Threads.cpp revision 7e1139c0377b6806942fb2a043737b3b9cf0ae91
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 // FIXME Need to understand why this has be done asynchronously 377 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 378 true /*asynchronous*/); 379 if (err != 0) { 380 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 381 "error %d", 382 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 383 } 384 } break; 385 case CFG_EVENT_IO: { 386 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 387 mAudioFlinger->mLock.lock(); 388 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 389 mAudioFlinger->mLock.unlock(); 390 } break; 391 default: 392 ALOGE("processConfigEvents() unknown event type %d", event->type()); 393 break; 394 } 395 delete event; 396 mLock.lock(); 397 } 398 mLock.unlock(); 399} 400 401void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 402{ 403 const size_t SIZE = 256; 404 char buffer[SIZE]; 405 String8 result; 406 407 bool locked = AudioFlinger::dumpTryLock(mLock); 408 if (!locked) { 409 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 410 write(fd, buffer, strlen(buffer)); 411 } 412 413 snprintf(buffer, SIZE, "io handle: %d\n", mId); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 432 result.append(buffer); 433 434 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 435 result.append(buffer); 436 result.append(" Index Command"); 437 for (size_t i = 0; i < mNewParameters.size(); ++i) { 438 snprintf(buffer, SIZE, "\n %02d ", i); 439 result.append(buffer); 440 result.append(mNewParameters[i]); 441 } 442 443 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 444 result.append(buffer); 445 for (size_t i = 0; i < mConfigEvents.size(); i++) { 446 mConfigEvents[i]->dump(buffer, SIZE); 447 result.append(buffer); 448 } 449 result.append("\n"); 450 451 write(fd, result.string(), result.size()); 452 453 if (locked) { 454 mLock.unlock(); 455 } 456} 457 458void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 459{ 460 const size_t SIZE = 256; 461 char buffer[SIZE]; 462 String8 result; 463 464 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 465 write(fd, buffer, strlen(buffer)); 466 467 for (size_t i = 0; i < mEffectChains.size(); ++i) { 468 sp<EffectChain> chain = mEffectChains[i]; 469 if (chain != 0) { 470 chain->dump(fd, args); 471 } 472 } 473} 474 475void AudioFlinger::ThreadBase::acquireWakeLock() 476{ 477 Mutex::Autolock _l(mLock); 478 acquireWakeLock_l(); 479} 480 481void AudioFlinger::ThreadBase::acquireWakeLock_l() 482{ 483 if (mPowerManager == 0) { 484 // use checkService() to avoid blocking if power service is not up yet 485 sp<IBinder> binder = 486 defaultServiceManager()->checkService(String16("power")); 487 if (binder == 0) { 488 ALOGW("Thread %s cannot connect to the power manager service", mName); 489 } else { 490 mPowerManager = interface_cast<IPowerManager>(binder); 491 binder->linkToDeath(mDeathRecipient); 492 } 493 } 494 if (mPowerManager != 0) { 495 sp<IBinder> binder = new BBinder(); 496 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 497 binder, 498 String16(mName)); 499 if (status == NO_ERROR) { 500 mWakeLockToken = binder; 501 } 502 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 503 } 504} 505 506void AudioFlinger::ThreadBase::releaseWakeLock() 507{ 508 Mutex::Autolock _l(mLock); 509 releaseWakeLock_l(); 510} 511 512void AudioFlinger::ThreadBase::releaseWakeLock_l() 513{ 514 if (mWakeLockToken != 0) { 515 ALOGV("releaseWakeLock_l() %s", mName); 516 if (mPowerManager != 0) { 517 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 518 } 519 mWakeLockToken.clear(); 520 } 521} 522 523void AudioFlinger::ThreadBase::clearPowerManager() 524{ 525 Mutex::Autolock _l(mLock); 526 releaseWakeLock_l(); 527 mPowerManager.clear(); 528} 529 530void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 531{ 532 sp<ThreadBase> thread = mThread.promote(); 533 if (thread != 0) { 534 thread->clearPowerManager(); 535 } 536 ALOGW("power manager service died !!!"); 537} 538 539void AudioFlinger::ThreadBase::setEffectSuspended( 540 const effect_uuid_t *type, bool suspend, int sessionId) 541{ 542 Mutex::Autolock _l(mLock); 543 setEffectSuspended_l(type, suspend, sessionId); 544} 545 546void AudioFlinger::ThreadBase::setEffectSuspended_l( 547 const effect_uuid_t *type, bool suspend, int sessionId) 548{ 549 sp<EffectChain> chain = getEffectChain_l(sessionId); 550 if (chain != 0) { 551 if (type != NULL) { 552 chain->setEffectSuspended_l(type, suspend); 553 } else { 554 chain->setEffectSuspendedAll_l(suspend); 555 } 556 } 557 558 updateSuspendedSessions_l(type, suspend, sessionId); 559} 560 561void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 562{ 563 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 564 if (index < 0) { 565 return; 566 } 567 568 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 569 mSuspendedSessions.valueAt(index); 570 571 for (size_t i = 0; i < sessionEffects.size(); i++) { 572 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 573 for (int j = 0; j < desc->mRefCount; j++) { 574 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 575 chain->setEffectSuspendedAll_l(true); 576 } else { 577 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 578 desc->mType.timeLow); 579 chain->setEffectSuspended_l(&desc->mType, true); 580 } 581 } 582 } 583} 584 585void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 586 bool suspend, 587 int sessionId) 588{ 589 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 590 591 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 592 593 if (suspend) { 594 if (index >= 0) { 595 sessionEffects = mSuspendedSessions.valueAt(index); 596 } else { 597 mSuspendedSessions.add(sessionId, sessionEffects); 598 } 599 } else { 600 if (index < 0) { 601 return; 602 } 603 sessionEffects = mSuspendedSessions.valueAt(index); 604 } 605 606 607 int key = EffectChain::kKeyForSuspendAll; 608 if (type != NULL) { 609 key = type->timeLow; 610 } 611 index = sessionEffects.indexOfKey(key); 612 613 sp<SuspendedSessionDesc> desc; 614 if (suspend) { 615 if (index >= 0) { 616 desc = sessionEffects.valueAt(index); 617 } else { 618 desc = new SuspendedSessionDesc(); 619 if (type != NULL) { 620 desc->mType = *type; 621 } 622 sessionEffects.add(key, desc); 623 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 624 } 625 desc->mRefCount++; 626 } else { 627 if (index < 0) { 628 return; 629 } 630 desc = sessionEffects.valueAt(index); 631 if (--desc->mRefCount == 0) { 632 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 633 sessionEffects.removeItemsAt(index); 634 if (sessionEffects.isEmpty()) { 635 ALOGV("updateSuspendedSessions_l() restore removing session %d", 636 sessionId); 637 mSuspendedSessions.removeItem(sessionId); 638 } 639 } 640 } 641 if (!sessionEffects.isEmpty()) { 642 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 643 } 644} 645 646void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 647 bool enabled, 648 int sessionId) 649{ 650 Mutex::Autolock _l(mLock); 651 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 652} 653 654void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 655 bool enabled, 656 int sessionId) 657{ 658 if (mType != RECORD) { 659 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 660 // another session. This gives the priority to well behaved effect control panels 661 // and applications not using global effects. 662 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 663 // global effects 664 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 665 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 666 } 667 } 668 669 sp<EffectChain> chain = getEffectChain_l(sessionId); 670 if (chain != 0) { 671 chain->checkSuspendOnEffectEnabled(effect, enabled); 672 } 673} 674 675// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 676sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 677 const sp<AudioFlinger::Client>& client, 678 const sp<IEffectClient>& effectClient, 679 int32_t priority, 680 int sessionId, 681 effect_descriptor_t *desc, 682 int *enabled, 683 status_t *status 684 ) 685{ 686 sp<EffectModule> effect; 687 sp<EffectHandle> handle; 688 status_t lStatus; 689 sp<EffectChain> chain; 690 bool chainCreated = false; 691 bool effectCreated = false; 692 bool effectRegistered = false; 693 694 lStatus = initCheck(); 695 if (lStatus != NO_ERROR) { 696 ALOGW("createEffect_l() Audio driver not initialized."); 697 goto Exit; 698 } 699 700 // Do not allow effects with session ID 0 on direct output or duplicating threads 701 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 702 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 703 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 704 desc->name, sessionId); 705 lStatus = BAD_VALUE; 706 goto Exit; 707 } 708 // Only Pre processor effects are allowed on input threads and only on input threads 709 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 710 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 711 desc->name, desc->flags, mType); 712 lStatus = BAD_VALUE; 713 goto Exit; 714 } 715 716 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 717 718 { // scope for mLock 719 Mutex::Autolock _l(mLock); 720 721 // check for existing effect chain with the requested audio session 722 chain = getEffectChain_l(sessionId); 723 if (chain == 0) { 724 // create a new chain for this session 725 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 726 chain = new EffectChain(this, sessionId); 727 addEffectChain_l(chain); 728 chain->setStrategy(getStrategyForSession_l(sessionId)); 729 chainCreated = true; 730 } else { 731 effect = chain->getEffectFromDesc_l(desc); 732 } 733 734 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 735 736 if (effect == 0) { 737 int id = mAudioFlinger->nextUniqueId(); 738 // Check CPU and memory usage 739 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 740 if (lStatus != NO_ERROR) { 741 goto Exit; 742 } 743 effectRegistered = true; 744 // create a new effect module if none present in the chain 745 effect = new EffectModule(this, chain, desc, id, sessionId); 746 lStatus = effect->status(); 747 if (lStatus != NO_ERROR) { 748 goto Exit; 749 } 750 lStatus = chain->addEffect_l(effect); 751 if (lStatus != NO_ERROR) { 752 goto Exit; 753 } 754 effectCreated = true; 755 756 effect->setDevice(mOutDevice); 757 effect->setDevice(mInDevice); 758 effect->setMode(mAudioFlinger->getMode()); 759 effect->setAudioSource(mAudioSource); 760 } 761 // create effect handle and connect it to effect module 762 handle = new EffectHandle(effect, client, effectClient, priority); 763 lStatus = effect->addHandle(handle.get()); 764 if (enabled != NULL) { 765 *enabled = (int)effect->isEnabled(); 766 } 767 } 768 769Exit: 770 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 771 Mutex::Autolock _l(mLock); 772 if (effectCreated) { 773 chain->removeEffect_l(effect); 774 } 775 if (effectRegistered) { 776 AudioSystem::unregisterEffect(effect->id()); 777 } 778 if (chainCreated) { 779 removeEffectChain_l(chain); 780 } 781 handle.clear(); 782 } 783 784 if (status != NULL) { 785 *status = lStatus; 786 } 787 return handle; 788} 789 790sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 791{ 792 Mutex::Autolock _l(mLock); 793 return getEffect_l(sessionId, effectId); 794} 795 796sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 797{ 798 sp<EffectChain> chain = getEffectChain_l(sessionId); 799 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 800} 801 802// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 803// PlaybackThread::mLock held 804status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 805{ 806 // check for existing effect chain with the requested audio session 807 int sessionId = effect->sessionId(); 808 sp<EffectChain> chain = getEffectChain_l(sessionId); 809 bool chainCreated = false; 810 811 if (chain == 0) { 812 // create a new chain for this session 813 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 814 chain = new EffectChain(this, sessionId); 815 addEffectChain_l(chain); 816 chain->setStrategy(getStrategyForSession_l(sessionId)); 817 chainCreated = true; 818 } 819 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 820 821 if (chain->getEffectFromId_l(effect->id()) != 0) { 822 ALOGW("addEffect_l() %p effect %s already present in chain %p", 823 this, effect->desc().name, chain.get()); 824 return BAD_VALUE; 825 } 826 827 status_t status = chain->addEffect_l(effect); 828 if (status != NO_ERROR) { 829 if (chainCreated) { 830 removeEffectChain_l(chain); 831 } 832 return status; 833 } 834 835 effect->setDevice(mOutDevice); 836 effect->setDevice(mInDevice); 837 effect->setMode(mAudioFlinger->getMode()); 838 effect->setAudioSource(mAudioSource); 839 return NO_ERROR; 840} 841 842void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 843 844 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 845 effect_descriptor_t desc = effect->desc(); 846 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 847 detachAuxEffect_l(effect->id()); 848 } 849 850 sp<EffectChain> chain = effect->chain().promote(); 851 if (chain != 0) { 852 // remove effect chain if removing last effect 853 if (chain->removeEffect_l(effect) == 0) { 854 removeEffectChain_l(chain); 855 } 856 } else { 857 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 858 } 859} 860 861void AudioFlinger::ThreadBase::lockEffectChains_l( 862 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 863{ 864 effectChains = mEffectChains; 865 for (size_t i = 0; i < mEffectChains.size(); i++) { 866 mEffectChains[i]->lock(); 867 } 868} 869 870void AudioFlinger::ThreadBase::unlockEffectChains( 871 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 872{ 873 for (size_t i = 0; i < effectChains.size(); i++) { 874 effectChains[i]->unlock(); 875 } 876} 877 878sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 879{ 880 Mutex::Autolock _l(mLock); 881 return getEffectChain_l(sessionId); 882} 883 884sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 885{ 886 size_t size = mEffectChains.size(); 887 for (size_t i = 0; i < size; i++) { 888 if (mEffectChains[i]->sessionId() == sessionId) { 889 return mEffectChains[i]; 890 } 891 } 892 return 0; 893} 894 895void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 896{ 897 Mutex::Autolock _l(mLock); 898 size_t size = mEffectChains.size(); 899 for (size_t i = 0; i < size; i++) { 900 mEffectChains[i]->setMode_l(mode); 901 } 902} 903 904void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 905 EffectHandle *handle, 906 bool unpinIfLast) { 907 908 Mutex::Autolock _l(mLock); 909 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 910 // delete the effect module if removing last handle on it 911 if (effect->removeHandle(handle) == 0) { 912 if (!effect->isPinned() || unpinIfLast) { 913 removeEffect_l(effect); 914 AudioSystem::unregisterEffect(effect->id()); 915 } 916 } 917} 918 919// ---------------------------------------------------------------------------- 920// Playback 921// ---------------------------------------------------------------------------- 922 923AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 924 AudioStreamOut* output, 925 audio_io_handle_t id, 926 audio_devices_t device, 927 type_t type) 928 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 929 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 930 // mStreamTypes[] initialized in constructor body 931 mOutput(output), 932 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 933 mMixerStatus(MIXER_IDLE), 934 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 935 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 936 mScreenState(AudioFlinger::mScreenState), 937 // index 0 is reserved for normal mixer's submix 938 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 939{ 940 snprintf(mName, kNameLength, "AudioOut_%X", id); 941 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 942 943 // Assumes constructor is called by AudioFlinger with it's mLock held, but 944 // it would be safer to explicitly pass initial masterVolume/masterMute as 945 // parameter. 