Threads.cpp revision 6bf9ae20b3bd2dbb8f2e89ee167a6785222301cf
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 // mLatchD, mLatchQ, 948 mLatchDValid(false), mLatchQValid(false) 949{ 950 snprintf(mName, kNameLength, "AudioOut_%X", id); 951 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 952 953 // Assumes constructor is called by AudioFlinger with it's mLock held, but 954 // it would be safer to explicitly pass initial masterVolume/masterMute as 955 // parameter. 956 // 957 // If the HAL we are using has support for master volume or master mute, 958 // then do not attenuate or mute during mixing (just leave the volume at 1.0 959 // and the mute set to false). 960 mMasterVolume = audioFlinger->masterVolume_l(); 961 mMasterMute = audioFlinger->masterMute_l(); 962 if (mOutput && mOutput->audioHwDev) { 963 if (mOutput->audioHwDev->canSetMasterVolume()) { 964 mMasterVolume = 1.0; 965 } 966 967 if (mOutput->audioHwDev->canSetMasterMute()) { 968 mMasterMute = false; 969 } 970 } 971 972 readOutputParameters(); 973 974 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 975 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 976 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 977 stream = (audio_stream_type_t) (stream + 1)) { 978 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 979 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 980 } 981 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 982 // because mAudioFlinger doesn't have one to copy from 983} 984 985AudioFlinger::PlaybackThread::~PlaybackThread() 986{ 987 mAudioFlinger->unregisterWriter(mNBLogWriter); 988 delete [] mAllocMixBuffer; 989} 990 991void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 992{ 993 dumpInternals(fd, args); 994 dumpTracks(fd, args); 995 dumpEffectChains(fd, args); 996} 997 998void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 999{ 1000 const size_t SIZE = 256; 1001 char buffer[SIZE]; 1002 String8 result; 1003 1004 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1005 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1006 const stream_type_t *st = &mStreamTypes[i]; 1007 if (i > 0) { 1008 result.appendFormat(", "); 1009 } 1010 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1011 if (st->mute) { 1012 result.append("M"); 1013 } 1014 } 1015 result.append("\n"); 1016 write(fd, result.string(), result.length()); 1017 result.clear(); 1018 1019 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1020 result.append(buffer); 1021 Track::appendDumpHeader(result); 1022 for (size_t i = 0; i < mTracks.size(); ++i) { 1023 sp<Track> track = mTracks[i]; 1024 if (track != 0) { 1025 track->dump(buffer, SIZE); 1026 result.append(buffer); 1027 } 1028 } 1029 1030 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1031 result.append(buffer); 1032 Track::appendDumpHeader(result); 1033 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1034 sp<Track> track = mActiveTracks[i].promote(); 1035 if (track != 0) { 1036 track->dump(buffer, SIZE); 1037 result.append(buffer); 1038 } 1039 } 1040 write(fd, result.string(), result.size()); 1041 1042 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1043 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1044 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1045 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1046} 1047 1048void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1049{ 1050 const size_t SIZE = 256; 1051 char buffer[SIZE]; 1052 String8 result; 1053 1054 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1055 result.append(buffer); 1056 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1057 result.append(buffer); 1058 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1059 ns2ms(systemTime() - mLastWriteTime)); 1060 result.append(buffer); 1061 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1062 result.append(buffer); 1063 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1064 result.append(buffer); 1065 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1066 result.append(buffer); 1067 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1068 result.append(buffer); 1069 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1070 result.append(buffer); 1071 write(fd, result.string(), result.size()); 1072 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1073 1074 dumpBase(fd, args); 1075} 1076 1077// Thread virtuals 1078status_t AudioFlinger::PlaybackThread::readyToRun() 1079{ 1080 status_t status = initCheck(); 1081 if (status == NO_ERROR) { 1082 ALOGI("AudioFlinger's thread %p ready to run", this); 1083 } else { 1084 ALOGE("No working audio driver found."); 1085 } 1086 return status; 1087} 1088 1089void AudioFlinger::PlaybackThread::onFirstRef() 1090{ 1091 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1092} 1093 1094// ThreadBase virtuals 1095void AudioFlinger::PlaybackThread::preExit() 1096{ 1097 ALOGV(" preExit()"); 1098 // FIXME this is using hard-coded strings but in the future, this functionality will be 1099 // converted to use audio HAL extensions required to support tunneling 1100 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1101} 1102 1103// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1104sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1105 const sp<AudioFlinger::Client>& client, 1106 audio_stream_type_t streamType, 1107 uint32_t sampleRate, 1108 audio_format_t format, 1109 audio_channel_mask_t channelMask, 1110 size_t frameCount, 1111 const sp<IMemory>& sharedBuffer, 1112 int sessionId, 1113 IAudioFlinger::track_flags_t *flags, 1114 pid_t tid, 1115 status_t *status) 1116{ 1117 sp<Track> track; 1118 status_t lStatus; 1119 1120 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1121 1122 // client expresses a preference for FAST, but we get the final say 1123 if (*flags & IAudioFlinger::TRACK_FAST) { 1124 if ( 1125 // not timed 1126 (!isTimed) && 1127 // either of these use cases: 1128 ( 1129 // use case 1: shared buffer with any frame count 1130 ( 1131 (sharedBuffer != 0) 1132 ) || 1133 // use case 2: callback handler and frame count is default or at least as large as HAL 1134 ( 1135 (tid != -1) && 1136 ((frameCount == 0) || 1137 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1138 ) 1139 ) && 1140 // PCM data 1141 audio_is_linear_pcm(format) && 1142 // mono or stereo 1143 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1144 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1145#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1146 // hardware sample rate 1147 (sampleRate == mSampleRate) && 1148#endif 1149 // normal mixer has an associated fast mixer 1150 hasFastMixer() && 1151 // there are sufficient fast track slots available 1152 (mFastTrackAvailMask != 0) 1153 // FIXME test that MixerThread for this fast track has a capable output HAL 1154 // FIXME add a permission test also? 1155 ) { 1156 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1157 if (frameCount == 0) { 1158 frameCount = mFrameCount * kFastTrackMultiplier; 1159 } 1160 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1161 frameCount, mFrameCount); 1162 } else { 1163 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1164 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1165 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1166 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1167 audio_is_linear_pcm(format), 1168 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1169 *flags &= ~IAudioFlinger::TRACK_FAST; 1170 // For compatibility with AudioTrack calculation, buffer depth is forced 1171 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1172 // This is probably too conservative, but legacy application code may depend on it. 1173 // If you change this calculation, also review the start threshold which is related. 1174 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1175 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1176 if (minBufCount < 2) { 1177 minBufCount = 2; 1178 } 1179 size_t minFrameCount = mNormalFrameCount * minBufCount; 1180 if (frameCount < minFrameCount) { 1181 frameCount = minFrameCount; 1182 } 1183 } 1184 } 1185 1186 if (mType == DIRECT) { 1187 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1188 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1189 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1190 "for output %p with format %d", 1191 sampleRate, format, channelMask, mOutput, mFormat); 1192 lStatus = BAD_VALUE; 1193 goto Exit; 1194 } 1195 } 1196 } else if (mType == OFFLOAD) { 1197 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1198 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1199 "for output %p with format %d", 1200 sampleRate, format, channelMask, mOutput, mFormat); 1201 lStatus = BAD_VALUE; 1202 goto Exit; 1203 } 1204 } else { 1205 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1206 ALOGE("createTrack_l() Bad parameter: format %d \"" 1207 "for output %p with format %d", 1208 format, mOutput, mFormat); 1209 lStatus = BAD_VALUE; 1210 goto Exit; 1211 } 1212 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1213 if (sampleRate > mSampleRate*2) { 1214 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1215 lStatus = BAD_VALUE; 1216 goto Exit; 1217 } 1218 } 1219 1220 lStatus = initCheck(); 1221 if (lStatus != NO_ERROR) { 1222 ALOGE("Audio driver not initialized."); 1223 goto Exit; 1224 } 1225 1226 { // scope for mLock 1227 Mutex::Autolock _l(mLock); 1228 1229 // all tracks in same audio session must share the same routing strategy otherwise 1230 // conflicts will happen when tracks are moved from one output to another by audio policy 1231 // manager 1232 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1233 for (size_t i = 0; i < mTracks.size(); ++i) { 1234 sp<Track> t = mTracks[i]; 1235 if (t != 0 && !t->isOutputTrack()) { 1236 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1237 if (sessionId == t->sessionId() && strategy != actual) { 1238 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1239 strategy, actual); 1240 lStatus = BAD_VALUE; 1241 goto Exit; 1242 } 1243 } 1244 } 1245 1246 if (!isTimed) { 1247 track = new Track(this, client, streamType, sampleRate, format, 1248 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1249 } else { 1250 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1251 channelMask, frameCount, sharedBuffer, sessionId); 1252 } 1253 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1254 lStatus = NO_MEMORY; 1255 goto Exit; 1256 } 1257 1258 mTracks.add(track); 1259 1260 sp<EffectChain> chain = getEffectChain_l(sessionId); 1261 if (chain != 0) { 1262 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1263 track->setMainBuffer(chain->inBuffer()); 1264 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1265 chain->incTrackCnt(); 1266 } 1267 1268 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1269 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1270 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1271 // so ask activity manager to do this on our behalf 1272 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1273 } 1274 } 1275 1276 lStatus = NO_ERROR; 1277 1278Exit: 1279 if (status) { 1280 *status = lStatus; 1281 } 1282 return track; 1283} 1284 1285uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1286{ 1287 return latency; 1288} 1289 1290uint32_t AudioFlinger::PlaybackThread::latency() const 1291{ 1292 Mutex::Autolock _l(mLock); 1293 return latency_l(); 1294} 1295uint32_t AudioFlinger::PlaybackThread::latency_l() const 1296{ 1297 if (initCheck() == NO_ERROR) { 1298 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1299 } else { 1300 return 0; 1301 } 1302} 1303 1304void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1305{ 1306 Mutex::Autolock _l(mLock); 1307 // Don't apply master volume in SW if our HAL can do it for us. 1308 if (mOutput && mOutput->audioHwDev && 1309 mOutput->audioHwDev->canSetMasterVolume()) { 1310 mMasterVolume = 1.0; 1311 } else { 1312 mMasterVolume = value; 1313 } 1314} 1315 1316void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1317{ 1318 Mutex::Autolock _l(mLock); 1319 // Don't apply master mute in SW if our HAL can do it for us. 1320 if (mOutput && mOutput->audioHwDev && 1321 mOutput->audioHwDev->canSetMasterMute()) { 1322 mMasterMute = false; 1323 } else { 1324 mMasterMute = muted; 1325 } 1326} 1327 1328void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1329{ 1330 Mutex::Autolock _l(mLock); 1331 mStreamTypes[stream].volume = value; 1332 signal_l(); 1333} 1334 1335void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1336{ 1337 Mutex::Autolock _l(mLock); 1338 mStreamTypes[stream].mute = muted; 1339 signal_l(); 1340} 1341 1342float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1343{ 1344 Mutex::Autolock _l(mLock); 1345 return mStreamTypes[stream].volume; 1346} 1347 1348// addTrack_l() must be called with ThreadBase::mLock held 1349status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1350{ 1351 status_t status = ALREADY_EXISTS; 1352 1353 // set retry count for buffer fill 1354 track->mRetryCount = kMaxTrackStartupRetries; 1355 if (mActiveTracks.indexOf(track) < 0) { 1356 // the track is newly added, make sure it fills up all its 1357 // buffers before playing. This is to ensure the client will 1358 // effectively get the latency it requested. 1359 if (!track->isOutputTrack()) { 1360 TrackBase::track_state state = track->mState; 1361 mLock.unlock(); 1362 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1363 mLock.lock(); 1364 // abort track was stopped/paused while we released the lock 1365 if (state != track->mState) { 1366 if (status == NO_ERROR) { 1367 mLock.unlock(); 1368 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1369 mLock.lock(); 1370 } 1371 return INVALID_OPERATION; 1372 } 1373 // abort if start is rejected by audio policy manager 1374 if (status != NO_ERROR) { 1375 return PERMISSION_DENIED; 1376 } 1377#ifdef ADD_BATTERY_DATA 1378 // to track the speaker usage 1379 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1380#endif 1381 } 1382 1383 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1384 track->mResetDone = false; 1385 track->mPresentationCompleteFrames = 0; 1386 mActiveTracks.add(track); 1387 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1388 if (chain != 0) { 1389 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1390 track->sessionId()); 1391 chain->incActiveTrackCnt(); 1392 } 1393 1394 status = NO_ERROR; 1395 } 1396 1397 ALOGV("mWaitWorkCV.broadcast"); 1398 mWaitWorkCV.broadcast(); 1399 1400 return status; 1401} 1402 1403bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1404{ 1405 track->terminate(); 1406 // active tracks are removed by threadLoop() 1407 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1408 track->mState = TrackBase::STOPPED; 1409 if (!trackActive) { 1410 removeTrack_l(track); 1411 } else if (track->isFastTrack() || track->isOffloaded()) { 1412 track->mState = TrackBase::STOPPING_1; 1413 } 1414 1415 return trackActive; 1416} 1417 1418void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1419{ 1420 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1421 mTracks.remove(track); 1422 deleteTrackName_l(track->name()); 1423 // redundant as track is about to be destroyed, for dumpsys only 1424 track->mName = -1; 1425 if (track->isFastTrack()) { 1426 int index = track->mFastIndex; 1427 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1428 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1429 mFastTrackAvailMask |= 1 << index; 1430 // redundant as track is about to be destroyed, for dumpsys only 1431 track->mFastIndex = -1; 1432 } 1433 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1434 if (chain != 0) { 1435 chain->decTrackCnt(); 1436 } 1437} 1438 1439void AudioFlinger::PlaybackThread::signal_l() 1440{ 1441 // Thread could be blocked waiting for async 1442 // so signal it to handle state changes immediately 1443 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1444 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1445 mSignalPending = true; 1446 mWaitWorkCV.signal(); 1447} 1448 1449String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1450{ 1451 Mutex::Autolock _l(mLock); 1452 if (initCheck() != NO_ERROR) { 1453 return String8(); 1454 } 1455 1456 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1457 const String8 out_s8(s); 1458 free(s); 1459 return out_s8; 1460} 1461 