Threads.cpp revision 9f80dd223d83d9bb9077fb6baee056cee4eaf7e5
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 = 1; 143 144// ---------------------------------------------------------------------------- 145 146#ifdef ADD_BATTERY_DATA 147// To collect the amplifier usage 148static void addBatteryData(uint32_t params) { 149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150 if (service == NULL) { 151 // it already logged 152 return; 153 } 154 155 service->addBatteryData(params); 156} 157#endif 158 159 160// ---------------------------------------------------------------------------- 161// CPU Stats 162// ---------------------------------------------------------------------------- 163 164class CpuStats { 165public: 166 CpuStats(); 167 void sample(const String8 &title); 168#ifdef DEBUG_CPU_USAGE 169private: 170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172 173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174 175 int mCpuNum; // thread's current CPU number 176 int mCpukHz; // frequency of thread's current CPU in kHz 177#endif 178}; 179 180CpuStats::CpuStats() 181#ifdef DEBUG_CPU_USAGE 182 : mCpuNum(-1), mCpukHz(-1) 183#endif 184{ 185} 186 187void CpuStats::sample(const String8 &title) { 188#ifdef DEBUG_CPU_USAGE 189 // get current thread's delta CPU time in wall clock ns 190 double wcNs; 191 bool valid = mCpuUsage.sampleAndEnable(wcNs); 192 193 // record sample for wall clock statistics 194 if (valid) { 195 mWcStats.sample(wcNs); 196 } 197 198 // get the current CPU number 199 int cpuNum = sched_getcpu(); 200 201 // get the current CPU frequency in kHz 202 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203 204 // check if either CPU number or frequency changed 205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 206 mCpuNum = cpuNum; 207 mCpukHz = cpukHz; 208 // ignore sample for purposes of cycles 209 valid = false; 210 } 211 212 // if no change in CPU number or frequency, then record sample for cycle statistics 213 if (valid && mCpukHz > 0) { 214 double cycles = wcNs * cpukHz * 0.000001; 215 mHzStats.sample(cycles); 216 } 217 218 unsigned n = mWcStats.n(); 219 // mCpuUsage.elapsed() is expensive, so don't call it every loop 220 if ((n & 127) == 1) { 221 long long elapsed = mCpuUsage.elapsed(); 222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 223 double perLoop = elapsed / (double) n; 224 double perLoop100 = perLoop * 0.01; 225 double perLoop1k = perLoop * 0.001; 226 double mean = mWcStats.mean(); 227 double stddev = mWcStats.stddev(); 228 double minimum = mWcStats.minimum(); 229 double maximum = mWcStats.maximum(); 230 double meanCycles = mHzStats.mean(); 231 double stddevCycles = mHzStats.stddev(); 232 double minCycles = mHzStats.minimum(); 233 double maxCycles = mHzStats.maximum(); 234 mCpuUsage.resetElapsed(); 235 mWcStats.reset(); 236 mHzStats.reset(); 237 ALOGD("CPU usage for %s over past %.1f secs\n" 238 " (%u mixer loops at %.1f mean ms per loop):\n" 239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 242 title.string(), 243 elapsed * .000000001, n, perLoop * .000001, 244 mean * .001, 245 stddev * .001, 246 minimum * .001, 247 maximum * .001, 248 mean / perLoop100, 249 stddev / perLoop100, 250 minimum / perLoop100, 251 maximum / perLoop100, 252 meanCycles / perLoop1k, 253 stddevCycles / perLoop1k, 254 minCycles / perLoop1k, 255 maxCycles / perLoop1k); 256 257 } 258 } 259#endif 260}; 261 262// ---------------------------------------------------------------------------- 263// ThreadBase 264// ---------------------------------------------------------------------------- 265 266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 268 : Thread(false /*canCallJava*/), 269 mType(type), 270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 271 // mChannelMask 272 mChannelCount(0), 273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 mParamCond.broadcast(); 285 // do not lock the mutex in destructor 286 releaseWakeLock_l(); 287 if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 // FIXME Need to understand why this has be done asynchronously 377 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 378 true /*asynchronous*/); 379 if (err != 0) { 380 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 381 "error %d", 382 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 383 } 384 } break; 385 case CFG_EVENT_IO: { 386 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 387 mAudioFlinger->mLock.lock(); 388 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 389 mAudioFlinger->mLock.unlock(); 390 } break; 391 default: 392 ALOGE("processConfigEvents() unknown event type %d", event->type()); 393 break; 394 } 395 delete event; 396 mLock.lock(); 397 } 398 mLock.unlock(); 399} 400 401void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 402{ 403 const size_t SIZE = 256; 404 char buffer[SIZE]; 405 String8 result; 406 407 bool locked = AudioFlinger::dumpTryLock(mLock); 408 if (!locked) { 409 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 410 write(fd, buffer, strlen(buffer)); 411 } 412 413 snprintf(buffer, SIZE, "io handle: %d\n", mId); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 432 result.append(buffer); 433 434 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 435 result.append(buffer); 436 result.append(" Index Command"); 437 for (size_t i = 0; i < mNewParameters.size(); ++i) { 438 snprintf(buffer, SIZE, "\n %02d ", i); 439 result.append(buffer); 440 result.append(mNewParameters[i]); 441 } 442 443 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 444 result.append(buffer); 445 for (size_t i = 0; i < mConfigEvents.size(); i++) { 446 mConfigEvents[i]->dump(buffer, SIZE); 447 result.append(buffer); 448 } 449 result.append("\n"); 450 451 write(fd, result.string(), result.size()); 452 453 if (locked) { 454 mLock.unlock(); 455 } 456} 457 458void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 459{ 460 const size_t SIZE = 256; 461 char buffer[SIZE]; 462 String8 result; 463 464 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 465 write(fd, buffer, strlen(buffer)); 466 467 for (size_t i = 0; i < mEffectChains.size(); ++i) { 468 sp<EffectChain> chain = mEffectChains[i]; 469 if (chain != 0) { 470 chain->dump(fd, args); 471 } 472 } 473} 474 475void AudioFlinger::ThreadBase::acquireWakeLock() 476{ 477 Mutex::Autolock _l(mLock); 478 acquireWakeLock_l(); 479} 480 481void AudioFlinger::ThreadBase::acquireWakeLock_l() 482{ 483 if (mPowerManager == 0) { 484 // use checkService() to avoid blocking if power service is not up yet 485 sp<IBinder> binder = 486 defaultServiceManager()->checkService(String16("power")); 487 if (binder == 0) { 488 ALOGW("Thread %s cannot connect to the power manager service", mName); 489 } else { 490 mPowerManager = interface_cast<IPowerManager>(binder); 491 binder->linkToDeath(mDeathRecipient); 492 } 493 } 494 if (mPowerManager != 0) { 495 sp<IBinder> binder = new BBinder(); 496 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 497 binder, 498 String16(mName), 499 String16("media")); 500 if (status == NO_ERROR) { 501 mWakeLockToken = binder; 502 } 503 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 504 } 505} 506 507void AudioFlinger::ThreadBase::releaseWakeLock() 508{ 509 Mutex::Autolock _l(mLock); 510 releaseWakeLock_l(); 511} 512 513void AudioFlinger::ThreadBase::releaseWakeLock_l() 514{ 515 if (mWakeLockToken != 0) { 516 ALOGV("releaseWakeLock_l() %s", mName); 517 if (mPowerManager != 0) { 518 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 519 } 520 mWakeLockToken.clear(); 521 } 522} 523 524void AudioFlinger::ThreadBase::clearPowerManager() 525{ 526 Mutex::Autolock _l(mLock); 527 releaseWakeLock_l(); 528 mPowerManager.clear(); 529} 530 531void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 532{ 533 sp<ThreadBase> thread = mThread.promote(); 534 if (thread != 0) { 535 thread->clearPowerManager(); 536 } 537 ALOGW("power manager service died !!!"); 538} 539 540void AudioFlinger::ThreadBase::setEffectSuspended( 541 const effect_uuid_t *type, bool suspend, int sessionId) 542{ 543 Mutex::Autolock _l(mLock); 544 setEffectSuspended_l(type, suspend, sessionId); 545} 546 547void AudioFlinger::ThreadBase::setEffectSuspended_l( 548 const effect_uuid_t *type, bool suspend, int sessionId) 549{ 550 sp<EffectChain> chain = getEffectChain_l(sessionId); 551 if (chain != 0) { 552 if (type != NULL) { 553 chain->setEffectSuspended_l(type, suspend); 554 } else { 555 chain->setEffectSuspendedAll_l(suspend); 556 } 557 } 558 559 updateSuspendedSessions_l(type, suspend, sessionId); 560} 561 562void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 563{ 564 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 565 if (index < 0) { 566 return; 567 } 568 569 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 570 mSuspendedSessions.valueAt(index); 571 572 for (size_t i = 0; i < sessionEffects.size(); i++) { 573 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 574 for (int j = 0; j < desc->mRefCount; j++) { 575 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 576 chain->setEffectSuspendedAll_l(true); 577 } else { 578 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 579 desc->mType.timeLow); 580 chain->setEffectSuspended_l(&desc->mType, true); 581 } 582 } 583 } 584} 585 586void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 587 bool suspend, 588 int sessionId) 589{ 590 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 591 592 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 593 594 if (suspend) { 595 if (index >= 0) { 596 sessionEffects = mSuspendedSessions.valueAt(index); 597 } else { 598 mSuspendedSessions.add(sessionId, sessionEffects); 599 } 600 } else { 601 if (index < 0) { 602 return; 603 } 604 sessionEffects = mSuspendedSessions.valueAt(index); 605 } 606 607 608 int key = EffectChain::kKeyForSuspendAll; 609 if (type != NULL) { 610 key = type->timeLow; 611 } 612 index = sessionEffects.indexOfKey(key); 613 614 sp<SuspendedSessionDesc> desc; 615 if (suspend) { 616 if (index >= 0) { 617 desc = sessionEffects.valueAt(index); 618 } else { 619 desc = new SuspendedSessionDesc(); 620 if (type != NULL) { 621 desc->mType = *type; 622 } 623 sessionEffects.add(key, desc); 624 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 625 } 626 desc->mRefCount++; 627 } else { 628 if (index < 0) { 629 return; 630 } 631 desc = sessionEffects.valueAt(index); 632 if (--desc->mRefCount == 0) { 633 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 634 sessionEffects.removeItemsAt(index); 635 if (sessionEffects.isEmpty()) { 636 ALOGV("updateSuspendedSessions_l() restore removing session %d", 637 sessionId); 638 mSuspendedSessions.removeItem(sessionId); 639 } 640 } 641 } 642 if (!sessionEffects.isEmpty()) { 643 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 644 } 645} 646 647void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 648 bool enabled, 649 int sessionId) 650{ 651 Mutex::Autolock _l(mLock); 652 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 653} 654 655void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 656 bool enabled, 657 int sessionId) 658{ 659 if (mType != RECORD) { 660 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 661 // another session. This gives the priority to well behaved effect control panels 662 // and applications not using global effects. 663 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 664 // global effects 665 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 666 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 667 } 668 } 669 670 sp<EffectChain> chain = getEffectChain_l(sessionId); 671 if (chain != 0) { 672 chain->checkSuspendOnEffectEnabled(effect, enabled); 673 } 674} 675 676// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 677sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 678 const sp<AudioFlinger::Client>& client, 679 const sp<IEffectClient>& effectClient, 680 int32_t priority, 681 int sessionId, 682 effect_descriptor_t *desc, 683 int *enabled, 684 status_t *status 685 ) 686{ 687 sp<EffectModule> effect; 688 sp<EffectHandle> handle; 689 status_t lStatus; 690 sp<EffectChain> chain; 691 bool chainCreated = false; 692 bool effectCreated = false; 693 bool effectRegistered = false; 694 695 lStatus = initCheck(); 696 if (lStatus != NO_ERROR) { 697 ALOGW("createEffect_l() Audio driver not initialized."); 698 goto Exit; 699 } 700 701 // Do not allow effects with session ID 0 on direct output or duplicating threads 702 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 703 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 704 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 705 desc->name, sessionId); 706 lStatus = BAD_VALUE; 707 goto Exit; 708 } 709 // Only Pre processor effects are allowed on input threads and only on input threads 710 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 711 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 712 desc->name, desc->flags, mType); 713 lStatus = BAD_VALUE; 714 goto Exit; 715 } 716 717 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 718 719 { // scope for mLock 720 Mutex::Autolock _l(mLock); 721 722 // check for existing effect chain with the requested audio session 723 chain = getEffectChain_l(sessionId); 724 if (chain == 0) { 725 // create a new chain for this session 726 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 727 chain = new EffectChain(this, sessionId); 728 addEffectChain_l(chain); 729 chain->setStrategy(getStrategyForSession_l(sessionId)); 730 chainCreated = true; 731 } else { 732 effect = chain->getEffectFromDesc_l(desc); 733 } 734 735 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 736 737 if (effect == 0) { 738 int id = mAudioFlinger->nextUniqueId(); 739 // Check CPU and memory usage 740 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 741 if (lStatus != NO_ERROR) { 742 goto Exit; 743 } 744 effectRegistered = true; 745 // create a new effect module if none present in the chain 746 effect = new EffectModule(this, chain, desc, id, sessionId); 747 lStatus = effect->status(); 748 if (lStatus != NO_ERROR) { 749 goto Exit; 750 } 751 lStatus = chain->addEffect_l(effect); 752 if (lStatus != NO_ERROR) { 753 goto Exit; 754 } 755 effectCreated = true; 756 757 effect->setDevice(mOutDevice); 758 effect->setDevice(mInDevice); 759 effect->setMode(mAudioFlinger->getMode()); 760 effect->setAudioSource(mAudioSource); 761 } 762 // create effect handle and connect it to effect module 763 handle = new EffectHandle(effect, client, effectClient, priority); 764 lStatus = effect->addHandle(handle.get()); 765 if (enabled != NULL) { 766 *enabled = (int)effect->isEnabled(); 767 } 768 } 769 770Exit: 771 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 772 Mutex::Autolock _l(mLock); 773 if (effectCreated) { 774 chain->removeEffect_l(effect); 775 } 776 if (effectRegistered) { 777 AudioSystem::unregisterEffect(effect->id()); 778 } 779 if (chainCreated) { 780 removeEffectChain_l(chain); 781 } 782 handle.clear(); 783 } 784 785 if (status != NULL) { 786 *status = lStatus; 787 } 788 return handle; 789} 790 791sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 792{ 793 Mutex::Autolock _l(mLock); 794 return getEffect_l(sessionId, effectId); 795} 796 797sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 798{ 799 sp<EffectChain> chain = getEffectChain_l(sessionId); 800 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 801} 802 803// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 804// PlaybackThread::mLock held 805status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 806{ 807 // check for existing effect chain with the requested audio session 808 int sessionId = effect->sessionId(); 809 sp<EffectChain> chain = getEffectChain_l(sessionId); 810 bool chainCreated = false; 811 812 if (chain == 0) { 813 // create a new chain for this session 814 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 815 chain = new EffectChain(this, sessionId); 816 addEffectChain_l(chain); 817 chain->setStrategy(getStrategyForSession_l(sessionId)); 818 chainCreated = true; 819 } 820 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 821 822 if (chain->getEffectFromId_l(effect->id()) != 0) { 823 ALOGW("addEffect_l() %p effect %s already present in chain %p", 824 this, effect->desc().name, chain.get()); 825 return BAD_VALUE; 826 } 827 828 status_t status = chain->addEffect_l(effect); 829 if (status != NO_ERROR) { 830 if (chainCreated) { 831 removeEffectChain_l(chain); 832 } 833 return status; 834 } 835 836 effect->setDevice(mOutDevice); 837 effect->setDevice(mInDevice); 838 effect->setMode(mAudioFlinger->getMode()); 839 effect->setAudioSource(mAudioSource); 840 return NO_ERROR; 841} 842 843void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 844 845 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 846 effect_descriptor_t desc = effect->desc(); 847 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 848 detachAuxEffect_l(effect->id()); 849 } 850 851 sp<EffectChain> chain = effect->chain().promote(); 852 if (chain != 0) { 853 // remove effect chain if removing last effect 854 if (chain->removeEffect_l(effect) == 0) { 855 removeEffectChain_l(chain); 856 } 857 } else { 858 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 859 } 860} 861 862void AudioFlinger::ThreadBase::lockEffectChains_l( 863 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 864{ 865 effectChains = mEffectChains; 866 for (size_t i = 0; i < mEffectChains.size(); i++) { 867 mEffectChains[i]->lock(); 868 } 869} 870 871void AudioFlinger::ThreadBase::unlockEffectChains( 872 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 873{ 874 for (size_t i = 0; i < effectChains.size(); i++) { 875 effectChains[i]->unlock(); 876 } 877} 878 879sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 880{ 881 Mutex::Autolock _l(mLock); 882 return getEffectChain_l(sessionId); 883} 884 885sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 886{ 887 size_t size = mEffectChains.size(); 888 for (size_t i = 0; i < size; i++) { 889 if (mEffectChains[i]->sessionId() == sessionId) { 890 return mEffectChains[i]; 891 } 892 } 893 return 0; 894} 895 896void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 897{ 898 Mutex::Autolock _l(mLock); 899 size_t size = mEffectChains.size(); 900 for (size_t i = 0; i < size; i++) { 901 mEffectChains[i]->setMode_l(mode); 902 } 903} 904 905void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 906 EffectHandle *handle, 907 bool unpinIfLast) { 908 909 Mutex::Autolock _l(mLock); 910 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 911 // delete the effect module if removing last handle on it 912 if (effect->removeHandle(handle) == 0) { 913 if (!effect->isPinned() || unpinIfLast) { 914 removeEffect_l(effect); 915 AudioSystem::unregisterEffect(effect->id()); 916 } 917 } 918} 919 920// ---------------------------------------------------------------------------- 921// Playback 922// ---------------------------------------------------------------------------- 923 924AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 925 AudioStreamOut* output, 926 audio_io_handle_t id, 927 audio_devices_t device, 928 type_t type) 929 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 930 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 931 // mStreamTypes[] initialized in constructor body 932 mOutput(output), 933 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 934 mMixerStatus(MIXER_IDLE), 935 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 936 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 937 mScreenState(AudioFlinger::mScreenState), 938 // index 0 is reserved for normal mixer's submix 939 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 940{ 941 snprintf(mName, kNameLength, "AudioOut_%X", id); 942 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 943 944 // Assumes constructor is called by AudioFlinger with it's mLock held, but 945 // it would be safer to explicitly pass initial masterVolume/masterMute as 946 // parameter. 947 // 948 // If the HAL we are using has support for master volume or master mute, 949 // then do not attenuate or mute during mixing (just leave the volume at 1.0 950 // and the mute set to false). 951 mMasterVolume = audioFlinger->masterVolume_l(); 952 mMasterMute = audioFlinger->masterMute_l(); 953 if (mOutput && mOutput->audioHwDev) { 954 if (mOutput->audioHwDev->canSetMasterVolume()) { 955 mMasterVolume = 1.0; 956 } 957 958 if (mOutput->audioHwDev->canSetMasterMute()) { 959 mMasterMute = false; 960 } 961 } 962 963 readOutputParameters(); 964 965 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 966 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 967 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 968 stream = (audio_stream_type_t) (stream + 1)) { 969 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 970 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 971 } 972 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 973 // because mAudioFlinger doesn't have one to copy from 974} 975 976AudioFlinger::PlaybackThread::~PlaybackThread() 977{ 978 mAudioFlinger->unregisterWriter(mNBLogWriter); 979 delete [] mMixBuffer; 980} 981 982void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 983{ 984 dumpInternals(fd, args); 985 dumpTracks(fd, args); 986 dumpEffectChains(fd, args); 987} 988 989void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 990{ 991 const size_t SIZE = 256; 992 char buffer[SIZE]; 993 String8 result; 994 995 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 996 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 997 const stream_type_t *st = &mStreamTypes[i]; 998 if (i > 0) { 999 result.appendFormat(", "); 1000 } 1001 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1002 if (st->mute) { 1003 result.append("M"); 1004 } 1005 } 1006 result.append("\n"); 1007 write(fd, result.string(), result.length()); 1008 result.clear(); 1009 1010 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1011 result.append(buffer); 1012 Track::appendDumpHeader(result); 1013 for (size_t i = 0; i < mTracks.size(); ++i) { 1014 sp<Track> track = mTracks[i]; 1015 if (track != 0) { 1016 track->dump(buffer, SIZE); 1017 result.append(buffer); 1018 } 1019 } 1020 1021 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1022 result.append(buffer); 1023 Track::appendDumpHeader(result); 1024 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1025 sp<Track> track = mActiveTracks[i].promote(); 1026 if (track != 0) { 1027 track->dump(buffer, SIZE); 1028 result.append(buffer); 1029 } 1030 } 1031 write(fd, result.string(), result.size()); 1032 1033 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1034 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1035 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1036 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1037} 1038 1039void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1040{ 1041 const size_t SIZE = 256; 1042 char buffer[SIZE]; 1043 String8 result; 1044 1045 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1048 ns2ms(systemTime() - mLastWriteTime)); 1049 result.append(buffer); 1050 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1051 result.append(buffer); 1052 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1053 result.append(buffer); 1054 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1055 result.append(buffer); 1056 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1057 result.append(buffer); 1058 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1059 result.append(buffer); 1060 write(fd, result.string(), result.size()); 1061 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1062 1063 dumpBase(fd, args); 1064} 1065 1066// Thread virtuals 1067status_t AudioFlinger::PlaybackThread::readyToRun() 1068{ 1069 status_t status = initCheck(); 1070 if (status == NO_ERROR) { 1071 ALOGI("AudioFlinger's thread %p ready to run", this); 1072 } else { 1073 ALOGE("No working audio driver found."); 1074 } 1075 return status; 1076} 1077 1078void AudioFlinger::PlaybackThread::onFirstRef() 1079{ 1080 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1081} 1082 1083// ThreadBase virtuals 1084void AudioFlinger::PlaybackThread::preExit() 1085{ 1086 ALOGV(" preExit()"); 1087 // FIXME this is using hard-coded strings but in the future, this functionality will be 1088 // converted to use audio HAL extensions required to support tunneling 1089 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1090} 1091 1092// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1093sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1094 const sp<AudioFlinger::Client>& client, 1095 audio_stream_type_t streamType, 1096 uint32_t sampleRate, 1097 audio_format_t format, 1098 audio_channel_mask_t channelMask, 1099 size_t frameCount, 1100 const sp<IMemory>& sharedBuffer, 1101 int sessionId, 1102 IAudioFlinger::track_flags_t *flags, 1103 pid_t tid, 1104 status_t *status) 1105{ 1106 sp<Track> track; 1107 status_t lStatus; 1108 1109 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1110 1111 // client expresses a preference for FAST, but we get the final say 1112 if (*flags & IAudioFlinger::TRACK_FAST) { 1113 if ( 1114 // not timed 1115 (!isTimed) && 1116 // either of these use cases: 1117 ( 1118 // use case 1: shared buffer with any frame count 1119 ( 1120 (sharedBuffer != 0) 1121 ) || 1122 // use case 2: callback handler and frame count is default or at least as large as HAL 1123 ( 1124 (tid != -1) && 1125 ((frameCount == 0) || 1126 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1127 ) 1128 ) && 1129 // PCM data 1130 audio_is_linear_pcm(format) && 1131 // mono or stereo 1132 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1133 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1134#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1135 // hardware sample rate 1136 (sampleRate == mSampleRate) && 1137#endif 1138 // normal mixer has an associated fast mixer 1139 hasFastMixer() && 1140 // there are sufficient fast track slots available 1141 (mFastTrackAvailMask != 0) 1142 // FIXME test that MixerThread for this fast track has a capable output HAL 1143 // FIXME add a permission test also? 1144 ) { 1145 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1146 if (frameCount == 0) { 1147 frameCount = mFrameCount * kFastTrackMultiplier; 1148 } 1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1150 frameCount, mFrameCount); 1151 } else { 1152 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1153 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1154 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1155 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1156 audio_is_linear_pcm(format), 1157 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1158 *flags &= ~IAudioFlinger::TRACK_FAST; 1159 // For compatibility with AudioTrack calculation, buffer depth is forced 1160 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1161 // This is probably too conservative, but legacy application code may depend on it. 1162 // If you change this calculation, also review the start threshold which is related. 1163 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1164 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1165 if (minBufCount < 2) { 1166 minBufCount = 2; 1167 } 1168 size_t minFrameCount = mNormalFrameCount * minBufCount; 1169 if (frameCount < minFrameCount) { 1170 frameCount = minFrameCount; 1171 } 1172 } 1173 } 1174 1175 if (mType == DIRECT) { 1176 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1177 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1178 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1179 "for output %p with format %d", 1180 sampleRate, format, channelMask, mOutput, mFormat); 1181 lStatus = BAD_VALUE; 1182 goto Exit; 1183 } 1184 } 1185 } else { 1186 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1187 if (sampleRate > mSampleRate*2) { 1188 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1189 lStatus = BAD_VALUE; 1190 goto Exit; 1191 } 1192 } 1193 1194 lStatus = initCheck(); 1195 if (lStatus != NO_ERROR) { 1196 ALOGE("Audio driver not initialized."); 1197 goto Exit; 1198 } 1199 1200 { // scope for mLock 1201 Mutex::Autolock _l(mLock); 1202 1203 // all tracks in same audio session must share the same routing strategy otherwise 1204 // conflicts will happen when tracks are moved from one output to another by audio policy 1205 // manager 1206 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1207 for (size_t i = 0; i < mTracks.size(); ++i) { 1208 sp<Track> t = mTracks[i]; 1209 if (t != 0 && !t->isOutputTrack()) { 1210 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1211 if (sessionId == t->sessionId() && strategy != actual) { 1212 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1213 strategy, actual); 1214 lStatus = BAD_VALUE; 1215 goto Exit; 1216 } 1217 } 1218 } 1219 1220 if (!isTimed) { 1221 track = new Track(this, client, streamType, sampleRate, format, 1222 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1223 } else { 1224 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1225 channelMask, frameCount, sharedBuffer, sessionId); 1226 } 1227 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1228 lStatus = NO_MEMORY; 1229 goto Exit; 1230 } 1231 mTracks.add(track); 1232 1233 sp<EffectChain> chain = getEffectChain_l(sessionId); 1234 if (chain != 0) { 1235 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1236 track->setMainBuffer(chain->inBuffer()); 1237 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1238 chain->incTrackCnt(); 1239 } 1240 1241 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1242 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1243 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1244 // so ask activity manager to do this on our behalf 1245 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1246 } 1247 } 1248 1249 lStatus = NO_ERROR; 1250 1251Exit: 1252 if (status) { 1253 *status = lStatus; 1254 } 1255 return track; 1256} 1257 1258uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1259{ 1260 return latency; 1261} 1262 1263uint32_t AudioFlinger::PlaybackThread::latency() const 1264{ 1265 Mutex::Autolock _l(mLock); 1266 return latency_l(); 1267} 1268uint32_t AudioFlinger::PlaybackThread::latency_l() const 1269{ 1270 if (initCheck() == NO_ERROR) { 1271 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1272 } else { 1273 return 0; 1274 } 1275} 1276 1277void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1278{ 1279 Mutex::Autolock _l(mLock); 1280 // Don't apply master volume in SW if our HAL can do it for us. 1281 if (mOutput && mOutput->audioHwDev && 1282 mOutput->audioHwDev->canSetMasterVolume()) { 1283 mMasterVolume = 1.0; 1284 } else { 1285 mMasterVolume = value; 1286 } 1287} 1288 1289void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1290{ 1291 Mutex::Autolock _l(mLock); 1292 // Don't apply master mute in SW if our HAL can do it for us. 1293 if (mOutput && mOutput->audioHwDev && 1294 mOutput->audioHwDev->canSetMasterMute()) { 1295 mMasterMute = false; 1296 } else { 1297 mMasterMute = muted; 1298 } 1299} 1300 1301void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1302{ 1303 Mutex::Autolock _l(mLock); 1304 mStreamTypes[stream].volume = value; 1305} 1306 1307void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1308{ 1309 Mutex::Autolock _l(mLock); 1310 mStreamTypes[stream].mute = muted; 1311} 1312 1313float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1314{ 1315 Mutex::Autolock _l(mLock); 1316 return mStreamTypes[stream].volume; 1317} 1318 1319// addTrack_l() must be called with ThreadBase::mLock held 1320status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1321{ 1322 status_t status = ALREADY_EXISTS; 1323 1324 // set retry count for buffer fill 1325 track->mRetryCount = kMaxTrackStartupRetries; 1326 if (mActiveTracks.indexOf(track) < 0) { 1327 // the track is newly added, make sure it fills up all its 1328 // buffers before playing. This is to ensure the client will 1329 // effectively get the latency it requested. 