946 // 947 // If the HAL we are using has support for master volume or master mute, 948 // then do not attenuate or mute during mixing (just leave the volume at 1.0 949 // and the mute set to false). 950 mMasterVolume = audioFlinger->masterVolume_l(); 951 mMasterMute = audioFlinger->masterMute_l(); 952 if (mOutput && mOutput->audioHwDev) { 953 if (mOutput->audioHwDev->canSetMasterVolume()) { 954 mMasterVolume = 1.0; 955 } 956 957 if (mOutput->audioHwDev->canSetMasterMute()) { 958 mMasterMute = false; 959 } 960 } 961 962 readOutputParameters(); 963 964 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 965 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 966 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 967 stream = (audio_stream_type_t) (stream + 1)) { 968 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 969 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 970 } 971 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 972 // because mAudioFlinger doesn't have one to copy from 973} 974 975AudioFlinger::PlaybackThread::~PlaybackThread() 976{ 977 mAudioFlinger->unregisterWriter(mNBLogWriter); 978 delete [] mMixBuffer; 979} 980 981void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 982{ 983 dumpInternals(fd, args); 984 dumpTracks(fd, args); 985 dumpEffectChains(fd, args); 986} 987 988void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 989{ 990 const size_t SIZE = 256; 991 char buffer[SIZE]; 992 String8 result; 993 994 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 995 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 996 const stream_type_t *st = &mStreamTypes[i]; 997 if (i > 0) { 998 result.appendFormat(", "); 999 } 1000 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1001 if (st->mute) { 1002 result.append("M"); 1003 } 1004 } 1005 result.append("\n"); 1006 write(fd, result.string(), result.length()); 1007 result.clear(); 1008 1009 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1010 result.append(buffer); 1011 Track::appendDumpHeader(result); 1012 for (size_t i = 0; i < mTracks.size(); ++i) { 1013 sp<Track> track = mTracks[i]; 1014 if (track != 0) { 1015 track->dump(buffer, SIZE); 1016 result.append(buffer); 1017 } 1018 } 1019 1020 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1021 result.append(buffer); 1022 Track::appendDumpHeader(result); 1023 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1024 sp<Track> track = mActiveTracks[i].promote(); 1025 if (track != 0) { 1026 track->dump(buffer, SIZE); 1027 result.append(buffer); 1028 } 1029 } 1030 write(fd, result.string(), result.size()); 1031 1032 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1033 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1034 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1035 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1036} 1037 1038void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1039{ 1040 const size_t SIZE = 256; 1041 char buffer[SIZE]; 1042 String8 result; 1043 1044 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1045 result.append(buffer); 1046 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1047 ns2ms(systemTime() - mLastWriteTime)); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1056 result.append(buffer); 1057 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1058 result.append(buffer); 1059 write(fd, result.string(), result.size()); 1060 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1061 1062 dumpBase(fd, args); 1063} 1064 1065// Thread virtuals 1066status_t AudioFlinger::PlaybackThread::readyToRun() 1067{ 1068 status_t status = initCheck(); 1069 if (status == NO_ERROR) { 1070 ALOGI("AudioFlinger's thread %p ready to run", this); 1071 } else { 1072 ALOGE("No working audio driver found."); 1073 } 1074 return status; 1075} 1076 1077void AudioFlinger::PlaybackThread::onFirstRef() 1078{ 1079 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1080} 1081 1082// ThreadBase virtuals 1083void AudioFlinger::PlaybackThread::preExit() 1084{ 1085 ALOGV(" preExit()"); 1086 // FIXME this is using hard-coded strings but in the future, this functionality will be 1087 // converted to use audio HAL extensions required to support tunneling 1088 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1089} 1090 1091// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1092sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1093 const sp<AudioFlinger::Client>& client, 1094 audio_stream_type_t streamType, 1095 uint32_t sampleRate, 1096 audio_format_t format, 1097 audio_channel_mask_t channelMask, 1098 size_t frameCount, 1099 const sp<IMemory>& sharedBuffer, 1100 int sessionId, 1101 IAudioFlinger::track_flags_t *flags, 1102 pid_t tid, 1103 status_t *status) 1104{ 1105 sp<Track> track; 1106 status_t lStatus; 1107 1108 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1109 1110 // client expresses a preference for FAST, but we get the final say 1111 if (*flags & IAudioFlinger::TRACK_FAST) { 1112 if ( 1113 // not timed 1114 (!isTimed) && 1115 // either of these use cases: 1116 ( 1117 // use case 1: shared buffer with any frame count 1118 ( 1119 (sharedBuffer != 0) 1120 ) || 1121 // use case 2: callback handler and frame count is default or at least as large as HAL 1122 ( 1123 (tid != -1) && 1124 ((frameCount == 0) || 1125 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1126 ) 1127 ) && 1128 // PCM data 1129 audio_is_linear_pcm(format) && 1130 // mono or stereo 1131 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1132 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1133#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1134 // hardware sample rate 1135 (sampleRate == mSampleRate) && 1136#endif 1137 // normal mixer has an associated fast mixer 1138 hasFastMixer() && 1139 // there are sufficient fast track slots available 1140 (mFastTrackAvailMask != 0) 1141 // FIXME test that MixerThread for this fast track has a capable output HAL 1142 // FIXME add a permission test also? 1143 ) { 1144 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1145 if (frameCount == 0) { 1146 frameCount = mFrameCount * kFastTrackMultiplier; 1147 } 1148 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1149 frameCount, mFrameCount); 1150 } else { 1151 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1152 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1153 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1154 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1155 audio_is_linear_pcm(format), 1156 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1157 *flags &= ~IAudioFlinger::TRACK_FAST; 1158 // For compatibility with AudioTrack calculation, buffer depth is forced 1159 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1160 // This is probably too conservative, but legacy application code may depend on it. 1161 // If you change this calculation, also review the start threshold which is related. 1162 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1163 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1164 if (minBufCount < 2) { 1165 minBufCount = 2; 1166 } 1167 size_t minFrameCount = mNormalFrameCount * minBufCount; 1168 if (frameCount < minFrameCount) { 1169 frameCount = minFrameCount; 1170 } 1171 } 1172 } 1173 1174 if (mType == DIRECT) { 1175 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1176 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1177 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1178 "for output %p with format %d", 1179 sampleRate, format, channelMask, mOutput, mFormat); 1180 lStatus = BAD_VALUE; 1181 goto Exit; 1182 } 1183 } 1184 } else { 1185 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1186 if (sampleRate > mSampleRate*2) { 1187 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1188 lStatus = BAD_VALUE; 1189 goto Exit; 1190 } 1191 } 1192 1193 lStatus = initCheck(); 1194 if (lStatus != NO_ERROR) { 1195 ALOGE("Audio driver not initialized."); 1196 goto Exit; 1197 } 1198 1199 { // scope for mLock 1200 Mutex::Autolock _l(mLock); 1201 1202 // all tracks in same audio session must share the same routing strategy otherwise 1203 // conflicts will happen when tracks are moved from one output to another by audio policy 1204 // manager 1205 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1206 for (size_t i = 0; i < mTracks.size(); ++i) { 1207 sp<Track> t = mTracks[i]; 1208 if (t != 0 && !t->isOutputTrack()) { 1209 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1210 if (sessionId == t->sessionId() && strategy != actual) { 1211 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1212 strategy, actual); 1213 lStatus = BAD_VALUE; 1214 goto Exit; 1215 } 1216 } 1217 } 1218 1219 if (!isTimed) { 1220 track = new Track(this, client, streamType, sampleRate, format, 1221 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1222 } else { 1223 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1224 channelMask, frameCount, sharedBuffer, sessionId); 1225 } 1226 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1227 lStatus = NO_MEMORY; 1228 goto Exit; 1229 } 1230 mTracks.add(track); 1231 1232 sp<EffectChain> chain = getEffectChain_l(sessionId); 1233 if (chain != 0) { 1234 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1235 track->setMainBuffer(chain->inBuffer()); 1236 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1237 chain->incTrackCnt(); 1238 } 1239 1240 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1241 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1242 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1243 // so ask activity manager to do this on our behalf 1244 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1245 } 1246 } 1247 1248 lStatus = NO_ERROR; 1249 1250Exit: 1251 if (status) { 1252 *status = lStatus; 1253 } 1254 return track; 1255} 1256 1257uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1258{ 1259 return latency; 1260} 1261 1262uint32_t AudioFlinger::PlaybackThread::latency() const 1263{ 1264 Mutex::Autolock _l(mLock); 1265 return latency_l(); 1266} 1267uint32_t AudioFlinger::PlaybackThread::latency_l() const 1268{ 1269 if (initCheck() == NO_ERROR) { 1270 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1271 } else { 1272 return 0; 1273 } 1274} 1275 1276void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1277{ 1278 Mutex::Autolock _l(mLock); 1279 // Don't apply master volume in SW if our HAL can do it for us. 1280 if (mOutput && mOutput->audioHwDev && 1281 mOutput->audioHwDev->canSetMasterVolume()) { 1282 mMasterVolume = 1.0; 1283 } else { 1284 mMasterVolume = value; 1285 } 1286} 1287 1288void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1289{ 1290 Mutex::Autolock _l(mLock); 1291 // Don't apply master mute in SW if our HAL can do it for us. 1292 if (mOutput && mOutput->audioHwDev && 1293 mOutput->audioHwDev->canSetMasterMute()) { 1294 mMasterMute = false; 1295 } else { 1296 mMasterMute = muted; 1297 } 1298} 1299 1300void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1301{ 1302 Mutex::Autolock _l(mLock); 1303 mStreamTypes[stream].volume = value; 1304} 1305 1306void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1307{ 1308 Mutex::Autolock _l(mLock); 1309 mStreamTypes[stream].mute = muted; 1310} 1311 1312float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1313{ 1314 Mutex::Autolock _l(mLock); 1315 return mStreamTypes[stream].volume; 1316} 1317 1318// addTrack_l() must be called with ThreadBase::mLock held 1319status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1320{ 1321 status_t status = ALREADY_EXISTS; 1322 1323 // set retry count for buffer fill 1324 track->mRetryCount = kMaxTrackStartupRetries; 1325 if (mActiveTracks.indexOf(track) < 0) { 1326 // the track is newly added, make sure it fills up all its 1327 // buffers before playing. This is to ensure the client will 1328 // effectively get the latency it requested. 1329 track->mFillingUpStatus = Track::FS_FILLING; 1330 track->mResetDone = false; 1331 track->mPresentationCompleteFrames = 0; 1332 mActiveTracks.add(track); 1333 if (track->mainBuffer() != mMixBuffer) { 1334 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1335 if (chain != 0) { 1336 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1337 track->sessionId()); 1338 chain->incActiveTrackCnt(); 1339 } 1340 } 1341 1342 status = NO_ERROR; 1343 } 1344 1345 ALOGV("mWaitWorkCV.broadcast"); 1346 mWaitWorkCV.broadcast(); 1347 1348 return status; 1349} 1350 1351// destroyTrack_l() must be called with ThreadBase::mLock held 1352void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1353{ 1354 track->mState = TrackBase::TERMINATED; 1355 // active tracks are removed by threadLoop() 1356 if (mActiveTracks.indexOf(track) < 0) { 1357 removeTrack_l(track); 1358 } 1359} 1360 1361void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1362{ 1363 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1364 mTracks.remove(track); 1365 deleteTrackName_l(track->name()); 1366 // redundant as track is about to be destroyed, for dumpsys only 1367 track->mName = -1; 1368 if (track->isFastTrack()) { 1369 int index = track->mFastIndex; 1370 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1371 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1372 mFastTrackAvailMask |= 1 << index; 1373 // redundant as track is about to be destroyed, for dumpsys only 1374 track->mFastIndex = -1; 1375 } 1376 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1377 if (chain != 0) { 1378 chain->decTrackCnt(); 1379 } 1380} 1381 1382String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1383{ 1384 String8 out_s8 = String8(""); 1385 char *s; 1386 1387 Mutex::Autolock _l(mLock); 1388 if (initCheck() != NO_ERROR) { 1389 return out_s8; 1390 } 1391 1392 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1393 out_s8 = String8(s); 1394 free(s); 1395 return out_s8; 1396} 1397 1398// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1399void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1400 AudioSystem::OutputDescriptor desc; 1401 void *param2 = NULL; 1402 1403 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1404 param); 1405 1406 switch (event) { 1407 case AudioSystem::OUTPUT_OPENED: 1408 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1409 desc.channels = mChannelMask; 1410 desc.samplingRate = mSampleRate; 1411 desc.format = mFormat; 1412 desc.frameCount = mNormalFrameCount; // FIXME see 1413 // AudioFlinger::frameCount(audio_io_handle_t) 1414 desc.latency = latency(); 1415 param2 = &desc; 1416 break; 1417 1418 case AudioSystem::STREAM_CONFIG_CHANGED: 1419 param2 = ¶m; 1420 case AudioSystem::OUTPUT_CLOSED: 1421 default: 1422 break; 1423 } 1424 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1425} 1426 1427void AudioFlinger::PlaybackThread::readOutputParameters() 1428{ 1429 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1430 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1431 mChannelCount = (uint16_t)popcount(mChannelMask); 1432 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1433 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1434 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1435 if (mFrameCount & 15) { 1436 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1437 mFrameCount); 1438 } 1439 1440 // Calculate size of normal mix buffer relative to the HAL output buffer size 1441 double multiplier = 1.0; 1442 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1443 kUseFastMixer == FastMixer_Dynamic)) { 1444 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1445 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1446 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1447 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1448 maxNormalFrameCount = maxNormalFrameCount & ~15; 1449 if (maxNormalFrameCount < minNormalFrameCount) { 1450 maxNormalFrameCount = minNormalFrameCount; 1451 } 1452 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1453 if (multiplier <= 1.0) { 1454 multiplier = 1.0; 1455 } else if (multiplier <= 2.0) { 1456 if (2 * mFrameCount <= maxNormalFrameCount) { 1457 multiplier = 2.0; 1458 } else { 1459 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1460 } 1461 } else { 1462 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1463 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1464 // track, but we sometimes have to do this to satisfy the maximum frame count 1465 // constraint) 1466 // FIXME this rounding up should not be done if no HAL SRC 1467 uint32_t truncMult = (uint32_t) multiplier; 1468 if ((truncMult & 1)) { 1469 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1470 ++truncMult; 1471 } 1472 } 1473 multiplier = (double) truncMult; 1474 } 1475 } 1476 mNormalFrameCount = multiplier * mFrameCount; 1477 // round up to nearest 16 frames to satisfy AudioMixer 1478 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1479 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1480 mNormalFrameCount); 1481 1482 delete[] mMixBuffer; 1483 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1484 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1485 1486 // force reconfiguration of effect chains and engines to take new buffer size and audio 1487 // parameters into account 1488 // Note that mLock is not held when readOutputParameters() is called from the constructor 1489 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1490 // matter. 