1462// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1463void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1464 AudioSystem::OutputDescriptor desc; 1465 void *param2 = NULL; 1466 1467 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1468 param); 1469 1470 switch (event) { 1471 case AudioSystem::OUTPUT_OPENED: 1472 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1473 desc.channelMask = mChannelMask; 1474 desc.samplingRate = mSampleRate; 1475 desc.format = mFormat; 1476 desc.frameCount = mNormalFrameCount; // FIXME see 1477 // AudioFlinger::frameCount(audio_io_handle_t) 1478 desc.latency = latency(); 1479 param2 = &desc; 1480 break; 1481 1482 case AudioSystem::STREAM_CONFIG_CHANGED: 1483 param2 = ¶m; 1484 case AudioSystem::OUTPUT_CLOSED: 1485 default: 1486 break; 1487 } 1488 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1489} 1490 1491void AudioFlinger::PlaybackThread::writeCallback() 1492{ 1493 ALOG_ASSERT(mCallbackThread != 0); 1494 mCallbackThread->setWriteBlocked(false); 1495} 1496 1497void AudioFlinger::PlaybackThread::drainCallback() 1498{ 1499 ALOG_ASSERT(mCallbackThread != 0); 1500 mCallbackThread->setDraining(false); 1501} 1502 1503void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1504{ 1505 Mutex::Autolock _l(mLock); 1506 mWriteBlocked = value; 1507 if (!value) { 1508 mWaitWorkCV.signal(); 1509 } 1510} 1511 1512void AudioFlinger::PlaybackThread::setDraining(bool value) 1513{ 1514 Mutex::Autolock _l(mLock); 1515 mDraining = value; 1516 if (!value) { 1517 mWaitWorkCV.signal(); 1518 } 1519} 1520 1521// static 1522int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1523 void *param, 1524 void *cookie) 1525{ 1526 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1527 ALOGV("asyncCallback() event %d", event); 1528 switch (event) { 1529 case STREAM_CBK_EVENT_WRITE_READY: 1530 me->writeCallback(); 1531 break; 1532 case STREAM_CBK_EVENT_DRAIN_READY: 1533 me->drainCallback(); 1534 break; 1535 default: 1536 ALOGW("asyncCallback() unknown event %d", event); 1537 break; 1538 } 1539 return 0; 1540} 1541 1542void AudioFlinger::PlaybackThread::readOutputParameters() 1543{ 1544 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1545 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1546 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1547 if (!audio_is_output_channel(mChannelMask)) { 1548 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1549 } 1550 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1551 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1552 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1553 } 1554 mChannelCount = popcount(mChannelMask); 1555 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1556 if (!audio_is_valid_format(mFormat)) { 1557 LOG_FATAL("HAL format %d not valid for output", mFormat); 1558 } 1559 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1560 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1561 mFormat); 1562 } 1563 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1564 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1565 if (mFrameCount & 15) { 1566 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1567 mFrameCount); 1568 } 1569 1570 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1571 (mOutput->stream->set_callback != NULL)) { 1572 if (mOutput->stream->set_callback(mOutput->stream, 1573 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1574 mUseAsyncWrite = true; 1575 } 1576 } 1577 1578 // Calculate size of normal mix buffer relative to the HAL output buffer size 1579 double multiplier = 1.0; 1580 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1581 kUseFastMixer == FastMixer_Dynamic)) { 1582 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1583 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1584 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1585 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1586 maxNormalFrameCount = maxNormalFrameCount & ~15; 1587 if (maxNormalFrameCount < minNormalFrameCount) { 1588 maxNormalFrameCount = minNormalFrameCount; 1589 } 1590 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1591 if (multiplier <= 1.0) { 1592 multiplier = 1.0; 1593 } else if (multiplier <= 2.0) { 1594 if (2 * mFrameCount <= maxNormalFrameCount) { 1595 multiplier = 2.0; 1596 } else { 1597 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1598 } 1599 } else { 1600 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1601 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1602 // track, but we sometimes have to do this to satisfy the maximum frame count 1603 // constraint) 1604 // FIXME this rounding up should not be done if no HAL SRC 1605 uint32_t truncMult = (uint32_t) multiplier; 1606 if ((truncMult & 1)) { 1607 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1608 ++truncMult; 1609 } 1610 } 1611 multiplier = (double) truncMult; 1612 } 1613 } 1614 mNormalFrameCount = multiplier * mFrameCount; 1615 // round up to nearest 16 frames to satisfy AudioMixer 1616 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1617 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1618 mNormalFrameCount); 1619 1620 delete[] mAllocMixBuffer; 1621 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1622 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1623 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1624 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1625 1626 // force reconfiguration of effect chains and engines to take new buffer size and audio 1627 // parameters into account 1628 // Note that mLock is not held when readOutputParameters() is called from the constructor 1629 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1630 // matter. 1631 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1632 Vector< sp<EffectChain> > effectChains = mEffectChains; 1633 for (size_t i = 0; i < effectChains.size(); i ++) { 1634 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1635 } 1636} 1637 1638 1639status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1640{ 1641 if (halFrames == NULL || dspFrames == NULL) { 1642 return BAD_VALUE; 1643 } 1644 Mutex::Autolock _l(mLock); 1645 if (initCheck() != NO_ERROR) { 1646 return INVALID_OPERATION; 1647 } 1648 size_t framesWritten = mBytesWritten / mFrameSize; 1649 *halFrames = framesWritten; 1650 1651 if (isSuspended()) { 1652 // return an estimation of rendered frames when the output is suspended 1653 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1654 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1655 return NO_ERROR; 1656 } else { 1657 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1658 } 1659} 1660 1661uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1662{ 1663 Mutex::Autolock _l(mLock); 1664 uint32_t result = 0; 1665 if (getEffectChain_l(sessionId) != 0) { 1666 result = EFFECT_SESSION; 1667 } 1668 1669 for (size_t i = 0; i < mTracks.size(); ++i) { 1670 sp<Track> track = mTracks[i]; 1671 if (sessionId == track->sessionId() && !track->isInvalid()) { 1672 result |= TRACK_SESSION; 1673 break; 1674 } 1675 } 1676 1677 return result; 1678} 1679 1680uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1681{ 1682 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1683 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1684 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1685 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1686 } 1687 for (size_t i = 0; i < mTracks.size(); i++) { 1688 sp<Track> track = mTracks[i]; 1689 if (sessionId == track->sessionId() && !track->isInvalid()) { 1690 return AudioSystem::getStrategyForStream(track->streamType()); 1691 } 1692 } 1693 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1694} 1695 1696 1697AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1698{ 1699 Mutex::Autolock _l(mLock); 1700 return mOutput; 1701} 1702 1703AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1704{ 1705 Mutex::Autolock _l(mLock); 1706 AudioStreamOut *output = mOutput; 1707 mOutput = NULL; 1708 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1709 // must push a NULL and wait for ack 1710 mOutputSink.clear(); 1711 mPipeSink.clear(); 1712 mNormalSink.clear(); 1713 return output; 1714} 1715 1716// this method must always be called either with ThreadBase mLock held or inside the thread loop 1717audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1718{ 1719 if (mOutput == NULL) { 1720 return NULL; 1721 } 1722 return &mOutput->stream->common; 1723} 1724 1725uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1726{ 1727 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1728} 1729 1730status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1731{ 1732 if (!isValidSyncEvent(event)) { 1733 return BAD_VALUE; 1734 } 1735 1736 Mutex::Autolock _l(mLock); 1737 1738 for (size_t i = 0; i < mTracks.size(); ++i) { 1739 sp<Track> track = mTracks[i]; 1740 if (event->triggerSession() == track->sessionId()) { 1741 (void) track->setSyncEvent(event); 1742 return NO_ERROR; 1743 } 1744 } 1745 1746 return NAME_NOT_FOUND; 1747} 1748 1749bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1750{ 1751 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1752} 1753 1754void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1755 const Vector< sp<Track> >& tracksToRemove) 1756{ 1757 size_t count = tracksToRemove.size(); 1758 if (count) { 1759 for (size_t i = 0 ; i < count ; i++) { 1760 const sp<Track>& track = tracksToRemove.itemAt(i); 1761 if (!track->isOutputTrack()) { 1762 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1763#ifdef ADD_BATTERY_DATA 1764 // to track the speaker usage 1765 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1766#endif 1767 if (track->isTerminated()) { 1768 AudioSystem::releaseOutput(mId); 1769 } 1770 } 1771 } 1772 } 1773} 1774 1775void AudioFlinger::PlaybackThread::checkSilentMode_l() 1776{ 1777 if (!mMasterMute) { 1778 char value[PROPERTY_VALUE_MAX]; 1779 if (property_get("ro.audio.silent", value, "0") > 0) { 1780 char *endptr; 1781 unsigned long ul = strtoul(value, &endptr, 0); 1782 if (*endptr == '\0' && ul != 0) { 1783 ALOGD("Silence is golden"); 1784 // The setprop command will not allow a property to be changed after 1785 // the first time it is set, so we don't have to worry about un-muting. 1786 setMasterMute_l(true); 1787 } 1788 } 1789 } 1790} 1791 1792// shared by MIXER and DIRECT, overridden by DUPLICATING 1793ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1794{ 1795 // FIXME rewrite to reduce number of system calls 1796 mLastWriteTime = systemTime(); 1797 mInWrite = true; 1798 ssize_t bytesWritten; 1799 1800 // If an NBAIO sink is present, use it to write the normal mixer's submix 1801 if (mNormalSink != 0) { 1802#define mBitShift 2 // FIXME 1803 size_t count = mBytesRemaining >> mBitShift; 1804 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1805 ATRACE_BEGIN("write"); 1806 // update the setpoint when AudioFlinger::mScreenState changes 1807 uint32_t screenState = AudioFlinger::mScreenState; 1808 if (screenState != mScreenState) { 1809 mScreenState = screenState; 1810 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1811 if (pipe != NULL) { 1812 pipe->setAvgFrames((mScreenState & 1) ? 1813 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1814 } 1815 } 1816 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1817 ATRACE_END(); 1818 if (framesWritten > 0) { 1819 bytesWritten = framesWritten << mBitShift; 1820 } else { 1821 bytesWritten = framesWritten; 1822 } 1823 status_t status = INVALID_OPERATION; // mLatchD.mTimestamp is invalid 1824 if (status == NO_ERROR) { 1825 size_t totalFramesWritten = mNormalSink->framesWritten(); 1826 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1827 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1828 mLatchDValid = true; 1829 } 1830 } 1831 // otherwise use the HAL / AudioStreamOut directly 1832 } else { 1833 // Direct output and offload threads 1834 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1835 if (mUseAsyncWrite) { 1836 mWriteBlocked = true; 1837 ALOG_ASSERT(mCallbackThread != 0); 1838 mCallbackThread->setWriteBlocked(true); 1839 } 1840 bytesWritten = mOutput->stream->write(mOutput->stream, 1841 mMixBuffer + offset, mBytesRemaining); 1842 if (mUseAsyncWrite && 1843 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1844 // do not wait for async callback in case of error of full write 1845 mWriteBlocked = false; 1846 ALOG_ASSERT(mCallbackThread != 0); 1847 mCallbackThread->setWriteBlocked(false); 1848 } 1849 } 1850 1851 mNumWrites++; 1852 mInWrite = false; 1853 1854 return bytesWritten; 1855} 1856 1857void AudioFlinger::PlaybackThread::threadLoop_drain() 1858{ 1859 if (mOutput->stream->drain) { 1860 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1861 if (mUseAsyncWrite) { 1862 mDraining = true; 1863 ALOG_ASSERT(mCallbackThread != 0); 1864 mCallbackThread->setDraining(true); 1865 } 1866 mOutput->stream->drain(mOutput->stream, 1867 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1868 : AUDIO_DRAIN_ALL); 1869 } 1870} 1871 1872void AudioFlinger::PlaybackThread::threadLoop_exit() 1873{ 1874 // Default implementation has nothing to do 1875} 1876 1877/* 1878The derived values that are cached: 1879 - mixBufferSize from frame count * frame size 1880 - activeSleepTime from activeSleepTimeUs() 1881 - idleSleepTime from idleSleepTimeUs() 1882 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1883 - maxPeriod from frame count and sample rate (MIXER only) 1884 1885The parameters that affect these derived values are: 1886 - frame count 1887 - frame size 1888 - sample rate 1889 - device type: A2DP or not 1890 - device latency 1891 - format: PCM or not 1892 - active sleep time 1893 - idle sleep time 1894*/ 1895 1896void AudioFlinger::PlaybackThread::cacheParameters_l() 1897{ 1898 mixBufferSize = mNormalFrameCount * mFrameSize; 1899 activeSleepTime = activeSleepTimeUs(); 1900 idleSleepTime = idleSleepTimeUs(); 1901} 1902 1903void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1904{ 1905 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1906 this, streamType, mTracks.size()); 1907 Mutex::Autolock _l(mLock); 1908 1909 size_t size = mTracks.size(); 1910 for (size_t i = 0; i < size; i++) { 1911 sp<Track> t = mTracks[i]; 1912 if (t->streamType() == streamType) { 1913 t->invalidate(); 1914 } 1915 } 1916} 1917 1918status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1919{ 1920 int session = chain->sessionId(); 1921 int16_t *buffer = mMixBuffer; 1922 bool ownsBuffer = false; 1923 1924 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1925 if (session > 0) { 1926 // Only one effect chain can be present in direct output thread and it uses 1927 // the mix buffer as input 1928 if (mType != DIRECT) { 1929 size_t numSamples = mNormalFrameCount * mChannelCount; 1930 buffer = new int16_t[numSamples]; 1931 memset(buffer, 0, numSamples * sizeof(int16_t)); 1932 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1933 ownsBuffer = true; 1934 } 1935 1936 // Attach all tracks with same session ID to this chain. 