1330 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1331 track->mResetDone = false; 1332 track->mPresentationCompleteFrames = 0; 1333 mActiveTracks.add(track); 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 status = NO_ERROR; 1342 } 1343 1344 ALOGV("mWaitWorkCV.broadcast"); 1345 mWaitWorkCV.broadcast(); 1346 1347 return status; 1348} 1349 1350// destroyTrack_l() must be called with ThreadBase::mLock held 1351void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1352{ 1353 track->mState = TrackBase::TERMINATED; 1354 // active tracks are removed by threadLoop() 1355 if (mActiveTracks.indexOf(track) < 0) { 1356 removeTrack_l(track); 1357 } 1358} 1359 1360void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1361{ 1362 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1363 mTracks.remove(track); 1364 deleteTrackName_l(track->name()); 1365 // redundant as track is about to be destroyed, for dumpsys only 1366 track->mName = -1; 1367 if (track->isFastTrack()) { 1368 int index = track->mFastIndex; 1369 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1370 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1371 mFastTrackAvailMask |= 1 << index; 1372 // redundant as track is about to be destroyed, for dumpsys only 1373 track->mFastIndex = -1; 1374 } 1375 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1376 if (chain != 0) { 1377 chain->decTrackCnt(); 1378 } 1379} 1380 1381String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1382{ 1383 String8 out_s8 = String8(""); 1384 char *s; 1385 1386 Mutex::Autolock _l(mLock); 1387 if (initCheck() != NO_ERROR) { 1388 return out_s8; 1389 } 1390 1391 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1392 out_s8 = String8(s); 1393 free(s); 1394 return out_s8; 1395} 1396 1397// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1398void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1399 AudioSystem::OutputDescriptor desc; 1400 void *param2 = NULL; 1401 1402 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1403 param); 1404 1405 switch (event) { 1406 case AudioSystem::OUTPUT_OPENED: 1407 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1408 desc.channels = mChannelMask; 1409 desc.samplingRate = mSampleRate; 1410 desc.format = mFormat; 1411 desc.frameCount = mNormalFrameCount; // FIXME see 1412 // AudioFlinger::frameCount(audio_io_handle_t) 1413 desc.latency = latency(); 1414 param2 = &desc; 1415 break; 1416 1417 case AudioSystem::STREAM_CONFIG_CHANGED: 1418 param2 = ¶m; 1419 case AudioSystem::OUTPUT_CLOSED: 1420 default: 1421 break; 1422 } 1423 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1424} 1425 1426void AudioFlinger::PlaybackThread::readOutputParameters() 1427{ 1428 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1429 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1430 mChannelCount = (uint16_t)popcount(mChannelMask); 1431 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1432 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1433 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1434 if (mFrameCount & 15) { 1435 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1436 mFrameCount); 1437 } 1438 1439 // Calculate size of normal mix buffer relative to the HAL output buffer size 1440 double multiplier = 1.0; 1441 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1442 kUseFastMixer == FastMixer_Dynamic)) { 1443 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1444 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1445 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1446 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1447 maxNormalFrameCount = maxNormalFrameCount & ~15; 1448 if (maxNormalFrameCount < minNormalFrameCount) { 1449 maxNormalFrameCount = minNormalFrameCount; 1450 } 1451 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1452 if (multiplier <= 1.0) { 1453 multiplier = 1.0; 1454 } else if (multiplier <= 2.0) { 1455 if (2 * mFrameCount <= maxNormalFrameCount) { 1456 multiplier = 2.0; 1457 } else { 1458 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1459 } 1460 } else { 1461 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1462 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1463 // track, but we sometimes have to do this to satisfy the maximum frame count 1464 // constraint) 1465 // FIXME this rounding up should not be done if no HAL SRC 1466 uint32_t truncMult = (uint32_t) multiplier; 1467 if ((truncMult & 1)) { 1468 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1469 ++truncMult; 1470 } 1471 } 1472 multiplier = (double) truncMult; 1473 } 1474 } 1475 mNormalFrameCount = multiplier * mFrameCount; 1476 // round up to nearest 16 frames to satisfy AudioMixer 1477 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1478 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1479 mNormalFrameCount); 1480 1481 delete[] mMixBuffer; 1482 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1483 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1484 1485 // force reconfiguration of effect chains and engines to take new buffer size and audio 1486 // parameters into account 1487 // Note that mLock is not held when readOutputParameters() is called from the constructor 1488 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1489 // matter. 1490 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1491 Vector< sp<EffectChain> > effectChains = mEffectChains; 1492 for (size_t i = 0; i < effectChains.size(); i ++) { 1493 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1494 } 1495} 1496 1497 1498status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1499{ 1500 if (halFrames == NULL || dspFrames == NULL) { 1501 return BAD_VALUE; 1502 } 1503 Mutex::Autolock _l(mLock); 1504 if (initCheck() != NO_ERROR) { 1505 return INVALID_OPERATION; 1506 } 1507 size_t framesWritten = mBytesWritten / mFrameSize; 1508 *halFrames = framesWritten; 1509 1510 if (isSuspended()) { 1511 // return an estimation of rendered frames when the output is suspended 1512 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1513 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1514 return NO_ERROR; 1515 } else { 1516 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1517 } 1518} 1519 1520uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1521{ 1522 Mutex::Autolock _l(mLock); 1523 uint32_t result = 0; 1524 if (getEffectChain_l(sessionId) != 0) { 1525 result = EFFECT_SESSION; 1526 } 1527 1528 for (size_t i = 0; i < mTracks.size(); ++i) { 1529 sp<Track> track = mTracks[i]; 1530 if (sessionId == track->sessionId() && !track->isInvalid()) { 1531 result |= TRACK_SESSION; 1532 break; 1533 } 1534 } 1535 1536 return result; 1537} 1538 1539uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1540{ 1541 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1542 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1543 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1544 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1545 } 1546 for (size_t i = 0; i < mTracks.size(); i++) { 1547 sp<Track> track = mTracks[i]; 1548 if (sessionId == track->sessionId() && !track->isInvalid()) { 1549 return AudioSystem::getStrategyForStream(track->streamType()); 1550 } 1551 } 1552 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1553} 1554 1555 1556AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1557{ 1558 Mutex::Autolock _l(mLock); 1559 return mOutput; 1560} 1561 1562AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1563{ 1564 Mutex::Autolock _l(mLock); 1565 AudioStreamOut *output = mOutput; 1566 mOutput = NULL; 1567 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1568 // must push a NULL and wait for ack 1569 mOutputSink.clear(); 1570 mPipeSink.clear(); 1571 mNormalSink.clear(); 1572 return output; 1573} 1574 1575// this method must always be called either with ThreadBase mLock held or inside the thread loop 1576audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1577{ 1578 if (mOutput == NULL) { 1579 return NULL; 1580 } 1581 return &mOutput->stream->common; 1582} 1583 1584uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1585{ 1586 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1587} 1588 1589status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1590{ 1591 if (!isValidSyncEvent(event)) { 1592 return BAD_VALUE; 1593 } 1594 1595 Mutex::Autolock _l(mLock); 1596 1597 for (size_t i = 0; i < mTracks.size(); ++i) { 1598 sp<Track> track = mTracks[i]; 1599 if (event->triggerSession() == track->sessionId()) { 1600 (void) track->setSyncEvent(event); 1601 return NO_ERROR; 1602 } 1603 } 1604 1605 return NAME_NOT_FOUND; 1606} 1607 1608bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1609{ 1610 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1611} 1612 1613void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1614 const Vector< sp<Track> >& tracksToRemove) 1615{ 1616 size_t count = tracksToRemove.size(); 1617 if (CC_UNLIKELY(count)) { 1618 for (size_t i = 0 ; i < count ; i++) { 1619 const sp<Track>& track = tracksToRemove.itemAt(i); 1620 if ((track->sharedBuffer() != 0) && 1621 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1622 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1623 } 1624 } 1625 } 1626 1627} 1628 1629void AudioFlinger::PlaybackThread::checkSilentMode_l() 1630{ 1631 if (!mMasterMute) { 1632 char value[PROPERTY_VALUE_MAX]; 1633 if (property_get("ro.audio.silent", value, "0") > 0) { 1634 char *endptr; 1635 unsigned long ul = strtoul(value, &endptr, 0); 1636 if (*endptr == '\0' && ul != 0) { 1637 ALOGD("Silence is golden"); 1638 // The setprop command will not allow a property to be changed after 1639 // the first time it is set, so we don't have to worry about un-muting. 1640 setMasterMute_l(true); 1641 } 1642 } 1643 } 1644} 1645 1646// shared by MIXER and DIRECT, overridden by DUPLICATING 1647void AudioFlinger::PlaybackThread::threadLoop_write() 1648{ 1649 // FIXME rewrite to reduce number of system calls 1650 mLastWriteTime = systemTime(); 1651 mInWrite = true; 1652 int bytesWritten; 1653 1654 // If an NBAIO sink is present, use it to write the normal mixer's submix 1655 if (mNormalSink != 0) { 1656#define mBitShift 2 // FIXME 1657 size_t count = mixBufferSize >> mBitShift; 1658 ATRACE_BEGIN("write"); 1659 // update the setpoint when AudioFlinger::mScreenState changes 1660 uint32_t screenState = AudioFlinger::mScreenState; 1661 if (screenState != mScreenState) { 1662 mScreenState = screenState; 1663 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1664 if (pipe != NULL) { 1665 pipe->setAvgFrames((mScreenState & 1) ? 1666 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1667 } 1668 } 1669 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1670 ATRACE_END(); 1671 if (framesWritten > 0) { 1672 bytesWritten = framesWritten << mBitShift; 1673 } else { 1674 bytesWritten = framesWritten; 1675 } 1676 // otherwise use the HAL / AudioStreamOut directly 1677 } else { 1678 // Direct output thread. 1679 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1680 } 1681 1682 if (bytesWritten > 0) { 1683 mBytesWritten += mixBufferSize; 1684 } 1685 mNumWrites++; 1686 mInWrite = false; 1687} 1688 1689/* 1690The derived values that are cached: 1691 - mixBufferSize from frame count * frame size 1692 - activeSleepTime from activeSleepTimeUs() 1693 - idleSleepTime from idleSleepTimeUs() 1694 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1695 - maxPeriod from frame count and sample rate (MIXER only) 1696 1697The parameters that affect these derived values are: 1698 - frame count 1699 - frame size 1700 - sample rate 1701 - device type: A2DP or not 1702 - device latency 1703 - format: PCM or not 1704 - active sleep time 1705 - idle sleep time 1706*/ 1707 1708void AudioFlinger::PlaybackThread::cacheParameters_l() 1709{ 1710 mixBufferSize = mNormalFrameCount * mFrameSize; 1711 activeSleepTime = activeSleepTimeUs(); 1712 idleSleepTime = idleSleepTimeUs(); 1713} 1714 1715void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1716{ 1717 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1718 this, streamType, mTracks.size()); 1719 Mutex::Autolock _l(mLock); 1720 1721 size_t size = mTracks.size(); 1722 for (size_t i = 0; i < size; i++) { 1723 sp<Track> t = mTracks[i]; 1724 if (t->streamType() == streamType) { 1725 t->invalidate(); 1726 } 1727 } 1728} 1729 1730status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1731{ 1732 int session = chain->sessionId(); 1733 int16_t *buffer = mMixBuffer; 1734 bool ownsBuffer = false; 1735 1736 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1737 if (session > 0) { 1738 // Only one effect chain can be present in direct output thread and it uses 1739 // the mix buffer as input 1740 if (mType != DIRECT) { 1741 size_t numSamples = mNormalFrameCount * mChannelCount; 1742 buffer = new int16_t[numSamples]; 1743 memset(buffer, 0, numSamples * sizeof(int16_t)); 1744 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1745 ownsBuffer = true; 1746 } 1747 1748 // Attach all tracks with same session ID to this chain. 1749 for (size_t i = 0; i < mTracks.size(); ++i) { 1750 sp<Track> track = mTracks[i]; 1751 if (session == track->sessionId()) { 1752 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1753 buffer); 1754 track->setMainBuffer(buffer); 1755 chain->incTrackCnt(); 1756 } 1757 } 1758 1759 // indicate all active tracks in the chain 1760 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1761 sp<Track> track = mActiveTracks[i].promote(); 1762 if (track == 0) { 1763 continue; 1764 } 1765 if (session == track->sessionId()) { 1766 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1767 chain->incActiveTrackCnt(); 1768 } 1769 } 1770 } 1771 1772 chain->setInBuffer(buffer, ownsBuffer); 1773 chain->setOutBuffer(mMixBuffer); 1774 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1775 // chains list in order to be processed last as it contains output stage effects 1776 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1777 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1778 // after track specific effects and before output stage 1779 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1780 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1781 // Effect chain for other sessions are inserted at beginning of effect 1782 // chains list to be processed before output mix effects. Relative order between other 1783 // sessions is not important 1784 size_t size = mEffectChains.size(); 1785 size_t i = 0; 1786 for (i = 0; i < size; i++) { 1787 if (mEffectChains[i]->sessionId() < session) { 1788 break; 1789 } 1790 } 1791 mEffectChains.insertAt(chain, i); 1792 checkSuspendOnAddEffectChain_l(chain); 1793 1794 return NO_ERROR; 1795} 1796 1797size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1798{ 1799 int session = chain->sessionId(); 1800 1801 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1802 1803 for (size_t i = 0; i < mEffectChains.size(); i++) { 1804 if (chain == mEffectChains[i]) { 1805 mEffectChains.removeAt(i); 1806 // detach all active tracks from the chain 1807 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1808 sp<Track> track = mActiveTracks[i].promote(); 1809 if (track == 0) { 1810 continue; 1811 } 1812 if (session == track->sessionId()) { 1813 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1814 chain.get(), session); 1815 chain->decActiveTrackCnt(); 1816 } 1817 } 1818 1819 // detach all tracks with same session ID from this chain 1820 for (size_t i = 0; i < mTracks.size(); ++i) { 1821 sp<Track> track = mTracks[i]; 1822 if (session == track->sessionId()) { 1823 track->setMainBuffer(mMixBuffer); 1824 chain->decTrackCnt(); 1825 } 1826 } 1827 break; 1828 } 1829 } 1830 return mEffectChains.size(); 1831} 1832 1833status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1834 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1835{ 1836 Mutex::Autolock _l(mLock); 1837 return attachAuxEffect_l(track, EffectId); 1838} 1839 1840status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1841 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1842{ 1843 status_t status = NO_ERROR; 1844 1845 if (EffectId == 0) { 1846 track->setAuxBuffer(0, NULL); 1847 } else { 1848 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1849 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1850 if (effect != 0) { 1851 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1852 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1853 } else { 1854 status = INVALID_OPERATION; 1855 } 1856 } else { 1857 status = BAD_VALUE; 1858 } 1859 } 1860 return status; 1861} 1862 1863void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1864{ 1865 for (size_t i = 0; i < mTracks.size(); ++i) { 1866 sp<Track> track = mTracks[i]; 1867 if (track->auxEffectId() == effectId) { 1868 attachAuxEffect_l(track, 0); 1869 } 1870 } 1871} 1872 1873bool AudioFlinger::PlaybackThread::threadLoop() 1874{ 1875 Vector< sp<Track> > tracksToRemove; 1876 1877 standbyTime = systemTime(); 1878 1879 // MIXER 1880 nsecs_t lastWarning = 0; 1881 1882 // DUPLICATING 1883 // FIXME could this be made local to while loop? 1884 writeFrames = 0; 1885 1886 cacheParameters_l(); 1887 sleepTime = idleSleepTime; 1888 1889 if (mType == MIXER) { 1890 sleepTimeShift = 0; 1891 } 1892 1893 CpuStats cpuStats; 1894 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1895 1896 acquireWakeLock(); 1897 1898 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1899 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1900 // and then that string will be logged at the next convenient opportunity. 