1491 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1492 Vector< sp<EffectChain> > effectChains = mEffectChains; 1493 for (size_t i = 0; i < effectChains.size(); i ++) { 1494 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1495 } 1496} 1497 1498 1499status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1500{ 1501 if (halFrames == NULL || dspFrames == NULL) { 1502 return BAD_VALUE; 1503 } 1504 Mutex::Autolock _l(mLock); 1505 if (initCheck() != NO_ERROR) { 1506 return INVALID_OPERATION; 1507 } 1508 size_t framesWritten = mBytesWritten / mFrameSize; 1509 *halFrames = framesWritten; 1510 1511 if (isSuspended()) { 1512 // return an estimation of rendered frames when the output is suspended 1513 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1514 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1515 return NO_ERROR; 1516 } else { 1517 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1518 } 1519} 1520 1521uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1522{ 1523 Mutex::Autolock _l(mLock); 1524 uint32_t result = 0; 1525 if (getEffectChain_l(sessionId) != 0) { 1526 result = EFFECT_SESSION; 1527 } 1528 1529 for (size_t i = 0; i < mTracks.size(); ++i) { 1530 sp<Track> track = mTracks[i]; 1531 if (sessionId == track->sessionId() && !track->isInvalid()) { 1532 result |= TRACK_SESSION; 1533 break; 1534 } 1535 } 1536 1537 return result; 1538} 1539 1540uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1541{ 1542 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1543 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1544 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1545 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1546 } 1547 for (size_t i = 0; i < mTracks.size(); i++) { 1548 sp<Track> track = mTracks[i]; 1549 if (sessionId == track->sessionId() && !track->isInvalid()) { 1550 return AudioSystem::getStrategyForStream(track->streamType()); 1551 } 1552 } 1553 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1554} 1555 1556 1557AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1558{ 1559 Mutex::Autolock _l(mLock); 1560 return mOutput; 1561} 1562 1563AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1564{ 1565 Mutex::Autolock _l(mLock); 1566 AudioStreamOut *output = mOutput; 1567 mOutput = NULL; 1568 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1569 // must push a NULL and wait for ack 1570 mOutputSink.clear(); 1571 mPipeSink.clear(); 1572 mNormalSink.clear(); 1573 return output; 1574} 1575 1576// this method must always be called either with ThreadBase mLock held or inside the thread loop 1577audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1578{ 1579 if (mOutput == NULL) { 1580 return NULL; 1581 } 1582 return &mOutput->stream->common; 1583} 1584 1585uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1586{ 1587 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1588} 1589 1590status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1591{ 1592 if (!isValidSyncEvent(event)) { 1593 return BAD_VALUE; 1594 } 1595 1596 Mutex::Autolock _l(mLock); 1597 1598 for (size_t i = 0; i < mTracks.size(); ++i) { 1599 sp<Track> track = mTracks[i]; 1600 if (event->triggerSession() == track->sessionId()) { 1601 (void) track->setSyncEvent(event); 1602 return NO_ERROR; 1603 } 1604 } 1605 1606 return NAME_NOT_FOUND; 1607} 1608 1609bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1610{ 1611 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1612} 1613 1614void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1615 const Vector< sp<Track> >& tracksToRemove) 1616{ 1617 size_t count = tracksToRemove.size(); 1618 if (CC_UNLIKELY(count)) { 1619 for (size_t i = 0 ; i < count ; i++) { 1620 const sp<Track>& track = tracksToRemove.itemAt(i); 1621 if ((track->sharedBuffer() != 0) && 1622 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1623 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1624 } 1625 } 1626 } 1627 1628} 1629 1630void AudioFlinger::PlaybackThread::checkSilentMode_l() 1631{ 1632 if (!mMasterMute) { 1633 char value[PROPERTY_VALUE_MAX]; 1634 if (property_get("ro.audio.silent", value, "0") > 0) { 1635 char *endptr; 1636 unsigned long ul = strtoul(value, &endptr, 0); 1637 if (*endptr == '\0' && ul != 0) { 1638 ALOGD("Silence is golden"); 1639 // The setprop command will not allow a property to be changed after 1640 // the first time it is set, so we don't have to worry about un-muting. 1641 setMasterMute_l(true); 1642 } 1643 } 1644 } 1645} 1646 1647// shared by MIXER and DIRECT, overridden by DUPLICATING 1648void AudioFlinger::PlaybackThread::threadLoop_write() 1649{ 1650 // FIXME rewrite to reduce number of system calls 1651 mLastWriteTime = systemTime(); 1652 mInWrite = true; 1653 int bytesWritten; 1654 1655 // If an NBAIO sink is present, use it to write the normal mixer's submix 1656 if (mNormalSink != 0) { 1657#define mBitShift 2 // FIXME 1658 size_t count = mixBufferSize >> mBitShift; 1659 ATRACE_BEGIN("write"); 1660 // update the setpoint when AudioFlinger::mScreenState changes 1661 uint32_t screenState = AudioFlinger::mScreenState; 1662 if (screenState != mScreenState) { 1663 mScreenState = screenState; 1664 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1665 if (pipe != NULL) { 1666 pipe->setAvgFrames((mScreenState & 1) ? 1667 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1668 } 1669 } 1670 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1671 ATRACE_END(); 1672 if (framesWritten > 0) { 1673 bytesWritten = framesWritten << mBitShift; 1674 } else { 1675 bytesWritten = framesWritten; 1676 } 1677 // otherwise use the HAL / AudioStreamOut directly 1678 } else { 1679 // Direct output thread. 1680 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1681 } 1682 1683 if (bytesWritten > 0) { 1684 mBytesWritten += mixBufferSize; 1685 } 1686 mNumWrites++; 1687 mInWrite = false; 1688} 1689 1690/* 1691The derived values that are cached: 1692 - mixBufferSize from frame count * frame size 1693 - activeSleepTime from activeSleepTimeUs() 1694 - idleSleepTime from idleSleepTimeUs() 1695 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1696 - maxPeriod from frame count and sample rate (MIXER only) 1697 1698The parameters that affect these derived values are: 1699 - frame count 1700 - frame size 1701 - sample rate 1702 - device type: A2DP or not 1703 - device latency 1704 - format: PCM or not 1705 - active sleep time 1706 - idle sleep time 1707*/ 1708 1709void AudioFlinger::PlaybackThread::cacheParameters_l() 1710{ 1711 mixBufferSize = mNormalFrameCount * mFrameSize; 1712 activeSleepTime = activeSleepTimeUs(); 1713 idleSleepTime = idleSleepTimeUs(); 1714} 1715 1716void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1717{ 1718 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1719 this, streamType, mTracks.size()); 1720 Mutex::Autolock _l(mLock); 1721 1722 size_t size = mTracks.size(); 1723 for (size_t i = 0; i < size; i++) { 1724 sp<Track> t = mTracks[i]; 1725 if (t->streamType() == streamType) { 1726 t->invalidate(); 1727 } 1728 } 1729} 1730 1731status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1732{ 1733 int session = chain->sessionId(); 1734 int16_t *buffer = mMixBuffer; 1735 bool ownsBuffer = false; 1736 1737 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1738 if (session > 0) { 1739 // Only one effect chain can be present in direct output thread and it uses 1740 // the mix buffer as input 1741 if (mType != DIRECT) { 1742 size_t numSamples = mNormalFrameCount * mChannelCount; 1743 buffer = new int16_t[numSamples]; 1744 memset(buffer, 0, numSamples * sizeof(int16_t)); 1745 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1746 ownsBuffer = true; 1747 } 1748 1749 // Attach all tracks with same session ID to this chain. 1750 for (size_t i = 0; i < mTracks.size(); ++i) { 1751 sp<Track> track = mTracks[i]; 1752 if (session == track->sessionId()) { 1753 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1754 buffer); 1755 track->setMainBuffer(buffer); 1756 chain->incTrackCnt(); 1757 } 1758 } 1759 1760 // indicate all active tracks in the chain 1761 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1762 sp<Track> track = mActiveTracks[i].promote(); 1763 if (track == 0) { 1764 continue; 1765 } 1766 if (session == track->sessionId()) { 1767 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1768 chain->incActiveTrackCnt(); 1769 } 1770 } 1771 } 1772 1773 chain->setInBuffer(buffer, ownsBuffer); 1774 chain->setOutBuffer(mMixBuffer); 1775 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1776 // chains list in order to be processed last as it contains output stage effects 1777 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1778 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1779 // after track specific effects and before output stage 1780 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1781 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1782 // Effect chain for other sessions are inserted at beginning of effect 1783 // chains list to be processed before output mix effects. Relative order between other 1784 // sessions is not important 1785 size_t size = mEffectChains.size(); 1786 size_t i = 0; 1787 for (i = 0; i < size; i++) { 1788 if (mEffectChains[i]->sessionId() < session) { 1789 break; 1790 } 1791 } 1792 mEffectChains.insertAt(chain, i); 1793 checkSuspendOnAddEffectChain_l(chain); 1794 1795 return NO_ERROR; 1796} 1797 1798size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1799{ 1800 int session = chain->sessionId(); 1801 1802 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1803 1804 for (size_t i = 0; i < mEffectChains.size(); i++) { 1805 if (chain == mEffectChains[i]) { 1806 mEffectChains.removeAt(i); 1807 // detach all active tracks from the chain 1808 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1809 sp<Track> track = mActiveTracks[i].promote(); 1810 if (track == 0) { 1811 continue; 1812 } 1813 if (session == track->sessionId()) { 1814 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1815 chain.get(), session); 1816 chain->decActiveTrackCnt(); 1817 } 1818 } 1819 1820 // detach all tracks with same session ID from this chain 1821 for (size_t i = 0; i < mTracks.size(); ++i) { 1822 sp<Track> track = mTracks[i]; 1823 if (session == track->sessionId()) { 1824 track->setMainBuffer(mMixBuffer); 1825 chain->decTrackCnt(); 1826 } 1827 } 1828 break; 1829 } 1830 } 1831 return mEffectChains.size(); 1832} 1833 1834status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1835 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1836{ 1837 Mutex::Autolock _l(mLock); 1838 return attachAuxEffect_l(track, EffectId); 1839} 1840 1841status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1842 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1843{ 1844 status_t status = NO_ERROR; 1845 1846 if (EffectId == 0) { 1847 track->setAuxBuffer(0, NULL); 1848 } else { 1849 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1850 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1851 if (effect != 0) { 1852 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1853 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1854 } else { 1855 status = INVALID_OPERATION; 1856 } 1857 } else { 1858 status = BAD_VALUE; 1859 } 1860 } 1861 return status; 1862} 1863 1864void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1865{ 1866 for (size_t i = 0; i < mTracks.size(); ++i) { 1867 sp<Track> track = mTracks[i]; 1868 if (track->auxEffectId() == effectId) { 1869 attachAuxEffect_l(track, 0); 1870 } 1871 } 1872} 1873 1874bool AudioFlinger::PlaybackThread::threadLoop() 1875{ 1876 Vector< sp<Track> > tracksToRemove; 1877 1878 standbyTime = systemTime(); 1879 1880 // MIXER 1881 nsecs_t lastWarning = 0; 1882 1883 // DUPLICATING 1884 // FIXME could this be made local to while loop? 1885 writeFrames = 0; 1886 1887 cacheParameters_l(); 1888 sleepTime = idleSleepTime; 1889 1890 if (mType == MIXER) { 1891 sleepTimeShift = 0; 1892 } 1893 1894 CpuStats cpuStats; 1895 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1896 1897 acquireWakeLock(); 1898 1899 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1900 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1901 // and then that string will be logged at the next convenient opportunity. 1902 const char *logString = NULL; 1903 1904 while (!exitPending()) 1905 { 1906 cpuStats.sample(myName); 1907 1908 Vector< sp<EffectChain> > effectChains; 1909 1910 processConfigEvents(); 1911 1912 { // scope for mLock 1913 1914 Mutex::Autolock _l(mLock); 1915 1916 if (logString != NULL) { 1917 mNBLogWriter->logTimestamp(); 1918 mNBLogWriter->log(logString); 1919 logString = NULL; 1920 } 1921 1922 if (checkForNewParameters_l()) { 1923 cacheParameters_l(); 1924 } 1925 1926 saveOutputTracks(); 1927 1928 // put audio hardware into standby after short delay 1929 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1930 isSuspended())) { 1931 if (!mStandby) { 1932 1933 threadLoop_standby(); 1934 1935 mStandby = true; 1936 } 1937 1938 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1939 // we're about to wait, flush the binder command buffer 1940 IPCThreadState::self()->flushCommands(); 1941 1942 clearOutputTracks(); 1943 1944 if (exitPending()) { 1945 break; 1946 } 1947 1948 releaseWakeLock_l(); 1949 // wait until we have something to do... 1950 ALOGV("%s going to sleep", myName.string()); 1951 mWaitWorkCV.wait(mLock); 1952 ALOGV("%s waking up", myName.string()); 1953 acquireWakeLock_l(); 1954 1955 mMixerStatus = MIXER_IDLE; 1956 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1957 mBytesWritten = 0; 1958 1959 checkSilentMode_l(); 1960 1961 standbyTime = systemTime() + standbyDelay; 1962 sleepTime = idleSleepTime; 1963 if (mType == MIXER) { 1964 sleepTimeShift = 0; 1965 } 1966 1967 continue; 1968 } 1969 } 1970 1971 // mMixerStatusIgnoringFastTracks is also updated internally 1972 mMixerStatus = prepareTracks_l(&tracksToRemove); 1973 1974 // prevent any changes in effect chain list and in each effect chain 1975 // during mixing and effect process as the audio buffers could be deleted 1976 // or modified if an effect is created or deleted 1977 lockEffectChains_l(effectChains); 1978 } 1979 1980 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1981 threadLoop_mix(); 1982 } else { 1983 threadLoop_sleepTime(); 1984 } 1985 1986 if (isSuspended()) { 1987 sleepTime = suspendSleepTimeUs(); 1988 mBytesWritten += mixBufferSize; 1989 } 1990 1991 // only process effects if we're going to write 1992 if (sleepTime == 0) { 1993 for (size_t i = 0; i < effectChains.size(); i ++) { 1994 effectChains[i]->process_l(); 1995 } 1996 } 1997 1998 // enable changes in effect chain 1999 unlockEffectChains(effectChains); 2000 2001 // sleepTime == 0 means we must write to audio hardware 2002 if (sleepTime == 0) { 2003 2004 threadLoop_write(); 2005 2006if (mType == MIXER) { 2007 // write blocked detection 2008 nsecs_t now = systemTime(); 2009 nsecs_t delta = now - mLastWriteTime; 2010 if (!mStandby && delta > maxPeriod) { 2011 mNumDelayedWrites++; 2012 if ((now - lastWarning) > kWarningThrottleNs) { 2013 ATRACE_NAME("underrun"); 2014 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2015 ns2ms(delta), mNumDelayedWrites, this); 2016 lastWarning = now; 2017 } 2018 } 2019} 2020 2021 mStandby = false; 2022 } else { 2023 usleep(sleepTime); 2024 } 2025 2026 // Finally let go of removed track(s), without the lock held 2027 // since we can't guarantee the destructors won't acquire that 2028 // same lock. This will also mutate and push a new fast mixer state. 