Threads.cpp revision a30e75897934da2ce7b1b03bcb4b58e139d3e81e
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, and mFormat are 270 // 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, "Channel Count: %u\n", mChannelCount); 427 result.append(buffer); 428 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 429 result.append(buffer); 430 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 433 result.append(buffer); 434 435 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 436 result.append(buffer); 437 result.append(" Index Command"); 438 for (size_t i = 0; i < mNewParameters.size(); ++i) { 439 snprintf(buffer, SIZE, "\n %02d ", i); 440 result.append(buffer); 441 result.append(mNewParameters[i]); 442 } 443 444 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 445 result.append(buffer); 446 for (size_t i = 0; i < mConfigEvents.size(); i++) { 447 mConfigEvents[i]->dump(buffer, SIZE); 448 result.append(buffer); 449 } 450 result.append("\n"); 451 452 write(fd, result.string(), result.size()); 453 454 if (locked) { 455 mLock.unlock(); 456 } 457} 458 459void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 460{ 461 const size_t SIZE = 256; 462 char buffer[SIZE]; 463 String8 result; 464 465 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 466 write(fd, buffer, strlen(buffer)); 467 468 for (size_t i = 0; i < mEffectChains.size(); ++i) { 469 sp<EffectChain> chain = mEffectChains[i]; 470 if (chain != 0) { 471 chain->dump(fd, args); 472 } 473 } 474} 475 476void AudioFlinger::ThreadBase::acquireWakeLock() 477{ 478 Mutex::Autolock _l(mLock); 479 acquireWakeLock_l(); 480} 481 482void AudioFlinger::ThreadBase::acquireWakeLock_l() 483{ 484 if (mPowerManager == 0) { 485 // use checkService() to avoid blocking if power service is not up yet 486 sp<IBinder> binder = 487 defaultServiceManager()->checkService(String16("power")); 488 if (binder == 0) { 489 ALOGW("Thread %s cannot connect to the power manager service", mName); 490 } else { 491 mPowerManager = interface_cast<IPowerManager>(binder); 492 binder->linkToDeath(mDeathRecipient); 493 } 494 } 495 if (mPowerManager != 0) { 496 sp<IBinder> binder = new BBinder(); 497 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 498 binder, 499 String16(mName), 500 String16("media")); 501 if (status == NO_ERROR) { 502 mWakeLockToken = binder; 503 } 504 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 505 } 506} 507 508void AudioFlinger::ThreadBase::releaseWakeLock() 509{ 510 Mutex::Autolock _l(mLock); 511 releaseWakeLock_l(); 512} 513 514void AudioFlinger::ThreadBase::releaseWakeLock_l() 515{ 516 if (mWakeLockToken != 0) { 517 ALOGV("releaseWakeLock_l() %s", mName); 518 if (mPowerManager != 0) { 519 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 520 } 521 mWakeLockToken.clear(); 522 } 523} 524 525void AudioFlinger::ThreadBase::clearPowerManager() 526{ 527 Mutex::Autolock _l(mLock); 528 releaseWakeLock_l(); 529 mPowerManager.clear(); 530} 531 532void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 533{ 534 sp<ThreadBase> thread = mThread.promote(); 535 if (thread != 0) { 536 thread->clearPowerManager(); 537 } 538 ALOGW("power manager service died !!!"); 539} 540 541void AudioFlinger::ThreadBase::setEffectSuspended( 542 const effect_uuid_t *type, bool suspend, int sessionId) 543{ 544 Mutex::Autolock _l(mLock); 545 setEffectSuspended_l(type, suspend, sessionId); 546} 547 548void AudioFlinger::ThreadBase::setEffectSuspended_l( 549 const effect_uuid_t *type, bool suspend, int sessionId) 550{ 551 sp<EffectChain> chain = getEffectChain_l(sessionId); 552 if (chain != 0) { 553 if (type != NULL) { 554 chain->setEffectSuspended_l(type, suspend); 555 } else { 556 chain->setEffectSuspendedAll_l(suspend); 557 } 558 } 559 560 updateSuspendedSessions_l(type, suspend, sessionId); 561} 562 563void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 564{ 565 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 566 if (index < 0) { 567 return; 568 } 569 570 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 571 mSuspendedSessions.valueAt(index); 572 573 for (size_t i = 0; i < sessionEffects.size(); i++) { 574 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 575 for (int j = 0; j < desc->mRefCount; j++) { 576 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 577 chain->setEffectSuspendedAll_l(true); 578 } else { 579 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 580 desc->mType.timeLow); 581 chain->setEffectSuspended_l(&desc->mType, true); 582 } 583 } 584 } 585} 586 587void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 588 bool suspend, 589 int sessionId) 590{ 591 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 592 593 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 594 595 if (suspend) { 596 if (index >= 0) { 597 sessionEffects = mSuspendedSessions.valueAt(index); 598 } else { 599 mSuspendedSessions.add(sessionId, sessionEffects); 600 } 601 } else { 602 if (index < 0) { 603 return; 604 } 605 sessionEffects = mSuspendedSessions.valueAt(index); 606 } 607 608 609 int key = EffectChain::kKeyForSuspendAll; 610 if (type != NULL) { 611 key = type->timeLow; 612 } 613 index = sessionEffects.indexOfKey(key); 614 615 sp<SuspendedSessionDesc> desc; 616 if (suspend) { 617 if (index >= 0) { 618 desc = sessionEffects.valueAt(index); 619 } else { 620 desc = new SuspendedSessionDesc(); 621 if (type != NULL) { 622 desc->mType = *type; 623 } 624 sessionEffects.add(key, desc); 625 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 626 } 627 desc->mRefCount++; 628 } else { 629 if (index < 0) { 630 return; 631 } 632 desc = sessionEffects.valueAt(index); 633 if (--desc->mRefCount == 0) { 634 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 635 sessionEffects.removeItemsAt(index); 636 if (sessionEffects.isEmpty()) { 637 ALOGV("updateSuspendedSessions_l() restore removing session %d", 638 sessionId); 639 mSuspendedSessions.removeItem(sessionId); 640 } 641 } 642 } 643 if (!sessionEffects.isEmpty()) { 644 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 645 } 646} 647 648void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 649 bool enabled, 650 int sessionId) 651{ 652 Mutex::Autolock _l(mLock); 653 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 654} 655 656void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 657 bool enabled, 658 int sessionId) 659{ 660 if (mType != RECORD) { 661 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 662 // another session. This gives the priority to well behaved effect control panels 663 // and applications not using global effects. 664 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 665 // global effects 666 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 667 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 668 } 669 } 670 671 sp<EffectChain> chain = getEffectChain_l(sessionId); 672 if (chain != 0) { 673 chain->checkSuspendOnEffectEnabled(effect, enabled); 674 } 675} 676 677// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 678sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 679 const sp<AudioFlinger::Client>& client, 680 const sp<IEffectClient>& effectClient, 681 int32_t priority, 682 int sessionId, 683 effect_descriptor_t *desc, 684 int *enabled, 685 status_t *status 686 ) 687{ 688 sp<EffectModule> effect; 689 sp<EffectHandle> handle; 690 status_t lStatus; 691 sp<EffectChain> chain; 692 bool chainCreated = false; 693 bool effectCreated = false; 694 bool effectRegistered = false; 695 696 lStatus = initCheck(); 697 if (lStatus != NO_ERROR) { 698 ALOGW("createEffect_l() Audio driver not initialized."); 699 goto Exit; 700 } 701 702 // Do not allow effects with session ID 0 on direct output or duplicating threads 703 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 704 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 705 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 706 desc->name, sessionId); 707 lStatus = BAD_VALUE; 708 goto Exit; 709 } 710 // Only Pre processor effects are allowed on input threads and only on input threads 711 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 712 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 713 desc->name, desc->flags, mType); 714 lStatus = BAD_VALUE; 715 goto Exit; 716 } 717 718 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 719 720 { // scope for mLock 721 Mutex::Autolock _l(mLock); 722 723 // check for existing effect chain with the requested audio session 724 chain = getEffectChain_l(sessionId); 725 if (chain == 0) { 726 // create a new chain for this session 727 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 728 chain = new EffectChain(this, sessionId); 729 addEffectChain_l(chain); 730 chain->setStrategy(getStrategyForSession_l(sessionId)); 731 chainCreated = true; 732 } else { 733 effect = chain->getEffectFromDesc_l(desc); 734 } 735 736 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 737 738 if (effect == 0) { 739 int id = mAudioFlinger->nextUniqueId(); 740 // Check CPU and memory usage 741 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 742 if (lStatus != NO_ERROR) { 743 goto Exit; 744 } 745 effectRegistered = true; 746 // create a new effect module if none present in the chain 747 effect = new EffectModule(this, chain, desc, id, sessionId); 748 lStatus = effect->status(); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 lStatus = chain->addEffect_l(effect); 753 if (lStatus != NO_ERROR) { 754 goto Exit; 755 } 756 effectCreated = true; 757 758 effect->setDevice(mOutDevice); 759 effect->setDevice(mInDevice); 760 effect->setMode(mAudioFlinger->getMode()); 761 effect->setAudioSource(mAudioSource); 762 } 763 // create effect handle and connect it to effect module 764 handle = new EffectHandle(effect, client, effectClient, priority); 765 lStatus = effect->addHandle(handle.get()); 766 if (enabled != NULL) { 767 *enabled = (int)effect->isEnabled(); 768 } 769 } 770 771Exit: 772 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 773 Mutex::Autolock _l(mLock); 774 if (effectCreated) { 775 chain->removeEffect_l(effect); 776 } 777 if (effectRegistered) { 778 AudioSystem::unregisterEffect(effect->id()); 779 } 780 if (chainCreated) { 781 removeEffectChain_l(chain); 782 } 783 handle.clear(); 784 } 785 786 if (status != NULL) { 787 *status = lStatus; 788 } 789 return handle; 790} 791 792sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 793{ 794 Mutex::Autolock _l(mLock); 795 return getEffect_l(sessionId, effectId); 796} 797 798sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 799{ 800 sp<EffectChain> chain = getEffectChain_l(sessionId); 801 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 802} 803 804// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 805// PlaybackThread::mLock held 806status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 807{ 808 // check for existing effect chain with the requested audio session 809 int sessionId = effect->sessionId(); 810 sp<EffectChain> chain = getEffectChain_l(sessionId); 811 bool chainCreated = false; 812 813 if (chain == 0) { 814 // create a new chain for this session 815 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 816 chain = new EffectChain(this, sessionId); 817 addEffectChain_l(chain); 818 chain->setStrategy(getStrategyForSession_l(sessionId)); 819 chainCreated = true; 820 } 821 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 822 823 if (chain->getEffectFromId_l(effect->id()) != 0) { 824 ALOGW("addEffect_l() %p effect %s already present in chain %p", 825 this, effect->desc().name, chain.get()); 826 return BAD_VALUE; 827 } 828 829 status_t status = chain->addEffect_l(effect); 830 if (status != NO_ERROR) { 831 if (chainCreated) { 832 removeEffectChain_l(chain); 833 } 834 return status; 835 } 836 837 effect->setDevice(mOutDevice); 838 effect->setDevice(mInDevice); 839 effect->setMode(mAudioFlinger->getMode()); 840 effect->setAudioSource(mAudioSource); 841 return NO_ERROR; 842} 843 844void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 845 846 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 847 effect_descriptor_t desc = effect->desc(); 848 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 849 detachAuxEffect_l(effect->id()); 850 } 851 852 sp<EffectChain> chain = effect->chain().promote(); 853 if (chain != 0) { 854 // remove effect chain if removing last effect 855 if (chain->removeEffect_l(effect) == 0) { 856 removeEffectChain_l(chain); 857 } 858 } else { 859 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 860 } 861} 862 863void AudioFlinger::ThreadBase::lockEffectChains_l( 864 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 865{ 866 effectChains = mEffectChains; 867 for (size_t i = 0; i < mEffectChains.size(); i++) { 868 mEffectChains[i]->lock(); 869 } 870} 871 872void AudioFlinger::ThreadBase::unlockEffectChains( 873 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 874{ 875 for (size_t i = 0; i < effectChains.size(); i++) { 876 effectChains[i]->unlock(); 877 } 878} 879 880sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 881{ 882 Mutex::Autolock _l(mLock); 883 return getEffectChain_l(sessionId); 884} 885 886sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 887{ 888 size_t size = mEffectChains.size(); 889 for (size_t i = 0; i < size; i++) { 890 if (mEffectChains[i]->sessionId() == sessionId) { 891 return mEffectChains[i]; 892 } 893 } 894 return 0; 895} 896 897void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 898{ 899 Mutex::Autolock _l(mLock); 900 size_t size = mEffectChains.size(); 901 for (size_t i = 0; i < size; i++) { 902 mEffectChains[i]->setMode_l(mode); 903 } 904} 905 906void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 907 EffectHandle *handle, 908 bool unpinIfLast) { 909 910 Mutex::Autolock _l(mLock); 911 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 912 // delete the effect module if removing last handle on it 913 if (effect->removeHandle(handle) == 0) { 914 if (!effect->isPinned() || unpinIfLast) { 915 removeEffect_l(effect); 916 AudioSystem::unregisterEffect(effect->id()); 917 } 918 } 919} 920 921// ---------------------------------------------------------------------------- 922// Playback 923// ---------------------------------------------------------------------------- 924 925AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 926 AudioStreamOut* output, 927 audio_io_handle_t id, 928 audio_devices_t device, 929 type_t type) 930 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 931 mNormalFrameCount(0), mMixBuffer(NULL), 932 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 933 // mStreamTypes[] initialized in constructor body 934 mOutput(output), 935 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 936 mMixerStatus(MIXER_IDLE), 937 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 938 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 939 mBytesRemaining(0), 940 mCurrentWriteLength(0), 941 mUseAsyncWrite(false), 942 mWriteBlocked(false), 943 mDraining(false), 944 mScreenState(AudioFlinger::mScreenState), 945 // index 0 is reserved for normal mixer's submix 946 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 947{ 948 snprintf(mName, kNameLength, "AudioOut_%X", id); 949 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 950 951 // Assumes constructor is called by AudioFlinger with it's mLock held, but 952 // it would be safer to explicitly pass initial masterVolume/masterMute as 953 // parameter. 954 // 955 // If the HAL we are using has support for master volume or master mute, 956 // then do not attenuate or mute during mixing (just leave the volume at 1.0 957 // and the mute set to false). 