Threads.cpp revision 7a90bc9265782788675af577c7b1c56e5d5be709
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 22#include <math.h> 23#include <fcntl.h> 24#include <sys/stat.h> 25#include <cutils/properties.h> 26#include <cutils/compiler.h> 27#include <utils/Log.h> 28 29#include <private/media/AudioTrackShared.h> 30#include <hardware/audio.h> 31#include <audio_effects/effect_ns.h> 32#include <audio_effects/effect_aec.h> 33#include <audio_utils/primitives.h> 34 35// NBAIO implementations 36#include <media/nbaio/AudioStreamOutSink.h> 37#include <media/nbaio/MonoPipe.h> 38#include <media/nbaio/MonoPipeReader.h> 39#include <media/nbaio/Pipe.h> 40#include <media/nbaio/PipeReader.h> 41#include <media/nbaio/SourceAudioBufferProvider.h> 42 43#include <powermanager/PowerManager.h> 44 45#include <common_time/cc_helper.h> 46#include <common_time/local_clock.h> 47 48#include "AudioFlinger.h" 49#include "AudioMixer.h" 50#include "FastMixer.h" 51#include "ServiceUtilities.h" 52#include "SchedulingPolicyService.h" 53 54#undef ADD_BATTERY_DATA 55 56#ifdef ADD_BATTERY_DATA 57#include <media/IMediaPlayerService.h> 58#include <media/IMediaDeathNotifier.h> 59#endif 60 61// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Whether to use fast mixer 113static const enum { 114 FastMixer_Never, // never initialize or use: for debugging only 115 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 116 // normal mixer multiplier is 1 117 FastMixer_Static, // initialize if needed, then use all the time if initialized, 118 // multiplier is calculated based on min & max normal mixer buffer size 119 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 // FIXME for FastMixer_Dynamic: 122 // Supporting this option will require fixing HALs that can't handle large writes. 123 // For example, one HAL implementation returns an error from a large write, 124 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 125 // We could either fix the HAL implementations, or provide a wrapper that breaks 126 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 127} kUseFastMixer = FastMixer_Static; 128 129// Priorities for requestPriority 130static const int kPriorityAudioApp = 2; 131static const int kPriorityFastMixer = 3; 132 133// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 134// for the track. The client then sub-divides this into smaller buffers for its use. 135// Currently the client uses double-buffering by default, but doesn't tell us about that. 136// So for now we just assume that client is double-buffered. 137// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 138// N-buffering, so AudioFlinger could allocate the right amount of memory. 139// See the client's minBufCount and mNotificationFramesAct calculations for details. 140static const int kFastTrackMultiplier = 2; 141 142// ---------------------------------------------------------------------------- 143 144#ifdef ADD_BATTERY_DATA 145// To collect the amplifier usage 146static void addBatteryData(uint32_t params) { 147 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 148 if (service == NULL) { 149 // it already logged 150 return; 151 } 152 153 service->addBatteryData(params); 154} 155#endif 156 157 158// ---------------------------------------------------------------------------- 159// CPU Stats 160// ---------------------------------------------------------------------------- 161 162class CpuStats { 163public: 164 CpuStats(); 165 void sample(const String8 &title); 166#ifdef DEBUG_CPU_USAGE 167private: 168 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 169 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 170 171 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 172 173 int mCpuNum; // thread's current CPU number 174 int mCpukHz; // frequency of thread's current CPU in kHz 175#endif 176}; 177 178CpuStats::CpuStats() 179#ifdef DEBUG_CPU_USAGE 180 : mCpuNum(-1), mCpukHz(-1) 181#endif 182{ 183} 184 185void CpuStats::sample(const String8 &title) { 186#ifdef DEBUG_CPU_USAGE 187 // get current thread's delta CPU time in wall clock ns 188 double wcNs; 189 bool valid = mCpuUsage.sampleAndEnable(wcNs); 190 191 // record sample for wall clock statistics 192 if (valid) { 193 mWcStats.sample(wcNs); 194 } 195 196 // get the current CPU number 197 int cpuNum = sched_getcpu(); 198 199 // get the current CPU frequency in kHz 200 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 201 202 // check if either CPU number or frequency changed 203 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 204 mCpuNum = cpuNum; 205 mCpukHz = cpukHz; 206 // ignore sample for purposes of cycles 207 valid = false; 208 } 209 210 // if no change in CPU number or frequency, then record sample for cycle statistics 211 if (valid && mCpukHz > 0) { 212 double cycles = wcNs * cpukHz * 0.000001; 213 mHzStats.sample(cycles); 214 } 215 216 unsigned n = mWcStats.n(); 217 // mCpuUsage.elapsed() is expensive, so don't call it every loop 218 if ((n & 127) == 1) { 219 long long elapsed = mCpuUsage.elapsed(); 220 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 221 double perLoop = elapsed / (double) n; 222 double perLoop100 = perLoop * 0.01; 223 double perLoop1k = perLoop * 0.001; 224 double mean = mWcStats.mean(); 225 double stddev = mWcStats.stddev(); 226 double minimum = mWcStats.minimum(); 227 double maximum = mWcStats.maximum(); 228 double meanCycles = mHzStats.mean(); 229 double stddevCycles = mHzStats.stddev(); 230 double minCycles = mHzStats.minimum(); 231 double maxCycles = mHzStats.maximum(); 232 mCpuUsage.resetElapsed(); 233 mWcStats.reset(); 234 mHzStats.reset(); 235 ALOGD("CPU usage for %s over past %.1f secs\n" 236 " (%u mixer loops at %.1f mean ms per loop):\n" 237 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 238 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 239 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 240 title.string(), 241 elapsed * .000000001, n, perLoop * .000001, 242 mean * .001, 243 stddev * .001, 244 minimum * .001, 245 maximum * .001, 246 mean / perLoop100, 247 stddev / perLoop100, 248 minimum / perLoop100, 249 maximum / perLoop100, 250 meanCycles / perLoop1k, 251 stddevCycles / perLoop1k, 252 minCycles / perLoop1k, 253 maxCycles / perLoop1k); 254 255 } 256 } 257#endif 258}; 259 260// ---------------------------------------------------------------------------- 261// ThreadBase 262// ---------------------------------------------------------------------------- 263 264AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 265 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 266 : Thread(false /*canCallJava*/), 267 mType(type), 268 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 269 // mChannelMask 270 mChannelCount(0), 271 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 272 mParamStatus(NO_ERROR), 273 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 274 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 275 // mName will be set by concrete (non-virtual) subclass 276 mDeathRecipient(new PMDeathRecipient(this)) 277{ 278} 279 280AudioFlinger::ThreadBase::~ThreadBase() 281{ 282 mParamCond.broadcast(); 283 // do not lock the mutex in destructor 284 releaseWakeLock_l(); 285 if (mPowerManager != 0) { 286 sp<IBinder> binder = mPowerManager->asBinder(); 287 binder->unlinkToDeath(mDeathRecipient); 288 } 289} 290 291void AudioFlinger::ThreadBase::exit() 292{ 293 ALOGV("ThreadBase::exit"); 294 // do any cleanup required for exit to succeed 295 preExit(); 296 { 297 // This lock prevents the following race in thread (uniprocessor for illustration): 298 // if (!exitPending()) { 299 // // context switch from here to exit() 300 // // exit() calls requestExit(), what exitPending() observes 301 // // exit() calls signal(), which is dropped since no waiters 302 // // context switch back from exit() to here 303 // mWaitWorkCV.wait(...); 304 // // now thread is hung 305 // } 306 AutoMutex lock(mLock); 307 requestExit(); 308 mWaitWorkCV.broadcast(); 309 } 310 // When Thread::requestExitAndWait is made virtual and this method is renamed to 311 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 312 requestExitAndWait(); 313} 314 315status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 316{ 317 status_t status; 318 319 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 320 Mutex::Autolock _l(mLock); 321 322 mNewParameters.add(keyValuePairs); 323 mWaitWorkCV.signal(); 324 // wait condition with timeout in case the thread loop has exited 325 // before the request could be processed 326 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 327 status = mParamStatus; 328 mWaitWorkCV.signal(); 329 } else { 330 status = TIMED_OUT; 331 } 332 return status; 333} 334 335void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 336{ 337 Mutex::Autolock _l(mLock); 338 sendIoConfigEvent_l(event, param); 339} 340 341// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 342void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 343{ 344 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 345 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 346 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 347 param); 348 mWaitWorkCV.signal(); 349} 350 351// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 352void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 353{ 354 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 355 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 356 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 357 mConfigEvents.size(), pid, tid, prio); 358 mWaitWorkCV.signal(); 359} 360 361void AudioFlinger::ThreadBase::processConfigEvents() 362{ 363 mLock.lock(); 364 while (!mConfigEvents.isEmpty()) { 365 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 366 ConfigEvent *event = mConfigEvents[0]; 367 mConfigEvents.removeAt(0); 368 // release mLock before locking AudioFlinger mLock: lock order is always 369 // AudioFlinger then ThreadBase to avoid cross deadlock 370 mLock.unlock(); 371 switch(event->type()) { 372 case CFG_EVENT_PRIO: { 373 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 374 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 375 if (err != 0) { 376 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 377 "error %d", 378 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 379 } 380 } break; 381 case CFG_EVENT_IO: { 382 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 383 mAudioFlinger->mLock.lock(); 384 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 385 mAudioFlinger->mLock.unlock(); 386 } break; 387 default: 388 ALOGE("processConfigEvents() unknown event type %d", event->type()); 389 break; 390 } 391 delete event; 392 mLock.lock(); 393 } 394 mLock.unlock(); 395} 396 397void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 398{ 399 const size_t SIZE = 256; 400 char buffer[SIZE]; 401 String8 result; 402 403 bool locked = AudioFlinger::dumpTryLock(mLock); 404 if (!locked) { 405 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 406 write(fd, buffer, strlen(buffer)); 407 } 408 409 snprintf(buffer, SIZE, "io handle: %d\n", mId); 410 result.append(buffer); 411 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 428 result.append(buffer); 429 430 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 431 result.append(buffer); 432 result.append(" Index Command"); 433 for (size_t i = 0; i < mNewParameters.size(); ++i) { 434 snprintf(buffer, SIZE, "\n %02d ", i); 435 result.append(buffer); 436 result.append(mNewParameters[i]); 437 } 438 439 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 440 result.append(buffer); 441 for (size_t i = 0; i < mConfigEvents.size(); i++) { 442 mConfigEvents[i]->dump(buffer, SIZE); 443 result.append(buffer); 444 } 445 result.append("\n"); 446 447 write(fd, result.string(), result.size()); 448 449 if (locked) { 450 mLock.unlock(); 451 } 452} 453 454void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 455{ 456 const size_t SIZE = 256; 457 char buffer[SIZE]; 458 String8 result; 459 460 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 461 write(fd, buffer, strlen(buffer)); 462 463 for (size_t i = 0; i < mEffectChains.size(); ++i) { 464 sp<EffectChain> chain = mEffectChains[i]; 465 if (chain != 0) { 466 chain->dump(fd, args); 467 } 468 } 469} 470 471void AudioFlinger::ThreadBase::acquireWakeLock() 472{ 473 Mutex::Autolock _l(mLock); 474 acquireWakeLock_l(); 475} 476 477void AudioFlinger::ThreadBase::acquireWakeLock_l() 478{ 479 if (mPowerManager == 0) { 480 // use checkService() to avoid blocking if power service is not up yet 481 sp<IBinder> binder = 482 defaultServiceManager()->checkService(String16("power")); 483 if (binder == 0) { 484 ALOGW("Thread %s cannot connect to the power manager service", mName); 485 } else { 486 mPowerManager = interface_cast<IPowerManager>(binder); 487 binder->linkToDeath(mDeathRecipient); 488 } 489 } 490 if (mPowerManager != 0) { 491 sp<IBinder> binder = new BBinder(); 492 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 493 binder, 494 String16(mName)); 495 if (status == NO_ERROR) { 496 mWakeLockToken = binder; 497 } 498 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 499 } 500} 501 502void AudioFlinger::ThreadBase::releaseWakeLock() 503{ 504 Mutex::Autolock _l(mLock); 505 releaseWakeLock_l(); 506} 507 508void AudioFlinger::ThreadBase::releaseWakeLock_l() 509{ 510 if (mWakeLockToken != 0) { 511 ALOGV("releaseWakeLock_l() %s", mName); 512 if (mPowerManager != 0) { 513 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 514 } 515 mWakeLockToken.clear(); 516 } 517} 518 519void AudioFlinger::ThreadBase::clearPowerManager() 520{ 521 Mutex::Autolock _l(mLock); 522 releaseWakeLock_l(); 523 mPowerManager.clear(); 524} 525 526void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 527{ 528 sp<ThreadBase> thread = mThread.promote(); 529 if (thread != 0) { 530 thread->clearPowerManager(); 531 } 532 ALOGW("power manager service died !!!"); 533} 534 535void AudioFlinger::ThreadBase::setEffectSuspended( 536 const effect_uuid_t *type, bool suspend, int sessionId) 537{ 538 Mutex::Autolock _l(mLock); 539 setEffectSuspended_l(type, suspend, sessionId); 540} 541 542void AudioFlinger::ThreadBase::setEffectSuspended_l( 543 const effect_uuid_t *type, bool suspend, int sessionId) 544{ 545 sp<EffectChain> chain = getEffectChain_l(sessionId); 546 if (chain != 0) { 547 if (type != NULL) { 548 chain->setEffectSuspended_l(type, suspend); 549 } else { 550 chain->setEffectSuspendedAll_l(suspend); 551 } 552 } 553 554 updateSuspendedSessions_l(type, suspend, sessionId); 555} 556 557void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 558{ 559 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 560 if (index < 0) { 561 return; 562 } 563 564 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 565 mSuspendedSessions.valueAt(index); 566 567 for (size_t i = 0; i < sessionEffects.size(); i++) { 568 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 569 for (int j = 0; j < desc->mRefCount; j++) { 570 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 571 chain->setEffectSuspendedAll_l(true); 572 } else { 573 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 574 desc->mType.timeLow); 575 chain->setEffectSuspended_l(&desc->mType, true); 576 } 577 } 578 } 579} 580 581void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 582 bool suspend, 583 int sessionId) 584{ 585 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 586 587 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 588 589 if (suspend) { 590 if (index >= 0) { 591 sessionEffects = mSuspendedSessions.valueAt(index); 592 } else { 593 mSuspendedSessions.add(sessionId, sessionEffects); 594 } 595 } else { 596 if (index < 0) { 597 return; 598 } 599 sessionEffects = mSuspendedSessions.valueAt(index); 600 } 601 602 603 int key = EffectChain::kKeyForSuspendAll; 604 if (type != NULL) { 605 key = type->timeLow; 606 } 607 index = sessionEffects.indexOfKey(key); 608 609 sp<SuspendedSessionDesc> desc; 610 if (suspend) { 611 if (index >= 0) { 612 desc = sessionEffects.valueAt(index); 613 } else { 614 desc = new SuspendedSessionDesc(); 615 if (type != NULL) { 616 desc->mType = *type; 617 } 618 sessionEffects.add(key, desc); 619 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 620 } 621 desc->mRefCount++; 622 } else { 623 if (index < 0) { 624 return; 625 } 626 desc = sessionEffects.valueAt(index); 627 if (--desc->mRefCount == 0) { 628 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 629 sessionEffects.removeItemsAt(index); 630 if (sessionEffects.isEmpty()) { 631 ALOGV("updateSuspendedSessions_l() restore removing session %d", 632 sessionId); 633 mSuspendedSessions.removeItem(sessionId); 634 } 635 } 636 } 637 if (!sessionEffects.isEmpty()) { 638 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 639 } 640} 641 642void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 643 bool enabled, 644 int sessionId) 645{ 646 Mutex::Autolock _l(mLock); 647 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 648} 649 650void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 651 bool enabled, 652 int sessionId) 653{ 654 if (mType != RECORD) { 655 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 656 // another session. This gives the priority to well behaved effect control panels 657 // and applications not using global effects. 