Threads.cpp revision 1abbdb4429479975718421c4fef3f79ce7c820e3
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 mWriteAckSequence(0), 943 mDrainSequence(0), 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->resetWriteBlocked(); 1495} 1496 1497void AudioFlinger::PlaybackThread::drainCallback() 1498{ 1499 ALOG_ASSERT(mCallbackThread != 0); 1500 mCallbackThread->resetDraining(); 1501} 1502 1503void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1504{ 1505 Mutex::Autolock _l(mLock); 1506 // reject out of sequence requests 1507 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1508 mWriteAckSequence &= ~1; 1509 mWaitWorkCV.signal(); 1510 } 1511} 1512 1513void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1514{ 1515 Mutex::Autolock _l(mLock); 1516 // reject out of sequence requests 1517 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1518 mDrainSequence &= ~1; 1519 mWaitWorkCV.signal(); 1520 } 1521} 1522 1523// static 1524int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1525 void *param, 1526 void *cookie) 1527{ 1528 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1529 ALOGV("asyncCallback() event %d", event); 1530 switch (event) { 1531 case STREAM_CBK_EVENT_WRITE_READY: 1532 me->writeCallback(); 1533 break; 1534 case STREAM_CBK_EVENT_DRAIN_READY: 1535 me->drainCallback(); 1536 break; 1537 default: 1538 ALOGW("asyncCallback() unknown event %d", event); 1539 break; 1540 } 1541 return 0; 1542} 1543 1544void AudioFlinger::PlaybackThread::readOutputParameters() 1545{ 1546 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1547 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1548 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1549 if (!audio_is_output_channel(mChannelMask)) { 1550 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1551 } 1552 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1553 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1554 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1555 } 1556 mChannelCount = popcount(mChannelMask); 1557 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1558 if (!audio_is_valid_format(mFormat)) { 1559 LOG_FATAL("HAL format %d not valid for output", mFormat); 1560 } 1561 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1562 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1563 mFormat); 1564 } 1565 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1566 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1567 if (mFrameCount & 15) { 1568 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1569 mFrameCount); 1570 } 1571 1572 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1573 (mOutput->stream->set_callback != NULL)) { 1574 if (mOutput->stream->set_callback(mOutput->stream, 1575 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1576 mUseAsyncWrite = true; 1577 } 1578 } 1579 1580 // Calculate size of normal mix buffer relative to the HAL output buffer size 1581 double multiplier = 1.0; 1582 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1583 kUseFastMixer == FastMixer_Dynamic)) { 1584 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1585 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1586 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1587 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1588 maxNormalFrameCount = maxNormalFrameCount & ~15; 1589 if (maxNormalFrameCount < minNormalFrameCount) { 1590 maxNormalFrameCount = minNormalFrameCount; 1591 } 1592 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1593 if (multiplier <= 1.0) { 1594 multiplier = 1.0; 1595 } else if (multiplier <= 2.0) { 1596 if (2 * mFrameCount <= maxNormalFrameCount) { 1597 multiplier = 2.0; 1598 } else { 1599 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1600 } 1601 } else { 1602 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1603 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1604 // track, but we sometimes have to do this to satisfy the maximum frame count 1605 // constraint) 1606 // FIXME this rounding up should not be done if no HAL SRC 1607 uint32_t truncMult = (uint32_t) multiplier; 1608 if ((truncMult & 1)) { 1609 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1610 ++truncMult; 1611 } 1612 } 1613 multiplier = (double) truncMult; 1614 } 1615 } 1616 mNormalFrameCount = multiplier * mFrameCount; 1617 // round up to nearest 16 frames to satisfy AudioMixer 1618 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1619 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1620 mNormalFrameCount); 1621 1622 delete[] mAllocMixBuffer; 1623 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1624 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1625 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1626 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1627 1628 // force reconfiguration of effect chains and engines to take new buffer size and audio 1629 // parameters into account 1630 // Note that mLock is not held when readOutputParameters() is called from the constructor 1631 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1632 // matter. 1633 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1634 Vector< sp<EffectChain> > effectChains = mEffectChains; 1635 for (size_t i = 0; i < effectChains.size(); i ++) { 1636 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1637 } 1638} 1639 1640 1641status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1642{ 1643 if (halFrames == NULL || dspFrames == NULL) { 1644 return BAD_VALUE; 1645 } 1646 Mutex::Autolock _l(mLock); 1647 if (initCheck() != NO_ERROR) { 1648 return INVALID_OPERATION; 1649 } 1650 size_t framesWritten = mBytesWritten / mFrameSize; 1651 *halFrames = framesWritten; 1652 1653 if (isSuspended()) { 1654 // return an estimation of rendered frames when the output is suspended 1655 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1656 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1657 return NO_ERROR; 1658 } else { 1659 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1660 } 1661} 1662 1663uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1664{ 1665 Mutex::Autolock _l(mLock); 1666 uint32_t result = 0; 1667 if (getEffectChain_l(sessionId) != 0) { 1668 result = EFFECT_SESSION; 1669 } 1670 1671 for (size_t i = 0; i < mTracks.size(); ++i) { 1672 sp<Track> track = mTracks[i]; 1673 if (sessionId == track->sessionId() && !track->isInvalid()) { 1674 result |= TRACK_SESSION; 1675 break; 1676 } 1677 } 1678 1679 return result; 1680} 1681 1682uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1683{ 1684 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1685 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1686 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1687 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1688 } 1689 for (size_t i = 0; i < mTracks.size(); i++) { 1690 sp<Track> track = mTracks[i]; 1691 if (sessionId == track->sessionId() && !track->isInvalid()) { 1692 return AudioSystem::getStrategyForStream(track->streamType()); 1693 } 1694 } 1695 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1696} 1697 1698 1699AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1700{ 1701 Mutex::Autolock _l(mLock); 1702 return mOutput; 1703} 1704 1705AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1706{ 1707 Mutex::Autolock _l(mLock); 1708 AudioStreamOut *output = mOutput; 1709 mOutput = NULL; 1710 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1711 // must push a NULL and wait for ack 1712 mOutputSink.clear(); 1713 mPipeSink.clear(); 1714 mNormalSink.clear(); 1715 return output; 1716} 1717 1718// this method must always be called either with ThreadBase mLock held or inside the thread loop 1719audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1720{ 1721 if (mOutput == NULL) { 1722 return NULL; 1723 } 1724 return &mOutput->stream->common; 1725} 1726 1727uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1728{ 1729 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1730} 1731 1732status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1733{ 1734 if (!isValidSyncEvent(event)) { 1735 return BAD_VALUE; 1736 } 1737 1738 Mutex::Autolock _l(mLock); 1739 1740 for (size_t i = 0; i < mTracks.size(); ++i) { 1741 sp<Track> track = mTracks[i]; 1742 if (event->triggerSession() == track->sessionId()) { 1743 (void) track->setSyncEvent(event); 1744 return NO_ERROR; 1745 } 1746 } 1747 1748 return NAME_NOT_FOUND; 1749} 1750 1751bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1752{ 1753 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1754} 1755 1756void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1757 const Vector< sp<Track> >& tracksToRemove) 1758{ 1759 size_t count = tracksToRemove.size(); 1760 if (count) { 1761 for (size_t i = 0 ; i < count ; i++) { 1762 const sp<Track>& track = tracksToRemove.itemAt(i); 1763 if (!track->isOutputTrack()) { 1764 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1765#ifdef ADD_BATTERY_DATA 1766 // to track the speaker usage 1767 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1768#endif 1769 if (track->isTerminated()) { 1770 AudioSystem::releaseOutput(mId); 1771 } 1772 } 1773 } 1774 } 1775} 1776 1777void AudioFlinger::PlaybackThread::checkSilentMode_l() 1778{ 1779 if (!mMasterMute) { 1780 char value[PROPERTY_VALUE_MAX]; 1781 if (property_get("ro.audio.silent", value, "0") > 0) { 1782 char *endptr; 1783 unsigned long ul = strtoul(value, &endptr, 0); 1784 if (*endptr == '\0' && ul != 0) { 1785 ALOGD("Silence is golden"); 1786 // The setprop command will not allow a property to be changed after 1787 // the first time it is set, so we don't have to worry about un-muting. 1788 setMasterMute_l(true); 1789 } 1790 } 1791 } 1792} 1793 1794// shared by MIXER and DIRECT, overridden by DUPLICATING 1795ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1796{ 1797 // FIXME rewrite to reduce number of system calls 1798 mLastWriteTime = systemTime(); 1799 mInWrite = true; 1800 ssize_t bytesWritten; 1801 1802 // If an NBAIO sink is present, use it to write the normal mixer's submix 1803 if (mNormalSink != 0) { 1804#define mBitShift 2 // FIXME 1805 size_t count = mBytesRemaining >> mBitShift; 1806 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1807 ATRACE_BEGIN("write"); 1808 // update the setpoint when AudioFlinger::mScreenState changes 1809 uint32_t screenState = AudioFlinger::mScreenState; 1810 if (screenState != mScreenState) { 1811 mScreenState = screenState; 1812 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1813 if (pipe != NULL) { 1814 pipe->setAvgFrames((mScreenState & 1) ? 1815 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1816 } 1817 } 1818 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1819 ATRACE_END(); 1820 if (framesWritten > 0) { 1821 bytesWritten = framesWritten << mBitShift; 1822 } else { 1823 bytesWritten = framesWritten; 1824 } 1825 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1826 if (status == NO_ERROR) { 1827 size_t totalFramesWritten = mNormalSink->framesWritten(); 1828 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1829 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1830 mLatchDValid = true; 1831 } 1832 } 1833 // otherwise use the HAL / AudioStreamOut directly 1834 } else { 1835 // Direct output and offload threads 1836 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1837 if (mUseAsyncWrite) { 1838 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1839 mWriteAckSequence += 2; 1840 mWriteAckSequence |= 1; 1841 ALOG_ASSERT(mCallbackThread != 0); 1842 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1843 } 1844 // FIXME We should have an implementation of timestamps for direct output threads. 1845 // They are used e.g for multichannel PCM playback over HDMI. 1846 bytesWritten = mOutput->stream->write(mOutput->stream, 1847 mMixBuffer + offset, mBytesRemaining); 1848 if (mUseAsyncWrite && 1849 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1850 // do not wait for async callback in case of error of full write 1851 mWriteAckSequence &= ~1; 1852 ALOG_ASSERT(mCallbackThread != 0); 1853 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1854 } 1855 } 1856 1857 mNumWrites++; 1858 mInWrite = false; 1859 1860 return bytesWritten; 1861} 1862 1863void AudioFlinger::PlaybackThread::threadLoop_drain() 1864{ 1865 if (mOutput->stream->drain) { 1866 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1867 if (mUseAsyncWrite) { 1868 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1869 mDrainSequence |= 1; 1870 ALOG_ASSERT(mCallbackThread != 0); 1871 mCallbackThread->setDraining(mDrainSequence); 1872 } 1873 mOutput->stream->drain(mOutput->stream, 1874 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1875 : AUDIO_DRAIN_ALL); 1876 } 1877} 1878 1879void AudioFlinger::PlaybackThread::threadLoop_exit() 1880{ 1881 // Default implementation has nothing to do 1882} 1883 1884/* 1885The derived values that are cached: 1886 - mixBufferSize from frame count * frame size 1887 - activeSleepTime from activeSleepTimeUs() 1888 - idleSleepTime from idleSleepTimeUs() 1889 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1890 - maxPeriod from frame count and sample rate (MIXER only) 1891 1892The parameters that affect these derived values are: 1893 - frame count 1894 - frame size 1895 - sample rate 1896 - device type: A2DP or not 1897 - device latency 1898 - format: PCM or not 1899 - active sleep time 1900 - idle sleep time 1901*/ 1902 1903void AudioFlinger::PlaybackThread::cacheParameters_l() 1904{ 1905 mixBufferSize = mNormalFrameCount * mFrameSize; 1906 activeSleepTime = activeSleepTimeUs(); 1907 idleSleepTime = idleSleepTimeUs(); 1908} 1909 1910void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1911{ 1912 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1913 this, streamType, mTracks.size()); 1914 Mutex::Autolock _l(mLock); 1915 1916 size_t size = mTracks.size(); 1917 for (size_t i = 0; i < size; i++) { 1918 sp<Track> t = mTracks[i]; 1919 if (t->streamType() == streamType) { 1920 t->invalidate(); 1921 } 1922 } 1923} 1924 1925status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1926{ 1927 int session = chain->sessionId(); 1928 int16_t *buffer = mMixBuffer; 1929 bool ownsBuffer = false; 1930 1931 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1932 if (session > 0) { 1933 // Only one effect chain can be present in direct output thread and it uses 1934 // the mix buffer as input 1935 if (mType != DIRECT) { 1936 size_t numSamples = mNormalFrameCount * mChannelCount; 1937 buffer = new int16_t[numSamples]; 1938 memset(buffer, 0, numSamples * sizeof(int16_t)); 1939 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1940 ownsBuffer = true; 1941 } 1942 1943 // Attach all tracks with same session ID to this chain. 