Threads.cpp revision 04b035e3ccbf2919e4447c66e6483c11f2889f01
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 // FIXME Need to understand why this has be done asynchronously 377 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 378 true /*asynchronous*/); 379 if (err != 0) { 380 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 381 "error %d", 382 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 383 } 384 } break; 385 case CFG_EVENT_IO: { 386 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 387 mAudioFlinger->mLock.lock(); 388 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 389 mAudioFlinger->mLock.unlock(); 390 } break; 391 default: 392 ALOGE("processConfigEvents() unknown event type %d", event->type()); 393 break; 394 } 395 delete event; 396 mLock.lock(); 397 } 398 mLock.unlock(); 399} 400 401void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 402{ 403 const size_t SIZE = 256; 404 char buffer[SIZE]; 405 String8 result; 406 407 bool locked = AudioFlinger::dumpTryLock(mLock); 408 if (!locked) { 409 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 410 write(fd, buffer, strlen(buffer)); 411 } 412 413 snprintf(buffer, SIZE, "io handle: %d\n", mId); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 432 result.append(buffer); 433 434 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 435 result.append(buffer); 436 result.append(" Index Command"); 437 for (size_t i = 0; i < mNewParameters.size(); ++i) { 438 snprintf(buffer, SIZE, "\n %02d ", i); 439 result.append(buffer); 440 result.append(mNewParameters[i]); 441 } 442 443 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 444 result.append(buffer); 445 for (size_t i = 0; i < mConfigEvents.size(); i++) { 446 mConfigEvents[i]->dump(buffer, SIZE); 447 result.append(buffer); 448 } 449 result.append("\n"); 450 451 write(fd, result.string(), result.size()); 452 453 if (locked) { 454 mLock.unlock(); 455 } 456} 457 458void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 459{ 460 const size_t SIZE = 256; 461 char buffer[SIZE]; 462 String8 result; 463 464 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 465 write(fd, buffer, strlen(buffer)); 466 467 for (size_t i = 0; i < mEffectChains.size(); ++i) { 468 sp<EffectChain> chain = mEffectChains[i]; 469 if (chain != 0) { 470 chain->dump(fd, args); 471 } 472 } 473} 474 475void AudioFlinger::ThreadBase::acquireWakeLock() 476{ 477 Mutex::Autolock _l(mLock); 478 acquireWakeLock_l(); 479} 480 481void AudioFlinger::ThreadBase::acquireWakeLock_l() 482{ 483 if (mPowerManager == 0) { 484 // use checkService() to avoid blocking if power service is not up yet 485 sp<IBinder> binder = 486 defaultServiceManager()->checkService(String16("power")); 487 if (binder == 0) { 488 ALOGW("Thread %s cannot connect to the power manager service", mName); 489 } else { 490 mPowerManager = interface_cast<IPowerManager>(binder); 491 binder->linkToDeath(mDeathRecipient); 492 } 493 } 494 if (mPowerManager != 0) { 495 sp<IBinder> binder = new BBinder(); 496 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 497 binder, 498 String16(mName), 499 String16("media")); 500 if (status == NO_ERROR) { 501 mWakeLockToken = binder; 502 } 503 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 504 } 505} 506 507void AudioFlinger::ThreadBase::releaseWakeLock() 508{ 509 Mutex::Autolock _l(mLock); 510 releaseWakeLock_l(); 511} 512 513void AudioFlinger::ThreadBase::releaseWakeLock_l() 514{ 515 if (mWakeLockToken != 0) { 516 ALOGV("releaseWakeLock_l() %s", mName); 517 if (mPowerManager != 0) { 518 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 519 } 520 mWakeLockToken.clear(); 521 } 522} 523 524void AudioFlinger::ThreadBase::clearPowerManager() 525{ 526 Mutex::Autolock _l(mLock); 527 releaseWakeLock_l(); 528 mPowerManager.clear(); 529} 530 531void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 532{ 533 sp<ThreadBase> thread = mThread.promote(); 534 if (thread != 0) { 535 thread->clearPowerManager(); 536 } 537 ALOGW("power manager service died !!!"); 538} 539 540void AudioFlinger::ThreadBase::setEffectSuspended( 541 const effect_uuid_t *type, bool suspend, int sessionId) 542{ 543 Mutex::Autolock _l(mLock); 544 setEffectSuspended_l(type, suspend, sessionId); 545} 546 547void AudioFlinger::ThreadBase::setEffectSuspended_l( 548 const effect_uuid_t *type, bool suspend, int sessionId) 549{ 550 sp<EffectChain> chain = getEffectChain_l(sessionId); 551 if (chain != 0) { 552 if (type != NULL) { 553 chain->setEffectSuspended_l(type, suspend); 554 } else { 555 chain->setEffectSuspendedAll_l(suspend); 556 } 557 } 558 559 updateSuspendedSessions_l(type, suspend, sessionId); 560} 561 562void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 563{ 564 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 565 if (index < 0) { 566 return; 567 } 568 569 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 570 mSuspendedSessions.valueAt(index); 571 572 for (size_t i = 0; i < sessionEffects.size(); i++) { 573 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 574 for (int j = 0; j < desc->mRefCount; j++) { 575 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 576 chain->setEffectSuspendedAll_l(true); 577 } else { 578 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 579 desc->mType.timeLow); 580 chain->setEffectSuspended_l(&desc->mType, true); 581 } 582 } 583 } 584} 585 586void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 587 bool suspend, 588 int sessionId) 589{ 590 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 591 592 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 593 594 if (suspend) { 595 if (index >= 0) { 596 sessionEffects = mSuspendedSessions.valueAt(index); 597 } else { 598 mSuspendedSessions.add(sessionId, sessionEffects); 599 } 600 } else { 601 if (index < 0) { 602 return; 603 } 604 sessionEffects = mSuspendedSessions.valueAt(index); 605 } 606 607 608 int key = EffectChain::kKeyForSuspendAll; 609 if (type != NULL) { 610 key = type->timeLow; 611 } 612 index = sessionEffects.indexOfKey(key); 613 614 sp<SuspendedSessionDesc> desc; 615 if (suspend) { 616 if (index >= 0) { 617 desc = sessionEffects.valueAt(index); 618 } else { 619 desc = new SuspendedSessionDesc(); 620 if (type != NULL) { 621 desc->mType = *type; 622 } 623 sessionEffects.add(key, desc); 624 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 625 } 626 desc->mRefCount++; 627 } else { 628 if (index < 0) { 629 return; 630 } 631 desc = sessionEffects.valueAt(index); 632 if (--desc->mRefCount == 0) { 633 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 634 sessionEffects.removeItemsAt(index); 635 if (sessionEffects.isEmpty()) { 636 ALOGV("updateSuspendedSessions_l() restore removing session %d", 637 sessionId); 638 mSuspendedSessions.removeItem(sessionId); 639 } 640 } 641 } 642 if (!sessionEffects.isEmpty()) { 643 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 644 } 645} 646 647void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 648 bool enabled, 649 int sessionId) 650{ 651 Mutex::Autolock _l(mLock); 652 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 653} 654 655void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 656 bool enabled, 657 int sessionId) 658{ 659 if (mType != RECORD) { 660 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 661 // another session. This gives the priority to well behaved effect control panels 662 // and applications not using global effects. 663 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 664 // global effects 665 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 666 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 667 } 668 } 669 670 sp<EffectChain> chain = getEffectChain_l(sessionId); 671 if (chain != 0) { 672 chain->checkSuspendOnEffectEnabled(effect, enabled); 673 } 674} 675 676// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 677sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 678 const sp<AudioFlinger::Client>& client, 679 const sp<IEffectClient>& effectClient, 680 int32_t priority, 681 int sessionId, 682 effect_descriptor_t *desc, 683 int *enabled, 684 status_t *status 685 ) 686{ 687 sp<EffectModule> effect; 688 sp<EffectHandle> handle; 689 status_t lStatus; 690 sp<EffectChain> chain; 691 bool chainCreated = false; 692 bool effectCreated = false; 693 bool effectRegistered = false; 694 695 lStatus = initCheck(); 696 if (lStatus != NO_ERROR) { 697 ALOGW("createEffect_l() Audio driver not initialized."); 698 goto Exit; 699 } 700 701 // Do not allow effects with session ID 0 on direct output or duplicating threads 702 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 703 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 704 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 705 desc->name, sessionId); 706 lStatus = BAD_VALUE; 707 goto Exit; 708 } 709 // Only Pre processor effects are allowed on input threads and only on input threads 710 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 711 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 712 desc->name, desc->flags, mType); 713 lStatus = BAD_VALUE; 714 goto Exit; 715 } 716 717 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 718 719 { // scope for mLock 720 Mutex::Autolock _l(mLock); 721 722 // check for existing effect chain with the requested audio session 723 chain = getEffectChain_l(sessionId); 724 if (chain == 0) { 725 // create a new chain for this session 726 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 727 chain = new EffectChain(this, sessionId); 728 addEffectChain_l(chain); 729 chain->setStrategy(getStrategyForSession_l(sessionId)); 730 chainCreated = true; 731 } else { 732 effect = chain->getEffectFromDesc_l(desc); 733 } 734 735 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 736 737 if (effect == 0) { 738 int id = mAudioFlinger->nextUniqueId(); 739 // Check CPU and memory usage 740 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 741 if (lStatus != NO_ERROR) { 742 goto Exit; 743 } 744 effectRegistered = true; 745 // create a new effect module if none present in the chain 746 effect = new EffectModule(this, chain, desc, id, sessionId); 747 lStatus = effect->status(); 748 if (lStatus != NO_ERROR) { 749 goto Exit; 750 } 751 lStatus = chain->addEffect_l(effect); 752 if (lStatus != NO_ERROR) { 753 goto Exit; 754 } 755 effectCreated = true; 756 757 effect->setDevice(mOutDevice); 758 effect->setDevice(mInDevice); 759 effect->setMode(mAudioFlinger->getMode()); 760 effect->setAudioSource(mAudioSource); 761 } 762 // create effect handle and connect it to effect module 763 handle = new EffectHandle(effect, client, effectClient, priority); 764 lStatus = effect->addHandle(handle.get()); 765 if (enabled != NULL) { 766 *enabled = (int)effect->isEnabled(); 767 } 768 } 769 770Exit: 771 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 772 Mutex::Autolock _l(mLock); 773 if (effectCreated) { 774 chain->removeEffect_l(effect); 775 } 776 if (effectRegistered) { 777 AudioSystem::unregisterEffect(effect->id()); 778 } 779 if (chainCreated) { 780 removeEffectChain_l(chain); 781 } 782 handle.clear(); 783 } 784 785 if (status != NULL) { 786 *status = lStatus; 787 } 788 return handle; 789} 790 791sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 792{ 793 Mutex::Autolock _l(mLock); 794 return getEffect_l(sessionId, effectId); 795} 796 797sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 798{ 799 sp<EffectChain> chain = getEffectChain_l(sessionId); 800 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 801} 802 803// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 804// PlaybackThread::mLock held 805status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 806{ 807 // check for existing effect chain with the requested audio session 808 int sessionId = effect->sessionId(); 809 sp<EffectChain> chain = getEffectChain_l(sessionId); 810 bool chainCreated = false; 811 812 if (chain == 0) { 813 // create a new chain for this session 814 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 815 chain = new EffectChain(this, sessionId); 816 addEffectChain_l(chain); 817 chain->setStrategy(getStrategyForSession_l(sessionId)); 818 chainCreated = true; 819 } 820 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 821 822 if (chain->getEffectFromId_l(effect->id()) != 0) { 823 ALOGW("addEffect_l() %p effect %s already present in chain %p", 824 this, effect->desc().name, chain.get()); 825 return BAD_VALUE; 826 } 827 828 status_t status = chain->addEffect_l(effect); 829 if (status != NO_ERROR) { 830 if (chainCreated) { 831 removeEffectChain_l(chain); 832 } 833 return status; 834 } 835 836 effect->setDevice(mOutDevice); 837 effect->setDevice(mInDevice); 838 effect->setMode(mAudioFlinger->getMode()); 839 effect->setAudioSource(mAudioSource); 840 return NO_ERROR; 841} 842 843void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 844 845 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 846 effect_descriptor_t desc = effect->desc(); 847 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 848 detachAuxEffect_l(effect->id()); 849 } 850 851 sp<EffectChain> chain = effect->chain().promote(); 852 if (chain != 0) { 853 // remove effect chain if removing last effect 854 if (chain->removeEffect_l(effect) == 0) { 855 removeEffectChain_l(chain); 856 } 857 } else { 858 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 859 } 860} 861 862void AudioFlinger::ThreadBase::lockEffectChains_l( 863 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 864{ 865 effectChains = mEffectChains; 866 for (size_t i = 0; i < mEffectChains.size(); i++) { 867 mEffectChains[i]->lock(); 868 } 869} 870 871void AudioFlinger::ThreadBase::unlockEffectChains( 872 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 873{ 874 for (size_t i = 0; i < effectChains.size(); i++) { 875 effectChains[i]->unlock(); 876 } 877} 878 879sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 880{ 881 Mutex::Autolock _l(mLock); 882 return getEffectChain_l(sessionId); 883} 884 885sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 886{ 887 size_t size = mEffectChains.size(); 888 for (size_t i = 0; i < size; i++) { 889 if (mEffectChains[i]->sessionId() == sessionId) { 890 return mEffectChains[i]; 891 } 892 } 893 return 0; 894} 895 896void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 897{ 898 Mutex::Autolock _l(mLock); 899 size_t size = mEffectChains.size(); 900 for (size_t i = 0; i < size; i++) { 901 mEffectChains[i]->setMode_l(mode); 902 } 903} 904 905void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 906 EffectHandle *handle, 907 bool unpinIfLast) { 908 909 Mutex::Autolock _l(mLock); 910 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 911 // delete the effect module if removing last handle on it 912 if (effect->removeHandle(handle) == 0) { 913 if (!effect->isPinned() || unpinIfLast) { 914 removeEffect_l(effect); 915 AudioSystem::unregisterEffect(effect->id()); 916 } 917 } 918} 919 920// ---------------------------------------------------------------------------- 921// Playback 922// ---------------------------------------------------------------------------- 923 924AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 925 AudioStreamOut* output, 926 audio_io_handle_t id, 927 audio_devices_t device, 928 type_t type) 929 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 930 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 931 // mStreamTypes[] initialized in constructor body 932 mOutput(output), 933 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 934 mMixerStatus(MIXER_IDLE), 935 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 936 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 937 mScreenState(AudioFlinger::mScreenState), 938 // index 0 is reserved for normal mixer's submix 939 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 940{ 941 snprintf(mName, kNameLength, "AudioOut_%X", id); 942 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 943 944 // Assumes constructor is called by AudioFlinger with it's mLock held, but 945 // it would be safer to explicitly pass initial masterVolume/masterMute as 946 // parameter. 