Threads.cpp revision 70949c47fbae3f836d15f040551d7631be3ed7c2
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Whether to use fast mixer 113static const enum { 114 FastMixer_Never, // never initialize or use: for debugging only 115 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 116 // normal mixer multiplier is 1 117 FastMixer_Static, // initialize if needed, then use all the time if initialized, 118 // multiplier is calculated based on min & max normal mixer buffer size 119 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 // FIXME for FastMixer_Dynamic: 122 // Supporting this option will require fixing HALs that can't handle large writes. 123 // For example, one HAL implementation returns an error from a large write, 124 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 125 // We could either fix the HAL implementations, or provide a wrapper that breaks 126 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 127} kUseFastMixer = FastMixer_Static; 128 129// Priorities for requestPriority 130static const int kPriorityAudioApp = 2; 131static const int kPriorityFastMixer = 3; 132 133// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 134// for the track. The client then sub-divides this into smaller buffers for its use. 135// Currently the client uses double-buffering by default, but doesn't tell us about that. 136// So for now we just assume that client is double-buffered. 137// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 138// N-buffering, so AudioFlinger could allocate the right amount of memory. 139// See the client's minBufCount and mNotificationFramesAct calculations for details. 140static const int kFastTrackMultiplier = 1; 141 142// ---------------------------------------------------------------------------- 143 144#ifdef ADD_BATTERY_DATA 145// To collect the amplifier usage 146static void addBatteryData(uint32_t params) { 147 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 148 if (service == NULL) { 149 // it already logged 150 return; 151 } 152 153 service->addBatteryData(params); 154} 155#endif 156 157 158// ---------------------------------------------------------------------------- 159// CPU Stats 160// ---------------------------------------------------------------------------- 161 162class CpuStats { 163public: 164 CpuStats(); 165 void sample(const String8 &title); 166#ifdef DEBUG_CPU_USAGE 167private: 168 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 169 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 170 171 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 172 173 int mCpuNum; // thread's current CPU number 174 int mCpukHz; // frequency of thread's current CPU in kHz 175#endif 176}; 177 178CpuStats::CpuStats() 179#ifdef DEBUG_CPU_USAGE 180 : mCpuNum(-1), mCpukHz(-1) 181#endif 182{ 183} 184 185void CpuStats::sample(const String8 &title) { 186#ifdef DEBUG_CPU_USAGE 187 // get current thread's delta CPU time in wall clock ns 188 double wcNs; 189 bool valid = mCpuUsage.sampleAndEnable(wcNs); 190 191 // record sample for wall clock statistics 192 if (valid) { 193 mWcStats.sample(wcNs); 194 } 195 196 // get the current CPU number 197 int cpuNum = sched_getcpu(); 198 199 // get the current CPU frequency in kHz 200 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 201 202 // check if either CPU number or frequency changed 203 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 204 mCpuNum = cpuNum; 205 mCpukHz = cpukHz; 206 // ignore sample for purposes of cycles 207 valid = false; 208 } 209 210 // if no change in CPU number or frequency, then record sample for cycle statistics 211 if (valid && mCpukHz > 0) { 212 double cycles = wcNs * cpukHz * 0.000001; 213 mHzStats.sample(cycles); 214 } 215 216 unsigned n = mWcStats.n(); 217 // mCpuUsage.elapsed() is expensive, so don't call it every loop 218 if ((n & 127) == 1) { 219 long long elapsed = mCpuUsage.elapsed(); 220 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 221 double perLoop = elapsed / (double) n; 222 double perLoop100 = perLoop * 0.01; 223 double perLoop1k = perLoop * 0.001; 224 double mean = mWcStats.mean(); 225 double stddev = mWcStats.stddev(); 226 double minimum = mWcStats.minimum(); 227 double maximum = mWcStats.maximum(); 228 double meanCycles = mHzStats.mean(); 229 double stddevCycles = mHzStats.stddev(); 230 double minCycles = mHzStats.minimum(); 231 double maxCycles = mHzStats.maximum(); 232 mCpuUsage.resetElapsed(); 233 mWcStats.reset(); 234 mHzStats.reset(); 235 ALOGD("CPU usage for %s over past %.1f secs\n" 236 " (%u mixer loops at %.1f mean ms per loop):\n" 237 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 238 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 239 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 240 title.string(), 241 elapsed * .000000001, n, perLoop * .000001, 242 mean * .001, 243 stddev * .001, 244 minimum * .001, 245 maximum * .001, 246 mean / perLoop100, 247 stddev / perLoop100, 248 minimum / perLoop100, 249 maximum / perLoop100, 250 meanCycles / perLoop1k, 251 stddevCycles / perLoop1k, 252 minCycles / perLoop1k, 253 maxCycles / perLoop1k); 254 255 } 256 } 257#endif 258}; 259 260// ---------------------------------------------------------------------------- 261// ThreadBase 262// ---------------------------------------------------------------------------- 263 264AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 265 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 266 : Thread(false /*canCallJava*/), 267 mType(type), 268 mAudioFlinger(audioFlinger), 269 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 270 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 271 mParamStatus(NO_ERROR), 272 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 273 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 274 // mName will be set by concrete (non-virtual) subclass 275 mDeathRecipient(new PMDeathRecipient(this)) 276{ 277} 278 279AudioFlinger::ThreadBase::~ThreadBase() 280{ 281 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 282 for (size_t i = 0; i < mConfigEvents.size(); i++) { 283 delete mConfigEvents[i]; 284 } 285 mConfigEvents.clear(); 286 287 mParamCond.broadcast(); 288 // do not lock the mutex in destructor 289 releaseWakeLock_l(); 290 if (mPowerManager != 0) { 291 sp<IBinder> binder = mPowerManager->asBinder(); 292 binder->unlinkToDeath(mDeathRecipient); 293 } 294} 295 296void AudioFlinger::ThreadBase::exit() 297{ 298 ALOGV("ThreadBase::exit"); 299 // do any cleanup required for exit to succeed 300 preExit(); 301 { 302 // This lock prevents the following race in thread (uniprocessor for illustration): 303 // if (!exitPending()) { 304 // // context switch from here to exit() 305 // // exit() calls requestExit(), what exitPending() observes 306 // // exit() calls signal(), which is dropped since no waiters 307 // // context switch back from exit() to here 308 // mWaitWorkCV.wait(...); 309 // // now thread is hung 310 // } 311 AutoMutex lock(mLock); 312 requestExit(); 313 mWaitWorkCV.broadcast(); 314 } 315 // When Thread::requestExitAndWait is made virtual and this method is renamed to 316 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 317 requestExitAndWait(); 318} 319 320status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 321{ 322 status_t status; 323 324 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 325 Mutex::Autolock _l(mLock); 326 327 mNewParameters.add(keyValuePairs); 328 mWaitWorkCV.signal(); 329 // wait condition with timeout in case the thread loop has exited 330 // before the request could be processed 331 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 332 status = mParamStatus; 333 mWaitWorkCV.signal(); 334 } else { 335 status = TIMED_OUT; 336 } 337 return status; 338} 339 340void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 341{ 342 Mutex::Autolock _l(mLock); 343 sendIoConfigEvent_l(event, param); 344} 345 346// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 347void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 348{ 349 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 350 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 351 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 352 param); 353 mWaitWorkCV.signal(); 354} 355 356// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 357void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 358{ 359 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 360 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 361 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 362 mConfigEvents.size(), pid, tid, prio); 363 mWaitWorkCV.signal(); 364} 365 366void AudioFlinger::ThreadBase::processConfigEvents() 367{ 368 mLock.lock(); 369 while (!mConfigEvents.isEmpty()) { 370 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 371 ConfigEvent *event = mConfigEvents[0]; 372 mConfigEvents.removeAt(0); 373 // release mLock before locking AudioFlinger mLock: lock order is always 374 // AudioFlinger then ThreadBase to avoid cross deadlock 375 mLock.unlock(); 376 switch(event->type()) { 377 case CFG_EVENT_PRIO: { 378 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 379 // FIXME Need to understand why this has be done asynchronously 380 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 381 true /*asynchronous*/); 382 if (err != 0) { 383 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 384 "error %d", 385 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 386 } 387 } break; 388 case CFG_EVENT_IO: { 389 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 390 mAudioFlinger->mLock.lock(); 391 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 392 mAudioFlinger->mLock.unlock(); 393 } break; 394 default: 395 ALOGE("processConfigEvents() unknown event type %d", event->type()); 396 break; 397 } 398 delete event; 399 mLock.lock(); 400 } 401 mLock.unlock(); 402} 403 404void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 405{ 406 const size_t SIZE = 256; 407 char buffer[SIZE]; 408 String8 result; 409 410 bool locked = AudioFlinger::dumpTryLock(mLock); 411 if (!locked) { 412 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 413 write(fd, buffer, strlen(buffer)); 414 } 415 416 snprintf(buffer, SIZE, "io handle: %d\n", mId); 417 result.append(buffer); 418 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 419 result.append(buffer); 420 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 421 result.append(buffer); 422 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 423 result.append(buffer); 424 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 425 result.append(buffer); 426 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 435 result.append(buffer); 436 437 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 438 result.append(buffer); 439 result.append(" Index Command"); 440 for (size_t i = 0; i < mNewParameters.size(); ++i) { 441 snprintf(buffer, SIZE, "\n %02d ", i); 442 result.append(buffer); 443 result.append(mNewParameters[i]); 444 } 445 446 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 447 result.append(buffer); 448 for (size_t i = 0; i < mConfigEvents.size(); i++) { 449 mConfigEvents[i]->dump(buffer, SIZE); 450 result.append(buffer); 451 } 452 result.append("\n"); 453 454 write(fd, result.string(), result.size()); 455 456 if (locked) { 457 mLock.unlock(); 458 } 459} 460 461void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 462{ 463 const size_t SIZE = 256; 464 char buffer[SIZE]; 465 String8 result; 466 467 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 468 write(fd, buffer, strlen(buffer)); 469 470 for (size_t i = 0; i < mEffectChains.size(); ++i) { 471 sp<EffectChain> chain = mEffectChains[i]; 472 if (chain != 0) { 473 chain->dump(fd, args); 474 } 475 } 476} 477 478void AudioFlinger::ThreadBase::acquireWakeLock() 479{ 480 Mutex::Autolock _l(mLock); 481 acquireWakeLock_l(); 482} 483 484void AudioFlinger::ThreadBase::acquireWakeLock_l() 485{ 486 if (mPowerManager == 0) { 487 // use checkService() to avoid blocking if power service is not up yet 488 sp<IBinder> binder = 489 defaultServiceManager()->checkService(String16("power")); 490 if (binder == 0) { 491 ALOGW("Thread %s cannot connect to the power manager service", mName); 492 } else { 493 mPowerManager = interface_cast<IPowerManager>(binder); 494 binder->linkToDeath(mDeathRecipient); 495 } 496 } 497 if (mPowerManager != 0) { 498 sp<IBinder> binder = new BBinder(); 499 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 500 binder, 501 String16(mName), 502 String16("media")); 503 if (status == NO_ERROR) { 504 mWakeLockToken = binder; 505 } 506 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 507 } 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock() 511{ 512 Mutex::Autolock _l(mLock); 513 releaseWakeLock_l(); 514} 515 516void AudioFlinger::ThreadBase::releaseWakeLock_l() 517{ 518 if (mWakeLockToken != 0) { 519 ALOGV("releaseWakeLock_l() %s", mName); 520 if (mPowerManager != 0) { 521 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 522 } 523 mWakeLockToken.clear(); 524 } 525} 526 527void AudioFlinger::ThreadBase::clearPowerManager() 528{ 529 Mutex::Autolock _l(mLock); 530 releaseWakeLock_l(); 531 mPowerManager.clear(); 532} 533 534void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 535{ 536 sp<ThreadBase> thread = mThread.promote(); 537 if (thread != 0) { 538 thread->clearPowerManager(); 539 } 540 ALOGW("power manager service died !!!"); 541} 542 543void AudioFlinger::ThreadBase::setEffectSuspended( 544 const effect_uuid_t *type, bool suspend, int sessionId) 545{ 546 Mutex::Autolock _l(mLock); 547 setEffectSuspended_l(type, suspend, sessionId); 548} 549 550void AudioFlinger::ThreadBase::setEffectSuspended_l( 551 const effect_uuid_t *type, bool suspend, int sessionId) 552{ 553 sp<EffectChain> chain = getEffectChain_l(sessionId); 554 if (chain != 0) { 555 if (type != NULL) { 556 chain->setEffectSuspended_l(type, suspend); 557 } else { 558 chain->setEffectSuspendedAll_l(suspend); 559 } 560 } 561 562 updateSuspendedSessions_l(type, suspend, sessionId); 563} 564 565void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 566{ 567 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 568 if (index < 0) { 569 return; 570 } 571 572 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 573 mSuspendedSessions.valueAt(index); 574 575 for (size_t i = 0; i < sessionEffects.size(); i++) { 576 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 577 for (int j = 0; j < desc->mRefCount; j++) { 578 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 579 chain->setEffectSuspendedAll_l(true); 580 } else { 581 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 582 desc->mType.timeLow); 583 chain->setEffectSuspended_l(&desc->mType, true); 584 } 585 } 586 } 587} 588 589void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 590 bool suspend, 591 int sessionId) 592{ 593 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 594 595 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 596 597 if (suspend) { 598 if (index >= 0) { 599 sessionEffects = mSuspendedSessions.valueAt(index); 600 } else { 601 mSuspendedSessions.add(sessionId, sessionEffects); 602 } 603 } else { 604 if (index < 0) { 605 return; 606 } 607 sessionEffects = mSuspendedSessions.valueAt(index); 608 } 609 610 611 int key = EffectChain::kKeyForSuspendAll; 612 if (type != NULL) { 613 key = type->timeLow; 614 } 615 index = sessionEffects.indexOfKey(key); 616 617 sp<SuspendedSessionDesc> desc; 618 if (suspend) { 619 if (index >= 0) { 620 desc = sessionEffects.valueAt(index); 621 } else { 622 desc = new SuspendedSessionDesc(); 623 if (type != NULL) { 624 desc->mType = *type; 625 } 626 sessionEffects.add(key, desc); 627 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 628 } 629 desc->mRefCount++; 630 } else { 631 if (index < 0) { 632 return; 633 } 634 desc = sessionEffects.valueAt(index); 635 if (--desc->mRefCount == 0) { 636 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 637 sessionEffects.removeItemsAt(index); 638 if (sessionEffects.isEmpty()) { 639 ALOGV("updateSuspendedSessions_l() restore removing session %d", 640 sessionId); 641 mSuspendedSessions.removeItem(sessionId); 642 } 643 } 644 } 645 if (!sessionEffects.isEmpty()) { 646 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 647 } 648} 649 650void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 651 bool enabled, 652 int sessionId) 653{ 654 Mutex::Autolock _l(mLock); 655 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 656} 657 658void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 659 bool enabled, 660 int sessionId) 661{ 662 if (mType != RECORD) { 663 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 664 // another session. This gives the priority to well behaved effect control panels 665 // and applications not using global effects. 