658 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 659 // global effects 660 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 661 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 662 } 663 } 664 665 sp<EffectChain> chain = getEffectChain_l(sessionId); 666 if (chain != 0) { 667 chain->checkSuspendOnEffectEnabled(effect, enabled); 668 } 669} 670 671// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 672sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 673 const sp<AudioFlinger::Client>& client, 674 const sp<IEffectClient>& effectClient, 675 int32_t priority, 676 int sessionId, 677 effect_descriptor_t *desc, 678 int *enabled, 679 status_t *status 680 ) 681{ 682 sp<EffectModule> effect; 683 sp<EffectHandle> handle; 684 status_t lStatus; 685 sp<EffectChain> chain; 686 bool chainCreated = false; 687 bool effectCreated = false; 688 bool effectRegistered = false; 689 690 lStatus = initCheck(); 691 if (lStatus != NO_ERROR) { 692 ALOGW("createEffect_l() Audio driver not initialized."); 693 goto Exit; 694 } 695 696 // Do not allow effects with session ID 0 on direct output or duplicating threads 697 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 698 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 699 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 700 desc->name, sessionId); 701 lStatus = BAD_VALUE; 702 goto Exit; 703 } 704 // Only Pre processor effects are allowed on input threads and only on input threads 705 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 706 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 707 desc->name, desc->flags, mType); 708 lStatus = BAD_VALUE; 709 goto Exit; 710 } 711 712 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 713 714 { // scope for mLock 715 Mutex::Autolock _l(mLock); 716 717 // check for existing effect chain with the requested audio session 718 chain = getEffectChain_l(sessionId); 719 if (chain == 0) { 720 // create a new chain for this session 721 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 722 chain = new EffectChain(this, sessionId); 723 addEffectChain_l(chain); 724 chain->setStrategy(getStrategyForSession_l(sessionId)); 725 chainCreated = true; 726 } else { 727 effect = chain->getEffectFromDesc_l(desc); 728 } 729 730 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 731 732 if (effect == 0) { 733 int id = mAudioFlinger->nextUniqueId(); 734 // Check CPU and memory usage 735 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 736 if (lStatus != NO_ERROR) { 737 goto Exit; 738 } 739 effectRegistered = true; 740 // create a new effect module if none present in the chain 741 effect = new EffectModule(this, chain, desc, id, sessionId); 742 lStatus = effect->status(); 743 if (lStatus != NO_ERROR) { 744 goto Exit; 745 } 746 lStatus = chain->addEffect_l(effect); 747 if (lStatus != NO_ERROR) { 748 goto Exit; 749 } 750 effectCreated = true; 751 752 effect->setDevice(mOutDevice); 753 effect->setDevice(mInDevice); 754 effect->setMode(mAudioFlinger->getMode()); 755 effect->setAudioSource(mAudioSource); 756 } 757 // create effect handle and connect it to effect module 758 handle = new EffectHandle(effect, client, effectClient, priority); 759 lStatus = effect->addHandle(handle.get()); 760 if (enabled != NULL) { 761 *enabled = (int)effect->isEnabled(); 762 } 763 } 764 765Exit: 766 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 767 Mutex::Autolock _l(mLock); 768 if (effectCreated) { 769 chain->removeEffect_l(effect); 770 } 771 if (effectRegistered) { 772 AudioSystem::unregisterEffect(effect->id()); 773 } 774 if (chainCreated) { 775 removeEffectChain_l(chain); 776 } 777 handle.clear(); 778 } 779 780 if (status != NULL) { 781 *status = lStatus; 782 } 783 return handle; 784} 785 786sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 787{ 788 Mutex::Autolock _l(mLock); 789 return getEffect_l(sessionId, effectId); 790} 791 792sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 793{ 794 sp<EffectChain> chain = getEffectChain_l(sessionId); 795 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 796} 797 798// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 799// PlaybackThread::mLock held 800status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 801{ 802 // check for existing effect chain with the requested audio session 803 int sessionId = effect->sessionId(); 804 sp<EffectChain> chain = getEffectChain_l(sessionId); 805 bool chainCreated = false; 806 807 if (chain == 0) { 808 // create a new chain for this session 809 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 810 chain = new EffectChain(this, sessionId); 811 addEffectChain_l(chain); 812 chain->setStrategy(getStrategyForSession_l(sessionId)); 813 chainCreated = true; 814 } 815 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 816 817 if (chain->getEffectFromId_l(effect->id()) != 0) { 818 ALOGW("addEffect_l() %p effect %s already present in chain %p", 819 this, effect->desc().name, chain.get()); 820 return BAD_VALUE; 821 } 822 823 status_t status = chain->addEffect_l(effect); 824 if (status != NO_ERROR) { 825 if (chainCreated) { 826 removeEffectChain_l(chain); 827 } 828 return status; 829 } 830 831 effect->setDevice(mOutDevice); 832 effect->setDevice(mInDevice); 833 effect->setMode(mAudioFlinger->getMode()); 834 effect->setAudioSource(mAudioSource); 835 return NO_ERROR; 836} 837 838void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 839 840 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 841 effect_descriptor_t desc = effect->desc(); 842 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 843 detachAuxEffect_l(effect->id()); 844 } 845 846 sp<EffectChain> chain = effect->chain().promote(); 847 if (chain != 0) { 848 // remove effect chain if removing last effect 849 if (chain->removeEffect_l(effect) == 0) { 850 removeEffectChain_l(chain); 851 } 852 } else { 853 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 854 } 855} 856 857void AudioFlinger::ThreadBase::lockEffectChains_l( 858 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 859{ 860 effectChains = mEffectChains; 861 for (size_t i = 0; i < mEffectChains.size(); i++) { 862 mEffectChains[i]->lock(); 863 } 864} 865 866void AudioFlinger::ThreadBase::unlockEffectChains( 867 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 868{ 869 for (size_t i = 0; i < effectChains.size(); i++) { 870 effectChains[i]->unlock(); 871 } 872} 873 874sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 875{ 876 Mutex::Autolock _l(mLock); 877 return getEffectChain_l(sessionId); 878} 879 880sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 881{ 882 size_t size = mEffectChains.size(); 883 for (size_t i = 0; i < size; i++) { 884 if (mEffectChains[i]->sessionId() == sessionId) { 885 return mEffectChains[i]; 886 } 887 } 888 return 0; 889} 890 891void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 892{ 893 Mutex::Autolock _l(mLock); 894 size_t size = mEffectChains.size(); 895 for (size_t i = 0; i < size; i++) { 896 mEffectChains[i]->setMode_l(mode); 897 } 898} 899 900void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 901 EffectHandle *handle, 902 bool unpinIfLast) { 903 904 Mutex::Autolock _l(mLock); 905 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 906 // delete the effect module if removing last handle on it 907 if (effect->removeHandle(handle) == 0) { 908 if (!effect->isPinned() || unpinIfLast) { 909 removeEffect_l(effect); 910 AudioSystem::unregisterEffect(effect->id()); 911 } 912 } 913} 914 915// ---------------------------------------------------------------------------- 916// Playback 917// ---------------------------------------------------------------------------- 918 919AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 920 AudioStreamOut* output, 921 audio_io_handle_t id, 922 audio_devices_t device, 923 type_t type) 924 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 925 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 926 // mStreamTypes[] initialized in constructor body 927 mOutput(output), 928 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 929 mMixerStatus(MIXER_IDLE), 930 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 931 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 932 mScreenState(AudioFlinger::mScreenState), 933 // index 0 is reserved for normal mixer's submix 934 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 935{ 936 snprintf(mName, kNameLength, "AudioOut_%X", id); 937 938 // Assumes constructor is called by AudioFlinger with it's mLock held, but 939 // it would be safer to explicitly pass initial masterVolume/masterMute as 940 // parameter. 941 // 942 // If the HAL we are using has support for master volume or master mute, 943 // then do not attenuate or mute during mixing (just leave the volume at 1.0 944 // and the mute set to false). 945 mMasterVolume = audioFlinger->masterVolume_l(); 946 mMasterMute = audioFlinger->masterMute_l(); 947 if (mOutput && mOutput->audioHwDev) { 948 if (mOutput->audioHwDev->canSetMasterVolume()) { 949 mMasterVolume = 1.0; 950 } 951 952 if (mOutput->audioHwDev->canSetMasterMute()) { 953 mMasterMute = false; 954 } 955 } 956 957 readOutputParameters(); 958 959 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 960 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 961 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 962 stream = (audio_stream_type_t) (stream + 1)) { 963 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 964 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 965 } 966 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 967 // because mAudioFlinger doesn't have one to copy from 968} 969 970AudioFlinger::PlaybackThread::~PlaybackThread() 971{ 972 delete [] mMixBuffer; 973} 974 975void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 976{ 977 dumpInternals(fd, args); 978 dumpTracks(fd, args); 979 dumpEffectChains(fd, args); 980} 981 982void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 983{ 984 const size_t SIZE = 256; 985 char buffer[SIZE]; 986 String8 result; 987 988 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 989 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 990 const stream_type_t *st = &mStreamTypes[i]; 991 if (i > 0) { 992 result.appendFormat(", "); 993 } 994 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 995 if (st->mute) { 996 result.append("M"); 997 } 998 } 999 result.append("\n"); 1000 write(fd, result.string(), result.length()); 1001 result.clear(); 1002 1003 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1004 result.append(buffer); 1005 Track::appendDumpHeader(result); 1006 for (size_t i = 0; i < mTracks.size(); ++i) { 1007 sp<Track> track = mTracks[i]; 1008 if (track != 0) { 1009 track->dump(buffer, SIZE); 1010 result.append(buffer); 1011 } 1012 } 1013 1014 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1015 result.append(buffer); 1016 Track::appendDumpHeader(result); 1017 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1018 sp<Track> track = mActiveTracks[i].promote(); 1019 if (track != 0) { 1020 track->dump(buffer, SIZE); 1021 result.append(buffer); 1022 } 1023 } 1024 write(fd, result.string(), result.size()); 1025 1026 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1027 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1028 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1029 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1030} 1031 1032void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1033{ 1034 const size_t SIZE = 256; 1035 char buffer[SIZE]; 1036 String8 result; 1037 1038 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1039 result.append(buffer); 1040 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1041 ns2ms(systemTime() - mLastWriteTime)); 1042 result.append(buffer); 1043 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1044 result.append(buffer); 1045 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1052 result.append(buffer); 1053 write(fd, result.string(), result.size()); 1054 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1055 1056 dumpBase(fd, args); 1057} 1058 1059// Thread virtuals 1060status_t AudioFlinger::PlaybackThread::readyToRun() 1061{ 1062 status_t status = initCheck(); 1063 if (status == NO_ERROR) { 1064 ALOGI("AudioFlinger's thread %p ready to run", this); 1065 } else { 1066 ALOGE("No working audio driver found."); 1067 } 1068 return status; 1069} 1070 1071void AudioFlinger::PlaybackThread::onFirstRef() 1072{ 1073 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1074} 1075 1076// ThreadBase virtuals 1077void AudioFlinger::PlaybackThread::preExit() 1078{ 1079 ALOGV(" preExit()"); 1080 // FIXME this is using hard-coded strings but in the future, this functionality will be 1081 // converted to use audio HAL extensions required to support tunneling 1082 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1083} 1084 1085// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1086sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1087 const sp<AudioFlinger::Client>& client, 1088 audio_stream_type_t streamType, 1089 uint32_t sampleRate, 1090 audio_format_t format, 1091 audio_channel_mask_t channelMask, 1092 size_t frameCount, 1093 const sp<IMemory>& sharedBuffer, 1094 int sessionId, 1095 IAudioFlinger::track_flags_t *flags, 1096 pid_t tid, 1097 status_t *status) 1098{ 1099 sp<Track> track; 1100 status_t lStatus; 1101 1102 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1103 1104 // client expresses a preference for FAST, but we get the final say 1105 if (*flags & IAudioFlinger::TRACK_FAST) { 1106 if ( 1107 // not timed 1108 (!isTimed) && 1109 // either of these use cases: 1110 ( 1111 // use case 1: shared buffer with any frame count 1112 ( 1113 (sharedBuffer != 0) 1114 ) || 1115 // use case 2: callback handler and frame count is default or at least as large as HAL 1116 ( 1117 (tid != -1) && 1118 ((frameCount == 0) || 1119 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1120 ) 1121 ) && 1122 // PCM data 1123 audio_is_linear_pcm(format) && 1124 // mono or stereo 1125 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1126 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1127#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1128 // hardware sample rate 1129 (sampleRate == mSampleRate) && 1130#endif 1131 // normal mixer has an associated fast mixer 1132 hasFastMixer() && 1133 // there are sufficient fast track slots available 1134 (mFastTrackAvailMask != 0) 1135 // FIXME test that MixerThread for this fast track has a capable output HAL 1136 // FIXME add a permission test also? 1137 ) { 1138 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1139 if (frameCount == 0) { 1140 frameCount = mFrameCount * kFastTrackMultiplier; 1141 } 1142 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1143 frameCount, mFrameCount); 1144 } else { 1145 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1146 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1147 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1148 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1149 audio_is_linear_pcm(format), 1150 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1151 *flags &= ~IAudioFlinger::TRACK_FAST; 1152 // For compatibility with AudioTrack calculation, buffer depth is forced 1153 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1154 // This is probably too conservative, but legacy application code may depend on it. 1155 // If you change this calculation, also review the start threshold which is related. 