerStatus; 2894 if (fastTracks > 0) { 2895 mixerStatus = MIXER_TRACKS_READY; 2896 } 2897 return mixerStatus; 2898} 2899 2900// getTrackName_l() must be called with ThreadBase::mLock held 2901int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2902{ 2903 return mAudioMixer->getTrackName(channelMask, sessionId); 2904} 2905 2906// deleteTrackName_l() must be called with ThreadBase::mLock held 2907void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2908{ 2909 ALOGV("remove track (%d) and delete from mixer", name); 2910 mAudioMixer->deleteTrackName(name); 2911} 2912 2913// checkForNewParameters_l() must be called with ThreadBase::mLock held 2914bool AudioFlinger::MixerThread::checkForNewParameters_l() 2915{ 2916 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2917 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2918 bool reconfig = false; 2919 2920 while (!mNewParameters.isEmpty()) { 2921 2922 if (mFastMixer != NULL) { 2923 FastMixerStateQueue *sq = mFastMixer->sq(); 2924 FastMixerState *state = sq->begin(); 2925 if (!(state->mCommand & FastMixerState::IDLE)) { 2926 previousCommand = state->mCommand; 2927 state->mCommand = FastMixerState::HOT_IDLE; 2928 sq->end(); 2929 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2930 } else { 2931 sq->end(false /*didModify*/); 2932 } 2933 } 2934 2935 status_t status = NO_ERROR; 2936 String8 keyValuePair = mNewParameters[0]; 2937 AudioParameter param = AudioParameter(keyValuePair); 2938 int value; 2939 2940 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2941 reconfig = true; 2942 } 2943 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2944 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2945 status = BAD_VALUE; 2946 } else { 2947 reconfig = true; 2948 } 2949 } 2950 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2951 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2952 status = BAD_VALUE; 2953 } else { 2954 reconfig = true; 2955 } 2956 } 2957 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2958 // do not accept frame count changes if tracks are open as the track buffer 2959 // size depends on frame count and correct behavior would not be guaranteed 2960 // if frame count is changed after track creation 2961 if (!mTracks.isEmpty()) { 2962 status = INVALID_OPERATION; 2963 } else { 2964 reconfig = true; 2965 } 2966 } 2967 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2968#ifdef ADD_BATTERY_DATA 2969 // when changing the audio output device, call addBatteryData to notify 2970 // the change 2971 if (mOutDevice != value) { 2972 uint32_t params = 0; 2973 // check whether speaker is on 2974 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2975 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2976 } 2977 2978 audio_devices_t deviceWithoutSpeaker 2979 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2980 // check if any other device (except speaker) is on 2981 if (value & deviceWithoutSpeaker ) { 2982 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2983 } 2984 2985 if (params != 0) { 2986 addBatteryData(params); 2987 } 2988 } 2989#endif 2990 2991 // forward device change to effects that have requested to be 2992 // aware of attached audio device. 2993 mOutDevice = value; 2994 for (size_t i = 0; i < mEffectChains.size(); i++) { 2995 mEffectChains[i]->setDevice_l(mOutDevice); 2996 } 2997 } 2998 2999 if (status == NO_ERROR) { 3000 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3001 keyValuePair.string()); 3002 if (!mStandby && status == INVALID_OPERATION) { 3003 mOutput->stream->common.standby(&mOutput->stream->common); 3004 mStandby = true; 3005 mBytesWritten = 0; 3006 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3007 keyValuePair.string()); 3008 } 3009 if (status == NO_ERROR && reconfig) { 3010 delete mAudioMixer; 3011 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3012 mAudioMixer = NULL; 3013 readOutputParameters(); 3014 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3015 for (size_t i = 0; i < mTracks.size() ; i++) { 3016 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3017 if (name < 0) { 3018 break; 3019 } 3020 mTracks[i]->mName = name; 3021 } 3022 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3023 } 3024 } 3025 3026 mNewParameters.removeAt(0); 3027 3028 mParamStatus = status; 3029 mParamCond.signal(); 3030 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3031 // already timed out waiting for the status and will never signal the condition. 3032 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3033 } 3034 3035 if (!(previousCommand & FastMixerState::IDLE)) { 3036 ALOG_ASSERT(mFastMixer != NULL); 3037 FastMixerStateQueue *sq = mFastMixer->sq(); 3038 FastMixerState *state = sq->begin(); 3039 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3040 state->mCommand = previousCommand; 3041 sq->end(); 3042 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3043 } 3044 3045 return reconfig; 3046} 3047 3048 3049void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3050{ 3051 const size_t SIZE = 256; 3052 char buffer[SIZE]; 3053 String8 result; 3054 3055 PlaybackThread::dumpInternals(fd, args); 3056 3057 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3058 result.append(buffer); 3059 write(fd, result.string(), result.size()); 3060 3061 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3062 FastMixerDumpState copy = mFastMixerDumpState; 3063 copy.dump(fd); 3064 3065#ifdef STATE_QUEUE_DUMP 3066 // Similar for state queue 3067 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3068 observerCopy.dump(fd); 3069 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3070 mutatorCopy.dump(fd); 3071#endif 3072 3073#ifdef TEE_SINK 3074 // Write the tee output to a .wav file 3075 dumpTee(fd, mTeeSource, mId); 3076#endif 3077 