Threads.cpp revision e4756fe3a387615acb63c6a05788c8db9b5786cb
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 <math.h> 24#include <fcntl.h> 25#include <sys/stat.h> 26#include <cutils/properties.h> 27#include <cutils/compiler.h> 28#include <utils/Log.h> 29#include <utils/Trace.h> 30 31#include <private/media/AudioTrackShared.h> 32#include <hardware/audio.h> 33#include <audio_effects/effect_ns.h> 34#include <audio_effects/effect_aec.h> 35#include <audio_utils/primitives.h> 36 37// NBAIO implementations 38#include <media/nbaio/AudioStreamOutSink.h> 39#include <media/nbaio/MonoPipe.h> 40#include <media/nbaio/MonoPipeReader.h> 41#include <media/nbaio/Pipe.h> 42#include <media/nbaio/PipeReader.h> 43#include <media/nbaio/SourceAudioBufferProvider.h> 44 45#include <powermanager/PowerManager.h> 46 47#include <common_time/cc_helper.h> 48#include <common_time/local_clock.h> 49 50#include "AudioFlinger.h" 51#include "AudioMixer.h" 52#include "FastMixer.h" 53#include "ServiceUtilities.h" 54#include "SchedulingPolicyService.h" 55 56#undef ADD_BATTERY_DATA 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait for setParameters to complete 102static const nsecs_t kSetParametersTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal mix buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalMixBufferSizeMs = 20; 111// maximum normal mix buffer size 112static const uint32_t kMaxNormalMixBufferSizeMs = 24; 113 114// Whether to use fast mixer 115static const enum { 116 FastMixer_Never, // never initialize or use: for debugging only 117 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 118 // normal mixer multiplier is 1 119 FastMixer_Static, // initialize if needed, then use all the time if initialized, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 // FIXME for FastMixer_Dynamic: 124 // Supporting this option will require fixing HALs that can't handle large writes. 125 // For example, one HAL implementation returns an error from a large write, 126 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 127 // We could either fix the HAL implementations, or provide a wrapper that breaks 128 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 129} kUseFastMixer = FastMixer_Static; 130 131// Priorities for requestPriority 132static const int kPriorityAudioApp = 2; 133static const int kPriorityFastMixer = 3; 134 135// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 136// for the track. The client then sub-divides this into smaller buffers for its use. 137// Currently the client uses double-buffering by default, but doesn't tell us about that. 138// So for now we just assume that client is double-buffered. 139// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 140// N-buffering, so AudioFlinger could allocate the right amount of memory. 141// See the client's minBufCount and mNotificationFramesAct calculations for details. 142static const int kFastTrackMultiplier = 2; 143 144// ---------------------------------------------------------------------------- 145 146#ifdef ADD_BATTERY_DATA 147// To collect the amplifier usage 148static void addBatteryData(uint32_t params) { 149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150 if (service == NULL) { 151 // it already logged 152 return; 153 } 154 155 service->addBatteryData(params); 156} 157#endif 158 159 160// ---------------------------------------------------------------------------- 161// CPU Stats 162// ---------------------------------------------------------------------------- 163 164class CpuStats { 165public: 166 CpuStats(); 167 void sample(const String8 &title); 168#ifdef DEBUG_CPU_USAGE 169private: 170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172 173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174 175 int mCpuNum; // thread's current CPU number 176 int mCpukHz; // frequency of thread's current CPU in kHz 177#endif 178}; 179 180CpuStats::CpuStats() 181#ifdef DEBUG_CPU_USAGE 182 : mCpuNum(-1), mCpukHz(-1) 183#endif 184{ 185} 186 187void CpuStats::sample(const String8 &title) { 188#ifdef DEBUG_CPU_USAGE 189 // get current thread's delta CPU time in wall clock ns 190 double wcNs; 191 bool valid = mCpuUsage.sampleAndEnable(wcNs); 192 193 // record sample for wall clock statistics 194 if (valid) { 195 mWcStats.sample(wcNs); 196 } 197 198 // get the current CPU number 199 int cpuNum = sched_getcpu(); 200 201 // get the current CPU frequency in kHz 202 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203 204 // check if either CPU number or frequency changed 205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 206 mCpuNum = cpuNum; 207 mCpukHz = cpukHz; 208 // ignore sample for purposes of cycles 209 valid = false; 210 } 211 212 // if no change in CPU number or frequency, then record sample for cycle statistics 213 if (valid && mCpukHz > 0) { 214 double cycles = wcNs * cpukHz * 0.000001; 215 mHzStats.sample(cycles); 216 } 217 218 unsigned n = mWcStats.n(); 219 // mCpuUsage.elapsed() is expensive, so don't call it every loop 220 if ((n & 127) == 1) { 221 long long elapsed = mCpuUsage.elapsed(); 222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 223 double perLoop = elapsed / (double) n; 224 double perLoop100 = perLoop * 0.01; 225 double perLoop1k = perLoop * 0.001; 226 double mean = mWcStats.mean(); 227 double stddev = mWcStats.stddev(); 228 double minimum = mWcStats.minimum(); 229 double maximum = mWcStats.maximum(); 230 double meanCycles = mHzStats.mean(); 231 double stddevCycles = mHzStats.stddev(); 232 double minCycles = mHzStats.minimum(); 233 double maxCycles = mHzStats.maximum(); 234 mCpuUsage.resetElapsed(); 235 mWcStats.reset(); 236 mHzStats.reset(); 237 ALOGD("CPU usage for %s over past %.1f secs\n" 238 " (%u mixer loops at %.1f mean ms per loop):\n" 239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 242 title.string(), 243 elapsed * .000000001, n, perLoop * .000001, 244 mean * .001, 245 stddev * .001, 246 minimum * .001, 247 maximum * .001, 248 mean / perLoop100, 249 stddev / perLoop100, 250 minimum / perLoop100, 251 maximum / perLoop100, 252 meanCycles / perLoop1k, 253 stddevCycles / perLoop1k, 254 minCycles / perLoop1k, 255 maxCycles / perLoop1k); 256 257 } 258 } 259#endif 260}; 261 262// ---------------------------------------------------------------------------- 263// ThreadBase 264// ---------------------------------------------------------------------------- 265 266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 268 : Thread(false /*canCallJava*/), 269 mType(type), 270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 271 // mChannelMask 272 mChannelCount(0), 273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 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 mParamCond.broadcast(); 285 // do not lock the mutex in destructor 286 releaseWakeLock_l(); 287 if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 377 if (err != 0) { 378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 379 "error %d", 380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 381 } 382 } break; 383 case CFG_EVENT_IO: { 384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 385 mAudioFlinger->mLock.lock(); 386 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 387 mAudioFlinger->mLock.unlock(); 388 } break; 389 default: 390 ALOGE("processConfigEvents() unknown event type %d", event->type()); 391 break; 392 } 393 delete event; 394 mLock.lock(); 395 } 396 mLock.unlock(); 397} 398 399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 400{ 401 const size_t SIZE = 256; 402 char buffer[SIZE]; 403 String8 result; 404 405 bool locked = AudioFlinger::dumpTryLock(mLock); 406 if (!locked) { 407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 408 write(fd, buffer, strlen(buffer)); 409 } 410 411 snprintf(buffer, SIZE, "io handle: %d\n", mId); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 430 result.append(buffer); 431 432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 433 result.append(buffer); 434 result.append(" Index Command"); 435 for (size_t i = 0; i < mNewParameters.size(); ++i) { 436 snprintf(buffer, SIZE, "\n %02d ", i); 437 result.append(buffer); 438 result.append(mNewParameters[i]); 439 } 440 441 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 442 result.append(buffer); 443 for (size_t i = 0; i < mConfigEvents.size(); i++) { 444 mConfigEvents[i]->dump(buffer, SIZE); 445 result.append(buffer); 446 } 447 result.append("\n"); 448 449 write(fd, result.string(), result.size()); 450 451 if (locked) { 452 mLock.unlock(); 453 } 454} 455 456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 457{ 458 const size_t SIZE = 256; 459 char buffer[SIZE]; 460 String8 result; 461 462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 463 write(fd, buffer, strlen(buffer)); 464 465 for (size_t i = 0; i < mEffectChains.size(); ++i) { 466 sp<EffectChain> chain = mEffectChains[i]; 467 if (chain != 0) { 468 chain->dump(fd, args); 469 } 470 } 471} 472 473void AudioFlinger::ThreadBase::acquireWakeLock() 474{ 475 Mutex::Autolock _l(mLock); 476 acquireWakeLock_l(); 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock_l() 480{ 481 if (mPowerManager == 0) { 482 // use checkService() to avoid blocking if power service is not up yet 483 sp<IBinder> binder = 484 defaultServiceManager()->checkService(String16("power")); 485 if (binder == 0) { 486 ALOGW("Thread %s cannot connect to the power manager service", mName); 487 } else { 488 mPowerManager = interface_cast<IPowerManager>(binder); 489 binder->linkToDeath(mDeathRecipient); 490 } 491 } 492 if (mPowerManager != 0) { 493 sp<IBinder> binder = new BBinder(); 494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 495 binder, 496 String16(mName)); 497 if (status == NO_ERROR) { 498 mWakeLockToken = binder; 499 } 500 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 501 } 502} 503 504void AudioFlinger::ThreadBase::releaseWakeLock() 505{ 506 Mutex::Autolock _l(mLock); 507 releaseWakeLock_l(); 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock_l() 511{ 512 if (mWakeLockToken != 0) { 513 ALOGV("releaseWakeLock_l() %s", mName); 514 if (mPowerManager != 0) { 515 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 516 } 517 mWakeLockToken.clear(); 518 } 519} 520 521void AudioFlinger::ThreadBase::clearPowerManager() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525 mPowerManager.clear(); 526} 527 528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 529{ 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 thread->clearPowerManager(); 533 } 534 ALOGW("power manager service died !!!"); 535} 536 537void AudioFlinger::ThreadBase::setEffectSuspended( 538 const effect_uuid_t *type, bool suspend, int sessionId) 539{ 540 Mutex::Autolock _l(mLock); 541 setEffectSuspended_l(type, suspend, sessionId); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended_l( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 sp<EffectChain> chain = getEffectChain_l(sessionId); 548 if (chain != 0) { 549 if (type != NULL) { 550 chain->setEffectSuspended_l(type, suspend); 551 } else { 552 chain->setEffectSuspendedAll_l(suspend); 553 } 554 } 555 556 updateSuspendedSessions_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 560{ 561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 562 if (index < 0) { 563 return; 564 } 565 566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 567 mSuspendedSessions.valueAt(index); 568 569 for (size_t i = 0; i < sessionEffects.size(); i++) { 570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 571 for (int j = 0; j < desc->mRefCount; j++) { 572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 573 chain->setEffectSuspendedAll_l(true); 574 } else { 575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 576 desc->mType.timeLow); 577 chain->setEffectSuspended_l(&desc->mType, true); 578 } 579 } 580 } 581} 582 583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId) 586{ 587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 588 589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 590 591 if (suspend) { 592 if (index >= 0) { 593 sessionEffects = mSuspendedSessions.valueAt(index); 594 } else { 595 mSuspendedSessions.add(sessionId, sessionEffects); 596 } 597 } else { 598 if (index < 0) { 599 return; 600 } 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } 603 604 605 int key = EffectChain::kKeyForSuspendAll; 606 if (type != NULL) { 607 key = type->timeLow; 608 } 609 index = sessionEffects.indexOfKey(key); 610 611 sp<SuspendedSessionDesc> desc; 612 if (suspend) { 613 if (index >= 0) { 614 desc = sessionEffects.valueAt(index); 615 } else { 616 desc = new SuspendedSessionDesc(); 617 if (type != NULL) { 618 desc->mType = *type; 619 } 620 sessionEffects.add(key, desc); 621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 622 } 623 desc->mRefCount++; 624 } else { 625 if (index < 0) { 626 return; 627 } 628 desc = sessionEffects.valueAt(index); 629 if (--desc->mRefCount == 0) { 630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 631 sessionEffects.removeItemsAt(index); 632 if (sessionEffects.isEmpty()) { 633 ALOGV("updateSuspendedSessions_l() restore removing session %d", 634 sessionId); 635 mSuspendedSessions.removeItem(sessionId); 636 } 637 } 638 } 639 if (!sessionEffects.isEmpty()) { 640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 641 } 642} 643 644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 645 bool enabled, 646 int sessionId) 647{ 648 Mutex::Autolock _l(mLock); 649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 if (mType != RECORD) { 657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 658 // another session. This gives the priority to well behaved effect control panels 659 // and applications not using global effects. 660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 661 // global effects 662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 664 } 665 } 666 667 sp<EffectChain> chain = getEffectChain_l(sessionId); 668 if (chain != 0) { 669 chain->checkSuspendOnEffectEnabled(effect, enabled); 670 } 671} 672 673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 675 const sp<AudioFlinger::Client>& client, 676 const sp<IEffectClient>& effectClient, 677 int32_t priority, 678 int sessionId, 679 effect_descriptor_t *desc, 680 int *enabled, 681 status_t *status 682 ) 683{ 684 sp<EffectModule> effect; 685 sp<EffectHandle> handle; 686 status_t lStatus; 687 sp<EffectChain> chain; 688 bool chainCreated = false; 689 bool effectCreated = false; 690 bool effectRegistered = false; 691 692 lStatus = initCheck(); 693 if (lStatus != NO_ERROR) { 694 ALOGW("createEffect_l() Audio driver not initialized."); 695 goto Exit; 696 } 697 698 // Do not allow effects with session ID 0 on direct output or duplicating threads 699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 702 desc->name, sessionId); 703 lStatus = BAD_VALUE; 704 goto Exit; 705 } 706 // Only Pre processor effects are allowed on input threads and only on input threads 707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 709 desc->name, desc->flags, mType); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 715 716 { // scope for mLock 717 Mutex::Autolock _l(mLock); 718 719 // check for existing effect chain with the requested audio session 720 chain = getEffectChain_l(sessionId); 721 if (chain == 0) { 722 // create a new chain for this session 723 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 724 chain = new EffectChain(this, sessionId); 725 addEffectChain_l(chain); 726 chain->setStrategy(getStrategyForSession_l(sessionId)); 727 chainCreated = true; 728 } else { 729 effect = chain->getEffectFromDesc_l(desc); 730 } 731 732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 733 734 if (effect == 0) { 735 int id = mAudioFlinger->nextUniqueId(); 736 // Check CPU and memory usage 737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 738 if (lStatus != NO_ERROR) { 739 goto Exit; 740 } 741 effectRegistered = true; 742 // create a new effect module if none present in the chain 743 effect = new EffectModule(this, chain, desc, id, sessionId); 744 lStatus = effect->status(); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 lStatus = chain->addEffect_l(effect); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 effectCreated = true; 753 754 effect->setDevice(mOutDevice); 755 effect->setDevice(mInDevice); 756 effect->setMode(mAudioFlinger->getMode()); 