666 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 667 // global effects 668 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 669 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 670 } 671 } 672 673 sp<EffectChain> chain = getEffectChain_l(sessionId); 674 if (chain != 0) { 675 chain->checkSuspendOnEffectEnabled(effect, enabled); 676 } 677} 678 679// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 680sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 681 const sp<AudioFlinger::Client>& client, 682 const sp<IEffectClient>& effectClient, 683 int32_t priority, 684 int sessionId, 685 effect_descriptor_t *desc, 686 int *enabled, 687 status_t *status 688 ) 689{ 690 sp<EffectModule> effect; 691 sp<EffectHandle> handle; 692 status_t lStatus; 693 sp<EffectChain> chain; 694 bool chainCreated = false; 695 bool effectCreated = false; 696 bool effectRegistered = false; 697 698 lStatus = initCheck(); 699 if (lStatus != NO_ERROR) { 700 ALOGW("createEffect_l() Audio driver not initialized."); 701 goto Exit; 702 } 703 704 // Do not allow effects with session ID 0 on direct output or duplicating threads 705 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 706 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 707 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 708 desc->name, sessionId); 709 lStatus = BAD_VALUE; 710 goto Exit; 711 } 712 // Only Pre processor effects are allowed on input threads and only on input threads 713 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 714 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 715 desc->name, desc->flags, mType); 716 lStatus = BAD_VALUE; 717 goto Exit; 718 } 719 720 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 721 722 { // scope for mLock 723 Mutex::Autolock _l(mLock); 724 725 // check for existing effect chain with the requested audio session 726 chain = getEffectChain_l(sessionId); 727 if (chain == 0) { 728 // create a new chain for this session 729 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 730 chain = new EffectChain(this, sessionId); 731 addEffectChain_l(chain); 732 chain->setStrategy(getStrategyForSession_l(sessionId)); 733 chainCreated = true; 734 } else { 735 effect = chain->getEffectFromDesc_l(desc); 736 } 737 738 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 739 740 if (effect == 0) { 741 int id = mAudioFlinger->nextUniqueId(); 742 // Check CPU and memory usage 743 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 744 if (lStatus != NO_ERROR) { 745 goto Exit; 746 } 747 effectRegistered = true; 748 // create a new effect module if none present in the chain 749 effect = new EffectModule(this, chain, desc, id, sessionId); 750 lStatus = effect->status(); 751 if (lStatus != NO_ERROR) { 752 goto Exit; 753 } 754 lStatus = chain->addEffect_l(effect); 755 if (lStatus != NO_ERROR) { 756 goto Exit; 757 } 758 effectCreated = true; 759 760 effect->setDevice(mOutDevice); 761 effect->setDevice(mInDevice); 762 effect->setMode(mAudioFlinger->getMode()); 763 effect->setAudioSource(mAudioSource); 764 } 765 // create effect handle and connect it to effect module 766 handle = new EffectHandle(effect, client, effectClient, priority); 767 lStatus = effect->addHandle(handle.get()); 768 if (enabled != NULL) { 769 *enabled = (int)effect->isEnabled(); 770 } 771 } 772 773Exit: 774 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 775 Mutex::Autolock _l(mLock); 776 if (effectCreated) { 777 chain->removeEffect_l(effect); 778 } 779 if (effectRegistered) { 780 AudioSystem::unregisterEffect(effect->id()); 781 } 782 if (chainCreated) { 783 removeEffectChain_l(chain); 784 } 785 handle.clear(); 786 } 787 788 if (status != NULL) { 789 *status = lStatus; 790 } 791 return handle; 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 795{ 796 Mutex::Autolock _l(mLock); 797 return getEffect_l(sessionId, effectId); 798} 799 800sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 801{ 802 sp<EffectChain> chain = getEffectChain_l(sessionId); 803 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 804} 805 806// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 807// PlaybackThread::mLock held 808status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 809{ 810 // check for existing effect chain with the requested audio session 811 int sessionId = effect->sessionId(); 812 sp<EffectChain> chain = getEffectChain_l(sessionId); 813 bool chainCreated = false; 814 815 if (chain == 0) { 816 // create a new chain for this session 817 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 818 chain = new EffectChain(this, sessionId); 819 addEffectChain_l(chain); 820 chain->setStrategy(getStrategyForSession_l(sessionId)); 821 chainCreated = true; 822 } 823 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 824 825 if (chain->getEffectFromId_l(effect->id()) != 0) { 826 ALOGW("addEffect_l() %p effect %s already present in chain %p", 827 this, effect->desc().name, chain.get()); 828 return BAD_VALUE; 829 } 830 831 status_t status = chain->addEffect_l(effect); 832 if (status != NO_ERROR) { 833 if (chainCreated) { 834 removeEffectChain_l(chain); 835 } 836 return status; 837 } 838 839 effect->setDevice(mOutDevice); 840 effect->setDevice(mInDevice); 841 effect->setMode(mAudioFlinger->getMode()); 842 effect->setAudioSource(mAudioSource); 843 return NO_ERROR; 844} 845 846void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 847 848 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 849 effect_descriptor_t desc = effect->desc(); 850 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 851 detachAuxEffect_l(effect->id()); 852 } 853 854 sp<EffectChain> chain = effect->chain().promote(); 855 if (chain != 0) { 856 // remove effect chain if removing last effect 857 if (chain->removeEffect_l(effect) == 0) { 858 removeEffectChain_l(chain); 859 } 860 } else { 861 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 862 } 863} 864 865void AudioFlinger::ThreadBase::lockEffectChains_l( 866 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 867{ 868 effectChains = mEffectChains; 869 for (size_t i = 0; i < mEffectChains.size(); i++) { 870 mEffectChains[i]->lock(); 871 } 872} 873 874void AudioFlinger::ThreadBase::unlockEffectChains( 875 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 876{ 877 for (size_t i = 0; i < effectChains.size(); i++) { 878 effectChains[i]->unlock(); 879 } 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 883{ 884 Mutex::Autolock _l(mLock); 885 return getEffectChain_l(sessionId); 886} 887 888sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 889{ 890 size_t size = mEffectChains.size(); 891 for (size_t i = 0; i < size; i++) { 892 if (mEffectChains[i]->sessionId() == sessionId) { 893 return mEffectChains[i]; 894 } 895 } 896 return 0; 897} 898 899void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 900{ 901 Mutex::Autolock _l(mLock); 902 size_t size = mEffectChains.size(); 903 for (size_t i = 0; i < size; i++) { 904 mEffectChains[i]->setMode_l(mode); 905 } 906} 907 908void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 909 EffectHandle *handle, 910 bool unpinIfLast) { 911 912 Mutex::Autolock _l(mLock); 913 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 914 // delete the effect module if removing last handle on it 915 if (effect->removeHandle(handle) == 0) { 916 if (!effect->isPinned() || unpinIfLast) { 917 removeEffect_l(effect); 918 AudioSystem::unregisterEffect(effect->id()); 919 } 920 } 921} 922 923// ---------------------------------------------------------------------------- 924// Playback 925// ---------------------------------------------------------------------------- 926 927AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 928 AudioStreamOut* output, 929 audio_io_handle_t id, 930 audio_devices_t device, 931 type_t type) 932 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 933 mNormalFrameCount(0), mMixBuffer(NULL), 934 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 935 // mStreamTypes[] initialized in constructor body 936 mOutput(output), 937 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 938 mMixerStatus(MIXER_IDLE), 939 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 940 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 941 mBytesRemaining(0), 942 mCurrentWriteLength(0), 943 mUseAsyncWrite(false), 944 mWriteBlocked(false), 945 mDraining(false), 946 mScreenState(AudioFlinger::mScreenState), 947 // index 0 is reserved for normal mixer's submix 948 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 949{ 950 snprintf(mName, kNameLength, "AudioOut_%X", id); 951 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 952 953 // Assumes constructor is called by AudioFlinger with it's mLock held, but 954 // it would be safer to explicitly pass initial masterVolume/masterMute as 955 // parameter. 956 // 957 // If the HAL we are using has support for master volume or master mute, 958 // then do not attenuate or mute during mixing (just leave the volume at 1.0 959 // and the mute set to false). 960 mMasterVolume = audioFlinger->masterVolume_l(); 961 mMasterMute = audioFlinger->masterMute_l(); 962 if (mOutput && mOutput->audioHwDev) { 963 if (mOutput->audioHwDev->canSetMasterVolume()) { 964 mMasterVolume = 1.0; 965 } 966 967 if (mOutput->audioHwDev->canSetMasterMute()) { 968 mMasterMute = false; 969 } 970 } 971 972 readOutputParameters(); 973 974 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 975 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 976 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 977 stream = (audio_stream_type_t) (stream + 1)) { 978 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 979 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 980 } 981 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 982 // because mAudioFlinger doesn't have one to copy from 983} 984 985AudioFlinger::PlaybackThread::~PlaybackThread() 986{ 987 mAudioFlinger->unregisterWriter(mNBLogWriter); 988 delete [] mAllocMixBuffer; 989} 990 991void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 992{ 993 dumpInternals(fd, args); 994 dumpTracks(fd, args); 995 dumpEffectChains(fd, args); 996} 997 998void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 999{ 1000 const size_t SIZE = 256; 1001 char buffer[SIZE]; 1002 String8 result; 1003 1004 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1005 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1006 const stream_type_t *st = &mStreamTypes[i]; 1007 if (i > 0) { 1008 result.appendFormat(", "); 1009 } 1010 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1011 if (st->mute) { 1012 result.append("M"); 1013 } 1014 } 1015 result.append("\n"); 1016 write(fd, result.string(), result.length()); 1017 result.clear(); 1018 1019 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1020 result.append(buffer); 1021 Track::appendDumpHeader(result); 1022 for (size_t i = 0; i < mTracks.size(); ++i) { 1023 sp<Track> track = mTracks[i]; 1024 if (track != 0) { 1025 track->dump(buffer, SIZE); 1026 result.append(buffer); 1027 } 1028 } 1029 1030 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1031 result.append(buffer); 1032 Track::appendDumpHeader(result); 1033 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1034 sp<Track> track = mActiveTracks[i].promote(); 1035 if (track != 0) { 1036 track->dump(buffer, SIZE); 1037 result.append(buffer); 1038 } 1039 } 1040 write(fd, result.string(), result.size()); 1041 1042 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1043 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1044 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1045 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1046} 1047 1048void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1049{ 1050 const size_t SIZE = 256; 1051 char buffer[SIZE]; 1052 String8 result; 1053 1054 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1055 result.append(buffer); 1056 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1057 result.append(buffer); 1058 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1059 ns2ms(systemTime() - mLastWriteTime)); 1060 result.append(buffer); 1061 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1062 result.append(buffer); 1063 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1064 result.append(buffer); 1065 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1066 result.append(buffer); 1067 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1068 result.append(buffer); 1069 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1070 result.append(buffer); 1071 write(fd, result.string(), result.size()); 1072 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1073 1074 dumpBase(fd, args); 1075} 1076 1077// Thread virtuals 1078status_t AudioFlinger::PlaybackThread::readyToRun() 1079{ 1080 status_t status = initCheck(); 1081 if (status == NO_ERROR) { 1082 ALOGI("AudioFlinger's thread %p ready to run", this); 1083 } else { 1084 ALOGE("No working audio driver found."); 1085 } 1086 return status; 1087} 1088 1089void AudioFlinger::PlaybackThread::onFirstRef() 1090{ 1091 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1092} 1093 1094// ThreadBase virtuals 1095void AudioFlinger::PlaybackThread::preExit() 1096{ 1097 ALOGV(" preExit()"); 1098 // FIXME this is using hard-coded strings but in the future, this functionality will be 1099 // converted to use audio HAL extensions required to support tunneling 1100 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1101} 1102 1103// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1104sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1105 const sp<AudioFlinger::Client>& client, 1106 audio_stream_type_t streamType, 1107 uint32_t sampleRate, 1108 audio_format_t format, 1109 audio_channel_mask_t channelMask, 1110 size_t frameCount, 1111 const sp<IMemory>& sharedBuffer, 1112 int sessionId, 1113 IAudioFlinger::track_flags_t *flags, 1114 pid_t tid, 1115 status_t *status) 1116{ 1117 sp<Track> track; 1118 status_t lStatus; 1119 1120 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1121 1122 // client expresses a preference for FAST, but we get the final say 1123 if (*flags & IAudioFlinger::TRACK_FAST) { 1124 if ( 1125 // not timed 1126 (!isTimed) && 1127 // either of these use cases: 1128 ( 1129 // use case 1: shared buffer with any frame count 1130 ( 1131 (sharedBuffer != 0) 1132 ) || 1133 // use case 2: callback handler and frame count is default or at least as large as HAL 1134 ( 1135 (tid != -1) && 1136 ((frameCount == 0) || 1137 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1138 ) 1139 ) && 1140 // PCM data 1141 audio_is_linear_pcm(format) && 1142 // mono or stereo 1143 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1144 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1145#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1146 // hardware sample rate 1147 (sampleRate == mSampleRate) && 1148#endif 1149 // normal mixer has an associated fast mixer 1150 hasFastMixer() && 1151 // there are sufficient fast track slots available 1152 (mFastTrackAvailMask != 0) 1153 // FIXME test that MixerThread for this fast track has a capable output HAL 1154 // FIXME add a permission test also? 1155 ) { 1156 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1157 if (frameCount == 0) { 1158 frameCount = mFrameCount * kFastTrackMultiplier; 1159 } 1160 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1161 frameCount, mFrameCount); 1162 } else { 1163 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1164 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1165 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1166 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1167 audio_is_linear_pcm(format), 1168 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1169 *flags &= ~IAudioFlinger::TRACK_FAST; 1170 // For compatibility with AudioTrack calculation, buffer depth is forced 1171 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1172 // This is probably too conservative, but legacy application code may depend on it. 1173 // If you change this calculation, also review the start threshold which is related. 1174 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1175 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1176 if (minBufCount < 2) { 1177 minBufCount = 2; 1178 } 1179 size_t minFrameCount = mNormalFrameCount * minBufCount; 1180 if (frameCount < minFrameCount) { 1181 frameCount = minFrameCount; 1182 } 1183 } 1184 } 1185 1186 if (mType == DIRECT) { 1187 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1188 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1189 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1190 "for output %p with format %d", 1191 sampleRate, format, channelMask, mOutput, mFormat); 1192 lStatus = BAD_VALUE; 1193 goto Exit; 1194 } 1195 } 1196 } else if (mType == OFFLOAD) { 1197 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1198 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1199 "for output %p with format %d", 1200 sampleRate, format, channelMask, mOutput, mFormat); 1201 lStatus = BAD_VALUE; 1202 goto Exit; 1203 } 1204 } else { 1205 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1206 ALOGE("createTrack_l() Bad parameter: format %d \"" 1207 "for output %p with format %d", 1208 format, mOutput, mFormat); 1209 lStatus = BAD_VALUE; 1210 goto Exit; 1211 } 1212 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1213 if (sampleRate > mSampleRate*2) { 1214 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1215 lStatus = BAD_VALUE; 1216 goto Exit; 1217 } 1218 } 1219 1220 lStatus = initCheck(); 1221 if (lStatus != NO_ERROR) { 1222 ALOGE("Audio driver not initialized."); 1223 goto Exit; 1224 } 1225 1226 { // scope for mLock 1227 Mutex::Autolock _l(mLock); 1228 1229 // all tracks in same audio session must share the same routing strategy otherwise 1230 // conflicts will happen when tracks are moved from one output to another by audio policy 1231 // manager 1232 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1233 for (size_t i = 0; i < mTracks.size(); ++i) { 1234 sp<Track> t = mTracks[i]; 1235 if (t != 0 && !t->isOutputTrack()) { 1236 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1237 if (sessionId == t->sessionId() && strategy != actual) { 1238 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1239 strategy, actual); 1240 lStatus = BAD_VALUE; 1241 goto Exit; 1242 } 1243 } 1244 } 1245 1246 if (!isTimed) { 1247 track = new Track(this, client, streamType, sampleRate, format, 1248 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1249 } else { 1250 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1251 channelMask, frameCount, sharedBuffer, sessionId); 1252 } 1253 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1254 lStatus = NO_MEMORY; 1255 goto Exit; 1256 } 1257 1258 mTracks.add(track); 1259 1260 sp<EffectChain> chain = getEffectChain_l(sessionId); 1261 if (chain != 0) { 1262 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1263 track->setMainBuffer(chain->inBuffer()); 1264 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1265 chain->incTrackCnt(); 1266 } 1267 1268 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1269 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1270 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1271 // so ask activity manager to do this on our behalf 1272 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1273 } 1274 } 1275 1276 lStatus = NO_ERROR; 1277 1278Exit: 1279 if (status) { 1280 *status = lStatus; 1281 } 1282 return track; 1283} 1284 1285uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1286{ 1287 return latency; 1288} 1289 1290uint32_t AudioFlinger::PlaybackThread::latency() const 1291{ 1292 Mutex::Autolock _l(mLock); 1293 return latency_l(); 1294} 1295uint32_t AudioFlinger::PlaybackThread::latency_l() const 1296{ 1297 if (initCheck() == NO_ERROR) { 1298 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1299 } else { 1300 return 0; 1301 } 1302} 1303 1304void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1305{ 1306 Mutex::Autolock _l(mLock); 1307 // Don't apply master volume in SW if our HAL can do it for us. 1308 if (mOutput && mOutput->audioHwDev && 1309 mOutput->audioHwDev->canSetMasterVolume()) { 1310 mMasterVolume = 1.0; 1311 } else { 1312 mMasterVolume = value; 1313 } 1314} 1315 1316void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1317{ 1318 Mutex::Autolock _l(mLock); 1319 // Don't apply master mute in SW if our HAL can do it for us. 1320 if (mOutput && mOutput->audioHwDev && 1321 mOutput->audioHwDev->canSetMasterMute()) { 1322 mMasterMute = false; 1323 } else { 1324 mMasterMute = muted; 1325 } 1326} 1327 1328void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1329{ 1330 Mutex::Autolock _l(mLock); 1331 mStreamTypes[stream].volume = value; 1332 signal_l(); 1333} 1334 1335void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1336{ 1337 Mutex::Autolock _l(mLock); 1338 mStreamTypes[stream].mute = muted; 1339 signal_l(); 1340} 1341 1342float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1343{ 1344 Mutex::Autolock _l(mLock); 1345 return mStreamTypes[stream].volume; 1346} 1347 1348// addTrack_l() must be called with ThreadBase::mLock held 1349status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1350{ 1351 status_t status = ALREADY_EXISTS; 1352 1353 // set retry count for buffer fill 1354 track->mRetryCount = kMaxTrackStartupRetries; 1355 if (mActiveTracks.indexOf(track) < 0) { 1356 // the track is newly added, make sure it fills up all its 1357 // buffers before playing. This is to ensure the client will 1358 // effectively get the latency it requested. 