2029 threadLoop_removeTracks(tracksToRemove); 2030 tracksToRemove.clear(); 2031 2032 // FIXME I don't understand the need for this here; 2033 // it was in the original code but maybe the 2034 // assignment in saveOutputTracks() makes this unnecessary? 2035 clearOutputTracks(); 2036 2037 // Effect chains will be actually deleted here if they were removed from 2038 // mEffectChains list during mixing or effects processing 2039 effectChains.clear(); 2040 2041 // FIXME Note that the above .clear() is no longer necessary since effectChains 2042 // is now local to this block, but will keep it for now (at least until merge done). 2043 } 2044 2045 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2046 if (mType == MIXER || mType == DIRECT) { 2047 // put output stream into standby mode 2048 if (!mStandby) { 2049 mOutput->stream->common.standby(&mOutput->stream->common); 2050 } 2051 } 2052 2053 releaseWakeLock(); 2054 2055 ALOGV("Thread %p type %d exiting", this, mType); 2056 return false; 2057} 2058 2059 2060// ---------------------------------------------------------------------------- 2061 2062AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2063 audio_io_handle_t id, audio_devices_t device, type_t type) 2064 : PlaybackThread(audioFlinger, output, id, device, type), 2065 // mAudioMixer below 2066 // mFastMixer below 2067 mFastMixerFutex(0) 2068 // mOutputSink below 2069 // mPipeSink below 2070 // mNormalSink below 2071{ 2072 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2073 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2074 "mFrameCount=%d, mNormalFrameCount=%d", 2075 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2076 mNormalFrameCount); 2077 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2078 2079 // FIXME - Current mixer implementation only supports stereo output 2080 if (mChannelCount != FCC_2) { 2081 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2082 } 2083 2084 // create an NBAIO sink for the HAL output stream, and negotiate 2085 mOutputSink = new AudioStreamOutSink(output->stream); 2086 size_t numCounterOffers = 0; 2087 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2088 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2089 ALOG_ASSERT(index == 0); 2090 2091 // initialize fast mixer depending on configuration 2092 bool initFastMixer; 2093 switch (kUseFastMixer) { 2094 case FastMixer_Never: 2095 initFastMixer = false; 2096 break; 2097 case FastMixer_Always: 2098 initFastMixer = true; 2099 break; 2100 case FastMixer_Static: 2101 case FastMixer_Dynamic: 2102 initFastMixer = mFrameCount < mNormalFrameCount; 2103 break; 2104 } 2105 if (initFastMixer) { 2106 2107 // create a MonoPipe to connect our submix to FastMixer 2108 NBAIO_Format format = mOutputSink->format(); 2109 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2110 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2111 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2112 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2113 const NBAIO_Format offers[1] = {format}; 2114 size_t numCounterOffers = 0; 2115 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2116 ALOG_ASSERT(index == 0); 2117 monoPipe->setAvgFrames((mScreenState & 1) ? 2118 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2119 mPipeSink = monoPipe; 2120 2121#ifdef TEE_SINK 2122 if (mTeeSinkOutputEnabled) { 2123 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2124 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2125 numCounterOffers = 0; 2126 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2127 ALOG_ASSERT(index == 0); 2128 mTeeSink = teeSink; 2129 PipeReader *teeSource = new PipeReader(*teeSink); 2130 numCounterOffers = 0; 2131 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2132 ALOG_ASSERT(index == 0); 2133 mTeeSource = teeSource; 2134 } 2135#endif 2136 2137 // create fast mixer and configure it initially with just one fast track for our submix 2138 mFastMixer = new FastMixer(); 2139 FastMixerStateQueue *sq = mFastMixer->sq(); 2140#ifdef STATE_QUEUE_DUMP 2141 sq->setObserverDump(&mStateQueueObserverDump); 2142 sq->setMutatorDump(&mStateQueueMutatorDump); 2143#endif 2144 FastMixerState *state = sq->begin(); 2145 FastTrack *fastTrack = &state->mFastTracks[0]; 2146 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2147 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2148 fastTrack->mVolumeProvider = NULL; 2149 fastTrack->mGeneration++; 2150 state->mFastTracksGen++; 2151 state->mTrackMask = 1; 2152 // fast mixer will use the HAL output sink 2153 state->mOutputSink = mOutputSink.get(); 2154 state->mOutputSinkGen++; 2155 state->mFrameCount = mFrameCount; 2156 state->mCommand = FastMixerState::COLD_IDLE; 2157 // already done in constructor initialization list 2158 //mFastMixerFutex = 0; 2159 state->mColdFutexAddr = &mFastMixerFutex; 2160 state->mColdGen++; 2161 state->mDumpState = &mFastMixerDumpState; 2162#ifdef TEE_SINK 2163 state->mTeeSink = mTeeSink.get(); 2164#endif 2165 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2166 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2167 sq->end(); 2168 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2169 2170 // start the fast mixer 2171 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2172 pid_t tid = mFastMixer->getTid(); 2173 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2174 if (err != 0) { 2175 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2176 kPriorityFastMixer, getpid_cached, tid, err); 2177 } 2178 2179#ifdef AUDIO_WATCHDOG 2180 // create and start the watchdog 2181 mAudioWatchdog = new AudioWatchdog(); 2182 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2183 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2184 tid = mAudioWatchdog->getTid(); 2185 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2186 if (err != 0) { 2187 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2188 kPriorityFastMixer, getpid_cached, tid, err); 2189 } 2190#endif 2191 2192 } else { 2193 mFastMixer = NULL; 2194 } 2195 2196 switch (kUseFastMixer) { 2197 case FastMixer_Never: 2198 case FastMixer_Dynamic: 2199 mNormalSink = mOutputSink; 2200 break; 2201 case FastMixer_Always: 2202 mNormalSink = mPipeSink; 2203 break; 2204 case FastMixer_Static: 2205 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2206 break; 2207 } 2208} 2209 2210AudioFlinger::MixerThread::~MixerThread() 2211{ 2212 if (mFastMixer != NULL) { 2213 FastMixerStateQueue *sq = mFastMixer->sq(); 2214 FastMixerState *state = sq->begin(); 2215 if (state->mCommand == FastMixerState::COLD_IDLE) { 2216 int32_t old = android_atomic_inc(&mFastMixerFutex); 2217 if (old == -1) { 2218 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2219 } 2220 } 2221 state->mCommand = FastMixerState::EXIT; 2222 sq->end(); 2223 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2224 mFastMixer->join(); 2225 // Though the fast mixer thread has exited, it's state queue is still valid. 2226 // We'll use that extract the final state which contains one remaining fast track 2227 // corresponding to our sub-mix. 2228 state = sq->begin(); 2229 ALOG_ASSERT(state->mTrackMask == 1); 2230 FastTrack *fastTrack = &state->mFastTracks[0]; 2231 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2232 delete fastTrack->mBufferProvider; 2233 sq->end(false /*didModify*/); 2234 delete mFastMixer; 2235#ifdef AUDIO_WATCHDOG 2236 if (mAudioWatchdog != 0) { 2237 mAudioWatchdog->requestExit(); 2238 mAudioWatchdog->requestExitAndWait(); 2239 mAudioWatchdog.clear(); 2240 } 2241#endif 2242 } 2243 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2244 delete mAudioMixer; 2245} 2246 2247 2248uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2249{ 2250 if (mFastMixer != NULL) { 2251 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2252 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2253 } 2254 return latency; 2255} 2256 2257 2258void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2259{ 2260 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2261} 2262 2263void AudioFlinger::MixerThread::threadLoop_write() 2264{ 2265 // FIXME we should only do one push per cycle; confirm this is true 2266 // Start the fast mixer if it's not already running 2267 if (mFastMixer != NULL) { 2268 FastMixerStateQueue *sq = mFastMixer->sq(); 2269 FastMixerState *state = sq->begin(); 2270 if (state->mCommand != FastMixerState::MIX_WRITE && 2271 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2272 if (state->mCommand == FastMixerState::COLD_IDLE) { 2273 int32_t old = android_atomic_inc(&mFastMixerFutex); 2274 if (old == -1) { 2275 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2276 } 2277#ifdef AUDIO_WATCHDOG 2278 if (mAudioWatchdog != 0) { 2279 mAudioWatchdog->resume(); 2280 } 2281#endif 2282 } 2283 state->mCommand = FastMixerState::MIX_WRITE; 2284 sq->end(); 2285 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2286 if (kUseFastMixer == FastMixer_Dynamic) { 2287 mNormalSink = mPipeSink; 2288 } 2289 } else { 2290 sq->end(false /*didModify*/); 2291 } 2292 } 2293 PlaybackThread::threadLoop_write(); 2294} 2295 2296void AudioFlinger::MixerThread::threadLoop_standby() 2297{ 2298 // Idle the fast mixer if it's currently running 2299 if (mFastMixer != NULL) { 2300 FastMixerStateQueue *sq = mFastMixer->sq(); 2301 FastMixerState *state = sq->begin(); 2302 if (!(state->mCommand & FastMixerState::IDLE)) { 2303 state->mCommand = FastMixerState::COLD_IDLE; 2304 state->mColdFutexAddr = &mFastMixerFutex; 2305 state->mColdGen++; 2306 mFastMixerFutex = 0; 2307 sq->end(); 2308 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2309 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2310 if (kUseFastMixer == FastMixer_Dynamic) { 2311 mNormalSink = mOutputSink; 2312 } 2313#ifdef AUDIO_WATCHDOG 2314 if (mAudioWatchdog != 0) { 2315 mAudioWatchdog->pause(); 2316 } 2317#endif 2318 } else { 2319 sq->end(false /*didModify*/); 2320 } 2321 } 2322 PlaybackThread::threadLoop_standby(); 2323} 2324 2325// shared by MIXER and DIRECT, overridden by DUPLICATING 2326void AudioFlinger::PlaybackThread::threadLoop_standby() 2327{ 2328 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2329 mOutput->stream->common.standby(&mOutput->stream->common); 2330} 2331 2332void AudioFlinger::MixerThread::threadLoop_mix() 2333{ 2334 // obtain the presentation timestamp of the next output buffer 2335 int64_t pts; 2336 status_t status = INVALID_OPERATION; 2337 2338 if (mNormalSink != 0) { 2339 status = mNormalSink->getNextWriteTimestamp(&pts); 2340 } else { 2341 status = mOutputSink->getNextWriteTimestamp(&pts); 2342 } 2343 2344 if (status != NO_ERROR) { 2345 pts = AudioBufferProvider::kInvalidPTS; 2346 } 2347 2348 // mix buffers... 2349 mAudioMixer->process(pts); 2350 // increase sleep time progressively when application underrun condition clears. 2351 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2352 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2353 // such that we would underrun the audio HAL. 2354 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2355 sleepTimeShift--; 2356 } 2357 sleepTime = 0; 2358 standbyTime = systemTime() + standbyDelay; 2359 //TODO: delay standby when effects have a tail 2360} 2361 2362void AudioFlinger::MixerThread::threadLoop_sleepTime() 2363{ 2364 // If no tracks are ready, sleep once for the duration of an output 2365 // buffer size, then write 0s to the output 2366 if (sleepTime == 0) { 2367 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2368 sleepTime = activeSleepTime >> sleepTimeShift; 2369 if (sleepTime < kMinThreadSleepTimeUs) { 2370 sleepTime = kMinThreadSleepTimeUs; 2371 } 2372 // reduce sleep time in case of consecutive application underruns to avoid 2373 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2374 // duration we would end up writing less data than needed by the audio HAL if 2375 // the condition persists. 2376 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2377 sleepTimeShift++; 2378 } 2379 } else { 2380 sleepTime = idleSleepTime; 2381 } 2382 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2383 memset (mMixBuffer, 0, mixBufferSize); 2384 sleepTime = 0; 2385 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2386 "anticipated start"); 2387 } 2388 // TODO add standby time extension fct of effect tail 2389} 2390 2391// prepareTracks_l() must be called with ThreadBase::mLock held 2392AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2393 Vector< sp<Track> > *tracksToRemove) 2394{ 2395 2396 mixer_state mixerStatus = MIXER_IDLE; 2397 // find out which tracks need to be processed 2398 size_t count = mActiveTracks.size(); 2399 size_t mixedTracks = 0; 2400 size_t tracksWithEffect = 0; 2401 // counts only _active_ fast tracks 2402 size_t fastTracks = 0; 2403 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2404 2405 float masterVolume = mMasterVolume; 2406 bool masterMute = mMasterMute; 2407 2408 if (masterMute) { 2409 masterVolume = 0; 2410 } 2411 // Delegate master volume control to effect in output mix effect chain if needed 2412 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2413 if (chain != 0) { 2414 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2415 chain->setVolume_l(&v, &v); 2416 masterVolume = (float)((v + (1 << 23)) >> 24); 2417 chain.clear(); 2418 } 2419 2420 // prepare a new state to push 2421 FastMixerStateQueue *sq = NULL; 2422 FastMixerState *state = NULL; 2423 bool didModify = false; 2424 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2425 if (mFastMixer != NULL) { 2426 sq = mFastMixer->sq(); 2427 state = sq->begin(); 2428 } 2429 2430 for (size_t i=0 ; i<count ; i++) { 2431 sp<Track> t = mActiveTracks[i].promote(); 2432 if (t == 0) { 2433 continue; 2434 } 2435 2436 // this const just means the local variable doesn't change 2437 Track* const track = t.get(); 2438 2439 // process fast tracks 2440 if (track->isFastTrack()) { 2441 2442 // It's theoretically possible (though unlikely) for a fast track to be created 2443 // and then removed within the same normal mix cycle. This is not a problem, as 2444 // the track never becomes active so it's fast mixer slot is never touched. 