757 effect->setAudioSource(mAudioSource); 758 } 759 // create effect handle and connect it to effect module 760 handle = new EffectHandle(effect, client, effectClient, priority); 761 lStatus = effect->addHandle(handle.get()); 762 if (enabled != NULL) { 763 *enabled = (int)effect->isEnabled(); 764 } 765 } 766 767Exit: 768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 769 Mutex::Autolock _l(mLock); 770 if (effectCreated) { 771 chain->removeEffect_l(effect); 772 } 773 if (effectRegistered) { 774 AudioSystem::unregisterEffect(effect->id()); 775 } 776 if (chainCreated) { 777 removeEffectChain_l(chain); 778 } 779 handle.clear(); 780 } 781 782 if (status != NULL) { 783 *status = lStatus; 784 } 785 return handle; 786} 787 788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 789{ 790 Mutex::Autolock _l(mLock); 791 return getEffect_l(sessionId, effectId); 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 795{ 796 sp<EffectChain> chain = getEffectChain_l(sessionId); 797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 798} 799 800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 801// PlaybackThread::mLock held 802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 803{ 804 // check for existing effect chain with the requested audio session 805 int sessionId = effect->sessionId(); 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 bool chainCreated = false; 808 809 if (chain == 0) { 810 // create a new chain for this session 811 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 812 chain = new EffectChain(this, sessionId); 813 addEffectChain_l(chain); 814 chain->setStrategy(getStrategyForSession_l(sessionId)); 815 chainCreated = true; 816 } 817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 818 819 if (chain->getEffectFromId_l(effect->id()) != 0) { 820 ALOGW("addEffect_l() %p effect %s already present in chain %p", 821 this, effect->desc().name, chain.get()); 822 return BAD_VALUE; 823 } 824 825 status_t status = chain->addEffect_l(effect); 826 if (status != NO_ERROR) { 827 if (chainCreated) { 828 removeEffectChain_l(chain); 829 } 830 return status; 831 } 832 833 effect->setDevice(mOutDevice); 834 effect->setDevice(mInDevice); 835 effect->setMode(mAudioFlinger->getMode()); 836 effect->setAudioSource(mAudioSource); 837 return NO_ERROR; 838} 839 840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 841 842 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 843 effect_descriptor_t desc = effect->desc(); 844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 845 detachAuxEffect_l(effect->id()); 846 } 847 848 sp<EffectChain> chain = effect->chain().promote(); 849 if (chain != 0) { 850 // remove effect chain if removing last effect 851 if (chain->removeEffect_l(effect) == 0) { 852 removeEffectChain_l(chain); 853 } 854 } else { 855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 856 } 857} 858 859void AudioFlinger::ThreadBase::lockEffectChains_l( 860 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 861{ 862 effectChains = mEffectChains; 863 for (size_t i = 0; i < mEffectChains.size(); i++) { 864 mEffectChains[i]->lock(); 865 } 866} 867 868void AudioFlinger::ThreadBase::unlockEffectChains( 869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 870{ 871 for (size_t i = 0; i < effectChains.size(); i++) { 872 effectChains[i]->unlock(); 873 } 874} 875 876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 877{ 878 Mutex::Autolock _l(mLock); 879 return getEffectChain_l(sessionId); 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 883{ 884 size_t size = mEffectChains.size(); 885 for (size_t i = 0; i < size; i++) { 886 if (mEffectChains[i]->sessionId() == sessionId) { 887 return mEffectChains[i]; 888 } 889 } 890 return 0; 891} 892 893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 894{ 895 Mutex::Autolock _l(mLock); 896 size_t size = mEffectChains.size(); 897 for (size_t i = 0; i < size; i++) { 898 mEffectChains[i]->setMode_l(mode); 899 } 900} 901 902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 903 EffectHandle *handle, 904 bool unpinIfLast) { 905 906 Mutex::Autolock _l(mLock); 907 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 908 // delete the effect module if removing last handle on it 909 if (effect->removeHandle(handle) == 0) { 910 if (!effect->isPinned() || unpinIfLast) { 911 removeEffect_l(effect); 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 } 915} 916 917// ---------------------------------------------------------------------------- 918// Playback 919// ---------------------------------------------------------------------------- 920 921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 922 AudioStreamOut* output, 923 audio_io_handle_t id, 924 audio_devices_t device, 925 type_t type) 926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 928 // mStreamTypes[] initialized in constructor body 929 mOutput(output), 930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 931 mMixerStatus(MIXER_IDLE), 932 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 934 mScreenState(AudioFlinger::mScreenState), 935 // index 0 is reserved for normal mixer's submix 936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 937{ 938 snprintf(mName, kNameLength, "AudioOut_%X", id); 939 940 // Assumes constructor is called by AudioFlinger with it's mLock held, but 941 // it would be safer to explicitly pass initial masterVolume/masterMute as 942 // parameter. 943 // 944 // If the HAL we are using has support for master volume or master mute, 945 // then do not attenuate or mute during mixing (just leave the volume at 1.0 946 // and the mute set to false). 947 mMasterVolume = audioFlinger->masterVolume_l(); 948 mMasterMute = audioFlinger->masterMute_l(); 949 if (mOutput && mOutput->audioHwDev) { 950 if (mOutput->audioHwDev->canSetMasterVolume()) { 951 mMasterVolume = 1.0; 952 } 953 954 if (mOutput->audioHwDev->canSetMasterMute()) { 955 mMasterMute = false; 956 } 957 } 958 959 readOutputParameters(); 960 961 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 962 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 963 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 964 stream = (audio_stream_type_t) (stream + 1)) { 965 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 966 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 967 } 968 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 969 // because mAudioFlinger doesn't have one to copy from 970} 971 972AudioFlinger::PlaybackThread::~PlaybackThread() 973{ 974 delete [] mMixBuffer; 975} 976 977void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 978{ 979 dumpInternals(fd, args); 980 dumpTracks(fd, args); 981 dumpEffectChains(fd, args); 982} 983 984void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 985{ 986 const size_t SIZE = 256; 987 char buffer[SIZE]; 988 String8 result; 989 990 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 991 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 992 const stream_type_t *st = &mStreamTypes[i]; 993 if (i > 0) { 994 result.appendFormat(", "); 995 } 996 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 997 if (st->mute) { 998 result.append("M"); 999 } 1000 } 1001 result.append("\n"); 1002 write(fd, result.string(), result.length()); 1003 result.clear(); 1004 1005 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1006 result.append(buffer); 1007 Track::appendDumpHeader(result); 1008 for (size_t i = 0; i < mTracks.size(); ++i) { 1009 sp<Track> track = mTracks[i]; 1010 if (track != 0) { 1011 track->dump(buffer, SIZE); 1012 result.append(buffer); 1013 } 1014 } 1015 1016 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1017 result.append(buffer); 1018 Track::appendDumpHeader(result); 1019 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1020 sp<Track> track = mActiveTracks[i].promote(); 1021 if (track != 0) { 1022 track->dump(buffer, SIZE); 1023 result.append(buffer); 1024 } 1025 } 1026 write(fd, result.string(), result.size()); 1027 1028 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1029 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1030 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1031 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1032} 1033 1034void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1035{ 1036 const size_t SIZE = 256; 1037 char buffer[SIZE]; 1038 String8 result; 1039 1040 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1041 result.append(buffer); 1042 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1043 ns2ms(systemTime() - mLastWriteTime)); 1044 result.append(buffer); 1045 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1054 result.append(buffer); 1055 write(fd, result.string(), result.size()); 1056 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1057 1058 dumpBase(fd, args); 1059} 1060 1061// Thread virtuals 1062status_t AudioFlinger::PlaybackThread::readyToRun() 1063{ 1064 status_t status = initCheck(); 1065 if (status == NO_ERROR) { 1066 ALOGI("AudioFlinger's thread %p ready to run", this); 1067 } else { 1068 ALOGE("No working audio driver found."); 1069 } 1070 return status; 1071} 1072 1073void AudioFlinger::PlaybackThread::onFirstRef() 1074{ 1075 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1076} 1077 1078// ThreadBase virtuals 1079void AudioFlinger::PlaybackThread::preExit() 1080{ 1081 ALOGV(" preExit()"); 1082 // FIXME this is using hard-coded strings but in the future, this functionality will be 1083 // converted to use audio HAL extensions required to support tunneling 1084 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1085} 1086 1087// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1088sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1089 const sp<AudioFlinger::Client>& client, 1090 audio_stream_type_t streamType, 1091 uint32_t sampleRate, 1092 audio_format_t format, 1093 audio_channel_mask_t channelMask, 1094 size_t frameCount, 1095 const sp<IMemory>& sharedBuffer, 1096 int sessionId, 1097 IAudioFlinger::track_flags_t *flags, 1098 pid_t tid, 1099 status_t *status) 1100{ 1101 sp<Track> track; 1102 status_t lStatus; 1103 1104 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1105 1106 // client expresses a preference for FAST, but we get the final say 1107 if (*flags & IAudioFlinger::TRACK_FAST) { 1108 if ( 1109 // not timed 1110 (!isTimed) && 1111 // either of these use cases: 1112 ( 1113 // use case 1: shared buffer with any frame count 1114 ( 1115 (sharedBuffer != 0) 1116 ) || 1117 // use case 2: callback handler and frame count is default or at least as large as HAL 1118 ( 1119 (tid != -1) && 1120 ((frameCount == 0) || 1121 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1122 ) 1123 ) && 1124 // PCM data 1125 audio_is_linear_pcm(format) && 1126 // mono or stereo 1127 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1128 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1129#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1130 // hardware sample rate 1131 (sampleRate == mSampleRate) && 1132#endif 1133 // normal mixer has an associated fast mixer 1134 hasFastMixer() && 1135 // there are sufficient fast track slots available 1136 (mFastTrackAvailMask != 0) 1137 // FIXME test that MixerThread for this fast track has a capable output HAL 1138 // FIXME add a permission test also? 1139 ) { 1140 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1141 if (frameCount == 0) { 1142 frameCount = mFrameCount * kFastTrackMultiplier; 1143 } 1144 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1145 frameCount, mFrameCount); 1146 } else { 1147 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1148 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1149 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1150 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1151 audio_is_linear_pcm(format), 1152 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1153 *flags &= ~IAudioFlinger::TRACK_FAST; 1154 // For compatibility with AudioTrack calculation, buffer depth is forced 1155 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1156 // This is probably too conservative, but legacy application code may depend on it. 1157 // If you change this calculation, also review the start threshold which is related. 1158 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1159 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1160 if (minBufCount < 2) { 1161 minBufCount = 2; 1162 } 1163 size_t minFrameCount = mNormalFrameCount * minBufCount; 1164 if (frameCount < minFrameCount) { 1165 frameCount = minFrameCount; 1166 } 1167 } 1168 } 1169 1170 if (mType == DIRECT) { 1171 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1172 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1173 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1174 "for output %p with format %d", 1175 sampleRate, format, channelMask, mOutput, mFormat); 1176 lStatus = BAD_VALUE; 1177 goto Exit; 1178 } 1179 } 1180 } else { 1181 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1182 if (sampleRate > mSampleRate*2) { 1183 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1184 lStatus = BAD_VALUE; 1185 goto Exit; 1186 } 1187 } 1188 1189 lStatus = initCheck(); 1190 if (lStatus != NO_ERROR) { 1191 ALOGE("Audio driver not initialized."); 1192 goto Exit; 1193 } 1194 1195 { // scope for mLock 1196 Mutex::Autolock _l(mLock); 1197 1198 // all tracks in same audio session must share the same routing strategy otherwise 1199 // conflicts will happen when tracks are moved from one output to another by audio policy 1200 // manager 1201 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1202 for (size_t i = 0; i < mTracks.size(); ++i) { 1203 sp<Track> t = mTracks[i]; 1204 if (t != 0 && !t->isOutputTrack()) { 1205 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1206 if (sessionId == t->sessionId() && strategy != actual) { 1207 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1208 strategy, actual); 1209 lStatus = BAD_VALUE; 1210 goto Exit; 1211 } 1212 } 1213 } 1214 1215 if (!isTimed) { 1216 track = new Track(this, client, streamType, sampleRate, format, 1217 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1218 } else { 1219 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1220 channelMask, frameCount, sharedBuffer, sessionId); 1221 } 1222 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1223 lStatus = NO_MEMORY; 1224 goto Exit; 1225 } 1226 mTracks.add(track); 1227 1228 sp<EffectChain> chain = getEffectChain_l(sessionId); 1229 if (chain != 0) { 1230 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1231 track->setMainBuffer(chain->inBuffer()); 1232 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1233 chain->incTrackCnt(); 1234 } 1235 1236 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1237 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1238 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1239 // so ask activity manager to do this on our behalf 1240 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1241 } 1242 } 1243 1244 lStatus = NO_ERROR; 1245 1246Exit: 1247 if (status) { 1248 *status = lStatus; 1249 } 1250 return track; 1251} 1252 1253uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1254{ 1255 return latency; 1256} 1257 1258uint32_t AudioFlinger::PlaybackThread::latency() const 1259{ 1260 Mutex::Autolock _l(mLock); 1261 return latency_l(); 1262} 1263uint32_t AudioFlinger::PlaybackThread::latency_l() const 1264{ 1265 if (initCheck() == NO_ERROR) { 1266 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1267 } else { 1268 return 0; 1269 } 1270} 1271 1272void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1273{ 1274 Mutex::Autolock _l(mLock); 1275 // Don't apply master volume in SW if our HAL can do it for us. 1276 if (mOutput && mOutput->audioHwDev && 1277 mOutput->audioHwDev->canSetMasterVolume()) { 1278 mMasterVolume = 1.0; 1279 } else { 1280 mMasterVolume = value; 1281 } 1282} 1283 1284void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1285{ 1286 Mutex::Autolock _l(mLock); 1287 // Don't apply master mute in SW if our HAL can do it for us. 1288 if (mOutput && mOutput->audioHwDev && 1289 mOutput->audioHwDev->canSetMasterMute()) { 1290 mMasterMute = false; 1291 } else { 1292 mMasterMute = muted; 1293 } 1294} 1295 1296void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1297{ 1298 Mutex::Autolock _l(mLock); 1299 mStreamTypes[stream].volume = value; 1300} 1301 1302void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1303{ 1304 Mutex::Autolock _l(mLock); 1305 mStreamTypes[stream].mute = muted; 1306} 1307 1308float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1309{ 1310 Mutex::Autolock _l(mLock); 1311 return mStreamTypes[stream].volume; 1312} 1313 1314// addTrack_l() must be called with ThreadBase::mLock held 1315status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1316{ 1317 status_t status = ALREADY_EXISTS; 1318 1319 // set retry count for buffer fill 1320 track->mRetryCount = kMaxTrackStartupRetries; 1321 if (mActiveTracks.indexOf(track) < 0) { 1322 // the track is newly added, make sure it fills up all its 1323 // buffers before playing. This is to ensure the client will 1324 // effectively get the latency it requested. 