Threads.cpp revision e14a5d6d2cc91dd2fc09ffdf7aa670b37da0795d
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// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses double-buffering by default, but doesn't tell us about that. 139// So for now we just assume that client is double-buffered. 140// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 141// N-buffering, so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 1; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title) { 189#ifdef DEBUG_CPU_USAGE 190 // get current thread's delta CPU time in wall clock ns 191 double wcNs; 192 bool valid = mCpuUsage.sampleAndEnable(wcNs); 193 194 // record sample for wall clock statistics 195 if (valid) { 196 mWcStats.sample(wcNs); 197 } 198 199 // get the current CPU number 200 int cpuNum = sched_getcpu(); 201 202 // get the current CPU frequency in kHz 203 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 204 205 // check if either CPU number or frequency changed 206 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 207 mCpuNum = cpuNum; 208 mCpukHz = cpukHz; 209 // ignore sample for purposes of cycles 210 valid = false; 211 } 212 213 // if no change in CPU number or frequency, then record sample for cycle statistics 214 if (valid && mCpukHz > 0) { 215 double cycles = wcNs * cpukHz * 0.000001; 216 mHzStats.sample(cycles); 217 } 218 219 unsigned n = mWcStats.n(); 220 // mCpuUsage.elapsed() is expensive, so don't call it every loop 221 if ((n & 127) == 1) { 222 long long elapsed = mCpuUsage.elapsed(); 223 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 224 double perLoop = elapsed / (double) n; 225 double perLoop100 = perLoop * 0.01; 226 double perLoop1k = perLoop * 0.001; 227 double mean = mWcStats.mean(); 228 double stddev = mWcStats.stddev(); 229 double minimum = mWcStats.minimum(); 230 double maximum = mWcStats.maximum(); 231 double meanCycles = mHzStats.mean(); 232 double stddevCycles = mHzStats.stddev(); 233 double minCycles = mHzStats.minimum(); 234 double maxCycles = mHzStats.maximum(); 235 mCpuUsage.resetElapsed(); 236 mWcStats.reset(); 237 mHzStats.reset(); 238 ALOGD("CPU usage for %s over past %.1f secs\n" 239 " (%u mixer loops at %.1f mean ms per loop):\n" 240 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 241 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 242 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 243 title.string(), 244 elapsed * .000000001, n, perLoop * .000001, 245 mean * .001, 246 stddev * .001, 247 minimum * .001, 248 maximum * .001, 249 mean / perLoop100, 250 stddev / perLoop100, 251 minimum / perLoop100, 252 maximum / perLoop100, 253 meanCycles / perLoop1k, 254 stddevCycles / perLoop1k, 255 minCycles / perLoop1k, 256 maxCycles / perLoop1k); 257 258 } 259 } 260#endif 261}; 262 263// ---------------------------------------------------------------------------- 264// ThreadBase 265// ---------------------------------------------------------------------------- 266 267AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 268 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 269 : Thread(false /*canCallJava*/), 270 mType(type), 271 mAudioFlinger(audioFlinger), 272 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are 273 // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 285 for (size_t i = 0; i < mConfigEvents.size(); i++) { 286 delete mConfigEvents[i]; 287 } 288 mConfigEvents.clear(); 289 290 mParamCond.broadcast(); 291 // do not lock the mutex in destructor 292 releaseWakeLock_l(); 293 if (mPowerManager != 0) { 294 sp<IBinder> binder = mPowerManager->asBinder(); 295 binder->unlinkToDeath(mDeathRecipient); 296 } 297} 298 299void AudioFlinger::ThreadBase::exit() 300{ 301 ALOGV("ThreadBase::exit"); 302 // do any cleanup required for exit to succeed 303 preExit(); 304 { 305 // This lock prevents the following race in thread (uniprocessor for illustration): 306 // if (!exitPending()) { 307 // // context switch from here to exit() 308 // // exit() calls requestExit(), what exitPending() observes 309 // // exit() calls signal(), which is dropped since no waiters 310 // // context switch back from exit() to here 311 // mWaitWorkCV.wait(...); 312 // // now thread is hung 313 // } 314 AutoMutex lock(mLock); 315 requestExit(); 316 mWaitWorkCV.broadcast(); 317 } 318 // When Thread::requestExitAndWait is made virtual and this method is renamed to 319 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 320 requestExitAndWait(); 321} 322 323status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 324{ 325 status_t status; 326 327 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 328 Mutex::Autolock _l(mLock); 329 330 mNewParameters.add(keyValuePairs); 331 mWaitWorkCV.signal(); 332 // wait condition with timeout in case the thread loop has exited 333 // before the request could be processed 334 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 335 status = mParamStatus; 336 mWaitWorkCV.signal(); 337 } else { 338 status = TIMED_OUT; 339 } 340 return status; 341} 342 343void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 344{ 345 Mutex::Autolock _l(mLock); 346 sendIoConfigEvent_l(event, param); 347} 348 349// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 350void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 351{ 352 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 353 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 354 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 355 param); 356 mWaitWorkCV.signal(); 357} 358 359// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 360void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 361{ 362 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 363 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 364 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 365 mConfigEvents.size(), pid, tid, prio); 366 mWaitWorkCV.signal(); 367} 368 369void AudioFlinger::ThreadBase::processConfigEvents() 370{ 371 mLock.lock(); 372 while (!mConfigEvents.isEmpty()) { 373 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 374 ConfigEvent *event = mConfigEvents[0]; 375 mConfigEvents.removeAt(0); 376 // release mLock before locking AudioFlinger mLock: lock order is always 377 // AudioFlinger then ThreadBase to avoid cross deadlock 378 mLock.unlock(); 379 switch(event->type()) { 380 case CFG_EVENT_PRIO: { 381 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 382 // FIXME Need to understand why this has be done asynchronously 383 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 384 true /*asynchronous*/); 385 if (err != 0) { 386 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 387 "error %d", 388 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 389 } 390 } break; 391 case CFG_EVENT_IO: { 392 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 393 mAudioFlinger->mLock.lock(); 394 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 395 mAudioFlinger->mLock.unlock(); 396 } break; 397 default: 398 ALOGE("processConfigEvents() unknown event type %d", event->type()); 399 break; 400 } 401 delete event; 402 mLock.lock(); 403 } 404 mLock.unlock(); 405} 406 407void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 408{ 409 const size_t SIZE = 256; 410 char buffer[SIZE]; 411 String8 result; 412 413 bool locked = AudioFlinger::dumpTryLock(mLock); 414 if (!locked) { 415 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 416 write(fd, buffer, strlen(buffer)); 417 } 418 419 snprintf(buffer, SIZE, "io handle: %d\n", mId); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 432 result.append(buffer); 433 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 434 result.append(buffer); 435 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 436 result.append(buffer); 437 438 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 439 result.append(buffer); 440 result.append(" Index Command"); 441 for (size_t i = 0; i < mNewParameters.size(); ++i) { 442 snprintf(buffer, SIZE, "\n %02d ", i); 443 result.append(buffer); 444 result.append(mNewParameters[i]); 445 } 446 447 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 448 result.append(buffer); 449 for (size_t i = 0; i < mConfigEvents.size(); i++) { 450 mConfigEvents[i]->dump(buffer, SIZE); 451 result.append(buffer); 452 } 453 result.append("\n"); 454 455 write(fd, result.string(), result.size()); 456 457 if (locked) { 458 mLock.unlock(); 459 } 460} 461 462void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 463{ 464 const size_t SIZE = 256; 465 char buffer[SIZE]; 466 String8 result; 467 468 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 469 write(fd, buffer, strlen(buffer)); 470 471 for (size_t i = 0; i < mEffectChains.size(); ++i) { 472 sp<EffectChain> chain = mEffectChains[i]; 473 if (chain != 0) { 474 chain->dump(fd, args); 475 } 476 } 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 480{ 481 Mutex::Autolock _l(mLock); 482 acquireWakeLock_l(uid); 483} 484 485void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 486{ 487 if (mPowerManager == 0) { 488 // use checkService() to avoid blocking if power service is not up yet 489 sp<IBinder> binder = 490 defaultServiceManager()->checkService(String16("power")); 491 if (binder == 0) { 492 ALOGW("Thread %s cannot connect to the power manager service", mName); 493 } else { 494 mPowerManager = interface_cast<IPowerManager>(binder); 495 binder->linkToDeath(mDeathRecipient); 496 } 497 } 498 if (mPowerManager != 0) { 499 sp<IBinder> binder = new BBinder(); 500 status_t status; 501 if (uid >= 0) { 502 mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 503 binder, 504 String16(mName), 505 String16("media"), 506 uid); 507 } else { 508 mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 509 binder, 510 String16(mName), 511 String16("media")); 512 } 513 if (status == NO_ERROR) { 514 mWakeLockToken = binder; 515 } 516 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 517 } 518} 519 520void AudioFlinger::ThreadBase::releaseWakeLock() 521{ 522 Mutex::Autolock _l(mLock); 523 releaseWakeLock_l(); 524} 525 526void AudioFlinger::ThreadBase::releaseWakeLock_l() 527{ 528 if (mWakeLockToken != 0) { 529 ALOGV("releaseWakeLock_l() %s", mName); 530 if (mPowerManager != 0) { 531 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 532 } 533 mWakeLockToken.clear(); 534 } 535} 536 537void AudioFlinger::ThreadBase::clearPowerManager() 538{ 539 Mutex::Autolock _l(mLock); 540 releaseWakeLock_l(); 541 mPowerManager.clear(); 542} 543 544void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 545{ 546 sp<ThreadBase> thread = mThread.promote(); 547 if (thread != 0) { 548 thread->clearPowerManager(); 549 } 550 ALOGW("power manager service died !!!"); 551} 552 553void AudioFlinger::ThreadBase::setEffectSuspended( 554 const effect_uuid_t *type, bool suspend, int sessionId) 555{ 556 Mutex::Autolock _l(mLock); 557 setEffectSuspended_l(type, suspend, sessionId); 558} 559 560void AudioFlinger::ThreadBase::setEffectSuspended_l( 561 const effect_uuid_t *type, bool suspend, int sessionId) 562{ 563 sp<EffectChain> chain = getEffectChain_l(sessionId); 564 if (chain != 0) { 565 if (type != NULL) { 566 chain->setEffectSuspended_l(type, suspend); 567 } else { 568 chain->setEffectSuspendedAll_l(suspend); 569 } 570 } 571 572 updateSuspendedSessions_l(type, suspend, sessionId); 573} 574 575void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 576{ 577 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 578 if (index < 0) { 579 return; 580 } 581 582 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 583 mSuspendedSessions.valueAt(index); 584 585 for (size_t i = 0; i < sessionEffects.size(); i++) { 586 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 587 for (int j = 0; j < desc->mRefCount; j++) { 588 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 589 chain->setEffectSuspendedAll_l(true); 590 } else { 591 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 592 desc->mType.timeLow); 593 chain->setEffectSuspended_l(&desc->mType, true); 594 } 595 } 596 } 597} 598 599void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 600 bool suspend, 601 int sessionId) 602{ 603 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 604 605 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 606 607 if (suspend) { 608 if (index >= 0) { 609 sessionEffects = mSuspendedSessions.valueAt(index); 610 } else { 611 mSuspendedSessions.add(sessionId, sessionEffects); 612 } 613 } else { 614 if (index < 0) { 615 return; 616 } 617 sessionEffects = mSuspendedSessions.valueAt(index); 618 } 619 620 621 int key = EffectChain::kKeyForSuspendAll; 622 if (type != NULL) { 623 key = type->timeLow; 624 } 625 index = sessionEffects.indexOfKey(key); 626 627 sp<SuspendedSessionDesc> desc; 628 if (suspend) { 629 if (index >= 0) { 630 desc = sessionEffects.valueAt(index); 631 } else { 632 desc = new SuspendedSessionDesc(); 633 if (type != NULL) { 634 desc->mType = *type; 635 } 636 sessionEffects.add(key, desc); 637 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 638 } 639 desc->mRefCount++; 640 } else { 641 if (index < 0) { 642 return; 643 } 644 desc = sessionEffects.valueAt(index); 645 if (--desc->mRefCount == 0) { 646 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 647 sessionEffects.removeItemsAt(index); 648 if (sessionEffects.isEmpty()) { 649 ALOGV("updateSuspendedSessions_l() restore removing session %d", 650 sessionId); 651 mSuspendedSessions.removeItem(sessionId); 652 } 653 } 654 } 655 if (!sessionEffects.isEmpty()) { 656 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 657 } 658} 659 660void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 661 bool enabled, 662 int sessionId) 663{ 664 Mutex::Autolock _l(mLock); 665 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 666} 667 668void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 669 bool enabled, 670 int sessionId) 671{ 672 if (mType != RECORD) { 673 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 674 // another session. This gives the priority to well behaved effect control panels 675 // and applications not using global effects. 676 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 677 // global effects 678 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 679 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 680 } 681 } 682 683 sp<EffectChain> chain = getEffectChain_l(sessionId); 684 if (chain != 0) { 685 chain->checkSuspendOnEffectEnabled(effect, enabled); 686 } 687} 688 689// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 690sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 691 const sp<AudioFlinger::Client>& client, 692 const sp<IEffectClient>& effectClient, 693 int32_t priority, 694 int sessionId, 695 effect_descriptor_t *desc, 696 int *enabled, 697 status_t *status 698 ) 699{ 700 sp<EffectModule> effect; 701 sp<EffectHandle> handle; 702 status_t lStatus; 703 sp<EffectChain> chain; 704 bool chainCreated = false; 705 bool effectCreated = false; 706 bool effectRegistered = false; 707 708 lStatus = initCheck(); 709 if (lStatus != NO_ERROR) { 710 ALOGW("createEffect_l() Audio driver not initialized."); 711 goto Exit; 712 } 713 714 // Allow global effects only on offloaded and mixer threads 715 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 716 switch (mType) { 717 case MIXER: 718 case OFFLOAD: 719 break; 720 case DIRECT: 721 case DUPLICATING: 722 case RECORD: 723 default: 724 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 725 lStatus = BAD_VALUE; 726 goto Exit; 727 } 728 } 729 730 // Only Pre processor effects are allowed on input threads and only on input threads 731 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 732 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 733 desc->name, desc->flags, mType); 734 lStatus = BAD_VALUE; 735 goto Exit; 736 } 737 738 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 739 740 { // scope for mLock 741 Mutex::Autolock _l(mLock); 742 743 // check for existing effect chain with the requested audio session 744 chain = getEffectChain_l(sessionId); 745 if (chain == 0) { 746 // create a new chain for this session 747 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 748 chain = new EffectChain(this, sessionId); 749 addEffectChain_l(chain); 750 chain->setStrategy(getStrategyForSession_l(sessionId)); 751 chainCreated = true; 752 } else { 753 effect = chain->getEffectFromDesc_l(desc); 754 } 755 756 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 757 758 if (effect == 0) { 759 int id = mAudioFlinger->nextUniqueId(); 760 // Check CPU and memory usage 761 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 762 if (lStatus != NO_ERROR) { 763 goto Exit; 764 } 765 effectRegistered = true; 766 // create a new effect module if none present in the chain 767 effect = new EffectModule(this, chain, desc, id, sessionId); 768 lStatus = effect->status(); 769 if (lStatus != NO_ERROR) { 770 goto Exit; 771 } 772 effect->setOffloaded(mType == OFFLOAD, mId); 773 774 lStatus = chain->addEffect_l(effect); 775 if (lStatus != NO_ERROR) { 776 goto Exit; 777 } 778 effectCreated = true; 779 780 effect->setDevice(mOutDevice); 781 effect->setDevice(mInDevice); 782 effect->setMode(mAudioFlinger->getMode()); 783 effect->setAudioSource(mAudioSource); 784 } 785 // create effect handle and connect it to effect module 786 handle = new EffectHandle(effect, client, effectClient, priority); 787 lStatus = effect->addHandle(handle.get()); 788 if (enabled != NULL) { 789 *enabled = (int)effect->isEnabled(); 790 } 791 } 792 793Exit: 794 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 795 Mutex::Autolock _l(mLock); 796 if (effectCreated) { 797 chain->removeEffect_l(effect); 798 } 799 if (effectRegistered) { 800 AudioSystem::unregisterEffect(effect->id()); 801 } 802 if (chainCreated) { 803 removeEffectChain_l(chain); 804 } 805 handle.clear(); 806 } 807 808 if (status != NULL) { 809 *status = lStatus; 810 } 811 return handle; 812} 813 814sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 815{ 816 Mutex::Autolock _l(mLock); 817 return getEffect_l(sessionId, effectId); 818} 819 820sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 821{ 822 sp<EffectChain> chain = getEffectChain_l(sessionId); 823 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 824} 825 826// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 827// PlaybackThread::mLock held 828status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 829{ 830 // check for existing effect chain with the requested audio session 831 int sessionId = effect->sessionId(); 832 sp<EffectChain> chain = getEffectChain_l(sessionId); 833 bool chainCreated = false; 834 835 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 836 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 837 this, effect->desc().name, effect->desc().flags); 838 839 if (chain == 0) { 840 // create a new chain for this session 841 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 842 chain = new EffectChain(this, sessionId); 843 addEffectChain_l(chain); 844 chain->setStrategy(getStrategyForSession_l(sessionId)); 845 chainCreated = true; 846 } 847 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 848 849 if (chain->getEffectFromId_l(effect->id()) != 0) { 850 ALOGW("addEffect_l() %p effect %s already present in chain %p", 851 this, effect->desc().name, chain.get()); 852 return BAD_VALUE; 853 } 854 855 effect->setOffloaded(mType == OFFLOAD, mId); 856 857 status_t status = chain->addEffect_l(effect); 858 if (status != NO_ERROR) { 859 if (chainCreated) { 860 removeEffectChain_l(chain); 861 } 862 return status; 863 } 864 865 effect->setDevice(mOutDevice); 866 effect->setDevice(mInDevice); 867 