1156 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1157 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1158 if (minBufCount < 2) { 1159 minBufCount = 2; 1160 } 1161 size_t minFrameCount = mNormalFrameCount * minBufCount; 1162 if (frameCount < minFrameCount) { 1163 frameCount = minFrameCount; 1164 } 1165 } 1166 } 1167 1168 if (mType == DIRECT) { 1169 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1170 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1171 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1172 "for output %p with format %d", 1173 sampleRate, format, channelMask, mOutput, mFormat); 1174 lStatus = BAD_VALUE; 1175 goto Exit; 1176 } 1177 } 1178 } else { 1179 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1180 if (sampleRate > mSampleRate*2) { 1181 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1182 lStatus = BAD_VALUE; 1183 goto Exit; 1184 } 1185 } 1186 1187 lStatus = initCheck(); 1188 if (lStatus != NO_ERROR) { 1189 ALOGE("Audio driver not initialized."); 1190 goto Exit; 1191 } 1192 1193 { // scope for mLock 1194 Mutex::Autolock _l(mLock); 1195 1196 // all tracks in same audio session must share the same routing strategy otherwise 1197 // conflicts will happen when tracks are moved from one output to another by audio policy 1198 // manager 1199 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1200 for (size_t i = 0; i < mTracks.size(); ++i) { 1201 sp<Track> t = mTracks[i]; 1202 if (t != 0 && !t->isOutputTrack()) { 1203 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1204 if (sessionId == t->sessionId() && strategy != actual) { 1205 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1206 strategy, actual); 1207 lStatus = BAD_VALUE; 1208 goto Exit; 1209 } 1210 } 1211 } 1212 1213 if (!isTimed) { 1214 track = new Track(this, client, streamType, sampleRate, format, 1215 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1216 } else { 1217 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1218 channelMask, frameCount, sharedBuffer, sessionId); 1219 } 1220 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1221 lStatus = NO_MEMORY; 1222 goto Exit; 1223 } 1224 mTracks.add(track); 1225 1226 sp<EffectChain> chain = getEffectChain_l(sessionId); 1227 if (chain != 0) { 1228 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1229 track->setMainBuffer(chain->inBuffer()); 1230 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1231 chain->incTrackCnt(); 1232 } 1233 1234 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1235 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1236 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1237 // so ask activity manager to do this on our behalf 1238 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1239 } 1240 } 1241 1242 lStatus = NO_ERROR; 1243 1244Exit: 1245 if (status) { 1246 *status = lStatus; 1247 } 1248 return track; 1249} 1250 1251uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1252{ 1253 return latency; 1254} 1255 1256uint32_t AudioFlinger::PlaybackThread::latency() const 1257{ 1258 Mutex::Autolock _l(mLock); 1259 return latency_l(); 1260} 1261uint32_t AudioFlinger::PlaybackThread::latency_l() const 1262{ 1263 if (initCheck() == NO_ERROR) { 1264 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1265 } else { 1266 return 0; 1267 } 1268} 1269 1270void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1271{ 1272 Mutex::Autolock _l(mLock); 1273 // Don't apply master volume in SW if our HAL can do it for us. 1274 if (mOutput && mOutput->audioHwDev && 1275 mOutput->audioHwDev->canSetMasterVolume()) { 1276 mMasterVolume = 1.0; 1277 } else { 1278 mMasterVolume = value; 1279 } 1280} 1281 1282void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1283{ 1284 Mutex::Autolock _l(mLock); 1285 // Don't apply master mute in SW if our HAL can do it for us. 1286 if (mOutput && mOutput->audioHwDev && 1287 mOutput->audioHwDev->canSetMasterMute()) { 1288 mMasterMute = false; 1289 } else { 1290 mMasterMute = muted; 1291 } 1292} 1293 1294void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1295{ 1296 Mutex::Autolock _l(mLock); 1297 mStreamTypes[stream].volume = value; 1298} 1299 1300void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1301{ 1302 Mutex::Autolock _l(mLock); 1303 mStreamTypes[stream].mute = muted; 1304} 1305 1306float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1307{ 1308 Mutex::Autolock _l(mLock); 1309 return mStreamTypes[stream].volume; 1310} 1311 1312// addTrack_l() must be called with ThreadBase::mLock held 1313status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1314{ 1315 status_t status = ALREADY_EXISTS; 1316 1317 // set retry count for buffer fill 1318 track->mRetryCount = kMaxTrackStartupRetries; 1319 if (mActiveTracks.indexOf(track) < 0) { 1320 // the track is newly added, make sure it fills up all its 1321 // buffers before playing. This is to ensure the client will 1322 // effectively get the latency it requested. 1323 track->mFillingUpStatus = Track::FS_FILLING; 1324 track->mResetDone = false; 1325 track->mPresentationCompleteFrames = 0; 1326 mActiveTracks.add(track); 1327 if (track->mainBuffer() != mMixBuffer) { 1328 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1329 if (chain != 0) { 1330 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1331 track->sessionId()); 1332 chain->incActiveTrackCnt(); 1333 } 1334 } 1335 1336 status = NO_ERROR; 1337 } 1338 1339 ALOGV("mWaitWorkCV.broadcast"); 1340 mWaitWorkCV.broadcast(); 1341 1342 return status; 1343} 1344 1345// destroyTrack_l() must be called with ThreadBase::mLock held 1346void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1347{ 1348 track->mState = TrackBase::TERMINATED; 1349 // active tracks are removed by threadLoop() 1350 if (mActiveTracks.indexOf(track) < 0) { 1351 removeTrack_l(track); 1352 } 1353} 1354 1355void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1356{ 1357 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1358 mTracks.remove(track); 1359 deleteTrackName_l(track->name()); 1360 // redundant as track is about to be destroyed, for dumpsys only 1361 track->mName = -1; 1362 if (track->isFastTrack()) { 1363 int index = track->mFastIndex; 1364 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1365 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1366 mFastTrackAvailMask |= 1 << index; 1367 // redundant as track is about to be destroyed, for dumpsys only 1368 track->mFastIndex = -1; 1369 } 1370 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1371 if (chain != 0) { 1372 chain->decTrackCnt(); 1373 } 1374} 1375 1376String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1377{ 1378 String8 out_s8 = String8(""); 1379 char *s; 1380 1381 Mutex::Autolock _l(mLock); 1382 if (initCheck() != NO_ERROR) { 1383 return out_s8; 1384 } 1385 1386 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1387 out_s8 = String8(s); 1388 free(s); 1389 return out_s8; 1390} 1391 1392// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1393void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1394 AudioSystem::OutputDescriptor desc; 1395 void *param2 = NULL; 1396 1397 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1398 param); 1399 1400 switch (event) { 1401 case AudioSystem::OUTPUT_OPENED: 1402 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1403 desc.channels = mChannelMask; 1404 desc.samplingRate = mSampleRate; 1405 desc.format = mFormat; 1406 desc.frameCount = mNormalFrameCount; // FIXME see 1407 // AudioFlinger::frameCount(audio_io_handle_t) 1408 desc.latency = latency(); 1409 param2 = &desc; 1410 break; 1411 1412 case AudioSystem::STREAM_CONFIG_CHANGED: 1413 param2 = ¶m; 1414 case AudioSystem::OUTPUT_CLOSED: 1415 default: 1416 break; 1417 } 1418 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1419} 1420 1421void AudioFlinger::PlaybackThread::readOutputParameters() 1422{ 1423 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1424 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1425 mChannelCount = (uint16_t)popcount(mChannelMask); 1426 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1427 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1428 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1429 if (mFrameCount & 15) { 1430 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1431 mFrameCount); 1432 } 1433 1434 // Calculate size of normal mix buffer relative to the HAL output buffer size 1435 double multiplier = 1.0; 1436 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1437 kUseFastMixer == FastMixer_Dynamic)) { 1438 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1439 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1440 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1441 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1442 maxNormalFrameCount = maxNormalFrameCount & ~15; 1443 if (maxNormalFrameCount < minNormalFrameCount) { 1444 maxNormalFrameCount = minNormalFrameCount; 1445 } 1446 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1447 if (multiplier <= 1.0) { 1448 multiplier = 1.0; 1449 } else if (multiplier <= 2.0) { 1450 if (2 * mFrameCount <= maxNormalFrameCount) { 1451 multiplier = 2.0; 1452 } else { 1453 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1454 } 1455 } else { 1456 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1457 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1458 // track, but we sometimes have to do this to satisfy the maximum frame count 1459 // constraint) 1460 // FIXME this rounding up should not be done if no HAL SRC 1461 uint32_t truncMult = (uint32_t) multiplier; 1462 if ((truncMult & 1)) { 1463 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1464 ++truncMult; 1465 } 1466 } 1467 multiplier = (double) truncMult; 1468 } 1469 } 1470 mNormalFrameCount = multiplier * mFrameCount; 1471 // round up to nearest 16 frames to satisfy AudioMixer 1472 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1473 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1474 mNormalFrameCount); 1475 1476 delete[] mMixBuffer; 1477 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1478 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1479 1480 // force reconfiguration of effect chains and engines to take new buffer size and audio 1481 // parameters into account 1482 // Note that mLock is not held when readOutputParameters() is called from the constructor 1483 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1484 // matter. 1485 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1486 Vector< sp<EffectChain> > effectChains = mEffectChains; 1487 for (size_t i = 0; i < effectChains.size(); i ++) { 1488 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1489 } 1490} 1491 1492 1493status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1494{ 1495 if (halFrames == NULL || dspFrames == NULL) { 1496 return BAD_VALUE; 1497 } 1498 Mutex::Autolock _l(mLock); 1499 if (initCheck() != NO_ERROR) { 1500 return INVALID_OPERATION; 1501 } 1502 size_t framesWritten = mBytesWritten / mFrameSize; 1503 *halFrames = framesWritten; 1504 1505 if (isSuspended()) { 1506 // return an estimation of rendered frames when the output is suspended 1507 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1508 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1509 return NO_ERROR; 1510 } else { 1511 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1512 } 1513} 1514 1515uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1516{ 1517 Mutex::Autolock _l(mLock); 1518 uint32_t result = 0; 1519 if (getEffectChain_l(sessionId) != 0) { 1520 result = EFFECT_SESSION; 1521 } 1522 1523 for (size_t i = 0; i < mTracks.size(); ++i) { 1524 sp<Track> track = mTracks[i]; 1525 if (sessionId == track->sessionId() && 1526 !(track->mCblk->flags & CBLK_INVALID)) { 1527 result |= TRACK_SESSION; 1528 break; 1529 } 1530 } 1531 1532 return result; 1533} 1534 1535uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1536{ 1537 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1538 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1539 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1540 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1541 } 1542 for (size_t i = 0; i < mTracks.size(); i++) { 1543 sp<Track> track = mTracks[i]; 1544 if (sessionId == track->sessionId() && 1545 !(track->mCblk->flags & CBLK_INVALID)) { 1546 return AudioSystem::getStrategyForStream(track->streamType()); 1547 } 1548 } 1549 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1550} 1551 1552 1553AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1554{ 1555 Mutex::Autolock _l(mLock); 1556 return mOutput; 1557} 1558 1559AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1560{ 1561 Mutex::Autolock _l(mLock); 1562 AudioStreamOut *output = mOutput; 1563 mOutput = NULL; 1564 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1565 // must push a NULL and wait for ack 1566 mOutputSink.clear(); 1567 mPipeSink.clear(); 1568 mNormalSink.clear(); 1569 return output; 1570} 1571 1572// this method must always be called either with ThreadBase mLock held or inside the thread loop 1573audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1574{ 1575 if (mOutput == NULL) { 1576 return NULL; 1577 } 1578 return &mOutput->stream->common; 1579} 1580 1581uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1582{ 1583 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1584} 1585 1586status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1587{ 1588 if (!isValidSyncEvent(event)) { 1589 return BAD_VALUE; 1590 } 1591 1592 Mutex::Autolock _l(mLock); 1593 1594 for (size_t i = 0; i < mTracks.size(); ++i) { 1595 sp<Track> track = mTracks[i]; 1596 if (event->triggerSession() == track->sessionId()) { 1597 (void) track->setSyncEvent(event); 1598 return NO_ERROR; 1599 } 1600 } 1601 1602 return NAME_NOT_FOUND; 1603} 1604 1605bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1606{ 1607 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1608} 1609 1610void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1611 const Vector< sp<Track> >& tracksToRemove) 1612{ 1613 size_t count = tracksToRemove.size(); 1614 if (CC_UNLIKELY(count)) { 1615 for (size_t i = 0 ; i < count ; i++) { 1616 const sp<Track>& track = tracksToRemove.itemAt(i); 1617 if ((track->sharedBuffer() != 0) && 1618 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1619 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1620 } 1621 } 1622 } 1623 1624} 1625 1626void AudioFlinger::PlaybackThread::checkSilentMode_l() 1627{ 1628 if (!mMasterMute) { 1629 char value[PROPERTY_VALUE_MAX]; 1630 if (property_get("ro.audio.silent", value, "0") > 0) { 1631 char *endptr; 1632 unsigned long ul = strtoul(value, &endptr, 0); 1633 if (*endptr == '\0' && ul != 0) { 1634 ALOGD("Silence is golden"); 1635 // The setprop command will not allow a property to be changed after 1636 // the first time it is set, so we don't have to worry about un-muting. 