1325 track->mFillingUpStatus = Track::FS_FILLING; 1326 track->mResetDone = false; 1327 track->mPresentationCompleteFrames = 0; 1328 mActiveTracks.add(track); 1329 if (track->mainBuffer() != mMixBuffer) { 1330 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1331 if (chain != 0) { 1332 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1333 track->sessionId()); 1334 chain->incActiveTrackCnt(); 1335 } 1336 } 1337 1338 status = NO_ERROR; 1339 } 1340 1341 ALOGV("mWaitWorkCV.broadcast"); 1342 mWaitWorkCV.broadcast(); 1343 1344 return status; 1345} 1346 1347// destroyTrack_l() must be called with ThreadBase::mLock held 1348void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1349{ 1350 track->mState = TrackBase::TERMINATED; 1351 // active tracks are removed by threadLoop() 1352 if (mActiveTracks.indexOf(track) < 0) { 1353 removeTrack_l(track); 1354 } 1355} 1356 1357void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1358{ 1359 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1360 mTracks.remove(track); 1361 deleteTrackName_l(track->name()); 1362 // redundant as track is about to be destroyed, for dumpsys only 1363 track->mName = -1; 1364 if (track->isFastTrack()) { 1365 int index = track->mFastIndex; 1366 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1367 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1368 mFastTrackAvailMask |= 1 << index; 1369 // redundant as track is about to be destroyed, for dumpsys only 1370 track->mFastIndex = -1; 1371 } 1372 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1373 if (chain != 0) { 1374 chain->decTrackCnt(); 1375 } 1376} 1377 1378String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1379{ 1380 String8 out_s8 = String8(""); 1381 char *s; 1382 1383 Mutex::Autolock _l(mLock); 1384 if (initCheck() != NO_ERROR) { 1385 return out_s8; 1386 } 1387 1388 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1389 out_s8 = String8(s); 1390 free(s); 1391 return out_s8; 1392} 1393 1394// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1395void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1396 AudioSystem::OutputDescriptor desc; 1397 void *param2 = NULL; 1398 1399 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1400 param); 1401 1402 switch (event) { 1403 case AudioSystem::OUTPUT_OPENED: 1404 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1405 desc.channels = mChannelMask; 1406 desc.samplingRate = mSampleRate; 1407 desc.format = mFormat; 1408 desc.frameCount = mNormalFrameCount; // FIXME see 1409 // AudioFlinger::frameCount(audio_io_handle_t) 1410 desc.latency = latency(); 1411 param2 = &desc; 1412 break; 1413 1414 case AudioSystem::STREAM_CONFIG_CHANGED: 1415 param2 = ¶m; 1416 case AudioSystem::OUTPUT_CLOSED: 1417 default: 1418 break; 1419 } 1420 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1421} 1422 1423void AudioFlinger::PlaybackThread::readOutputParameters() 1424{ 1425 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1426 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1427 mChannelCount = (uint16_t)popcount(mChannelMask); 1428 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1429 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1430 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1431 if (mFrameCount & 15) { 1432 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1433 mFrameCount); 1434 } 1435 1436 // Calculate size of normal mix buffer relative to the HAL output buffer size 1437 double multiplier = 1.0; 1438 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1439 kUseFastMixer == FastMixer_Dynamic)) { 1440 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1441 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1442 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1443 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1444 maxNormalFrameCount = maxNormalFrameCount & ~15; 1445 if (maxNormalFrameCount < minNormalFrameCount) { 1446 maxNormalFrameCount = minNormalFrameCount; 1447 } 1448 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1449 if (multiplier <= 1.0) { 1450 multiplier = 1.0; 1451 } else if (multiplier <= 2.0) { 1452 if (2 * mFrameCount <= maxNormalFrameCount) { 1453 multiplier = 2.0; 1454 } else { 1455 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1456 } 1457 } else { 1458 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1459 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1460 // track, but we sometimes have to do this to satisfy the maximum frame count 1461 // constraint) 1462 // FIXME this rounding up should not be done if no HAL SRC 1463 uint32_t truncMult = (uint32_t) multiplier; 1464 if ((truncMult & 1)) { 1465 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1466 ++truncMult; 1467 } 1468 } 1469 multiplier = (double) truncMult; 1470 } 1471 } 1472 mNormalFrameCount = multiplier * mFrameCount; 1473 // round up to nearest 16 frames to satisfy AudioMixer 1474 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1475 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1476 mNormalFrameCount); 1477 1478 delete[] mMixBuffer; 1479 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1480 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1481 1482 // force reconfiguration of effect chains and engines to take new buffer size and audio 1483 // parameters into account 1484 // Note that mLock is not held when readOutputParameters() is called from the constructor 1485 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1486 // matter. 1487 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1488 Vector< sp<EffectChain> > effectChains = mEffectChains; 1489 for (size_t i = 0; i < effectChains.size(); i ++) { 1490 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1491 } 1492} 1493 1494 1495status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1496{ 1497 if (halFrames == NULL || dspFrames == NULL) { 1498 return BAD_VALUE; 1499 } 1500 Mutex::Autolock _l(mLock); 1501 if (initCheck() != NO_ERROR) { 1502 return INVALID_OPERATION; 1503 } 1504 size_t framesWritten = mBytesWritten / mFrameSize; 1505 *halFrames = framesWritten; 1506 1507 if (isSuspended()) { 1508 // return an estimation of rendered frames when the output is suspended 1509 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1510 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1511 return NO_ERROR; 1512 } else { 1513 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1514 } 1515} 1516 1517uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1518{ 1519 Mutex::Autolock _l(mLock); 1520 uint32_t result = 0; 1521 if (getEffectChain_l(sessionId) != 0) { 1522 result = EFFECT_SESSION; 1523 } 1524 1525 for (size_t i = 0; i < mTracks.size(); ++i) { 1526 sp<Track> track = mTracks[i]; 1527 if (sessionId == track->sessionId() && 1528 !(track->mCblk->flags & CBLK_INVALID)) { 1529 result |= TRACK_SESSION; 1530 break; 1531 } 1532 } 1533 1534 return result; 1535} 1536 1537uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1538{ 1539 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1540 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1541 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1542 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1543 } 1544 for (size_t i = 0; i < mTracks.size(); i++) { 1545 sp<Track> track = mTracks[i]; 1546 if (sessionId == track->sessionId() && 1547 !(track->mCblk->flags & CBLK_INVALID)) { 1548 return AudioSystem::getStrategyForStream(track->streamType()); 1549 } 1550 } 1551 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1552} 1553 1554 1555AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1556{ 1557 Mutex::Autolock _l(mLock); 1558 return mOutput; 1559} 1560 1561AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1562{ 1563 Mutex::Autolock _l(mLock); 1564 AudioStreamOut *output = mOutput; 1565 mOutput = NULL; 1566 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1567 // must push a NULL and wait for ack 1568 mOutputSink.clear(); 1569 mPipeSink.clear(); 1570 mNormalSink.clear(); 1571 return output; 1572} 1573 1574// this method must always be called either with ThreadBase mLock held or inside the thread loop 1575audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1576{ 1577 if (mOutput == NULL) { 1578 return NULL; 1579 } 1580 return &mOutput->stream->common; 1581} 1582 1583uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1584{ 1585 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1586} 1587 1588status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1589{ 1590 if (!isValidSyncEvent(event)) { 1591 return BAD_VALUE; 1592 } 1593 1594 Mutex::Autolock _l(mLock); 1595 1596 for (size_t i = 0; i < mTracks.size(); ++i) { 1597 sp<Track> track = mTracks[i]; 1598 if (event->triggerSession() == track->sessionId()) { 1599 (void) track->setSyncEvent(event); 1600 return NO_ERROR; 1601 } 1602 } 1603 1604 return NAME_NOT_FOUND; 1605} 1606 1607bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1608{ 1609 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1610} 1611 1612void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1613 const Vector< sp<Track> >& tracksToRemove) 1614{ 1615 size_t count = tracksToRemove.size(); 1616 if (CC_UNLIKELY(count)) { 1617 for (size_t i = 0 ; i < count ; i++) { 1618 const sp<Track>& track = tracksToRemove.itemAt(i); 1619 if ((track->sharedBuffer() != 0) && 1620 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1621 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1622 } 1623 } 1624 } 1625 1626} 1627 1628void AudioFlinger::PlaybackThread::checkSilentMode_l() 1629{ 1630 if (!mMasterMute) { 1631 char value[PROPERTY_VALUE_MAX]; 1632 if (property_get("ro.audio.silent", value, "0") > 0) { 1633 char *endptr; 1634 unsigned long ul = strtoul(value, &endptr, 0); 1635 if (*endptr == '\0' && ul != 0) { 1636 ALOGD("Silence is golden"); 1637 // The setprop command will not allow a property to be changed after 1638 // the first time it is set, so we don't have to worry about un-muting. 1639 setMasterMute_l(true); 1640 } 1641 } 1642 } 1643} 1644 1645// shared by MIXER and DIRECT, overridden by DUPLICATING 1646void AudioFlinger::PlaybackThread::threadLoop_write() 1647{ 1648 // FIXME rewrite to reduce number of system calls 1649 mLastWriteTime = systemTime(); 1650 mInWrite = true; 1651 int bytesWritten; 1652 1653 // If an NBAIO sink is present, use it to write the normal mixer's submix 1654 if (mNormalSink != 0) { 1655#define mBitShift 2 // FIXME 1656 size_t count = mixBufferSize >> mBitShift; 1657 ATRACE_BEGIN("write"); 1658 // update the setpoint when AudioFlinger::mScreenState changes 1659 uint32_t screenState = AudioFlinger::mScreenState; 1660 if (screenState != mScreenState) { 1661 mScreenState = screenState; 1662 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1663 if (pipe != NULL) { 1664 pipe->setAvgFrames((mScreenState & 1) ? 1665 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1666 } 1667 } 1668 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1669 ATRACE_END(); 1670 if (framesWritten > 0) { 1671 bytesWritten = framesWritten << mBitShift; 1672 } else { 1673 bytesWritten = framesWritten; 1674 } 1675 // otherwise use the HAL / AudioStreamOut directly 1676 } else { 1677 // Direct output thread. 1678 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1679 } 1680 1681 if (bytesWritten > 0) { 1682 mBytesWritten += mixBufferSize; 1683 } 1684 mNumWrites++; 1685 mInWrite = false; 1686} 1687 1688/* 1689The derived values that are cached: 1690 - mixBufferSize from frame count * frame size 1691 - activeSleepTime from activeSleepTimeUs() 1692 - idleSleepTime from idleSleepTimeUs() 1693 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1694 - maxPeriod from frame count and sample rate (MIXER only) 1695 1696The parameters that affect these derived values are: 1697 - frame count 1698 - frame size 1699 - sample rate 1700 - device type: A2DP or not 1701 - device latency 1702 - format: PCM or not 1703 - active sleep time 1704 - idle sleep time 1705*/ 1706 1707void AudioFlinger::PlaybackThread::cacheParameters_l() 1708{ 1709 mixBufferSize = mNormalFrameCount * mFrameSize; 1710 activeSleepTime = activeSleepTimeUs(); 1711 idleSleepTime = idleSleepTimeUs(); 1712} 1713 1714void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1715{ 1716 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1717 this, streamType, mTracks.size()); 1718 Mutex::Autolock _l(mLock); 1719 1720 size_t size = mTracks.size(); 1721 for (size_t i = 0; i < size; i++) { 1722 sp<Track> t = mTracks[i]; 1723 if (t->streamType() == streamType) { 1724 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 1725 t->mCblk->cv.signal(); 1726 } 1727 } 1728} 1729 1730status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1731{ 1732 int session = chain->sessionId(); 1733 int16_t *buffer = mMixBuffer; 1734 bool ownsBuffer = false; 1735 1736 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1737 if (session > 0) { 1738 // Only one effect chain can be present in direct output thread and it uses 1739 // the mix buffer as input 1740 if (mType != DIRECT) { 1741 size_t numSamples = mNormalFrameCount * mChannelCount; 1742 buffer = new int16_t[numSamples]; 1743 memset(buffer, 0, numSamples * sizeof(int16_t)); 1744 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1745 ownsBuffer = true; 1746 } 1747 1748 // Attach all tracks with same session ID to this chain. 