effect->setMode(mAudioFlinger->getMode()); 868 effect->setAudioSource(mAudioSource); 869 return NO_ERROR; 870} 871 872void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 873 874 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 875 effect_descriptor_t desc = effect->desc(); 876 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 877 detachAuxEffect_l(effect->id()); 878 } 879 880 sp<EffectChain> chain = effect->chain().promote(); 881 if (chain != 0) { 882 // remove effect chain if removing last effect 883 if (chain->removeEffect_l(effect) == 0) { 884 removeEffectChain_l(chain); 885 } 886 } else { 887 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 888 } 889} 890 891void AudioFlinger::ThreadBase::lockEffectChains_l( 892 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 893{ 894 effectChains = mEffectChains; 895 for (size_t i = 0; i < mEffectChains.size(); i++) { 896 mEffectChains[i]->lock(); 897 } 898} 899 900void AudioFlinger::ThreadBase::unlockEffectChains( 901 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 902{ 903 for (size_t i = 0; i < effectChains.size(); i++) { 904 effectChains[i]->unlock(); 905 } 906} 907 908sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 909{ 910 Mutex::Autolock _l(mLock); 911 return getEffectChain_l(sessionId); 912} 913 914sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 915{ 916 size_t size = mEffectChains.size(); 917 for (size_t i = 0; i < size; i++) { 918 if (mEffectChains[i]->sessionId() == sessionId) { 919 return mEffectChains[i]; 920 } 921 } 922 return 0; 923} 924 925void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 926{ 927 Mutex::Autolock _l(mLock); 928 size_t size = mEffectChains.size(); 929 for (size_t i = 0; i < size; i++) { 930 mEffectChains[i]->setMode_l(mode); 931 } 932} 933 934void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 935 EffectHandle *handle, 936 bool unpinIfLast) { 937 938 Mutex::Autolock _l(mLock); 939 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 940 // delete the effect module if removing last handle on it 941 if (effect->removeHandle(handle) == 0) { 942 if (!effect->isPinned() || unpinIfLast) { 943 removeEffect_l(effect); 944 AudioSystem::unregisterEffect(effect->id()); 945 } 946 } 947} 948 949// ---------------------------------------------------------------------------- 950// Playback 951// ---------------------------------------------------------------------------- 952 953AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 954 AudioStreamOut* output, 955 audio_io_handle_t id, 956 audio_devices_t device, 957 type_t type) 958 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 959 mNormalFrameCount(0), mMixBuffer(NULL), 960 mAllocMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 961 // mStreamTypes[] initialized in constructor body 962 mOutput(output), 963 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 964 mMixerStatus(MIXER_IDLE), 965 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 966 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 967 mBytesRemaining(0), 968 mCurrentWriteLength(0), 969 mUseAsyncWrite(false), 970 mWriteAckSequence(0), 971 mDrainSequence(0), 972 mSignalPending(false), 973 mScreenState(AudioFlinger::mScreenState), 974 // index 0 is reserved for normal mixer's submix 975 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 976 // mLatchD, mLatchQ, 977 mLatchDValid(false), mLatchQValid(false) 978{ 979 snprintf(mName, kNameLength, "AudioOut_%X", id); 980 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 981 982 // Assumes constructor is called by AudioFlinger with it's mLock held, but 983 // it would be safer to explicitly pass initial masterVolume/masterMute as 984 // parameter. 985 // 986 // If the HAL we are using has support for master volume or master mute, 987 // then do not attenuate or mute during mixing (just leave the volume at 1.0 988 // and the mute set to false). 989 mMasterVolume = audioFlinger->masterVolume_l(); 990 mMasterMute = audioFlinger->masterMute_l(); 991 if (mOutput && mOutput->audioHwDev) { 992 if (mOutput->audioHwDev->canSetMasterVolume()) { 993 mMasterVolume = 1.0; 994 } 995 996 if (mOutput->audioHwDev->canSetMasterMute()) { 997 mMasterMute = false; 998 } 999 } 1000 1001 readOutputParameters(); 1002 1003 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1004 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1005 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1006 stream = (audio_stream_type_t) (stream + 1)) { 1007 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1008 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1009 } 1010 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1011 // because mAudioFlinger doesn't have one to copy from 1012} 1013 1014AudioFlinger::PlaybackThread::~PlaybackThread() 1015{ 1016 mAudioFlinger->unregisterWriter(mNBLogWriter); 1017 delete [] mAllocMixBuffer; 1018} 1019 1020void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1021{ 1022 dumpInternals(fd, args); 1023 dumpTracks(fd, args); 1024 dumpEffectChains(fd, args); 1025} 1026 1027void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1028{ 1029 const size_t SIZE = 256; 1030 char buffer[SIZE]; 1031 String8 result; 1032 1033 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1034 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1035 const stream_type_t *st = &mStreamTypes[i]; 1036 if (i > 0) { 1037 result.appendFormat(", "); 1038 } 1039 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1040 if (st->mute) { 1041 result.append("M"); 1042 } 1043 } 1044 result.append("\n"); 1045 write(fd, result.string(), result.length()); 1046 result.clear(); 1047 1048 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1049 result.append(buffer); 1050 Track::appendDumpHeader(result); 1051 for (size_t i = 0; i < mTracks.size(); ++i) { 1052 sp<Track> track = mTracks[i]; 1053 if (track != 0) { 1054 track->dump(buffer, SIZE); 1055 result.append(buffer); 1056 } 1057 } 1058 1059 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1060 result.append(buffer); 1061 Track::appendDumpHeader(result); 1062 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1063 sp<Track> track = mActiveTracks[i].promote(); 1064 if (track != 0) { 1065 track->dump(buffer, SIZE); 1066 result.append(buffer); 1067 } 1068 } 1069 write(fd, result.string(), result.size()); 1070 1071 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1072 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1073 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1074 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1075} 1076 1077void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1078{ 1079 const size_t SIZE = 256; 1080 char buffer[SIZE]; 1081 String8 result; 1082 1083 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1084 result.append(buffer); 1085 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1086 result.append(buffer); 1087 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1088 ns2ms(systemTime() - mLastWriteTime)); 1089 result.append(buffer); 1090 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1091 result.append(buffer); 1092 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1093 result.append(buffer); 1094 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1095 result.append(buffer); 1096 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1097 result.append(buffer); 1098 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1099 result.append(buffer); 1100 write(fd, result.string(), result.size()); 1101 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1102 1103 dumpBase(fd, args); 1104} 1105 1106// Thread virtuals 1107status_t AudioFlinger::PlaybackThread::readyToRun() 1108{ 1109 status_t status = initCheck(); 1110 if (status == NO_ERROR) { 1111 ALOGI("AudioFlinger's thread %p ready to run", this); 1112 } else { 1113 ALOGE("No working audio driver found."); 1114 } 1115 return status; 1116} 1117 1118void AudioFlinger::PlaybackThread::onFirstRef() 1119{ 1120 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1121} 1122 1123// ThreadBase virtuals 1124void AudioFlinger::PlaybackThread::preExit() 1125{ 1126 ALOGV(" preExit()"); 1127 // FIXME this is using hard-coded strings but in the future, this functionality will be 1128 // converted to use audio HAL extensions required to support tunneling 1129 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1130} 1131 1132// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1133sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1134 const sp<AudioFlinger::Client>& client, 1135 audio_stream_type_t streamType, 1136 uint32_t sampleRate, 1137 audio_format_t format, 1138 audio_channel_mask_t channelMask, 1139 size_t frameCount, 1140 const sp<IMemory>& sharedBuffer, 1141 int sessionId, 1142 IAudioFlinger::track_flags_t *flags, 1143 pid_t tid, 1144 status_t *status) 1145{ 1146 sp<Track> track; 1147 status_t lStatus; 1148 1149 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1150 1151 // client expresses a preference for FAST, but we get the final say 1152 if (*flags & IAudioFlinger::TRACK_FAST) { 1153 if ( 1154 // not timed 1155 (!isTimed) && 1156 // either of these use cases: 1157 ( 1158 // use case 1: shared buffer with any frame count 1159 ( 1160 (sharedBuffer != 0) 1161 ) || 1162 // use case 2: callback handler and frame count is default or at least as large as HAL 1163 ( 1164 (tid != -1) && 1165 ((frameCount == 0) || 1166 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1167 ) 1168 ) && 1169 // PCM data 1170 audio_is_linear_pcm(format) && 1171 // mono or stereo 1172 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1173 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1174#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1175 // hardware sample rate 1176 (sampleRate == mSampleRate) && 1177#endif 1178 // normal mixer has an associated fast mixer 1179 hasFastMixer() && 1180 // there are sufficient fast track slots available 1181 (mFastTrackAvailMask != 0) 1182 // FIXME test that MixerThread for this fast track has a capable output HAL 1183 // FIXME add a permission test also? 1184 ) { 1185 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1186 if (frameCount == 0) { 1187 frameCount = mFrameCount * kFastTrackMultiplier; 1188 } 1189 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1190 frameCount, mFrameCount); 1191 } else { 1192 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1193 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1194 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1195 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1196 audio_is_linear_pcm(format), 1197 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1198 *flags &= ~IAudioFlinger::TRACK_FAST; 1199 // For compatibility with AudioTrack calculation, buffer depth is forced 1200 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1201 // This is probably too conservative, but legacy application code may depend on it. 1202 // If you change this calculation, also review the start threshold which is related. 1203 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1204 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1205 if (minBufCount < 2) { 1206 minBufCount = 2; 1207 } 1208 size_t minFrameCount = mNormalFrameCount * minBufCount; 1209 if (frameCount < minFrameCount) { 1210 frameCount = minFrameCount; 1211 } 1212 } 1213 } 1214 1215 if (mType == DIRECT) { 1216 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1217 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1218 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1219 "for output %p with format %d", 1220 sampleRate, format, channelMask, mOutput, mFormat); 1221 lStatus = BAD_VALUE; 1222 goto Exit; 1223 } 1224 } 1225 } else if (mType == OFFLOAD) { 1226 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1227 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1228 "for output %p with format %d", 1229 sampleRate, format, channelMask, mOutput, mFormat); 1230 lStatus = BAD_VALUE; 1231 goto Exit; 1232 } 1233 } else { 1234 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1235 ALOGE("createTrack_l() Bad parameter: format %d \"" 1236 "for output %p with format %d", 1237 format, mOutput, mFormat); 1238 lStatus = BAD_VALUE; 1239 goto Exit; 1240 } 1241 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1242 if (sampleRate > mSampleRate*2) { 1243 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1244 lStatus = BAD_VALUE; 1245 goto Exit; 1246 } 1247 } 1248 1249 lStatus = initCheck(); 1250 if (lStatus != NO_ERROR) { 1251 ALOGE("Audio driver not initialized."); 1252 goto Exit; 1253 } 1254 1255 { // scope for mLock 1256 Mutex::Autolock _l(mLock); 1257 1258 // all tracks in same audio session must share the same routing strategy otherwise 1259 // conflicts will happen when tracks are moved from one output to another by audio policy 1260 // manager 1261 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1262 for (size_t i = 0; i < mTracks.size(); ++i) { 1263 sp<Track> t = mTracks[i]; 1264 if (t != 0 && !t->isOutputTrack()) { 1265 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1266 if (sessionId == t->sessionId() && strategy != actual) { 1267 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1268 strategy, actual); 1269 lStatus = BAD_VALUE; 1270 goto Exit; 1271 } 1272 } 1273 } 1274 1275 if (!isTimed) { 1276 track = new Track(this, client, streamType, sampleRate, format, 1277 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1278 } else { 1279 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1280 channelMask, frameCount, sharedBuffer, sessionId); 1281 } 1282 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1283 lStatus = NO_MEMORY; 1284 goto Exit; 1285 } 1286 1287 mTracks.add(track); 1288 1289 sp<EffectChain> chain = getEffectChain_l(sessionId); 1290 if (chain != 0) { 1291 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1292 track->setMainBuffer(chain->inBuffer()); 1293 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1294 chain->incTrackCnt(); 1295 } 1296 1297 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1298 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1299 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1300 // so ask activity manager to do this on our behalf 1301 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1302 } 1303 } 1304 1305 lStatus = NO_ERROR; 1306 1307Exit: 1308 if (status) { 1309 *status = lStatus; 1310 } 1311 return track; 1312} 1313 1314uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1315{ 1316 return latency; 1317} 1318 1319uint32_t AudioFlinger::PlaybackThread::latency() const 1320{ 1321 Mutex::Autolock _l(mLock); 1322 return latency_l(); 1323} 1324uint32_t AudioFlinger::PlaybackThread::latency_l() const 1325{ 1326 if (initCheck() == NO_ERROR) { 1327 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1328 } else { 1329 return 0; 1330 } 1331} 1332 1333void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1334{ 1335 Mutex::Autolock _l(mLock); 1336 // Don't apply master volume in SW if our HAL can do it for us. 1337 if (mOutput && mOutput->audioHwDev && 1338 mOutput->audioHwDev->canSetMasterVolume()) { 1339 mMasterVolume = 1.0; 1340 } else { 1341 mMasterVolume = value; 1342 } 1343} 1344 1345void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1346{ 1347 Mutex::Autolock _l(mLock); 1348 // Don't apply master mute in SW if our HAL can do it for us. 1349 if (mOutput && mOutput->audioHwDev && 1350 mOutput->audioHwDev->canSetMasterMute()) { 1351 mMasterMute = false; 1352 } else { 1353 mMasterMute = muted; 1354 } 1355} 1356 1357void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1358{ 1359 Mutex::Autolock _l(mLock); 1360 mStreamTypes[stream].volume = value; 1361 broadcast_l(); 1362} 1363 1364void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1365{ 1366 Mutex::Autolock _l(mLock); 1367 mStreamTypes[stream].mute = muted; 1368 broadcast_l(); 1369} 1370 1371float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1372{ 1373 Mutex::Autolock _l(mLock); 1374 return mStreamTypes[stream].volume; 1375} 1376 1377// addTrack_l() must be called with ThreadBase::mLock held 1378status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1379{ 1380 status_t status = ALREADY_EXISTS; 1381 1382 // set retry count for buffer fill 1383 track->mRetryCount = kMaxTrackStartupRetries; 1384 if (mActiveTracks.indexOf(track) < 0) { 1385 // the track is newly added, make sure it fills up all its 1386 // buffers before playing. This is to ensure the client will 1387 // effectively get the latency it requested. 1388 if (!track->isOutputTrack()) { 1389 TrackBase::track_state state = track->mState; 1390 mLock.unlock(); 1391 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1392 mLock.lock(); 1393 // abort track was stopped/paused while we released the lock 1394 if (state != track->mState) { 1395 if (status == NO_ERROR) { 1396 mLock.unlock(); 1397 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1398 mLock.lock(); 1399 } 1400 return INVALID_OPERATION; 1401 } 1402 // abort if start is rejected by audio policy manager 1403 if (status != NO_ERROR) { 1404 return PERMISSION_DENIED; 1405 } 1406#ifdef ADD_BATTERY_DATA 1407 // to track the speaker usage 1408 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1409#endif 1410 } 1411 1412 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1413 track->mResetDone = false; 1414 track->mPresentationCompleteFrames = 0; 1415 mActiveTracks.add(track); 1416 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1417 if (chain != 0) { 1418 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1419 track->sessionId()); 1420 chain->incActiveTrackCnt(); 1421 } 1422 1423 status = NO_ERROR; 1424 } 1425 1426 ALOGV("signal playback thread"); 1427 broadcast_l(); 1428 1429 return status; 1430} 1431 1432bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1433{ 1434 track->terminate(); 1435 // active tracks are removed by threadLoop() 1436 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1437 track->mState = TrackBase::STOPPED; 1438 if (!trackActive) { 1439 removeTrack_l(track); 1440 } else if (track->isFastTrack() || track->isOffloaded()) { 1441 track->mState = TrackBase::STOPPING_1; 1442 } 1443 1444 return trackActive; 1445} 1446 1447void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1448{ 1449 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1450 mTracks.remove(track); 1451 deleteTrackName_l(track->name()); 1452 // redundant as track is about to be destroyed, for dumpsys only 1453 track->mName = -1; 1454 if (track->isFastTrack()) { 1455 int index = track->mFastIndex; 1456 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1457 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1458 mFastTrackAvailMask |= 1 << index; 1459 // redundant as track is about to be destroyed, for dumpsys only 1460 track->mFastIndex = -1; 1461 } 1462 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1463 if (chain != 0) { 1464 chain->decTrackCnt(); 1465 } 1466} 1467 1468void AudioFlinger::PlaybackThread::broadcast_l() 1469{ 1470 // Thread could be blocked waiting for async 1471 // so signal it to handle state changes immediately 1472 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1473 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1474 mSignalPending = true; 1475 mWaitWorkCV.broadcast(); 1476} 1477 1478String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1479{ 1480 Mutex::Autolock _l(mLock); 1481 if (initCheck() != NO_ERROR) { 1482 return String8(); 1483 } 1484 1485 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1486 const String8 out_s8(s); 1487 free(s); 1488 return out_s8; 1489} 1490 