1944 for (size_t i = 0; i < mTracks.size(); ++i) { 1945 sp<Track> track = mTracks[i]; 1946 if (session == track->sessionId()) { 1947 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1948 buffer); 1949 track->setMainBuffer(buffer); 1950 chain->incTrackCnt(); 1951 } 1952 } 1953 1954 // indicate all active tracks in the chain 1955 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1956 sp<Track> track = mActiveTracks[i].promote(); 1957 if (track == 0) { 1958 continue; 1959 } 1960 if (session == track->sessionId()) { 1961 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1962 chain->incActiveTrackCnt(); 1963 } 1964 } 1965 } 1966 1967 chain->setInBuffer(buffer, ownsBuffer); 1968 chain->setOutBuffer(mMixBuffer); 1969 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1970 // chains list in order to be processed last as it contains output stage effects 1971 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1972 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1973 // after track specific effects and before output stage 1974 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1975 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1976 // Effect chain for other sessions are inserted at beginning of effect 1977 // chains list to be processed before output mix effects. Relative order between other 1978 // sessions is not important 1979 size_t size = mEffectChains.size(); 1980 size_t i = 0; 1981 for (i = 0; i < size; i++) { 1982 if (mEffectChains[i]->sessionId() < session) { 1983 break; 1984 } 1985 } 1986 mEffectChains.insertAt(chain, i); 1987 checkSuspendOnAddEffectChain_l(chain); 1988 1989 return NO_ERROR; 1990} 1991 1992size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1993{ 1994 int session = chain->sessionId(); 1995 1996 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1997 1998 for (size_t i = 0; i < mEffectChains.size(); i++) { 1999 if (chain == mEffectChains[i]) { 2000 mEffectChains.removeAt(i); 2001 // detach all active tracks from the chain 2002 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2003 sp<Track> track = mActiveTracks[i].promote(); 2004 if (track == 0) { 2005 continue; 2006 } 2007 if (session == track->sessionId()) { 2008 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2009 chain.get(), session); 2010 chain->decActiveTrackCnt(); 2011 } 2012 } 2013 2014 // detach all tracks with same session ID from this chain 2015 for (size_t i = 0; i < mTracks.size(); ++i) { 2016 sp<Track> track = mTracks[i]; 2017 if (session == track->sessionId()) { 2018 track->setMainBuffer(mMixBuffer); 2019 chain->decTrackCnt(); 2020 } 2021 } 2022 break; 2023 } 2024 } 2025 return mEffectChains.size(); 2026} 2027 2028status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2029 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2030{ 2031 Mutex::Autolock _l(mLock); 2032 return attachAuxEffect_l(track, EffectId); 2033} 2034 2035status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2036 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2037{ 2038 status_t status = NO_ERROR; 2039 2040 if (EffectId == 0) { 2041 track->setAuxBuffer(0, NULL); 2042 } else { 2043 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2044 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2045 if (effect != 0) { 2046 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2047 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2048 } else { 2049 status = INVALID_OPERATION; 2050 } 2051 } else { 2052 status = BAD_VALUE; 2053 } 2054 } 2055 return status; 2056} 2057 2058void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2059{ 2060 for (size_t i = 0; i < mTracks.size(); ++i) { 2061 sp<Track> track = mTracks[i]; 2062 if (track->auxEffectId() == effectId) { 2063 attachAuxEffect_l(track, 0); 2064 } 2065 } 2066} 2067 2068bool AudioFlinger::PlaybackThread::threadLoop() 2069{ 2070 Vector< sp<Track> > tracksToRemove; 2071 2072 standbyTime = systemTime(); 2073 2074 // MIXER 2075 nsecs_t lastWarning = 0; 2076 2077 // DUPLICATING 2078 // FIXME could this be made local to while loop? 2079 writeFrames = 0; 2080 2081 cacheParameters_l(); 2082 sleepTime = idleSleepTime; 2083 2084 if (mType == MIXER) { 2085 sleepTimeShift = 0; 2086 } 2087 2088 CpuStats cpuStats; 2089 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2090 2091 acquireWakeLock(); 2092 2093 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2094 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2095 // and then that string will be logged at the next convenient opportunity. 2096 const char *logString = NULL; 2097 2098 while (!exitPending()) 2099 { 2100 cpuStats.sample(myName); 2101 2102 Vector< sp<EffectChain> > effectChains; 2103 2104 processConfigEvents(); 2105 2106 { // scope for mLock 2107 2108 Mutex::Autolock _l(mLock); 2109 2110 if (logString != NULL) { 2111 mNBLogWriter->logTimestamp(); 2112 mNBLogWriter->log(logString); 2113 logString = NULL; 2114 } 2115 2116 if (mLatchDValid) { 2117 mLatchQ = mLatchD; 2118 mLatchDValid = false; 2119 mLatchQValid = true; 2120 } 2121 2122 if (checkForNewParameters_l()) { 2123 cacheParameters_l(); 2124 } 2125 2126 saveOutputTracks(); 2127 2128 if (mSignalPending) { 2129 // A signal was raised while we were unlocked 2130 mSignalPending = false; 2131 } else if (waitingAsyncCallback_l()) { 2132 if (exitPending()) { 2133 break; 2134 } 2135 releaseWakeLock_l(); 2136 ALOGV("wait async completion"); 2137 mWaitWorkCV.wait(mLock); 2138 ALOGV("async completion/wake"); 2139 acquireWakeLock_l(); 2140 if (exitPending()) { 2141 break; 2142 } 2143 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2144 continue; 2145 } 2146 sleepTime = 0; 2147 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2148 isSuspended()) { 2149 // put audio hardware into standby after short delay 2150 if (shouldStandby_l()) { 2151 2152 threadLoop_standby(); 2153 2154 mStandby = true; 2155 } 2156 2157 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2158 // we're about to wait, flush the binder command buffer 2159 IPCThreadState::self()->flushCommands(); 2160 2161 clearOutputTracks(); 2162 2163 if (exitPending()) { 2164 break; 2165 } 2166 2167 releaseWakeLock_l(); 2168 // wait until we have something to do... 2169 ALOGV("%s going to sleep", myName.string()); 2170 mWaitWorkCV.wait(mLock); 2171 ALOGV("%s waking up", myName.string()); 2172 acquireWakeLock_l(); 2173 2174 mMixerStatus = MIXER_IDLE; 2175 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2176 mBytesWritten = 0; 2177 mBytesRemaining = 0; 2178 checkSilentMode_l(); 2179 2180 standbyTime = systemTime() + standbyDelay; 2181 sleepTime = idleSleepTime; 2182 if (mType == MIXER) { 2183 sleepTimeShift = 0; 2184 } 2185 2186 continue; 2187 } 2188 } 2189 2190 // mMixerStatusIgnoringFastTracks is also updated internally 2191 mMixerStatus = prepareTracks_l(&tracksToRemove); 2192 2193 // prevent any changes in effect chain list and in each effect chain 2194 // during mixing and effect process as the audio buffers could be deleted 2195 // or modified if an effect is created or deleted 2196 lockEffectChains_l(effectChains); 2197 } 2198 2199 if (mBytesRemaining == 0) { 2200 mCurrentWriteLength = 0; 2201 if (mMixerStatus == MIXER_TRACKS_READY) { 2202 // threadLoop_mix() sets mCurrentWriteLength 2203 threadLoop_mix(); 2204 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2205 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2206 // threadLoop_sleepTime sets sleepTime to 0 if data 2207 // must be written to HAL 2208 threadLoop_sleepTime(); 2209 if (sleepTime == 0) { 2210 mCurrentWriteLength = mixBufferSize; 2211 } 2212 } 2213 mBytesRemaining = mCurrentWriteLength; 2214 if (isSuspended()) { 2215 sleepTime = suspendSleepTimeUs(); 2216 // simulate write to HAL when suspended 2217 mBytesWritten += mixBufferSize; 2218 mBytesRemaining = 0; 2219 } 2220 2221 // only process effects if we're going to write 2222 if (sleepTime == 0) { 2223 for (size_t i = 0; i < effectChains.size(); i ++) { 2224 effectChains[i]->process_l(); 2225 } 2226 } 2227 } 2228 2229 // enable changes in effect chain 2230 unlockEffectChains(effectChains); 2231 2232 if (!waitingAsyncCallback()) { 2233 // sleepTime == 0 means we must write to audio hardware 2234 if (sleepTime == 0) { 2235 if (mBytesRemaining) { 2236 ssize_t ret = threadLoop_write(); 2237 if (ret < 0) { 2238 mBytesRemaining = 0; 2239 } else { 2240 mBytesWritten += ret; 2241 mBytesRemaining -= ret; 2242 } 2243 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2244 (mMixerStatus == MIXER_DRAIN_ALL)) { 2245 threadLoop_drain(); 2246 } 2247if (mType == MIXER) { 2248 // write blocked detection 2249 nsecs_t now = systemTime(); 2250 nsecs_t delta = now - mLastWriteTime; 2251 if (!mStandby && delta > maxPeriod) { 2252 mNumDelayedWrites++; 2253 if ((now - lastWarning) > kWarningThrottleNs) { 2254 ATRACE_NAME("underrun"); 2255 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2256 ns2ms(delta), mNumDelayedWrites, this); 2257 lastWarning = now; 2258 } 2259 } 2260} 2261 2262 mStandby = false; 2263 } else { 2264 usleep(sleepTime); 2265 } 2266 } 2267 2268 // Finally let go of removed track(s), without the lock held 2269 // since we can't guarantee the destructors won't acquire that 2270 // same lock. This will also mutate and push a new fast mixer state. 2271 threadLoop_removeTracks(tracksToRemove); 2272 tracksToRemove.clear(); 2273 2274 // FIXME I don't understand the need for this here; 2275 // it was in the original code but maybe the 2276 // assignment in saveOutputTracks() makes this unnecessary? 2277 clearOutputTracks(); 2278 2279 // Effect chains will be actually deleted here if they were removed from 2280 // mEffectChains list during mixing or effects processing 2281 effectChains.clear(); 2282 2283 // FIXME Note that the above .clear() is no longer necessary since effectChains 2284 // is now local to this block, but will keep it for now (at least until merge done). 2285 } 2286 2287 threadLoop_exit(); 2288 2289 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2290 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2291 // put output stream into standby mode 2292 if (!mStandby) { 2293 mOutput->stream->common.standby(&mOutput->stream->common); 2294 } 2295 } 2296 2297 releaseWakeLock(); 2298 2299 ALOGV("Thread %p type %d exiting", this, mType); 2300 return false; 2301} 2302 2303// removeTracks_l() must be called with ThreadBase::mLock held 2304void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2305{ 2306 size_t count = tracksToRemove.size(); 2307 if (count) { 2308 for (size_t i=0 ; i<count ; i++) { 2309 const sp<Track>& track = tracksToRemove.itemAt(i); 2310 mActiveTracks.remove(track); 2311 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2312 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2313 if (chain != 0) { 2314 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2315 track->sessionId()); 2316 chain->decActiveTrackCnt(); 2317 } 2318 if (track->isTerminated()) { 2319 removeTrack_l(track); 2320 } 2321 } 2322 } 2323 2324} 2325 2326// ---------------------------------------------------------------------------- 2327 2328AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2329 audio_io_handle_t id, audio_devices_t device, type_t type) 2330 : PlaybackThread(audioFlinger, output, id, device, type), 2331 // mAudioMixer below 2332 // mFastMixer below 2333 mFastMixerFutex(0) 2334 // mOutputSink below 2335 // mPipeSink below 2336 // mNormalSink below 2337{ 2338 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2339 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2340 "mFrameCount=%d, mNormalFrameCount=%d", 2341 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2342 mNormalFrameCount); 2343 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2344 2345 // FIXME - Current mixer implementation only supports stereo output 2346 if (mChannelCount != FCC_2) { 2347 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2348 } 2349 2350 // create an NBAIO sink for the HAL output stream, and negotiate 2351 mOutputSink = new AudioStreamOutSink(output->stream); 2352 size_t numCounterOffers = 0; 2353 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2354 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2355 ALOG_ASSERT(index == 0); 2356 2357 // initialize fast mixer depending on configuration 2358 bool initFastMixer; 2359 switch (kUseFastMixer) { 2360 case FastMixer_Never: 2361 initFastMixer = false; 2362 break; 2363 case FastMixer_Always: 2364 initFastMixer = true; 2365 break; 2366 case FastMixer_Static: 2367 case FastMixer_Dynamic: 2368 initFastMixer = mFrameCount < mNormalFrameCount; 2369 break; 2370 } 2371 if (initFastMixer) { 2372 2373 // create a MonoPipe to connect our submix to FastMixer 2374 NBAIO_Format format = mOutputSink->format(); 2375 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2376 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2377 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2378 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2379 const NBAIO_Format offers[1] = {format}; 2380 size_t numCounterOffers = 0; 2381 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2382 ALOG_ASSERT(index == 0); 2383 monoPipe->setAvgFrames((mScreenState & 1) ? 