Threads.cpp revision 46909e7eb074ce1b95b8a411eb71154f53f84f77
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include <math.h> 24#include <fcntl.h> 25#include <sys/stat.h> 26#include <cutils/properties.h> 27#include <cutils/compiler.h> 28#include <utils/Log.h> 29#include <utils/Trace.h> 30 31#include <private/media/AudioTrackShared.h> 32#include <hardware/audio.h> 33#include <audio_effects/effect_ns.h> 34#include <audio_effects/effect_aec.h> 35#include <audio_utils/primitives.h> 36 37// NBAIO implementations 38#include <media/nbaio/AudioStreamOutSink.h> 39#include <media/nbaio/MonoPipe.h> 40#include <media/nbaio/MonoPipeReader.h> 41#include <media/nbaio/Pipe.h> 42#include <media/nbaio/PipeReader.h> 43#include <media/nbaio/SourceAudioBufferProvider.h> 44 45#include <powermanager/PowerManager.h> 46 47#include <common_time/cc_helper.h> 48#include <common_time/local_clock.h> 49 50#include "AudioFlinger.h" 51#include "AudioMixer.h" 52#include "FastMixer.h" 53#include "ServiceUtilities.h" 54#include "SchedulingPolicyService.h" 55 56#undef ADD_BATTERY_DATA 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait for setParameters to complete 102static const nsecs_t kSetParametersTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal mix buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalMixBufferSizeMs = 20; 111// maximum normal mix buffer size 112static const uint32_t kMaxNormalMixBufferSizeMs = 24; 113 114// Whether to use fast mixer 115static const enum { 116 FastMixer_Never, // never initialize or use: for debugging only 117 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 118 // normal mixer multiplier is 1 119 FastMixer_Static, // initialize if needed, then use all the time if initialized, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 // FIXME for FastMixer_Dynamic: 124 // Supporting this option will require fixing HALs that can't handle large writes. 125 // For example, one HAL implementation returns an error from a large write, 126 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 127 // We could either fix the HAL implementations, or provide a wrapper that breaks 128 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 129} kUseFastMixer = FastMixer_Static; 130 131// Priorities for requestPriority 132static const int kPriorityAudioApp = 2; 133static const int kPriorityFastMixer = 3; 134 135// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 136// for the track. The client then sub-divides this into smaller buffers for its use. 137// Currently the client uses double-buffering by default, but doesn't tell us about that. 138// So for now we just assume that client is double-buffered. 139// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 140// N-buffering, so AudioFlinger could allocate the right amount of memory. 141// See the client's minBufCount and mNotificationFramesAct calculations for details. 142static const int kFastTrackMultiplier = 2; 143 144// ---------------------------------------------------------------------------- 145 146#ifdef ADD_BATTERY_DATA 147// To collect the amplifier usage 148static void addBatteryData(uint32_t params) { 149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150 if (service == NULL) { 151 // it already logged 152 return; 153 } 154 155 service->addBatteryData(params); 156} 157#endif 158 159 160// ---------------------------------------------------------------------------- 161// CPU Stats 162// ---------------------------------------------------------------------------- 163 164class CpuStats { 165public: 166 CpuStats(); 167 void sample(const String8 &title); 168#ifdef DEBUG_CPU_USAGE 169private: 170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172 173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174 175 int mCpuNum; // thread's current CPU number 176 int mCpukHz; // frequency of thread's current CPU in kHz 177#endif 178}; 179 180CpuStats::CpuStats() 181#ifdef DEBUG_CPU_USAGE 182 : mCpuNum(-1), mCpukHz(-1) 183#endif 184{ 185} 186 187void CpuStats::sample(const String8 &title) { 188#ifdef DEBUG_CPU_USAGE 189 // get current thread's delta CPU time in wall clock ns 190 double wcNs; 191 bool valid = mCpuUsage.sampleAndEnable(wcNs); 192 193 // record sample for wall clock statistics 194 if (valid) { 195 mWcStats.sample(wcNs); 196 } 197 198 // get the current CPU number 199 int cpuNum = sched_getcpu(); 200 201 // get the current CPU frequency in kHz 202 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203 204 // check if either CPU number or frequency changed 205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 206 mCpuNum = cpuNum; 207 mCpukHz = cpukHz; 208 // ignore sample for purposes of cycles 209 valid = false; 210 } 211 212 // if no change in CPU number or frequency, then record sample for cycle statistics 213 if (valid && mCpukHz > 0) { 214 double cycles = wcNs * cpukHz * 0.000001; 215 mHzStats.sample(cycles); 216 } 217 218 unsigned n = mWcStats.n(); 219 // mCpuUsage.elapsed() is expensive, so don't call it every loop 220 if ((n & 127) == 1) { 221 long long elapsed = mCpuUsage.elapsed(); 222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 223 double perLoop = elapsed / (double) n; 224 double perLoop100 = perLoop * 0.01; 225 double perLoop1k = perLoop * 0.001; 226 double mean = mWcStats.mean(); 227 double stddev = mWcStats.stddev(); 228 double minimum = mWcStats.minimum(); 229 double maximum = mWcStats.maximum(); 230 double meanCycles = mHzStats.mean(); 231 double stddevCycles = mHzStats.stddev(); 232 double minCycles = mHzStats.minimum(); 233 double maxCycles = mHzStats.maximum(); 234 mCpuUsage.resetElapsed(); 235 mWcStats.reset(); 236 mHzStats.reset(); 237 ALOGD("CPU usage for %s over past %.1f secs\n" 238 " (%u mixer loops at %.1f mean ms per loop):\n" 239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 242 title.string(), 243 elapsed * .000000001, n, perLoop * .000001, 244 mean * .001, 245 stddev * .001, 246 minimum * .001, 247 maximum * .001, 248 mean / perLoop100, 249 stddev / perLoop100, 250 minimum / perLoop100, 251 maximum / perLoop100, 252 meanCycles / perLoop1k, 253 stddevCycles / perLoop1k, 254 minCycles / perLoop1k, 255 maxCycles / perLoop1k); 256 257 } 258 } 259#endif 260}; 261 262// ---------------------------------------------------------------------------- 263// ThreadBase 264// ---------------------------------------------------------------------------- 265 266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 268 : Thread(false /*canCallJava*/), 269 mType(type), 270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 271 // mChannelMask 272 mChannelCount(0), 273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 mParamCond.broadcast(); 285 // do not lock the mutex in destructor 286 releaseWakeLock_l(); 287 if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 377 if (err != 0) { 378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 379 "error %d", 380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 381 } 382 } break; 383 case CFG_EVENT_IO: { 384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 385 mAudioFlinger->mLock.lock(); 386 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 387 mAudioFlinger->mLock.unlock(); 388 } break; 389 default: 390 ALOGE("processConfigEvents() unknown event type %d", event->type()); 391 break; 392 } 393 delete event; 394 mLock.lock(); 395 } 396 mLock.unlock(); 397} 398 399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 400{ 401 const size_t SIZE = 256; 402 char buffer[SIZE]; 403 String8 result; 404 405 bool locked = AudioFlinger::dumpTryLock(mLock); 406 if (!locked) { 407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 408 write(fd, buffer, strlen(buffer)); 409 } 410 411 snprintf(buffer, SIZE, "io handle: %d\n", mId); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 430 result.append(buffer); 431 432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 433 result.append(buffer); 434 result.append(" Index Command"); 435 for (size_t i = 0; i < mNewParameters.size(); ++i) { 436 snprintf(buffer, SIZE, "\n %02d ", i); 437 result.append(buffer); 438 result.append(mNewParameters[i]); 439 } 440 441 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 442 result.append(buffer); 443 for (size_t i = 0; i < mConfigEvents.size(); i++) { 444 mConfigEvents[i]->dump(buffer, SIZE); 445 result.append(buffer); 446 } 447 result.append("\n"); 448 449 write(fd, result.string(), result.size()); 450 451 if (locked) { 452 mLock.unlock(); 453 } 454} 455 456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 457{ 458 const size_t SIZE = 256; 459 char buffer[SIZE]; 460 String8 result; 461 462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 463 write(fd, buffer, strlen(buffer)); 464 465 for (size_t i = 0; i < mEffectChains.size(); ++i) { 466 sp<EffectChain> chain = mEffectChains[i]; 467 if (chain != 0) { 468 chain->dump(fd, args); 469 } 470 } 471} 472 473void AudioFlinger::ThreadBase::acquireWakeLock() 474{ 475 Mutex::Autolock _l(mLock); 476 acquireWakeLock_l(); 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock_l() 480{ 481 if (mPowerManager == 0) { 482 // use checkService() to avoid blocking if power service is not up yet 483 sp<IBinder> binder = 484 defaultServiceManager()->checkService(String16("power")); 485 if (binder == 0) { 486 ALOGW("Thread %s cannot connect to the power manager service", mName); 487 } else { 488 mPowerManager = interface_cast<IPowerManager>(binder); 489 binder->linkToDeath(mDeathRecipient); 490 } 491 } 492 if (mPowerManager != 0) { 493 sp<IBinder> binder = new BBinder(); 494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 495 binder, 496 String16(mName)); 497 if (status == NO_ERROR) { 498 mWakeLockToken = binder; 499 } 500 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 501 } 502} 503 504void AudioFlinger::ThreadBase::releaseWakeLock() 505{ 506 Mutex::Autolock _l(mLock); 507 releaseWakeLock_l(); 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock_l() 511{ 512 if (mWakeLockToken != 0) { 513 ALOGV("releaseWakeLock_l() %s", mName); 514 if (mPowerManager != 0) { 515 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 516 } 517 mWakeLockToken.clear(); 518 } 519} 520 521void AudioFlinger::ThreadBase::clearPowerManager() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525 mPowerManager.clear(); 526} 527 528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 529{ 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 thread->clearPowerManager(); 533 } 534 ALOGW("power manager service died !!!"); 535} 536 537void AudioFlinger::ThreadBase::setEffectSuspended( 538 const effect_uuid_t *type, bool suspend, int sessionId) 539{ 540 Mutex::Autolock _l(mLock); 541 setEffectSuspended_l(type, suspend, sessionId); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended_l( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 sp<EffectChain> chain = getEffectChain_l(sessionId); 548 if (chain != 0) { 549 if (type != NULL) { 550 chain->setEffectSuspended_l(type, suspend); 551 } else { 552 chain->setEffectSuspendedAll_l(suspend); 553 } 554 } 555 556 updateSuspendedSessions_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 560{ 561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 562 if (index < 0) { 563 return; 564 } 565 566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 567 mSuspendedSessions.valueAt(index); 568 569 for (size_t i = 0; i < sessionEffects.size(); i++) { 570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 571 for (int j = 0; j < desc->mRefCount; j++) { 572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 573 chain->setEffectSuspendedAll_l(true); 574 } else { 575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 576 desc->mType.timeLow); 577 chain->setEffectSuspended_l(&desc->mType, true); 578 } 579 } 580 } 581} 582 583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId) 586{ 587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 588 589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 590 591 if (suspend) { 592 if (index >= 0) { 593 sessionEffects = mSuspendedSessions.valueAt(index); 594 } else { 595 mSuspendedSessions.add(sessionId, sessionEffects); 596 } 597 } else { 598 if (index < 0) { 599 return; 600 } 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } 603 604 605 int key = EffectChain::kKeyForSuspendAll; 606 if (type != NULL) { 607 key = type->timeLow; 608 } 609 index = sessionEffects.indexOfKey(key); 610 611 sp<SuspendedSessionDesc> desc; 612 if (suspend) { 613 if (index >= 0) { 614 desc = sessionEffects.valueAt(index); 615 } else { 616 desc = new SuspendedSessionDesc(); 617 if (type != NULL) { 618 desc->mType = *type; 619 } 620 sessionEffects.add(key, desc); 621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 622 } 623 desc->mRefCount++; 624 } else { 625 if (index < 0) { 626 return; 627 } 628 desc = sessionEffects.valueAt(index); 629 if (--desc->mRefCount == 0) { 630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 631 sessionEffects.removeItemsAt(index); 632 if (sessionEffects.isEmpty()) { 633 ALOGV("updateSuspendedSessions_l() restore removing session %d", 634 sessionId); 635 mSuspendedSessions.removeItem(sessionId); 636 } 637 } 638 } 639 if (!sessionEffects.isEmpty()) { 640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 641 } 642} 643 644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 645 bool enabled, 646 int sessionId) 647{ 648 Mutex::Autolock _l(mLock); 649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 if (mType != RECORD) { 657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 658 // another session. This gives the priority to well behaved effect control panels 659 // and applications not using global effects. 660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 661 // global effects 662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 664 } 665 } 666 667 sp<EffectChain> chain = getEffectChain_l(sessionId); 668 if (chain != 0) { 669 chain->checkSuspendOnEffectEnabled(effect, enabled); 670 } 671} 672 673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 675 const sp<AudioFlinger::Client>& client, 676 const sp<IEffectClient>& effectClient, 677 int32_t priority, 678 int sessionId, 679 effect_descriptor_t *desc, 680 int *enabled, 681 status_t *status 682 ) 683{ 684 sp<EffectModule> effect; 685 sp<EffectHandle> handle; 686 status_t lStatus; 687 sp<EffectChain> chain; 688 bool chainCreated = false; 689 bool effectCreated = false; 690 bool effectRegistered = false; 691 692 lStatus = initCheck(); 693 if (lStatus != NO_ERROR) { 694 ALOGW("createEffect_l() Audio driver not initialized."); 695 goto Exit; 696 } 697 698 // Do not allow effects with session ID 0 on direct output or duplicating threads 699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 702 desc->name, sessionId); 703 lStatus = BAD_VALUE; 704 goto Exit; 705 } 706 // Only Pre processor effects are allowed on input threads and only on input threads 707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 709 desc->name, desc->flags, mType); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 715 716 { // scope for mLock 717 Mutex::Autolock _l(mLock); 718 719 // check for existing effect chain with the requested audio session 720 chain = getEffectChain_l(sessionId); 721 if (chain == 0) { 722 // create a new chain for this session 723 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 724 chain = new EffectChain(this, sessionId); 725 addEffectChain_l(chain); 726 chain->setStrategy(getStrategyForSession_l(sessionId)); 727 chainCreated = true; 728 } else { 729 effect = chain->getEffectFromDesc_l(desc); 730 } 731 732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 733 734 if (effect == 0) { 735 int id = mAudioFlinger->nextUniqueId(); 736 // Check CPU and memory usage 737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 738 if (lStatus != NO_ERROR) { 739 goto Exit; 740 } 741 effectRegistered = true; 742 // create a new effect module if none present in the chain 743 effect = new EffectModule(this, chain, desc, id, sessionId); 744 lStatus = effect->status(); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 lStatus = chain->addEffect_l(effect); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 effectCreated = true; 753 754 effect->setDevice(mOutDevice); 755 effect->setDevice(mInDevice); 756 effect->setMode(mAudioFlinger->getMode()); 757 effect->setAudioSource(mAudioSource); 758 } 759 // create effect