958 mMasterVolume = audioFlinger->masterVolume_l(); 959 mMasterMute = audioFlinger->masterMute_l(); 960 if (mOutput && mOutput->audioHwDev) { 961 if (mOutput->audioHwDev->canSetMasterVolume()) { 962 mMasterVolume = 1.0; 963 } 964 965 if (mOutput->audioHwDev->canSetMasterMute()) { 966 mMasterMute = false; 967 } 968 } 969 970 readOutputParameters(); 971 972 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 973 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 974 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 975 stream = (audio_stream_type_t) (stream + 1)) { 976 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 977 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 978 } 979 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 980 // because mAudioFlinger doesn't have one to copy from 981} 982 983AudioFlinger::PlaybackThread::~PlaybackThread() 984{ 985 mAudioFlinger->unregisterWriter(mNBLogWriter); 986 delete [] mAllocMixBuffer; 987} 988 989void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 990{ 991 dumpInternals(fd, args); 992 dumpTracks(fd, args); 993 dumpEffectChains(fd, args); 994} 995 996void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 997{ 998 const size_t SIZE = 256; 999 char buffer[SIZE]; 1000 String8 result; 1001 1002 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1003 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1004 const stream_type_t *st = &mStreamTypes[i]; 1005 if (i > 0) { 1006 result.appendFormat(", "); 1007 } 1008 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1009 if (st->mute) { 1010 result.append("M"); 1011 } 1012 } 1013 result.append("\n"); 1014 write(fd, result.string(), result.length()); 1015 result.clear(); 1016 1017 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1018 result.append(buffer); 1019 Track::appendDumpHeader(result); 1020 for (size_t i = 0; i < mTracks.size(); ++i) { 1021 sp<Track> track = mTracks[i]; 1022 if (track != 0) { 1023 track->dump(buffer, SIZE); 1024 result.append(buffer); 1025 } 1026 } 1027 1028 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1029 result.append(buffer); 1030 Track::appendDumpHeader(result); 1031 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1032 sp<Track> track = mActiveTracks[i].promote(); 1033 if (track != 0) { 1034 track->dump(buffer, SIZE); 1035 result.append(buffer); 1036 } 1037 } 1038 write(fd, result.string(), result.size()); 1039 1040 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1041 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1042 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1043 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1044} 1045 1046void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1047{ 1048 const size_t SIZE = 256; 1049 char buffer[SIZE]; 1050 String8 result; 1051 1052 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1053 result.append(buffer); 1054 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1055 result.append(buffer); 1056 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1057 ns2ms(systemTime() - mLastWriteTime)); 1058 result.append(buffer); 1059 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1060 result.append(buffer); 1061 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1062 result.append(buffer); 1063 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1064 result.append(buffer); 1065 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1066 result.append(buffer); 1067 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1068 result.append(buffer); 1069 write(fd, result.string(), result.size()); 1070 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1071 1072 dumpBase(fd, args); 1073} 1074 1075// Thread virtuals 1076status_t AudioFlinger::PlaybackThread::readyToRun() 1077{ 1078 status_t status = initCheck(); 1079 if (status == NO_ERROR) { 1080 ALOGI("AudioFlinger's thread %p ready to run", this); 1081 } else { 1082 ALOGE("No working audio driver found."); 1083 } 1084 return status; 1085} 1086 1087void AudioFlinger::PlaybackThread::onFirstRef() 1088{ 1089 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1090} 1091 1092// ThreadBase virtuals 1093void AudioFlinger::PlaybackThread::preExit() 1094{ 1095 ALOGV(" preExit()"); 1096 // FIXME this is using hard-coded strings but in the future, this functionality will be 1097 // converted to use audio HAL extensions required to support tunneling 1098 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1099} 1100 1101// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1102sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1103 const sp<AudioFlinger::Client>& client, 1104 audio_stream_type_t streamType, 1105 uint32_t sampleRate, 1106 audio_format_t format, 1107 audio_channel_mask_t channelMask, 1108 size_t frameCount, 1109 const sp<IMemory>& sharedBuffer, 1110 int sessionId, 1111 IAudioFlinger::track_flags_t *flags, 1112 pid_t tid, 1113 status_t *status) 1114{ 1115 sp<Track> track; 1116 status_t lStatus; 1117 1118 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1119 1120 // client expresses a preference for FAST, but we get the final say 1121 if (*flags & IAudioFlinger::TRACK_FAST) { 1122 if ( 1123 // not timed 1124 (!isTimed) && 1125 // either of these use cases: 1126 ( 1127 // use case 1: shared buffer with any frame count 1128 ( 1129 (sharedBuffer != 0) 1130 ) || 1131 // use case 2: callback handler and frame count is default or at least as large as HAL 1132 ( 1133 (tid != -1) && 1134 ((frameCount == 0) || 1135 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1136 ) 1137 ) && 1138 // PCM data 1139 audio_is_linear_pcm(format) && 1140 // mono or stereo 1141 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1142 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1143#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1144 // hardware sample rate 1145 (sampleRate == mSampleRate) && 1146#endif 1147 // normal mixer has an associated fast mixer 1148 hasFastMixer() && 1149 // there are sufficient fast track slots available 1150 (mFastTrackAvailMask != 0) 1151 // FIXME test that MixerThread for this fast track has a capable output HAL 1152 // FIXME add a permission test also? 1153 ) { 1154 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1155 if (frameCount == 0) { 1156 frameCount = mFrameCount * kFastTrackMultiplier; 1157 } 1158 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1159 frameCount, mFrameCount); 1160 } else { 1161 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1162 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1163 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1164 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1165 audio_is_linear_pcm(format), 1166 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1167 *flags &= ~IAudioFlinger::TRACK_FAST; 1168 // For compatibility with AudioTrack calculation, buffer depth is forced 1169 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1170 // This is probably too conservative, but legacy application code may depend on it. 1171 // If you change this calculation, also review the start threshold which is related. 1172 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1173 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1174 if (minBufCount < 2) { 1175 minBufCount = 2; 1176 } 1177 size_t minFrameCount = mNormalFrameCount * minBufCount; 1178 if (frameCount < minFrameCount) { 1179 frameCount = minFrameCount; 1180 } 1181 } 1182 } 1183 1184 if (mType == DIRECT) { 1185 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1186 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1187 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1188 "for output %p with format %d", 1189 sampleRate, format, channelMask, mOutput, mFormat); 1190 lStatus = BAD_VALUE; 1191 goto Exit; 1192 } 1193 } 1194 } else if (mType == OFFLOAD) { 1195 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1196 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1197 "for output %p with format %d", 1198 sampleRate, format, channelMask, mOutput, mFormat); 1199 lStatus = BAD_VALUE; 1200 goto Exit; 1201 } 1202 } else { 1203 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1204 ALOGE("createTrack_l() Bad parameter: format %d \"" 1205 "for output %p with format %d", 1206 format, mOutput, mFormat); 1207 lStatus = BAD_VALUE; 1208 goto Exit; 1209 } 1210 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1211 if (sampleRate > mSampleRate*2) { 1212 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1213 lStatus = BAD_VALUE; 1214 goto Exit; 1215 } 1216 } 1217 1218 lStatus = initCheck(); 1219 if (lStatus != NO_ERROR) { 1220 ALOGE("Audio driver not initialized."); 1221 goto Exit; 1222 } 1223 1224 { // scope for mLock 1225 Mutex::Autolock _l(mLock); 1226 1227 // all tracks in same audio session must share the same routing strategy otherwise 1228 // conflicts will happen when tracks are moved from one output to another by audio policy 1229 // manager 1230 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1231 for (size_t i = 0; i < mTracks.size(); ++i) { 1232 sp<Track> t = mTracks[i]; 1233 if (t != 0 && !t->isOutputTrack()) { 1234 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1235 if (sessionId == t->sessionId() && strategy != actual) { 1236 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1237 strategy, actual); 1238 lStatus = BAD_VALUE; 1239 goto Exit; 1240 } 1241 } 1242 } 1243 1244 if (!isTimed) { 1245 track = new Track(this, client, streamType, sampleRate, format, 1246 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1247 } else { 1248 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1249 channelMask, frameCount, sharedBuffer, sessionId); 1250 } 1251 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1252 lStatus = NO_MEMORY; 1253 goto Exit; 1254 } 1255 1256 mTracks.add(track); 1257 1258 sp<EffectChain> chain = getEffectChain_l(sessionId); 1259 if (chain != 0) { 1260 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1261 track->setMainBuffer(chain->inBuffer()); 1262 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1263 chain->incTrackCnt(); 1264 } 1265 1266 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1267 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1268 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1269 // so ask activity manager to do this on our behalf 1270 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1271 } 1272 } 1273 1274 lStatus = NO_ERROR; 1275 1276Exit: 1277 if (status) { 1278 *status = lStatus; 1279 } 1280 return track; 1281} 1282 1283uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1284{ 1285 return latency; 1286} 1287 1288uint32_t AudioFlinger::PlaybackThread::latency() const 1289{ 1290 Mutex::Autolock _l(mLock); 1291 return latency_l(); 1292} 1293uint32_t AudioFlinger::PlaybackThread::latency_l() const 1294{ 1295 if (initCheck() == NO_ERROR) { 1296 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1297 } else { 1298 return 0; 1299 } 1300} 1301 1302void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1303{ 1304 Mutex::Autolock _l(mLock); 1305 // Don't apply master volume in SW if our HAL can do it for us. 1306 if (mOutput && mOutput->audioHwDev && 1307 mOutput->audioHwDev->canSetMasterVolume()) { 1308 mMasterVolume = 1.0; 1309 } else { 1310 mMasterVolume = value; 1311 } 1312} 1313 1314void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 // Don't apply master mute in SW if our HAL can do it for us. 1318 if (mOutput && mOutput->audioHwDev && 1319 mOutput->audioHwDev->canSetMasterMute()) { 1320 mMasterMute = false; 1321 } else { 1322 mMasterMute = muted; 1323 } 1324} 1325 1326void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1327{ 1328 Mutex::Autolock _l(mLock); 1329 mStreamTypes[stream].volume = value; 1330 signal_l(); 1331} 1332 1333void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1334{ 1335 Mutex::Autolock _l(mLock); 1336 mStreamTypes[stream].mute = muted; 1337 signal_l(); 1338} 1339 1340float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1341{ 1342 Mutex::Autolock _l(mLock); 1343 return mStreamTypes[stream].volume; 1344} 1345 1346// addTrack_l() must be called with ThreadBase::mLock held 1347status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1348{ 1349 status_t status = ALREADY_EXISTS; 1350 1351 // set retry count for buffer fill 1352 track->mRetryCount = kMaxTrackStartupRetries; 1353 if (mActiveTracks.indexOf(track) < 0) { 1354 // the track is newly added, make sure it fills up all its 1355 // buffers before playing. This is to ensure the client will 1356 // effectively get the latency it requested. 1357 if (!track->isOutputTrack()) { 1358 TrackBase::track_state state = track->mState; 1359 mLock.unlock(); 1360 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1361 mLock.lock(); 1362 // abort track was stopped/paused while we released the lock 1363 if (state != track->mState) { 1364 if (status == NO_ERROR) { 1365 mLock.unlock(); 1366 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1367 mLock.lock(); 1368 } 1369 return INVALID_OPERATION; 1370 } 1371 // abort if start is rejected by audio policy manager 1372 if (status != NO_ERROR) { 1373 return PERMISSION_DENIED; 1374 } 1375#ifdef ADD_BATTERY_DATA 1376 // to track the speaker usage 1377 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1378#endif 1379 } 1380 1381 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1382 track->mResetDone = false; 1383 track->mPresentationCompleteFrames = 0; 1384 mActiveTracks.add(track); 1385 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1386 if (chain != 0) { 1387 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1388 track->sessionId()); 1389 chain->incActiveTrackCnt(); 1390 } 1391 1392 status = NO_ERROR; 1393 } 1394 1395 ALOGV("mWaitWorkCV.broadcast"); 1396 mWaitWorkCV.broadcast(); 1397 1398 return status; 1399} 1400 1401bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1402{ 1403 track->terminate(); 1404 // active tracks are removed by threadLoop() 1405 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1406 track->mState = TrackBase::STOPPED; 1407 if (!trackActive) { 1408 removeTrack_l(track); 1409 } else if (track->isFastTrack() || track->isOffloaded()) { 1410 track->mState = TrackBase::STOPPING_1; 1411 } 1412 1413 return trackActive; 1414} 1415 1416void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1417{ 1418 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1419 mTracks.remove(track); 1420 deleteTrackName_l(track->name()); 1421 // redundant as track is about to be destroyed, for dumpsys only 1422 track->mName = -1; 1423 if (track->isFastTrack()) { 1424 int index = track->mFastIndex; 1425 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1426 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1427 mFastTrackAvailMask |= 1 << index; 1428 // redundant as track is about to be destroyed, for dumpsys only 1429 track->mFastIndex = -1; 1430 } 1431 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1432 if (chain != 0) { 1433 chain->decTrackCnt(); 1434 } 1435} 1436 1437void AudioFlinger::PlaybackThread::signal_l() 1438{ 1439 // Thread could be blocked waiting for async 1440 // so signal it to handle state changes immediately 1441 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1442 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1443 mSignalPending = true; 1444 mWaitWorkCV.signal(); 1445} 1446 1447String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1448{ 1449 Mutex::Autolock _l(mLock); 1450 if (initCheck() != NO_ERROR) { 1451 return String8(); 1452 } 1453 1454 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1455 const String8 out_s8(s); 1456 free(s); 1457 return out_s8; 1458} 1459 1460// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1461void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1462 AudioSystem::OutputDescriptor desc; 1463 void *param2 = NULL; 1464 1465 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1466 param); 1467 1468 switch (event) { 1469 case AudioSystem::OUTPUT_OPENED: 1470 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1471 desc.channelMask = mChannelMask; 1472 desc.samplingRate = mSampleRate; 1473 desc.format = mFormat; 1474 desc.frameCount = mNormalFrameCount; // FIXME see 1475 // AudioFlinger::frameCount(audio_io_handle_t) 1476 desc.latency = latency(); 1477 param2 = &desc; 1478 break; 1479 1480 case AudioSystem::STREAM_CONFIG_CHANGED: 1481 param2 = ¶m; 1482 case AudioSystem::OUTPUT_CLOSED: 1483 default: 1484 break; 1485 } 1486 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1487} 1488 1489void AudioFlinger::PlaybackThread::writeCallback() 1490{ 1491 ALOG_ASSERT(mCallbackThread != 0); 1492 mCallbackThread->setWriteBlocked(false); 1493} 1494 1495void AudioFlinger::PlaybackThread::drainCallback() 1496{ 1497 ALOG_ASSERT(mCallbackThread != 0); 1498 mCallbackThread->setDraining(false); 1499} 1500 1501void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1502{ 1503 Mutex::Autolock _l(mLock); 1504 mWriteBlocked = value; 1505 if (!value) { 1506 mWaitWorkCV.signal(); 1507 } 1508} 1509 1510void AudioFlinger::PlaybackThread::setDraining(bool value) 1511{ 1512 Mutex::Autolock _l(mLock); 1513 mDraining = value; 1514 if (!value) { 1515 mWaitWorkCV.signal(); 1516 } 1517} 1518 1519// static 1520int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1521 void *param, 1522 void *cookie) 1523{ 1524 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1525 ALOGV("asyncCallback() event %d", event); 1526 switch (event) { 1527 case STREAM_CBK_EVENT_WRITE_READY: 1528 me->writeCallback(); 1529 break; 1530 case STREAM_CBK_EVENT_DRAIN_READY: 1531 me->drainCallback(); 1532 break; 1533 default: 1534 ALOGW("asyncCallback() unknown event %d", event); 1535 break; 1536 } 1537 return 0; 1538} 1539 1540void AudioFlinger::PlaybackThread::readOutputParameters() 1541{ 1542 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1543 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1544 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1545 if (!audio_is_output_channel(mChannelMask)) { 1546 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1547 } 1548 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1549 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1550 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1551 } 1552 mChannelCount = popcount(mChannelMask); 1553 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1554 if (!audio_is_valid_format(mFormat)) { 1555 LOG_FATAL("HAL format %d not valid for output", mFormat); 1556 } 1557 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1558 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1559 mFormat); 1560 } 1561 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1562 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1563 if (mFrameCount & 15) { 1564 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1565 mFrameCount); 1566 } 1567 1568 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1569 (mOutput->stream->set_callback != NULL)) { 1570 if (mOutput->stream->set_callback(mOutput->stream, 1571 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1572 mUseAsyncWrite = true; 1573 } 1574 } 1575 1576 // Calculate size of normal mix buffer relative to the HAL output buffer size 1577 double multiplier = 1.0; 1578 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1579 kUseFastMixer == FastMixer_Dynamic)) { 1580 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1581 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1582 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1583 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1584 maxNormalFrameCount = maxNormalFrameCount & ~15; 1585 if (maxNormalFrameCount < minNormalFrameCount) { 1586 maxNormalFrameCount = minNormalFrameCount; 1587 } 1588 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1589 if (multiplier <= 1.0) { 1590 multiplier = 1.0; 1591 } else if (multiplier <= 2.0) { 1592 if (2 * mFrameCount <= maxNormalFrameCount) { 1593 multiplier = 2.0; 1594 } else { 1595 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1596 } 1597 } else { 1598 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1599 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1600 // track, but we sometimes have to do this to satisfy the maximum frame count 1601 // constraint) 1602 // FIXME this rounding up should not be done if no HAL SRC 1603 uint32_t truncMult = (uint32_t) multiplier; 1604 if ((truncMult & 1)) { 1605 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1606 ++truncMult; 1607 } 1608 } 1609 multiplier = (double) truncMult; 1610 } 1611 } 1612 mNormalFrameCount = multiplier * mFrameCount; 1613 // round up to nearest 16 frames to satisfy AudioMixer 1614 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1615 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1616 mNormalFrameCount); 1617 1618 delete[] mAllocMixBuffer; 1619 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1620 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1621 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1622 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1623 1624 // force reconfiguration of effect chains and engines to take new buffer size and audio 1625 // parameters into account 1626 // Note that mLock is not held when readOutputParameters() is called from the constructor 1627 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1628 // matter. 