1491// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1492void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1493 AudioSystem::OutputDescriptor desc; 1494 void *param2 = NULL; 1495 1496 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1497 param); 1498 1499 switch (event) { 1500 case AudioSystem::OUTPUT_OPENED: 1501 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1502 desc.channelMask = mChannelMask; 1503 desc.samplingRate = mSampleRate; 1504 desc.format = mFormat; 1505 desc.frameCount = mNormalFrameCount; // FIXME see 1506 // AudioFlinger::frameCount(audio_io_handle_t) 1507 desc.latency = latency(); 1508 param2 = &desc; 1509 break; 1510 1511 case AudioSystem::STREAM_CONFIG_CHANGED: 1512 param2 = ¶m; 1513 case AudioSystem::OUTPUT_CLOSED: 1514 default: 1515 break; 1516 } 1517 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1518} 1519 1520void AudioFlinger::PlaybackThread::writeCallback() 1521{ 1522 ALOG_ASSERT(mCallbackThread != 0); 1523 mCallbackThread->resetWriteBlocked(); 1524} 1525 1526void AudioFlinger::PlaybackThread::drainCallback() 1527{ 1528 ALOG_ASSERT(mCallbackThread != 0); 1529 mCallbackThread->resetDraining(); 1530} 1531 1532void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1533{ 1534 Mutex::Autolock _l(mLock); 1535 // reject out of sequence requests 1536 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1537 mWriteAckSequence &= ~1; 1538 mWaitWorkCV.signal(); 1539 } 1540} 1541 1542void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1543{ 1544 Mutex::Autolock _l(mLock); 1545 // reject out of sequence requests 1546 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1547 mDrainSequence &= ~1; 1548 mWaitWorkCV.signal(); 1549 } 1550} 1551 1552// static 1553int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1554 void *param, 1555 void *cookie) 1556{ 1557 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1558 ALOGV("asyncCallback() event %d", event); 1559 switch (event) { 1560 case STREAM_CBK_EVENT_WRITE_READY: 1561 me->writeCallback(); 1562 break; 1563 case STREAM_CBK_EVENT_DRAIN_READY: 1564 me->drainCallback(); 1565 break; 1566 default: 1567 ALOGW("asyncCallback() unknown event %d", event); 1568 break; 1569 } 1570 return 0; 1571} 1572 1573void AudioFlinger::PlaybackThread::readOutputParameters() 1574{ 1575 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1576 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1577 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1578 if (!audio_is_output_channel(mChannelMask)) { 1579 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1580 } 1581 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1582 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1583 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1584 } 1585 mChannelCount = popcount(mChannelMask); 1586 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1587 if (!audio_is_valid_format(mFormat)) { 1588 LOG_FATAL("HAL format %d not valid for output", mFormat); 1589 } 1590 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1591 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1592 mFormat); 1593 } 1594 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1595 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1596 if (mFrameCount & 15) { 1597 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1598 mFrameCount); 1599 } 1600 1601 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1602 (mOutput->stream->set_callback != NULL)) { 1603 if (mOutput->stream->set_callback(mOutput->stream, 1604 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1605 mUseAsyncWrite = true; 1606 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1607 } 1608 } 1609 1610 // Calculate size of normal mix buffer relative to the HAL output buffer size 1611 double multiplier = 1.0; 1612 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1613 kUseFastMixer == FastMixer_Dynamic)) { 1614 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1615 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1616 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1617 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1618 maxNormalFrameCount = maxNormalFrameCount & ~15; 1619 if (maxNormalFrameCount < minNormalFrameCount) { 1620 maxNormalFrameCount = minNormalFrameCount; 1621 } 1622 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1623 if (multiplier <= 1.0) { 1624 multiplier = 1.0; 1625 } else if (multiplier <= 2.0) { 1626 if (2 * mFrameCount <= maxNormalFrameCount) { 1627 multiplier = 2.0; 1628 } else { 1629 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1630 } 1631 } else { 1632 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1633 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1634 // track, but we sometimes have to do this to satisfy the maximum frame count 1635 // constraint) 1636 // FIXME this rounding up should not be done if no HAL SRC 1637 uint32_t truncMult = (uint32_t) multiplier; 1638 if ((truncMult & 1)) { 1639 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1640 ++truncMult; 1641 } 1642 } 1643 multiplier = (double) truncMult; 1644 } 1645 } 1646 mNormalFrameCount = multiplier * mFrameCount; 1647 // round up to nearest 16 frames to satisfy AudioMixer 1648 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1649 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1650 mNormalFrameCount); 1651 1652 delete[] mAllocMixBuffer; 1653 size_t align = (mFrameSize < sizeof(int16_t)) ? sizeof(int16_t) : mFrameSize; 1654 mAllocMixBuffer = new int8_t[mNormalFrameCount * mFrameSize + align - 1]; 1655 mMixBuffer = (int16_t *) ((((size_t)mAllocMixBuffer + align - 1) / align) * align); 1656 memset(mMixBuffer, 0, mNormalFrameCount * mFrameSize); 1657 1658 // force reconfiguration of effect chains and engines to take new buffer size and audio 1659 // parameters into account 1660 // Note that mLock is not held when readOutputParameters() is called from the constructor 1661 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1662 // matter. 1663 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1664 Vector< sp<EffectChain> > effectChains = mEffectChains; 1665 for (size_t i = 0; i < effectChains.size(); i ++) { 1666 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1667 } 1668} 1669 1670 1671status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1672{ 1673 if (halFrames == NULL || dspFrames == NULL) { 1674 return BAD_VALUE; 1675 } 1676 Mutex::Autolock _l(mLock); 1677 if (initCheck() != NO_ERROR) { 1678 return INVALID_OPERATION; 1679 } 1680 size_t framesWritten = mBytesWritten / mFrameSize; 1681 *halFrames = framesWritten; 1682 1683 if (isSuspended()) { 1684 // return an estimation of rendered frames when the output is suspended 1685 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1686 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1687 return NO_ERROR; 1688 } else { 1689 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1690 } 1691} 1692 1693uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1694{ 1695 Mutex::Autolock _l(mLock); 1696 uint32_t result = 0; 1697 if (getEffectChain_l(sessionId) != 0) { 1698 result = EFFECT_SESSION; 1699 } 1700 1701 for (size_t i = 0; i < mTracks.size(); ++i) { 1702 sp<Track> track = mTracks[i]; 1703 if (sessionId == track->sessionId() && !track->isInvalid()) { 1704 result |= TRACK_SESSION; 1705 break; 1706 } 1707 } 1708 1709 return result; 1710} 1711 1712uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1713{ 1714 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1715 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1716 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1717 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1718 } 1719 for (size_t i = 0; i < mTracks.size(); i++) { 1720 sp<Track> track = mTracks[i]; 1721 if (sessionId == track->sessionId() && !track->isInvalid()) { 1722 return AudioSystem::getStrategyForStream(track->streamType()); 1723 } 1724 } 1725 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1726} 1727 1728 1729AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1730{ 1731 Mutex::Autolock _l(mLock); 1732 return mOutput; 1733} 1734 1735AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1736{ 1737 Mutex::Autolock _l(mLock); 1738 AudioStreamOut *output = mOutput; 1739 mOutput = NULL; 1740 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1741 // must push a NULL and wait for ack 1742 mOutputSink.clear(); 1743 mPipeSink.clear(); 1744 mNormalSink.clear(); 1745 return output; 1746} 1747 1748// this method must always be called either with ThreadBase mLock held or inside the thread loop 1749audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1750{ 1751 if (mOutput == NULL) { 1752 return NULL; 1753 } 1754 return &mOutput->stream->common; 1755} 1756 1757uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1758{ 1759 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1760} 1761 1762status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1763{ 1764 if (!isValidSyncEvent(event)) { 1765 return BAD_VALUE; 1766 } 1767 1768 Mutex::Autolock _l(mLock); 1769 1770 for (size_t i = 0; i < mTracks.size(); ++i) { 1771 sp<Track> track = mTracks[i]; 1772 if (event->triggerSession() == track->sessionId()) { 1773 (void) track->setSyncEvent(event); 1774 return NO_ERROR; 1775 } 1776 } 1777 1778 return NAME_NOT_FOUND; 1779} 1780 1781bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1782{ 1783 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1784} 1785 1786void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1787 const Vector< sp<Track> >& tracksToRemove) 1788{ 1789 size_t count = tracksToRemove.size(); 1790 if (count) { 1791 for (size_t i = 0 ; i < count ; i++) { 1792 const sp<Track>& track = tracksToRemove.itemAt(i); 1793 if (!track->isOutputTrack()) { 1794 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1795#ifdef ADD_BATTERY_DATA 1796 // to track the speaker usage 1797 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1798#endif 1799 if (track->isTerminated()) { 1800 AudioSystem::releaseOutput(mId); 1801 } 1802 } 1803 } 1804 } 1805} 1806 1807void AudioFlinger::PlaybackThread::checkSilentMode_l() 1808{ 1809 if (!mMasterMute) { 1810 char value[PROPERTY_VALUE_MAX]; 1811 if (property_get("ro.audio.silent", value, "0") > 0) { 1812 char *endptr; 1813 unsigned long ul = strtoul(value, &endptr, 0); 1814 if (*endptr == '\0' && ul != 0) { 1815 ALOGD("Silence is golden"); 1816 // The setprop command will not allow a property to be changed after 1817 // the first time it is set, so we don't have to worry about un-muting. 1818 setMasterMute_l(true); 1819 } 1820 } 1821 } 1822} 1823 1824// shared by MIXER and DIRECT, overridden by DUPLICATING 1825ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1826{ 1827 // FIXME rewrite to reduce number of system calls 1828 mLastWriteTime = systemTime(); 1829 mInWrite = true; 1830 ssize_t bytesWritten; 1831 1832 // If an NBAIO sink is present, use it to write the normal mixer's submix 1833 if (mNormalSink != 0) { 1834#define mBitShift 2 // FIXME 1835 size_t count = mBytesRemaining >> mBitShift; 1836 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1837 ATRACE_BEGIN("write"); 1838 // update the setpoint when AudioFlinger::mScreenState changes 1839 uint32_t screenState = AudioFlinger::mScreenState; 1840 if (screenState != mScreenState) { 1841 mScreenState = screenState; 1842 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1843 if (pipe != NULL) { 1844 pipe->setAvgFrames((mScreenState & 1) ? 1845 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1846 } 1847 } 1848 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1849 ATRACE_END(); 1850 if (framesWritten > 0) { 1851 bytesWritten = framesWritten << mBitShift; 1852 } else { 1853 bytesWritten = framesWritten; 1854 } 1855 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1856 if (status == NO_ERROR) { 1857 size_t totalFramesWritten = mNormalSink->framesWritten(); 1858 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1859 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1860 mLatchDValid = true; 1861 } 1862 } 1863 // otherwise use the HAL / AudioStreamOut directly 1864 } else { 1865 // Direct output and offload threads 1866 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1867 if (mUseAsyncWrite) { 1868 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 1869 mWriteAckSequence += 2; 1870 mWriteAckSequence |= 1; 1871 ALOG_ASSERT(mCallbackThread != 0); 1872 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1873 } 1874 // FIXME We should have an implementation of timestamps for direct output threads. 1875 // They are used e.g for multichannel PCM playback over HDMI. 1876 bytesWritten = mOutput->stream->write(mOutput->stream, 1877 mMixBuffer + offset, mBytesRemaining); 1878 if (mUseAsyncWrite && 1879 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1880 // do not wait for async callback in case of error of full write 1881 mWriteAckSequence &= ~1; 1882 ALOG_ASSERT(mCallbackThread != 0); 1883 mCallbackThread->setWriteBlocked(mWriteAckSequence); 1884 } 1885 } 1886 1887 mNumWrites++; 1888 mInWrite = false; 1889 1890 return bytesWritten; 1891} 1892 1893void AudioFlinger::PlaybackThread::threadLoop_drain() 1894{ 1895 if (mOutput->stream->drain) { 1896 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1897 if (mUseAsyncWrite) { 1898 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 1899 mDrainSequence |= 1; 1900 ALOG_ASSERT(mCallbackThread != 0); 1901 mCallbackThread->setDraining(mDrainSequence); 1902 } 1903 mOutput->stream->drain(mOutput->stream, 1904 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1905 : AUDIO_DRAIN_ALL); 1906 } 1907} 1908 1909void AudioFlinger::PlaybackThread::threadLoop_exit() 1910{ 1911 // Default implementation has nothing to do 1912} 1913 1914/* 1915The derived values that are cached: 1916 - mixBufferSize from frame count * frame size 1917 - activeSleepTime from activeSleepTimeUs() 1918 - idleSleepTime from idleSleepTimeUs() 1919 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1920 - maxPeriod from frame count and sample rate (MIXER only) 1921 1922The parameters that affect these derived values are: 1923 - frame count 1924 - frame size 1925 - sample rate 1926 - device type: A2DP or not 1927 - device latency 1928 - format: PCM or not 1929 - active sleep time 1930 - idle sleep time 1931*/ 1932 1933void AudioFlinger::PlaybackThread::cacheParameters_l() 1934{ 1935 mixBufferSize = mNormalFrameCount * mFrameSize; 1936 activeSleepTime = activeSleepTimeUs(); 1937 idleSleepTime = idleSleepTimeUs(); 1938} 1939 1940void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1941{ 1942 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1943 this, streamType, mTracks.size()); 1944 Mutex::Autolock _l(mLock); 1945 1946 size_t size = mTracks.size(); 1947 for (size_t i = 0; i < size; i++) { 1948 sp<Track> t = mTracks[i]; 1949 if (t->streamType() == streamType) { 1950 t->invalidate(); 1951 } 1952 } 1953} 1954 1955status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1956{ 1957 int session = chain->sessionId(); 1958 int16_t *buffer = mMixBuffer; 1959 bool ownsBuffer = false; 1960 1961 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1962 if (session > 0) { 1963 // Only one effect chain can be present in direct output thread and it uses 1964 // the mix buffer as input 1965 if (mType != DIRECT) { 1966 size_t numSamples = mNormalFrameCount * mChannelCount; 1967 buffer = new int16_t[numSamples]; 1968 memset(buffer, 0, numSamples * sizeof(int16_t)); 1969 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1970 ownsBuffer = true; 1971 } 1972 1973 // Attach all tracks with same session ID to this chain. 1974 for (size_t i = 0; i < mTracks.size(); ++i) { 1975 sp<Track> track = mTracks[i]; 1976 if (session == track->sessionId()) { 1977 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1978 buffer); 1979 track->setMainBuffer(buffer); 1980 chain->incTrackCnt(); 1981 } 1982 } 1983 1984 // indicate all active tracks in the chain 1985 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1986 sp<Track> track = mActiveTracks[i].promote(); 1987 if (track == 0) { 1988 continue; 1989 } 1990 if (session == track->sessionId()) { 1991 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1992 chain->incActiveTrackCnt(); 1993 } 1994 } 1995 } 1996 1997 chain->setInBuffer(buffer, ownsBuffer); 1998 chain->setOutBuffer(mMixBuffer); 1999 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2000 // chains list in order to be processed last as it contains output stage effects 2001 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2002 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2003 // after track specific effects and before output stage 2004 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2005 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2006 // Effect chain for other sessions are inserted at beginning of effect 2007 // chains list to be processed before output mix effects. Relative order between other 2008 // sessions is not important 2009 size_t size = mEffectChains.size(); 2010 size_t i = 0; 2011 for (i = 0; i < size; i++) { 2012 if (mEffectChains[i]->sessionId() < session) { 2013 break; 2014 } 2015 } 2016 mEffectChains.insertAt(chain, i); 2017 checkSuspendOnAddEffectChain_l(chain); 2018 2019 return NO_ERROR; 2020} 2021 2022size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2023{ 2024 int session = chain->sessionId(); 2025 2026 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2027 2028 for (size_t i = 0; i < mEffectChains.size(); i++) { 2029 if (chain == mEffectChains[i]) { 2030 mEffectChains.removeAt(i); 2031 // detach all active tracks from the chain 2032 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2033 sp<Track> track = mActiveTracks[i].promote(); 2034 if (track == 0) { 2035 continue; 2036 } 2037 if (session == track->sessionId()) { 2038 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2039 chain.get(), session); 2040 chain->decActiveTrackCnt(); 2041 } 2042 } 2043 2044 // detach all tracks with same session ID from this chain 2045 for (size_t i = 0; i < mTracks.size(); ++i) { 2046 sp<Track> track = mTracks[i]; 2047 if (session == track->sessionId()) { 2048 track->setMainBuffer(mMixBuffer); 2049 chain->decTrackCnt(); 2050 } 2051 } 2052 break; 2053 } 2054 } 2055 return mEffectChains.size(); 2056} 2057 2058status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2059 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2060{ 2061 Mutex::Autolock _l(mLock); 2062 return attachAuxEffect_l(track, EffectId); 2063} 2064 2065status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2066 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2067{ 2068 status_t status = NO_ERROR; 2069 2070 if (EffectId == 0) { 2071 track->setAuxBuffer(0, NULL); 2072 } else { 2073 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2074 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2075 if (effect != 0) { 2076 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2077 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2078 } else { 2079 status = INVALID_OPERATION; 2080 } 2081 } else { 2082 status = BAD_VALUE; 2083 } 2084 } 2085 return status; 2086} 2087 2088void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2089{ 2090 for (size_t i = 0; i < mTracks.size(); ++i) { 2091 sp<Track> track = mTracks[i]; 2092 if (track->auxEffectId() == effectId) { 2093 attachAuxEffect_l(track, 0); 2094 } 2095 } 2096} 2097 2098bool AudioFlinger::PlaybackThread::threadLoop() 2099{ 2100 Vector< sp<Track> > tracksToRemove; 2101 2102 standbyTime = systemTime(); 2103 2104 // MIXER 2105 nsecs_t lastWarning = 0; 2106 2107 // DUPLICATING 2108 // FIXME could this be made local to while loop? 2109 writeFrames = 0; 2110 2111 cacheParameters_l(); 2112 sleepTime = idleSleepTime; 2113 2114 if (mType == MIXER) { 2115 sleepTimeShift = 0; 2116 } 2117 2118 CpuStats cpuStats; 2119 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2120 2121 acquireWakeLock(); 2122 2123 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2124 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2125 // and then that string will be logged at the next convenient opportunity. 