2384 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2385 mPipeSink = monoPipe; 2386 2387#ifdef TEE_SINK 2388 if (mTeeSinkOutputEnabled) { 2389 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2390 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2391 numCounterOffers = 0; 2392 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2393 ALOG_ASSERT(index == 0); 2394 mTeeSink = teeSink; 2395 PipeReader *teeSource = new PipeReader(*teeSink); 2396 numCounterOffers = 0; 2397 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2398 ALOG_ASSERT(index == 0); 2399 mTeeSource = teeSource; 2400 } 2401#endif 2402 2403 // create fast mixer and configure it initially with just one fast track for our submix 2404 mFastMixer = new FastMixer(); 2405 FastMixerStateQueue *sq = mFastMixer->sq(); 2406#ifdef STATE_QUEUE_DUMP 2407 sq->setObserverDump(&mStateQueueObserverDump); 2408 sq->setMutatorDump(&mStateQueueMutatorDump); 2409#endif 2410 FastMixerState *state = sq->begin(); 2411 FastTrack *fastTrack = &state->mFastTracks[0]; 2412 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2413 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2414 fastTrack->mVolumeProvider = NULL; 2415 fastTrack->mGeneration++; 2416 state->mFastTracksGen++; 2417 state->mTrackMask = 1; 2418 // fast mixer will use the HAL output sink 2419 state->mOutputSink = mOutputSink.get(); 2420 state->mOutputSinkGen++; 2421 state->mFrameCount = mFrameCount; 2422 state->mCommand = FastMixerState::COLD_IDLE; 2423 // already done in constructor initialization list 2424 //mFastMixerFutex = 0; 2425 state->mColdFutexAddr = &mFastMixerFutex; 2426 state->mColdGen++; 2427 state->mDumpState = &mFastMixerDumpState; 2428#ifdef TEE_SINK 2429 state->mTeeSink = mTeeSink.get(); 2430#endif 2431 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2432 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2433 sq->end(); 2434 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2435 2436 // start the fast mixer 2437 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2438 pid_t tid = mFastMixer->getTid(); 2439 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2440 if (err != 0) { 2441 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2442 kPriorityFastMixer, getpid_cached, tid, err); 2443 } 2444 2445#ifdef AUDIO_WATCHDOG 2446 // create and start the watchdog 2447 mAudioWatchdog = new AudioWatchdog(); 2448 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2449 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2450 tid = mAudioWatchdog->getTid(); 2451 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2452 if (err != 0) { 2453 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2454 kPriorityFastMixer, getpid_cached, tid, err); 2455 } 2456#endif 2457 2458 } else { 2459 mFastMixer = NULL; 2460 } 2461 2462 switch (kUseFastMixer) { 2463 case FastMixer_Never: 2464 case FastMixer_Dynamic: 2465 mNormalSink = mOutputSink; 2466 break; 2467 case FastMixer_Always: 2468 mNormalSink = mPipeSink; 2469 break; 2470 case FastMixer_Static: 2471 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2472 break; 2473 } 2474} 2475 2476AudioFlinger::MixerThread::~MixerThread() 2477{ 2478 if (mFastMixer != NULL) { 2479 FastMixerStateQueue *sq = mFastMixer->sq(); 2480 FastMixerState *state = sq->begin(); 2481 if (state->mCommand == FastMixerState::COLD_IDLE) { 2482 int32_t old = android_atomic_inc(&mFastMixerFutex); 2483 if (old == -1) { 2484 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2485 } 2486 } 2487 state->mCommand = FastMixerState::EXIT; 2488 sq->end(); 2489 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2490 mFastMixer->join(); 2491 // Though the fast mixer thread has exited, it's state queue is still valid. 2492 // We'll use that extract the final state which contains one remaining fast track 2493 // corresponding to our sub-mix. 2494 state = sq->begin(); 2495 ALOG_ASSERT(state->mTrackMask == 1); 2496 FastTrack *fastTrack = &state->mFastTracks[0]; 2497 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2498 delete fastTrack->mBufferProvider; 2499 sq->end(false /*didModify*/); 2500 delete mFastMixer; 2501#ifdef AUDIO_WATCHDOG 2502 if (mAudioWatchdog != 0) { 2503 mAudioWatchdog->requestExit(); 2504 mAudioWatchdog->requestExitAndWait(); 2505 mAudioWatchdog.clear(); 2506 } 2507#endif 2508 } 2509 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2510 delete mAudioMixer; 2511} 2512 2513 2514uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2515{ 2516 if (mFastMixer != NULL) { 2517 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2518 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2519 } 2520 return latency; 2521} 2522 2523 2524void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2525{ 2526 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2527} 2528 2529ssize_t AudioFlinger::MixerThread::threadLoop_write() 2530{ 2531 // FIXME we should only do one push per cycle; confirm this is true 2532 // Start the fast mixer if it's not already running 2533 if (mFastMixer != NULL) { 2534 FastMixerStateQueue *sq = mFastMixer->sq(); 2535 FastMixerState *state = sq->begin(); 2536 if (state->mCommand != FastMixerState::MIX_WRITE && 2537 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2538 if (state->mCommand == FastMixerState::COLD_IDLE) { 2539 int32_t old = android_atomic_inc(&mFastMixerFutex); 2540 if (old == -1) { 2541 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2542 } 2543#ifdef AUDIO_WATCHDOG 2544 if (mAudioWatchdog != 0) { 2545 mAudioWatchdog->resume(); 2546 } 2547#endif 2548 } 2549 state->mCommand = FastMixerState::MIX_WRITE; 2550 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2551 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2552 sq->end(); 2553 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2554 if (kUseFastMixer == FastMixer_Dynamic) { 2555 mNormalSink = mPipeSink; 2556 } 2557 } else { 2558 sq->end(false /*didModify*/); 2559 } 2560 } 2561 return PlaybackThread::threadLoop_write(); 2562} 2563 2564void AudioFlinger::MixerThread::threadLoop_standby() 2565{ 2566 // Idle the fast mixer if it's currently running 2567 if (mFastMixer != NULL) { 2568 FastMixerStateQueue *sq = mFastMixer->sq(); 2569 FastMixerState *state = sq->begin(); 2570 if (!(state->mCommand & FastMixerState::IDLE)) { 2571 state->mCommand = FastMixerState::COLD_IDLE; 2572 state->mColdFutexAddr = &mFastMixerFutex; 2573 state->mColdGen++; 2574 mFastMixerFutex = 0; 2575 sq->end(); 2576 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2577 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2578 if (kUseFastMixer == FastMixer_Dynamic) { 2579 mNormalSink = mOutputSink; 2580 } 2581#ifdef AUDIO_WATCHDOG 2582 if (mAudioWatchdog != 0) { 2583 mAudioWatchdog->pause(); 2584 } 2585#endif 2586 } else { 2587 sq->end(false /*didModify*/); 2588 } 2589 } 2590 PlaybackThread::threadLoop_standby(); 2591} 2592 2593// Empty implementation for standard mixer 2594// Overridden for offloaded playback 2595void AudioFlinger::PlaybackThread::flushOutput_l() 2596{ 2597} 2598 2599bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2600{ 2601 return false; 2602} 2603 2604bool AudioFlinger::PlaybackThread::shouldStandby_l() 2605{ 2606 return !mStandby; 2607} 2608 2609bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2610{ 2611 Mutex::Autolock _l(mLock); 2612 return waitingAsyncCallback_l(); 2613} 2614 2615// shared by MIXER and DIRECT, overridden by DUPLICATING 2616void AudioFlinger::PlaybackThread::threadLoop_standby() 2617{ 2618 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2619 mOutput->stream->common.standby(&mOutput->stream->common); 2620 if (mUseAsyncWrite != 0) { 2621 // discard any pending drain or write ack by incrementing sequence 2622 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2623 mDrainSequence = (mDrainSequence + 2) & ~1; 2624 ALOG_ASSERT(mCallbackThread != 0); 2625 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2626 mCallbackThread->setDraining(mDrainSequence); 2627 } 2628} 2629 2630void AudioFlinger::MixerThread::threadLoop_mix() 2631{ 2632 // obtain the presentation timestamp of the next output buffer 2633 int64_t pts; 2634 status_t status = INVALID_OPERATION; 2635 2636 if (mNormalSink != 0) { 2637 status = mNormalSink->getNextWriteTimestamp(&pts); 2638 } else { 2639 status = mOutputSink->getNextWriteTimestamp(&pts); 2640 } 2641 2642 if (status != NO_ERROR) { 2643 pts = AudioBufferProvider::kInvalidPTS; 2644 } 2645 2646 // mix buffers... 2647 mAudioMixer->process(pts); 2648 mCurrentWriteLength = mixBufferSize; 2649 // increase sleep time progressively when application underrun condition clears. 2650 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2651 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2652 // such that we would underrun the audio HAL. 2653 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2654 sleepTimeShift--; 2655 } 2656 sleepTime = 0; 2657 standbyTime = systemTime() + standbyDelay; 2658 //TODO: delay standby when effects have a tail 2659} 2660 2661void AudioFlinger::MixerThread::threadLoop_sleepTime() 2662{ 2663 // If no tracks are ready, sleep once for the duration of an output 2664 // buffer size, then write 0s to the output 2665 if (sleepTime == 0) { 2666 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2667 sleepTime = activeSleepTime >> sleepTimeShift; 2668 if (sleepTime < kMinThreadSleepTimeUs) { 2669 sleepTime = kMinThreadSleepTimeUs; 2670 } 2671 // reduce sleep time in case of consecutive application underruns to avoid 2672 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2673 // duration we would end up writing less data than needed by the audio HAL if 2674 // the condition persists. 2675 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2676 sleepTimeShift++; 2677 } 2678 } else { 2679 sleepTime = idleSleepTime; 2680 } 2681 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2682 memset (mMixBuffer, 0, mixBufferSize); 2683 sleepTime = 0; 2684 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2685 "anticipated start"); 2686 } 2687 // TODO add standby time extension fct of effect tail 2688} 2689 2690// prepareTracks_l() must be called with ThreadBase::mLock held 2691AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2692 Vector< sp<Track> > *tracksToRemove) 2693{ 2694 2695 mixer_state mixerStatus = MIXER_IDLE; 2696 // find out which tracks need to be processed 2697 size_t count = mActiveTracks.size(); 2698 size_t mixedTracks = 0; 2699 size_t tracksWithEffect = 0; 2700 // counts only _active_ fast tracks 2701 size_t fastTracks = 0; 2702 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2703 2704 float masterVolume = mMasterVolume; 2705 bool masterMute = mMasterMute; 2706 2707 if (masterMute) { 2708 masterVolume = 0; 2709 } 2710 // Delegate master volume control to effect in output mix effect chain if needed 2711 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2712 if (chain != 0) { 2713 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2714 chain->setVolume_l(&v, &v); 2715 masterVolume = (float)((v + (1 << 23)) >> 24); 2716 chain.clear(); 2717 } 2718 2719 // prepare a new state to push 2720 FastMixerStateQueue *sq = NULL; 2721 FastMixerState *state = NULL; 2722 bool didModify = false; 2723 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2724 if (mFastMixer != NULL) { 2725 sq = mFastMixer->sq(); 2726 state = sq->begin(); 2727 } 2728 2729 for (size_t i=0 ; i<count ; i++) { 2730 const sp<Track> t = mActiveTracks[i].promote(); 2731 if (t == 0) { 2732 continue; 2733 } 2734 2735 // this const just means the local variable doesn't change 2736 Track* const track = t.get(); 2737 2738 // process fast tracks 2739 if (track->isFastTrack()) { 2740 2741 // It's theoretically possible (though unlikely) for a fast track to be created 2742 // and then removed within the same normal mix cycle. This is not a problem, as 2743 // the track never becomes active so it's fast mixer slot is never touched. 2744 // The converse, of removing an (active) track and then creating a new track 2745 // at the identical fast mixer slot within the same normal mix cycle, 2746 // is impossible because the slot isn't marked available until the end of each cycle. 2747 int j = track->mFastIndex; 2748 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2749 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2750 FastTrack *fastTrack = &state->mFastTracks[j]; 2751 2752 // Determine whether the track is currently in underrun condition, 2753 // and whether it had a recent underrun. 2754 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2755 FastTrackUnderruns underruns = ftDump->mUnderruns; 2756 uint32_t recentFull = (underruns.mBitFields.mFull - 2757 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2758 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2759 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2760 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2761 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2762 uint32_t recentUnderruns = recentPartial + recentEmpty; 2763 track->mObservedUnderruns = underruns; 2764 // don't count underruns that occur while stopping or pausing 2765 // or stopped which can occur when flush() is called while active 2766 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2767 recentUnderruns > 0) { 2768 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2769 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2770 } 2771 2772 // This is similar to the state machine for normal tracks, 2773 // with a few modifications for fast tracks. 2774 bool isActive = true; 2775 switch (track->mState) { 2776 case TrackBase::STOPPING_1: 2777 // track stays active in STOPPING_1 state until first underrun 2778 if (recentUnderruns > 0 || track->isTerminated()) { 2779 track->mState = TrackBase::STOPPING_2; 2780 } 2781 break; 2782 case TrackBase::PAUSING: 2783 // ramp down is not yet implemented 2784 track->setPaused(); 2785 break; 2786 case TrackBase::RESUMING: 2787 // ramp up is not yet implemented 2788 track->mState = TrackBase::ACTIVE; 2789 break; 2790 case TrackBase::ACTIVE: 2791 if (recentFull > 0 || recentPartial > 0) { 2792 // track has provided at least some frames recently: reset retry count 2793 track->mRetryCount = kMaxTrackRetries; 2794 } 2795 if (recentUnderruns == 0) { 2796 // no recent underruns: stay active 2797 break; 2798 } 2799 // there has recently been an underrun of some kind 2800 if (track->sharedBuffer() == 0) { 2801 // were any of the recent underruns "empty" (no frames available)? 2802 if (recentEmpty == 0) { 2803 // no, then ignore the partial underruns as they are allowed indefinitely 2804 break; 2805 } 2806 // there has recently been an "empty" underrun: decrement the retry counter 2807 if (--(track->mRetryCount) > 0) { 2808 break; 2809 } 2810 // indicate to client process that the track was disabled because of underrun; 2811 // it will then automatically call start() when data is available 2812 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2813 // remove from active list, but state remains ACTIVE [confusing but true] 2814 isActive = false; 2815 break; 2816 } 2817 // fall through 2818 case TrackBase::STOPPING_2: 2819 case TrackBase::PAUSED: 2820 case TrackBase::STOPPED: 2821 case TrackBase::FLUSHED: // flush() while active 2822 // Check for presentation complete if track is inactive 2823 // We have consumed all the buffers of this track. 