1637 setMasterMute_l(true); 1638 } 1639 } 1640 } 1641} 1642 1643// shared by MIXER and DIRECT, overridden by DUPLICATING 1644void AudioFlinger::PlaybackThread::threadLoop_write() 1645{ 1646 // FIXME rewrite to reduce number of system calls 1647 mLastWriteTime = systemTime(); 1648 mInWrite = true; 1649 int bytesWritten; 1650 1651 // If an NBAIO sink is present, use it to write the normal mixer's submix 1652 if (mNormalSink != 0) { 1653#define mBitShift 2 // FIXME 1654 size_t count = mixBufferSize >> mBitShift; 1655#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 1656 ATRACE_BEGIN("write"); 1657#endif 1658 // update the setpoint when AudioFlinger::mScreenState changes 1659 uint32_t screenState = AudioFlinger::mScreenState; 1660 if (screenState != mScreenState) { 1661 mScreenState = screenState; 1662 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1663 if (pipe != NULL) { 1664 pipe->setAvgFrames((mScreenState & 1) ? 1665 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1666 } 1667 } 1668 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1669#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 1670 ATRACE_END(); 1671#endif 1672 if (framesWritten > 0) { 1673 bytesWritten = framesWritten << mBitShift; 1674 } else { 1675 bytesWritten = framesWritten; 1676 } 1677 // otherwise use the HAL / AudioStreamOut directly 1678 } else { 1679 // Direct output thread. 1680 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1681 } 1682 1683 if (bytesWritten > 0) { 1684 mBytesWritten += mixBufferSize; 1685 } 1686 mNumWrites++; 1687 mInWrite = false; 1688} 1689 1690/* 1691The derived values that are cached: 1692 - mixBufferSize from frame count * frame size 1693 - activeSleepTime from activeSleepTimeUs() 1694 - idleSleepTime from idleSleepTimeUs() 1695 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1696 - maxPeriod from frame count and sample rate (MIXER only) 1697 1698The parameters that affect these derived values are: 1699 - frame count 1700 - frame size 1701 - sample rate 1702 - device type: A2DP or not 1703 - device latency 1704 - format: PCM or not 1705 - active sleep time 1706 - idle sleep time 1707*/ 1708 1709void AudioFlinger::PlaybackThread::cacheParameters_l() 1710{ 1711 mixBufferSize = mNormalFrameCount * mFrameSize; 1712 activeSleepTime = activeSleepTimeUs(); 1713 idleSleepTime = idleSleepTimeUs(); 1714} 1715 1716void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1717{ 1718 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1719 this, streamType, mTracks.size()); 1720 Mutex::Autolock _l(mLock); 1721 1722 size_t size = mTracks.size(); 1723 for (size_t i = 0; i < size; i++) { 1724 sp<Track> t = mTracks[i]; 1725 if (t->streamType() == streamType) { 1726 android_atomic_or(CBLK_INVALID, &t->mCblk->flags); 1727 t->mCblk->cv.signal(); 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 while (!exitPending()) 1901 { 1902 cpuStats.sample(myName); 1903 1904 Vector< sp<EffectChain> > effectChains; 1905 1906 processConfigEvents(); 1907 1908 { // scope for mLock 1909 1910 Mutex::Autolock _l(mLock); 1911 1912 if (checkForNewParameters_l()) { 1913 cacheParameters_l(); 1914 } 1915 1916 saveOutputTracks(); 1917 1918 // put audio hardware into standby after short delay 1919 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1920 isSuspended())) { 1921 if (!mStandby) { 1922 1923 threadLoop_standby(); 1924 1925 mStandby = true; 1926 } 1927 1928 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1929 // we're about to wait, flush the binder command buffer 1930 IPCThreadState::self()->flushCommands(); 1931 1932 clearOutputTracks(); 1933 1934 if (exitPending()) { 1935 break; 1936 } 1937 1938 releaseWakeLock_l(); 1939 // wait until we have something to do... 1940 ALOGV("%s going to sleep", myName.string()); 1941 mWaitWorkCV.wait(mLock); 1942 ALOGV("%s waking up", myName.string()); 1943 acquireWakeLock_l(); 1944 1945 mMixerStatus = MIXER_IDLE; 1946 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1947 mBytesWritten = 0; 1948 1949 checkSilentMode_l(); 1950 1951 standbyTime = systemTime() + standbyDelay; 1952 sleepTime = idleSleepTime; 1953 if (mType == MIXER) { 1954 sleepTimeShift = 0; 1955 } 1956 1957 continue; 1958 } 1959 } 1960 1961 // mMixerStatusIgnoringFastTracks is also updated internally 1962 mMixerStatus = prepareTracks_l(&tracksToRemove); 1963 1964 // prevent any changes in effect chain list and in each effect chain 1965 // during mixing and effect process as the audio buffers could be deleted 1966 // or modified if an effect is created or deleted 1967 lockEffectChains_l(effectChains); 1968 } 1969 1970 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1971 threadLoop_mix(); 1972 } else { 1973 threadLoop_sleepTime(); 1974 } 1975 1976 if (isSuspended()) { 1977 sleepTime = suspendSleepTimeUs(); 1978 mBytesWritten += mixBufferSize; 1979 } 1980 1981 // only process effects if we're going to write 1982 if (sleepTime == 0) { 1983 for (size_t i = 0; i < effectChains.size(); i ++) { 1984 effectChains[i]->process_l(); 1985 } 1986 } 1987 1988 // enable changes in effect chain 1989 unlockEffectChains(effectChains); 1990 1991 // sleepTime == 0 means we must write to audio hardware 1992 if (sleepTime == 0) { 1993 1994 threadLoop_write(); 1995 1996if (mType == MIXER) { 1997 // write blocked detection 1998 nsecs_t now = systemTime(); 1999 nsecs_t delta = now - mLastWriteTime; 2000 if (!mStandby && delta > maxPeriod) { 2001 mNumDelayedWrites++; 2002 if ((now - lastWarning) > kWarningThrottleNs) { 2003#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2004 ScopedTrace st(ATRACE_TAG, "underrun"); 2005#endif 2006 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2007 ns2ms(delta), mNumDelayedWrites, this); 2008 lastWarning = now; 2009 } 2010 } 2011} 2012 2013 mStandby = false; 2014 } else { 2015 usleep(sleepTime); 2016 } 2017 2018 // Finally let go of removed track(s), without the lock held 2019 // since we can't guarantee the destructors won't acquire that 2020 // same lock. This will also mutate and push a new fast mixer state. 2021 threadLoop_removeTracks(tracksToRemove); 2022 tracksToRemove.clear(); 2023 2024 // FIXME I don't understand the need for this here; 2025 // it was in the original code but maybe the 2026 // assignment in saveOutputTracks() makes this unnecessary? 2027 clearOutputTracks(); 2028 2029 // Effect chains will be actually deleted here if they were removed from 2030 // mEffectChains list during mixing or effects processing 2031 effectChains.clear(); 2032 2033 // FIXME Note that the above .clear() is no longer necessary since effectChains 2034 // is now local to this block, but will keep it for now (at least until merge done). 2035 } 2036 2037 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2038 if (mType == MIXER || mType == DIRECT) { 2039 // put output stream into standby mode 2040 if (!mStandby) { 2041 mOutput->stream->common.standby(&mOutput->stream->common); 2042 } 2043 } 2044 2045 releaseWakeLock(); 2046 2047 ALOGV("Thread %p type %d exiting", this, mType); 2048 return false; 2049} 2050 2051 2052// ---------------------------------------------------------------------------- 2053 2054AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2055 audio_io_handle_t id, audio_devices_t device, type_t type) 2056 : PlaybackThread(audioFlinger, output, id, device, type), 2057 // mAudioMixer below 2058 // mFastMixer below 2059 mFastMixerFutex(0) 2060 // mOutputSink below 2061 // mPipeSink below 2062 // mNormalSink below 2063{ 2064 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2065 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2066 "mFrameCount=%d, mNormalFrameCount=%d", 2067 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2068 mNormalFrameCount); 2069 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2070 2071 // FIXME - Current mixer implementation only supports stereo output 2072 if (mChannelCount != FCC_2) { 2073 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2074 } 2075 2076 // create an NBAIO sink for the HAL output stream, and negotiate 2077 mOutputSink = new AudioStreamOutSink(output->stream); 2078 size_t numCounterOffers = 0; 2079 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2080 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2081 ALOG_ASSERT(index == 0); 2082 2083 // initialize fast mixer depending on configuration 2084 bool initFastMixer; 2085 switch (kUseFastMixer) { 2086 case FastMixer_Never: 2087 initFastMixer = false; 2088 break; 2089 case FastMixer_Always: 2090 initFastMixer = true; 2091 break; 2092 case FastMixer_Static: 2093 case FastMixer_Dynamic: 2094 initFastMixer = mFrameCount < mNormalFrameCount; 2095 break; 2096 } 2097 if (initFastMixer) { 2098 2099 // create a MonoPipe to connect our submix to FastMixer 2100 NBAIO_Format format = mOutputSink->format(); 2101 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2102 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2103 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2104 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2105 const NBAIO_Format offers[1] = {format}; 2106 size_t numCounterOffers = 0; 2107 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2108 ALOG_ASSERT(index == 0); 2109 monoPipe->setAvgFrames((mScreenState & 1) ? 2110 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2111 mPipeSink = monoPipe; 2112 2113#ifdef TEE_SINK_FRAMES 2114 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2115 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2116 numCounterOffers = 0; 2117 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2118 ALOG_ASSERT(index == 0); 2119 mTeeSink = teeSink; 2120 PipeReader *teeSource = new PipeReader(*teeSink); 2121 numCounterOffers = 0; 2122 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2123 ALOG_ASSERT(index == 0); 2124 mTeeSource = teeSource; 2125#endif 2126 2127 // create fast mixer and configure it initially with just one fast track for our submix 2128 mFastMixer = new FastMixer(); 2129 FastMixerStateQueue *sq = mFastMixer->sq(); 2130#ifdef STATE_QUEUE_DUMP 2131 sq->setObserverDump(&mStateQueueObserverDump); 2132 sq->setMutatorDump(&mStateQueueMutatorDump); 2133#endif 2134 FastMixerState *state = sq->begin(); 2135 FastTrack *fastTrack = &state->mFastTracks[0]; 2136 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2137 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2138 fastTrack->mVolumeProvider = NULL; 2139 fastTrack->mGeneration++; 2140 state->mFastTracksGen++; 2141 state->mTrackMask = 1; 2142 // fast mixer will use the HAL output sink 2143 state->mOutputSink = mOutputSink.get(); 2144 state->mOutputSinkGen++; 2145 state->mFrameCount = mFrameCount; 2146 state->mCommand = FastMixerState::COLD_IDLE; 2147 // already done in constructor initialization list 2148 //mFastMixerFutex = 0; 2149 state->mColdFutexAddr = &mFastMixerFutex; 2150 state->mColdGen++; 2151 state->mDumpState = &mFastMixerDumpState; 2152 state->mTeeSink = mTeeSink.get(); 2153 sq->end(); 2154 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2155 2156 // start the fast mixer 2157 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2158 pid_t tid = mFastMixer->getTid(); 2159 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2160 if (err != 0) { 2161 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2162 kPriorityFastMixer, getpid_cached, tid, err); 2163 } 2164 2165#ifdef AUDIO_WATCHDOG 2166 // create and start the watchdog 2167 mAudioWatchdog = new AudioWatchdog(); 2168 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2169 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2170 tid = mAudioWatchdog->getTid(); 2171 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2172 if (err != 0) { 2173 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2174 kPriorityFastMixer, getpid_cached, tid, err); 2175 } 2176#endif 2177 2178 } else { 2179 mFastMixer = NULL; 2180 } 2181 2182 switch (kUseFastMixer) { 2183 case FastMixer_Never: 2184 case FastMixer_Dynamic: 2185 mNormalSink = mOutputSink; 2186 break; 2187 case FastMixer_Always: 2188 mNormalSink = mPipeSink; 2189 break; 2190 case FastMixer_Static: 2191 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2192 break; 2193 } 2194} 2195 2196AudioFlinger::MixerThread::~MixerThread() 2197{ 2198 if (mFastMixer != NULL) { 2199 FastMixerStateQueue *sq = mFastMixer->sq(); 2200 FastMixerState *state = sq->begin(); 2201 if (state->mCommand == FastMixerState::COLD_IDLE) { 2202 int32_t old = android_atomic_inc(&mFastMixerFutex); 2203 if (old == -1) { 2204 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2205 } 2206 } 2207 state->mCommand = FastMixerState::EXIT; 2208 sq->end(); 2209 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2210 mFastMixer->join(); 2211 // Though the fast mixer thread has exited, it's state queue is still valid. 2212 // We'll use that extract the final state which contains one remaining fast track 2213 // corresponding to our sub-mix. 2214 state = sq->begin(); 2215 ALOG_ASSERT(state->mTrackMask == 1); 2216 FastTrack *fastTrack = &state->mFastTracks[0]; 2217 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2218 delete fastTrack->mBufferProvider; 2219 sq->end(false /*didModify*/); 2220 delete mFastMixer; 2221#ifdef AUDIO_WATCHDOG 2222 if (mAudioWatchdog != 0) { 2223 mAudioWatchdog->requestExit(); 2224 mAudioWatchdog->requestExitAndWait(); 2225 mAudioWatchdog.clear(); 2226 } 2227#endif 2228 } 2229 delete mAudioMixer; 2230} 2231 2232 2233uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2234{ 2235 if (mFastMixer != NULL) { 2236 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2237 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2238 } 2239 return latency; 2240} 2241 2242 2243void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2244{ 2245 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2246} 2247 2248void AudioFlinger::MixerThread::threadLoop_write() 2249{ 2250 // FIXME we should only do one push per cycle; confirm this is true 2251 // Start the fast mixer if it's not already running 2252 if (mFastMixer != NULL) { 2253 FastMixerStateQueue *sq = mFastMixer->sq(); 2254 FastMixerState *state = sq->begin(); 2255 if (state->mCommand != FastMixerState::MIX_WRITE && 2256 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2257 if (state->mCommand == FastMixerState::COLD_IDLE) { 2258 int32_t old = android_atomic_inc(&mFastMixerFutex); 2259 if (old == -1) { 2260 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2261 } 2262#ifdef AUDIO_WATCHDOG 2263 if (mAudioWatchdog != 0) { 2264 mAudioWatchdog->resume(); 2265 } 2266#endif 2267 } 2268 state->mCommand = FastMixerState::MIX_WRITE; 2269 sq->end(); 2270 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2271 if (kUseFastMixer == FastMixer_Dynamic) { 2272 mNormalSink = mPipeSink; 2273 } 2274 } else { 2275 sq->end(false /*didModify*/); 2276 } 2277 } 2278 PlaybackThread::threadLoop_write(); 2279} 2280 2281void AudioFlinger::MixerThread::threadLoop_standby() 2282{ 2283 // Idle the fast mixer if it's currently running 2284 if (mFastMixer != NULL) { 2285 FastMixerStateQueue *sq = mFastMixer->sq(); 2286 FastMixerState *state = sq->begin(); 2287 if (!