947 // 948 // If the HAL we are using has support for master volume or master mute, 949 // then do not attenuate or mute during mixing (just leave the volume at 1.0 950 // and the mute set to false). 951 mMasterVolume = audioFlinger->masterVolume_l(); 952 mMasterMute = audioFlinger->masterMute_l(); 953 if (mOutput && mOutput->audioHwDev) { 954 if (mOutput->audioHwDev->canSetMasterVolume()) { 955 mMasterVolume = 1.0; 956 } 957 958 if (mOutput->audioHwDev->canSetMasterMute()) { 959 mMasterMute = false; 960 } 961 } 962 963 readOutputParameters(); 964 965 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 966 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 967 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 968 stream = (audio_stream_type_t) (stream + 1)) { 969 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 970 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 971 } 972 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 973 // because mAudioFlinger doesn't have one to copy from 974} 975 976AudioFlinger::PlaybackThread::~PlaybackThread() 977{ 978 mAudioFlinger->unregisterWriter(mNBLogWriter); 979 delete [] mMixBuffer; 980} 981 982void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 983{ 984 dumpInternals(fd, args); 985 dumpTracks(fd, args); 986 dumpEffectChains(fd, args); 987} 988 989void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 990{ 991 const size_t SIZE = 256; 992 char buffer[SIZE]; 993 String8 result; 994 995 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 996 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 997 const stream_type_t *st = &mStreamTypes[i]; 998 if (i > 0) { 999 result.appendFormat(", "); 1000 } 1001 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1002 if (st->mute) { 1003 result.append("M"); 1004 } 1005 } 1006 result.append("\n"); 1007 write(fd, result.string(), result.length()); 1008 result.clear(); 1009 1010 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1011 result.append(buffer); 1012 Track::appendDumpHeader(result); 1013 for (size_t i = 0; i < mTracks.size(); ++i) { 1014 sp<Track> track = mTracks[i]; 1015 if (track != 0) { 1016 track->dump(buffer, SIZE); 1017 result.append(buffer); 1018 } 1019 } 1020 1021 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1022 result.append(buffer); 1023 Track::appendDumpHeader(result); 1024 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1025 sp<Track> track = mActiveTracks[i].promote(); 1026 if (track != 0) { 1027 track->dump(buffer, SIZE); 1028 result.append(buffer); 1029 } 1030 } 1031 write(fd, result.string(), result.size()); 1032 1033 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1034 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1035 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1036 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1037} 1038 1039void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1040{ 1041 const size_t SIZE = 256; 1042 char buffer[SIZE]; 1043 String8 result; 1044 1045 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1048 ns2ms(systemTime() - mLastWriteTime)); 1049 result.append(buffer); 1050 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1051 result.append(buffer); 1052 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1053 result.append(buffer); 1054 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1055 result.append(buffer); 1056 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1057 result.append(buffer); 1058 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1059 result.append(buffer); 1060 write(fd, result.string(), result.size()); 1061 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1062 1063 dumpBase(fd, args); 1064} 1065 1066// Thread virtuals 1067status_t AudioFlinger::PlaybackThread::readyToRun() 1068{ 1069 status_t status = initCheck(); 1070 if (status == NO_ERROR) { 1071 ALOGI("AudioFlinger's thread %p ready to run", this); 1072 } else { 1073 ALOGE("No working audio driver found."); 1074 } 1075 return status; 1076} 1077 1078void AudioFlinger::PlaybackThread::onFirstRef() 1079{ 1080 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1081} 1082 1083// ThreadBase virtuals 1084void AudioFlinger::PlaybackThread::preExit() 1085{ 1086 ALOGV(" preExit()"); 1087 // FIXME this is using hard-coded strings but in the future, this functionality will be 1088 // converted to use audio HAL extensions required to support tunneling 1089 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1090} 1091 1092// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1093sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1094 const sp<AudioFlinger::Client>& client, 1095 audio_stream_type_t streamType, 1096 uint32_t sampleRate, 1097 audio_format_t format, 1098 audio_channel_mask_t channelMask, 1099 size_t frameCount, 1100 const sp<IMemory>& sharedBuffer, 1101 int sessionId, 1102 IAudioFlinger::track_flags_t *flags, 1103 pid_t tid, 1104 status_t *status) 1105{ 1106 sp<Track> track; 1107 status_t lStatus; 1108 1109 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1110 1111 // client expresses a preference for FAST, but we get the final say 1112 if (*flags & IAudioFlinger::TRACK_FAST) { 1113 if ( 1114 // not timed 1115 (!isTimed) && 1116 // either of these use cases: 1117 ( 1118 // use case 1: shared buffer with any frame count 1119 ( 1120 (sharedBuffer != 0) 1121 ) || 1122 // use case 2: callback handler and frame count is default or at least as large as HAL 1123 ( 1124 (tid != -1) && 1125 ((frameCount == 0) || 1126 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1127 ) 1128 ) && 1129 // PCM data 1130 audio_is_linear_pcm(format) && 1131 // mono or stereo 1132 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1133 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1134#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1135 // hardware sample rate 1136 (sampleRate == mSampleRate) && 1137#endif 1138 // normal mixer has an associated fast mixer 1139 hasFastMixer() && 1140 // there are sufficient fast track slots available 1141 (mFastTrackAvailMask != 0) 1142 // FIXME test that MixerThread for this fast track has a capable output HAL 1143 // FIXME add a permission test also? 1144 ) { 1145 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1146 if (frameCount == 0) { 1147 frameCount = mFrameCount * kFastTrackMultiplier; 1148 } 1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1150 frameCount, mFrameCount); 1151 } else { 1152 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1153 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1154 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1155 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1156 audio_is_linear_pcm(format), 1157 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1158 *flags &= ~IAudioFlinger::TRACK_FAST; 1159 // For compatibility with AudioTrack calculation, buffer depth is forced 1160 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1161 // This is probably too conservative, but legacy application code may depend on it. 1162 // If you change this calculation, also review the start threshold which is related. 1163 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1164 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1165 if (minBufCount < 2) { 1166 minBufCount = 2; 1167 } 1168 size_t minFrameCount = mNormalFrameCount * minBufCount; 1169 if (frameCount < minFrameCount) { 1170 frameCount = minFrameCount; 1171 } 1172 } 1173 } 1174 1175 if (mType == DIRECT) { 1176 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1177 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1178 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1179 "for output %p with format %d", 1180 sampleRate, format, channelMask, mOutput, mFormat); 1181 lStatus = BAD_VALUE; 1182 goto Exit; 1183 } 1184 } 1185 } else { 1186 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1187 if (sampleRate > mSampleRate*2) { 1188 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1189 lStatus = BAD_VALUE; 1190 goto Exit; 1191 } 1192 } 1193 1194 lStatus = initCheck(); 1195 if (lStatus != NO_ERROR) { 1196 ALOGE("Audio driver not initialized."); 1197 goto Exit; 1198 } 1199 1200 { // scope for mLock 1201 Mutex::Autolock _l(mLock); 1202 1203 // all tracks in same audio session must share the same routing strategy otherwise 1204 // conflicts will happen when tracks are moved from one output to another by audio policy 1205 // manager 1206 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1207 for (size_t i = 0; i < mTracks.size(); ++i) { 1208 sp<Track> t = mTracks[i]; 1209 if (t != 0 && !t->isOutputTrack()) { 1210 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1211 if (sessionId == t->sessionId() && strategy != actual) { 1212 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1213 strategy, actual); 1214 lStatus = BAD_VALUE; 1215 goto Exit; 1216 } 1217 } 1218 } 1219 1220 if (!isTimed) { 1221 track = new Track(this, client, streamType, sampleRate, format, 1222 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1223 } else { 1224 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1225 channelMask, frameCount, sharedBuffer, sessionId); 1226 } 1227 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1228 lStatus = NO_MEMORY; 1229 goto Exit; 1230 } 1231 mTracks.add(track); 1232 1233 sp<EffectChain> chain = getEffectChain_l(sessionId); 1234 if (chain != 0) { 1235 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1236 track->setMainBuffer(chain->inBuffer()); 1237 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1238 chain->incTrackCnt(); 1239 } 1240 1241 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1242 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1243 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1244 // so ask activity manager to do this on our behalf 1245 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1246 } 1247 } 1248 1249 lStatus = NO_ERROR; 1250 1251Exit: 1252 if (status) { 1253 *status = lStatus; 1254 } 1255 return track; 1256} 1257 1258uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1259{ 1260 return latency; 1261} 1262 1263uint32_t AudioFlinger::PlaybackThread::latency() const 1264{ 1265 Mutex::Autolock _l(mLock); 1266 return latency_l(); 1267} 1268uint32_t AudioFlinger::PlaybackThread::latency_l() const 1269{ 1270 if (initCheck() == NO_ERROR) { 1271 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1272 } else { 1273 return 0; 1274 } 1275} 1276 1277void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1278{ 1279 Mutex::Autolock _l(mLock); 1280 // Don't apply master volume in SW if our HAL can do it for us. 1281 if (mOutput && mOutput->audioHwDev && 1282 mOutput->audioHwDev->canSetMasterVolume()) { 1283 mMasterVolume = 1.0; 1284 } else { 1285 mMasterVolume = value; 1286 } 1287} 1288 1289void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1290{ 1291 Mutex::Autolock _l(mLock); 1292 // Don't apply master mute in SW if our HAL can do it for us. 1293 if (mOutput && mOutput->audioHwDev && 1294 mOutput->audioHwDev->canSetMasterMute()) { 1295 mMasterMute = false; 1296 } else { 1297 mMasterMute = muted; 1298 } 1299} 1300 1301void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1302{ 1303 Mutex::Autolock _l(mLock); 1304 mStreamTypes[stream].volume = value; 1305} 1306 1307void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1308{ 1309 Mutex::Autolock _l(mLock); 1310 mStreamTypes[stream].mute = muted; 1311} 1312 1313float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1314{ 1315 Mutex::Autolock _l(mLock); 1316 return mStreamTypes[stream].volume; 1317} 1318 1319// addTrack_l() must be called with ThreadBase::mLock held 1320status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1321{ 1322 status_t status = ALREADY_EXISTS; 1323 1324 // set retry count for buffer fill 1325 track->mRetryCount = kMaxTrackStartupRetries; 1326 if (mActiveTracks.indexOf(track) < 0) { 1327 // the track is newly added, make sure it fills up all its 1328 // buffers before playing. This is to ensure the client will 1329 // effectively get the latency it requested. 1330 track->mFillingUpStatus = Track::FS_FILLING; 1331 track->mResetDone = false; 1332 track->mPresentationCompleteFrames = 0; 1333 mActiveTracks.add(track); 1334 if (track->mainBuffer() != mMixBuffer) { 1335 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1336 if (chain != 0) { 1337 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1338 track->sessionId()); 1339 chain->incActiveTrackCnt(); 1340 } 1341 } 1342 1343 status = NO_ERROR; 1344 } 1345 1346 ALOGV("mWaitWorkCV.broadcast"); 1347 mWaitWorkCV.broadcast(); 1348 1349 return status; 1350} 1351 1352// destroyTrack_l() must be called with ThreadBase::mLock held 1353void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1354{ 1355 track->mState = TrackBase::TERMINATED; 1356 // active tracks are removed by threadLoop() 1357 if (mActiveTracks.indexOf(track) < 0) { 1358 removeTrack_l(track); 1359 } 1360} 1361 1362void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1363{ 1364 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1365 mTracks.remove(track); 1366 deleteTrackName_l(track->name()); 1367 // redundant as track is about to be destroyed, for dumpsys only 1368 track->mName = -1; 1369 if (track->isFastTrack()) { 1370 int index = track->mFastIndex; 1371 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1372 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1373 mFastTrackAvailMask |= 1 << index; 1374 // redundant as track is about to be destroyed, for dumpsys only 1375 track->mFastIndex = -1; 1376 } 1377 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1378 if (chain != 0) { 1379 chain->decTrackCnt(); 1380 } 1381} 1382 1383String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1384{ 1385 String8 out_s8 = String8(""); 1386 char *s; 1387 1388 Mutex::Autolock _l(mLock); 1389 if (initCheck() != NO_ERROR) { 1390 return out_s8; 1391 } 1392 1393 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1394 out_s8 = String8(s); 1395 free(s); 1396 return out_s8; 1397} 1398 1399// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1400void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1401 AudioSystem::OutputDescriptor desc; 1402 void *param2 = NULL; 1403 1404 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1405 param); 1406 1407 switch (event) { 1408 case AudioSystem::OUTPUT_OPENED: 1409 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1410 desc.channels = mChannelMask; 1411 desc.samplingRate = mSampleRate; 1412 desc.format = mFormat; 1413 desc.frameCount = mNormalFrameCount; // FIXME see 1414 // AudioFlinger::frameCount(audio_io_handle_t) 1415 desc.latency = latency(); 1416 param2 = &desc; 1417 break; 1418 1419 case AudioSystem::STREAM_CONFIG_CHANGED: 1420 param2 = ¶m; 1421 case AudioSystem::OUTPUT_CLOSED: 1422 default: 1423 break; 1424 } 1425 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1426} 1427 1428void AudioFlinger::PlaybackThread::readOutputParameters() 1429{ 1430 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1431 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1432 mChannelCount = (uint16_t)popcount(mChannelMask); 1433 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1434 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1435 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1436 if (mFrameCount & 15) { 1437 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1438 mFrameCount); 1439 } 1440 1441 // Calculate size of normal mix buffer relative to the HAL output buffer size 1442 double multiplier = 1.0; 1443 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1444 kUseFastMixer == FastMixer_Dynamic)) { 1445 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1446 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1447 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1448 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1449 maxNormalFrameCount = maxNormalFrameCount & ~15; 1450 if (maxNormalFrameCount < minNormalFrameCount) { 1451 maxNormalFrameCount = minNormalFrameCount; 1452 } 1453 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1454 if (multiplier <= 1.0) { 1455 multiplier = 1.0; 1456 } else if (multiplier <= 2.0) { 1457 if (2 * mFrameCount <= maxNormalFrameCount) { 1458 multiplier = 2.0; 1459 } else { 1460 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1461 } 1462 } else { 1463 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1464 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1465 // track, but we sometimes have to do this to satisfy the maximum frame count 1466 // constraint) 1467 // FIXME this rounding up should not be done if no HAL SRC 1468 uint32_t truncMult = (uint32_t) multiplier; 1469 if ((truncMult & 1)) { 1470 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1471 ++truncMult; 1472 } 1473 } 1474 multiplier = (double) truncMult; 1475 } 1476 } 1477 mNormalFrameCount = multiplier * mFrameCount; 1478 // round up to nearest 16 frames to satisfy AudioMixer 1479 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1480 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1481 mNormalFrameCount); 1482 1483 delete[] mMixBuffer; 1484 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1485 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1486 1487 // force reconfiguration of effect chains and engines to take new buffer size and audio 1488 // parameters into account 1489 // Note that mLock is not held when readOutputParameters() is called from the constructor 1490 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1491 // matter. 