1937 for (size_t i = 0; i < mTracks.size(); ++i) { 1938 sp<Track> track = mTracks[i]; 1939 if (session == track->sessionId()) { 1940 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1941 buffer); 1942 track->setMainBuffer(buffer); 1943 chain->incTrackCnt(); 1944 } 1945 } 1946 1947 // indicate all active tracks in the chain 1948 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1949 sp<Track> track = mActiveTracks[i].promote(); 1950 if (track == 0) { 1951 continue; 1952 } 1953 if (session == track->sessionId()) { 1954 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1955 chain->incActiveTrackCnt(); 1956 } 1957 } 1958 } 1959 1960 chain->setInBuffer(buffer, ownsBuffer); 1961 chain->setOutBuffer(mMixBuffer); 1962 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1963 // chains list in order to be processed last as it contains output stage effects 1964 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1965 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1966 // after track specific effects and before output stage 1967 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1968 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1969 // Effect chain for other sessions are inserted at beginning of effect 1970 // chains list to be processed before output mix effects. Relative order between other 1971 // sessions is not important 1972 size_t size = mEffectChains.size(); 1973 size_t i = 0; 1974 for (i = 0; i < size; i++) { 1975 if (mEffectChains[i]->sessionId() < session) { 1976 break; 1977 } 1978 } 1979 mEffectChains.insertAt(chain, i); 1980 checkSuspendOnAddEffectChain_l(chain); 1981 1982 return NO_ERROR; 1983} 1984 1985size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1986{ 1987 int session = chain->sessionId(); 1988 1989 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1990 1991 for (size_t i = 0; i < mEffectChains.size(); i++) { 1992 if (chain == mEffectChains[i]) { 1993 mEffectChains.removeAt(i); 1994 // detach all active tracks from the chain 1995 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1996 sp<Track> track = mActiveTracks[i].promote(); 1997 if (track == 0) { 1998 continue; 1999 } 2000 if (session == track->sessionId()) { 2001 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2002 chain.get(), session); 2003 chain->decActiveTrackCnt(); 2004 } 2005 } 2006 2007 // detach all tracks with same session ID from this chain 2008 for (size_t i = 0; i < mTracks.size(); ++i) { 2009 sp<Track> track = mTracks[i]; 2010 if (session == track->sessionId()) { 2011 track->setMainBuffer(mMixBuffer); 2012 chain->decTrackCnt(); 2013 } 2014 } 2015 break; 2016 } 2017 } 2018 return mEffectChains.size(); 2019} 2020 2021status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2022 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2023{ 2024 Mutex::Autolock _l(mLock); 2025 return attachAuxEffect_l(track, EffectId); 2026} 2027 2028status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2029 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2030{ 2031 status_t status = NO_ERROR; 2032 2033 if (EffectId == 0) { 2034 track->setAuxBuffer(0, NULL); 2035 } else { 2036 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2037 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2038 if (effect != 0) { 2039 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2040 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2041 } else { 2042 status = INVALID_OPERATION; 2043 } 2044 } else { 2045 status = BAD_VALUE; 2046 } 2047 } 2048 return status; 2049} 2050 2051void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2052{ 2053 for (size_t i = 0; i < mTracks.size(); ++i) { 2054 sp<Track> track = mTracks[i]; 2055 if (track->auxEffectId() == effectId) { 2056 attachAuxEffect_l(track, 0); 2057 } 2058 } 2059} 2060 2061bool AudioFlinger::PlaybackThread::threadLoop() 2062{ 2063 Vector< sp<Track> > tracksToRemove; 2064 2065 standbyTime = systemTime(); 2066 2067 // MIXER 2068 nsecs_t lastWarning = 0; 2069 2070 // DUPLICATING 2071 // FIXME could this be made local to while loop? 2072 writeFrames = 0; 2073 2074 cacheParameters_l(); 2075 sleepTime = idleSleepTime; 2076 2077 if (mType == MIXER) { 2078 sleepTimeShift = 0; 2079 } 2080 2081 CpuStats cpuStats; 2082 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2083 2084 acquireWakeLock(); 2085 2086 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2087 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2088 // and then that string will be logged at the next convenient opportunity. 2089 const char *logString = NULL; 2090 2091 while (!exitPending()) 2092 { 2093 cpuStats.sample(myName); 2094 2095 Vector< sp<EffectChain> > effectChains; 2096 2097 processConfigEvents(); 2098 2099 { // scope for mLock 2100 2101 Mutex::Autolock _l(mLock); 2102 2103 if (logString != NULL) { 2104 mNBLogWriter->logTimestamp(); 2105 mNBLogWriter->log(logString); 2106 logString = NULL; 2107 } 2108 2109 if (mLatchDValid) { 2110 mLatchQ = mLatchD; 2111 mLatchDValid = false; 2112 mLatchQValid = true; 2113 } 2114 2115 if (checkForNewParameters_l()) { 2116 cacheParameters_l(); 2117 } 2118 2119 saveOutputTracks(); 2120 2121 if (mSignalPending) { 2122 // A signal was raised while we were unlocked 2123 mSignalPending = false; 2124 } else if (waitingAsyncCallback_l()) { 2125 if (exitPending()) { 2126 break; 2127 } 2128 releaseWakeLock_l(); 2129 ALOGV("wait async completion"); 2130 mWaitWorkCV.wait(mLock); 2131 ALOGV("async completion/wake"); 2132 acquireWakeLock_l(); 2133 if (exitPending()) { 2134 break; 2135 } 2136 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2137 continue; 2138 } 2139 sleepTime = 0; 2140 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2141 isSuspended()) { 2142 // put audio hardware into standby after short delay 2143 if (shouldStandby_l()) { 2144 2145 threadLoop_standby(); 2146 2147 mStandby = true; 2148 } 2149 2150 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2151 // we're about to wait, flush the binder command buffer 2152 IPCThreadState::self()->flushCommands(); 2153 2154 clearOutputTracks(); 2155 2156 if (exitPending()) { 2157 break; 2158 } 2159 2160 releaseWakeLock_l(); 2161 // wait until we have something to do... 2162 ALOGV("%s going to sleep", myName.string()); 2163 mWaitWorkCV.wait(mLock); 2164 ALOGV("%s waking up", myName.string()); 2165 acquireWakeLock_l(); 2166 2167 mMixerStatus = MIXER_IDLE; 2168 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2169 mBytesWritten = 0; 2170 mBytesRemaining = 0; 2171 checkSilentMode_l(); 2172 2173 standbyTime = systemTime() + standbyDelay; 2174 sleepTime = idleSleepTime; 2175 if (mType == MIXER) { 2176 sleepTimeShift = 0; 2177 } 2178 2179 continue; 2180 } 2181 } 2182 2183 // mMixerStatusIgnoringFastTracks is also updated internally 2184 mMixerStatus = prepareTracks_l(&tracksToRemove); 2185 2186 // prevent any changes in effect chain list and in each effect chain 2187 // during mixing and effect process as the audio buffers could be deleted 2188 // or modified if an effect is created or deleted 2189 lockEffectChains_l(effectChains); 2190 } 2191 2192 if (mBytesRemaining == 0) { 2193 mCurrentWriteLength = 0; 2194 if (mMixerStatus == MIXER_TRACKS_READY) { 2195 // threadLoop_mix() sets mCurrentWriteLength 2196 threadLoop_mix(); 2197 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2198 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2199 // threadLoop_sleepTime sets sleepTime to 0 if data 2200 // must be written to HAL 2201 threadLoop_sleepTime(); 2202 if (sleepTime == 0) { 2203 mCurrentWriteLength = mixBufferSize; 2204 } 2205 } 2206 mBytesRemaining = mCurrentWriteLength; 2207 if (isSuspended()) { 2208 sleepTime = suspendSleepTimeUs(); 2209 // simulate write to HAL when suspended 2210 mBytesWritten += mixBufferSize; 2211 mBytesRemaining = 0; 2212 } 2213 2214 // only process effects if we're going to write 2215 if (sleepTime == 0) { 2216 for (size_t i = 0; i < effectChains.size(); i ++) { 2217 effectChains[i]->process_l(); 2218 } 2219 } 2220 } 2221 2222 // enable changes in effect chain 2223 unlockEffectChains(effectChains); 2224 2225 if (!waitingAsyncCallback()) { 2226 // sleepTime == 0 means we must write to audio hardware 2227 if (sleepTime == 0) { 2228 if (mBytesRemaining) { 2229 ssize_t ret = threadLoop_write(); 2230 if (ret < 0) { 2231 mBytesRemaining = 0; 2232 } else { 2233 mBytesWritten += ret; 2234 mBytesRemaining -= ret; 2235 } 2236 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2237 (mMixerStatus == MIXER_DRAIN_ALL)) { 2238 threadLoop_drain(); 2239 } 2240if (mType == MIXER) { 2241 // write blocked detection 2242 nsecs_t now = systemTime(); 2243 nsecs_t delta = now - mLastWriteTime; 2244 if (!mStandby && delta > maxPeriod) { 2245 mNumDelayedWrites++; 2246 if ((now - lastWarning) > kWarningThrottleNs) { 2247 ATRACE_NAME("underrun"); 2248 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2249 ns2ms(delta), mNumDelayedWrites, this); 2250 lastWarning = now; 2251 } 2252 } 2253} 2254 2255 mStandby = false; 2256 } else { 2257 usleep(sleepTime); 2258 } 2259 } 2260 2261 // Finally let go of removed track(s), without the lock held 2262 // since we can't guarantee the destructors won't acquire that 2263 // same lock. This will also mutate and push a new fast mixer state. 2264 threadLoop_removeTracks(tracksToRemove); 2265 tracksToRemove.clear(); 2266 2267 // FIXME I don't understand the need for this here; 2268 // it was in the original code but maybe the 2269 // assignment in saveOutputTracks() makes this unnecessary? 2270 clearOutputTracks(); 2271 2272 // Effect chains will be actually deleted here if they were removed from 2273 // mEffectChains list during mixing or effects processing 2274 effectChains.clear(); 2275 2276 // FIXME Note that the above .clear() is no longer necessary since effectChains 2277 // is now local to this block, but will keep it for now (at least until merge done). 2278 } 2279 2280 threadLoop_exit(); 2281 2282 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2283 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2284 // put output stream into standby mode 2285 if (!mStandby) { 2286 mOutput->stream->common.standby(&mOutput->stream->common); 2287 } 2288 } 2289 2290 releaseWakeLock(); 2291 2292 ALOGV("Thread %p type %d exiting", this, mType); 2293 return false; 2294} 2295 2296// removeTracks_l() must be called with ThreadBase::mLock held 2297void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2298{ 2299 size_t count = tracksToRemove.size(); 2300 if (count) { 2301 for (size_t i=0 ; i<count ; i++) { 2302 const sp<Track>& track = tracksToRemove.itemAt(i); 2303 mActiveTracks.remove(track); 2304 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2305 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2306 if (chain != 0) { 2307 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2308 track->sessionId()); 2309 chain->decActiveTrackCnt(); 2310 } 2311 if (track->isTerminated()) { 2312 removeTrack_l(track); 2313 } 2314 } 2315 } 2316 2317} 2318 2319// ---------------------------------------------------------------------------- 2320 2321AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2322 audio_io_handle_t id, audio_devices_t device, type_t type) 2323 : PlaybackThread(audioFlinger, output, id, device, type), 2324 // mAudioMixer below 2325 // mFastMixer below 2326 mFastMixerFutex(0) 2327 // mOutputSink below 2328 // mPipeSink below 2329 // mNormalSink below 2330{ 2331 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2332 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2333 "mFrameCount=%d, mNormalFrameCount=%d", 2334 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2335 mNormalFrameCount); 2336 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2337 2338 // FIXME - Current mixer implementation only supports stereo output 2339 if (mChannelCount != FCC_2) { 2340 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2341 } 2342 2343 // create an NBAIO sink for the HAL output stream, and negotiate 2344 mOutputSink = new AudioStreamOutSink(output->stream); 2345 size_t numCounterOffers = 0; 2346 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2347 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2348 ALOG_ASSERT(index == 0); 2349 2350 // initialize fast mixer depending on configuration 2351 bool initFastMixer; 2352 switch (kUseFastMixer) { 2353 case FastMixer_Never: 2354 initFastMixer = false; 2355 break; 2356 case FastMixer_Always: 2357 initFastMixer = true; 2358 break; 2359 case FastMixer_Static: 2360 case FastMixer_Dynamic: 2361 initFastMixer = mFrameCount < mNormalFrameCount; 2362 break; 2363 } 2364 if (initFastMixer) { 2365 2366 // create a MonoPipe to connect our submix to FastMixer 2367 NBAIO_Format format = mOutputSink->format(); 2368 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2369 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2370 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2371 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2372 const NBAIO_Format offers[1] = {format}; 2373 size_t numCounterOffers = 0; 2374 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2375 ALOG_ASSERT(index == 0); 2376 monoPipe->setAvgFrames((mScreenState & 1) ? 2377 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2378 mPipeSink = monoPipe; 2379 2380#ifdef TEE_SINK 2381 if (mTeeSinkOutputEnabled) { 2382 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2383 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2384 numCounterOffers = 0; 2385 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2386 ALOG_ASSERT(index == 0); 2387 mTeeSink = teeSink; 2388 PipeReader *teeSource = new PipeReader(*teeSink); 2389 numCounterOffers = 0; 2390 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2391 ALOG_ASSERT(index == 0); 2392 mTeeSource = teeSource; 2393 } 2394#endif 2395 2396 // create fast mixer and configure it initially with just one fast track for our submix 2397 mFastMixer = new FastMixer(); 2398 FastMixerStateQueue *sq = mFastMixer->sq(); 2399#ifdef STATE_QUEUE_DUMP 2400 sq->setObserverDump(&mStateQueueObserverDump); 2401 sq->setMutatorDump(&mStateQueueMutatorDump); 2402#endif 2403 FastMixerState *state = sq->begin(); 2404 FastTrack *fastTrack = &state->mFastTracks[0]; 2405 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2406 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2407 fastTrack->mVolumeProvider = NULL; 2408 fastTrack->mGeneration++; 2409 state->mFastTracksGen++; 2410 state->mTrackMask = 1; 2411 // fast mixer will use the HAL output sink 2412 state->mOutputSink = mOutputSink.get(); 2413 state->mOutputSinkGen++; 2414 state->mFrameCount = mFrameCount; 2415 state->mCommand = FastMixerState::COLD_IDLE; 2416 // already done in constructor initialization list 2417 //mFastMixerFutex = 0; 2418 state->mColdFutexAddr = &mFastMixerFutex; 2419 state->mColdGen++; 2420 state->mDumpState = &mFastMixerDumpState; 2421#ifdef TEE_SINK 2422 state->mTeeSink = mTeeSink.get(); 2423#endif 2424 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2425 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2426 sq->end(); 2427 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2428 2429 // start the fast mixer 2430 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2431 pid_t tid = mFastMixer->getTid(); 2432 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2433 if (err != 0) { 2434 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2435 kPriorityFastMixer, getpid_cached, tid, err); 2436 } 2437 2438#ifdef AUDIO_WATCHDOG 2439 // create and start the watchdog 2440 mAudioWatchdog = new AudioWatchdog(); 2441 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2442 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2443 tid = mAudioWatchdog->getTid(); 2444 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2445 if (err != 0) { 2446 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2447 kPriorityFastMixer, getpid_cached, tid, err); 2448 } 2449#endif 2450 2451 } else { 2452 mFastMixer = NULL; 2453 } 2454 2455 switch (kUseFastMixer) { 2456 case FastMixer_Never: 2457 case FastMixer_Dynamic: 2458 mNormalSink = mOutputSink; 2459 break; 2460 case FastMixer_Always: 2461 mNormalSink = mPipeSink; 2462 break; 2463 case FastMixer_Static: 2464 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2465 break; 2466 } 2467} 2468 2469AudioFlinger::MixerThread::~MixerThread() 2470{ 2471 if (mFastMixer != NULL) { 2472 FastMixerStateQueue *sq = mFastMixer->sq(); 2473 FastMixerState *state = sq->begin(); 2474 if (state->mCommand == FastMixerState::COLD_IDLE) { 2475 int32_t old = android_atomic_inc(&mFastMixerFutex); 2476 if (old == -1) { 2477 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2478 } 2479 } 2480 state->mCommand = FastMixerState::EXIT; 2481 sq->end(); 2482 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2483 mFastMixer->join(); 2484 // Though the fast mixer thread has exited, it's state queue is still valid. 