2824 // This would be incomplete if we auto-paused on underrun 2825 { 2826 size_t audioHALFrames = 2827 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2828 size_t framesWritten = mBytesWritten / mFrameSize; 2829 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2830 // track stays in active list until presentation is complete 2831 break; 2832 } 2833 } 2834 if (track->isStopping_2()) { 2835 track->mState = TrackBase::STOPPED; 2836 } 2837 if (track->isStopped()) { 2838 // Can't reset directly, as fast mixer is still polling this track 2839 // track->reset(); 2840 // So instead mark this track as needing to be reset after push with ack 2841 resetMask |= 1 << i; 2842 } 2843 isActive = false; 2844 break; 2845 case TrackBase::IDLE: 2846 default: 2847 LOG_FATAL("unexpected track state %d", track->mState); 2848 } 2849 2850 if (isActive) { 2851 // was it previously inactive? 2852 if (!(state->mTrackMask & (1 << j))) { 2853 ExtendedAudioBufferProvider *eabp = track; 2854 VolumeProvider *vp = track; 2855 fastTrack->mBufferProvider = eabp; 2856 fastTrack->mVolumeProvider = vp; 2857 fastTrack->mSampleRate = track->mSampleRate; 2858 fastTrack->mChannelMask = track->mChannelMask; 2859 fastTrack->mGeneration++; 2860 state->mTrackMask |= 1 << j; 2861 didModify = true; 2862 // no acknowledgement required for newly active tracks 2863 } 2864 // cache the combined master volume and stream type volume for fast mixer; this 2865 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2866 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2867 ++fastTracks; 2868 } else { 2869 // was it previously active? 2870 if (state->mTrackMask & (1 << j)) { 2871 fastTrack->mBufferProvider = NULL; 2872 fastTrack->mGeneration++; 2873 state->mTrackMask &= ~(1 << j); 2874 didModify = true; 2875 // If any fast tracks were removed, we must wait for acknowledgement 2876 // because we're about to decrement the last sp<> on those tracks. 2877 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2878 } else { 2879 LOG_FATAL("fast track %d should have been active", j); 2880 } 2881 tracksToRemove->add(track); 2882 // Avoids a misleading display in dumpsys 2883 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2884 } 2885 continue; 2886 } 2887 2888 { // local variable scope to avoid goto warning 2889 2890 audio_track_cblk_t* cblk = track->cblk(); 2891 2892 // The first time a track is added we wait 2893 // for all its buffers to be filled before processing it 2894 int name = track->name(); 2895 // make sure that we have enough frames to mix one full buffer. 2896 // enforce this condition only once to enable draining the buffer in case the client 2897 // app does not call stop() and relies on underrun to stop: 2898 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2899 // during last round 2900 size_t desiredFrames; 2901 uint32_t sr = track->sampleRate(); 2902 if (sr == mSampleRate) { 2903 desiredFrames = mNormalFrameCount; 2904 } else { 2905 // +1 for rounding and +1 for additional sample needed for interpolation 2906 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2907 // add frames already consumed but not yet released by the resampler 2908 // because cblk->framesReady() will include these frames 2909 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2910 // the minimum track buffer size is normally twice the number of frames necessary 2911 // to fill one buffer and the resampler should not leave more than one buffer worth 2912 // of unreleased frames after each pass, but just in case... 2913 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2914 } 2915 uint32_t minFrames = 1; 2916 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2917 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2918 minFrames = desiredFrames; 2919 } 2920 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2921 size_t framesReady; 2922 if (track->sharedBuffer() == 0) { 2923 framesReady = track->framesReady(); 2924 } else if (track->isStopped()) { 2925 framesReady = 0; 2926 } else { 2927 framesReady = 1; 2928 } 2929 if ((framesReady >= minFrames) && track->isReady() && 2930 !track->isPaused() && !track->isTerminated()) 2931 { 2932 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2933 2934 mixedTracks++; 2935 2936 // track->mainBuffer() != mMixBuffer means there is an effect chain 2937 // connected to the track 2938 chain.clear(); 2939 if (track->mainBuffer() != mMixBuffer) { 2940 chain = getEffectChain_l(track->sessionId()); 2941 // Delegate volume control to effect in track effect chain if needed 2942 if (chain != 0) { 2943 tracksWithEffect++; 2944 } else { 2945 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2946 "session %d", 2947 name, track->sessionId()); 2948 } 2949 } 2950 2951 2952 int param = AudioMixer::VOLUME; 2953 if (track->mFillingUpStatus == Track::FS_FILLED) { 2954 // no ramp for the first volume setting 2955 track->mFillingUpStatus = Track::FS_ACTIVE; 2956 if (track->mState == TrackBase::RESUMING) { 2957 track->mState = TrackBase::ACTIVE; 2958 param = AudioMixer::RAMP_VOLUME; 2959 } 2960 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2961 // FIXME should not make a decision based on mServer 2962 } else if (cblk->mServer != 0) { 2963 // If the track is stopped before the first frame was mixed, 2964 // do not apply ramp 2965 param = AudioMixer::RAMP_VOLUME; 2966 } 2967 2968 // compute volume for this track 2969 uint32_t vl, vr, va; 2970 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2971 vl = vr = va = 0; 2972 if (track->isPausing()) { 2973 track->setPaused(); 2974 } 2975 } else { 2976 2977 // read original volumes with volume control 2978 float typeVolume = mStreamTypes[track->streamType()].volume; 2979 float v = masterVolume * typeVolume; 2980 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2981 uint32_t vlr = proxy->getVolumeLR(); 2982 vl = vlr & 0xFFFF; 2983 vr = vlr >> 16; 2984 // track volumes come from shared memory, so can't be trusted and must be clamped 2985 if (vl > MAX_GAIN_INT) { 2986 ALOGV("Track left volume out of range: %04X", vl); 2987 vl = MAX_GAIN_INT; 2988 } 2989 if (vr > MAX_GAIN_INT) { 2990 ALOGV("Track right volume out of range: %04X", vr); 2991 vr = MAX_GAIN_INT; 2992 } 2993 // now apply the master volume and stream type volume 2994 vl = (uint32_t)(v * vl) << 12; 2995 vr = (uint32_t)(v * vr) << 12; 2996 // assuming master volume and stream type volume each go up to 1.0, 2997 // vl and vr are now in 8.24 format 2998 2999 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3000 // send level comes from shared memory and so may be corrupt 3001 if (sendLevel > MAX_GAIN_INT) { 3002 ALOGV("Track send level out of range: %04X", sendLevel); 3003 sendLevel = MAX_GAIN_INT; 3004 } 3005 va = (uint32_t)(v * sendLevel); 3006 } 3007 3008 // Delegate volume control to effect in track effect chain if needed 3009 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3010 // Do not ramp volume if volume is controlled by effect 3011 param = AudioMixer::VOLUME; 3012 track->mHasVolumeController = true; 3013 } else { 3014 // force no volume ramp when volume controller was just disabled or removed 3015 // from effect chain to avoid volume spike 3016 if (track->mHasVolumeController) { 3017 param = AudioMixer::VOLUME; 3018 } 3019 track->mHasVolumeController = false; 3020 } 3021 3022 // Convert volumes from 8.24 to 4.12 format 3023 // This additional clamping is needed in case chain->setVolume_l() overshot 3024 vl = (vl + (1 << 11)) >> 12; 3025 if (vl > MAX_GAIN_INT) { 3026 vl = MAX_GAIN_INT; 3027 } 3028 vr = (vr + (1 << 11)) >> 12; 3029 if (vr > MAX_GAIN_INT) { 3030 vr = MAX_GAIN_INT; 3031 } 3032 3033 if (va > MAX_GAIN_INT) { 3034 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3035 } 3036 3037 // XXX: these things DON'T need to be done each time 3038 mAudioMixer->setBufferProvider(name, track); 3039 mAudioMixer->enable(name); 3040 3041 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3042 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3043 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3044 mAudioMixer->setParameter( 3045 name, 3046 AudioMixer::TRACK, 3047 AudioMixer::FORMAT, (void *)track->format()); 3048 mAudioMixer->setParameter( 3049 name, 3050 AudioMixer::TRACK, 3051 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3052 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3053 uint32_t maxSampleRate = mSampleRate * 2; 3054 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3055 if (reqSampleRate == 0) { 3056 reqSampleRate = mSampleRate; 3057 } else if (reqSampleRate > maxSampleRate) { 3058 reqSampleRate = maxSampleRate; 3059 } 3060 mAudioMixer->setParameter( 3061 name, 3062 AudioMixer::RESAMPLE, 3063 AudioMixer::SAMPLE_RATE, 3064 (void *)reqSampleRate); 3065 mAudioMixer->setParameter( 3066 name, 3067 AudioMixer::TRACK, 3068 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3069 mAudioMixer->setParameter( 3070 name, 3071 AudioMixer::TRACK, 3072 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3073 3074 // reset retry count 3075 track->mRetryCount = kMaxTrackRetries; 3076 3077 // If one track is ready, set the mixer ready if: 3078 // - the mixer was not ready during previous round OR 3079 // - no other track is not ready 3080 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3081 mixerStatus != MIXER_TRACKS_ENABLED) { 3082 mixerStatus = MIXER_TRACKS_READY; 3083 } 3084 } else { 3085 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3086 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3087 } 3088 // clear effect chain input buffer if an active track underruns to avoid sending 3089 // previous audio buffer again to effects 3090 chain = getEffectChain_l(track->sessionId()); 3091 if (chain != 0) { 3092 chain->clearInputBuffer(); 3093 } 3094 3095 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3096 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3097 track->isStopped() || track->isPaused()) { 3098 // We have consumed all the buffers of this track. 3099 // Remove it from the list of active tracks. 3100 // TODO: use actual buffer filling status instead of latency when available from 3101 // audio HAL 3102 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3103 size_t framesWritten = mBytesWritten / mFrameSize; 3104 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3105 if (track->isStopped()) { 3106 track->reset(); 3107 } 3108 tracksToRemove->add(track); 3109 } 3110 } else { 3111 // No buffers for this track. Give it a few chances to 3112 // fill a buffer, then remove it from active list. 3113 if (--(track->mRetryCount) <= 0) { 3114 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3115 tracksToRemove->add(track); 3116 // indicate to client process that the track was disabled because of underrun; 3117 // it will then automatically call start() when data is available 3118 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3119 // If one track is not ready, mark the mixer also not ready if: 3120 // - the mixer was ready during previous round OR 3121 // - no other track is ready 3122 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3123 mixerStatus != MIXER_TRACKS_READY) { 3124 mixerStatus = MIXER_TRACKS_ENABLED; 3125 } 3126 } 3127 mAudioMixer->disable(name); 3128 } 3129 3130 } // local variable scope to avoid goto warning 3131track_is_ready: ; 3132 3133 } 3134 3135 // Push the new FastMixer state if necessary 3136 bool pauseAudioWatchdog = false; 3137 if (didModify) { 3138 state->mFastTracksGen++; 3139 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3140 if (kUseFastMixer == FastMixer_Dynamic && 3141 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3142 state->mCommand = FastMixerState::COLD_IDLE; 3143 state->mColdFutexAddr = &mFastMixerFutex; 3144 state->mColdGen++; 3145 mFastMixerFutex = 0; 3146 if (kUseFastMixer == FastMixer_Dynamic) { 3147 mNormalSink = mOutputSink; 3148 } 3149 // If we go into cold idle, need to wait for acknowledgement 3150 // so that fast mixer stops doing I/O. 3151 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3152 pauseAudioWatchdog = true; 3153 } 3154 } 3155 if (sq != NULL) { 3156 sq->end(didModify); 3157 sq->push(block); 3158 } 3159#ifdef AUDIO_WATCHDOG 3160 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3161 mAudioWatchdog->pause(); 3162 } 3163#endif 3164 3165 // Now perform the deferred reset on fast tracks that have stopped 3166 while (resetMask != 0) { 3167 size_t i = __builtin_ctz(resetMask); 3168 ALOG_ASSERT(i < count); 3169 resetMask &= ~(1 << i); 3170 sp<Track> t = mActiveTracks[i].promote(); 3171 if (t == 0) { 3172 continue; 3173 } 3174 Track* track = t.get(); 3175 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3176 track->reset(); 3177 } 3178 3179 // remove all the tracks that need to be... 3180 removeTracks_l(*tracksToRemove); 3181 3182 // mix buffer must be cleared if all tracks are connected to an 3183 // effect chain as in this case the mixer will not write to 3184 // mix buffer and track effects will accumulate into it 3185 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3186 (mixedTracks == 0 && fastTracks > 0))) { 3187 // FIXME as a performance optimization, should remember previous zero status 3188 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3189 } 3190 3191 // if any fast tracks, then status is ready 3192 mMixerStatusIgnoringFastTracks = mixerStatus; 3193 if (fastTracks > 0) { 3194 mixerStatus = MIXER_TRACKS_READY; 3195 } 3196 return mixerStatus; 3197} 3198 3199// getTrackName_l() must be called with ThreadBase::mLock held 3200int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3201{ 3202 return mAudioMixer->getTrackName(channelMask, sessionId); 3203} 3204 3205// deleteTrackName_l() must be called with ThreadBase::mLock held 3206void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3207{ 3208 ALOGV("remove track (%d) and delete from mixer", name); 3209 mAudioMixer->deleteTrackName(name); 3210} 3211 3212// checkForNewParameters_l() must be called with ThreadBase::mLock held 3213bool AudioFlinger::MixerThread::checkForNewParameters_l() 3214{ 3215 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3216 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3217 bool reconfig = false; 3218 3219 while (!mNewParameters.isEmpty()) { 3220 3221 if (mFastMixer != NULL) { 3222 FastMixerStateQueue *sq = mFastMixer->sq(); 3223 FastMixerState *state = sq->begin(); 3224 if (!