handle and connect it to effect module 760 handle = new EffectHandle(effect, client, effectClient, priority); 761 lStatus = effect->addHandle(handle.get()); 762 if (enabled != NULL) { 763 *enabled = (int)effect->isEnabled(); 764 } 765 } 766 767Exit: 768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 769 Mutex::Autolock _l(mLock); 770 if (effectCreated) { 771 chain->removeEffect_l(effect); 772 } 773 if (effectRegistered) { 774 AudioSystem::unregisterEffect(effect->id()); 775 } 776 if (chainCreated) { 777 removeEffectChain_l(chain); 778 } 779 handle.clear(); 780 } 781 782 if (status != NULL) { 783 *status = lStatus; 784 } 785 return handle; 786} 787 788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 789{ 790 Mutex::Autolock _l(mLock); 791 return getEffect_l(sessionId, effectId); 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 795{ 796 sp<EffectChain> chain = getEffectChain_l(sessionId); 797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 798} 799 800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 801// PlaybackThread::mLock held 802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 803{ 804 // check for existing effect chain with the requested audio session 805 int sessionId = effect->sessionId(); 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 bool chainCreated = false; 808 809 if (chain == 0) { 810 // create a new chain for this session 811 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 812 chain = new EffectChain(this, sessionId); 813 addEffectChain_l(chain); 814 chain->setStrategy(getStrategyForSession_l(sessionId)); 815 chainCreated = true; 816 } 817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 818 819 if (chain->getEffectFromId_l(effect->id()) != 0) { 820 ALOGW("addEffect_l() %p effect %s already present in chain %p", 821 this, effect->desc().name, chain.get()); 822 return BAD_VALUE; 823 } 824 825 status_t status = chain->addEffect_l(effect); 826 if (status != NO_ERROR) { 827 if (chainCreated) { 828 removeEffectChain_l(chain); 829 } 830 return status; 831 } 832 833 effect->setDevice(mOutDevice); 834 effect->setDevice(mInDevice); 835 effect->setMode(mAudioFlinger->getMode()); 836 effect->setAudioSource(mAudioSource); 837 return NO_ERROR; 838} 839 840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 841 842 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 843 effect_descriptor_t desc = effect->desc(); 844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 845 detachAuxEffect_l(effect->id()); 846 } 847 848 sp<EffectChain> chain = effect->chain().promote(); 849 if (chain != 0) { 850 // remove effect chain if removing last effect 851 if (chain->removeEffect_l(effect) == 0) { 852 removeEffectChain_l(chain); 853 } 854 } else { 855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 856 } 857} 858 859void AudioFlinger::ThreadBase::lockEffectChains_l( 860 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 861{ 862 effectChains = mEffectChains; 863 for (size_t i = 0; i < mEffectChains.size(); i++) { 864 mEffectChains[i]->lock(); 865 } 866} 867 868void AudioFlinger::ThreadBase::unlockEffectChains( 869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 870{ 871 for (size_t i = 0; i < effectChains.size(); i++) { 872 effectChains[i]->unlock(); 873 } 874} 875 876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 877{ 878 Mutex::Autolock _l(mLock); 879 return getEffectChain_l(sessionId); 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 883{ 884 size_t size = mEffectChains.size(); 885 for (size_t i = 0; i < size; i++) { 886 if (mEffectChains[i]->sessionId() == sessionId) { 887 return mEffectChains[i]; 888 } 889 } 890 return 0; 891} 892 893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 894{ 895 Mutex::Autolock _l(mLock); 896 size_t size = mEffectChains.size(); 897 for (size_t i = 0; i < size; i++) { 898 mEffectChains[i]->setMode_l(mode); 899 } 900} 901 902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 903 EffectHandle *handle, 904 bool unpinIfLast) { 905 906 Mutex::Autolock _l(mLock); 907 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 908 // delete the effect module if removing last handle on it 909 if (effect->removeHandle(handle) == 0) { 910 if (!effect->isPinned() || unpinIfLast) { 911 removeEffect_l(effect); 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 } 915} 916 917// ---------------------------------------------------------------------------- 918// Playback 919// ---------------------------------------------------------------------------- 920 921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 922 AudioStreamOut* output, 923 audio_io_handle_t id, 924 audio_devices_t device, 925 type_t type) 926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 928 // mStreamTypes[] initialized in constructor body 929 mOutput(output), 930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 931 mMixerStatus(MIXER_IDLE), 932 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 934 mScreenState(AudioFlinger::mScreenState), 935 // index 0 is reserved for normal mixer's submix 936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 937{ 938 snprintf(mName, kNameLength, "AudioOut_%X", id); 939 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 940 941 // Assumes constructor is called by AudioFlinger with it's mLock held, but 942 // it would be safer to explicitly pass initial masterVolume/masterMute as 943 // parameter. 944 // 945 // If the HAL we are using has support for master volume or master mute, 946 // then do not attenuate or mute during mixing (just leave the volume at 1.0 947 // and the mute set to false). 948 mMasterVolume = audioFlinger->masterVolume_l(); 949 mMasterMute = audioFlinger->masterMute_l(); 950 if (mOutput && mOutput->audioHwDev) { 951 if (mOutput->audioHwDev->canSetMasterVolume()) { 952 mMasterVolume = 1.0; 953 } 954 955 if (mOutput->audioHwDev->canSetMasterMute()) { 956 mMasterMute = false; 957 } 958 } 959 960 readOutputParameters(); 961 962 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 963 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 964 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 965 stream = (audio_stream_type_t) (stream + 1)) { 966 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 967 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 968 } 969 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 970 // because mAudioFlinger doesn't have one to copy from 971} 972 973AudioFlinger::PlaybackThread::~PlaybackThread() 974{ 975 mAudioFlinger->unregisterWriter(mNBLogWriter); 976 delete [] mMixBuffer; 977} 978 979void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 980{ 981 dumpInternals(fd, args); 982 dumpTracks(fd, args); 983 dumpEffectChains(fd, args); 984} 985 986void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 987{ 988 const size_t SIZE = 256; 989 char buffer[SIZE]; 990 String8 result; 991 992 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 993 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 994 const stream_type_t *st = &mStreamTypes[i]; 995 if (i > 0) { 996 result.appendFormat(", "); 997 } 998 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 999 if (st->mute) { 1000 result.append("M"); 1001 } 1002 } 1003 result.append("\n"); 1004 write(fd, result.string(), result.length()); 1005 result.clear(); 1006 1007 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1008 result.append(buffer); 1009 Track::appendDumpHeader(result); 1010 for (size_t i = 0; i < mTracks.size(); ++i) { 1011 sp<Track> track = mTracks[i]; 1012 if (track != 0) { 1013 track->dump(buffer, SIZE); 1014 result.append(buffer); 1015 } 1016 } 1017 1018 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1019 result.append(buffer); 1020 Track::appendDumpHeader(result); 1021 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1022 sp<Track> track = mActiveTracks[i].promote(); 1023 if (track != 0) { 1024 track->dump(buffer, SIZE); 1025 result.append(buffer); 1026 } 1027 } 1028 write(fd, result.string(), result.size()); 1029 1030 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1031 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1032 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1033 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1034} 1035 1036void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1037{ 1038 const size_t SIZE = 256; 1039 char buffer[SIZE]; 1040 String8 result; 1041 1042 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1043 result.append(buffer); 1044 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1045 ns2ms(systemTime() - mLastWriteTime)); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1056 result.append(buffer); 1057 write(fd, result.string(), result.size()); 1058 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1059 1060 dumpBase(fd, args); 1061} 1062 1063// Thread virtuals 1064status_t AudioFlinger::PlaybackThread::readyToRun() 1065{ 1066 status_t status = initCheck(); 1067 if (status == NO_ERROR) { 1068 ALOGI("AudioFlinger's thread %p ready to run", this); 1069 } else { 1070 ALOGE("No working audio driver found."); 1071 } 1072 return status; 1073} 1074 1075void AudioFlinger::PlaybackThread::onFirstRef() 1076{ 1077 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1078} 1079 1080// ThreadBase virtuals 1081void AudioFlinger::PlaybackThread::preExit() 1082{ 1083 ALOGV(" preExit()"); 1084 // FIXME this is using hard-coded strings but in the future, this functionality will be 1085 // converted to use audio HAL extensions required to support tunneling 1086 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1087} 1088 1089// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1090sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1091 const sp<AudioFlinger::Client>& client, 1092 audio_stream_type_t streamType, 1093 uint32_t sampleRate, 1094 audio_format_t format, 1095 audio_channel_mask_t channelMask, 1096 size_t frameCount, 1097 const sp<IMemory>& sharedBuffer, 1098 int sessionId, 1099 IAudioFlinger::track_flags_t *flags, 1100 pid_t tid, 1101 status_t *status) 1102{ 1103 sp<Track> track; 1104 status_t lStatus; 1105 1106 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1107 1108 // client expresses a preference for FAST, but we get the final say 1109 if (*flags & IAudioFlinger::TRACK_FAST) { 1110 if ( 1111 // not timed 1112 (!isTimed) && 1113 // either of these use cases: 1114 ( 1115 // use case 1: shared buffer with any frame count 1116 ( 1117 (sharedBuffer != 0) 1118 ) || 1119 // use case 2: callback handler and frame count is default or at least as large as HAL 1120 ( 1121 (tid != -1) && 1122 ((frameCount == 0) || 1123 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1124 ) 1125 ) && 1126 // PCM data 1127 audio_is_linear_pcm(format) && 1128 // mono or stereo 1129 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1130 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1131#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1132 // hardware sample rate 1133 (sampleRate == mSampleRate) && 1134#endif 1135 // normal mixer has an associated fast mixer 1136 hasFastMixer() && 1137 // there are sufficient fast track slots available 1138 (mFastTrackAvailMask != 0) 1139 // FIXME test that MixerThread for this fast track has a capable output HAL 1140 // FIXME add a permission test also? 1141 ) { 1142 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1143 if (frameCount == 0) { 1144 frameCount = mFrameCount * kFastTrackMultiplier; 1145 } 1146 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1147 frameCount, mFrameCount); 1148 } else { 1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1150 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1151 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1152 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1153 audio_is_linear_pcm(format), 1154 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1155 *flags &= ~IAudioFlinger::TRACK_FAST; 1156 // For compatibility with AudioTrack calculation, buffer depth is forced 1157 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1158 // This is probably too conservative, but legacy application code may depend on it. 1159 // If you change this calculation, also review the start threshold which is related. 1160 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1161 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1162 if (minBufCount < 2) { 1163 minBufCount = 2; 1164 } 1165 size_t minFrameCount = mNormalFrameCount * minBufCount; 1166 if (frameCount < minFrameCount) { 1167 frameCount = minFrameCount; 1168 } 1169 } 1170 } 1171 1172 if (mType == DIRECT) { 1173 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1174 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1175 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1176 "for output %p with format %d", 1177 sampleRate, format, channelMask, mOutput, mFormat); 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } else { 1183 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1184 if (sampleRate > mSampleRate*2) { 1185 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 } 1190 1191 lStatus = initCheck(); 1192 if (lStatus != NO_ERROR) { 1193 ALOGE("Audio driver not initialized."); 1194 goto Exit; 1195 } 1196 1197 { // scope for mLock 1198 Mutex::Autolock _l(mLock); 1199 1200 // all tracks in same audio session must share the same routing strategy otherwise 1201 // conflicts will happen when tracks are moved from one output to another by audio policy 1202 // manager 1203 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1204 for (size_t i = 0; i < mTracks.size(); ++i) { 1205 sp<Track> t = mTracks[i]; 1206 if (t != 0 && !t->isOutputTrack()) { 1207 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1208 if (sessionId == t->sessionId() && strategy != actual) { 1209 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1210 strategy, actual); 1211 lStatus = BAD_VALUE; 1212 goto Exit; 1213 } 1214 } 1215 } 1216 1217 if (!isTimed) { 1218 track = new Track(this, client, streamType, sampleRate, format, 1219 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1220 } else { 1221 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1222 channelMask, frameCount, sharedBuffer, sessionId); 1223 } 1224 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1225 lStatus = NO_MEMORY; 1226 goto Exit; 1227 } 1228 mTracks.add(track); 1229 1230 sp<EffectChain> chain = getEffectChain_l(sessionId); 1231 if (chain != 0) { 1232 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1233 track->setMainBuffer(chain->inBuffer()); 1234 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1235 chain->incTrackCnt(); 1236 } 1237 1238 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1239 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1240 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1241 // so ask activity manager to do this on our behalf 1242 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1243 } 1244 } 1245 1246 lStatus = NO_ERROR; 1247 1248Exit: 1249 if (status) { 1250 *status = lStatus; 1251 } 1252 mNBLogWriter->logf("createTrack_l"); 1253 return track; 1254} 1255 1256uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1257{ 1258 return latency; 1259} 1260 1261uint32_t AudioFlinger::PlaybackThread::latency() const 1262{ 1263 Mutex::Autolock _l(mLock); 1264 return latency_l(); 1265} 1266uint32_t AudioFlinger::PlaybackThread::latency_l() const 1267{ 1268 if (initCheck() == NO_ERROR) { 1269 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1270 } else { 1271 return 0; 1272 } 1273} 1274 1275void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1276{ 1277 Mutex::Autolock _l(mLock); 1278 // Don't apply master volume in SW if our HAL can do it for us. 1279 if (mOutput && mOutput->audioHwDev && 1280 mOutput->audioHwDev->canSetMasterVolume()) { 1281 mMasterVolume = 1.0; 1282 } else { 1283 mMasterVolume = value; 1284 } 1285} 1286 1287void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1288{ 1289 Mutex::Autolock _l(mLock); 1290 // Don't apply master mute in SW if our HAL can do it for us. 1291 if (mOutput && mOutput->audioHwDev && 1292 mOutput->audioHwDev->canSetMasterMute()) { 1293 mMasterMute = false; 1294 } else { 1295 mMasterMute = muted; 1296 } 1297} 1298 1299void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1300{ 1301 Mutex::Autolock _l(mLock); 1302 mStreamTypes[stream].volume = value; 1303} 1304 