2485 // We'll use that extract the final state which contains one remaining fast track 2486 // corresponding to our sub-mix. 2487 state = sq->begin(); 2488 ALOG_ASSERT(state->mTrackMask == 1); 2489 FastTrack *fastTrack = &state->mFastTracks[0]; 2490 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2491 delete fastTrack->mBufferProvider; 2492 sq->end(false /*didModify*/); 2493 delete mFastMixer; 2494#ifdef AUDIO_WATCHDOG 2495 if (mAudioWatchdog != 0) { 2496 mAudioWatchdog->requestExit(); 2497 mAudioWatchdog->requestExitAndWait(); 2498 mAudioWatchdog.clear(); 2499 } 2500#endif 2501 } 2502 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2503 delete mAudioMixer; 2504} 2505 2506 2507uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2508{ 2509 if (mFastMixer != NULL) { 2510 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2511 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2512 } 2513 return latency; 2514} 2515 2516 2517void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2518{ 2519 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2520} 2521 2522ssize_t AudioFlinger::MixerThread::threadLoop_write() 2523{ 2524 // FIXME we should only do one push per cycle; confirm this is true 2525 // Start the fast mixer if it's not already running 2526 if (mFastMixer != NULL) { 2527 FastMixerStateQueue *sq = mFastMixer->sq(); 2528 FastMixerState *state = sq->begin(); 2529 if (state->mCommand != FastMixerState::MIX_WRITE && 2530 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2531 if (state->mCommand == FastMixerState::COLD_IDLE) { 2532 int32_t old = android_atomic_inc(&mFastMixerFutex); 2533 if (old == -1) { 2534 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2535 } 2536#ifdef AUDIO_WATCHDOG 2537 if (mAudioWatchdog != 0) { 2538 mAudioWatchdog->resume(); 2539 } 2540#endif 2541 } 2542 state->mCommand = FastMixerState::MIX_WRITE; 2543 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2544 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2545 sq->end(); 2546 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2547 if (kUseFastMixer == FastMixer_Dynamic) { 2548 mNormalSink = mPipeSink; 2549 } 2550 } else { 2551 sq->end(false /*didModify*/); 2552 } 2553 } 2554 return PlaybackThread::threadLoop_write(); 2555} 2556 2557void AudioFlinger::MixerThread::threadLoop_standby() 2558{ 2559 // Idle the fast mixer if it's currently running 2560 if (mFastMixer != NULL) { 2561 FastMixerStateQueue *sq = mFastMixer->sq(); 2562 FastMixerState *state = sq->begin(); 2563 if (!(state->mCommand & FastMixerState::IDLE)) { 2564 state->mCommand = FastMixerState::COLD_IDLE; 2565 state->mColdFutexAddr = &mFastMixerFutex; 2566 state->mColdGen++; 2567 mFastMixerFutex = 0; 2568 sq->end(); 2569 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2570 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2571 if (kUseFastMixer == FastMixer_Dynamic) { 2572 mNormalSink = mOutputSink; 2573 } 2574#ifdef AUDIO_WATCHDOG 2575 if (mAudioWatchdog != 0) { 2576 mAudioWatchdog->pause(); 2577 } 2578#endif 2579 } else { 2580 sq->end(false /*didModify*/); 2581 } 2582 } 2583 PlaybackThread::threadLoop_standby(); 2584} 2585 2586// Empty implementation for standard mixer 2587// Overridden for offloaded playback 2588void AudioFlinger::PlaybackThread::flushOutput_l() 2589{ 2590} 2591 2592bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2593{ 2594 return false; 2595} 2596 2597bool AudioFlinger::PlaybackThread::shouldStandby_l() 2598{ 2599 return !mStandby; 2600} 2601 2602bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2603{ 2604 Mutex::Autolock _l(mLock); 2605 return waitingAsyncCallback_l(); 2606} 2607 2608// shared by MIXER and DIRECT, overridden by DUPLICATING 2609void AudioFlinger::PlaybackThread::threadLoop_standby() 2610{ 2611 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2612 mOutput->stream->common.standby(&mOutput->stream->common); 2613 if (mUseAsyncWrite != 0) { 2614 mWriteBlocked = false; 2615 mDraining = false; 2616 ALOG_ASSERT(mCallbackThread != 0); 2617 mCallbackThread->setWriteBlocked(false); 2618 mCallbackThread->setDraining(false); 2619 } 2620} 2621 2622void AudioFlinger::MixerThread::threadLoop_mix() 2623{ 2624 // obtain the presentation timestamp of the next output buffer 2625 int64_t pts; 2626 status_t status = INVALID_OPERATION; 2627 2628 if (mNormalSink != 0) { 2629 status = mNormalSink->getNextWriteTimestamp(&pts); 2630 } else { 2631 status = mOutputSink->getNextWriteTimestamp(&pts); 2632 } 2633 2634 if (status != NO_ERROR) { 2635 pts = AudioBufferProvider::kInvalidPTS; 2636 } 2637 2638 // mix buffers... 2639 mAudioMixer->process(pts); 2640 mCurrentWriteLength = mixBufferSize; 2641 // increase sleep time progressively when application underrun condition clears. 2642 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2643 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2644 // such that we would underrun the audio HAL. 2645 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2646 sleepTimeShift--; 2647 } 2648 sleepTime = 0; 2649 standbyTime = systemTime() + standbyDelay; 2650 //TODO: delay standby when effects have a tail 2651} 2652 2653void AudioFlinger::MixerThread::threadLoop_sleepTime() 2654{ 2655 // If no tracks are ready, sleep once for the duration of an output 2656 // buffer size, then write 0s to the output 2657 if (sleepTime == 0) { 2658 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2659 sleepTime = activeSleepTime >> sleepTimeShift; 2660 if (sleepTime < kMinThreadSleepTimeUs) { 2661 sleepTime = kMinThreadSleepTimeUs; 2662 } 2663 // reduce sleep time in case of consecutive application underruns to avoid 2664 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2665 // duration we would end up writing less data than needed by the audio HAL if 2666 // the condition persists. 2667 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2668 sleepTimeShift++; 2669 } 2670 } else { 2671 sleepTime = idleSleepTime; 2672 } 2673 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2674 memset (mMixBuffer, 0, mixBufferSize); 2675 sleepTime = 0; 2676 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2677 "anticipated start"); 2678 } 2679 // TODO add standby time extension fct of effect tail 2680} 2681 2682// prepareTracks_l() must be called with ThreadBase::mLock held 2683AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2684 Vector< sp<Track> > *tracksToRemove) 2685{ 2686 2687 mixer_state mixerStatus = MIXER_IDLE; 2688 // find out which tracks need to be processed 2689 size_t count = mActiveTracks.size(); 2690 size_t mixedTracks = 0; 2691 size_t tracksWithEffect = 0; 2692 // counts only _active_ fast tracks 2693 size_t fastTracks = 0; 2694 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2695 2696 float masterVolume = mMasterVolume; 2697 bool masterMute = mMasterMute; 2698 2699 if (masterMute) { 2700 masterVolume = 0; 2701 } 2702 // Delegate master volume control to effect in output mix effect chain if needed 2703 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2704 if (chain != 0) { 2705 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2706 chain->setVolume_l(&v, &v); 2707 masterVolume = (float)((v + (1 << 23)) >> 24); 2708 chain.clear(); 2709 } 2710 2711 // prepare a new state to push 2712 FastMixerStateQueue *sq = NULL; 2713 FastMixerState *state = NULL; 2714 bool didModify = false; 2715 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2716 if (mFastMixer != NULL) { 2717 sq = mFastMixer->sq(); 2718 state = sq->begin(); 2719 } 2720 2721 for (size_t i=0 ; i<count ; i++) { 2722 const sp<Track> t = mActiveTracks[i].promote(); 2723 if (t == 0) { 2724 continue; 2725 } 2726 2727 // this const just means the local variable doesn't change 2728 Track* const track = t.get(); 2729 2730 // process fast tracks 2731 if (track->isFastTrack()) { 2732 2733 // It's theoretically possible (though unlikely) for a fast track to be created 2734 // and then removed within the same normal mix cycle. This is not a problem, as 2735 // the track never becomes active so it's fast mixer slot is never touched. 2736 // The converse, of removing an (active) track and then creating a new track 2737 // at the identical fast mixer slot within the same normal mix cycle, 2738 // is impossible because the slot isn't marked available until the end of each cycle. 2739 int j = track->mFastIndex; 2740 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2741 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2742 FastTrack *fastTrack = &state->mFastTracks[j]; 2743 2744 // Determine whether the track is currently in underrun condition, 2745 // and whether it had a recent underrun. 2746 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2747 FastTrackUnderruns underruns = ftDump->mUnderruns; 2748 uint32_t recentFull = (underruns.mBitFields.mFull - 2749 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2750 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2751 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2752 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2753 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2754 uint32_t recentUnderruns = recentPartial + recentEmpty; 2755 track->mObservedUnderruns = underruns; 2756 // don't count underruns that occur while stopping or pausing 2757 // or stopped which can occur when flush() is called while active 2758 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2759 recentUnderruns > 0) { 2760 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2761 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2762 } 2763 2764 // This is similar to the state machine for normal tracks, 2765 // with a few modifications for fast tracks. 2766 bool isActive = true; 2767 switch (track->mState) { 2768 case TrackBase::STOPPING_1: 2769 // track stays active in STOPPING_1 state until first underrun 2770 if (recentUnderruns > 0 || track->isTerminated()) { 2771 track->mState = TrackBase::STOPPING_2; 2772 } 2773 break; 2774 case TrackBase::PAUSING: 2775 // ramp down is not yet implemented 2776 track->setPaused(); 2777 break; 2778 case TrackBase::RESUMING: 2779 // ramp up is not yet implemented 2780 track->mState = TrackBase::ACTIVE; 2781 break; 2782 case TrackBase::ACTIVE: 2783 if (recentFull > 0 || recentPartial > 0) { 2784 // track has provided at least some frames recently: reset retry count 2785 track->mRetryCount = kMaxTrackRetries; 2786 } 2787 if (recentUnderruns == 0) { 2788 // no recent underruns: stay active 2789 break; 2790 } 2791 // there has recently been an underrun of some kind 2792 if (track->sharedBuffer() == 0) { 2793 // were any of the recent underruns "empty" (no frames available)? 2794 if (recentEmpty == 0) { 2795 // no, then ignore the partial underruns as they are allowed indefinitely 2796 break; 2797 } 2798 // there has recently been an "empty" underrun: decrement the retry counter 2799 if (--(track->mRetryCount) > 0) { 2800 break; 2801 } 2802 // indicate to client process that the track was disabled because of underrun; 2803 // it will then automatically call start() when data is available 2804 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2805 // remove from active list, but state remains ACTIVE [confusing but true] 2806 isActive = false; 2807 break; 2808 } 2809 // fall through 2810 case TrackBase::STOPPING_2: 2811 case TrackBase::PAUSED: 2812 case TrackBase::STOPPED: 2813 case TrackBase::FLUSHED: // flush() while active 2814 // Check for presentation complete if track is inactive 2815 // We have consumed all the buffers of this track. 2816 // This would be incomplete if we auto-paused on underrun 2817 { 2818 size_t audioHALFrames = 2819 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2820 size_t framesWritten = mBytesWritten / mFrameSize; 2821 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2822 // track stays in active list until presentation is complete 2823 break; 2824 } 2825 } 2826 if (track->isStopping_2()) { 2827 track->mState = TrackBase::STOPPED; 2828 } 2829 if (track->isStopped()) { 2830 // Can't reset directly, as fast mixer is still polling this track 2831 // track->reset(); 2832 // So instead mark this track as needing to be reset after push with ack 2833 resetMask |= 1 << i; 2834 } 2835 isActive = false; 2836 break; 2837 case TrackBase::IDLE: 2838 default: 2839 LOG_FATAL("unexpected track state %d", track->mState); 2840 } 2841 2842 if (isActive) { 2843 // was it previously inactive? 2844 if (!(state->mTrackMask & (1 << j))) { 2845 ExtendedAudioBufferProvider *eabp = track; 2846 VolumeProvider *vp = track; 2847 fastTrack->mBufferProvider = eabp; 2848 fastTrack->mVolumeProvider = vp; 2849 fastTrack->mSampleRate = track->mSampleRate; 2850 fastTrack->mChannelMask = track->mChannelMask; 2851 fastTrack->mGeneration++; 2852 state->mTrackMask |= 1 << j; 2853 didModify = true; 2854 // no acknowledgement required for newly active tracks 2855 } 2856 // cache the combined master volume and stream type volume for fast mixer; this 2857 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2858 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2859 ++fastTracks; 2860 } else { 2861 // was it previously active? 2862 if (state->mTrackMask & (1 << j)) { 2863 fastTrack->mBufferProvider = NULL; 2864 fastTrack->mGeneration++; 2865 state->mTrackMask &= ~(1 << j); 2866 didModify = true; 2867 // If any fast tracks were removed, we must wait for acknowledgement 2868 // because we're about to decrement the last sp<> on those tracks. 2869 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2870 } else { 2871 LOG_FATAL("fast track %d should have been active", j); 2872 } 2873 tracksToRemove->add(track); 2874 // Avoids a misleading display in dumpsys 2875 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2876 } 2877 continue; 2878 } 2879 2880 { // local variable scope to avoid goto warning 2881 2882 audio_track_cblk_t* cblk = track->cblk(); 2883 2884 // The first time a track is added we wait 2885 // for all its buffers to be filled before processing it 2886 int name = track->name(); 2887 // make sure that we have enough frames to mix one full buffer. 2888 // enforce this condition only once to enable draining the buffer in case the client 2889 // app does not call stop() and relies on underrun to stop: 2890 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2891 // during last round 2892 size_t desiredFrames; 2893 uint32_t sr = track->sampleRate(); 2894 if (sr == mSampleRate) { 2895 desiredFrames = mNormalFrameCount; 2896 } else { 2897 // +1 for rounding and +1 for additional sample needed for interpolation 2898 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2899 // add frames already consumed but not yet released by the resampler 2900 // because cblk->framesReady() will include these frames 2901 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2902 // the minimum track buffer size is normally twice the number of frames necessary 2903 // to fill one buffer and the resampler should not leave more than one buffer worth 2904 // of unreleased frames after each pass, but just in case... 2905 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2906 } 2907 uint32_t minFrames = 1; 2908 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2909 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2910 minFrames = desiredFrames; 2911 } 2912 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2913 size_t framesReady; 2914 if (track->sharedBuffer() == 0) { 2915 framesReady = track->framesReady(); 2916 } else if (track->isStopped()) { 2917 framesReady = 0; 2918 } else { 2919 framesReady = 1; 2920 } 2921 if ((framesReady >= minFrames) && track->isReady() && 2922 !track->isPaused() && !track->isTerminated()) 2923 { 2924 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2925 2926 mixedTracks++; 2927 2928 // track->mainBuffer() != mMixBuffer means there is an effect chain 2929 // connected to the track 2930 chain.clear(); 2931 if (track->mainBuffer() != mMixBuffer) { 2932 chain = getEffectChain_l(track->sessionId()); 2933 // Delegate volume control to effect in track effect chain if needed 2934 if (chain != 0) { 2935 tracksWithEffect++; 2936 } else { 2937 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2938 "session %d", 2939 name, track->sessionId()); 2940 } 2941 } 2942 2943 2944 int param = AudioMixer::VOLUME; 2945 if (track->mFillingUpStatus == Track::FS_FILLED) { 2946 // no ramp for the first volume setting 2947 track->mFillingUpStatus = Track::FS_ACTIVE; 2948 if (track->mState == TrackBase::RESUMING) { 2949 track->mState = TrackBase::ACTIVE; 2950 param = AudioMixer::RAMP_VOLUME; 2951 } 2952 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2953 // FIXME should not make a decision based on mServer 2954 } else if (cblk->mServer != 0) { 2955 // If the track is stopped before the first frame was mixed, 2956 // do not apply ramp 2957 param = AudioMixer::RAMP_VOLUME; 2958 } 2959 2960 // compute volume for this track 2961 uint32_t vl, vr, va; 2962 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2963 vl = vr = va = 0; 2964 if (track->isPausing()) { 2965 track->setPaused(); 2966 } 2967 } else { 2968 2969 // read original volumes with volume control 2970 float typeVolume = mStreamTypes[track->streamType()].volume; 2971 float v = masterVolume * typeVolume; 2972 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2973 uint32_t vlr = proxy->getVolumeLR(); 2974 