(state->mCommand & FastMixerState::IDLE)) { 3225 previousCommand = state->mCommand; 3226 state->mCommand = FastMixerState::HOT_IDLE; 3227 sq->end(); 3228 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3229 } else { 3230 sq->end(false /*didModify*/); 3231 } 3232 } 3233 3234 status_t status = NO_ERROR; 3235 String8 keyValuePair = mNewParameters[0]; 3236 AudioParameter param = AudioParameter(keyValuePair); 3237 int value; 3238 3239 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3240 reconfig = true; 3241 } 3242 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3243 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3244 status = BAD_VALUE; 3245 } else { 3246 reconfig = true; 3247 } 3248 } 3249 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3250 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3251 status = BAD_VALUE; 3252 } else { 3253 reconfig = true; 3254 } 3255 } 3256 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3257 // do not accept frame count changes if tracks are open as the track buffer 3258 // size depends on frame count and correct behavior would not be guaranteed 3259 // if frame count is changed after track creation 3260 if (!mTracks.isEmpty()) { 3261 status = INVALID_OPERATION; 3262 } else { 3263 reconfig = true; 3264 } 3265 } 3266 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3267#ifdef ADD_BATTERY_DATA 3268 // when changing the audio output device, call addBatteryData to notify 3269 // the change 3270 if (mOutDevice != value) { 3271 uint32_t params = 0; 3272 // check whether speaker is on 3273 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3274 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3275 } 3276 3277 audio_devices_t deviceWithoutSpeaker 3278 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3279 // check if any other device (except speaker) is on 3280 if (value & deviceWithoutSpeaker ) { 3281 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3282 } 3283 3284 if (params != 0) { 3285 addBatteryData(params); 3286 } 3287 } 3288#endif 3289 3290 // forward device change to effects that have requested to be 3291 // aware of attached audio device. 3292 if (value != AUDIO_DEVICE_NONE) { 3293 mOutDevice = value; 3294 for (size_t i = 0; i < mEffectChains.size(); i++) { 3295 mEffectChains[i]->setDevice_l(mOutDevice); 3296 } 3297 } 3298 } 3299 3300 if (status == NO_ERROR) { 3301 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3302 keyValuePair.string()); 3303 if (!mStandby && status == INVALID_OPERATION) { 3304 mOutput->stream->common.standby(&mOutput->stream->common); 3305 mStandby = true; 3306 mBytesWritten = 0; 3307 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3308 keyValuePair.string()); 3309 } 3310 if (status == NO_ERROR && reconfig) { 3311 readOutputParameters(); 3312 delete mAudioMixer; 3313 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3314 for (size_t i = 0; i < mTracks.size() ; i++) { 3315 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3316 if (name < 0) { 3317 break; 3318 } 3319 mTracks[i]->mName = name; 3320 } 3321 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3322 } 3323 } 3324 3325 mNewParameters.removeAt(0); 3326 3327 mParamStatus = status; 3328 mParamCond.signal(); 3329 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3330 // already timed out waiting for the status and will never signal the condition. 3331 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3332 } 3333 3334 if (!(previousCommand & FastMixerState::IDLE)) { 3335 ALOG_ASSERT(mFastMixer != NULL); 3336 FastMixerStateQueue *sq = mFastMixer->sq(); 3337 FastMixerState *state = sq->begin(); 3338 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3339 state->mCommand = previousCommand; 3340 sq->end(); 3341 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3342 } 3343 3344 return reconfig; 3345} 3346 3347 3348void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3349{ 3350 const size_t SIZE = 256; 3351 char buffer[SIZE]; 3352 String8 result; 3353 3354 PlaybackThread::dumpInternals(fd, args); 3355 3356 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3357 result.append(buffer); 3358 write(fd, result.string(), result.size()); 3359 3360 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3361 const FastMixerDumpState copy(mFastMixerDumpState); 3362 copy.dump(fd); 3363 3364#ifdef STATE_QUEUE_DUMP 3365 // Similar for state queue 3366 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3367 observerCopy.dump(fd); 3368 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3369 mutatorCopy.dump(fd); 3370#endif 3371 3372#ifdef TEE_SINK 3373 // Write the tee output to a .wav file 3374 dumpTee(fd, mTeeSource, mId); 3375#endif 3376 3377#ifdef AUDIO_WATCHDOG 3378 if (mAudioWatchdog != 0) { 3379 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3380 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3381 wdCopy.dump(fd); 3382 } 3383#endif 3384} 3385 3386uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3387{ 3388 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3389} 3390 3391uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3392{ 3393 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3394} 3395 3396void AudioFlinger::MixerThread::cacheParameters_l() 3397{ 3398 PlaybackThread::cacheParameters_l(); 3399 3400 // FIXME: Relaxed timing because of a certain device that can't meet latency 3401 // Should be reduced to 2x after the vendor fixes the driver issue 3402 // increase threshold again due to low power audio mode. The way this warning 3403 // threshold is calculated and its usefulness should be reconsidered anyway. 3404 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3405} 3406 3407// ---------------------------------------------------------------------------- 3408 3409AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3410 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3411 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3412 // mLeftVolFloat, mRightVolFloat 3413{ 3414} 3415 3416AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3417 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3418 ThreadBase::type_t type) 3419 : PlaybackThread(audioFlinger, output, id, device, type) 3420 // mLeftVolFloat, mRightVolFloat 3421{ 3422} 3423 3424AudioFlinger::DirectOutputThread::~DirectOutputThread() 3425{ 3426} 3427 3428void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3429{ 3430 audio_track_cblk_t* cblk = track->cblk(); 3431 float left, right; 3432 3433 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3434 left = right = 0; 3435 } else { 3436 float typeVolume = mStreamTypes[track->streamType()].volume; 3437 float v = mMasterVolume * typeVolume; 3438 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3439 uint32_t vlr = proxy->getVolumeLR(); 3440 float v_clamped = v * (vlr & 0xFFFF); 3441 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3442 left = v_clamped/MAX_GAIN; 3443 v_clamped = v * (vlr >> 16); 3444 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3445 right = v_clamped/MAX_GAIN; 3446 } 3447 3448 if (lastTrack) { 3449 if (left != mLeftVolFloat || right != mRightVolFloat) { 3450 mLeftVolFloat = left; 3451 mRightVolFloat = right; 3452 3453 // Convert volumes from float to 8.24 3454 uint32_t vl = (uint32_t)(left * (1 << 24)); 3455 uint32_t vr = (uint32_t)(right * (1 << 24)); 3456 3457 // Delegate volume control to effect in track effect chain if needed 3458 // only one effect chain can be present on DirectOutputThread, so if 3459 // there is one, the track is connected to it 3460 if (!mEffectChains.isEmpty()) { 3461 mEffectChains[0]->setVolume_l(&vl, &vr); 3462 left = (float)vl / (1 << 24); 3463 right = (float)vr / (1 << 24); 3464 } 3465 if (mOutput->stream->set_volume) { 3466 mOutput->stream->set_volume(mOutput->stream, left, right); 3467 } 3468 } 3469 } 3470} 3471 3472 3473AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3474 Vector< sp<Track> > *tracksToRemove 3475) 3476{ 3477 size_t count = mActiveTracks.size(); 3478 mixer_state mixerStatus = MIXER_IDLE; 3479 3480 // find out which tracks need to be processed 3481 for (size_t i = 0; i < count; i++) { 3482 sp<Track> t = mActiveTracks[i].promote(); 3483 // The track died recently 3484 if (t == 0) { 3485 continue; 3486 } 3487 3488 Track* const track = t.get(); 3489 audio_track_cblk_t* cblk = track->cblk(); 3490 3491 // The first time a track is added we wait 3492 // for all its buffers to be filled before processing it 3493 uint32_t minFrames; 3494 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3495 minFrames = mNormalFrameCount; 3496 } else { 3497 minFrames = 1; 3498 } 3499 // Only consider last track started for volume and mixer state control. 3500 // This is the last entry in mActiveTracks unless a track underruns. 3501 // As we only care about the transition phase between two tracks on a 3502 // direct output, it is not a problem to ignore the underrun case. 3503 bool last = (i == (count - 1)); 3504 3505 if ((track->framesReady() >= minFrames) && track->isReady() && 3506 !track->isPaused() && !track->isTerminated()) 3507 { 3508 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3509 3510 if (track->mFillingUpStatus == Track::FS_FILLED) { 3511 track->mFillingUpStatus = Track::FS_ACTIVE; 3512 // make sure processVolume_l() will apply new volume even if 0 3513 mLeftVolFloat = mRightVolFloat = -1.0; 3514 if (track->mState == TrackBase::RESUMING) { 3515 track->mState = TrackBase::ACTIVE; 3516 } 3517 } 3518 3519 // compute volume for this track 3520 processVolume_l(track, last); 3521 if (last) { 3522 // reset retry count 3523 track->mRetryCount = kMaxTrackRetriesDirect; 3524 mActiveTrack = t; 3525 mixerStatus = MIXER_TRACKS_READY; 3526 } 3527 } else { 3528 // clear effect chain input buffer if the last active track started underruns 3529 // to avoid sending previous audio buffer again to effects 3530 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3531 mEffectChains[0]->clearInputBuffer(); 3532 } 3533 3534 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3535 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3536 track->isStopped() || track->isPaused()) { 3537 // We have consumed all the buffers of this track. 3538 // Remove it from the list of active tracks. 3539 // TODO: implement behavior for compressed audio 3540 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3541 size_t framesWritten = mBytesWritten / mFrameSize; 3542 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3543 if (track->isStopped()) { 3544 track->reset(); 3545 } 3546 tracksToRemove->add(track); 3547 } 3548 } else { 3549 // No buffers for this track. Give it a few chances to 3550 // fill a buffer, then remove it from active list. 3551 // Only consider last track started for mixer state control 3552 if (--(track->mRetryCount) <= 0) { 3553 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3554 tracksToRemove->add(track); 3555 } else if (last) { 3556 mixerStatus = MIXER_TRACKS_ENABLED; 3557 } 3558 } 3559 } 3560 } 3561 3562 // remove all the tracks that need to be... 3563 removeTracks_l(*tracksToRemove); 3564 3565 return mixerStatus; 3566} 3567 3568void AudioFlinger::DirectOutputThread::threadLoop_mix() 3569{ 3570 size_t frameCount = mFrameCount; 3571 int8_t *curBuf = (int8_t *)mMixBuffer; 3572 // output audio to hardware 3573 while (frameCount) { 3574 AudioBufferProvider::Buffer buffer; 3575 buffer.frameCount = frameCount; 3576 mActiveTrack->getNextBuffer(&buffer); 3577 if (buffer.raw == NULL) { 3578 memset(curBuf, 0, frameCount * mFrameSize); 3579 break; 3580 } 3581 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3582 frameCount -= buffer.frameCount; 3583 curBuf += buffer.frameCount * mFrameSize; 3584 mActiveTrack->releaseBuffer(&buffer); 3585 } 3586 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3587 sleepTime = 0; 3588 standbyTime = systemTime() + standbyDelay; 3589 mActiveTrack.clear(); 3590} 3591 3592void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3593{ 3594 if (sleepTime == 0) { 3595 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3596 sleepTime = activeSleepTime; 3597 } else { 3598 sleepTime = idleSleepTime; 3599 } 3600 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3601 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3602 sleepTime = 0; 3603 } 3604} 3605 3606// getTrackName_l() must be called with ThreadBase::mLock held 3607int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3608 int sessionId) 3609{ 3610 return 0; 3611} 3612 3613// deleteTrackName_l() must be called with ThreadBase::mLock held 3614void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3615{ 3616} 3617 3618// checkForNewParameters_l() must be called with ThreadBase::mLock held 3619bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3620{ 3621 bool reconfig = false; 3622 3623 while (!mNewParameters.isEmpty()) { 3624 status_t status = NO_ERROR; 3625 String8 keyValuePair = mNewParameters[0]; 3626 AudioParameter param = AudioParameter(keyValuePair); 3627 int value; 3628 3629 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3630 // do not accept frame count changes if tracks are open as the track buffer 3631 // size depends on frame count and correct behavior would not be garantied 3632 // if frame count is changed after track creation 3633 if (!mTracks.isEmpty()) { 3634 status = INVALID_OPERATION; 3635 } else { 3636 reconfig = true; 3637 } 3638 } 3639 if (status == NO_ERROR) { 3640 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3641 keyValuePair.string()); 3642 if (!mStandby && status == INVALID_OPERATION) { 3643 mOutput->stream->common.standby(&mOutput->stream->common); 3644 mStandby = true; 3645 mBytesWritten = 0; 3646 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3647 keyValuePair.string()); 3648 } 3649 if (status == NO_ERROR && reconfig) { 3650 readOutputParameters(); 3651 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3652 } 3653 } 3654 3655 mNewParameters.removeAt(0); 3656 3657 mParamStatus = status; 3658 mParamCond.signal(); 3659 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3660 // already timed out waiting for the status and will never signal the condition. 