3078#ifdef AUDIO_WATCHDOG 3079 if (mAudioWatchdog != 0) { 3080 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3081 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3082 wdCopy.dump(fd); 3083 } 3084#endif 3085} 3086 3087uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3088{ 3089 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3090} 3091 3092uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3093{ 3094 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3095} 3096 3097void AudioFlinger::MixerThread::cacheParameters_l() 3098{ 3099 PlaybackThread::cacheParameters_l(); 3100 3101 // FIXME: Relaxed timing because of a certain device that can't meet latency 3102 // Should be reduced to 2x after the vendor fixes the driver issue 3103 // increase threshold again due to low power audio mode. The way this warning 3104 // threshold is calculated and its usefulness should be reconsidered anyway. 3105 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3106} 3107 3108// ---------------------------------------------------------------------------- 3109 3110AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3111 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3112 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3113 // mLeftVolFloat, mRightVolFloat 3114{ 3115} 3116 3117AudioFlinger::DirectOutputThread::~DirectOutputThread() 3118{ 3119} 3120 3121AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3122 Vector< sp<Track> > *tracksToRemove 3123) 3124{ 3125 sp<Track> trackToRemove; 3126 3127 mixer_state mixerStatus = MIXER_IDLE; 3128 3129 // find out which tracks need to be processed 3130 if (mActiveTracks.size() != 0) { 3131 sp<Track> t = mActiveTracks[0].promote(); 3132 // The track died recently 3133 if (t == 0) { 3134 return MIXER_IDLE; 3135 } 3136 3137 Track* const track = t.get(); 3138 audio_track_cblk_t* cblk = track->cblk(); 3139 3140 // The first time a track is added we wait 3141 // for all its buffers to be filled before processing it 3142 uint32_t minFrames; 3143 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3144 minFrames = mNormalFrameCount; 3145 } else { 3146 minFrames = 1; 3147 } 3148 if ((track->framesReady() >= minFrames) && track->isReady() && 3149 !track->isPaused() && !track->isTerminated()) 3150 { 3151 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3152 3153 if (track->mFillingUpStatus == Track::FS_FILLED) { 3154 track->mFillingUpStatus = Track::FS_ACTIVE; 3155 mLeftVolFloat = mRightVolFloat = 0; 3156 if (track->mState == TrackBase::RESUMING) { 3157 track->mState = TrackBase::ACTIVE; 3158 } 3159 } 3160 3161 // compute volume for this track 3162 float left, right; 3163 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3164 left = right = 0; 3165 if (track->isPausing()) { 3166 track->setPaused(); 3167 } 3168 } else { 3169 float typeVolume = mStreamTypes[track->streamType()].volume; 3170 float v = mMasterVolume * typeVolume; 3171 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3172 float v_clamped = v * (vlr & 0xFFFF); 3173 if (v_clamped > MAX_GAIN) { 3174 v_clamped = MAX_GAIN; 3175 } 3176 left = v_clamped/MAX_GAIN; 3177 v_clamped = v * (vlr >> 16); 3178 if (v_clamped > MAX_GAIN) { 3179 v_clamped = MAX_GAIN; 3180 } 3181 right = v_clamped/MAX_GAIN; 3182 } 3183 3184 if (left != mLeftVolFloat || right != mRightVolFloat) { 3185 mLeftVolFloat = left; 3186 mRightVolFloat = right; 3187 3188 // Convert volumes from float to 8.24 3189 uint32_t vl = (uint32_t)(left * (1 << 24)); 3190 uint32_t vr = (uint32_t)(right * (1 << 24)); 3191 3192 // Delegate volume control to effect in track effect chain if needed 3193 // only one effect chain can be present on DirectOutputThread, so if 3194 // there is one, the track is connected to it 3195 if (!mEffectChains.isEmpty()) { 3196 // Do not ramp volume if volume is controlled by effect 3197 mEffectChains[0]->setVolume_l(&vl, &vr); 3198 left = (float)vl / (1 << 24); 3199 right = (float)vr / (1 << 24); 3200 } 3201 mOutput->stream->set_volume(mOutput->stream, left, right); 3202 } 3203 3204 // reset retry count 3205 track->mRetryCount = kMaxTrackRetriesDirect; 3206 mActiveTrack = t; 3207 mixerStatus = MIXER_TRACKS_READY; 3208 } else { 3209 // clear effect chain input buffer if an active track underruns to avoid sending 3210 // previous audio buffer again to effects 3211 if (!mEffectChains.isEmpty()) { 3212 mEffectChains[0]->clearInputBuffer(); 3213 } 3214 3215 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3216 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3217 track->isStopped() || track->isPaused()) { 3218 // We have consumed all the buffers of this track. 3219 // Remove it from the list of active tracks. 3220 // TODO: implement behavior for compressed audio 3221 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3222 size_t framesWritten = mBytesWritten / mFrameSize; 3223 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3224 if (track->isStopped()) { 3225 track->reset(); 3226 } 3227 trackToRemove = track; 3228 } 3229 } else { 3230 // No buffers for this track. Give it a few chances to 3231 // fill a buffer, then remove it from active list. 3232 if (--(track->mRetryCount) <= 0) { 3233 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3234 trackToRemove = track; 3235 } else { 3236 mixerStatus = MIXER_TRACKS_ENABLED; 3237 } 3238 } 3239 } 3240 } 3241 3242 // FIXME merge this with similar code for removing multiple tracks 3243 // remove all the tracks that need to be... 3244 if (CC_UNLIKELY(trackToRemove != 0)) { 3245 tracksToRemove->add(trackToRemove); 3246 mActiveTracks.remove(trackToRemove); 3247 if (!mEffectChains.isEmpty()) { 3248 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3249 trackToRemove->sessionId()); 3250 mEffectChains[0]->decActiveTrackCnt(); 3251 } 3252 if (trackToRemove->isTerminated()) { 3253 removeTrack_l(trackToRemove); 3254 } 3255 } 3256 3257 return mixerStatus; 3258} 3259 3260void AudioFlinger::DirectOutputThread::threadLoop_mix() 3261{ 3262 AudioBufferProvider::Buffer buffer; 3263 size_t frameCount = mFrameCount; 3264 int8_t *curBuf = (int8_t *)mMixBuffer; 3265 // output audio to hardware 3266 while (frameCount) { 3267 buffer.frameCount = frameCount; 3268 mActiveTrack->getNextBuffer(&buffer); 3269 if (CC_UNLIKELY(buffer.raw == NULL)) { 3270 memset(curBuf, 0, frameCount * mFrameSize); 3271 break; 3272 } 3273 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3274 frameCount -= buffer.frameCount; 3275 curBuf += buffer.frameCount * mFrameSize; 3276 mActiveTrack->releaseBuffer(&buffer); 3277 } 3278 sleepTime = 0; 3279 standbyTime = systemTime() + standbyDelay; 3280 mActiveTrack.clear(); 3281 3282} 3283 3284void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3285{ 3286 if (sleepTime == 0) { 3287 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3288 sleepTime = activeSleepTime; 3289 } else { 3290 sleepTime = idleSleepTime; 3291 } 3292 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3293 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3294 sleepTime = 0; 3295 } 3296} 3297 3298// getTrackName_l() must be called with ThreadBase::mLock held 3299int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3300 int sessionId) 3301{ 3302 return 0; 3303} 3304 3305// deleteTrackName_l() must be called with ThreadBase::mLock held 3306void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3307{ 3308} 3309 3310// checkForNewParameters_l() must be called with ThreadBase::mLock held 3311bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3312{ 3313 bool reconfig = false; 3314 3315 while (!mNewParameters.isEmpty()) { 3316 status_t status = NO_ERROR; 3317 String8 keyValuePair = mNewParameters[0]; 3318 AudioParameter param = AudioParameter(keyValuePair); 3319 int value; 3320 3321 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3322 // do not accept frame count changes if tracks are open as the track buffer 3323 // size depends on frame count and correct behavior would not be garantied 3324 // if frame count is changed after track creation 3325 if (!mTracks.isEmpty()) { 3326 status = INVALID_OPERATION; 3327 } else { 3328 reconfig = true; 3329 } 3330 } 3331 if (status == NO_ERROR) { 3332 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3333 keyValuePair.string()); 3334 if (!mStandby && status == INVALID_OPERATION) { 3335 mOutput->stream->common.standby(&mOutput->stream->common); 3336 mStandby = true; 3337 mBytesWritten = 0; 3338 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3339 keyValuePair.string()); 3340 } 3341 if (status == NO_ERROR && reconfig) { 3342 readOutputParameters(); 3343 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3344 } 3345 } 3346 3347 mNewParameters.removeAt(0); 3348 3349 mParamStatus = status; 3350 mParamCond.signal(); 3351 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3352 // already timed out waiting for the status and will never signal the condition. 3353 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3354 } 3355 return reconfig; 3356} 3357 3358uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3359{ 3360 uint32_t time; 3361 if (audio_is_linear_pcm(mFormat)) { 3362 time = PlaybackThread::activeSleepTimeUs(); 3363 } else { 3364 time = 10000; 3365 } 3366 return time; 3367} 3368 3369uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3370{ 3371 uint32_t time; 3372 if (audio_is_linear_pcm(mFormat)) { 3373 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3374 } else { 3375 time = 10000; 3376 } 3377 return time; 3378} 3379 3380uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3381{ 3382 uint32_t time; 3383 if (audio_is_linear_pcm(mFormat)) { 3384 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3385 } else { 3386 time = 10000; 3387 } 3388 return time; 3389} 3390 3391void AudioFlinger::DirectOutputThread::cacheParameters_l() 3392{ 3393 PlaybackThread::cacheParameters_l(); 3394 3395 // use shorter standby delay as on normal output to release 3396 // hardware resources as soon as possible 3397 standbyDelay = microseconds(activeSleepTime*2); 3398} 3399 3400// ---------------------------------------------------------------------------- 3401 3402AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3403 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3404 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3405 DUPLICATING), 3406 mWaitTimeMs(UINT_MAX) 3407{ 3408 addOutputTrack(mainThread); 3409} 3410 3411AudioFlinger::DuplicatingThread::~DuplicatingThread() 3412{ 3413 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3414 mOutputTracks[i]->destroy(); 3415 } 3416} 3417 3418void AudioFlinger::DuplicatingThread::threadLoop_mix() 3419{ 3420 // mix buffers... 3421 if (outputsReady(outputTracks)) { 3422 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3423 } else { 3424 memset(mMixBuffer, 0, mixBufferSize); 3425 } 3426 sleepTime = 0; 3427 writeFrames = mNormalFrameCount; 3428 standbyTime = systemTime() + standbyDelay; 3429} 3430 3431void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3432{ 3433 if (sleepTime == 0) { 3434 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3435 sleepTime = activeSleepTime; 3436 } else { 3437 sleepTime = idleSleepTime; 3438 } 3439 } else if (mBytesWritten != 0) { 3440 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3441 writeFrames = mNormalFrameCount; 3442 memset(mMixBuffer, 0, mixBufferSize); 3443 } else { 3444 // flush remaining overflow buffers in output tracks 3445 writeFrames = 0; 3446 } 3447 sleepTime = 0; 3448 } 3449} 3450 3451void AudioFlinger::DuplicatingThread::threadLoop_write() 3452{ 