1359 if (!track->isOutputTrack()) { 1360 TrackBase::track_state state = track->mState; 1361 mLock.unlock(); 1362 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1363 mLock.lock(); 1364 // abort track was stopped/paused while we released the lock 1365 if (state != track->mState) { 1366 if (status == NO_ERROR) { 1367 mLock.unlock(); 1368 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1369 mLock.lock(); 1370 } 1371 return INVALID_OPERATION; 1372 } 1373 // abort if start is rejected by audio policy manager 1374 if (status != NO_ERROR) { 1375 return PERMISSION_DENIED; 1376 } 1377#ifdef ADD_BATTERY_DATA 1378 // to track the speaker usage 1379 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1380#endif 1381 } 1382 1383 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1384 track->mResetDone = false; 1385 track->mPresentationCompleteFrames = 0; 1386 mActiveTracks.add(track); 1387 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1388 if (chain != 0) { 1389 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1390 track->sessionId()); 1391 chain->incActiveTrackCnt(); 1392 } 1393 1394 status = NO_ERROR; 1395 } 1396 1397 ALOGV("mWaitWorkCV.broadcast"); 1398 mWaitWorkCV.broadcast(); 1399 1400 return status; 1401} 1402 1403bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1404{ 1405 track->terminate(); 1406 // active tracks are removed by threadLoop() 1407 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1408 track->mState = TrackBase::STOPPED; 1409 if (!trackActive) { 1410 removeTrack_l(track); 1411 } else if (track->isFastTrack() || track->isOffloaded()) { 1412 track->mState = TrackBase::STOPPING_1; 1413 } 1414 1415 return trackActive; 1416} 1417 1418void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1419{ 1420 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1421 mTracks.remove(track); 1422 deleteTrackName_l(track->name()); 1423 // redundant as track is about to be destroyed, for dumpsys only 1424 track->mName = -1; 1425 if (track->isFastTrack()) { 1426 int index = track->mFastIndex; 1427 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1428 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1429 mFastTrackAvailMask |= 1 << index; 1430 // redundant as track is about to be destroyed, for dumpsys only 1431 track->mFastIndex = -1; 1432 } 1433 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1434 if (chain != 0) { 1435 chain->decTrackCnt(); 1436 } 1437} 1438 1439void AudioFlinger::PlaybackThread::signal_l() 1440{ 1441 // Thread could be blocked waiting for async 1442 // so signal it to handle state changes immediately 1443 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1444 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1445 mSignalPending = true; 1446 mWaitWorkCV.signal(); 1447} 1448 1449String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1450{ 1451 Mutex::Autolock _l(mLock); 1452 if (initCheck() != NO_ERROR) { 1453 return String8(); 1454 } 1455 1456 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1457 const String8 out_s8(s); 1458 free(s); 1459 return out_s8; 1460} 1461 1462// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1463void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1464 AudioSystem::OutputDescriptor desc; 1465 void *param2 = NULL; 1466 1467 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1468 param); 1469 1470 switch (event) { 1471 case AudioSystem::OUTPUT_OPENED: 1472 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1473 desc.channelMask = mChannelMask; 1474 desc.samplingRate = mSampleRate; 1475 desc.format = mFormat; 1476 desc.frameCount = mNormalFrameCount; // FIXME see 1477 // AudioFlinger::frameCount(audio_io_handle_t) 1478 desc.latency = latency(); 1479 param2 = &desc; 1480 break; 1481 1482 case AudioSystem::STREAM_CONFIG_CHANGED: 1483 param2 = ¶m; 1484 case AudioSystem::OUTPUT_CLOSED: 1485 default: 1486 break; 1487 } 1488 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1489} 1490 1491void AudioFlinger::PlaybackThread::writeCallback() 1492{ 1493 ALOG_ASSERT(mCallbackThread != 0); 1494 mCallbackThread->setWriteBlocked(false); 1495} 1496 1497void AudioFlinger::PlaybackThread::drainCallback() 1498{ 1499 ALOG_ASSERT(mCallbackThread != 0); 1500 mCallbackThread->setDraining(false); 1501} 1502 1503void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1504{ 1505 Mutex::Autolock _l(mLock); 1506 mWriteBlocked = value; 1507 if (!value) { 1508 mWaitWorkCV.signal(); 1509 } 1510} 1511 1512void AudioFlinger::PlaybackThread::setDraining(bool value) 1513{ 1514 Mutex::Autolock _l(mLock); 1515 mDraining = value; 1516 if (!value) { 1517 mWaitWorkCV.signal(); 1518 } 1519} 1520 1521// static 1522int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1523 void *param, 1524 void *cookie) 1525{ 1526 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1527 ALOGV("asyncCallback() event %d", event); 1528 switch (event) { 1529 case STREAM_CBK_EVENT_WRITE_READY: 1530 me->writeCallback(); 1531 break; 1532 case STREAM_CBK_EVENT_DRAIN_READY: 1533 me->drainCallback(); 1534 break; 1535 default: 1536 ALOGW("asyncCallback() unknown event %d", event); 1537 break; 1538 } 1539 return 0; 1540} 1541 1542void AudioFlinger::PlaybackThread::readOutputParameters() 1543{ 1544 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1545 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1546 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1547 if (!audio_is_output_channel(mChannelMask)) { 1548 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1549 } 1550 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1551 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1552 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1553 } 1554 mChannelCount = popcount(mChannelMask); 1555 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1556 if (!audio_is_valid_format(mFormat)) { 1557 LOG_FATAL("HAL format %d not valid for output", mFormat); 1558 } 1559 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1560 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1561 mFormat); 1562 } 1563 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1564 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1565 mFrameCount = mBufferSize / mFrameSize; 1566 if (mFrameCount & 15) { 1567 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1568 mFrameCount); 1569 } 1570 1571 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1572 (mOutput->stream->set_callback != NULL)) { 1573 if (mOutput->stream->set_callback(mOutput->stream, 1574 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1575 mUseAsyncWrite = true; 1576 } 1577 } 1578 1579 // Calculate size of normal mix buffer relative to the HAL output buffer size 1580 double multiplier = 1.0; 1581 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1582 kUseFastMixer == FastMixer_Dynamic)) { 1583 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1584 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1585 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1586 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1587 maxNormalFrameCount = maxNormalFrameCount & ~15; 1588 if (maxNormalFrameCount < minNormalFrameCount) { 1589 maxNormalFrameCount = minNormalFrameCount; 1590 } 1591 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1592 if (multiplier <= 1.0) { 1593 multiplier = 1.0; 1594 } else if (multiplier <= 2.0) { 1595 if (2 * mFrameCount <= maxNormalFrameCount) { 1596 multiplier = 2.0; 1597 } else { 1598 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1599 } 1600 } else { 1601 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1602 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1603 // track, but we sometimes have to do this to satisfy the maximum frame count 1604 // constraint) 1605 // FIXME this rounding up should not be done if no HAL SRC 1606 uint32_t truncMult = (uint32_t) multiplier; 1607 if ((truncMult & 1)) { 1608 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1609 ++truncMult; 1610 } 1611 } 1612 multiplier = (double) truncMult; 1613 } 1614 } 1615 mNormalFrameCount = multiplier * mFrameCount; 1616 // round up to nearest 16 frames to satisfy AudioMixer 1617 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1618 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1619 mNormalFrameCount); 1620 1621 delete[] mAllocMixBuffer; 1622 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1623 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1624 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1625 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1626 1627 // force reconfiguration of effect chains and engines to take new buffer size and audio 1628 // parameters into account 1629 // Note that mLock is not held when readOutputParameters() is called from the constructor 1630 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1631 // matter. 1632 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1633 Vector< sp<EffectChain> > effectChains = mEffectChains; 1634 for (size_t i = 0; i < effectChains.size(); i ++) { 1635 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1636 } 1637} 1638 1639 1640status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1641{ 1642 if (halFrames == NULL || dspFrames == NULL) { 1643 return BAD_VALUE; 1644 } 1645 Mutex::Autolock _l(mLock); 1646 if (initCheck() != NO_ERROR) { 1647 return INVALID_OPERATION; 1648 } 1649 size_t framesWritten = mBytesWritten / mFrameSize; 1650 *halFrames = framesWritten; 1651 1652 if (isSuspended()) { 1653 // return an estimation of rendered frames when the output is suspended 1654 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1655 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1656 return NO_ERROR; 1657 } else { 1658 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1659 } 1660} 1661 1662uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1663{ 1664 Mutex::Autolock _l(mLock); 1665 uint32_t result = 0; 1666 if (getEffectChain_l(sessionId) != 0) { 1667 result = EFFECT_SESSION; 1668 } 1669 1670 for (size_t i = 0; i < mTracks.size(); ++i) { 1671 sp<Track> track = mTracks[i]; 1672 if (sessionId == track->sessionId() && !track->isInvalid()) { 1673 result |= TRACK_SESSION; 1674 break; 1675 } 1676 } 1677 1678 return result; 1679} 1680 1681uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1682{ 1683 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1684 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1685 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1686 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1687 } 1688 for (size_t i = 0; i < mTracks.size(); i++) { 1689 sp<Track> track = mTracks[i]; 1690 if (sessionId == track->sessionId() && !track->isInvalid()) { 1691 return AudioSystem::getStrategyForStream(track->streamType()); 1692 } 1693 } 1694 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1695} 1696 1697 1698AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1699{ 1700 Mutex::Autolock _l(mLock); 1701 return mOutput; 1702} 1703 1704AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1705{ 1706 Mutex::Autolock _l(mLock); 1707 AudioStreamOut *output = mOutput; 1708 mOutput = NULL; 1709 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1710 // must push a NULL and wait for ack 1711 mOutputSink.clear(); 1712 mPipeSink.clear(); 1713 mNormalSink.clear(); 1714 return output; 1715} 1716 1717// this method must always be called either with ThreadBase mLock held or inside the thread loop 1718audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1719{ 1720 if (mOutput == NULL) { 1721 return NULL; 1722 } 1723 return &mOutput->stream->common; 1724} 1725 1726uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1727{ 1728 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1729} 1730 1731status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1732{ 1733 if (!isValidSyncEvent(event)) { 1734 return BAD_VALUE; 1735 } 1736 1737 Mutex::Autolock _l(mLock); 1738 1739 for (size_t i = 0; i < mTracks.size(); ++i) { 1740 sp<Track> track = mTracks[i]; 1741 if (event->triggerSession() == track->sessionId()) { 1742 (void) track->setSyncEvent(event); 1743 return NO_ERROR; 1744 } 1745 } 1746 1747 return NAME_NOT_FOUND; 1748} 1749 1750bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1751{ 1752 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1753} 1754 1755void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1756 const Vector< sp<Track> >& tracksToRemove) 1757{ 1758 size_t count = tracksToRemove.size(); 1759 if (count) { 1760 for (size_t i = 0 ; i < count ; i++) { 1761 const sp<Track>& track = tracksToRemove.itemAt(i); 1762 if (!track->isOutputTrack()) { 1763 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1764#ifdef ADD_BATTERY_DATA 1765 // to track the speaker usage 1766 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1767#endif 1768 if (track->isTerminated()) { 1769 AudioSystem::releaseOutput(mId); 1770 } 1771 } 1772 } 1773 } 1774} 1775 1776void AudioFlinger::PlaybackThread::checkSilentMode_l() 1777{ 1778 if (!mMasterMute) { 1779 char value[PROPERTY_VALUE_MAX]; 1780 if (property_get("ro.audio.silent", value, "0") > 0) { 1781 char *endptr; 1782 unsigned long ul = strtoul(value, &endptr, 0); 1783 if (*endptr == '\0' && ul != 0) { 1784 ALOGD("Silence is golden"); 1785 // The setprop command will not allow a property to be changed after 1786 // the first time it is set, so we don't have to worry about un-muting. 1787 setMasterMute_l(true); 1788 } 1789 } 1790 } 1791} 1792 1793// shared by MIXER and DIRECT, overridden by DUPLICATING 1794ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1795{ 1796 // FIXME rewrite to reduce number of system calls 1797 mLastWriteTime = systemTime(); 1798 mInWrite = true; 1799 ssize_t bytesWritten; 1800 1801 // If an NBAIO sink is present, use it to write the normal mixer's submix 1802 if (mNormalSink != 0) { 1803#define mBitShift 2 // FIXME 1804 size_t count = mBytesRemaining >> mBitShift; 1805 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1806 ATRACE_BEGIN("write"); 1807 // update the setpoint when AudioFlinger::mScreenState changes 1808 uint32_t screenState = AudioFlinger::mScreenState; 1809 if (screenState != mScreenState) { 1810 mScreenState = screenState; 1811 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1812 if (pipe != NULL) { 1813 pipe->setAvgFrames((mScreenState & 1) ? 1814 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1815 } 1816 } 1817 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1818 ATRACE_END(); 1819 if (framesWritten > 0) { 1820 bytesWritten = framesWritten << mBitShift; 1821 } else { 1822 bytesWritten = framesWritten; 1823 } 1824 // otherwise use the HAL / AudioStreamOut directly 1825 } else { 1826 // Direct output and offload threads 1827 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1828 if (mUseAsyncWrite) { 1829 mWriteBlocked = true; 1830 ALOG_ASSERT(mCallbackThread != 0); 1831 mCallbackThread->setWriteBlocked(true); 1832 } 1833 bytesWritten = mOutput->stream->write(mOutput->stream, 1834 mMixBuffer + offset, mBytesRemaining); 1835 if (mUseAsyncWrite && 1836 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1837 // do not wait for async callback in case of error of full write 1838 mWriteBlocked = false; 1839 ALOG_ASSERT(mCallbackThread != 0); 1840 mCallbackThread->setWriteBlocked(false); 1841 } 1842 } 1843 1844 mNumWrites++; 1845 mInWrite = false; 1846 1847 return bytesWritten; 1848} 1849 1850void AudioFlinger::PlaybackThread::threadLoop_drain() 1851{ 1852 if (mOutput->stream->drain) { 1853 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1854 if (mUseAsyncWrite) { 1855 mDraining = true; 1856 ALOG_ASSERT(mCallbackThread != 0); 1857 mCallbackThread->setDraining(true); 1858 } 1859 mOutput->stream->drain(mOutput->stream, 1860 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1861 : AUDIO_DRAIN_ALL); 1862 } 1863} 1864 1865void AudioFlinger::PlaybackThread::threadLoop_exit() 1866{ 1867 // Default implementation has nothing to do 1868} 1869 1870/* 1871The derived values that are cached: 1872 - mixBufferSize from frame count * frame size 1873 - activeSleepTime from activeSleepTimeUs() 1874 - idleSleepTime from idleSleepTimeUs() 1875 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1876 - maxPeriod from frame count and sample rate (MIXER only) 1877 1878The parameters that affect these derived values are: 1879 - frame count 1880 - frame size 1881 - sample rate 1882 - device type: A2DP or not 1883 - device latency 1884 - format: PCM or not 1885 - active sleep time 1886 - idle sleep time 1887*/ 1888 1889void AudioFlinger::PlaybackThread::cacheParameters_l() 1890{ 1891 mixBufferSize = mNormalFrameCount * mFrameSize; 1892 activeSleepTime = activeSleepTimeUs(); 1893 idleSleepTime = idleSleepTimeUs(); 1894} 1895 1896void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1897{ 1898 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1899 this, streamType, mTracks.size()); 1900 Mutex::Autolock _l(mLock); 1901 1902 size_t size = mTracks.size(); 1903 for (size_t i = 0; i < size; i++) { 1904 sp<Track> t = mTracks[i]; 1905 if (t->streamType() == streamType) { 1906 t->invalidate(); 1907 } 1908 } 1909} 1910 1911status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1912{ 1913 int session = chain->sessionId(); 1914 int16_t *buffer = mMixBuffer; 1915 bool ownsBuffer = false; 1916 1917 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1918 if (session > 0) { 1919 // Only one effect chain can be present in direct output thread and it uses 1920 // the mix buffer as input 1921 if (mType != DIRECT) { 1922 size_t numSamples = mNormalFrameCount * mChannelCount; 1923 buffer = new int16_t[numSamples]; 1924 memset(buffer, 0, numSamples * sizeof(int16_t)); 1925 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1926 ownsBuffer = true; 1927 } 1928 1929 // Attach all tracks with same session ID to this chain. 