3661 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3662 } 3663 return reconfig; 3664} 3665 3666uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3667{ 3668 uint32_t time; 3669 if (audio_is_linear_pcm(mFormat)) { 3670 time = PlaybackThread::activeSleepTimeUs(); 3671 } else { 3672 time = 10000; 3673 } 3674 return time; 3675} 3676 3677uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3678{ 3679 uint32_t time; 3680 if (audio_is_linear_pcm(mFormat)) { 3681 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3682 } else { 3683 time = 10000; 3684 } 3685 return time; 3686} 3687 3688uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3689{ 3690 uint32_t time; 3691 if (audio_is_linear_pcm(mFormat)) { 3692 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3693 } else { 3694 time = 10000; 3695 } 3696 return time; 3697} 3698 3699void AudioFlinger::DirectOutputThread::cacheParameters_l() 3700{ 3701 PlaybackThread::cacheParameters_l(); 3702 3703 // use shorter standby delay as on normal output to release 3704 // hardware resources as soon as possible 3705 standbyDelay = microseconds(activeSleepTime*2); 3706} 3707 3708// ---------------------------------------------------------------------------- 3709 3710AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3711 const sp<AudioFlinger::OffloadThread>& offloadThread) 3712 : Thread(false /*canCallJava*/), 3713 mOffloadThread(offloadThread), 3714 mWriteAckSequence(0), 3715 mDrainSequence(0) 3716{ 3717} 3718 3719AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3720{ 3721} 3722 3723void AudioFlinger::AsyncCallbackThread::onFirstRef() 3724{ 3725 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3726} 3727 3728bool AudioFlinger::AsyncCallbackThread::threadLoop() 3729{ 3730 while (!exitPending()) { 3731 uint32_t writeAckSequence; 3732 uint32_t drainSequence; 3733 3734 { 3735 Mutex::Autolock _l(mLock); 3736 mWaitWorkCV.wait(mLock); 3737 if (exitPending()) { 3738 break; 3739 } 3740 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3741 mWriteAckSequence, mDrainSequence); 3742 writeAckSequence = mWriteAckSequence; 3743 mWriteAckSequence &= ~1; 3744 drainSequence = mDrainSequence; 3745 mDrainSequence &= ~1; 3746 } 3747 { 3748 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3749 if (offloadThread != 0) { 3750 if (writeAckSequence & 1) { 3751 offloadThread->resetWriteBlocked(writeAckSequence >> 1); 3752 } 3753 if (drainSequence & 1) { 3754 offloadThread->resetDraining(drainSequence >> 1); 3755 } 3756 } 3757 } 3758 } 3759 return false; 3760} 3761 3762void AudioFlinger::AsyncCallbackThread::exit() 3763{ 3764 ALOGV("AsyncCallbackThread::exit"); 3765 Mutex::Autolock _l(mLock); 3766 requestExit(); 3767 mWaitWorkCV.broadcast(); 3768} 3769 3770void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3771{ 3772 Mutex::Autolock _l(mLock); 3773 // bit 0 is cleared 3774 mWriteAckSequence = sequence << 1; 3775} 3776 3777void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3778{ 3779 Mutex::Autolock _l(mLock); 3780 // ignore unexpected callbacks 3781 if (mWriteAckSequence & 2) { 3782 mWriteAckSequence |= 1; 3783 mWaitWorkCV.signal(); 3784 } 3785} 3786 3787void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3788{ 3789 Mutex::Autolock _l(mLock); 3790 // bit 0 is cleared 3791 mDrainSequence = sequence << 1; 3792} 3793 3794void AudioFlinger::AsyncCallbackThread::resetDraining() 3795{ 3796 Mutex::Autolock _l(mLock); 3797 // ignore unexpected callbacks 3798 if (mDrainSequence & 2) { 3799 mDrainSequence |= 1; 3800 mWaitWorkCV.signal(); 3801 } 3802} 3803 3804 3805// ---------------------------------------------------------------------------- 3806AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3807 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3808 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3809 mHwPaused(false), 3810 mPausedBytesRemaining(0) 3811{ 3812 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3813} 3814 3815AudioFlinger::OffloadThread::~OffloadThread() 3816{ 3817 mPreviousTrack.clear(); 3818} 3819 3820void AudioFlinger::OffloadThread::threadLoop_exit() 3821{ 3822 if (mFlushPending || mHwPaused) { 3823 // If a flush is pending or track was paused, just discard buffered data 3824 flushHw_l(); 3825 } else { 3826 mMixerStatus = MIXER_DRAIN_ALL; 3827 threadLoop_drain(); 3828 } 3829 mCallbackThread->exit(); 3830 PlaybackThread::threadLoop_exit(); 3831} 3832 3833AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3834 Vector< sp<Track> > *tracksToRemove 3835) 3836{ 3837 ALOGV("OffloadThread::prepareTracks_l"); 3838 size_t count = mActiveTracks.size(); 3839 3840 mixer_state mixerStatus = MIXER_IDLE; 3841 // find out which tracks need to be processed 3842 for (size_t i = 0; i < count; i++) { 3843 sp<Track> t = mActiveTracks[i].promote(); 3844 // The track died recently 3845 if (t == 0) { 3846 continue; 3847 } 3848 Track* const track = t.get(); 3849 audio_track_cblk_t* cblk = track->cblk(); 3850 if (mPreviousTrack != NULL) { 3851 if (t != mPreviousTrack) { 3852 // Flush any data still being written from last track 3853 mBytesRemaining = 0; 3854 if (mPausedBytesRemaining) { 3855 // Last track was paused so we also need to flush saved 3856 // mixbuffer state and invalidate track so that it will 3857 // re-submit that unwritten data when it is next resumed 3858 mPausedBytesRemaining = 0; 3859 // Invalidate is a bit drastic - would be more efficient 3860 // to have a flag to tell client that some of the 3861 // previously written data was lost 3862 mPreviousTrack->invalidate(); 3863 } 3864 } 3865 } 3866 mPreviousTrack = t; 3867 bool last = (i == (count - 1)); 3868 if (track->isPausing()) { 3869 track->setPaused(); 3870 if (last) { 3871 if (!mHwPaused) { 3872 mOutput->stream->pause(mOutput->stream); 3873 mHwPaused = true; 3874 } 3875 // If we were part way through writing the mixbuffer to 3876 // the HAL we must save this until we resume 3877 // BUG - this will be wrong if a different track is made active, 3878 // in that case we want to discard the pending data in the 3879 // mixbuffer and tell the client to present it again when the 3880 // track is resumed 3881 mPausedWriteLength = mCurrentWriteLength; 3882 mPausedBytesRemaining = mBytesRemaining; 3883 mBytesRemaining = 0; // stop writing 3884 } 3885 tracksToRemove->add(track); 3886 } else if (track->framesReady() && track->isReady() && 3887 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3888 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3889 if (track->mFillingUpStatus == Track::FS_FILLED) { 3890 track->mFillingUpStatus = Track::FS_ACTIVE; 3891 // make sure processVolume_l() will apply new volume even if 0 3892 mLeftVolFloat = mRightVolFloat = -1.0; 3893 if (track->mState == TrackBase::RESUMING) { 3894 if (mPausedBytesRemaining) { 3895 // Need to continue write that was interrupted 3896 mCurrentWriteLength = mPausedWriteLength; 3897 mBytesRemaining = mPausedBytesRemaining; 3898 mPausedBytesRemaining = 0; 3899 } 3900 track->mState = TrackBase::ACTIVE; 3901 } 3902 } 3903 3904 if (last) { 3905 if (mHwPaused) { 3906 mOutput->stream->resume(mOutput->stream); 3907 mHwPaused = false; 3908 // threadLoop_mix() will handle the case that we need to 3909 // resume an interrupted write 3910 } 3911 // reset retry count 3912 track->mRetryCount = kMaxTrackRetriesOffload; 3913 mActiveTrack = t; 3914 mixerStatus = MIXER_TRACKS_READY; 3915 } 3916 } else { 3917 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3918 if (track->isStopping_1()) { 3919 // Hardware buffer can hold a large amount of audio so we must 3920 // wait for all current track's data to drain before we say 3921 // that the track is stopped. 3922 if (mBytesRemaining == 0) { 3923 // Only start draining when all data in mixbuffer 3924 // has been written 3925 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3926 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3927 sleepTime = 0; 3928 standbyTime = systemTime() + standbyDelay; 3929 if (last) { 3930 mixerStatus = MIXER_DRAIN_TRACK; 3931 mDrainSequence += 2; 3932 if (mHwPaused) { 3933 // It is possible to move from PAUSED to STOPPING_1 without 3934 // a resume so we must ensure hardware is running 3935 mOutput->stream->resume(mOutput->stream); 3936 mHwPaused = false; 3937 } 3938 } 3939 } 3940 } else if (track->isStopping_2()) { 3941 // Drain has completed, signal presentation complete 3942 if (!(mDrainSequence & 1) || !last) { 3943 track->mState = TrackBase::STOPPED; 3944 size_t audioHALFrames = 3945 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3946 size_t framesWritten = 3947 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3948 track->presentationComplete(framesWritten, audioHALFrames); 3949 track->reset(); 3950 tracksToRemove->add(track); 3951 } 3952 } else { 3953 // No buffers for this track. Give it a few chances to 3954 // fill a buffer, then remove it from active list. 3955 if (--(track->mRetryCount) <= 0) { 3956 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3957 track->name()); 3958 tracksToRemove->add(track); 3959 } else if (last){ 3960 mixerStatus = MIXER_TRACKS_ENABLED; 3961 } 3962 } 3963 } 3964 // compute volume for this track 3965 processVolume_l(track, last); 3966 } 3967 3968 if (mFlushPending) { 3969 flushHw_l(); 3970 mFlushPending = false; 3971 } 3972 3973 // remove all the tracks that need to be... 3974 removeTracks_l(*tracksToRemove); 3975 3976 return mixerStatus; 3977} 3978 3979void AudioFlinger::OffloadThread::flushOutput_l() 3980{ 3981 mFlushPending = true; 3982} 3983 3984// must be called with thread mutex locked 3985bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3986{ 3987 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 3988 mWriteAckSequence, mDrainSequence); 3989 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 3990 return true; 3991 } 3992 return false; 3993} 3994 3995// must be called with thread mutex locked 3996bool AudioFlinger::OffloadThread::shouldStandby_l() 3997{ 3998 bool TrackPaused = false; 3999 4000 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4001 // after a timeout and we will enter standby then. 4002 if (mTracks.size() > 0) { 4003 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4004 } 4005 4006 return !mStandby && !TrackPaused; 4007} 4008 4009 4010bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4011{ 4012 Mutex::Autolock _l(mLock); 4013 return waitingAsyncCallback_l(); 4014} 4015 4016void AudioFlinger::OffloadThread::flushHw_l() 4017{ 4018 mOutput->stream->flush(mOutput->stream); 4019 // Flush anything still waiting in the mixbuffer 4020 mCurrentWriteLength = 0; 4021 mBytesRemaining = 0; 4022 mPausedWriteLength = 0; 4023 mPausedBytesRemaining = 0; 4024 if (mUseAsyncWrite) { 4025 // discard any pending drain or write ack by incrementing sequence 4026 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4027 mDrainSequence = (mDrainSequence + 2) & ~1; 4028 ALOG_ASSERT(mCallbackThread != 0); 4029 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4030 mCallbackThread->setDraining(mDrainSequence); 4031 } 4032} 4033 4034// ---------------------------------------------------------------------------- 4035 4036AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4037 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4038 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4039 DUPLICATING), 4040 mWaitTimeMs(UINT_MAX) 4041{ 4042 addOutputTrack(mainThread); 4043} 4044 4045AudioFlinger::DuplicatingThread::~DuplicatingThread() 4046{ 4047 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4048 mOutputTracks[i]->destroy(); 4049 } 4050} 4051 4052void AudioFlinger::DuplicatingThread::threadLoop_mix() 4053{ 4054 // mix buffers... 4055 if (outputsReady(outputTracks)) { 4056 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4057 } else { 4058 memset(mMixBuffer, 0, mixBufferSize); 4059 } 4060 sleepTime = 0; 4061 writeFrames = mNormalFrameCount; 4062 mCurrentWriteLength = mixBufferSize; 4063 standbyTime = systemTime() + standbyDelay; 4064} 4065 4066void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4067{ 4068 if (sleepTime == 0) { 4069 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4070 sleepTime = activeSleepTime; 4071 } else { 4072 sleepTime = idleSleepTime; 4073 } 4074 } else if (mBytesWritten != 0) { 4075 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4076 writeFrames = mNormalFrameCount; 4077 memset(mMixBuffer, 0, mixBufferSize); 4078 } else { 4079 // flush remaining overflow buffers in output tracks 4080 writeFrames = 0; 4081 } 4082 sleepTime = 0; 4083 } 4084} 4085 4086ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4087{ 4088 for (size_t i = 0; i < outputTracks.size(); i++) { 4089 outputTracks[i]->write(mMixBuffer, writeFrames); 4090 } 4091 return (ssize_t)mixBufferSize; 4092} 4093 4094void AudioFlinger::DuplicatingThread::threadLoop_standby() 4095{ 4096 // DuplicatingThread implements standby by stopping all tracks 4097 for (size_t i = 0; i < outputTracks.size(); i++) { 4098 outputTracks[i]->stop(); 4099 } 4100} 4101 4102void AudioFlinger::DuplicatingThread::saveOutputTracks() 4103{ 4104 outputTracks = mOutputTracks; 4105} 4106 4107void AudioFlinger::DuplicatingThread::clearOutputTracks() 4108{ 4109 outputTracks.clear(); 4110} 4111 4112void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4113{ 4114 Mutex::Autolock _l(mLock); 4115 // FIXME explain this formula 4116 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4117 OutputTrack *outputTrack = new OutputTrack(thread, 4118 this, 4119 mSampleRate, 4120 mFormat, 4121 mChannelMask, 4122 frameCount); 4123 if (outputTrack->cblk() != NULL) { 4124 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4125 mOutputTracks.add(outputTrack); 4126 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4127 updateWaitTime_l(); 4128 } 4129} 4130 4131void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4132{ 4133 Mutex::Autolock _l(mLock); 4134 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4135 if (mOutputTracks[i]->thread() == thread) { 4136 mOutputTracks[i]->destroy(); 4137 mOutputTracks.removeAt(i); 4138 updateWaitTime_l(); 4139 return; 4140 } 4141 } 4142 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4143} 4144 4145// caller must hold mLock 4146void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4147{ 4148 mWaitTimeMs = UINT_MAX; 4149 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4150 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4151 if (strong != 0) { 4152 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4153 if (waitTimeMs < mWaitTimeMs) { 4154 mWaitTimeMs = waitTimeMs; 4155 } 4156 } 4157 } 4158} 4159 4160 4161bool AudioFlinger::DuplicatingThread::outputsReady( 4162 const SortedVector< sp<OutputTrack> > &outputTracks) 4163{ 4164 for (size_t i = 0; i < outputTracks.size(); i++) { 4165 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4166 if (thread == 0) { 4167 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4168 outputTracks[i].get()); 4169 return false; 4170 } 4171 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4172 // see note at standby() declaration 4173 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4174 