vl = vlr & 0xFFFF; 2975 vr = vlr >> 16; 2976 // track volumes come from shared memory, so can't be trusted and must be clamped 2977 if (vl > MAX_GAIN_INT) { 2978 ALOGV("Track left volume out of range: %04X", vl); 2979 vl = MAX_GAIN_INT; 2980 } 2981 if (vr > MAX_GAIN_INT) { 2982 ALOGV("Track right volume out of range: %04X", vr); 2983 vr = MAX_GAIN_INT; 2984 } 2985 // now apply the master volume and stream type volume 2986 vl = (uint32_t)(v * vl) << 12; 2987 vr = (uint32_t)(v * vr) << 12; 2988 // assuming master volume and stream type volume each go up to 1.0, 2989 // vl and vr are now in 8.24 format 2990 2991 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2992 // send level comes from shared memory and so may be corrupt 2993 if (sendLevel > MAX_GAIN_INT) { 2994 ALOGV("Track send level out of range: %04X", sendLevel); 2995 sendLevel = MAX_GAIN_INT; 2996 } 2997 va = (uint32_t)(v * sendLevel); 2998 } 2999 3000 // Delegate volume control to effect in track effect chain if needed 3001 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3002 // Do not ramp volume if volume is controlled by effect 3003 param = AudioMixer::VOLUME; 3004 track->mHasVolumeController = true; 3005 } else { 3006 // force no volume ramp when volume controller was just disabled or removed 3007 // from effect chain to avoid volume spike 3008 if (track->mHasVolumeController) { 3009 param = AudioMixer::VOLUME; 3010 } 3011 track->mHasVolumeController = false; 3012 } 3013 3014 // Convert volumes from 8.24 to 4.12 format 3015 // This additional clamping is needed in case chain->setVolume_l() overshot 3016 vl = (vl + (1 << 11)) >> 12; 3017 if (vl > MAX_GAIN_INT) { 3018 vl = MAX_GAIN_INT; 3019 } 3020 vr = (vr + (1 << 11)) >> 12; 3021 if (vr > MAX_GAIN_INT) { 3022 vr = MAX_GAIN_INT; 3023 } 3024 3025 if (va > MAX_GAIN_INT) { 3026 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3027 } 3028 3029 // XXX: these things DON'T need to be done each time 3030 mAudioMixer->setBufferProvider(name, track); 3031 mAudioMixer->enable(name); 3032 3033 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3034 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3035 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3036 mAudioMixer->setParameter( 3037 name, 3038 AudioMixer::TRACK, 3039 AudioMixer::FORMAT, (void *)track->format()); 3040 mAudioMixer->setParameter( 3041 name, 3042 AudioMixer::TRACK, 3043 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3044 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3045 uint32_t maxSampleRate = mSampleRate * 2; 3046 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3047 if (reqSampleRate == 0) { 3048 reqSampleRate = mSampleRate; 3049 } else if (reqSampleRate > maxSampleRate) { 3050 reqSampleRate = maxSampleRate; 3051 } 3052 mAudioMixer->setParameter( 3053 name, 3054 AudioMixer::RESAMPLE, 3055 AudioMixer::SAMPLE_RATE, 3056 (void *)reqSampleRate); 3057 mAudioMixer->setParameter( 3058 name, 3059 AudioMixer::TRACK, 3060 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3061 mAudioMixer->setParameter( 3062 name, 3063 AudioMixer::TRACK, 3064 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3065 3066 // reset retry count 3067 track->mRetryCount = kMaxTrackRetries; 3068 3069 // If one track is ready, set the mixer ready if: 3070 // - the mixer was not ready during previous round OR 3071 // - no other track is not ready 3072 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3073 mixerStatus != MIXER_TRACKS_ENABLED) { 3074 mixerStatus = MIXER_TRACKS_READY; 3075 } 3076 } else { 3077 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3078 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3079 } 3080 // clear effect chain input buffer if an active track underruns to avoid sending 3081 // previous audio buffer again to effects 3082 chain = getEffectChain_l(track->sessionId()); 3083 if (chain != 0) { 3084 chain->clearInputBuffer(); 3085 } 3086 3087 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3088 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3089 track->isStopped() || track->isPaused()) { 3090 // We have consumed all the buffers of this track. 3091 // Remove it from the list of active tracks. 3092 // TODO: use actual buffer filling status instead of latency when available from 3093 // audio HAL 3094 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3095 size_t framesWritten = mBytesWritten / mFrameSize; 3096 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3097 if (track->isStopped()) { 3098 track->reset(); 3099 } 3100 tracksToRemove->add(track); 3101 } 3102 } else { 3103 // No buffers for this track. Give it a few chances to 3104 // fill a buffer, then remove it from active list. 3105 if (--(track->mRetryCount) <= 0) { 3106 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3107 tracksToRemove->add(track); 3108 // indicate to client process that the track was disabled because of underrun; 3109 // it will then automatically call start() when data is available 3110 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3111 // If one track is not ready, mark the mixer also not ready if: 3112 // - the mixer was ready during previous round OR 3113 // - no other track is ready 3114 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3115 mixerStatus != MIXER_TRACKS_READY) { 3116 mixerStatus = MIXER_TRACKS_ENABLED; 3117 } 3118 } 3119 mAudioMixer->disable(name); 3120 } 3121 3122 } // local variable scope to avoid goto warning 3123track_is_ready: ; 3124 3125 } 3126 3127 // Push the new FastMixer state if necessary 3128 bool pauseAudioWatchdog = false; 3129 if (didModify) { 3130 state->mFastTracksGen++; 3131 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3132 if (kUseFastMixer == FastMixer_Dynamic && 3133 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3134 state->mCommand = FastMixerState::COLD_IDLE; 3135 state->mColdFutexAddr = &mFastMixerFutex; 3136 state->mColdGen++; 3137 mFastMixerFutex = 0; 3138 if (kUseFastMixer == FastMixer_Dynamic) { 3139 mNormalSink = mOutputSink; 3140 } 3141 // If we go into cold idle, need to wait for acknowledgement 3142 // so that fast mixer stops doing I/O. 3143 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3144 pauseAudioWatchdog = true; 3145 } 3146 } 3147 if (sq != NULL) { 3148 sq->end(didModify); 3149 sq->push(block); 3150 } 3151#ifdef AUDIO_WATCHDOG 3152 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3153 mAudioWatchdog->pause(); 3154 } 3155#endif 3156 3157 // Now perform the deferred reset on fast tracks that have stopped 3158 while (resetMask != 0) { 3159 size_t i = __builtin_ctz(resetMask); 3160 ALOG_ASSERT(i < count); 3161 resetMask &= ~(1 << i); 3162 sp<Track> t = mActiveTracks[i].promote(); 3163 if (t == 0) { 3164 continue; 3165 } 3166 Track* track = t.get(); 3167 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3168 track->reset(); 3169 } 3170 3171 // remove all the tracks that need to be... 3172 removeTracks_l(*tracksToRemove); 3173 3174 // mix buffer must be cleared if all tracks are connected to an 3175 // effect chain as in this case the mixer will not write to 3176 // mix buffer and track effects will accumulate into it 3177 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3178 (mixedTracks == 0 && fastTracks > 0))) { 3179 // FIXME as a performance optimization, should remember previous zero status 3180 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3181 } 3182 3183 // if any fast tracks, then status is ready 3184 mMixerStatusIgnoringFastTracks = mixerStatus; 3185 if (fastTracks > 0) { 3186 mixerStatus = MIXER_TRACKS_READY; 3187 } 3188 return mixerStatus; 3189} 3190 3191// getTrackName_l() must be called with ThreadBase::mLock held 3192int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3193{ 3194 return mAudioMixer->getTrackName(channelMask, sessionId); 3195} 3196 3197// deleteTrackName_l() must be called with ThreadBase::mLock held 3198void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3199{ 3200 ALOGV("remove track (%d) and delete from mixer", name); 3201 mAudioMixer->deleteTrackName(name); 3202} 3203 3204// checkForNewParameters_l() must be called with ThreadBase::mLock held 3205bool AudioFlinger::MixerThread::checkForNewParameters_l() 3206{ 3207 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3208 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3209 bool reconfig = false; 3210 3211 while (!mNewParameters.isEmpty()) { 3212 3213 if (mFastMixer != NULL) { 3214 FastMixerStateQueue *sq = mFastMixer->sq(); 3215 FastMixerState *state = sq->begin(); 3216 if (!(state->mCommand & FastMixerState::IDLE)) { 3217 previousCommand = state->mCommand; 3218 state->mCommand = FastMixerState::HOT_IDLE; 3219 sq->end(); 3220 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3221 } else { 3222 sq->end(false /*didModify*/); 3223 } 3224 } 3225 3226 status_t status = NO_ERROR; 3227 String8 keyValuePair = mNewParameters[0]; 3228 AudioParameter param = AudioParameter(keyValuePair); 3229 int value; 3230 3231 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3232 reconfig = true; 3233 } 3234 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3235 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3236 status = BAD_VALUE; 3237 } else { 3238 reconfig = true; 3239 } 3240 } 3241 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3242 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3243 status = BAD_VALUE; 3244 } else { 3245 reconfig = true; 3246 } 3247 } 3248 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3249 // do not accept frame count changes if tracks are open as the track buffer 3250 // size depends on frame count and correct behavior would not be guaranteed 3251 // if frame count is changed after track creation 3252 if (!mTracks.isEmpty()) { 3253 status = INVALID_OPERATION; 3254 } else { 3255 reconfig = true; 3256 } 3257 } 3258 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3259#ifdef ADD_BATTERY_DATA 3260 // when changing the audio output device, call addBatteryData to notify 3261 // the change 3262 if (mOutDevice != value) { 3263 uint32_t params = 0; 3264 // check whether speaker is on 3265 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3266 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3267 } 3268 3269 audio_devices_t deviceWithoutSpeaker 3270 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3271 // check if any other device (except speaker) is on 3272 if (value & deviceWithoutSpeaker ) { 3273 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3274 } 3275 3276 if (params != 0) { 3277 addBatteryData(params); 3278 } 3279 } 3280#endif 3281 3282 // forward device change to effects that have requested to be 3283 // aware of attached audio device. 3284 if (value != AUDIO_DEVICE_NONE) { 3285 mOutDevice = value; 3286 for (size_t i = 0; i < mEffectChains.size(); i++) { 3287 mEffectChains[i]->setDevice_l(mOutDevice); 3288 } 3289 } 3290 } 3291 3292 if (status == NO_ERROR) { 3293 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3294 keyValuePair.string()); 3295 if (!mStandby && status == INVALID_OPERATION) { 3296 mOutput->stream->common.standby(&mOutput->stream->common); 3297 mStandby = true; 3298 mBytesWritten = 0; 3299 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3300 keyValuePair.string()); 3301 } 3302 if (status == NO_ERROR && reconfig) { 3303 readOutputParameters(); 3304 delete mAudioMixer; 3305 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3306 for (size_t i = 0; i < mTracks.size() ; i++) { 3307 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3308 if (name < 0) { 3309 break; 3310 } 3311 mTracks[i]->mName = name; 3312 } 3313 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3314 } 3315 } 3316 3317 mNewParameters.removeAt(0); 3318 3319 mParamStatus = status; 3320 mParamCond.signal(); 3321 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3322 // already timed out waiting for the status and will never signal the condition. 3323 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3324 } 3325 3326 if (!(previousCommand & FastMixerState::IDLE)) { 3327 ALOG_ASSERT(mFastMixer != NULL); 3328 FastMixerStateQueue *sq = mFastMixer->sq(); 3329 FastMixerState *state = sq->begin(); 3330 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3331 state->mCommand = previousCommand; 3332 sq->end(); 3333 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3334 } 3335 3336 return reconfig; 3337} 3338 3339 3340void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3341{ 3342 const size_t SIZE = 256; 3343 char buffer[SIZE]; 3344 String8 result; 3345 3346 PlaybackThread::dumpInternals(fd, args); 3347 3348 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3349 result.append(buffer); 3350 write(fd, result.string(), result.size()); 3351 3352 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3353 const FastMixerDumpState copy(mFastMixerDumpState); 3354 copy.dump(fd); 3355 3356#ifdef STATE_QUEUE_DUMP 3357 // Similar for state queue 3358 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3359 observerCopy.dump(fd); 3360 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3361 mutatorCopy.dump(fd); 3362#endif 3363 3364#ifdef TEE_SINK 3365 // Write the tee output to a .wav file 3366 dumpTee(fd, mTeeSource, mId); 3367#endif 3368 3369#ifdef AUDIO_WATCHDOG 3370 if (mAudioWatchdog != 0) { 3371 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3372 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3373 wdCopy.dump(fd); 3374 } 3375#endif 3376} 3377 3378uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3379{ 3380 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3381} 3382 3383uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3384{ 3385 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3386} 3387 3388void AudioFlinger::MixerThread::cacheParameters_l() 3389{ 3390 PlaybackThread::cacheParameters_l(); 3391 3392 // FIXME: Relaxed timing because of a certain device that can't meet latency 3393 // Should be reduced to 2x after the vendor fixes the driver issue 3394 // increase threshold again due to low power audio mode. The way this warning 3395 // threshold is calculated and its usefulness should be reconsidered anyway. 3396 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3397} 3398 3399// ---------------------------------------------------------------------------- 3400 3401AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3402 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3403 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3404 // mLeftVolFloat, mRightVolFloat 3405{ 3406} 3407 3408AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3409 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3410 ThreadBase::type_t type) 3411 : PlaybackThread(audioFlinger, output, id, device, type) 3412 // mLeftVolFloat, mRightVolFloat 3413{ 3414} 3415 3416AudioFlinger::DirectOutputThread::~DirectOutputThread() 3417{ 3418} 3419 3420void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3421{ 3422 audio_track_cblk_t* cblk = track->cblk(); 3423 float left, right; 3424 3425 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3426 left = right = 0; 3427 } else { 3428 float typeVolume = mStreamTypes[track->streamType()].volume; 3429 float v = mMasterVolume * typeVolume; 3430 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3431 uint32_t vlr = proxy->getVolumeLR(); 3432 float v_clamped = v * (vlr & 0xFFFF); 3433 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3434 left = v_clamped/MAX_GAIN; 3435 v_clamped = v * (vlr >> 16); 3436 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3437 right = v_clamped/MAX_GAIN; 3438 } 3439 3440 if (lastTrack) { 3441 if (left != mLeftVolFloat || right != mRightVolFloat) { 3442 mLeftVolFloat = left; 3443 mRightVolFloat = right; 3444 3445 // Convert volumes from float to 8.24 3446 uint32_t vl = (uint32_t)(left * (1 << 24)); 3447 uint32_t vr = (uint32_t)(right * (1 << 24)); 3448 3449 // Delegate volume control to effect in track effect chain if needed 3450 // only one effect chain can be present on DirectOutputThread, so if 3451 // there is one, the track is connected to it 3452 if (!mEffectChains.isEmpty()) { 3453 mEffectChains[0]->setVolume_l(&vl, &vr); 3454 left = (float)vl / (1 << 24); 3455 right = (float)vr / (1 << 24); 3456 } 3457 if (mOutput->stream->set_volume) { 3458 mOutput->stream->set_volume(mOutput->stream, left, right); 3459 } 3460 } 3461 } 3462} 3463 3464 3465AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3466 Vector< sp<Track> > *tracksToRemove 3467) 3468{ 3469 size_t count = mActiveTracks.size(); 3470 mixer_state mixerStatus = MIXER_IDLE; 3471 3472 // find out which tracks need to be processed 3473 for (size_t i = 0; i < count; i++) { 3474 sp<Track> t = mActiveTracks[i].promote(); 3475 // The track died recently 3476 if (t == 0) { 3477 continue; 3478 } 3479 3480 Track* const track = t.get(); 3481 audio_track_cblk_t* cblk = track->cblk(); 3482 3483 // The first time a track is added we wait 3484 // for all its buffers to be filled before processing it 3485 uint32_t minFrames; 3486 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3487 minFrames = mNormalFrameCount; 3488 } else { 3489 minFrames = 1; 3490 } 3491 // Only consider last track started for volume and mixer state control. 