3453 for (size_t i = 0; i < outputTracks.size(); i++) { 3454 outputTracks[i]->write(mMixBuffer, writeFrames); 3455 } 3456 mBytesWritten += mixBufferSize; 3457} 3458 3459void AudioFlinger::DuplicatingThread::threadLoop_standby() 3460{ 3461 // DuplicatingThread implements standby by stopping all tracks 3462 for (size_t i = 0; i < outputTracks.size(); i++) { 3463 outputTracks[i]->stop(); 3464 } 3465} 3466 3467void AudioFlinger::DuplicatingThread::saveOutputTracks() 3468{ 3469 outputTracks = mOutputTracks; 3470} 3471 3472void AudioFlinger::DuplicatingThread::clearOutputTracks() 3473{ 3474 outputTracks.clear(); 3475} 3476 3477void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3478{ 3479 Mutex::Autolock _l(mLock); 3480 // FIXME explain this formula 3481 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3482 OutputTrack *outputTrack = new OutputTrack(thread, 3483 this, 3484 mSampleRate, 3485 mFormat, 3486 mChannelMask, 3487 frameCount); 3488 if (outputTrack->cblk() != NULL) { 3489 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3490 mOutputTracks.add(outputTrack); 3491 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3492 updateWaitTime_l(); 3493 } 3494} 3495 3496void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3497{ 3498 Mutex::Autolock _l(mLock); 3499 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3500 if (mOutputTracks[i]->thread() == thread) { 3501 mOutputTracks[i]->destroy(); 3502 mOutputTracks.removeAt(i); 3503 updateWaitTime_l(); 3504 return; 3505 } 3506 } 3507 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3508} 3509 3510// caller must hold mLock 3511void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3512{ 3513 mWaitTimeMs = UINT_MAX; 3514 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3515 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3516 if (strong != 0) { 3517 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3518 if (waitTimeMs < mWaitTimeMs) { 3519 mWaitTimeMs = waitTimeMs; 3520 } 3521 } 3522 } 3523} 3524 3525 3526bool AudioFlinger::DuplicatingThread::outputsReady( 3527 const SortedVector< sp<OutputTrack> > &outputTracks) 3528{ 3529 for (size_t i = 0; i < outputTracks.size(); i++) { 3530 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3531 if (thread == 0) { 3532 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3533 outputTracks[i].get()); 3534 return false; 3535 } 3536 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3537 // see note at standby() declaration 3538 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3539 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3540 thread.get()); 3541 return false; 3542 } 3543 } 3544 return true; 3545} 3546 3547uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3548{ 3549 return (mWaitTimeMs * 1000) / 2; 3550} 3551 3552void AudioFlinger::DuplicatingThread::cacheParameters_l() 3553{ 3554 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3555 updateWaitTime_l(); 3556 3557 MixerThread::cacheParameters_l(); 3558} 3559 3560// ---------------------------------------------------------------------------- 3561// Record 3562// ---------------------------------------------------------------------------- 3563 3564AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3565 AudioStreamIn *input, 3566 uint32_t sampleRate, 3567 audio_channel_mask_t channelMask, 3568 audio_io_handle_t id, 3569 audio_devices_t outDevice, 3570 audio_devices_t inDevice 3571#ifdef TEE_SINK 3572 , const sp<NBAIO_Sink>& teeSink 3573#endif 3574 ) : 3575 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3576 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3577 // mRsmpInIndex and mInputBytes set by readInputParameters() 3578 mReqChannelCount(popcount(channelMask)), 3579 mReqSampleRate(sampleRate) 3580 // mBytesRead is only meaningful while active, and so is cleared in start() 3581 // (but might be better to also clear here for dump?) 3582#ifdef TEE_SINK 3583 , mTeeSink(teeSink) 3584#endif 3585{ 3586 snprintf(mName, kNameLength, "AudioIn_%X", id); 3587 3588 readInputParameters(); 3589 3590} 3591 3592 3593AudioFlinger::RecordThread::~RecordThread() 3594{ 3595 delete[] mRsmpInBuffer; 3596 delete mResampler; 3597 delete[] mRsmpOutBuffer; 3598} 3599 3600void AudioFlinger::RecordThread::onFirstRef() 3601{ 3602 run(mName, PRIORITY_URGENT_AUDIO); 3603} 3604 3605status_t AudioFlinger::RecordThread::readyToRun() 3606{ 3607 status_t status = initCheck(); 3608 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3609 return status; 3610} 3611 3612bool AudioFlinger::RecordThread::threadLoop() 3613{ 3614 AudioBufferProvider::Buffer buffer; 3615 sp<RecordTrack> activeTrack; 3616 Vector< sp<EffectChain> > effectChains; 3617 3618 nsecs_t lastWarning = 0; 3619 3620 inputStandBy(); 3621 acquireWakeLock(); 3622 3623 // used to verify we've read at least once before evaluating how many bytes were read 3624 bool readOnce = false; 3625 3626 // start recording 3627 while (!exitPending()) { 3628 3629 processConfigEvents(); 3630 3631 { // scope for mLock 3632 Mutex::Autolock _l(mLock); 3633 checkForNewParameters_l(); 3634 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3635 standby(); 3636 3637 if (exitPending()) { 3638 break; 3639 } 3640 3641 releaseWakeLock_l(); 3642 ALOGV("RecordThread: loop stopping"); 3643 // go to sleep 3644 mWaitWorkCV.wait(mLock); 3645 ALOGV("RecordThread: loop starting"); 3646 acquireWakeLock_l(); 3647 continue; 3648 } 3649 if (mActiveTrack != 0) { 3650 if (mActiveTrack->mState == TrackBase::PAUSING) { 3651 standby(); 3652 mActiveTrack.clear(); 3653 mStartStopCond.broadcast(); 3654 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3655 if (mReqChannelCount != mActiveTrack->channelCount()) { 3656 mActiveTrack.clear(); 3657 mStartStopCond.broadcast(); 3658 } else