1901 const char *logString = NULL; 1902 1903 while (!exitPending()) 1904 { 1905 cpuStats.sample(myName); 1906 1907 Vector< sp<EffectChain> > effectChains; 1908 1909 processConfigEvents(); 1910 1911 { // scope for mLock 1912 1913 Mutex::Autolock _l(mLock); 1914 1915 if (logString != NULL) { 1916 mNBLogWriter->logTimestamp(); 1917 mNBLogWriter->log(logString); 1918 logString = NULL; 1919 } 1920 1921 if (checkForNewParameters_l()) { 1922 cacheParameters_l(); 1923 } 1924 1925 saveOutputTracks(); 1926 1927 // put audio hardware into standby after short delay 1928 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1929 isSuspended())) { 1930 if (!mStandby) { 1931 1932 threadLoop_standby(); 1933 1934 mStandby = true; 1935 } 1936 1937 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1938 // we're about to wait, flush the binder command buffer 1939 IPCThreadState::self()->flushCommands(); 1940 1941 clearOutputTracks(); 1942 1943 if (exitPending()) { 1944 break; 1945 } 1946 1947 releaseWakeLock_l(); 1948 // wait until we have something to do... 1949 ALOGV("%s going to sleep", myName.string()); 1950 mWaitWorkCV.wait(mLock); 1951 ALOGV("%s waking up", myName.string()); 1952 acquireWakeLock_l(); 1953 1954 mMixerStatus = MIXER_IDLE; 1955 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1956 mBytesWritten = 0; 1957 1958 checkSilentMode_l(); 1959 1960 standbyTime = systemTime() + standbyDelay; 1961 sleepTime = idleSleepTime; 1962 if (mType == MIXER) { 1963 sleepTimeShift = 0; 1964 } 1965 1966 continue; 1967 } 1968 } 1969 1970 // mMixerStatusIgnoringFastTracks is also updated internally 1971 mMixerStatus = prepareTracks_l(&tracksToRemove); 1972 1973 // prevent any changes in effect chain list and in each effect chain 1974 // during mixing and effect process as the audio buffers could be deleted 1975 // or modified if an effect is created or deleted 1976 lockEffectChains_l(effectChains); 1977 } 1978 1979 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1980 threadLoop_mix(); 1981 } else { 1982 threadLoop_sleepTime(); 1983 } 1984 1985 if (isSuspended()) { 1986 sleepTime = suspendSleepTimeUs(); 1987 mBytesWritten += mixBufferSize; 1988 } 1989 1990 // only process effects if we're going to write 1991 if (sleepTime == 0) { 1992 for (size_t i = 0; i < effectChains.size(); i ++) { 1993 effectChains[i]->process_l(); 1994 } 1995 } 1996 1997 // enable changes in effect chain 1998 unlockEffectChains(effectChains); 1999 2000 // sleepTime == 0 means we must write to audio hardware 2001 if (sleepTime == 0) { 2002 2003 threadLoop_write(); 2004 2005if (mType == MIXER) { 2006 // write blocked detection 2007 nsecs_t now = systemTime(); 2008 nsecs_t delta = now - mLastWriteTime; 2009 if (!mStandby && delta > maxPeriod) { 2010 mNumDelayedWrites++; 2011 if ((now - lastWarning) > kWarningThrottleNs) { 2012 ATRACE_NAME("underrun"); 2013 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2014 ns2ms(delta), mNumDelayedWrites, this); 2015 lastWarning = now; 2016 } 2017 } 2018} 2019 2020 mStandby = false; 2021 } else { 2022 usleep(sleepTime); 2023 } 2024 2025 // Finally let go of removed track(s), without the lock held 2026 // since we can't guarantee the destructors won't acquire that 2027 // same lock. This will also mutate and push a new fast mixer state. 2028 threadLoop_removeTracks(tracksToRemove); 2029 tracksToRemove.clear(); 2030 2031 // FIXME I don't understand the need for this here; 2032 // it was in the original code but maybe the 2033 // assignment in saveOutputTracks() makes this unnecessary? 2034 clearOutputTracks(); 2035 2036 // Effect chains will be actually deleted here if they were removed from 2037 // mEffectChains list during mixing or effects processing 2038 effectChains.clear(); 2039 2040 // FIXME Note that the above .clear() is no longer necessary since effectChains 2041 // is now local to this block, but will keep it for now (at least until merge done). 2042 } 2043 2044 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2045 if (mType == MIXER || mType == DIRECT) { 2046 // put output stream into standby mode 2047 if (!mStandby) { 2048 mOutput->stream->common.standby(&mOutput->stream->common); 2049 } 2050 } 2051 2052 releaseWakeLock(); 2053 2054 ALOGV("Thread %p type %d exiting", this, mType); 2055 return false; 2056} 2057 2058 2059// ---------------------------------------------------------------------------- 2060 2061AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2062 audio_io_handle_t id, audio_devices_t device, type_t type) 2063 : PlaybackThread(audioFlinger, output, id, device, type), 2064 // mAudioMixer below 2065 // mFastMixer below 2066 mFastMixerFutex(0) 2067 // mOutputSink below 2068 // mPipeSink below 2069 // mNormalSink below 2070{ 2071 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2072 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2073 "mFrameCount=%d, mNormalFrameCount=%d", 2074 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2075 mNormalFrameCount); 2076 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2077 2078 // FIXME - Current mixer implementation only supports stereo output 2079 if (mChannelCount != FCC_2) { 2080 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2081 } 2082 2083 // create an NBAIO sink for the HAL output stream, and negotiate 2084 mOutputSink = new AudioStreamOutSink(output->stream); 2085 size_t numCounterOffers = 0; 2086 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2087 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2088 ALOG_ASSERT(index == 0); 2089 2090 // initialize fast mixer depending on configuration 2091 bool initFastMixer; 2092 switch (kUseFastMixer) { 2093 case FastMixer_Never: 2094 initFastMixer = false; 2095 break; 2096 case FastMixer_Always: 2097 initFastMixer = true; 2098 break; 2099 case FastMixer_Static: 2100 case FastMixer_Dynamic: 2101 initFastMixer = mFrameCount < mNormalFrameCount; 2102 break; 2103 } 2104 if (initFastMixer) { 2105 2106 // create a MonoPipe to connect our submix to FastMixer 2107 NBAIO_Format format = mOutputSink->format(); 2108 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2109 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2110 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2111 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2112 const NBAIO_Format offers[1] = {format}; 2113 size_t numCounterOffers = 0; 2114 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2115 ALOG_ASSERT(index == 0); 2116 monoPipe->setAvgFrames((mScreenState & 1) ? 2117 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2118 mPipeSink = monoPipe; 2119 2120#ifdef TEE_SINK 2121 if (mTeeSinkOutputEnabled) { 2122 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2123 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2124 numCounterOffers = 0; 2125 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2126 ALOG_ASSERT(index == 0); 2127 mTeeSink = teeSink; 2128 PipeReader *teeSource = new PipeReader(*teeSink); 2129 numCounterOffers = 0; 2130 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2131 ALOG_ASSERT(index == 0); 2132 mTeeSource = teeSource; 2133 } 2134#endif 2135 2136 // create fast mixer and configure it initially with just one fast track for our submix 2137 mFastMixer = new FastMixer(); 2138 FastMixerStateQueue *sq = mFastMixer->sq(); 2139#ifdef STATE_QUEUE_DUMP 2140 sq->setObserverDump(&mStateQueueObserverDump); 2141 sq->setMutatorDump(&mStateQueueMutatorDump); 2142#endif 2143 FastMixerState *state = sq->begin(); 2144 FastTrack *fastTrack = &state->mFastTracks[0]; 2145 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2146 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2147 fastTrack->mVolumeProvider = NULL; 2148 fastTrack->mGeneration++; 2149 state->mFastTracksGen++; 2150 state->mTrackMask = 1; 2151 // fast mixer will use the HAL output sink 2152 state->mOutputSink = mOutputSink.get(); 2153 state->mOutputSinkGen++; 2154 state->mFrameCount = mFrameCount; 2155 state->mCommand = FastMixerState::COLD_IDLE; 2156 // already done in constructor initialization list 2157 //mFastMixerFutex = 0; 2158 state->mColdFutexAddr = &mFastMixerFutex; 2159 state->mColdGen++; 2160 state->mDumpState = &mFastMixerDumpState; 2161#ifdef TEE_SINK 2162 state->mTeeSink = mTeeSink.get(); 2163#endif 2164 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2165 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2166 sq->end(); 2167 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2168 2169 // start the fast mixer 2170 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2171 pid_t tid = mFastMixer->getTid(); 2172 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2173 if (err != 0) { 2174 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2175 kPriorityFastMixer, getpid_cached, tid, err); 2176 } 2177 2178#ifdef AUDIO_WATCHDOG 2179 // create and start the watchdog 2180 mAudioWatchdog = new AudioWatchdog(); 2181 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2182 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2183 tid = mAudioWatchdog->getTid(); 2184 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2185 if (err != 0) { 2186 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2187 kPriorityFastMixer, getpid_cached, tid, err); 2188 } 2189#endif 2190 2191 } else { 2192 mFastMixer = NULL; 2193 } 2194 2195 switch (kUseFastMixer) { 2196 case FastMixer_Never: 2197 case FastMixer_Dynamic: 2198 mNormalSink = mOutputSink; 2199 break; 2200 case FastMixer_Always: 2201 mNormalSink = mPipeSink; 2202 break; 2203 case FastMixer_Static: 2204 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2205 break; 2206 } 2207} 2208 2209AudioFlinger::MixerThread::~MixerThread() 2210{ 2211 if (mFastMixer != NULL) { 2212 FastMixerStateQueue *sq = mFastMixer->sq(); 2213 FastMixerState *state = sq->begin(); 2214 if (state->mCommand == FastMixerState::COLD_IDLE) { 2215 int32_t old = android_atomic_inc(&mFastMixerFutex); 2216 if (old == -1) { 2217 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2218 } 2219 } 2220 state->mCommand = FastMixerState::EXIT; 2221 sq->end(); 2222 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2223 mFastMixer->join(); 2224 // Though the fast mixer thread has exited, it's state queue is still valid. 2225 // We'll use that extract the final state which contains one remaining fast track 2226 // corresponding to our sub-mix. 2227 state = sq->begin(); 2228 ALOG_ASSERT(state->mTrackMask == 1); 2229 FastTrack *fastTrack = &state->mFastTracks[0]; 2230 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2231 delete fastTrack->mBufferProvider; 2232 sq->end(false /*didModify*/); 2233 delete mFastMixer; 2234#ifdef AUDIO_WATCHDOG 2235 if (mAudioWatchdog != 0) { 2236 mAudioWatchdog->requestExit(); 2237 mAudioWatchdog->requestExitAndWait(); 2238 mAudioWatchdog.clear(); 2239 } 2240#endif 2241 } 2242 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2243 delete mAudioMixer; 2244} 2245 2246 2247uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2248{ 2249 if (mFastMixer != NULL) { 2250 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2251 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2252 } 2253 return latency; 2254} 2255 2256 2257void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2258{ 2259 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2260} 2261 2262void AudioFlinger::MixerThread::threadLoop_write() 2263{ 2264 // FIXME we should only do one push per cycle; confirm this is true 2265 // Start the fast mixer if it's not already running 2266 if (mFastMixer != NULL) { 2267 FastMixerStateQueue *sq = mFastMixer->sq(); 2268 FastMixerState *state = sq->begin(); 2269 if (state->mCommand != FastMixerState::MIX_WRITE && 2270 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2271 if (state->mCommand == FastMixerState::COLD_IDLE) { 2272 int32_t old = android_atomic_inc(&mFastMixerFutex); 2273 if (old == -1) { 2274 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2275 } 2276#ifdef AUDIO_WATCHDOG 2277 if (mAudioWatchdog != 0) { 2278 mAudioWatchdog->resume(); 2279 } 2280#endif 2281 } 2282 state->mCommand = FastMixerState::MIX_WRITE; 2283 sq->end(); 2284 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2285 if (kUseFastMixer == FastMixer_Dynamic) { 2286 mNormalSink = mPipeSink; 2287 } 2288 } else { 2289 sq->end(false /*didModify*/); 2290 } 2291 } 2292 PlaybackThread::threadLoop_write(); 2293} 2294 2295void AudioFlinger::MixerThread::threadLoop_standby() 2296{ 2297 // Idle the fast mixer if it's currently running 2298 if (mFastMixer != NULL) { 2299 FastMixerStateQueue *sq = mFastMixer->sq(); 2300 FastMixerState *state = sq->begin(); 2301 if (!(state->mCommand & FastMixerState::IDLE)) { 2302 state->mCommand = FastMixerState::COLD_IDLE; 2303 state->mColdFutexAddr = &mFastMixerFutex; 2304 state->mColdGen++; 2305 mFastMixerFutex = 0; 2306 sq->end(); 2307 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2308 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2309 if (kUseFastMixer == FastMixer_Dynamic) { 2310 mNormalSink = mOutputSink; 2311 } 2312#ifdef AUDIO_WATCHDOG 2313 if (mAudioWatchdog != 0) { 2314 mAudioWatchdog->pause(); 2315 } 2316#endif 2317 } else { 2318 sq->end(false /*didModify*/); 2319 } 2320 } 2321 PlaybackThread::threadLoop_standby(); 2322} 2323 2324// shared by MIXER and DIRECT, overridden by DUPLICATING 2325void AudioFlinger::PlaybackThread::threadLoop_standby() 2326{ 2327 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2328 mOutput->stream->common.standby(&mOutput->stream->common); 2329} 2330 2331void AudioFlinger::MixerThread::threadLoop_mix() 2332{ 2333 // obtain the presentation timestamp of the next output buffer 2334 int64_t pts; 2335 status_t status = INVALID_OPERATION; 2336 2337 if (mNormalSink != 0) { 2338 status = mNormalSink->getNextWriteTimestamp(&pts); 2339 } else { 2340 status = mOutputSink->getNextWriteTimestamp(&pts); 2341 } 2342 2343 if (status != NO_ERROR) { 2344 pts = AudioBufferProvider::kInvalidPTS; 2345 } 2346 2347 // mix buffers... 2348 mAudioMixer->process(pts); 2349 // increase sleep time progressively when application underrun condition clears. 2350 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2351 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2352 // such that we would underrun the audio HAL. 2353 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2354 sleepTimeShift--; 2355 } 2356 sleepTime = 0; 2357 standbyTime = systemTime() + standbyDelay; 2358 //TODO: delay standby when effects have a tail 2359} 2360 2361void AudioFlinger::MixerThread::threadLoop_sleepTime() 2362{ 2363 // If no tracks are ready, sleep once for the duration of an output 2364 // buffer size, then write 0s to the output 2365 if (sleepTime == 0) { 2366 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2367 sleepTime = activeSleepTime >> sleepTimeShift; 2368 if (sleepTime < kMinThreadSleepTimeUs) { 2369 sleepTime = kMinThreadSleepTimeUs; 2370 } 2371 // reduce sleep time in case of consecutive application underruns to avoid 2372 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2373 // duration we would end up writing less data than needed by the audio HAL if 2374 // the condition persists. 