1305void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1306{ 1307 Mutex::Autolock _l(mLock); 1308 mStreamTypes[stream].mute = muted; 1309} 1310 1311float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1312{ 1313 Mutex::Autolock _l(mLock); 1314 return mStreamTypes[stream].volume; 1315} 1316 1317// addTrack_l() must be called with ThreadBase::mLock held 1318status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1319{ 1320 mNBLogWriter->logf("addTrack_l mName=%d", track->mName); 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 mNBLogWriter->logf("destroyTrack_l mName=%d", track->mName); 1355 track->mState = TrackBase::TERMINATED; 1356 // active tracks are removed by threadLoop() 1357 if (mActiveTracks.indexOf(track) < 0) { 1358 removeTrack_l(track); 1359 } 1360} 1361 1362void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1363{ 1364 mNBLogWriter->logf("removeTrack_l mName=%d", track->mName); 1365 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1366 mTracks.remove(track); 1367 deleteTrackName_l(track->name()); 1368 // redundant as track is about to be destroyed, for dumpsys only 1369 track->mName = -1; 1370 if (track->isFastTrack()) { 1371 int index = track->mFastIndex; 1372 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1373 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1374 mFastTrackAvailMask |= 1 << index; 1375 // redundant as track is about to be destroyed, for dumpsys only 1376 track->mFastIndex = -1; 1377 } 1378 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1379 if (chain != 0) { 1380 chain->decTrackCnt(); 1381 } 1382} 1383 1384String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1385{ 1386 String8 out_s8 = String8(""); 1387 char *s; 1388 1389 Mutex::Autolock _l(mLock); 1390 if (initCheck() != NO_ERROR) { 1391 return out_s8; 1392 } 1393 1394 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1395 out_s8 = String8(s); 1396 free(s); 1397 return out_s8; 1398} 1399 1400// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1401void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1402 AudioSystem::OutputDescriptor desc; 1403 void *param2 = NULL; 1404 1405 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1406 param); 1407 1408 switch (event) { 1409 case AudioSystem::OUTPUT_OPENED: 1410 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1411 desc.channels = mChannelMask; 1412 desc.samplingRate = mSampleRate; 1413 desc.format = mFormat; 1414 desc.frameCount = mNormalFrameCount; // FIXME see 1415 // AudioFlinger::frameCount(audio_io_handle_t) 1416 desc.latency = latency(); 1417 param2 = &desc; 1418 break; 1419 1420 case AudioSystem::STREAM_CONFIG_CHANGED: 1421 param2 = ¶m; 1422 case AudioSystem::OUTPUT_CLOSED: 1423 default: 1424 break; 1425 } 1426 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1427} 1428 1429void AudioFlinger::PlaybackThread::readOutputParameters() 1430{ 1431 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1432 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1433 mChannelCount = (uint16_t)popcount(mChannelMask); 1434 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1435 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1436 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1437 if (mFrameCount & 15) { 1438 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1439 mFrameCount); 1440 } 1441 1442 // Calculate size of normal mix buffer relative to the HAL output buffer size 1443 double multiplier = 1.0; 1444 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1445 kUseFastMixer == FastMixer_Dynamic)) { 1446 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1447 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1448 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1449 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1450 maxNormalFrameCount = maxNormalFrameCount & ~15; 1451 if (maxNormalFrameCount < minNormalFrameCount) { 1452 maxNormalFrameCount = minNormalFrameCount; 1453 } 1454 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1455 if (multiplier <= 1.0) { 1456 multiplier = 1.0; 1457 } else if (multiplier <= 2.0) { 1458 if (2 * mFrameCount <= maxNormalFrameCount) { 1459 multiplier = 2.0; 1460 } else { 1461 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1462 } 1463 } else { 1464 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1465 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1466 // track, but we sometimes have to do this to satisfy the maximum frame count 1467 // constraint) 1468 // FIXME this rounding up should not be done if no HAL SRC 1469 uint32_t truncMult = (uint32_t) multiplier; 1470 if ((truncMult & 1)) { 1471 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1472 ++truncMult; 1473 } 1474 } 1475 multiplier = (double) truncMult; 1476 } 1477 } 1478 mNormalFrameCount = multiplier * mFrameCount; 1479 // round up to nearest 16 frames to satisfy AudioMixer 1480 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1481 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1482 mNormalFrameCount); 1483 1484 delete[] mMixBuffer; 1485 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1486 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1487 1488 // force reconfiguration of effect chains and engines to take new buffer size and audio 1489 // parameters into account 1490 // Note that mLock is not held when readOutputParameters() is called from the constructor 1491 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1492 // matter. 1493 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1494 Vector< sp<EffectChain> > effectChains = mEffectChains; 1495 for (size_t i = 0; i < effectChains.size(); i ++) { 1496 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1497 } 1498} 1499 1500 1501status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1502{ 1503 if (halFrames == NULL || dspFrames == NULL) { 1504 return BAD_VALUE; 1505 } 1506 Mutex::Autolock _l(mLock); 1507 if (initCheck() != NO_ERROR) { 1508 return INVALID_OPERATION; 1509 } 1510 size_t framesWritten = mBytesWritten / mFrameSize; 1511 *halFrames = framesWritten; 1512 1513 if (isSuspended()) { 1514 // return an estimation of rendered frames when the output is suspended 1515 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1516 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1517 return NO_ERROR; 1518 } else { 1519 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1520 } 1521} 1522 1523uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1524{ 1525 Mutex::Autolock _l(mLock); 1526 uint32_t result = 0; 1527 if (getEffectChain_l(sessionId) != 0) { 1528 result = EFFECT_SESSION; 1529 } 1530 1531 for (size_t i = 0; i < mTracks.size(); ++i) { 1532 sp<Track> track = mTracks[i]; 1533 if (sessionId == track->sessionId() && !track->isInvalid()) { 1534 result |= TRACK_SESSION; 1535 break; 1536 } 1537 } 1538 1539 return result; 1540} 1541 1542uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1543{ 1544 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1545 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1546 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1547 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1548 } 1549 for (size_t i = 0; i < mTracks.size(); i++) { 1550 sp<Track> track = mTracks[i]; 1551 if (sessionId == track->sessionId() && !track->isInvalid()) { 1552 return AudioSystem::getStrategyForStream(track->streamType()); 1553 } 1554 } 1555 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1556} 1557 1558 1559AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1560{ 1561 Mutex::Autolock _l(mLock); 1562 return mOutput; 1563} 1564 1565AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1566{ 1567 Mutex::Autolock _l(mLock); 1568 AudioStreamOut *output = mOutput; 1569 mOutput = NULL; 1570 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1571 // must push a NULL and wait for ack 1572 mOutputSink.clear(); 1573 mPipeSink.clear(); 1574 mNormalSink.clear(); 1575 return output; 1576} 1577 1578// this method must always be called either with ThreadBase mLock held or inside the thread loop 1579audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1580{ 1581 if (mOutput == NULL) { 1582 return NULL; 1583 } 1584 return &mOutput->stream->common; 1585} 1586 1587uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1588{ 1589 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1590} 1591 1592status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1593{ 1594 if (!isValidSyncEvent(event)) { 1595 return BAD_VALUE; 1596 } 1597 1598 Mutex::Autolock _l(mLock); 1599 1600 for (size_t i = 0; i < mTracks.size(); ++i) { 1601 sp<Track> track = mTracks[i]; 1602 if (event->triggerSession() == track->sessionId()) { 1603 (void) track->setSyncEvent(event); 1604 return NO_ERROR; 1605 } 1606 } 1607 1608 return NAME_NOT_FOUND; 1609} 1610 1611bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1612{ 1613 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1614} 1615 1616void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1617 const Vector< sp<Track> >& tracksToRemove) 1618{ 1619 size_t count = tracksToRemove.size(); 1620 if (CC_UNLIKELY(count)) { 1621 for (size_t i = 0 ; i < count ; i++) { 1622 const sp<Track>& track = tracksToRemove.itemAt(i); 1623 if ((track->sharedBuffer() != 0) && 1624 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1625 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1626 } 1627 } 1628 } 1629 1630} 1631 1632void AudioFlinger::PlaybackThread::checkSilentMode_l() 1633{ 1634 if (!mMasterMute) { 1635 char value[PROPERTY_VALUE_MAX]; 1636 if (property_get("ro.audio.silent", value, "0") > 0) { 1637 char *endptr; 1638 unsigned long ul = strtoul(value, &endptr, 0); 1639 if (*endptr == '\0' && ul != 0) { 1640 ALOGD("Silence is golden"); 1641 // The setprop command will not allow a property to be changed after 1642 // the first time it is set, so we don't have to worry about un-muting. 1643 setMasterMute_l(true); 1644 } 1645 } 1646 } 1647} 1648 1649// shared by MIXER and DIRECT, overridden by DUPLICATING 1650void AudioFlinger::PlaybackThread::threadLoop_write() 1651{ 1652 // FIXME rewrite to reduce number of system calls 1653 mLastWriteTime = systemTime(); 1654 mInWrite = true; 1655 int bytesWritten; 1656 1657 // If an NBAIO sink is present, use it to write the normal mixer's submix 1658 if (mNormalSink != 0) { 1659#define mBitShift 2 // FIXME 1660 size_t count = mixBufferSize >> mBitShift; 1661 ATRACE_BEGIN("write"); 1662 // update the setpoint when AudioFlinger::mScreenState changes 1663 uint32_t screenState = AudioFlinger::mScreenState; 1664 if (screenState != mScreenState) { 1665 mScreenState = screenState; 1666 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1667 if (pipe != NULL) { 1668 pipe->setAvgFrames((mScreenState & 1) ? 1669 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1670 } 1671 } 1672 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1673 ATRACE_END(); 1674 if (framesWritten > 0) { 1675 bytesWritten = framesWritten << mBitShift; 1676 } else { 1677 bytesWritten = framesWritten; 1678 } 1679 // otherwise use the HAL / AudioStreamOut directly 1680 } else { 1681 // Direct output thread. 1682 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1683 } 1684 1685 if (bytesWritten > 0) { 1686 mBytesWritten += mixBufferSize; 1687 } 1688 mNumWrites++; 1689 mInWrite = false; 1690} 1691 1692/* 1693The derived values that are cached: 1694 - mixBufferSize from frame count * frame size 1695 - activeSleepTime from activeSleepTimeUs() 1696 - idleSleepTime from idleSleepTimeUs() 1697 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1698 - maxPeriod from frame count and sample rate (MIXER only) 1699 1700The parameters that affect these derived values are: 1701 - frame count 1702 - frame size 1703 - sample rate 1704 - device type: A2DP or not 1705 - device latency 1706 - format: PCM or not 1707 - active sleep time 1708 - idle sleep time 1709*/ 1710 1711void AudioFlinger::PlaybackThread::cacheParameters_l() 1712{ 1713 mixBufferSize = mNormalFrameCount * mFrameSize; 1714 activeSleepTime = activeSleepTimeUs(); 1715 idleSleepTime = idleSleepTimeUs(); 1716} 1717 1718void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1719{ 1720 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1721 this, streamType, mTracks.size()); 1722 Mutex::Autolock _l(mLock); 1723 1724 size_t size = mTracks.size(); 1725 for (size_t i = 0; i < size; i++) { 1726 sp<Track> t = mTracks[i]; 1727 if (t->streamType() == streamType) { 1728 t->invalidate(); 1729 } 1730 } 1731} 1732 1733status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1734{ 1735 int session = chain->sessionId(); 1736 int16_t *buffer = mMixBuffer; 1737 bool ownsBuffer = false; 1738 1739 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1740 if (session > 0) { 1741 // Only one effect chain can be present in direct output thread and it uses 1742 // the mix buffer as input 1743 if (mType != DIRECT) { 1744 size_t numSamples = mNormalFrameCount * mChannelCount; 1745 buffer = new int16_t[numSamples]; 1746 memset(buffer, 0, numSamples * sizeof(int16_t)); 1747 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1748 ownsBuffer = true; 1749 } 1750 1751 // Attach all tracks with same session ID to this chain. 1752 for (size_t i = 0; i < mTracks.size(); ++i) { 1753 sp<Track> track = mTracks[i]; 1754 if (session == track->sessionId()) { 1755 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1756 buffer); 1757 track->setMainBuffer(buffer); 1758 chain->incTrackCnt(); 1759 } 1760 } 1761 1762 // indicate all active tracks in the chain 1763 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1764 sp<Track> track = mActiveTracks[i].promote(); 1765 if (track == 0) { 1766 continue; 1767 } 1768 if (session == track->sessionId()) { 1769 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1770 chain->incActiveTrackCnt(); 1771 } 1772 } 1773 } 1774 1775 chain->setInBuffer(buffer, ownsBuffer); 1776 chain->setOutBuffer(mMixBuffer); 1777 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1778 // chains list in order to be processed last as it contains output stage effects 1779 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1780 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1781 // after track specific effects and before output stage 1782 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1783 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1784 // Effect chain for other sessions are inserted at beginning of effect 1785 // chains list to be processed before output mix effects. Relative order between other 1786 // sessions is not important 1787 size_t size = mEffectChains.size(); 1788 size_t i = 0; 1789 for (i = 0; i < size; i++) { 1790 if (mEffectChains[i]->sessionId() < session) { 1791 break; 1792 } 1793 } 1794 mEffectChains.insertAt(chain, i); 1795 checkSuspendOnAddEffectChain_l(chain); 1796 1797 return NO_ERROR; 1798} 1799 1800size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1801{ 1802 int session = chain->sessionId(); 1803 1804 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1805 1806 for (size_t i = 0; i < mEffectChains.size(); i++) { 1807 if (chain == mEffectChains[i]) { 1808 mEffectChains.removeAt(i); 1809 // detach all active tracks from the chain 1810 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1811 sp<Track> track = mActiveTracks[i].promote(); 1812 if (track == 0) { 1813 continue; 1814 } 1815 if (session == track->sessionId()) { 1816 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1817 chain.get(), session); 1818 chain->decActiveTrackCnt(); 1819 } 1820 } 1821 1822 // detach all tracks with same session ID from this chain 1823 for (size_t i = 0; i < mTracks.size(); ++i) { 1824 sp<Track> track = mTracks[i]; 1825 if (session == track->sessionId()) { 1826 track->setMainBuffer(mMixBuffer); 1827 chain->decTrackCnt(); 1828 } 1829 } 1830 break; 1831 } 1832 } 1833 return mEffectChains.size(); 1834} 1835 1836status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1837 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1838{ 1839 Mutex::Autolock _l(mLock); 1840 return attachAuxEffect_l(track, EffectId); 1841} 1842 1843status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1844 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1845{ 1846 status_t status = NO_ERROR; 1847 1848 if (EffectId == 0) { 1849 track->setAuxBuffer(0, NULL); 1850 } else { 1851 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1852 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1853 if (effect != 0) { 1854 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1855 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1856 } else { 1857 status = INVALID_OPERATION; 1858 } 1859 } else { 1860 status = BAD_VALUE; 1861 } 1862 } 1863 return status; 1864} 1865 1866void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1867{ 1868 for (size_t i = 0; i < mTracks.size(); ++i) { 1869 sp<Track> track = mTracks[i]; 1870 if (track->auxEffectId() == effectId) { 1871 attachAuxEffect_l(track, 0); 1872 } 1873 } 1874} 1875 1876bool AudioFlinger::PlaybackThread::threadLoop() 1877{ 1878 Vector< sp<Track> > tracksToRemove; 1879 1880 standbyTime = systemTime(); 1881 1882 // MIXER 1883 nsecs_t lastWarning = 0; 1884 1885 // DUPLICATING 1886 // FIXME could this be made local to while loop? 1887 writeFrames = 0; 1888 1889 cacheParameters_l(); 1890 sleepTime = idleSleepTime; 1891 1892 if (mType == MIXER) { 1893 sleepTimeShift = 0; 1894 } 1895 1896 CpuStats cpuStats; 1897 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1898 1899 acquireWakeLock(); 1900 1901 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1902 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1903 // and then that string will be logged at the next convenient opportunity. 