2126 const char *logString = NULL; 2127 2128 checkSilentMode_l(); 2129 2130 while (!exitPending()) 2131 { 2132 cpuStats.sample(myName); 2133 2134 Vector< sp<EffectChain> > effectChains; 2135 2136 processConfigEvents(); 2137 2138 { // scope for mLock 2139 2140 Mutex::Autolock _l(mLock); 2141 2142 if (logString != NULL) { 2143 mNBLogWriter->logTimestamp(); 2144 mNBLogWriter->log(logString); 2145 logString = NULL; 2146 } 2147 2148 if (mLatchDValid) { 2149 mLatchQ = mLatchD; 2150 mLatchDValid = false; 2151 mLatchQValid = true; 2152 } 2153 2154 if (checkForNewParameters_l()) { 2155 cacheParameters_l(); 2156 } 2157 2158 saveOutputTracks(); 2159 if (mSignalPending) { 2160 // A signal was raised while we were unlocked 2161 mSignalPending = false; 2162 } else if (waitingAsyncCallback_l()) { 2163 if (exitPending()) { 2164 break; 2165 } 2166 releaseWakeLock_l(); 2167 ALOGV("wait async completion"); 2168 mWaitWorkCV.wait(mLock); 2169 ALOGV("async completion/wake"); 2170 acquireWakeLock_l(); 2171 standbyTime = systemTime() + standbyDelay; 2172 sleepTime = 0; 2173 2174 continue; 2175 } 2176 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2177 isSuspended()) { 2178 // put audio hardware into standby after short delay 2179 if (shouldStandby_l()) { 2180 2181 threadLoop_standby(); 2182 2183 mStandby = true; 2184 } 2185 2186 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2187 // we're about to wait, flush the binder command buffer 2188 IPCThreadState::self()->flushCommands(); 2189 2190 clearOutputTracks(); 2191 2192 if (exitPending()) { 2193 break; 2194 } 2195 2196 releaseWakeLock_l(); 2197 // wait until we have something to do... 2198 ALOGV("%s going to sleep", myName.string()); 2199 mWaitWorkCV.wait(mLock); 2200 ALOGV("%s waking up", myName.string()); 2201 acquireWakeLock_l(); 2202 2203 mMixerStatus = MIXER_IDLE; 2204 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2205 mBytesWritten = 0; 2206 mBytesRemaining = 0; 2207 checkSilentMode_l(); 2208 2209 standbyTime = systemTime() + standbyDelay; 2210 sleepTime = idleSleepTime; 2211 if (mType == MIXER) { 2212 sleepTimeShift = 0; 2213 } 2214 2215 continue; 2216 } 2217 } 2218 // mMixerStatusIgnoringFastTracks is also updated internally 2219 mMixerStatus = prepareTracks_l(&tracksToRemove); 2220 2221 // prevent any changes in effect chain list and in each effect chain 2222 // during mixing and effect process as the audio buffers could be deleted 2223 // or modified if an effect is created or deleted 2224 lockEffectChains_l(effectChains); 2225 } 2226 2227 if (mBytesRemaining == 0) { 2228 mCurrentWriteLength = 0; 2229 if (mMixerStatus == MIXER_TRACKS_READY) { 2230 // threadLoop_mix() sets mCurrentWriteLength 2231 threadLoop_mix(); 2232 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2233 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2234 // threadLoop_sleepTime sets sleepTime to 0 if data 2235 // must be written to HAL 2236 threadLoop_sleepTime(); 2237 if (sleepTime == 0) { 2238 mCurrentWriteLength = mixBufferSize; 2239 } 2240 } 2241 mBytesRemaining = mCurrentWriteLength; 2242 if (isSuspended()) { 2243 sleepTime = suspendSleepTimeUs(); 2244 // simulate write to HAL when suspended 2245 mBytesWritten += mixBufferSize; 2246 mBytesRemaining = 0; 2247 } 2248 2249 // only process effects if we're going to write 2250 if (sleepTime == 0) { 2251 for (size_t i = 0; i < effectChains.size(); i ++) { 2252 effectChains[i]->process_l(); 2253 } 2254 } 2255 } 2256 2257 // enable changes in effect chain 2258 unlockEffectChains(effectChains); 2259 2260 if (!waitingAsyncCallback()) { 2261 // sleepTime == 0 means we must write to audio hardware 2262 if (sleepTime == 0) { 2263 if (mBytesRemaining) { 2264 ssize_t ret = threadLoop_write(); 2265 if (ret < 0) { 2266 mBytesRemaining = 0; 2267 } else { 2268 mBytesWritten += ret; 2269 mBytesRemaining -= ret; 2270 } 2271 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2272 (mMixerStatus == MIXER_DRAIN_ALL)) { 2273 threadLoop_drain(); 2274 } 2275if (mType == MIXER) { 2276 // write blocked detection 2277 nsecs_t now = systemTime(); 2278 nsecs_t delta = now - mLastWriteTime; 2279 if (!mStandby && delta > maxPeriod) { 2280 mNumDelayedWrites++; 2281 if ((now - lastWarning) > kWarningThrottleNs) { 2282 ATRACE_NAME("underrun"); 2283 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2284 ns2ms(delta), mNumDelayedWrites, this); 2285 lastWarning = now; 2286 } 2287 } 2288} 2289 2290 mStandby = false; 2291 } else { 2292 usleep(sleepTime); 2293 } 2294 } 2295 2296 // Finally let go of removed track(s), without the lock held 2297 // since we can't guarantee the destructors won't acquire that 2298 // same lock. This will also mutate and push a new fast mixer state. 2299 threadLoop_removeTracks(tracksToRemove); 2300 tracksToRemove.clear(); 2301 2302 // FIXME I don't understand the need for this here; 2303 // it was in the original code but maybe the 2304 // assignment in saveOutputTracks() makes this unnecessary? 2305 clearOutputTracks(); 2306 2307 // Effect chains will be actually deleted here if they were removed from 2308 // mEffectChains list during mixing or effects processing 2309 effectChains.clear(); 2310 2311 // FIXME Note that the above .clear() is no longer necessary since effectChains 2312 // is now local to this block, but will keep it for now (at least until merge done). 2313 } 2314 2315 threadLoop_exit(); 2316 2317 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2318 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2319 // put output stream into standby mode 2320 if (!mStandby) { 2321 mOutput->stream->common.standby(&mOutput->stream->common); 2322 } 2323 } 2324 2325 releaseWakeLock(); 2326 2327 ALOGV("Thread %p type %d exiting", this, mType); 2328 return false; 2329} 2330 2331// removeTracks_l() must be called with ThreadBase::mLock held 2332void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2333{ 2334 size_t count = tracksToRemove.size(); 2335 if (count) { 2336 for (size_t i=0 ; i<count ; i++) { 2337 const sp<Track>& track = tracksToRemove.itemAt(i); 2338 mActiveTracks.remove(track); 2339 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2340 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2341 if (chain != 0) { 2342 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2343 track->sessionId()); 2344 chain->decActiveTrackCnt(); 2345 } 2346 if (track->isTerminated()) { 2347 removeTrack_l(track); 2348 } 2349 } 2350 } 2351 2352} 2353 2354status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2355{ 2356 if (mNormalSink != 0) { 2357 return mNormalSink->getTimestamp(timestamp); 2358 } 2359 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2360 uint64_t position64; 2361 int ret = mOutput->stream->get_presentation_position( 2362 mOutput->stream, &position64, ×tamp.mTime); 2363 if (ret == 0) { 2364 timestamp.mPosition = (uint32_t)position64; 2365 return NO_ERROR; 2366 } 2367 } 2368 return INVALID_OPERATION; 2369} 2370// ---------------------------------------------------------------------------- 2371 2372AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2373 audio_io_handle_t id, audio_devices_t device, type_t type) 2374 : PlaybackThread(audioFlinger, output, id, device, type), 2375 // mAudioMixer below 2376 // mFastMixer below 2377 mFastMixerFutex(0) 2378 // mOutputSink below 2379 // mPipeSink below 2380 // mNormalSink below 2381{ 2382 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2383 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2384 "mFrameCount=%d, mNormalFrameCount=%d", 2385 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2386 mNormalFrameCount); 2387 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2388 2389 // FIXME - Current mixer implementation only supports stereo output 2390 if (mChannelCount != FCC_2) { 2391 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2392 } 2393 2394 // create an NBAIO sink for the HAL output stream, and negotiate 2395 mOutputSink = new AudioStreamOutSink(output->stream); 2396 size_t numCounterOffers = 0; 2397 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2398 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2399 ALOG_ASSERT(index == 0); 2400 2401 // initialize fast mixer depending on configuration 2402 bool initFastMixer; 2403 switch (kUseFastMixer) { 2404 case FastMixer_Never: 2405 initFastMixer = false; 2406 break; 2407 case FastMixer_Always: 2408 initFastMixer = true; 2409 break; 2410 case FastMixer_Static: 2411 case FastMixer_Dynamic: 2412 initFastMixer = mFrameCount < mNormalFrameCount; 2413 break; 2414 } 2415 if (initFastMixer) { 2416 2417 // create a MonoPipe to connect our submix to FastMixer 2418 NBAIO_Format format = mOutputSink->format(); 2419 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2420 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2421 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2422 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2423 const NBAIO_Format offers[1] = {format}; 2424 size_t numCounterOffers = 0; 2425 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2426 ALOG_ASSERT(index == 0); 2427 monoPipe->setAvgFrames((mScreenState & 1) ? 2428 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2429 mPipeSink = monoPipe; 2430 2431#ifdef TEE_SINK 2432 if (mTeeSinkOutputEnabled) { 2433 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2434 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2435 numCounterOffers = 0; 2436 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2437 ALOG_ASSERT(index == 0); 2438 mTeeSink = teeSink; 2439 PipeReader *teeSource = new PipeReader(*teeSink); 2440 numCounterOffers = 0; 2441 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2442 ALOG_ASSERT(index == 0); 2443 mTeeSource = teeSource; 2444 } 2445#endif 2446 2447 // create fast mixer and configure it initially with just one fast track for our submix 2448 mFastMixer = new FastMixer(); 2449 FastMixerStateQueue *sq = mFastMixer->sq(); 2450#ifdef STATE_QUEUE_DUMP 2451 sq->setObserverDump(&mStateQueueObserverDump); 2452 sq->setMutatorDump(&mStateQueueMutatorDump); 2453#endif 2454 FastMixerState *state = sq->begin(); 2455 FastTrack *fastTrack = &state->mFastTracks[0]; 2456 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2457 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2458 fastTrack->mVolumeProvider = NULL; 2459 fastTrack->mGeneration++; 2460 state->mFastTracksGen++; 2461 state->mTrackMask = 1; 2462 // fast mixer will use the HAL output sink 2463 state->mOutputSink = mOutputSink.get(); 2464 state->mOutputSinkGen++; 2465 state->mFrameCount = mFrameCount; 2466 state->mCommand = FastMixerState::COLD_IDLE; 2467 // already done in constructor initialization list 2468 //mFastMixerFutex = 0; 2469 state->mColdFutexAddr = &mFastMixerFutex; 2470 state->mColdGen++; 2471 state->mDumpState = &mFastMixerDumpState; 2472#ifdef TEE_SINK 2473 state->mTeeSink = mTeeSink.get(); 2474#endif 2475 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2476 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2477 sq->end(); 2478 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2479 2480 // start the fast mixer 2481 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2482 pid_t tid = mFastMixer->getTid(); 2483 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2484 if (err != 0) { 2485 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2486 kPriorityFastMixer, getpid_cached, tid, err); 2487 } 2488 2489#ifdef AUDIO_WATCHDOG 2490 // create and start the watchdog 2491 mAudioWatchdog = new AudioWatchdog(); 2492 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2493 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2494 tid = mAudioWatchdog->getTid(); 2495 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2496 if (err != 0) { 2497 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2498 kPriorityFastMixer, getpid_cached, tid, err); 2499 } 2500#endif 2501 2502 } else { 2503 mFastMixer = NULL; 2504 } 2505 2506 switch (kUseFastMixer) { 2507 case FastMixer_Never: 2508 case FastMixer_Dynamic: 2509 mNormalSink = mOutputSink; 2510 break; 2511 case FastMixer_Always: 2512 mNormalSink = mPipeSink; 2513 break; 2514 case FastMixer_Static: 2515 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2516 break; 2517 } 2518} 2519 2520AudioFlinger::MixerThread::~MixerThread() 2521{ 2522 if (mFastMixer != NULL) { 2523 FastMixerStateQueue *sq = mFastMixer->sq(); 2524 FastMixerState *state = sq->begin(); 2525 if (state->mCommand == FastMixerState::COLD_IDLE) { 2526 int32_t old = android_atomic_inc(&mFastMixerFutex); 2527 if (old == -1) { 2528 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2529 } 2530 } 2531 state->mCommand = FastMixerState::EXIT; 2532 sq->end(); 2533 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2534 mFastMixer->join(); 2535 // Though the fast mixer thread has exited, it's state queue is still valid. 2536 // We'll use that extract the final state which contains one remaining fast track 2537 // corresponding to our sub-mix. 2538 state = sq->begin(); 2539 ALOG_ASSERT(state->mTrackMask == 1); 2540 FastTrack *fastTrack = &state->mFastTracks[0]; 2541 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2542 delete fastTrack->mBufferProvider; 2543 sq->end(false /*didModify*/); 2544 delete mFastMixer; 2545#ifdef AUDIO_WATCHDOG 2546 if (mAudioWatchdog != 0) { 2547 mAudioWatchdog->requestExit(); 2548 mAudioWatchdog->requestExitAndWait(); 2549 mAudioWatchdog.clear(); 2550 } 2551#endif 2552 } 2553 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2554 delete mAudioMixer; 2555} 2556 2557 2558uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2559{ 2560 if (mFastMixer != NULL) { 2561 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2562 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2563 } 2564 return latency; 2565} 2566 2567 2568void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2569{ 2570 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2571} 2572 2573ssize_t AudioFlinger::MixerThread::threadLoop_write() 2574{ 2575 // FIXME we should only do one push per cycle; confirm this is true 2576 // Start the fast mixer if it's not already running 2577 if (mFastMixer != NULL) { 2578 FastMixerStateQueue *sq = mFastMixer->sq(); 2579 FastMixerState *state = sq->begin(); 2580 if (state->mCommand != FastMixerState::MIX_WRITE && 2581 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2582 if (state->mCommand == FastMixerState::COLD_IDLE) { 2583 int32_t old = android_atomic_inc(&mFastMixerFutex); 2584 if (old == -1) { 2585 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2586 } 2587#ifdef AUDIO_WATCHDOG 2588 if (mAudioWatchdog != 0) { 2589 mAudioWatchdog->resume(); 2590 } 2591#endif 2592 } 2593 state->mCommand = FastMixerState::MIX_WRITE; 2594 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2595 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2596 sq->end(); 2597 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2598 if (kUseFastMixer == FastMixer_Dynamic) { 2599 mNormalSink = mPipeSink; 2600 } 2601 } else { 2602 sq->end(false /*didModify*/); 2603 } 2604 } 2605 return PlaybackThread::threadLoop_write(); 2606} 2607 2608void AudioFlinger::MixerThread::threadLoop_standby() 2609{ 2610 // Idle the fast mixer if it's currently running 2611 if (mFastMixer != NULL) { 2612 FastMixerStateQueue *sq = mFastMixer->sq(); 2613 FastMixerState *state = sq->begin(); 2614 if (!(state->mCommand & FastMixerState::IDLE)) { 2615 state->mCommand = FastMixerState::COLD_IDLE; 2616 state->mColdFutexAddr = &mFastMixerFutex; 2617 state->mColdGen++; 2618 mFastMixerFutex = 0; 2619 sq->end(); 2620 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2621 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2622 if (kUseFastMixer == FastMixer_Dynamic) { 2623 mNormalSink = mOutputSink; 2624 } 2625#ifdef AUDIO_WATCHDOG 2626 if (mAudioWatchdog != 0) { 2627 mAudioWatchdog->pause(); 2628 } 2629#endif 2630 } else { 2631 sq->end(false /*didModify*/); 2632 } 2633 } 2634 PlaybackThread::threadLoop_standby(); 2635} 2636 2637// Empty implementation for standard mixer 2638// Overridden for offloaded playback 2639void AudioFlinger::PlaybackThread::flushOutput_l() 2640{ 2641} 2642 2643bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2644{ 2645 return false; 2646} 2647 2648bool AudioFlinger::PlaybackThread::shouldStandby_l() 2649{ 2650 return !mStandby; 2651} 2652 2653bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2654{ 2655 Mutex::Autolock _l(mLock); 2656 return waitingAsyncCallback_l(); 2657} 2658 2659// shared by MIXER and DIRECT, overridden by DUPLICATING 2660void AudioFlinger::PlaybackThread::threadLoop_standby() 2661{ 2662 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2663 mOutput->stream->common.standby(&mOutput->stream->common); 2664 if (mUseAsyncWrite != 0) { 2665 // discard any pending drain or write ack by incrementing sequence 2666 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2667 mDrainSequence = (mDrainSequence + 2) & ~1; 2668 ALOG_ASSERT(mCallbackThread != 0); 2669 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2670 mCallbackThread->setDraining(mDrainSequence); 2671 } 2672} 2673 2674void AudioFlinger::MixerThread::threadLoop_mix() 2675{ 2676 // obtain the presentation timestamp of the next output buffer 2677 int64_t pts; 2678 status_t status = INVALID_OPERATION; 2679 2680 if (mNormalSink != 0) { 2681 status = mNormalSink->getNextWriteTimestamp(&pts); 2682 } else { 2683 status = mOutputSink->getNextWriteTimestamp(&pts); 2684 } 2685 2686 if (status != NO_ERROR) { 2687 pts = AudioBufferProvider::kInvalidPTS; 2688 } 2689 2690 // mix buffers... 2691 mAudioMixer->process(pts); 2692 mCurrentWriteLength = mixBufferSize; 2693 // increase sleep time progressively when application underrun condition clears. 2694 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2695 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2696 // such that we would underrun the audio HAL. 2697 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2698 sleepTimeShift--; 2699 } 2700 sleepTime = 0; 2701 standbyTime = systemTime() + standbyDelay; 2702 //TODO: delay standby when effects have a tail 2703} 2704 2705void AudioFlinger::MixerThread::threadLoop_sleepTime() 2706{ 2707 // If no tracks are ready, sleep once for the duration of an output 2708 // buffer size, then write 0s to the output 2709 if (sleepTime == 0) { 2710 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2711 sleepTime = activeSleepTime >> sleepTimeShift; 2712 if (sleepTime < kMinThreadSleepTimeUs) { 2713 sleepTime = kMinThreadSleepTimeUs; 2714 } 2715 // reduce sleep time in case of consecutive application underruns to avoid 2716 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2717 // duration we would end up writing less data than needed by the audio HAL if 2718 // the condition persists. 