if (readOnce) { 3659 // record start succeeds only if first read from audio input 3660 // succeeds 3661 if (mBytesRead >= 0) { 3662 mActiveTrack->mState = TrackBase::ACTIVE; 3663 } else { 3664 mActiveTrack.clear(); 3665 } 3666 mStartStopCond.broadcast(); 3667 } 3668 mStandby = false; 3669 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3670 removeTrack_l(mActiveTrack); 3671 mActiveTrack.clear(); 3672 } 3673 } 3674 lockEffectChains_l(effectChains); 3675 } 3676 3677 if (mActiveTrack != 0) { 3678 if (mActiveTrack->mState != TrackBase::ACTIVE && 3679 mActiveTrack->mState != TrackBase::RESUMING) { 3680 unlockEffectChains(effectChains); 3681 usleep(kRecordThreadSleepUs); 3682 continue; 3683 } 3684 for (size_t i = 0; i < effectChains.size(); i ++) { 3685 effectChains[i]->process_l(); 3686 } 3687 3688 buffer.frameCount = mFrameCount; 3689 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3690 readOnce = true; 3691 size_t framesOut = buffer.frameCount; 3692 if (mResampler == NULL) { 3693 // no resampling 3694 while (framesOut) { 3695 size_t framesIn = mFrameCount - mRsmpInIndex; 3696 if (framesIn) { 3697 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3698 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3699 mActiveTrack->mFrameSize; 3700 if (framesIn > framesOut) 3701 framesIn = framesOut; 3702 mRsmpInIndex += framesIn; 3703 framesOut -= framesIn; 3704 if (mChannelCount == mReqChannelCount || 3705 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3706 memcpy(dst, src, framesIn * mFrameSize); 3707 } else { 3708 if (mChannelCount == 1) { 3709 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3710 (int16_t *)src, framesIn); 3711 } else { 3712 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3713 (int16_t *)src, framesIn); 3714 } 3715 } 3716 } 3717 if (framesOut && mFrameCount == mRsmpInIndex) { 3718 void *readInto; 3719 if (framesOut == mFrameCount && 3720 (mChannelCount == mReqChannelCount || 3721 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3722 readInto = buffer.raw; 3723 framesOut = 0; 3724 } else { 3725 readInto = mRsmpInBuffer; 3726 mRsmpInIndex = 0; 3727 } 3728 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3729 if (mBytesRead <= 0) { 3730 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3731 { 3732 ALOGE("Error reading audio input"); 3733 // Force input into standby so that it tries to 3734 // recover at next read attempt 3735 inputStandBy(); 3736 usleep(kRecordThreadSleepUs); 3737 } 3738 mRsmpInIndex = mFrameCount; 3739 framesOut = 0; 3740 buffer.frameCount = 0; 3741 } 3742#ifdef TEE_SINK 3743 else if (mTeeSink != 0) { 3744 (void) mTeeSink->write(readInto, 3745 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3746 } 3747#endif 3748 } 3749 } 3750 } else { 3751 // resampling 3752 3753 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3754 // alter output frame count as if we were expecting stereo samples 3755 if (mChannelCount == 1 && mReqChannelCount == 1) { 3756 framesOut >>= 1; 3757 } 3758 mResampler->resample(mRsmpOutBuffer, framesOut, 3759 this /* AudioBufferProvider* */); 3760 // ditherAndClamp() works as long as all buffers returned by 3761 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3762 if (mChannelCount == 2 && mReqChannelCount == 1) { 3763 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3764 // the resampler always outputs stereo samples: 3765 // do post stereo to mono conversion 3766 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3767 framesOut); 3768 } else { 3769 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3770 } 3771 3772 } 3773 if (mFramestoDrop == 0) { 3774 mActiveTrack->releaseBuffer(&buffer); 3775 } else { 3776 if (mFramestoDrop > 0) { 3777 mFramestoDrop -= buffer.frameCount; 3778 if (mFramestoDrop <= 0) { 3779 clearSyncStartEvent(); 3780 } 3781 } else { 3782 mFramestoDrop += buffer.frameCount; 3783 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3784 mSyncStartEvent->isCancelled()) { 3785 ALOGW("Synced record %s, session %d, trigger session %d", 3786 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3787 mActiveTrack->sessionId(), 3788 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3789 clearSyncStartEvent(); 3790 } 3791 } 3792 } 3793 mActiveTrack->clearOverflow(); 3794 } 3795 // client isn't retrieving buffers fast enough 3796 else { 3797 if (!mActiveTrack->setOverflow()) { 3798 nsecs_t now = systemTime(); 3799 if ((now - lastWarning) > kWarningThrottleNs) { 3800 ALOGW("RecordThread: buffer overflow"); 3801 lastWarning = now; 3802 } 3803 } 3804 // Release the processor for a while before asking for a new buffer. 3805 // This will give the application more chance to read from the buffer and 3806 // clear the overflow. 3807 usleep(kRecordThreadSleepUs); 3808 } 3809 } 3810 // enable changes in effect chain 3811 unlockEffectChains(effectChains); 3812 effectChains.clear(); 3813 } 3814 3815 standby(); 3816 3817 { 3818 Mutex::Autolock _l(mLock); 3819 mActiveTrack.clear(); 3820 mStartStopCond.broadcast(); 3821 } 3822 3823 releaseWakeLock(); 3824 3825 ALOGV("RecordThread %p exiting", this); 3826 return false; 3827} 3828 3829void AudioFlinger::RecordThread::standby() 3830{ 3831 if (!mStandby) { 3832 inputStandBy(); 3833 mStandby = true; 3834 } 3835} 3836 3837void AudioFlinger::RecordThread::inputStandBy() 3838{ 3839 mInput->stream->common.standby(&mInput->stream->common); 3840} 3841 3842sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3843 const sp<AudioFlinger::Client>& client, 3844 uint32_t sampleRate, 3845 audio_format_t format, 3846 audio_channel_mask_t channelMask, 3847 size_t frameCount, 3848 int sessionId, 3849 IAudioFlinger::track_flags_t flags, 3850 pid_t tid, 3851 status_t *status) 3852{ 3853 sp<RecordTrack> track; 3854 status_t lStatus; 3855 3856 lStatus = initCheck(); 3857 if (lStatus != NO_ERROR) { 3858 ALOGE("Audio driver not initialized."); 3859 goto Exit; 3860 } 3861 3862 // FIXME use flags and tid similar to createTrack_l() 3863 3864 { // scope for mLock 3865 Mutex::Autolock _l(mLock); 3866 3867 track = new RecordTrack(this, client, sampleRate, 3868 format, channelMask, frameCount, sessionId); 3869 3870 if (track->getCblk() == 0) { 3871 lStatus = NO_MEMORY; 3872 goto Exit; 3873 } 3874 mTracks.add(track); 3875 3876 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3877 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3878 mAudioFlinger->btNrecIsOff(); 3879 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3880 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3881 } 3882 lStatus = NO_ERROR; 3883 3884Exit: 3885 if (status) { 3886 *status = lStatus; 3887 } 3888 return track; 3889} 3890 3891status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3892 AudioSystem::sync_event_t event, 3893 int triggerSession) 3894{ 3895 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3896 sp<ThreadBase> strongMe = this; 3897 status_t status = NO_ERROR; 3898 3899 if (event == AudioSystem::SYNC_EVENT_NONE) { 3900 clearSyncStartEvent(); 3901 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3902 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3903 triggerSession, 3904 recordTrack->sessionId(), 3905 syncStartEventCallback, 3906 this); 3907 // Sync event can be cancelled by the trigger session if the track is not in a 3908 // compatible state in which case we start record immediately 3909 if (mSyncStartEvent->isCancelled()) { 3910 clearSyncStartEvent(); 3911 } else { 3912 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3913 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3914 } 3915 } 3916 3917 { 3918 AutoMutex lock(mLock); 3919 if (mActiveTrack != 0) { 3920 if (recordTrack != mActiveTrack.get()) { 3921 status = -EBUSY; 3922 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3923 mActiveTrack->mState = TrackBase::ACTIVE; 3924 } 3925 return status; 3926 } 3927 3928 recordTrack->mState = TrackBase::IDLE; 3929 mActiveTrack = recordTrack; 3930 mLock.unlock(); 3931 status_t status = AudioSystem::startInput(mId); 3932 mLock.lock(); 3933 if (status != NO_ERROR) { 3934 mActiveTrack.clear(); 3935 clearSyncStartEvent(); 3936 return status; 3937 } 3938 mRsmpInIndex = mFrameCount; 3939 mBytesRead = 0; 3940 if (mResampler != NULL) { 3941 mResampler->reset(); 3942 } 3943 mActiveTrack->mState = TrackBase::RESUMING; 3944 // signal thread to start 3945 ALOGV("Signal record thread"); 3946 mWaitWorkCV.broadcast(); 3947 // do not wait for mStartStopCond if exiting 3948 if (exitPending()) { 3949 mActiveTrack.clear(); 3950 status = INVALID_OPERATION; 3951 goto startError; 3952 } 3953 mStartStopCond.wait(mLock); 3954 if (mActiveTrack == 0) { 3955 ALOGV("Record failed to start"); 3956 status = BAD_VALUE; 3957 goto startError; 3958 } 3959 ALOGV("Record started OK"); 3960 return status; 3961 } 3962startError: 3963 AudioSystem::stopInput(mId); 3964 clearSyncStartEvent(); 3965 return status; 3966} 3967 3968void AudioFlinger::RecordThread::clearSyncStartEvent() 3969{ 3970 if (mSyncStartEvent != 0) { 3971 mSyncStartEvent->cancel(); 3972 } 3973 mSyncStartEvent.clear(); 3974 mFramestoDrop = 0; 3975} 3976 3977void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3978{ 3979 sp<SyncEvent> strongEvent = event.promote(); 3980 3981 if (strongEvent != 0) { 3982 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3983 me->handleSyncStartEvent(strongEvent); 3984 } 3985} 3986 3987void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 3988{ 3989 if (event == mSyncStartEvent) { 3990 // TODO: use actual buffer filling status instead of 2 buffers when info is available 3991 // from audio HAL 3992 mFramestoDrop = mFrameCount * 2; 3993 } 3994} 3995 3996bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 3997 ALOGV("RecordThread::stop"); 3998 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 3999 return false; 4000 } 4001 recordTrack->mState = TrackBase::PAUSING; 4002 // do not wait for mStartStopCond if exiting 4003 if (exitPending()) { 4004 return true; 4005 } 4006 mStartStopCond.wait(mLock); 4007 // if we have been restarted, recordTrack == mActiveTrack.get() here 4008 if (exitPending() || recordTrack != mActiveTrack.get()) { 4009 ALOGV("Record stopped OK"); 4010 return true; 4011 } 4012 return false; 4013} 4014 4015bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4016{ 4017 return false; 4018} 4019 4020status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4021{ 4022#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4023 if (!isValidSyncEvent(event)) { 4024 return BAD_VALUE; 4025 } 4026 4027 int eventSession = event->triggerSession(); 4028 status_t ret = NAME_NOT_FOUND; 4029 4030 Mutex::Autolock _l(mLock); 4031 4032 for (size_t i = 0; i < mTracks.size(); i++) { 4033 sp<RecordTrack> track = mTracks[i]; 4034 if (eventSession == track->sessionId()) { 4035 (void) track->setSyncEvent(event); 4036 ret = NO_ERROR; 4037 } 4038 } 4039 return ret; 4040#else 4041 return BAD_VALUE; 4042#endif 4043} 4044 4045// destroyTrack_l() must be called with ThreadBase::mLock held 4046void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4047{ 4048 track->mState = TrackBase::TERMINATED; 4049 // active tracks are removed by threadLoop() 4050 if (mActiveTrack != track) { 4051 removeTrack_l(track); 4052 } 4053} 4054 4055void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4056{ 4057 mTracks.remove(track); 4058 // need anything related to effects here? 4059} 4060 4061void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4062{ 4063 dumpInternals(fd, args); 4064 dumpTracks(fd, args); 4065 dumpEffectChains(fd, args); 4066} 4067 