1904 const char *logString = NULL; 1905 1906 while (!exitPending()) 1907 { 1908 cpuStats.sample(myName); 1909 1910 Vector< sp<EffectChain> > effectChains; 1911 1912 processConfigEvents(); 1913 1914 { // scope for mLock 1915 1916 Mutex::Autolock _l(mLock); 1917 1918 if (logString != NULL) { 1919 mNBLogWriter->logTimestamp(); 1920 mNBLogWriter->log(logString); 1921 logString = NULL; 1922 } 1923 1924 if (checkForNewParameters_l()) { 1925 cacheParameters_l(); 1926 } 1927 1928 saveOutputTracks(); 1929 1930 // put audio hardware into standby after short delay 1931 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1932 isSuspended())) { 1933 if (!mStandby) { 1934 1935 threadLoop_standby(); 1936 1937 mNBLogWriter->log("standby"); 1938 mStandby = true; 1939 } 1940 1941 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1942 // we're about to wait, flush the binder command buffer 1943 IPCThreadState::self()->flushCommands(); 1944 1945 clearOutputTracks(); 1946 1947 if (exitPending()) { 1948 break; 1949 } 1950 1951 releaseWakeLock_l(); 1952 // wait until we have something to do... 1953 ALOGV("%s going to sleep", myName.string()); 1954 mWaitWorkCV.wait(mLock); 1955 ALOGV("%s waking up", myName.string()); 1956 acquireWakeLock_l(); 1957 1958 mMixerStatus = MIXER_IDLE; 1959 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1960 mBytesWritten = 0; 1961 1962 checkSilentMode_l(); 1963 1964 standbyTime = systemTime() + standbyDelay; 1965 sleepTime = idleSleepTime; 1966 if (mType == MIXER) { 1967 sleepTimeShift = 0; 1968 } 1969 1970 continue; 1971 } 1972 } 1973 1974 // mMixerStatusIgnoringFastTracks is also updated internally 1975 mMixerStatus = prepareTracks_l(&tracksToRemove); 1976 1977 // prevent any changes in effect chain list and in each effect chain 1978 // during mixing and effect process as the audio buffers could be deleted 1979 // or modified if an effect is created or deleted 1980 lockEffectChains_l(effectChains); 1981 } 1982 1983 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1984 threadLoop_mix(); 1985 } else { 1986 threadLoop_sleepTime(); 1987 } 1988 1989 if (isSuspended()) { 1990 sleepTime = suspendSleepTimeUs(); 1991 mBytesWritten += mixBufferSize; 1992 } 1993 1994 // only process effects if we're going to write 1995 if (sleepTime == 0) { 1996 for (size_t i = 0; i < effectChains.size(); i ++) { 1997 effectChains[i]->process_l(); 1998 } 1999 } 2000 2001 // enable changes in effect chain 2002 unlockEffectChains(effectChains); 2003 2004 // sleepTime == 0 means we must write to audio hardware 2005 if (sleepTime == 0) { 2006 2007 threadLoop_write(); 2008 2009if (mType == MIXER) { 2010 // write blocked detection 2011 nsecs_t now = systemTime(); 2012 nsecs_t delta = now - mLastWriteTime; 2013 if (!mStandby && delta > maxPeriod) { 2014 mNumDelayedWrites++; 2015 if ((now - lastWarning) > kWarningThrottleNs) { 2016 ATRACE_NAME("underrun"); 2017 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2018 ns2ms(delta), mNumDelayedWrites, this); 2019 lastWarning = now; 2020 } 2021 } 2022} 2023 2024 mStandby = false; 2025 } else { 2026 usleep(sleepTime); 2027 } 2028 2029 // Finally let go of removed track(s), without the lock held 2030 // since we can't guarantee the destructors won't acquire that 2031 // same lock. This will also mutate and push a new fast mixer state. 2032 threadLoop_removeTracks(tracksToRemove); 2033 if (tracksToRemove.size() > 0) { 2034 logString = "remove"; 2035 } 2036 tracksToRemove.clear(); 2037 2038 // FIXME I don't understand the need for this here; 2039 // it was in the original code but maybe the 2040 // assignment in saveOutputTracks() makes this unnecessary? 2041 clearOutputTracks(); 2042 2043 // Effect chains will be actually deleted here if they were removed from 2044 // mEffectChains list during mixing or effects processing 2045 effectChains.clear(); 2046 2047 // FIXME Note that the above .clear() is no longer necessary since effectChains 2048 // is now local to this block, but will keep it for now (at least until merge done). 2049 } 2050 2051 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2052 if (mType == MIXER || mType == DIRECT) { 2053 // put output stream into standby mode 2054 if (!mStandby) { 2055 mOutput->stream->common.standby(&mOutput->stream->common); 2056 } 2057 } 2058 2059 releaseWakeLock(); 2060 2061 ALOGV("Thread %p type %d exiting", this, mType); 2062 return false; 2063} 2064 2065 2066// ---------------------------------------------------------------------------- 2067 2068AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2069 audio_io_handle_t id, audio_devices_t device, type_t type) 2070 : PlaybackThread(audioFlinger, output, id, device, type), 2071 // mAudioMixer below 2072 // mFastMixer below 2073 mFastMixerFutex(0) 2074 // mOutputSink below 2075 // mPipeSink below 2076 // mNormalSink below 2077{ 2078 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2079 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2080 "mFrameCount=%d, mNormalFrameCount=%d", 2081 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2082 mNormalFrameCount); 2083 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2084 2085 // FIXME - Current mixer implementation only supports stereo output 2086 if (mChannelCount != FCC_2) { 2087 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2088 } 2089 2090 // create an NBAIO sink for the HAL output stream, and negotiate 2091 mOutputSink = new AudioStreamOutSink(output->stream); 2092 size_t numCounterOffers = 0; 2093 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2094 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2095 ALOG_ASSERT(index == 0); 2096 2097 // initialize fast mixer depending on configuration 2098 bool initFastMixer; 2099 switch (kUseFastMixer) { 2100 case FastMixer_Never: 2101 initFastMixer = false; 2102 break; 2103 case FastMixer_Always: 2104 initFastMixer = true; 2105 break; 2106 case FastMixer_Static: 2107 case FastMixer_Dynamic: 2108 initFastMixer = mFrameCount < mNormalFrameCount; 2109 break; 2110 } 2111 if (initFastMixer) { 2112 2113 // create a MonoPipe to connect our submix to FastMixer 2114 NBAIO_Format format = mOutputSink->format(); 2115 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2116 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2117 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2118 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2119 const NBAIO_Format offers[1] = {format}; 2120 size_t numCounterOffers = 0; 2121 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2122 ALOG_ASSERT(index == 0); 2123 monoPipe->setAvgFrames((mScreenState & 1) ? 2124 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2125 mPipeSink = monoPipe; 2126 2127#ifdef TEE_SINK 2128 if (mTeeSinkOutputEnabled) { 2129 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2130 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2131 numCounterOffers = 0; 2132 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2133 ALOG_ASSERT(index == 0); 2134 mTeeSink = teeSink; 2135 PipeReader *teeSource = new PipeReader(*teeSink); 2136 numCounterOffers = 0; 2137 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2138 ALOG_ASSERT(index == 0); 2139 mTeeSource = teeSource; 2140 } 2141#endif 2142 2143 // create fast mixer and configure it initially with just one fast track for our submix 2144 mFastMixer = new FastMixer(); 2145 FastMixerStateQueue *sq = mFastMixer->sq(); 2146#ifdef STATE_QUEUE_DUMP 2147 sq->setObserverDump(&mStateQueueObserverDump); 2148 sq->setMutatorDump(&mStateQueueMutatorDump); 2149#endif 2150 FastMixerState *state = sq->begin(); 2151 FastTrack *fastTrack = &state->mFastTracks[0]; 2152 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2153 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2154 fastTrack->mVolumeProvider = NULL; 2155 fastTrack->mGeneration++; 2156 state->mFastTracksGen++; 2157 state->mTrackMask = 1; 2158 // fast mixer will use the HAL output sink 2159 state->mOutputSink = mOutputSink.get(); 2160 state->mOutputSinkGen++; 2161 state->mFrameCount = mFrameCount; 2162 state->mCommand = FastMixerState::COLD_IDLE; 2163 // already done in constructor initialization list 2164 //mFastMixerFutex = 0; 2165 state->mColdFutexAddr = &mFastMixerFutex; 2166 state->mColdGen++; 2167 state->mDumpState = &mFastMixerDumpState; 2168#ifdef TEE_SINK 2169 state->mTeeSink = mTeeSink.get(); 2170#endif 2171 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2172 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2173 sq->end(); 2174 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2175 2176 // start the fast mixer 2177 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2178 pid_t tid = mFastMixer->getTid(); 2179 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2180 if (err != 0) { 2181 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2182 kPriorityFastMixer, getpid_cached, tid, err); 2183 } 2184 2185#ifdef AUDIO_WATCHDOG 2186 // create and start the watchdog 2187 mAudioWatchdog = new AudioWatchdog(); 2188 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2189 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2190 tid = mAudioWatchdog->getTid(); 2191 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2192 if (err != 0) { 2193 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2194 kPriorityFastMixer, getpid_cached, tid, err); 2195 } 2196#endif 2197 2198 } else { 2199 mFastMixer = NULL; 2200 } 2201 2202 switch (kUseFastMixer) { 2203 case FastMixer_Never: 2204 case FastMixer_Dynamic: 2205 mNormalSink = mOutputSink; 2206 break; 2207 case FastMixer_Always: 2208 mNormalSink = mPipeSink; 2209 break; 2210 case FastMixer_Static: 2211 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2212 break; 2213 } 2214} 2215 2216AudioFlinger::MixerThread::~MixerThread() 2217{ 2218 if (mFastMixer != NULL) { 2219 FastMixerStateQueue *sq = mFastMixer->sq(); 2220 FastMixerState *state = sq->begin(); 2221 if (state->mCommand == FastMixerState::COLD_IDLE) { 2222 int32_t old = android_atomic_inc(&mFastMixerFutex); 2223 if (old == -1) { 2224 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2225 } 2226 } 2227 state->mCommand = FastMixerState::EXIT; 2228 sq->end(); 2229 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2230 mFastMixer->join(); 2231 // Though the fast mixer thread has exited, it's state queue is still valid. 2232 // We'll use that extract the final state which contains one remaining fast track 2233 // corresponding to our sub-mix. 2234 state = sq->begin(); 2235 ALOG_ASSERT(state->mTrackMask == 1); 2236 FastTrack *fastTrack = &state->mFastTracks[0]; 2237 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2238 delete fastTrack->mBufferProvider; 2239 sq->end(false /*didModify*/); 2240 delete mFastMixer; 2241#ifdef AUDIO_WATCHDOG 2242 if (mAudioWatchdog != 0) { 2243 mAudioWatchdog->requestExit(); 2244 mAudioWatchdog->requestExitAndWait(); 2245 mAudioWatchdog.clear(); 2246 } 2247#endif 2248 } 2249 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2250 delete mAudioMixer; 2251} 2252 2253 2254uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2255{ 2256 if (mFastMixer != NULL) { 2257 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2258 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2259 } 2260 return latency; 2261} 2262 2263 2264void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2265{ 2266 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2267} 2268 2269void AudioFlinger::MixerThread::threadLoop_write() 2270{ 2271 // FIXME we should only do one push per cycle; confirm this is true 2272 // Start the fast mixer if it's not already running 2273 if (mFastMixer != NULL) { 2274 FastMixerStateQueue *sq = mFastMixer->sq(); 2275 FastMixerState *state = sq->begin(); 2276 if (state->mCommand != FastMixerState::MIX_WRITE && 2277 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2278 if (state->mCommand == FastMixerState::COLD_IDLE) { 2279 int32_t old = android_atomic_inc(&mFastMixerFutex); 2280 if (old == -1) { 2281 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2282 } 2283#ifdef AUDIO_WATCHDOG 2284 if (mAudioWatchdog != 0) { 2285 mAudioWatchdog->resume(); 2286 } 2287#endif 2288 } 2289 state->mCommand = FastMixerState::MIX_WRITE; 2290 sq->end(); 2291 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2292 if (kUseFastMixer == FastMixer_Dynamic) { 2293 mNormalSink = mPipeSink; 2294 } 2295 } else { 2296 sq->end(false /*didModify*/); 2297 } 2298 } 2299 PlaybackThread::threadLoop_write(); 2300} 2301 2302void AudioFlinger::MixerThread::threadLoop_standby() 2303{ 2304 // Idle the fast mixer if it's currently running 2305 if (mFastMixer != NULL) { 2306 FastMixerStateQueue *sq = mFastMixer->sq(); 2307 FastMixerState *state = sq->begin(); 2308 if (!(state->mCommand & FastMixerState::IDLE)) { 2309 state->mCommand = FastMixerState::COLD_IDLE; 2310 state->mColdFutexAddr = &mFastMixerFutex; 2311 state->mColdGen++; 2312 mFastMixerFutex = 0; 2313 sq->end(); 2314 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2315 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2316 if (kUseFastMixer == FastMixer_Dynamic) { 2317 mNormalSink = mOutputSink; 2318 } 2319#ifdef AUDIO_WATCHDOG 2320 if (mAudioWatchdog != 0) { 2321 mAudioWatchdog->pause(); 2322 } 2323#endif 2324 } else { 2325 sq->end(false /*didModify*/); 2326 } 2327 } 2328 PlaybackThread::threadLoop_standby(); 2329} 2330 2331// shared by MIXER and DIRECT, overridden by DUPLICATING 2332void AudioFlinger::PlaybackThread::threadLoop_standby() 2333{ 2334 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2335 mOutput->stream->common.standby(&mOutput->stream->common); 2336} 2337 2338void AudioFlinger::MixerThread::threadLoop_mix() 2339{ 2340 // obtain the presentation timestamp of the next output buffer 2341 int64_t pts; 2342 status_t status = INVALID_OPERATION; 2343 2344 if (mNormalSink != 0) { 2345 status = mNormalSink->getNextWriteTimestamp(&pts); 2346 } else { 2347 status = mOutputSink->getNextWriteTimestamp(&pts); 2348 } 2349 2350 if (status != NO_ERROR) { 2351 pts = AudioBufferProvider::kInvalidPTS; 2352 } 2353 2354 // mix buffers... 2355 mAudioMixer->process(pts); 2356 // increase sleep time progressively when application underrun condition clears. 2357 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2358 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2359 // such that we would underrun the audio HAL. 2360 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2361 sleepTimeShift--; 2362 } 2363 sleepTime = 0; 2364 standbyTime = systemTime() + standbyDelay; 2365 //TODO: delay standby when effects have a tail 2366} 2367 2368void AudioFlinger::MixerThread::threadLoop_sleepTime() 2369{ 2370 // If no tracks are ready, sleep once for the duration of an output 2371 // buffer size, then write 0s to the output 2372 if (sleepTime == 0) { 2373 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2374 sleepTime = activeSleepTime >> sleepTimeShift; 2375 if (sleepTime < kMinThreadSleepTimeUs) { 2376 sleepTime = kMinThreadSleepTimeUs; 2377 } 2378 // reduce sleep time in case of consecutive application underruns to avoid 2379 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2380 // duration we would end up writing less data than needed by the audio HAL if 2381 // the condition persists. 