1492 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1493 Vector< sp<EffectChain> > effectChains = mEffectChains; 1494 for (size_t i = 0; i < effectChains.size(); i ++) { 1495 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1496 } 1497} 1498 1499 1500status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1501{ 1502 if (halFrames == NULL || dspFrames == NULL) { 1503 return BAD_VALUE; 1504 } 1505 Mutex::Autolock _l(mLock); 1506 if (initCheck() != NO_ERROR) { 1507 return INVALID_OPERATION; 1508 } 1509 size_t framesWritten = mBytesWritten / mFrameSize; 1510 *halFrames = framesWritten; 1511 1512 if (isSuspended()) { 1513 // return an estimation of rendered frames when the output is suspended 1514 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1515 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1516 return NO_ERROR; 1517 } else { 1518 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1519 } 1520} 1521 1522uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1523{ 1524 Mutex::Autolock _l(mLock); 1525 uint32_t result = 0; 1526 if (getEffectChain_l(sessionId) != 0) { 1527 result = EFFECT_SESSION; 1528 } 1529 1530 for (size_t i = 0; i < mTracks.size(); ++i) { 1531 sp<Track> track = mTracks[i]; 1532 if (sessionId == track->sessionId() && !track->isInvalid()) { 1533 result |= TRACK_SESSION; 1534 break; 1535 } 1536 } 1537 1538 return result; 1539} 1540 1541uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1542{ 1543 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1544 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1545 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1546 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1547 } 1548 for (size_t i = 0; i < mTracks.size(); i++) { 1549 sp<Track> track = mTracks[i]; 1550 if (sessionId == track->sessionId() && !track->isInvalid()) { 1551 return AudioSystem::getStrategyForStream(track->streamType()); 1552 } 1553 } 1554 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1555} 1556 1557 1558AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1559{ 1560 Mutex::Autolock _l(mLock); 1561 return mOutput; 1562} 1563 1564AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1565{ 1566 Mutex::Autolock _l(mLock); 1567 AudioStreamOut *output = mOutput; 1568 mOutput = NULL; 1569 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1570 // must push a NULL and wait for ack 1571 mOutputSink.clear(); 1572 mPipeSink.clear(); 1573 mNormalSink.clear(); 1574 return output; 1575} 1576 1577// this method must always be called either with ThreadBase mLock held or inside the thread loop 1578audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1579{ 1580 if (mOutput == NULL) { 1581 return NULL; 1582 } 1583 return &mOutput->stream->common; 1584} 1585 1586uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1587{ 1588 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1589} 1590 1591status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1592{ 1593 if (!isValidSyncEvent(event)) { 1594 return BAD_VALUE; 1595 } 1596 1597 Mutex::Autolock _l(mLock); 1598 1599 for (size_t i = 0; i < mTracks.size(); ++i) { 1600 sp<Track> track = mTracks[i]; 1601 if (event->triggerSession() == track->sessionId()) { 1602 (void) track->setSyncEvent(event); 1603 return NO_ERROR; 1604 } 1605 } 1606 1607 return NAME_NOT_FOUND; 1608} 1609 1610bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1611{ 1612 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1613} 1614 1615void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1616 const Vector< sp<Track> >& tracksToRemove) 1617{ 1618 size_t count = tracksToRemove.size(); 1619 if (CC_UNLIKELY(count)) { 1620 for (size_t i = 0 ; i < count ; i++) { 1621 const sp<Track>& track = tracksToRemove.itemAt(i); 1622 if ((track->sharedBuffer() != 0) && 1623 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1624 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1625 } 1626 } 1627 } 1628 1629} 1630 1631void AudioFlinger::PlaybackThread::checkSilentMode_l() 1632{ 1633 if (!mMasterMute) { 1634 char value[PROPERTY_VALUE_MAX]; 1635 if (property_get("ro.audio.silent", value, "0") > 0) { 1636 char *endptr; 1637 unsigned long ul = strtoul(value, &endptr, 0); 1638 if (*endptr == '\0' && ul != 0) { 1639 ALOGD("Silence is golden"); 1640 // The setprop command will not allow a property to be changed after 1641 // the first time it is set, so we don't have to worry about un-muting. 1642 setMasterMute_l(true); 1643 } 1644 } 1645 } 1646} 1647 1648// shared by MIXER and DIRECT, overridden by DUPLICATING 1649void AudioFlinger::PlaybackThread::threadLoop_write() 1650{ 1651 // FIXME rewrite to reduce number of system calls 1652 mLastWriteTime = systemTime(); 1653 mInWrite = true; 1654 int bytesWritten; 1655 1656 // If an NBAIO sink is present, use it to write the normal mixer's submix 1657 if (mNormalSink != 0) { 1658#define mBitShift 2 // FIXME 1659 size_t count = mixBufferSize >> mBitShift; 1660 ATRACE_BEGIN("write"); 1661 // update the setpoint when AudioFlinger::mScreenState changes 1662 uint32_t screenState = AudioFlinger::mScreenState; 1663 if (screenState != mScreenState) { 1664 mScreenState = screenState; 1665 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1666 if (pipe != NULL) { 1667 pipe->setAvgFrames((mScreenState & 1) ? 1668 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1669 } 1670 } 1671 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1672 ATRACE_END(); 1673 if (framesWritten > 0) { 1674 bytesWritten = framesWritten << mBitShift; 1675 } else { 1676 bytesWritten = framesWritten; 1677 } 1678 // otherwise use the HAL / AudioStreamOut directly 1679 } else { 1680 // Direct output thread. 1681 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1682 } 1683 1684 if (bytesWritten > 0) { 1685 mBytesWritten += mixBufferSize; 1686 } 1687 mNumWrites++; 1688 mInWrite = false; 1689} 1690 1691/* 1692The derived values that are cached: 1693 - mixBufferSize from frame count * frame size 1694 - activeSleepTime from activeSleepTimeUs() 1695 - idleSleepTime from idleSleepTimeUs() 1696 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1697 - maxPeriod from frame count and sample rate (MIXER only) 1698 1699The parameters that affect these derived values are: 1700 - frame count 1701 - frame size 1702 - sample rate 1703 - device type: A2DP or not 1704 - device latency 1705 - format: PCM or not 1706 - active sleep time 1707 - idle sleep time 1708*/ 1709 1710void AudioFlinger::PlaybackThread::cacheParameters_l() 1711{ 1712 mixBufferSize = mNormalFrameCount * mFrameSize; 1713 activeSleepTime = activeSleepTimeUs(); 1714 idleSleepTime = idleSleepTimeUs(); 1715} 1716 1717void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1718{ 1719 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1720 this, streamType, mTracks.size()); 1721 Mutex::Autolock _l(mLock); 1722 1723 size_t size = mTracks.size(); 1724 for (size_t i = 0; i < size; i++) { 1725 sp<Track> t = mTracks[i]; 1726 if (t->streamType() == streamType) { 1727 t->invalidate(); 1728 } 1729 } 1730} 1731 1732status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1733{ 1734 int session = chain->sessionId(); 1735 int16_t *buffer = mMixBuffer; 1736 bool ownsBuffer = false; 1737 1738 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1739 if (session > 0) { 1740 // Only one effect chain can be present in direct output thread and it uses 1741 // the mix buffer as input 1742 if (mType != DIRECT) { 1743 size_t numSamples = mNormalFrameCount * mChannelCount; 1744 buffer = new int16_t[numSamples]; 1745 memset(buffer, 0, numSamples * sizeof(int16_t)); 1746 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1747 ownsBuffer = true; 1748 } 1749 1750 // Attach all tracks with same session ID to this chain. 1751 for (size_t i = 0; i < mTracks.size(); ++i) { 1752 sp<Track> track = mTracks[i]; 1753 if (session == track->sessionId()) { 1754 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1755 buffer); 1756 track->setMainBuffer(buffer); 1757 chain->incTrackCnt(); 1758 } 1759 } 1760 1761 // indicate all active tracks in the chain 1762 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1763 sp<Track> track = mActiveTracks[i].promote(); 1764 if (track == 0) { 1765 continue; 1766 } 1767 if (session == track->sessionId()) { 1768 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1769 chain->incActiveTrackCnt(); 1770 } 1771 } 1772 } 1773 1774 chain->setInBuffer(buffer, ownsBuffer); 1775 chain->setOutBuffer(mMixBuffer); 1776 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1777 // chains list in order to be processed last as it contains output stage effects 1778 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1779 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1780 // after track specific effects and before output stage 1781 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1782 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1783 // Effect chain for other sessions are inserted at beginning of effect 1784 // chains list to be processed before output mix effects. Relative order between other 1785 // sessions is not important 1786 size_t size = mEffectChains.size(); 1787 size_t i = 0; 1788 for (i = 0; i < size; i++) { 1789 if (mEffectChains[i]->sessionId() < session) { 1790 break; 1791 } 1792 } 1793 mEffectChains.insertAt(chain, i); 1794 checkSuspendOnAddEffectChain_l(chain); 1795 1796 return NO_ERROR; 1797} 1798 1799size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1800{ 1801 int session = chain->sessionId(); 1802 1803 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1804 1805 for (size_t i = 0; i < mEffectChains.size(); i++) { 1806 if (chain == mEffectChains[i]) { 1807 mEffectChains.removeAt(i); 1808 // detach all active tracks from the chain 1809 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1810 sp<Track> track = mActiveTracks[i].promote(); 1811 if (track == 0) { 1812 continue; 1813 } 1814 if (session == track->sessionId()) { 1815 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1816 chain.get(), session); 1817 chain->decActiveTrackCnt(); 1818 } 1819 } 1820 1821 // detach all tracks with same session ID from this chain 1822 for (size_t i = 0; i < mTracks.size(); ++i) { 1823 sp<Track> track = mTracks[i]; 1824 if (session == track->sessionId()) { 1825 track->setMainBuffer(mMixBuffer); 1826 chain->decTrackCnt(); 1827 } 1828 } 1829 break; 1830 } 1831 } 1832 return mEffectChains.size(); 1833} 1834 1835status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1836 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1837{ 1838 Mutex::Autolock _l(mLock); 1839 return attachAuxEffect_l(track, EffectId); 1840} 1841 1842status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1843 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1844{ 1845 status_t status = NO_ERROR; 1846 1847 if (EffectId == 0) { 1848 track->setAuxBuffer(0, NULL); 1849 } else { 1850 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1851 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1852 if (effect != 0) { 1853 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1854 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1855 } else { 1856 status = INVALID_OPERATION; 1857 } 1858 } else { 1859 status = BAD_VALUE; 1860 } 1861 } 1862 return status; 1863} 1864 1865void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1866{ 1867 for (size_t i = 0; i < mTracks.size(); ++i) { 1868 sp<Track> track = mTracks[i]; 1869 if (track->auxEffectId() == effectId) { 1870 attachAuxEffect_l(track, 0); 1871 } 1872 } 1873} 1874 1875bool AudioFlinger::PlaybackThread::threadLoop() 1876{ 1877 Vector< sp<Track> > tracksToRemove; 1878 1879 standbyTime = systemTime(); 1880 1881 // MIXER 1882 nsecs_t lastWarning = 0; 1883 1884 // DUPLICATING 1885 // FIXME could this be made local to while loop? 1886 writeFrames = 0; 1887 1888 cacheParameters_l(); 1889 sleepTime = idleSleepTime; 1890 1891 if (mType == MIXER) { 1892 sleepTimeShift = 0; 1893 } 1894 1895 CpuStats cpuStats; 1896 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1897 1898 acquireWakeLock(); 1899 1900 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1901 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1902 // and then that string will be logged at the next convenient opportunity. 