3492 // This is the last entry in mActiveTracks unless a track underruns. 3493 // As we only care about the transition phase between two tracks on a 3494 // direct output, it is not a problem to ignore the underrun case. 3495 bool last = (i == (count - 1)); 3496 3497 if ((track->framesReady() >= minFrames) && track->isReady() && 3498 !track->isPaused() && !track->isTerminated()) 3499 { 3500 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3501 3502 if (track->mFillingUpStatus == Track::FS_FILLED) { 3503 track->mFillingUpStatus = Track::FS_ACTIVE; 3504 mLeftVolFloat = mRightVolFloat = 0; 3505 if (track->mState == TrackBase::RESUMING) { 3506 track->mState = TrackBase::ACTIVE; 3507 } 3508 } 3509 3510 // compute volume for this track 3511 processVolume_l(track, last); 3512 if (last) { 3513 // reset retry count 3514 track->mRetryCount = kMaxTrackRetriesDirect; 3515 mActiveTrack = t; 3516 mixerStatus = MIXER_TRACKS_READY; 3517 } 3518 } else { 3519 // clear effect chain input buffer if the last active track started underruns 3520 // to avoid sending previous audio buffer again to effects 3521 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3522 mEffectChains[0]->clearInputBuffer(); 3523 } 3524 3525 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3526 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3527 track->isStopped() || track->isPaused()) { 3528 // We have consumed all the buffers of this track. 3529 // Remove it from the list of active tracks. 3530 // TODO: implement behavior for compressed audio 3531 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3532 size_t framesWritten = mBytesWritten / mFrameSize; 3533 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3534 if (track->isStopped()) { 3535 track->reset(); 3536 } 3537 tracksToRemove->add(track); 3538 } 3539 } else { 3540 // No buffers for this track. Give it a few chances to 3541 // fill a buffer, then remove it from active list. 3542 // Only consider last track started for mixer state control 3543 if (--(track->mRetryCount) <= 0) { 3544 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3545 tracksToRemove->add(track); 3546 } else if (last) { 3547 mixerStatus = MIXER_TRACKS_ENABLED; 3548 } 3549 } 3550 } 3551 } 3552 3553 // remove all the tracks that need to be... 3554 removeTracks_l(*tracksToRemove); 3555 3556 return mixerStatus; 3557} 3558 3559void AudioFlinger::DirectOutputThread::threadLoop_mix() 3560{ 3561 size_t frameCount = mFrameCount; 3562 int8_t *curBuf = (int8_t *)mMixBuffer; 3563 // output audio to hardware 3564 while (frameCount) { 3565 AudioBufferProvider::Buffer buffer; 3566 buffer.frameCount = frameCount; 3567 mActiveTrack->getNextBuffer(&buffer); 3568 if (buffer.raw == NULL) { 3569 memset(curBuf, 0, frameCount * mFrameSize); 3570 break; 3571 } 3572 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3573 frameCount -= buffer.frameCount; 3574 curBuf += buffer.frameCount * mFrameSize; 3575 mActiveTrack->releaseBuffer(&buffer); 3576 } 3577 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3578 sleepTime = 0; 3579 standbyTime = systemTime() + standbyDelay; 3580 mActiveTrack.clear(); 3581} 3582 3583void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3584{ 3585 if (sleepTime == 0) { 3586 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3587 sleepTime = activeSleepTime; 3588 } else { 3589 sleepTime = idleSleepTime; 3590 } 3591 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3592 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3593 sleepTime = 0; 3594 } 3595} 3596 3597// getTrackName_l() must be called with ThreadBase::mLock held 3598int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3599 int sessionId) 3600{ 3601 return 0; 3602} 3603 3604// deleteTrackName_l() must be called with ThreadBase::mLock held 3605void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3606{ 3607} 3608 3609// checkForNewParameters_l() must be called with ThreadBase::mLock held 3610bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3611{ 3612 bool reconfig = false; 3613 3614 while (!mNewParameters.isEmpty()) { 3615 status_t status = NO_ERROR; 3616 String8 keyValuePair = mNewParameters[0]; 3617 AudioParameter param = AudioParameter(keyValuePair); 3618 int value; 3619 3620 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3621 // do not accept frame count changes if tracks are open as the track buffer 3622 // size depends on frame count and correct behavior would not be garantied 3623 // if frame count is changed after track creation 3624 if (!mTracks.isEmpty()) { 3625 status = INVALID_OPERATION; 3626 } else { 3627 reconfig = true; 3628 } 3629 } 3630 if (status == NO_ERROR) { 3631 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3632 keyValuePair.string()); 3633 if (!mStandby && status == INVALID_OPERATION) { 3634 mOutput->stream->common.standby(&mOutput->stream->common); 3635 mStandby = true; 3636 mBytesWritten = 0; 3637 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3638 keyValuePair.string()); 3639 } 3640 if (status == NO_ERROR && reconfig) { 3641 readOutputParameters(); 3642 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3643 } 3644 } 3645 3646 mNewParameters.removeAt(0); 3647 3648 mParamStatus = status; 3649 mParamCond.signal(); 3650 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3651 // already timed out waiting for the status and will never signal the condition. 3652 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3653 } 3654 return reconfig; 3655} 3656 3657uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3658{ 3659 uint32_t time; 3660 if (audio_is_linear_pcm(mFormat)) { 3661 time = PlaybackThread::activeSleepTimeUs(); 3662 } else { 3663 time = 10000; 3664 } 3665 return time; 3666} 3667 3668uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3669{ 3670 uint32_t time; 3671 if (audio_is_linear_pcm(mFormat)) { 3672 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3673 } else { 3674 time = 10000; 3675 } 3676 return time; 3677} 3678 3679uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3680{ 3681 uint32_t time; 3682 if (audio_is_linear_pcm(mFormat)) { 3683 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3684 } else { 3685 time = 10000; 3686 } 3687 return time; 3688} 3689 3690void AudioFlinger::DirectOutputThread::cacheParameters_l() 3691{ 3692 PlaybackThread::cacheParameters_l(); 3693 3694 // use shorter standby delay as on normal output to release 3695 // hardware resources as soon as possible 3696 standbyDelay = microseconds(activeSleepTime*2); 3697} 3698 3699// ---------------------------------------------------------------------------- 3700 3701AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3702 const sp<AudioFlinger::OffloadThread>& offloadThread) 3703 : Thread(false /*canCallJava*/), 3704 mOffloadThread(offloadThread), 3705 mWriteBlocked(false), 3706 mDraining(false) 3707{ 3708} 3709 3710AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3711{ 3712} 3713 3714void AudioFlinger::AsyncCallbackThread::onFirstRef() 3715{ 3716 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3717} 3718 3719bool AudioFlinger::AsyncCallbackThread::threadLoop() 3720{ 3721 while (!exitPending()) { 3722 bool writeBlocked; 3723 bool draining; 3724 3725 { 3726 Mutex::Autolock _l(mLock); 3727 mWaitWorkCV.wait(mLock); 3728 if (exitPending()) { 3729 break; 3730 } 3731 writeBlocked = mWriteBlocked; 3732 draining = mDraining; 3733 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3734 } 3735 { 3736 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3737 if (offloadThread != 0) { 3738 if (writeBlocked == false) { 3739 offloadThread->setWriteBlocked(false); 3740 } 3741 if (draining == false) { 3742 offloadThread->setDraining(false); 3743 } 3744 } 3745 } 3746 } 3747 return false; 3748} 3749 3750void AudioFlinger::AsyncCallbackThread::exit() 3751{ 3752 ALOGV("AsyncCallbackThread::exit"); 3753 Mutex::Autolock _l(mLock); 3754 requestExit(); 3755 mWaitWorkCV.broadcast(); 3756} 3757 3758void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3759{ 3760 Mutex::Autolock _l(mLock); 3761 mWriteBlocked = value; 3762 if (!value) { 3763 mWaitWorkCV.signal(); 3764 } 3765} 3766 3767void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3768{ 3769 Mutex::Autolock _l(mLock); 3770 mDraining = value; 3771 if (!value) { 3772 mWaitWorkCV.signal(); 3773 } 3774} 3775 3776 3777// ---------------------------------------------------------------------------- 3778AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3779 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3780 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3781 mHwPaused(false), 3782 mPausedBytesRemaining(0) 3783{ 3784 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3785} 3786 3787AudioFlinger::OffloadThread::~OffloadThread() 3788{ 3789 mPreviousTrack.clear(); 3790} 3791 3792void AudioFlinger::OffloadThread::threadLoop_exit() 3793{ 3794 if (mFlushPending || mHwPaused) { 3795 // If a flush is pending or track was paused, just discard buffered data 3796 flushHw_l(); 3797 } else { 3798 mMixerStatus = MIXER_DRAIN_ALL; 3799 threadLoop_drain(); 3800 } 3801 mCallbackThread->exit(); 3802 PlaybackThread::threadLoop_exit(); 3803} 3804 3805AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3806 Vector< sp<Track> > *tracksToRemove 3807) 3808{ 3809 ALOGV("OffloadThread::prepareTracks_l"); 3810 size_t count = mActiveTracks.size(); 3811 3812 mixer_state mixerStatus = MIXER_IDLE; 3813 // find out which tracks need to be processed 3814 for (size_t i = 0; i < count; i++) { 3815 sp<Track> t = mActiveTracks[i].promote(); 3816 // The track died recently 3817 if (t == 0) { 3818 continue; 3819 } 3820 Track* const track = t.get(); 3821 audio_track_cblk_t* cblk = track->cblk(); 3822 if (mPreviousTrack != NULL) { 3823 if (t != mPreviousTrack) { 3824 // Flush any data still being written from last track 3825 mBytesRemaining = 0; 3826 if (mPausedBytesRemaining) { 3827 // Last track was paused so we also need to flush saved 3828 // mixbuffer state and invalidate track so that it will 3829 // re-submit that unwritten data when it is next resumed 3830 mPausedBytesRemaining = 0; 3831 // Invalidate is a bit drastic - would be more efficient 3832 // to have a flag to tell client that some of the 3833 // previously written data was lost 3834 mPreviousTrack->invalidate(); 3835 } 3836 } 3837 } 3838 mPreviousTrack = t; 3839 bool last = (i == (count - 1)); 3840 if (track->isPausing()) { 3841 track->setPaused(); 3842 if (last) { 3843 if (!mHwPaused) { 3844 mOutput->stream->pause(mOutput->stream); 3845 mHwPaused = true; 3846 } 3847 // If we were part way through writing the mixbuffer to 3848 // the HAL we must save this until we resume 3849 // BUG - this will be wrong if a different track is made active, 3850 // in that case we want to discard the pending data in the 3851 // mixbuffer and tell the client to present it again when the 3852 // track is resumed 3853 mPausedWriteLength = mCurrentWriteLength; 3854 mPausedBytesRemaining = mBytesRemaining; 3855 mBytesRemaining = 0; // stop writing 3856 } 3857 tracksToRemove->add(track); 3858 } else if (track->framesReady() && track->isReady() && 3859 !track->isPaused() && !track->isTerminated()) { 3860 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3861 if (track->mFillingUpStatus == Track::FS_FILLED) { 3862 track->mFillingUpStatus = Track::FS_ACTIVE; 3863 mLeftVolFloat = mRightVolFloat = 0; 3864 if (track->mState == TrackBase::RESUMING) { 3865 if (mPausedBytesRemaining) { 3866 // Need to continue write that was interrupted 3867 mCurrentWriteLength = mPausedWriteLength; 3868 mBytesRemaining = mPausedBytesRemaining; 3869 mPausedBytesRemaining = 0; 3870 } 3871 track->mState = TrackBase::ACTIVE; 3872 } 3873 } 3874 3875 if (last) { 3876 if (mHwPaused) { 3877 mOutput->stream->resume(mOutput->stream); 3878 mHwPaused = false; 3879 // threadLoop_mix() will handle the case that we need to 3880 // resume an interrupted write 3881 } 3882 // reset retry count 3883 track->mRetryCount = kMaxTrackRetriesOffload; 3884 mActiveTrack = t; 3885 mixerStatus = MIXER_TRACKS_READY; 3886 } 3887 } else { 3888 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3889 if (track->isStopping_1()) { 3890 // Hardware buffer can hold a large amount of audio so we must 3891 // wait for all current track's data to drain before we say 3892 // that the track is stopped. 3893 if (mBytesRemaining == 0) { 3894 // Only start draining when all data in mixbuffer 3895 // has been written 3896 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3897 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3898 sleepTime = 0; 3899 standbyTime = systemTime() + standbyDelay; 3900 if (last) { 3901 mixerStatus = MIXER_DRAIN_TRACK; 3902 if (mHwPaused) { 3903 // It is possible to move from PAUSED to STOPPING_1 without 3904 // a resume so we must ensure hardware is running 3905 mOutput->stream->resume(mOutput->stream); 3906 mHwPaused = false; 3907 } 3908 } 3909 } 3910 } else if (track->isStopping_2()) { 3911 // Drain has completed, signal presentation complete 3912 if (!mDraining || !last) { 3913 track->mState = TrackBase::STOPPED; 3914 size_t audioHALFrames = 3915 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3916 size_t framesWritten = 3917 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3918 track->presentationComplete(framesWritten, audioHALFrames); 3919 track->reset(); 3920 tracksToRemove->add(track); 3921 } 3922 } else { 3923 // No buffers for this track. Give it a few chances to 3924 // fill a buffer, then remove it from active list. 3925 if (--(track->mRetryCount) <= 0) { 3926 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3927 track->name()); 3928 tracksToRemove->add(track); 3929 } else if (last){ 3930 mixerStatus = MIXER_TRACKS_ENABLED; 3931 } 3932 } 3933 } 3934 // compute volume for this track 3935 processVolume_l(track, last); 3936 } 3937 3938 if (mFlushPending) { 3939 flushHw_l(); 3940 mFlushPending = false; 3941 } 3942 3943 // remove all the tracks that need to be... 3944 removeTracks_l(*tracksToRemove); 3945 3946 return mixerStatus; 3947} 3948 3949void AudioFlinger::OffloadThread::flushOutput_l() 3950{ 3951 mFlushPending = true; 3952} 3953 3954// must be called with thread mutex locked 3955bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3956{ 3957 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3958 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3959 return true; 3960 } 3961 return false; 3962} 3963 3964// must be called with thread mutex locked 3965bool AudioFlinger::OffloadThread::shouldStandby_l() 3966{ 3967 bool TrackPaused = false; 3968 3969 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3970 // after a timeout and we will enter standby then. 3971 if (mTracks.size() > 0) { 3972 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3973 } 3974 3975 return !mStandby && !TrackPaused; 3976} 3977 3978 3979bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3980{ 3981 Mutex::Autolock _l(mLock); 3982 return waitingAsyncCallback_l(); 3983} 3984 3985void AudioFlinger::OffloadThread::flushHw_l() 3986{ 3987 mOutput->stream->flush(mOutput->stream); 3988 // Flush anything still waiting in the mixbuffer 3989 mCurrentWriteLength = 0; 3990 mBytesRemaining = 0; 3991 mPausedWriteLength = 0; 3992 mPausedBytesRemaining = 0; 3993 if (mUseAsyncWrite) { 3994 mWriteBlocked = false; 3995 mDraining = false; 3996 ALOG_ASSERT(mCallbackThread != 0); 3997 mCallbackThread->setWriteBlocked(false); 3998 mCallbackThread->setDraining(false); 3999 } 4000} 4001 4002// ---------------------------------------------------------------------------- 4003 4004AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4005 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4006 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4007 DUPLICATING), 4008 mWaitTimeMs(UINT_MAX) 4009{ 4010 addOutputTrack(mainThread); 4011} 4012 4013AudioFlinger::DuplicatingThread::~DuplicatingThread() 4014{ 4015 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4016 mOutputTracks[i]->destroy(); 4017 } 4018} 4019 4020void AudioFlinger::DuplicatingThread::threadLoop_mix() 4021{ 4022 // mix buffers... 