2382 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2383 sleepTimeShift++; 2384 } 2385 } else { 2386 sleepTime = idleSleepTime; 2387 } 2388 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2389 memset (mMixBuffer, 0, mixBufferSize); 2390 sleepTime = 0; 2391 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2392 "anticipated start"); 2393 } 2394 // TODO add standby time extension fct of effect tail 2395} 2396 2397// prepareTracks_l() must be called with ThreadBase::mLock held 2398AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2399 Vector< sp<Track> > *tracksToRemove) 2400{ 2401 2402 mixer_state mixerStatus = MIXER_IDLE; 2403 // find out which tracks need to be processed 2404 size_t count = mActiveTracks.size(); 2405 size_t mixedTracks = 0; 2406 size_t tracksWithEffect = 0; 2407 // counts only _active_ fast tracks 2408 size_t fastTracks = 0; 2409 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2410 2411 float masterVolume = mMasterVolume; 2412 bool masterMute = mMasterMute; 2413 2414 if (masterMute) { 2415 masterVolume = 0; 2416 } 2417 // Delegate master volume control to effect in output mix effect chain if needed 2418 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2419 if (chain != 0) { 2420 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2421 chain->setVolume_l(&v, &v); 2422 masterVolume = (float)((v + (1 << 23)) >> 24); 2423 chain.clear(); 2424 } 2425 2426 // prepare a new state to push 2427 FastMixerStateQueue *sq = NULL; 2428 FastMixerState *state = NULL; 2429 bool didModify = false; 2430 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2431 if (mFastMixer != NULL) { 2432 sq = mFastMixer->sq(); 2433 state = sq->begin(); 2434 } 2435 2436 for (size_t i=0 ; i<count ; i++) { 2437 sp<Track> t = mActiveTracks[i].promote(); 2438 if (t == 0) { 2439 continue; 2440 } 2441 2442 // this const just means the local variable doesn't change 2443 Track* const track = t.get(); 2444 2445 // process fast tracks 2446 if (track->isFastTrack()) { 2447 2448 // It's theoretically possible (though unlikely) for a fast track to be created 2449 // and then removed within the same normal mix cycle. This is not a problem, as 2450 // the track never becomes active so it's fast mixer slot is never touched. 2451 // The converse, of removing an (active) track and then creating a new track 2452 // at the identical fast mixer slot within the same normal mix cycle, 2453 // is impossible because the slot isn't marked available until the end of each cycle. 2454 int j = track->mFastIndex; 2455 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2456 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2457 FastTrack *fastTrack = &state->mFastTracks[j]; 2458 2459 // Determine whether the track is currently in underrun condition, 2460 // and whether it had a recent underrun. 2461 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2462 FastTrackUnderruns underruns = ftDump->mUnderruns; 2463 uint32_t recentFull = (underruns.mBitFields.mFull - 2464 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2465 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2466 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2467 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2468 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2469 uint32_t recentUnderruns = recentPartial + recentEmpty; 2470 track->mObservedUnderruns = underruns; 2471 // don't count underruns that occur while stopping or pausing 2472 // or stopped which can occur when flush() is called while active 2473 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2474 track->mUnderrunCount += recentUnderruns; 2475 } 2476 2477 // This is similar to the state machine for normal tracks, 2478 // with a few modifications for fast tracks. 2479 bool isActive = true; 2480 switch (track->mState) { 2481 case TrackBase::STOPPING_1: 2482 // track stays active in STOPPING_1 state until first underrun 2483 if (recentUnderruns > 0) { 2484 track->mState = TrackBase::STOPPING_2; 2485 } 2486 break; 2487 case TrackBase::PAUSING: 2488 // ramp down is not yet implemented 2489 track->setPaused(); 2490 break; 2491 case TrackBase::RESUMING: 2492 // ramp up is not yet implemented 2493 track->mState = TrackBase::ACTIVE; 2494 break; 2495 case TrackBase::ACTIVE: 2496 if (recentFull > 0 || recentPartial > 0) { 2497 // track has provided at least some frames recently: reset retry count 2498 track->mRetryCount = kMaxTrackRetries; 2499 } 2500 if (recentUnderruns == 0) { 2501 // no recent underruns: stay active 2502 break; 2503 } 2504 // there has recently been an underrun of some kind 2505 if (track->sharedBuffer() == 0) { 2506 // were any of the recent underruns "empty" (no frames available)? 2507 if (recentEmpty == 0) { 2508 // no, then ignore the partial underruns as they are allowed indefinitely 2509 break; 2510 } 2511 // there has recently been an "empty" underrun: decrement the retry counter 2512 if (--(track->mRetryCount) > 0) { 2513 break; 2514 } 2515 // indicate to client process that the track was disabled because of underrun; 2516 // it will then automatically call start() when data is available 2517 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2518 // remove from active list, but state remains ACTIVE [confusing but true] 2519 isActive = false; 2520 break; 2521 } 2522 // fall through 2523 case TrackBase::STOPPING_2: 2524 case TrackBase::PAUSED: 2525 case TrackBase::TERMINATED: 2526 case TrackBase::STOPPED: 2527 case TrackBase::FLUSHED: // flush() while active 2528 // Check for presentation complete if track is inactive 2529 // We have consumed all the buffers of this track. 2530 // This would be incomplete if we auto-paused on underrun 2531 { 2532 size_t audioHALFrames = 2533 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2534 size_t framesWritten = mBytesWritten / mFrameSize; 2535 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2536 // track stays in active list until presentation is complete 2537 break; 2538 } 2539 } 2540 if (track->isStopping_2()) { 2541 track->mState = TrackBase::STOPPED; 2542 } 2543 if (track->isStopped()) { 2544 // Can't reset directly, as fast mixer is still polling this track 2545 // track->reset(); 2546 // So instead mark this track as needing to be reset after push with ack 2547 resetMask |= 1 << i; 2548 } 2549 isActive = false; 2550 break; 2551 case TrackBase::IDLE: 2552 default: 2553 LOG_FATAL("unexpected track state %d", track->mState); 2554 } 2555 2556 if (isActive) { 2557 // was it previously inactive? 2558 if (!(state->mTrackMask & (1 << j))) { 2559 ExtendedAudioBufferProvider *eabp = track; 2560 VolumeProvider *vp = track; 2561 fastTrack->mBufferProvider = eabp; 2562 fastTrack->mVolumeProvider = vp; 2563 fastTrack->mSampleRate = track->mSampleRate; 2564 fastTrack->mChannelMask = track->mChannelMask; 2565 fastTrack->mGeneration++; 2566 state->mTrackMask |= 1 << j; 2567 didModify = true; 2568 // no acknowledgement required for newly active tracks 2569 } 2570 // cache the combined master volume and stream type volume for fast mixer; this 2571 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2572 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2573 ++fastTracks; 2574 } else { 2575 // was it previously active? 2576 if (state->mTrackMask & (1 << j)) { 2577 fastTrack->mBufferProvider = NULL; 2578 fastTrack->mGeneration++; 2579 state->mTrackMask &= ~(1 << j); 2580 didModify = true; 2581 // If any fast tracks were removed, we must wait for acknowledgement 2582 // because we're about to decrement the last sp<> on those tracks. 2583 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2584 } else { 2585 LOG_FATAL("fast track %d should have been active", j); 2586 } 2587 tracksToRemove->add(track); 2588 // Avoids a misleading display in dumpsys 2589 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2590 } 2591 continue; 2592 } 2593 2594 { // local variable scope to avoid goto warning 2595 2596 audio_track_cblk_t* cblk = track->cblk(); 2597 2598 // The first time a track is added we wait 2599 // for all its buffers to be filled before processing it 2600 int name = track->name(); 2601 // make sure that we have enough frames to mix one full buffer. 2602 // enforce this condition only once to enable draining the buffer in case the client 2603 // app does not call stop() and relies on underrun to stop: 2604 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2605 // during last round 2606 uint32_t minFrames = 1; 2607 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2608 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2609 if (t->sampleRate() == mSampleRate) { 2610 minFrames = mNormalFrameCount; 2611 } else { 2612 // +1 for rounding and +1 for additional sample needed for interpolation 2613 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2614 // add frames already consumed but not yet released by the resampler 2615 // because cblk->framesReady() will include these frames 2616 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2617 // the minimum track buffer size is normally twice the number of frames necessary 2618 // to fill one buffer and the resampler should not leave more than one buffer worth 2619 // of unreleased frames after each pass, but just in case... 2620 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2621 } 2622 } 2623 if ((track->framesReady() >= minFrames) && track->isReady() && 2624 !track->isPaused() && !track->isTerminated()) 2625 { 2626 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2627 this); 2628 2629 mixedTracks++; 2630 2631 // track->mainBuffer() != mMixBuffer means there is an effect chain 2632 // connected to the track 2633 chain.clear(); 2634 if (track->mainBuffer() != mMixBuffer) { 2635 chain = getEffectChain_l(track->sessionId()); 2636 // Delegate volume control to effect in track effect chain if needed 2637 if (chain != 0) { 2638 tracksWithEffect++; 2639 } else { 2640 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2641 "session %d", 2642 name, track->sessionId()); 2643 } 2644 } 2645 2646 2647 int param = AudioMixer::VOLUME; 2648 if (track->mFillingUpStatus == Track::FS_FILLED) { 2649 // no ramp for the first volume setting 2650 track->mFillingUpStatus = Track::FS_ACTIVE; 2651 if (track->mState == TrackBase::RESUMING) { 2652 track->mState = TrackBase::ACTIVE; 2653 param = AudioMixer::RAMP_VOLUME; 2654 } 2655 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2656 } else if (cblk->server != 0) { 2657 // If the track is stopped before the first frame was mixed, 2658 // do not apply ramp 2659 param = AudioMixer::RAMP_VOLUME; 2660 } 2661 2662 // compute volume for this track 2663 uint32_t vl, vr, va; 2664 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2665 vl = vr = va = 0; 2666 if (track->isPausing()) { 2667 track->setPaused(); 2668 } 2669 } else { 2670 2671 // read original volumes with volume control 2672 float typeVolume = mStreamTypes[track->streamType()].volume; 2673 float v = masterVolume * typeVolume; 2674 ServerProxy *proxy = track->mServerProxy; 2675 uint32_t vlr = proxy->getVolumeLR(); 2676 vl = vlr & 0xFFFF; 2677 vr = vlr >> 16; 2678 // track volumes come from shared memory, so can't be trusted and must be clamped 2679 if (vl > MAX_GAIN_INT) { 2680 ALOGV("Track left volume out of range: %04X", vl); 2681 vl = MAX_GAIN_INT; 2682 } 2683 if (vr > MAX_GAIN_INT) { 2684 ALOGV("Track right volume out of range: %04X", vr); 2685 vr = MAX_GAIN_INT; 2686 } 2687 // now apply the master volume and stream type volume 2688 vl = (uint32_t)(v * vl) << 12; 2689 vr = (uint32_t)(v * vr) << 12; 2690 // assuming master volume and stream type volume each go up to 1.0, 2691 // vl and vr are now in 8.24 format 2692 2693 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2694 // send level comes from shared memory and so may be corrupt 2695 if (sendLevel > MAX_GAIN_INT) { 2696 ALOGV("Track send level out of range: %04X", sendLevel); 2697 sendLevel = MAX_GAIN_INT; 2698 } 2699 va = (uint32_t)(v * sendLevel); 2700 } 2701 // Delegate volume control to effect in track effect chain if needed 2702 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2703 // Do not ramp volume if volume is controlled by effect 2704 param = AudioMixer::VOLUME; 2705 track->mHasVolumeController = true; 2706 } else { 2707 // force no volume ramp when volume controller was just disabled or removed 2708 // from effect chain to avoid volume spike 2709 if (track->mHasVolumeController) { 2710 param = AudioMixer::VOLUME; 2711 } 2712 track->mHasVolumeController = false; 2713 } 2714 2715 // Convert volumes from 8.24 to 4.12 format 2716 // This additional clamping is needed in case chain->setVolume_l() overshot 2717 vl = (vl + (1 << 11)) >> 12; 2718 if (vl > MAX_GAIN_INT) { 2719 vl = MAX_GAIN_INT; 2720 } 2721 vr = (vr + (1 << 11)) >> 12; 2722 if (vr > MAX_GAIN_INT) { 2723 vr = MAX_GAIN_INT; 2724 } 2725 2726 if (va > MAX_GAIN_INT) { 2727 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2728 } 2729 2730 // XXX: these things DON'T need to be done each time 2731 mAudioMixer->setBufferProvider(name, track); 2732 mAudioMixer->enable(name); 2733 2734 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2735 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2736 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2737 mAudioMixer->setParameter( 2738 name, 2739 AudioMixer::TRACK, 2740 AudioMixer::FORMAT, (void *)track->format()); 2741 mAudioMixer->setParameter( 2742 name, 2743 AudioMixer::TRACK, 2744 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2745 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2746 uint32_t maxSampleRate = mSampleRate * 2; 2747 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2748 if (reqSampleRate == 0) { 2749 reqSampleRate = mSampleRate; 2750 } else if (reqSampleRate > maxSampleRate) { 2751 reqSampleRate = maxSampleRate; 2752 } 2753 mAudioMixer->setParameter( 2754 name, 2755 AudioMixer::RESAMPLE, 2756 AudioMixer::SAMPLE_RATE, 2757 (void *)reqSampleRate); 2758 mAudioMixer->setParameter( 2759 name, 2760 AudioMixer::TRACK, 2761 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2762 mAudioMixer->setParameter( 2763 name, 2764 AudioMixer::TRACK, 2765 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2766 2767 // reset retry count 2768 track->mRetryCount = kMaxTrackRetries; 2769 2770 // If one track is ready, set the mixer ready if: 2771 // - the mixer was not ready during previous round OR 2772 // - no other track is not ready 2773 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2774 mixerStatus != MIXER_TRACKS_ENABLED) { 2775 mixerStatus = MIXER_TRACKS_READY; 2776 } 2777 } else { 2778 // clear effect chain input buffer if an active track underruns to avoid sending 2779 // previous audio buffer again to effects 2780 chain = getEffectChain_l(track->sessionId()); 2781 if (chain != 0) { 2782 chain->clearInputBuffer(); 2783 } 2784 2785 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2786 cblk->server, this); 2787 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2788 track->isStopped() || track->isPaused()) { 2789 // We have consumed all the buffers of this track. 2790 // Remove it from the list of active tracks. 