1903 const char *logString = NULL; 1904 1905 while (!exitPending()) 1906 { 1907 cpuStats.sample(myName); 1908 1909 Vector< sp<EffectChain> > effectChains; 1910 1911 processConfigEvents(); 1912 1913 { // scope for mLock 1914 1915 Mutex::Autolock _l(mLock); 1916 1917 if (logString != NULL) { 1918 mNBLogWriter->logTimestamp(); 1919 mNBLogWriter->log(logString); 1920 logString = NULL; 1921 } 1922 1923 if (checkForNewParameters_l()) { 1924 cacheParameters_l(); 1925 } 1926 1927 saveOutputTracks(); 1928 1929 // put audio hardware into standby after short delay 1930 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1931 isSuspended())) { 1932 if (!mStandby) { 1933 1934 threadLoop_standby(); 1935 1936 mStandby = true; 1937 } 1938 1939 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1940 // we're about to wait, flush the binder command buffer 1941 IPCThreadState::self()->flushCommands(); 1942 1943 clearOutputTracks(); 1944 1945 if (exitPending()) { 1946 break; 1947 } 1948 1949 releaseWakeLock_l(); 1950 // wait until we have something to do... 1951 ALOGV("%s going to sleep", myName.string()); 1952 mWaitWorkCV.wait(mLock); 1953 ALOGV("%s waking up", myName.string()); 1954 acquireWakeLock_l(); 1955 1956 mMixerStatus = MIXER_IDLE; 1957 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1958 mBytesWritten = 0; 1959 1960 checkSilentMode_l(); 1961 1962 standbyTime = systemTime() + standbyDelay; 1963 sleepTime = idleSleepTime; 1964 if (mType == MIXER) { 1965 sleepTimeShift = 0; 1966 } 1967 1968 continue; 1969 } 1970 } 1971 1972 // mMixerStatusIgnoringFastTracks is also updated internally 1973 mMixerStatus = prepareTracks_l(&tracksToRemove); 1974 1975 // prevent any changes in effect chain list and in each effect chain 1976 // during mixing and effect process as the audio buffers could be deleted 1977 // or modified if an effect is created or deleted 1978 lockEffectChains_l(effectChains); 1979 } 1980 1981 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1982 threadLoop_mix(); 1983 } else { 1984 threadLoop_sleepTime(); 1985 } 1986 1987 if (isSuspended()) { 1988 sleepTime = suspendSleepTimeUs(); 1989 mBytesWritten += mixBufferSize; 1990 } 1991 1992 // only process effects if we're going to write 1993 if (sleepTime == 0) { 1994 for (size_t i = 0; i < effectChains.size(); i ++) { 1995 effectChains[i]->process_l(); 1996 } 1997 } 1998 1999 // enable changes in effect chain 2000 unlockEffectChains(effectChains); 2001 2002 // sleepTime == 0 means we must write to audio hardware 2003 if (sleepTime == 0) { 2004 2005 threadLoop_write(); 2006 2007if (mType == MIXER) { 2008 // write blocked detection 2009 nsecs_t now = systemTime(); 2010 nsecs_t delta = now - mLastWriteTime; 2011 if (!mStandby && delta > maxPeriod) { 2012 mNumDelayedWrites++; 2013 if ((now - lastWarning) > kWarningThrottleNs) { 2014 ATRACE_NAME("underrun"); 2015 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2016 ns2ms(delta), mNumDelayedWrites, this); 2017 lastWarning = now; 2018 } 2019 } 2020} 2021 2022 mStandby = false; 2023 } else { 2024 usleep(sleepTime); 2025 } 2026 2027 // Finally let go of removed track(s), without the lock held 2028 // since we can't guarantee the destructors won't acquire that 2029 // same lock. This will also mutate and push a new fast mixer state. 2030 threadLoop_removeTracks(tracksToRemove); 2031 tracksToRemove.clear(); 2032 2033 // FIXME I don't understand the need for this here; 2034 // it was in the original code but maybe the 2035 // assignment in saveOutputTracks() makes this unnecessary? 2036 clearOutputTracks(); 2037 2038 // Effect chains will be actually deleted here if they were removed from 2039 // mEffectChains list during mixing or effects processing 2040 effectChains.clear(); 2041 2042 // FIXME Note that the above .clear() is no longer necessary since effectChains 2043 // is now local to this block, but will keep it for now (at least until merge done). 2044 } 2045 2046 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2047 if (mType == MIXER || mType == DIRECT) { 2048 // put output stream into standby mode 2049 if (!mStandby) { 2050 mOutput->stream->common.standby(&mOutput->stream->common); 2051 } 2052 } 2053 2054 releaseWakeLock(); 2055 2056 ALOGV("Thread %p type %d exiting", this, mType); 2057 return false; 2058} 2059 2060 2061// ---------------------------------------------------------------------------- 2062 2063AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2064 audio_io_handle_t id, audio_devices_t device, type_t type) 2065 : PlaybackThread(audioFlinger, output, id, device, type), 2066 // mAudioMixer below 2067 // mFastMixer below 2068 mFastMixerFutex(0) 2069 // mOutputSink below 2070 // mPipeSink below 2071 // mNormalSink below 2072{ 2073 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2074 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2075 "mFrameCount=%d, mNormalFrameCount=%d", 2076 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2077 mNormalFrameCount); 2078 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2079 2080 // FIXME - Current mixer implementation only supports stereo output 2081 if (mChannelCount != FCC_2) { 2082 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2083 } 2084 2085 // create an NBAIO sink for the HAL output stream, and negotiate 2086 mOutputSink = new AudioStreamOutSink(output->stream); 2087 size_t numCounterOffers = 0; 2088 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2089 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2090 ALOG_ASSERT(index == 0); 2091 2092 // initialize fast mixer depending on configuration 2093 bool initFastMixer; 2094 switch (kUseFastMixer) { 2095 case FastMixer_Never: 2096 initFastMixer = false; 2097 break; 2098 case FastMixer_Always: 2099 initFastMixer = true; 2100 break; 2101 case FastMixer_Static: 2102 case FastMixer_Dynamic: 2103 initFastMixer = mFrameCount < mNormalFrameCount; 2104 break; 2105 } 2106 if (initFastMixer) { 2107 2108 // create a MonoPipe to connect our submix to FastMixer 2109 NBAIO_Format format = mOutputSink->format(); 2110 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2111 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2112 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2113 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2114 const NBAIO_Format offers[1] = {format}; 2115 size_t numCounterOffers = 0; 2116 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2117 ALOG_ASSERT(index == 0); 2118 monoPipe->setAvgFrames((mScreenState & 1) ? 2119 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2120 mPipeSink = monoPipe; 2121 2122#ifdef TEE_SINK 2123 if (mTeeSinkOutputEnabled) { 2124 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2125 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2126 numCounterOffers = 0; 2127 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2128 ALOG_ASSERT(index == 0); 2129 mTeeSink = teeSink; 2130 PipeReader *teeSource = new PipeReader(*teeSink); 2131 numCounterOffers = 0; 2132 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2133 ALOG_ASSERT(index == 0); 2134 mTeeSource = teeSource; 2135 } 2136#endif 2137 2138 // create fast mixer and configure it initially with just one fast track for our submix 2139 mFastMixer = new FastMixer(); 2140 FastMixerStateQueue *sq = mFastMixer->sq(); 2141#ifdef STATE_QUEUE_DUMP 2142 sq->setObserverDump(&mStateQueueObserverDump); 2143 sq->setMutatorDump(&mStateQueueMutatorDump); 2144#endif 2145 FastMixerState *state = sq->begin(); 2146 FastTrack *fastTrack = &state->mFastTracks[0]; 2147 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2148 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2149 fastTrack->mVolumeProvider = NULL; 2150 fastTrack->mGeneration++; 2151 state->mFastTracksGen++; 2152 state->mTrackMask = 1; 2153 // fast mixer will use the HAL output sink 2154 state->mOutputSink = mOutputSink.get(); 2155 state->mOutputSinkGen++; 2156 state->mFrameCount = mFrameCount; 2157 state->mCommand = FastMixerState::COLD_IDLE; 2158 // already done in constructor initialization list 2159 //mFastMixerFutex = 0; 2160 state->mColdFutexAddr = &mFastMixerFutex; 2161 state->mColdGen++; 2162 state->mDumpState = &mFastMixerDumpState; 2163#ifdef TEE_SINK 2164 state->mTeeSink = mTeeSink.get(); 2165#endif 2166 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2167 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2168 sq->end(); 2169 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2170 2171 // start the fast mixer 2172 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2173 pid_t tid = mFastMixer->getTid(); 2174 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2175 if (err != 0) { 2176 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2177 kPriorityFastMixer, getpid_cached, tid, err); 2178 } 2179 2180#ifdef AUDIO_WATCHDOG 2181 // create and start the watchdog 2182 mAudioWatchdog = new AudioWatchdog(); 2183 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2184 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2185 tid = mAudioWatchdog->getTid(); 2186 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2187 if (err != 0) { 2188 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2189 kPriorityFastMixer, getpid_cached, tid, err); 2190 } 2191#endif 2192 2193 } else { 2194 mFastMixer = NULL; 2195 } 2196 2197 switch (kUseFastMixer) { 2198 case FastMixer_Never: 2199 case FastMixer_Dynamic: 2200 mNormalSink = mOutputSink; 2201 break; 2202 case FastMixer_Always: 2203 mNormalSink = mPipeSink; 2204 break; 2205 case FastMixer_Static: 2206 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2207 break; 2208 } 2209} 2210 2211AudioFlinger::MixerThread::~MixerThread() 2212{ 2213 if (mFastMixer != NULL) { 2214 FastMixerStateQueue *sq = mFastMixer->sq(); 2215 FastMixerState *state = sq->begin(); 2216 if (state->mCommand == FastMixerState::COLD_IDLE) { 2217 int32_t old = android_atomic_inc(&mFastMixerFutex); 2218 if (old == -1) { 2219 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2220 } 2221 } 2222 state->mCommand = FastMixerState::EXIT; 2223 sq->end(); 2224 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2225 mFastMixer->join(); 2226 // Though the fast mixer thread has exited, it's state queue is still valid. 2227 // We'll use that extract the final state which contains one remaining fast track 2228 // corresponding to our sub-mix. 2229 state = sq->begin(); 2230 ALOG_ASSERT(state->mTrackMask == 1); 2231 FastTrack *fastTrack = &state->mFastTracks[0]; 2232 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2233 delete fastTrack->mBufferProvider; 2234 sq->end(false /*didModify*/); 2235 delete mFastMixer; 2236#ifdef AUDIO_WATCHDOG 2237 if (mAudioWatchdog != 0) { 2238 mAudioWatchdog->requestExit(); 2239 mAudioWatchdog->requestExitAndWait(); 2240 mAudioWatchdog.clear(); 2241 } 2242#endif 2243 } 2244 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2245 delete mAudioMixer; 2246} 2247 2248 2249uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2250{ 2251 if (mFastMixer != NULL) { 2252 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2253 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2254 } 2255 return latency; 2256} 2257 2258 2259void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2260{ 2261 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2262} 2263 2264void AudioFlinger::MixerThread::threadLoop_write() 2265{ 2266 // FIXME we should only do one push per cycle; confirm this is true 2267 // Start the fast mixer if it's not already running 2268 if (mFastMixer != NULL) { 2269 FastMixerStateQueue *sq = mFastMixer->sq(); 2270 FastMixerState *state = sq->begin(); 2271 if (state->mCommand != FastMixerState::MIX_WRITE && 2272 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2273 if (state->mCommand == FastMixerState::COLD_IDLE) { 2274 int32_t old = android_atomic_inc(&mFastMixerFutex); 2275 if (old == -1) { 2276 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2277 } 2278#ifdef AUDIO_WATCHDOG 2279 if (mAudioWatchdog != 0) { 2280 mAudioWatchdog->resume(); 2281 } 2282#endif 2283 } 2284 state->mCommand = FastMixerState::MIX_WRITE; 2285 sq->end(); 2286 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2287 if (kUseFastMixer == FastMixer_Dynamic) { 2288 mNormalSink = mPipeSink; 2289 } 2290 } else { 2291 sq->end(false /*didModify*/); 2292 } 2293 } 2294 PlaybackThread::threadLoop_write(); 2295} 2296 2297void AudioFlinger::MixerThread::threadLoop_standby() 2298{ 2299 // Idle the fast mixer if it's currently running 2300 if (mFastMixer != NULL) { 2301 FastMixerStateQueue *sq = mFastMixer->sq(); 2302 FastMixerState *state = sq->begin(); 2303 if (!(state->mCommand & FastMixerState::IDLE)) { 2304 state->mCommand = FastMixerState::COLD_IDLE; 2305 state->mColdFutexAddr = &mFastMixerFutex; 2306 state->mColdGen++; 2307 mFastMixerFutex = 0; 2308 sq->end(); 2309 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2310 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2311 if (kUseFastMixer == FastMixer_Dynamic) { 2312 mNormalSink = mOutputSink; 2313 } 2314#ifdef AUDIO_WATCHDOG 2315 if (mAudioWatchdog != 0) { 2316 mAudioWatchdog->pause(); 2317 } 2318#endif 2319 } else { 2320 sq->end(false /*didModify*/); 2321 } 2322 } 2323 PlaybackThread::threadLoop_standby(); 2324} 2325 2326// shared by MIXER and DIRECT, overridden by DUPLICATING 2327void AudioFlinger::PlaybackThread::threadLoop_standby() 2328{ 2329 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2330 mOutput->stream->common.standby(&mOutput->stream->common); 2331} 2332 2333void AudioFlinger::MixerThread::threadLoop_mix() 2334{ 2335 // obtain the presentation timestamp of the next output buffer 2336 int64_t pts; 2337 status_t status = INVALID_OPERATION; 2338 2339 if (mNormalSink != 0) { 2340 status = mNormalSink->getNextWriteTimestamp(&pts); 2341 } else { 2342 status = mOutputSink->getNextWriteTimestamp(&pts); 2343 } 2344 2345 if (status != NO_ERROR) { 2346 pts = AudioBufferProvider::kInvalidPTS; 2347 } 2348 2349 // mix buffers... 2350 mAudioMixer->process(pts); 2351 // increase sleep time progressively when application underrun condition clears. 2352 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2353 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2354 // such that we would underrun the audio HAL. 2355 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2356 sleepTimeShift--; 2357 } 2358 sleepTime = 0; 2359 standbyTime = systemTime() + standbyDelay; 2360 //TODO: delay standby when effects have a tail 2361} 2362 2363void AudioFlinger::MixerThread::threadLoop_sleepTime() 2364{ 2365 // If no tracks are ready, sleep once for the duration of an output 2366 // buffer size, then write 0s to the output 2367 if (sleepTime == 0) { 2368 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2369 sleepTime = activeSleepTime >> sleepTimeShift; 2370 if (sleepTime < kMinThreadSleepTimeUs) { 2371 sleepTime = kMinThreadSleepTimeUs; 2372 } 2373 // reduce sleep time in case of consecutive application underruns to avoid 2374 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2375 // duration we would end up writing less data than needed by the audio HAL if 2376 // the condition persists. 2377 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2378 sleepTimeShift++; 2379 } 2380 } else { 2381 sleepTime = idleSleepTime; 2382 } 2383 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2384 memset (mMixBuffer, 0, mixBufferSize); 2385 sleepTime = 0; 2386 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2387 "anticipated start"); 2388 } 2389 // TODO add standby time extension fct of effect tail 2390} 2391 2392// prepareTracks_l() must be called with ThreadBase::mLock held 2393AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2394 Vector< sp<Track> > *tracksToRemove) 2395{ 2396 2397 mixer_state mixerStatus = MIXER_IDLE; 2398 // find out which tracks need to be processed 2399 size_t count = mActiveTracks.size(); 2400 size_t mixedTracks = 0; 2401 size_t tracksWithEffect = 0; 2402 // counts only _active_ fast tracks 2403 size_t fastTracks = 0; 2404 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2405 2406 float masterVolume = mMasterVolume; 2407 bool masterMute = mMasterMute; 2408 2409 if (masterMute) { 2410 masterVolume = 0; 2411 } 2412 // Delegate master volume control to effect in output mix effect chain if needed 2413 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2414 if (chain != 0) { 2415 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2416 chain->setVolume_l(&v, &v); 2417 masterVolume = (float)((v + (1 << 23)) >> 24); 2418 chain.clear(); 2419 } 2420 2421 // prepare a new state to push 2422 FastMixerStateQueue *sq = NULL; 2423 FastMixerState *state = NULL; 2424 bool didModify = false; 2425 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2426 if (mFastMixer != NULL) { 2427 sq = mFastMixer->sq(); 2428 state = sq->begin(); 2429 } 2430 2431 for (size_t i=0 ; i<count ; i++) { 2432 sp<Track> t = mActiveTracks[i].promote(); 2433 if (t == 0) { 2434 continue; 2435 } 2436 2437 // this const just means the local variable doesn't change 2438 Track* const track = t.get(); 2439 2440 // process fast tracks 2441 if (track->isFastTrack()) { 2442 2443 // It's theoretically possible (though unlikely) for a fast track to be created 2444 // and then removed within the same normal mix cycle. This is not a problem, as 2445 // the track never becomes active so it's fast mixer slot is never touched. 