1930 for (size_t i = 0; i < mTracks.size(); ++i) { 1931 sp<Track> track = mTracks[i]; 1932 if (session == track->sessionId()) { 1933 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1934 buffer); 1935 track->setMainBuffer(buffer); 1936 chain->incTrackCnt(); 1937 } 1938 } 1939 1940 // indicate all active tracks in the chain 1941 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1942 sp<Track> track = mActiveTracks[i].promote(); 1943 if (track == 0) { 1944 continue; 1945 } 1946 if (session == track->sessionId()) { 1947 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1948 chain->incActiveTrackCnt(); 1949 } 1950 } 1951 } 1952 1953 chain->setInBuffer(buffer, ownsBuffer); 1954 chain->setOutBuffer(mMixBuffer); 1955 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1956 // chains list in order to be processed last as it contains output stage effects 1957 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1958 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1959 // after track specific effects and before output stage 1960 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1961 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1962 // Effect chain for other sessions are inserted at beginning of effect 1963 // chains list to be processed before output mix effects. Relative order between other 1964 // sessions is not important 1965 size_t size = mEffectChains.size(); 1966 size_t i = 0; 1967 for (i = 0; i < size; i++) { 1968 if (mEffectChains[i]->sessionId() < session) { 1969 break; 1970 } 1971 } 1972 mEffectChains.insertAt(chain, i); 1973 checkSuspendOnAddEffectChain_l(chain); 1974 1975 return NO_ERROR; 1976} 1977 1978size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1979{ 1980 int session = chain->sessionId(); 1981 1982 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1983 1984 for (size_t i = 0; i < mEffectChains.size(); i++) { 1985 if (chain == mEffectChains[i]) { 1986 mEffectChains.removeAt(i); 1987 // detach all active tracks from the chain 1988 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1989 sp<Track> track = mActiveTracks[i].promote(); 1990 if (track == 0) { 1991 continue; 1992 } 1993 if (session == track->sessionId()) { 1994 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1995 chain.get(), session); 1996 chain->decActiveTrackCnt(); 1997 } 1998 } 1999 2000 // detach all tracks with same session ID from this chain 2001 for (size_t i = 0; i < mTracks.size(); ++i) { 2002 sp<Track> track = mTracks[i]; 2003 if (session == track->sessionId()) { 2004 track->setMainBuffer(mMixBuffer); 2005 chain->decTrackCnt(); 2006 } 2007 } 2008 break; 2009 } 2010 } 2011 return mEffectChains.size(); 2012} 2013 2014status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2015 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2016{ 2017 Mutex::Autolock _l(mLock); 2018 return attachAuxEffect_l(track, EffectId); 2019} 2020 2021status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2022 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2023{ 2024 status_t status = NO_ERROR; 2025 2026 if (EffectId == 0) { 2027 track->setAuxBuffer(0, NULL); 2028 } else { 2029 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2030 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2031 if (effect != 0) { 2032 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2033 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2034 } else { 2035 status = INVALID_OPERATION; 2036 } 2037 } else { 2038 status = BAD_VALUE; 2039 } 2040 } 2041 return status; 2042} 2043 2044void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2045{ 2046 for (size_t i = 0; i < mTracks.size(); ++i) { 2047 sp<Track> track = mTracks[i]; 2048 if (track->auxEffectId() == effectId) { 2049 attachAuxEffect_l(track, 0); 2050 } 2051 } 2052} 2053 2054bool AudioFlinger::PlaybackThread::threadLoop() 2055{ 2056 Vector< sp<Track> > tracksToRemove; 2057 2058 standbyTime = systemTime(); 2059 2060 // MIXER 2061 nsecs_t lastWarning = 0; 2062 2063 // DUPLICATING 2064 // FIXME could this be made local to while loop? 2065 writeFrames = 0; 2066 2067 cacheParameters_l(); 2068 sleepTime = idleSleepTime; 2069 2070 if (mType == MIXER) { 2071 sleepTimeShift = 0; 2072 } 2073 2074 CpuStats cpuStats; 2075 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2076 2077 acquireWakeLock(); 2078 2079 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2080 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2081 // and then that string will be logged at the next convenient opportunity. 2082 const char *logString = NULL; 2083 2084 while (!exitPending()) 2085 { 2086 cpuStats.sample(myName); 2087 2088 Vector< sp<EffectChain> > effectChains; 2089 2090 processConfigEvents(); 2091 2092 { // scope for mLock 2093 2094 Mutex::Autolock _l(mLock); 2095 2096 if (logString != NULL) { 2097 mNBLogWriter->logTimestamp(); 2098 mNBLogWriter->log(logString); 2099 logString = NULL; 2100 } 2101 2102 if (checkForNewParameters_l()) { 2103 cacheParameters_l(); 2104 } 2105 2106 saveOutputTracks(); 2107 2108 if (mSignalPending) { 2109 // A signal was raised while we were unlocked 2110 mSignalPending = false; 2111 } else if (waitingAsyncCallback_l()) { 2112 if (exitPending()) { 2113 break; 2114 } 2115 releaseWakeLock_l(); 2116 ALOGV("wait async completion"); 2117 mWaitWorkCV.wait(mLock); 2118 ALOGV("async completion/wake"); 2119 acquireWakeLock_l(); 2120 if (exitPending()) { 2121 break; 2122 } 2123 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2124 continue; 2125 } 2126 sleepTime = 0; 2127 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2128 isSuspended()) { 2129 // put audio hardware into standby after short delay 2130 if (shouldStandby_l()) { 2131 2132 threadLoop_standby(); 2133 2134 mStandby = true; 2135 } 2136 2137 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2138 // we're about to wait, flush the binder command buffer 2139 IPCThreadState::self()->flushCommands(); 2140 2141 clearOutputTracks(); 2142 2143 if (exitPending()) { 2144 break; 2145 } 2146 2147 releaseWakeLock_l(); 2148 // wait until we have something to do... 2149 ALOGV("%s going to sleep", myName.string()); 2150 mWaitWorkCV.wait(mLock); 2151 ALOGV("%s waking up", myName.string()); 2152 acquireWakeLock_l(); 2153 2154 mMixerStatus = MIXER_IDLE; 2155 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2156 mBytesWritten = 0; 2157 mBytesRemaining = 0; 2158 checkSilentMode_l(); 2159 2160 standbyTime = systemTime() + standbyDelay; 2161 sleepTime = idleSleepTime; 2162 if (mType == MIXER) { 2163 sleepTimeShift = 0; 2164 } 2165 2166 continue; 2167 } 2168 } 2169 2170 // mMixerStatusIgnoringFastTracks is also updated internally 2171 mMixerStatus = prepareTracks_l(&tracksToRemove); 2172 2173 // prevent any changes in effect chain list and in each effect chain 2174 // during mixing and effect process as the audio buffers could be deleted 2175 // or modified if an effect is created or deleted 2176 lockEffectChains_l(effectChains); 2177 } 2178 2179 if (mBytesRemaining == 0) { 2180 mCurrentWriteLength = 0; 2181 if (mMixerStatus == MIXER_TRACKS_READY) { 2182 // threadLoop_mix() sets mCurrentWriteLength 2183 threadLoop_mix(); 2184 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2185 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2186 // threadLoop_sleepTime sets sleepTime to 0 if data 2187 // must be written to HAL 2188 threadLoop_sleepTime(); 2189 if (sleepTime == 0) { 2190 mCurrentWriteLength = mixBufferSize; 2191 } 2192 } 2193 mBytesRemaining = mCurrentWriteLength; 2194 if (isSuspended()) { 2195 sleepTime = suspendSleepTimeUs(); 2196 // simulate write to HAL when suspended 2197 mBytesWritten += mixBufferSize; 2198 mBytesRemaining = 0; 2199 } 2200 2201 // only process effects if we're going to write 2202 if (sleepTime == 0) { 2203 for (size_t i = 0; i < effectChains.size(); i ++) { 2204 effectChains[i]->process_l(); 2205 } 2206 } 2207 } 2208 2209 // enable changes in effect chain 2210 unlockEffectChains(effectChains); 2211 2212 if (!waitingAsyncCallback()) { 2213 // sleepTime == 0 means we must write to audio hardware 2214 if (sleepTime == 0) { 2215 if (mBytesRemaining) { 2216 ssize_t ret = threadLoop_write(); 2217 if (ret < 0) { 2218 mBytesRemaining = 0; 2219 } else { 2220 mBytesWritten += ret; 2221 mBytesRemaining -= ret; 2222 } 2223 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2224 (mMixerStatus == MIXER_DRAIN_ALL)) { 2225 threadLoop_drain(); 2226 } 2227if (mType == MIXER) { 2228 // write blocked detection 2229 nsecs_t now = systemTime(); 2230 nsecs_t delta = now - mLastWriteTime; 2231 if (!mStandby && delta > maxPeriod) { 2232 mNumDelayedWrites++; 2233 if ((now - lastWarning) > kWarningThrottleNs) { 2234 ATRACE_NAME("underrun"); 2235 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2236 ns2ms(delta), mNumDelayedWrites, this); 2237 lastWarning = now; 2238 } 2239 } 2240} 2241 2242 mStandby = false; 2243 } else { 2244 usleep(sleepTime); 2245 } 2246 } 2247 2248 // Finally let go of removed track(s), without the lock held 2249 // since we can't guarantee the destructors won't acquire that 2250 // same lock. This will also mutate and push a new fast mixer state. 2251 threadLoop_removeTracks(tracksToRemove); 2252 tracksToRemove.clear(); 2253 2254 // FIXME I don't understand the need for this here; 2255 // it was in the original code but maybe the 2256 // assignment in saveOutputTracks() makes this unnecessary? 2257 clearOutputTracks(); 2258 2259 // Effect chains will be actually deleted here if they were removed from 2260 // mEffectChains list during mixing or effects processing 2261 effectChains.clear(); 2262 2263 // FIXME Note that the above .clear() is no longer necessary since effectChains 2264 // is now local to this block, but will keep it for now (at least until merge done). 2265 } 2266 2267 threadLoop_exit(); 2268 2269 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2270 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2271 // put output stream into standby mode 2272 if (!mStandby) { 2273 mOutput->stream->common.standby(&mOutput->stream->common); 2274 } 2275 } 2276 2277 releaseWakeLock(); 2278 2279 ALOGV("Thread %p type %d exiting", this, mType); 2280 return false; 2281} 2282 2283// removeTracks_l() must be called with ThreadBase::mLock held 2284void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2285{ 2286 size_t count = tracksToRemove.size(); 2287 if (count) { 2288 for (size_t i=0 ; i<count ; i++) { 2289 const sp<Track>& track = tracksToRemove.itemAt(i); 2290 mActiveTracks.remove(track); 2291 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2292 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2293 if (chain != 0) { 2294 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2295 track->sessionId()); 2296 chain->decActiveTrackCnt(); 2297 } 2298 if (track->isTerminated()) { 2299 removeTrack_l(track); 2300 } 2301 } 2302 } 2303 2304} 2305 2306// ---------------------------------------------------------------------------- 2307 2308AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2309 audio_io_handle_t id, audio_devices_t device, type_t type) 2310 : PlaybackThread(audioFlinger, output, id, device, type), 2311 // mAudioMixer below 2312 // mFastMixer below 2313 mFastMixerFutex(0) 2314 // mOutputSink below 2315 // mPipeSink below 2316 // mNormalSink below 2317{ 2318 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2319 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2320 "mFrameCount=%d, mNormalFrameCount=%d", 2321 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2322 mNormalFrameCount); 2323 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2324 2325 // FIXME - Current mixer implementation only supports stereo output 2326 if (mChannelCount != FCC_2) { 2327 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2328 } 2329 2330 // create an NBAIO sink for the HAL output stream, and negotiate 2331 mOutputSink = new AudioStreamOutSink(output->stream); 2332 size_t numCounterOffers = 0; 2333 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2334 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2335 ALOG_ASSERT(index == 0); 2336 2337 // initialize fast mixer depending on configuration 2338 bool initFastMixer; 2339 switch (kUseFastMixer) { 2340 case FastMixer_Never: 2341 initFastMixer = false; 2342 break; 2343 case FastMixer_Always: 2344 initFastMixer = true; 2345 break; 2346 case FastMixer_Static: 2347 case FastMixer_Dynamic: 2348 initFastMixer = mFrameCount < mNormalFrameCount; 2349 break; 2350 } 2351 if (initFastMixer) { 2352 2353 // create a MonoPipe to connect our submix to FastMixer 2354 NBAIO_Format format = mOutputSink->format(); 2355 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2356 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2357 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2358 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2359 const NBAIO_Format offers[1] = {format}; 2360 size_t numCounterOffers = 0; 2361 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2362 ALOG_ASSERT(index == 0); 2363 monoPipe->setAvgFrames((mScreenState & 1) ? 2364 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2365 mPipeSink = monoPipe; 2366 2367#ifdef TEE_SINK 2368 if (mTeeSinkOutputEnabled) { 2369 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2370 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2371 numCounterOffers = 0; 2372 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2373 ALOG_ASSERT(index == 0); 2374 mTeeSink = teeSink; 2375 PipeReader *teeSource = new PipeReader(*teeSink); 2376 numCounterOffers = 0; 2377 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2378 ALOG_ASSERT(index == 0); 2379 mTeeSource = teeSource; 2380 } 2381#endif 2382 2383 // create fast mixer and configure it initially with just one fast track for our submix 2384 mFastMixer = new FastMixer(); 2385 FastMixerStateQueue *sq = mFastMixer->sq(); 2386#ifdef STATE_QUEUE_DUMP 2387 sq->setObserverDump(&mStateQueueObserverDump); 2388 sq->setMutatorDump(&mStateQueueMutatorDump); 2389#endif 2390 FastMixerState *state = sq->begin(); 2391 FastTrack *fastTrack = &state->mFastTracks[0]; 2392 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2393 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2394 fastTrack->mVolumeProvider = NULL; 2395 fastTrack->mGeneration++; 2396 state->mFastTracksGen++; 2397 state->mTrackMask = 1; 2398 // fast mixer will use the HAL output sink 2399 state->mOutputSink = mOutputSink.get(); 2400 state->mOutputSinkGen++; 2401 state->mFrameCount = mFrameCount; 2402 state->mCommand = FastMixerState::COLD_IDLE; 2403 // already done in constructor initialization list 2404 //mFastMixerFutex = 0; 2405 state->mColdFutexAddr = &mFastMixerFutex; 2406 state->mColdGen++; 2407 state->mDumpState = &mFastMixerDumpState; 2408#ifdef TEE_SINK 2409 state->mTeeSink = mTeeSink.get(); 2410#endif 2411 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2412 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2413 sq->end(); 2414 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2415 2416 // start the fast mixer 2417 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2418 pid_t tid = mFastMixer->getTid(); 2419 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2420 if (err != 0) { 2421 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2422 kPriorityFastMixer, getpid_cached, tid, err); 2423 } 2424 2425#ifdef AUDIO_WATCHDOG 2426 // create and start the watchdog 2427 mAudioWatchdog = new AudioWatchdog(); 2428 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2429 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2430 tid = mAudioWatchdog->getTid(); 2431 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2432 if (err != 0) { 2433 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2434 kPriorityFastMixer, getpid_cached, tid, err); 2435 } 2436#endif 2437 2438 } else { 2439 mFastMixer = NULL; 2440 } 2441 2442 switch (kUseFastMixer) { 2443 case FastMixer_Never: 2444 case FastMixer_Dynamic: 2445 mNormalSink = mOutputSink; 2446 break; 2447 case FastMixer_Always: 2448 mNormalSink = mPipeSink; 2449 break; 2450 case FastMixer_Static: 2451 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2452 break; 2453 } 2454} 2455 2456AudioFlinger::MixerThread::~MixerThread() 2457{ 2458 if (mFastMixer != NULL) { 2459 FastMixerStateQueue *sq = mFastMixer->sq(); 2460 FastMixerState *state = sq->begin(); 2461 if (state->mCommand == FastMixerState::COLD_IDLE) { 2462 int32_t old = android_atomic_inc(&mFastMixerFutex); 2463 if (old == -1) { 2464 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2465 } 2466 } 2467 state->mCommand = FastMixerState::EXIT; 2468 sq->end(); 2469 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2470 mFastMixer->join(); 2471 // Though the fast mixer thread has exited, it's state queue is still valid. 2472 // We'll use that extract the final state which contains one remaining fast track 2473 // corresponding to our sub-mix. 2474 state = sq->begin(); 2475 ALOG_ASSERT(state->mTrackMask == 1); 2476 FastTrack *fastTrack = &state->mFastTracks[0]; 2477 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2478 delete fastTrack->mBufferProvider; 2479 sq->end(false /*didModify*/); 2480 delete mFastMixer; 2481#ifdef AUDIO_WATCHDOG 2482 if (mAudioWatchdog != 0) { 2483 mAudioWatchdog->requestExit(); 2484 mAudioWatchdog->requestExitAndWait(); 2485 mAudioWatchdog.clear(); 2486 } 2487#endif 2488 } 2489 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2490 delete mAudioMixer; 2491} 2492 2493 2494uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2495{ 2496 if (mFastMixer != NULL) { 2497 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2498 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2499 } 2500 return latency; 2501} 2502 2503 2504void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2505{ 2506 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2507} 2508 2509ssize_t AudioFlinger::MixerThread::threadLoop_write() 2510{ 2511 // FIXME we should only do one push per cycle; confirm this is true 2512 // Start the fast mixer if it's not already running 2513 if (mFastMixer != NULL) { 2514 FastMixerStateQueue *sq = mFastMixer->sq(); 2515 FastMixerState *state = sq->begin(); 2516 if (state->mCommand != FastMixerState::MIX_WRITE && 2517 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2518 if (state->mCommand == FastMixerState::COLD_IDLE) { 2519 int32_t old = android_atomic_inc(&mFastMixerFutex); 2520 if (old == -1) { 2521 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2522 } 2523#ifdef AUDIO_WATCHDOG 2524 if (mAudioWatchdog != 0) { 2525 mAudioWatchdog->resume(); 2526 } 2527#endif 2528 } 2529 state->mCommand = FastMixerState::MIX_WRITE; 2530 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2531 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2532 sq->end(); 2533 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2534 if (kUseFastMixer == FastMixer_Dynamic) { 2535 mNormalSink = mPipeSink; 2536 } 2537 } else { 2538 sq->end(false /*didModify*/); 2539 } 2540 } 2541 return PlaybackThread::threadLoop_write(); 2542} 2543 2544void AudioFlinger::MixerThread::threadLoop_standby() 2545{ 2546 // Idle the fast mixer if it's currently running 2547 if (mFastMixer != NULL) { 2548 FastMixerStateQueue *sq = mFastMixer->sq(); 2549 FastMixerState *state = sq->begin(); 2550 if (!(state->mCommand & FastMixerState::IDLE)) { 2551 state->mCommand = FastMixerState::COLD_IDLE; 2552 state->mColdFutexAddr = &mFastMixerFutex; 2553 state->mColdGen++; 2554 mFastMixerFutex = 0; 2555 sq->end(); 2556 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2557 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2558 if (kUseFastMixer == FastMixer_Dynamic) { 2559 mNormalSink = mOutputSink; 2560 } 2561#ifdef AUDIO_WATCHDOG 2562 if (mAudioWatchdog != 0) { 2563 mAudioWatchdog->pause(); 2564 } 2565#endif 2566 } else { 2567 sq->end(false /*didModify*/); 2568 } 2569 } 2570 PlaybackThread::threadLoop_standby(); 2571} 2572 2573// Empty implementation for standard mixer 2574// Overridden for offloaded playback 2575void AudioFlinger::PlaybackThread::flushOutput_l() 2576{ 2577} 2578 2579bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2580{ 2581 return false; 2582} 2583 2584bool AudioFlinger::PlaybackThread::shouldStandby_l() 2585{ 2586 return !mStandby; 2587} 2588 2589bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2590{ 2591 Mutex::Autolock _l(mLock); 2592 return waitingAsyncCallback_l(); 2593} 2594 2595// shared by MIXER and DIRECT, overridden by DUPLICATING 2596void AudioFlinger::PlaybackThread::threadLoop_standby() 2597{ 2598 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2599 mOutput->stream->common.standby(&mOutput->stream->common); 2600 if (mUseAsyncWrite != 0) { 2601 mWriteBlocked = false; 2602 mDraining = false; 2603 ALOG_ASSERT(mCallbackThread != 0); 2604 mCallbackThread->setWriteBlocked(false); 2605 mCallbackThread->setDraining(false); 2606 } 2607} 2608 2609void AudioFlinger::MixerThread::threadLoop_mix() 2610{ 2611 // obtain the presentation timestamp of the next output buffer 2612 int64_t pts; 2613 status_t status = INVALID_OPERATION; 2614 2615 if (mNormalSink != 0) { 2616 status = mNormalSink->getNextWriteTimestamp(&pts); 2617 } else { 2618 status = mOutputSink->getNextWriteTimestamp(&pts); 2619 } 2620 2621 if (status != NO_ERROR) { 2622 pts = AudioBufferProvider::kInvalidPTS; 2623 } 2624 2625 // mix buffers... 