2445 // The converse, of removing an (active) track and then creating a new track 2446 // at the identical fast mixer slot within the same normal mix cycle, 2447 // is impossible because the slot isn't marked available until the end of each cycle. 2448 int j = track->mFastIndex; 2449 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2450 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2451 FastTrack *fastTrack = &state->mFastTracks[j]; 2452 2453 // Determine whether the track is currently in underrun condition, 2454 // and whether it had a recent underrun. 2455 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2456 FastTrackUnderruns underruns = ftDump->mUnderruns; 2457 uint32_t recentFull = (underruns.mBitFields.mFull - 2458 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2459 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2460 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2461 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2462 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2463 uint32_t recentUnderruns = recentPartial + recentEmpty; 2464 track->mObservedUnderruns = underruns; 2465 // don't count underruns that occur while stopping or pausing 2466 // or stopped which can occur when flush() is called while active 2467 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2468 track->mUnderrunCount += recentUnderruns; 2469 } 2470 2471 // This is similar to the state machine for normal tracks, 2472 // with a few modifications for fast tracks. 2473 bool isActive = true; 2474 switch (track->mState) { 2475 case TrackBase::STOPPING_1: 2476 // track stays active in STOPPING_1 state until first underrun 2477 if (recentUnderruns > 0) { 2478 track->mState = TrackBase::STOPPING_2; 2479 } 2480 break; 2481 case TrackBase::PAUSING: 2482 // ramp down is not yet implemented 2483 track->setPaused(); 2484 break; 2485 case TrackBase::RESUMING: 2486 // ramp up is not yet implemented 2487 track->mState = TrackBase::ACTIVE; 2488 break; 2489 case TrackBase::ACTIVE: 2490 if (recentFull > 0 || recentPartial > 0) { 2491 // track has provided at least some frames recently: reset retry count 2492 track->mRetryCount = kMaxTrackRetries; 2493 } 2494 if (recentUnderruns == 0) { 2495 // no recent underruns: stay active 2496 break; 2497 } 2498 // there has recently been an underrun of some kind 2499 if (track->sharedBuffer() == 0) { 2500 // were any of the recent underruns "empty" (no frames available)? 2501 if (recentEmpty == 0) { 2502 // no, then ignore the partial underruns as they are allowed indefinitely 2503 break; 2504 } 2505 // there has recently been an "empty" underrun: decrement the retry counter 2506 if (--(track->mRetryCount) > 0) { 2507 break; 2508 } 2509 // indicate to client process that the track was disabled because of underrun; 2510 // it will then automatically call start() when data is available 2511 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2512 // remove from active list, but state remains ACTIVE [confusing but true] 2513 isActive = false; 2514 break; 2515 } 2516 // fall through 2517 case TrackBase::STOPPING_2: 2518 case TrackBase::PAUSED: 2519 case TrackBase::TERMINATED: 2520 case TrackBase::STOPPED: 2521 case TrackBase::FLUSHED: // flush() while active 2522 // Check for presentation complete if track is inactive 2523 // We have consumed all the buffers of this track. 2524 // This would be incomplete if we auto-paused on underrun 2525 { 2526 size_t audioHALFrames = 2527 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2528 size_t framesWritten = mBytesWritten / mFrameSize; 2529 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2530 // track stays in active list until presentation is complete 2531 break; 2532 } 2533 } 2534 if (track->isStopping_2()) { 2535 track->mState = TrackBase::STOPPED; 2536 } 2537 if (track->isStopped()) { 2538 // Can't reset directly, as fast mixer is still polling this track 2539 // track->reset(); 2540 // So instead mark this track as needing to be reset after push with ack 2541 resetMask |= 1 << i; 2542 } 2543 isActive = false; 2544 break; 2545 case TrackBase::IDLE: 2546 default: 2547 LOG_FATAL("unexpected track state %d", track->mState); 2548 } 2549 2550 if (isActive) { 2551 // was it previously inactive? 2552 if (!(state->mTrackMask & (1 << j))) { 2553 ExtendedAudioBufferProvider *eabp = track; 2554 VolumeProvider *vp = track; 2555 fastTrack->mBufferProvider = eabp; 2556 fastTrack->mVolumeProvider = vp; 2557 fastTrack->mSampleRate = track->mSampleRate; 2558 fastTrack->mChannelMask = track->mChannelMask; 2559 fastTrack->mGeneration++; 2560 state->mTrackMask |= 1 << j; 2561 didModify = true; 2562 // no acknowledgement required for newly active tracks 2563 } 2564 // cache the combined master volume and stream type volume for fast mixer; this 2565 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2566 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2567 ++fastTracks; 2568 } else { 2569 // was it previously active? 2570 if (state->mTrackMask & (1 << j)) { 2571 fastTrack->mBufferProvider = NULL; 2572 fastTrack->mGeneration++; 2573 state->mTrackMask &= ~(1 << j); 2574 didModify = true; 2575 // If any fast tracks were removed, we must wait for acknowledgement 2576 // because we're about to decrement the last sp<> on those tracks. 2577 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2578 } else { 2579 LOG_FATAL("fast track %d should have been active", j); 2580 } 2581 tracksToRemove->add(track); 2582 // Avoids a misleading display in dumpsys 2583 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2584 } 2585 continue; 2586 } 2587 2588 { // local variable scope to avoid goto warning 2589 2590 audio_track_cblk_t* cblk = track->cblk(); 2591 2592 // The first time a track is added we wait 2593 // for all its buffers to be filled before processing it 2594 int name = track->name(); 2595 // make sure that we have enough frames to mix one full buffer. 2596 // enforce this condition only once to enable draining the buffer in case the client 2597 // app does not call stop() and relies on underrun to stop: 2598 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2599 // during last round 2600 uint32_t minFrames = 1; 2601 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2602 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2603 if (t->sampleRate() == mSampleRate) { 2604 minFrames = mNormalFrameCount; 2605 } else { 2606 // +1 for rounding and +1 for additional sample needed for interpolation 2607 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2608 // add frames already consumed but not yet released by the resampler 2609 // because cblk->framesReady() will include these frames 2610 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2611 // the minimum track buffer size is normally twice the number of frames necessary 2612 // to fill one buffer and the resampler should not leave more than one buffer worth 2613 // of unreleased frames after each pass, but just in case... 2614 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2615 } 2616 } 2617 if ((track->framesReady() >= minFrames) && track->isReady() && 2618 !track->isPaused() && !track->isTerminated()) 2619 { 2620 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2621 this); 2622 2623 mixedTracks++; 2624 2625 // track->mainBuffer() != mMixBuffer means there is an effect chain 2626 // connected to the track 2627 chain.clear(); 2628 if (track->mainBuffer() != mMixBuffer) { 2629 chain = getEffectChain_l(track->sessionId()); 2630 // Delegate volume control to effect in track effect chain if needed 2631 if (chain != 0) { 2632 tracksWithEffect++; 2633 } else { 2634 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2635 "session %d", 2636 name, track->sessionId()); 2637 } 2638 } 2639 2640 2641 int param = AudioMixer::VOLUME; 2642 if (track->mFillingUpStatus == Track::FS_FILLED) { 2643 // no ramp for the first volume setting 2644 track->mFillingUpStatus = Track::FS_ACTIVE; 2645 if (track->mState == TrackBase::RESUMING) { 2646 track->mState = TrackBase::ACTIVE; 2647 param = AudioMixer::RAMP_VOLUME; 2648 } 2649 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2650 } else if (cblk->server != 0) { 2651 // If the track is stopped before the first frame was mixed, 2652 // do not apply ramp 2653 param = AudioMixer::RAMP_VOLUME; 2654 } 2655 2656 // compute volume for this track 2657 uint32_t vl, vr, va; 2658 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2659 vl = vr = va = 0; 2660 if (track->isPausing()) { 2661 track->setPaused(); 2662 } 2663 } else { 2664 2665 // read original volumes with volume control 2666 float typeVolume = mStreamTypes[track->streamType()].volume; 2667 float v = masterVolume * typeVolume; 2668 ServerProxy *proxy = track->mServerProxy; 2669 uint32_t vlr = proxy->getVolumeLR(); 2670 vl = vlr & 0xFFFF; 2671 vr = vlr >> 16; 2672 // track volumes come from shared memory, so can't be trusted and must be clamped 2673 if (vl > MAX_GAIN_INT) { 2674 ALOGV("Track left volume out of range: %04X", vl); 2675 vl = MAX_GAIN_INT; 2676 } 2677 if (vr > MAX_GAIN_INT) { 2678 ALOGV("Track right volume out of range: %04X", vr); 2679 vr = MAX_GAIN_INT; 2680 } 2681 // now apply the master volume and stream type volume 2682 vl = (uint32_t)(v * vl) << 12; 2683 vr = (uint32_t)(v * vr) << 12; 2684 // assuming master volume and stream type volume each go up to 1.0, 2685 // vl and vr are now in 8.24 format 2686 2687 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2688 // send level comes from shared memory and so may be corrupt 2689 if (sendLevel > MAX_GAIN_INT) { 2690 ALOGV("Track send level out of range: %04X", sendLevel); 2691 sendLevel = MAX_GAIN_INT; 2692 } 2693 va = (uint32_t)(v * sendLevel); 2694 } 2695 // Delegate volume control to effect in track effect chain if needed 2696 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2697 // Do not ramp volume if volume is controlled by effect 2698 param = AudioMixer::VOLUME; 2699 track->mHasVolumeController = true; 2700 } else { 2701 // force no volume ramp when volume controller was just disabled or removed 2702 // from effect chain to avoid volume spike 2703 if (track->mHasVolumeController) { 2704 param = AudioMixer::VOLUME; 2705 } 2706 track->mHasVolumeController = false; 2707 } 2708 2709 // Convert volumes from 8.24 to 4.12 format 2710 // This additional clamping is needed in case chain->setVolume_l() overshot 2711 vl = (vl + (1 << 11)) >> 12; 2712 if (vl > MAX_GAIN_INT) { 2713 vl = MAX_GAIN_INT; 2714 } 2715 vr = (vr + (1 << 11)) >> 12; 2716 if (vr > MAX_GAIN_INT) { 2717 vr = MAX_GAIN_INT; 2718 } 2719 2720 if (va > MAX_GAIN_INT) { 2721 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2722 } 2723 2724 // XXX: these things DON'T need to be done each time 2725 mAudioMixer->setBufferProvider(name, track); 2726 mAudioMixer->enable(name); 2727 2728 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2729 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2730 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2731 mAudioMixer->setParameter( 2732 name, 2733 AudioMixer::TRACK, 2734 AudioMixer::FORMAT, (void *)track->format()); 2735 mAudioMixer->setParameter( 2736 name, 2737 AudioMixer::TRACK, 2738 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2739 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2740 uint32_t maxSampleRate = mSampleRate * 2; 2741 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2742 if (reqSampleRate == 0) { 2743 reqSampleRate = mSampleRate; 2744 } else if (reqSampleRate > maxSampleRate) { 2745 reqSampleRate = maxSampleRate; 2746 } 2747 mAudioMixer->setParameter( 2748 name, 2749 AudioMixer::RESAMPLE, 2750 AudioMixer::SAMPLE_RATE, 2751 (void *)reqSampleRate); 2752 mAudioMixer->setParameter( 2753 name, 2754 AudioMixer::TRACK, 2755 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2756 mAudioMixer->setParameter( 2757 name, 2758 AudioMixer::TRACK, 2759 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2760 2761 // reset retry count 2762 track->mRetryCount = kMaxTrackRetries; 2763 2764 // If one track is ready, set the mixer ready if: 2765 // - the mixer was not ready during previous round OR 2766 // - no other track is not ready 2767 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2768 mixerStatus != MIXER_TRACKS_ENABLED) { 2769 mixerStatus = MIXER_TRACKS_READY; 2770 } 2771 } else { 2772 // clear effect chain input buffer if an active track underruns to avoid sending 2773 // previous audio buffer again to effects 2774 chain = getEffectChain_l(track->sessionId()); 2775 if (chain != 0) { 2776 chain->clearInputBuffer(); 2777 } 2778 2779 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2780 cblk->server, this); 2781 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2782 track->isStopped() || track->isPaused()) { 2783 // We have consumed all the buffers of this track. 2784 // Remove it from the list of active tracks. 2785 // TODO: use actual buffer filling status instead of latency when available from 2786 // audio HAL 2787 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2788 size_t framesWritten = mBytesWritten / mFrameSize; 2789 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2790 if (track->isStopped()) { 2791 track->reset(); 2792 } 2793 tracksToRemove->add(track); 2794 } 2795 } else { 2796 track->mUnderrunCount++; 2797 // No buffers for this track. Give it a few chances to 2798 // fill a buffer, then remove it from active list. 2799 if (--(track->mRetryCount) <= 0) { 2800 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2801 tracksToRemove->add(track); 2802 // indicate to client process that the track was disabled because of underrun; 2803 // it will then automatically call start() when data is available 2804 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2805 // If one track is not ready, mark the mixer also not ready if: 2806 // - the mixer was ready during previous round OR 2807 // - no other track is ready 2808 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2809 mixerStatus != MIXER_TRACKS_READY) { 2810 mixerStatus = MIXER_TRACKS_ENABLED; 2811 } 2812 } 2813 mAudioMixer->disable(name); 2814 } 2815 2816 } // local variable scope to avoid goto warning 2817track_is_ready: ; 2818 2819 } 2820 2821 // Push the new FastMixer state if necessary 2822 bool pauseAudioWatchdog = false; 2823 if (didModify) { 2824 state->mFastTracksGen++; 2825 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2826 if (kUseFastMixer == FastMixer_Dynamic && 2827 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2828 state->mCommand = FastMixerState::COLD_IDLE; 2829 state->mColdFutexAddr = &mFastMixerFutex; 2830 state->mColdGen++; 2831 mFastMixerFutex = 0; 2832 if (kUseFastMixer == FastMixer_Dynamic) { 2833 mNormalSink = mOutputSink; 2834 } 2835 // If we go into cold idle, need to wait for acknowledgement 2836 // so that fast mixer stops doing I/O. 2837 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2838 pauseAudioWatchdog = true; 2839 } 2840 } 2841 if (sq != NULL) { 2842 sq->end(didModify); 2843 sq->push(block); 2844 } 2845#ifdef AUDIO_WATCHDOG 2846 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2847 mAudioWatchdog->pause(); 2848 } 2849#endif 2850 2851 // Now perform the deferred reset on fast tracks that have stopped 2852 while (resetMask != 0) { 2853 size_t i = __builtin_ctz(resetMask); 2854 ALOG_ASSERT(i < count); 2855 resetMask &= ~(1 << i); 2856 sp<Track> t = mActiveTracks[i].promote(); 2857 if (t == 0) { 2858 continue; 2859 } 2860 Track* track = t.get(); 2861 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2862 track->reset(); 2863 } 2864 2865 // remove all the tracks that need to be... 2866 count = tracksToRemove->size(); 2867 if (CC_UNLIKELY(count)) { 2868 for (size_t i=0 ; i<count ; i++) { 2869 const sp<Track>& track = tracksToRemove->itemAt(i); 2870 mActiveTracks.remove(track); 2871 if (track->mainBuffer() != mMixBuffer) { 2872 chain = getEffectChain_l(track->sessionId()); 2873 if (chain != 0) { 2874 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2875 track->sessionId()); 2876 chain->decActiveTrackCnt(); 2877 } 2878 } 2879 if (track->isTerminated()) { 2880 removeTrack_l(track); 2881 } 2882 } 2883 } 2884 2885 // mix buffer must be cleared if all tracks are connected to an 2886 // effect chain as in this case the mixer will not write to 2887 // mix buffer and track effects will accumulate into it 2888 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2889 (mixedTracks == 0 && fastTracks > 0)) { 2890 // FIXME as a performance optimization, should remember previous zero status 2891 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2892 } 2893 2894 // if any fast tracks, then status is ready 2895 mMixerStatusIgnoringFastTracks = mixerStatus; 2896 if (fastTracks > 0) { 2897 mixerStatus = MIXER_TRACKS_READY; 2898 } 2899 return mixerStatus; 2900} 2901 2902// getTrackName_l() must be called with ThreadBase::mLock held 2903int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2904{ 2905 return mAudioMixer->getTrackName(channelMask, sessionId); 2906} 2907 2908// deleteTrackName_l() must be called with ThreadBase::mLock held 2909void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2910{ 2911 ALOGV("remove track (%d) and delete from mixer", name); 2912 mAudioMixer->deleteTrackName(name); 2913} 2914 2915// checkForNewParameters_l() must be called with ThreadBase::mLock held 2916bool AudioFlinger::MixerThread::checkForNewParameters_l() 2917{ 2918 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2919 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2920 bool reconfig = false; 2921 2922 while (!mNewParameters.isEmpty()) { 2923 2924 if (mFastMixer != NULL) { 2925 FastMixerStateQueue *sq = mFastMixer->sq(); 2926 FastMixerState *state = sq->begin(); 2927 if (!