2719 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2720 sleepTimeShift++; 2721 } 2722 } else { 2723 sleepTime = idleSleepTime; 2724 } 2725 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2726 memset (mMixBuffer, 0, mixBufferSize); 2727 sleepTime = 0; 2728 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2729 "anticipated start"); 2730 } 2731 // TODO add standby time extension fct of effect tail 2732} 2733 2734// prepareTracks_l() must be called with ThreadBase::mLock held 2735AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2736 Vector< sp<Track> > *tracksToRemove) 2737{ 2738 2739 mixer_state mixerStatus = MIXER_IDLE; 2740 // find out which tracks need to be processed 2741 size_t count = mActiveTracks.size(); 2742 size_t mixedTracks = 0; 2743 size_t tracksWithEffect = 0; 2744 // counts only _active_ fast tracks 2745 size_t fastTracks = 0; 2746 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2747 2748 float masterVolume = mMasterVolume; 2749 bool masterMute = mMasterMute; 2750 2751 if (masterMute) { 2752 masterVolume = 0; 2753 } 2754 // Delegate master volume control to effect in output mix effect chain if needed 2755 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2756 if (chain != 0) { 2757 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2758 chain->setVolume_l(&v, &v); 2759 masterVolume = (float)((v + (1 << 23)) >> 24); 2760 chain.clear(); 2761 } 2762 2763 // prepare a new state to push 2764 FastMixerStateQueue *sq = NULL; 2765 FastMixerState *state = NULL; 2766 bool didModify = false; 2767 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2768 if (mFastMixer != NULL) { 2769 sq = mFastMixer->sq(); 2770 state = sq->begin(); 2771 } 2772 2773 for (size_t i=0 ; i<count ; i++) { 2774 const sp<Track> t = mActiveTracks[i].promote(); 2775 if (t == 0) { 2776 continue; 2777 } 2778 2779 // this const just means the local variable doesn't change 2780 Track* const track = t.get(); 2781 2782 // process fast tracks 2783 if (track->isFastTrack()) { 2784 2785 // It's theoretically possible (though unlikely) for a fast track to be created 2786 // and then removed within the same normal mix cycle. This is not a problem, as 2787 // the track never becomes active so it's fast mixer slot is never touched. 2788 // The converse, of removing an (active) track and then creating a new track 2789 // at the identical fast mixer slot within the same normal mix cycle, 2790 // is impossible because the slot isn't marked available until the end of each cycle. 2791 int j = track->mFastIndex; 2792 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2793 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2794 FastTrack *fastTrack = &state->mFastTracks[j]; 2795 2796 // Determine whether the track is currently in underrun condition, 2797 // and whether it had a recent underrun. 2798 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2799 FastTrackUnderruns underruns = ftDump->mUnderruns; 2800 uint32_t recentFull = (underruns.mBitFields.mFull - 2801 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2802 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2803 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2804 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2805 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2806 uint32_t recentUnderruns = recentPartial + recentEmpty; 2807 track->mObservedUnderruns = underruns; 2808 // don't count underruns that occur while stopping or pausing 2809 // or stopped which can occur when flush() is called while active 2810 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2811 recentUnderruns > 0) { 2812 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2813 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2814 } 2815 2816 // This is similar to the state machine for normal tracks, 2817 // with a few modifications for fast tracks. 2818 bool isActive = true; 2819 switch (track->mState) { 2820 case TrackBase::STOPPING_1: 2821 // track stays active in STOPPING_1 state until first underrun 2822 if (recentUnderruns > 0 || track->isTerminated()) { 2823 track->mState = TrackBase::STOPPING_2; 2824 } 2825 break; 2826 case TrackBase::PAUSING: 2827 // ramp down is not yet implemented 2828 track->setPaused(); 2829 break; 2830 case TrackBase::RESUMING: 2831 // ramp up is not yet implemented 2832 track->mState = TrackBase::ACTIVE; 2833 break; 2834 case TrackBase::ACTIVE: 2835 if (recentFull > 0 || recentPartial > 0) { 2836 // track has provided at least some frames recently: reset retry count 2837 track->mRetryCount = kMaxTrackRetries; 2838 } 2839 if (recentUnderruns == 0) { 2840 // no recent underruns: stay active 2841 break; 2842 } 2843 // there has recently been an underrun of some kind 2844 if (track->sharedBuffer() == 0) { 2845 // were any of the recent underruns "empty" (no frames available)? 2846 if (recentEmpty == 0) { 2847 // no, then ignore the partial underruns as they are allowed indefinitely 2848 break; 2849 } 2850 // there has recently been an "empty" underrun: decrement the retry counter 2851 if (--(track->mRetryCount) > 0) { 2852 break; 2853 } 2854 // indicate to client process that the track was disabled because of underrun; 2855 // it will then automatically call start() when data is available 2856 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2857 // remove from active list, but state remains ACTIVE [confusing but true] 2858 isActive = false; 2859 break; 2860 } 2861 // fall through 2862 case TrackBase::STOPPING_2: 2863 case TrackBase::PAUSED: 2864 case TrackBase::STOPPED: 2865 case TrackBase::FLUSHED: // flush() while active 2866 // Check for presentation complete if track is inactive 2867 // We have consumed all the buffers of this track. 2868 // This would be incomplete if we auto-paused on underrun 2869 { 2870 size_t audioHALFrames = 2871 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2872 size_t framesWritten = mBytesWritten / mFrameSize; 2873 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2874 // track stays in active list until presentation is complete 2875 break; 2876 } 2877 } 2878 if (track->isStopping_2()) { 2879 track->mState = TrackBase::STOPPED; 2880 } 2881 if (track->isStopped()) { 2882 // Can't reset directly, as fast mixer is still polling this track 2883 // track->reset(); 2884 // So instead mark this track as needing to be reset after push with ack 2885 resetMask |= 1 << i; 2886 } 2887 isActive = false; 2888 break; 2889 case TrackBase::IDLE: 2890 default: 2891 LOG_FATAL("unexpected track state %d", track->mState); 2892 } 2893 2894 if (isActive) { 2895 // was it previously inactive? 2896 if (!(state->mTrackMask & (1 << j))) { 2897 ExtendedAudioBufferProvider *eabp = track; 2898 VolumeProvider *vp = track; 2899 fastTrack->mBufferProvider = eabp; 2900 fastTrack->mVolumeProvider = vp; 2901 fastTrack->mSampleRate = track->mSampleRate; 2902 fastTrack->mChannelMask = track->mChannelMask; 2903 fastTrack->mGeneration++; 2904 state->mTrackMask |= 1 << j; 2905 didModify = true; 2906 // no acknowledgement required for newly active tracks 2907 } 2908 // cache the combined master volume and stream type volume for fast mixer; this 2909 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2910 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2911 ++fastTracks; 2912 } else { 2913 // was it previously active? 2914 if (state->mTrackMask & (1 << j)) { 2915 fastTrack->mBufferProvider = NULL; 2916 fastTrack->mGeneration++; 2917 state->mTrackMask &= ~(1 << j); 2918 didModify = true; 2919 // If any fast tracks were removed, we must wait for acknowledgement 2920 // because we're about to decrement the last sp<> on those tracks. 2921 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2922 } else { 2923 LOG_FATAL("fast track %d should have been active", j); 2924 } 2925 tracksToRemove->add(track); 2926 // Avoids a misleading display in dumpsys 2927 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2928 } 2929 continue; 2930 } 2931 2932 { // local variable scope to avoid goto warning 2933 2934 audio_track_cblk_t* cblk = track->cblk(); 2935 2936 // The first time a track is added we wait 2937 // for all its buffers to be filled before processing it 2938 int name = track->name(); 2939 // make sure that we have enough frames to mix one full buffer. 2940 // enforce this condition only once to enable draining the buffer in case the client 2941 // app does not call stop() and relies on underrun to stop: 2942 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2943 // during last round 2944 size_t desiredFrames; 2945 uint32_t sr = track->sampleRate(); 2946 if (sr == mSampleRate) { 2947 desiredFrames = mNormalFrameCount; 2948 } else { 2949 // +1 for rounding and +1 for additional sample needed for interpolation 2950 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2951 // add frames already consumed but not yet released by the resampler 2952 // because cblk->framesReady() will include these frames 2953 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2954 // the minimum track buffer size is normally twice the number of frames necessary 2955 // to fill one buffer and the resampler should not leave more than one buffer worth 2956 // of unreleased frames after each pass, but just in case... 2957 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2958 } 2959 uint32_t minFrames = 1; 2960 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2961 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2962 minFrames = desiredFrames; 2963 } 2964 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2965 size_t framesReady; 2966 if (track->sharedBuffer() == 0) { 2967 framesReady = track->framesReady(); 2968 } else if (track->isStopped()) { 2969 framesReady = 0; 2970 } else { 2971 framesReady = 1; 2972 } 2973 if ((framesReady >= minFrames) && track->isReady() && 2974 !track->isPaused() && !track->isTerminated()) 2975 { 2976 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2977 2978 mixedTracks++; 2979 2980 // track->mainBuffer() != mMixBuffer means there is an effect chain 2981 // connected to the track 2982 chain.clear(); 2983 if (track->mainBuffer() != mMixBuffer) { 2984 chain = getEffectChain_l(track->sessionId()); 2985 // Delegate volume control to effect in track effect chain if needed 2986 if (chain != 0) { 2987 tracksWithEffect++; 2988 } else { 2989 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2990 "session %d", 2991 name, track->sessionId()); 2992 } 2993 } 2994 2995 2996 int param = AudioMixer::VOLUME; 2997 if (track->mFillingUpStatus == Track::FS_FILLED) { 2998 // no ramp for the first volume setting 2999 track->mFillingUpStatus = Track::FS_ACTIVE; 3000 if (track->mState == TrackBase::RESUMING) { 3001 track->mState = TrackBase::ACTIVE; 3002 param = AudioMixer::RAMP_VOLUME; 3003 } 3004 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3005 // FIXME should not make a decision based on mServer 3006 } else if (cblk->mServer != 0) { 3007 // If the track is stopped before the first frame was mixed, 3008 // do not apply ramp 3009 param = AudioMixer::RAMP_VOLUME; 3010 } 3011 3012 // compute volume for this track 3013 uint32_t vl, vr, va; 3014 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3015 vl = vr = va = 0; 3016 if (track->isPausing()) { 3017 track->setPaused(); 3018 } 3019 } else { 3020 3021 // read original volumes with volume control 3022 float typeVolume = mStreamTypes[track->streamType()].volume; 3023 float v = masterVolume * typeVolume; 3024 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3025 uint32_t vlr = proxy->getVolumeLR(); 3026 vl = vlr & 0xFFFF; 3027 vr = vlr >> 16; 3028 // track volumes come from shared memory, so can't be trusted and must be clamped 3029 if (vl > MAX_GAIN_INT) { 3030 ALOGV("Track left volume out of range: %04X", vl); 3031 vl = MAX_GAIN_INT; 3032 } 3033 if (vr > MAX_GAIN_INT) { 3034 ALOGV("Track right volume out of range: %04X", vr); 3035 vr = MAX_GAIN_INT; 3036 } 3037 // now apply the master volume and stream type volume 3038 vl = (uint32_t)(v * vl) << 12; 3039 vr = (uint32_t)(v * vr) << 12; 3040 // assuming master volume and stream type volume each go up to 1.0, 3041 // vl and vr are now in 8.24 format 3042 3043 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3044 // send level comes from shared memory and so may be corrupt 3045 if (sendLevel > MAX_GAIN_INT) { 3046 ALOGV("Track send level out of range: %04X", sendLevel); 3047 sendLevel = MAX_GAIN_INT; 3048 } 3049 va = (uint32_t)(v * sendLevel); 3050 } 3051 3052 // Delegate volume control to effect in track effect chain if needed 3053 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3054 // Do not ramp volume if volume is controlled by effect 3055 param = AudioMixer::VOLUME; 3056 track->mHasVolumeController = true; 3057 } else { 3058 // force no volume ramp when volume controller was just disabled or removed 3059 // from effect chain to avoid volume spike 3060 if (track->mHasVolumeController) { 3061 param = AudioMixer::VOLUME; 3062 } 3063 track->mHasVolumeController = false; 3064 } 3065 3066 // Convert volumes from 8.24 to 4.12 format 3067 // This additional clamping is needed in case chain->setVolume_l() overshot 3068 vl = (vl + (1 << 11)) >> 12; 3069 if (vl > MAX_GAIN_INT) { 3070 vl = MAX_GAIN_INT; 3071 } 3072 vr = (vr + (1 << 11)) >> 12; 3073 if (vr > MAX_GAIN_INT) { 3074 vr = MAX_GAIN_INT; 3075 } 3076 3077 if (va > MAX_GAIN_INT) { 3078 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3079 } 3080 3081 // XXX: these things DON'T need to be done each time 3082 mAudioMixer->setBufferProvider(name, track); 3083 mAudioMixer->enable(name); 3084 3085 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3086 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3087 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3088 mAudioMixer->setParameter( 3089 name, 3090 AudioMixer::TRACK, 3091 AudioMixer::FORMAT, (void *)track->format()); 3092 mAudioMixer->setParameter( 3093 name, 3094 AudioMixer::TRACK, 3095 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3096 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3097 uint32_t maxSampleRate = mSampleRate * 2; 3098 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3099 if (reqSampleRate == 0) { 3100 reqSampleRate = mSampleRate; 3101 } else if (reqSampleRate > maxSampleRate) { 3102 reqSampleRate = maxSampleRate; 3103 } 3104 mAudioMixer->setParameter( 3105 name, 3106 AudioMixer::RESAMPLE, 3107 AudioMixer::SAMPLE_RATE, 3108 (void *)reqSampleRate); 3109 mAudioMixer->setParameter( 3110 name, 3111 AudioMixer::TRACK, 3112 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3113 mAudioMixer->setParameter( 3114 name, 3115 AudioMixer::TRACK, 3116 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3117 3118 // reset retry count 3119 track->mRetryCount = kMaxTrackRetries; 3120 3121 // If one track is ready, set the mixer ready if: 3122 // - the mixer was not ready during previous round OR 3123 // - no other track is not ready 3124 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3125 mixerStatus != MIXER_TRACKS_ENABLED) { 3126 mixerStatus = MIXER_TRACKS_READY; 3127 } 3128 } else { 3129 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3130 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3131 } 3132 // clear effect chain input buffer if an active track underruns to avoid sending 3133 // previous audio buffer again to effects 3134 chain = getEffectChain_l(track->sessionId()); 3135 if (chain != 0) { 3136 chain->clearInputBuffer(); 3137 } 3138 3139 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3140 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3141 track->isStopped() || track->isPaused()) { 3142 // We have consumed all the buffers of this track. 3143 // Remove it from the list of active tracks. 3144 // TODO: use actual buffer filling status instead of latency when available from 3145 // audio HAL 3146 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3147 size_t framesWritten = mBytesWritten / mFrameSize; 3148 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3149 if (track->isStopped()) { 3150 track->reset(); 3151 } 3152 tracksToRemove->add(track); 3153 } 3154 } else { 3155 // No buffers for this track. Give it a few chances to 3156 // fill a buffer, then remove it from active list. 3157 if (--(track->mRetryCount) <= 0) { 3158 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3159 tracksToRemove->add(track); 3160 // indicate to client process that the track was disabled because of underrun; 3161 // it will then automatically call start() when data is available 3162 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3163 // If one track is not ready, mark the mixer also not ready if: 3164 // - the mixer was ready during previous round OR 3165 // - no other track is ready 3166 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3167 mixerStatus != MIXER_TRACKS_READY) { 3168 mixerStatus = MIXER_TRACKS_ENABLED; 3169 } 3170 } 3171 mAudioMixer->disable(name); 3172 } 3173 3174 } // local variable scope to avoid goto warning 3175track_is_ready: ; 3176 3177 } 3178 3179 // Push the new FastMixer state if necessary 3180 bool pauseAudioWatchdog = false; 3181 if (didModify) { 3182 state->mFastTracksGen++; 3183 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3184 if (kUseFastMixer == FastMixer_Dynamic && 3185 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3186 state->mCommand = FastMixerState::COLD_IDLE; 3187 state->mColdFutexAddr = &mFastMixerFutex; 3188 state->mColdGen++; 3189 mFastMixerFutex = 0; 3190 if (kUseFastMixer == FastMixer_Dynamic) { 3191 mNormalSink = mOutputSink; 3192 } 3193 // If we go into cold idle, need to wait for acknowledgement 3194 // so that fast mixer stops doing I/O. 3195 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3196 pauseAudioWatchdog = true; 3197 } 3198 } 3199 if (sq != NULL) { 3200 sq->end(didModify); 3201 sq->push(block); 3202 } 3203#ifdef AUDIO_WATCHDOG 3204 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3205 mAudioWatchdog->pause(); 3206 } 3207#endif 3208 3209 // Now perform the deferred reset on fast tracks that have stopped 3210 while (resetMask != 0) { 3211 size_t i = __builtin_ctz(resetMask); 3212 ALOG_ASSERT(i < count); 3213 resetMask &= ~(1 << i); 3214 sp<Track> t = mActiveTracks[i].promote(); 3215 if (t == 0) { 3216 continue; 3217 } 3218 Track* track = t.get(); 3219 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3220 track->reset(); 3221 } 3222 3223 // remove all the tracks that need to be... 3224 removeTracks_l(*tracksToRemove); 3225 3226 // mix buffer must be cleared if all tracks are connected to an 3227 // effect chain as in this case the mixer will not write to 3228 // mix buffer and track effects will accumulate into it 3229 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3230 (mixedTracks == 0 && fastTracks > 0))) { 3231 // FIXME as a performance optimization, should remember previous zero status 3232 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3233 } 3234 3235 // if any fast tracks, then status is ready 3236 mMixerStatusIgnoringFastTracks = mixerStatus; 3237 if (fastTracks > 0) { 3238 mixerStatus = MIXER_TRACKS_READY; 3239 } 3240 return mixerStatus; 3241} 3242 3243// getTrackName_l() must be called with ThreadBase::mLock held 3244int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3245{ 3246 return mAudioMixer->getTrackName(channelMask, sessionId); 3247} 3248 3249// deleteTrackName_l() must be called with ThreadBase::mLock held 3250void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3251{ 3252 ALOGV("remove track (%d) and delete from mixer", name); 3253 mAudioMixer->deleteTrackName(name); 3254} 3255 3256// checkForNewParameters_l() must be called with ThreadBase::mLock held 3257bool AudioFlinger::MixerThread::checkForNewParameters_l() 3258{ 3259 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3260 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3261 bool reconfig = false; 3262 3263 while (!mNewParameters.isEmpty()) { 3264 3265 if (mFastMixer != NULL) { 3266 FastMixerStateQueue *sq = mFastMixer->sq(); 3267 FastMixerState *state = sq->begin(); 3268 if (!