4023 if (outputsReady(outputTracks)) { 4024 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4025 } else { 4026 memset(mMixBuffer, 0, mixBufferSize); 4027 } 4028 sleepTime = 0; 4029 writeFrames = mNormalFrameCount; 4030 mCurrentWriteLength = mixBufferSize; 4031 standbyTime = systemTime() + standbyDelay; 4032} 4033 4034void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4035{ 4036 if (sleepTime == 0) { 4037 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4038 sleepTime = activeSleepTime; 4039 } else { 4040 sleepTime = idleSleepTime; 4041 } 4042 } else if (mBytesWritten != 0) { 4043 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4044 writeFrames = mNormalFrameCount; 4045 memset(mMixBuffer, 0, mixBufferSize); 4046 } else { 4047 // flush remaining overflow buffers in output tracks 4048 writeFrames = 0; 4049 } 4050 sleepTime = 0; 4051 } 4052} 4053 4054ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4055{ 4056 for (size_t i = 0; i < outputTracks.size(); i++) { 4057 outputTracks[i]->write(mMixBuffer, writeFrames); 4058 } 4059 return (ssize_t)mixBufferSize; 4060} 4061 4062void AudioFlinger::DuplicatingThread::threadLoop_standby() 4063{ 4064 // DuplicatingThread implements standby by stopping all tracks 4065 for (size_t i = 0; i < outputTracks.size(); i++) { 4066 outputTracks[i]->stop(); 4067 } 4068} 4069 4070void AudioFlinger::DuplicatingThread::saveOutputTracks() 4071{ 4072 outputTracks = mOutputTracks; 4073} 4074 4075void AudioFlinger::DuplicatingThread::clearOutputTracks() 4076{ 4077 outputTracks.clear(); 4078} 4079 4080void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4081{ 4082 Mutex::Autolock _l(mLock); 4083 // FIXME explain this formula 4084 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4085 OutputTrack *outputTrack = new OutputTrack(thread, 4086 this, 4087 mSampleRate, 4088 mFormat, 4089 mChannelMask, 4090 frameCount); 4091 if (outputTrack->cblk() != NULL) { 4092 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4093 mOutputTracks.add(outputTrack); 4094 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4095 updateWaitTime_l(); 4096 } 4097} 4098 4099void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4100{ 4101 Mutex::Autolock _l(mLock); 4102 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4103 if (mOutputTracks[i]->thread() == thread) { 4104 mOutputTracks[i]->destroy(); 4105 mOutputTracks.removeAt(i); 4106 updateWaitTime_l(); 4107 return; 4108 } 4109 } 4110 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4111} 4112 4113// caller must hold mLock 4114void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4115{ 4116 mWaitTimeMs = UINT_MAX; 4117 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4118 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4119 if (strong != 0) { 4120 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4121 if (waitTimeMs < mWaitTimeMs) { 4122 mWaitTimeMs = waitTimeMs; 4123 } 4124 } 4125 } 4126} 4127 4128 4129bool AudioFlinger::DuplicatingThread::outputsReady( 4130 const SortedVector< sp<OutputTrack> > &outputTracks) 4131{ 4132 for (size_t i = 0; i < outputTracks.size(); i++) { 4133 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4134 if (thread == 0) { 4135 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4136 outputTracks[i].get()); 4137 return false; 4138 } 4139 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4140 // see note at standby() declaration 4141 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4142 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4143 thread.get()); 4144 return false; 4145 } 4146 } 4147 return true; 4148} 4149 4150uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4151{ 4152 return (mWaitTimeMs * 1000) / 2; 4153} 4154 4155void AudioFlinger::DuplicatingThread::cacheParameters_l() 4156{ 4157 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4158 updateWaitTime_l(); 4159 4160 MixerThread::cacheParameters_l(); 4161} 4162 4163// ---------------------------------------------------------------------------- 4164// Record 4165// ---------------------------------------------------------------------------- 4166 4167AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4168 AudioStreamIn *input, 4169 uint32_t sampleRate, 4170 audio_channel_mask_t channelMask, 4171 audio_io_handle_t id, 4172 audio_devices_t outDevice, 4173 audio_devices_t inDevice 4174#ifdef TEE_SINK 4175 , const sp<NBAIO_Sink>& teeSink 4176#endif 4177 ) : 4178 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4179 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4180 // mRsmpInIndex and mBufferSize set by readInputParameters() 4181 mReqChannelCount(popcount(channelMask)), 4182 mReqSampleRate(sampleRate) 4183 // mBytesRead is only meaningful while active, and so is cleared in start() 4184 // (but might be better to also clear here for dump?) 4185#ifdef TEE_SINK 4186 , mTeeSink(teeSink) 4187#endif 4188{ 4189 snprintf(mName, kNameLength, "AudioIn_%X", id); 4190 4191 readInputParameters(); 4192 4193} 4194 4195 4196AudioFlinger::RecordThread::~RecordThread() 4197{ 4198 delete[] mRsmpInBuffer; 4199 delete mResampler; 4200 delete[] mRsmpOutBuffer; 4201} 4202 4203void AudioFlinger::RecordThread::onFirstRef() 4204{ 4205 run(mName, PRIORITY_URGENT_AUDIO); 4206} 4207 4208status_t AudioFlinger::RecordThread::readyToRun() 4209{ 4210 status_t status = initCheck(); 4211 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4212 return status; 4213} 4214 4215bool AudioFlinger::RecordThread::threadLoop() 4216{ 4217 AudioBufferProvider::Buffer buffer; 4218 sp<RecordTrack> activeTrack; 4219 Vector< sp<EffectChain> > effectChains; 4220 4221 nsecs_t lastWarning = 0; 4222 4223 inputStandBy(); 4224 acquireWakeLock(); 4225 4226 // used to verify we've read at least once before evaluating how many bytes were read 4227 bool readOnce = false; 4228 4229 // start recording 4230 while (!exitPending()) { 4231 4232 processConfigEvents(); 4233 4234 { // scope for mLock 4235 Mutex::Autolock _l(mLock); 4236 checkForNewParameters_l(); 4237 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4238 standby(); 4239 4240 if (exitPending()) { 4241 break; 4242 } 4243 4244 releaseWakeLock_l(); 4245 ALOGV("RecordThread: loop stopping"); 4246 // go to sleep 4247 mWaitWorkCV.wait(mLock); 4248 ALOGV("RecordThread: loop starting"); 4249 acquireWakeLock_l(); 4250 continue; 4251 } 4252 if (mActiveTrack != 0) { 4253 if (mActiveTrack->isTerminated()) { 4254 removeTrack_l(mActiveTrack); 4255 mActiveTrack.clear(); 4256 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4257 standby(); 4258 mActiveTrack.clear(); 4259 mStartStopCond.broadcast(); 4260 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4261 if (mReqChannelCount != mActiveTrack->channelCount()) { 4262 mActiveTrack.clear(); 4263 mStartStopCond.broadcast(); 4264 } else if (readOnce) { 4265 // record start succeeds only if first read from audio input 4266 // succeeds 4267 if (mBytesRead >= 0) { 4268 mActiveTrack->mState = TrackBase::ACTIVE; 4269 } else { 4270 mActiveTrack.clear(); 4271 } 4272 mStartStopCond.broadcast(); 4273 } 4274 mStandby = false; 4275 } 4276 } 4277 lockEffectChains_l(effectChains); 4278 } 4279 4280 if (mActiveTrack != 0) { 4281 if (mActiveTrack->mState != TrackBase::ACTIVE && 4282 mActiveTrack->mState != TrackBase::RESUMING) { 4283 unlockEffectChains(effectChains); 4284 usleep(kRecordThreadSleepUs); 4285 continue; 4286 } 4287 for (size_t i = 0; i < effectChains.size(); i ++) { 4288 effectChains[i]->process_l(); 4289 } 4290 4291 buffer.frameCount = mFrameCount; 4292 status_t status = mActiveTrack->getNextBuffer(&buffer); 4293 if (status == NO_ERROR) { 4294 readOnce = true; 4295 size_t framesOut = buffer.frameCount; 4296 if (mResampler == NULL) { 4297 // no resampling 4298 while (framesOut) { 4299 size_t framesIn = mFrameCount - mRsmpInIndex; 4300 if (framesIn) { 4301 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4302 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4303 mActiveTrack->mFrameSize; 4304 if (framesIn > framesOut) 4305 framesIn = framesOut; 4306 mRsmpInIndex += framesIn; 4307 framesOut -= framesIn; 4308 if (mChannelCount == mReqChannelCount) { 4309 memcpy(dst, src, framesIn * mFrameSize); 4310 } else { 4311 if (mChannelCount == 1) { 4312 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4313 (int16_t *)src, framesIn); 4314 } else { 4315 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4316 (int16_t *)src, framesIn); 4317 } 4318 } 4319 } 4320 if (framesOut && mFrameCount == mRsmpInIndex) { 4321 void *readInto; 4322 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4323 readInto = buffer.raw; 4324 framesOut = 0; 4325 } else { 4326 readInto = mRsmpInBuffer; 4327 mRsmpInIndex = 0; 4328 } 4329 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4330 mBufferSize); 4331 if (mBytesRead <= 0) { 4332 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4333 { 4334 ALOGE("Error reading audio input"); 4335 // Force input into standby so that it tries to 4336 // recover at next read attempt 4337 inputStandBy(); 4338 usleep(kRecordThreadSleepUs); 4339 } 4340 mRsmpInIndex = mFrameCount; 4341 framesOut = 0; 4342 buffer.frameCount = 0; 4343 } 4344#ifdef TEE_SINK 4345 else if (mTeeSink != 0) { 4346 (void) mTeeSink->write(readInto, 4347 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4348 } 4349#endif 4350 } 4351 } 4352 } else { 4353 // resampling 4354 4355 // resampler accumulates, but we only have one source track 4356 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4357 // alter output frame count as if we were expecting stereo samples 4358 if (mChannelCount == 1 && mReqChannelCount == 1) { 4359 framesOut >>= 1; 4360 } 4361 mResampler->resample(mRsmpOutBuffer, framesOut, 4362 this /* AudioBufferProvider* */); 4363 // ditherAndClamp() works as long as all buffers returned by 4364 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4365 if (mChannelCount == 2 && mReqChannelCount == 1) { 4366 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4367 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4368 // the resampler always outputs stereo samples: 4369 // do post stereo to mono conversion 4370 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4371 framesOut); 4372 } else { 4373 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4374 } 4375 // now done with mRsmpOutBuffer 4376 4377 } 4378 if (mFramestoDrop == 0) { 4379 mActiveTrack->releaseBuffer(&buffer); 4380 } else { 4381 if (mFramestoDrop > 0) { 4382 mFramestoDrop -= buffer.frameCount; 4383 if (mFramestoDrop <= 0) { 4384 clearSyncStartEvent(); 4385 } 4386 } else { 4387 mFramestoDrop += buffer.frameCount; 4388 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4389 mSyncStartEvent->isCancelled()) { 4390 ALOGW("Synced record %s, session %d, trigger session %d", 4391 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4392 mActiveTrack->sessionId(), 4393 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4394 clearSyncStartEvent(); 4395 } 4396 } 4397 } 4398 mActiveTrack->clearOverflow(); 4399 } 4400 // client isn't retrieving buffers fast enough 4401 else { 4402 if (!mActiveTrack->setOverflow()) { 4403 nsecs_t now = systemTime(); 4404 if ((now - lastWarning) > kWarningThrottleNs) { 4405 ALOGW("RecordThread: buffer overflow"); 4406 lastWarning = now; 4407 } 4408 } 4409 // Release the processor for a while before asking for a new buffer. 4410 // This will give the application more chance to read from the buffer and 4411 // clear the overflow. 4412 usleep(kRecordThreadSleepUs); 4413 } 4414 } 4415 // enable changes in effect chain 4416 unlockEffectChains(effectChains); 4417 effectChains.clear(); 4418 } 4419 4420 standby(); 4421 4422 { 4423 Mutex::Autolock _l(mLock); 4424 mActiveTrack.clear(); 4425 mStartStopCond.broadcast(); 4426 } 4427 4428 releaseWakeLock(); 4429 4430 ALOGV("RecordThread %p exiting", this); 4431 return false; 4432} 4433 4434void AudioFlinger::RecordThread::standby() 4435{ 4436 if (!mStandby) { 4437 inputStandBy(); 4438 mStandby = true; 4439 } 4440} 4441 4442void AudioFlinger::RecordThread::inputStandBy() 4443{ 4444 mInput->stream->common.standby(&mInput->stream->common); 4445} 4446 4447sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4448 const sp<AudioFlinger::Client>& client, 4449 uint32_t sampleRate, 4450 audio_format_t format, 4451 audio_channel_mask_t channelMask, 4452 size_t frameCount, 4453 int sessionId, 4454 IAudioFlinger::track_flags_t *flags, 4455 pid_t tid, 4456 status_t *status) 4457{ 4458 sp<RecordTrack> track; 4459 status_t lStatus; 4460 4461 lStatus = initCheck(); 4462 if (lStatus != NO_ERROR) { 4463 ALOGE("Audio driver not initialized."); 4464 goto Exit; 4465 } 4466 4467 // client expresses a preference for FAST, but we get the final say 4468 if (*flags & IAudioFlinger::TRACK_FAST) { 4469 if ( 4470 // use case: callback handler and frame count is default or at least as large as HAL 4471 ( 4472 (tid != -1) && 4473 ((frameCount == 0) || 4474 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4475 ) && 4476 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4477 // mono or stereo 4478 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4479 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4480 // hardware sample rate 4481 (sampleRate == mSampleRate) && 4482 // record thread has an associated fast recorder 4483 hasFastRecorder() 4484 // FIXME test that RecordThread for this fast track has a capable output HAL 4485 // FIXME add a permission test also? 4486 ) { 4487 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4488 if (frameCount == 0) { 4489 frameCount = mFrameCount * kFastTrackMultiplier; 4490 } 4491 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4492 frameCount, mFrameCount); 4493 } else { 4494 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4495 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4496 "hasFastRecorder=%d tid=%d", 4497 frameCount, mFrameCount, format, 4498 audio_is_linear_pcm(format), 4499 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4500 *flags &= ~IAudioFlinger::TRACK_FAST; 4501 // For compatibility with AudioRecord calculation, buffer depth is forced 4502 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4503 // This is probably too conservative, but legacy application code may depend on it. 4504 // If you change this calculation, also review the start threshold which is related. 