(state->mCommand & FastMixerState::IDLE)) { 2288 state->mCommand = FastMixerState::COLD_IDLE; 2289 state->mColdFutexAddr = &mFastMixerFutex; 2290 state->mColdGen++; 2291 mFastMixerFutex = 0; 2292 sq->end(); 2293 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2294 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2295 if (kUseFastMixer == FastMixer_Dynamic) { 2296 mNormalSink = mOutputSink; 2297 } 2298#ifdef AUDIO_WATCHDOG 2299 if (mAudioWatchdog != 0) { 2300 mAudioWatchdog->pause(); 2301 } 2302#endif 2303 } else { 2304 sq->end(false /*didModify*/); 2305 } 2306 } 2307 PlaybackThread::threadLoop_standby(); 2308} 2309 2310// shared by MIXER and DIRECT, overridden by DUPLICATING 2311void AudioFlinger::PlaybackThread::threadLoop_standby() 2312{ 2313 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2314 mOutput->stream->common.standby(&mOutput->stream->common); 2315} 2316 2317void AudioFlinger::MixerThread::threadLoop_mix() 2318{ 2319 // obtain the presentation timestamp of the next output buffer 2320 int64_t pts; 2321 status_t status = INVALID_OPERATION; 2322 2323 if (mNormalSink != 0) { 2324 status = mNormalSink->getNextWriteTimestamp(&pts); 2325 } else { 2326 status = mOutputSink->getNextWriteTimestamp(&pts); 2327 } 2328 2329 if (status != NO_ERROR) { 2330 pts = AudioBufferProvider::kInvalidPTS; 2331 } 2332 2333 // mix buffers... 2334 mAudioMixer->process(pts); 2335 // increase sleep time progressively when application underrun condition clears. 2336 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2337 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2338 // such that we would underrun the audio HAL. 2339 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2340 sleepTimeShift--; 2341 } 2342 sleepTime = 0; 2343 standbyTime = systemTime() + standbyDelay; 2344 //TODO: delay standby when effects have a tail 2345} 2346 2347void AudioFlinger::MixerThread::threadLoop_sleepTime() 2348{ 2349 // If no tracks are ready, sleep once for the duration of an output 2350 // buffer size, then write 0s to the output 2351 if (sleepTime == 0) { 2352 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2353 sleepTime = activeSleepTime >> sleepTimeShift; 2354 if (sleepTime < kMinThreadSleepTimeUs) { 2355 sleepTime = kMinThreadSleepTimeUs; 2356 } 2357 // reduce sleep time in case of consecutive application underruns to avoid 2358 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2359 // duration we would end up writing less data than needed by the audio HAL if 2360 // the condition persists. 2361 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2362 sleepTimeShift++; 2363 } 2364 } else { 2365 sleepTime = idleSleepTime; 2366 } 2367 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2368 memset (mMixBuffer, 0, mixBufferSize); 2369 sleepTime = 0; 2370 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2371 "anticipated start"); 2372 } 2373 // TODO add standby time extension fct of effect tail 2374} 2375 2376// prepareTracks_l() must be called with ThreadBase::mLock held 2377AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2378 Vector< sp<Track> > *tracksToRemove) 2379{ 2380 2381 mixer_state mixerStatus = MIXER_IDLE; 2382 // find out which tracks need to be processed 2383 size_t count = mActiveTracks.size(); 2384 size_t mixedTracks = 0; 2385 size_t tracksWithEffect = 0; 2386 // counts only _active_ fast tracks 2387 size_t fastTracks = 0; 2388 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2389 2390 float masterVolume = mMasterVolume; 2391 bool masterMute = mMasterMute; 2392 2393 if (masterMute) { 2394 masterVolume = 0; 2395 } 2396 // Delegate master volume control to effect in output mix effect chain if needed 2397 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2398 if (chain != 0) { 2399 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2400 chain->setVolume_l(&v, &v); 2401 masterVolume = (float)((v + (1 << 23)) >> 24); 2402 chain.clear(); 2403 } 2404 2405 // prepare a new state to push 2406 FastMixerStateQueue *sq = NULL; 2407 FastMixerState *state = NULL; 2408 bool didModify = false; 2409 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2410 if (mFastMixer != NULL) { 2411 sq = mFastMixer->sq(); 2412 state = sq->begin(); 2413 } 2414 2415 for (size_t i=0 ; i<count ; i++) { 2416 sp<Track> t = mActiveTracks[i].promote(); 2417 if (t == 0) { 2418 continue; 2419 } 2420 2421 // this const just means the local variable doesn't change 2422 Track* const track = t.get(); 2423 2424 // process fast tracks 2425 if (track->isFastTrack()) { 2426 2427 // It's theoretically possible (though unlikely) for a fast track to be created 2428 // and then removed within the same normal mix cycle. This is not a problem, as 2429 // the track never becomes active so it's fast mixer slot is never touched. 2430 // The converse, of removing an (active) track and then creating a new track 2431 // at the identical fast mixer slot within the same normal mix cycle, 2432 // is impossible because the slot isn't marked available until the end of each cycle. 2433 int j = track->mFastIndex; 2434 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2435 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2436 FastTrack *fastTrack = &state->mFastTracks[j]; 2437 2438 // Determine whether the track is currently in underrun condition, 2439 // and whether it had a recent underrun. 2440 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2441 FastTrackUnderruns underruns = ftDump->mUnderruns; 2442 uint32_t recentFull = (underruns.mBitFields.mFull - 2443 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2444 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2445 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2446 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2447 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2448 uint32_t recentUnderruns = recentPartial + recentEmpty; 2449 track->mObservedUnderruns = underruns; 2450 // don't count underruns that occur while stopping or pausing 2451 // or stopped which can occur when flush() is called while active 2452 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2453 track->mUnderrunCount += recentUnderruns; 2454 } 2455 2456 // This is similar to the state machine for normal tracks, 2457 // with a few modifications for fast tracks. 2458 bool isActive = true; 2459 switch (track->mState) { 2460 case TrackBase::STOPPING_1: 2461 // track stays active in STOPPING_1 state until first underrun 2462 if (recentUnderruns > 0) { 2463 track->mState = TrackBase::STOPPING_2; 2464 } 2465 break; 2466 case TrackBase::PAUSING: 2467 // ramp down is not yet implemented 2468 track->setPaused(); 2469 break; 2470 case TrackBase::RESUMING: 2471 // ramp up is not yet implemented 2472 track->mState = TrackBase::ACTIVE; 2473 break; 2474 case TrackBase::ACTIVE: 2475 if (recentFull > 0 || recentPartial > 0) { 2476 // track has provided at least some frames recently: reset retry count 2477 track->mRetryCount = kMaxTrackRetries; 2478 } 2479 if (recentUnderruns == 0) { 2480 // no recent underruns: stay active 2481 break; 2482 } 2483 // there has recently been an underrun of some kind 2484 if (track->sharedBuffer() == 0) { 2485 // were any of the recent underruns "empty" (no frames available)? 2486 if (recentEmpty == 0) { 2487 // no, then ignore the partial underruns as they are allowed indefinitely 2488 break; 2489 } 2490 // there has recently been an "empty" underrun: decrement the retry counter 2491 if (--(track->mRetryCount) > 0) { 2492 break; 2493 } 2494 // indicate to client process that the track was disabled because of underrun; 2495 // it will then automatically call start() when data is available 2496 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2497 // remove from active list, but state remains ACTIVE [confusing but true] 2498 isActive = false; 2499 break; 2500 } 2501 // fall through 2502 case TrackBase::STOPPING_2: 2503 case TrackBase::PAUSED: 2504 case TrackBase::TERMINATED: 2505 case TrackBase::STOPPED: 2506 case TrackBase::FLUSHED: // flush() while active 2507 // Check for presentation complete if track is inactive 2508 // We have consumed all the buffers of this track. 2509 // This would be incomplete if we auto-paused on underrun 2510 { 2511 size_t audioHALFrames = 2512 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2513 size_t framesWritten = mBytesWritten / mFrameSize; 2514 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2515 // track stays in active list until presentation is complete 2516 break; 2517 } 2518 } 2519 if (track->isStopping_2()) { 2520 track->mState = TrackBase::STOPPED; 2521 } 2522 if (track->isStopped()) { 2523 // Can't reset directly, as fast mixer is still polling this track 2524 // track->reset(); 2525 // So instead mark this track as needing to be reset after push with ack 2526 resetMask |= 1 << i; 2527 } 2528 isActive = false; 2529 break; 2530 case TrackBase::IDLE: 2531 default: 2532 LOG_FATAL("unexpected track state %d", track->mState); 2533 } 2534 2535 if (isActive) { 2536 // was it previously inactive? 2537 if (!(state->mTrackMask & (1 << j))) { 2538 ExtendedAudioBufferProvider *eabp = track; 2539 VolumeProvider *vp = track; 2540 fastTrack->mBufferProvider = eabp; 2541 fastTrack->mVolumeProvider = vp; 2542 fastTrack->mSampleRate = track->mSampleRate; 2543 fastTrack->mChannelMask = track->mChannelMask; 2544 fastTrack->mGeneration++; 2545 state->mTrackMask |= 1 << j; 2546 didModify = true; 2547 // no acknowledgement required for newly active tracks 2548 } 2549 // cache the combined master volume and stream type volume for fast mixer; this 2550 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2551 track->mCachedVolume = track->isMuted() ? 2552 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2553 ++fastTracks; 2554 } else { 2555 // was it previously active? 2556 if (state->mTrackMask & (1 << j)) { 2557 fastTrack->mBufferProvider = NULL; 2558 fastTrack->mGeneration++; 2559 state->mTrackMask &= ~(1 << j); 2560 didModify = true; 2561 // If any fast tracks were removed, we must wait for acknowledgement 2562 // because we're about to decrement the last sp<> on those tracks. 2563 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2564 } else { 2565 LOG_FATAL("fast track %d should have been active", j); 2566 } 2567 tracksToRemove->add(track); 2568 // Avoids a misleading display in dumpsys 2569 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2570 } 2571 continue; 2572 } 2573 2574 { // local variable scope to avoid goto warning 2575 2576 audio_track_cblk_t* cblk = track->cblk(); 2577 2578 // The first time a track is added we wait 2579 // for all its buffers to be filled before processing it 2580 int name = track->name(); 2581 // make sure that we have enough frames to mix one full buffer. 2582 // enforce this condition only once to enable draining the buffer in case the client 2583 // app does not call stop() and relies on underrun to stop: 2584 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2585 // during last round 2586 uint32_t minFrames = 1; 2587 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2588 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2589 if (t->sampleRate() == mSampleRate) { 2590 minFrames = mNormalFrameCount; 2591 } else { 2592 // +1 for rounding and +1 for additional sample needed for interpolation 2593 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2594 // add frames already consumed but not yet released by the resampler 2595 // because cblk->framesReady() will include these frames 2596 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2597 // the minimum track buffer size is normally twice the number of frames necessary 2598 // to fill one buffer and the resampler should not leave more than one buffer worth 2599 // of unreleased frames after each pass, but just in case... 2600 ALOG_ASSERT(minFrames <= cblk->frameCount); 2601 } 2602 } 2603 if ((track->framesReady() >= minFrames) && track->isReady() && 2604 !track->isPaused() && !track->isTerminated()) 2605 { 2606 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2607 this); 2608 2609 mixedTracks++; 2610 2611 // track->mainBuffer() != mMixBuffer means there is an effect chain 2612 // connected to the track 2613 chain.clear(); 2614 if (track->mainBuffer() != mMixBuffer) { 2615 chain = getEffectChain_l(track->sessionId()); 2616 // Delegate volume control to effect in track effect chain if needed 2617 if (chain != 0) { 2618 tracksWithEffect++; 2619 } else { 2620 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2621 "session %d", 2622 name, track->sessionId()); 2623 } 2624 } 2625 2626 2627 int param = AudioMixer::VOLUME; 2628 if (track->mFillingUpStatus == Track::FS_FILLED) { 2629 // no ramp for the first volume setting 2630 track->mFillingUpStatus = Track::FS_ACTIVE; 2631 if (track->mState == TrackBase::RESUMING) { 2632 track->mState = TrackBase::ACTIVE; 2633 param = AudioMixer::RAMP_VOLUME; 2634 } 2635 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2636 } else if (cblk->server != 0) { 2637 // If the track is stopped before the first frame was mixed, 2638 // do not apply ramp 2639 param = AudioMixer::RAMP_VOLUME; 2640 } 2641 2642 // compute volume for this track 2643 uint32_t vl, vr, va; 2644 if (track->isMuted() || track->isPausing() || 2645 mStreamTypes[track->streamType()].mute) { 2646 vl = vr = va = 0; 2647 if (track->isPausing()) { 2648 track->setPaused(); 2649 } 2650 } else { 2651 2652 // read original volumes with volume control 2653 float typeVolume = mStreamTypes[track->streamType()].volume; 2654 float v = masterVolume * typeVolume; 2655 uint32_t vlr = cblk->getVolumeLR(); 2656 vl = vlr & 0xFFFF; 2657 vr = vlr >> 16; 2658 // track volumes come from shared memory, so can't be trusted and must be clamped 2659 if (vl > MAX_GAIN_INT) { 2660 ALOGV("Track left volume out of range: %04X", vl); 2661 vl = MAX_GAIN_INT; 2662 } 2663 if (vr > MAX_GAIN_INT) { 2664 ALOGV("Track right volume out of range: %04X", vr); 2665 vr = MAX_GAIN_INT; 2666 } 2667 // now apply the master volume and stream type volume 2668 vl = (uint32_t)(v * vl) << 12; 2669 vr = (uint32_t)(v * vr) << 12; 2670 // assuming master volume and stream type volume each go up to 1.0, 2671 // vl and vr are now in 8.24 format 2672 2673 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2674 // send level comes from shared memory and so may be corrupt 2675 if (sendLevel > MAX_GAIN_INT) { 2676 ALOGV("Track send level out of range: %04X", sendLevel); 2677 sendLevel = MAX_GAIN_INT; 2678 } 2679 va = (uint32_t)(v * sendLevel); 2680 } 2681 // Delegate volume control to effect in track effect chain if needed 2682 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2683 // Do not ramp volume if volume is controlled by effect 2684 param = AudioMixer::VOLUME; 2685 track->mHasVolumeController = true; 2686 } else { 2687 // force no volume ramp when volume controller was just disabled or removed 2688 // from effect chain to avoid volume spike 2689 if (track->mHasVolumeController) { 2690 param = AudioMixer::VOLUME; 2691 } 2692 track->mHasVolumeController = false; 2693 } 2694 2695 // Convert volumes from 8.24 to 4.12 format 2696 // This additional clamping is needed in case chain->setVolume_l() overshot 2697 vl = (vl + (1 << 11)) >> 12; 2698 if (vl > MAX_GAIN_INT) { 2699 vl = MAX_GAIN_INT; 2700 } 2701 vr = (vr + (1 << 11)) >> 12; 2702 if (vr > MAX_GAIN_INT) { 2703 vr = MAX_GAIN_INT; 2704 } 2705 2706 if (va > MAX_GAIN_INT) { 2707 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2708 } 2709 2710 // XXX: these things DON'T need to be done each time 2711 mAudioMixer->setBufferProvider(name, track); 2712 mAudioMixer->enable(name); 2713 2714 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2715 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2716 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2717 mAudioMixer->setParameter( 2718 name, 2719 AudioMixer::TRACK, 2720 AudioMixer::FORMAT, (void *)track->format()); 2721 mAudioMixer->setParameter( 2722 name, 2723 AudioMixer::TRACK, 2724 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2725 mAudioMixer->setParameter( 2726 name, 2727 AudioMixer::RESAMPLE, 2728 AudioMixer::SAMPLE_RATE, 2729 (void *)(cblk->sampleRate)); 2730 mAudioMixer->setParameter( 2731 name, 2732 AudioMixer::TRACK, 2733 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2734 mAudioMixer->setParameter( 2735 name, 2736 AudioMixer::TRACK, 2737 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2738 2739 // reset retry count 2740 track->mRetryCount = kMaxTrackRetries; 2741 2742 // If one track is ready, set the mixer ready if: 2743 // - the mixer was not ready during previous round OR 2744 // - no other track is not ready 2745 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2746 mixerStatus != MIXER_TRACKS_ENABLED) { 2747 mixerStatus = MIXER_TRACKS_READY; 2748 } 2749 } else { 2750 // clear effect chain input buffer if an active track underruns to avoid sending 2751 // previous audio buffer again to effects 2752 chain = getEffectChain_l(track->sessionId()); 2753 if (chain != 0) { 2754 chain->clearInputBuffer(); 2755 } 2756 2757 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2758 cblk->server, this); 2759 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2760 track->isStopped() || track->isPaused()) { 2761 // We have consumed all the buffers of this track. 