2626 mAudioMixer->process(pts); 2627 mCurrentWriteLength = mixBufferSize; 2628 // increase sleep time progressively when application underrun condition clears. 2629 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2630 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2631 // such that we would underrun the audio HAL. 2632 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2633 sleepTimeShift--; 2634 } 2635 sleepTime = 0; 2636 standbyTime = systemTime() + standbyDelay; 2637 //TODO: delay standby when effects have a tail 2638} 2639 2640void AudioFlinger::MixerThread::threadLoop_sleepTime() 2641{ 2642 // If no tracks are ready, sleep once for the duration of an output 2643 // buffer size, then write 0s to the output 2644 if (sleepTime == 0) { 2645 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2646 sleepTime = activeSleepTime >> sleepTimeShift; 2647 if (sleepTime < kMinThreadSleepTimeUs) { 2648 sleepTime = kMinThreadSleepTimeUs; 2649 } 2650 // reduce sleep time in case of consecutive application underruns to avoid 2651 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2652 // duration we would end up writing less data than needed by the audio HAL if 2653 // the condition persists. 2654 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2655 sleepTimeShift++; 2656 } 2657 } else { 2658 sleepTime = idleSleepTime; 2659 } 2660 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2661 memset (mMixBuffer, 0, mixBufferSize); 2662 sleepTime = 0; 2663 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2664 "anticipated start"); 2665 } 2666 // TODO add standby time extension fct of effect tail 2667} 2668 2669// prepareTracks_l() must be called with ThreadBase::mLock held 2670AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2671 Vector< sp<Track> > *tracksToRemove) 2672{ 2673 2674 mixer_state mixerStatus = MIXER_IDLE; 2675 // find out which tracks need to be processed 2676 size_t count = mActiveTracks.size(); 2677 size_t mixedTracks = 0; 2678 size_t tracksWithEffect = 0; 2679 // counts only _active_ fast tracks 2680 size_t fastTracks = 0; 2681 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2682 2683 float masterVolume = mMasterVolume; 2684 bool masterMute = mMasterMute; 2685 2686 if (masterMute) { 2687 masterVolume = 0; 2688 } 2689 // Delegate master volume control to effect in output mix effect chain if needed 2690 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2691 if (chain != 0) { 2692 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2693 chain->setVolume_l(&v, &v); 2694 masterVolume = (float)((v + (1 << 23)) >> 24); 2695 chain.clear(); 2696 } 2697 2698 // prepare a new state to push 2699 FastMixerStateQueue *sq = NULL; 2700 FastMixerState *state = NULL; 2701 bool didModify = false; 2702 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2703 if (mFastMixer != NULL) { 2704 sq = mFastMixer->sq(); 2705 state = sq->begin(); 2706 } 2707 2708 for (size_t i=0 ; i<count ; i++) { 2709 const sp<Track> t = mActiveTracks[i].promote(); 2710 if (t == 0) { 2711 continue; 2712 } 2713 2714 // this const just means the local variable doesn't change 2715 Track* const track = t.get(); 2716 2717 // process fast tracks 2718 if (track->isFastTrack()) { 2719 2720 // It's theoretically possible (though unlikely) for a fast track to be created 2721 // and then removed within the same normal mix cycle. This is not a problem, as 2722 // the track never becomes active so it's fast mixer slot is never touched. 2723 // The converse, of removing an (active) track and then creating a new track 2724 // at the identical fast mixer slot within the same normal mix cycle, 2725 // is impossible because the slot isn't marked available until the end of each cycle. 2726 int j = track->mFastIndex; 2727 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2728 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2729 FastTrack *fastTrack = &state->mFastTracks[j]; 2730 2731 // Determine whether the track is currently in underrun condition, 2732 // and whether it had a recent underrun. 2733 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2734 FastTrackUnderruns underruns = ftDump->mUnderruns; 2735 uint32_t recentFull = (underruns.mBitFields.mFull - 2736 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2737 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2738 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2739 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2740 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2741 uint32_t recentUnderruns = recentPartial + recentEmpty; 2742 track->mObservedUnderruns = underruns; 2743 // don't count underruns that occur while stopping or pausing 2744 // or stopped which can occur when flush() is called while active 2745 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2746 recentUnderruns > 0) { 2747 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2748 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2749 } 2750 2751 // This is similar to the state machine for normal tracks, 2752 // with a few modifications for fast tracks. 2753 bool isActive = true; 2754 switch (track->mState) { 2755 case TrackBase::STOPPING_1: 2756 // track stays active in STOPPING_1 state until first underrun 2757 if (recentUnderruns > 0 || track->isTerminated()) { 2758 track->mState = TrackBase::STOPPING_2; 2759 } 2760 break; 2761 case TrackBase::PAUSING: 2762 // ramp down is not yet implemented 2763 track->setPaused(); 2764 break; 2765 case TrackBase::RESUMING: 2766 // ramp up is not yet implemented 2767 track->mState = TrackBase::ACTIVE; 2768 break; 2769 case TrackBase::ACTIVE: 2770 if (recentFull > 0 || recentPartial > 0) { 2771 // track has provided at least some frames recently: reset retry count 2772 track->mRetryCount = kMaxTrackRetries; 2773 } 2774 if (recentUnderruns == 0) { 2775 // no recent underruns: stay active 2776 break; 2777 } 2778 // there has recently been an underrun of some kind 2779 if (track->sharedBuffer() == 0) { 2780 // were any of the recent underruns "empty" (no frames available)? 2781 if (recentEmpty == 0) { 2782 // no, then ignore the partial underruns as they are allowed indefinitely 2783 break; 2784 } 2785 // there has recently been an "empty" underrun: decrement the retry counter 2786 if (--(track->mRetryCount) > 0) { 2787 break; 2788 } 2789 // indicate to client process that the track was disabled because of underrun; 2790 // it will then automatically call start() when data is available 2791 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2792 // remove from active list, but state remains ACTIVE [confusing but true] 2793 isActive = false; 2794 break; 2795 } 2796 // fall through 2797 case TrackBase::STOPPING_2: 2798 case TrackBase::PAUSED: 2799 case TrackBase::STOPPED: 2800 case TrackBase::FLUSHED: // flush() while active 2801 // Check for presentation complete if track is inactive 2802 // We have consumed all the buffers of this track. 2803 // This would be incomplete if we auto-paused on underrun 2804 { 2805 size_t audioHALFrames = 2806 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2807 size_t framesWritten = mBytesWritten / mFrameSize; 2808 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2809 // track stays in active list until presentation is complete 2810 break; 2811 } 2812 } 2813 if (track->isStopping_2()) { 2814 track->mState = TrackBase::STOPPED; 2815 } 2816 if (track->isStopped()) { 2817 // Can't reset directly, as fast mixer is still polling this track 2818 // track->reset(); 2819 // So instead mark this track as needing to be reset after push with ack 2820 resetMask |= 1 << i; 2821 } 2822 isActive = false; 2823 break; 2824 case TrackBase::IDLE: 2825 default: 2826 LOG_FATAL("unexpected track state %d", track->mState); 2827 } 2828 2829 if (isActive) { 2830 // was it previously inactive? 2831 if (!(state->mTrackMask & (1 << j))) { 2832 ExtendedAudioBufferProvider *eabp = track; 2833 VolumeProvider *vp = track; 2834 fastTrack->mBufferProvider = eabp; 2835 fastTrack->mVolumeProvider = vp; 2836 fastTrack->mSampleRate = track->mSampleRate; 2837 fastTrack->mChannelMask = track->mChannelMask; 2838 fastTrack->mGeneration++; 2839 state->mTrackMask |= 1 << j; 2840 didModify = true; 2841 // no acknowledgement required for newly active tracks 2842 } 2843 // cache the combined master volume and stream type volume for fast mixer; this 2844 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2845 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2846 ++fastTracks; 2847 } else { 2848 // was it previously active? 2849 if (state->mTrackMask & (1 << j)) { 2850 fastTrack->mBufferProvider = NULL; 2851 fastTrack->mGeneration++; 2852 state->mTrackMask &= ~(1 << j); 2853 didModify = true; 2854 // If any fast tracks were removed, we must wait for acknowledgement 2855 // because we're about to decrement the last sp<> on those tracks. 2856 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2857 } else { 2858 LOG_FATAL("fast track %d should have been active", j); 2859 } 2860 tracksToRemove->add(track); 2861 // Avoids a misleading display in dumpsys 2862 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2863 } 2864 continue; 2865 } 2866 2867 { // local variable scope to avoid goto warning 2868 2869 audio_track_cblk_t* cblk = track->cblk(); 2870 2871 // The first time a track is added we wait 2872 // for all its buffers to be filled before processing it 2873 int name = track->name(); 2874 // make sure that we have enough frames to mix one full buffer. 2875 // enforce this condition only once to enable draining the buffer in case the client 2876 // app does not call stop() and relies on underrun to stop: 2877 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2878 // during last round 2879 size_t desiredFrames; 2880 uint32_t sr = track->sampleRate(); 2881 if (sr == mSampleRate) { 2882 desiredFrames = mNormalFrameCount; 2883 } else { 2884 // +1 for rounding and +1 for additional sample needed for interpolation 2885 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2886 // add frames already consumed but not yet released by the resampler 2887 // because cblk->framesReady() will include these frames 2888 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2889 // the minimum track buffer size is normally twice the number of frames necessary 2890 // to fill one buffer and the resampler should not leave more than one buffer worth 2891 // of unreleased frames after each pass, but just in case... 2892 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2893 } 2894 uint32_t minFrames = 1; 2895 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2896 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2897 minFrames = desiredFrames; 2898 } 2899 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2900 size_t framesReady; 2901 if (track->sharedBuffer() == 0) { 2902 framesReady = track->framesReady(); 2903 } else if (track->isStopped()) { 2904 framesReady = 0; 2905 } else { 2906 framesReady = 1; 2907 } 2908 if ((framesReady >= minFrames) && track->isReady() && 2909 !track->isPaused() && !track->isTerminated()) 2910 { 2911 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2912 2913 mixedTracks++; 2914 2915 // track->mainBuffer() != mMixBuffer means there is an effect chain 2916 // connected to the track 2917 chain.clear(); 2918 if (track->mainBuffer() != mMixBuffer) { 2919 chain = getEffectChain_l(track->sessionId()); 2920 // Delegate volume control to effect in track effect chain if needed 2921 if (chain != 0) { 2922 tracksWithEffect++; 2923 } else { 2924 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2925 "session %d", 2926 name, track->sessionId()); 2927 } 2928 } 2929 2930 2931 int param = AudioMixer::VOLUME; 2932 if (track->mFillingUpStatus == Track::FS_FILLED) { 2933 // no ramp for the first volume setting 2934 track->mFillingUpStatus = Track::FS_ACTIVE; 2935 if (track->mState == TrackBase::RESUMING) { 2936 track->mState = TrackBase::ACTIVE; 2937 param = AudioMixer::RAMP_VOLUME; 2938 } 2939 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2940 // FIXME should not make a decision based on mServer 2941 } else if (cblk->mServer != 0) { 2942 // If the track is stopped before the first frame was mixed, 2943 // do not apply ramp 2944 param = AudioMixer::RAMP_VOLUME; 2945 } 2946 2947 // compute volume for this track 2948 uint32_t vl, vr, va; 2949 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2950 vl = vr = va = 0; 2951 if (track->isPausing()) { 2952 track->setPaused(); 2953 } 2954 } else { 2955 2956 // read original volumes with volume control 2957 float typeVolume = mStreamTypes[track->streamType()].volume; 2958 float v = masterVolume * typeVolume; 2959 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2960 uint32_t vlr = proxy->getVolumeLR(); 2961 vl = vlr & 0xFFFF; 2962 vr = vlr >> 16; 2963 // track volumes come from shared memory, so can't be trusted and must be clamped 2964 if (vl > MAX_GAIN_INT) { 2965 ALOGV("Track left volume out of range: %04X", vl); 2966 vl = MAX_GAIN_INT; 2967 } 2968 if (vr > MAX_GAIN_INT) { 2969 ALOGV("Track right volume out of range: %04X", vr); 2970 vr = MAX_GAIN_INT; 2971 } 2972 // now apply the master volume and stream type volume 2973 vl = (uint32_t)(v * vl) << 12; 2974 vr = (uint32_t)(v * vr) << 12; 2975 // assuming master volume and stream type volume each go up to 1.0, 2976 // vl and vr are now in 8.24 format 2977 2978 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2979 // send level comes from shared memory and so may be corrupt 2980 if (sendLevel > MAX_GAIN_INT) { 2981 ALOGV("Track send level out of range: %04X", sendLevel); 2982 sendLevel = MAX_GAIN_INT; 2983 } 2984 va = (uint32_t)(v * sendLevel); 2985 } 2986 2987 // Delegate volume control to effect in track effect chain if needed 2988 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2989 // Do not ramp volume if volume is controlled by effect 2990 param = AudioMixer::VOLUME; 2991 track->mHasVolumeController = true; 2992 } else { 2993 // force no volume ramp when volume controller was just disabled or removed 2994 // from effect chain to avoid volume spike 2995 if (track->mHasVolumeController) { 2996 param = AudioMixer::VOLUME; 2997 } 2998 track->mHasVolumeController = false; 2999 } 3000 3001 // Convert volumes from 8.24 to 4.12 format 3002 // This additional clamping is needed in case chain->setVolume_l() overshot 3003 vl = (vl + (1 << 11)) >> 12; 3004 if (vl > MAX_GAIN_INT) { 3005 vl = MAX_GAIN_INT; 3006 } 3007 vr = (vr + (1 << 11)) >> 12; 3008 if (vr > MAX_GAIN_INT) { 3009 vr = MAX_GAIN_INT; 3010 } 3011 3012 if (va > MAX_GAIN_INT) { 3013 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3014 } 3015 3016 // XXX: these things DON'T need to be done each time 3017 mAudioMixer->setBufferProvider(name, track); 3018 mAudioMixer->enable(name); 3019 3020 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3021 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3022 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3023 mAudioMixer->setParameter( 3024 name, 3025 AudioMixer::TRACK, 3026 AudioMixer::FORMAT, (void *)track->format()); 3027 mAudioMixer->setParameter( 3028 name, 3029 AudioMixer::TRACK, 3030 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3031 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3032 uint32_t maxSampleRate = mSampleRate * 2; 3033 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3034 if (reqSampleRate == 0) { 3035 reqSampleRate = mSampleRate; 3036 } else if (reqSampleRate > maxSampleRate) { 3037 reqSampleRate = maxSampleRate; 3038 } 3039 mAudioMixer->setParameter( 3040 name, 3041 AudioMixer::RESAMPLE, 3042 AudioMixer::SAMPLE_RATE, 3043 (void *)reqSampleRate); 3044 mAudioMixer->setParameter( 3045 name, 3046 AudioMixer::TRACK, 3047 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3048 mAudioMixer->setParameter( 3049 name, 3050 AudioMixer::TRACK, 3051 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3052 3053 // reset retry count 3054 track->mRetryCount = kMaxTrackRetries; 3055 3056 // If one track is ready, set the mixer ready if: 3057 // - the mixer was not ready during previous round OR 3058 // - no other track is not ready 3059 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3060 mixerStatus != MIXER_TRACKS_ENABLED) { 3061 mixerStatus = MIXER_TRACKS_READY; 3062 } 3063 } else { 3064 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3065 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3066 } 3067 // clear effect chain input buffer if an active track underruns to avoid sending 3068 // previous audio buffer again to effects 3069 chain = getEffectChain_l(track->sessionId()); 3070 if (chain != 0) { 3071 chain->clearInputBuffer(); 3072 } 3073 3074 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3075 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3076 track->isStopped() || track->isPaused()) { 3077 // We have consumed all the buffers of this track. 3078 // Remove it from the list of active tracks. 