1629 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1630 Vector< sp<EffectChain> > effectChains = mEffectChains; 1631 for (size_t i = 0; i < effectChains.size(); i ++) { 1632 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1633 } 1634} 1635 1636 1637status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1638{ 1639 if (halFrames == NULL || dspFrames == NULL) { 1640 return BAD_VALUE; 1641 } 1642 Mutex::Autolock _l(mLock); 1643 if (initCheck() != NO_ERROR) { 1644 return INVALID_OPERATION; 1645 } 1646 size_t framesWritten = mBytesWritten / mFrameSize; 1647 *halFrames = framesWritten; 1648 1649 if (isSuspended()) { 1650 // return an estimation of rendered frames when the output is suspended 1651 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1652 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1653 return NO_ERROR; 1654 } else { 1655 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1656 } 1657} 1658 1659uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1660{ 1661 Mutex::Autolock _l(mLock); 1662 uint32_t result = 0; 1663 if (getEffectChain_l(sessionId) != 0) { 1664 result = EFFECT_SESSION; 1665 } 1666 1667 for (size_t i = 0; i < mTracks.size(); ++i) { 1668 sp<Track> track = mTracks[i]; 1669 if (sessionId == track->sessionId() && !track->isInvalid()) { 1670 result |= TRACK_SESSION; 1671 break; 1672 } 1673 } 1674 1675 return result; 1676} 1677 1678uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1679{ 1680 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1681 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1682 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1683 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1684 } 1685 for (size_t i = 0; i < mTracks.size(); i++) { 1686 sp<Track> track = mTracks[i]; 1687 if (sessionId == track->sessionId() && !track->isInvalid()) { 1688 return AudioSystem::getStrategyForStream(track->streamType()); 1689 } 1690 } 1691 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1692} 1693 1694 1695AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1696{ 1697 Mutex::Autolock _l(mLock); 1698 return mOutput; 1699} 1700 1701AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1702{ 1703 Mutex::Autolock _l(mLock); 1704 AudioStreamOut *output = mOutput; 1705 mOutput = NULL; 1706 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1707 // must push a NULL and wait for ack 1708 mOutputSink.clear(); 1709 mPipeSink.clear(); 1710 mNormalSink.clear(); 1711 return output; 1712} 1713 1714// this method must always be called either with ThreadBase mLock held or inside the thread loop 1715audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1716{ 1717 if (mOutput == NULL) { 1718 return NULL; 1719 } 1720 return &mOutput->stream->common; 1721} 1722 1723uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1724{ 1725 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1726} 1727 1728status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1729{ 1730 if (!isValidSyncEvent(event)) { 1731 return BAD_VALUE; 1732 } 1733 1734 Mutex::Autolock _l(mLock); 1735 1736 for (size_t i = 0; i < mTracks.size(); ++i) { 1737 sp<Track> track = mTracks[i]; 1738 if (event->triggerSession() == track->sessionId()) { 1739 (void) track->setSyncEvent(event); 1740 return NO_ERROR; 1741 } 1742 } 1743 1744 return NAME_NOT_FOUND; 1745} 1746 1747bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1748{ 1749 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1750} 1751 1752void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1753 const Vector< sp<Track> >& tracksToRemove) 1754{ 1755 size_t count = tracksToRemove.size(); 1756 if (count) { 1757 for (size_t i = 0 ; i < count ; i++) { 1758 const sp<Track>& track = tracksToRemove.itemAt(i); 1759 if (!track->isOutputTrack()) { 1760 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1761#ifdef ADD_BATTERY_DATA 1762 // to track the speaker usage 1763 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1764#endif 1765 if (track->isTerminated()) { 1766 AudioSystem::releaseOutput(mId); 1767 } 1768 } 1769 } 1770 } 1771} 1772 1773void AudioFlinger::PlaybackThread::checkSilentMode_l() 1774{ 1775 if (!mMasterMute) { 1776 char value[PROPERTY_VALUE_MAX]; 1777 if (property_get("ro.audio.silent", value, "0") > 0) { 1778 char *endptr; 1779 unsigned long ul = strtoul(value, &endptr, 0); 1780 if (*endptr == '\0' && ul != 0) { 1781 ALOGD("Silence is golden"); 1782 // The setprop command will not allow a property to be changed after 1783 // the first time it is set, so we don't have to worry about un-muting. 1784 setMasterMute_l(true); 1785 } 1786 } 1787 } 1788} 1789 1790// shared by MIXER and DIRECT, overridden by DUPLICATING 1791ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1792{ 1793 // FIXME rewrite to reduce number of system calls 1794 mLastWriteTime = systemTime(); 1795 mInWrite = true; 1796 ssize_t bytesWritten; 1797 1798 // If an NBAIO sink is present, use it to write the normal mixer's submix 1799 if (mNormalSink != 0) { 1800#define mBitShift 2 // FIXME 1801 size_t count = mBytesRemaining >> mBitShift; 1802 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1803 ATRACE_BEGIN("write"); 1804 // update the setpoint when AudioFlinger::mScreenState changes 1805 uint32_t screenState = AudioFlinger::mScreenState; 1806 if (screenState != mScreenState) { 1807 mScreenState = screenState; 1808 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1809 if (pipe != NULL) { 1810 pipe->setAvgFrames((mScreenState & 1) ? 1811 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1812 } 1813 } 1814 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1815 ATRACE_END(); 1816 if (framesWritten > 0) { 1817 bytesWritten = framesWritten << mBitShift; 1818 } else { 1819 bytesWritten = framesWritten; 1820 } 1821 // otherwise use the HAL / AudioStreamOut directly 1822 } else { 1823 // Direct output and offload threads 1824 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1825 if (mUseAsyncWrite) { 1826 mWriteBlocked = true; 1827 ALOG_ASSERT(mCallbackThread != 0); 1828 mCallbackThread->setWriteBlocked(true); 1829 } 1830 bytesWritten = mOutput->stream->write(mOutput->stream, 1831 mMixBuffer + offset, mBytesRemaining); 1832 if (mUseAsyncWrite && 1833 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1834 // do not wait for async callback in case of error of full write 1835 mWriteBlocked = false; 1836 ALOG_ASSERT(mCallbackThread != 0); 1837 mCallbackThread->setWriteBlocked(false); 1838 } 1839 } 1840 1841 mNumWrites++; 1842 mInWrite = false; 1843 1844 return bytesWritten; 1845} 1846 1847void AudioFlinger::PlaybackThread::threadLoop_drain() 1848{ 1849 if (mOutput->stream->drain) { 1850 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1851 if (mUseAsyncWrite) { 1852 mDraining = true; 1853 ALOG_ASSERT(mCallbackThread != 0); 1854 mCallbackThread->setDraining(true); 1855 } 1856 mOutput->stream->drain(mOutput->stream, 1857 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1858 : AUDIO_DRAIN_ALL); 1859 } 1860} 1861 1862void AudioFlinger::PlaybackThread::threadLoop_exit() 1863{ 1864 // Default implementation has nothing to do 1865} 1866 1867/* 1868The derived values that are cached: 1869 - mixBufferSize from frame count * frame size 1870 - activeSleepTime from activeSleepTimeUs() 1871 - idleSleepTime from idleSleepTimeUs() 1872 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1873 - maxPeriod from frame count and sample rate (MIXER only) 1874 1875The parameters that affect these derived values are: 1876 - frame count 1877 - frame size 1878 - sample rate 1879 - device type: A2DP or not 1880 - device latency 1881 - format: PCM or not 1882 - active sleep time 1883 - idle sleep time 1884*/ 1885 1886void AudioFlinger::PlaybackThread::cacheParameters_l() 1887{ 1888 mixBufferSize = mNormalFrameCount * mFrameSize; 1889 activeSleepTime = activeSleepTimeUs(); 1890 idleSleepTime = idleSleepTimeUs(); 1891} 1892 1893void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1894{ 1895 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1896 this, streamType, mTracks.size()); 1897 Mutex::Autolock _l(mLock); 1898 1899 size_t size = mTracks.size(); 1900 for (size_t i = 0; i < size; i++) { 1901 sp<Track> t = mTracks[i]; 1902 if (t->streamType() == streamType) { 1903 t->invalidate(); 1904 } 1905 } 1906} 1907 1908status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1909{ 1910 int session = chain->sessionId(); 1911 int16_t *buffer = mMixBuffer; 1912 bool ownsBuffer = false; 1913 1914 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1915 if (session > 0) { 1916 // Only one effect chain can be present in direct output thread and it uses 1917 // the mix buffer as input 1918 if (mType != DIRECT) { 1919 size_t numSamples = mNormalFrameCount * mChannelCount; 1920 buffer = new int16_t[numSamples]; 1921 memset(buffer, 0, numSamples * sizeof(int16_t)); 1922 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1923 ownsBuffer = true; 1924 } 1925 1926 // Attach all tracks with same session ID to this chain. 1927 for (size_t i = 0; i < mTracks.size(); ++i) { 1928 sp<Track> track = mTracks[i]; 1929 if (session == track->sessionId()) { 1930 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1931 buffer); 1932 track->setMainBuffer(buffer); 1933 chain->incTrackCnt(); 1934 } 1935 } 1936 1937 // indicate all active tracks in the chain 1938 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1939 sp<Track> track = mActiveTracks[i].promote(); 1940 if (track == 0) { 1941 continue; 1942 } 1943 if (session == track->sessionId()) { 1944 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1945 chain->incActiveTrackCnt(); 1946 } 1947 } 1948 } 1949 1950 chain->setInBuffer(buffer, ownsBuffer); 1951 chain->setOutBuffer(mMixBuffer); 1952 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1953 // chains list in order to be processed last as it contains output stage effects 1954 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1955 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1956 // after track specific effects and before output stage 1957 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1958 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1959 // Effect chain for other sessions are inserted at beginning of effect 1960 // chains list to be processed before output mix effects. Relative order between other 1961 // sessions is not important 1962 size_t size = mEffectChains.size(); 1963 size_t i = 0; 1964 for (i = 0; i < size; i++) { 1965 if (mEffectChains[i]->sessionId() < session) { 1966 break; 1967 } 1968 } 1969 mEffectChains.insertAt(chain, i); 1970 checkSuspendOnAddEffectChain_l(chain); 1971 1972 return NO_ERROR; 1973} 1974 1975size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1976{ 1977 int session = chain->sessionId(); 1978 1979 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1980 1981 for (size_t i = 0; i < mEffectChains.size(); i++) { 1982 if (chain == mEffectChains[i]) { 1983 mEffectChains.removeAt(i); 1984 // detach all active tracks from the chain 1985 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1986 sp<Track> track = mActiveTracks[i].promote(); 1987 if (track == 0) { 1988 continue; 1989 } 1990 if (session == track->sessionId()) { 1991 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1992 chain.get(), session); 1993 chain->decActiveTrackCnt(); 1994 } 1995 } 1996 1997 // detach all tracks with same session ID from this chain 1998 for (size_t i = 0; i < mTracks.size(); ++i) { 1999 sp<Track> track = mTracks[i]; 2000 if (session == track->sessionId()) { 2001 track->setMainBuffer(mMixBuffer); 2002 chain->decTrackCnt(); 2003 } 2004 } 2005 break; 2006 } 2007 } 2008 return mEffectChains.size(); 2009} 2010 2011status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2012 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2013{ 2014 Mutex::Autolock _l(mLock); 2015 return attachAuxEffect_l(track, EffectId); 2016} 2017 2018status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2019 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2020{ 2021 status_t status = NO_ERROR; 2022 2023 if (EffectId == 0) { 2024 track->setAuxBuffer(0, NULL); 2025 } else { 2026 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2027 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2028 if (effect != 0) { 2029 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2030 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2031 } else { 2032 status = INVALID_OPERATION; 2033 } 2034 } else { 2035 status = BAD_VALUE; 2036 } 2037 } 2038 return status; 2039} 2040 2041void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2042{ 2043 for (size_t i = 0; i < mTracks.size(); ++i) { 2044 sp<Track> track = mTracks[i]; 2045 if (track->auxEffectId() == effectId) { 2046 attachAuxEffect_l(track, 0); 2047 } 2048 } 2049} 2050 2051bool AudioFlinger::PlaybackThread::threadLoop() 2052{ 2053 Vector< sp<Track> > tracksToRemove; 2054 2055 standbyTime = systemTime(); 2056 2057 // MIXER 2058 nsecs_t lastWarning = 0; 2059 2060 // DUPLICATING 2061 // FIXME could this be made local to while loop? 2062 writeFrames = 0; 2063 2064 cacheParameters_l(); 2065 sleepTime = idleSleepTime; 2066 2067 if (mType == MIXER) { 2068 sleepTimeShift = 0; 2069 } 2070 2071 CpuStats cpuStats; 2072 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2073 2074 acquireWakeLock(); 2075 2076 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2077 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2078 // and then that string will be logged at the next convenient opportunity. 2079 const char *logString = NULL; 2080 2081 while (!exitPending()) 2082 { 2083 cpuStats.sample(myName); 2084 2085 Vector< sp<EffectChain> > effectChains; 2086 2087 processConfigEvents(); 2088 2089 { // scope for mLock 2090 2091 Mutex::Autolock _l(mLock); 2092 2093 if (logString != NULL) { 2094 mNBLogWriter->logTimestamp(); 2095 mNBLogWriter->log(logString); 2096 logString = NULL; 2097 } 2098 2099 if (checkForNewParameters_l()) { 2100 cacheParameters_l(); 2101 } 2102 2103 saveOutputTracks(); 2104 2105 if (mSignalPending) { 2106 // A signal was raised while we were unlocked 2107 mSignalPending = false; 2108 } else if (waitingAsyncCallback_l()) { 2109 if (exitPending()) { 2110 break; 2111 } 2112 releaseWakeLock_l(); 2113 ALOGV("wait async completion"); 2114 mWaitWorkCV.wait(mLock); 2115 ALOGV("async completion/wake"); 2116 acquireWakeLock_l(); 2117 if (exitPending()) { 2118 break; 2119 } 2120 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2121 continue; 2122 } 2123 sleepTime = 0; 2124 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2125 isSuspended()) { 2126 // put audio hardware into standby after short delay 2127 if (shouldStandby_l()) { 2128 2129 threadLoop_standby(); 2130 2131 mStandby = true; 2132 } 2133 2134 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2135 // we're about to wait, flush the binder command buffer 2136 IPCThreadState::self()->flushCommands(); 2137 2138 clearOutputTracks(); 2139 2140 if (exitPending()) { 2141 break; 2142 } 2143 2144 releaseWakeLock_l(); 2145 // wait until we have something to do... 2146 ALOGV("%s going to sleep", myName.string()); 2147 mWaitWorkCV.wait(mLock); 2148 ALOGV("%s waking up", myName.string()); 2149 acquireWakeLock_l(); 2150 2151 mMixerStatus = MIXER_IDLE; 2152 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2153 mBytesWritten = 0; 2154 mBytesRemaining = 0; 2155 checkSilentMode_l(); 2156 2157 standbyTime = systemTime() + standbyDelay; 2158 sleepTime = idleSleepTime; 2159 if (mType == MIXER) { 2160 sleepTimeShift = 0; 2161 } 2162 2163 continue; 2164 } 2165 } 2166 2167 // mMixerStatusIgnoringFastTracks is also updated internally 2168 mMixerStatus = prepareTracks_l(&tracksToRemove); 2169 2170 // prevent any changes in effect chain list and in each effect chain 2171 // during mixing and effect process as the audio buffers could be deleted 2172 // or modified if an effect is created or deleted 2173 lockEffectChains_l(effectChains); 2174 } 2175 2176 if (mBytesRemaining == 0) { 2177 mCurrentWriteLength = 0; 2178 if (mMixerStatus == MIXER_TRACKS_READY) { 2179 // threadLoop_mix() sets mCurrentWriteLength 2180 threadLoop_mix(); 2181 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2182 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2183 // threadLoop_sleepTime sets sleepTime to 0 if data 2184 // must be written to HAL 2185 threadLoop_sleepTime(); 2186 if (sleepTime == 0) { 2187 mCurrentWriteLength = mixBufferSize; 2188 } 2189 } 2190 mBytesRemaining = mCurrentWriteLength; 2191 if (isSuspended()) { 2192 sleepTime = suspendSleepTimeUs(); 2193 // simulate write to HAL when suspended 2194 mBytesWritten += mixBufferSize; 2195 mBytesRemaining = 0; 2196 } 2197 2198 // only process effects if we're going to write 2199 if (sleepTime == 0) { 2200 for (size_t i = 0; i < effectChains.size(); i ++) { 2201 effectChains[i]->process_l(); 2202 } 2203 } 2204 } 2205 2206 // enable changes in effect chain 2207 unlockEffectChains(effectChains); 2208 2209 if (!waitingAsyncCallback()) { 2210 // sleepTime == 0 means we must write to audio hardware 2211 if (sleepTime == 0) { 2212 if (mBytesRemaining) { 2213 ssize_t ret = threadLoop_write(); 2214 if (ret < 0) { 2215 mBytesRemaining = 0; 2216 } else { 2217 mBytesWritten += ret; 2218 mBytesRemaining -= ret; 2219 } 2220 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2221 (mMixerStatus == MIXER_DRAIN_ALL)) { 2222 threadLoop_drain(); 2223 } 2224if (mType == MIXER) { 2225 // write blocked detection 2226 nsecs_t now = systemTime(); 2227 nsecs_t delta = now - mLastWriteTime; 2228 if (!mStandby && delta > maxPeriod) { 2229 mNumDelayedWrites++; 2230 if ((now - lastWarning) > kWarningThrottleNs) { 2231 ATRACE_NAME("underrun"); 2232 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2233 ns2ms(delta), mNumDelayedWrites, this); 2234 lastWarning = now; 2235 } 2236 } 2237} 2238 2239 mStandby = false; 2240 } else { 2241 usleep(sleepTime); 2242 } 2243 } 2244 2245 // Finally let go of removed track(s), without the lock held 2246 // since we can't guarantee the destructors won't acquire that 2247 // same lock. This will also mutate and push a new fast mixer state. 