4068void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4069{ 4070 const size_t SIZE = 256; 4071 char buffer[SIZE]; 4072 String8 result; 4073 4074 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4075 result.append(buffer); 4076 4077 if (mActiveTrack != 0) { 4078 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4079 result.append(buffer); 4080 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4081 result.append(buffer); 4082 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4083 result.append(buffer); 4084 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4085 result.append(buffer); 4086 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4087 result.append(buffer); 4088 } else { 4089 result.append("No active record client\n"); 4090 } 4091 4092 write(fd, result.string(), result.size()); 4093 4094 dumpBase(fd, args); 4095} 4096 4097void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4098{ 4099 const size_t SIZE = 256; 4100 char buffer[SIZE]; 4101 String8 result; 4102 4103 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4104 result.append(buffer); 4105 RecordTrack::appendDumpHeader(result); 4106 for (size_t i = 0; i < mTracks.size(); ++i) { 4107 sp<RecordTrack> track = mTracks[i]; 4108 if (track != 0) { 4109 track->dump(buffer, SIZE); 4110 result.append(buffer); 4111 } 4112 } 4113 4114 if (mActiveTrack != 0) { 4115 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4116 result.append(buffer); 4117 RecordTrack::appendDumpHeader(result); 4118 mActiveTrack->dump(buffer, SIZE); 4119 result.append(buffer); 4120 4121 } 4122 write(fd, result.string(), result.size()); 4123} 4124 4125// AudioBufferProvider interface 4126status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4127{ 4128 size_t framesReq = buffer->frameCount; 4129 size_t framesReady = mFrameCount - mRsmpInIndex; 4130 int channelCount; 4131 4132 if (framesReady == 0) { 4133 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4134 if (mBytesRead <= 0) { 4135 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4136 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4137 // Force input into standby so that it tries to 4138 // recover at next read attempt 4139 inputStandBy(); 4140 usleep(kRecordThreadSleepUs); 4141 } 4142 buffer->raw = NULL; 4143 buffer->frameCount = 0; 4144 return NOT_ENOUGH_DATA; 4145 } 4146 mRsmpInIndex = 0; 4147 framesReady = mFrameCount; 4148 } 4149 4150 if (framesReq > framesReady) { 4151 framesReq = framesReady; 4152 } 4153 4154 if (mChannelCount == 1 && mReqChannelCount == 2) { 4155 channelCount = 1; 4156 } else { 4157 channelCount = 2; 4158 } 4159 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4160 buffer->frameCount = framesReq; 4161 return NO_ERROR; 4162} 4163 4164// AudioBufferProvider interface 4165void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4166{ 4167 mRsmpInIndex += buffer->frameCount; 4168 buffer->frameCount = 0; 4169} 4170 4171bool AudioFlinger::RecordThread::checkForNewParameters_l() 4172{ 4173 bool reconfig = false; 4174 4175 while (!mNewParameters.isEmpty()) { 4176 status_t status = NO_ERROR; 4177 String8 keyValuePair = mNewParameters[0]; 4178 AudioParameter param = AudioParameter(keyValuePair); 4179 int value; 4180 audio_format_t reqFormat = mFormat; 4181 uint32_t reqSamplingRate = mReqSampleRate; 4182 uint32_t reqChannelCount = mReqChannelCount; 4183 4184 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4185 reqSamplingRate = value; 4186 reconfig = true; 4187 } 4188 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4189 reqFormat = (audio_format_t) value; 4190 reconfig = true; 4191 } 4192 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4193 reqChannelCount = popcount(value); 4194 reconfig = true; 4195 } 4196 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4197 // do not accept frame count changes if tracks are open as the track buffer 4198 // size depends on frame count and correct behavior would not be guaranteed 4199 // if frame count is changed after track creation 4200 if (mActiveTrack != 0) { 4201 status = INVALID_OPERATION; 4202 } else { 4203 reconfig = true; 4204 } 4205 } 4206 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4207 // forward device change to effects that have requested to be 4208 // aware of attached audio device. 4209 for (size_t i = 0; i < mEffectChains.size(); i++) { 4210 mEffectChains[i]->setDevice_l(value); 4211 } 4212 4213 // store input device and output device but do not forward output device to audio HAL. 4214 // Note that status is ignored by the caller for output device 4215 // (see AudioFlinger::setParameters() 4216 if (audio_is_output_devices(value)) { 4217 mOutDevice = value; 4218 status = BAD_VALUE; 4219 } else { 4220 mInDevice = value; 4221 // disable AEC and NS if the device is a BT SCO headset supporting those 4222 // pre processings 4223 if (mTracks.size() > 0) { 4224 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4225 mAudioFlinger->btNrecIsOff(); 4226 for (size_t i = 0; i < mTracks.size(); i++) { 4227 sp<RecordTrack> track = mTracks[i]; 4228 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4229 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4230 } 4231 } 4232 } 4233 } 4234 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4235 mAudioSource != (audio_source_t)value) { 4236 // forward device change to effects that have requested to be 4237 // aware of attached audio device. 