2375 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2376 sleepTimeShift++; 2377 } 2378 } else { 2379 sleepTime = idleSleepTime; 2380 } 2381 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2382 memset (mMixBuffer, 0, mixBufferSize); 2383 sleepTime = 0; 2384 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2385 "anticipated start"); 2386 } 2387 // TODO add standby time extension fct of effect tail 2388} 2389 2390// prepareTracks_l() must be called with ThreadBase::mLock held 2391AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2392 Vector< sp<Track> > *tracksToRemove) 2393{ 2394 2395 mixer_state mixerStatus = MIXER_IDLE; 2396 // find out which tracks need to be processed 2397 size_t count = mActiveTracks.size(); 2398 size_t mixedTracks = 0; 2399 size_t tracksWithEffect = 0; 2400 // counts only _active_ fast tracks 2401 size_t fastTracks = 0; 2402 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2403 2404 float masterVolume = mMasterVolume; 2405 bool masterMute = mMasterMute; 2406 2407 if (masterMute) { 2408 masterVolume = 0; 2409 } 2410 // Delegate master volume control to effect in output mix effect chain if needed 2411 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2412 if (chain != 0) { 2413 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2414 chain->setVolume_l(&v, &v); 2415 masterVolume = (float)((v + (1 << 23)) >> 24); 2416 chain.clear(); 2417 } 2418 2419 // prepare a new state to push 2420 FastMixerStateQueue *sq = NULL; 2421 FastMixerState *state = NULL; 2422 bool didModify = false; 2423 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2424 if (mFastMixer != NULL) { 2425 sq = mFastMixer->sq(); 2426 state = sq->begin(); 2427 } 2428 2429 for (size_t i=0 ; i<count ; i++) { 2430 sp<Track> t = mActiveTracks[i].promote(); 2431 if (t == 0) { 2432 continue; 2433 } 2434 2435 // this const just means the local variable doesn't change 2436 Track* const track = t.get(); 2437 2438 // process fast tracks 2439 if (track->isFastTrack()) { 2440 2441 // It's theoretically possible (though unlikely) for a fast track to be created 2442 // and then removed within the same normal mix cycle. This is not a problem, as 2443 // the track never becomes active so it's fast mixer slot is never touched. 2444 // The converse, of removing an (active) track and then creating a new track 2445 // at the identical fast mixer slot within the same normal mix cycle, 2446 // is impossible because the slot isn't marked available until the end of each cycle. 2447 int j = track->mFastIndex; 2448 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2449 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2450 FastTrack *fastTrack = &state->mFastTracks[j]; 2451 2452 // Determine whether the track is currently in underrun condition, 2453 // and whether it had a recent underrun. 2454 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2455 FastTrackUnderruns underruns = ftDump->mUnderruns; 2456 uint32_t recentFull = (underruns.mBitFields.mFull - 2457 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2458 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2459 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2460 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2461 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2462 uint32_t recentUnderruns = recentPartial + recentEmpty; 2463 track->mObservedUnderruns = underruns; 2464 // don't count underruns that occur while stopping or pausing 2465 // or stopped which can occur when flush() is called while active 2466 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2467 track->mUnderrunCount += recentUnderruns; 2468 } 2469 2470 // This is similar to the state machine for normal tracks, 2471 // with a few modifications for fast tracks. 2472 bool isActive = true; 2473 switch (track->mState) { 2474 case TrackBase::STOPPING_1: 2475 // track stays active in STOPPING_1 state until first underrun 2476 if (recentUnderruns > 0) { 2477 track->mState = TrackBase::STOPPING_2; 2478 } 2479 break; 2480 case TrackBase::PAUSING: 2481 // ramp down is not yet implemented 2482 track->setPaused(); 2483 break; 2484 case TrackBase::RESUMING: 2485 // ramp up is not yet implemented 2486 track->mState = TrackBase::ACTIVE; 2487 break; 2488 case TrackBase::ACTIVE: 2489 if (recentFull > 0 || recentPartial > 0) { 2490 // track has provided at least some frames recently: reset retry count 2491 track->mRetryCount = kMaxTrackRetries; 2492 } 2493 if (recentUnderruns == 0) { 2494 // no recent underruns: stay active 2495 break; 2496 } 2497 // there has recently been an underrun of some kind 2498 if (track->sharedBuffer() == 0) { 2499 // were any of the recent underruns "empty" (no frames available)? 2500 if (recentEmpty == 0) { 2501 // no, then ignore the partial underruns as they are allowed indefinitely 2502 break; 2503 } 2504 // there has recently been an "empty" underrun: decrement the retry counter 2505 if (--(track->mRetryCount) > 0) { 2506 break; 2507 } 2508 // indicate to client process that the track was disabled because of underrun; 2509 // it will then automatically call start() when data is available 2510 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2511 // remove from active list, but state remains ACTIVE [confusing but true] 2512 isActive = false; 2513 break; 2514 } 2515 // fall through 2516 case TrackBase::STOPPING_2: 2517 case TrackBase::PAUSED: 2518 case TrackBase::TERMINATED: 2519 case TrackBase::STOPPED: 2520 case TrackBase::FLUSHED: // flush() while active 2521 // Check for presentation complete if track is inactive 2522 // We have consumed all the buffers of this track. 2523 // This would be incomplete if we auto-paused on underrun 2524 { 2525 size_t audioHALFrames = 2526 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2527 size_t framesWritten = mBytesWritten / mFrameSize; 2528 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2529 // track stays in active list until presentation is complete 2530 break; 2531 } 2532 } 2533 if (track->isStopping_2()) { 2534 track->mState = TrackBase::STOPPED; 2535 } 2536 if (track->isStopped()) { 2537 // Can't reset directly, as fast mixer is still polling this track 2538 // track->reset(); 2539 // So instead mark this track as needing to be reset after push with ack 2540 resetMask |= 1 << i; 2541 } 2542 isActive = false; 2543 break; 2544 case TrackBase::IDLE: 2545 default: 2546 LOG_FATAL("unexpected track state %d", track->mState); 2547 } 2548 2549 if (isActive) { 2550 // was it previously inactive? 2551 if (!(state->mTrackMask & (1 << j))) { 2552 ExtendedAudioBufferProvider *eabp = track; 2553 VolumeProvider *vp = track; 2554 fastTrack->mBufferProvider = eabp; 2555 fastTrack->mVolumeProvider = vp; 2556 fastTrack->mSampleRate = track->mSampleRate; 2557 fastTrack->mChannelMask = track->mChannelMask; 2558 fastTrack->mGeneration++; 2559 state->mTrackMask |= 1 << j; 2560 didModify = true; 2561 // no acknowledgement required for newly active tracks 2562 } 2563 // cache the combined master volume and stream type volume for fast mixer; this 2564 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2565 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2566 ++fastTracks; 2567 } else { 2568 // was it previously active? 2569 if (state->mTrackMask & (1 << j)) { 2570 fastTrack->mBufferProvider = NULL; 2571 fastTrack->mGeneration++; 2572 state->mTrackMask &= ~(1 << j); 2573 didModify = true; 2574 // If any fast tracks were removed, we must wait for acknowledgement 2575 // because we're about to decrement the last sp<> on those tracks. 2576 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2577 } else { 2578 LOG_FATAL("fast track %d should have been active", j); 2579 } 2580 tracksToRemove->add(track); 2581 // Avoids a misleading display in dumpsys 2582 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2583 } 2584 continue; 2585 } 2586 2587 { // local variable scope to avoid goto warning 2588 2589 audio_track_cblk_t* cblk = track->cblk(); 2590 2591 // The first time a track is added we wait 2592 // for all its buffers to be filled before processing it 2593 int name = track->name(); 2594 // make sure that we have enough frames to mix one full buffer. 2595 // enforce this condition only once to enable draining the buffer in case the client 2596 // app does not call stop() and relies on underrun to stop: 2597 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2598 // during last round 2599 size_t desiredFrames; 2600 if (t->sampleRate() == mSampleRate) { 2601 desiredFrames = mNormalFrameCount; 2602 } else { 2603 // +1 for rounding and +1 for additional sample needed for interpolation 2604 desiredFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2605 // add frames already consumed but not yet released by the resampler 2606 // because cblk->framesReady() will include these frames 2607 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2608 // the minimum track buffer size is normally twice the number of frames necessary 2609 // to fill one buffer and the resampler should not leave more than one buffer worth 2610 // of unreleased frames after each pass, but just in case... 2611 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2612 } 2613 uint32_t minFrames = 1; 2614 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2615 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2616 minFrames = desiredFrames; 2617 } 2618 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2619 size_t framesReady; 2620 if (track->sharedBuffer() == 0) { 2621 framesReady = track->framesReady(); 2622 } else if (track->isStopped()) { 2623 framesReady = 0; 2624 } else { 2625 framesReady = 1; 2626 } 2627 if ((framesReady >= minFrames) && track->isReady() && 2628 !track->isPaused() && !track->isTerminated()) 2629 { 2630 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2631 this); 2632 2633 mixedTracks++; 2634 2635 // track->mainBuffer() != mMixBuffer means there is an effect chain 2636 // connected to the track 2637 chain.clear(); 2638 if (track->mainBuffer() != mMixBuffer) { 2639 chain = getEffectChain_l(track->sessionId()); 2640 // Delegate volume control to effect in track effect chain if needed 2641 if (chain != 0) { 2642 tracksWithEffect++; 2643 } else { 2644 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2645 "session %d", 2646 name, track->sessionId()); 2647 } 2648 } 2649 2650 2651 int param = AudioMixer::VOLUME; 2652 if (track->mFillingUpStatus == Track::FS_FILLED) { 2653 // no ramp for the first volume setting 2654 track->mFillingUpStatus = Track::FS_ACTIVE; 2655 if (track->mState == TrackBase::RESUMING) { 2656 track->mState = TrackBase::ACTIVE; 2657 param = AudioMixer::RAMP_VOLUME; 2658 } 2659 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2660 } else if (cblk->server != 0) { 2661 // If the track is stopped before the first frame was mixed, 2662 // do not apply ramp 2663 param = AudioMixer::RAMP_VOLUME; 2664 } 2665 2666 // compute volume for this track 2667 uint32_t vl, vr, va; 2668 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2669 vl = vr = va = 0; 2670 if (track->isPausing()) { 2671 track->setPaused(); 2672 } 2673 } else { 2674 2675 // read original volumes with volume control 2676 float typeVolume = mStreamTypes[track->streamType()].volume; 2677 float v = masterVolume * typeVolume; 2678 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2679 uint32_t vlr = proxy->getVolumeLR(); 2680 vl = vlr & 0xFFFF; 2681 vr = vlr >> 16; 2682 // track volumes come from shared memory, so can't be trusted and must be clamped 2683 if (vl > MAX_GAIN_INT) { 2684 ALOGV("Track left volume out of range: %04X", vl); 2685 vl = MAX_GAIN_INT; 2686 } 2687 if (vr > MAX_GAIN_INT) { 2688 ALOGV("Track right volume out of range: %04X", vr); 2689 vr = MAX_GAIN_INT; 2690 } 2691 // now apply the master volume and stream type volume 2692 vl = (uint32_t)(v * vl) << 12; 2693 vr = (uint32_t)(v * vr) << 12; 2694 // assuming master volume and stream type volume each go up to 1.0, 2695 // vl and vr are now in 8.24 format 2696 2697 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2698 // send level comes from shared memory and so may be corrupt 2699 if (sendLevel > MAX_GAIN_INT) { 2700 ALOGV("Track send level out of range: %04X", sendLevel); 2701 sendLevel = MAX_GAIN_INT; 2702 } 2703 va = (uint32_t)(v * sendLevel); 2704 } 2705 // Delegate volume control to effect in track effect chain if needed 2706 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2707 // Do not ramp volume if volume is controlled by effect 2708 param = AudioMixer::VOLUME; 2709 track->mHasVolumeController = true; 2710 } else { 2711 // force no volume ramp when volume controller was just disabled or removed 2712 // from effect chain to avoid volume spike 2713 if (track->mHasVolumeController) { 2714 param = AudioMixer::VOLUME; 2715 } 2716 track->mHasVolumeController = false; 2717 } 2718 2719 // Convert volumes from 8.24 to 4.12 format 2720 // This additional clamping is needed in case chain->setVolume_l() overshot 2721 vl = (vl + (1 << 11)) >> 12; 2722 if (vl > MAX_GAIN_INT) { 2723 vl = MAX_GAIN_INT; 2724 } 2725 vr = (vr + (1 << 11)) >> 12; 2726 if (vr > MAX_GAIN_INT) { 2727 vr = MAX_GAIN_INT; 2728 } 2729 2730 if (va > MAX_GAIN_INT) { 2731 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2732 } 2733 2734 // XXX: these things DON'T need to be done each time 2735 mAudioMixer->setBufferProvider(name, track); 2736 mAudioMixer->enable(name); 2737 2738 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2739 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2740 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2741 mAudioMixer->setParameter( 2742 name, 2743 AudioMixer::TRACK, 2744 AudioMixer::FORMAT, (void *)track->format()); 2745 mAudioMixer->setParameter( 2746 name, 2747 AudioMixer::TRACK, 2748 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2749 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2750 uint32_t maxSampleRate = mSampleRate * 2; 2751 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 2752 if (reqSampleRate == 0) { 2753 reqSampleRate = mSampleRate; 2754 } else if (reqSampleRate > maxSampleRate) { 2755 reqSampleRate = maxSampleRate; 2756 } 2757 mAudioMixer->setParameter( 2758 name, 2759 AudioMixer::RESAMPLE, 2760 AudioMixer::SAMPLE_RATE, 2761 (void *)reqSampleRate); 2762 mAudioMixer->setParameter( 2763 name, 2764 AudioMixer::TRACK, 2765 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2766 mAudioMixer->setParameter( 2767 name, 2768 AudioMixer::TRACK, 2769 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2770 2771 // reset retry count 2772 track->mRetryCount = kMaxTrackRetries; 2773 2774 // If one track is ready, set the mixer ready if: 2775 // - the mixer was not ready during previous round OR 2776 // - no other track is not ready 2777 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2778 mixerStatus != MIXER_TRACKS_ENABLED) { 2779 mixerStatus = MIXER_TRACKS_READY; 2780 } 2781 } else { 2782 // only implemented for normal tracks, not fast tracks 2783 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 2784 // we missed desiredFrames whatever the actual number of frames missing was 2785 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 2786 // FIXME also wake futex so that underrun is noticed more quickly 2787 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 2788 } 2789 // clear effect chain input buffer if an active track underruns to avoid sending 2790 // previous audio buffer again to effects 2791 chain = getEffectChain_l(track->sessionId()); 2792 if (chain != 0) { 2793 chain->clearInputBuffer(); 2794 } 2795 2796 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2797 cblk->server, this); 2798 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2799 track->isStopped() || track->isPaused()) { 2800 // We have consumed all the buffers of this track. 2801 // Remove it from the list of active tracks. 