(state->mCommand & FastMixerState::IDLE)) { 3269 previousCommand = state->mCommand; 3270 state->mCommand = FastMixerState::HOT_IDLE; 3271 sq->end(); 3272 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3273 } else { 3274 sq->end(false /*didModify*/); 3275 } 3276 } 3277 3278 status_t status = NO_ERROR; 3279 String8 keyValuePair = mNewParameters[0]; 3280 AudioParameter param = AudioParameter(keyValuePair); 3281 int value; 3282 3283 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3284 reconfig = true; 3285 } 3286 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3287 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3288 status = BAD_VALUE; 3289 } else { 3290 reconfig = true; 3291 } 3292 } 3293 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3294 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3295 status = BAD_VALUE; 3296 } else { 3297 reconfig = true; 3298 } 3299 } 3300 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3301 // do not accept frame count changes if tracks are open as the track buffer 3302 // size depends on frame count and correct behavior would not be guaranteed 3303 // if frame count is changed after track creation 3304 if (!mTracks.isEmpty()) { 3305 status = INVALID_OPERATION; 3306 } else { 3307 reconfig = true; 3308 } 3309 } 3310 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3311#ifdef ADD_BATTERY_DATA 3312 // when changing the audio output device, call addBatteryData to notify 3313 // the change 3314 if (mOutDevice != value) { 3315 uint32_t params = 0; 3316 // check whether speaker is on 3317 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3318 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3319 } 3320 3321 audio_devices_t deviceWithoutSpeaker 3322 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3323 // check if any other device (except speaker) is on 3324 if (value & deviceWithoutSpeaker ) { 3325 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3326 } 3327 3328 if (params != 0) { 3329 addBatteryData(params); 3330 } 3331 } 3332#endif 3333 3334 // forward device change to effects that have requested to be 3335 // aware of attached audio device. 3336 if (value != AUDIO_DEVICE_NONE) { 3337 mOutDevice = value; 3338 for (size_t i = 0; i < mEffectChains.size(); i++) { 3339 mEffectChains[i]->setDevice_l(mOutDevice); 3340 } 3341 } 3342 } 3343 3344 if (status == NO_ERROR) { 3345 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3346 keyValuePair.string()); 3347 if (!mStandby && status == INVALID_OPERATION) { 3348 mOutput->stream->common.standby(&mOutput->stream->common); 3349 mStandby = true; 3350 mBytesWritten = 0; 3351 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3352 keyValuePair.string()); 3353 } 3354 if (status == NO_ERROR && reconfig) { 3355 readOutputParameters(); 3356 delete mAudioMixer; 3357 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3358 for (size_t i = 0; i < mTracks.size() ; i++) { 3359 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3360 if (name < 0) { 3361 break; 3362 } 3363 mTracks[i]->mName = name; 3364 } 3365 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3366 } 3367 } 3368 3369 mNewParameters.removeAt(0); 3370 3371 mParamStatus = status; 3372 mParamCond.signal(); 3373 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3374 // already timed out waiting for the status and will never signal the condition. 3375 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3376 } 3377 3378 if (!(previousCommand & FastMixerState::IDLE)) { 3379 ALOG_ASSERT(mFastMixer != NULL); 3380 FastMixerStateQueue *sq = mFastMixer->sq(); 3381 FastMixerState *state = sq->begin(); 3382 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3383 state->mCommand = previousCommand; 3384 sq->end(); 3385 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3386 } 3387 3388 return reconfig; 3389} 3390 3391 3392void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3393{ 3394 const size_t SIZE = 256; 3395 char buffer[SIZE]; 3396 String8 result; 3397 3398 PlaybackThread::dumpInternals(fd, args); 3399 3400 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3401 result.append(buffer); 3402 write(fd, result.string(), result.size()); 3403 3404 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3405 const FastMixerDumpState copy(mFastMixerDumpState); 3406 copy.dump(fd); 3407 3408#ifdef STATE_QUEUE_DUMP 3409 // Similar for state queue 3410 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3411 observerCopy.dump(fd); 3412 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3413 mutatorCopy.dump(fd); 3414#endif 3415 3416#ifdef TEE_SINK 3417 // Write the tee output to a .wav file 3418 dumpTee(fd, mTeeSource, mId); 3419#endif 3420 3421#ifdef AUDIO_WATCHDOG 3422 if (mAudioWatchdog != 0) { 3423 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3424 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3425 wdCopy.dump(fd); 3426 } 3427#endif 3428} 3429 3430uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3431{ 3432 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3433} 3434 3435uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3436{ 3437 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3438} 3439 3440void AudioFlinger::MixerThread::cacheParameters_l() 3441{ 3442 PlaybackThread::cacheParameters_l(); 3443 3444 // FIXME: Relaxed timing because of a certain device that can't meet latency 3445 // Should be reduced to 2x after the vendor fixes the driver issue 3446 // increase threshold again due to low power audio mode. The way this warning 3447 // threshold is calculated and its usefulness should be reconsidered anyway. 3448 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3449} 3450 3451// ---------------------------------------------------------------------------- 3452 3453AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3454 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3455 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3456 // mLeftVolFloat, mRightVolFloat 3457{ 3458} 3459 3460AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3461 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3462 ThreadBase::type_t type) 3463 : PlaybackThread(audioFlinger, output, id, device, type) 3464 // mLeftVolFloat, mRightVolFloat 3465{ 3466} 3467 3468AudioFlinger::DirectOutputThread::~DirectOutputThread() 3469{ 3470} 3471 3472void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3473{ 3474 audio_track_cblk_t* cblk = track->cblk(); 3475 float left, right; 3476 3477 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3478 left = right = 0; 3479 } else { 3480 float typeVolume = mStreamTypes[track->streamType()].volume; 3481 float v = mMasterVolume * typeVolume; 3482 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3483 uint32_t vlr = proxy->getVolumeLR(); 3484 float v_clamped = v * (vlr & 0xFFFF); 3485 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3486 left = v_clamped/MAX_GAIN; 3487 v_clamped = v * (vlr >> 16); 3488 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3489 right = v_clamped/MAX_GAIN; 3490 } 3491 3492 if (lastTrack) { 3493 if (left != mLeftVolFloat || right != mRightVolFloat) { 3494 mLeftVolFloat = left; 3495 mRightVolFloat = right; 3496 3497 // Convert volumes from float to 8.24 3498 uint32_t vl = (uint32_t)(left * (1 << 24)); 3499 uint32_t vr = (uint32_t)(right * (1 << 24)); 3500 3501 // Delegate volume control to effect in track effect chain if needed 3502 // only one effect chain can be present on DirectOutputThread, so if 3503 // there is one, the track is connected to it 3504 if (!mEffectChains.isEmpty()) { 3505 mEffectChains[0]->setVolume_l(&vl, &vr); 3506 left = (float)vl / (1 << 24); 3507 right = (float)vr / (1 << 24); 3508 } 3509 if (mOutput->stream->set_volume) { 3510 mOutput->stream->set_volume(mOutput->stream, left, right); 3511 } 3512 } 3513 } 3514} 3515 3516 3517AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3518 Vector< sp<Track> > *tracksToRemove 3519) 3520{ 3521 size_t count = mActiveTracks.size(); 3522 mixer_state mixerStatus = MIXER_IDLE; 3523 3524 // find out which tracks need to be processed 3525 for (size_t i = 0; i < count; i++) { 3526 sp<Track> t = mActiveTracks[i].promote(); 3527 // The track died recently 3528 if (t == 0) { 3529 continue; 3530 } 3531 3532 Track* const track = t.get(); 3533 audio_track_cblk_t* cblk = track->cblk(); 3534 3535 // The first time a track is added we wait 3536 // for all its buffers to be filled before processing it 3537 uint32_t minFrames; 3538 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3539 minFrames = mNormalFrameCount; 3540 } else { 3541 minFrames = 1; 3542 } 3543 // Only consider last track started for volume and mixer state control. 3544 // This is the last entry in mActiveTracks unless a track underruns. 3545 // As we only care about the transition phase between two tracks on a 3546 // direct output, it is not a problem to ignore the underrun case. 3547 bool last = (i == (count - 1)); 3548 3549 if ((track->framesReady() >= minFrames) && track->isReady() && 3550 !track->isPaused() && !track->isTerminated()) 3551 { 3552 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3553 3554 if (track->mFillingUpStatus == Track::FS_FILLED) { 3555 track->mFillingUpStatus = Track::FS_ACTIVE; 3556 // make sure processVolume_l() will apply new volume even if 0 3557 mLeftVolFloat = mRightVolFloat = -1.0; 3558 if (track->mState == TrackBase::RESUMING) { 3559 track->mState = TrackBase::ACTIVE; 3560 } 3561 } 3562 3563 // compute volume for this track 3564 processVolume_l(track, last); 3565 if (last) { 3566 // reset retry count 3567 track->mRetryCount = kMaxTrackRetriesDirect; 3568 mActiveTrack = t; 3569 mixerStatus = MIXER_TRACKS_READY; 3570 } 3571 } else { 3572 // clear effect chain input buffer if the last active track started underruns 3573 // to avoid sending previous audio buffer again to effects 3574 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3575 mEffectChains[0]->clearInputBuffer(); 3576 } 3577 3578 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3579 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3580 track->isStopped() || track->isPaused()) { 3581 // We have consumed all the buffers of this track. 3582 // Remove it from the list of active tracks. 3583 // TODO: implement behavior for compressed audio 3584 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3585 size_t framesWritten = mBytesWritten / mFrameSize; 3586 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3587 if (track->isStopped()) { 3588 track->reset(); 3589 } 3590 tracksToRemove->add(track); 3591 } 3592 } else { 3593 // No buffers for this track. Give it a few chances to 3594 // fill a buffer, then remove it from active list. 3595 // Only consider last track started for mixer state control 3596 if (--(track->mRetryCount) <= 0) { 3597 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3598 tracksToRemove->add(track); 3599 } else if (last) { 3600 mixerStatus = MIXER_TRACKS_ENABLED; 3601 } 3602 } 3603 } 3604 } 3605 3606 // remove all the tracks that need to be... 3607 removeTracks_l(*tracksToRemove); 3608 3609 return mixerStatus; 3610} 3611 3612void AudioFlinger::DirectOutputThread::threadLoop_mix() 3613{ 3614 size_t frameCount = mFrameCount; 3615 int8_t *curBuf = (int8_t *)mMixBuffer; 3616 // output audio to hardware 3617 while (frameCount) { 3618 AudioBufferProvider::Buffer buffer; 3619 buffer.frameCount = frameCount; 3620 mActiveTrack->getNextBuffer(&buffer); 3621 if (buffer.raw == NULL) { 3622 memset(curBuf, 0, frameCount * mFrameSize); 3623 break; 3624 } 3625 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3626 frameCount -= buffer.frameCount; 3627 curBuf += buffer.frameCount * mFrameSize; 3628 mActiveTrack->releaseBuffer(&buffer); 3629 } 3630 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3631 sleepTime = 0; 3632 standbyTime = systemTime() + standbyDelay; 3633 mActiveTrack.clear(); 3634} 3635 3636void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3637{ 3638 if (sleepTime == 0) { 3639 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3640 sleepTime = activeSleepTime; 3641 } else { 3642 sleepTime = idleSleepTime; 3643 } 3644 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3645 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3646 sleepTime = 0; 3647 } 3648} 3649 3650// getTrackName_l() must be called with ThreadBase::mLock held 3651int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3652 int sessionId) 3653{ 3654 return 0; 3655} 3656 3657// deleteTrackName_l() must be called with ThreadBase::mLock held 3658void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3659{ 3660} 3661 3662// checkForNewParameters_l() must be called with ThreadBase::mLock held 3663bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3664{ 3665 bool reconfig = false; 3666 3667 while (!mNewParameters.isEmpty()) { 3668 status_t status = NO_ERROR; 3669 String8 keyValuePair = mNewParameters[0]; 3670 AudioParameter param = AudioParameter(keyValuePair); 3671 int value; 3672 3673 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3674 // do not accept frame count changes if tracks are open as the track buffer 3675 // size depends on frame count and correct behavior would not be garantied 3676 // if frame count is changed after track creation 3677 if (!mTracks.isEmpty()) { 3678 status = INVALID_OPERATION; 3679 } else { 3680 reconfig = true; 3681 } 3682 } 3683 if (status == NO_ERROR) { 3684 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3685 keyValuePair.string()); 3686 if (!mStandby && status == INVALID_OPERATION) { 3687 mOutput->stream->common.standby(&mOutput->stream->common); 3688 mStandby = true; 3689 mBytesWritten = 0; 3690 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3691 keyValuePair.string()); 3692 } 3693 if (status == NO_ERROR && reconfig) { 3694 readOutputParameters(); 3695 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3696 } 3697 } 3698 3699 mNewParameters.removeAt(0); 3700 3701 mParamStatus = status; 3702 mParamCond.signal(); 3703 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3704 // already timed out waiting for the status and will never signal the condition. 3705 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3706 } 3707 return reconfig; 3708} 3709 3710uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3711{ 3712 uint32_t time; 3713 if (audio_is_linear_pcm(mFormat)) { 3714 time = PlaybackThread::activeSleepTimeUs(); 3715 } else { 3716 time = 10000; 3717 } 3718 return time; 3719} 3720 3721uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3722{ 3723 uint32_t time; 3724 if (audio_is_linear_pcm(mFormat)) { 3725 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3726 } else { 3727 time = 10000; 3728 } 3729 return time; 3730} 3731 3732uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3733{ 3734 uint32_t time; 3735 if (audio_is_linear_pcm(mFormat)) { 3736 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3737 } else { 3738 time = 10000; 3739 } 3740 return time; 3741} 3742 3743void AudioFlinger::DirectOutputThread::cacheParameters_l() 3744{ 3745 PlaybackThread::cacheParameters_l(); 3746 3747 // use shorter standby delay as on normal output to release 3748 // hardware resources as soon as possible 3749 if (audio_is_linear_pcm(mFormat)) { 3750 standbyDelay = microseconds(activeSleepTime*2); 3751 } else { 3752 standbyDelay = kOffloadStandbyDelayNs; 3753 } 3754} 3755 3756// ---------------------------------------------------------------------------- 3757 3758AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3759 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3760 : Thread(false /*canCallJava*/), 3761 mPlaybackThread(playbackThread), 3762 mWriteAckSequence(0), 3763 mDrainSequence(0) 3764{ 3765} 3766 3767AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3768{ 3769} 3770 3771void AudioFlinger::AsyncCallbackThread::onFirstRef() 3772{ 3773 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3774} 3775 3776bool AudioFlinger::AsyncCallbackThread::threadLoop() 3777{ 3778 while (!exitPending()) { 3779 uint32_t writeAckSequence; 3780 uint32_t drainSequence; 3781 3782 { 3783 Mutex::Autolock _l(mLock); 3784 mWaitWorkCV.wait(mLock); 3785 if (exitPending()) { 3786 break; 3787 } 3788 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 3789 mWriteAckSequence, mDrainSequence); 3790 writeAckSequence = mWriteAckSequence; 3791 mWriteAckSequence &= ~1; 3792 drainSequence = mDrainSequence; 3793 mDrainSequence &= ~1; 3794 } 3795 { 3796 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 3797 if (playbackThread != 0) { 3798 if (writeAckSequence & 1) { 3799 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 3800 } 3801 if (drainSequence & 1) { 3802 playbackThread->resetDraining(drainSequence >> 1); 3803 } 3804 } 3805 } 3806 } 3807 return false; 3808} 3809 3810void AudioFlinger::AsyncCallbackThread::exit() 3811{ 3812 ALOGV("AsyncCallbackThread::exit"); 3813 Mutex::Autolock _l(mLock); 3814 requestExit(); 3815 mWaitWorkCV.broadcast(); 3816} 3817 3818void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 3819{ 3820 Mutex::Autolock _l(mLock); 3821 // bit 0 is cleared 3822 mWriteAckSequence = sequence << 1; 3823} 3824 3825void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 3826{ 3827 Mutex::Autolock _l(mLock); 3828 // ignore unexpected callbacks 3829 if (mWriteAckSequence & 2) { 3830 mWriteAckSequence |= 1; 3831 mWaitWorkCV.signal(); 3832 } 3833} 3834 3835void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 3836{ 3837 Mutex::Autolock _l(mLock); 3838 // bit 0 is cleared 3839 mDrainSequence = sequence << 1; 3840} 3841 3842void AudioFlinger::AsyncCallbackThread::resetDraining() 3843{ 3844 Mutex::Autolock _l(mLock); 3845 // ignore unexpected callbacks 3846 if (mDrainSequence & 2) { 3847 mDrainSequence |= 1; 3848 mWaitWorkCV.signal(); 3849 } 3850} 3851 3852 3853// ---------------------------------------------------------------------------- 3854AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3855 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3856 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3857 mHwPaused(false), 3858 mPausedBytesRemaining(0) 3859{ 3860} 3861 3862AudioFlinger::OffloadThread::~OffloadThread() 3863{ 3864 mPreviousTrack.clear(); 3865} 3866 3867void AudioFlinger::OffloadThread::threadLoop_exit() 3868{ 3869 if (mFlushPending || mHwPaused) { 3870 // If a flush is pending or track was paused, just discard buffered data 3871 flushHw_l(); 3872 } else { 3873 mMixerStatus = MIXER_DRAIN_ALL; 3874 threadLoop_drain(); 3875 } 3876 mCallbackThread->exit(); 3877 PlaybackThread::threadLoop_exit(); 3878} 3879 3880AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3881 Vector< sp<Track> > *tracksToRemove 3882) 3883{ 3884 size_t count = mActiveTracks.size(); 3885 3886 mixer_state mixerStatus = MIXER_IDLE; 3887 bool doHwPause = false; 3888 bool doHwResume = false; 3889 3890 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 3891 3892 // find out which tracks need to be processed 3893 for (size_t i = 0; i < count; i++) { 3894 sp<Track> t = mActiveTracks[i].promote(); 3895 // The track died recently 3896 if (t == 0) { 3897 continue; 3898 } 3899 Track* const track = t.get(); 3900 audio_track_cblk_t* cblk = track->cblk(); 3901 if (mPreviousTrack != NULL) { 3902 if (t != mPreviousTrack) { 3903 // Flush any data still being written from last track 3904 mBytesRemaining = 0; 3905 if (mPausedBytesRemaining) { 3906 // Last track was paused so we also need to flush saved 3907 // mixbuffer state and invalidate track so that it will 3908 // re-submit that unwritten data when it is next resumed 3909 mPausedBytesRemaining = 0; 3910 // Invalidate is a bit drastic - would be more efficient 3911 // to have a flag to tell client that some of the 3912 // previously written data was lost 3913 mPreviousTrack->invalidate(); 3914 } 3915 } 3916 } 3917 mPreviousTrack = t; 3918 bool last = (i == (count - 1)); 3919 if (track->isPausing()) { 3920 track->setPaused(); 3921 if (last) { 3922 if (!mHwPaused) { 3923 doHwPause = true; 3924 mHwPaused = true; 3925 } 3926 // If we were part way through writing the mixbuffer to 3927 // the HAL we must save this until we resume 3928 // BUG - this will be wrong if a different track is made active, 3929 // in that case we want to discard the pending data in the 3930 // mixbuffer and tell the client to present it again when the 3931 // track is resumed 3932 mPausedWriteLength = mCurrentWriteLength; 3933 mPausedBytesRemaining = mBytesRemaining; 3934 mBytesRemaining = 0; // stop writing 3935 } 3936 tracksToRemove->add(track); 3937 } else if (track->framesReady() && track->isReady() && 3938 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 3939 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3940 if (track->mFillingUpStatus == Track::FS_FILLED) { 3941 track->mFillingUpStatus = Track::FS_ACTIVE; 3942 // make sure processVolume_l() will apply new volume even if 0 3943 mLeftVolFloat = mRightVolFloat = -1.0; 3944 if (track->mState == TrackBase::RESUMING) { 3945 track->mState = TrackBase::ACTIVE; 3946 if (last) { 3947 if (mPausedBytesRemaining) { 3948 // Need to continue write that was interrupted 3949 mCurrentWriteLength = mPausedWriteLength; 3950 mBytesRemaining = mPausedBytesRemaining; 3951 mPausedBytesRemaining = 0; 3952 } 3953 if (mHwPaused) { 3954 doHwResume = true; 3955 mHwPaused = false; 3956 // threadLoop_mix() will handle the case that we need to 3957 // resume an interrupted write 3958 } 3959 // enable write to audio HAL 3960 sleepTime = 0; 3961 } 3962 } 3963 } 3964 3965 if (last) { 3966 // reset retry count 3967 track->mRetryCount = kMaxTrackRetriesOffload; 3968 mActiveTrack = t; 3969 mixerStatus = MIXER_TRACKS_READY; 3970 } 3971 } else { 3972 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3973 if (track->isStopping_1()) { 3974 // Hardware buffer can hold a large amount of audio so we must 3975 // wait for all current track's data to drain before we say 3976 // that the track is stopped. 