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4175 thread.get()); 4176 return false; 4177 } 4178 } 4179 return true; 4180} 4181 4182uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4183{ 4184 return (mWaitTimeMs * 1000) / 2; 4185} 4186 4187void AudioFlinger::DuplicatingThread::cacheParameters_l() 4188{ 4189 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4190 updateWaitTime_l(); 4191 4192 MixerThread::cacheParameters_l(); 4193} 4194 4195// ---------------------------------------------------------------------------- 4196// Record 4197// ---------------------------------------------------------------------------- 4198 4199AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4200 AudioStreamIn *input, 4201 uint32_t sampleRate, 4202 audio_channel_mask_t channelMask, 4203 audio_io_handle_t id, 4204 audio_devices_t outDevice, 4205 audio_devices_t inDevice 4206#ifdef TEE_SINK 4207 , const sp<NBAIO_Sink>& teeSink 4208#endif 4209 ) : 4210 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4211 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4212 // mRsmpInIndex and mBufferSize set by readInputParameters() 4213 mReqChannelCount(popcount(channelMask)), 4214 mReqSampleRate(sampleRate) 4215 // mBytesRead is only meaningful while active, and so is cleared in start() 4216 // (but might be better to also clear here for dump?) 4217#ifdef TEE_SINK 4218 , mTeeSink(teeSink) 4219#endif 4220{ 4221 snprintf(mName, kNameLength, "AudioIn_%X", id); 4222 4223 readInputParameters(); 4224 4225} 4226 4227 4228AudioFlinger::RecordThread::~RecordThread() 4229{ 4230 delete[] mRsmpInBuffer; 4231 delete mResampler; 4232 delete[] mRsmpOutBuffer; 4233} 4234 4235void AudioFlinger::RecordThread::onFirstRef() 4236{ 4237 run(mName, PRIORITY_URGENT_AUDIO); 4238} 4239 4240status_t AudioFlinger::RecordThread::readyToRun() 4241{ 4242 status_t status = initCheck(); 4243 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4244 return status; 4245} 4246 4247bool AudioFlinger::RecordThread::threadLoop() 4248{ 4249 AudioBufferProvider::Buffer buffer; 4250 sp<RecordTrack> activeTrack; 4251 Vector< sp<EffectChain> > effectChains; 4252 4253 nsecs_t lastWarning = 0; 4254 4255 inputStandBy(); 4256 acquireWakeLock(); 4257 4258 // used to verify we've read at least once before evaluating how many bytes were read 4259 bool readOnce = false; 4260 4261 // start recording 4262 while (!exitPending()) { 4263 4264 processConfigEvents(); 4265 4266 { // scope for mLock 4267 Mutex::Autolock _l(mLock); 4268 checkForNewParameters_l(); 4269 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4270 standby(); 4271 4272 if (exitPending()) { 4273 break; 4274 } 4275 4276 releaseWakeLock_l(); 4277 ALOGV("RecordThread: loop stopping"); 4278 // go to sleep 4279 mWaitWorkCV.wait(mLock); 4280 ALOGV("RecordThread: loop starting"); 4281 acquireWakeLock_l(); 4282 continue; 4283 } 4284 if (mActiveTrack != 0) { 4285 if (mActiveTrack->isTerminated()) { 4286 removeTrack_l(mActiveTrack); 4287 mActiveTrack.clear(); 4288 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4289 standby(); 4290 mActiveTrack.clear(); 4291 mStartStopCond.broadcast(); 4292 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4293 if (mReqChannelCount != mActiveTrack->channelCount()) { 4294 mActiveTrack.clear(); 4295 mStartStopCond.broadcast(); 4296 } else if (readOnce) { 4297 // record start succeeds only if first read from audio input 4298 // succeeds 4299 if (mBytesRead >= 0) { 4300 mActiveTrack->mState = TrackBase::ACTIVE; 4301 } else { 4302 mActiveTrack.clear(); 4303 } 4304 mStartStopCond.broadcast(); 4305 } 4306 mStandby = false; 4307 } 4308 } 4309 lockEffectChains_l(effectChains); 4310 } 4311 4312 if (mActiveTrack != 0) { 4313 if (mActiveTrack->mState != TrackBase::ACTIVE && 4314 mActiveTrack->mState != TrackBase::RESUMING) { 4315 unlockEffectChains(effectChains); 4316 usleep(kRecordThreadSleepUs); 4317 continue; 4318 } 4319 for (size_t i = 0; i < effectChains.size(); i ++) { 4320 effectChains[i]->process_l(); 4321 } 4322 4323 buffer.frameCount = mFrameCount; 4324 status_t status = mActiveTrack->getNextBuffer(&buffer); 4325 if (status == NO_ERROR) { 4326 readOnce = true; 4327 size_t framesOut = buffer.frameCount; 4328 if (mResampler == NULL) { 4329 // no resampling 4330 while (framesOut) { 4331 size_t framesIn = mFrameCount - mRsmpInIndex; 4332 if (framesIn) { 4333 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4334 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4335 mActiveTrack->mFrameSize; 4336 if (framesIn > framesOut) 4337 framesIn = framesOut; 4338 mRsmpInIndex += framesIn; 4339 framesOut -= framesIn; 4340 if (mChannelCount == mReqChannelCount) { 4341 memcpy(dst, src, framesIn * mFrameSize); 4342 } else { 4343 if (mChannelCount == 1) { 4344 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4345 (int16_t *)src, framesIn); 4346 } else { 4347 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4348 (int16_t *)src, framesIn); 4349 } 4350 } 4351 } 4352 if (framesOut && mFrameCount == mRsmpInIndex) { 4353 void *readInto; 4354 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4355 readInto = buffer.raw; 4356 framesOut = 0; 4357 } else { 4358 readInto = mRsmpInBuffer; 4359 mRsmpInIndex = 0; 4360 } 4361 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4362 mBufferSize); 4363 if (mBytesRead <= 0) { 4364 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4365 { 4366 ALOGE("Error reading audio input"); 4367 // Force input into standby so that it tries to 4368 // recover at next read attempt 4369 inputStandBy(); 4370 usleep(kRecordThreadSleepUs); 4371 } 4372 mRsmpInIndex = mFrameCount; 4373 framesOut = 0; 4374 buffer.frameCount = 0; 4375 } 4376#ifdef TEE_SINK 4377 else if (mTeeSink != 0) { 4378 (void) mTeeSink->write(readInto, 4379 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4380 } 4381#endif 4382 } 4383 } 4384 } else { 4385 // resampling 4386 4387 // resampler accumulates, but we only have one source track 4388 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4389 // alter output frame count as if we were expecting stereo samples 4390 if (mChannelCount == 1 && mReqChannelCount == 1) { 4391 framesOut >>= 1; 4392 } 4393 mResampler->resample(mRsmpOutBuffer, framesOut, 4394 this /* AudioBufferProvider* */); 4395 // ditherAndClamp() works as long as all buffers returned by 4396 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4397 if (mChannelCount == 2 && mReqChannelCount == 1) { 4398 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4399 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4400 // the resampler always outputs stereo samples: 4401 // do post stereo to mono conversion 4402 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4403 framesOut); 4404 } else { 4405 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4406 } 4407 // now done with mRsmpOutBuffer 4408 4409 } 4410 if (mFramestoDrop == 0) { 4411 mActiveTrack->releaseBuffer(&buffer); 4412 } else { 4413 if (mFramestoDrop > 0) { 4414 mFramestoDrop -= buffer.frameCount; 4415 if (mFramestoDrop <= 0) { 4416 clearSyncStartEvent(); 4417 } 4418 } else { 4419 mFramestoDrop += buffer.frameCount; 4420 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4421 mSyncStartEvent->isCancelled()) { 4422 ALOGW("Synced record %s, session %d, trigger session %d", 4423 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4424 mActiveTrack->sessionId(), 4425 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4426 clearSyncStartEvent(); 4427 } 4428 } 4429 } 4430 mActiveTrack->clearOverflow(); 4431 } 4432 // client isn't retrieving buffers fast enough 4433 else { 4434 if (!mActiveTrack->setOverflow()) { 4435 nsecs_t now = systemTime(); 4436 if ((now - lastWarning) > kWarningThrottleNs) { 4437 ALOGW("RecordThread: buffer overflow"); 4438 lastWarning = now; 4439 } 4440 } 4441 // Release the processor for a while before asking for a new buffer. 4442 // This will give the application more chance to read from the buffer and 4443 // clear the overflow. 4444 usleep(kRecordThreadSleepUs); 4445 } 4446 } 4447 // enable changes in effect chain 4448 unlockEffectChains(effectChains); 4449 effectChains.clear(); 4450 } 4451 4452 standby(); 4453 4454 { 4455 Mutex::Autolock _l(mLock); 4456 for (size_t i = 0; i < mTracks.size(); i++) { 4457 sp<RecordTrack> track = mTracks[i]; 4458 track->invalidate(); 4459 } 4460 mActiveTrack.clear(); 4461 mStartStopCond.broadcast(); 4462 } 4463 4464 releaseWakeLock(); 4465 4466 ALOGV("RecordThread %p exiting", this); 4467 return false; 4468} 4469 4470void AudioFlinger::RecordThread::standby() 4471{ 4472 if (!mStandby) { 4473 inputStandBy(); 4474 mStandby = true; 4475 } 4476} 4477 4478void AudioFlinger::RecordThread::inputStandBy() 4479{ 4480 mInput->stream->common.standby(&mInput->stream->common); 4481} 4482 4483sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4484 const sp<AudioFlinger::Client>& client, 4485 uint32_t sampleRate, 4486 audio_format_t format, 4487 audio_channel_mask_t channelMask, 4488 size_t frameCount, 4489 int sessionId, 4490 IAudioFlinger::track_flags_t *flags, 4491 pid_t tid, 4492 status_t *status) 4493{ 4494 sp<RecordTrack> track; 4495 status_t lStatus; 4496 4497 lStatus = initCheck(); 4498 if (lStatus != NO_ERROR) { 4499 ALOGE("Audio driver not initialized."); 4500 goto Exit; 4501 } 4502 4503 // client expresses a preference for FAST, but we get the final say 4504 if (*flags & IAudioFlinger::TRACK_FAST) { 4505 if ( 4506 // use case: callback handler and frame count is default or at least as large as HAL 4507 ( 4508 (tid != -1) && 4509 ((frameCount == 0) || 4510 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4511 ) && 4512 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4513 // mono or stereo 4514 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4515 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4516 // hardware sample rate 4517 (sampleRate == mSampleRate) && 4518 // record thread has an associated fast recorder 4519 hasFastRecorder() 4520 // FIXME test that RecordThread for this fast track has a capable output HAL 4521 // FIXME add a permission test also? 4522 ) { 4523 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4524 if (frameCount == 0) { 4525 frameCount = mFrameCount * kFastTrackMultiplier; 4526 } 4527 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4528 frameCount, mFrameCount); 4529 } else { 4530 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4531 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4532 "hasFastRecorder=%d tid=%d", 4533 frameCount, mFrameCount, format, 4534 audio_is_linear_pcm(format), 4535 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4536 *flags &= ~IAudioFlinger::TRACK_FAST; 4537 // For compatibility with AudioRecord calculation, buffer depth is forced 4538 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4539 // This is probably too conservative, but legacy application code may depend on it. 4540 // If you change this calculation, also review the start threshold which is related. 4541 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4542 size_t mNormalFrameCount = 2048; // FIXME 4543 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4544 if (minBufCount < 2) { 4545 minBufCount = 2; 4546 } 4547 size_t minFrameCount = mNormalFrameCount * minBufCount; 4548 if (frameCount < minFrameCount) { 4549 frameCount = minFrameCount; 4550 } 4551 } 4552 } 4553 4554 // FIXME use flags and tid similar to createTrack_l() 4555 4556 { // scope for mLock 4557 Mutex::Autolock _l(mLock); 4558 4559 track = new RecordTrack(this, client, sampleRate, 4560 format, channelMask, frameCount, sessionId); 4561 4562 if (track->getCblk() == 0) { 4563 lStatus = NO_MEMORY; 4564 goto Exit; 4565 } 4566 mTracks.add(track); 4567 4568 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4569 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4570 mAudioFlinger->btNrecIsOff(); 4571 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4572 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4573 4574 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4575 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4576 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4577 // so ask activity manager to do this on our behalf 4578 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4579 } 4580 } 4581 lStatus = NO_ERROR; 4582 4583Exit: 4584 if (status) { 4585 *status = lStatus; 4586 } 4587 return track; 4588} 4589 4590status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4591 AudioSystem::sync_event_t event, 4592 int triggerSession) 4593{ 4594 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4595 sp<ThreadBase> strongMe = this; 4596 status_t status = NO_ERROR; 4597 4598 if (event == AudioSystem::SYNC_EVENT_NONE) { 4599 clearSyncStartEvent(); 4600 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4601 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4602 triggerSession, 4603 recordTrack->sessionId(), 4604 syncStartEventCallback, 4605 this); 4606 // Sync event can be cancelled by the trigger session if the track is not in a 4607 // compatible state in which case we start record immediately 4608 if (mSyncStartEvent->isCancelled()) { 4609 clearSyncStartEvent(); 4610 } else { 4611 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4612 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4613 } 4614 } 4615 4616 { 4617 AutoMutex lock(mLock); 4618 if (mActiveTrack != 0) { 4619 if (recordTrack != mActiveTrack.get()) { 4620 status = -EBUSY; 4621 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4622 mActiveTrack->mState = TrackBase::ACTIVE; 4623 } 4624 return status; 4625 } 4626 4627 recordTrack->mState = TrackBase::IDLE; 4628 mActiveTrack = recordTrack; 4629 mLock.unlock(); 4630 status_t status = AudioSystem::startInput(mId); 4631 mLock.lock(); 4632 if (status != NO_ERROR) { 4633 mActiveTrack.clear(); 4634 clearSyncStartEvent(); 4635 return status; 4636 } 4637 mRsmpInIndex = mFrameCount; 4638 mBytesRead = 0; 4639 if (mResampler != NULL) { 4640 mResampler->reset(); 4641 } 4642 mActiveTrack->mState = TrackBase::RESUMING; 4643 // signal thread to start 4644 ALOGV("Signal record thread"); 4645 mWaitWorkCV.broadcast(); 4646 // do not wait for mStartStopCond if exiting 4647 if (exitPending()) { 4648 mActiveTrack.clear(); 4649 status = INVALID_OPERATION; 4650 goto startError; 4651 } 4652 mStartStopCond.wait(mLock); 4653 if (mActiveTrack == 0) { 4654 ALOGV("Record failed to start"); 4655 status = BAD_VALUE; 4656 goto startError; 4657 } 4658 ALOGV("Record