2791 // TODO: use actual buffer filling status instead of latency when available from 2792 // audio HAL 2793 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2794 size_t framesWritten = mBytesWritten / mFrameSize; 2795 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2796 if (track->isStopped()) { 2797 track->reset(); 2798 } 2799 tracksToRemove->add(track); 2800 } 2801 } else { 2802 track->mUnderrunCount++; 2803 // No buffers for this track. Give it a few chances to 2804 // fill a buffer, then remove it from active list. 2805 if (--(track->mRetryCount) <= 0) { 2806 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2807 tracksToRemove->add(track); 2808 // indicate to client process that the track was disabled because of underrun; 2809 // it will then automatically call start() when data is available 2810 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2811 // If one track is not ready, mark the mixer also not ready if: 2812 // - the mixer was ready during previous round OR 2813 // - no other track is ready 2814 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2815 mixerStatus != MIXER_TRACKS_READY) { 2816 mixerStatus = MIXER_TRACKS_ENABLED; 2817 } 2818 } 2819 mAudioMixer->disable(name); 2820 } 2821 2822 } // local variable scope to avoid goto warning 2823track_is_ready: ; 2824 2825 } 2826 2827 // Push the new FastMixer state if necessary 2828 bool pauseAudioWatchdog = false; 2829 if (didModify) { 2830 state->mFastTracksGen++; 2831 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2832 if (kUseFastMixer == FastMixer_Dynamic && 2833 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2834 state->mCommand = FastMixerState::COLD_IDLE; 2835 state->mColdFutexAddr = &mFastMixerFutex; 2836 state->mColdGen++; 2837 mFastMixerFutex = 0; 2838 if (kUseFastMixer == FastMixer_Dynamic) { 2839 mNormalSink = mOutputSink; 2840 } 2841 // If we go into cold idle, need to wait for acknowledgement 2842 // so that fast mixer stops doing I/O. 2843 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2844 pauseAudioWatchdog = true; 2845 } 2846 sq->end(); 2847 } 2848 if (sq != NULL) { 2849 sq->end(didModify); 2850 sq->push(block); 2851 } 2852#ifdef AUDIO_WATCHDOG 2853 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2854 mAudioWatchdog->pause(); 2855 } 2856#endif 2857 2858 // Now perform the deferred reset on fast tracks that have stopped 2859 while (resetMask != 0) { 2860 size_t i = __builtin_ctz(resetMask); 2861 ALOG_ASSERT(i < count); 2862 resetMask &= ~(1 << i); 2863 sp<Track> t = mActiveTracks[i].promote(); 2864 if (t == 0) { 2865 continue; 2866 } 2867 Track* track = t.get(); 2868 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2869 track->reset(); 2870 } 2871 2872 // remove all the tracks that need to be... 2873 count = tracksToRemove->size(); 2874 if (CC_UNLIKELY(count)) { 2875 for (size_t i=0 ; i<count ; i++) { 2876 const sp<Track>& track = tracksToRemove->itemAt(i); 2877 mNBLogWriter->logf("prepareTracks_l remove name=%u", track->name()); 2878 mActiveTracks.remove(track); 2879 if (track->mainBuffer() != mMixBuffer) { 2880 chain = getEffectChain_l(track->sessionId()); 2881 if (chain != 0) { 2882 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2883 track->sessionId()); 2884 chain->decActiveTrackCnt(); 2885 } 2886 } 2887 if (track->isTerminated()) { 2888 removeTrack_l(track); 2889 } 2890 } 2891 } 2892 2893 // mix buffer must be cleared if all tracks are connected to an 2894 // effect chain as in this case the mixer will not write to 2895 // mix buffer and track effects will accumulate into it 2896 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2897 (mixedTracks == 0 && fastTracks > 0)) { 2898 // FIXME as a performance optimization, should remember previous zero status 2899 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2900 } 2901 2902 // if any fast tracks, then status is ready 2903 mMixerStatusIgnoringFastTracks = mixerStatus; 2904 if (fastTracks > 0) { 2905 mixerStatus = MIXER_TRACKS_READY; 2906 } 2907 return mixerStatus; 2908} 2909 2910// getTrackName_l() must be called with ThreadBase::mLock held 2911int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2912{ 2913 return mAudioMixer->getTrackName(channelMask, sessionId); 2914} 2915 2916// deleteTrackName_l() must be called with ThreadBase::mLock held 2917void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2918{ 2919 ALOGV("remove track (%d) and delete from mixer", name); 2920 mAudioMixer->deleteTrackName(name); 2921} 2922 2923// checkForNewParameters_l() must be called with ThreadBase::mLock held 2924bool AudioFlinger::MixerThread::checkForNewParameters_l() 2925{ 2926 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2927 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2928 bool reconfig = false; 2929 2930 while (!mNewParameters.isEmpty()) { 2931 2932 if (mFastMixer != NULL) { 2933 FastMixerStateQueue *sq = mFastMixer->sq(); 2934 FastMixerState *state = sq->begin(); 2935 if (!(state->mCommand & FastMixerState::IDLE)) { 2936 previousCommand = state->mCommand; 2937 state->mCommand = FastMixerState::HOT_IDLE; 2938 sq->end(); 2939 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2940 } else { 2941 sq->end(false /*didModify*/); 2942 } 2943 } 2944 2945 status_t status = NO_ERROR; 2946 String8 keyValuePair = mNewParameters[0]; 2947 AudioParameter param = AudioParameter(keyValuePair); 2948 int value; 2949 2950 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2951 reconfig = true; 2952 } 2953 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2954 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2955 status = BAD_VALUE; 2956 } else { 2957 reconfig = true; 2958 } 2959 } 2960 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2961 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2962 status = BAD_VALUE; 2963 } else { 2964 reconfig = true; 2965 } 2966 } 2967 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2968 // do not accept frame count changes if tracks are open as the track buffer 2969 // size depends on frame count and correct behavior would not be guaranteed 2970 // if frame count is changed after track creation 2971 if (!mTracks.isEmpty()) { 2972 status = INVALID_OPERATION; 2973 } else { 2974 reconfig = true; 2975 } 2976 } 2977 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2978#ifdef ADD_BATTERY_DATA 2979 // when changing the audio output device, call addBatteryData to notify 2980 // the change 2981 if (mOutDevice != value) { 2982 uint32_t params = 0; 2983 // check whether speaker is on 2984 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2985 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2986 } 2987 2988 audio_devices_t deviceWithoutSpeaker 2989 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2990 // check if any other device (except speaker) is on 2991 if (value & deviceWithoutSpeaker ) { 2992 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2993 } 2994 2995 if (params != 0) { 2996 addBatteryData(params); 2997 } 2998 } 2999#endif 3000 3001 // forward device change to effects that have requested to be 3002 // aware of attached audio device. 3003 mOutDevice = value; 3004 for (size_t i = 0; i < mEffectChains.size(); i++) { 3005 mEffectChains[i]->setDevice_l(mOutDevice); 3006 } 3007 } 3008 3009 if (status == NO_ERROR) { 3010 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3011 keyValuePair.string()); 3012 if (!mStandby && status == INVALID_OPERATION) { 3013 mOutput->stream->common.standby(&mOutput->stream->common); 3014 mStandby = true; 3015 mBytesWritten = 0; 3016 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3017 keyValuePair.string()); 3018 } 3019 if (status == NO_ERROR && reconfig) { 3020 delete mAudioMixer; 3021 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3022 mAudioMixer = NULL; 3023 readOutputParameters(); 3024 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3025 for (size_t i = 0; i < mTracks.size() ; i++) { 3026 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3027 if (name < 0) { 3028 break; 3029 } 3030 mTracks[i]->mName = name; 3031 } 3032 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3033 } 3034 } 3035 3036 mNewParameters.removeAt(0); 3037 3038 mParamStatus = status; 3039 mParamCond.signal(); 3040 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3041 // already timed out waiting for the status and will never signal the condition. 3042 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3043 } 3044 3045 if (!(previousCommand & FastMixerState::IDLE)) { 3046 ALOG_ASSERT(mFastMixer != NULL); 3047 FastMixerStateQueue *sq = mFastMixer->sq(); 3048 FastMixerState *state = sq->begin(); 3049 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3050 state->mCommand = previousCommand; 3051 sq->end(); 3052 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3053 } 3054 3055 return reconfig; 3056} 3057 3058 3059void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3060{ 3061 const size_t SIZE = 256; 3062 char buffer[SIZE]; 3063 String8 result; 3064 3065 PlaybackThread::dumpInternals(fd, args); 3066 3067 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3068 result.append(buffer); 3069 write(fd, result.string(), result.size()); 3070 3071 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3072 FastMixerDumpState copy = mFastMixerDumpState; 3073 copy.dump(fd); 3074 3075#ifdef STATE_QUEUE_DUMP 3076 // Similar for state queue 3077 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3078 observerCopy.dump(fd); 3079 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3080 mutatorCopy.dump(fd); 3081#endif 3082 3083#ifdef TEE_SINK 3084 // Write the tee output to a .wav file 3085 dumpTee(fd, mTeeSource, mId); 3086#endif 3087 3088#ifdef AUDIO_WATCHDOG 3089 if (mAudioWatchdog != 0) { 3090 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3091 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3092 wdCopy.dump(fd); 3093 } 3094#endif 3095} 3096 3097uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3098{ 3099 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3100} 3101 3102uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3103{ 3104 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3105} 3106 3107void AudioFlinger::MixerThread::cacheParameters_l() 3108{ 3109 PlaybackThread::cacheParameters_l(); 3110 3111 // FIXME: Relaxed timing because of a certain device that can't meet latency 3112 // Should be reduced to 2x after the vendor fixes the driver issue 3113 // increase threshold again due to low power audio mode. The way this warning 3114 // threshold is calculated and its usefulness should be reconsidered anyway. 3115 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3116} 3117 3118// ---------------------------------------------------------------------------- 3119 3120AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3121 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3122 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3123 // mLeftVolFloat, mRightVolFloat 3124{ 3125} 3126 3127AudioFlinger::DirectOutputThread::~DirectOutputThread() 3128{ 3129} 3130 3131AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3132 Vector< sp<Track> > *tracksToRemove 3133) 3134{ 3135 sp<Track> trackToRemove; 3136 3137 mixer_state mixerStatus = MIXER_IDLE; 3138 3139 // find out which tracks need to be processed 3140 if (mActiveTracks.size() != 0) { 3141 sp<Track> t = mActiveTracks[0].promote(); 3142 // The track died recently 3143 if (t == 0) { 3144 return MIXER_IDLE; 3145 } 3146 3147 Track* const track = t.get(); 3148 audio_track_cblk_t* cblk = track->cblk(); 3149 3150 // The first time a track is added we wait 3151 // for all its buffers to be filled before processing it 3152 uint32_t minFrames; 3153 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3154 minFrames = mNormalFrameCount; 3155 } else { 3156 minFrames = 1; 3157 } 3158 if ((track->framesReady() >= minFrames) && track->isReady() && 3159 !track->isPaused() && !track->isTerminated()) 3160 { 3161 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3162 3163 if (track->mFillingUpStatus == Track::FS_FILLED) { 3164 track->mFillingUpStatus = Track::FS_ACTIVE; 3165 mLeftVolFloat = mRightVolFloat = 0; 3166 if (track->mState == TrackBase::RESUMING) { 3167 track->mState = TrackBase::ACTIVE; 3168 } 3169 } 3170 3171 // compute volume for this track 3172 float left, right; 3173 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3174 left = right = 0; 3175 if (track->isPausing()) { 3176 track->setPaused(); 3177 } 3178 } else { 3179 float typeVolume = mStreamTypes[track->streamType()].volume; 3180 float v = mMasterVolume * typeVolume; 3181 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3182 float v_clamped = v * (vlr & 0xFFFF); 3183 if (v_clamped > MAX_GAIN) { 3184 v_clamped = MAX_GAIN; 3185 } 3186 left = v_clamped/MAX_GAIN; 3187 v_clamped = v * (vlr >> 16); 3188 if (v_clamped > MAX_GAIN) { 3189 v_clamped = MAX_GAIN; 3190 } 3191 right = v_clamped/MAX_GAIN; 3192 } 3193 3194 if (left != mLeftVolFloat || right != mRightVolFloat) { 3195 mLeftVolFloat = left; 3196 mRightVolFloat = right; 3197 3198 // Convert volumes from float to 8.24 3199 uint32_t vl = (uint32_t)(left * (1 << 24)); 3200 uint32_t vr = (uint32_t)(right * (1 << 24)); 3201 3202 // Delegate volume control to effect in track effect chain if needed 3203 // only one effect chain can be present on DirectOutputThread, so if 3204 // there is one, the track is connected to it 3205 if (!mEffectChains.isEmpty()) { 3206 // Do not ramp volume if volume is controlled by effect 3207 mEffectChains[0]->setVolume_l(&vl, &vr); 3208 left = (float)vl / (1 << 24); 3209 right = (float)vr / (1 << 24); 3210 } 3211 mOutput->stream->set_volume(mOutput->stream, left, right); 3212 } 3213 3214 // reset retry count 3215 track->mRetryCount = kMaxTrackRetriesDirect; 3216 mActiveTrack = t; 3217 mixerStatus = MIXER_TRACKS_READY; 3218 } else { 3219 // clear effect chain input buffer if an active track underruns to avoid sending 3220 // previous audio buffer again to effects 3221 if (!mEffectChains.isEmpty()) { 3222 mEffectChains[0]->clearInputBuffer(); 3223 } 3224 3225 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3226 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3227 track->isStopped() || track->isPaused()) { 3228 // We have consumed all the buffers of this track. 3229 // Remove it from the list of active tracks. 3230 // TODO: implement behavior for compressed audio 3231 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3232 size_t framesWritten = mBytesWritten / mFrameSize; 3233 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3234 if (track->isStopped()) { 3235 track->reset(); 3236 } 3237 trackToRemove = track; 3238 } 3239 } else { 3240 // No buffers for this track. Give it a few chances to 3241 // fill a buffer, then remove it from active list. 3242 if (--(track->mRetryCount) <= 0) { 3243 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3244 trackToRemove = track; 3245 } else { 3246 mixerStatus = MIXER_TRACKS_ENABLED; 3247 } 3248 } 3249 } 3250 } 3251 3252 // FIXME merge this with similar code for removing multiple tracks 3253 // remove all the tracks that need to be... 3254 if (CC_UNLIKELY(trackToRemove != 0)) { 3255 tracksToRemove->add(trackToRemove); 3256#if 0 3257 mNBLogWriter->logf("prepareTracks_l remove name=%u", trackToRemove->name()); 3258#endif 3259 mActiveTracks.remove(trackToRemove); 3260 if (!mEffectChains.isEmpty()) { 3261 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3262 trackToRemove->sessionId()); 3263 mEffectChains[0]->decActiveTrackCnt(); 3264 } 3265 if (trackToRemove->isTerminated()) { 3266 removeTrack_l(trackToRemove); 3267 } 3268 } 3269 3270 return mixerStatus; 3271} 3272 3273void AudioFlinger::DirectOutputThread::threadLoop_mix() 3274{ 3275 AudioBufferProvider::Buffer buffer; 3276 size_t frameCount = mFrameCount; 3277 int8_t *curBuf = (int8_t *)mMixBuffer; 3278 // output audio to hardware 3279 while (frameCount) { 3280 buffer.frameCount = frameCount; 3281 mActiveTrack->getNextBuffer(&buffer); 3282 if (CC_UNLIKELY(buffer.raw == NULL)) { 3283 memset(curBuf, 0, frameCount * mFrameSize); 3284 break; 3285 } 3286 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3287 frameCount -= buffer.frameCount; 3288 curBuf += buffer.frameCount * mFrameSize; 3289 mActiveTrack->releaseBuffer(&buffer); 3290 } 3291 sleepTime = 0; 3292 standbyTime = systemTime() + standbyDelay; 3293 mActiveTrack.clear(); 3294 3295} 3296 3297void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3298{ 3299 if (sleepTime == 0) { 3300 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3301 sleepTime = activeSleepTime; 3302 } else { 3303 sleepTime = idleSleepTime; 3304 } 3305 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3306 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3307 sleepTime = 0; 3308 } 3309} 3310 3311// getTrackName_l() must be called with ThreadBase::mLock held 3312int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3313 int sessionId) 3314{ 3315 return 0; 3316} 3317 3318// deleteTrackName_l() must be called with ThreadBase::mLock held 3319void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3320{ 3321} 3322 3323// checkForNewParameters_l() must be called with ThreadBase::mLock held 3324bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3325{ 3326 bool reconfig = false; 3327 3328 while (!mNewParameters.isEmpty()) { 3329 status_t status = NO_ERROR; 3330 String8 keyValuePair = mNewParameters[0]; 3331 AudioParameter param = AudioParameter(keyValuePair); 3332 int value; 3333 3334 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3335 // do not accept frame count changes if tracks are open as the track buffer 3336 // size depends on frame count and correct behavior would not be garantied 3337 // if frame count is changed after track creation 3338 if (!mTracks.isEmpty()) { 3339 status = INVALID_OPERATION; 3340 } else { 3341 reconfig = true; 3342 } 3343 } 3344 if (status == NO_ERROR) { 3345 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3346 keyValuePair.string()); 3347 if (!mStandby && status == INVALID_OPERATION) { 3348 mOutput->stream->common.standby(&mOutput->stream->common); 3349 mStandby = true; 3350 mBytesWritten = 0; 3351 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3352 keyValuePair.string()); 3353 } 3354 if (status == NO_ERROR && reconfig) { 3355 readOutputParameters(); 3356 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3357 } 3358 } 3359 3360 mNewParameters.removeAt(0); 3361 3362 mParamStatus = status; 3363 mParamCond.signal(); 3364 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3365 // already timed out waiting for the status and will never signal the condition. 