3977 if (mBytesRemaining == 0) { 3978 // Only start draining when all data in mixbuffer 3979 // has been written 3980 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3981 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3982 if (last) { 3983 sleepTime = 0; 3984 standbyTime = systemTime() + standbyDelay; 3985 mixerStatus = MIXER_DRAIN_TRACK; 3986 mDrainSequence += 2; 3987 if (mHwPaused) { 3988 // It is possible to move from PAUSED to STOPPING_1 without 3989 // a resume so we must ensure hardware is running 3990 mOutput->stream->resume(mOutput->stream); 3991 mHwPaused = false; 3992 } 3993 } 3994 } 3995 } else if (track->isStopping_2()) { 3996 // Drain has completed, signal presentation complete 3997 if (!(mDrainSequence & 1) || !last) { 3998 track->mState = TrackBase::STOPPED; 3999 size_t audioHALFrames = 4000 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4001 size_t framesWritten = 4002 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4003 track->presentationComplete(framesWritten, audioHALFrames); 4004 track->reset(); 4005 tracksToRemove->add(track); 4006 } 4007 } else { 4008 // No buffers for this track. Give it a few chances to 4009 // fill a buffer, then remove it from active list. 4010 if (--(track->mRetryCount) <= 0) { 4011 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4012 track->name()); 4013 tracksToRemove->add(track); 4014 } else if (last){ 4015 mixerStatus = MIXER_TRACKS_ENABLED; 4016 } 4017 } 4018 } 4019 // compute volume for this track 4020 processVolume_l(track, last); 4021 } 4022 4023 // make sure the pause/flush/resume sequence is executed in the right order 4024 if (doHwPause) { 4025 mOutput->stream->pause(mOutput->stream); 4026 } 4027 if (mFlushPending) { 4028 flushHw_l(); 4029 mFlushPending = false; 4030 } 4031 if (doHwResume) { 4032 mOutput->stream->resume(mOutput->stream); 4033 } 4034 4035 // remove all the tracks that need to be... 4036 removeTracks_l(*tracksToRemove); 4037 4038 return mixerStatus; 4039} 4040 4041void AudioFlinger::OffloadThread::flushOutput_l() 4042{ 4043 mFlushPending = true; 4044} 4045 4046// must be called with thread mutex locked 4047bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4048{ 4049 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4050 mWriteAckSequence, mDrainSequence); 4051 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4052 return true; 4053 } 4054 return false; 4055} 4056 4057// must be called with thread mutex locked 4058bool AudioFlinger::OffloadThread::shouldStandby_l() 4059{ 4060 bool TrackPaused = false; 4061 4062 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4063 // after a timeout and we will enter standby then. 4064 if (mTracks.size() > 0) { 4065 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4066 } 4067 4068 return !mStandby && !TrackPaused; 4069} 4070 4071 4072bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4073{ 4074 Mutex::Autolock _l(mLock); 4075 return waitingAsyncCallback_l(); 4076} 4077 4078void AudioFlinger::OffloadThread::flushHw_l() 4079{ 4080 mOutput->stream->flush(mOutput->stream); 4081 // Flush anything still waiting in the mixbuffer 4082 mCurrentWriteLength = 0; 4083 mBytesRemaining = 0; 4084 mPausedWriteLength = 0; 4085 mPausedBytesRemaining = 0; 4086 if (mUseAsyncWrite) { 4087 // discard any pending drain or write ack by incrementing sequence 4088 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4089 mDrainSequence = (mDrainSequence + 2) & ~1; 4090 ALOG_ASSERT(mCallbackThread != 0); 4091 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4092 mCallbackThread->setDraining(mDrainSequence); 4093 } 4094} 4095 4096// ---------------------------------------------------------------------------- 4097 4098AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4099 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4100 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4101 DUPLICATING), 4102 mWaitTimeMs(UINT_MAX) 4103{ 4104 addOutputTrack(mainThread); 4105} 4106 4107AudioFlinger::DuplicatingThread::~DuplicatingThread() 4108{ 4109 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4110 mOutputTracks[i]->destroy(); 4111 } 4112} 4113 4114void AudioFlinger::DuplicatingThread::threadLoop_mix() 4115{ 4116 // mix buffers... 4117 if (outputsReady(outputTracks)) { 4118 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4119 } else { 4120 memset(mMixBuffer, 0, mixBufferSize); 4121 } 4122 sleepTime = 0; 4123 writeFrames = mNormalFrameCount; 4124 mCurrentWriteLength = mixBufferSize; 4125 standbyTime = systemTime() + standbyDelay; 4126} 4127 4128void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4129{ 4130 if (sleepTime == 0) { 4131 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4132 sleepTime = activeSleepTime; 4133 } else { 4134 sleepTime = idleSleepTime; 4135 } 4136 } else if (mBytesWritten != 0) { 4137 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4138 writeFrames = mNormalFrameCount; 4139 memset(mMixBuffer, 0, mixBufferSize); 4140 } else { 4141 // flush remaining overflow buffers in output tracks 4142 writeFrames = 0; 4143 } 4144 sleepTime = 0; 4145 } 4146} 4147 4148ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4149{ 4150 for (size_t i = 0; i < outputTracks.size(); i++) { 4151 outputTracks[i]->write(mMixBuffer, writeFrames); 4152 } 4153 return (ssize_t)mixBufferSize; 4154} 4155 4156void AudioFlinger::DuplicatingThread::threadLoop_standby() 4157{ 4158 // DuplicatingThread implements standby by stopping all tracks 4159 for (size_t i = 0; i < outputTracks.size(); i++) { 4160 outputTracks[i]->stop(); 4161 } 4162} 4163 4164void AudioFlinger::DuplicatingThread::saveOutputTracks() 4165{ 4166 outputTracks = mOutputTracks; 4167} 4168 4169void AudioFlinger::DuplicatingThread::clearOutputTracks() 4170{ 4171 outputTracks.clear(); 4172} 4173 4174void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4175{ 4176 Mutex::Autolock _l(mLock); 4177 // FIXME explain this formula 4178 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4179 OutputTrack *outputTrack = new OutputTrack(thread, 4180 this, 4181 mSampleRate, 4182 mFormat, 4183 mChannelMask, 4184 frameCount); 4185 if (outputTrack->cblk() != NULL) { 4186 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4187 mOutputTracks.add(outputTrack); 4188 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4189 updateWaitTime_l(); 4190 } 4191} 4192 4193void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4194{ 4195 Mutex::Autolock _l(mLock); 4196 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4197 if (mOutputTracks[i]->thread() == thread) { 4198 mOutputTracks[i]->destroy(); 4199 mOutputTracks.removeAt(i); 4200 updateWaitTime_l(); 4201 return; 4202 } 4203 } 4204 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4205} 4206 4207// caller must hold mLock 4208void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4209{ 4210 mWaitTimeMs = UINT_MAX; 4211 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4212 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4213 if (strong != 0) { 4214 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4215 if (waitTimeMs < mWaitTimeMs) { 4216 mWaitTimeMs = waitTimeMs; 4217 } 4218 } 4219 } 4220} 4221 4222 4223bool AudioFlinger::DuplicatingThread::outputsReady( 4224 const SortedVector< sp<OutputTrack> > &outputTracks) 4225{ 4226 for (size_t i = 0; i < outputTracks.size(); i++) { 4227 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4228 if (thread == 0) { 4229 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4230 outputTracks[i].get()); 4231 return false; 4232 } 4233 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4234 // see note at standby() declaration 4235 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4236 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4237 thread.get()); 4238 return false; 4239 } 4240 } 4241 return true; 4242} 4243 4244uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4245{ 4246 return (mWaitTimeMs * 1000) / 2; 4247} 4248 4249void AudioFlinger::DuplicatingThread::cacheParameters_l() 4250{ 4251 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4252 updateWaitTime_l(); 4253 4254 MixerThread::cacheParameters_l(); 4255} 4256 4257// ---------------------------------------------------------------------------- 4258// Record 4259// ---------------------------------------------------------------------------- 4260 4261AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4262 AudioStreamIn *input, 4263 uint32_t sampleRate, 4264 audio_channel_mask_t channelMask, 4265 audio_io_handle_t id, 4266 audio_devices_t outDevice, 4267 audio_devices_t inDevice 4268#ifdef TEE_SINK 4269 , const sp<NBAIO_Sink>& teeSink 4270#endif 4271 ) : 4272 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4273 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4274 // mRsmpInIndex and mBufferSize set by readInputParameters() 4275 mReqChannelCount(popcount(channelMask)), 4276 mReqSampleRate(sampleRate) 4277 // mBytesRead is only meaningful while active, and so is cleared in start() 4278 // (but might be better to also clear here for dump?) 4279#ifdef TEE_SINK 4280 , mTeeSink(teeSink) 4281#endif 4282{ 4283 snprintf(mName, kNameLength, "AudioIn_%X", id); 4284 4285 readInputParameters(); 4286 mClientUid = IPCThreadState::self()->getCallingUid(); 4287} 4288 4289 4290AudioFlinger::RecordThread::~RecordThread() 4291{ 4292 delete[] mRsmpInBuffer; 4293 delete mResampler; 4294 delete[] mRsmpOutBuffer; 4295} 4296 4297void AudioFlinger::RecordThread::onFirstRef() 4298{ 4299 run(mName, PRIORITY_URGENT_AUDIO); 4300} 4301 4302status_t AudioFlinger::RecordThread::readyToRun() 4303{ 4304 status_t status = initCheck(); 4305 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4306 return status; 4307} 4308 4309bool AudioFlinger::RecordThread::threadLoop() 4310{ 4311 AudioBufferProvider::Buffer buffer; 4312 sp<RecordTrack> activeTrack; 4313 Vector< sp<EffectChain> > effectChains; 4314 4315 nsecs_t lastWarning = 0; 4316 4317 inputStandBy(); 4318 acquireWakeLock(mClientUid); 4319 4320 // used to verify we've read at least once before evaluating how many bytes were read 4321 bool readOnce = false; 4322 4323 // start recording 4324 while (!exitPending()) { 4325 4326 processConfigEvents(); 4327 4328 { // scope for mLock 4329 Mutex::Autolock _l(mLock); 4330 checkForNewParameters_l(); 4331 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4332 standby(); 4333 4334 if (exitPending()) { 4335 break; 4336 } 4337 4338 releaseWakeLock_l(); 4339 ALOGV("RecordThread: loop stopping"); 4340 // go to sleep 4341 mWaitWorkCV.wait(mLock); 4342 ALOGV("RecordThread: loop starting"); 4343 acquireWakeLock_l(mClientUid); 4344 continue; 4345 } 4346 if (mActiveTrack != 0) { 4347 if (mActiveTrack->isTerminated()) { 4348 removeTrack_l(mActiveTrack); 4349 mActiveTrack.clear(); 4350 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4351 standby(); 4352 mActiveTrack.clear(); 4353 mStartStopCond.broadcast(); 4354 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4355 if (mReqChannelCount != mActiveTrack->channelCount()) { 4356 mActiveTrack.clear(); 4357 mStartStopCond.broadcast(); 4358 } else if (readOnce) { 4359 // record start succeeds only if first read from audio input 4360 // succeeds 4361 if (mBytesRead >= 0) { 4362 mActiveTrack->mState = TrackBase::ACTIVE; 4363 } else { 4364 mActiveTrack.clear(); 4365 } 4366 mStartStopCond.broadcast(); 4367 } 4368 mStandby = false; 4369 } 4370 } 4371 4372 lockEffectChains_l(effectChains); 4373 } 4374 4375 if (mActiveTrack != 0) { 4376 if (mActiveTrack->mState != TrackBase::ACTIVE && 4377 mActiveTrack->mState != TrackBase::RESUMING) { 4378 unlockEffectChains(effectChains); 4379 usleep(kRecordThreadSleepUs); 4380 continue; 4381 } 4382 for (size_t i = 0; i < effectChains.size(); i ++) { 4383 effectChains[i]->process_l(); 4384 } 4385 4386 buffer.frameCount = mFrameCount; 4387 status_t status = mActiveTrack->getNextBuffer(&buffer); 4388 if (status == NO_ERROR) { 4389 readOnce = true; 4390 size_t framesOut = buffer.frameCount; 4391 if (mResampler == NULL) { 4392 // no resampling 4393 while (framesOut) { 4394 size_t framesIn = mFrameCount - mRsmpInIndex; 4395 if (framesIn) { 4396 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4397 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4398 mActiveTrack->mFrameSize; 4399 if (framesIn > framesOut) 4400 framesIn = framesOut; 4401 mRsmpInIndex += framesIn; 4402 framesOut -= framesIn; 4403 if (mChannelCount == mReqChannelCount) { 4404 memcpy(dst, src, framesIn * mFrameSize); 4405 } else { 4406 if (mChannelCount == 1) { 4407 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4408 (int16_t *)src, framesIn); 4409 } else { 4410 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4411 (int16_t *)src, framesIn); 4412 } 4413 } 4414 } 4415 if (framesOut && mFrameCount == mRsmpInIndex) { 4416 void *readInto; 4417 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4418 readInto = buffer.raw; 4419 framesOut = 0; 4420 } else { 4421 readInto = mRsmpInBuffer; 4422 mRsmpInIndex = 0; 4423 } 4424 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4425 mBufferSize); 4426 if (mBytesRead <= 0) { 4427 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4428 { 4429 ALOGE("Error reading audio input"); 4430 // Force input into standby so that it tries to 4431 // recover at next read attempt 4432 inputStandBy(); 4433 usleep(kRecordThreadSleepUs); 4434 } 4435 mRsmpInIndex = mFrameCount; 4436 framesOut = 0; 4437 buffer.frameCount = 0; 4438 } 4439#ifdef TEE_SINK 4440 else if (mTeeSink != 0) { 4441 (void) mTeeSink->write(readInto, 4442 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4443 } 4444#endif 4445 } 4446 } 4447 } else { 4448 // resampling 4449 4450 // resampler accumulates, but we only have one source track 4451 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4452 // alter output frame count as if we were expecting stereo samples 4453 if (mChannelCount == 1 && mReqChannelCount == 1) { 4454 framesOut >>= 1; 4455 } 4456 mResampler->resample(mRsmpOutBuffer, framesOut, 4457 this /* AudioBufferProvider* */); 4458 // ditherAndClamp() works as long as all buffers returned by 4459 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4460 if (mChannelCount == 2 && mReqChannelCount == 1) { 4461 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4462 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4463 // the resampler always outputs stereo samples: 4464 // do post stereo to mono conversion 4465 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4466 framesOut); 4467 } else { 4468 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4469 } 4470 // now done with mRsmpOutBuffer 4471 4472 } 4473 if (mFramestoDrop == 0) { 4474 mActiveTrack->releaseBuffer(&buffer); 4475 } else { 4476 if (mFramestoDrop > 0) { 4477 mFramestoDrop -= buffer.frameCount; 4478 if (mFramestoDrop <= 0) { 4479 clearSyncStartEvent(); 4480 } 4481 } else { 4482 mFramestoDrop += buffer.frameCount; 4483 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4484 mSyncStartEvent->isCancelled()) { 4485 ALOGW("Synced record %s, session %d, trigger session %d", 4486 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4487 mActiveTrack->sessionId(), 4488 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4489 clearSyncStartEvent(); 4490 } 4491 } 4492 } 4493 mActiveTrack->clearOverflow(); 4494 } 4495 // client isn't retrieving buffers fast enough 4496 else { 4497 if (!mActiveTrack->setOverflow()) { 4498 nsecs_t now = systemTime(); 4499 if ((now - lastWarning) > kWarningThrottleNs) { 4500 ALOGW("RecordThread: buffer overflow"); 4501 lastWarning = now; 4502 } 4503 } 4504 // Release the processor for a while before asking for a new buffer. 4505 // This will give the application more chance to read from the buffer and 4506 // clear the overflow. 