3079 // TODO: use actual buffer filling status instead of latency when available from 3080 // audio HAL 3081 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3082 size_t framesWritten = mBytesWritten / mFrameSize; 3083 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3084 if (track->isStopped()) { 3085 track->reset(); 3086 } 3087 tracksToRemove->add(track); 3088 } 3089 } else { 3090 // No buffers for this track. Give it a few chances to 3091 // fill a buffer, then remove it from active list. 3092 if (--(track->mRetryCount) <= 0) { 3093 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3094 tracksToRemove->add(track); 3095 // indicate to client process that the track was disabled because of underrun; 3096 // it will then automatically call start() when data is available 3097 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3098 // If one track is not ready, mark the mixer also not ready if: 3099 // - the mixer was ready during previous round OR 3100 // - no other track is ready 3101 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3102 mixerStatus != MIXER_TRACKS_READY) { 3103 mixerStatus = MIXER_TRACKS_ENABLED; 3104 } 3105 } 3106 mAudioMixer->disable(name); 3107 } 3108 3109 } // local variable scope to avoid goto warning 3110track_is_ready: ; 3111 3112 } 3113 3114 // Push the new FastMixer state if necessary 3115 bool pauseAudioWatchdog = false; 3116 if (didModify) { 3117 state->mFastTracksGen++; 3118 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3119 if (kUseFastMixer == FastMixer_Dynamic && 3120 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3121 state->mCommand = FastMixerState::COLD_IDLE; 3122 state->mColdFutexAddr = &mFastMixerFutex; 3123 state->mColdGen++; 3124 mFastMixerFutex = 0; 3125 if (kUseFastMixer == FastMixer_Dynamic) { 3126 mNormalSink = mOutputSink; 3127 } 3128 // If we go into cold idle, need to wait for acknowledgement 3129 // so that fast mixer stops doing I/O. 3130 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3131 pauseAudioWatchdog = true; 3132 } 3133 } 3134 if (sq != NULL) { 3135 sq->end(didModify); 3136 sq->push(block); 3137 } 3138#ifdef AUDIO_WATCHDOG 3139 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3140 mAudioWatchdog->pause(); 3141 } 3142#endif 3143 3144 // Now perform the deferred reset on fast tracks that have stopped 3145 while (resetMask != 0) { 3146 size_t i = __builtin_ctz(resetMask); 3147 ALOG_ASSERT(i < count); 3148 resetMask &= ~(1 << i); 3149 sp<Track> t = mActiveTracks[i].promote(); 3150 if (t == 0) { 3151 continue; 3152 } 3153 Track* track = t.get(); 3154 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3155 track->reset(); 3156 } 3157 3158 // remove all the tracks that need to be... 3159 removeTracks_l(*tracksToRemove); 3160 3161 // mix buffer must be cleared if all tracks are connected to an 3162 // effect chain as in this case the mixer will not write to 3163 // mix buffer and track effects will accumulate into it 3164 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3165 (mixedTracks == 0 && fastTracks > 0))) { 3166 // FIXME as a performance optimization, should remember previous zero status 3167 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3168 } 3169 3170 // if any fast tracks, then status is ready 3171 mMixerStatusIgnoringFastTracks = mixerStatus; 3172 if (fastTracks > 0) { 3173 mixerStatus = MIXER_TRACKS_READY; 3174 } 3175 return mixerStatus; 3176} 3177 3178// getTrackName_l() must be called with ThreadBase::mLock held 3179int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3180{ 3181 return mAudioMixer->getTrackName(channelMask, sessionId); 3182} 3183 3184// deleteTrackName_l() must be called with ThreadBase::mLock held 3185void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3186{ 3187 ALOGV("remove track (%d) and delete from mixer", name); 3188 mAudioMixer->deleteTrackName(name); 3189} 3190 3191// checkForNewParameters_l() must be called with ThreadBase::mLock held 3192bool AudioFlinger::MixerThread::checkForNewParameters_l() 3193{ 3194 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3195 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3196 bool reconfig = false; 3197 3198 while (!mNewParameters.isEmpty()) { 3199 3200 if (mFastMixer != NULL) { 3201 FastMixerStateQueue *sq = mFastMixer->sq(); 3202 FastMixerState *state = sq->begin(); 3203 if (!(state->mCommand & FastMixerState::IDLE)) { 3204 previousCommand = state->mCommand; 3205 state->mCommand = FastMixerState::HOT_IDLE; 3206 sq->end(); 3207 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3208 } else { 3209 sq->end(false /*didModify*/); 3210 } 3211 } 3212 3213 status_t status = NO_ERROR; 3214 String8 keyValuePair = mNewParameters[0]; 3215 AudioParameter param = AudioParameter(keyValuePair); 3216 int value; 3217 3218 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3219 reconfig = true; 3220 } 3221 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3222 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3223 status = BAD_VALUE; 3224 } else { 3225 reconfig = true; 3226 } 3227 } 3228 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3229 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3230 status = BAD_VALUE; 3231 } else { 3232 reconfig = true; 3233 } 3234 } 3235 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3236 // do not accept frame count changes if tracks are open as the track buffer 3237 // size depends on frame count and correct behavior would not be guaranteed 3238 // if frame count is changed after track creation 3239 if (!mTracks.isEmpty()) { 3240 status = INVALID_OPERATION; 3241 } else { 3242 reconfig = true; 3243 } 3244 } 3245 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3246#ifdef ADD_BATTERY_DATA 3247 // when changing the audio output device, call addBatteryData to notify 3248 // the change 3249 if (mOutDevice != value) { 3250 uint32_t params = 0; 3251 // check whether speaker is on 3252 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3253 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3254 } 3255 3256 audio_devices_t deviceWithoutSpeaker 3257 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3258 // check if any other device (except speaker) is on 3259 if (value & deviceWithoutSpeaker ) { 3260 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3261 } 3262 3263 if (params != 0) { 3264 addBatteryData(params); 3265 } 3266 } 3267#endif 3268 3269 // forward device change to effects that have requested to be 3270 // aware of attached audio device. 3271 if (value != AUDIO_DEVICE_NONE) { 3272 mOutDevice = value; 3273 for (size_t i = 0; i < mEffectChains.size(); i++) { 3274 mEffectChains[i]->setDevice_l(mOutDevice); 3275 } 3276 } 3277 } 3278 3279 if (status == NO_ERROR) { 3280 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3281 keyValuePair.string()); 3282 if (!mStandby && status == INVALID_OPERATION) { 3283 mOutput->stream->common.standby(&mOutput->stream->common); 3284 mStandby = true; 3285 mBytesWritten = 0; 3286 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3287 keyValuePair.string()); 3288 } 3289 if (status == NO_ERROR && reconfig) { 3290 readOutputParameters(); 3291 delete mAudioMixer; 3292 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3293 for (size_t i = 0; i < mTracks.size() ; i++) { 3294 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3295 if (name < 0) { 3296 break; 3297 } 3298 mTracks[i]->mName = name; 3299 } 3300 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3301 } 3302 } 3303 3304 mNewParameters.removeAt(0); 3305 3306 mParamStatus = status; 3307 mParamCond.signal(); 3308 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3309 // already timed out waiting for the status and will never signal the condition. 3310 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3311 } 3312 3313 if (!(previousCommand & FastMixerState::IDLE)) { 3314 ALOG_ASSERT(mFastMixer != NULL); 3315 FastMixerStateQueue *sq = mFastMixer->sq(); 3316 FastMixerState *state = sq->begin(); 3317 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3318 state->mCommand = previousCommand; 3319 sq->end(); 3320 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3321 } 3322 3323 return reconfig; 3324} 3325 3326 3327void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3328{ 3329 const size_t SIZE = 256; 3330 char buffer[SIZE]; 3331 String8 result; 3332 3333 PlaybackThread::dumpInternals(fd, args); 3334 3335 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3336 result.append(buffer); 3337 write(fd, result.string(), result.size()); 3338 3339 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3340 const FastMixerDumpState copy(mFastMixerDumpState); 3341 copy.dump(fd); 3342 3343#ifdef STATE_QUEUE_DUMP 3344 // Similar for state queue 3345 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3346 observerCopy.dump(fd); 3347 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3348 mutatorCopy.dump(fd); 3349#endif 3350 3351#ifdef TEE_SINK 3352 // Write the tee output to a .wav file 3353 dumpTee(fd, mTeeSource, mId); 3354#endif 3355 3356#ifdef AUDIO_WATCHDOG 3357 if (mAudioWatchdog != 0) { 3358 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3359 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3360 wdCopy.dump(fd); 3361 } 3362#endif 3363} 3364 3365uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3366{ 3367 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3368} 3369 3370uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3371{ 3372 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3373} 3374 3375void AudioFlinger::MixerThread::cacheParameters_l() 3376{ 3377 PlaybackThread::cacheParameters_l(); 3378 3379 // FIXME: Relaxed timing because of a certain device that can't meet latency 3380 // Should be reduced to 2x after the vendor fixes the driver issue 3381 // increase threshold again due to low power audio mode. The way this warning 3382 // threshold is calculated and its usefulness should be reconsidered anyway. 3383 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3384} 3385 3386// ---------------------------------------------------------------------------- 3387 3388AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3389 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3390 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3391 // mLeftVolFloat, mRightVolFloat 3392{ 3393} 3394 3395AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3396 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3397 ThreadBase::type_t type) 3398 : PlaybackThread(audioFlinger, output, id, device, type) 3399 // mLeftVolFloat, mRightVolFloat 3400{ 3401} 3402 3403AudioFlinger::DirectOutputThread::~DirectOutputThread() 3404{ 3405} 3406 3407void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3408{ 3409 audio_track_cblk_t* cblk = track->cblk(); 3410 float left, right; 3411 3412 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3413 left = right = 0; 3414 } else { 3415 float typeVolume = mStreamTypes[track->streamType()].volume; 3416 float v = mMasterVolume * typeVolume; 3417 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3418 uint32_t vlr = proxy->getVolumeLR(); 3419 float v_clamped = v * (vlr & 0xFFFF); 3420 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3421 left = v_clamped/MAX_GAIN; 3422 v_clamped = v * (vlr >> 16); 3423 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3424 right = v_clamped/MAX_GAIN; 3425 } 3426 3427 if (lastTrack) { 3428 if (left != mLeftVolFloat || right != mRightVolFloat) { 3429 mLeftVolFloat = left; 3430 mRightVolFloat = right; 3431 3432 // Convert volumes from float to 8.24 3433 uint32_t vl = (uint32_t)(left * (1 << 24)); 3434 uint32_t vr = (uint32_t)(right * (1 << 24)); 3435 3436 // Delegate volume control to effect in track effect chain if needed 3437 // only one effect chain can be present on DirectOutputThread, so if 3438 // there is one, the track is connected to it 3439 if (!mEffectChains.isEmpty()) { 3440 mEffectChains[0]->setVolume_l(&vl, &vr); 3441 left = (float)vl / (1 << 24); 3442 right = (float)vr / (1 << 24); 3443 } 3444 if (mOutput->stream->set_volume) { 3445 mOutput->stream->set_volume(mOutput->stream, left, right); 3446 } 3447 } 3448 } 3449} 3450 3451 3452AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3453 Vector< sp<Track> > *tracksToRemove 3454) 3455{ 3456 size_t count = mActiveTracks.size(); 3457 mixer_state mixerStatus = MIXER_IDLE; 3458 3459 // find out which tracks need to be processed 3460 for (size_t i = 0; i < count; i++) { 3461 sp<Track> t = mActiveTracks[i].promote(); 3462 // The track died recently 3463 if (t == 0) { 3464 continue; 3465 } 3466 3467 Track* const track = t.get(); 3468 audio_track_cblk_t* cblk = track->cblk(); 3469 3470 // The first time a track is added we wait 3471 // for all its buffers to be filled before processing it 3472 uint32_t minFrames; 3473 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3474 minFrames = mNormalFrameCount; 3475 } else { 3476 minFrames = 1; 3477 } 3478 // Only consider last track started for volume and mixer state control. 3479 // This is the last entry in mActiveTracks unless a track underruns. 3480 // As we only care about the transition phase between two tracks on a 3481 // direct output, it is not a problem to ignore the underrun case. 3482 bool last = (i == (count - 1)); 3483 3484 if ((track->framesReady() >= minFrames) && track->isReady() && 3485 !track->isPaused() && !track->isTerminated()) 3486 { 3487 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3488 3489 if (track->mFillingUpStatus == Track::FS_FILLED) { 3490 track->mFillingUpStatus = Track::FS_ACTIVE; 3491 mLeftVolFloat = mRightVolFloat = 0; 3492 if (track->mState == TrackBase::RESUMING) { 3493 track->mState = TrackBase::ACTIVE; 3494 } 3495 } 3496 3497 // compute volume for this track 3498 processVolume_l(track, last); 3499 if (last) { 3500 // reset retry count 3501 track->mRetryCount = kMaxTrackRetriesDirect; 3502 mActiveTrack = t; 3503 mixerStatus = MIXER_TRACKS_READY; 3504 } 3505 } else { 3506 // clear effect chain input buffer if the last active track started underruns 3507 // to avoid sending previous audio buffer again to effects 3508 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3509 mEffectChains[0]->clearInputBuffer(); 3510 } 3511 3512 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3513 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3514 track->isStopped() || track->isPaused()) { 3515 // We have consumed all the buffers of this track. 3516 // Remove it from the list of active tracks. 3517 // TODO: implement behavior for compressed audio 3518 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3519 size_t framesWritten = mBytesWritten / mFrameSize; 3520 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3521 if (track->isStopped()) { 3522 track->reset(); 3523 } 3524 tracksToRemove->add(track); 3525 } 3526 } else { 3527 // No buffers for this track. Give it a few chances to 3528 // fill a buffer, then remove it from active list. 3529 // Only consider last track started for mixer state control 3530 if (--(track->mRetryCount) <= 0) { 3531 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3532 tracksToRemove->add(track); 3533 } else if (last) { 3534 mixerStatus = MIXER_TRACKS_ENABLED; 3535 } 3536 } 3537 } 3538 } 3539 3540 // remove all the tracks that need to be... 3541 removeTracks_l(*tracksToRemove); 3542 3543 return mixerStatus; 3544} 3545 3546void AudioFlinger::DirectOutputThread::threadLoop_mix() 3547{ 3548 size_t frameCount = mFrameCount; 3549 int8_t *curBuf = (int8_t *)mMixBuffer; 3550 // output audio to hardware 3551 while (frameCount) { 3552 AudioBufferProvider::Buffer buffer; 3553 buffer.frameCount = frameCount; 3554 mActiveTrack->getNextBuffer(&buffer); 3555 if (buffer.raw == NULL) { 3556 memset(curBuf, 0, frameCount * mFrameSize); 3557 break; 3558 } 3559 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3560 frameCount -= buffer.frameCount; 3561 curBuf += buffer.frameCount * mFrameSize; 3562 mActiveTrack->releaseBuffer(&buffer); 3563 } 3564 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3565 sleepTime = 0; 3566 standbyTime = systemTime() + standbyDelay; 3567 mActiveTrack.clear(); 3568} 3569 3570void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3571{ 3572 if (sleepTime == 0) { 3573 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3574 sleepTime = activeSleepTime; 3575 } else { 3576 sleepTime = idleSleepTime; 3577 } 3578 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3579 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3580 sleepTime = 0; 3581 } 3582} 3583 3584// getTrackName_l() must be called with ThreadBase::mLock held 3585int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3586 int sessionId) 3587{ 3588 return 0; 3589} 3590 3591// deleteTrackName_l() must be called with ThreadBase::mLock held 3592void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3593{ 3594} 3595 3596// checkForNewParameters_l() must be called with ThreadBase::mLock held 3597bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3598{ 3599 bool reconfig = false; 3600 3601 while (!mNewParameters.isEmpty()) { 3602 status_t status = NO_ERROR; 3603 String8 keyValuePair = mNewParameters[0]; 3604 AudioParameter param = AudioParameter(keyValuePair); 3605 int value; 3606 3607 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3608 // do not accept frame count changes if tracks are open as the track buffer 3609 // size depends on frame count and correct behavior would not be garantied 3610 // if frame count is changed after track creation 3611 if (!mTracks.isEmpty()) { 3612 status = INVALID_OPERATION; 3613 } else { 3614 reconfig = true; 3615 } 3616 } 3617 if (status == NO_ERROR) { 3618 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3619 keyValuePair.string()); 3620 if (!mStandby && status == INVALID_OPERATION) { 3621 mOutput->stream->common.standby(&mOutput->stream->common); 3622 mStandby = true; 3623 mBytesWritten = 0; 3624 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3625 keyValuePair.string()); 3626 } 3627 if (status == NO_ERROR && reconfig) { 3628 readOutputParameters(); 3629 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3630 } 3631 } 3632 3633 mNewParameters.removeAt(0); 3634 3635 mParamStatus = status; 3636 mParamCond.signal(); 3637 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3638 // already timed out waiting for the status and will never signal the condition. 