3366 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3367 } 3368 return reconfig; 3369} 3370 3371uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3372{ 3373 uint32_t time; 3374 if (audio_is_linear_pcm(mFormat)) { 3375 time = PlaybackThread::activeSleepTimeUs(); 3376 } else { 3377 time = 10000; 3378 } 3379 return time; 3380} 3381 3382uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3383{ 3384 uint32_t time; 3385 if (audio_is_linear_pcm(mFormat)) { 3386 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3387 } else { 3388 time = 10000; 3389 } 3390 return time; 3391} 3392 3393uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3394{ 3395 uint32_t time; 3396 if (audio_is_linear_pcm(mFormat)) { 3397 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3398 } else { 3399 time = 10000; 3400 } 3401 return time; 3402} 3403 3404void AudioFlinger::DirectOutputThread::cacheParameters_l() 3405{ 3406 PlaybackThread::cacheParameters_l(); 3407 3408 // use shorter standby delay as on normal output to release 3409 // hardware resources as soon as possible 3410 standbyDelay = microseconds(activeSleepTime*2); 3411} 3412 3413// ---------------------------------------------------------------------------- 3414 3415AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3416 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3417 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3418 DUPLICATING), 3419 mWaitTimeMs(UINT_MAX) 3420{ 3421 addOutputTrack(mainThread); 3422} 3423 3424AudioFlinger::DuplicatingThread::~DuplicatingThread() 3425{ 3426 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3427 mOutputTracks[i]->destroy(); 3428 } 3429} 3430 3431void AudioFlinger::DuplicatingThread::threadLoop_mix() 3432{ 3433 // mix buffers... 3434 if (outputsReady(outputTracks)) { 3435 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3436 } else { 3437 memset(mMixBuffer, 0, mixBufferSize); 3438 } 3439 sleepTime = 0; 3440 writeFrames = mNormalFrameCount; 3441 standbyTime = systemTime() + standbyDelay; 3442} 3443 3444void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3445{ 3446 if (sleepTime == 0) { 3447 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3448 sleepTime = activeSleepTime; 3449 } else { 3450 sleepTime = idleSleepTime; 3451 } 3452 } else if (mBytesWritten != 0) { 3453 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3454 writeFrames = mNormalFrameCount; 3455 memset(mMixBuffer, 0, mixBufferSize); 3456 } else { 3457 // flush remaining overflow buffers in output tracks 3458 writeFrames = 0; 3459 } 3460 sleepTime = 0; 3461 } 3462} 3463 3464void AudioFlinger::DuplicatingThread::threadLoop_write() 3465{ 3466 for (size_t i = 0; i < outputTracks.size(); i++) { 3467 outputTracks[i]->write(mMixBuffer, writeFrames); 3468 } 3469 mBytesWritten += mixBufferSize; 3470} 3471 3472void AudioFlinger::DuplicatingThread::threadLoop_standby() 3473{ 3474 // DuplicatingThread implements standby by stopping all tracks 3475 for (size_t i = 0; i < outputTracks.size(); i++) { 3476 outputTracks[i]->stop(); 3477 } 3478} 3479 3480void AudioFlinger::DuplicatingThread::saveOutputTracks() 3481{ 3482 outputTracks = mOutputTracks; 3483} 3484 3485void AudioFlinger::DuplicatingThread::clearOutputTracks() 3486{ 3487 outputTracks.clear(); 3488} 3489 3490void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3491{ 3492 Mutex::Autolock _l(mLock); 3493 // FIXME explain this formula 3494 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3495 OutputTrack *outputTrack = new OutputTrack(thread, 3496 this, 3497 mSampleRate, 3498 mFormat, 3499 mChannelMask, 3500 frameCount); 3501 if (outputTrack->cblk() != NULL) { 3502 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3503 mOutputTracks.add(outputTrack); 3504 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3505 updateWaitTime_l(); 3506 } 3507} 3508 3509void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3510{ 3511 Mutex::Autolock _l(mLock); 3512 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3513 if (mOutputTracks[i]->thread() == thread) { 3514 mOutputTracks[i]->destroy(); 3515 mOutputTracks.removeAt(i); 3516 updateWaitTime_l(); 3517 return; 3518 } 3519 } 3520 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3521} 3522 3523// caller must hold mLock 3524void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3525{ 3526 mWaitTimeMs = UINT_MAX; 3527 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3528 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3529 if (strong != 0) { 3530 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3531 if (waitTimeMs < mWaitTimeMs) { 3532 mWaitTimeMs = waitTimeMs; 3533 } 3534 } 3535 } 3536} 3537 3538 3539bool AudioFlinger::DuplicatingThread::outputsReady( 3540 const SortedVector< sp<OutputTrack> > &outputTracks) 3541{ 3542 for (size_t i = 0; i < outputTracks.size(); i++) { 3543 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3544 if (thread == 0) { 3545 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3546 outputTracks[i].get()); 3547 return false; 3548 } 3549 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3550 // see note at standby() declaration 3551 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3552 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3553 thread.get()); 3554 return false; 3555 } 3556 } 3557 return true; 3558} 3559 3560uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3561{ 3562 return (mWaitTimeMs * 1000) / 2; 3563} 3564 3565void AudioFlinger::DuplicatingThread::cacheParameters_l() 3566{ 3567 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3568 updateWaitTime_l(); 3569 3570 MixerThread::cacheParameters_l(); 3571} 3572 3573// ---------------------------------------------------------------------------- 3574// Record 3575// ---------------------------------------------------------------------------- 3576 3577AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3578 AudioStreamIn *input, 3579 uint32_t sampleRate, 3580 audio_channel_mask_t channelMask, 3581 audio_io_handle_t id, 3582 audio_devices_t outDevice, 3583 audio_devices_t inDevice 3584#ifdef TEE_SINK 3585 , const sp<NBAIO_Sink>& teeSink 3586#endif 3587 ) : 3588 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3589 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3590 // mRsmpInIndex and mInputBytes set by readInputParameters() 3591 mReqChannelCount(popcount(channelMask)), 3592 mReqSampleRate(sampleRate) 3593 // mBytesRead is only meaningful while active, and so is cleared in start() 3594 // (but might be better to also clear here for dump?) 3595#ifdef TEE_SINK 3596 , mTeeSink(teeSink) 3597#endif 3598{ 3599 snprintf(mName, kNameLength, "AudioIn_%X", id); 3600 3601 readInputParameters(); 3602 3603} 3604 3605 3606AudioFlinger::RecordThread::~RecordThread() 3607{ 3608 delete[] mRsmpInBuffer; 3609 delete mResampler; 3610 delete[] mRsmpOutBuffer; 3611} 3612 3613void AudioFlinger::RecordThread::onFirstRef() 3614{ 3615 run(mName, PRIORITY_URGENT_AUDIO); 3616} 3617 3618status_t AudioFlinger::RecordThread::readyToRun() 3619{ 3620 status_t status = initCheck(); 3621 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3622 return status; 3623} 3624 3625bool AudioFlinger::RecordThread::threadLoop() 3626{ 3627 AudioBufferProvider::Buffer buffer; 3628 sp<RecordTrack> activeTrack; 3629 Vector< sp<EffectChain> > effectChains; 3630 3631 nsecs_t lastWarning = 0; 3632 3633 inputStandBy(); 3634 acquireWakeLock(); 3635 3636 // used to verify we've read at least once before evaluating how many bytes were read 3637 bool readOnce = false; 3638 3639 // start recording 3640 while (!exitPending()) { 3641 3642 processConfigEvents(); 3643 3644 { // scope for mLock 3645 Mutex::Autolock _l(mLock); 3646 checkForNewParameters_l(); 3647 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3648 standby(); 3649 3650 if (exitPending()) { 3651 break; 3652 } 3653 3654 releaseWakeLock_l(); 3655 ALOGV("RecordThread: loop stopping"); 3656 // go to sleep 3657 mWaitWorkCV.wait(mLock); 3658 ALOGV("RecordThread: loop starting"); 3659 acquireWakeLock_l(); 3660 continue; 3661 } 3662 if (mActiveTrack != 0) { 3663 if (mActiveTrack->mState == TrackBase::PAUSING) { 3664 standby(); 3665 mActiveTrack.clear(); 3666 mStartStopCond.broadcast(); 3667 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3668 if (mReqChannelCount != mActiveTrack->channelCount()) { 3669 mActiveTrack.clear(); 3670 mStartStopCond.broadcast(); 3671 } else if (readOnce) { 3672 // record start succeeds only if first read from audio input 3673 // succeeds 3674 if (mBytesRead >= 0) { 3675 mActiveTrack->mState = TrackBase::ACTIVE; 3676 } else { 3677 mActiveTrack.clear(); 3678 } 3679 mStartStopCond.broadcast(); 3680 } 3681 mStandby = false; 3682 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3683 removeTrack_l(mActiveTrack); 3684 mActiveTrack.clear(); 3685 } 3686 } 3687 lockEffectChains_l(effectChains); 3688 } 3689 3690 if (mActiveTrack != 0) { 3691 if (mActiveTrack->mState != TrackBase::ACTIVE && 3692 mActiveTrack->mState != TrackBase::RESUMING) { 3693 unlockEffectChains(effectChains); 3694 usleep(kRecordThreadSleepUs); 3695 continue; 3696 } 3697 for (size_t i = 0; i < effectChains.size(); i ++) { 3698 effectChains[i]->process_l(); 3699 } 3700 3701 buffer.frameCount = mFrameCount; 3702 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3703 readOnce = true; 3704 size_t framesOut = buffer.frameCount; 3705 if (mResampler == NULL) { 3706 // no resampling 3707 while (framesOut) { 3708 size_t framesIn = mFrameCount - mRsmpInIndex; 3709 if (framesIn) { 3710 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3711 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3712 mActiveTrack->mFrameSize; 3713 if (framesIn > framesOut) 3714 framesIn = framesOut; 3715 mRsmpInIndex += framesIn; 3716 framesOut -= framesIn; 3717 if (mChannelCount == mReqChannelCount || 3718 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3719 memcpy(dst, src, framesIn * mFrameSize); 3720 } else { 3721 if (mChannelCount == 1) { 3722 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3723 (int16_t *)src, framesIn); 3724 } else { 3725 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3726 (int16_t *)src, framesIn); 3727 } 3728 } 3729 } 3730 if (framesOut && mFrameCount == mRsmpInIndex) { 3731 void *readInto; 3732 if (framesOut == mFrameCount && 3733 (mChannelCount == mReqChannelCount || 3734 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3735 readInto = buffer.raw; 3736 framesOut = 0; 3737 } else { 3738 readInto = mRsmpInBuffer; 3739 mRsmpInIndex = 0; 3740 } 3741 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3742 if (mBytesRead <= 0) { 3743 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3744 { 3745 ALOGE("Error reading audio input"); 3746 // Force input into standby so that it tries to 3747 // recover at next read attempt 3748 inputStandBy(); 3749 usleep(kRecordThreadSleepUs); 3750 } 3751 mRsmpInIndex = mFrameCount; 3752 framesOut = 0; 3753 buffer.frameCount = 0; 3754 } 3755#ifdef TEE_SINK 3756 else if (mTeeSink != 0) { 3757 (void) mTeeSink->write(readInto, 3758 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3759 } 3760#endif 3761 } 3762 } 3763 } else { 3764 // resampling 3765 3766 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3767 // alter output frame count as if we were expecting stereo samples 3768 if (mChannelCount == 1 && mReqChannelCount == 1) { 3769 framesOut >>= 1; 3770 } 3771 mResampler->resample(mRsmpOutBuffer, framesOut, 3772 this /* AudioBufferProvider* */); 3773 // ditherAndClamp() works as long as all buffers returned by 3774 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3775 if (mChannelCount == 2 && mReqChannelCount == 1) { 3776 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3777 // the resampler always outputs stereo samples: 3778 // do post stereo to mono conversion 3779 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3780 framesOut); 3781 } else { 3782 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3783 } 3784 3785 } 3786 if (mFramestoDrop == 0) { 3787 mActiveTrack->releaseBuffer(&buffer); 3788 } else { 3789 if (mFramestoDrop > 0) { 3790 mFramestoDrop -= buffer.frameCount; 3791 if (mFramestoDrop <= 0) { 3792 clearSyncStartEvent(); 3793 } 3794 } else { 3795 mFramestoDrop += buffer.frameCount; 3796 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3797 mSyncStartEvent->isCancelled()) { 3798 ALOGW("Synced record %s, session %d, trigger session %d", 3799 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3800 mActiveTrack->sessionId(), 3801 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3802 clearSyncStartEvent(); 3803 } 3804 } 3805 } 3806 mActiveTrack->clearOverflow(); 3807 } 3808 // client isn't retrieving buffers fast enough 3809 else { 3810 if (!mActiveTrack->setOverflow()) { 3811 nsecs_t now = systemTime(); 3812 if ((now - lastWarning) > kWarningThrottleNs) { 3813 ALOGW("RecordThread: buffer overflow"); 3814 lastWarning = now; 3815 } 3816 } 3817 // Release the processor for a while before asking for a new buffer. 3818 // This will give the application more chance to read from the buffer and 3819 // clear the overflow. 