(state->mCommand & FastMixerState::IDLE)) { 2928 previousCommand = state->mCommand; 2929 state->mCommand = FastMixerState::HOT_IDLE; 2930 sq->end(); 2931 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2932 } else { 2933 sq->end(false /*didModify*/); 2934 } 2935 } 2936 2937 status_t status = NO_ERROR; 2938 String8 keyValuePair = mNewParameters[0]; 2939 AudioParameter param = AudioParameter(keyValuePair); 2940 int value; 2941 2942 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2943 reconfig = true; 2944 } 2945 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2946 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2947 status = BAD_VALUE; 2948 } else { 2949 reconfig = true; 2950 } 2951 } 2952 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2953 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2954 status = BAD_VALUE; 2955 } else { 2956 reconfig = true; 2957 } 2958 } 2959 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2960 // do not accept frame count changes if tracks are open as the track buffer 2961 // size depends on frame count and correct behavior would not be guaranteed 2962 // if frame count is changed after track creation 2963 if (!mTracks.isEmpty()) { 2964 status = INVALID_OPERATION; 2965 } else { 2966 reconfig = true; 2967 } 2968 } 2969 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2970#ifdef ADD_BATTERY_DATA 2971 // when changing the audio output device, call addBatteryData to notify 2972 // the change 2973 if (mOutDevice != value) { 2974 uint32_t params = 0; 2975 // check whether speaker is on 2976 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2977 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2978 } 2979 2980 audio_devices_t deviceWithoutSpeaker 2981 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2982 // check if any other device (except speaker) is on 2983 if (value & deviceWithoutSpeaker ) { 2984 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2985 } 2986 2987 if (params != 0) { 2988 addBatteryData(params); 2989 } 2990 } 2991#endif 2992 2993 // forward device change to effects that have requested to be 2994 // aware of attached audio device. 2995 if (value != AUDIO_DEVICE_NONE) { 2996 mOutDevice = value; 2997 for (size_t i = 0; i < mEffectChains.size(); i++) { 2998 mEffectChains[i]->setDevice_l(mOutDevice); 2999 } 3000 } 3001 } 3002 3003 if (status == NO_ERROR) { 3004 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3005 keyValuePair.string()); 3006 if (!mStandby && status == INVALID_OPERATION) { 3007 mOutput->stream->common.standby(&mOutput->stream->common); 3008 mStandby = true; 3009 mBytesWritten = 0; 3010 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3011 keyValuePair.string()); 3012 } 3013 if (status == NO_ERROR && reconfig) { 3014 delete mAudioMixer; 3015 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3016 mAudioMixer = NULL; 3017 readOutputParameters(); 3018 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3019 for (size_t i = 0; i < mTracks.size() ; i++) { 3020 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3021 if (name < 0) { 3022 break; 3023 } 3024 mTracks[i]->mName = name; 3025 } 3026 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3027 } 3028 } 3029 3030 mNewParameters.removeAt(0); 3031 3032 mParamStatus = status; 3033 mParamCond.signal(); 3034 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3035 // already timed out waiting for the status and will never signal the condition. 3036 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3037 } 3038 3039 if (!(previousCommand & FastMixerState::IDLE)) { 3040 ALOG_ASSERT(mFastMixer != NULL); 3041 FastMixerStateQueue *sq = mFastMixer->sq(); 3042 FastMixerState *state = sq->begin(); 3043 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3044 state->mCommand = previousCommand; 3045 sq->end(); 3046 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3047 } 3048 3049 return reconfig; 3050} 3051 3052 3053void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3054{ 3055 const size_t SIZE = 256; 3056 char buffer[SIZE]; 3057 String8 result; 3058 3059 PlaybackThread::dumpInternals(fd, args); 3060 3061 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3062 result.append(buffer); 3063 write(fd, result.string(), result.size()); 3064 3065 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3066 FastMixerDumpState copy = mFastMixerDumpState; 3067 copy.dump(fd); 3068 3069#ifdef STATE_QUEUE_DUMP 3070 // Similar for state queue 3071 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3072 observerCopy.dump(fd); 3073 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3074 mutatorCopy.dump(fd); 3075#endif 3076 3077#ifdef TEE_SINK 3078 // Write the tee output to a .wav file 3079 dumpTee(fd, mTeeSource, mId); 3080#endif 3081 3082#ifdef AUDIO_WATCHDOG 3083 if (mAudioWatchdog != 0) { 3084 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3085 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3086 wdCopy.dump(fd); 3087 } 3088#endif 3089} 3090 3091uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3092{ 3093 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3094} 3095 3096uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3097{ 3098 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3099} 3100 3101void AudioFlinger::MixerThread::cacheParameters_l() 3102{ 3103 PlaybackThread::cacheParameters_l(); 3104 3105 // FIXME: Relaxed timing because of a certain device that can't meet latency 3106 // Should be reduced to 2x after the vendor fixes the driver issue 3107 // increase threshold again due to low power audio mode. The way this warning 3108 // threshold is calculated and its usefulness should be reconsidered anyway. 3109 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3110} 3111 3112// ---------------------------------------------------------------------------- 3113 3114AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3115 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3116 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3117 // mLeftVolFloat, mRightVolFloat 3118{ 3119} 3120 3121AudioFlinger::DirectOutputThread::~DirectOutputThread() 3122{ 3123} 3124 3125AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3126 Vector< sp<Track> > *tracksToRemove 3127) 3128{ 3129 size_t count = mActiveTracks.size(); 3130 mixer_state mixerStatus = MIXER_IDLE; 3131 3132 // find out which tracks need to be processed 3133 for (size_t i = 0; i < count; i++) { 3134 sp<Track> t = mActiveTracks[i].promote(); 3135 // The track died recently 3136 if (t == 0) { 3137 continue; 3138 } 3139 3140 Track* const track = t.get(); 3141 audio_track_cblk_t* cblk = track->cblk(); 3142 3143 // The first time a track is added we wait 3144 // for all its buffers to be filled before processing it 3145 uint32_t minFrames; 3146 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3147 minFrames = mNormalFrameCount; 3148 } else { 3149 minFrames = 1; 3150 } 3151 if ((track->framesReady() >= minFrames) && track->isReady() && 3152 !track->isPaused() && !track->isTerminated()) 3153 { 3154 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3155 3156 if (track->mFillingUpStatus == Track::FS_FILLED) { 3157 track->mFillingUpStatus = Track::FS_ACTIVE; 3158 mLeftVolFloat = mRightVolFloat = 0; 3159 if (track->mState == TrackBase::RESUMING) { 3160 track->mState = TrackBase::ACTIVE; 3161 } 3162 } 3163 3164 // compute volume for this track 3165 float left, right; 3166 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3167 left = right = 0; 3168 if (track->isPausing()) { 3169 track->setPaused(); 3170 } 3171 } else { 3172 float typeVolume = mStreamTypes[track->streamType()].volume; 3173 float v = mMasterVolume * typeVolume; 3174 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3175 float v_clamped = v * (vlr & 0xFFFF); 3176 if (v_clamped > MAX_GAIN) { 3177 v_clamped = MAX_GAIN; 3178 } 3179 left = v_clamped/MAX_GAIN; 3180 v_clamped = v * (vlr >> 16); 3181 if (v_clamped > MAX_GAIN) { 3182 v_clamped = MAX_GAIN; 3183 } 3184 right = v_clamped/MAX_GAIN; 3185 } 3186 // Only consider last track started for volume and mixer state control. 3187 // This is the last entry in mActiveTracks unless a track underruns. 3188 // As we only care about the transition phase between two tracks on a 3189 // direct output, it is not a problem to ignore the underrun case. 3190 if (i == (count - 1)) { 3191 if (left != mLeftVolFloat || right != mRightVolFloat) { 3192 mLeftVolFloat = left; 3193 mRightVolFloat = right; 3194 3195 // Convert volumes from float to 8.24 3196 uint32_t vl = (uint32_t)(left * (1 << 24)); 3197 uint32_t vr = (uint32_t)(right * (1 << 24)); 3198 3199 // Delegate volume control to effect in track effect chain if needed 3200 // only one effect chain can be present on DirectOutputThread, so if 3201 // there is one, the track is connected to it 3202 if (!mEffectChains.isEmpty()) { 3203 // Do not ramp volume if volume is controlled by effect 3204 mEffectChains[0]->setVolume_l(&vl, &vr); 3205 left = (float)vl / (1 << 24); 3206 right = (float)vr / (1 << 24); 3207 } 3208 mOutput->stream->set_volume(mOutput->stream, left, right); 3209 } 3210 3211 // reset retry count 3212 track->mRetryCount = kMaxTrackRetriesDirect; 3213 mActiveTrack = t; 3214 mixerStatus = MIXER_TRACKS_READY; 3215 } 3216 } else { 3217 // clear effect chain input buffer if the last active track started underruns 3218 // to avoid sending previous audio buffer again to effects 3219 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3220 mEffectChains[0]->clearInputBuffer(); 3221 } 3222 3223 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3224 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3225 track->isStopped() || track->isPaused()) { 3226 // We have consumed all the buffers of this track. 3227 // Remove it from the list of active tracks. 3228 // TODO: implement behavior for compressed audio 3229 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3230 size_t framesWritten = mBytesWritten / mFrameSize; 3231 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3232 if (track->isStopped()) { 3233 track->reset(); 3234 } 3235 tracksToRemove->add(track); 3236 } 3237 } else { 3238 // No buffers for this track. Give it a few chances to 3239 // fill a buffer, then remove it from active list. 3240 // Only consider last track started for mixer state control 3241 if (--(track->mRetryCount) <= 0) { 3242 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3243 tracksToRemove->add(track); 3244 } else if (i == (count -1)){ 3245 mixerStatus = MIXER_TRACKS_ENABLED; 3246 } 3247 } 3248 } 3249 } 3250 3251 // remove all the tracks that need to be... 3252 count = tracksToRemove->size(); 3253 if (CC_UNLIKELY(count)) { 3254 for (size_t i = 0 ; i < count ; i++) { 3255 const sp<Track>& track = tracksToRemove->itemAt(i); 3256 mActiveTracks.remove(track); 3257 if (!mEffectChains.isEmpty()) { 3258 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3259 track->sessionId()); 3260 mEffectChains[0]->decActiveTrackCnt(); 3261 } 3262 if (track->isTerminated()) { 3263 removeTrack_l(track); 3264 } 3265 } 3266 } 3267 3268 return mixerStatus; 3269} 3270 3271void AudioFlinger::DirectOutputThread::threadLoop_mix() 3272{ 3273 AudioBufferProvider::Buffer buffer; 3274 size_t frameCount = mFrameCount; 3275 int8_t *curBuf = (int8_t *)mMixBuffer; 3276 // output audio to hardware 3277 while (frameCount) { 3278 buffer.frameCount = frameCount; 3279 mActiveTrack->getNextBuffer(&buffer); 3280 if (CC_UNLIKELY(buffer.raw == NULL)) { 3281 memset(curBuf, 0, frameCount * mFrameSize); 3282 break; 3283 } 3284 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3285 frameCount -= buffer.frameCount; 3286 curBuf += buffer.frameCount * mFrameSize; 3287 mActiveTrack->releaseBuffer(&buffer); 3288 } 3289 sleepTime = 0; 3290 standbyTime = systemTime() + standbyDelay; 3291 mActiveTrack.clear(); 3292 3293} 3294 3295void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3296{ 3297 if (sleepTime == 0) { 3298 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3299 sleepTime = activeSleepTime; 3300 } else { 3301 sleepTime = idleSleepTime; 3302 } 3303 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3304 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3305 sleepTime = 0; 3306 } 3307} 3308 3309// getTrackName_l() must be called with ThreadBase::mLock held 3310int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3311 int sessionId) 3312{ 3313 return 0; 3314} 3315 3316// deleteTrackName_l() must be called with ThreadBase::mLock held 3317void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3318{ 3319} 3320 3321// checkForNewParameters_l() must be called with ThreadBase::mLock held 3322bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3323{ 3324 bool reconfig = false; 3325 3326 while (!mNewParameters.isEmpty()) { 3327 status_t status = NO_ERROR; 3328 String8 keyValuePair = mNewParameters[0]; 3329 AudioParameter param = AudioParameter(keyValuePair); 3330 int value; 3331 3332 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3333 // do not accept frame count changes if tracks are open as the track buffer 3334 // size depends on frame count and correct behavior would not be garantied 3335 // if frame count is changed after track creation 3336 if (!mTracks.isEmpty()) { 3337 status = INVALID_OPERATION; 3338 } else { 3339 reconfig = true; 3340 } 3341 } 3342 if (status == NO_ERROR) { 3343 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3344 keyValuePair.string()); 3345 if (!mStandby && status == INVALID_OPERATION) { 3346 mOutput->stream->common.standby(&mOutput->stream->common); 3347 mStandby = true; 3348 mBytesWritten = 0; 3349 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3350 keyValuePair.string()); 3351 } 3352 if (status == NO_ERROR && reconfig) { 3353 readOutputParameters(); 3354 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3355 } 3356 } 3357 3358 mNewParameters.removeAt(0); 3359 3360 mParamStatus = status; 3361 mParamCond.signal(); 3362 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3363 // already timed out waiting for the status and will never signal the condition. 