1749 for (size_t i = 0; i < mTracks.size(); ++i) { 1750 sp<Track> track = mTracks[i]; 1751 if (session == track->sessionId()) { 1752 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1753 buffer); 1754 track->setMainBuffer(buffer); 1755 chain->incTrackCnt(); 1756 } 1757 } 1758 1759 // indicate all active tracks in the chain 1760 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1761 sp<Track> track = mActiveTracks[i].promote(); 1762 if (track == 0) { 1763 continue; 1764 } 1765 if (session == track->sessionId()) { 1766 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1767 chain->incActiveTrackCnt(); 1768 } 1769 } 1770 } 1771 1772 chain->setInBuffer(buffer, ownsBuffer); 1773 chain->setOutBuffer(mMixBuffer); 1774 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1775 // chains list in order to be processed last as it contains output stage effects 1776 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1777 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1778 // after track specific effects and before output stage 1779 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1780 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1781 // Effect chain for other sessions are inserted at beginning of effect 1782 // chains list to be processed before output mix effects. Relative order between other 1783 // sessions is not important 1784 size_t size = mEffectChains.size(); 1785 size_t i = 0; 1786 for (i = 0; i < size; i++) { 1787 if (mEffectChains[i]->sessionId() < session) { 1788 break; 1789 } 1790 } 1791 mEffectChains.insertAt(chain, i); 1792 checkSuspendOnAddEffectChain_l(chain); 1793 1794 return NO_ERROR; 1795} 1796 1797size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1798{ 1799 int session = chain->sessionId(); 1800 1801 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1802 1803 for (size_t i = 0; i < mEffectChains.size(); i++) { 1804 if (chain == mEffectChains[i]) { 1805 mEffectChains.removeAt(i); 1806 // detach all active tracks from the chain 1807 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1808 sp<Track> track = mActiveTracks[i].promote(); 1809 if (track == 0) { 1810 continue; 1811 } 1812 if (session == track->sessionId()) { 1813 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1814 chain.get(), session); 1815 chain->decActiveTrackCnt(); 1816 } 1817 } 1818 1819 // detach all tracks with same session ID from this chain 1820 for (size_t i = 0; i < mTracks.size(); ++i) { 1821 sp<Track> track = mTracks[i]; 1822 if (session == track->sessionId()) { 1823 track->setMainBuffer(mMixBuffer); 1824 chain->decTrackCnt(); 1825 } 1826 } 1827 break; 1828 } 1829 } 1830 return mEffectChains.size(); 1831} 1832 1833status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1834 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1835{ 1836 Mutex::Autolock _l(mLock); 1837 return attachAuxEffect_l(track, EffectId); 1838} 1839 1840status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1841 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1842{ 1843 status_t status = NO_ERROR; 1844 1845 if (EffectId == 0) { 1846 track->setAuxBuffer(0, NULL); 1847 } else { 1848 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1849 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1850 if (effect != 0) { 1851 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1852 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1853 } else { 1854 status = INVALID_OPERATION; 1855 } 1856 } else { 1857 status = BAD_VALUE; 1858 } 1859 } 1860 return status; 1861} 1862 1863void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1864{ 1865 for (size_t i = 0; i < mTracks.size(); ++i) { 1866 sp<Track> track = mTracks[i]; 1867 if (track->auxEffectId() == effectId) { 1868 attachAuxEffect_l(track, 0); 1869 } 1870 } 1871} 1872 1873bool AudioFlinger::PlaybackThread::threadLoop() 1874{ 1875 Vector< sp<Track> > tracksToRemove; 1876 1877 standbyTime = systemTime(); 1878 1879 // MIXER 1880 nsecs_t lastWarning = 0; 1881 1882 // DUPLICATING 1883 // FIXME could this be made local to while loop? 1884 writeFrames = 0; 1885 1886 cacheParameters_l(); 1887 sleepTime = idleSleepTime; 1888 1889 if (mType == MIXER) { 1890 sleepTimeShift = 0; 1891 } 1892 1893 CpuStats cpuStats; 1894 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1895 1896 acquireWakeLock(); 1897 1898 while (!exitPending()) 1899 { 1900 cpuStats.sample(myName); 1901 1902 Vector< sp<EffectChain> > effectChains; 1903 1904 processConfigEvents(); 1905 1906 { // scope for mLock 1907 1908 Mutex::Autolock _l(mLock); 1909 1910 if (checkForNewParameters_l()) { 1911 cacheParameters_l(); 1912 } 1913 1914 saveOutputTracks(); 1915 1916 // put audio hardware into standby after short delay 1917 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1918 isSuspended())) { 1919 if (!mStandby) { 1920 1921 threadLoop_standby(); 1922 1923 mStandby = true; 1924 } 1925 1926 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1927 // we're about to wait, flush the binder command buffer 1928 IPCThreadState::self()->flushCommands(); 1929 1930 clearOutputTracks(); 1931 1932 if (exitPending()) { 1933 break; 1934 } 1935 1936 releaseWakeLock_l(); 1937 // wait until we have something to do... 1938 ALOGV("%s going to sleep", myName.string()); 1939 mWaitWorkCV.wait(mLock); 1940 ALOGV("%s waking up", myName.string()); 1941 acquireWakeLock_l(); 1942 1943 mMixerStatus = MIXER_IDLE; 1944 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1945 mBytesWritten = 0; 1946 1947 checkSilentMode_l(); 1948 1949 standbyTime = systemTime() + standbyDelay; 1950 sleepTime = idleSleepTime; 1951 if (mType == MIXER) { 1952 sleepTimeShift = 0; 1953 } 1954 1955 continue; 1956 } 1957 } 1958 1959 // mMixerStatusIgnoringFastTracks is also updated internally 1960 mMixerStatus = prepareTracks_l(&tracksToRemove); 1961 1962 // prevent any changes in effect chain list and in each effect chain 1963 // during mixing and effect process as the audio buffers could be deleted 1964 // or modified if an effect is created or deleted 1965 lockEffectChains_l(effectChains); 1966 } 1967 1968 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1969 threadLoop_mix(); 1970 } else { 1971 threadLoop_sleepTime(); 1972 } 1973 1974 if (isSuspended()) { 1975 sleepTime = suspendSleepTimeUs(); 1976 mBytesWritten += mixBufferSize; 1977 } 1978 1979 // only process effects if we're going to write 1980 if (sleepTime == 0) { 1981 for (size_t i = 0; i < effectChains.size(); i ++) { 1982 effectChains[i]->process_l(); 1983 } 1984 } 1985 1986 // enable changes in effect chain 1987 unlockEffectChains(effectChains); 1988 1989 // sleepTime == 0 means we must write to audio hardware 1990 if (sleepTime == 0) { 1991 1992 threadLoop_write(); 1993 1994if (mType == MIXER) { 1995 // write blocked detection 1996 nsecs_t now = systemTime(); 1997 nsecs_t delta = now - mLastWriteTime; 1998 if (!mStandby && delta > maxPeriod) { 1999 mNumDelayedWrites++; 2000 if ((now - lastWarning) > kWarningThrottleNs) { 2001 ATRACE_NAME("underrun"); 2002 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2003 ns2ms(delta), mNumDelayedWrites, this); 2004 lastWarning = now; 2005 } 2006 } 2007} 2008 2009 mStandby = false; 2010 } else { 2011 usleep(sleepTime); 2012 } 2013 2014 // Finally let go of removed track(s), without the lock held 2015 // since we can't guarantee the destructors won't acquire that 2016 // same lock. This will also mutate and push a new fast mixer state. 2017 threadLoop_removeTracks(tracksToRemove); 2018 tracksToRemove.clear(); 2019 2020 // FIXME I don't understand the need for this here; 2021 // it was in the original code but maybe the 2022 // assignment in saveOutputTracks() makes this unnecessary? 2023 clearOutputTracks(); 2024 2025 // Effect chains will be actually deleted here if they were removed from 2026 // mEffectChains list during mixing or effects processing 2027 effectChains.clear(); 2028 2029 // FIXME Note that the above .clear() is no longer necessary since effectChains 2030 // is now local to this block, but will keep it for now (at least until merge done). 2031 } 2032 2033 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2034 if (mType == MIXER || mType == DIRECT) { 2035 // put output stream into standby mode 2036 if (!mStandby) { 2037 mOutput->stream->common.standby(&mOutput->stream->common); 2038 } 2039 } 2040 2041 releaseWakeLock(); 2042 2043 ALOGV("Thread %p type %d exiting", this, mType); 2044 return false; 2045} 2046 2047 2048// ---------------------------------------------------------------------------- 2049 2050AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2051 audio_io_handle_t id, audio_devices_t device, type_t type) 2052 : PlaybackThread(audioFlinger, output, id, device, type), 2053 // mAudioMixer below 2054 // mFastMixer below 2055 mFastMixerFutex(0) 2056 // mOutputSink below 2057 // mPipeSink below 2058 // mNormalSink below 2059{ 2060 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2061 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2062 "mFrameCount=%d, mNormalFrameCount=%d", 2063 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2064 mNormalFrameCount); 2065 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2066 2067 // FIXME - Current mixer implementation only supports stereo output 2068 if (mChannelCount != FCC_2) { 2069 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2070 } 2071 2072 // create an NBAIO sink for the HAL output stream, and negotiate 2073 mOutputSink = new AudioStreamOutSink(output->stream); 2074 size_t numCounterOffers = 0; 2075 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2076 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2077 ALOG_ASSERT(index == 0); 2078 2079 // initialize fast mixer depending on configuration 2080 bool initFastMixer; 2081 switch (kUseFastMixer) { 2082 case FastMixer_Never: 2083 initFastMixer = false; 2084 break; 2085 case FastMixer_Always: 2086 initFastMixer = true; 2087 break; 2088 case FastMixer_Static: 2089 case FastMixer_Dynamic: 2090 initFastMixer = mFrameCount < mNormalFrameCount; 2091 break; 2092 } 2093 if (initFastMixer) { 2094 2095 // create a MonoPipe to connect our submix to FastMixer 2096 NBAIO_Format format = mOutputSink->format(); 2097 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2098 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2099 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2100 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2101 const NBAIO_Format offers[1] = {format}; 2102 size_t numCounterOffers = 0; 2103 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2104 ALOG_ASSERT(index == 0); 2105 monoPipe->setAvgFrames((mScreenState & 1) ? 2106 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2107 mPipeSink = monoPipe; 2108 2109#ifdef TEE_SINK_FRAMES 2110 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2111 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2112 numCounterOffers = 0; 2113 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2114 ALOG_ASSERT(index == 0); 2115 mTeeSink = teeSink; 2116 PipeReader *teeSource = new PipeReader(*teeSink); 2117 numCounterOffers = 0; 2118 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2119 ALOG_ASSERT(index == 0); 2120 mTeeSource = teeSource; 2121#endif 2122 2123 // create fast mixer and configure it initially with just one fast track for our submix 2124 mFastMixer = new FastMixer(); 2125 FastMixerStateQueue *sq = mFastMixer->sq(); 2126#ifdef STATE_QUEUE_DUMP 2127 sq->setObserverDump(&mStateQueueObserverDump); 2128 sq->setMutatorDump(&mStateQueueMutatorDump); 2129#endif 2130 FastMixerState *state = sq->begin(); 2131 FastTrack *fastTrack = &state->mFastTracks[0]; 2132 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2133 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2134 fastTrack->mVolumeProvider = NULL; 2135 fastTrack->mGeneration++; 2136 state->mFastTracksGen++; 2137 state->mTrackMask = 1; 2138 // fast mixer will use the HAL output sink 2139 state->mOutputSink = mOutputSink.get(); 2140 state->mOutputSinkGen++; 2141 state->mFrameCount = mFrameCount; 2142 state->mCommand = FastMixerState::COLD_IDLE; 2143 // already done in constructor initialization list 2144 //mFastMixerFutex = 0; 2145 state->mColdFutexAddr = &mFastMixerFutex; 2146 state->mColdGen++; 2147 state->mDumpState = &mFastMixerDumpState; 2148 state->mTeeSink = mTeeSink.get(); 2149 sq->end(); 2150 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2151 2152 // start the fast mixer 2153 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2154 pid_t tid = mFastMixer->getTid(); 2155 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2156 if (err != 0) { 2157 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2158 kPriorityFastMixer, getpid_cached, tid, err); 2159 } 2160 2161#ifdef AUDIO_WATCHDOG 2162 // create and start the watchdog 2163 mAudioWatchdog = new AudioWatchdog(); 2164 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2165 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2166 tid = mAudioWatchdog->getTid(); 2167 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2168 if (err != 0) { 2169 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2170 kPriorityFastMixer, getpid_cached, tid, err); 2171 } 2172#endif 2173 2174 } else { 2175 mFastMixer = NULL; 2176 } 2177 2178 switch (kUseFastMixer) { 2179 case FastMixer_Never: 2180 case FastMixer_Dynamic: 2181 mNormalSink = mOutputSink; 2182 break; 2183 case FastMixer_Always: 2184 mNormalSink = mPipeSink; 2185 break; 2186 case FastMixer_Static: 2187 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2188 break; 2189 } 2190} 2191 2192AudioFlinger::MixerThread::~MixerThread() 2193{ 2194 if (mFastMixer != NULL) { 2195 FastMixerStateQueue *sq = mFastMixer->sq(); 2196 FastMixerState *state = sq->begin(); 2197 if (state->mCommand == FastMixerState::COLD_IDLE) { 2198 int32_t old = android_atomic_inc(&mFastMixerFutex); 2199 if (old == -1) { 2200 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2201 } 2202 } 2203 state->mCommand = FastMixerState::EXIT; 2204 sq->end(); 2205 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2206 mFastMixer->join(); 2207 // Though the fast mixer thread has exited, it's state queue is still valid. 2208 // We'll use that extract the final state which contains one remaining fast track 2209 // corresponding to our sub-mix. 2210 state = sq->begin(); 2211 ALOG_ASSERT(state->mTrackMask == 1); 2212 FastTrack *fastTrack = &state->mFastTracks[0]; 2213 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2214 delete fastTrack->mBufferProvider; 2215 sq->end(false /*didModify*/); 2216 delete mFastMixer; 2217#ifdef AUDIO_WATCHDOG 2218 if (mAudioWatchdog != 0) { 2219 mAudioWatchdog->requestExit(); 2220 mAudioWatchdog->requestExitAndWait(); 2221 mAudioWatchdog.clear(); 2222 } 2223#endif 2224 } 2225 delete mAudioMixer; 2226} 2227 2228 2229uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2230{ 2231 if (mFastMixer != NULL) { 2232 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2233 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2234 } 2235 return latency; 2236} 2237 2238 2239void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2240{ 2241 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2242} 2243 2244void AudioFlinger::MixerThread::threadLoop_write() 2245{ 2246 // FIXME we should only do one push per cycle; confirm this is true 2247 // Start the fast mixer if it's not already running 2248 if (mFastMixer != NULL) { 2249 FastMixerStateQueue *sq = mFastMixer->sq(); 2250 FastMixerState *state = sq->begin(); 2251 if (state->mCommand != FastMixerState::MIX_WRITE && 2252 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2253 if (state->mCommand == FastMixerState::COLD_IDLE) { 2254 int32_t old = android_atomic_inc(&mFastMixerFutex); 2255 if (old == -1) { 2256 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2257 } 2258#ifdef AUDIO_WATCHDOG 2259 if (mAudioWatchdog != 0) { 2260 mAudioWatchdog->resume(); 2261 } 2262#endif 2263 } 2264 state->mCommand = FastMixerState::MIX_WRITE; 2265 sq->end(); 2266 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2267 if (kUseFastMixer == FastMixer_Dynamic) { 2268 mNormalSink = mPipeSink; 2269 } 2270 } else { 2271 sq->end(false /*didModify*/); 2272 } 2273 } 2274 PlaybackThread::threadLoop_write(); 2275} 2276 2277void AudioFlinger::MixerThread::threadLoop_standby() 2278{ 2279 // Idle the fast mixer if it's currently running 2280 if (mFastMixer != NULL) { 2281 FastMixerStateQueue *sq = mFastMixer->sq(); 2282 FastMixerState *state = sq->begin(); 2283 if (!(state->mCommand & FastMixerState::IDLE)) { 2284 state->mCommand = FastMixerState::COLD_IDLE; 2285 state->mColdFutexAddr = &mFastMixerFutex; 2286 state->mColdGen++; 2287 mFastMixerFutex = 0; 2288 sq->end(); 2289 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2290 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2291 if (kUseFastMixer == FastMixer_Dynamic) { 2292 mNormalSink = mOutputSink; 2293 } 2294#ifdef AUDIO_WATCHDOG 2295 if (mAudioWatchdog != 0) { 2296 mAudioWatchdog->pause(); 2297 } 2298#endif 2299 } else { 2300 sq->end(false /*didModify*/); 2301 } 2302 } 2303 PlaybackThread::threadLoop_standby(); 2304} 2305 2306// shared by MIXER and DIRECT, overridden by DUPLICATING 2307void AudioFlinger::PlaybackThread::threadLoop_standby() 2308{ 2309 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2310 mOutput->stream->common.standby(&mOutput->stream->common); 2311} 2312 2313void AudioFlinger::MixerThread::threadLoop_mix() 2314{ 2315 // obtain the presentation timestamp of the next output buffer 2316 int64_t pts; 2317 status_t status = INVALID_OPERATION; 2318 2319 if (mNormalSink != 0) { 2320 status = mNormalSink->getNextWriteTimestamp(&pts); 2321 } else { 2322 status = mOutputSink->getNextWriteTimestamp(&pts); 2323 } 2324 2325 if (status != NO_ERROR) { 2326 pts = AudioBufferProvider::kInvalidPTS; 2327 } 2328 2329 // mix buffers... 2330 mAudioMixer->process(pts); 2331 // increase sleep time progressively when application underrun condition clears. 