2248 threadLoop_removeTracks(tracksToRemove); 2249 tracksToRemove.clear(); 2250 2251 // FIXME I don't understand the need for this here; 2252 // it was in the original code but maybe the 2253 // assignment in saveOutputTracks() makes this unnecessary? 2254 clearOutputTracks(); 2255 2256 // Effect chains will be actually deleted here if they were removed from 2257 // mEffectChains list during mixing or effects processing 2258 effectChains.clear(); 2259 2260 // FIXME Note that the above .clear() is no longer necessary since effectChains 2261 // is now local to this block, but will keep it for now (at least until merge done). 2262 } 2263 2264 threadLoop_exit(); 2265 2266 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2267 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2268 // put output stream into standby mode 2269 if (!mStandby) { 2270 mOutput->stream->common.standby(&mOutput->stream->common); 2271 } 2272 } 2273 2274 releaseWakeLock(); 2275 2276 ALOGV("Thread %p type %d exiting", this, mType); 2277 return false; 2278} 2279 2280// removeTracks_l() must be called with ThreadBase::mLock held 2281void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2282{ 2283 size_t count = tracksToRemove.size(); 2284 if (count) { 2285 for (size_t i=0 ; i<count ; i++) { 2286 const sp<Track>& track = tracksToRemove.itemAt(i); 2287 mActiveTracks.remove(track); 2288 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2289 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2290 if (chain != 0) { 2291 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2292 track->sessionId()); 2293 chain->decActiveTrackCnt(); 2294 } 2295 if (track->isTerminated()) { 2296 removeTrack_l(track); 2297 } 2298 } 2299 } 2300 2301} 2302 2303// ---------------------------------------------------------------------------- 2304 2305AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2306 audio_io_handle_t id, audio_devices_t device, type_t type) 2307 : PlaybackThread(audioFlinger, output, id, device, type), 2308 // mAudioMixer below 2309 // mFastMixer below 2310 mFastMixerFutex(0) 2311 // mOutputSink below 2312 // mPipeSink below 2313 // mNormalSink below 2314{ 2315 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2316 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2317 "mFrameCount=%d, mNormalFrameCount=%d", 2318 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2319 mNormalFrameCount); 2320 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2321 2322 // FIXME - Current mixer implementation only supports stereo output 2323 if (mChannelCount != FCC_2) { 2324 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2325 } 2326 2327 // create an NBAIO sink for the HAL output stream, and negotiate 2328 mOutputSink = new AudioStreamOutSink(output->stream); 2329 size_t numCounterOffers = 0; 2330 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2331 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2332 ALOG_ASSERT(index == 0); 2333 2334 // initialize fast mixer depending on configuration 2335 bool initFastMixer; 2336 switch (kUseFastMixer) { 2337 case FastMixer_Never: 2338 initFastMixer = false; 2339 break; 2340 case FastMixer_Always: 2341 initFastMixer = true; 2342 break; 2343 case FastMixer_Static: 2344 case FastMixer_Dynamic: 2345 initFastMixer = mFrameCount < mNormalFrameCount; 2346 break; 2347 } 2348 if (initFastMixer) { 2349 2350 // create a MonoPipe to connect our submix to FastMixer 2351 NBAIO_Format format = mOutputSink->format(); 2352 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2353 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2354 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2355 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2356 const NBAIO_Format offers[1] = {format}; 2357 size_t numCounterOffers = 0; 2358 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2359 ALOG_ASSERT(index == 0); 2360 monoPipe->setAvgFrames((mScreenState & 1) ? 2361 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2362 mPipeSink = monoPipe; 2363 2364#ifdef TEE_SINK 2365 if (mTeeSinkOutputEnabled) { 2366 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2367 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2368 numCounterOffers = 0; 2369 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2370 ALOG_ASSERT(index == 0); 2371 mTeeSink = teeSink; 2372 PipeReader *teeSource = new PipeReader(*teeSink); 2373 numCounterOffers = 0; 2374 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2375 ALOG_ASSERT(index == 0); 2376 mTeeSource = teeSource; 2377 } 2378#endif 2379 2380 // create fast mixer and configure it initially with just one fast track for our submix 2381 mFastMixer = new FastMixer(); 2382 FastMixerStateQueue *sq = mFastMixer->sq(); 2383#ifdef STATE_QUEUE_DUMP 2384 sq->setObserverDump(&mStateQueueObserverDump); 2385 sq->setMutatorDump(&mStateQueueMutatorDump); 2386#endif 2387 FastMixerState *state = sq->begin(); 2388 FastTrack *fastTrack = &state->mFastTracks[0]; 2389 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2390 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2391 fastTrack->mVolumeProvider = NULL; 2392 fastTrack->mGeneration++; 2393 state->mFastTracksGen++; 2394 state->mTrackMask = 1; 2395 // fast mixer will use the HAL output sink 2396 state->mOutputSink = mOutputSink.get(); 2397 state->mOutputSinkGen++; 2398 state->mFrameCount = mFrameCount; 2399 state->mCommand = FastMixerState::COLD_IDLE; 2400 // already done in constructor initialization list 2401 //mFastMixerFutex = 0; 2402 state->mColdFutexAddr = &mFastMixerFutex; 2403 state->mColdGen++; 2404 state->mDumpState = &mFastMixerDumpState; 2405#ifdef TEE_SINK 2406 state->mTeeSink = mTeeSink.get(); 2407#endif 2408 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2409 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2410 sq->end(); 2411 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2412 2413 // start the fast mixer 2414 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2415 pid_t tid = mFastMixer->getTid(); 2416 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2417 if (err != 0) { 2418 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2419 kPriorityFastMixer, getpid_cached, tid, err); 2420 } 2421 2422#ifdef AUDIO_WATCHDOG 2423 // create and start the watchdog 2424 mAudioWatchdog = new AudioWatchdog(); 2425 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2426 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2427 tid = mAudioWatchdog->getTid(); 2428 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2429 if (err != 0) { 2430 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2431 kPriorityFastMixer, getpid_cached, tid, err); 2432 } 2433#endif 2434 2435 } else { 2436 mFastMixer = NULL; 2437 } 2438 2439 switch (kUseFastMixer) { 2440 case FastMixer_Never: 2441 case FastMixer_Dynamic: 2442 mNormalSink = mOutputSink; 2443 break; 2444 case FastMixer_Always: 2445 mNormalSink = mPipeSink; 2446 break; 2447 case FastMixer_Static: 2448 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2449 break; 2450 } 2451} 2452 2453AudioFlinger::MixerThread::~MixerThread() 2454{ 2455 if (mFastMixer != NULL) { 2456 FastMixerStateQueue *sq = mFastMixer->sq(); 2457 FastMixerState *state = sq->begin(); 2458 if (state->mCommand == FastMixerState::COLD_IDLE) { 2459 int32_t old = android_atomic_inc(&mFastMixerFutex); 2460 if (old == -1) { 2461 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2462 } 2463 } 2464 state->mCommand = FastMixerState::EXIT; 2465 sq->end(); 2466 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2467 mFastMixer->join(); 2468 // Though the fast mixer thread has exited, it's state queue is still valid. 2469 // We'll use that extract the final state which contains one remaining fast track 2470 // corresponding to our sub-mix. 2471 state = sq->begin(); 2472 ALOG_ASSERT(state->mTrackMask == 1); 2473 FastTrack *fastTrack = &state->mFastTracks[0]; 2474 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2475 delete fastTrack->mBufferProvider; 2476 sq->end(false /*didModify*/); 2477 delete mFastMixer; 2478#ifdef AUDIO_WATCHDOG 2479 if (mAudioWatchdog != 0) { 2480 mAudioWatchdog->requestExit(); 2481 mAudioWatchdog->requestExitAndWait(); 2482 mAudioWatchdog.clear(); 2483 } 2484#endif 2485 } 2486 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2487 delete mAudioMixer; 2488} 2489 2490 2491uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2492{ 2493 if (mFastMixer != NULL) { 2494 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2495 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2496 } 2497 return latency; 2498} 2499 2500 2501void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2502{ 2503 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2504} 2505 2506ssize_t AudioFlinger::MixerThread::threadLoop_write() 2507{ 2508 // FIXME we should only do one push per cycle; confirm this is true 2509 // Start the fast mixer if it's not already running 2510 if (mFastMixer != NULL) { 2511 FastMixerStateQueue *sq = mFastMixer->sq(); 2512 FastMixerState *state = sq->begin(); 2513 if (state->mCommand != FastMixerState::MIX_WRITE && 2514 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2515 if (state->mCommand == FastMixerState::COLD_IDLE) { 2516 int32_t old = android_atomic_inc(&mFastMixerFutex); 2517 if (old == -1) { 2518 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2519 } 2520#ifdef AUDIO_WATCHDOG 2521 if (mAudioWatchdog != 0) { 2522 mAudioWatchdog->resume(); 2523 } 2524#endif 2525 } 2526 state->mCommand = FastMixerState::MIX_WRITE; 2527 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2528 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2529 sq->end(); 2530 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2531 if (kUseFastMixer == FastMixer_Dynamic) { 2532 mNormalSink = mPipeSink; 2533 } 2534 } else { 2535 sq->end(false /*didModify*/); 2536 } 2537 } 2538 return PlaybackThread::threadLoop_write(); 2539} 2540 2541void AudioFlinger::MixerThread::threadLoop_standby() 2542{ 2543 // Idle the fast mixer if it's currently running 2544 if (mFastMixer != NULL) { 2545 FastMixerStateQueue *sq = mFastMixer->sq(); 2546 FastMixerState *state = sq->begin(); 2547 if (!(state->mCommand & FastMixerState::IDLE)) { 2548 state->mCommand = FastMixerState::COLD_IDLE; 2549 state->mColdFutexAddr = &mFastMixerFutex; 2550 state->mColdGen++; 2551 mFastMixerFutex = 0; 2552 sq->end(); 2553 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2554 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2555 if (kUseFastMixer == FastMixer_Dynamic) { 2556 mNormalSink = mOutputSink; 2557 } 2558#ifdef AUDIO_WATCHDOG 2559 if (mAudioWatchdog != 0) { 2560 mAudioWatchdog->pause(); 2561 } 2562#endif 2563 } else { 2564 sq->end(false /*didModify*/); 2565 } 2566 } 2567 PlaybackThread::threadLoop_standby(); 2568} 2569 2570// Empty implementation for standard mixer 2571// Overridden for offloaded playback 2572void AudioFlinger::PlaybackThread::flushOutput_l() 2573{ 2574} 2575 2576bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2577{ 2578 return false; 2579} 2580 2581bool AudioFlinger::PlaybackThread::shouldStandby_l() 2582{ 2583 return !mStandby; 2584} 2585 2586bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2587{ 2588 Mutex::Autolock _l(mLock); 2589 return waitingAsyncCallback_l(); 2590} 2591 2592// shared by MIXER and DIRECT, overridden by DUPLICATING 2593void AudioFlinger::PlaybackThread::threadLoop_standby() 2594{ 2595 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2596 mOutput->stream->common.standby(&mOutput->stream->common); 2597 if (mUseAsyncWrite != 0) { 2598 mWriteBlocked = false; 2599 mDraining = false; 2600 ALOG_ASSERT(mCallbackThread != 0); 2601 mCallbackThread->setWriteBlocked(false); 2602 mCallbackThread->setDraining(false); 2603 } 2604} 2605 2606void AudioFlinger::MixerThread::threadLoop_mix() 2607{ 2608 // obtain the presentation timestamp of the next output buffer 2609 int64_t pts; 2610 status_t status = INVALID_OPERATION; 2611 2612 if (mNormalSink != 0) { 2613 status = mNormalSink->getNextWriteTimestamp(&pts); 2614 } else { 2615 status = mOutputSink->getNextWriteTimestamp(&pts); 2616 } 2617 2618 if (status != NO_ERROR) { 2619 pts = AudioBufferProvider::kInvalidPTS; 2620 } 2621 2622 // mix buffers... 2623 mAudioMixer->process(pts); 2624 mCurrentWriteLength = mixBufferSize; 2625 // increase sleep time progressively when application underrun condition clears. 2626 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2627 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2628 // such that we would underrun the audio HAL. 2629 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2630 sleepTimeShift--; 2631 } 2632 sleepTime = 0; 2633 standbyTime = systemTime() + standbyDelay; 2634 //TODO: delay standby when effects have a tail 2635} 2636 2637void AudioFlinger::MixerThread::threadLoop_sleepTime() 2638{ 2639 // If no tracks are ready, sleep once for the duration of an output 2640 // buffer size, then write 0s to the output 2641 if (sleepTime == 0) { 2642 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2643 sleepTime = activeSleepTime >> sleepTimeShift; 2644 if (sleepTime < kMinThreadSleepTimeUs) { 2645 sleepTime = kMinThreadSleepTimeUs; 2646 } 2647 // reduce sleep time in case of consecutive application underruns to avoid 2648 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2649 // duration we would end up writing less data than needed by the audio HAL if 2650 // the condition persists. 2651 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2652 sleepTimeShift++; 2653 } 2654 } else { 2655 sleepTime = idleSleepTime; 2656 } 2657 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2658 memset (mMixBuffer, 0, mixBufferSize); 2659 sleepTime = 0; 2660 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2661 "anticipated start"); 2662 } 2663 // TODO add standby time extension fct of effect tail 2664} 2665 2666// prepareTracks_l() must be called with ThreadBase::mLock held 2667AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2668 Vector< sp<Track> > *tracksToRemove) 2669{ 2670 2671 mixer_state mixerStatus = MIXER_IDLE; 2672 // find out which tracks need to be processed 2673 size_t count = mActiveTracks.size(); 2674 size_t mixedTracks = 0; 2675 size_t tracksWithEffect = 0; 2676 // counts only _active_ fast tracks 2677 size_t fastTracks = 0; 2678 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2679 2680 float masterVolume = mMasterVolume; 2681 bool masterMute = mMasterMute; 2682 2683 if (masterMute) { 2684 masterVolume = 0; 2685 } 2686 // Delegate master volume control to effect in output mix effect chain if needed 2687 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2688 if (chain != 0) { 2689 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2690 chain->setVolume_l(&v, &v); 2691 masterVolume = (float)((v + (1 << 23)) >> 24); 2692 chain.clear(); 2693 } 2694 2695 // prepare a new state to push 2696 FastMixerStateQueue *sq = NULL; 2697 FastMixerState *state = NULL; 2698 bool didModify = false; 2699 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2700 if (mFastMixer != NULL) { 2701 sq = mFastMixer->sq(); 2702 state = sq->begin(); 2703 } 2704 2705 for (size_t i=0 ; i<count ; i++) { 2706 const sp<Track> t = mActiveTracks[i].promote(); 2707 if (t == 0) { 2708 continue; 2709 } 2710 2711 // this const just means the local variable doesn't change 2712 Track* const track = t.get(); 2713 2714 // process fast tracks 2715 if (track->isFastTrack()) { 2716 2717 // It's theoretically possible (though unlikely) for a fast track to be created 2718 // and then removed within the same normal mix cycle. This is not a problem, as 2719 // the track never becomes active so it's fast mixer slot is never touched. 2720 // The converse, of removing an (active) track and then creating a new track 2721 // at the identical fast mixer slot within the same normal mix cycle, 2722 // is impossible because the slot isn't marked available until the end of each cycle. 2723 int j = track->mFastIndex; 2724 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2725 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2726 FastTrack *fastTrack = &state->mFastTracks[j]; 2727 2728 // Determine whether the track is currently in underrun condition, 2729 // and whether it had a recent underrun. 