2762 // Remove it from the list of active tracks. 2763 // TODO: use actual buffer filling status instead of latency when available from 2764 // audio HAL 2765 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2766 size_t framesWritten = mBytesWritten / mFrameSize; 2767 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2768 if (track->isStopped()) { 2769 track->reset(); 2770 } 2771 tracksToRemove->add(track); 2772 } 2773 } else { 2774 track->mUnderrunCount++; 2775 // No buffers for this track. Give it a few chances to 2776 // fill a buffer, then remove it from active list. 2777 if (--(track->mRetryCount) <= 0) { 2778 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2779 tracksToRemove->add(track); 2780 // indicate to client process that the track was disabled because of underrun; 2781 // it will then automatically call start() when data is available 2782 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2783 // If one track is not ready, mark the mixer also not ready if: 2784 // - the mixer was ready during previous round OR 2785 // - no other track is ready 2786 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2787 mixerStatus != MIXER_TRACKS_READY) { 2788 mixerStatus = MIXER_TRACKS_ENABLED; 2789 } 2790 } 2791 mAudioMixer->disable(name); 2792 } 2793 2794 } // local variable scope to avoid goto warning 2795track_is_ready: ; 2796 2797 } 2798 2799 // Push the new FastMixer state if necessary 2800 bool pauseAudioWatchdog = false; 2801 if (didModify) { 2802 state->mFastTracksGen++; 2803 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2804 if (kUseFastMixer == FastMixer_Dynamic && 2805 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2806 state->mCommand = FastMixerState::COLD_IDLE; 2807 state->mColdFutexAddr = &mFastMixerFutex; 2808 state->mColdGen++; 2809 mFastMixerFutex = 0; 2810 if (kUseFastMixer == FastMixer_Dynamic) { 2811 mNormalSink = mOutputSink; 2812 } 2813 // If we go into cold idle, need to wait for acknowledgement 2814 // so that fast mixer stops doing I/O. 2815 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2816 pauseAudioWatchdog = true; 2817 } 2818 sq->end(); 2819 } 2820 if (sq != NULL) { 2821 sq->end(didModify); 2822 sq->push(block); 2823 } 2824#ifdef AUDIO_WATCHDOG 2825 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2826 mAudioWatchdog->pause(); 2827 } 2828#endif 2829 2830 // Now perform the deferred reset on fast tracks that have stopped 2831 while (resetMask != 0) { 2832 size_t i = __builtin_ctz(resetMask); 2833 ALOG_ASSERT(i < count); 2834 resetMask &= ~(1 << i); 2835 sp<Track> t = mActiveTracks[i].promote(); 2836 if (t == 0) { 2837 continue; 2838 } 2839 Track* track = t.get(); 2840 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2841 track->reset(); 2842 } 2843 2844 // remove all the tracks that need to be... 2845 count = tracksToRemove->size(); 2846 if (CC_UNLIKELY(count)) { 2847 for (size_t i=0 ; i<count ; i++) { 2848 const sp<Track>& track = tracksToRemove->itemAt(i); 2849 mActiveTracks.remove(track); 2850 if (track->mainBuffer() != mMixBuffer) { 2851 chain = getEffectChain_l(track->sessionId()); 2852 if (chain != 0) { 2853 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2854 track->sessionId()); 2855 chain->decActiveTrackCnt(); 2856 } 2857 } 2858 if (track->isTerminated()) { 2859 removeTrack_l(track); 2860 } 2861 } 2862 } 2863 2864 // mix buffer must be cleared if all tracks are connected to an 2865 // effect chain as in this case the mixer will not write to 2866 // mix buffer and track effects will accumulate into it 2867 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2868 (mixedTracks == 0 && fastTracks > 0)) { 2869 // FIXME as a performance optimization, should remember previous zero status 2870 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2871 } 2872 2873 // if any fast tracks, then status is ready 2874 mMixerStatusIgnoringFastTracks = mixerStatus; 2875 if (fastTracks > 0) { 2876 mixerStatus = MIXER_TRACKS_READY; 2877 } 2878 return mixerStatus; 2879} 2880 2881// getTrackName_l() must be called with ThreadBase::mLock held 2882int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2883{ 2884 return mAudioMixer->getTrackName(channelMask, sessionId); 2885} 2886 2887// deleteTrackName_l() must be called with ThreadBase::mLock held 2888void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2889{ 2890 ALOGV("remove track (%d) and delete from mixer", name); 2891 mAudioMixer->deleteTrackName(name); 2892} 2893 2894// checkForNewParameters_l() must be called with ThreadBase::mLock held 2895bool AudioFlinger::MixerThread::checkForNewParameters_l() 2896{ 2897 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2898 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2899 bool reconfig = false; 2900 2901 while (!mNewParameters.isEmpty()) { 2902 2903 if (mFastMixer != NULL) { 2904 FastMixerStateQueue *sq = mFastMixer->sq(); 2905 FastMixerState *state = sq->begin(); 2906 if (!(state->mCommand & FastMixerState::IDLE)) { 2907 previousCommand = state->mCommand; 2908 state->mCommand = FastMixerState::HOT_IDLE; 2909 sq->end(); 2910 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2911 } else { 2912 sq->end(false /*didModify*/); 2913 } 2914 } 2915 2916 status_t status = NO_ERROR; 2917 String8 keyValuePair = mNewParameters[0]; 2918 AudioParameter param = AudioParameter(keyValuePair); 2919 int value; 2920 2921 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2922 reconfig = true; 2923 } 2924 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2925 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2926 status = BAD_VALUE; 2927 } else { 2928 reconfig = true; 2929 } 2930 } 2931 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2932 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2933 status = BAD_VALUE; 2934 } else { 2935 reconfig = true; 2936 } 2937 } 2938 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2939 // do not accept frame count changes if tracks are open as the track buffer 2940 // size depends on frame count and correct behavior would not be guaranteed 2941 // if frame count is changed after track creation 2942 if (!mTracks.isEmpty()) { 2943 status = INVALID_OPERATION; 2944 } else { 2945 reconfig = true; 2946 } 2947 } 2948 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2949#ifdef ADD_BATTERY_DATA 2950 // when changing the audio output device, call addBatteryData to notify 2951 // the change 2952 if (mOutDevice != value) { 2953 uint32_t params = 0; 2954 // check whether speaker is on 2955 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2956 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2957 } 2958 2959 audio_devices_t deviceWithoutSpeaker 2960 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2961 // check if any other device (except speaker) is on 2962 if (value & deviceWithoutSpeaker ) { 2963 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2964 } 2965 2966 if (params != 0) { 2967 addBatteryData(params); 2968 } 2969 } 2970#endif 2971 2972 // forward device change to effects that have requested to be 2973 // aware of attached audio device. 2974 mOutDevice = value; 2975 for (size_t i = 0; i < mEffectChains.size(); i++) { 2976 mEffectChains[i]->setDevice_l(mOutDevice); 2977 } 2978 } 2979 2980 if (status == NO_ERROR) { 2981 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2982 keyValuePair.string()); 2983 if (!mStandby && status == INVALID_OPERATION) { 2984 mOutput->stream->common.standby(&mOutput->stream->common); 2985 mStandby = true; 2986 mBytesWritten = 0; 2987 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2988 keyValuePair.string()); 2989 } 2990 if (status == NO_ERROR && reconfig) { 2991 delete mAudioMixer; 2992 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2993 mAudioMixer = NULL; 2994 readOutputParameters(); 2995 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2996 for (size_t i = 0; i < mTracks.size() ; i++) { 2997 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 2998 if (name < 0) { 2999 break; 3000 } 3001 mTracks[i]->mName = name; 3002 // limit track sample rate to 2 x new output sample rate 3003 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3004 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3005 } 3006 } 3007 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3008 } 3009 } 3010 3011 mNewParameters.removeAt(0); 3012 3013 mParamStatus = status; 3014 mParamCond.signal(); 3015 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3016 // already timed out waiting for the status and will never signal the condition. 3017 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3018 } 3019 3020 if (!(previousCommand & FastMixerState::IDLE)) { 3021 ALOG_ASSERT(mFastMixer != NULL); 3022 FastMixerStateQueue *sq = mFastMixer->sq(); 3023 FastMixerState *state = sq->begin(); 3024 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3025 state->mCommand = previousCommand; 3026 sq->end(); 3027 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3028 } 3029 3030 return reconfig; 3031} 3032 3033 3034void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3035{ 3036 const size_t SIZE = 256; 3037 char buffer[SIZE]; 3038 String8 result; 3039 3040 PlaybackThread::dumpInternals(fd, args); 3041 3042 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3043 result.append(buffer); 3044 write(fd, result.string(), result.size()); 3045 3046 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3047 FastMixerDumpState copy = mFastMixerDumpState; 3048 copy.dump(fd); 3049 3050#ifdef STATE_QUEUE_DUMP 3051 // Similar for state queue 3052 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3053 observerCopy.dump(fd); 3054 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3055 mutatorCopy.dump(fd); 3056#endif 3057 3058 // Write the tee output to a .wav file 3059 dumpTee(fd, mTeeSource, mId); 3060 3061#ifdef AUDIO_WATCHDOG 3062 if (mAudioWatchdog != 0) { 3063 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3064 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3065 wdCopy.dump(fd); 3066 } 3067#endif 3068} 3069 3070uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3071{ 3072 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3073} 3074 3075uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3076{ 3077 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3078} 3079 3080void AudioFlinger::MixerThread::cacheParameters_l() 3081{ 3082 PlaybackThread::cacheParameters_l(); 3083 3084 // FIXME: Relaxed timing because of a certain device that can't meet latency 3085 // Should be reduced to 2x after the vendor fixes the driver issue 3086 // increase threshold again due to low power audio mode. The way this warning 3087 // threshold is calculated and its usefulness should be reconsidered anyway. 3088 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3089} 3090 3091// ---------------------------------------------------------------------------- 3092 3093AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3094 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3095 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3096 // mLeftVolFloat, mRightVolFloat 3097{ 3098} 3099 3100AudioFlinger::DirectOutputThread::~DirectOutputThread() 3101{ 3102} 3103 3104AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3105 Vector< sp<Track> > *tracksToRemove 3106) 3107{ 3108 sp<Track> trackToRemove; 3109 3110 mixer_state mixerStatus = MIXER_IDLE; 3111 3112 // find out which tracks need to be processed 3113 if (mActiveTracks.size() != 0) { 3114 sp<Track> t = mActiveTracks[0].promote(); 3115 // The track died recently 3116 if (t == 0) { 3117 return MIXER_IDLE; 3118 } 3119 3120 Track* const track = t.get(); 3121 audio_track_cblk_t* cblk = track->cblk(); 3122 3123 // The first time a track is added we wait 3124 // for all its buffers to be filled before processing it 3125 uint32_t minFrames; 3126 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3127 minFrames = mNormalFrameCount; 3128 } else { 3129 minFrames = 1; 3130 } 3131 if ((track->framesReady() >= minFrames) && track->isReady() && 3132 !track->isPaused() && !track->isTerminated()) 3133 { 3134 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3135 3136 if (track->mFillingUpStatus == Track::FS_FILLED) { 3137 track->mFillingUpStatus = Track::FS_ACTIVE; 3138 mLeftVolFloat = mRightVolFloat = 0; 3139 if (track->mState == TrackBase::RESUMING) { 3140 track->mState = TrackBase::ACTIVE; 3141 } 3142 } 3143 3144 // compute volume for this track 3145 float left, right; 3146 if (track->isMuted() || mMasterMute || track->isPausing() || 3147 mStreamTypes[track->streamType()].mute) { 3148 left = right = 0; 3149 if (track->isPausing()) { 3150 track->setPaused(); 3151 } 3152 } else { 3153 float typeVolume = mStreamTypes[track->streamType()].volume; 3154 float v = mMasterVolume * typeVolume; 3155 uint32_t vlr = cblk->getVolumeLR(); 3156 float v_clamped = v * (vlr & 0xFFFF); 3157 if (v_clamped > MAX_GAIN) { 3158 v_clamped = MAX_GAIN; 3159 } 3160 left = v_clamped/MAX_GAIN; 3161 v_clamped = v * (vlr >> 16); 3162 if (v_clamped > MAX_GAIN) { 3163 v_clamped = MAX_GAIN; 3164 } 3165 right = v_clamped/MAX_GAIN; 3166 } 3167 3168 if (left != mLeftVolFloat || right != mRightVolFloat) { 3169 mLeftVolFloat = left; 3170 mRightVolFloat = right; 3171 3172 // Convert volumes from float to 8.24 3173 uint32_t vl = (uint32_t)(left * (1 << 24)); 3174 uint32_t vr = (uint32_t)(right * (1 << 24)); 3175 3176 // Delegate volume control to effect in track effect chain if needed 3177 // only one effect chain can be present on DirectOutputThread, so if 3178 // there is one, the track is connected to it 3179 if (!mEffectChains.isEmpty()) { 3180 // Do not ramp volume if volume is controlled by effect 3181 mEffectChains[0]->setVolume_l(&vl, &vr); 3182 left = (float)vl / (1 << 24); 3183 right = (float)vr / (1 << 24); 3184 } 3185 mOutput->stream->set_volume(mOutput->stream, left, right); 3186 } 3187 3188 // reset retry count 3189 track->mRetryCount = kMaxTrackRetriesDirect; 3190 mActiveTrack = t; 3191 mixerStatus = MIXER_TRACKS_READY; 3192 } else { 3193 // clear effect chain input buffer if an active track underruns to avoid sending 3194 // previous audio buffer again to effects 3195 if (!mEffectChains.isEmpty()) { 3196 mEffectChains[0]->clearInputBuffer(); 3197 } 3198 3199 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3200 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3201 track->isStopped() || track->isPaused()) { 3202 // We have consumed all the buffers of this track. 