2332 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2333 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2334 // such that we would underrun the audio HAL. 2335 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2336 sleepTimeShift--; 2337 } 2338 sleepTime = 0; 2339 standbyTime = systemTime() + standbyDelay; 2340 //TODO: delay standby when effects have a tail 2341} 2342 2343void AudioFlinger::MixerThread::threadLoop_sleepTime() 2344{ 2345 // If no tracks are ready, sleep once for the duration of an output 2346 // buffer size, then write 0s to the output 2347 if (sleepTime == 0) { 2348 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2349 sleepTime = activeSleepTime >> sleepTimeShift; 2350 if (sleepTime < kMinThreadSleepTimeUs) { 2351 sleepTime = kMinThreadSleepTimeUs; 2352 } 2353 // reduce sleep time in case of consecutive application underruns to avoid 2354 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2355 // duration we would end up writing less data than needed by the audio HAL if 2356 // the condition persists. 2357 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2358 sleepTimeShift++; 2359 } 2360 } else { 2361 sleepTime = idleSleepTime; 2362 } 2363 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2364 memset (mMixBuffer, 0, mixBufferSize); 2365 sleepTime = 0; 2366 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2367 "anticipated start"); 2368 } 2369 // TODO add standby time extension fct of effect tail 2370} 2371 2372// prepareTracks_l() must be called with ThreadBase::mLock held 2373AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2374 Vector< sp<Track> > *tracksToRemove) 2375{ 2376 2377 mixer_state mixerStatus = MIXER_IDLE; 2378 // find out which tracks need to be processed 2379 size_t count = mActiveTracks.size(); 2380 size_t mixedTracks = 0; 2381 size_t tracksWithEffect = 0; 2382 // counts only _active_ fast tracks 2383 size_t fastTracks = 0; 2384 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2385 2386 float masterVolume = mMasterVolume; 2387 bool masterMute = mMasterMute; 2388 2389 if (masterMute) { 2390 masterVolume = 0; 2391 } 2392 // Delegate master volume control to effect in output mix effect chain if needed 2393 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2394 if (chain != 0) { 2395 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2396 chain->setVolume_l(&v, &v); 2397 masterVolume = (float)((v + (1 << 23)) >> 24); 2398 chain.clear(); 2399 } 2400 2401 // prepare a new state to push 2402 FastMixerStateQueue *sq = NULL; 2403 FastMixerState *state = NULL; 2404 bool didModify = false; 2405 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2406 if (mFastMixer != NULL) { 2407 sq = mFastMixer->sq(); 2408 state = sq->begin(); 2409 } 2410 2411 for (size_t i=0 ; i<count ; i++) { 2412 sp<Track> t = mActiveTracks[i].promote(); 2413 if (t == 0) { 2414 continue; 2415 } 2416 2417 // this const just means the local variable doesn't change 2418 Track* const track = t.get(); 2419 2420 // process fast tracks 2421 if (track->isFastTrack()) { 2422 2423 // It's theoretically possible (though unlikely) for a fast track to be created 2424 // and then removed within the same normal mix cycle. This is not a problem, as 2425 // the track never becomes active so it's fast mixer slot is never touched. 2426 // The converse, of removing an (active) track and then creating a new track 2427 // at the identical fast mixer slot within the same normal mix cycle, 2428 // is impossible because the slot isn't marked available until the end of each cycle. 2429 int j = track->mFastIndex; 2430 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2431 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2432 FastTrack *fastTrack = &state->mFastTracks[j]; 2433 2434 // Determine whether the track is currently in underrun condition, 2435 // and whether it had a recent underrun. 2436 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2437 FastTrackUnderruns underruns = ftDump->mUnderruns; 2438 uint32_t recentFull = (underruns.mBitFields.mFull - 2439 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2440 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2441 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2442 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2443 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2444 uint32_t recentUnderruns = recentPartial + recentEmpty; 2445 track->mObservedUnderruns = underruns; 2446 // don't count underruns that occur while stopping or pausing 2447 // or stopped which can occur when flush() is called while active 2448 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2449 track->mUnderrunCount += recentUnderruns; 2450 } 2451 2452 // This is similar to the state machine for normal tracks, 2453 // with a few modifications for fast tracks. 2454 bool isActive = true; 2455 switch (track->mState) { 2456 case TrackBase::STOPPING_1: 2457 // track stays active in STOPPING_1 state until first underrun 2458 if (recentUnderruns > 0) { 2459 track->mState = TrackBase::STOPPING_2; 2460 } 2461 break; 2462 case TrackBase::PAUSING: 2463 // ramp down is not yet implemented 2464 track->setPaused(); 2465 break; 2466 case TrackBase::RESUMING: 2467 // ramp up is not yet implemented 2468 track->mState = TrackBase::ACTIVE; 2469 break; 2470 case TrackBase::ACTIVE: 2471 if (recentFull > 0 || recentPartial > 0) { 2472 // track has provided at least some frames recently: reset retry count 2473 track->mRetryCount = kMaxTrackRetries; 2474 } 2475 if (recentUnderruns == 0) { 2476 // no recent underruns: stay active 2477 break; 2478 } 2479 // there has recently been an underrun of some kind 2480 if (track->sharedBuffer() == 0) { 2481 // were any of the recent underruns "empty" (no frames available)? 2482 if (recentEmpty == 0) { 2483 // no, then ignore the partial underruns as they are allowed indefinitely 2484 break; 2485 } 2486 // there has recently been an "empty" underrun: decrement the retry counter 2487 if (--(track->mRetryCount) > 0) { 2488 break; 2489 } 2490 // indicate to client process that the track was disabled because of underrun; 2491 // it will then automatically call start() when data is available 2492 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2493 // remove from active list, but state remains ACTIVE [confusing but true] 2494 isActive = false; 2495 break; 2496 } 2497 // fall through 2498 case TrackBase::STOPPING_2: 2499 case TrackBase::PAUSED: 2500 case TrackBase::TERMINATED: 2501 case TrackBase::STOPPED: 2502 case TrackBase::FLUSHED: // flush() while active 2503 // Check for presentation complete if track is inactive 2504 // We have consumed all the buffers of this track. 2505 // This would be incomplete if we auto-paused on underrun 2506 { 2507 size_t audioHALFrames = 2508 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2509 size_t framesWritten = mBytesWritten / mFrameSize; 2510 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2511 // track stays in active list until presentation is complete 2512 break; 2513 } 2514 } 2515 if (track->isStopping_2()) { 2516 track->mState = TrackBase::STOPPED; 2517 } 2518 if (track->isStopped()) { 2519 // Can't reset directly, as fast mixer is still polling this track 2520 // track->reset(); 2521 // So instead mark this track as needing to be reset after push with ack 2522 resetMask |= 1 << i; 2523 } 2524 isActive = false; 2525 break; 2526 case TrackBase::IDLE: 2527 default: 2528 LOG_FATAL("unexpected track state %d", track->mState); 2529 } 2530 2531 if (isActive) { 2532 // was it previously inactive? 2533 if (!(state->mTrackMask & (1 << j))) { 2534 ExtendedAudioBufferProvider *eabp = track; 2535 VolumeProvider *vp = track; 2536 fastTrack->mBufferProvider = eabp; 2537 fastTrack->mVolumeProvider = vp; 2538 fastTrack->mSampleRate = track->mSampleRate; 2539 fastTrack->mChannelMask = track->mChannelMask; 2540 fastTrack->mGeneration++; 2541 state->mTrackMask |= 1 << j; 2542 didModify = true; 2543 // no acknowledgement required for newly active tracks 2544 } 2545 // cache the combined master volume and stream type volume for fast mixer; this 2546 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2547 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2548 ++fastTracks; 2549 } else { 2550 // was it previously active? 2551 if (state->mTrackMask & (1 << j)) { 2552 fastTrack->mBufferProvider = NULL; 2553 fastTrack->mGeneration++; 2554 state->mTrackMask &= ~(1 << j); 2555 didModify = true; 2556 // If any fast tracks were removed, we must wait for acknowledgement 2557 // because we're about to decrement the last sp<> on those tracks. 2558 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2559 } else { 2560 LOG_FATAL("fast track %d should have been active", j); 2561 } 2562 tracksToRemove->add(track); 2563 // Avoids a misleading display in dumpsys 2564 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2565 } 2566 continue; 2567 } 2568 2569 { // local variable scope to avoid goto warning 2570 2571 audio_track_cblk_t* cblk = track->cblk(); 2572 2573 // The first time a track is added we wait 2574 // for all its buffers to be filled before processing it 2575 int name = track->name(); 2576 // make sure that we have enough frames to mix one full buffer. 2577 // enforce this condition only once to enable draining the buffer in case the client 2578 // app does not call stop() and relies on underrun to stop: 2579 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2580 // during last round 2581 uint32_t minFrames = 1; 2582 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2583 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2584 if (t->sampleRate() == mSampleRate) { 2585 minFrames = mNormalFrameCount; 2586 } else { 2587 // +1 for rounding and +1 for additional sample needed for interpolation 2588 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2589 // add frames already consumed but not yet released by the resampler 2590 // because cblk->framesReady() will include these frames 2591 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2592 // the minimum track buffer size is normally twice the number of frames necessary 2593 // to fill one buffer and the resampler should not leave more than one buffer worth 2594 // of unreleased frames after each pass, but just in case... 2595 ALOG_ASSERT(minFrames <= cblk->frameCount); 2596 } 2597 } 2598 if ((track->framesReady() >= minFrames) && track->isReady() && 2599 !track->isPaused() && !track->isTerminated()) 2600 { 2601 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2602 this); 2603 2604 mixedTracks++; 2605 2606 // track->mainBuffer() != mMixBuffer means there is an effect chain 2607 // connected to the track 2608 chain.clear(); 2609 if (track->mainBuffer() != mMixBuffer) { 2610 chain = getEffectChain_l(track->sessionId()); 2611 // Delegate volume control to effect in track effect chain if needed 2612 if (chain != 0) { 2613 tracksWithEffect++; 2614 } else { 2615 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2616 "session %d", 2617 name, track->sessionId()); 2618 } 2619 } 2620 2621 2622 int param = AudioMixer::VOLUME; 2623 if (track->mFillingUpStatus == Track::FS_FILLED) { 2624 // no ramp for the first volume setting 2625 track->mFillingUpStatus = Track::FS_ACTIVE; 2626 if (track->mState == TrackBase::RESUMING) { 2627 track->mState = TrackBase::ACTIVE; 2628 param = AudioMixer::RAMP_VOLUME; 2629 } 2630 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2631 } else if (cblk->server != 0) { 2632 // If the track is stopped before the first frame was mixed, 2633 // do not apply ramp 2634 param = AudioMixer::RAMP_VOLUME; 2635 } 2636 2637 // compute volume for this track 2638 uint32_t vl, vr, va; 2639 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2640 vl = vr = va = 0; 2641 if (track->isPausing()) { 2642 track->setPaused(); 2643 } 2644 } else { 2645 2646 // read original volumes with volume control 2647 float typeVolume = mStreamTypes[track->streamType()].volume; 2648 float v = masterVolume * typeVolume; 2649 uint32_t vlr = cblk->getVolumeLR(); 2650 vl = vlr & 0xFFFF; 2651 vr = vlr >> 16; 2652 // track volumes come from shared memory, so can't be trusted and must be clamped 2653 if (vl > MAX_GAIN_INT) { 2654 ALOGV("Track left volume out of range: %04X", vl); 2655 vl = MAX_GAIN_INT; 2656 } 2657 if (vr > MAX_GAIN_INT) { 2658 ALOGV("Track right volume out of range: %04X", vr); 2659 vr = MAX_GAIN_INT; 2660 } 2661 // now apply the master volume and stream type volume 2662 vl = (uint32_t)(v * vl) << 12; 2663 vr = (uint32_t)(v * vr) << 12; 2664 // assuming master volume and stream type volume each go up to 1.0, 2665 // vl and vr are now in 8.24 format 2666 2667 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2668 // send level comes from shared memory and so may be corrupt 2669 if (sendLevel > MAX_GAIN_INT) { 2670 ALOGV("Track send level out of range: %04X", sendLevel); 2671 sendLevel = MAX_GAIN_INT; 2672 } 2673 va = (uint32_t)(v * sendLevel); 2674 } 2675 // Delegate volume control to effect in track effect chain if needed 2676 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2677 // Do not ramp volume if volume is controlled by effect 2678 param = AudioMixer::VOLUME; 2679 track->mHasVolumeController = true; 2680 } else { 2681 // force no volume ramp when volume controller was just disabled or removed 2682 // from effect chain to avoid volume spike 2683 if (track->mHasVolumeController) { 2684 param = AudioMixer::VOLUME; 2685 } 2686 track->mHasVolumeController = false; 2687 } 2688 2689 // Convert volumes from 8.24 to 4.12 format 2690 // This additional clamping is needed in case chain->setVolume_l() overshot 2691 vl = (vl + (1 << 11)) >> 12; 2692 if (vl > MAX_GAIN_INT) { 2693 vl = MAX_GAIN_INT; 2694 } 2695 vr = (vr + (1 << 11)) >> 12; 2696 if (vr > MAX_GAIN_INT) { 2697 vr = MAX_GAIN_INT; 2698 } 2699 2700 if (va > MAX_GAIN_INT) { 2701 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2702 } 2703 2704 // XXX: these things DON'T need to be done each time 2705 mAudioMixer->setBufferProvider(name, track); 2706 mAudioMixer->enable(name); 2707 2708 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2709 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2710 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2711 mAudioMixer->setParameter( 2712 name, 2713 AudioMixer::TRACK, 2714 AudioMixer::FORMAT, (void *)track->format()); 2715 mAudioMixer->setParameter( 2716 name, 2717 AudioMixer::TRACK, 2718 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2719 mAudioMixer->setParameter( 2720 name, 2721 AudioMixer::RESAMPLE, 2722 AudioMixer::SAMPLE_RATE, 2723 (void *)(cblk->sampleRate)); 2724 mAudioMixer->setParameter( 2725 name, 2726 AudioMixer::TRACK, 2727 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2728 mAudioMixer->setParameter( 2729 name, 2730 AudioMixer::TRACK, 2731 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2732 2733 // reset retry count 2734 track->mRetryCount = kMaxTrackRetries; 2735 2736 // If one track is ready, set the mixer ready if: 2737 // - the mixer was not ready during previous round OR 2738 // - no other track is not ready 2739 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2740 mixerStatus != MIXER_TRACKS_ENABLED) { 2741 mixerStatus = MIXER_TRACKS_READY; 2742 } 2743 } else { 2744 // clear effect chain input buffer if an active track underruns to avoid sending 2745 // previous audio buffer again to effects 2746 chain = getEffectChain_l(track->sessionId()); 2747 if (chain != 0) { 2748 chain->clearInputBuffer(); 2749 } 2750 2751 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2752 cblk->server, this); 2753 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2754 track->isStopped() || track->isPaused()) { 2755 // We have consumed all the buffers of this track. 2756 // Remove it from the list of active tracks. 