2802 // TODO: use actual buffer filling status instead of latency when available from 2803 // audio HAL 2804 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2805 size_t framesWritten = mBytesWritten / mFrameSize; 2806 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2807 if (track->isStopped()) { 2808 track->reset(); 2809 } 2810 tracksToRemove->add(track); 2811 } 2812 } else { 2813 track->mUnderrunCount++; 2814 // No buffers for this track. Give it a few chances to 2815 // fill a buffer, then remove it from active list. 2816 if (--(track->mRetryCount) <= 0) { 2817 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2818 tracksToRemove->add(track); 2819 // indicate to client process that the track was disabled because of underrun; 2820 // it will then automatically call start() when data is available 2821 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2822 // If one track is not ready, mark the mixer also not ready if: 2823 // - the mixer was ready during previous round OR 2824 // - no other track is ready 2825 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2826 mixerStatus != MIXER_TRACKS_READY) { 2827 mixerStatus = MIXER_TRACKS_ENABLED; 2828 } 2829 } 2830 mAudioMixer->disable(name); 2831 } 2832 2833 } // local variable scope to avoid goto warning 2834track_is_ready: ; 2835 2836 } 2837 2838 // Push the new FastMixer state if necessary 2839 bool pauseAudioWatchdog = false; 2840 if (didModify) { 2841 state->mFastTracksGen++; 2842 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2843 if (kUseFastMixer == FastMixer_Dynamic && 2844 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2845 state->mCommand = FastMixerState::COLD_IDLE; 2846 state->mColdFutexAddr = &mFastMixerFutex; 2847 state->mColdGen++; 2848 mFastMixerFutex = 0; 2849 if (kUseFastMixer == FastMixer_Dynamic) { 2850 mNormalSink = mOutputSink; 2851 } 2852 // If we go into cold idle, need to wait for acknowledgement 2853 // so that fast mixer stops doing I/O. 2854 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2855 pauseAudioWatchdog = true; 2856 } 2857 } 2858 if (sq != NULL) { 2859 sq->end(didModify); 2860 sq->push(block); 2861 } 2862#ifdef AUDIO_WATCHDOG 2863 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2864 mAudioWatchdog->pause(); 2865 } 2866#endif 2867 2868 // Now perform the deferred reset on fast tracks that have stopped 2869 while (resetMask != 0) { 2870 size_t i = __builtin_ctz(resetMask); 2871 ALOG_ASSERT(i < count); 2872 resetMask &= ~(1 << i); 2873 sp<Track> t = mActiveTracks[i].promote(); 2874 if (t == 0) { 2875 continue; 2876 } 2877 Track* track = t.get(); 2878 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2879 track->reset(); 2880 } 2881 2882 // remove all the tracks that need to be... 2883 count = tracksToRemove->size(); 2884 if (CC_UNLIKELY(count)) { 2885 for (size_t i=0 ; i<count ; i++) { 2886 const sp<Track>& track = tracksToRemove->itemAt(i); 2887 mActiveTracks.remove(track); 2888 if (track->mainBuffer() != mMixBuffer) { 2889 chain = getEffectChain_l(track->sessionId()); 2890 if (chain != 0) { 2891 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2892 track->sessionId()); 2893 chain->decActiveTrackCnt(); 2894 } 2895 } 2896 if (track->isTerminated()) { 2897 removeTrack_l(track); 2898 } 2899 } 2900 } 2901 2902 // mix buffer must be cleared if all tracks are connected to an 2903 // effect chain as in this case the mixer will not write to 2904 // mix buffer and track effects will accumulate into it 2905 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2906 (mixedTracks == 0 && fastTracks > 0)) { 2907 // FIXME as a performance optimization, should remember previous zero status 2908 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2909 } 2910 2911 // if any fast tracks, then status is ready 2912 mMixerStatusIgnoringFastTracks = mixerStatus; 2913 if (fastTracks > 0) { 2914 mixerStatus = MIXER_TRACKS_READY; 2915 } 2916 return mixerStatus; 2917} 2918 2919// getTrackName_l() must be called with ThreadBase::mLock held 2920int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2921{ 2922 return mAudioMixer->getTrackName(channelMask, sessionId); 2923} 2924 2925// deleteTrackName_l() must be called with ThreadBase::mLock held 2926void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2927{ 2928 ALOGV("remove track (%d) and delete from mixer", name); 2929 mAudioMixer->deleteTrackName(name); 2930} 2931 2932// checkForNewParameters_l() must be called with ThreadBase::mLock held 2933bool AudioFlinger::MixerThread::checkForNewParameters_l() 2934{ 2935 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2936 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2937 bool reconfig = false; 2938 2939 while (!mNewParameters.isEmpty()) { 2940 2941 if (mFastMixer != NULL) { 2942 FastMixerStateQueue *sq = mFastMixer->sq(); 2943 FastMixerState *state = sq->begin(); 2944 if (!(state->mCommand & FastMixerState::IDLE)) { 2945 previousCommand = state->mCommand; 2946 state->mCommand = FastMixerState::HOT_IDLE; 2947 sq->end(); 2948 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2949 } else { 2950 sq->end(false /*didModify*/); 2951 } 2952 } 2953 2954 status_t status = NO_ERROR; 2955 String8 keyValuePair = mNewParameters[0]; 2956 AudioParameter param = AudioParameter(keyValuePair); 2957 int value; 2958 2959 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2960 reconfig = true; 2961 } 2962 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2963 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2964 status = BAD_VALUE; 2965 } else { 2966 reconfig = true; 2967 } 2968 } 2969 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2970 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2971 status = BAD_VALUE; 2972 } else { 2973 reconfig = true; 2974 } 2975 } 2976 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2977 // do not accept frame count changes if tracks are open as the track buffer 2978 // size depends on frame count and correct behavior would not be guaranteed 2979 // if frame count is changed after track creation 2980 if (!mTracks.isEmpty()) { 2981 status = INVALID_OPERATION; 2982 } else { 2983 reconfig = true; 2984 } 2985 } 2986 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2987#ifdef ADD_BATTERY_DATA 2988 // when changing the audio output device, call addBatteryData to notify 2989 // the change 2990 if (mOutDevice != value) { 2991 uint32_t params = 0; 2992 // check whether speaker is on 2993 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2994 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2995 } 2996 2997 audio_devices_t deviceWithoutSpeaker 2998 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2999 // check if any other device (except speaker) is on 3000 if (value & deviceWithoutSpeaker ) { 3001 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3002 } 3003 3004 if (params != 0) { 3005 addBatteryData(params); 3006 } 3007 } 3008#endif 3009 3010 // forward device change to effects that have requested to be 3011 // aware of attached audio device. 3012 if (value != AUDIO_DEVICE_NONE) { 3013 mOutDevice = value; 3014 for (size_t i = 0; i < mEffectChains.size(); i++) { 3015 mEffectChains[i]->setDevice_l(mOutDevice); 3016 } 3017 } 3018 } 3019 3020 if (status == NO_ERROR) { 3021 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3022 keyValuePair.string()); 3023 if (!mStandby && status == INVALID_OPERATION) { 3024 mOutput->stream->common.standby(&mOutput->stream->common); 3025 mStandby = true; 3026 mBytesWritten = 0; 3027 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3028 keyValuePair.string()); 3029 } 3030 if (status == NO_ERROR && reconfig) { 3031 delete mAudioMixer; 3032 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3033 mAudioMixer = NULL; 3034 readOutputParameters(); 3035 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3036 for (size_t i = 0; i < mTracks.size() ; i++) { 3037 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3038 if (name < 0) { 3039 break; 3040 } 3041 mTracks[i]->mName = name; 3042 } 3043 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3044 } 3045 } 3046 3047 mNewParameters.removeAt(0); 3048 3049 mParamStatus = status; 3050 mParamCond.signal(); 3051 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3052 // already timed out waiting for the status and will never signal the condition. 3053 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3054 } 3055 3056 if (!(previousCommand & FastMixerState::IDLE)) { 3057 ALOG_ASSERT(mFastMixer != NULL); 3058 FastMixerStateQueue *sq = mFastMixer->sq(); 3059 FastMixerState *state = sq->begin(); 3060 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3061 state->mCommand = previousCommand; 3062 sq->end(); 3063 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3064 } 3065 3066 return reconfig; 3067} 3068 3069 3070void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3071{ 3072 const size_t SIZE = 256; 3073 char buffer[SIZE]; 3074 String8 result; 3075 3076 PlaybackThread::dumpInternals(fd, args); 3077 3078 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3079 result.append(buffer); 3080 write(fd, result.string(), result.size()); 3081 3082 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3083 FastMixerDumpState copy = mFastMixerDumpState; 3084 copy.dump(fd); 3085 3086#ifdef STATE_QUEUE_DUMP 3087 // Similar for state queue 3088 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3089 observerCopy.dump(fd); 3090 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3091 mutatorCopy.dump(fd); 3092#endif 3093 3094#ifdef TEE_SINK 3095 // Write the tee output to a .wav file 3096 dumpTee(fd, mTeeSource, mId); 3097#endif 3098 3099#ifdef AUDIO_WATCHDOG 3100 if (mAudioWatchdog != 0) { 3101 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3102 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3103 wdCopy.dump(fd); 3104 } 3105#endif 3106} 3107 3108uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3109{ 3110 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3111} 3112 3113uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3114{ 3115 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3116} 3117 3118void AudioFlinger::MixerThread::cacheParameters_l() 3119{ 3120 PlaybackThread::cacheParameters_l(); 3121 3122 // FIXME: Relaxed timing because of a certain device that can't meet latency 3123 // Should be reduced to 2x after the vendor fixes the driver issue 3124 // increase threshold again due to low power audio mode. The way this warning 3125 // threshold is calculated and its usefulness should be reconsidered anyway. 3126 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3127} 3128 3129// ---------------------------------------------------------------------------- 3130 3131AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3132 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3133 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3134 // mLeftVolFloat, mRightVolFloat 3135{ 3136} 3137 3138AudioFlinger::DirectOutputThread::~DirectOutputThread() 3139{ 3140} 3141 3142AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3143 Vector< sp<Track> > *tracksToRemove 3144) 3145{ 3146 size_t count = mActiveTracks.size(); 3147 mixer_state mixerStatus = MIXER_IDLE; 3148 3149 // find out which tracks need to be processed 3150 for (size_t i = 0; i < count; i++) { 3151 sp<Track> t = mActiveTracks[i].promote(); 3152 // The track died recently 3153 if (t == 0) { 3154 continue; 3155 } 3156 3157 Track* const track = t.get(); 3158 audio_track_cblk_t* cblk = track->cblk(); 3159 3160 // The first time a track is added we wait 3161 // for all its buffers to be filled before processing it 3162 uint32_t minFrames; 3163 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3164 minFrames = mNormalFrameCount; 3165 } else { 3166 minFrames = 1; 3167 } 3168 if ((track->framesReady() >= minFrames) && track->isReady() && 3169 !track->isPaused() && !track->isTerminated()) 3170 { 3171 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3172 3173 if (track->mFillingUpStatus == Track::FS_FILLED) { 3174 track->mFillingUpStatus = Track::FS_ACTIVE; 3175 mLeftVolFloat = mRightVolFloat = 0; 3176 if (track->mState == TrackBase::RESUMING) { 3177 track->mState = TrackBase::ACTIVE; 3178 } 3179 } 3180 3181 // compute volume for this track 3182 float left, right; 3183 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3184 left = right = 0; 3185 if (track->isPausing()) { 3186 track->setPaused(); 3187 } 3188 } else { 3189 float typeVolume = mStreamTypes[track->streamType()].volume; 3190 float v = mMasterVolume * typeVolume; 3191 uint32_t vlr = track->mAudioTrackServerProxy->getVolumeLR(); 3192 float v_clamped = v * (vlr & 0xFFFF); 3193 if (v_clamped > MAX_GAIN) { 3194 v_clamped = MAX_GAIN; 3195 } 3196 left = v_clamped/MAX_GAIN; 3197 v_clamped = v * (vlr >> 16); 3198 if (v_clamped > MAX_GAIN) { 3199 v_clamped = MAX_GAIN; 3200 } 3201 right = v_clamped/MAX_GAIN; 3202 } 3203 // Only consider last track started for volume and mixer state control. 3204 // This is the last entry in mActiveTracks unless a track underruns. 3205 // As we only care about the transition phase between two tracks on a 3206 // direct output, it is not a problem to ignore the underrun case. 3207 if (i == (count - 1)) { 3208 if (left != mLeftVolFloat || right != mRightVolFloat) { 3209 mLeftVolFloat = left; 3210 mRightVolFloat = right; 3211 3212 // Convert volumes from float to 8.24 3213 uint32_t vl = (uint32_t)(left * (1 << 24)); 3214 uint32_t vr = (uint32_t)(right * (1 << 24)); 3215 3216 // Delegate volume control to effect in track effect chain if needed 3217 // only one effect chain can be present on DirectOutputThread, so if 3218 // there is one, the track is connected to it 3219 if (!mEffectChains.isEmpty()) { 3220 // Do not ramp volume if volume is controlled by effect 3221 mEffectChains[0]->setVolume_l(&vl, &vr); 3222 left = (float)vl / (1 << 24); 3223 right = (float)vr / (1 << 24); 3224 } 3225 mOutput->stream->set_volume(mOutput->stream, left, right); 3226 } 3227 3228 // reset retry count 3229 track->mRetryCount = kMaxTrackRetriesDirect; 3230 mActiveTrack = t; 3231 mixerStatus = MIXER_TRACKS_READY; 3232 } 3233 } else { 3234 // clear effect chain input buffer if the last active track started underruns 3235 // to avoid sending previous audio buffer again to effects 3236 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3237 mEffectChains[0]->clearInputBuffer(); 3238 } 3239 3240 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3241 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3242 track->isStopped() || track->isPaused()) { 3243 // We have consumed all the buffers of this track. 3244 // Remove it from the list of active tracks. 3245 // TODO: implement behavior for compressed audio 3246 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3247 size_t framesWritten = mBytesWritten / mFrameSize; 3248 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3249 if (track->isStopped()) { 3250 track->reset(); 3251 } 3252 tracksToRemove->add(track); 3253 } 3254 } else { 3255 // No buffers for this track. Give it a few chances to 3256 // fill a buffer, then remove it from active list. 3257 // Only consider last track started for mixer state control 3258 if (--(track->mRetryCount) <= 0) { 3259 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3260 tracksToRemove->add(track); 3261 } else if (i == (count -1)){ 3262 mixerStatus = MIXER_TRACKS_ENABLED; 3263 } 3264 } 3265 } 3266 } 3267 3268 // remove all the tracks that need to be... 3269 count = tracksToRemove->size(); 3270 if (CC_UNLIKELY(count)) { 3271 for (size_t i = 0 ; i < count ; i++) { 3272 const sp<Track>& track = tracksToRemove->itemAt(i); 3273 mActiveTracks.remove(track); 3274 if (!mEffectChains.isEmpty()) { 3275 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3276 track->sessionId()); 3277 mEffectChains[0]->decActiveTrackCnt(); 3278 } 3279 if (track->isTerminated()) { 3280 removeTrack_l(track); 3281 } 3282 } 3283 } 3284 3285 return mixerStatus; 3286} 3287 3288void AudioFlinger::DirectOutputThread::threadLoop_mix() 3289{ 3290 AudioBufferProvider::Buffer buffer; 3291 size_t frameCount = mFrameCount; 3292 int8_t *curBuf = (int8_t *)mMixBuffer; 3293 // output audio to hardware 3294 while (frameCount) { 3295 buffer.frameCount = frameCount; 3296 mActiveTrack->getNextBuffer(&buffer); 3297 if (CC_UNLIKELY(buffer.raw == NULL)) { 3298 memset(curBuf, 0, frameCount * mFrameSize); 3299 break; 3300 } 3301 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3302 frameCount -= buffer.frameCount; 3303 curBuf += buffer.frameCount * mFrameSize; 3304 mActiveTrack->releaseBuffer(&buffer); 3305 } 3306 sleepTime = 0; 3307 standbyTime = systemTime() + standbyDelay; 3308 mActiveTrack.clear(); 3309 3310} 3311 3312void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3313{ 3314 if (sleepTime == 0) { 3315 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3316 sleepTime = activeSleepTime; 3317 } else { 3318 sleepTime = idleSleepTime; 3319 } 3320 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3321 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3322 sleepTime = 0; 3323 } 3324} 3325 3326// getTrackName_l() must be called with ThreadBase::mLock held 3327int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3328 int sessionId) 3329{ 3330 return 0; 3331} 3332 3333// deleteTrackName_l() must be called with ThreadBase::mLock held 3334void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3335{ 3336} 3337 3338// checkForNewParameters_l() must be called with ThreadBase::mLock held 3339bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3340{ 3341 bool reconfig = false; 3342 3343 while (!mNewParameters.isEmpty()) { 3344 status_t status = NO_ERROR; 3345 String8 keyValuePair = mNewParameters[0]; 3346 AudioParameter param = AudioParameter(keyValuePair); 3347 int value; 3348 3349 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3350 // do not accept frame count changes if tracks are open as the track buffer 3351 // size depends on frame count and correct behavior would not be garantied 3352 // if frame count is changed after track creation 3353 if (!mTracks.isEmpty()) { 3354 status = INVALID_OPERATION; 3355 } else { 3356 reconfig = true; 3357 } 3358 } 3359 if (status == NO_ERROR) { 3360 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3361 keyValuePair.string()); 3362 if (!mStandby && status == INVALID_OPERATION) { 3363 mOutput->stream->common.standby(&mOutput->stream->common); 3364 mStandby = true; 3365 mBytesWritten = 0; 3366 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3367 keyValuePair.string()); 3368 } 3369 if (status == NO_ERROR && reconfig) { 3370 readOutputParameters(); 3371 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3372 } 3373 } 3374 3375 mNewParameters.removeAt(0); 3376 3377 mParamStatus = status; 3378 mParamCond.signal(); 3379 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3380 // already timed out waiting for the status and will never signal the condition. 