3203 // Remove it from the list of active tracks. 3204 // TODO: implement behavior for compressed audio 3205 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3206 size_t framesWritten = mBytesWritten / mFrameSize; 3207 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3208 if (track->isStopped()) { 3209 track->reset(); 3210 } 3211 trackToRemove = track; 3212 } 3213 } else { 3214 // No buffers for this track. Give it a few chances to 3215 // fill a buffer, then remove it from active list. 3216 if (--(track->mRetryCount) <= 0) { 3217 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3218 trackToRemove = track; 3219 } else { 3220 mixerStatus = MIXER_TRACKS_ENABLED; 3221 } 3222 } 3223 } 3224 } 3225 3226 // FIXME merge this with similar code for removing multiple tracks 3227 // remove all the tracks that need to be... 3228 if (CC_UNLIKELY(trackToRemove != 0)) { 3229 tracksToRemove->add(trackToRemove); 3230 mActiveTracks.remove(trackToRemove); 3231 if (!mEffectChains.isEmpty()) { 3232 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3233 trackToRemove->sessionId()); 3234 mEffectChains[0]->decActiveTrackCnt(); 3235 } 3236 if (trackToRemove->isTerminated()) { 3237 removeTrack_l(trackToRemove); 3238 } 3239 } 3240 3241 return mixerStatus; 3242} 3243 3244void AudioFlinger::DirectOutputThread::threadLoop_mix() 3245{ 3246 AudioBufferProvider::Buffer buffer; 3247 size_t frameCount = mFrameCount; 3248 int8_t *curBuf = (int8_t *)mMixBuffer; 3249 // output audio to hardware 3250 while (frameCount) { 3251 buffer.frameCount = frameCount; 3252 mActiveTrack->getNextBuffer(&buffer); 3253 if (CC_UNLIKELY(buffer.raw == NULL)) { 3254 memset(curBuf, 0, frameCount * mFrameSize); 3255 break; 3256 } 3257 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3258 frameCount -= buffer.frameCount; 3259 curBuf += buffer.frameCount * mFrameSize; 3260 mActiveTrack->releaseBuffer(&buffer); 3261 } 3262 sleepTime = 0; 3263 standbyTime = systemTime() + standbyDelay; 3264 mActiveTrack.clear(); 3265 3266} 3267 3268void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3269{ 3270 if (sleepTime == 0) { 3271 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3272 sleepTime = activeSleepTime; 3273 } else { 3274 sleepTime = idleSleepTime; 3275 } 3276 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3277 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3278 sleepTime = 0; 3279 } 3280} 3281 3282// getTrackName_l() must be called with ThreadBase::mLock held 3283int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3284 int sessionId) 3285{ 3286 return 0; 3287} 3288 3289// deleteTrackName_l() must be called with ThreadBase::mLock held 3290void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3291{ 3292} 3293 3294// checkForNewParameters_l() must be called with ThreadBase::mLock held 3295bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3296{ 3297 bool reconfig = false; 3298 3299 while (!mNewParameters.isEmpty()) { 3300 status_t status = NO_ERROR; 3301 String8 keyValuePair = mNewParameters[0]; 3302 AudioParameter param = AudioParameter(keyValuePair); 3303 int value; 3304 3305 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3306 // do not accept frame count changes if tracks are open as the track buffer 3307 // size depends on frame count and correct behavior would not be garantied 3308 // if frame count is changed after track creation 3309 if (!mTracks.isEmpty()) { 3310 status = INVALID_OPERATION; 3311 } else { 3312 reconfig = true; 3313 } 3314 } 3315 if (status == NO_ERROR) { 3316 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3317 keyValuePair.string()); 3318 if (!mStandby && status == INVALID_OPERATION) { 3319 mOutput->stream->common.standby(&mOutput->stream->common); 3320 mStandby = true; 3321 mBytesWritten = 0; 3322 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3323 keyValuePair.string()); 3324 } 3325 if (status == NO_ERROR && reconfig) { 3326 readOutputParameters(); 3327 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3328 } 3329 } 3330 3331 mNewParameters.removeAt(0); 3332 3333 mParamStatus = status; 3334 mParamCond.signal(); 3335 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3336 // already timed out waiting for the status and will never signal the condition. 3337 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3338 } 3339 return reconfig; 3340} 3341 3342uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3343{ 3344 uint32_t time; 3345 if (audio_is_linear_pcm(mFormat)) { 3346 time = PlaybackThread::activeSleepTimeUs(); 3347 } else { 3348 time = 10000; 3349 } 3350 return time; 3351} 3352 3353uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3354{ 3355 uint32_t time; 3356 if (audio_is_linear_pcm(mFormat)) { 3357 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3358 } else { 3359 time = 10000; 3360 } 3361 return time; 3362} 3363 3364uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3365{ 3366 uint32_t time; 3367 if (audio_is_linear_pcm(mFormat)) { 3368 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3369 } else { 3370 time = 10000; 3371 } 3372 return time; 3373} 3374 3375void AudioFlinger::DirectOutputThread::cacheParameters_l() 3376{ 3377 PlaybackThread::cacheParameters_l(); 3378 3379 // use shorter standby delay as on normal output to release 3380 // hardware resources as soon as possible 3381 standbyDelay = microseconds(activeSleepTime*2); 3382} 3383 3384// ---------------------------------------------------------------------------- 3385 3386AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3387 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3388 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3389 DUPLICATING), 3390 mWaitTimeMs(UINT_MAX) 3391{ 3392 addOutputTrack(mainThread); 3393} 3394 3395AudioFlinger::DuplicatingThread::~DuplicatingThread() 3396{ 3397 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3398 mOutputTracks[i]->destroy(); 3399 } 3400} 3401 3402void AudioFlinger::DuplicatingThread::threadLoop_mix() 3403{ 3404 // mix buffers... 3405 if (outputsReady(outputTracks)) { 3406 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3407 } else { 3408 memset(mMixBuffer, 0, mixBufferSize); 3409 } 3410 sleepTime = 0; 3411 writeFrames = mNormalFrameCount; 3412 standbyTime = systemTime() + standbyDelay; 3413} 3414 3415void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3416{ 3417 if (sleepTime == 0) { 3418 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3419 sleepTime = activeSleepTime; 3420 } else { 3421 sleepTime = idleSleepTime; 3422 } 3423 } else if (mBytesWritten != 0) { 3424 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3425 writeFrames = mNormalFrameCount; 3426 memset(mMixBuffer, 0, mixBufferSize); 3427 } else { 3428 // flush remaining overflow buffers in output tracks 3429 writeFrames = 0; 3430 } 3431 sleepTime = 0; 3432 } 3433} 3434 3435void AudioFlinger::DuplicatingThread::threadLoop_write() 3436{ 3437 for (size_t i = 0; i < outputTracks.size(); i++) { 3438 outputTracks[i]->write(mMixBuffer, writeFrames); 3439 } 3440 mBytesWritten += mixBufferSize; 3441} 3442 3443void AudioFlinger::DuplicatingThread::threadLoop_standby() 3444{ 3445 // DuplicatingThread implements standby by stopping all tracks 3446 for (size_t i = 0; i < outputTracks.size(); i++) { 3447 outputTracks[i]->stop(); 3448 } 3449} 3450 3451void AudioFlinger::DuplicatingThread::saveOutputTracks() 3452{ 3453 outputTracks = mOutputTracks; 3454} 3455 3456void AudioFlinger::DuplicatingThread::clearOutputTracks() 3457{ 3458 outputTracks.clear(); 3459} 3460 3461void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3462{ 3463 Mutex::Autolock _l(mLock); 3464 // FIXME explain this formula 3465 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3466 OutputTrack *outputTrack = new OutputTrack(thread, 3467 this, 3468 mSampleRate, 3469 mFormat, 3470 mChannelMask, 3471 frameCount); 3472 if (outputTrack->cblk() != NULL) { 3473 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3474 mOutputTracks.add(outputTrack); 3475 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3476 updateWaitTime_l(); 3477 } 3478} 3479 3480void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3481{ 3482 Mutex::Autolock _l(mLock); 3483 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3484 if (mOutputTracks[i]->thread() == thread) { 3485 mOutputTracks[i]->destroy(); 3486 mOutputTracks.removeAt(i); 3487 updateWaitTime_l(); 3488 return; 3489 } 3490 } 3491 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3492} 3493 3494// caller must hold mLock 3495void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3496{ 3497 mWaitTimeMs = UINT_MAX; 3498 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3499 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3500 if (strong != 0) { 3501 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3502 if (waitTimeMs < mWaitTimeMs) { 3503 mWaitTimeMs = waitTimeMs; 3504 } 3505 } 3506 } 3507} 3508 3509 3510bool AudioFlinger::DuplicatingThread::outputsReady( 3511 const SortedVector< sp<OutputTrack> > &outputTracks) 3512{ 3513 for (size_t i = 0; i < outputTracks.size(); i++) { 3514 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3515 if (thread == 0) { 3516 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3517 outputTracks[i].get()); 3518 return false; 3519 } 3520 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3521 // see note at standby() declaration 3522 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3523 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3524 thread.get()); 3525 return false; 3526 } 3527 } 3528 return true; 3529} 3530 3531uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3532{ 3533 return (mWaitTimeMs * 1000) / 2; 3534} 3535 3536void AudioFlinger::DuplicatingThread::cacheParameters_l() 3537{ 3538 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3539 updateWaitTime_l(); 3540 3541 MixerThread::cacheParameters_l(); 3542} 3543 3544// ---------------------------------------------------------------------------- 3545// Record 3546// ---------------------------------------------------------------------------- 3547 3548AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3549 AudioStreamIn *input, 3550 uint32_t sampleRate, 3551 audio_channel_mask_t channelMask, 3552 audio_io_handle_t id, 3553 audio_devices_t device, 3554 const sp<NBAIO_Sink>& teeSink) : 3555 ThreadBase(audioFlinger, id, AUDIO_DEVICE_NONE, device, RECORD), 3556 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3557 // mRsmpInIndex and mInputBytes set by readInputParameters() 3558 mReqChannelCount(popcount(channelMask)), 3559 mReqSampleRate(sampleRate), 3560 // mBytesRead is only meaningful while active, and so is cleared in start() 3561 // (but might be better to also clear here for dump?) 3562 mTeeSink(teeSink) 3563{ 3564 snprintf(mName, kNameLength, "AudioIn_%X", id); 3565 3566 readInputParameters(); 3567 3568} 3569 3570 3571AudioFlinger::RecordThread::~RecordThread() 3572{ 3573 delete[] mRsmpInBuffer; 3574 delete mResampler; 3575 delete[] mRsmpOutBuffer; 3576} 3577 3578void AudioFlinger::RecordThread::onFirstRef() 3579{ 3580 run(mName, PRIORITY_URGENT_AUDIO); 3581} 3582 3583status_t AudioFlinger::RecordThread::readyToRun() 3584{ 3585 status_t status = initCheck(); 3586 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3587 return status; 3588} 3589 3590bool AudioFlinger::RecordThread::threadLoop() 3591{ 3592 AudioBufferProvider::Buffer buffer; 3593 sp<RecordTrack> activeTrack; 3594 Vector< sp<EffectChain> > effectChains; 3595 3596 nsecs_t lastWarning = 0; 3597 3598 inputStandBy(); 3599 acquireWakeLock(); 3600 3601 // used to verify we've read at least once before evaluating how many bytes were read 3602 bool readOnce = false; 3603 3604 // start recording 3605 while (!exitPending()) { 3606 3607 processConfigEvents(); 3608 3609 { // scope for mLock 3610 Mutex::Autolock _l(mLock); 3611 checkForNewParameters_l(); 3612 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3613 standby(); 3614 3615 if (exitPending()) { 3616 break; 3617 } 3618 3619 releaseWakeLock_l(); 3620 ALOGV("RecordThread: loop stopping"); 3621 // go to sleep 3622 mWaitWorkCV.wait(mLock); 3623 ALOGV("RecordThread: loop starting"); 3624 acquireWakeLock_l(); 3625 continue; 3626 } 3627 if (mActiveTrack != 0) { 3628 if (mActiveTrack->mState == TrackBase::PAUSING) { 3629 standby(); 3630 mActiveTrack.clear(); 3631 mStartStopCond.broadcast(); 3632 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3633 if (mReqChannelCount != mActiveTrack->channelCount()) { 3634 mActiveTrack.clear(); 3635 mStartStopCond.broadcast(); 3636 } else if (readOnce) { 3637 // record start succeeds only if first read from audio input 3638 // succeeds 3639 if (mBytesRead >= 0) { 3640 mActiveTrack->mState = TrackBase::ACTIVE; 3641 } else { 3642 mActiveTrack.clear(); 3643 } 3644 mStartStopCond.broadcast(); 3645 } 3646 mStandby = false; 3647 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3648 removeTrack_l(mActiveTrack); 3649 mActiveTrack.clear(); 3650 } 3651 } 3652 lockEffectChains_l(effectChains); 3653 } 3654 3655 if (mActiveTrack != 0) { 3656 if (mActiveTrack->mState != TrackBase::ACTIVE && 3657 mActiveTrack->mState != TrackBase::RESUMING) { 3658 unlockEffectChains(effectChains); 3659 usleep(kRecordThreadSleepUs); 3660 continue; 3661 } 3662 for (size_t i = 0; i < effectChains.size(); i ++) { 3663 effectChains[i]->process_l(); 3664 } 3665 3666 buffer.frameCount = mFrameCount; 3667 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3668 readOnce = true; 3669 size_t framesOut = buffer.frameCount; 3670 if (mResampler == NULL) { 3671 // no resampling 3672 while (framesOut) { 3673 size_t framesIn = mFrameCount - mRsmpInIndex; 3674 if (framesIn) { 3675 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3676 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3677 mActiveTrack->mFrameSize; 3678 if (framesIn > framesOut) 3679 framesIn = framesOut; 3680 mRsmpInIndex += framesIn; 3681 framesOut -= framesIn; 3682 if (mChannelCount == mReqChannelCount || 3683 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3684 memcpy(dst, src, framesIn * mFrameSize); 3685 } else { 3686 if (mChannelCount == 1) { 3687 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3688 (int16_t *)src, framesIn); 3689 } else { 3690 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3691 (int16_t *)src, framesIn); 3692 } 3693 } 3694 } 3695 if (framesOut && mFrameCount == mRsmpInIndex) { 3696 void *readInto; 3697 if (framesOut == mFrameCount && 3698 (mChannelCount == mReqChannelCount || 3699 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3700 readInto = buffer.raw; 3701 framesOut = 0; 3702 } else { 3703 readInto = mRsmpInBuffer; 3704 mRsmpInIndex = 0; 3705 } 3706 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3707 if (mBytesRead <= 0) { 3708 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3709 { 3710 ALOGE("Error reading audio input"); 3711 // Force input into standby so that it tries to 3712 // recover at next read attempt 3713 inputStandBy(); 3714 usleep(kRecordThreadSleepUs); 3715 } 3716 mRsmpInIndex = mFrameCount; 3717 framesOut = 0; 3718 buffer.frameCount = 0; 3719 } else if (mTeeSink != 0) { 3720 (void) mTeeSink->write(readInto, 3721 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3722 } 3723 } 3724 } 3725 } else { 3726 // resampling 3727 3728 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3729 // alter output frame count as if we were expecting stereo samples 3730 if (mChannelCount == 1 && mReqChannelCount == 1) { 3731 framesOut >>= 1; 3732 } 3733 mResampler->resample(mRsmpOutBuffer, framesOut, 3734 this /* AudioBufferProvider* */); 3735 // ditherAndClamp() works as long as all buffers returned by 3736 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3737 if (mChannelCount == 2 && mReqChannelCount == 1) { 3738 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3739 // the resampler always outputs stereo samples: 3740 // do post stereo to mono conversion 3741 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3742 framesOut); 3743 } else { 3744 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3745 } 3746 3747 } 3748 if (mFramestoDrop == 0) { 3749 mActiveTrack->releaseBuffer(&buffer); 3750 } else { 3751 if (mFramestoDrop > 0) { 3752 mFramestoDrop -= buffer.frameCount; 3753 if (mFramestoDrop <= 0) { 3754 clearSyncStartEvent(); 3755 } 3756 } else { 3757 mFramestoDrop += buffer.frameCount; 3758 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3759 mSyncStartEvent->isCancelled()) { 3760 ALOGW("Synced record %s, session %d, trigger session %d", 3761 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3762 mActiveTrack->sessionId(), 3763 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3764 clearSyncStartEvent(); 3765 } 3766 } 3767 } 3768 mActiveTrack->clearOverflow(); 3769 } 3770 // client isn't retrieving buffers fast enough 3771 else { 3772 if (!mActiveTrack->setOverflow()) { 3773 nsecs_t now = systemTime(); 3774 if ((now - lastWarning) > kWarningThrottleNs) { 3775 ALOGW("RecordThread: buffer overflow"); 3776 lastWarning = now; 3777 } 3778 } 3779 // Release the processor for a while before asking for a new buffer. 