2730 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2731 FastTrackUnderruns underruns = ftDump->mUnderruns; 2732 uint32_t recentFull = (underruns.mBitFields.mFull - 2733 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2734 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2735 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2736 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2737 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2738 uint32_t recentUnderruns = recentPartial + recentEmpty; 2739 track->mObservedUnderruns = underruns; 2740 // don't count underruns that occur while stopping or pausing 2741 // or stopped which can occur when flush() is called while active 2742 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2743 track->mUnderrunCount += recentUnderruns; 2744 } 2745 2746 // This is similar to the state machine for normal tracks, 2747 // with a few modifications for fast tracks. 2748 bool isActive = true; 2749 switch (track->mState) { 2750 case TrackBase::STOPPING_1: 2751 // track stays active in STOPPING_1 state until first underrun 2752 if (recentUnderruns > 0 || track->isTerminated()) { 2753 track->mState = TrackBase::STOPPING_2; 2754 } 2755 break; 2756 case TrackBase::PAUSING: 2757 // ramp down is not yet implemented 2758 track->setPaused(); 2759 break; 2760 case TrackBase::RESUMING: 2761 // ramp up is not yet implemented 2762 track->mState = TrackBase::ACTIVE; 2763 break; 2764 case TrackBase::ACTIVE: 2765 if (recentFull > 0 || recentPartial > 0) { 2766 // track has provided at least some frames recently: reset retry count 2767 track->mRetryCount = kMaxTrackRetries; 2768 } 2769 if (recentUnderruns == 0) { 2770 // no recent underruns: stay active 2771 break; 2772 } 2773 // there has recently been an underrun of some kind 2774 if (track->sharedBuffer() == 0) { 2775 // were any of the recent underruns "empty" (no frames available)? 2776 if (recentEmpty == 0) { 2777 // no, then ignore the partial underruns as they are allowed indefinitely 2778 break; 2779 } 2780 // there has recently been an "empty" underrun: decrement the retry counter 2781 if (--(track->mRetryCount) > 0) { 2782 break; 2783 } 2784 // indicate to client process that the track was disabled because of underrun; 2785 // it will then automatically call start() when data is available 2786 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2787 // remove from active list, but state remains ACTIVE [confusing but true] 2788 isActive = false; 2789 break; 2790 } 2791 // fall through 2792 case TrackBase::STOPPING_2: 2793 case TrackBase::PAUSED: 2794 case TrackBase::STOPPED: 2795 case TrackBase::FLUSHED: // flush() while active 2796 // Check for presentation complete if track is inactive 2797 // We have consumed all the buffers of this track. 2798 // This would be incomplete if we auto-paused on underrun 2799 { 2800 size_t audioHALFrames = 2801 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2802 size_t framesWritten = mBytesWritten / mFrameSize; 2803 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2804 // track stays in active list until presentation is complete 2805 break; 2806 } 2807 } 2808 if (track->isStopping_2()) { 2809 track->mState = TrackBase::STOPPED; 2810 } 2811 if (track->isStopped()) { 2812 // Can't reset directly, as fast mixer is still polling this track 2813 // track->reset(); 2814 // So instead mark this track as needing to be reset after push with ack 2815 resetMask |= 1 << i; 2816 } 2817 isActive = false; 2818 break; 2819 case TrackBase::IDLE: 2820 default: 2821 LOG_FATAL("unexpected track state %d", track->mState); 2822 } 2823 2824 if (isActive) { 2825 // was it previously inactive? 2826 if (!(state->mTrackMask & (1 << j))) { 2827 ExtendedAudioBufferProvider *eabp = track; 2828 VolumeProvider *vp = track; 2829 fastTrack->mBufferProvider = eabp; 2830 fastTrack->mVolumeProvider = vp; 2831 fastTrack->mSampleRate = track->mSampleRate; 2832 fastTrack->mChannelMask = track->mChannelMask; 2833 fastTrack->mGeneration++; 2834 state->mTrackMask |= 1 << j; 2835 didModify = true; 2836 // no acknowledgement required for newly active tracks 2837 } 2838 // cache the combined master volume and stream type volume for fast mixer; this 2839 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2840 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2841 ++fastTracks; 2842 } else { 2843 // was it previously active? 2844 if (state->mTrackMask & (1 << j)) { 2845 fastTrack->mBufferProvider = NULL; 2846 fastTrack->mGeneration++; 2847 state->mTrackMask &= ~(1 << j); 2848 didModify = true; 2849 // If any fast tracks were removed, we must wait for acknowledgement 2850 // because we're about to decrement the last sp<> on those tracks. 2851 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2852 } else { 2853 LOG_FATAL("fast track %d should have been active", j); 2854 } 2855 tracksToRemove->add(track); 2856 // Avoids a misleading display in dumpsys 2857 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2858 } 2859 continue; 2860 } 2861 2862 { // local variable scope to avoid goto warning 2863 2864 audio_track_cblk_t* cblk = track->cblk(); 2865 2866 // The first time a track is added we wait 2867 // for all its buffers to be filled before processing it 2868 int name = track->name(); 2869 // make sure that we have enough frames to mix one full buffer. 2870 // enforce this condition only once to enable draining the buffer in case the client 2871 // app does not call stop() and relies on underrun to stop: 2872 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2873 // during last round 2874 size_t desiredFrames; 2875 uint32_t sr = track->sampleRate(); 2876 if (sr == mSampleRate) { 2877 desiredFrames = mNormalFrameCount; 2878 } else { 2879 // +1 for rounding and +1 for additional sample needed for interpolation 2880 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2881 // add frames already consumed but not yet released by the resampler 2882 // because cblk->framesReady() will include these frames 2883 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2884 // the minimum track buffer size is normally twice the number of frames necessary 2885 // to fill one buffer and the resampler should not leave more than one buffer worth 2886 // of unreleased frames after each pass, but just in case... 2887 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2888 } 2889 uint32_t minFrames = 1; 2890 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2891 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2892 minFrames = desiredFrames; 2893 } 2894 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2895 size_t framesReady; 2896 if (track->sharedBuffer() == 0) { 2897 framesReady = track->framesReady(); 2898 } else if (track->isStopped()) { 2899 framesReady = 0; 2900 } else { 2901 framesReady = 1; 2902 } 2903 if ((framesReady >= minFrames) && track->isReady() && 2904 !track->isPaused() && !track->isTerminated()) 2905 { 2906 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->server, this); 2907 2908 mixedTracks++; 2909 2910 // track->mainBuffer() != mMixBuffer means there is an effect chain 2911 // connected to the track 2912 chain.clear(); 2913 if (track->mainBuffer() != mMixBuffer) { 2914 chain = getEffectChain_l(track->sessionId()); 2915 // Delegate volume control to effect in track effect chain if needed 2916 if (chain != 0) { 2917 tracksWithEffect++; 2918 } else { 2919 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2920 "session %d", 2921 name, track->sessionId()); 2922 } 2923 } 2924 2925 2926 int param = AudioMixer::VOLUME; 2927 if (track->mFillingUpStatus == Track::FS_FILLED) { 2928 // no ramp for the first volume setting 2929 track->mFillingUpStatus = Track::FS_ACTIVE; 2930 if (track->mState == TrackBase::RESUMING) { 2931 track->mState = TrackBase::ACTIVE; 2932 param = AudioMixer::RAMP_VOLUME; 2933 } 2934 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2935 } else if (cblk->server != 0) { 2936 // If the track is stopped before the first frame was mixed, 2937 // do not apply ramp 2938 param = AudioMixer::RAMP_VOLUME; 2939 } 2940 2941 // compute volume for this track 2942 uint32_t vl, vr, va; 2943 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2944 vl = vr = va = 0; 2945 if (track->isPausing()) { 2946 track->setPaused(); 2947 } 2948 } else { 2949 2950 // read original volumes with volume control 2951 float typeVolume = mStreamTypes[track->streamType()].volume; 2952 float v = masterVolume * typeVolume; 2953 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2954 uint32_t vlr = proxy->getVolumeLR(); 2955 vl = vlr & 0xFFFF; 2956 vr = vlr >> 16; 2957 // track volumes come from shared memory, so can't be trusted and must be clamped 2958 if (vl > MAX_GAIN_INT) { 2959 ALOGV("Track left volume out of range: %04X", vl); 2960 vl = MAX_GAIN_INT; 2961 } 2962 if (vr > MAX_GAIN_INT) { 2963 ALOGV("Track right volume out of range: %04X", vr); 2964 vr = MAX_GAIN_INT; 2965 } 2966 // now apply the master volume and stream type volume 2967 vl = (uint32_t)(v * vl) << 12; 2968 vr = (uint32_t)(v * vr) << 12; 2969 // assuming master volume and stream type volume each go up to 1.0, 2970 // vl and vr are now in 8.24 format 2971 2972 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2973 // send level comes from shared memory and so may be corrupt 2974 if (sendLevel > MAX_GAIN_INT) { 2975 ALOGV("Track send level out of range: %04X", sendLevel); 2976 sendLevel = MAX_GAIN_INT; 2977 } 2978 va = (uint32_t)(v * sendLevel); 2979 } 2980 2981 // Delegate volume control to effect in track effect chain if needed 2982 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2983 // Do not ramp volume if volume is controlled by effect 2984 param = AudioMixer::VOLUME; 2985 track->mHasVolumeController = true; 2986 } else { 2987 // force no volume ramp when volume controller was just disabled or removed 2988 // from effect chain to avoid volume spike 2989 if (track->mHasVolumeController) { 2990 param = AudioMixer::VOLUME; 2991 } 2992 track->mHasVolumeController = false; 2993 } 2994 2995 // Convert volumes from 8.24 to 4.12 format 2996 // This additional clamping is needed in case chain->setVolume_l() overshot 2997 vl = (vl + (1 << 11)) >> 12; 2998 if (vl > MAX_GAIN_INT) { 2999 vl = MAX_GAIN_INT; 3000 } 3001 vr = (vr + (1 << 11)) >> 12; 3002 if (vr > MAX_GAIN_INT) { 3003 vr = MAX_GAIN_INT; 3004 } 3005 3006 if (va > MAX_GAIN_INT) { 3007 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3008 } 3009 3010 // XXX: these things DON'T need to be done each time 3011 mAudioMixer->setBufferProvider(name, track); 3012 mAudioMixer->enable(name); 3013 3014 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3015 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3016 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3017 mAudioMixer->setParameter( 3018 name, 3019 AudioMixer::TRACK, 3020 AudioMixer::FORMAT, (void *)track->format()); 3021 mAudioMixer->setParameter( 3022 name, 3023 AudioMixer::TRACK, 3024 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3025 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3026 uint32_t maxSampleRate = mSampleRate * 2; 3027 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3028 if (reqSampleRate == 0) { 3029 reqSampleRate = mSampleRate; 3030 } else if (reqSampleRate > maxSampleRate) { 3031 reqSampleRate = maxSampleRate; 3032 } 3033 mAudioMixer->setParameter( 3034 name, 3035 AudioMixer::RESAMPLE, 3036 AudioMixer::SAMPLE_RATE, 3037 (void *)reqSampleRate); 3038 mAudioMixer->setParameter( 3039 name, 3040 AudioMixer::TRACK, 3041 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3042 mAudioMixer->setParameter( 3043 name, 3044 AudioMixer::TRACK, 3045 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3046 3047 // reset retry count 3048 track->mRetryCount = kMaxTrackRetries; 3049 3050 // If one track is ready, set the mixer ready if: 3051 // - the mixer was not ready during previous round OR 3052 // - no other track is not ready 3053 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3054 mixerStatus != MIXER_TRACKS_ENABLED) { 3055 mixerStatus = MIXER_TRACKS_READY; 3056 } 3057 } else { 3058 // only implemented for normal tracks, not fast tracks 3059 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3060 // we missed desiredFrames whatever the actual number of frames missing was 3061 cblk->u.mStreaming.mUnderrunFrames += desiredFrames; 3062 // FIXME also wake futex so that underrun is noticed more quickly 3063 (void) android_atomic_or(CBLK_UNDERRUN, &cblk->flags); 3064 } 3065 // clear effect chain input buffer if an active track underruns to avoid sending 3066 // previous audio buffer again to effects 3067 chain = getEffectChain_l(track->sessionId()); 3068 if (chain != 0) { 3069 chain->clearInputBuffer(); 3070 } 3071 3072 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->server, this); 3073 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3074 track->isStopped() || track->isPaused()) { 3075 // We have consumed all the buffers of this track. 3076 // Remove it from the list of active tracks. 3077 // TODO: use actual buffer filling status instead of latency when available from 3078 // audio HAL 3079 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3080 size_t framesWritten = mBytesWritten / mFrameSize; 3081 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3082 if (track->isStopped()) { 3083 track->reset(); 3084 } 3085 tracksToRemove->add(track); 3086 } 3087 } else { 3088 track->mUnderrunCount++; 3089 // No buffers for this track. Give it a few chances to 3090 // fill a buffer, then remove it from active list. 3091 if (--(track->mRetryCount) <= 0) { 3092 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3093 tracksToRemove->add(track); 3094 // indicate to client process that the track was disabled because of underrun; 3095 // it will then automatically call start() when data is available 3096 android_atomic_or(CBLK_DISABLED, &cblk->flags); 3097 // If one track is not ready, mark the mixer also not ready if: 3098 // - the mixer was ready during previous round OR 3099 // - no other track is ready 3100 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3101 mixerStatus != MIXER_TRACKS_READY) { 3102 mixerStatus = MIXER_TRACKS_ENABLED; 3103 } 3104 } 3105 mAudioMixer->disable(name); 3106 } 3107 3108 } // local variable scope to avoid goto warning 3109track_is_ready: ; 3110 3111 } 3112 3113 // Push the new FastMixer state if necessary 3114 bool pauseAudioWatchdog = false; 3115 if (didModify) { 3116 state->mFastTracksGen++; 3117 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3118 if (kUseFastMixer == FastMixer_Dynamic && 3119 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3120 state->mCommand = FastMixerState::COLD_IDLE; 3121 state->mColdFutexAddr = &mFastMixerFutex; 3122 state->mColdGen++; 3123 mFastMixerFutex = 0; 3124 if (kUseFastMixer == FastMixer_Dynamic) { 3125 mNormalSink = mOutputSink; 3126 } 3127 // If we go into cold idle, need to wait for acknowledgement 3128 // so that fast mixer stops doing I/O. 3129 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3130 pauseAudioWatchdog = true; 3131 } 3132 } 3133 if (sq != NULL) { 3134 sq->end(didModify); 3135 sq->push(block); 3136 } 3137#ifdef AUDIO_WATCHDOG 3138 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3139 mAudioWatchdog->pause(); 3140 } 3141#endif 3142 3143 // Now perform the deferred reset on fast tracks that have stopped 3144 while (resetMask != 0) { 3145 size_t i = __builtin_ctz(resetMask); 3146 ALOG_ASSERT(i < count); 3147 resetMask &= ~(1 << i); 3148 sp<Track> t = mActiveTracks[i].promote(); 3149 if (t == 0) { 3150 continue; 3151 } 3152 Track* track = t.get(); 3153 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3154 track->reset(); 3155 } 3156 3157 // remove all the tracks that need to be... 3158 removeTracks_l(*tracksToRemove); 3159 3160 // mix buffer must be cleared if all tracks are connected to an 3161 // effect chain as in this case the mixer will not write to 3162 // mix buffer and track effects will accumulate into it 3163 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3164 (mixedTracks == 0 && fastTracks > 0))) { 3165 // FIXME as a performance optimization, should remember previous zero status 3166 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3167 } 3168 3169 // if any fast tracks, then status is ready 3170 mMixerStatusIgnoringFastTracks = mixerStatus; 3171 if (fastTracks > 0) { 3172 mixerStatus = MIXER_TRACKS_READY; 3173 } 3174 return mixerStatus; 3175} 3176 3177// getTrackName_l() must be called with ThreadBase::mLock held 3178int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3179{ 3180 return mAudioMixer->getTrackName(channelMask, sessionId); 3181} 3182 3183// deleteTrackName_l() must be called with ThreadBase::mLock held 3184void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3185{ 3186 ALOGV("remove track (%d) and delete from mixer", name); 3187 mAudioMixer->deleteTrackName(name); 3188} 3189 3190// checkForNewParameters_l() must be called with ThreadBase::mLock held 3191bool AudioFlinger::MixerThread::checkForNewParameters_l() 3192{ 3193 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3194 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3195 bool reconfig = false; 3196 3197 while (!mNewParameters.isEmpty()) { 3198 3199 if (mFastMixer != NULL) { 3200 FastMixerStateQueue *sq = mFastMixer->sq(); 3201 FastMixerState *state = sq->begin(); 3202 if (!