3364 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3365 } 3366 return reconfig; 3367} 3368 3369uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3370{ 3371 uint32_t time; 3372 if (audio_is_linear_pcm(mFormat)) { 3373 time = PlaybackThread::activeSleepTimeUs(); 3374 } else { 3375 time = 10000; 3376 } 3377 return time; 3378} 3379 3380uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3381{ 3382 uint32_t time; 3383 if (audio_is_linear_pcm(mFormat)) { 3384 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3385 } else { 3386 time = 10000; 3387 } 3388 return time; 3389} 3390 3391uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3392{ 3393 uint32_t time; 3394 if (audio_is_linear_pcm(mFormat)) { 3395 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3396 } else { 3397 time = 10000; 3398 } 3399 return time; 3400} 3401 3402void AudioFlinger::DirectOutputThread::cacheParameters_l() 3403{ 3404 PlaybackThread::cacheParameters_l(); 3405 3406 // use shorter standby delay as on normal output to release 3407 // hardware resources as soon as possible 3408 standbyDelay = microseconds(activeSleepTime*2); 3409} 3410 3411// ---------------------------------------------------------------------------- 3412 3413AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3414 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3415 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3416 DUPLICATING), 3417 mWaitTimeMs(UINT_MAX) 3418{ 3419 addOutputTrack(mainThread); 3420} 3421 3422AudioFlinger::DuplicatingThread::~DuplicatingThread() 3423{ 3424 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3425 mOutputTracks[i]->destroy(); 3426 } 3427} 3428 3429void AudioFlinger::DuplicatingThread::threadLoop_mix() 3430{ 3431 // mix buffers... 3432 if (outputsReady(outputTracks)) { 3433 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3434 } else { 3435 memset(mMixBuffer, 0, mixBufferSize); 3436 } 3437 sleepTime = 0; 3438 writeFrames = mNormalFrameCount; 3439 standbyTime = systemTime() + standbyDelay; 3440} 3441 3442void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3443{ 3444 if (sleepTime == 0) { 3445 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3446 sleepTime = activeSleepTime; 3447 } else { 3448 sleepTime = idleSleepTime; 3449 } 3450 } else if (mBytesWritten != 0) { 3451 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3452 writeFrames = mNormalFrameCount; 3453 memset(mMixBuffer, 0, mixBufferSize); 3454 } else { 3455 // flush remaining overflow buffers in output tracks 3456 writeFrames = 0; 3457 } 3458 sleepTime = 0; 3459 } 3460} 3461 3462void AudioFlinger::DuplicatingThread::threadLoop_write() 3463{ 3464 for (size_t i = 0; i < outputTracks.size(); i++) { 3465 outputTracks[i]->write(mMixBuffer, writeFrames); 3466 } 3467 mBytesWritten += mixBufferSize; 3468} 3469 3470void AudioFlinger::DuplicatingThread::threadLoop_standby() 3471{ 3472 // DuplicatingThread implements standby by stopping all tracks 3473 for (size_t i = 0; i < outputTracks.size(); i++) { 3474 outputTracks[i]->stop(); 3475 } 3476} 3477 3478void AudioFlinger::DuplicatingThread::saveOutputTracks() 3479{ 3480 outputTracks = mOutputTracks; 3481} 3482 3483void AudioFlinger::DuplicatingThread::clearOutputTracks() 3484{ 3485 outputTracks.clear(); 3486} 3487 3488void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3489{ 3490 Mutex::Autolock _l(mLock); 3491 // FIXME explain this formula 3492 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3493 OutputTrack *outputTrack = new OutputTrack(thread, 3494 this, 3495 mSampleRate, 3496 mFormat, 3497 mChannelMask, 3498 frameCount); 3499 if (outputTrack->cblk() != NULL) { 3500 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3501 mOutputTracks.add(outputTrack); 3502 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3503 updateWaitTime_l(); 3504 } 3505} 3506 3507void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3508{ 3509 Mutex::Autolock _l(mLock); 3510 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3511 if (mOutputTracks[i]->thread() == thread) { 3512 mOutputTracks[i]->destroy(); 3513 mOutputTracks.removeAt(i); 3514 updateWaitTime_l(); 3515 return; 3516 } 3517 } 3518 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3519} 3520 3521// caller must hold mLock 3522void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3523{ 3524 mWaitTimeMs = UINT_MAX; 3525 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3526 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3527 if (strong != 0) { 3528 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3529 if (waitTimeMs < mWaitTimeMs) { 3530 mWaitTimeMs = waitTimeMs; 3531 } 3532 } 3533 } 3534} 3535 3536 3537bool AudioFlinger::DuplicatingThread::outputsReady( 3538 const SortedVector< sp<OutputTrack> > &outputTracks) 3539{ 3540 for (size_t i = 0; i < outputTracks.size(); i++) { 3541 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3542 if (thread == 0) { 3543 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3544 outputTracks[i].get()); 3545 return false; 3546 } 3547 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3548 // see note at standby() declaration 3549 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3550 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3551 thread.get()); 3552 return false; 3553 } 3554 } 3555 return true; 3556} 3557 3558uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3559{ 3560 return (mWaitTimeMs * 1000) / 2; 3561} 3562 3563void AudioFlinger::DuplicatingThread::cacheParameters_l() 3564{ 3565 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3566 updateWaitTime_l(); 3567 3568 MixerThread::cacheParameters_l(); 3569} 3570 3571// ---------------------------------------------------------------------------- 3572// Record 3573// ---------------------------------------------------------------------------- 3574 3575AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3576 AudioStreamIn *input, 3577 uint32_t sampleRate, 3578 audio_channel_mask_t channelMask, 3579 audio_io_handle_t id, 3580 audio_devices_t outDevice, 3581 audio_devices_t inDevice 3582#ifdef TEE_SINK 3583 , const sp<NBAIO_Sink>& teeSink 3584#endif 3585 ) : 3586 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3587 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3588 // mRsmpInIndex and mInputBytes set by readInputParameters() 3589 mReqChannelCount(popcount(channelMask)), 3590 mReqSampleRate(sampleRate) 3591 // mBytesRead is only meaningful while active, and so is cleared in start() 3592 // (but might be better to also clear here for dump?) 3593#ifdef TEE_SINK 3594 , mTeeSink(teeSink) 3595#endif 3596{ 3597 snprintf(mName, kNameLength, "AudioIn_%X", id); 3598 3599 readInputParameters(); 3600 3601} 3602 3603 3604AudioFlinger::RecordThread::~RecordThread() 3605{ 3606 delete[] mRsmpInBuffer; 3607 delete mResampler; 3608 delete[] mRsmpOutBuffer; 3609} 3610 3611void AudioFlinger::RecordThread::onFirstRef() 3612{ 3613 run(mName, PRIORITY_URGENT_AUDIO); 3614} 3615 3616status_t AudioFlinger::RecordThread::readyToRun() 3617{ 3618 status_t status = initCheck(); 3619 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3620 return status; 3621} 3622 3623bool AudioFlinger::RecordThread::threadLoop() 3624{ 3625 AudioBufferProvider::Buffer buffer; 3626 sp<RecordTrack> activeTrack; 3627 Vector< sp<EffectChain> > effectChains; 3628 3629 nsecs_t lastWarning = 0; 3630 3631 inputStandBy(); 3632 acquireWakeLock(); 3633 3634 // used to verify we've read at least once before evaluating how many bytes were read 3635 bool readOnce = false; 3636 3637 // start recording 3638 while (!exitPending()) { 3639 3640 processConfigEvents(); 3641 3642 { // scope for mLock 3643 Mutex::Autolock _l(mLock); 3644 checkForNewParameters_l(); 3645 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3646 standby(); 3647 3648 if (exitPending()) { 3649 break; 3650 } 3651 3652 releaseWakeLock_l(); 3653 ALOGV("RecordThread: loop stopping"); 3654 // go to sleep 3655 mWaitWorkCV.wait(mLock); 3656 ALOGV("RecordThread: loop starting"); 3657 acquireWakeLock_l(); 3658 continue; 3659 } 3660 if (mActiveTrack != 0) { 3661 if (mActiveTrack->mState == TrackBase::PAUSING) { 3662 standby(); 3663 mActiveTrack.clear(); 3664 mStartStopCond.broadcast(); 3665 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3666 if (mReqChannelCount != mActiveTrack->channelCount()) { 3667 mActiveTrack.clear(); 3668 mStartStopCond.broadcast(); 3669 } else if (readOnce) { 3670 // record start succeeds only if first read from audio input 3671 // succeeds 3672 if (mBytesRead >= 0) { 3673 mActiveTrack->mState = TrackBase::ACTIVE; 3674 } else { 3675 mActiveTrack.clear(); 3676 } 3677 mStartStopCond.broadcast(); 3678 } 3679 mStandby = false; 3680 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3681 removeTrack_l(mActiveTrack); 3682 mActiveTrack.clear(); 3683 } 3684 } 3685 lockEffectChains_l(effectChains); 3686 } 3687 3688 if (mActiveTrack != 0) { 3689 if (mActiveTrack->mState != TrackBase::ACTIVE && 3690 mActiveTrack->mState != TrackBase::RESUMING) { 3691 unlockEffectChains(effectChains); 3692 usleep(kRecordThreadSleepUs); 3693 continue; 3694 } 3695 for (size_t i = 0; i < effectChains.size(); i ++) { 3696 effectChains[i]->process_l(); 3697 } 3698 3699 buffer.frameCount = mFrameCount; 3700 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3701 readOnce = true; 3702 size_t framesOut = buffer.frameCount; 3703 if (mResampler == NULL) { 3704 // no resampling 3705 while (framesOut) { 3706 size_t framesIn = mFrameCount - mRsmpInIndex; 3707 if (framesIn) { 3708 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3709 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3710 mActiveTrack->mFrameSize; 3711 if (framesIn > framesOut) 3712 framesIn = framesOut; 3713 mRsmpInIndex += framesIn; 3714 framesOut -= framesIn; 3715 if (mChannelCount == mReqChannelCount || 3716 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3717 memcpy(dst, src, framesIn * mFrameSize); 3718 } else { 3719 if (mChannelCount == 1) { 3720 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3721 (int16_t *)src, framesIn); 3722 } else { 3723 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3724 (int16_t *)src, framesIn); 3725 } 3726 } 3727 } 3728 if (framesOut && mFrameCount == mRsmpInIndex) { 3729 void *readInto; 3730 if (framesOut == mFrameCount && 3731 (mChannelCount == mReqChannelCount || 3732 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3733 readInto = buffer.raw; 3734 framesOut = 0; 3735 } else { 3736 readInto = mRsmpInBuffer; 3737 mRsmpInIndex = 0; 3738 } 3739 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3740 mInputBytes); 3741 if (mBytesRead <= 0) { 3742 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3743 { 3744 ALOGE("Error reading audio input"); 3745 // Force input into standby so that it tries to 3746 // recover at next read attempt 3747 inputStandBy(); 3748 usleep(kRecordThreadSleepUs); 3749 } 3750 mRsmpInIndex = mFrameCount; 3751 framesOut = 0; 3752 buffer.frameCount = 0; 3753 } 3754#ifdef TEE_SINK 3755 else if (mTeeSink != 0) { 3756 (void) mTeeSink->write(readInto, 3757 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3758 } 3759#endif 3760 } 3761 } 3762 } else { 3763 // resampling 3764 3765 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3766 // alter output frame count as if we were expecting stereo samples 3767 if (mChannelCount == 1 && mReqChannelCount == 1) { 3768 framesOut >>= 1; 3769 } 3770 mResampler->resample(mRsmpOutBuffer, framesOut, 3771 this /* AudioBufferProvider* */); 3772 // ditherAndClamp() works as long as all buffers returned by 3773 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3774 if (mChannelCount == 2 && mReqChannelCount == 1) { 3775 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3776 // the resampler always outputs stereo samples: 3777 // do post stereo to mono conversion 3778 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3779 framesOut); 3780 } else { 3781 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3782 } 3783 3784 } 3785 if (mFramestoDrop == 0) { 3786 mActiveTrack->releaseBuffer(&buffer); 3787 } else { 3788 if (mFramestoDrop > 0) { 3789 mFramestoDrop -= buffer.frameCount; 3790 if (mFramestoDrop <= 0) { 3791 clearSyncStartEvent(); 3792 } 3793 } else { 3794 mFramestoDrop += buffer.frameCount; 3795 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3796 mSyncStartEvent->isCancelled()) { 3797 ALOGW("Synced record %s, session %d, trigger session %d", 3798 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3799 mActiveTrack->sessionId(), 3800 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3801 clearSyncStartEvent(); 3802 } 3803 } 3804 } 3805 mActiveTrack->clearOverflow(); 3806 } 3807 // client isn't retrieving buffers fast enough 3808 else { 3809 if (!mActiveTrack->setOverflow()) { 3810 nsecs_t now = systemTime(); 3811 if ((now - lastWarning) > kWarningThrottleNs) { 3812 ALOGW("RecordThread: buffer overflow"); 3813 lastWarning = now; 3814 } 3815 } 3816 // Release the processor for a while before asking for a new buffer. 3817 // This will give the application more chance to read from the buffer and 3818 // clear the overflow. 3819 usleep(kRecordThreadSleepUs); 3820 } 3821 } 3822 // enable changes in effect chain 3823 unlockEffectChains(effectChains); 3824 effectChains.clear(); 3825 } 3826 3827 standby(); 3828 3829 { 3830 Mutex::Autolock _l(mLock); 3831 mActiveTrack.clear(); 3832 mStartStopCond.broadcast(); 3833 } 3834 3835 releaseWakeLock(); 3836 3837 ALOGV("RecordThread %p exiting", this); 3838 return false; 3839} 3840 3841void AudioFlinger::RecordThread::standby() 3842{ 3843 if (!mStandby) { 3844 inputStandBy(); 3845 mStandby = true; 3846 } 3847} 3848 3849void AudioFlinger::RecordThread::inputStandBy() 3850{ 3851 mInput->stream->common.standby(&mInput->stream->common); 3852} 3853 3854sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3855 const sp<AudioFlinger::Client>& client, 3856 uint32_t sampleRate, 3857 audio_format_t format, 3858 audio_channel_mask_t channelMask, 3859 size_t frameCount, 3860 int sessionId, 3861 IAudioFlinger::track_flags_t flags, 3862 pid_t tid, 3863 status_t *status) 3864{ 3865 sp<RecordTrack> track; 3866 status_t lStatus; 3867 3868 lStatus = initCheck(); 3869 if (lStatus != NO_ERROR) { 3870 ALOGE("Audio driver not initialized."); 3871 goto Exit; 3872 } 3873 3874 // FIXME use flags and tid similar to createTrack_l() 3875 3876 { // scope for mLock 3877 Mutex::Autolock _l(mLock); 3878 3879 track = new RecordTrack(this, client, sampleRate, 3880 format, channelMask, frameCount, sessionId); 3881 3882 if (track->getCblk() == 0) { 3883 lStatus = NO_MEMORY; 3884 goto Exit; 3885 } 3886 mTracks.add(track); 3887 3888 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3889 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3890 mAudioFlinger->btNrecIsOff(); 3891 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3892 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3893 } 3894 lStatus = NO_ERROR; 3895 3896Exit: 3897 if (status) { 3898 *status = lStatus; 3899 } 3900 return track; 3901} 3902 3903status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3904 AudioSystem::sync_event_t event, 3905 int triggerSession) 3906{ 3907 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3908 sp<ThreadBase> strongMe = this; 3909 status_t status = NO_ERROR; 3910 3911 if (event == AudioSystem::SYNC_EVENT_NONE) { 3912 clearSyncStartEvent(); 3913 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3914 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3915 triggerSession, 3916 recordTrack->sessionId(), 3917 syncStartEventCallback, 3918 this); 3919 // Sync event can be cancelled by the trigger