3381 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3382 } 3383 return reconfig; 3384} 3385 3386uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3387{ 3388 uint32_t time; 3389 if (audio_is_linear_pcm(mFormat)) { 3390 time = PlaybackThread::activeSleepTimeUs(); 3391 } else { 3392 time = 10000; 3393 } 3394 return time; 3395} 3396 3397uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3398{ 3399 uint32_t time; 3400 if (audio_is_linear_pcm(mFormat)) { 3401 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3402 } else { 3403 time = 10000; 3404 } 3405 return time; 3406} 3407 3408uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3409{ 3410 uint32_t time; 3411 if (audio_is_linear_pcm(mFormat)) { 3412 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3413 } else { 3414 time = 10000; 3415 } 3416 return time; 3417} 3418 3419void AudioFlinger::DirectOutputThread::cacheParameters_l() 3420{ 3421 PlaybackThread::cacheParameters_l(); 3422 3423 // use shorter standby delay as on normal output to release 3424 // hardware resources as soon as possible 3425 standbyDelay = microseconds(activeSleepTime*2); 3426} 3427 3428// ---------------------------------------------------------------------------- 3429 3430AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3431 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3432 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3433 DUPLICATING), 3434 mWaitTimeMs(UINT_MAX) 3435{ 3436 addOutputTrack(mainThread); 3437} 3438 3439AudioFlinger::DuplicatingThread::~DuplicatingThread() 3440{ 3441 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3442 mOutputTracks[i]->destroy(); 3443 } 3444} 3445 3446void AudioFlinger::DuplicatingThread::threadLoop_mix() 3447{ 3448 // mix buffers... 3449 if (outputsReady(outputTracks)) { 3450 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3451 } else { 3452 memset(mMixBuffer, 0, mixBufferSize); 3453 } 3454 sleepTime = 0; 3455 writeFrames = mNormalFrameCount; 3456 standbyTime = systemTime() + standbyDelay; 3457} 3458 3459void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3460{ 3461 if (sleepTime == 0) { 3462 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3463 sleepTime = activeSleepTime; 3464 } else { 3465 sleepTime = idleSleepTime; 3466 } 3467 } else if (mBytesWritten != 0) { 3468 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3469 writeFrames = mNormalFrameCount; 3470 memset(mMixBuffer, 0, mixBufferSize); 3471 } else { 3472 // flush remaining overflow buffers in output tracks 3473 writeFrames = 0; 3474 } 3475 sleepTime = 0; 3476 } 3477} 3478 3479void AudioFlinger::DuplicatingThread::threadLoop_write() 3480{ 3481 for (size_t i = 0; i < outputTracks.size(); i++) { 3482 outputTracks[i]->write(mMixBuffer, writeFrames); 3483 } 3484 mBytesWritten += mixBufferSize; 3485} 3486 3487void AudioFlinger::DuplicatingThread::threadLoop_standby() 3488{ 3489 // DuplicatingThread implements standby by stopping all tracks 3490 for (size_t i = 0; i < outputTracks.size(); i++) { 3491 outputTracks[i]->stop(); 3492 } 3493} 3494 3495void AudioFlinger::DuplicatingThread::saveOutputTracks() 3496{ 3497 outputTracks = mOutputTracks; 3498} 3499 3500void AudioFlinger::DuplicatingThread::clearOutputTracks() 3501{ 3502 outputTracks.clear(); 3503} 3504 3505void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3506{ 3507 Mutex::Autolock _l(mLock); 3508 // FIXME explain this formula 3509 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3510 OutputTrack *outputTrack = new OutputTrack(thread, 3511 this, 3512 mSampleRate, 3513 mFormat, 3514 mChannelMask, 3515 frameCount); 3516 if (outputTrack->cblk() != NULL) { 3517 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3518 mOutputTracks.add(outputTrack); 3519 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3520 updateWaitTime_l(); 3521 } 3522} 3523 3524void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3525{ 3526 Mutex::Autolock _l(mLock); 3527 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3528 if (mOutputTracks[i]->thread() == thread) { 3529 mOutputTracks[i]->destroy(); 3530 mOutputTracks.removeAt(i); 3531 updateWaitTime_l(); 3532 return; 3533 } 3534 } 3535 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3536} 3537 3538// caller must hold mLock 3539void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3540{ 3541 mWaitTimeMs = UINT_MAX; 3542 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3543 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3544 if (strong != 0) { 3545 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3546 if (waitTimeMs < mWaitTimeMs) { 3547 mWaitTimeMs = waitTimeMs; 3548 } 3549 } 3550 } 3551} 3552 3553 3554bool AudioFlinger::DuplicatingThread::outputsReady( 3555 const SortedVector< sp<OutputTrack> > &outputTracks) 3556{ 3557 for (size_t i = 0; i < outputTracks.size(); i++) { 3558 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3559 if (thread == 0) { 3560 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3561 outputTracks[i].get()); 3562 return false; 3563 } 3564 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3565 // see note at standby() declaration 3566 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3567 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3568 thread.get()); 3569 return false; 3570 } 3571 } 3572 return true; 3573} 3574 3575uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3576{ 3577 return (mWaitTimeMs * 1000) / 2; 3578} 3579 3580void AudioFlinger::DuplicatingThread::cacheParameters_l() 3581{ 3582 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3583 updateWaitTime_l(); 3584 3585 MixerThread::cacheParameters_l(); 3586} 3587 3588// ---------------------------------------------------------------------------- 3589// Record 3590// ---------------------------------------------------------------------------- 3591 3592AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3593 AudioStreamIn *input, 3594 uint32_t sampleRate, 3595 audio_channel_mask_t channelMask, 3596 audio_io_handle_t id, 3597 audio_devices_t outDevice, 3598 audio_devices_t inDevice 3599#ifdef TEE_SINK 3600 , const sp<NBAIO_Sink>& teeSink 3601#endif 3602 ) : 3603 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3604 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3605 // mRsmpInIndex and mInputBytes set by readInputParameters() 3606 mReqChannelCount(popcount(channelMask)), 3607 mReqSampleRate(sampleRate) 3608 // mBytesRead is only meaningful while active, and so is cleared in start() 3609 // (but might be better to also clear here for dump?) 3610#ifdef TEE_SINK 3611 , mTeeSink(teeSink) 3612#endif 3613{ 3614 snprintf(mName, kNameLength, "AudioIn_%X", id); 3615 3616 readInputParameters(); 3617 3618} 3619 3620 3621AudioFlinger::RecordThread::~RecordThread() 3622{ 3623 delete[] mRsmpInBuffer; 3624 delete mResampler; 3625 delete[] mRsmpOutBuffer; 3626} 3627 3628void AudioFlinger::RecordThread::onFirstRef() 3629{ 3630 run(mName, PRIORITY_URGENT_AUDIO); 3631} 3632 3633status_t AudioFlinger::RecordThread::readyToRun() 3634{ 3635 status_t status = initCheck(); 3636 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3637 return status; 3638} 3639 3640bool AudioFlinger::RecordThread::threadLoop() 3641{ 3642 AudioBufferProvider::Buffer buffer; 3643 sp<RecordTrack> activeTrack; 3644 Vector< sp<EffectChain> > effectChains; 3645 3646 nsecs_t lastWarning = 0; 3647 3648 inputStandBy(); 3649 acquireWakeLock(); 3650 3651 // used to verify we've read at least once before evaluating how many bytes were read 3652 bool readOnce = false; 3653 3654 // start recording 3655 while (!exitPending()) { 3656 3657 processConfigEvents(); 3658 3659 { // scope for mLock 3660 Mutex::Autolock _l(mLock); 3661 checkForNewParameters_l(); 3662 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3663 standby(); 3664 3665 if (exitPending()) { 3666 break; 3667 } 3668 3669 releaseWakeLock_l(); 3670 ALOGV("RecordThread: loop stopping"); 3671 // go to sleep 3672 mWaitWorkCV.wait(mLock); 3673 ALOGV("RecordThread: loop starting"); 3674 acquireWakeLock_l(); 3675 continue; 3676 } 3677 if (mActiveTrack != 0) { 3678 if (mActiveTrack->mState == TrackBase::PAUSING) { 3679 standby(); 3680 mActiveTrack.clear(); 3681 mStartStopCond.broadcast(); 3682 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3683 if (mReqChannelCount != mActiveTrack->channelCount()) { 3684 mActiveTrack.clear(); 3685 mStartStopCond.broadcast(); 3686 } else if (readOnce) { 3687 // record start succeeds only if first read from audio input 3688 // succeeds 3689 if (mBytesRead >= 0) { 3690 mActiveTrack->mState = TrackBase::ACTIVE; 3691 } else { 3692 mActiveTrack.clear(); 3693 } 3694 mStartStopCond.broadcast(); 3695 } 3696 mStandby = false; 3697 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3698 removeTrack_l(mActiveTrack); 3699 mActiveTrack.clear(); 3700 } 3701 } 3702 lockEffectChains_l(effectChains); 3703 } 3704 3705 if (mActiveTrack != 0) { 3706 if (mActiveTrack->mState != TrackBase::ACTIVE && 3707 mActiveTrack->mState != TrackBase::RESUMING) { 3708 unlockEffectChains(effectChains); 3709 usleep(kRecordThreadSleepUs); 3710 continue; 3711 } 3712 for (size_t i = 0; i < effectChains.size(); i ++) { 3713 effectChains[i]->process_l(); 3714 } 3715 3716 buffer.frameCount = mFrameCount; 3717 status_t status = mActiveTrack->getNextBuffer(&buffer); 3718 if (CC_LIKELY(status == NO_ERROR)) { 3719 readOnce = true; 3720 size_t framesOut = buffer.frameCount; 3721 if (mResampler == NULL) { 3722 // no resampling 3723 while (framesOut) { 3724 size_t framesIn = mFrameCount - mRsmpInIndex; 3725 if (framesIn) { 3726 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3727 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3728 mActiveTrack->mFrameSize; 3729 if (framesIn > framesOut) 3730 framesIn = framesOut; 3731 mRsmpInIndex += framesIn; 3732 framesOut -= framesIn; 3733 if (mChannelCount == mReqChannelCount || 3734 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3735 memcpy(dst, src, framesIn * mFrameSize); 3736 } else { 3737 if (mChannelCount == 1) { 3738 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3739 (int16_t *)src, framesIn); 3740 } else { 3741 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3742 (int16_t *)src, framesIn); 3743 } 3744 } 3745 } 3746 if (framesOut && mFrameCount == mRsmpInIndex) { 3747 void *readInto; 3748 if (framesOut == mFrameCount && 3749 (mChannelCount == mReqChannelCount || 3750 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3751 readInto = buffer.raw; 3752 framesOut = 0; 3753 } else { 3754 readInto = mRsmpInBuffer; 3755 mRsmpInIndex = 0; 3756 } 3757 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3758 mInputBytes); 3759 if (mBytesRead <= 0) { 3760 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3761 { 3762 ALOGE("Error reading audio input"); 3763 // Force input into standby so that it tries to 3764 // recover at next read attempt 3765 inputStandBy(); 3766 usleep(kRecordThreadSleepUs); 3767 } 3768 mRsmpInIndex = mFrameCount; 3769 framesOut = 0; 3770 buffer.frameCount = 0; 3771 } 3772#ifdef TEE_SINK 3773 else if (mTeeSink != 0) { 3774 (void) mTeeSink->write(readInto, 3775 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3776 } 3777#endif 3778 } 3779 } 3780 } else { 3781 // resampling 3782 3783 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3784 // alter output frame count as if we were expecting stereo samples 3785 if (mChannelCount == 1 && mReqChannelCount == 1) { 3786 framesOut >>= 1; 3787 } 3788 mResampler->resample(mRsmpOutBuffer, framesOut, 3789 this /* AudioBufferProvider* */); 3790 // ditherAndClamp() works as long as all buffers returned by 3791 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3792 if (mChannelCount == 2 && mReqChannelCount == 1) { 3793 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3794 // the resampler always outputs stereo samples: 3795 // do post stereo to mono conversion 3796 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3797 framesOut); 3798 } else { 3799 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3800 } 3801 3802 } 3803 if (mFramestoDrop == 0) { 3804 mActiveTrack->releaseBuffer(&buffer); 3805 } else { 3806 if (mFramestoDrop > 0) { 3807 mFramestoDrop -= buffer.frameCount; 3808 if (mFramestoDrop <= 0) { 3809 clearSyncStartEvent(); 3810 } 3811 } else { 3812 mFramestoDrop += buffer.frameCount; 3813 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3814 mSyncStartEvent->isCancelled()) { 3815 ALOGW("Synced record %s, session %d, trigger session %d", 3816 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3817 mActiveTrack->sessionId(), 3818 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3819 clearSyncStartEvent(); 3820 } 3821 } 3822 } 3823 mActiveTrack->clearOverflow(); 3824 } 3825 // client isn't retrieving buffers fast enough 3826 else { 3827 if (!mActiveTrack->setOverflow()) { 3828 nsecs_t now = systemTime(); 3829 if ((now - lastWarning) > kWarningThrottleNs) { 3830 ALOGW("RecordThread: buffer overflow"); 3831 lastWarning = now; 3832 } 3833 } 3834 // Release the processor for a while before asking for a new buffer. 3835 // This will give the application more chance to read from the buffer and 3836 // clear the overflow. 