3820 usleep(kRecordThreadSleepUs); 3821 } 3822 } 3823 // enable changes in effect chain 3824 unlockEffectChains(effectChains); 3825 effectChains.clear(); 3826 } 3827 3828 standby(); 3829 3830 { 3831 Mutex::Autolock _l(mLock); 3832 mActiveTrack.clear(); 3833 mStartStopCond.broadcast(); 3834 } 3835 3836 releaseWakeLock(); 3837 3838 ALOGV("RecordThread %p exiting", this); 3839 return false; 3840} 3841 3842void AudioFlinger::RecordThread::standby() 3843{ 3844 if (!mStandby) { 3845 inputStandBy(); 3846 mStandby = true; 3847 } 3848} 3849 3850void AudioFlinger::RecordThread::inputStandBy() 3851{ 3852 mInput->stream->common.standby(&mInput->stream->common); 3853} 3854 3855sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3856 const sp<AudioFlinger::Client>& client, 3857 uint32_t sampleRate, 3858 audio_format_t format, 3859 audio_channel_mask_t channelMask, 3860 size_t frameCount, 3861 int sessionId, 3862 IAudioFlinger::track_flags_t flags, 3863 pid_t tid, 3864 status_t *status) 3865{ 3866 sp<RecordTrack> track; 3867 status_t lStatus; 3868 3869 lStatus = initCheck(); 3870 if (lStatus != NO_ERROR) { 3871 ALOGE("Audio driver not initialized."); 3872 goto Exit; 3873 } 3874 3875 // FIXME use flags and tid similar to createTrack_l() 3876 3877 { // scope for mLock 3878 Mutex::Autolock _l(mLock); 3879 3880 track = new RecordTrack(this, client, sampleRate, 3881 format, channelMask, frameCount, sessionId); 3882 3883 if (track->getCblk() == 0) { 3884 lStatus = NO_MEMORY; 3885 goto Exit; 3886 } 3887 mTracks.add(track); 3888 3889 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3890 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3891 mAudioFlinger->btNrecIsOff(); 3892 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3893 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3894 } 3895 lStatus = NO_ERROR; 3896 3897Exit: 3898 if (status) { 3899 *status = lStatus; 3900 } 3901 return track; 3902} 3903 3904status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3905 AudioSystem::sync_event_t event, 3906 int triggerSession) 3907{ 3908 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3909 sp<ThreadBase> strongMe = this; 3910 status_t status = NO_ERROR; 3911 3912 if (event == AudioSystem::SYNC_EVENT_NONE) { 3913 clearSyncStartEvent(); 3914 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3915 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3916 triggerSession, 3917 recordTrack->sessionId(), 3918 syncStartEventCallback, 3919 this); 3920 // Sync event can be cancelled by the trigger session if the track is not in a 3921 // compatible state in which case we start record immediately 3922 if (mSyncStartEvent->isCancelled()) { 3923 clearSyncStartEvent(); 3924 } else { 3925 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3926 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3927 } 3928 } 3929 3930 { 3931 AutoMutex lock(mLock); 3932 if (mActiveTrack != 0) { 3933 if (recordTrack != mActiveTrack.get()) { 3934 status = -EBUSY; 3935 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3936 mActiveTrack->mState = TrackBase::ACTIVE; 3937 } 3938 return status; 3939 } 3940 3941 recordTrack->mState = TrackBase::IDLE; 3942 mActiveTrack = recordTrack; 3943 mLock.unlock(); 3944 status_t status = AudioSystem::startInput(mId); 3945 mLock.lock(); 3946 if (status != NO_ERROR) { 3947 mActiveTrack.clear(); 3948 clearSyncStartEvent(); 3949 return status; 3950 } 3951 mRsmpInIndex = mFrameCount; 3952 mBytesRead = 0; 3953 if (mResampler != NULL) { 3954 mResampler->reset(); 3955 } 3956 mActiveTrack->mState = TrackBase::RESUMING; 3957 // signal thread to start 3958 ALOGV("Signal record thread"); 3959 mWaitWorkCV.broadcast(); 3960 // do not wait for mStartStopCond if exiting 3961 if (exitPending()) { 3962 mActiveTrack.clear(); 3963 status = INVALID_OPERATION; 3964 goto startError; 3965 } 3966 mStartStopCond.wait(mLock); 3967 if (mActiveTrack == 0) { 3968 ALOGV("Record failed to start"); 3969 status = BAD_VALUE; 3970 goto startError; 3971 } 3972 ALOGV("Record started OK"); 3973 return status; 3974 } 3975startError: 3976 AudioSystem::stopInput(mId); 3977 clearSyncStartEvent(); 3978 return status; 3979} 3980 3981void AudioFlinger::RecordThread::clearSyncStartEvent() 3982{ 3983 if (mSyncStartEvent != 0) { 3984 mSyncStartEvent->cancel(); 3985 } 3986 mSyncStartEvent.clear(); 3987 mFramestoDrop = 0; 3988} 3989 3990void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3991{ 3992 sp<SyncEvent> strongEvent = event.promote(); 3993 3994 if (strongEvent != 0) { 3995 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3996 me->handleSyncStartEvent(strongEvent); 3997 } 3998} 3999 4000void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4001{ 4002 if (event == mSyncStartEvent) { 4003 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4004 // from audio HAL 4005 mFramestoDrop = mFrameCount * 2; 4006 } 4007} 4008 4009bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4010 ALOGV("RecordThread::stop"); 4011 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4012 return false; 4013 } 4014 recordTrack->mState = TrackBase::PAUSING; 4015 // do not wait for mStartStopCond if exiting 4016 if (exitPending()) { 4017 return true; 4018 } 4019 mStartStopCond.wait(mLock); 4020 // if we have been restarted, recordTrack == mActiveTrack.get() here 4021 if (exitPending() || recordTrack != mActiveTrack.get()) { 4022 ALOGV("Record stopped OK"); 4023 return true; 4024 } 4025 return false; 4026} 4027 4028bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4029{ 4030 return false; 4031} 4032 4033status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4034{ 4035#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4036 if (!isValidSyncEvent(event)) { 4037 return BAD_VALUE; 4038 } 4039 4040 int eventSession = event->triggerSession(); 4041 status_t ret = NAME_NOT_FOUND; 4042 4043 Mutex::Autolock _l(mLock); 4044 4045 for (size_t i = 0; i < mTracks.size(); i++) { 4046 sp<RecordTrack> track = mTracks[i]; 4047 if (eventSession == track->sessionId()) { 4048 (void) track->setSyncEvent(event); 4049 ret = NO_ERROR; 4050 } 4051 } 4052 return ret; 4053#else 4054 return BAD_VALUE; 4055#endif 4056} 4057 4058// destroyTrack_l() must be called with ThreadBase::mLock held 4059void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4060{ 4061 track->mState = TrackBase::TERMINATED; 4062 // active tracks are removed by threadLoop() 4063 if (mActiveTrack != track) { 4064 removeTrack_l(track); 4065 } 4066} 4067 4068void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4069{ 4070 mTracks.remove(track); 4071 // need anything related to effects here? 4072} 4073 4074void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4075{ 4076 dumpInternals(fd, args); 4077 dumpTracks(fd, args); 4078 dumpEffectChains(fd, args); 4079} 4080 4081void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4082{ 4083 const size_t SIZE = 256; 4084 char buffer[SIZE]; 4085 String8 result; 4086 4087 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4088 result.append(buffer); 4089 4090 if (mActiveTrack != 0) { 4091 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4092 result.append(buffer); 4093 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4094 result.append(buffer); 4095 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4096 result.append(buffer); 4097 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4098 result.append(buffer); 4099 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4100 result.append(buffer); 4101 } else { 4102 result.append("No active record client\n"); 4103 } 4104 4105 write(fd, result.string(), result.size()); 4106 4107 dumpBase(fd, args); 4108} 4109 4110void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4111{ 4112 const size_t SIZE = 256; 4113 char buffer[SIZE]; 4114 String8 result; 4115 4116 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4117 result.append(buffer); 4118 RecordTrack::appendDumpHeader(result); 4119 for (size_t i = 0; i < mTracks.size(); ++i) { 4120 sp<RecordTrack> track = mTracks[i]; 4121 if (track != 0) { 4122 track->dump(buffer, SIZE); 4123 result.append(buffer); 4124 } 4125 } 4126 4127 if (mActiveTrack != 0) { 4128 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4129 result.append(buffer); 4130 RecordTrack::appendDumpHeader(result); 4131 mActiveTrack->dump(buffer, SIZE); 4132 result.append(buffer); 4133 4134 } 4135 write(fd, result.string(), result.size()); 4136} 4137 4138// AudioBufferProvider interface 4139status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4140{ 4141 size_t framesReq = buffer->frameCount; 4142 size_t framesReady = mFrameCount - mRsmpInIndex; 4143 int channelCount; 4144 4145 if (framesReady == 0) { 4146 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4147 if (mBytesRead <= 0) { 4148 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4149 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4150 // Force input into standby so that it tries to 4151 // recover at next read attempt 4152 inputStandBy(); 4153 usleep(kRecordThreadSleepUs); 4154 } 4155 buffer->raw = NULL; 4156 buffer->frameCount = 0; 4157 return NOT_ENOUGH_DATA; 4158 } 4159 mRsmpInIndex = 0; 4160 framesReady = mFrameCount; 4161 } 4162 4163 if (framesReq > framesReady) { 4164 framesReq = framesReady; 4165 } 4166 4167 if (mChannelCount == 1 && mReqChannelCount == 2) { 4168 channelCount = 1; 4169 } else { 4170 channelCount = 2; 4171 } 4172 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4173 buffer->frameCount = framesReq; 4174 return NO_ERROR; 4175} 4176 4177// AudioBufferProvider interface 4178void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4179{ 4180 mRsmpInIndex += buffer->frameCount; 4181 buffer->frameCount = 0; 4182} 4183 4184bool AudioFlinger::RecordThread::checkForNewParameters_l() 4185{ 4186 bool reconfig = false; 4187 4188 while (!mNewParameters.isEmpty()) { 4189 status_t status = NO_ERROR; 4190 String8 keyValuePair = mNewParameters[0]; 4191 AudioParameter param = AudioParameter(keyValuePair); 4192 int value; 4193 audio_format_t reqFormat = mFormat; 4194 uint32_t reqSamplingRate = mReqSampleRate; 4195 uint32_t reqChannelCount = mReqChannelCount; 4196 4197 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4198 reqSamplingRate = value; 4199 reconfig = true; 4200 } 4201 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4202 reqFormat = (audio_format_t) value; 4203 reconfig = true; 4204 } 4205 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4206 reqChannelCount = popcount(value); 4207 reconfig = true; 4208 } 4209 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4210 // do not accept frame count changes if tracks are open as the track buffer 4211 // size depends on frame count and correct behavior would not be guaranteed 4212 // if frame count is changed after track creation 4213 if (mActiveTrack != 0) { 4214 status = INVALID_OPERATION; 4215 } else { 4216 reconfig = true; 4217 } 4218 } 4219 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4220 // forward device change to effects that have requested to be 4221 // aware of attached audio device. 4222 for (size_t i = 0; i < mEffectChains.size(); i++) { 4223 mEffectChains[i]->setDevice_l(value); 4224 } 4225 4226 // store input device and output device but do not forward output device to audio HAL. 4227 // Note that status is ignored by the caller for output device 4228 // (see AudioFlinger::setParameters() 4229 if (audio_is_output_devices(value)) { 4230 mOutDevice = value; 4231 status = BAD_VALUE; 4232 } else { 4233 mInDevice = value; 4234 // disable AEC and NS if the device is a BT SCO headset supporting those 4235 // pre processings 4236 if (mTracks.size() > 0) { 4237 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4238 mAudioFlinger->btNrecIsOff(); 4239 for (size_t i = 0; i < mTracks.size(); i++) { 4240 sp<RecordTrack> track = mTracks[i]; 4241 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4242 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4243 } 4244 } 4245 } 4246 } 4247 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4248 mAudioSource != (audio_source_t)value) { 4249 // forward device change to effects that have requested to be 4250 // aware of attached audio device. 4251 for (size_t i = 0; i < mEffectChains.size(); i++) { 4252 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4253 } 4254 mAudioSource = (audio_source_t)value; 4255 } 4256 if (status == NO_ERROR) { 4257 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4258 keyValuePair.string()); 4259 if (status == INVALID_OPERATION) { 4260 inputStandBy(); 4261 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4262 keyValuePair.string()); 4263 } 4264 if (reconfig) { 4265 if (status == BAD_VALUE && 4266 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4267 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4268 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4269 <= (2 * reqSamplingRate)) && 4270 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4271 <= FCC_2 && 4272 (reqChannelCount <= FCC_2)) { 4273 status = NO_ERROR; 4274 } 4275 if (status == NO_ERROR) { 4276 readInputParameters(); 4277 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4278 } 4279 } 4280 } 4281 4282 mNewParameters.removeAt(0); 4283 4284 mParamStatus = status; 4285 mParamCond.signal(); 4286 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4287 // already timed out waiting for the status and will never signal the condition. 4288 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4289 } 4290 return reconfig; 4291} 4292 4293String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4294{ 4295 char *s; 4296 String8 out_s8 = String8(); 4297 4298 Mutex::Autolock _l(mLock); 4299 if (initCheck() != NO_ERROR) { 4300 return out_s8; 4301 } 4302 4303 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4304 out_s8 = String8(s); 4305 free(s); 4306 return out_s8; 4307} 4308 4309void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4310 AudioSystem::OutputDescriptor desc; 4311 void *param2 = NULL; 4312 4313 switch (event) { 4314 case AudioSystem::INPUT_OPENED: 4315 case AudioSystem::INPUT_CONFIG_CHANGED: 4316 desc.channels = mChannelMask; 4317 desc.samplingRate = mSampleRate; 4318 desc.format = mFormat; 4319 desc.frameCount = mFrameCount; 4320 desc.latency = 0; 4321 param2 = &desc; 4322 break; 4323 4324 case AudioSystem::INPUT_CLOSED: 4325 default: 4326 break; 4327 } 4328 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4329} 4330 4331void AudioFlinger::RecordThread::readInputParameters() 4332{ 4333 delete mRsmpInBuffer; 4334 // mRsmpInBuffer is always assigned a new[] below 4335 delete mRsmpOutBuffer; 4336 mRsmpOutBuffer = NULL; 4337 delete mResampler; 4338 mResampler = NULL; 4339 4340 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4341 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4342 mChannelCount = (uint16_t)popcount(mChannelMask); 4343 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4344 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4345 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4346 mFrameCount = mInputBytes / mFrameSize; 4347 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4348 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4349 4350 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4351 { 4352 int channelCount; 4353 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4354 // stereo to mono post process as the resampler always outputs stereo. 4355 if (mChannelCount == 1 && mReqChannelCount == 2) { 4356 channelCount = 1; 4357 } else { 4358 channelCount = 2; 4359 } 4360 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4361 mResampler->setSampleRate(mSampleRate); 4362 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4363 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4364 4365 // optmization: if mono to mono, alter input frame count as if we were inputing 4366 // stereo samples 4367 if (mChannelCount == 1 && mReqChannelCount == 1) { 4368 mFrameCount >>= 1; 4369 } 4370 4371 } 4372 mRsmpInIndex = mFrameCount; 4373} 4374 4375unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4376{ 4377 Mutex::Autolock _l(mLock); 4378 if (initCheck() != NO_ERROR) { 4379 return 0; 4380 } 4381 4382 return mInput->stream->get_input_frames_lost(mInput->stream); 4383} 4384 4385uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4386{ 4387 Mutex::Autolock _l(mLock); 4388 uint32_t result = 0; 4389 if (getEffectChain_l(sessionId) != 0) { 4390 result = EFFECT_SESSION; 4391 } 4392 4393 for (size_t i = 0; i < mTracks.size(); ++i) { 4394 if (sessionId == mTracks[i]->sessionId()) { 4395 result |= TRACK_SESSION; 4396 break; 4397 } 4398 } 4399 4400 return result; 4401} 4402 4403KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4404{ 4405 KeyedVector<int, bool> ids; 4406 Mutex::Autolock _l(mLock); 4407 for (size_t j = 0; j < mTracks.size(); ++j) { 4408 sp<RecordThread::RecordTrack> track = mTracks[j]; 4409 int sessionId = track->sessionId(); 4410 if (ids.indexOfKey(sessionId) < 0) { 4411 ids.add(sessionId, true); 4412 } 4413 } 4414 return ids; 4415} 4416 4417AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4418{ 4419 Mutex::Autolock _l(mLock); 4420 AudioStreamIn *input = mInput; 4421 mInput = NULL; 4422 return input; 4423} 4424 4425// this method must always be called either with ThreadBase mLock held or inside the thread loop 4426audio_stream_t* AudioFlinger::RecordThread::stream() const 4427{ 4428 if (mInput == NULL) { 4429 return NULL; 4430 } 4431 return &mInput->stream->common; 4432} 4433 4434status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4435{ 4436 // only one chain per input thread 4437 if (mEffectChains.size() != 0) { 4438 return INVALID_OPERATION; 4439 } 4440 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4441 4442 chain->setInBuffer(NULL); 4443 chain->setOutBuffer(NULL); 4444 4445 checkSuspendOnAddEffectChain_l(chain); 4446 4447 mEffectChains.add(chain); 4448 4449 return NO_ERROR; 4450} 4451 4452size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4453{ 4454 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4455 ALOGW_IF(mEffectChains.size() != 1, 4456 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4457 chain.get(), mEffectChains.size(), this); 4458 if (mEffectChains.size() == 1) { 4459 mEffectChains.removeAt(0); 4460 } 4461 return 0; 4462} 4463 4464}; // namespace android 4465