(state->mCommand & FastMixerState::IDLE)) { 3203 previousCommand = state->mCommand; 3204 state->mCommand = FastMixerState::HOT_IDLE; 3205 sq->end(); 3206 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3207 } else { 3208 sq->end(false /*didModify*/); 3209 } 3210 } 3211 3212 status_t status = NO_ERROR; 3213 String8 keyValuePair = mNewParameters[0]; 3214 AudioParameter param = AudioParameter(keyValuePair); 3215 int value; 3216 3217 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3218 reconfig = true; 3219 } 3220 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3221 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3222 status = BAD_VALUE; 3223 } else { 3224 reconfig = true; 3225 } 3226 } 3227 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3228 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3229 status = BAD_VALUE; 3230 } else { 3231 reconfig = true; 3232 } 3233 } 3234 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3235 // do not accept frame count changes if tracks are open as the track buffer 3236 // size depends on frame count and correct behavior would not be guaranteed 3237 // if frame count is changed after track creation 3238 if (!mTracks.isEmpty()) { 3239 status = INVALID_OPERATION; 3240 } else { 3241 reconfig = true; 3242 } 3243 } 3244 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3245#ifdef ADD_BATTERY_DATA 3246 // when changing the audio output device, call addBatteryData to notify 3247 // the change 3248 if (mOutDevice != value) { 3249 uint32_t params = 0; 3250 // check whether speaker is on 3251 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3252 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3253 } 3254 3255 audio_devices_t deviceWithoutSpeaker 3256 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3257 // check if any other device (except speaker) is on 3258 if (value & deviceWithoutSpeaker ) { 3259 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3260 } 3261 3262 if (params != 0) { 3263 addBatteryData(params); 3264 } 3265 } 3266#endif 3267 3268 // forward device change to effects that have requested to be 3269 // aware of attached audio device. 3270 if (value != AUDIO_DEVICE_NONE) { 3271 mOutDevice = value; 3272 for (size_t i = 0; i < mEffectChains.size(); i++) { 3273 mEffectChains[i]->setDevice_l(mOutDevice); 3274 } 3275 } 3276 } 3277 3278 if (status == NO_ERROR) { 3279 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3280 keyValuePair.string()); 3281 if (!mStandby && status == INVALID_OPERATION) { 3282 mOutput->stream->common.standby(&mOutput->stream->common); 3283 mStandby = true; 3284 mBytesWritten = 0; 3285 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3286 keyValuePair.string()); 3287 } 3288 if (status == NO_ERROR && reconfig) { 3289 readOutputParameters(); 3290 delete mAudioMixer; 3291 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3292 for (size_t i = 0; i < mTracks.size() ; i++) { 3293 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3294 if (name < 0) { 3295 break; 3296 } 3297 mTracks[i]->mName = name; 3298 } 3299 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3300 } 3301 } 3302 3303 mNewParameters.removeAt(0); 3304 3305 mParamStatus = status; 3306 mParamCond.signal(); 3307 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3308 // already timed out waiting for the status and will never signal the condition. 3309 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3310 } 3311 3312 if (!(previousCommand & FastMixerState::IDLE)) { 3313 ALOG_ASSERT(mFastMixer != NULL); 3314 FastMixerStateQueue *sq = mFastMixer->sq(); 3315 FastMixerState *state = sq->begin(); 3316 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3317 state->mCommand = previousCommand; 3318 sq->end(); 3319 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3320 } 3321 3322 return reconfig; 3323} 3324 3325 3326void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3327{ 3328 const size_t SIZE = 256; 3329 char buffer[SIZE]; 3330 String8 result; 3331 3332 PlaybackThread::dumpInternals(fd, args); 3333 3334 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3335 result.append(buffer); 3336 write(fd, result.string(), result.size()); 3337 3338 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3339 const FastMixerDumpState copy(mFastMixerDumpState); 3340 copy.dump(fd); 3341 3342#ifdef STATE_QUEUE_DUMP 3343 // Similar for state queue 3344 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3345 observerCopy.dump(fd); 3346 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3347 mutatorCopy.dump(fd); 3348#endif 3349 3350#ifdef TEE_SINK 3351 // Write the tee output to a .wav file 3352 dumpTee(fd, mTeeSource, mId); 3353#endif 3354 3355#ifdef AUDIO_WATCHDOG 3356 if (mAudioWatchdog != 0) { 3357 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3358 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3359 wdCopy.dump(fd); 3360 } 3361#endif 3362} 3363 3364uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3365{ 3366 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3367} 3368 3369uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3370{ 3371 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3372} 3373 3374void AudioFlinger::MixerThread::cacheParameters_l() 3375{ 3376 PlaybackThread::cacheParameters_l(); 3377 3378 // FIXME: Relaxed timing because of a certain device that can't meet latency 3379 // Should be reduced to 2x after the vendor fixes the driver issue 3380 // increase threshold again due to low power audio mode. The way this warning 3381 // threshold is calculated and its usefulness should be reconsidered anyway. 3382 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3383} 3384 3385// ---------------------------------------------------------------------------- 3386 3387AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3388 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3389 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3390 // mLeftVolFloat, mRightVolFloat 3391{ 3392} 3393 3394AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3395 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3396 ThreadBase::type_t type) 3397 : PlaybackThread(audioFlinger, output, id, device, type) 3398 // mLeftVolFloat, mRightVolFloat 3399{ 3400} 3401 3402AudioFlinger::DirectOutputThread::~DirectOutputThread() 3403{ 3404} 3405 3406void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3407{ 3408 audio_track_cblk_t* cblk = track->cblk(); 3409 float left, right; 3410 3411 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3412 left = right = 0; 3413 } else { 3414 float typeVolume = mStreamTypes[track->streamType()].volume; 3415 float v = mMasterVolume * typeVolume; 3416 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3417 uint32_t vlr = proxy->getVolumeLR(); 3418 float v_clamped = v * (vlr & 0xFFFF); 3419 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3420 left = v_clamped/MAX_GAIN; 3421 v_clamped = v * (vlr >> 16); 3422 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3423 right = v_clamped/MAX_GAIN; 3424 } 3425 3426 if (lastTrack) { 3427 if (left != mLeftVolFloat || right != mRightVolFloat) { 3428 mLeftVolFloat = left; 3429 mRightVolFloat = right; 3430 3431 // Convert volumes from float to 8.24 3432 uint32_t vl = (uint32_t)(left * (1 << 24)); 3433 uint32_t vr = (uint32_t)(right * (1 << 24)); 3434 3435 // Delegate volume control to effect in track effect chain if needed 3436 // only one effect chain can be present on DirectOutputThread, so if 3437 // there is one, the track is connected to it 3438 if (!mEffectChains.isEmpty()) { 3439 mEffectChains[0]->setVolume_l(&vl, &vr); 3440 left = (float)vl / (1 << 24); 3441 right = (float)vr / (1 << 24); 3442 } 3443 if (mOutput->stream->set_volume) { 3444 mOutput->stream->set_volume(mOutput->stream, left, right); 3445 } 3446 } 3447 } 3448} 3449 3450 3451AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3452 Vector< sp<Track> > *tracksToRemove 3453) 3454{ 3455 size_t count = mActiveTracks.size(); 3456 mixer_state mixerStatus = MIXER_IDLE; 3457 3458 // find out which tracks need to be processed 3459 for (size_t i = 0; i < count; i++) { 3460 sp<Track> t = mActiveTracks[i].promote(); 3461 // The track died recently 3462 if (t == 0) { 3463 continue; 3464 } 3465 3466 Track* const track = t.get(); 3467 audio_track_cblk_t* cblk = track->cblk(); 3468 3469 // The first time a track is added we wait 3470 // for all its buffers to be filled before processing it 3471 uint32_t minFrames; 3472 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3473 minFrames = mNormalFrameCount; 3474 } else { 3475 minFrames = 1; 3476 } 3477 // Only consider last track started for volume and mixer state control. 3478 // This is the last entry in mActiveTracks unless a track underruns. 3479 // As we only care about the transition phase between two tracks on a 3480 // direct output, it is not a problem to ignore the underrun case. 3481 bool last = (i == (count - 1)); 3482 3483 if ((track->framesReady() >= minFrames) && track->isReady() && 3484 !track->isPaused() && !track->isTerminated()) 3485 { 3486 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3487 3488 if (track->mFillingUpStatus == Track::FS_FILLED) { 3489 track->mFillingUpStatus = Track::FS_ACTIVE; 3490 mLeftVolFloat = mRightVolFloat = 0; 3491 if (track->mState == TrackBase::RESUMING) { 3492 track->mState = TrackBase::ACTIVE; 3493 } 3494 } 3495 3496 // compute volume for this track 3497 processVolume_l(track, last); 3498 if (last) { 3499 // reset retry count 3500 track->mRetryCount = kMaxTrackRetriesDirect; 3501 mActiveTrack = t; 3502 mixerStatus = MIXER_TRACKS_READY; 3503 } 3504 } else { 3505 // clear effect chain input buffer if the last active track started underruns 3506 // to avoid sending previous audio buffer again to effects 3507 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3508 mEffectChains[0]->clearInputBuffer(); 3509 } 3510 3511 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3512 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3513 track->isStopped() || track->isPaused()) { 3514 // We have consumed all the buffers of this track. 3515 // Remove it from the list of active tracks. 3516 // TODO: implement behavior for compressed audio 3517 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3518 size_t framesWritten = mBytesWritten / mFrameSize; 3519 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3520 if (track->isStopped()) { 3521 track->reset(); 3522 } 3523 tracksToRemove->add(track); 3524 } 3525 } else { 3526 // No buffers for this track. Give it a few chances to 3527 // fill a buffer, then remove it from active list. 3528 // Only consider last track started for mixer state control 3529 if (--(track->mRetryCount) <= 0) { 3530 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3531 tracksToRemove->add(track); 3532 } else if (last) { 3533 mixerStatus = MIXER_TRACKS_ENABLED; 3534 } 3535 } 3536 } 3537 } 3538 3539 // remove all the tracks that need to be... 3540 removeTracks_l(*tracksToRemove); 3541 3542 return mixerStatus; 3543} 3544 3545void AudioFlinger::DirectOutputThread::threadLoop_mix() 3546{ 3547 size_t frameCount = mFrameCount; 3548 int8_t *curBuf = (int8_t *)mMixBuffer; 3549 // output audio to hardware 3550 while (frameCount) { 3551 AudioBufferProvider::Buffer buffer; 3552 buffer.frameCount = frameCount; 3553 mActiveTrack->getNextBuffer(&buffer); 3554 if (buffer.raw == NULL) { 3555 memset(curBuf, 0, frameCount * mFrameSize); 3556 break; 3557 } 3558 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3559 frameCount -= buffer.frameCount; 3560 curBuf += buffer.frameCount * mFrameSize; 3561 mActiveTrack->releaseBuffer(&buffer); 3562 } 3563 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3564 sleepTime = 0; 3565 standbyTime = systemTime() + standbyDelay; 3566 mActiveTrack.clear(); 3567} 3568 3569void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3570{ 3571 if (sleepTime == 0) { 3572 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3573 sleepTime = activeSleepTime; 3574 } else { 3575 sleepTime = idleSleepTime; 3576 } 3577 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3578 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3579 sleepTime = 0; 3580 } 3581} 3582 3583// getTrackName_l() must be called with ThreadBase::mLock held 3584int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3585 int sessionId) 3586{ 3587 return 0; 3588} 3589 3590// deleteTrackName_l() must be called with ThreadBase::mLock held 3591void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3592{ 3593} 3594 3595// checkForNewParameters_l() must be called with ThreadBase::mLock held 3596bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3597{ 3598 bool reconfig = false; 3599 3600 while (!mNewParameters.isEmpty()) { 3601 status_t status = NO_ERROR; 3602 String8 keyValuePair = mNewParameters[0]; 3603 AudioParameter param = AudioParameter(keyValuePair); 3604 int value; 3605 3606 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3607 // do not accept frame count changes if tracks are open as the track buffer 3608 // size depends on frame count and correct behavior would not be garantied 3609 // if frame count is changed after track creation 3610 if (!mTracks.isEmpty()) { 3611 status = INVALID_OPERATION; 3612 } else { 3613 reconfig = true; 3614 } 3615 } 3616 if (status == NO_ERROR) { 3617 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3618 keyValuePair.string()); 3619 if (!mStandby && status == INVALID_OPERATION) { 3620 mOutput->stream->common.standby(&mOutput->stream->common); 3621 mStandby = true; 3622 mBytesWritten = 0; 3623 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3624 keyValuePair.string()); 3625 } 3626 if (status == NO_ERROR && reconfig) { 3627 readOutputParameters(); 3628 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3629 } 3630 } 3631 3632 mNewParameters.removeAt(0); 3633 3634 mParamStatus = status; 3635 mParamCond.signal(); 3636 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3637 // already timed out waiting for the status and will never signal the condition. 3638 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3639 } 3640 return reconfig; 3641} 3642 3643uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3644{ 3645 uint32_t time; 3646 if (audio_is_linear_pcm(mFormat)) { 3647 time = PlaybackThread::activeSleepTimeUs(); 3648 } else { 3649 time = 10000; 3650 } 3651 return time; 3652} 3653 3654uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3655{ 3656 uint32_t time; 3657 if (audio_is_linear_pcm(mFormat)) { 3658 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3659 } else { 3660 time = 10000; 3661 } 3662 return time; 3663} 3664 3665uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3666{ 3667 uint32_t time; 3668 if (audio_is_linear_pcm(mFormat)) { 3669 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3670 } else { 3671 time = 10000; 3672 } 3673 return time; 3674} 3675 3676void AudioFlinger::DirectOutputThread::cacheParameters_l() 3677{ 3678 PlaybackThread::cacheParameters_l(); 3679 3680 // use shorter standby delay as on normal output to release 3681 // hardware resources as soon as possible 3682 standbyDelay = microseconds(activeSleepTime*2); 3683} 3684 3685// ---------------------------------------------------------------------------- 3686 3687AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3688 const sp<AudioFlinger::OffloadThread>& offloadThread) 3689 : Thread(false /*canCallJava*/), 3690 mOffloadThread(offloadThread), 3691 mWriteBlocked(false), 3692 mDraining(false) 3693{ 3694} 3695 3696AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3697{ 3698} 3699 3700void AudioFlinger::AsyncCallbackThread::onFirstRef() 3701{ 3702 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3703} 3704 3705bool AudioFlinger::AsyncCallbackThread::threadLoop() 3706{ 3707 while (!exitPending()) { 3708 bool writeBlocked; 3709 bool draining; 3710 3711 { 3712 Mutex::Autolock _l(mLock); 3713 mWaitWorkCV.wait(mLock); 3714 if (exitPending()) { 3715 break; 3716 } 3717 writeBlocked = mWriteBlocked; 3718 draining = mDraining; 3719 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3720 } 3721 { 3722 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3723 if (offloadThread != 0) { 3724 if (writeBlocked == false) { 3725 offloadThread->setWriteBlocked(false); 3726 } 3727 if (draining == false) { 3728 offloadThread->setDraining(false); 3729 } 3730 } 3731 } 3732 } 3733 return false; 3734} 3735 3736void AudioFlinger::AsyncCallbackThread::exit() 3737{ 3738 ALOGV("AsyncCallbackThread::exit"); 3739 Mutex::Autolock _l(mLock); 3740 requestExit(); 3741 mWaitWorkCV.broadcast(); 3742} 3743 3744void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3745{ 3746 Mutex::Autolock _l(mLock); 3747 mWriteBlocked = value; 3748 if (!value) { 3749 mWaitWorkCV.signal(); 3750 } 3751} 3752 3753void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3754{ 3755 Mutex::Autolock _l(mLock); 3756 mDraining = value; 3757 if (!value) { 3758 mWaitWorkCV.signal(); 3759 } 3760} 3761 3762 3763// ---------------------------------------------------------------------------- 3764AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3765 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3766 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3767 mHwPaused(false), 3768 mPausedBytesRemaining(0) 3769{ 3770 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3771} 3772 3773AudioFlinger::OffloadThread::~OffloadThread() 3774{ 3775 mPreviousTrack.clear(); 3776} 3777 3778void AudioFlinger::OffloadThread::threadLoop_exit() 3779{ 3780 if (mFlushPending || mHwPaused) { 3781 // If a flush is pending or track was paused, just discard buffered data 3782 flushHw_l(); 3783 } else { 3784 mMixerStatus = MIXER_DRAIN_ALL; 3785 threadLoop_drain(); 3786 } 3787 mCallbackThread->exit(); 3788 PlaybackThread::threadLoop_exit(); 3789} 3790 3791AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3792 Vector< sp<Track> > *tracksToRemove 3793) 3794{ 3795 ALOGV("OffloadThread::prepareTracks_l"); 3796 size_t count = mActiveTracks.size(); 3797 3798 mixer_state mixerStatus = MIXER_IDLE; 3799 if (mFlushPending) { 3800 flushHw_l(); 3801 mFlushPending = false; 3802 } 3803 // find out which tracks need to be processed 