2757 // TODO: use actual buffer filling status instead of latency when available from 2758 // audio HAL 2759 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2760 size_t framesWritten = mBytesWritten / mFrameSize; 2761 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2762 if (track->isStopped()) { 2763 track->reset(); 2764 } 2765 tracksToRemove->add(track); 2766 } 2767 } else { 2768 track->mUnderrunCount++; 2769 // No buffers for this track. Give it a few chances to 2770 // fill a buffer, then remove it from active list. 2771 if (--(track->mRetryCount) <= 0) { 2772 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2773 tracksToRemove->add(track); 2774 // indicate to client process that the track was disabled because of underrun; 2775 // it will then automatically call start() when data is available 2776 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2777 // If one track is not ready, mark the mixer also not ready if: 2778 // - the mixer was ready during previous round OR 2779 // - no other track is ready 2780 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2781 mixerStatus != MIXER_TRACKS_READY) { 2782 mixerStatus = MIXER_TRACKS_ENABLED; 2783 } 2784 } 2785 mAudioMixer->disable(name); 2786 } 2787 2788 } // local variable scope to avoid goto warning 2789track_is_ready: ; 2790 2791 } 2792 2793 // Push the new FastMixer state if necessary 2794 bool pauseAudioWatchdog = false; 2795 if (didModify) { 2796 state->mFastTracksGen++; 2797 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2798 if (kUseFastMixer == FastMixer_Dynamic && 2799 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2800 state->mCommand = FastMixerState::COLD_IDLE; 2801 state->mColdFutexAddr = &mFastMixerFutex; 2802 state->mColdGen++; 2803 mFastMixerFutex = 0; 2804 if (kUseFastMixer == FastMixer_Dynamic) { 2805 mNormalSink = mOutputSink; 2806 } 2807 // If we go into cold idle, need to wait for acknowledgement 2808 // so that fast mixer stops doing I/O. 2809 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2810 pauseAudioWatchdog = true; 2811 } 2812 sq->end(); 2813 } 2814 if (sq != NULL) { 2815 sq->end(didModify); 2816 sq->push(block); 2817 } 2818#ifdef AUDIO_WATCHDOG 2819 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2820 mAudioWatchdog->pause(); 2821 } 2822#endif 2823 2824 // Now perform the deferred reset on fast tracks that have stopped 2825 while (resetMask != 0) { 2826 size_t i = __builtin_ctz(resetMask); 2827 ALOG_ASSERT(i < count); 2828 resetMask &= ~(1 << i); 2829 sp<Track> t = mActiveTracks[i].promote(); 2830 if (t == 0) { 2831 continue; 2832 } 2833 Track* track = t.get(); 2834 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2835 track->reset(); 2836 } 2837 2838 // remove all the tracks that need to be... 2839 count = tracksToRemove->size(); 2840 if (CC_UNLIKELY(count)) { 2841 for (size_t i=0 ; i<count ; i++) { 2842 const sp<Track>& track = tracksToRemove->itemAt(i); 2843 mActiveTracks.remove(track); 2844 if (track->mainBuffer() != mMixBuffer) { 2845 chain = getEffectChain_l(track->sessionId()); 2846 if (chain != 0) { 2847 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2848 track->sessionId()); 2849 chain->decActiveTrackCnt(); 2850 } 2851 } 2852 if (track->isTerminated()) { 2853 removeTrack_l(track); 2854 } 2855 } 2856 } 2857 2858 // mix buffer must be cleared if all tracks are connected to an 2859 // effect chain as in this case the mixer will not write to 2860 // mix buffer and track effects will accumulate into it 2861 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2862 (mixedTracks == 0 && fastTracks > 0)) { 2863 // FIXME as a performance optimization, should remember previous zero status 2864 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2865 } 2866 2867 // if any fast tracks, then status is ready 2868 mMixerStatusIgnoringFastTracks = mixerStatus; 2869 if (fastTracks > 0) { 2870 mixerStatus = MIXER_TRACKS_READY; 2871 } 2872 return mixerStatus; 2873} 2874 2875// getTrackName_l() must be called with ThreadBase::mLock held 2876int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2877{ 2878 return mAudioMixer->getTrackName(channelMask, sessionId); 2879} 2880 2881// deleteTrackName_l() must be called with ThreadBase::mLock held 2882void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2883{ 2884 ALOGV("remove track (%d) and delete from mixer", name); 2885 mAudioMixer->deleteTrackName(name); 2886} 2887 2888// checkForNewParameters_l() must be called with ThreadBase::mLock held 2889bool AudioFlinger::MixerThread::checkForNewParameters_l() 2890{ 2891 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2892 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2893 bool reconfig = false; 2894 2895 while (!mNewParameters.isEmpty()) { 2896 2897 if (mFastMixer != NULL) { 2898 FastMixerStateQueue *sq = mFastMixer->sq(); 2899 FastMixerState *state = sq->begin(); 2900 if (!(state->mCommand & FastMixerState::IDLE)) { 2901 previousCommand = state->mCommand; 2902 state->mCommand = FastMixerState::HOT_IDLE; 2903 sq->end(); 2904 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2905 } else { 2906 sq->end(false /*didModify*/); 2907 } 2908 } 2909 2910 status_t status = NO_ERROR; 2911 String8 keyValuePair = mNewParameters[0]; 2912 AudioParameter param = AudioParameter(keyValuePair); 2913 int value; 2914 2915 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2916 reconfig = true; 2917 } 2918 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2919 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2920 status = BAD_VALUE; 2921 } else { 2922 reconfig = true; 2923 } 2924 } 2925 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2926 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2927 status = BAD_VALUE; 2928 } else { 2929 reconfig = true; 2930 } 2931 } 2932 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2933 // do not accept frame count changes if tracks are open as the track buffer 2934 // size depends on frame count and correct behavior would not be guaranteed 2935 // if frame count is changed after track creation 2936 if (!mTracks.isEmpty()) { 2937 status = INVALID_OPERATION; 2938 } else { 2939 reconfig = true; 2940 } 2941 } 2942 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2943#ifdef ADD_BATTERY_DATA 2944 // when changing the audio output device, call addBatteryData to notify 2945 // the change 2946 if (mOutDevice != value) { 2947 uint32_t params = 0; 2948 // check whether speaker is on 2949 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2950 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2951 } 2952 2953 audio_devices_t deviceWithoutSpeaker 2954 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2955 // check if any other device (except speaker) is on 2956 if (value & deviceWithoutSpeaker ) { 2957 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2958 } 2959 2960 if (params != 0) { 2961 addBatteryData(params); 2962 } 2963 } 2964#endif 2965 2966 // forward device change to effects that have requested to be 2967 // aware of attached audio device. 2968 mOutDevice = value; 2969 for (size_t i = 0; i < mEffectChains.size(); i++) { 2970 mEffectChains[i]->setDevice_l(mOutDevice); 2971 } 2972 } 2973 2974 if (status == NO_ERROR) { 2975 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2976 keyValuePair.string()); 2977 if (!mStandby && status == INVALID_OPERATION) { 2978 mOutput->stream->common.standby(&mOutput->stream->common); 2979 mStandby = true; 2980 mBytesWritten = 0; 2981 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2982 keyValuePair.string()); 2983 } 2984 if (status == NO_ERROR && reconfig) { 2985 delete mAudioMixer; 2986 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2987 mAudioMixer = NULL; 2988 readOutputParameters(); 2989 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2990 for (size_t i = 0; i < mTracks.size() ; i++) { 2991 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 2992 if (name < 0) { 2993 break; 2994 } 2995 mTracks[i]->mName = name; 2996 // limit track sample rate to 2 x new output sample rate 2997 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2998 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2999 } 3000 } 3001 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3002 } 3003 } 3004 3005 mNewParameters.removeAt(0); 3006 3007 mParamStatus = status; 3008 mParamCond.signal(); 3009 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3010 // already timed out waiting for the status and will never signal the condition. 3011 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3012 } 3013 3014 if (!(previousCommand & FastMixerState::IDLE)) { 3015 ALOG_ASSERT(mFastMixer != NULL); 3016 FastMixerStateQueue *sq = mFastMixer->sq(); 3017 FastMixerState *state = sq->begin(); 3018 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3019 state->mCommand = previousCommand; 3020 sq->end(); 3021 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3022 } 3023 3024 return reconfig; 3025} 3026 3027 3028void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3029{ 3030 const size_t SIZE = 256; 3031 char buffer[SIZE]; 3032 String8 result; 3033 3034 PlaybackThread::dumpInternals(fd, args); 3035 3036 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3037 result.append(buffer); 3038 write(fd, result.string(), result.size()); 3039 3040 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3041 FastMixerDumpState copy = mFastMixerDumpState; 3042 copy.dump(fd); 3043 3044#ifdef STATE_QUEUE_DUMP 3045 // Similar for state queue 3046 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3047 observerCopy.dump(fd); 3048 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3049 mutatorCopy.dump(fd); 3050#endif 3051 3052 // Write the tee output to a .wav file 3053 dumpTee(fd, mTeeSource, mId); 3054 3055#ifdef AUDIO_WATCHDOG 3056 if (mAudioWatchdog != 0) { 3057 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3058 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3059 wdCopy.dump(fd); 3060 } 3061#endif 3062} 3063 3064uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3065{ 3066 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3067} 3068 3069uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3070{ 3071 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3072} 3073 3074void AudioFlinger::MixerThread::cacheParameters_l() 3075{ 3076 PlaybackThread::cacheParameters_l(); 3077 3078 // FIXME: Relaxed timing because of a certain device that can't meet latency 3079 // Should be reduced to 2x after the vendor fixes the driver issue 3080 // increase threshold again due to low power audio mode. The way this warning 3081 // threshold is calculated and its usefulness should be reconsidered anyway. 3082 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3083} 3084 3085// ---------------------------------------------------------------------------- 3086 3087AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3088 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3089 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3090 // mLeftVolFloat, mRightVolFloat 3091{ 3092} 3093 3094AudioFlinger::DirectOutputThread::~DirectOutputThread() 3095{ 3096} 3097 3098AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3099 Vector< sp<Track> > *tracksToRemove 3100) 3101{ 3102 sp<Track> trackToRemove; 3103 3104 mixer_state mixerStatus = MIXER_IDLE; 3105 3106 // find out which tracks need to be processed 3107 if (mActiveTracks.size() != 0) { 3108 sp<Track> t = mActiveTracks[0].promote(); 3109 // The track died recently 3110 if (t == 0) { 3111 return MIXER_IDLE; 3112 } 3113 3114 Track* const track = t.get(); 3115 audio_track_cblk_t* cblk = track->cblk(); 3116 3117 // The first time a track is added we wait 3118 // for all its buffers to be filled before processing it 3119 uint32_t minFrames; 3120 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3121 minFrames = mNormalFrameCount; 3122 } else { 3123 minFrames = 1; 3124 } 3125 if ((track->framesReady() >= minFrames) && track->isReady() && 3126 !track->isPaused() && !track->isTerminated()) 3127 { 3128 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3129 3130 if (track->mFillingUpStatus == Track::FS_FILLED) { 3131 track->mFillingUpStatus = Track::FS_ACTIVE; 3132 mLeftVolFloat = mRightVolFloat = 0; 3133 if (track->mState == TrackBase::RESUMING) { 3134 track->mState = TrackBase::ACTIVE; 3135 } 3136 } 3137 3138 // compute volume for this track 3139 float left, right; 3140 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3141 left = right = 0; 3142 if (track->isPausing()) { 3143 track->setPaused(); 3144 } 3145 } else { 3146 float typeVolume = mStreamTypes[track->streamType()].volume; 3147 float v = mMasterVolume * typeVolume; 3148 uint32_t vlr = cblk->getVolumeLR(); 3149 float v_clamped = v * (vlr & 0xFFFF); 3150 if (v_clamped > MAX_GAIN) { 3151 v_clamped = MAX_GAIN; 3152 } 3153 left = v_clamped/MAX_GAIN; 3154 v_clamped = v * (vlr >> 16); 3155 if (v_clamped > MAX_GAIN) { 3156 v_clamped = MAX_GAIN; 3157 } 3158 right = v_clamped/MAX_GAIN; 3159 } 3160 3161 if (left != mLeftVolFloat || right != mRightVolFloat) { 3162 mLeftVolFloat = left; 3163 mRightVolFloat = right; 3164 3165 // Convert volumes from float to 8.24 3166 uint32_t vl = (uint32_t)(left * (1 << 24)); 3167 uint32_t vr = (uint32_t)(right * (1 << 24)); 3168 3169 // Delegate volume control to effect in track effect chain if needed 3170 // only one effect chain can be present on DirectOutputThread, so if 3171 // there is one, the track is connected to it 3172 if (!mEffectChains.isEmpty()) { 3173 // Do not ramp volume if volume is controlled by effect 3174 mEffectChains[0]->setVolume_l(&vl, &vr); 3175 left = (float)vl / (1 << 24); 3176 right = (float)vr / (1 << 24); 3177 } 3178 mOutput->stream->set_volume(mOutput->stream, left, right); 3179 } 3180 3181 // reset retry count 3182 track->mRetryCount = kMaxTrackRetriesDirect; 3183 mActiveTrack = t; 3184 mixerStatus = MIXER_TRACKS_READY; 3185 } else { 3186 // clear effect chain input buffer if an active track underruns to avoid sending 3187 // previous audio buffer again to effects 3188 if (!mEffectChains.isEmpty()) { 3189 mEffectChains[0]->clearInputBuffer(); 3190 } 3191 3192 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3193 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3194 track->isStopped() || track->isPaused()) { 3195 // We have consumed all the buffers of this track. 3196 // Remove it from the list of active tracks. 3197 // TODO: implement behavior for compressed audio 3198 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3199 size_t framesWritten = mBytesWritten / mFrameSize; 3200 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3201 if (track->isStopped()) { 3202 track->reset(); 3203 } 3204 trackToRemove = track; 3205 } 3206 } else { 3207 // No buffers for this track. Give it a few chances to 3208 // fill a buffer, then remove it from active list. 3209 if (--(track->mRetryCount) <= 0) { 3210 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3211 trackToRemove = track; 3212 } else { 3213 mixerStatus = MIXER_TRACKS_ENABLED; 3214 } 3215 } 3216 } 3217 } 3218 3219 // FIXME merge this with similar code for removing multiple tracks 3220 // remove all the tracks that need to be... 3221 if (CC_UNLIKELY(trackToRemove != 0)) { 3222 tracksToRemove->add(trackToRemove); 3223 mActiveTracks.remove(trackToRemove); 3224 if (!mEffectChains.isEmpty()) { 3225 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3226 trackToRemove->sessionId()); 3227 mEffectChains[0]->decActiveTrackCnt(); 3228 } 3229 if (trackToRemove->isTerminated()) { 3230 removeTrack_l(trackToRemove); 3231 } 3232 } 3233 3234 return mixerStatus; 3235} 3236 3237void AudioFlinger::DirectOutputThread::threadLoop_mix() 3238{ 3239 AudioBufferProvider::Buffer buffer; 3240 size_t frameCount = mFrameCount; 3241 int8_t *curBuf = (int8_t *)mMixBuffer; 3242 // output audio to hardware 3243 while (frameCount) { 3244 buffer.frameCount = frameCount; 3245 mActiveTrack->getNextBuffer(&buffer); 3246 if (CC_UNLIKELY(buffer.raw == NULL)) { 3247 memset(curBuf, 0, frameCount * mFrameSize); 3248 break; 3249 } 3250 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3251 frameCount -= buffer.frameCount; 3252 curBuf += buffer.frameCount * mFrameSize; 3253 mActiveTrack->releaseBuffer(&buffer); 3254 } 3255 sleepTime = 0; 3256 standbyTime = systemTime() + standbyDelay; 3257 mActiveTrack.clear(); 3258 3259} 3260 3261void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3262{ 3263 if (sleepTime == 0) { 3264 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3265 sleepTime = activeSleepTime; 3266 } else { 3267 sleepTime = idleSleepTime; 3268 } 3269 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3270 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3271 sleepTime = 0; 3272 } 3273} 3274 3275// getTrackName_l() must be called with ThreadBase::mLock held 3276int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3277 int sessionId) 3278{ 3279 return 0; 3280} 3281 3282// deleteTrackName_l() must be called with ThreadBase::mLock held 3283void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3284{ 3285} 3286 3287// checkForNewParameters_l() must be called with ThreadBase::mLock held 3288bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3289{ 3290 bool reconfig = false; 3291 3292 while (!mNewParameters.isEmpty()) { 3293 status_t status = NO_ERROR; 3294 String8 keyValuePair = mNewParameters[0]; 3295 AudioParameter param = AudioParameter(keyValuePair); 3296 int value; 3297 3298 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3299 // do not accept frame count changes if tracks are open as the track buffer 3300 // size depends on frame count and correct behavior would not be garantied 3301 // if frame count is changed after track creation 3302 if (!mTracks.isEmpty()) { 3303 status = INVALID_OPERATION; 3304 } else { 3305 reconfig = true; 3306 } 3307 } 3308 if (status == NO_ERROR) { 3309 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3310 keyValuePair.string()); 3311 if (!mStandby && status == INVALID_OPERATION) { 3312 mOutput->stream->common.standby(&mOutput->stream->common); 3313 mStandby = true; 3314 mBytesWritten = 0; 3315 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3316 keyValuePair.string()); 3317 } 3318 if (status == NO_ERROR && reconfig) { 3319 readOutputParameters(); 3320 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3321 } 3322 } 3323 3324 mNewParameters.removeAt(0); 3325 3326 mParamStatus = status; 3327 mParamCond.signal(); 3328 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3329 // already timed out waiting for the status and will never signal the condition. 