started OK"); 4659 return status; 4660 } 4661 4662startError: 4663 AudioSystem::stopInput(mId); 4664 clearSyncStartEvent(); 4665 return status; 4666} 4667 4668void AudioFlinger::RecordThread::clearSyncStartEvent() 4669{ 4670 if (mSyncStartEvent != 0) { 4671 mSyncStartEvent->cancel(); 4672 } 4673 mSyncStartEvent.clear(); 4674 mFramestoDrop = 0; 4675} 4676 4677void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4678{ 4679 sp<SyncEvent> strongEvent = event.promote(); 4680 4681 if (strongEvent != 0) { 4682 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4683 me->handleSyncStartEvent(strongEvent); 4684 } 4685} 4686 4687void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4688{ 4689 if (event == mSyncStartEvent) { 4690 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4691 // from audio HAL 4692 mFramestoDrop = mFrameCount * 2; 4693 } 4694} 4695 4696bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4697 ALOGV("RecordThread::stop"); 4698 AutoMutex _l(mLock); 4699 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4700 return false; 4701 } 4702 recordTrack->mState = TrackBase::PAUSING; 4703 // do not wait for mStartStopCond if exiting 4704 if (exitPending()) { 4705 return true; 4706 } 4707 mStartStopCond.wait(mLock); 4708 // if we have been restarted, recordTrack == mActiveTrack.get() here 4709 if (exitPending() || recordTrack != mActiveTrack.get()) { 4710 ALOGV("Record stopped OK"); 4711 return true; 4712 } 4713 return false; 4714} 4715 4716bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4717{ 4718 return false; 4719} 4720 4721status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4722{ 4723#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4724 if (!isValidSyncEvent(event)) { 4725 return BAD_VALUE; 4726 } 4727 4728 int eventSession = event->triggerSession(); 4729 status_t ret = NAME_NOT_FOUND; 4730 4731 Mutex::Autolock _l(mLock); 4732 4733 for (size_t i = 0; i < mTracks.size(); i++) { 4734 sp<RecordTrack> track = mTracks[i]; 4735 if (eventSession == track->sessionId()) { 4736 (void) track->setSyncEvent(event); 4737 ret = NO_ERROR; 4738 } 4739 } 4740 return ret; 4741#else 4742 return BAD_VALUE; 4743#endif 4744} 4745 4746// destroyTrack_l() must be called with ThreadBase::mLock held 4747void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4748{ 4749 track->terminate(); 4750 track->mState = TrackBase::STOPPED; 4751 // active tracks are removed by threadLoop() 4752 if (mActiveTrack != track) { 4753 removeTrack_l(track); 4754 } 4755} 4756 4757void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4758{ 4759 mTracks.remove(track); 4760 // need anything related to effects here? 4761} 4762 4763void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4764{ 4765 dumpInternals(fd, args); 4766 dumpTracks(fd, args); 4767 dumpEffectChains(fd, args); 4768} 4769 4770void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4771{ 4772 const size_t SIZE = 256; 4773 char buffer[SIZE]; 4774 String8 result; 4775 4776 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4777 result.append(buffer); 4778 4779 if (mActiveTrack != 0) { 4780 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4781 result.append(buffer); 4782 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4783 result.append(buffer); 4784 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4785 result.append(buffer); 4786 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4787 result.append(buffer); 4788 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4789 result.append(buffer); 4790 } else { 4791 result.append("No active record client\n"); 4792 } 4793 4794 write(fd, result.string(), result.size()); 4795 4796 dumpBase(fd, args); 4797} 4798 4799void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4800{ 4801 const size_t SIZE = 256; 4802 char buffer[SIZE]; 4803 String8 result; 4804 4805 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4806 result.append(buffer); 4807 RecordTrack::appendDumpHeader(result); 4808 for (size_t i = 0; i < mTracks.size(); ++i) { 4809 sp<RecordTrack> track = mTracks[i]; 4810 if (track != 0) { 4811 track->dump(buffer, SIZE); 4812 result.append(buffer); 4813 } 4814 } 4815 4816 if (mActiveTrack != 0) { 4817 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4818 result.append(buffer); 4819 RecordTrack::appendDumpHeader(result); 4820 mActiveTrack->dump(buffer, SIZE); 4821 result.append(buffer); 4822 4823 } 4824 write(fd, result.string(), result.size()); 4825} 4826 4827// AudioBufferProvider interface 4828status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4829{ 4830 size_t framesReq = buffer->frameCount; 4831 size_t framesReady = mFrameCount - mRsmpInIndex; 4832 int channelCount; 4833 4834 if (framesReady == 0) { 4835 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4836 if (mBytesRead <= 0) { 4837 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4838 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4839 // Force input into standby so that it tries to 4840 // recover at next read attempt 4841 inputStandBy(); 4842 usleep(kRecordThreadSleepUs); 4843 } 4844 buffer->raw = NULL; 4845 buffer->frameCount = 0; 4846 return NOT_ENOUGH_DATA; 4847 } 4848 mRsmpInIndex = 0; 4849 framesReady = mFrameCount; 4850 } 4851 4852 if (framesReq > framesReady) { 4853 framesReq = framesReady; 4854 } 4855 4856 if (mChannelCount == 1 && mReqChannelCount == 2) { 4857 channelCount = 1; 4858 } else { 4859 channelCount = 2; 4860 } 4861 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4862 buffer->frameCount = framesReq; 4863 return NO_ERROR; 4864} 4865 4866// AudioBufferProvider interface 4867void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4868{ 4869 mRsmpInIndex += buffer->frameCount; 4870 buffer->frameCount = 0; 4871} 4872 4873bool AudioFlinger::RecordThread::checkForNewParameters_l() 4874{ 4875 bool reconfig = false; 4876 4877 while (!mNewParameters.isEmpty()) { 4878 status_t status = NO_ERROR; 4879 String8 keyValuePair = mNewParameters[0]; 4880 AudioParameter param = AudioParameter(keyValuePair); 4881 int value; 4882 audio_format_t reqFormat = mFormat; 4883 uint32_t reqSamplingRate = mReqSampleRate; 4884 uint32_t reqChannelCount = mReqChannelCount; 4885 4886 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4887 reqSamplingRate = value; 4888 reconfig = true; 4889 } 4890 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4891 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4892 status = BAD_VALUE; 4893 } else { 4894 reqFormat = (audio_format_t) value; 4895 reconfig = true; 4896 } 4897 } 4898 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4899 reqChannelCount = popcount(value); 4900 reconfig = true; 4901 } 4902 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4903 // do not accept frame count changes if tracks are open as the track buffer 4904 // size depends on frame count and correct behavior would not be guaranteed 4905 // if frame count is changed after track creation 4906 if (mActiveTrack != 0) { 4907 status = INVALID_OPERATION; 4908 } else { 4909 reconfig = true; 4910 } 4911 } 4912 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4913 // forward device change to effects that have requested to be 4914 // aware of attached audio device. 4915 for (size_t i = 0; i < mEffectChains.size(); i++) { 4916 mEffectChains[i]->setDevice_l(value); 4917 } 4918 4919 // store input device and output device but do not forward output device to audio HAL. 4920 // Note that status is ignored by the caller for output device 4921 // (see AudioFlinger::setParameters() 4922 if (audio_is_output_devices(value)) { 4923 mOutDevice = value; 4924 status = BAD_VALUE; 4925 } else { 4926 mInDevice = value; 4927 // disable AEC and NS if the device is a BT SCO headset supporting those 4928 // pre processings 4929 if (mTracks.size() > 0) { 4930 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4931 mAudioFlinger->btNrecIsOff(); 4932 for (size_t i = 0; i < mTracks.size(); i++) { 4933 sp<RecordTrack> track = mTracks[i]; 4934 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4935 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4936 } 4937 } 4938 } 4939 } 4940 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4941 mAudioSource != (audio_source_t)value) { 4942 // forward device change to effects that have requested to be 4943 // aware of attached audio device. 4944 for (size_t i = 0; i < mEffectChains.size(); i++) { 4945 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4946 } 4947 mAudioSource = (audio_source_t)value; 4948 } 4949 if (status == NO_ERROR) { 4950 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4951 keyValuePair.string()); 4952 if (status == INVALID_OPERATION) { 4953 inputStandBy(); 4954 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4955 keyValuePair.string()); 4956 } 4957 if (reconfig) { 4958 if (status == BAD_VALUE && 4959 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4960 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4961 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4962 <= (2 * reqSamplingRate)) && 4963 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4964 <= FCC_2 && 4965 (reqChannelCount <= FCC_2)) { 4966 status = NO_ERROR; 4967 } 4968 if (status == NO_ERROR) { 4969 readInputParameters(); 4970 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4971 } 4972 } 4973 } 4974 4975 mNewParameters.removeAt(0); 4976 4977 mParamStatus = status; 4978 mParamCond.signal(); 4979 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4980 // already timed out waiting for the status and will never signal the condition. 4981 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4982 } 4983 return reconfig; 4984} 4985 4986String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4987{ 4988 Mutex::Autolock _l(mLock); 4989 if (initCheck() != NO_ERROR) { 4990 return String8(); 4991 } 4992 4993 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4994 const String8 out_s8(s); 4995 free(s); 4996 return out_s8; 4997} 4998 4999void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5000 AudioSystem::OutputDescriptor desc; 5001 void *param2 = NULL; 5002 5003 switch (event) { 5004 case AudioSystem::INPUT_OPENED: 5005 case AudioSystem::INPUT_CONFIG_CHANGED: 5006 desc.channelMask = mChannelMask; 5007 desc.samplingRate = mSampleRate; 5008 desc.format = mFormat; 5009 desc.frameCount = mFrameCount; 5010 desc.latency = 0; 5011 param2 = &desc; 5012 break; 5013 5014 case AudioSystem::INPUT_CLOSED: 5015 default: 5016 break; 5017 } 5018 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5019} 5020 5021void AudioFlinger::RecordThread::readInputParameters() 5022{ 5023 delete[] mRsmpInBuffer; 5024 // mRsmpInBuffer is always assigned a new[] below 5025 delete[] mRsmpOutBuffer; 5026 mRsmpOutBuffer = NULL; 5027 delete mResampler; 5028 mResampler = NULL; 5029 5030 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5031 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5032 mChannelCount = popcount(mChannelMask); 5033 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5034 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5035 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5036 } 5037 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5038 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5039 mFrameCount = mBufferSize / mFrameSize; 5040 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5041 5042 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5043 { 5044 int channelCount; 5045 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5046 // stereo to mono post process as the resampler always outputs stereo. 5047 if (mChannelCount == 1 && mReqChannelCount == 2) { 5048 channelCount = 1; 5049 } else { 5050 channelCount = 2; 5051 } 5052 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5053 mResampler->setSampleRate(mSampleRate); 5054 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5055 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5056 5057 // optmization: if mono to mono, alter input frame count as if we were inputing 5058 // stereo samples 5059 if (mChannelCount == 1 && mReqChannelCount == 1) { 5060 mFrameCount >>= 1; 5061 } 5062 5063 } 5064 mRsmpInIndex = mFrameCount; 5065} 5066 5067unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5068{ 5069 Mutex::Autolock _l(mLock); 5070 if (initCheck() != NO_ERROR) { 5071 return 0; 5072 } 5073 5074 return mInput->stream->get_input_frames_lost(mInput->stream); 5075} 5076 5077uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5078{ 5079 Mutex::Autolock _l(mLock); 5080 uint32_t result = 0; 5081 if (getEffectChain_l(sessionId) != 0) { 5082 result = EFFECT_SESSION; 5083 } 5084 5085 for (size_t i = 0; i < mTracks.size(); ++i) { 5086 if (sessionId == mTracks[i]->sessionId()) { 5087 result |= TRACK_SESSION; 5088 break; 5089 } 5090 } 5091 5092 return result; 5093} 5094 5095KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5096{ 5097 KeyedVector<int, bool> ids; 5098 Mutex::Autolock _l(mLock); 5099 for (size_t j = 0; j < mTracks.size(); ++j) { 5100 sp<RecordThread::RecordTrack> track = mTracks[j]; 5101 int sessionId = track->sessionId(); 5102 if (ids.indexOfKey(sessionId) < 0) { 5103 ids.add(sessionId, true); 5104 } 5105 } 5106 return ids; 5107} 5108 5109AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5110{ 5111 Mutex::Autolock _l(mLock); 5112 AudioStreamIn *input = mInput; 5113 mInput = NULL; 5114 return input; 5115} 5116 5117// this method must always be called either with ThreadBase mLock held or inside the thread loop 5118audio_stream_t* AudioFlinger::RecordThread::stream() const 5119{ 5120 if (mInput == NULL) { 5121 return NULL; 5122 } 5123 return &mInput->stream->common; 5124} 5125 5126status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5127{ 5128 // only one chain per input thread 5129 if (mEffectChains.size() != 0) { 5130 return INVALID_OPERATION; 5131 } 5132 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5133 5134 chain->setInBuffer(NULL); 5135 chain->setOutBuffer(NULL); 5136 5137 checkSuspendOnAddEffectChain_l(chain); 5138 5139 mEffectChains.add(chain); 5140 5141 return NO_ERROR; 5142} 5143 5144size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5145{ 5146 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5147 ALOGW_IF(mEffectChains.size() != 1, 5148 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5149 chain.get(), mEffectChains.size(), this); 5150 if (mEffectChains.size() == 1) { 5151 mEffectChains.removeAt(0); 5152 } 5153 return 0; 5154} 5155 5156}; // namespace android 5157