2446 // The converse, of removing an (active) track and then creating a new track 2447 // at the identical fast mixer slot within the same normal mix cycle, 2448 // is impossible because the slot isn't marked available until the end of each cycle. 2449 int j = track->mFastIndex; 2450 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2451 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2452 FastTrack *fastTrack = &state->mFastTracks[j]; 2453 2454 // Determine whether the track is currently in underrun condition, 2455 // and whether it had a recent underrun. 2456 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2457 FastTrackUnderruns underruns = ftDump->mUnderruns; 2458 uint32_t recentFull = (underruns.mBitFields.mFull - 2459 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2460 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2461 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2462 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2463 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2464 uint32_t recentUnderruns = recentPartial + recentEmpty; 2465 track->mObservedUnderruns = underruns; 2466 // don't count underruns that occur while stopping or pausing 2467 // or stopped which can occur when flush() is called while active 2468 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2469 track->mUnderrunCount += recentUnderruns; 2470 } 2471 2472 // This is similar to the state machine for normal tracks, 2473 // with a few modifications for fast tracks. 2474 bool isActive = true; 2475 switch (track->mState) { 2476 case TrackBase::STOPPING_1: 2477 // track stays active in STOPPING_1 state until first underrun 2478 if (recentUnderruns > 0) { 2479 track->mState = TrackBase::STOPPING_2; 2480 } 2481 break; 2482 case TrackBase::PAUSING: 2483 // ramp down is not yet implemented 2484 track->setPaused(); 2485 break; 2486 case TrackBase::RESUMING: 2487 // ramp up is not yet implemented 2488 track->mState = TrackBase::ACTIVE; 2489 break; 2490 case TrackBase::ACTIVE: 2491 if (recentFull > 0 || recentPartial > 0) { 2492 // track has provided at least some frames recently: reset retry count 2493 track->mRetryCount = kMaxTrackRetries; 2494 } 2495 if (recentUnderruns == 0) { 2496 // no recent underruns: stay active 2497 break; 2498 } 2499 // there has recently been an underrun of some kind 2500 if (track->sharedBuffer() == 0) { 2501 // were any of the recent underruns "empty" (no frames available)? 2502 if (recentEmpty == 0) { 2503 // no, then ignore the partial underruns as they are allowed indefinitely 2504 break; 2505 } 2506 // there has recently been an "empty" underrun: decrement the retry counter 2507 if (--(track->mRetryCount) > 0) { 2508 break; 2509 } 2510 // indicate to client process that the track was disabled because of underrun; 2511 // it will then automatically call start() when data is available 2512 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2513 // remove from active list, but state remains ACTIVE [confusing but true] 2514 isActive = false; 2515 break; 2516 } 2517 // fall through 2518 case TrackBase::STOPPING_2: 2519 case TrackBase::PAUSED: 2520 case TrackBase::TERMINATED: 2521 case TrackBase::STOPPED: 2522 case TrackBase::FLUSHED: // flush() while active 2523 // Check for presentation complete if track is inactive 2524 // We have consumed all the buffers of this track. 2525 // This would be incomplete if we auto-paused on underrun 2526 { 2527 size_t audioHALFrames = 2528 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2529 size_t framesWritten = mBytesWritten / mFrameSize; 2530 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2531 // track stays in active list until presentation is complete 2532 break; 2533 } 2534 } 2535 if (track->isStopping_2()) { 2536 track->mState = TrackBase::STOPPED; 2537 } 2538 if (track->isStopped()) { 2539 // Can't reset directly, as fast mixer is still polling this track 2540 // track->reset(); 2541 // So instead mark this track as needing to be reset after push with ack 2542 resetMask |= 1 << i; 2543 } 2544 isActive = false; 2545 break; 2546 case TrackBase::IDLE: 2547 default: 2548 LOG_FATAL("unexpected track state %d", track->mState); 2549 } 2550 2551 if (isActive) { 2552 // was it previously inactive? 2553 if (!(state->mTrackMask & (1 << j))) { 2554 ExtendedAudioBufferProvider *eabp = track; 2555 VolumeProvider *vp = track; 2556 fastTrack->mBufferProvider = eabp; 2557 fastTrack->mVolumeProvider = vp; 2558 fastTrack->mSampleRate = track->mSampleRate; 2559 fastTrack->mChannelMask = track->mChannelMask; 2560 fastTrack->mGeneration++; 2561 state->mTrackMask |= 1 << j; 2562 didModify = true; 2563 // no acknowledgement required for newly active tracks 2564 } 2565 // cache the combined master volume and stream type volume for fast mixer; this 2566 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2567 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2568 ++fastTracks; 2569 } else { 2570 // was it previously active? 2571 if (state->mTrackMask & (1 << j)) { 2572 fastTrack->mBufferProvider = NULL; 2573 fastTrack->mGeneration++; 2574 state->mTrackMask &= ~(1 << j); 2575 didModify = true; 2576 // If any fast tracks were removed, we must wait for acknowledgement 2577 // because we're about to decrement the last sp<> on those tracks. 2578 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2579 } else { 2580 LOG_FATAL("fast track %d should have been active", j); 2581 } 2582 tracksToRemove->add(track); 2583 // Avoids a misleading display in dumpsys 2584 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2585 } 2586 continue; 2587 } 2588 2589 { // local variable scope to avoid goto warning 2590 2591 audio_track_cblk_t* cblk = track->cblk(); 2592 2593 // The first time a track is added we wait 2594 // for all its buffers to be filled before processing it 2595 int name = track->name(); 2596 // make sure that we have enough frames to mix one full buffer. 2597 // enforce this condition only once to enable draining the buffer in case the client 2598 // app does not call stop() and relies on underrun to stop: 2599 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2600 // during last round 2601 uint32_t minFrames = 1; 2602 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2603 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2604 if (t->sampleRate() == mSampleRate) { 2605 minFrames = mNormalFrameCount; 2606 } else { 2607 // +1 for rounding and +1 for additional sample needed for interpolation 2608 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2609 // add frames already consumed but not yet released by the resampler 2610 // because cblk->framesReady() will include these frames 2611 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2612 // the minimum track buffer size is normally twice the number of frames necessary 2613 // to fill one buffer and the resampler should not leave more than one buffer worth 2614 // of unreleased frames after each pass, but just in case... 2615 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2616 } 2617 } 2618 if ((track->framesReady() >= minFrames) && track->isReady() && 2619 !track->isPaused() && !track->isTerminated()) 2620 { 2621 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2622 this); 2623 2624 mixedTracks++; 2625 2626 // track->mainBuffer() != mMixBuffer means there is an effect chain 2627 // connected to the track 2628 chain.clear(); 2629 if (track->mainBuffer() != mMixBuffer) { 2630 chain = getEffectChain_l(track->sessionId()); 2631 // Delegate volume control to effect in track effect chain if needed 2632 if (chain != 0) { 2633 tracksWithEffect++; 2634 } else { 2635 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2636 "session %d", 2637 name, track->sessionId()); 2638 } 2639 } 2640 2641 2642 int param = AudioMixer::VOLUME; 2643 if (track->mFillingUpStatus == Track::FS_FILLED) { 2644 // no ramp for the first volume setting 2645 track->mFillingUpStatus = Track::FS_ACTIVE; 2646 if (track->mState == TrackBase::RESUMING) { 2647 track->mState = TrackBase::ACTIVE; 2648 param = AudioMixer::RAMP_VOLUME; 2649 } 2650 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2651 } else if (cblk->server != 0) { 2652 // If the track is stopped before the first frame was mixed, 2653 // do not apply ramp 2654 param = AudioMixer::RAMP_VOLUME; 2655 } 2656 2657 // compute volume for this track 2658 uint32_t vl, vr, va; 2659 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2660 vl = vr = va = 0; 2661 if (track->isPausing()) { 2662 track->setPaused(); 2663 } 2664 } else { 2665 2666 // read original volumes with volume control 2667 float typeVolume = mStreamTypes[track->streamType()].volume; 2668 float v = masterVolume * typeVolume; 2669 ServerProxy *proxy = track->mServerProxy; 2670 uint32_t vlr = proxy->getVolumeLR(); 2671 vl = vlr & 0xFFFF; 2672 vr = vlr >> 16; 2673 // track volumes come from shared memory, so can't be trusted and must be clamped 2674 if (vl > MAX_GAIN_INT) { 2675 ALOGV("Track left volume out of range: %04X", vl); 2676 vl = MAX_GAIN_INT; 2677 } 2678 if (vr > MAX_GAIN_INT) { 2679 ALOGV("Track right volume out of range: %04X", vr); 2680 vr = MAX_GAIN_INT; 2681 } 2682 // now apply the master volume and stream type volume 2683 vl = (uint32_t)(v * vl) << 12; 2684 vr = (uint32_t)(v * vr) << 12; 2685 // assuming master volume and stream type volume each go up to 1.0, 2686 // vl and vr are now in 8.24 format 2687 2688 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2689 // send level comes from shared memory and so may be corrupt 2690 if (sendLevel > MAX_GAIN_INT) { 2691 ALOGV("Track send level out of range: %04X", sendLevel); 2692 sendLevel = MAX_GAIN_INT; 2693 } 2694 va = (uint32_t)(v * sendLevel); 2695 } 2696 // Delegate volume control to effect in track effect chain if needed 2697 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2698 // Do not ramp volume if volume is controlled by effect 2699 param = AudioMixer::VOLUME; 2700 track->mHasVolumeController = true; 2701 } else { 2702 // force no volume ramp when volume controller was just disabled or removed 2703 // from effect chain to avoid volume spike 2704 if (track->mHasVolumeController) { 2705 param = AudioMixer::VOLUME; 2706 } 2707 track->mHasVolumeController = false; 2708 } 2709 2710 // Convert volumes from 8.24 to 4.12 format 2711 // This additional clamping is needed in case chain->setVolume_l() overshot 2712 vl = (vl + (1 << 11)) >> 12; 2713 if (vl > MAX_GAIN_INT) { 2714 vl = MAX_GAIN_INT; 2715 } 2716 vr = (vr + (1 << 11)) >> 12; 2717 if (vr > MAX_GAIN_INT) { 2718 vr = MAX_GAIN_INT; 2719 } 2720 2721 if (va > MAX_GAIN_INT) { 2722 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2723 } 2724 2725 // XXX: these things DON'T need to be done each time 2726 mAudioMixer->setBufferProvider(name, track); 2727 mAudioMixer->enable(name); 2728 2729 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2730 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2731 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2732 mAudioMixer->setParameter( 2733 name, 2734 AudioMixer::TRACK, 2735 AudioMixer::FORMAT, (void *)track->format()); 2736 mAudioMixer->setParameter( 2737 name, 2738 AudioMixer::TRACK, 2739 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2740 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2741 uint32_t maxSampleRate = mSampleRate * 2; 2742 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2743 if (reqSampleRate == 0) { 2744 reqSampleRate = mSampleRate; 2745 } else if (reqSampleRate > maxSampleRate) { 2746 reqSampleRate = maxSampleRate; 2747 } 2748 mAudioMixer->setParameter( 2749 name, 2750 AudioMixer::RESAMPLE, 2751 AudioMixer::SAMPLE_RATE, 2752 (void *)reqSampleRate); 2753 mAudioMixer->setParameter( 2754 name, 2755 AudioMixer::TRACK, 2756 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2757 mAudioMixer->setParameter( 2758 name, 2759 AudioMixer::TRACK, 2760 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2761 2762 // reset retry count 2763 track->mRetryCount = kMaxTrackRetries; 2764 2765 // If one track is ready, set the mixer ready if: 2766 // - the mixer was not ready during previous round OR 2767 // - no other track is not ready 2768 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2769 mixerStatus != MIXER_TRACKS_ENABLED) { 2770 mixerStatus = MIXER_TRACKS_READY; 2771 } 2772 } else { 2773 // clear effect chain input buffer if an active track underruns to avoid sending 2774 // previous audio buffer again to effects 2775 chain = getEffectChain_l(track->sessionId()); 2776 if (chain != 0) { 2777 chain->clearInputBuffer(); 2778 } 2779 2780 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2781 cblk->server, this); 2782 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2783 track->isStopped() || track->isPaused()) { 2784 // We have consumed all the buffers of this track. 2785 // Remove it from the list of active tracks. 2786 // TODO: use actual buffer filling status instead of latency when available from 2787 // audio HAL 2788 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2789 size_t framesWritten = mBytesWritten / mFrameSize; 2790 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2791 if (track->isStopped()) { 2792 track->reset(); 2793 } 2794 tracksToRemove->add(track); 2795 } 2796 } else { 2797 track->mUnderrunCount++; 2798 // No buffers for this track. Give it a few chances to 2799 // fill a buffer, then remove it from active list. 2800 if (--(track->mRetryCount) <= 0) { 2801 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2802 tracksToRemove->add(track); 2803 // indicate to client process that the track was disabled because of underrun; 2804 // it will then automatically call start() when data is available 2805 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2806 // If one track is not ready, mark the mixer also not ready if: 2807 // - the mixer was ready during previous round OR 2808 // - no other track is ready 2809 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2810 mixerStatus != MIXER_TRACKS_READY) { 2811 mixerStatus = MIXER_TRACKS_ENABLED; 2812 } 2813 } 2814 mAudioMixer->disable(name); 2815 } 2816 2817 } // local variable scope to avoid goto warning 2818track_is_ready: ; 2819 2820 } 2821 2822 // Push the new FastMixer state if necessary 2823 bool pauseAudioWatchdog = false; 2824 if (didModify) { 2825 state->mFastTracksGen++; 2826 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2827 if (kUseFastMixer == FastMixer_Dynamic && 2828 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2829 state->mCommand = FastMixerState::COLD_IDLE; 2830 state->mColdFutexAddr = &mFastMixerFutex; 2831 state->mColdGen++; 2832 mFastMixerFutex = 0; 2833 if (kUseFastMixer == FastMixer_Dynamic) { 2834 mNormalSink = mOutputSink; 2835 } 2836 // If we go into cold idle, need to wait for acknowledgement 2837 // so that fast mixer stops doing I/O. 2838 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2839 pauseAudioWatchdog = true; 2840 } 2841 } 2842 if (sq != NULL) { 2843 sq->end(didModify); 2844 sq->push(block); 2845 } 2846#ifdef AUDIO_WATCHDOG 2847 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2848 mAudioWatchdog->pause(); 2849 } 2850#endif 2851 2852 // Now perform the deferred reset on fast tracks that have stopped 2853 while (resetMask != 0) { 2854 size_t i = __builtin_ctz(resetMask); 2855 ALOG_ASSERT(i < count); 2856 resetMask &= ~(1 << i); 2857 sp<Track> t = mActiveTracks[i].promote(); 2858 if (t == 0) { 2859 continue; 2860 } 2861 Track* track = t.get(); 2862 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2863 track->reset(); 2864 } 2865 2866 // remove all the tracks that need to be... 2867 count = tracksToRemove->size(); 2868 if (CC_UNLIKELY(count)) { 2869 for (size_t i=0 ; i<count ; i++) { 2870 const sp<Track>& track = tracksToRemove->itemAt(i); 2871 mActiveTracks.remove(track); 2872 if (track->mainBuffer() != mMixBuffer) { 2873 chain = getEffectChain_l(track->sessionId()); 2874 if (chain != 0) { 2875 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2876 track->sessionId()); 2877 chain->decActiveTrackCnt(); 2878 } 2879 } 2880 if (track->isTerminated()) { 2881 removeTrack_l(track); 2882 } 2883 } 2884 } 2885 2886 // mix buffer must be cleared if all tracks are connected to an 2887 // effect chain as in this case the mixer will not write to 2888 // mix buffer and track effects will accumulate into it 2889 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2890 (mixedTracks == 0 && fastTracks > 0)) { 2891 // FIXME as a performance optimization, should remember previous zero status 2892 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2893 } 2894 2895 // if any fast tracks, then status is ready 2896 mMixerStatusIgnoringFastTracks = mixerStatus; 2897 if (fastTracks > 0) { 2898 mixerStatus = MIXER_TRACKS_READY; 2899 } 2900 return mixerStatus; 2901} 2902 2903// getTrackName_l() must be called with ThreadBase::mLock held 2904int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2905{ 2906 return mAudioMixer->getTrackName(channelMask, sessionId); 2907} 2908 2909// deleteTrackName_l() must be called with ThreadBase::mLock held 2910void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2911{ 2912 ALOGV("remove track (%d) and delete from mixer", name); 2913 mAudioMixer->deleteTrackName(name); 2914} 2915 2916// checkForNewParameters_l() must be called with ThreadBase::mLock held 2917bool AudioFlinger::MixerThread::checkForNewParameters_l() 2918{ 2919 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2920 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2921 bool reconfig = false; 2922 2923 while (!mNewParameters.isEmpty()) { 2924 2925 if (mFastMixer != NULL) { 2926 FastMixerStateQueue *sq = mFastMixer->sq(); 2927 FastMixerState *state = sq->begin(); 2928 if (!