3330 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3331 } 3332 return reconfig; 3333} 3334 3335uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3336{ 3337 uint32_t time; 3338 if (audio_is_linear_pcm(mFormat)) { 3339 time = PlaybackThread::activeSleepTimeUs(); 3340 } else { 3341 time = 10000; 3342 } 3343 return time; 3344} 3345 3346uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3347{ 3348 uint32_t time; 3349 if (audio_is_linear_pcm(mFormat)) { 3350 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3351 } else { 3352 time = 10000; 3353 } 3354 return time; 3355} 3356 3357uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3358{ 3359 uint32_t time; 3360 if (audio_is_linear_pcm(mFormat)) { 3361 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3362 } else { 3363 time = 10000; 3364 } 3365 return time; 3366} 3367 3368void AudioFlinger::DirectOutputThread::cacheParameters_l() 3369{ 3370 PlaybackThread::cacheParameters_l(); 3371 3372 // use shorter standby delay as on normal output to release 3373 // hardware resources as soon as possible 3374 standbyDelay = microseconds(activeSleepTime*2); 3375} 3376 3377// ---------------------------------------------------------------------------- 3378 3379AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3380 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3381 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3382 DUPLICATING), 3383 mWaitTimeMs(UINT_MAX) 3384{ 3385 addOutputTrack(mainThread); 3386} 3387 3388AudioFlinger::DuplicatingThread::~DuplicatingThread() 3389{ 3390 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3391 mOutputTracks[i]->destroy(); 3392 } 3393} 3394 3395void AudioFlinger::DuplicatingThread::threadLoop_mix() 3396{ 3397 // mix buffers... 3398 if (outputsReady(outputTracks)) { 3399 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3400 } else { 3401 memset(mMixBuffer, 0, mixBufferSize); 3402 } 3403 sleepTime = 0; 3404 writeFrames = mNormalFrameCount; 3405 standbyTime = systemTime() + standbyDelay; 3406} 3407 3408void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3409{ 3410 if (sleepTime == 0) { 3411 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3412 sleepTime = activeSleepTime; 3413 } else { 3414 sleepTime = idleSleepTime; 3415 } 3416 } else if (mBytesWritten != 0) { 3417 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3418 writeFrames = mNormalFrameCount; 3419 memset(mMixBuffer, 0, mixBufferSize); 3420 } else { 3421 // flush remaining overflow buffers in output tracks 3422 writeFrames = 0; 3423 } 3424 sleepTime = 0; 3425 } 3426} 3427 3428void AudioFlinger::DuplicatingThread::threadLoop_write() 3429{ 3430 for (size_t i = 0; i < outputTracks.size(); i++) { 3431 outputTracks[i]->write(mMixBuffer, writeFrames); 3432 } 3433 mBytesWritten += mixBufferSize; 3434} 3435 3436void AudioFlinger::DuplicatingThread::threadLoop_standby() 3437{ 3438 // DuplicatingThread implements standby by stopping all tracks 3439 for (size_t i = 0; i < outputTracks.size(); i++) { 3440 outputTracks[i]->stop(); 3441 } 3442} 3443 3444void AudioFlinger::DuplicatingThread::saveOutputTracks() 3445{ 3446 outputTracks = mOutputTracks; 3447} 3448 3449void AudioFlinger::DuplicatingThread::clearOutputTracks() 3450{ 3451 outputTracks.clear(); 3452} 3453 3454void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3455{ 3456 Mutex::Autolock _l(mLock); 3457 // FIXME explain this formula 3458 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3459 OutputTrack *outputTrack = new OutputTrack(thread, 3460 this, 3461 mSampleRate, 3462 mFormat, 3463 mChannelMask, 3464 frameCount); 3465 if (outputTrack->cblk() != NULL) { 3466 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3467 mOutputTracks.add(outputTrack); 3468 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3469 updateWaitTime_l(); 3470 } 3471} 3472 3473void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3474{ 3475 Mutex::Autolock _l(mLock); 3476 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3477 if (mOutputTracks[i]->thread() == thread) { 3478 mOutputTracks[i]->destroy(); 3479 mOutputTracks.removeAt(i); 3480 updateWaitTime_l(); 3481 return; 3482 } 3483 } 3484 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3485} 3486 3487// caller must hold mLock 3488void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3489{ 3490 mWaitTimeMs = UINT_MAX; 3491 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3492 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3493 if (strong != 0) { 3494 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3495 if (waitTimeMs < mWaitTimeMs) { 3496 mWaitTimeMs = waitTimeMs; 3497 } 3498 } 3499 } 3500} 3501 3502 3503bool AudioFlinger::DuplicatingThread::outputsReady( 3504 const SortedVector< sp<OutputTrack> > &outputTracks) 3505{ 3506 for (size_t i = 0; i < outputTracks.size(); i++) { 3507 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3508 if (thread == 0) { 3509 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3510 outputTracks[i].get()); 3511 return false; 3512 } 3513 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3514 // see note at standby() declaration 3515 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3516 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3517 thread.get()); 3518 return false; 3519 } 3520 } 3521 return true; 3522} 3523 3524uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3525{ 3526 return (mWaitTimeMs * 1000) / 2; 3527} 3528 3529void AudioFlinger::DuplicatingThread::cacheParameters_l() 3530{ 3531 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3532 updateWaitTime_l(); 3533 3534 MixerThread::cacheParameters_l(); 3535} 3536 3537// ---------------------------------------------------------------------------- 3538// Record 3539// ---------------------------------------------------------------------------- 3540 3541AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3542 AudioStreamIn *input, 3543 uint32_t sampleRate, 3544 audio_channel_mask_t channelMask, 3545 audio_io_handle_t id, 3546 audio_devices_t device, 3547 const sp<NBAIO_Sink>& teeSink) : 3548 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 3549 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3550 // mRsmpInIndex and mInputBytes set by readInputParameters() 3551 mReqChannelCount(popcount(channelMask)), 3552 mReqSampleRate(sampleRate), 3553 // mBytesRead is only meaningful while active, and so is cleared in start() 3554 // (but might be better to also clear here for dump?) 3555 mTeeSink(teeSink) 3556{ 3557 snprintf(mName, kNameLength, "AudioIn_%X", id); 3558 3559 readInputParameters(); 3560 3561} 3562 3563 3564AudioFlinger::RecordThread::~RecordThread() 3565{ 3566 delete[] mRsmpInBuffer; 3567 delete mResampler; 3568 delete[] mRsmpOutBuffer; 3569} 3570 3571void AudioFlinger::RecordThread::onFirstRef() 3572{ 3573 run(mName, PRIORITY_URGENT_AUDIO); 3574} 3575 3576status_t AudioFlinger::RecordThread::readyToRun() 3577{ 3578 status_t status = initCheck(); 3579 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3580 return status; 3581} 3582 3583bool AudioFlinger::RecordThread::threadLoop() 3584{ 3585 AudioBufferProvider::Buffer buffer; 3586 sp<RecordTrack> activeTrack; 3587 Vector< sp<EffectChain> > effectChains; 3588 3589 nsecs_t lastWarning = 0; 3590 3591 inputStandBy(); 3592 acquireWakeLock(); 3593 3594 // used to verify we've read at least once before evaluating how many bytes were read 3595 bool readOnce = false; 3596 3597 // start recording 3598 while (!exitPending()) { 3599 3600 processConfigEvents(); 3601 3602 { // scope for mLock 3603 Mutex::Autolock _l(mLock); 3604 checkForNewParameters_l(); 3605 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3606 standby(); 3607 3608 if (exitPending()) { 3609 break; 3610 } 3611 3612 releaseWakeLock_l(); 3613 ALOGV("RecordThread: loop stopping"); 3614 // go to sleep 3615 mWaitWorkCV.wait(mLock); 3616 ALOGV("RecordThread: loop starting"); 3617 acquireWakeLock_l(); 3618 continue; 3619 } 3620 if (mActiveTrack != 0) { 3621 if (mActiveTrack->mState == TrackBase::PAUSING) { 3622 standby(); 3623 mActiveTrack.clear(); 3624 mStartStopCond.broadcast(); 3625 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3626 if (mReqChannelCount != mActiveTrack->channelCount()) { 3627 mActiveTrack.clear(); 3628 mStartStopCond.broadcast(); 3629 } else if (readOnce) { 3630 // record start succeeds only if first read from audio input 3631 // succeeds 3632 if (mBytesRead >= 0) { 3633 mActiveTrack->mState = TrackBase::ACTIVE; 3634 } else { 3635 mActiveTrack.clear(); 3636 } 3637 mStartStopCond.broadcast(); 3638 } 3639 mStandby = false; 3640 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3641 removeTrack_l(mActiveTrack); 3642 mActiveTrack.clear(); 3643 } 3644 } 3645 lockEffectChains_l(effectChains); 3646 } 3647 3648 if (mActiveTrack != 0) { 3649 if (mActiveTrack->mState != TrackBase::ACTIVE && 3650 mActiveTrack->mState != TrackBase::RESUMING) { 3651 unlockEffectChains(effectChains); 3652 usleep(kRecordThreadSleepUs); 3653 continue; 3654 } 3655 for (size_t i = 0; i < effectChains.size(); i ++) { 3656 effectChains[i]->process_l(); 3657 } 3658 3659 buffer.frameCount = mFrameCount; 3660 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3661 readOnce = true; 3662 size_t framesOut = buffer.frameCount; 3663 if (mResampler == NULL) { 3664 // no resampling 3665 while (framesOut) { 3666 size_t framesIn = mFrameCount - mRsmpInIndex; 3667 if (framesIn) { 3668 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3669 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3670 mActiveTrack->mFrameSize; 3671 if (framesIn > framesOut) 3672 framesIn = framesOut; 3673 mRsmpInIndex += framesIn; 3674 framesOut -= framesIn; 3675 if (mChannelCount == mReqChannelCount || 3676 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3677 memcpy(dst, src, framesIn * mFrameSize); 3678 } else { 3679 if (mChannelCount == 1) { 3680 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3681 (int16_t *)src, framesIn); 3682 } else { 3683 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3684 (int16_t *)src, framesIn); 3685 } 3686 } 3687 } 3688 if (framesOut && mFrameCount == mRsmpInIndex) { 3689 void *readInto; 3690 if (framesOut == mFrameCount && 3691 (mChannelCount == mReqChannelCount || 3692 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3693 readInto = buffer.raw; 3694 framesOut = 0; 3695 } else { 3696 readInto = mRsmpInBuffer; 3697 mRsmpInIndex = 0; 3698 } 3699 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3700 if (mBytesRead <= 0) { 3701 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3702 { 3703 ALOGE("Error reading audio input"); 3704 // Force input into standby so that it tries to 3705 // recover at next read attempt 3706 inputStandBy(); 3707 usleep(kRecordThreadSleepUs); 3708 } 3709 mRsmpInIndex = mFrameCount; 3710 framesOut = 0; 3711 buffer.frameCount = 0; 3712 } else if (mTeeSink != 0) { 3713 (void) mTeeSink->write(readInto, 3714 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3715 } 3716 } 3717 } 3718 } else { 3719 // resampling 3720 3721 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3722 // alter output frame count as if we were expecting stereo samples 3723 if (mChannelCount == 1 && mReqChannelCount == 1) { 3724 framesOut >>= 1; 3725 } 3726 mResampler->resample(mRsmpOutBuffer, framesOut, 3727 this /* AudioBufferProvider* */); 3728 // ditherAndClamp() works as long as all buffers returned by 3729 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3730 if (mChannelCount == 2 && mReqChannelCount == 1) { 3731 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3732 // the resampler always outputs stereo samples: 3733 // do post stereo to mono conversion 3734 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3735 framesOut); 3736 } else { 3737 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3738 } 3739 3740 } 3741 if (mFramestoDrop == 0) { 3742 mActiveTrack->releaseBuffer(&buffer); 3743 } else { 3744 if (mFramestoDrop > 0) { 3745 mFramestoDrop -= buffer.frameCount; 3746 if (mFramestoDrop <= 0) { 3747 clearSyncStartEvent(); 3748 } 3749 } else { 3750 mFramestoDrop += buffer.frameCount; 3751 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3752 mSyncStartEvent->isCancelled()) { 3753 ALOGW("Synced record %s, session %d, trigger session %d", 3754 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3755 mActiveTrack->sessionId(), 3756 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3757 clearSyncStartEvent(); 3758 } 3759 } 3760 } 3761 mActiveTrack->clearOverflow(); 3762 } 3763 // client isn't retrieving buffers fast enough 3764 else { 3765 if (!mActiveTrack->setOverflow()) { 3766 nsecs_t now = systemTime(); 3767 if ((now - lastWarning) > kWarningThrottleNs) { 3768 ALOGW("RecordThread: buffer overflow"); 3769 lastWarning = now; 3770 } 3771 } 3772 // Release the processor for a while before asking for a new buffer. 3773 // This will give the application more chance to read from the buffer and 3774 // clear the overflow. 