4238 for (size_t i = 0; i < mEffectChains.size(); i++) { 4239 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4240 } 4241 mAudioSource = (audio_source_t)value; 4242 } 4243 if (status == NO_ERROR) { 4244 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4245 keyValuePair.string()); 4246 if (status == INVALID_OPERATION) { 4247 inputStandBy(); 4248 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4249 keyValuePair.string()); 4250 } 4251 if (reconfig) { 4252 if (status == BAD_VALUE && 4253 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4254 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4255 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4256 <= (2 * reqSamplingRate)) && 4257 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4258 <= FCC_2 && 4259 (reqChannelCount <= FCC_2)) { 4260 status = NO_ERROR; 4261 } 4262 if (status == NO_ERROR) { 4263 readInputParameters(); 4264 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4265 } 4266 } 4267 } 4268 4269 mNewParameters.removeAt(0); 4270 4271 mParamStatus = status; 4272 mParamCond.signal(); 4273 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4274 // already timed out waiting for the status and will never signal the condition. 4275 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4276 } 4277 return reconfig; 4278} 4279 4280String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4281{ 4282 char *s; 4283 String8 out_s8 = String8(); 4284 4285 Mutex::Autolock _l(mLock); 4286 if (initCheck() != NO_ERROR) { 4287 return out_s8; 4288 } 4289 4290 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4291 out_s8 = String8(s); 4292 free(s); 4293 return out_s8; 4294} 4295 4296void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4297 AudioSystem::OutputDescriptor desc; 4298 void *param2 = NULL; 4299 4300 switch (event) { 4301 case AudioSystem::INPUT_OPENED: 4302 case AudioSystem::INPUT_CONFIG_CHANGED: 4303 desc.channels = mChannelMask; 4304 desc.samplingRate = mSampleRate; 4305 desc.format = mFormat; 4306 desc.frameCount = mFrameCount; 4307 desc.latency = 0; 4308 param2 = &desc; 4309 break; 4310 4311 case AudioSystem::INPUT_CLOSED: 4312 default: 4313 break; 4314 } 4315 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4316} 4317 4318void AudioFlinger::RecordThread::readInputParameters() 4319{ 4320 delete mRsmpInBuffer; 4321 // mRsmpInBuffer is always assigned a new[] below 4322 delete mRsmpOutBuffer; 4323 mRsmpOutBuffer = NULL; 4324 delete mResampler; 4325 mResampler = NULL; 4326 4327 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4328 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4329 mChannelCount = (uint16_t)popcount(mChannelMask); 4330 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4331 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4332 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4333 mFrameCount = mInputBytes / mFrameSize; 4334 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4335 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4336 4337 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4338 { 4339 int channelCount; 4340 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4341 // stereo to mono post process as the resampler always outputs stereo. 4342 if (mChannelCount == 1 && mReqChannelCount == 2) { 4343 channelCount = 1; 4344 } else { 4345 channelCount = 2; 4346 } 4347 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4348 mResampler->setSampleRate(mSampleRate); 4349 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4350 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4351 4352 // optmization: if mono to mono, alter input frame count as if we were inputing 4353 // stereo samples 4354 if (mChannelCount == 1 && mReqChannelCount == 1) { 4355 mFrameCount >>= 1; 4356 } 4357 4358 } 4359 mRsmpInIndex = mFrameCount; 4360} 4361 4362unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4363{ 4364 Mutex::Autolock _l(mLock); 4365 if (initCheck() != NO_ERROR) { 4366 return 0; 4367 } 4368 4369 return mInput->stream->get_input_frames_lost(mInput->stream); 4370} 4371 4372uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4373{ 4374 Mutex::Autolock _l(mLock); 4375 uint32_t result = 0; 4376 if (getEffectChain_l(sessionId) != 0) { 4377 result = EFFECT_SESSION; 4378 } 4379 4380 for (size_t i = 0; i < mTracks.size(); ++i) { 4381 if (sessionId == mTracks[i]->sessionId()) { 4382 result |= TRACK_SESSION; 4383 break; 4384 } 4385 } 4386 4387 return result; 4388} 4389 4390KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4391{ 4392 KeyedVector<int, bool> ids; 4393 Mutex::Autolock _l(mLock); 4394 for (size_t j = 0; j < mTracks.size(); ++j) { 4395 sp<RecordThread::RecordTrack> track = mTracks[j]; 4396 int sessionId = track->sessionId(); 4397 if (ids.indexOfKey(sessionId) < 0) { 4398 ids.add(sessionId, true); 4399 } 4400 } 4401 return ids; 4402} 4403 4404AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4405{ 4406 Mutex::Autolock _l(mLock); 4407 AudioStreamIn *input = mInput; 4408 mInput = NULL; 4409 return input; 4410} 4411 4412// this method must always be called either with ThreadBase mLock held or inside the thread loop 4413audio_stream_t* AudioFlinger::RecordThread::stream() const 4414{ 4415 if (mInput == NULL) { 4416 return NULL; 4417 } 4418 return &mInput->stream->common; 4419} 4420 4421status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4422{ 4423 // only one chain per input thread 4424 if (mEffectChains.size() != 0) { 4425 return INVALID_OPERATION; 4426 } 4427 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4428 4429 chain->setInBuffer(NULL); 4430 chain->setOutBuffer(NULL); 4431 4432 checkSuspendOnAddEffectChain_l(chain); 4433 4434 mEffectChains.add(chain); 4435 4436 return NO_ERROR; 4437} 4438 4439size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4440{ 4441 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4442 ALOGW_IF(mEffectChains.size() != 1, 4443 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4444 chain.get(), mEffectChains.size(), this); 4445 if (mEffectChains.size() == 1) { 4446 mEffectChains.removeAt(0); 4447 } 4448 return 0; 4449} 4450 4451}; // namespace android 4452