(state->mCommand & FastMixerState::IDLE)) { 2929 previousCommand = state->mCommand; 2930 state->mCommand = FastMixerState::HOT_IDLE; 2931 sq->end(); 2932 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2933 } else { 2934 sq->end(false /*didModify*/); 2935 } 2936 } 2937 2938 status_t status = NO_ERROR; 2939 String8 keyValuePair = mNewParameters[0]; 2940 AudioParameter param = AudioParameter(keyValuePair); 2941 int value; 2942 2943 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2944 reconfig = true; 2945 } 2946 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2947 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2948 status = BAD_VALUE; 2949 } else { 2950 reconfig = true; 2951 } 2952 } 2953 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2954 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2955 status = BAD_VALUE; 2956 } else { 2957 reconfig = true; 2958 } 2959 } 2960 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2961 // do not accept frame count changes if tracks are open as the track buffer 2962 // size depends on frame count and correct behavior would not be guaranteed 2963 // if frame count is changed after track creation 2964 if (!mTracks.isEmpty()) { 2965 status = INVALID_OPERATION; 2966 } else { 2967 reconfig = true; 2968 } 2969 } 2970 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2971#ifdef ADD_BATTERY_DATA 2972 // when changing the audio output device, call addBatteryData to notify 2973 // the change 2974 if (mOutDevice != value) { 2975 uint32_t params = 0; 2976 // check whether speaker is on 2977 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2978 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2979 } 2980 2981 audio_devices_t deviceWithoutSpeaker 2982 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2983 // check if any other device (except speaker) is on 2984 if (value & deviceWithoutSpeaker ) { 2985 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2986 } 2987 2988 if (params != 0) { 2989 addBatteryData(params); 2990 } 2991 } 2992#endif 2993 2994 // forward device change to effects that have requested to be 2995 // aware of attached audio device. 2996 if (value != AUDIO_DEVICE_NONE) { 2997 mOutDevice = value; 2998 for (size_t i = 0; i < mEffectChains.size(); i++) { 2999 mEffectChains[i]->setDevice_l(mOutDevice); 3000 } 3001 } 3002 } 3003 3004 if (status == NO_ERROR) { 3005 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3006 keyValuePair.string()); 3007 if (!mStandby && status == INVALID_OPERATION) { 3008 mOutput->stream->common.standby(&mOutput->stream->common); 3009 mStandby = true; 3010 mBytesWritten = 0; 3011 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3012 keyValuePair.string()); 3013 } 3014 if (status == NO_ERROR && reconfig) { 3015 delete mAudioMixer; 3016 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3017 mAudioMixer = NULL; 3018 readOutputParameters(); 3019 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3020 for (size_t i = 0; i < mTracks.size() ; i++) { 3021 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3022 if (name < 0) { 3023 break; 3024 } 3025 mTracks[i]->mName = name; 3026 } 3027 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3028 } 3029 } 3030 3031 mNewParameters.removeAt(0); 3032 3033 mParamStatus = status; 3034 mParamCond.signal(); 3035 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3036 // already timed out waiting for the status and will never signal the condition. 3037 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3038 } 3039 3040 if (!(previousCommand & FastMixerState::IDLE)) { 3041 ALOG_ASSERT(mFastMixer != NULL); 3042 FastMixerStateQueue *sq = mFastMixer->sq(); 3043 FastMixerState *state = sq->begin(); 3044 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3045 state->mCommand = previousCommand; 3046 sq->end(); 3047 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3048 } 3049 3050 return reconfig; 3051} 3052 3053 3054void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3055{ 3056 const size_t SIZE = 256; 3057 char buffer[SIZE]; 3058 String8 result; 3059 3060 PlaybackThread::dumpInternals(fd, args); 3061 3062 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3063 result.append(buffer); 3064 write(fd, result.string(), result.size()); 3065 3066 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3067 FastMixerDumpState copy = mFastMixerDumpState; 3068 copy.dump(fd); 3069 3070#ifdef STATE_QUEUE_DUMP 3071 // Similar for state queue 3072 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3073 observerCopy.dump(fd); 3074 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3075 mutatorCopy.dump(fd); 3076#endif 3077 3078#ifdef TEE_SINK 3079 // Write the tee output to a .wav file 3080 dumpTee(fd, mTeeSource, mId); 3081#endif 3082 3083#ifdef AUDIO_WATCHDOG 3084 if (mAudioWatchdog != 0) { 3085 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3086 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3087 wdCopy.dump(fd); 3088 } 3089#endif 3090} 3091 3092uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3093{ 3094 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3095} 3096 3097uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3098{ 3099 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3100} 3101 3102void AudioFlinger::MixerThread::cacheParameters_l() 3103{ 3104 PlaybackThread::cacheParameters_l(); 3105 3106 // FIXME: Relaxed timing because of a certain device that can't meet latency 3107 // Should be reduced to 2x after the vendor fixes the driver issue 3108 // increase threshold again due to low power audio mode. The way this warning 3109 // threshold is calculated and its usefulness should be reconsidered anyway. 3110 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3111} 3112 3113// ---------------------------------------------------------------------------- 3114 3115AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3116 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3117 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3118 // mLeftVolFloat, mRightVolFloat 3119{ 3120} 3121 3122AudioFlinger::DirectOutputThread::~DirectOutputThread() 3123{ 3124} 3125 3126AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3127 Vector< sp<Track> > *tracksToRemove 3128) 3129{ 3130 size_t count = mActiveTracks.size(); 3131 mixer_state mixerStatus = MIXER_IDLE; 3132 3133 // find out which tracks need to be processed 3134 for (size_t i = 0; i < count; i++) { 3135 sp<Track> t = mActiveTracks[i].promote(); 3136 // The track died recently 3137 if (t == 0) { 3138 continue; 3139 } 3140 3141 Track* const track = t.get(); 3142 audio_track_cblk_t* cblk = track->cblk(); 3143 3144 // The first time a track is added we wait 3145 // for all its buffers to be filled before processing it 3146 uint32_t minFrames; 3147 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3148 minFrames = mNormalFrameCount; 3149 } else { 3150 minFrames = 1; 3151 } 3152 if ((track->framesReady() >= minFrames) && track->isReady() && 3153 !track->isPaused() && !track->isTerminated()) 3154 { 3155 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3156 3157 if (track->mFillingUpStatus == Track::FS_FILLED) { 3158 track->mFillingUpStatus = Track::FS_ACTIVE; 3159 mLeftVolFloat = mRightVolFloat = 0; 3160 if (track->mState == TrackBase::RESUMING) { 3161 track->mState = TrackBase::ACTIVE; 3162 } 3163 } 3164 3165 // compute volume for this track 3166 float left, right; 3167 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3168 left = right = 0; 3169 if (track->isPausing()) { 3170 track->setPaused(); 3171 } 3172 } else { 3173 float typeVolume = mStreamTypes[track->streamType()].volume; 3174 float v = mMasterVolume * typeVolume; 3175 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3176 float v_clamped = v * (vlr & 0xFFFF); 3177 if (v_clamped > MAX_GAIN) { 3178 v_clamped = MAX_GAIN; 3179 } 3180 left = v_clamped/MAX_GAIN; 3181 v_clamped = v * (vlr >> 16); 3182 if (v_clamped > MAX_GAIN) { 3183 v_clamped = MAX_GAIN; 3184 } 3185 right = v_clamped/MAX_GAIN; 3186 } 3187 // Only consider last track started for volume and mixer state control. 3188 // This is the last entry in mActiveTracks unless a track underruns. 3189 // As we only care about the transition phase between two tracks on a 3190 // direct output, it is not a problem to ignore the underrun case. 3191 if (i == (count - 1)) { 3192 if (left != mLeftVolFloat || right != mRightVolFloat) { 3193 mLeftVolFloat = left; 3194 mRightVolFloat = right; 3195 3196 // Convert volumes from float to 8.24 3197 uint32_t vl = (uint32_t)(left * (1 << 24)); 3198 uint32_t vr = (uint32_t)(right * (1 << 24)); 3199 3200 // Delegate volume control to effect in track effect chain if needed 3201 // only one effect chain can be present on DirectOutputThread, so if 3202 // there is one, the track is connected to it 3203 if (!mEffectChains.isEmpty()) { 3204 // Do not ramp volume if volume is controlled by effect 3205 mEffectChains[0]->setVolume_l(&vl, &vr); 3206 left = (float)vl / (1 << 24); 3207 right = (float)vr / (1 << 24); 3208 } 3209 mOutput->stream->set_volume(mOutput->stream, left, right); 3210 } 3211 3212 // reset retry count 3213 track->mRetryCount = kMaxTrackRetriesDirect; 3214 mActiveTrack = t; 3215 mixerStatus = MIXER_TRACKS_READY; 3216 } 3217 } else { 3218 // clear effect chain input buffer if the last active track started underruns 3219 // to avoid sending previous audio buffer again to effects 3220 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3221 mEffectChains[0]->clearInputBuffer(); 3222 } 3223 3224 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3225 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3226 track->isStopped() || track->isPaused()) { 3227 // We have consumed all the buffers of this track. 3228 // Remove it from the list of active tracks. 3229 // TODO: implement behavior for compressed audio 3230 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3231 size_t framesWritten = mBytesWritten / mFrameSize; 3232 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3233 if (track->isStopped()) { 3234 track->reset(); 3235 } 3236 tracksToRemove->add(track); 3237 } 3238 } else { 3239 // No buffers for this track. Give it a few chances to 3240 // fill a buffer, then remove it from active list. 3241 // Only consider last track started for mixer state control 3242 if (--(track->mRetryCount) <= 0) { 3243 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3244 tracksToRemove->add(track); 3245 } else if (i == (count -1)){ 3246 mixerStatus = MIXER_TRACKS_ENABLED; 3247 } 3248 } 3249 } 3250 } 3251 3252 // remove all the tracks that need to be... 3253 count = tracksToRemove->size(); 3254 if (CC_UNLIKELY(count)) { 3255 for (size_t i = 0 ; i < count ; i++) { 3256 const sp<Track>& track = tracksToRemove->itemAt(i); 3257 mActiveTracks.remove(track); 3258 if (!mEffectChains.isEmpty()) { 3259 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3260 track->sessionId()); 3261 mEffectChains[0]->decActiveTrackCnt(); 3262 } 3263 if (track->isTerminated()) { 3264 removeTrack_l(track); 3265 } 3266 } 3267 } 3268 3269 return mixerStatus; 3270} 3271 3272void AudioFlinger::DirectOutputThread::threadLoop_mix() 3273{ 3274 AudioBufferProvider::Buffer buffer; 3275 size_t frameCount = mFrameCount; 3276 int8_t *curBuf = (int8_t *)mMixBuffer; 3277 // output audio to hardware 3278 while (frameCount) { 3279 buffer.frameCount = frameCount; 3280 mActiveTrack->getNextBuffer(&buffer); 3281 if (CC_UNLIKELY(buffer.raw == NULL)) { 3282 memset(curBuf, 0, frameCount * mFrameSize); 3283 break; 3284 } 3285 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3286 frameCount -= buffer.frameCount; 3287 curBuf += buffer.frameCount * mFrameSize; 3288 mActiveTrack->releaseBuffer(&buffer); 3289 } 3290 sleepTime = 0; 3291 standbyTime = systemTime() + standbyDelay; 3292 mActiveTrack.clear(); 3293 3294} 3295 3296void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3297{ 3298 if (sleepTime == 0) { 3299 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3300 sleepTime = activeSleepTime; 3301 } else { 3302 sleepTime = idleSleepTime; 3303 } 3304 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3305 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3306 sleepTime = 0; 3307 } 3308} 3309 3310// getTrackName_l() must be called with ThreadBase::mLock held 3311int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3312 int sessionId) 3313{ 3314 return 0; 3315} 3316 3317// deleteTrackName_l() must be called with ThreadBase::mLock held 3318void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3319{ 3320} 3321 3322// checkForNewParameters_l() must be called with ThreadBase::mLock held 3323bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3324{ 3325 bool reconfig = false; 3326 3327 while (!mNewParameters.isEmpty()) { 3328 status_t status = NO_ERROR; 3329 String8 keyValuePair = mNewParameters[0]; 3330 AudioParameter param = AudioParameter(keyValuePair); 3331 int value; 3332 3333 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3334 // do not accept frame count changes if tracks are open as the track buffer 3335 // size depends on frame count and correct behavior would not be garantied 3336 // if frame count is changed after track creation 3337 if (!mTracks.isEmpty()) { 3338 status = INVALID_OPERATION; 3339 } else { 3340 reconfig = true; 3341 } 3342 } 3343 if (status == NO_ERROR) { 3344 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3345 keyValuePair.string()); 3346 if (!mStandby && status == INVALID_OPERATION) { 3347 mOutput->stream->common.standby(&mOutput->stream->common); 3348 mStandby = true; 3349 mBytesWritten = 0; 3350 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3351 keyValuePair.string()); 3352 } 3353 if (status == NO_ERROR && reconfig) { 3354 readOutputParameters(); 3355 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3356 } 3357 } 3358 3359 mNewParameters.removeAt(0); 3360 3361 mParamStatus = status; 3362 mParamCond.signal(); 3363 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3364 // already timed out waiting for the status and will never signal the condition. 