4507 usleep(kRecordThreadSleepUs); 4508 } 4509 } 4510 // enable changes in effect chain 4511 unlockEffectChains(effectChains); 4512 effectChains.clear(); 4513 } 4514 4515 standby(); 4516 4517 { 4518 Mutex::Autolock _l(mLock); 4519 for (size_t i = 0; i < mTracks.size(); i++) { 4520 sp<RecordTrack> track = mTracks[i]; 4521 track->invalidate(); 4522 } 4523 mActiveTrack.clear(); 4524 mStartStopCond.broadcast(); 4525 } 4526 4527 releaseWakeLock(); 4528 4529 ALOGV("RecordThread %p exiting", this); 4530 return false; 4531} 4532 4533void AudioFlinger::RecordThread::standby() 4534{ 4535 if (!mStandby) { 4536 inputStandBy(); 4537 mStandby = true; 4538 } 4539} 4540 4541void AudioFlinger::RecordThread::inputStandBy() 4542{ 4543 mInput->stream->common.standby(&mInput->stream->common); 4544} 4545 4546sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4547 const sp<AudioFlinger::Client>& client, 4548 uint32_t sampleRate, 4549 audio_format_t format, 4550 audio_channel_mask_t channelMask, 4551 size_t frameCount, 4552 int sessionId, 4553 IAudioFlinger::track_flags_t *flags, 4554 pid_t tid, 4555 status_t *status) 4556{ 4557 sp<RecordTrack> track; 4558 status_t lStatus; 4559 4560 lStatus = initCheck(); 4561 if (lStatus != NO_ERROR) { 4562 ALOGE("Audio driver not initialized."); 4563 goto Exit; 4564 } 4565 // client expresses a preference for FAST, but we get the final say 4566 if (*flags & IAudioFlinger::TRACK_FAST) { 4567 if ( 4568 // use case: callback handler and frame count is default or at least as large as HAL 4569 ( 4570 (tid != -1) && 4571 ((frameCount == 0) || 4572 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4573 ) && 4574 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4575 // mono or stereo 4576 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4577 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4578 // hardware sample rate 4579 (sampleRate == mSampleRate) && 4580 // record thread has an associated fast recorder 4581 hasFastRecorder() 4582 // FIXME test that RecordThread for this fast track has a capable output HAL 4583 // FIXME add a permission test also? 4584 ) { 4585 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4586 if (frameCount == 0) { 4587 frameCount = mFrameCount * kFastTrackMultiplier; 4588 } 4589 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4590 frameCount, mFrameCount); 4591 } else { 4592 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4593 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4594 "hasFastRecorder=%d tid=%d", 4595 frameCount, mFrameCount, format, 4596 audio_is_linear_pcm(format), 4597 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4598 *flags &= ~IAudioFlinger::TRACK_FAST; 4599 // For compatibility with AudioRecord calculation, buffer depth is forced 4600 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4601 // This is probably too conservative, but legacy application code may depend on it. 4602 // If you change this calculation, also review the start threshold which is related. 4603 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4604 size_t mNormalFrameCount = 2048; // FIXME 4605 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4606 if (minBufCount < 2) { 4607 minBufCount = 2; 4608 } 4609 size_t minFrameCount = mNormalFrameCount * minBufCount; 4610 if (frameCount < minFrameCount) { 4611 frameCount = minFrameCount; 4612 } 4613 } 4614 } 4615 4616 // FIXME use flags and tid similar to createTrack_l() 4617 4618 { // scope for mLock 4619 Mutex::Autolock _l(mLock); 4620 4621 track = new RecordTrack(this, client, sampleRate, 4622 format, channelMask, frameCount, sessionId); 4623 4624 if (track->getCblk() == 0) { 4625 lStatus = NO_MEMORY; 4626 goto Exit; 4627 } 4628 mTracks.add(track); 4629 4630 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4631 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4632 mAudioFlinger->btNrecIsOff(); 4633 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4634 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4635 4636 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4637 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4638 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4639 // so ask activity manager to do this on our behalf 4640 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4641 } 4642 } 4643 lStatus = NO_ERROR; 4644 4645Exit: 4646 if (status) { 4647 *status = lStatus; 4648 } 4649 return track; 4650} 4651 4652status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4653 AudioSystem::sync_event_t event, 4654 int triggerSession) 4655{ 4656 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4657 sp<ThreadBase> strongMe = this; 4658 status_t status = NO_ERROR; 4659 4660 if (event == AudioSystem::SYNC_EVENT_NONE) { 4661 clearSyncStartEvent(); 4662 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4663 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4664 triggerSession, 4665 recordTrack->sessionId(), 4666 syncStartEventCallback, 4667 this); 4668 // Sync event can be cancelled by the trigger session if the track is not in a 4669 // compatible state in which case we start record immediately 4670 if (mSyncStartEvent->isCancelled()) { 4671 clearSyncStartEvent(); 4672 } else { 4673 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4674 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4675 } 4676 } 4677 4678 { 4679 AutoMutex lock(mLock); 4680 if (mActiveTrack != 0) { 4681 if (recordTrack != mActiveTrack.get()) { 4682 status = -EBUSY; 4683 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4684 mActiveTrack->mState = TrackBase::ACTIVE; 4685 } 4686 return status; 4687 } 4688 4689 recordTrack->mState = TrackBase::IDLE; 4690 mActiveTrack = recordTrack; 4691 mLock.unlock(); 4692 status_t status = AudioSystem::startInput(mId); 4693 mLock.lock(); 4694 if (status != NO_ERROR) { 4695 mActiveTrack.clear(); 4696 clearSyncStartEvent(); 4697 return status; 4698 } 4699 mRsmpInIndex = mFrameCount; 4700 mBytesRead = 0; 4701 if (mResampler != NULL) { 4702 mResampler->reset(); 4703 } 4704 mActiveTrack->mState = TrackBase::RESUMING; 4705 // signal thread to start 4706 ALOGV("Signal record thread"); 4707 mWaitWorkCV.broadcast(); 4708 // do not wait for mStartStopCond if exiting 4709 if (exitPending()) { 4710 mActiveTrack.clear(); 4711 status = INVALID_OPERATION; 4712 goto startError; 4713 } 4714 mStartStopCond.wait(mLock); 4715 if (mActiveTrack == 0) { 4716 ALOGV("Record failed to start"); 4717 status = BAD_VALUE; 4718 goto startError; 4719 } 4720 ALOGV("Record started OK"); 4721 return status; 4722 } 4723 4724startError: 4725 AudioSystem::stopInput(mId); 4726 clearSyncStartEvent(); 4727 return status; 4728} 4729 4730void AudioFlinger::RecordThread::clearSyncStartEvent() 4731{ 4732 if (mSyncStartEvent != 0) { 4733 mSyncStartEvent->cancel(); 4734 } 4735 mSyncStartEvent.clear(); 4736 mFramestoDrop = 0; 4737} 4738 4739void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4740{ 4741 sp<SyncEvent> strongEvent = event.promote(); 4742 4743 if (strongEvent != 0) { 4744 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4745 me->handleSyncStartEvent(strongEvent); 4746 } 4747} 4748 4749void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4750{ 4751 if (event == mSyncStartEvent) { 4752 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4753 // from audio HAL 4754 mFramestoDrop = mFrameCount * 2; 4755 } 4756} 4757 4758bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4759 ALOGV("RecordThread::stop"); 4760 AutoMutex _l(mLock); 4761 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4762 return false; 4763 } 4764 recordTrack->mState = TrackBase::PAUSING; 4765 // do not wait for mStartStopCond if exiting 4766 if (exitPending()) { 4767 return true; 4768 } 4769 mStartStopCond.wait(mLock); 4770 // if we have been restarted, recordTrack == mActiveTrack.get() here 4771 if (exitPending() || recordTrack != mActiveTrack.get()) { 4772 ALOGV("Record stopped OK"); 4773 return true; 4774 } 4775 return false; 4776} 4777 4778bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4779{ 4780 return false; 4781} 4782 4783status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4784{ 4785#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4786 if (!isValidSyncEvent(event)) { 4787 return BAD_VALUE; 4788 } 4789 4790 int eventSession = event->triggerSession(); 4791 status_t ret = NAME_NOT_FOUND; 4792 4793 Mutex::Autolock _l(mLock); 4794 4795 for (size_t i = 0; i < mTracks.size(); i++) { 4796 sp<RecordTrack> track = mTracks[i]; 4797 if (eventSession == track->sessionId()) { 4798 (void) track->setSyncEvent(event); 4799 ret = NO_ERROR; 4800 } 4801 } 4802 return ret; 4803#else 4804 return BAD_VALUE; 4805#endif 4806} 4807 4808// destroyTrack_l() must be called with ThreadBase::mLock held 4809void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4810{ 4811 track->terminate(); 4812 track->mState = TrackBase::STOPPED; 4813 // active tracks are removed by threadLoop() 4814 if (mActiveTrack != track) { 4815 removeTrack_l(track); 4816 } 4817} 4818 4819void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4820{ 4821 mTracks.remove(track); 4822 // need anything related to effects here? 4823} 4824 4825void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4826{ 4827 dumpInternals(fd, args); 4828 dumpTracks(fd, args); 4829 dumpEffectChains(fd, args); 4830} 4831 4832void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4833{ 4834 const size_t SIZE = 256; 4835 char buffer[SIZE]; 4836 String8 result; 4837 4838 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4839 result.append(buffer); 4840 4841 if (mActiveTrack != 0) { 4842 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4843 result.append(buffer); 4844 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4845 result.append(buffer); 4846 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4847 result.append(buffer); 4848 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4849 result.append(buffer); 4850 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4851 result.append(buffer); 4852 } else { 4853 result.append("No active record client\n"); 4854 } 4855 4856 write(fd, result.string(), result.size()); 4857 4858 dumpBase(fd, args); 4859} 4860 4861void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4862{ 4863 const size_t SIZE = 256; 4864 char buffer[SIZE]; 4865 String8 result; 4866 4867 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4868 result.append(buffer); 4869 RecordTrack::appendDumpHeader(result); 4870 for (size_t i = 0; i < mTracks.size(); ++i) { 4871 sp<RecordTrack> track = mTracks[i]; 4872 if (track != 0) { 4873 track->dump(buffer, SIZE); 4874 result.append(buffer); 4875 } 4876 } 4877 4878 if (mActiveTrack != 0) { 4879 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4880 result.append(buffer); 4881 RecordTrack::appendDumpHeader(result); 4882 mActiveTrack->dump(buffer, SIZE); 4883 result.append(buffer); 4884 4885 } 4886 write(fd, result.string(), result.size()); 4887} 4888 4889// AudioBufferProvider interface 4890status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4891{ 4892 size_t framesReq = buffer->frameCount; 4893 size_t framesReady = mFrameCount - mRsmpInIndex; 4894 int channelCount; 4895 4896 if (framesReady == 0) { 4897 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4898 if (mBytesRead <= 0) { 4899 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4900 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4901 // Force input into standby so that it tries to 4902 // recover at next read attempt 4903 inputStandBy(); 4904 usleep(kRecordThreadSleepUs); 4905 } 4906 buffer->raw = NULL; 4907 buffer->frameCount = 0; 4908 return NOT_ENOUGH_DATA; 4909 } 4910 mRsmpInIndex = 0; 4911 framesReady = mFrameCount; 4912 } 4913 4914 if (framesReq > framesReady) { 4915 framesReq = framesReady; 4916 } 4917 4918 if (mChannelCount == 1 && mReqChannelCount == 2) { 4919 channelCount = 1; 4920 } else { 4921 channelCount = 2; 4922 } 4923 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4924 buffer->frameCount = framesReq; 4925 return NO_ERROR; 4926} 4927 4928// AudioBufferProvider interface 4929void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4930{ 4931 mRsmpInIndex += buffer->frameCount; 4932 buffer->frameCount = 0; 4933} 4934 4935bool AudioFlinger::RecordThread::checkForNewParameters_l() 4936{ 4937 bool reconfig = false; 4938 4939 while (!mNewParameters.isEmpty()) { 4940 status_t status = NO_ERROR; 4941 String8 keyValuePair = mNewParameters[0]; 4942 AudioParameter param = AudioParameter(keyValuePair); 4943 int value; 4944 audio_format_t reqFormat = mFormat; 4945 uint32_t reqSamplingRate = mReqSampleRate; 4946 uint32_t reqChannelCount = mReqChannelCount; 4947 4948 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4949 reqSamplingRate = value; 4950 reconfig = true; 4951 } 4952 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4953 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4954 status = BAD_VALUE; 4955 } else { 4956 reqFormat = (audio_format_t) value; 4957 reconfig = true; 4958 } 4959 } 4960 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4961 reqChannelCount = popcount(value); 4962 reconfig = true; 4963 } 4964 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4965 // do not accept frame count changes if tracks are open as the track buffer 4966 // size depends on frame count and correct behavior would not be guaranteed 4967 // if frame count is changed after track creation 4968 if (mActiveTrack != 0) { 4969 status = INVALID_OPERATION; 4970 } else { 4971 reconfig = true; 4972 } 4973 } 4974 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4975 // forward device change to effects that have requested to be 4976 // aware of attached audio device. 4977 for (size_t i = 0; i < mEffectChains.size(); i++) { 4978 mEffectChains[i]->setDevice_l(value); 4979 } 4980 4981 // store input device and output device but do not forward output device to audio HAL. 4982 // Note that status is ignored by the caller for output device 4983 // (see AudioFlinger::setParameters() 4984 if (audio_is_output_devices(value)) { 4985 mOutDevice = value; 4986 status = BAD_VALUE; 4987 } else { 4988 mInDevice = value; 4989 // disable AEC and NS if the device is a BT SCO headset supporting those 4990 // pre processings 4991 if (mTracks.size() > 0) { 4992 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4993 mAudioFlinger->btNrecIsOff(); 4994 for (size_t i = 0; i < mTracks.size(); i++) { 4995 sp<RecordTrack> track = mTracks[i]; 4996 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4997 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4998 } 4999 } 5000 } 5001 } 5002 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5003 mAudioSource != (audio_source_t)value) { 5004 // forward device change to effects that have requested to be 5005 // aware of attached audio device. 5006 for (size_t i = 0; i < mEffectChains.size(); i++) { 5007 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5008 } 5009 mAudioSource = (audio_source_t)value; 5010 } 5011 if (status == NO_ERROR) { 5012 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5013 keyValuePair.string()); 5014 if (status == INVALID_OPERATION) { 5015 inputStandBy(); 5016 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5017 keyValuePair.string()); 5018 } 5019 if (reconfig) { 5020 if (status == BAD_VALUE && 5021 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5022 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5023 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5024 <= (2 * reqSamplingRate)) && 5025 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5026 <= FCC_2 && 5027 (reqChannelCount <= FCC_2)) { 5028 status = NO_ERROR; 5029 } 5030 if (status == NO_ERROR) { 5031 readInputParameters(); 5032 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5033 } 5034 } 5035 } 5036 5037 mNewParameters.removeAt(0); 5038 5039 mParamStatus = status; 5040 mParamCond.signal(); 5041 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5042 // already timed out waiting for the status and will never signal the condition. 5043 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5044 } 5045 return reconfig; 5046} 5047 5048String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5049{ 5050 Mutex::Autolock _l(mLock); 5051 if (initCheck() != NO_ERROR) { 5052 return String8(); 5053 } 5054 5055 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5056 const String8 out_s8(s); 5057 free(s); 5058 return out_s8; 5059} 5060 5061void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5062 AudioSystem::OutputDescriptor desc; 5063 void *param2 = NULL; 5064 5065 switch (event) { 5066 case AudioSystem::INPUT_OPENED: 5067 case AudioSystem::INPUT_CONFIG_CHANGED: 5068 desc.channelMask = mChannelMask; 5069 desc.samplingRate = mSampleRate; 5070 desc.format = mFormat; 5071 desc.frameCount = mFrameCount; 5072 desc.latency = 0; 5073 param2 = &desc; 5074 break; 5075 5076 case AudioSystem::INPUT_CLOSED: 5077 default: 5078 break; 5079 } 5080 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5081} 5082 5083void AudioFlinger::RecordThread::readInputParameters() 5084{ 5085 delete[] mRsmpInBuffer; 5086 // mRsmpInBuffer is always assigned a new[] below 5087 delete[] mRsmpOutBuffer; 5088 mRsmpOutBuffer = NULL; 5089 delete mResampler; 5090 mResampler = NULL; 5091 5092 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5093 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5094 mChannelCount = popcount(mChannelMask); 5095 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5096 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5097 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5098 } 5099 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5100 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5101 mFrameCount = mBufferSize / mFrameSize; 5102 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5103 5104 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5105 { 5106 int channelCount; 5107 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5108 // stereo to mono post process as the resampler always outputs stereo. 5109 if (mChannelCount == 1 && mReqChannelCount == 2) { 5110 channelCount = 1; 5111 } else { 5112 channelCount = 2; 5113 } 5114 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5115 mResampler->setSampleRate(mSampleRate); 5116 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5117 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5118 5119 // optmization: if mono to mono, alter input frame count as if we were inputing 5120 // stereo samples 5121 if (mChannelCount == 1 && mReqChannelCount == 1) { 5122 mFrameCount >>= 1; 5123 } 5124 5125 } 5126 mRsmpInIndex = mFrameCount; 5127} 5128 5129unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5130{ 5131 Mutex::Autolock _l(mLock); 5132 if (initCheck() != NO_ERROR) { 5133 return 0; 5134 } 5135 5136 return mInput->stream->get_input_frames_lost(mInput->stream); 5137} 5138 5139uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5140{ 5141 Mutex::Autolock _l(mLock); 5142 uint32_t result = 0; 5143 if (getEffectChain_l(sessionId) != 0) { 5144 result = EFFECT_SESSION; 5145 } 5146 5147 for (size_t i = 0; i < mTracks.size(); ++i) { 5148 if (sessionId == mTracks[i]->sessionId()) { 5149 result |= TRACK_SESSION; 5150 break; 5151 } 5152 } 5153 5154 return result; 5155} 5156 5157KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5158{ 5159 KeyedVector<int, bool> ids; 5160 Mutex::Autolock _l(mLock); 5161 for (size_t j = 0; j < mTracks.size(); ++j) { 5162 sp<RecordThread::RecordTrack> track = mTracks[j]; 5163 int sessionId = track->sessionId(); 5164 if (ids.indexOfKey(sessionId) < 0) { 5165 ids.add(sessionId, true); 5166 } 5167 } 5168 return ids; 5169} 5170 5171AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5172{ 5173 Mutex::Autolock _l(mLock); 5174 AudioStreamIn *input = mInput; 5175 mInput = NULL; 5176 return input; 5177} 5178 5179// this method must always be called either with ThreadBase mLock held or inside the thread loop 5180audio_stream_t* AudioFlinger::RecordThread::stream() const 5181{ 5182 if (mInput == NULL) { 5183 return NULL; 5184 } 5185 return &mInput->stream->common; 5186} 5187 5188status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5189{ 5190 // only one chain per input thread 5191 if (mEffectChains.size() != 0) { 5192 return INVALID_OPERATION; 5193 } 5194 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5195 5196 chain->setInBuffer(NULL); 5197 chain->setOutBuffer(NULL); 5198 5199 checkSuspendOnAddEffectChain_l(chain); 5200 5201 mEffectChains.add(chain); 5202 5203 return NO_ERROR; 5204} 5205 5206size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5207{ 5208 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5209 ALOGW_IF(mEffectChains.size() != 1, 5210 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5211 chain.get(), mEffectChains.size(), this); 5212 if (mEffectChains.size() == 1) { 5213 mEffectChains.removeAt(0); 5214 } 5215 return 0; 5216} 5217 5218}; // namespace android 5219