3639 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3640 } 3641 return reconfig; 3642} 3643 3644uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3645{ 3646 uint32_t time; 3647 if (audio_is_linear_pcm(mFormat)) { 3648 time = PlaybackThread::activeSleepTimeUs(); 3649 } else { 3650 time = 10000; 3651 } 3652 return time; 3653} 3654 3655uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3656{ 3657 uint32_t time; 3658 if (audio_is_linear_pcm(mFormat)) { 3659 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3660 } else { 3661 time = 10000; 3662 } 3663 return time; 3664} 3665 3666uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3667{ 3668 uint32_t time; 3669 if (audio_is_linear_pcm(mFormat)) { 3670 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3671 } else { 3672 time = 10000; 3673 } 3674 return time; 3675} 3676 3677void AudioFlinger::DirectOutputThread::cacheParameters_l() 3678{ 3679 PlaybackThread::cacheParameters_l(); 3680 3681 // use shorter standby delay as on normal output to release 3682 // hardware resources as soon as possible 3683 standbyDelay = microseconds(activeSleepTime*2); 3684} 3685 3686// ---------------------------------------------------------------------------- 3687 3688AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3689 const sp<AudioFlinger::OffloadThread>& offloadThread) 3690 : Thread(false /*canCallJava*/), 3691 mOffloadThread(offloadThread), 3692 mWriteBlocked(false), 3693 mDraining(false) 3694{ 3695} 3696 3697AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3698{ 3699} 3700 3701void AudioFlinger::AsyncCallbackThread::onFirstRef() 3702{ 3703 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3704} 3705 3706bool AudioFlinger::AsyncCallbackThread::threadLoop() 3707{ 3708 while (!exitPending()) { 3709 bool writeBlocked; 3710 bool draining; 3711 3712 { 3713 Mutex::Autolock _l(mLock); 3714 mWaitWorkCV.wait(mLock); 3715 if (exitPending()) { 3716 break; 3717 } 3718 writeBlocked = mWriteBlocked; 3719 draining = mDraining; 3720 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3721 } 3722 { 3723 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3724 if (offloadThread != 0) { 3725 if (writeBlocked == false) { 3726 offloadThread->setWriteBlocked(false); 3727 } 3728 if (draining == false) { 3729 offloadThread->setDraining(false); 3730 } 3731 } 3732 } 3733 } 3734 return false; 3735} 3736 3737void AudioFlinger::AsyncCallbackThread::exit() 3738{ 3739 ALOGV("AsyncCallbackThread::exit"); 3740 Mutex::Autolock _l(mLock); 3741 requestExit(); 3742 mWaitWorkCV.broadcast(); 3743} 3744 3745void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3746{ 3747 Mutex::Autolock _l(mLock); 3748 mWriteBlocked = value; 3749 if (!value) { 3750 mWaitWorkCV.signal(); 3751 } 3752} 3753 3754void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3755{ 3756 Mutex::Autolock _l(mLock); 3757 mDraining = value; 3758 if (!value) { 3759 mWaitWorkCV.signal(); 3760 } 3761} 3762 3763 3764// ---------------------------------------------------------------------------- 3765AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3766 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3767 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3768 mHwPaused(false), 3769 mPausedBytesRemaining(0) 3770{ 3771 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3772} 3773 3774AudioFlinger::OffloadThread::~OffloadThread() 3775{ 3776 mPreviousTrack.clear(); 3777} 3778 3779void AudioFlinger::OffloadThread::threadLoop_exit() 3780{ 3781 if (mFlushPending || mHwPaused) { 3782 // If a flush is pending or track was paused, just discard buffered data 3783 flushHw_l(); 3784 } else { 3785 mMixerStatus = MIXER_DRAIN_ALL; 3786 threadLoop_drain(); 3787 } 3788 mCallbackThread->exit(); 3789 PlaybackThread::threadLoop_exit(); 3790} 3791 3792AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3793 Vector< sp<Track> > *tracksToRemove 3794) 3795{ 3796 ALOGV("OffloadThread::prepareTracks_l"); 3797 size_t count = mActiveTracks.size(); 3798 3799 mixer_state mixerStatus = MIXER_IDLE; 3800 if (mFlushPending) { 3801 flushHw_l(); 3802 mFlushPending = false; 3803 } 3804 // find out which tracks need to be processed 3805 for (size_t i = 0; i < count; i++) { 3806 sp<Track> t = mActiveTracks[i].promote(); 3807 // The track died recently 3808 if (t == 0) { 3809 continue; 3810 } 3811 Track* const track = t.get(); 3812 audio_track_cblk_t* cblk = track->cblk(); 3813 if (mPreviousTrack != NULL) { 3814 if (t != mPreviousTrack) { 3815 // Flush any data still being written from last track 3816 mBytesRemaining = 0; 3817 if (mPausedBytesRemaining) { 3818 // Last track was paused so we also need to flush saved 3819 // mixbuffer state and invalidate track so that it will 3820 // re-submit that unwritten data when it is next resumed 3821 mPausedBytesRemaining = 0; 3822 // Invalidate is a bit drastic - would be more efficient 3823 // to have a flag to tell client that some of the 3824 // previously written data was lost 3825 mPreviousTrack->invalidate(); 3826 } 3827 } 3828 } 3829 mPreviousTrack = t; 3830 bool last = (i == (count - 1)); 3831 if (track->isPausing()) { 3832 track->setPaused(); 3833 if (last) { 3834 if (!mHwPaused) { 3835 mOutput->stream->pause(mOutput->stream); 3836 mHwPaused = true; 3837 } 3838 // If we were part way through writing the mixbuffer to 3839 // the HAL we must save this until we resume 3840 // BUG - this will be wrong if a different track is made active, 3841 // in that case we want to discard the pending data in the 3842 // mixbuffer and tell the client to present it again when the 3843 // track is resumed 3844 mPausedWriteLength = mCurrentWriteLength; 3845 mPausedBytesRemaining = mBytesRemaining; 3846 mBytesRemaining = 0; // stop writing 3847 } 3848 tracksToRemove->add(track); 3849 } else if (track->framesReady() && track->isReady() && 3850 !track->isPaused() && !track->isTerminated()) { 3851 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3852 if (track->mFillingUpStatus == Track::FS_FILLED) { 3853 track->mFillingUpStatus = Track::FS_ACTIVE; 3854 mLeftVolFloat = mRightVolFloat = 0; 3855 if (track->mState == TrackBase::RESUMING) { 3856 if (mPausedBytesRemaining) { 3857 // Need to continue write that was interrupted 3858 mCurrentWriteLength = mPausedWriteLength; 3859 mBytesRemaining = mPausedBytesRemaining; 3860 mPausedBytesRemaining = 0; 3861 } 3862 track->mState = TrackBase::ACTIVE; 3863 } 3864 } 3865 3866 if (last) { 3867 if (mHwPaused) { 3868 mOutput->stream->resume(mOutput->stream); 3869 mHwPaused = false; 3870 // threadLoop_mix() will handle the case that we need to 3871 // resume an interrupted write 3872 } 3873 // reset retry count 3874 track->mRetryCount = kMaxTrackRetriesOffload; 3875 mActiveTrack = t; 3876 mixerStatus = MIXER_TRACKS_READY; 3877 } 3878 } else { 3879 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3880 if (track->isStopping_1()) { 3881 // Hardware buffer can hold a large amount of audio so we must 3882 // wait for all current track's data to drain before we say 3883 // that the track is stopped. 3884 if (mBytesRemaining == 0) { 3885 // Only start draining when all data in mixbuffer 3886 // has been written 3887 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3888 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3889 sleepTime = 0; 3890 standbyTime = systemTime() + standbyDelay; 3891 if (last) { 3892 mixerStatus = MIXER_DRAIN_TRACK; 3893 if (mHwPaused) { 3894 // It is possible to move from PAUSED to STOPPING_1 without 3895 // a resume so we must ensure hardware is running 3896 mOutput->stream->resume(mOutput->stream); 3897 mHwPaused = false; 3898 } 3899 } 3900 } 3901 } else if (track->isStopping_2()) { 3902 // Drain has completed, signal presentation complete 3903 if (!mDraining || !last) { 3904 track->mState = TrackBase::STOPPED; 3905 size_t audioHALFrames = 3906 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3907 size_t framesWritten = 3908 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3909 track->presentationComplete(framesWritten, audioHALFrames); 3910 track->reset(); 3911 tracksToRemove->add(track); 3912 } 3913 } else { 3914 // No buffers for this track. Give it a few chances to 3915 // fill a buffer, then remove it from active list. 3916 if (--(track->mRetryCount) <= 0) { 3917 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3918 track->name()); 3919 tracksToRemove->add(track); 3920 } else if (last){ 3921 mixerStatus = MIXER_TRACKS_ENABLED; 3922 } 3923 } 3924 } 3925 // compute volume for this track 3926 processVolume_l(track, last); 3927 } 3928 // remove all the tracks that need to be... 3929 removeTracks_l(*tracksToRemove); 3930 3931 return mixerStatus; 3932} 3933 3934void AudioFlinger::OffloadThread::flushOutput_l() 3935{ 3936 mFlushPending = true; 3937} 3938 3939// must be called with thread mutex locked 3940bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3941{ 3942 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3943 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3944 return true; 3945 } 3946 return false; 3947} 3948 3949// must be called with thread mutex locked 3950bool AudioFlinger::OffloadThread::shouldStandby_l() 3951{ 3952 bool TrackPaused = false; 3953 3954 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3955 // after a timeout and we will enter standby then. 3956 if (mTracks.size() > 0) { 3957 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3958 } 3959 3960 return !mStandby && !TrackPaused; 3961} 3962 3963 3964bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3965{ 3966 Mutex::Autolock _l(mLock); 3967 return waitingAsyncCallback_l(); 3968} 3969 3970void AudioFlinger::OffloadThread::flushHw_l() 3971{ 3972 mOutput->stream->flush(mOutput->stream); 3973 // Flush anything still waiting in the mixbuffer 3974 mCurrentWriteLength = 0; 3975 mBytesRemaining = 0; 3976 mPausedWriteLength = 0; 3977 mPausedBytesRemaining = 0; 3978 if (mUseAsyncWrite) { 3979 mWriteBlocked = false; 3980 mDraining = false; 3981 ALOG_ASSERT(mCallbackThread != 0); 3982 mCallbackThread->setWriteBlocked(false); 3983 mCallbackThread->setDraining(false); 3984 } 3985} 3986 3987// ---------------------------------------------------------------------------- 3988 3989AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3990 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3991 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3992 DUPLICATING), 3993 mWaitTimeMs(UINT_MAX) 3994{ 3995 addOutputTrack(mainThread); 3996} 3997 3998AudioFlinger::DuplicatingThread::~DuplicatingThread() 3999{ 4000 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4001 mOutputTracks[i]->destroy(); 4002 } 4003} 4004 4005void AudioFlinger::DuplicatingThread::threadLoop_mix() 4006{ 4007 // mix buffers... 4008 if (outputsReady(outputTracks)) { 4009 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4010 } else { 4011 memset(mMixBuffer, 0, mixBufferSize); 4012 } 4013 sleepTime = 0; 4014 writeFrames = mNormalFrameCount; 4015 mCurrentWriteLength = mixBufferSize; 4016 standbyTime = systemTime() + standbyDelay; 4017} 4018 4019void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4020{ 4021 if (sleepTime == 0) { 4022 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4023 sleepTime = activeSleepTime; 4024 } else { 4025 sleepTime = idleSleepTime; 4026 } 4027 } else if (mBytesWritten != 0) { 4028 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4029 writeFrames = mNormalFrameCount; 4030 memset(mMixBuffer, 0, mixBufferSize); 4031 } else { 4032 // flush remaining overflow buffers in output tracks 4033 writeFrames = 0; 4034 } 4035 sleepTime = 0; 4036 } 4037} 4038 4039ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4040{ 4041 for (size_t i = 0; i < outputTracks.size(); i++) { 4042 outputTracks[i]->write(mMixBuffer, writeFrames); 4043 } 4044 return (ssize_t)mixBufferSize; 4045} 4046 4047void AudioFlinger::DuplicatingThread::threadLoop_standby() 4048{ 4049 // DuplicatingThread implements standby by stopping all tracks 4050 for (size_t i = 0; i < outputTracks.size(); i++) { 4051 outputTracks[i]->stop(); 4052 } 4053} 4054 4055void AudioFlinger::DuplicatingThread::saveOutputTracks() 4056{ 4057 outputTracks = mOutputTracks; 4058} 4059 4060void AudioFlinger::DuplicatingThread::clearOutputTracks() 4061{ 4062 outputTracks.clear(); 4063} 4064 4065void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4066{ 4067 Mutex::Autolock _l(mLock); 4068 // FIXME explain this formula 4069 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4070 OutputTrack *outputTrack = new OutputTrack(thread, 4071 this, 4072 mSampleRate, 4073 mFormat, 4074 mChannelMask, 4075 frameCount); 4076 if (outputTrack->cblk() != NULL) { 4077 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4078 mOutputTracks.add(outputTrack); 4079 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4080 updateWaitTime_l(); 4081 } 4082} 4083 4084void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4085{ 4086 Mutex::Autolock _l(mLock); 4087 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4088 if (mOutputTracks[i]->thread() == thread) { 4089 mOutputTracks[i]->destroy(); 4090 mOutputTracks.removeAt(i); 4091 updateWaitTime_l(); 4092 return; 4093 } 4094 } 4095 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4096} 4097 4098// caller must hold mLock 4099void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4100{ 4101 mWaitTimeMs = UINT_MAX; 4102 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4103 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4104 if (strong != 0) { 4105 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4106 if (waitTimeMs < mWaitTimeMs) { 4107 mWaitTimeMs = waitTimeMs; 4108 } 4109 } 4110 } 4111} 4112 4113 4114bool AudioFlinger::DuplicatingThread::outputsReady( 4115 const SortedVector< sp<OutputTrack> > &outputTracks) 4116{ 4117 for (size_t i = 0; i < outputTracks.size(); i++) { 4118 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4119 if (thread == 0) { 4120 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4121 outputTracks[i].get()); 4122 return false; 4123 } 4124 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4125 // see note at standby() declaration 4126 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4127 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4128 thread.get()); 4129 return false; 4130 } 4131 } 4132 return true; 4133} 4134 4135uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4136{ 4137 return (mWaitTimeMs * 1000) / 2; 4138} 4139 4140void AudioFlinger::DuplicatingThread::cacheParameters_l() 4141{ 4142 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4143 updateWaitTime_l(); 4144 4145 MixerThread::cacheParameters_l(); 4146} 4147 4148// ---------------------------------------------------------------------------- 4149// Record 4150// ---------------------------------------------------------------------------- 4151 4152AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4153 AudioStreamIn *input, 4154 uint32_t sampleRate, 4155 audio_channel_mask_t channelMask, 4156 audio_io_handle_t id, 4157 audio_devices_t outDevice, 4158 audio_devices_t inDevice 4159#ifdef TEE_SINK 4160 , const sp<NBAIO_Sink>& teeSink 4161#endif 4162 ) : 4163 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4164 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4165 // mRsmpInIndex set by readInputParameters() 4166 mReqChannelCount(popcount(channelMask)), 4167 mReqSampleRate(sampleRate) 4168 // mBytesRead is only meaningful while active, and so is cleared in start() 4169 // (but might be better to also clear here for dump?) 4170#ifdef TEE_SINK 4171 , mTeeSink(teeSink) 4172#endif 4173{ 4174 snprintf(mName, kNameLength, "AudioIn_%X", id); 4175 4176 readInputParameters(); 4177 4178} 4179 4180 4181AudioFlinger::RecordThread::~RecordThread() 4182{ 4183 delete[] mRsmpInBuffer; 4184 delete mResampler; 4185 delete[] mRsmpOutBuffer; 4186} 4187 4188void AudioFlinger::RecordThread::onFirstRef() 4189{ 4190 run(mName, PRIORITY_URGENT_AUDIO); 4191} 4192 4193status_t AudioFlinger::RecordThread::readyToRun() 4194{ 4195 status_t status = initCheck(); 4196 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4197 return status; 4198} 4199 4200bool AudioFlinger::RecordThread::threadLoop() 4201{ 4202 AudioBufferProvider::Buffer buffer; 4203 sp<RecordTrack> activeTrack; 4204 Vector< sp<EffectChain> > effectChains; 4205 4206 nsecs_t lastWarning = 0; 4207 4208 inputStandBy(); 4209 acquireWakeLock(); 4210 4211 // used to verify we've read at least once before evaluating how many bytes were read 4212 bool readOnce = false; 4213 4214 // start recording 4215 while (!exitPending()) { 4216 4217 processConfigEvents(); 4218 4219 { // scope for mLock 4220 Mutex::Autolock _l(mLock); 4221 checkForNewParameters_l(); 4222 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4223 standby(); 4224 4225 if (exitPending()) { 4226 break; 4227 } 4228 4229 releaseWakeLock_l(); 4230 ALOGV("RecordThread: loop stopping"); 4231 // go to sleep 4232 mWaitWorkCV.wait(mLock); 4233 ALOGV("RecordThread: loop starting"); 4234 acquireWakeLock_l(); 4235 continue; 4236 } 4237 if (mActiveTrack != 0) { 4238 if (mActiveTrack->isTerminated()) { 4239 removeTrack_l(mActiveTrack); 4240 mActiveTrack.clear(); 4241 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4242 standby(); 4243 mActiveTrack.clear(); 4244 mStartStopCond.broadcast(); 4245 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4246 if (mReqChannelCount != mActiveTrack->channelCount()) { 4247 mActiveTrack.clear(); 4248 mStartStopCond.broadcast(); 4249 } else if (readOnce) { 4250 // record start succeeds only if first read from audio input 4251 // succeeds 4252 if (mBytesRead >= 0) { 4253 mActiveTrack->mState = TrackBase::ACTIVE; 4254 } else { 4255 mActiveTrack.clear(); 4256 } 4257 mStartStopCond.broadcast(); 4258 } 4259 mStandby = false; 4260 } 4261 } 4262 lockEffectChains_l(effectChains); 4263 } 4264 4265 if (mActiveTrack != 0) { 4266 if (mActiveTrack->mState != TrackBase::ACTIVE && 4267 mActiveTrack->mState != TrackBase::RESUMING) { 4268 unlockEffectChains(effectChains); 4269 usleep(kRecordThreadSleepUs); 4270 continue; 4271 } 4272 for (size_t i = 0; i < effectChains.size(); i ++) { 4273 effectChains[i]->process_l(); 4274 } 4275 4276 buffer.frameCount = mFrameCount; 4277 status_t status = mActiveTrack->getNextBuffer(&buffer); 4278 if (status == NO_ERROR) { 4279 readOnce = true; 4280 size_t framesOut = buffer.frameCount; 4281 if (mResampler == NULL) { 4282 // no resampling 4283 while (framesOut) { 4284 size_t framesIn = mFrameCount - mRsmpInIndex; 4285 if (framesIn) { 4286 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4287 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4288 mActiveTrack->mFrameSize; 4289 if (framesIn > framesOut) 4290 framesIn = framesOut; 4291 mRsmpInIndex += framesIn; 4292 framesOut -= framesIn; 4293 if (mChannelCount == mReqChannelCount) { 4294 memcpy(dst, src, framesIn * mFrameSize); 4295 } else { 4296 if (mChannelCount == 1) { 4297 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4298 (int16_t *)src, framesIn); 4299 } else { 4300 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4301 (int16_t *)src, framesIn); 4302 } 4303 } 4304 } 4305 if (framesOut && mFrameCount == mRsmpInIndex) { 4306 void *readInto; 4307 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4308 readInto = buffer.raw; 4309 framesOut = 0; 4310 } else { 4311 readInto = mRsmpInBuffer; 4312 mRsmpInIndex = 0; 4313 } 4314 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4315 mBufferSize); 4316 if (mBytesRead <= 0) { 4317 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4318 { 4319 ALOGE("Error reading audio input"); 4320 // Force input into standby so that it tries to 4321 // recover at next read attempt 4322 inputStandBy(); 4323 usleep(kRecordThreadSleepUs); 4324 } 4325 mRsmpInIndex = mFrameCount; 4326 framesOut = 0; 4327 buffer.frameCount = 0; 4328 } 4329#ifdef TEE_SINK 4330 else if (mTeeSink != 0) { 4331 (void) mTeeSink->write(readInto, 4332 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4333 } 4334#endif 4335 } 4336 } 4337 } else { 4338 // resampling 4339 4340 // resampler accumulates, but we only have one source track 4341 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4342 // alter output frame count as if we were expecting stereo samples 4343 if (mChannelCount == 1 && mReqChannelCount == 1) { 4344 framesOut >>= 1; 4345 } 4346 mResampler->resample(mRsmpOutBuffer, framesOut, 4347 this /* AudioBufferProvider* */); 4348 // ditherAndClamp() works as long as all buffers returned by 4349 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4350 if (mChannelCount == 2 && mReqChannelCount == 1) { 4351 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4352 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4353 // the resampler always outputs stereo samples: 4354 // do post stereo to mono conversion 4355 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4356 framesOut); 4357 } else { 4358 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4359 } 4360 // now done with mRsmpOutBuffer 4361 4362 } 4363 if (mFramestoDrop == 0) { 4364 mActiveTrack->releaseBuffer(&buffer); 4365 } else { 4366 if (mFramestoDrop > 0) { 4367 mFramestoDrop -= buffer.frameCount; 4368 if (mFramestoDrop <= 0) { 4369 clearSyncStartEvent(); 4370 } 4371 } else { 4372 mFramestoDrop += buffer.frameCount; 4373 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4374 mSyncStartEvent->isCancelled()) { 4375 ALOGW("Synced record %s, session %d, trigger session %d", 4376 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4377 mActiveTrack->sessionId(), 4378 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4379 clearSyncStartEvent(); 4380 } 4381 } 4382 } 4383 mActiveTrack->clearOverflow(); 4384 } 4385 // client isn't retrieving buffers fast enough 4386 else { 4387 if (!mActiveTrack->setOverflow()) { 4388 nsecs_t now = systemTime(); 4389 if ((now - lastWarning) > kWarningThrottleNs) { 4390 ALOGW("RecordThread: buffer overflow"); 4391 lastWarning = now; 4392 } 4393 } 4394 // Release the processor for a while before asking for a new buffer. 