3365 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3366 } 3367 return reconfig; 3368} 3369 3370uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3371{ 3372 uint32_t time; 3373 if (audio_is_linear_pcm(mFormat)) { 3374 time = PlaybackThread::activeSleepTimeUs(); 3375 } else { 3376 time = 10000; 3377 } 3378 return time; 3379} 3380 3381uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3382{ 3383 uint32_t time; 3384 if (audio_is_linear_pcm(mFormat)) { 3385 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3386 } else { 3387 time = 10000; 3388 } 3389 return time; 3390} 3391 3392uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3393{ 3394 uint32_t time; 3395 if (audio_is_linear_pcm(mFormat)) { 3396 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3397 } else { 3398 time = 10000; 3399 } 3400 return time; 3401} 3402 3403void AudioFlinger::DirectOutputThread::cacheParameters_l() 3404{ 3405 PlaybackThread::cacheParameters_l(); 3406 3407 // use shorter standby delay as on normal output to release 3408 // hardware resources as soon as possible 3409 standbyDelay = microseconds(activeSleepTime*2); 3410} 3411 3412// ---------------------------------------------------------------------------- 3413 3414AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3415 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3416 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3417 DUPLICATING), 3418 mWaitTimeMs(UINT_MAX) 3419{ 3420 addOutputTrack(mainThread); 3421} 3422 3423AudioFlinger::DuplicatingThread::~DuplicatingThread() 3424{ 3425 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3426 mOutputTracks[i]->destroy(); 3427 } 3428} 3429 3430void AudioFlinger::DuplicatingThread::threadLoop_mix() 3431{ 3432 // mix buffers... 3433 if (outputsReady(outputTracks)) { 3434 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3435 } else { 3436 memset(mMixBuffer, 0, mixBufferSize); 3437 } 3438 sleepTime = 0; 3439 writeFrames = mNormalFrameCount; 3440 standbyTime = systemTime() + standbyDelay; 3441} 3442 3443void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3444{ 3445 if (sleepTime == 0) { 3446 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3447 sleepTime = activeSleepTime; 3448 } else { 3449 sleepTime = idleSleepTime; 3450 } 3451 } else if (mBytesWritten != 0) { 3452 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3453 writeFrames = mNormalFrameCount; 3454 memset(mMixBuffer, 0, mixBufferSize); 3455 } else { 3456 // flush remaining overflow buffers in output tracks 3457 writeFrames = 0; 3458 } 3459 sleepTime = 0; 3460 } 3461} 3462 3463void AudioFlinger::DuplicatingThread::threadLoop_write() 3464{ 3465 for (size_t i = 0; i < outputTracks.size(); i++) { 3466 outputTracks[i]->write(mMixBuffer, writeFrames); 3467 } 3468 mBytesWritten += mixBufferSize; 3469} 3470 3471void AudioFlinger::DuplicatingThread::threadLoop_standby() 3472{ 3473 // DuplicatingThread implements standby by stopping all tracks 3474 for (size_t i = 0; i < outputTracks.size(); i++) { 3475 outputTracks[i]->stop(); 3476 } 3477} 3478 3479void AudioFlinger::DuplicatingThread::saveOutputTracks() 3480{ 3481 outputTracks = mOutputTracks; 3482} 3483 3484void AudioFlinger::DuplicatingThread::clearOutputTracks() 3485{ 3486 outputTracks.clear(); 3487} 3488 3489void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3490{ 3491 Mutex::Autolock _l(mLock); 3492 // FIXME explain this formula 3493 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3494 OutputTrack *outputTrack = new OutputTrack(thread, 3495 this, 3496 mSampleRate, 3497 mFormat, 3498 mChannelMask, 3499 frameCount); 3500 if (outputTrack->cblk() != NULL) { 3501 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3502 mOutputTracks.add(outputTrack); 3503 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3504 updateWaitTime_l(); 3505 } 3506} 3507 3508void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3509{ 3510 Mutex::Autolock _l(mLock); 3511 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3512 if (mOutputTracks[i]->thread() == thread) { 3513 mOutputTracks[i]->destroy(); 3514 mOutputTracks.removeAt(i); 3515 updateWaitTime_l(); 3516 return; 3517 } 3518 } 3519 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3520} 3521 3522// caller must hold mLock 3523void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3524{ 3525 mWaitTimeMs = UINT_MAX; 3526 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3527 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3528 if (strong != 0) { 3529 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3530 if (waitTimeMs < mWaitTimeMs) { 3531 mWaitTimeMs = waitTimeMs; 3532 } 3533 } 3534 } 3535} 3536 3537 3538bool AudioFlinger::DuplicatingThread::outputsReady( 3539 const SortedVector< sp<OutputTrack> > &outputTracks) 3540{ 3541 for (size_t i = 0; i < outputTracks.size(); i++) { 3542 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3543 if (thread == 0) { 3544 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3545 outputTracks[i].get()); 3546 return false; 3547 } 3548 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3549 // see note at standby() declaration 3550 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3551 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3552 thread.get()); 3553 return false; 3554 } 3555 } 3556 return true; 3557} 3558 3559uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3560{ 3561 return (mWaitTimeMs * 1000) / 2; 3562} 3563 3564void AudioFlinger::DuplicatingThread::cacheParameters_l() 3565{ 3566 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3567 updateWaitTime_l(); 3568 3569 MixerThread::cacheParameters_l(); 3570} 3571 3572// ---------------------------------------------------------------------------- 3573// Record 3574// ---------------------------------------------------------------------------- 3575 3576AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3577 AudioStreamIn *input, 3578 uint32_t sampleRate, 3579 audio_channel_mask_t channelMask, 3580 audio_io_handle_t id, 3581 audio_devices_t outDevice, 3582 audio_devices_t inDevice 3583#ifdef TEE_SINK 3584 , const sp<NBAIO_Sink>& teeSink 3585#endif 3586 ) : 3587 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3588 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3589 // mRsmpInIndex and mInputBytes set by readInputParameters() 3590 mReqChannelCount(popcount(channelMask)), 3591 mReqSampleRate(sampleRate) 3592 // mBytesRead is only meaningful while active, and so is cleared in start() 3593 // (but might be better to also clear here for dump?) 3594#ifdef TEE_SINK 3595 , mTeeSink(teeSink) 3596#endif 3597{ 3598 snprintf(mName, kNameLength, "AudioIn_%X", id); 3599 3600 readInputParameters(); 3601 3602} 3603 3604 3605AudioFlinger::RecordThread::~RecordThread() 3606{ 3607 delete[] mRsmpInBuffer; 3608 delete mResampler; 3609 delete[] mRsmpOutBuffer; 3610} 3611 3612void AudioFlinger::RecordThread::onFirstRef() 3613{ 3614 run(mName, PRIORITY_URGENT_AUDIO); 3615} 3616 3617status_t AudioFlinger::RecordThread::readyToRun() 3618{ 3619 status_t status = initCheck(); 3620 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3621 return status; 3622} 3623 3624bool AudioFlinger::RecordThread::threadLoop() 3625{ 3626 AudioBufferProvider::Buffer buffer; 3627 sp<RecordTrack> activeTrack; 3628 Vector< sp<EffectChain> > effectChains; 3629 3630 nsecs_t lastWarning = 0; 3631 3632 inputStandBy(); 3633 acquireWakeLock(); 3634 3635 // used to verify we've read at least once before evaluating how many bytes were read 3636 bool readOnce = false; 3637 3638 // start recording 3639 while (!exitPending()) { 3640 3641 processConfigEvents(); 3642 3643 { // scope for mLock 3644 Mutex::Autolock _l(mLock); 3645 checkForNewParameters_l(); 3646 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3647 standby(); 3648 3649 if (exitPending()) { 3650 break; 3651 } 3652 3653 releaseWakeLock_l(); 3654 ALOGV("RecordThread: loop stopping"); 3655 // go to sleep 3656 mWaitWorkCV.wait(mLock); 3657 ALOGV("RecordThread: loop starting"); 3658 acquireWakeLock_l(); 3659 continue; 3660 } 3661 if (mActiveTrack != 0) { 3662 if (mActiveTrack->mState == TrackBase::PAUSING) { 3663 standby(); 3664 mActiveTrack.clear(); 3665 mStartStopCond.broadcast(); 3666 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3667 if (mReqChannelCount != mActiveTrack->channelCount()) { 3668 mActiveTrack.clear(); 3669 mStartStopCond.broadcast(); 3670 } else if (readOnce) { 3671 // record start succeeds only if first read from audio input 3672 // succeeds 3673 if (mBytesRead >= 0) { 3674 mActiveTrack->mState = TrackBase::ACTIVE; 3675 } else { 3676 mActiveTrack.clear(); 3677 } 3678 mStartStopCond.broadcast(); 3679 } 3680 mStandby = false; 3681 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3682 removeTrack_l(mActiveTrack); 3683 mActiveTrack.clear(); 3684 } 3685 } 3686 lockEffectChains_l(effectChains); 3687 } 3688 3689 if (mActiveTrack != 0) { 3690 if (mActiveTrack->mState != TrackBase::ACTIVE && 3691 mActiveTrack->mState != TrackBase::RESUMING) { 3692 unlockEffectChains(effectChains); 3693 usleep(kRecordThreadSleepUs); 3694 continue; 3695 } 3696 for (size_t i = 0; i < effectChains.size(); i ++) { 3697 effectChains[i]->process_l(); 3698 } 3699 3700 buffer.frameCount = mFrameCount; 3701 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3702 readOnce = true; 3703 size_t framesOut = buffer.frameCount; 3704 if (mResampler == NULL) { 3705 // no resampling 3706 while (framesOut) { 3707 size_t framesIn = mFrameCount - mRsmpInIndex; 3708 if (framesIn) { 3709 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3710 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3711 mActiveTrack->mFrameSize; 3712 if (framesIn > framesOut) 3713 framesIn = framesOut; 3714 mRsmpInIndex += framesIn; 3715 framesOut -= framesIn; 3716 if (mChannelCount == mReqChannelCount || 3717 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3718 memcpy(dst, src, framesIn * mFrameSize); 3719 } else { 3720 if (mChannelCount == 1) { 3721 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3722 (int16_t *)src, framesIn); 3723 } else { 3724 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3725 (int16_t *)src, framesIn); 3726 } 3727 } 3728 } 3729 if (framesOut && mFrameCount == mRsmpInIndex) { 3730 void *readInto; 3731 if (framesOut == mFrameCount && 3732 (mChannelCount == mReqChannelCount || 3733 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3734 readInto = buffer.raw; 3735 framesOut = 0; 3736 } else { 3737 readInto = mRsmpInBuffer; 3738 mRsmpInIndex = 0; 3739 } 3740 mBytesRead = mInput->stream->read(mInput->stream, readInto, 3741 mInputBytes); 3742 if (mBytesRead <= 0) { 3743 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3744 { 3745 ALOGE("Error reading audio input"); 3746 // Force input into standby so that it tries to 3747 // recover at next read attempt 3748 inputStandBy(); 3749 usleep(kRecordThreadSleepUs); 3750 } 3751 mRsmpInIndex = mFrameCount; 3752 framesOut = 0; 3753 buffer.frameCount = 0; 3754 } 3755#ifdef TEE_SINK 3756 else if (mTeeSink != 0) { 3757 (void) mTeeSink->write(readInto, 3758 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3759 } 3760#endif 3761 } 3762 } 3763 } else { 3764 // resampling 3765 3766 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3767 // alter output frame count as if we were expecting stereo samples 3768 if (mChannelCount == 1 && mReqChannelCount == 1) { 3769 framesOut >>= 1; 3770 } 3771 mResampler->resample(mRsmpOutBuffer, framesOut, 3772 this /* AudioBufferProvider* */); 3773 // ditherAndClamp() works as long as all buffers returned by 3774 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3775 if (mChannelCount == 2 && mReqChannelCount == 1) { 3776 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3777 // the resampler always outputs stereo samples: 3778 // do post stereo to mono conversion 3779 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3780 framesOut); 3781 } else { 3782 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3783 } 3784 3785 } 3786 if (mFramestoDrop == 0) { 3787 mActiveTrack->releaseBuffer(&buffer); 3788 } else { 3789 if (mFramestoDrop > 0) { 3790 mFramestoDrop -= buffer.frameCount; 3791 if (mFramestoDrop <= 0) { 3792 clearSyncStartEvent(); 3793 } 3794 } else { 3795 mFramestoDrop += buffer.frameCount; 3796 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3797 mSyncStartEvent->isCancelled()) { 3798 ALOGW("Synced record %s, session %d, trigger session %d", 3799 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3800 mActiveTrack->sessionId(), 3801 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3802 clearSyncStartEvent(); 3803 } 3804 } 3805 } 3806 mActiveTrack->clearOverflow(); 3807 } 3808 // client isn't retrieving buffers fast enough 3809 else { 3810 if (!mActiveTrack->setOverflow()) { 3811 nsecs_t now = systemTime(); 3812 if ((now - lastWarning) > kWarningThrottleNs) { 3813 ALOGW("RecordThread: buffer overflow"); 3814 lastWarning = now; 3815 } 3816 } 3817 // Release the processor for a while before asking for a new buffer. 3818 // This will give the application more chance to read from the buffer and 3819 // clear the overflow. 3820 usleep(kRecordThreadSleepUs); 3821 } 3822 } 3823 // enable changes in effect chain 3824 unlockEffectChains(effectChains); 3825 effectChains.clear(); 3826 } 3827 3828 standby(); 3829 3830 { 3831 Mutex::Autolock _l(mLock); 3832 mActiveTrack.clear(); 3833 mStartStopCond.broadcast(); 3834 } 3835 3836 releaseWakeLock(); 3837 3838 ALOGV("RecordThread %p exiting", this); 3839 return false; 3840} 3841 3842void AudioFlinger::RecordThread::standby() 3843{ 3844 if (!mStandby) { 3845 inputStandBy(); 3846 mStandby = true; 3847 } 3848} 3849 3850void AudioFlinger::RecordThread::inputStandBy() 3851{ 3852 mInput->stream->common.standby(&mInput->stream->common); 3853} 3854 3855sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3856 const sp<AudioFlinger::Client>& client, 3857 uint32_t sampleRate, 3858 audio_format_t format, 3859 audio_channel_mask_t channelMask, 3860 size_t frameCount, 3861 int sessionId, 3862 IAudioFlinger::track_flags_t flags, 3863 pid_t tid, 3864 status_t *status) 3865{ 3866 sp<RecordTrack> track; 3867 status_t lStatus; 3868 3869 lStatus = initCheck(); 3870 if (lStatus != NO_ERROR) { 3871 ALOGE("Audio driver not initialized."); 3872 goto Exit; 3873 } 3874 3875 // FIXME use flags and tid similar to createTrack_l() 3876 3877 { // scope for mLock 3878 Mutex::Autolock _l(mLock); 3879 3880 track = new RecordTrack(this, client, sampleRate, 3881 format, channelMask, frameCount, sessionId); 3882 3883 if (track->getCblk() == 0) { 3884 lStatus = NO_MEMORY; 3885 goto Exit; 3886 } 3887 mTracks.add(track); 3888 3889 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3890 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3891 mAudioFlinger->btNrecIsOff(); 3892 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3893 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3894 } 3895 lStatus = NO_ERROR; 3896 3897Exit: 3898 if (status) { 3899 *status = lStatus; 3900 } 3901 return track; 3902} 3903 3904status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3905 AudioSystem::sync_event_t event, 3906 int triggerSession) 3907{ 3908 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3909 sp<ThreadBase> strongMe = this; 3910 status_t status = NO_ERROR; 3911 3912 if (event == AudioSystem::SYNC_EVENT_NONE) { 3913 clearSyncStartEvent(); 3914 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3915 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3916 triggerSession, 3917 recordTrack->sessionId(), 3918 syncStartEventCallback, 3919 this); 3920 // Sync event can be