4505 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4506 size_t mNormalFrameCount = 2048; // FIXME 4507 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4508 if (minBufCount < 2) { 4509 minBufCount = 2; 4510 } 4511 size_t minFrameCount = mNormalFrameCount * minBufCount; 4512 if (frameCount < minFrameCount) { 4513 frameCount = minFrameCount; 4514 } 4515 } 4516 } 4517 4518 // FIXME use flags and tid similar to createTrack_l() 4519 4520 { // scope for mLock 4521 Mutex::Autolock _l(mLock); 4522 4523 track = new RecordTrack(this, client, sampleRate, 4524 format, channelMask, frameCount, sessionId); 4525 4526 if (track->getCblk() == 0) { 4527 lStatus = NO_MEMORY; 4528 goto Exit; 4529 } 4530 mTracks.add(track); 4531 4532 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4533 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4534 mAudioFlinger->btNrecIsOff(); 4535 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4536 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4537 4538 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4539 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4540 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4541 // so ask activity manager to do this on our behalf 4542 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4543 } 4544 } 4545 lStatus = NO_ERROR; 4546 4547Exit: 4548 if (status) { 4549 *status = lStatus; 4550 } 4551 return track; 4552} 4553 4554status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4555 AudioSystem::sync_event_t event, 4556 int triggerSession) 4557{ 4558 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4559 sp<ThreadBase> strongMe = this; 4560 status_t status = NO_ERROR; 4561 4562 if (event == AudioSystem::SYNC_EVENT_NONE) { 4563 clearSyncStartEvent(); 4564 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4565 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4566 triggerSession, 4567 recordTrack->sessionId(), 4568 syncStartEventCallback, 4569 this); 4570 // Sync event can be cancelled by the trigger session if the track is not in a 4571 // compatible state in which case we start record immediately 4572 if (mSyncStartEvent->isCancelled()) { 4573 clearSyncStartEvent(); 4574 } else { 4575 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4576 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4577 } 4578 } 4579 4580 { 4581 AutoMutex lock(mLock); 4582 if (mActiveTrack != 0) { 4583 if (recordTrack != mActiveTrack.get()) { 4584 status = -EBUSY; 4585 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4586 mActiveTrack->mState = TrackBase::ACTIVE; 4587 } 4588 return status; 4589 } 4590 4591 recordTrack->mState = TrackBase::IDLE; 4592 mActiveTrack = recordTrack; 4593 mLock.unlock(); 4594 status_t status = AudioSystem::startInput(mId); 4595 mLock.lock(); 4596 if (status != NO_ERROR) { 4597 mActiveTrack.clear(); 4598 clearSyncStartEvent(); 4599 return status; 4600 } 4601 mRsmpInIndex = mFrameCount; 4602 mBytesRead = 0; 4603 if (mResampler != NULL) { 4604 mResampler->reset(); 4605 } 4606 mActiveTrack->mState = TrackBase::RESUMING; 4607 // signal thread to start 4608 ALOGV("Signal record thread"); 4609 mWaitWorkCV.broadcast(); 4610 // do not wait for mStartStopCond if exiting 4611 if (exitPending()) { 4612 mActiveTrack.clear(); 4613 status = INVALID_OPERATION; 4614 goto startError; 4615 } 4616 mStartStopCond.wait(mLock); 4617 if (mActiveTrack == 0) { 4618 ALOGV("Record failed to start"); 4619 status = BAD_VALUE; 4620 goto startError; 4621 } 4622 ALOGV("Record started OK"); 4623 return status; 4624 } 4625 4626startError: 4627 AudioSystem::stopInput(mId); 4628 clearSyncStartEvent(); 4629 return status; 4630} 4631 4632void AudioFlinger::RecordThread::clearSyncStartEvent() 4633{ 4634 if (mSyncStartEvent != 0) { 4635 mSyncStartEvent->cancel(); 4636 } 4637 mSyncStartEvent.clear(); 4638 mFramestoDrop = 0; 4639} 4640 4641void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4642{ 4643 sp<SyncEvent> strongEvent = event.promote(); 4644 4645 if (strongEvent != 0) { 4646 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4647 me->handleSyncStartEvent(strongEvent); 4648 } 4649} 4650 4651void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4652{ 4653 if (event == mSyncStartEvent) { 4654 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4655 // from audio HAL 4656 mFramestoDrop = mFrameCount * 2; 4657 } 4658} 4659 4660bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4661 ALOGV("RecordThread::stop"); 4662 AutoMutex _l(mLock); 4663 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4664 return false; 4665 } 4666 recordTrack->mState = TrackBase::PAUSING; 4667 // do not wait for mStartStopCond if exiting 4668 if (exitPending()) { 4669 return true; 4670 } 4671 mStartStopCond.wait(mLock); 4672 // if we have been restarted, recordTrack == mActiveTrack.get() here 4673 if (exitPending() || recordTrack != mActiveTrack.get()) { 4674 ALOGV("Record stopped OK"); 4675 return true; 4676 } 4677 return false; 4678} 4679 4680bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4681{ 4682 return false; 4683} 4684 4685status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4686{ 4687#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4688 if (!isValidSyncEvent(event)) { 4689 return BAD_VALUE; 4690 } 4691 4692 int eventSession = event->triggerSession(); 4693 status_t ret = NAME_NOT_FOUND; 4694 4695 Mutex::Autolock _l(mLock); 4696 4697 for (size_t i = 0; i < mTracks.size(); i++) { 4698 sp<RecordTrack> track = mTracks[i]; 4699 if (eventSession == track->sessionId()) { 4700 (void) track->setSyncEvent(event); 4701 ret = NO_ERROR; 4702 } 4703 } 4704 return ret; 4705#else 4706 return BAD_VALUE; 4707#endif 4708} 4709 4710// destroyTrack_l() must be called with ThreadBase::mLock held 4711void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4712{ 4713 track->terminate(); 4714 track->mState = TrackBase::STOPPED; 4715 // active tracks are removed by threadLoop() 4716 if (mActiveTrack != track) { 4717 removeTrack_l(track); 4718 } 4719} 4720 4721void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4722{ 4723 mTracks.remove(track); 4724 // need anything related to effects here? 4725} 4726 4727void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4728{ 4729 dumpInternals(fd, args); 4730 dumpTracks(fd, args); 4731 dumpEffectChains(fd, args); 4732} 4733 4734void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4735{ 4736 const size_t SIZE = 256; 4737 char buffer[SIZE]; 4738 String8 result; 4739 4740 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4741 result.append(buffer); 4742 4743 if (mActiveTrack != 0) { 4744 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4745 result.append(buffer); 4746 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4747 result.append(buffer); 4748 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4749 result.append(buffer); 4750 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4751 result.append(buffer); 4752 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4753 result.append(buffer); 4754 } else { 4755 result.append("No active record client\n"); 4756 } 4757 4758 write(fd, result.string(), result.size()); 4759 4760 dumpBase(fd, args); 4761} 4762 4763void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4764{ 4765 const size_t SIZE = 256; 4766 char buffer[SIZE]; 4767 String8 result; 4768 4769 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4770 result.append(buffer); 4771 RecordTrack::appendDumpHeader(result); 4772 for (size_t i = 0; i < mTracks.size(); ++i) { 4773 sp<RecordTrack> track = mTracks[i]; 4774 if (track != 0) { 4775 track->dump(buffer, SIZE); 4776 result.append(buffer); 4777 } 4778 } 4779 4780 if (mActiveTrack != 0) { 4781 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4782 result.append(buffer); 4783 RecordTrack::appendDumpHeader(result); 4784 mActiveTrack->dump(buffer, SIZE); 4785 result.append(buffer); 4786 4787 } 4788 write(fd, result.string(), result.size()); 4789} 4790 4791// AudioBufferProvider interface 4792status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4793{ 4794 size_t framesReq = buffer->frameCount; 4795 size_t framesReady = mFrameCount - mRsmpInIndex; 4796 int channelCount; 4797 4798 if (framesReady == 0) { 4799 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4800 if (mBytesRead <= 0) { 4801 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4802 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4803 // Force input into standby so that it tries to 4804 // recover at next read attempt 4805 inputStandBy(); 4806 usleep(kRecordThreadSleepUs); 4807 } 4808 buffer->raw = NULL; 4809 buffer->frameCount = 0; 4810 return NOT_ENOUGH_DATA; 4811 } 4812 mRsmpInIndex = 0; 4813 framesReady = mFrameCount; 4814 } 4815 4816 if (framesReq > framesReady) { 4817 framesReq = framesReady; 4818 } 4819 4820 if (mChannelCount == 1 && mReqChannelCount == 2) { 4821 channelCount = 1; 4822 } else { 4823 channelCount = 2; 4824 } 4825 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4826 buffer->frameCount = framesReq; 4827 return NO_ERROR; 4828} 4829 4830// AudioBufferProvider interface 4831void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4832{ 4833 mRsmpInIndex += buffer->frameCount; 4834 buffer->frameCount = 0; 4835} 4836 4837bool AudioFlinger::RecordThread::checkForNewParameters_l() 4838{ 4839 bool reconfig = false; 4840 4841 while (!mNewParameters.isEmpty()) { 4842 status_t status = NO_ERROR; 4843 String8 keyValuePair = mNewParameters[0]; 4844 AudioParameter param = AudioParameter(keyValuePair); 4845 int value; 4846 audio_format_t reqFormat = mFormat; 4847 uint32_t reqSamplingRate = mReqSampleRate; 4848 uint32_t reqChannelCount = mReqChannelCount; 4849 4850 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4851 reqSamplingRate = value; 4852 reconfig = true; 4853 } 4854 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4855 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4856 status = BAD_VALUE; 4857 } else { 4858 reqFormat = (audio_format_t) value; 4859 reconfig = true; 4860 } 4861 } 4862 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4863 reqChannelCount = popcount(value); 4864 reconfig = true; 4865 } 4866 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4867 // do not accept frame count changes if tracks are open as the track buffer 4868 // size depends on frame count and correct behavior would not be guaranteed 4869 // if frame count is changed after track creation 4870 if (mActiveTrack != 0) { 4871 status = INVALID_OPERATION; 4872 } else { 4873 reconfig = true; 4874 } 4875 } 4876 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4877 // forward device change to effects that have requested to be 4878 // aware of attached audio device. 4879 for (size_t i = 0; i < mEffectChains.size(); i++) { 4880 mEffectChains[i]->setDevice_l(value); 4881 } 4882 4883 // store input device and output device but do not forward output device to audio HAL. 4884 // Note that status is ignored by the caller for output device 4885 // (see AudioFlinger::setParameters() 4886 if (audio_is_output_devices(value)) { 4887 mOutDevice = value; 4888 status = BAD_VALUE; 4889 } else { 4890 mInDevice = value; 4891 // disable AEC and NS if the device is a BT SCO headset supporting those 4892 // pre processings 4893 if (mTracks.size() > 0) { 4894 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4895 mAudioFlinger->btNrecIsOff(); 4896 for (size_t i = 0; i < mTracks.size(); i++) { 4897 sp<RecordTrack> track = mTracks[i]; 4898 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4899 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4900 } 4901 } 4902 } 4903 } 4904 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4905 mAudioSource != (audio_source_t)value) { 4906 // forward device change to effects that have requested to be 4907 // aware of attached audio device. 4908 for (size_t i = 0; i < mEffectChains.size(); i++) { 4909 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4910 } 4911 mAudioSource = (audio_source_t)value; 4912 } 4913 if (status == NO_ERROR) { 4914 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4915 keyValuePair.string()); 4916 if (status == INVALID_OPERATION) { 4917 inputStandBy(); 4918 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4919 keyValuePair.string()); 4920 } 4921 if (reconfig) { 4922 if (status == BAD_VALUE && 4923 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4924 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4925 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4926 <= (2 * reqSamplingRate)) && 4927 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4928 <= FCC_2 && 4929 (reqChannelCount <= FCC_2)) { 4930 status = NO_ERROR; 4931 } 4932 if (status == NO_ERROR) { 4933 readInputParameters(); 4934 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4935 } 4936 } 4937 } 4938 4939 mNewParameters.removeAt(0); 4940 4941 mParamStatus = status; 4942 mParamCond.signal(); 4943 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4944 // already timed out waiting for the status and will never signal the condition. 4945 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4946 } 4947 return reconfig; 4948} 4949 4950String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4951{ 4952 Mutex::Autolock _l(mLock); 4953 if (initCheck() != NO_ERROR) { 4954 return String8(); 4955 } 4956 4957 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4958 const String8 out_s8(s); 4959 free(s); 4960 return out_s8; 4961} 4962 4963void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4964 AudioSystem::OutputDescriptor desc; 4965 void *param2 = NULL; 4966 4967 switch (event) { 4968 case AudioSystem::INPUT_OPENED: 4969 case AudioSystem::INPUT_CONFIG_CHANGED: 4970 desc.channelMask = mChannelMask; 4971 desc.samplingRate = mSampleRate; 4972 desc.format = mFormat; 4973 desc.frameCount = mFrameCount; 4974 desc.latency = 0; 4975 param2 = &desc; 4976 break; 4977 4978 case AudioSystem::INPUT_CLOSED: 4979 default: 4980 break; 4981 } 4982 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4983} 4984 4985void AudioFlinger::RecordThread::readInputParameters() 4986{ 4987 delete[] mRsmpInBuffer; 4988 // mRsmpInBuffer is always assigned a new[] below 4989 delete[] mRsmpOutBuffer; 4990 mRsmpOutBuffer = NULL; 4991 delete mResampler; 4992 mResampler = NULL; 4993 4994 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4995 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4996 mChannelCount = popcount(mChannelMask); 4997 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4998 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4999 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5000 } 5001 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5002 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5003 mFrameCount = mBufferSize / mFrameSize; 5004 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5005 5006 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5007 { 5008 int channelCount; 5009 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5010 // stereo to mono post process as the resampler always outputs stereo. 5011 if (mChannelCount == 1 && mReqChannelCount == 2) { 5012 channelCount = 1; 5013 } else { 5014 channelCount = 2; 5015 } 5016 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5017 mResampler->setSampleRate(mSampleRate); 5018 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5019 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5020 5021 // optmization: if mono to mono, alter input frame count as if we were inputing 5022 // stereo samples 5023 if (mChannelCount == 1 && mReqChannelCount == 1) { 5024 mFrameCount >>= 1; 5025 } 5026 5027 } 5028 mRsmpInIndex = mFrameCount; 5029} 5030 5031unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5032{ 5033 Mutex::Autolock _l(mLock); 5034 if (initCheck() != NO_ERROR) { 5035 return 0; 5036 } 5037 5038 return mInput->stream->get_input_frames_lost(mInput->stream); 5039} 5040 5041uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5042{ 5043 Mutex::Autolock _l(mLock); 5044 uint32_t result = 0; 5045 if (getEffectChain_l(sessionId) != 0) { 5046 result = EFFECT_SESSION; 5047 } 5048 5049 for (size_t i = 0; i < mTracks.size(); ++i) { 5050 if (sessionId == mTracks[i]->sessionId()) { 5051 result |= TRACK_SESSION; 5052 break; 5053 } 5054 } 5055 5056 return result; 5057} 5058 5059KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5060{ 5061 KeyedVector<int, bool> ids; 5062 Mutex::Autolock _l(mLock); 5063 for (size_t j = 0; j < mTracks.size(); ++j) { 5064 sp<RecordThread::RecordTrack> track = mTracks[j]; 5065 int sessionId = track->sessionId(); 5066 if (ids.indexOfKey(sessionId) < 0) { 5067 ids.add(sessionId, true); 5068 } 5069 } 5070 return ids; 5071} 5072 5073AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5074{ 5075 Mutex::Autolock _l(mLock); 5076 AudioStreamIn *input = mInput; 5077 mInput = NULL; 5078 return input; 5079} 5080 5081// this method must always be called either with ThreadBase mLock held or inside the thread loop 5082audio_stream_t* AudioFlinger::RecordThread::stream() const 5083{ 5084 if (mInput == NULL) { 5085 return NULL; 5086 } 5087 return &mInput->stream->common; 5088} 5089 5090status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5091{ 5092 // only one chain per input thread 5093 if (mEffectChains.size() != 0) { 5094 return INVALID_OPERATION; 5095 } 5096 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5097 5098 chain->setInBuffer(NULL); 5099 chain->setOutBuffer(NULL); 5100 5101 checkSuspendOnAddEffectChain_l(chain); 5102 5103 mEffectChains.add(chain); 5104 5105 return NO_ERROR; 5106} 5107 5108size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5109{ 5110 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5111 ALOGW_IF(mEffectChains.size() != 1, 5112 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5113 chain.get(), mEffectChains.size(), this); 5114 if (mEffectChains.size() == 1) { 5115 mEffectChains.removeAt(0); 5116 } 5117 return 0; 5118} 5119 5120}; // namespace android 5121