session if the track is not in a 3920 // compatible state in which case we start record immediately 3921 if (mSyncStartEvent->isCancelled()) { 3922 clearSyncStartEvent(); 3923 } else { 3924 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3925 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3926 } 3927 } 3928 3929 { 3930 AutoMutex lock(mLock); 3931 if (mActiveTrack != 0) { 3932 if (recordTrack != mActiveTrack.get()) { 3933 status = -EBUSY; 3934 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3935 mActiveTrack->mState = TrackBase::ACTIVE; 3936 } 3937 return status; 3938 } 3939 3940 recordTrack->mState = TrackBase::IDLE; 3941 mActiveTrack = recordTrack; 3942 mLock.unlock(); 3943 status_t status = AudioSystem::startInput(mId); 3944 mLock.lock(); 3945 if (status != NO_ERROR) { 3946 mActiveTrack.clear(); 3947 clearSyncStartEvent(); 3948 return status; 3949 } 3950 mRsmpInIndex = mFrameCount; 3951 mBytesRead = 0; 3952 if (mResampler != NULL) { 3953 mResampler->reset(); 3954 } 3955 mActiveTrack->mState = TrackBase::RESUMING; 3956 // signal thread to start 3957 ALOGV("Signal record thread"); 3958 mWaitWorkCV.broadcast(); 3959 // do not wait for mStartStopCond if exiting 3960 if (exitPending()) { 3961 mActiveTrack.clear(); 3962 status = INVALID_OPERATION; 3963 goto startError; 3964 } 3965 mStartStopCond.wait(mLock); 3966 if (mActiveTrack == 0) { 3967 ALOGV("Record failed to start"); 3968 status = BAD_VALUE; 3969 goto startError; 3970 } 3971 ALOGV("Record started OK"); 3972 return status; 3973 } 3974startError: 3975 AudioSystem::stopInput(mId); 3976 clearSyncStartEvent(); 3977 return status; 3978} 3979 3980void AudioFlinger::RecordThread::clearSyncStartEvent() 3981{ 3982 if (mSyncStartEvent != 0) { 3983 mSyncStartEvent->cancel(); 3984 } 3985 mSyncStartEvent.clear(); 3986 mFramestoDrop = 0; 3987} 3988 3989void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3990{ 3991 sp<SyncEvent> strongEvent = event.promote(); 3992 3993 if (strongEvent != 0) { 3994 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3995 me->handleSyncStartEvent(strongEvent); 3996 } 3997} 3998 3999void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4000{ 4001 if (event == mSyncStartEvent) { 4002 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4003 // from audio HAL 4004 mFramestoDrop = mFrameCount * 2; 4005 } 4006} 4007 4008bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4009 ALOGV("RecordThread::stop"); 4010 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4011 return false; 4012 } 4013 recordTrack->mState = TrackBase::PAUSING; 4014 // do not wait for mStartStopCond if exiting 4015 if (exitPending()) { 4016 return true; 4017 } 4018 mStartStopCond.wait(mLock); 4019 // if we have been restarted, recordTrack == mActiveTrack.get() here 4020 if (exitPending() || recordTrack != mActiveTrack.get()) { 4021 ALOGV("Record stopped OK"); 4022 return true; 4023 } 4024 return false; 4025} 4026 4027bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4028{ 4029 return false; 4030} 4031 4032status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4033{ 4034#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4035 if (!isValidSyncEvent(event)) { 4036 return BAD_VALUE; 4037 } 4038 4039 int eventSession = event->triggerSession(); 4040 status_t ret = NAME_NOT_FOUND; 4041 4042 Mutex::Autolock _l(mLock); 4043 4044 for (size_t i = 0; i < mTracks.size(); i++) { 4045 sp<RecordTrack> track = mTracks[i]; 4046 if (eventSession == track->sessionId()) { 4047 (void) track->setSyncEvent(event); 4048 ret = NO_ERROR; 4049 } 4050 } 4051 return ret; 4052#else 4053 return BAD_VALUE; 4054#endif 4055} 4056 4057// destroyTrack_l() must be called with ThreadBase::mLock held 4058void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4059{ 4060 track->mState = TrackBase::TERMINATED; 4061 // active tracks are removed by threadLoop() 4062 if (mActiveTrack != track) { 4063 removeTrack_l(track); 4064 } 4065} 4066 4067void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4068{ 4069 mTracks.remove(track); 4070 // need anything related to effects here? 4071} 4072 4073void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4074{ 4075 dumpInternals(fd, args); 4076 dumpTracks(fd, args); 4077 dumpEffectChains(fd, args); 4078} 4079 4080void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4081{ 4082 const size_t SIZE = 256; 4083 char buffer[SIZE]; 4084 String8 result; 4085 4086 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4087 result.append(buffer); 4088 4089 if (mActiveTrack != 0) { 4090 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4091 result.append(buffer); 4092 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4093 result.append(buffer); 4094 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4095 result.append(buffer); 4096 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4097 result.append(buffer); 4098 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4099 result.append(buffer); 4100 } else { 4101 result.append("No active record client\n"); 4102 } 4103 4104 write(fd, result.string(), result.size()); 4105 4106 dumpBase(fd, args); 4107} 4108 4109void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4110{ 4111 const size_t SIZE = 256; 4112 char buffer[SIZE]; 4113 String8 result; 4114 4115 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4116 result.append(buffer); 4117 RecordTrack::appendDumpHeader(result); 4118 for (size_t i = 0; i < mTracks.size(); ++i) { 4119 sp<RecordTrack> track = mTracks[i]; 4120 if (track != 0) { 4121 track->dump(buffer, SIZE); 4122 result.append(buffer); 4123 } 4124 } 4125 4126 if (mActiveTrack != 0) { 4127 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4128 result.append(buffer); 4129 RecordTrack::appendDumpHeader(result); 4130 mActiveTrack->dump(buffer, SIZE); 4131 result.append(buffer); 4132 4133 } 4134 write(fd, result.string(), result.size()); 4135} 4136 4137// AudioBufferProvider interface 4138status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4139{ 4140 size_t framesReq = buffer->frameCount; 4141 size_t framesReady = mFrameCount - mRsmpInIndex; 4142 int channelCount; 4143 4144 if (framesReady == 0) { 4145 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4146 if (mBytesRead <= 0) { 4147 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4148 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4149 // Force input into standby so that it tries to 4150 // recover at next read attempt 4151 inputStandBy(); 4152 usleep(kRecordThreadSleepUs); 4153 } 4154 buffer->raw = NULL; 4155 buffer->frameCount = 0; 4156 return NOT_ENOUGH_DATA; 4157 } 4158 mRsmpInIndex = 0; 4159 framesReady = mFrameCount; 4160 } 4161 4162 if (framesReq > framesReady) { 4163 framesReq = framesReady; 4164 } 4165 4166 if (mChannelCount == 1 && mReqChannelCount == 2) { 4167 channelCount = 1; 4168 } else { 4169 channelCount = 2; 4170 } 4171 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4172 buffer->frameCount = framesReq; 4173 return NO_ERROR; 4174} 4175 4176// AudioBufferProvider interface 4177void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4178{ 4179 mRsmpInIndex += buffer->frameCount; 4180 buffer->frameCount = 0; 4181} 4182 4183bool AudioFlinger::RecordThread::checkForNewParameters_l() 4184{ 4185 bool reconfig = false; 4186 4187 while (!mNewParameters.isEmpty()) { 4188 status_t status = NO_ERROR; 4189 String8 keyValuePair = mNewParameters[0]; 4190 AudioParameter param = AudioParameter(keyValuePair); 4191 int value; 4192 audio_format_t reqFormat = mFormat; 4193 uint32_t reqSamplingRate = mReqSampleRate; 4194 uint32_t reqChannelCount = mReqChannelCount; 4195 4196 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4197 reqSamplingRate = value; 4198 reconfig = true; 4199 } 4200 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4201 reqFormat = (audio_format_t) value; 4202 reconfig = true; 4203 } 4204 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4205 reqChannelCount = popcount(value); 4206 reconfig = true; 4207 } 4208 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4209 // do not accept frame count changes if tracks are open as the track buffer 4210 // size depends on frame count and correct behavior would not be guaranteed 4211 // if frame count is changed after track creation 4212 if (mActiveTrack != 0) { 4213 status = INVALID_OPERATION; 4214 } else { 4215 reconfig = true; 4216 } 4217 } 4218 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4219 // forward device change to effects that have requested to be 4220 // aware of attached audio device. 4221 for (size_t i = 0; i < mEffectChains.size(); i++) { 4222 mEffectChains[i]->setDevice_l(value); 4223 } 4224 4225 // store input device and output device but do not forward output device to audio HAL. 4226 // Note that status is ignored by the caller for output device 4227 // (see AudioFlinger::setParameters() 4228 if (audio_is_output_devices(value)) { 4229 mOutDevice = value; 4230 status = BAD_VALUE; 4231 } else { 4232 mInDevice = value; 4233 // disable AEC and NS if the device is a BT SCO headset supporting those 4234 // pre processings 4235 if (mTracks.size() > 0) { 4236 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4237 mAudioFlinger->btNrecIsOff(); 4238 for (size_t i = 0; i < mTracks.size(); i++) { 4239 sp<RecordTrack> track = mTracks[i]; 4240 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4241 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4242 } 4243 } 4244 } 4245 } 4246 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4247 mAudioSource != (audio_source_t)value) { 4248 // forward device change to effects that have requested to be 4249 // aware of attached audio device. 4250 for (size_t i = 0; i < mEffectChains.size(); i++) { 4251 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4252 } 4253 mAudioSource = (audio_source_t)value; 4254 } 4255 if (status == NO_ERROR) { 4256 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4257 keyValuePair.string()); 4258 if (status == INVALID_OPERATION) { 4259 inputStandBy(); 4260 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4261 keyValuePair.string()); 4262 } 4263 if (reconfig) { 4264 if (status == BAD_VALUE && 4265 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4266 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4267 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4268 <= (2 * reqSamplingRate)) && 4269 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4270 <= FCC_2 && 4271 (reqChannelCount <= FCC_2)) { 4272 status = NO_ERROR; 4273 } 4274 if (status == NO_ERROR) { 4275 readInputParameters(); 4276 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4277 } 4278 } 4279 } 4280 4281 mNewParameters.removeAt(0); 4282 4283 mParamStatus = status; 4284 mParamCond.signal(); 4285 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4286 // already timed out waiting for the status and will never signal the condition. 4287 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4288 } 4289 return reconfig; 4290} 4291 4292String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4293{ 4294 char *s; 4295 String8 out_s8 = String8(); 4296 4297 Mutex::Autolock _l(mLock); 4298 if (initCheck() != NO_ERROR) { 4299 return out_s8; 4300 } 4301 4302 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4303 out_s8 = String8(s); 4304 free(s); 4305 return out_s8; 4306} 4307 4308void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4309 AudioSystem::OutputDescriptor desc; 4310 void *param2 = NULL; 4311 4312 switch (event) { 4313 case AudioSystem::INPUT_OPENED: 4314 case AudioSystem::INPUT_CONFIG_CHANGED: 4315 desc.channels = mChannelMask; 4316 desc.samplingRate = mSampleRate; 4317 desc.format = mFormat; 4318 desc.frameCount = mFrameCount; 4319 desc.latency = 0; 4320 param2 = &desc; 4321 break; 4322 4323 case AudioSystem::INPUT_CLOSED: 4324 default: 4325 break; 4326 } 4327 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4328} 4329 4330void AudioFlinger::RecordThread::readInputParameters() 4331{ 4332 delete mRsmpInBuffer; 4333 // mRsmpInBuffer is always assigned a new[] below 4334 delete mRsmpOutBuffer; 4335 mRsmpOutBuffer = NULL; 4336 delete mResampler; 4337 mResampler = NULL; 4338 4339 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4340 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4341 mChannelCount = (uint16_t)popcount(mChannelMask); 4342 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4343 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4344 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4345 mFrameCount = mInputBytes / mFrameSize; 4346 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4347 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4348 4349 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4350 { 4351 int channelCount; 4352 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4353 // stereo to mono post process as the resampler always outputs stereo. 4354 if (mChannelCount == 1 && mReqChannelCount == 2) { 4355 channelCount = 1; 4356 } else { 4357 channelCount = 2; 4358 } 4359 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4360 mResampler->setSampleRate(mSampleRate); 4361 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4362 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4363 4364 // optmization: if mono to mono, alter input frame count as if we were inputing 4365 // stereo samples 4366 if (mChannelCount == 1 && mReqChannelCount == 1) { 4367 mFrameCount >>= 1; 4368 } 4369 4370 } 4371 mRsmpInIndex = mFrameCount; 4372} 4373 4374unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4375{ 4376 Mutex::Autolock _l(mLock); 4377 if (initCheck() != NO_ERROR) { 4378 return 0; 4379 } 4380 4381 return mInput->stream->get_input_frames_lost(mInput->stream); 4382} 4383 4384uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4385{ 4386 Mutex::Autolock _l(mLock); 4387 uint32_t result = 0; 4388 if (getEffectChain_l(sessionId) != 0) { 4389 result = EFFECT_SESSION; 4390 } 4391 4392 for (size_t i = 0; i < mTracks.size(); ++i) { 4393 if (sessionId == mTracks[i]->sessionId()) { 4394 result |= TRACK_SESSION; 4395 break; 4396 } 4397 } 4398 4399 return result; 4400} 4401 4402KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4403{ 4404 KeyedVector<int, bool> ids; 4405 Mutex::Autolock _l(mLock); 4406 for (size_t j = 0; j < mTracks.size(); ++j) { 4407 sp<RecordThread::RecordTrack> track = mTracks[j]; 4408 int sessionId = track->sessionId(); 4409 if (ids.indexOfKey(sessionId) < 0) { 4410 ids.add(sessionId, true); 4411 } 4412 } 4413 return ids; 4414} 4415 4416AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4417{ 4418 Mutex::Autolock _l(mLock); 4419 AudioStreamIn *input = mInput; 4420 mInput = NULL; 4421 return input; 4422} 4423 4424// this method must always be called either with ThreadBase mLock held or inside the thread loop 4425audio_stream_t* AudioFlinger::RecordThread::stream() const 4426{ 4427 if (mInput == NULL) { 4428 return NULL; 4429 } 4430 return &mInput->stream->common; 4431} 4432 4433status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4434{ 4435 // only one chain per input thread 4436 if (mEffectChains.size() != 0) { 4437 return INVALID_OPERATION; 4438 } 4439 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4440 4441 chain->setInBuffer(NULL); 4442 chain->setOutBuffer(NULL); 4443 4444 checkSuspendOnAddEffectChain_l(chain); 4445 4446 mEffectChains.add(chain); 4447 4448 return NO_ERROR; 4449} 4450 4451size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4452{ 4453 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4454 ALOGW_IF(mEffectChains.size() != 1, 4455 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4456 chain.get(), mEffectChains.size(), this); 4457 if (mEffectChains.size() == 1) { 4458 mEffectChains.removeAt(0); 4459 } 4460 return 0; 4461} 4462 4463}; // namespace android 4464