3804 for (size_t i = 0; i < count; i++) { 3805 sp<Track> t = mActiveTracks[i].promote(); 3806 // The track died recently 3807 if (t == 0) { 3808 continue; 3809 } 3810 Track* const track = t.get(); 3811 audio_track_cblk_t* cblk = track->cblk(); 3812 if (mPreviousTrack != NULL) { 3813 if (t != mPreviousTrack) { 3814 // Flush any data still being written from last track 3815 mBytesRemaining = 0; 3816 if (mPausedBytesRemaining) { 3817 // Last track was paused so we also need to flush saved 3818 // mixbuffer state and invalidate track so that it will 3819 // re-submit that unwritten data when it is next resumed 3820 mPausedBytesRemaining = 0; 3821 // Invalidate is a bit drastic - would be more efficient 3822 // to have a flag to tell client that some of the 3823 // previously written data was lost 3824 mPreviousTrack->invalidate(); 3825 } 3826 } 3827 } 3828 mPreviousTrack = t; 3829 bool last = (i == (count - 1)); 3830 if (track->isPausing()) { 3831 track->setPaused(); 3832 if (last) { 3833 if (!mHwPaused) { 3834 mOutput->stream->pause(mOutput->stream); 3835 mHwPaused = true; 3836 } 3837 // If we were part way through writing the mixbuffer to 3838 // the HAL we must save this until we resume 3839 // BUG - this will be wrong if a different track is made active, 3840 // in that case we want to discard the pending data in the 3841 // mixbuffer and tell the client to present it again when the 3842 // track is resumed 3843 mPausedWriteLength = mCurrentWriteLength; 3844 mPausedBytesRemaining = mBytesRemaining; 3845 mBytesRemaining = 0; // stop writing 3846 } 3847 tracksToRemove->add(track); 3848 } else if (track->framesReady() && track->isReady() && 3849 !track->isPaused() && !track->isTerminated()) { 3850 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->server); 3851 if (track->mFillingUpStatus == Track::FS_FILLED) { 3852 track->mFillingUpStatus = Track::FS_ACTIVE; 3853 mLeftVolFloat = mRightVolFloat = 0; 3854 if (track->mState == TrackBase::RESUMING) { 3855 if (mPausedBytesRemaining) { 3856 // Need to continue write that was interrupted 3857 mCurrentWriteLength = mPausedWriteLength; 3858 mBytesRemaining = mPausedBytesRemaining; 3859 mPausedBytesRemaining = 0; 3860 } 3861 track->mState = TrackBase::ACTIVE; 3862 } 3863 } 3864 3865 if (last) { 3866 if (mHwPaused) { 3867 mOutput->stream->resume(mOutput->stream); 3868 mHwPaused = false; 3869 // threadLoop_mix() will handle the case that we need to 3870 // resume an interrupted write 3871 } 3872 // reset retry count 3873 track->mRetryCount = kMaxTrackRetriesOffload; 3874 mActiveTrack = t; 3875 mixerStatus = MIXER_TRACKS_READY; 3876 } 3877 } else { 3878 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->server); 3879 if (track->isStopping_1()) { 3880 // Hardware buffer can hold a large amount of audio so we must 3881 // wait for all current track's data to drain before we say 3882 // that the track is stopped. 3883 if (mBytesRemaining == 0) { 3884 // Only start draining when all data in mixbuffer 3885 // has been written 3886 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3887 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3888 sleepTime = 0; 3889 standbyTime = systemTime() + standbyDelay; 3890 if (last) { 3891 mixerStatus = MIXER_DRAIN_TRACK; 3892 if (mHwPaused) { 3893 // It is possible to move from PAUSED to STOPPING_1 without 3894 // a resume so we must ensure hardware is running 3895 mOutput->stream->resume(mOutput->stream); 3896 mHwPaused = false; 3897 } 3898 } 3899 } 3900 } else if (track->isStopping_2()) { 3901 // Drain has completed, signal presentation complete 3902 if (!mDraining || !last) { 3903 track->mState = TrackBase::STOPPED; 3904 size_t audioHALFrames = 3905 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3906 size_t framesWritten = 3907 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3908 track->presentationComplete(framesWritten, audioHALFrames); 3909 track->reset(); 3910 tracksToRemove->add(track); 3911 } 3912 } else { 3913 // No buffers for this track. Give it a few chances to 3914 // fill a buffer, then remove it from active list. 3915 if (--(track->mRetryCount) <= 0) { 3916 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3917 track->name()); 3918 tracksToRemove->add(track); 3919 } else if (last){ 3920 mixerStatus = MIXER_TRACKS_ENABLED; 3921 } 3922 } 3923 } 3924 // compute volume for this track 3925 processVolume_l(track, last); 3926 } 3927 // remove all the tracks that need to be... 3928 removeTracks_l(*tracksToRemove); 3929 3930 return mixerStatus; 3931} 3932 3933void AudioFlinger::OffloadThread::flushOutput_l() 3934{ 3935 mFlushPending = true; 3936} 3937 3938// must be called with thread mutex locked 3939bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3940{ 3941 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3942 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3943 return true; 3944 } 3945 return false; 3946} 3947 3948// must be called with thread mutex locked 3949bool AudioFlinger::OffloadThread::shouldStandby_l() 3950{ 3951 bool TrackPaused = false; 3952 3953 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3954 // after a timeout and we will enter standby then. 3955 if (mTracks.size() > 0) { 3956 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3957 } 3958 3959 return !mStandby && !TrackPaused; 3960} 3961 3962 3963bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3964{ 3965 Mutex::Autolock _l(mLock); 3966 return waitingAsyncCallback_l(); 3967} 3968 3969void AudioFlinger::OffloadThread::flushHw_l() 3970{ 3971 mOutput->stream->flush(mOutput->stream); 3972 // Flush anything still waiting in the mixbuffer 3973 mCurrentWriteLength = 0; 3974 mBytesRemaining = 0; 3975 mPausedWriteLength = 0; 3976 mPausedBytesRemaining = 0; 3977 if (mUseAsyncWrite) { 3978 mWriteBlocked = false; 3979 mDraining = false; 3980 ALOG_ASSERT(mCallbackThread != 0); 3981 mCallbackThread->setWriteBlocked(false); 3982 mCallbackThread->setDraining(false); 3983 } 3984} 3985 3986// ---------------------------------------------------------------------------- 3987 3988AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3989 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3990 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3991 DUPLICATING), 3992 mWaitTimeMs(UINT_MAX) 3993{ 3994 addOutputTrack(mainThread); 3995} 3996 3997AudioFlinger::DuplicatingThread::~DuplicatingThread() 3998{ 3999 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4000 mOutputTracks[i]->destroy(); 4001 } 4002} 4003 4004void AudioFlinger::DuplicatingThread::threadLoop_mix() 4005{ 4006 // mix buffers... 4007 if (outputsReady(outputTracks)) { 4008 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4009 } else { 4010 memset(mMixBuffer, 0, mixBufferSize); 4011 } 4012 sleepTime = 0; 4013 writeFrames = mNormalFrameCount; 4014 mCurrentWriteLength = mixBufferSize; 4015 standbyTime = systemTime() + standbyDelay; 4016} 4017 4018void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4019{ 4020 if (sleepTime == 0) { 4021 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4022 sleepTime = activeSleepTime; 4023 } else { 4024 sleepTime = idleSleepTime; 4025 } 4026 } else if (mBytesWritten != 0) { 4027 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4028 writeFrames = mNormalFrameCount; 4029 memset(mMixBuffer, 0, mixBufferSize); 4030 } else { 4031 // flush remaining overflow buffers in output tracks 4032 writeFrames = 0; 4033 } 4034 sleepTime = 0; 4035 } 4036} 4037 4038ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4039{ 4040 for (size_t i = 0; i < outputTracks.size(); i++) { 4041 outputTracks[i]->write(mMixBuffer, writeFrames); 4042 } 4043 return (ssize_t)mixBufferSize; 4044} 4045 4046void AudioFlinger::DuplicatingThread::threadLoop_standby() 4047{ 4048 // DuplicatingThread implements standby by stopping all tracks 4049 for (size_t i = 0; i < outputTracks.size(); i++) { 4050 outputTracks[i]->stop(); 4051 } 4052} 4053 4054void AudioFlinger::DuplicatingThread::saveOutputTracks() 4055{ 4056 outputTracks = mOutputTracks; 4057} 4058 4059void AudioFlinger::DuplicatingThread::clearOutputTracks() 4060{ 4061 outputTracks.clear(); 4062} 4063 4064void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4065{ 4066 Mutex::Autolock _l(mLock); 4067 // FIXME explain this formula 4068 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4069 OutputTrack *outputTrack = new OutputTrack(thread, 4070 this, 4071 mSampleRate, 4072 mFormat, 4073 mChannelMask, 4074 frameCount); 4075 if (outputTrack->cblk() != NULL) { 4076 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4077 mOutputTracks.add(outputTrack); 4078 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4079 updateWaitTime_l(); 4080 } 4081} 4082 4083void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4084{ 4085 Mutex::Autolock _l(mLock); 4086 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4087 if (mOutputTracks[i]->thread() == thread) { 4088 mOutputTracks[i]->destroy(); 4089 mOutputTracks.removeAt(i); 4090 updateWaitTime_l(); 4091 return; 4092 } 4093 } 4094 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4095} 4096 4097// caller must hold mLock 4098void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4099{ 4100 mWaitTimeMs = UINT_MAX; 4101 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4102 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4103 if (strong != 0) { 4104 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4105 if (waitTimeMs < mWaitTimeMs) { 4106 mWaitTimeMs = waitTimeMs; 4107 } 4108 } 4109 } 4110} 4111 4112 4113bool AudioFlinger::DuplicatingThread::outputsReady( 4114 const SortedVector< sp<OutputTrack> > &outputTracks) 4115{ 4116 for (size_t i = 0; i < outputTracks.size(); i++) { 4117 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4118 if (thread == 0) { 4119 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4120 outputTracks[i].get()); 4121 return false; 4122 } 4123 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4124 // see note at standby() declaration 4125 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4126 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4127 thread.get()); 4128 return false; 4129 } 4130 } 4131 return true; 4132} 4133 4134uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4135{ 4136 return (mWaitTimeMs * 1000) / 2; 4137} 4138 4139void AudioFlinger::DuplicatingThread::cacheParameters_l() 4140{ 4141 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4142 updateWaitTime_l(); 4143 4144 MixerThread::cacheParameters_l(); 4145} 4146 4147// ---------------------------------------------------------------------------- 4148// Record 4149// ---------------------------------------------------------------------------- 4150 4151AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4152 AudioStreamIn *input, 4153 uint32_t sampleRate, 4154 audio_channel_mask_t channelMask, 4155 audio_io_handle_t id, 4156 audio_devices_t outDevice, 4157 audio_devices_t inDevice 4158#ifdef TEE_SINK 4159 , const sp<NBAIO_Sink>& teeSink 4160#endif 4161 ) : 4162 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4163 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4164 // mRsmpInIndex and mBufferSize set by readInputParameters() 4165 mReqChannelCount(popcount(channelMask)), 4166 mReqSampleRate(sampleRate) 4167 // mBytesRead is only meaningful while active, and so is cleared in start() 4168 // (but might be better to also clear here for dump?) 4169#ifdef TEE_SINK 4170 , mTeeSink(teeSink) 4171#endif 4172{ 4173 snprintf(mName, kNameLength, "AudioIn_%X", id); 4174 4175 readInputParameters(); 4176 4177} 4178 4179 4180AudioFlinger::RecordThread::~RecordThread() 4181{ 4182 delete[] mRsmpInBuffer; 4183 delete mResampler; 4184 delete[] mRsmpOutBuffer; 4185} 4186 4187void AudioFlinger::RecordThread::onFirstRef() 4188{ 4189 run(mName, PRIORITY_URGENT_AUDIO); 4190} 4191 4192status_t AudioFlinger::RecordThread::readyToRun() 4193{ 4194 status_t status = initCheck(); 4195 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4196 return status; 4197} 4198 4199bool AudioFlinger::RecordThread::threadLoop() 4200{ 4201 AudioBufferProvider::Buffer buffer; 4202 sp<RecordTrack> activeTrack; 4203 Vector< sp<EffectChain> > effectChains; 4204 4205 nsecs_t lastWarning = 0; 4206 4207 inputStandBy(); 4208 acquireWakeLock(); 4209 4210 // used to verify we've read at least once before evaluating how many bytes were read 4211 bool readOnce = false; 4212 4213 // start recording 4214 while (!exitPending()) { 4215 4216 processConfigEvents(); 4217 4218 { // scope for mLock 4219 Mutex::Autolock _l(mLock); 4220 checkForNewParameters_l(); 4221 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4222 standby(); 4223 4224 if (exitPending()) { 4225 break; 4226 } 4227 4228 releaseWakeLock_l(); 4229 ALOGV("RecordThread: loop stopping"); 4230 // go to sleep 4231 mWaitWorkCV.wait(mLock); 4232 ALOGV("RecordThread: loop starting"); 4233 acquireWakeLock_l(); 4234 continue; 4235 } 4236 if (mActiveTrack != 0) { 4237 if (mActiveTrack->isTerminated()) { 4238 removeTrack_l(mActiveTrack); 4239 mActiveTrack.clear(); 4240 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4241 standby(); 4242 mActiveTrack.clear(); 4243 mStartStopCond.broadcast(); 4244 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4245 if (mReqChannelCount != mActiveTrack->channelCount()) { 4246 mActiveTrack.clear(); 4247 mStartStopCond.broadcast(); 4248 } else if (readOnce) { 4249 // record start succeeds only if first read from audio input 4250 // succeeds 4251 if (mBytesRead >= 0) { 4252 mActiveTrack->mState = TrackBase::ACTIVE; 4253 } else { 4254 mActiveTrack.clear(); 4255 } 4256 mStartStopCond.broadcast(); 4257 } 4258 mStandby = false; 4259 } 4260 } 4261 lockEffectChains_l(effectChains); 4262 } 4263 4264 if (mActiveTrack != 0) { 4265 if (mActiveTrack->mState != TrackBase::ACTIVE && 4266 mActiveTrack->mState != TrackBase::RESUMING) { 4267 unlockEffectChains(effectChains); 4268 usleep(kRecordThreadSleepUs); 4269 continue; 4270 } 4271 for (size_t i = 0; i < effectChains.size(); i ++) { 4272 effectChains[i]->process_l(); 4273 } 4274 4275 buffer.frameCount = mFrameCount; 4276 status_t status = mActiveTrack->getNextBuffer(&buffer); 4277 if (status == NO_ERROR) { 4278 readOnce = true; 4279 size_t framesOut = buffer.frameCount; 4280 if (mResampler == NULL) { 4281 // no resampling 4282 while (framesOut) { 4283 size_t framesIn = mFrameCount - mRsmpInIndex; 4284 if (framesIn) { 4285 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4286 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4287 mActiveTrack->mFrameSize; 4288 if (framesIn > framesOut) 4289 framesIn = framesOut; 4290 mRsmpInIndex += framesIn; 4291 framesOut -= framesIn; 4292 if (mChannelCount == mReqChannelCount || 4293 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 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 && 4308 (mChannelCount == mReqChannelCount || 4309 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4310 readInto = buffer.raw; 4311 framesOut = 0; 4312 } else { 4313 readInto = mRsmpInBuffer; 4314 mRsmpInIndex = 0; 4315 } 4316 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4317 mBufferSize); 4318 if (mBytesRead <= 0) { 4319 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4320 { 4321 ALOGE("Error reading audio input"); 4322 // Force input into standby so that it tries to 4323 // recover at next read attempt 4324 inputStandBy(); 4325 usleep(kRecordThreadSleepUs); 4326 } 4327 mRsmpInIndex = mFrameCount; 4328 framesOut = 0; 4329 buffer.frameCount = 0; 4330 } 4331#ifdef TEE_SINK 4332 else if (mTeeSink != 0) { 4333 (void) mTeeSink->write(readInto, 4334 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4335 } 4336#endif 4337 } 4338 } 4339 } else { 4340 // resampling 4341 4342 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4343 // alter output frame count as if we were expecting stereo samples 4344 if (mChannelCount == 1 && mReqChannelCount == 1) { 4345 framesOut >>= 1; 4346 } 4347 mResampler->resample(mRsmpOutBuffer, framesOut, 4348 this /* AudioBufferProvider* */); 4349 // ditherAndClamp() works as long as all buffers returned by 4350 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4351 if (mChannelCount == 2 && mReqChannelCount == 1) { 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 4361 } 4362 if (mFramestoDrop == 0) { 4363 mActiveTrack->releaseBuffer(&buffer); 4364 } else { 4365 if (mFramestoDrop > 0) { 4366 mFramestoDrop -= buffer.frameCount; 4367 if (mFramestoDrop <= 0) { 4368 clearSyncStartEvent(); 4369 } 4370 } else { 4371 mFramestoDrop += buffer.frameCount; 4372 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4373 mSyncStartEvent->isCancelled()) { 4374 ALOGW("Synced record %s, session %d, trigger session %d", 4375 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4376 mActiveTrack->sessionId(), 4377 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4378 clearSyncStartEvent(); 4379 } 4380 } 4381 } 4382 mActiveTrack->clearOverflow(); 4383 } 4384 // client isn't retrieving buffers fast enough 4385 else { 4386 if (!mActiveTrack->setOverflow()) { 4387 nsecs_t now = systemTime(); 4388 if ((now - lastWarning) > kWarningThrottleNs) { 4389 ALOGW("RecordThread: buffer overflow"); 4390 lastWarning = now; 4391 } 4392 } 4393 // Release the processor for a while before asking for a new buffer. 4394 // This will give the application more chance to read from the buffer and 4395 // clear the overflow. 