3837 usleep(kRecordThreadSleepUs); 3838 } 3839 } 3840 // enable changes in effect chain 3841 unlockEffectChains(effectChains); 3842 effectChains.clear(); 3843 } 3844 3845 standby(); 3846 3847 { 3848 Mutex::Autolock _l(mLock); 3849 mActiveTrack.clear(); 3850 mStartStopCond.broadcast(); 3851 } 3852 3853 releaseWakeLock(); 3854 3855 ALOGV("RecordThread %p exiting", this); 3856 return false; 3857} 3858 3859void AudioFlinger::RecordThread::standby() 3860{ 3861 if (!mStandby) { 3862 inputStandBy(); 3863 mStandby = true; 3864 } 3865} 3866 3867void AudioFlinger::RecordThread::inputStandBy() 3868{ 3869 mInput->stream->common.standby(&mInput->stream->common); 3870} 3871 3872sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3873 const sp<AudioFlinger::Client>& client, 3874 uint32_t sampleRate, 3875 audio_format_t format, 3876 audio_channel_mask_t channelMask, 3877 size_t frameCount, 3878 int sessionId, 3879 IAudioFlinger::track_flags_t flags, 3880 pid_t tid, 3881 status_t *status) 3882{ 3883 sp<RecordTrack> track; 3884 status_t lStatus; 3885 3886 lStatus = initCheck(); 3887 if (lStatus != NO_ERROR) { 3888 ALOGE("Audio driver not initialized."); 3889 goto Exit; 3890 } 3891 3892 // FIXME use flags and tid similar to createTrack_l() 3893 3894 { // scope for mLock 3895 Mutex::Autolock _l(mLock); 3896 3897 track = new RecordTrack(this, client, sampleRate, 3898 format, channelMask, frameCount, sessionId); 3899 3900 if (track->getCblk() == 0) { 3901 lStatus = NO_MEMORY; 3902 goto Exit; 3903 } 3904 mTracks.add(track); 3905 3906 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3907 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3908 mAudioFlinger->btNrecIsOff(); 3909 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3910 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3911 } 3912 lStatus = NO_ERROR; 3913 3914Exit: 3915 if (status) { 3916 *status = lStatus; 3917 } 3918 return track; 3919} 3920 3921status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3922 AudioSystem::sync_event_t event, 3923 int triggerSession) 3924{ 3925 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3926 sp<ThreadBase> strongMe = this; 3927 status_t status = NO_ERROR; 3928 3929 if (event == AudioSystem::SYNC_EVENT_NONE) { 3930 clearSyncStartEvent(); 3931 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3932 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3933 triggerSession, 3934 recordTrack->sessionId(), 3935 syncStartEventCallback, 3936 this); 3937 // Sync event can be cancelled by the trigger session if the track is not in a 3938 // compatible state in which case we start record immediately 3939 if (mSyncStartEvent->isCancelled()) { 3940 clearSyncStartEvent(); 3941 } else { 3942 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3943 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3944 } 3945 } 3946 3947 { 3948 AutoMutex lock(mLock); 3949 if (mActiveTrack != 0) { 3950 if (recordTrack != mActiveTrack.get()) { 3951 status = -EBUSY; 3952 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3953 mActiveTrack->mState = TrackBase::ACTIVE; 3954 } 3955 return status; 3956 } 3957 3958 recordTrack->mState = TrackBase::IDLE; 3959 mActiveTrack = recordTrack; 3960 mLock.unlock(); 3961 status_t status = AudioSystem::startInput(mId); 3962 mLock.lock(); 3963 if (status != NO_ERROR) { 3964 mActiveTrack.clear(); 3965 clearSyncStartEvent(); 3966 return status; 3967 } 3968 mRsmpInIndex = mFrameCount; 3969 mBytesRead = 0; 3970 if (mResampler != NULL) { 3971 mResampler->reset(); 3972 } 3973 mActiveTrack->mState = TrackBase::RESUMING; 3974 // signal thread to start 3975 ALOGV("Signal record thread"); 3976 mWaitWorkCV.broadcast(); 3977 // do not wait for mStartStopCond if exiting 3978 if (exitPending()) { 3979 mActiveTrack.clear(); 3980 status = INVALID_OPERATION; 3981 goto startError; 3982 } 3983 mStartStopCond.wait(mLock); 3984 if (mActiveTrack == 0) { 3985 ALOGV("Record failed to start"); 3986 status = BAD_VALUE; 3987 goto startError; 3988 } 3989 ALOGV("Record started OK"); 3990 return status; 3991 } 3992 3993startError: 3994 AudioSystem::stopInput(mId); 3995 clearSyncStartEvent(); 3996 return status; 3997} 3998 3999void AudioFlinger::RecordThread::clearSyncStartEvent() 4000{ 4001 if (mSyncStartEvent != 0) { 4002 mSyncStartEvent->cancel(); 4003 } 4004 mSyncStartEvent.clear(); 4005 mFramestoDrop = 0; 4006} 4007 4008void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4009{ 4010 sp<SyncEvent> strongEvent = event.promote(); 4011 4012 if (strongEvent != 0) { 4013 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4014 me->handleSyncStartEvent(strongEvent); 4015 } 4016} 4017 4018void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4019{ 4020 if (event == mSyncStartEvent) { 4021 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4022 // from audio HAL 4023 mFramestoDrop = mFrameCount * 2; 4024 } 4025} 4026 4027bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4028 ALOGV("RecordThread::stop"); 4029 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4030 return false; 4031 } 4032 recordTrack->mState = TrackBase::PAUSING; 4033 // do not wait for mStartStopCond if exiting 4034 if (exitPending()) { 4035 return true; 4036 } 4037 mStartStopCond.wait(mLock); 4038 // if we have been restarted, recordTrack == mActiveTrack.get() here 4039 if (exitPending() || recordTrack != mActiveTrack.get()) { 4040 ALOGV("Record stopped OK"); 4041 return true; 4042 } 4043 return false; 4044} 4045 4046bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4047{ 4048 return false; 4049} 4050 4051status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4052{ 4053#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4054 if (!isValidSyncEvent(event)) { 4055 return BAD_VALUE; 4056 } 4057 4058 int eventSession = event->triggerSession(); 4059 status_t ret = NAME_NOT_FOUND; 4060 4061 Mutex::Autolock _l(mLock); 4062 4063 for (size_t i = 0; i < mTracks.size(); i++) { 4064 sp<RecordTrack> track = mTracks[i]; 4065 if (eventSession == track->sessionId()) { 4066 (void) track->setSyncEvent(event); 4067 ret = NO_ERROR; 4068 } 4069 } 4070 return ret; 4071#else 4072 return BAD_VALUE; 4073#endif 4074} 4075 4076// destroyTrack_l() must be called with ThreadBase::mLock held 4077void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4078{ 4079 track->mState = TrackBase::TERMINATED; 4080 // active tracks are removed by threadLoop() 4081 if (mActiveTrack != track) { 4082 removeTrack_l(track); 4083 } 4084} 4085 4086void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4087{ 4088 mTracks.remove(track); 4089 // need anything related to effects here? 4090} 4091 4092void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4093{ 4094 dumpInternals(fd, args); 4095 dumpTracks(fd, args); 4096 dumpEffectChains(fd, args); 4097} 4098 4099void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4100{ 4101 const size_t SIZE = 256; 4102 char buffer[SIZE]; 4103 String8 result; 4104 4105 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4106 result.append(buffer); 4107 4108 if (mActiveTrack != 0) { 4109 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4110 result.append(buffer); 4111 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4112 result.append(buffer); 4113 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4114 result.append(buffer); 4115 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4116 result.append(buffer); 4117 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4118 result.append(buffer); 4119 } else { 4120 result.append("No active record client\n"); 4121 } 4122 4123 write(fd, result.string(), result.size()); 4124 4125 dumpBase(fd, args); 4126} 4127 4128void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4129{ 4130 const size_t SIZE = 256; 4131 char buffer[SIZE]; 4132 String8 result; 4133 4134 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4135 result.append(buffer); 4136 RecordTrack::appendDumpHeader(result); 4137 for (size_t i = 0; i < mTracks.size(); ++i) { 4138 sp<RecordTrack> track = mTracks[i]; 4139 if (track != 0) { 4140 track->dump(buffer, SIZE); 4141 result.append(buffer); 4142 } 4143 } 4144 4145 if (mActiveTrack != 0) { 4146 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4147 result.append(buffer); 4148 RecordTrack::appendDumpHeader(result); 4149 mActiveTrack->dump(buffer, SIZE); 4150 result.append(buffer); 4151 4152 } 4153 write(fd, result.string(), result.size()); 4154} 4155 4156// AudioBufferProvider interface 4157status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4158{ 4159 size_t framesReq = buffer->frameCount; 4160 size_t framesReady = mFrameCount - mRsmpInIndex; 4161 int channelCount; 4162 4163 if (framesReady == 0) { 4164 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4165 if (mBytesRead <= 0) { 4166 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4167 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4168 // Force input into standby so that it tries to 4169 // recover at next read attempt 4170 inputStandBy(); 4171 usleep(kRecordThreadSleepUs); 4172 } 4173 buffer->raw = NULL; 4174 buffer->frameCount = 0; 4175 return NOT_ENOUGH_DATA; 4176 } 4177 mRsmpInIndex = 0; 4178 framesReady = mFrameCount; 4179 } 4180 4181 if (framesReq > framesReady) { 4182 framesReq = framesReady; 4183 } 4184 4185 if (mChannelCount == 1 && mReqChannelCount == 2) { 4186 channelCount = 1; 4187 } else { 4188 channelCount = 2; 4189 } 4190 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4191 buffer->frameCount = framesReq; 4192 return NO_ERROR; 4193} 4194 4195// AudioBufferProvider interface 4196void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4197{ 4198 mRsmpInIndex += buffer->frameCount; 4199 buffer->frameCount = 0; 4200} 4201 4202bool AudioFlinger::RecordThread::checkForNewParameters_l() 4203{ 4204 bool reconfig = false; 4205 4206 while (!mNewParameters.isEmpty()) { 4207 status_t status = NO_ERROR; 4208 String8 keyValuePair = mNewParameters[0]; 4209 AudioParameter param = AudioParameter(keyValuePair); 4210 int value; 4211 audio_format_t reqFormat = mFormat; 4212 uint32_t reqSamplingRate = mReqSampleRate; 4213 uint32_t reqChannelCount = mReqChannelCount; 4214 4215 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4216 reqSamplingRate = value; 4217 reconfig = true; 4218 } 4219 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4220 reqFormat = (audio_format_t) value; 4221 reconfig = true; 4222 } 4223 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4224 reqChannelCount = popcount(value); 4225 reconfig = true; 4226 } 4227 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4228 // do not accept frame count changes if tracks are open as the track buffer 4229 // size depends on frame count and correct behavior would not be guaranteed 4230 // if frame count is changed after track creation 4231 if (mActiveTrack != 0) { 4232 status = INVALID_OPERATION; 4233 } else { 4234 reconfig = true; 4235 } 4236 } 4237 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4238 // forward device change to effects that have requested to be 4239 // aware of attached audio device. 4240 for (size_t i = 0; i < mEffectChains.size(); i++) { 4241 mEffectChains[i]->setDevice_l(value); 4242 } 4243 4244 // store input device and output device but do not forward output device to audio HAL. 4245 // Note that status is ignored by the caller for output device 4246 // (see AudioFlinger::setParameters() 4247 if (audio_is_output_devices(value)) { 4248 mOutDevice = value; 4249 status = BAD_VALUE; 4250 } else { 4251 mInDevice = value; 4252 // disable AEC and NS if the device is a BT SCO headset supporting those 4253 // pre processings 4254 if (mTracks.size() > 0) { 4255 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4256 mAudioFlinger->btNrecIsOff(); 4257 for (size_t i = 0; i < mTracks.size(); i++) { 4258 sp<RecordTrack> track = mTracks[i]; 4259 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4260 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4261 } 4262 } 4263 } 4264 } 4265 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4266 mAudioSource != (audio_source_t)value) { 4267 // forward device change to effects that have requested to be 4268 // aware of attached audio device. 4269 for (size_t i = 0; i < mEffectChains.size(); i++) { 4270 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4271 } 4272 mAudioSource = (audio_source_t)value; 4273 } 4274 if (status == NO_ERROR) { 4275 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4276 keyValuePair.string()); 4277 if (status == INVALID_OPERATION) { 4278 inputStandBy(); 4279 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4280 keyValuePair.string()); 4281 } 4282 if (reconfig) { 4283 if (status == BAD_VALUE && 4284 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4285 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4286 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4287 <= (2 * reqSamplingRate)) && 4288 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4289 <= FCC_2 && 4290 (reqChannelCount <= FCC_2)) { 4291 status = NO_ERROR; 4292 } 4293 if (status == NO_ERROR) { 4294 readInputParameters(); 4295 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4296 } 4297 } 4298 } 4299 4300 mNewParameters.removeAt(0); 4301 4302 mParamStatus = status; 4303 mParamCond.signal(); 4304 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4305 // already timed out waiting for the status and will never signal the condition. 4306 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4307 } 4308 return reconfig; 4309} 4310 4311String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4312{ 4313 char *s; 4314 String8 out_s8 = String8(); 4315 4316 Mutex::Autolock _l(mLock); 4317 if (initCheck() != NO_ERROR) { 4318 return out_s8; 4319 } 4320 4321 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4322 out_s8 = String8(s); 4323 free(s); 4324 return out_s8; 4325} 4326 4327void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4328 AudioSystem::OutputDescriptor desc; 4329 void *param2 = NULL; 4330 4331 switch (event) { 4332 case AudioSystem::INPUT_OPENED: 4333 case AudioSystem::INPUT_CONFIG_CHANGED: 4334 desc.channels = mChannelMask; 4335 desc.samplingRate = mSampleRate; 4336 desc.format = mFormat; 4337 desc.frameCount = mFrameCount; 4338 desc.latency = 0; 4339 param2 = &desc; 4340 break; 4341 4342 case AudioSystem::INPUT_CLOSED: 4343 default: 4344 break; 4345 } 4346 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4347} 4348 4349void AudioFlinger::RecordThread::readInputParameters() 4350{ 4351 delete mRsmpInBuffer; 4352 // mRsmpInBuffer is always assigned a new[] below 4353 delete mRsmpOutBuffer; 4354 mRsmpOutBuffer = NULL; 4355 delete mResampler; 4356 mResampler = NULL; 4357 4358 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4359 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4360 mChannelCount = (uint16_t)popcount(mChannelMask); 4361 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4362 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4363 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4364 mFrameCount = mInputBytes / mFrameSize; 4365 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4366 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4367 4368 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4369 { 4370 int channelCount; 4371 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4372 // stereo to mono post process as the resampler always outputs stereo. 4373 if (mChannelCount == 1 && mReqChannelCount == 2) { 4374 channelCount = 1; 4375 } else { 4376 channelCount = 2; 4377 } 4378 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4379 mResampler->setSampleRate(mSampleRate); 4380 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4381 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4382 4383 // optmization: if mono to mono, alter input frame count as if we were inputing 4384 // stereo samples 4385 if (mChannelCount == 1 && mReqChannelCount == 1) { 4386 mFrameCount >>= 1; 4387 } 4388 4389 } 4390 mRsmpInIndex = mFrameCount; 4391} 4392 4393unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4394{ 4395 Mutex::Autolock _l(mLock); 4396 if (initCheck() != NO_ERROR) { 4397 return 0; 4398 } 4399 4400 return mInput->stream->get_input_frames_lost(mInput->stream); 4401} 4402 4403uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4404{ 4405 Mutex::Autolock _l(mLock); 4406 uint32_t result = 0; 4407 if (getEffectChain_l(sessionId) != 0) { 4408 result = EFFECT_SESSION; 4409 } 4410 4411 for (size_t i = 0; i < mTracks.size(); ++i) { 4412 if (sessionId == mTracks[i]->sessionId()) { 4413 result |= TRACK_SESSION; 4414 break; 4415 } 4416 } 4417 4418 return result; 4419} 4420 4421KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4422{ 4423 KeyedVector<int, bool> ids; 4424 Mutex::Autolock _l(mLock); 4425 for (size_t j = 0; j < mTracks.size(); ++j) { 4426 sp<RecordThread::RecordTrack> track = mTracks[j]; 4427 int sessionId = track->sessionId(); 4428 if (ids.indexOfKey(sessionId) < 0) { 4429 ids.add(sessionId, true); 4430 } 4431 } 4432 return ids; 4433} 4434 4435AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4436{ 4437 Mutex::Autolock _l(mLock); 4438 AudioStreamIn *input = mInput; 4439 mInput = NULL; 4440 return input; 4441} 4442 4443// this method must always be called either with ThreadBase mLock held or inside the thread loop 4444audio_stream_t* AudioFlinger::RecordThread::stream() const 4445{ 4446 if (mInput == NULL) { 4447 return NULL; 4448 } 4449 return &mInput->stream->common; 4450} 4451 4452status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4453{ 4454 // only one chain per input thread 4455 if (mEffectChains.size() != 0) { 4456 return INVALID_OPERATION; 4457 } 4458 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4459 4460 chain->setInBuffer(NULL); 4461 chain->setOutBuffer(NULL); 4462 4463 checkSuspendOnAddEffectChain_l(chain); 4464 4465 mEffectChains.add(chain); 4466 4467 return NO_ERROR; 4468} 4469 4470size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4471{ 4472 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4473 ALOGW_IF(mEffectChains.size() != 1, 4474 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4475 chain.get(), mEffectChains.size(), this); 4476 if (mEffectChains.size() == 1) { 4477 mEffectChains.removeAt(0); 4478 } 4479 return 0; 4480} 4481 4482}; // namespace android 4483