cancelled by the trigger session if the track is not in a 3921 // compatible state in which case we start record immediately 3922 if (mSyncStartEvent->isCancelled()) { 3923 clearSyncStartEvent(); 3924 } else { 3925 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3926 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3927 } 3928 } 3929 3930 { 3931 AutoMutex lock(mLock); 3932 if (mActiveTrack != 0) { 3933 if (recordTrack != mActiveTrack.get()) { 3934 status = -EBUSY; 3935 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3936 mActiveTrack->mState = TrackBase::ACTIVE; 3937 } 3938 return status; 3939 } 3940 3941 recordTrack->mState = TrackBase::IDLE; 3942 mActiveTrack = recordTrack; 3943 mLock.unlock(); 3944 status_t status = AudioSystem::startInput(mId); 3945 mLock.lock(); 3946 if (status != NO_ERROR) { 3947 mActiveTrack.clear(); 3948 clearSyncStartEvent(); 3949 return status; 3950 } 3951 mRsmpInIndex = mFrameCount; 3952 mBytesRead = 0; 3953 if (mResampler != NULL) { 3954 mResampler->reset(); 3955 } 3956 mActiveTrack->mState = TrackBase::RESUMING; 3957 // signal thread to start 3958 ALOGV("Signal record thread"); 3959 mWaitWorkCV.broadcast(); 3960 // do not wait for mStartStopCond if exiting 3961 if (exitPending()) { 3962 mActiveTrack.clear(); 3963 status = INVALID_OPERATION; 3964 goto startError; 3965 } 3966 mStartStopCond.wait(mLock); 3967 if (mActiveTrack == 0) { 3968 ALOGV("Record failed to start"); 3969 status = BAD_VALUE; 3970 goto startError; 3971 } 3972 ALOGV("Record started OK"); 3973 return status; 3974 } 3975 3976startError: 3977 AudioSystem::stopInput(mId); 3978 clearSyncStartEvent(); 3979 return status; 3980} 3981 3982void AudioFlinger::RecordThread::clearSyncStartEvent() 3983{ 3984 if (mSyncStartEvent != 0) { 3985 mSyncStartEvent->cancel(); 3986 } 3987 mSyncStartEvent.clear(); 3988 mFramestoDrop = 0; 3989} 3990 3991void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3992{ 3993 sp<SyncEvent> strongEvent = event.promote(); 3994 3995 if (strongEvent != 0) { 3996 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3997 me->handleSyncStartEvent(strongEvent); 3998 } 3999} 4000 4001void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4002{ 4003 if (event == mSyncStartEvent) { 4004 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4005 // from audio HAL 4006 mFramestoDrop = mFrameCount * 2; 4007 } 4008} 4009 4010bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 4011 ALOGV("RecordThread::stop"); 4012 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4013 return false; 4014 } 4015 recordTrack->mState = TrackBase::PAUSING; 4016 // do not wait for mStartStopCond if exiting 4017 if (exitPending()) { 4018 return true; 4019 } 4020 mStartStopCond.wait(mLock); 4021 // if we have been restarted, recordTrack == mActiveTrack.get() here 4022 if (exitPending() || recordTrack != mActiveTrack.get()) { 4023 ALOGV("Record stopped OK"); 4024 return true; 4025 } 4026 return false; 4027} 4028 4029bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4030{ 4031 return false; 4032} 4033 4034status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4035{ 4036#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4037 if (!isValidSyncEvent(event)) { 4038 return BAD_VALUE; 4039 } 4040 4041 int eventSession = event->triggerSession(); 4042 status_t ret = NAME_NOT_FOUND; 4043 4044 Mutex::Autolock _l(mLock); 4045 4046 for (size_t i = 0; i < mTracks.size(); i++) { 4047 sp<RecordTrack> track = mTracks[i]; 4048 if (eventSession == track->sessionId()) { 4049 (void) track->setSyncEvent(event); 4050 ret = NO_ERROR; 4051 } 4052 } 4053 return ret; 4054#else 4055 return BAD_VALUE; 4056#endif 4057} 4058 4059// destroyTrack_l() must be called with ThreadBase::mLock held 4060void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4061{ 4062 track->mState = TrackBase::TERMINATED; 4063 // active tracks are removed by threadLoop() 4064 if (mActiveTrack != track) { 4065 removeTrack_l(track); 4066 } 4067} 4068 4069void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4070{ 4071 mTracks.remove(track); 4072 // need anything related to effects here? 4073} 4074 4075void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4076{ 4077 dumpInternals(fd, args); 4078 dumpTracks(fd, args); 4079 dumpEffectChains(fd, args); 4080} 4081 4082void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4083{ 4084 const size_t SIZE = 256; 4085 char buffer[SIZE]; 4086 String8 result; 4087 4088 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4089 result.append(buffer); 4090 4091 if (mActiveTrack != 0) { 4092 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4093 result.append(buffer); 4094 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4095 result.append(buffer); 4096 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4097 result.append(buffer); 4098 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4099 result.append(buffer); 4100 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4101 result.append(buffer); 4102 } else { 4103 result.append("No active record client\n"); 4104 } 4105 4106 write(fd, result.string(), result.size()); 4107 4108 dumpBase(fd, args); 4109} 4110 4111void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4112{ 4113 const size_t SIZE = 256; 4114 char buffer[SIZE]; 4115 String8 result; 4116 4117 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4118 result.append(buffer); 4119 RecordTrack::appendDumpHeader(result); 4120 for (size_t i = 0; i < mTracks.size(); ++i) { 4121 sp<RecordTrack> track = mTracks[i]; 4122 if (track != 0) { 4123 track->dump(buffer, SIZE); 4124 result.append(buffer); 4125 } 4126 } 4127 4128 if (mActiveTrack != 0) { 4129 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4130 result.append(buffer); 4131 RecordTrack::appendDumpHeader(result); 4132 mActiveTrack->dump(buffer, SIZE); 4133 result.append(buffer); 4134 4135 } 4136 write(fd, result.string(), result.size()); 4137} 4138 4139// AudioBufferProvider interface 4140status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4141{ 4142 size_t framesReq = buffer->frameCount; 4143 size_t framesReady = mFrameCount - mRsmpInIndex; 4144 int channelCount; 4145 4146 if (framesReady == 0) { 4147 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4148 if (mBytesRead <= 0) { 4149 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4150 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4151 // Force input into standby so that it tries to 4152 // recover at next read attempt 4153 inputStandBy(); 4154 usleep(kRecordThreadSleepUs); 4155 } 4156 buffer->raw = NULL; 4157 buffer->frameCount = 0; 4158 return NOT_ENOUGH_DATA; 4159 } 4160 mRsmpInIndex = 0; 4161 framesReady = mFrameCount; 4162 } 4163 4164 if (framesReq > framesReady) { 4165 framesReq = framesReady; 4166 } 4167 4168 if (mChannelCount == 1 && mReqChannelCount == 2) { 4169 channelCount = 1; 4170 } else { 4171 channelCount = 2; 4172 } 4173 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4174 buffer->frameCount = framesReq; 4175 return NO_ERROR; 4176} 4177 4178// AudioBufferProvider interface 4179void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4180{ 4181 mRsmpInIndex += buffer->frameCount; 4182 buffer->frameCount = 0; 4183} 4184 4185bool AudioFlinger::RecordThread::checkForNewParameters_l() 4186{ 4187 bool reconfig = false; 4188 4189 while (!mNewParameters.isEmpty()) { 4190 status_t status = NO_ERROR; 4191 String8 keyValuePair = mNewParameters[0]; 4192 AudioParameter param = AudioParameter(keyValuePair); 4193 int value; 4194 audio_format_t reqFormat = mFormat; 4195 uint32_t reqSamplingRate = mReqSampleRate; 4196 uint32_t reqChannelCount = mReqChannelCount; 4197 4198 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4199 reqSamplingRate = value; 4200 reconfig = true; 4201 } 4202 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4203 reqFormat = (audio_format_t) value; 4204 reconfig = true; 4205 } 4206 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4207 reqChannelCount = popcount(value); 4208 reconfig = true; 4209 } 4210 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4211 // do not accept frame count changes if tracks are open as the track buffer 4212 // size depends on frame count and correct behavior would not be guaranteed 4213 // if frame count is changed after track creation 4214 if (mActiveTrack != 0) { 4215 status = INVALID_OPERATION; 4216 } else { 4217 reconfig = true; 4218 } 4219 } 4220 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4221 // forward device change to effects that have requested to be 4222 // aware of attached audio device. 4223 for (size_t i = 0; i < mEffectChains.size(); i++) { 4224 mEffectChains[i]->setDevice_l(value); 4225 } 4226 4227 // store input device and output device but do not forward output device to audio HAL. 4228 // Note that status is ignored by the caller for output device 4229 // (see AudioFlinger::setParameters() 4230 if (audio_is_output_devices(value)) { 4231 mOutDevice = value; 4232 status = BAD_VALUE; 4233 } else { 4234 mInDevice = value; 4235 // disable AEC and NS if the device is a BT SCO headset supporting those 4236 // pre processings 4237 if (mTracks.size() > 0) { 4238 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4239 mAudioFlinger->btNrecIsOff(); 4240 for (size_t i = 0; i < mTracks.size(); i++) { 4241 sp<RecordTrack> track = mTracks[i]; 4242 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4243 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4244 } 4245 } 4246 } 4247 } 4248 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4249 mAudioSource != (audio_source_t)value) { 4250 // forward device change to effects that have requested to be 4251 // aware of attached audio device. 4252 for (size_t i = 0; i < mEffectChains.size(); i++) { 4253 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4254 } 4255 mAudioSource = (audio_source_t)value; 4256 } 4257 if (status == NO_ERROR) { 4258 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4259 keyValuePair.string()); 4260 if (status == INVALID_OPERATION) { 4261 inputStandBy(); 4262 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4263 keyValuePair.string()); 4264 } 4265 if (reconfig) { 4266 if (status == BAD_VALUE && 4267 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4268 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4269 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4270 <= (2 * reqSamplingRate)) && 4271 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4272 <= FCC_2 && 4273 (reqChannelCount <= FCC_2)) { 4274 status = NO_ERROR; 4275 } 4276 if (status == NO_ERROR) { 4277 readInputParameters(); 4278 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4279 } 4280 } 4281 } 4282 4283 mNewParameters.removeAt(0); 4284 4285 mParamStatus = status; 4286 mParamCond.signal(); 4287 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4288 // already timed out waiting for the status and will never signal the condition. 4289 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4290 } 4291 return reconfig; 4292} 4293 4294String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4295{ 4296 char *s; 4297 String8 out_s8 = String8(); 4298 4299 Mutex::Autolock _l(mLock); 4300 if (initCheck() != NO_ERROR) { 4301 return out_s8; 4302 } 4303 4304 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4305 out_s8 = String8(s); 4306 free(s); 4307 return out_s8; 4308} 4309 4310void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4311 AudioSystem::OutputDescriptor desc; 4312 void *param2 = NULL; 4313 4314 switch (event) { 4315 case AudioSystem::INPUT_OPENED: 4316 case AudioSystem::INPUT_CONFIG_CHANGED: 4317 desc.channels = mChannelMask; 4318 desc.samplingRate = mSampleRate; 4319 desc.format = mFormat; 4320 desc.frameCount = mFrameCount; 4321 desc.latency = 0; 4322 param2 = &desc; 4323 break; 4324 4325 case AudioSystem::INPUT_CLOSED: 4326 default: 4327 break; 4328 } 4329 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4330} 4331 4332void AudioFlinger::RecordThread::readInputParameters() 4333{ 4334 delete mRsmpInBuffer; 4335 // mRsmpInBuffer is always assigned a new[] below 4336 delete mRsmpOutBuffer; 4337 mRsmpOutBuffer = NULL; 4338 delete mResampler; 4339 mResampler = NULL; 4340 4341 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4342 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4343 mChannelCount = (uint16_t)popcount(mChannelMask); 4344 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4345 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4346 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4347 mFrameCount = mInputBytes / mFrameSize; 4348 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4349 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4350 4351 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4352 { 4353 int channelCount; 4354 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4355 // stereo to mono post process as the resampler always outputs stereo. 4356 if (mChannelCount == 1 && mReqChannelCount == 2) { 4357 channelCount = 1; 4358 } else { 4359 channelCount = 2; 4360 } 4361 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4362 mResampler->setSampleRate(mSampleRate); 4363 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4364 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4365 4366 // optmization: if mono to mono, alter input frame count as if we were inputing 4367 // stereo samples 4368 if (mChannelCount == 1 && mReqChannelCount == 1) { 4369 mFrameCount >>= 1; 4370 } 4371 4372 } 4373 mRsmpInIndex = mFrameCount; 4374} 4375 4376unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4377{ 4378 Mutex::Autolock _l(mLock); 4379 if (initCheck() != NO_ERROR) { 4380 return 0; 4381 } 4382 4383 return mInput->stream->get_input_frames_lost(mInput->stream); 4384} 4385 4386uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4387{ 4388 Mutex::Autolock _l(mLock); 4389 uint32_t result = 0; 4390 if (getEffectChain_l(sessionId) != 0) { 4391 result = EFFECT_SESSION; 4392 } 4393 4394 for (size_t i = 0; i < mTracks.size(); ++i) { 4395 if (sessionId == mTracks[i]->sessionId()) { 4396 result |= TRACK_SESSION; 4397 break; 4398 } 4399 } 4400 4401 return result; 4402} 4403 4404KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4405{ 4406 KeyedVector<int, bool> ids; 4407 Mutex::Autolock _l(mLock); 4408 for (size_t j = 0; j < mTracks.size(); ++j) { 4409 sp<RecordThread::RecordTrack> track = mTracks[j]; 4410 int sessionId = track->sessionId(); 4411 if (ids.indexOfKey(sessionId) < 0) { 4412 ids.add(sessionId, true); 4413 } 4414 } 4415 return ids; 4416} 4417 4418AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4419{ 4420 Mutex::Autolock _l(mLock); 4421 AudioStreamIn *input = mInput; 4422 mInput = NULL; 4423 return input; 4424} 4425 4426// this method must always be called either with ThreadBase mLock held or inside the thread loop 4427audio_stream_t* AudioFlinger::RecordThread::stream() const 4428{ 4429 if (mInput == NULL) { 4430 return NULL; 4431 } 4432 return &mInput->stream->common; 4433} 4434 4435status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4436{ 4437 // only one chain per input thread 4438 if (mEffectChains.size() != 0) { 4439 return INVALID_OPERATION; 4440 } 4441 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4442 4443 chain->setInBuffer(NULL); 4444 chain->setOutBuffer(NULL); 4445 4446 checkSuspendOnAddEffectChain_l(chain); 4447 4448 mEffectChains.add(chain); 4449 4450 return NO_ERROR; 4451} 4452 4453size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4454{ 4455 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4456 ALOGW_IF(mEffectChains.size() != 1, 4457 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4458 chain.get(), mEffectChains.size(), this); 4459 if (mEffectChains.size() == 1) { 4460 mEffectChains.removeAt(0); 4461 } 4462 return 0; 4463} 4464 4465}; // namespace android 4466