4396 usleep(kRecordThreadSleepUs); 4397 } 4398 } 4399 // enable changes in effect chain 4400 unlockEffectChains(effectChains); 4401 effectChains.clear(); 4402 } 4403 4404 standby(); 4405 4406 { 4407 Mutex::Autolock _l(mLock); 4408 mActiveTrack.clear(); 4409 mStartStopCond.broadcast(); 4410 } 4411 4412 releaseWakeLock(); 4413 4414 ALOGV("RecordThread %p exiting", this); 4415 return false; 4416} 4417 4418void AudioFlinger::RecordThread::standby() 4419{ 4420 if (!mStandby) { 4421 inputStandBy(); 4422 mStandby = true; 4423 } 4424} 4425 4426void AudioFlinger::RecordThread::inputStandBy() 4427{ 4428 mInput->stream->common.standby(&mInput->stream->common); 4429} 4430 4431sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4432 const sp<AudioFlinger::Client>& client, 4433 uint32_t sampleRate, 4434 audio_format_t format, 4435 audio_channel_mask_t channelMask, 4436 size_t frameCount, 4437 int sessionId, 4438 IAudioFlinger::track_flags_t flags, 4439 pid_t tid, 4440 status_t *status) 4441{ 4442 sp<RecordTrack> track; 4443 status_t lStatus; 4444 4445 lStatus = initCheck(); 4446 if (lStatus != NO_ERROR) { 4447 ALOGE("Audio driver not initialized."); 4448 goto Exit; 4449 } 4450 4451 // FIXME use flags and tid similar to createTrack_l() 4452 4453 { // scope for mLock 4454 Mutex::Autolock _l(mLock); 4455 4456 track = new RecordTrack(this, client, sampleRate, 4457 format, channelMask, frameCount, sessionId); 4458 4459 if (track->getCblk() == 0) { 4460 lStatus = NO_MEMORY; 4461 goto Exit; 4462 } 4463 mTracks.add(track); 4464 4465 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4466 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4467 mAudioFlinger->btNrecIsOff(); 4468 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4469 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4470 } 4471 lStatus = NO_ERROR; 4472 4473Exit: 4474 if (status) { 4475 *status = lStatus; 4476 } 4477 return track; 4478} 4479 4480status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4481 AudioSystem::sync_event_t event, 4482 int triggerSession) 4483{ 4484 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4485 sp<ThreadBase> strongMe = this; 4486 status_t status = NO_ERROR; 4487 4488 if (event == AudioSystem::SYNC_EVENT_NONE) { 4489 clearSyncStartEvent(); 4490 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4491 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4492 triggerSession, 4493 recordTrack->sessionId(), 4494 syncStartEventCallback, 4495 this); 4496 // Sync event can be cancelled by the trigger session if the track is not in a 4497 // compatible state in which case we start record immediately 4498 if (mSyncStartEvent->isCancelled()) { 4499 clearSyncStartEvent(); 4500 } else { 4501 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4502 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4503 } 4504 } 4505 4506 { 4507 AutoMutex lock(mLock); 4508 if (mActiveTrack != 0) { 4509 if (recordTrack != mActiveTrack.get()) { 4510 status = -EBUSY; 4511 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4512 mActiveTrack->mState = TrackBase::ACTIVE; 4513 } 4514 return status; 4515 } 4516 4517 recordTrack->mState = TrackBase::IDLE; 4518 mActiveTrack = recordTrack; 4519 mLock.unlock(); 4520 status_t status = AudioSystem::startInput(mId); 4521 mLock.lock(); 4522 if (status != NO_ERROR) { 4523 mActiveTrack.clear(); 4524 clearSyncStartEvent(); 4525 return status; 4526 } 4527 mRsmpInIndex = mFrameCount; 4528 mBytesRead = 0; 4529 if (mResampler != NULL) { 4530 mResampler->reset(); 4531 } 4532 mActiveTrack->mState = TrackBase::RESUMING; 4533 // signal thread to start 4534 ALOGV("Signal record thread"); 4535 mWaitWorkCV.broadcast(); 4536 // do not wait for mStartStopCond if exiting 4537 if (exitPending()) { 4538 mActiveTrack.clear(); 4539 status = INVALID_OPERATION; 4540 goto startError; 4541 } 4542 mStartStopCond.wait(mLock); 4543 if (mActiveTrack == 0) { 4544 ALOGV("Record failed to start"); 4545 status = BAD_VALUE; 4546 goto startError; 4547 } 4548 ALOGV("Record started OK"); 4549 return status; 4550 } 4551 4552startError: 4553 AudioSystem::stopInput(mId); 4554 clearSyncStartEvent(); 4555 return status; 4556} 4557 4558void AudioFlinger::RecordThread::clearSyncStartEvent() 4559{ 4560 if (mSyncStartEvent != 0) { 4561 mSyncStartEvent->cancel(); 4562 } 4563 mSyncStartEvent.clear(); 4564 mFramestoDrop = 0; 4565} 4566 4567void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4568{ 4569 sp<SyncEvent> strongEvent = event.promote(); 4570 4571 if (strongEvent != 0) { 4572 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4573 me->handleSyncStartEvent(strongEvent); 4574 } 4575} 4576 4577void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4578{ 4579 if (event == mSyncStartEvent) { 4580 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4581 // from audio HAL 4582 mFramestoDrop = mFrameCount * 2; 4583 } 4584} 4585 4586bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4587 ALOGV("RecordThread::stop"); 4588 AutoMutex _l(mLock); 4589 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4590 return false; 4591 } 4592 recordTrack->mState = TrackBase::PAUSING; 4593 // do not wait for mStartStopCond if exiting 4594 if (exitPending()) { 4595 return true; 4596 } 4597 mStartStopCond.wait(mLock); 4598 // if we have been restarted, recordTrack == mActiveTrack.get() here 4599 if (exitPending() || recordTrack != mActiveTrack.get()) { 4600 ALOGV("Record stopped OK"); 4601 return true; 4602 } 4603 return false; 4604} 4605 4606bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4607{ 4608 return false; 4609} 4610 4611status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4612{ 4613#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4614 if (!isValidSyncEvent(event)) { 4615 return BAD_VALUE; 4616 } 4617 4618 int eventSession = event->triggerSession(); 4619 status_t ret = NAME_NOT_FOUND; 4620 4621 Mutex::Autolock _l(mLock); 4622 4623 for (size_t i = 0; i < mTracks.size(); i++) { 4624 sp<RecordTrack> track = mTracks[i]; 4625 if (eventSession == track->sessionId()) { 4626 (void) track->setSyncEvent(event); 4627 ret = NO_ERROR; 4628 } 4629 } 4630 return ret; 4631#else 4632 return BAD_VALUE; 4633#endif 4634} 4635 4636// destroyTrack_l() must be called with ThreadBase::mLock held 4637void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4638{ 4639 track->terminate(); 4640 track->mState = TrackBase::STOPPED; 4641 // active tracks are removed by threadLoop() 4642 if (mActiveTrack != track) { 4643 removeTrack_l(track); 4644 } 4645} 4646 4647void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4648{ 4649 mTracks.remove(track); 4650 // need anything related to effects here? 4651} 4652 4653void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4654{ 4655 dumpInternals(fd, args); 4656 dumpTracks(fd, args); 4657 dumpEffectChains(fd, args); 4658} 4659 4660void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4661{ 4662 const size_t SIZE = 256; 4663 char buffer[SIZE]; 4664 String8 result; 4665 4666 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4667 result.append(buffer); 4668 4669 if (mActiveTrack != 0) { 4670 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4671 result.append(buffer); 4672 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4673 result.append(buffer); 4674 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4675 result.append(buffer); 4676 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4677 result.append(buffer); 4678 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4679 result.append(buffer); 4680 } else { 4681 result.append("No active record client\n"); 4682 } 4683 4684 write(fd, result.string(), result.size()); 4685 4686 dumpBase(fd, args); 4687} 4688 4689void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4690{ 4691 const size_t SIZE = 256; 4692 char buffer[SIZE]; 4693 String8 result; 4694 4695 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4696 result.append(buffer); 4697 RecordTrack::appendDumpHeader(result); 4698 for (size_t i = 0; i < mTracks.size(); ++i) { 4699 sp<RecordTrack> track = mTracks[i]; 4700 if (track != 0) { 4701 track->dump(buffer, SIZE); 4702 result.append(buffer); 4703 } 4704 } 4705 4706 if (mActiveTrack != 0) { 4707 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4708 result.append(buffer); 4709 RecordTrack::appendDumpHeader(result); 4710 mActiveTrack->dump(buffer, SIZE); 4711 result.append(buffer); 4712 4713 } 4714 write(fd, result.string(), result.size()); 4715} 4716 4717// AudioBufferProvider interface 4718status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4719{ 4720 size_t framesReq = buffer->frameCount; 4721 size_t framesReady = mFrameCount - mRsmpInIndex; 4722 int channelCount; 4723 4724 if (framesReady == 0) { 4725 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4726 if (mBytesRead <= 0) { 4727 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4728 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4729 // Force input into standby so that it tries to 4730 // recover at next read attempt 4731 inputStandBy(); 4732 usleep(kRecordThreadSleepUs); 4733 } 4734 buffer->raw = NULL; 4735 buffer->frameCount = 0; 4736 return NOT_ENOUGH_DATA; 4737 } 4738 mRsmpInIndex = 0; 4739 framesReady = mFrameCount; 4740 } 4741 4742 if (framesReq > framesReady) { 4743 framesReq = framesReady; 4744 } 4745 4746 if (mChannelCount == 1 && mReqChannelCount == 2) { 4747 channelCount = 1; 4748 } else { 4749 channelCount = 2; 4750 } 4751 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4752 buffer->frameCount = framesReq; 4753 return NO_ERROR; 4754} 4755 4756// AudioBufferProvider interface 4757void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4758{ 4759 mRsmpInIndex += buffer->frameCount; 4760 buffer->frameCount = 0; 4761} 4762 4763bool AudioFlinger::RecordThread::checkForNewParameters_l() 4764{ 4765 bool reconfig = false; 4766 4767 while (!mNewParameters.isEmpty()) { 4768 status_t status = NO_ERROR; 4769 String8 keyValuePair = mNewParameters[0]; 4770 AudioParameter param = AudioParameter(keyValuePair); 4771 int value; 4772 audio_format_t reqFormat = mFormat; 4773 uint32_t reqSamplingRate = mReqSampleRate; 4774 uint32_t reqChannelCount = mReqChannelCount; 4775 4776 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4777 reqSamplingRate = value; 4778 reconfig = true; 4779 } 4780 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4781 reqFormat = (audio_format_t) value; 4782 reconfig = true; 4783 } 4784 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4785 reqChannelCount = popcount(value); 4786 reconfig = true; 4787 } 4788 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4789 // do not accept frame count changes if tracks are open as the track buffer 4790 // size depends on frame count and correct behavior would not be guaranteed 4791 // if frame count is changed after track creation 4792 if (mActiveTrack != 0) { 4793 status = INVALID_OPERATION; 4794 } else { 4795 reconfig = true; 4796 } 4797 } 4798 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4799 // forward device change to effects that have requested to be 4800 // aware of attached audio device. 4801 for (size_t i = 0; i < mEffectChains.size(); i++) { 4802 mEffectChains[i]->setDevice_l(value); 4803 } 4804 4805 // store input device and output device but do not forward output device to audio HAL. 4806 // Note that status is ignored by the caller for output device 4807 // (see AudioFlinger::setParameters() 4808 if (audio_is_output_devices(value)) { 4809 mOutDevice = value; 4810 status = BAD_VALUE; 4811 } else { 4812 mInDevice = value; 4813 // disable AEC and NS if the device is a BT SCO headset supporting those 4814 // pre processings 4815 if (mTracks.size() > 0) { 4816 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4817 mAudioFlinger->btNrecIsOff(); 4818 for (size_t i = 0; i < mTracks.size(); i++) { 4819 sp<RecordTrack> track = mTracks[i]; 4820 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4821 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4822 } 4823 } 4824 } 4825 } 4826 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4827 mAudioSource != (audio_source_t)value) { 4828 // forward device change to effects that have requested to be 4829 // aware of attached audio device. 4830 for (size_t i = 0; i < mEffectChains.size(); i++) { 4831 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4832 } 4833 mAudioSource = (audio_source_t)value; 4834 } 4835 if (status == NO_ERROR) { 4836 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4837 keyValuePair.string()); 4838 if (status == INVALID_OPERATION) { 4839 inputStandBy(); 4840 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4841 keyValuePair.string()); 4842 } 4843 if (reconfig) { 4844 if (status == BAD_VALUE && 4845 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4846 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4847 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4848 <= (2 * reqSamplingRate)) && 4849 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4850 <= FCC_2 && 4851 (reqChannelCount <= FCC_2)) { 4852 status = NO_ERROR; 4853 } 4854 if (status == NO_ERROR) { 4855 readInputParameters(); 4856 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4857 } 4858 } 4859 } 4860 4861 mNewParameters.removeAt(0); 4862 4863 mParamStatus = status; 4864 mParamCond.signal(); 4865 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4866 // already timed out waiting for the status and will never signal the condition. 4867 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4868 } 4869 return reconfig; 4870} 4871 4872String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4873{ 4874 Mutex::Autolock _l(mLock); 4875 if (initCheck() != NO_ERROR) { 4876 return String8(); 4877 } 4878 4879 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4880 const String8 out_s8(s); 4881 free(s); 4882 return out_s8; 4883} 4884 4885void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4886 AudioSystem::OutputDescriptor desc; 4887 void *param2 = NULL; 4888 4889 switch (event) { 4890 case AudioSystem::INPUT_OPENED: 4891 case AudioSystem::INPUT_CONFIG_CHANGED: 4892 desc.channelMask = mChannelMask; 4893 desc.samplingRate = mSampleRate; 4894 desc.format = mFormat; 4895 desc.frameCount = mFrameCount; 4896 desc.latency = 0; 4897 param2 = &desc; 4898 break; 4899 4900 case AudioSystem::INPUT_CLOSED: 4901 default: 4902 break; 4903 } 4904 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4905} 4906 4907void AudioFlinger::RecordThread::readInputParameters() 4908{ 4909 delete mRsmpInBuffer; 4910 // mRsmpInBuffer is always assigned a new[] below 4911 delete mRsmpOutBuffer; 4912 mRsmpOutBuffer = NULL; 4913 delete mResampler; 4914 mResampler = NULL; 4915 4916 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4917 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4918 mChannelCount = popcount(mChannelMask); 4919 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4920 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4921 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4922 mFrameCount = mBufferSize / mFrameSize; 4923 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4924 4925 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4926 { 4927 int channelCount; 4928 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4929 // stereo to mono post process as the resampler always outputs stereo. 4930 if (mChannelCount == 1 && mReqChannelCount == 2) { 4931 channelCount = 1; 4932 } else { 4933 channelCount = 2; 4934 } 4935 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4936 mResampler->setSampleRate(mSampleRate); 4937 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4938 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4939 4940 // optmization: if mono to mono, alter input frame count as if we were inputing 4941 // stereo samples 4942 if (mChannelCount == 1 && mReqChannelCount == 1) { 4943 mFrameCount >>= 1; 4944 } 4945 4946 } 4947 mRsmpInIndex = mFrameCount; 4948} 4949 4950unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4951{ 4952 Mutex::Autolock _l(mLock); 4953 if (initCheck() != NO_ERROR) { 4954 return 0; 4955 } 4956 4957 return mInput->stream->get_input_frames_lost(mInput->stream); 4958} 4959 4960uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4961{ 4962 Mutex::Autolock _l(mLock); 4963 uint32_t result = 0; 4964 if (getEffectChain_l(sessionId) != 0) { 4965 result = EFFECT_SESSION; 4966 } 4967 4968 for (size_t i = 0; i < mTracks.size(); ++i) { 4969 if (sessionId == mTracks[i]->sessionId()) { 4970 result |= TRACK_SESSION; 4971 break; 4972 } 4973 } 4974 4975 return result; 4976} 4977 4978KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4979{ 4980 KeyedVector<int, bool> ids; 4981 Mutex::Autolock _l(mLock); 4982 for (size_t j = 0; j < mTracks.size(); ++j) { 4983 sp<RecordThread::RecordTrack> track = mTracks[j]; 4984 int sessionId = track->sessionId(); 4985 if (ids.indexOfKey(sessionId) < 0) { 4986 ids.add(sessionId, true); 4987 } 4988 } 4989 return ids; 4990} 4991 4992AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4993{ 4994 Mutex::Autolock _l(mLock); 4995 AudioStreamIn *input = mInput; 4996 mInput = NULL; 4997 return input; 4998} 4999 5000// this method must always be called either with ThreadBase mLock held or inside the thread loop 5001audio_stream_t* AudioFlinger::RecordThread::stream() const 5002{ 5003 if (mInput == NULL) { 5004 return NULL; 5005 } 5006 return &mInput->stream->common; 5007} 5008 5009status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5010{ 5011 // only one chain per input thread 5012 if (mEffectChains.size() != 0) { 5013 return INVALID_OPERATION; 5014 } 5015 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5016 5017 chain->setInBuffer(NULL); 5018 chain->setOutBuffer(NULL); 5019 5020 checkSuspendOnAddEffectChain_l(chain); 5021 5022 mEffectChains.add(chain); 5023 5024 return NO_ERROR; 5025} 5026 5027size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5028{ 5029 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5030 ALOGW_IF(mEffectChains.size() != 1, 5031 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5032 chain.get(), mEffectChains.size(), this); 5033 if (mEffectChains.size() == 1) { 5034 mEffectChains.removeAt(0); 5035 } 5036 return 0; 5037} 5038 5039}; // namespace android 5040