3775 usleep(kRecordThreadSleepUs); 3776 } 3777 } 3778 // enable changes in effect chain 3779 unlockEffectChains(effectChains); 3780 effectChains.clear(); 3781 } 3782 3783 standby(); 3784 3785 { 3786 Mutex::Autolock _l(mLock); 3787 mActiveTrack.clear(); 3788 mStartStopCond.broadcast(); 3789 } 3790 3791 releaseWakeLock(); 3792 3793 ALOGV("RecordThread %p exiting", this); 3794 return false; 3795} 3796 3797void AudioFlinger::RecordThread::standby() 3798{ 3799 if (!mStandby) { 3800 inputStandBy(); 3801 mStandby = true; 3802 } 3803} 3804 3805void AudioFlinger::RecordThread::inputStandBy() 3806{ 3807 mInput->stream->common.standby(&mInput->stream->common); 3808} 3809 3810sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3811 const sp<AudioFlinger::Client>& client, 3812 uint32_t sampleRate, 3813 audio_format_t format, 3814 audio_channel_mask_t channelMask, 3815 size_t frameCount, 3816 int sessionId, 3817 IAudioFlinger::track_flags_t flags, 3818 pid_t tid, 3819 status_t *status) 3820{ 3821 sp<RecordTrack> track; 3822 status_t lStatus; 3823 3824 lStatus = initCheck(); 3825 if (lStatus != NO_ERROR) { 3826 ALOGE("Audio driver not initialized."); 3827 goto Exit; 3828 } 3829 3830 // FIXME use flags and tid similar to createTrack_l() 3831 3832 { // scope for mLock 3833 Mutex::Autolock _l(mLock); 3834 3835 track = new RecordTrack(this, client, sampleRate, 3836 format, channelMask, frameCount, sessionId); 3837 3838 if (track->getCblk() == 0) { 3839 lStatus = NO_MEMORY; 3840 goto Exit; 3841 } 3842 mTracks.add(track); 3843 3844 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3845 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3846 mAudioFlinger->btNrecIsOff(); 3847 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3848 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3849 } 3850 lStatus = NO_ERROR; 3851 3852Exit: 3853 if (status) { 3854 *status = lStatus; 3855 } 3856 return track; 3857} 3858 3859status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3860 AudioSystem::sync_event_t event, 3861 int triggerSession) 3862{ 3863 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3864 sp<ThreadBase> strongMe = this; 3865 status_t status = NO_ERROR; 3866 3867 if (event == AudioSystem::SYNC_EVENT_NONE) { 3868 clearSyncStartEvent(); 3869 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3870 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3871 triggerSession, 3872 recordTrack->sessionId(), 3873 syncStartEventCallback, 3874 this); 3875 // Sync event can be cancelled by the trigger session if the track is not in a 3876 // compatible state in which case we start record immediately 3877 if (mSyncStartEvent->isCancelled()) { 3878 clearSyncStartEvent(); 3879 } else { 3880 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3881 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3882 } 3883 } 3884 3885 { 3886 AutoMutex lock(mLock); 3887 if (mActiveTrack != 0) { 3888 if (recordTrack != mActiveTrack.get()) { 3889 status = -EBUSY; 3890 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3891 mActiveTrack->mState = TrackBase::ACTIVE; 3892 } 3893 return status; 3894 } 3895 3896 recordTrack->mState = TrackBase::IDLE; 3897 mActiveTrack = recordTrack; 3898 mLock.unlock(); 3899 status_t status = AudioSystem::startInput(mId); 3900 mLock.lock(); 3901 if (status != NO_ERROR) { 3902 mActiveTrack.clear(); 3903 clearSyncStartEvent(); 3904 return status; 3905 } 3906 mRsmpInIndex = mFrameCount; 3907 mBytesRead = 0; 3908 if (mResampler != NULL) { 3909 mResampler->reset(); 3910 } 3911 mActiveTrack->mState = TrackBase::RESUMING; 3912 // signal thread to start 3913 ALOGV("Signal record thread"); 3914 mWaitWorkCV.broadcast(); 3915 // do not wait for mStartStopCond if exiting 3916 if (exitPending()) { 3917 mActiveTrack.clear(); 3918 status = INVALID_OPERATION; 3919 goto startError; 3920 } 3921 mStartStopCond.wait(mLock); 3922 if (mActiveTrack == 0) { 3923 ALOGV("Record failed to start"); 3924 status = BAD_VALUE; 3925 goto startError; 3926 } 3927 ALOGV("Record started OK"); 3928 return status; 3929 } 3930startError: 3931 AudioSystem::stopInput(mId); 3932 clearSyncStartEvent(); 3933 return status; 3934} 3935 3936void AudioFlinger::RecordThread::clearSyncStartEvent() 3937{ 3938 if (mSyncStartEvent != 0) { 3939 mSyncStartEvent->cancel(); 3940 } 3941 mSyncStartEvent.clear(); 3942 mFramestoDrop = 0; 3943} 3944 3945void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3946{ 3947 sp<SyncEvent> strongEvent = event.promote(); 3948 3949 if (strongEvent != 0) { 3950 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3951 me->handleSyncStartEvent(strongEvent); 3952 } 3953} 3954 3955void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 3956{ 3957 if (event == mSyncStartEvent) { 3958 // TODO: use actual buffer filling status instead of 2 buffers when info is available 3959 // from audio HAL 3960 mFramestoDrop = mFrameCount * 2; 3961 } 3962} 3963 3964bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 3965 ALOGV("RecordThread::stop"); 3966 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 3967 return false; 3968 } 3969 recordTrack->mState = TrackBase::PAUSING; 3970 // do not wait for mStartStopCond if exiting 3971 if (exitPending()) { 3972 return true; 3973 } 3974 mStartStopCond.wait(mLock); 3975 // if we have been restarted, recordTrack == mActiveTrack.get() here 3976 if (exitPending() || recordTrack != mActiveTrack.get()) { 3977 ALOGV("Record stopped OK"); 3978 return true; 3979 } 3980 return false; 3981} 3982 3983bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 3984{ 3985 return false; 3986} 3987 3988status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 3989{ 3990#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 3991 if (!isValidSyncEvent(event)) { 3992 return BAD_VALUE; 3993 } 3994 3995 int eventSession = event->triggerSession(); 3996 status_t ret = NAME_NOT_FOUND; 3997 3998 Mutex::Autolock _l(mLock); 3999 4000 for (size_t i = 0; i < mTracks.size(); i++) { 4001 sp<RecordTrack> track = mTracks[i]; 4002 if (eventSession == track->sessionId()) { 4003 (void) track->setSyncEvent(event); 4004 ret = NO_ERROR; 4005 } 4006 } 4007 return ret; 4008#else 4009 return BAD_VALUE; 4010#endif 4011} 4012 4013// destroyTrack_l() must be called with ThreadBase::mLock held 4014void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4015{ 4016 track->mState = TrackBase::TERMINATED; 4017 // active tracks are removed by threadLoop() 4018 if (mActiveTrack != track) { 4019 removeTrack_l(track); 4020 } 4021} 4022 4023void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4024{ 4025 mTracks.remove(track); 4026 // need anything related to effects here? 4027} 4028 4029void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4030{ 4031 dumpInternals(fd, args); 4032 dumpTracks(fd, args); 4033 dumpEffectChains(fd, args); 4034} 4035 4036void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4037{ 4038 const size_t SIZE = 256; 4039 char buffer[SIZE]; 4040 String8 result; 4041 4042 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4043 result.append(buffer); 4044 4045 if (mActiveTrack != 0) { 4046 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4047 result.append(buffer); 4048 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4049 result.append(buffer); 4050 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4051 result.append(buffer); 4052 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4053 result.append(buffer); 4054 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4055 result.append(buffer); 4056 } else { 4057 result.append("No active record client\n"); 4058 } 4059 4060 write(fd, result.string(), result.size()); 4061 4062 dumpBase(fd, args); 4063} 4064 4065void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4066{ 4067 const size_t SIZE = 256; 4068 char buffer[SIZE]; 4069 String8 result; 4070 4071 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4072 result.append(buffer); 4073 RecordTrack::appendDumpHeader(result); 4074 for (size_t i = 0; i < mTracks.size(); ++i) { 4075 sp<RecordTrack> track = mTracks[i]; 4076 if (track != 0) { 4077 track->dump(buffer, SIZE); 4078 result.append(buffer); 4079 } 4080 } 4081 4082 if (mActiveTrack != 0) { 4083 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4084 result.append(buffer); 4085 RecordTrack::appendDumpHeader(result); 4086 mActiveTrack->dump(buffer, SIZE); 4087 result.append(buffer); 4088 4089 } 4090 write(fd, result.string(), result.size()); 4091} 4092 4093// AudioBufferProvider interface 4094status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4095{ 4096 size_t framesReq = buffer->frameCount; 4097 size_t framesReady = mFrameCount - mRsmpInIndex; 4098 int channelCount; 4099 4100 if (framesReady == 0) { 4101 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4102 if (mBytesRead <= 0) { 4103 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4104 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4105 // Force input into standby so that it tries to 4106 // recover at next read attempt 4107 inputStandBy(); 4108 usleep(kRecordThreadSleepUs); 4109 } 4110 buffer->raw = NULL; 4111 buffer->frameCount = 0; 4112 return NOT_ENOUGH_DATA; 4113 } 4114 mRsmpInIndex = 0; 4115 framesReady = mFrameCount; 4116 } 4117 4118 if (framesReq > framesReady) { 4119 framesReq = framesReady; 4120 } 4121 4122 if (mChannelCount == 1 && mReqChannelCount == 2) { 4123 channelCount = 1; 4124 } else { 4125 channelCount = 2; 4126 } 4127 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4128 buffer->frameCount = framesReq; 4129 return NO_ERROR; 4130} 4131 4132// AudioBufferProvider interface 4133void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4134{ 4135 mRsmpInIndex += buffer->frameCount; 4136 buffer->frameCount = 0; 4137} 4138 4139bool AudioFlinger::RecordThread::checkForNewParameters_l() 4140{ 4141 bool reconfig = false; 4142 4143 while (!mNewParameters.isEmpty()) { 4144 status_t status = NO_ERROR; 4145 String8 keyValuePair = mNewParameters[0]; 4146 AudioParameter param = AudioParameter(keyValuePair); 4147 int value; 4148 audio_format_t reqFormat = mFormat; 4149 uint32_t reqSamplingRate = mReqSampleRate; 4150 uint32_t reqChannelCount = mReqChannelCount; 4151 4152 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4153 reqSamplingRate = value; 4154 reconfig = true; 4155 } 4156 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4157 reqFormat = (audio_format_t) value; 4158 reconfig = true; 4159 } 4160 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4161 reqChannelCount = popcount(value); 4162 reconfig = true; 4163 } 4164 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4165 // do not accept frame count changes if tracks are open as the track buffer 4166 // size depends on frame count and correct behavior would not be guaranteed 4167 // if frame count is changed after track creation 4168 if (mActiveTrack != 0) { 4169 status = INVALID_OPERATION; 4170 } else { 4171 reconfig = true; 4172 } 4173 } 4174 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4175 // forward device change to effects that have requested to be 4176 // aware of attached audio device. 4177 for (size_t i = 0; i < mEffectChains.size(); i++) { 4178 mEffectChains[i]->setDevice_l(value); 4179 } 4180 4181 // store input device and output device but do not forward output device to audio HAL. 4182 // Note that status is ignored by the caller for output device 4183 // (see AudioFlinger::setParameters() 4184 if (audio_is_output_devices(value)) { 4185 mOutDevice = value; 4186 status = BAD_VALUE; 4187 } else { 4188 mInDevice = value; 4189 // disable AEC and NS if the device is a BT SCO headset supporting those 4190 // pre processings 4191 if (mTracks.size() > 0) { 4192 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4193 mAudioFlinger->btNrecIsOff(); 4194 for (size_t i = 0; i < mTracks.size(); i++) { 4195 sp<RecordTrack> track = mTracks[i]; 4196 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4197 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4198 } 4199 } 4200 } 4201 } 4202 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4203 mAudioSource != (audio_source_t)value) { 4204 // forward device change to effects that have requested to be 4205 // aware of attached audio device. 4206 for (size_t i = 0; i < mEffectChains.size(); i++) { 4207 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4208 } 4209 mAudioSource = (audio_source_t)value; 4210 } 4211 if (status == NO_ERROR) { 4212 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4213 keyValuePair.string()); 4214 if (status == INVALID_OPERATION) { 4215 inputStandBy(); 4216 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4217 keyValuePair.string()); 4218 } 4219 if (reconfig) { 4220 if (status == BAD_VALUE && 4221 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4222 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4223 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 4224 <= (2 * reqSamplingRate)) && 4225 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4226 <= FCC_2 && 4227 (reqChannelCount <= FCC_2)) { 4228 status = NO_ERROR; 4229 } 4230 if (status == NO_ERROR) { 4231 readInputParameters(); 4232 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4233 } 4234 } 4235 } 4236 4237 mNewParameters.removeAt(0); 4238 4239 mParamStatus = status; 4240 mParamCond.signal(); 4241 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4242 // already timed out waiting for the status and will never signal the condition. 4243 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4244 } 4245 return reconfig; 4246} 4247 4248String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4249{ 4250 char *s; 4251 String8 out_s8 = String8(); 4252 4253 Mutex::Autolock _l(mLock); 4254 if (initCheck() != NO_ERROR) { 4255 return out_s8; 4256 } 4257 4258 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4259 out_s8 = String8(s); 4260 free(s); 4261 return out_s8; 4262} 4263 4264void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4265 AudioSystem::OutputDescriptor desc; 4266 void *param2 = NULL; 4267 4268 switch (event) { 4269 case AudioSystem::INPUT_OPENED: 4270 case AudioSystem::INPUT_CONFIG_CHANGED: 4271 desc.channels = mChannelMask; 4272 desc.samplingRate = mSampleRate; 4273 desc.format = mFormat; 4274 desc.frameCount = mFrameCount; 4275 desc.latency = 0; 4276 param2 = &desc; 4277 break; 4278 4279 case AudioSystem::INPUT_CLOSED: 4280 default: 4281 break; 4282 } 4283 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4284} 4285 4286void AudioFlinger::RecordThread::readInputParameters() 4287{ 4288 delete mRsmpInBuffer; 4289 // mRsmpInBuffer is always assigned a new[] below 4290 delete mRsmpOutBuffer; 4291 mRsmpOutBuffer = NULL; 4292 delete mResampler; 4293 mResampler = NULL; 4294 4295 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4296 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4297 mChannelCount = (uint16_t)popcount(mChannelMask); 4298 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4299 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4300 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4301 mFrameCount = mInputBytes / mFrameSize; 4302 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4303 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4304 4305 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4306 { 4307 int channelCount; 4308 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4309 // stereo to mono post process as the resampler always outputs stereo. 4310 if (mChannelCount == 1 && mReqChannelCount == 2) { 4311 channelCount = 1; 4312 } else { 4313 channelCount = 2; 4314 } 4315 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4316 mResampler->setSampleRate(mSampleRate); 4317 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4318 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4319 4320 // optmization: if mono to mono, alter input frame count as if we were inputing 4321 // stereo samples 4322 if (mChannelCount == 1 && mReqChannelCount == 1) { 4323 mFrameCount >>= 1; 4324 } 4325 4326 } 4327 mRsmpInIndex = mFrameCount; 4328} 4329 4330unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4331{ 4332 Mutex::Autolock _l(mLock); 4333 if (initCheck() != NO_ERROR) { 4334 return 0; 4335 } 4336 4337 return mInput->stream->get_input_frames_lost(mInput->stream); 4338} 4339 4340uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4341{ 4342 Mutex::Autolock _l(mLock); 4343 uint32_t result = 0; 4344 if (getEffectChain_l(sessionId) != 0) { 4345 result = EFFECT_SESSION; 4346 } 4347 4348 for (size_t i = 0; i < mTracks.size(); ++i) { 4349 if (sessionId == mTracks[i]->sessionId()) { 4350 result |= TRACK_SESSION; 4351 break; 4352 } 4353 } 4354 4355 return result; 4356} 4357 4358KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4359{ 4360 KeyedVector<int, bool> ids; 4361 Mutex::Autolock _l(mLock); 4362 for (size_t j = 0; j < mTracks.size(); ++j) { 4363 sp<RecordThread::RecordTrack> track = mTracks[j]; 4364 int sessionId = track->sessionId(); 4365 if (ids.indexOfKey(sessionId) < 0) { 4366 ids.add(sessionId, true); 4367 } 4368 } 4369 return ids; 4370} 4371 4372AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4373{ 4374 Mutex::Autolock _l(mLock); 4375 AudioStreamIn *input = mInput; 4376 mInput = NULL; 4377 return input; 4378} 4379 4380// this method must always be called either with ThreadBase mLock held or inside the thread loop 4381audio_stream_t* AudioFlinger::RecordThread::stream() const 4382{ 4383 if (mInput == NULL) { 4384 return NULL; 4385 } 4386 return &mInput->stream->common; 4387} 4388 4389status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4390{ 4391 // only one chain per input thread 4392 if (mEffectChains.size() != 0) { 4393 return INVALID_OPERATION; 4394 } 4395 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4396 4397 chain->setInBuffer(NULL); 4398 chain->setOutBuffer(NULL); 4399 4400 checkSuspendOnAddEffectChain_l(chain); 4401 4402 mEffectChains.add(chain); 4403 4404 return NO_ERROR; 4405} 4406 4407size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4408{ 4409 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4410 ALOGW_IF(mEffectChains.size() != 1, 4411 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4412 chain.get(), mEffectChains.size(), this); 4413 if (mEffectChains.size() == 1) { 4414 mEffectChains.removeAt(0); 4415 } 4416 return 0; 4417} 4418 4419}; // namespace android 4420