4395 // This will give the application more chance to read from the buffer and 4396 // clear the overflow. 4397 usleep(kRecordThreadSleepUs); 4398 } 4399 } 4400 // enable changes in effect chain 4401 unlockEffectChains(effectChains); 4402 effectChains.clear(); 4403 } 4404 4405 standby(); 4406 4407 { 4408 Mutex::Autolock _l(mLock); 4409 mActiveTrack.clear(); 4410 mStartStopCond.broadcast(); 4411 } 4412 4413 releaseWakeLock(); 4414 4415 ALOGV("RecordThread %p exiting", this); 4416 return false; 4417} 4418 4419void AudioFlinger::RecordThread::standby() 4420{ 4421 if (!mStandby) { 4422 inputStandBy(); 4423 mStandby = true; 4424 } 4425} 4426 4427void AudioFlinger::RecordThread::inputStandBy() 4428{ 4429 mInput->stream->common.standby(&mInput->stream->common); 4430} 4431 4432sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4433 const sp<AudioFlinger::Client>& client, 4434 uint32_t sampleRate, 4435 audio_format_t format, 4436 audio_channel_mask_t channelMask, 4437 size_t frameCount, 4438 int sessionId, 4439 IAudioFlinger::track_flags_t flags, 4440 pid_t tid, 4441 status_t *status) 4442{ 4443 sp<RecordTrack> track; 4444 status_t lStatus; 4445 4446 lStatus = initCheck(); 4447 if (lStatus != NO_ERROR) { 4448 ALOGE("Audio driver not initialized."); 4449 goto Exit; 4450 } 4451 4452 // FIXME use flags and tid similar to createTrack_l() 4453 4454 { // scope for mLock 4455 Mutex::Autolock _l(mLock); 4456 4457 track = new RecordTrack(this, client, sampleRate, 4458 format, channelMask, frameCount, sessionId); 4459 4460 if (track->getCblk() == 0) { 4461 lStatus = NO_MEMORY; 4462 goto Exit; 4463 } 4464 mTracks.add(track); 4465 4466 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4467 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4468 mAudioFlinger->btNrecIsOff(); 4469 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4470 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4471 } 4472 lStatus = NO_ERROR; 4473 4474Exit: 4475 if (status) { 4476 *status = lStatus; 4477 } 4478 return track; 4479} 4480 4481status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4482 AudioSystem::sync_event_t event, 4483 int triggerSession) 4484{ 4485 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4486 sp<ThreadBase> strongMe = this; 4487 status_t status = NO_ERROR; 4488 4489 if (event == AudioSystem::SYNC_EVENT_NONE) { 4490 clearSyncStartEvent(); 4491 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4492 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4493 triggerSession, 4494 recordTrack->sessionId(), 4495 syncStartEventCallback, 4496 this); 4497 // Sync event can be cancelled by the trigger session if the track is not in a 4498 // compatible state in which case we start record immediately 4499 if (mSyncStartEvent->isCancelled()) { 4500 clearSyncStartEvent(); 4501 } else { 4502 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4503 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4504 } 4505 } 4506 4507 { 4508 AutoMutex lock(mLock); 4509 if (mActiveTrack != 0) { 4510 if (recordTrack != mActiveTrack.get()) { 4511 status = -EBUSY; 4512 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4513 mActiveTrack->mState = TrackBase::ACTIVE; 4514 } 4515 return status; 4516 } 4517 4518 recordTrack->mState = TrackBase::IDLE; 4519 mActiveTrack = recordTrack; 4520 mLock.unlock(); 4521 status_t status = AudioSystem::startInput(mId); 4522 mLock.lock(); 4523 if (status != NO_ERROR) { 4524 mActiveTrack.clear(); 4525 clearSyncStartEvent(); 4526 return status; 4527 } 4528 mRsmpInIndex = mFrameCount; 4529 mBytesRead = 0; 4530 if (mResampler != NULL) { 4531 mResampler->reset(); 4532 } 4533 mActiveTrack->mState = TrackBase::RESUMING; 4534 // signal thread to start 4535 ALOGV("Signal record thread"); 4536 mWaitWorkCV.broadcast(); 4537 // do not wait for mStartStopCond if exiting 4538 if (exitPending()) { 4539 mActiveTrack.clear(); 4540 status = INVALID_OPERATION; 4541 goto startError; 4542 } 4543 mStartStopCond.wait(mLock); 4544 if (mActiveTrack == 0) { 4545 ALOGV("Record failed to start"); 4546 status = BAD_VALUE; 4547 goto startError; 4548 } 4549 ALOGV("Record started OK"); 4550 return status; 4551 } 4552 4553startError: 4554 AudioSystem::stopInput(mId); 4555 clearSyncStartEvent(); 4556 return status; 4557} 4558 4559void AudioFlinger::RecordThread::clearSyncStartEvent() 4560{ 4561 if (mSyncStartEvent != 0) { 4562 mSyncStartEvent->cancel(); 4563 } 4564 mSyncStartEvent.clear(); 4565 mFramestoDrop = 0; 4566} 4567 4568void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4569{ 4570 sp<SyncEvent> strongEvent = event.promote(); 4571 4572 if (strongEvent != 0) { 4573 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4574 me->handleSyncStartEvent(strongEvent); 4575 } 4576} 4577 4578void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4579{ 4580 if (event == mSyncStartEvent) { 4581 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4582 // from audio HAL 4583 mFramestoDrop = mFrameCount * 2; 4584 } 4585} 4586 4587bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4588 ALOGV("RecordThread::stop"); 4589 AutoMutex _l(mLock); 4590 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4591 return false; 4592 } 4593 recordTrack->mState = TrackBase::PAUSING; 4594 // do not wait for mStartStopCond if exiting 4595 if (exitPending()) { 4596 return true; 4597 } 4598 mStartStopCond.wait(mLock); 4599 // if we have been restarted, recordTrack == mActiveTrack.get() here 4600 if (exitPending() || recordTrack != mActiveTrack.get()) { 4601 ALOGV("Record stopped OK"); 4602 return true; 4603 } 4604 return false; 4605} 4606 4607bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4608{ 4609 return false; 4610} 4611 4612status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4613{ 4614#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4615 if (!isValidSyncEvent(event)) { 4616 return BAD_VALUE; 4617 } 4618 4619 int eventSession = event->triggerSession(); 4620 status_t ret = NAME_NOT_FOUND; 4621 4622 Mutex::Autolock _l(mLock); 4623 4624 for (size_t i = 0; i < mTracks.size(); i++) { 4625 sp<RecordTrack> track = mTracks[i]; 4626 if (eventSession == track->sessionId()) { 4627 (void) track->setSyncEvent(event); 4628 ret = NO_ERROR; 4629 } 4630 } 4631 return ret; 4632#else 4633 return BAD_VALUE; 4634#endif 4635} 4636 4637// destroyTrack_l() must be called with ThreadBase::mLock held 4638void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4639{ 4640 track->terminate(); 4641 track->mState = TrackBase::STOPPED; 4642 // active tracks are removed by threadLoop() 4643 if (mActiveTrack != track) { 4644 removeTrack_l(track); 4645 } 4646} 4647 4648void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4649{ 4650 mTracks.remove(track); 4651 // need anything related to effects here? 4652} 4653 4654void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4655{ 4656 dumpInternals(fd, args); 4657 dumpTracks(fd, args); 4658 dumpEffectChains(fd, args); 4659} 4660 4661void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4662{ 4663 const size_t SIZE = 256; 4664 char buffer[SIZE]; 4665 String8 result; 4666 4667 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4668 result.append(buffer); 4669 4670 if (mActiveTrack != 0) { 4671 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4672 result.append(buffer); 4673 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4674 result.append(buffer); 4675 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4676 result.append(buffer); 4677 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4678 result.append(buffer); 4679 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4680 result.append(buffer); 4681 } else { 4682 result.append("No active record client\n"); 4683 } 4684 4685 write(fd, result.string(), result.size()); 4686 4687 dumpBase(fd, args); 4688} 4689 4690void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4691{ 4692 const size_t SIZE = 256; 4693 char buffer[SIZE]; 4694 String8 result; 4695 4696 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4697 result.append(buffer); 4698 RecordTrack::appendDumpHeader(result); 4699 for (size_t i = 0; i < mTracks.size(); ++i) { 4700 sp<RecordTrack> track = mTracks[i]; 4701 if (track != 0) { 4702 track->dump(buffer, SIZE); 4703 result.append(buffer); 4704 } 4705 } 4706 4707 if (mActiveTrack != 0) { 4708 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4709 result.append(buffer); 4710 RecordTrack::appendDumpHeader(result); 4711 mActiveTrack->dump(buffer, SIZE); 4712 result.append(buffer); 4713 4714 } 4715 write(fd, result.string(), result.size()); 4716} 4717 4718// AudioBufferProvider interface 4719status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4720{ 4721 size_t framesReq = buffer->frameCount; 4722 size_t framesReady = mFrameCount - mRsmpInIndex; 4723 int channelCount; 4724 4725 if (framesReady == 0) { 4726 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4727 if (mBytesRead <= 0) { 4728 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4729 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4730 // Force input into standby so that it tries to 4731 // recover at next read attempt 4732 inputStandBy(); 4733 usleep(kRecordThreadSleepUs); 4734 } 4735 buffer->raw = NULL; 4736 buffer->frameCount = 0; 4737 return NOT_ENOUGH_DATA; 4738 } 4739 mRsmpInIndex = 0; 4740 framesReady = mFrameCount; 4741 } 4742 4743 if (framesReq > framesReady) { 4744 framesReq = framesReady; 4745 } 4746 4747 if (mChannelCount == 1 && mReqChannelCount == 2) { 4748 channelCount = 1; 4749 } else { 4750 channelCount = 2; 4751 } 4752 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4753 buffer->frameCount = framesReq; 4754 return NO_ERROR; 4755} 4756 4757// AudioBufferProvider interface 4758void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4759{ 4760 mRsmpInIndex += buffer->frameCount; 4761 buffer->frameCount = 0; 4762} 4763 4764bool AudioFlinger::RecordThread::checkForNewParameters_l() 4765{ 4766 bool reconfig = false; 4767 4768 while (!mNewParameters.isEmpty()) { 4769 status_t status = NO_ERROR; 4770 String8 keyValuePair = mNewParameters[0]; 4771 AudioParameter param = AudioParameter(keyValuePair); 4772 int value; 4773 audio_format_t reqFormat = mFormat; 4774 uint32_t reqSamplingRate = mReqSampleRate; 4775 uint32_t reqChannelCount = mReqChannelCount; 4776 4777 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4778 reqSamplingRate = value; 4779 reconfig = true; 4780 } 4781 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4782 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4783 status = BAD_VALUE; 4784 } else { 4785 reqFormat = (audio_format_t) value; 4786 reconfig = true; 4787 } 4788 } 4789 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4790 reqChannelCount = popcount(value); 4791 reconfig = true; 4792 } 4793 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4794 // do not accept frame count changes if tracks are open as the track buffer 4795 // size depends on frame count and correct behavior would not be guaranteed 4796 // if frame count is changed after track creation 4797 if (mActiveTrack != 0) { 4798 status = INVALID_OPERATION; 4799 } else { 4800 reconfig = true; 4801 } 4802 } 4803 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4804 // forward device change to effects that have requested to be 4805 // aware of attached audio device. 4806 for (size_t i = 0; i < mEffectChains.size(); i++) { 4807 mEffectChains[i]->setDevice_l(value); 4808 } 4809 4810 // store input device and output device but do not forward output device to audio HAL. 4811 // Note that status is ignored by the caller for output device 4812 // (see AudioFlinger::setParameters() 4813 if (audio_is_output_devices(value)) { 4814 mOutDevice = value; 4815 status = BAD_VALUE; 4816 } else { 4817 mInDevice = value; 4818 // disable AEC and NS if the device is a BT SCO headset supporting those 4819 // pre processings 4820 if (mTracks.size() > 0) { 4821 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4822 mAudioFlinger->btNrecIsOff(); 4823 for (size_t i = 0; i < mTracks.size(); i++) { 4824 sp<RecordTrack> track = mTracks[i]; 4825 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4826 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4827 } 4828 } 4829 } 4830 } 4831 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4832 mAudioSource != (audio_source_t)value) { 4833 // forward device change to effects that have requested to be 4834 // aware of attached audio device. 4835 for (size_t i = 0; i < mEffectChains.size(); i++) { 4836 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4837 } 4838 mAudioSource = (audio_source_t)value; 4839 } 4840 if (status == NO_ERROR) { 4841 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4842 keyValuePair.string()); 4843 if (status == INVALID_OPERATION) { 4844 inputStandBy(); 4845 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4846 keyValuePair.string()); 4847 } 4848 if (reconfig) { 4849 if (status == BAD_VALUE && 4850 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4851 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4852 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4853 <= (2 * reqSamplingRate)) && 4854 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4855 <= FCC_2 && 4856 (reqChannelCount <= FCC_2)) { 4857 status = NO_ERROR; 4858 } 4859 if (status == NO_ERROR) { 4860 readInputParameters(); 4861 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4862 } 4863 } 4864 } 4865 4866 mNewParameters.removeAt(0); 4867 4868 mParamStatus = status; 4869 mParamCond.signal(); 4870 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4871 // already timed out waiting for the status and will never signal the condition. 4872 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4873 } 4874 return reconfig; 4875} 4876 4877String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4878{ 4879 Mutex::Autolock _l(mLock); 4880 if (initCheck() != NO_ERROR) { 4881 return String8(); 4882 } 4883 4884 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4885 const String8 out_s8(s); 4886 free(s); 4887 return out_s8; 4888} 4889 4890void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4891 AudioSystem::OutputDescriptor desc; 4892 void *param2 = NULL; 4893 4894 switch (event) { 4895 case AudioSystem::INPUT_OPENED: 4896 case AudioSystem::INPUT_CONFIG_CHANGED: 4897 desc.channelMask = mChannelMask; 4898 desc.samplingRate = mSampleRate; 4899 desc.format = mFormat; 4900 desc.frameCount = mFrameCount; 4901 desc.latency = 0; 4902 param2 = &desc; 4903 break; 4904 4905 case AudioSystem::INPUT_CLOSED: 4906 default: 4907 break; 4908 } 4909 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4910} 4911 4912void AudioFlinger::RecordThread::readInputParameters() 4913{ 4914 delete[] mRsmpInBuffer; 4915 // mRsmpInBuffer is always assigned a new[] below 4916 delete[] mRsmpOutBuffer; 4917 mRsmpOutBuffer = NULL; 4918 delete mResampler; 4919 mResampler = NULL; 4920 4921 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4922 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4923 mChannelCount = popcount(mChannelMask); 4924 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4925 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4926 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 4927 } 4928 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4929 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4930 mFrameCount = mBufferSize / mFrameSize; 4931 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4932 4933 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4934 { 4935 int channelCount; 4936 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4937 // stereo to mono post process as the resampler always outputs stereo. 4938 if (mChannelCount == 1 && mReqChannelCount == 2) { 4939 channelCount = 1; 4940 } else { 4941 channelCount = 2; 4942 } 4943 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4944 mResampler->setSampleRate(mSampleRate); 4945 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4946 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 4947 4948 // optmization: if mono to mono, alter input frame count as if we were inputing 4949 // stereo samples 4950 if (mChannelCount == 1 && mReqChannelCount == 1) { 4951 mFrameCount >>= 1; 4952 } 4953 4954 } 4955 mRsmpInIndex = mFrameCount; 4956} 4957 4958unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4959{ 4960 Mutex::Autolock _l(mLock); 4961 if (initCheck() != NO_ERROR) { 4962 return 0; 4963 } 4964 4965 return mInput->stream->get_input_frames_lost(mInput->stream); 4966} 4967 4968uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4969{ 4970 Mutex::Autolock _l(mLock); 4971 uint32_t result = 0; 4972 if (getEffectChain_l(sessionId) != 0) { 4973 result = EFFECT_SESSION; 4974 } 4975 4976 for (size_t i = 0; i < mTracks.size(); ++i) { 4977 if (sessionId == mTracks[i]->sessionId()) { 4978 result |= TRACK_SESSION; 4979 break; 4980 } 4981 } 4982 4983 return result; 4984} 4985 4986KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4987{ 4988 KeyedVector<int, bool> ids; 4989 Mutex::Autolock _l(mLock); 4990 for (size_t j = 0; j < mTracks.size(); ++j) { 4991 sp<RecordThread::RecordTrack> track = mTracks[j]; 4992 int sessionId = track->sessionId(); 4993 if (ids.indexOfKey(sessionId) < 0) { 4994 ids.add(sessionId, true); 4995 } 4996 } 4997 return ids; 4998} 4999 5000AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5001{ 5002 Mutex::Autolock _l(mLock); 5003 AudioStreamIn *input = mInput; 5004 mInput = NULL; 5005 return input; 5006} 5007 5008// this method must always be called either with ThreadBase mLock held or inside the thread loop 5009audio_stream_t* AudioFlinger::RecordThread::stream() const 5010{ 5011 if (mInput == NULL) { 5012 return NULL; 5013 } 5014 return &mInput->stream->common; 5015} 5016 5017status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5018{ 5019 // only one chain per input thread 5020 if (mEffectChains.size() != 0) { 5021 return INVALID_OPERATION; 5022 } 5023 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5024 5025 chain->setInBuffer(NULL); 5026 chain->setOutBuffer(NULL); 5027 5028 checkSuspendOnAddEffectChain_l(chain); 5029 5030 mEffectChains.add(chain); 5031 5032 return NO_ERROR; 5033} 5034 5035size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5036{ 5037 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5038 ALOGW_IF(mEffectChains.size() != 1, 5039 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5040 chain.get(), mEffectChains.size(), this); 5041 if (mEffectChains.size() == 1) { 5042 mEffectChains.removeAt(0); 5043 } 5044 return 0; 5045} 5046 5047}; // namespace android 5048