3780 // This will give the application more chance to read from the buffer and 3781 // clear the overflow. 3782 usleep(kRecordThreadSleepUs); 3783 } 3784 } 3785 // enable changes in effect chain 3786 unlockEffectChains(effectChains); 3787 effectChains.clear(); 3788 } 3789 3790 standby(); 3791 3792 { 3793 Mutex::Autolock _l(mLock); 3794 mActiveTrack.clear(); 3795 mStartStopCond.broadcast(); 3796 } 3797 3798 releaseWakeLock(); 3799 3800 ALOGV("RecordThread %p exiting", this); 3801 return false; 3802} 3803 3804void AudioFlinger::RecordThread::standby() 3805{ 3806 if (!mStandby) { 3807 inputStandBy(); 3808 mStandby = true; 3809 } 3810} 3811 3812void AudioFlinger::RecordThread::inputStandBy() 3813{ 3814 mInput->stream->common.standby(&mInput->stream->common); 3815} 3816 3817sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3818 const sp<AudioFlinger::Client>& client, 3819 uint32_t sampleRate, 3820 audio_format_t format, 3821 audio_channel_mask_t channelMask, 3822 size_t frameCount, 3823 int sessionId, 3824 IAudioFlinger::track_flags_t flags, 3825 pid_t tid, 3826 status_t *status) 3827{ 3828 sp<RecordTrack> track; 3829 status_t lStatus; 3830 3831 lStatus = initCheck(); 3832 if (lStatus != NO_ERROR) { 3833 ALOGE("Audio driver not initialized."); 3834 goto Exit; 3835 } 3836 3837 // FIXME use flags and tid similar to createTrack_l() 3838 3839 { // scope for mLock 3840 Mutex::Autolock _l(mLock); 3841 3842 track = new RecordTrack(this, client, sampleRate, 3843 format, channelMask, frameCount, sessionId); 3844 3845 if (track->getCblk() == 0) { 3846 lStatus = NO_MEMORY; 3847 goto Exit; 3848 } 3849 mTracks.add(track); 3850 3851 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3852 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3853 mAudioFlinger->btNrecIsOff(); 3854 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3855 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3856 } 3857 lStatus = NO_ERROR; 3858 3859Exit: 3860 if (status) { 3861 *status = lStatus; 3862 } 3863 return track; 3864} 3865 3866status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3867 AudioSystem::sync_event_t event, 3868 int triggerSession) 3869{ 3870 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3871 sp<ThreadBase> strongMe = this; 3872 status_t status = NO_ERROR; 3873 3874 if (event == AudioSystem::SYNC_EVENT_NONE) { 3875 clearSyncStartEvent(); 3876 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3877 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3878 triggerSession, 3879 recordTrack->sessionId(), 3880 syncStartEventCallback, 3881 this); 3882 // Sync event can be cancelled by the trigger session if the track is not in a 3883 // compatible state in which case we start record immediately 3884 if (mSyncStartEvent->isCancelled()) { 3885 clearSyncStartEvent(); 3886 } else { 3887 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3888 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3889 } 3890 } 3891 3892 { 3893 AutoMutex lock(mLock); 3894 if (mActiveTrack != 0) { 3895 if (recordTrack != mActiveTrack.get()) { 3896 status = -EBUSY; 3897 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3898 mActiveTrack->mState = TrackBase::ACTIVE; 3899 } 3900 return status; 3901 } 3902 3903 recordTrack->mState = TrackBase::IDLE; 3904 mActiveTrack = recordTrack; 3905 mLock.unlock(); 3906 status_t status = AudioSystem::startInput(mId); 3907 mLock.lock(); 3908 if (status != NO_ERROR) { 3909 mActiveTrack.clear(); 3910 clearSyncStartEvent(); 3911 return status; 3912 } 3913 mRsmpInIndex = mFrameCount; 3914 mBytesRead = 0; 3915 if (mResampler != NULL) { 3916 mResampler->reset(); 3917 } 3918 mActiveTrack->mState = TrackBase::RESUMING; 3919 // signal thread to start 3920 ALOGV("Signal record thread"); 3921 mWaitWorkCV.broadcast(); 3922 // do not wait for mStartStopCond if exiting 3923 if (exitPending()) { 3924 mActiveTrack.clear(); 3925 status = INVALID_OPERATION; 3926 goto startError; 3927 } 3928 mStartStopCond.wait(mLock); 3929 if (mActiveTrack == 0) { 3930 ALOGV("Record failed to start"); 3931 status = BAD_VALUE; 3932 goto startError; 3933 } 3934 ALOGV("Record started OK"); 3935 return status; 3936 } 3937startError: 3938 AudioSystem::stopInput(mId); 3939 clearSyncStartEvent(); 3940 return status; 3941} 3942 3943void AudioFlinger::RecordThread::clearSyncStartEvent() 3944{ 3945 if (mSyncStartEvent != 0) { 3946 mSyncStartEvent->cancel(); 3947 } 3948 mSyncStartEvent.clear(); 3949 mFramestoDrop = 0; 3950} 3951 3952void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3953{ 3954 sp<SyncEvent> strongEvent = event.promote(); 3955 3956 if (strongEvent != 0) { 3957 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3958 me->handleSyncStartEvent(strongEvent); 3959 } 3960} 3961 3962void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 3963{ 3964 if (event == mSyncStartEvent) { 3965 // TODO: use actual buffer filling status instead of 2 buffers when info is available 3966 // from audio HAL 3967 mFramestoDrop = mFrameCount * 2; 3968 } 3969} 3970 3971bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 3972 ALOGV("RecordThread::stop"); 3973 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 3974 return false; 3975 } 3976 recordTrack->mState = TrackBase::PAUSING; 3977 // do not wait for mStartStopCond if exiting 3978 if (exitPending()) { 3979 return true; 3980 } 3981 mStartStopCond.wait(mLock); 3982 // if we have been restarted, recordTrack == mActiveTrack.get() here 3983 if (exitPending() || recordTrack != mActiveTrack.get()) { 3984 ALOGV("Record stopped OK"); 3985 return true; 3986 } 3987 return false; 3988} 3989 3990bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 3991{ 3992 return false; 3993} 3994 3995status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 3996{ 3997#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 3998 if (!isValidSyncEvent(event)) { 3999 return BAD_VALUE; 4000 } 4001 4002 int eventSession = event->triggerSession(); 4003 status_t ret = NAME_NOT_FOUND; 4004 4005 Mutex::Autolock _l(mLock); 4006 4007 for (size_t i = 0; i < mTracks.size(); i++) { 4008 sp<RecordTrack> track = mTracks[i]; 4009 if (eventSession == track->sessionId()) { 4010 (void) track->setSyncEvent(event); 4011 ret = NO_ERROR; 4012 } 4013 } 4014 return ret; 4015#else 4016 return BAD_VALUE; 4017#endif 4018} 4019 4020// destroyTrack_l() must be called with ThreadBase::mLock held 4021void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4022{ 4023 track->mState = TrackBase::TERMINATED; 4024 // active tracks are removed by threadLoop() 4025 if (mActiveTrack != track) { 4026 removeTrack_l(track); 4027 } 4028} 4029 4030void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4031{ 4032 mTracks.remove(track); 4033 // need anything related to effects here? 4034} 4035 4036void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4037{ 4038 dumpInternals(fd, args); 4039 dumpTracks(fd, args); 4040 dumpEffectChains(fd, args); 4041} 4042 4043void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4044{ 4045 const size_t SIZE = 256; 4046 char buffer[SIZE]; 4047 String8 result; 4048 4049 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4050 result.append(buffer); 4051 4052 if (mActiveTrack != 0) { 4053 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4054 result.append(buffer); 4055 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4056 result.append(buffer); 4057 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4058 result.append(buffer); 4059 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4060 result.append(buffer); 4061 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4062 result.append(buffer); 4063 } else { 4064 result.append("No active record client\n"); 4065 } 4066 4067 write(fd, result.string(), result.size()); 4068 4069 dumpBase(fd, args); 4070} 4071 4072void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4073{ 4074 const size_t SIZE = 256; 4075 char buffer[SIZE]; 4076 String8 result; 4077 4078 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4079 result.append(buffer); 4080 RecordTrack::appendDumpHeader(result); 4081 for (size_t i = 0; i < mTracks.size(); ++i) { 4082 sp<RecordTrack> track = mTracks[i]; 4083 if (track != 0) { 4084 track->dump(buffer, SIZE); 4085 result.append(buffer); 4086 } 4087 } 4088 4089 if (mActiveTrack != 0) { 4090 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4091 result.append(buffer); 4092 RecordTrack::appendDumpHeader(result); 4093 mActiveTrack->dump(buffer, SIZE); 4094 result.append(buffer); 4095 4096 } 4097 write(fd, result.string(), result.size()); 4098} 4099 4100// AudioBufferProvider interface 4101status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4102{ 4103 size_t framesReq = buffer->frameCount; 4104 size_t framesReady = mFrameCount - mRsmpInIndex; 4105 int channelCount; 4106 4107 if (framesReady == 0) { 4108 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4109 if (mBytesRead <= 0) { 4110 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4111 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4112 // Force input into standby so that it tries to 4113 // recover at next read attempt 4114 inputStandBy(); 4115 usleep(kRecordThreadSleepUs); 4116 } 4117 buffer->raw = NULL; 4118 buffer->frameCount = 0; 4119 return NOT_ENOUGH_DATA; 4120 } 4121 mRsmpInIndex = 0; 4122 framesReady = mFrameCount; 4123 } 4124 4125 if (framesReq > framesReady) { 4126 framesReq = framesReady; 4127 } 4128 4129 if (mChannelCount == 1 && mReqChannelCount == 2) { 4130 channelCount = 1; 4131 } else { 4132 channelCount = 2; 4133 } 4134 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4135 buffer->frameCount = framesReq; 4136 return NO_ERROR; 4137} 4138 4139// AudioBufferProvider interface 4140void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4141{ 4142 mRsmpInIndex += buffer->frameCount; 4143 buffer->frameCount = 0; 4144} 4145 4146bool AudioFlinger::RecordThread::checkForNewParameters_l() 4147{ 4148 bool reconfig = false; 4149 4150 while (!mNewParameters.isEmpty()) { 4151 status_t status = NO_ERROR; 4152 String8 keyValuePair = mNewParameters[0]; 4153 AudioParameter param = AudioParameter(keyValuePair); 4154 int value; 4155 audio_format_t reqFormat = mFormat; 4156 uint32_t reqSamplingRate = mReqSampleRate; 4157 uint32_t reqChannelCount = mReqChannelCount; 4158 4159 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4160 reqSamplingRate = value; 4161 reconfig = true; 4162 } 4163 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4164 reqFormat = (audio_format_t) value; 4165 reconfig = true; 4166 } 4167 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4168 reqChannelCount = popcount(value); 4169 reconfig = true; 4170 } 4171 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4172 // do not accept frame count changes if tracks are open as the track buffer 4173 // size depends on frame count and correct behavior would not be guaranteed 4174 // if frame count is changed after track creation 4175 if (mActiveTrack != 0) { 4176 status = INVALID_OPERATION; 4177 } else { 4178 reconfig = true; 4179 } 4180 } 4181 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4182 // forward device change to effects that have requested to be 4183 // aware of attached audio device. 4184 for (size_t i = 0; i < mEffectChains.size(); i++) { 4185 mEffectChains[i]->setDevice_l(value); 4186 } 4187 4188 // store input device and output device but do not forward output device to audio HAL. 4189 // Note that status is ignored by the caller for output device 4190 // (see AudioFlinger::setParameters() 4191 if (audio_is_output_devices(value)) { 4192 mOutDevice = value; 4193 status = BAD_VALUE; 4194 } else { 4195 mInDevice = value; 4196 // disable AEC and NS if the device is a BT SCO headset supporting those 4197 // pre processings 4198 if (mTracks.size() > 0) { 4199 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4200 mAudioFlinger->btNrecIsOff(); 4201 for (size_t i = 0; i < mTracks.size(); i++) { 4202 sp<RecordTrack> track = mTracks[i]; 4203 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4204 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4205 } 4206 } 4207 } 4208 } 4209 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4210 mAudioSource != (audio_source_t)value) { 4211 // forward device change to effects that have requested to be 4212 // aware of attached audio device. 4213 for (size_t i = 0; i < mEffectChains.size(); i++) { 4214 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4215 } 4216 mAudioSource = (audio_source_t)value; 4217 } 4218 if (status == NO_ERROR) { 4219 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4220 keyValuePair.string()); 4221 if (status == INVALID_OPERATION) { 4222 inputStandBy(); 4223 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4224 keyValuePair.string()); 4225 } 4226 if (reconfig) { 4227 if (status == BAD_VALUE && 4228 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4229 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4230 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) 4231 <= (2 * reqSamplingRate)) && 4232 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4233 <= FCC_2 && 4234 (reqChannelCount <= FCC_2)) { 4235 status = NO_ERROR; 4236 } 4237 if (status == NO_ERROR) { 4238 readInputParameters(); 4239 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4240 } 4241 } 4242 } 4243 4244 mNewParameters.removeAt(0); 4245 4246 mParamStatus = status; 4247 mParamCond.signal(); 4248 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4249 // already timed out waiting for the status and will never signal the condition. 4250 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4251 } 4252 return reconfig; 4253} 4254 4255String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4256{ 4257 char *s; 4258 String8 out_s8 = String8(); 4259 4260 Mutex::Autolock _l(mLock); 4261 if (initCheck() != NO_ERROR) { 4262 return out_s8; 4263 } 4264 4265 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4266 out_s8 = String8(s); 4267 free(s); 4268 return out_s8; 4269} 4270 4271void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4272 AudioSystem::OutputDescriptor desc; 4273 void *param2 = NULL; 4274 4275 switch (event) { 4276 case AudioSystem::INPUT_OPENED: 4277 case AudioSystem::INPUT_CONFIG_CHANGED: 4278 desc.channels = mChannelMask; 4279 desc.samplingRate = mSampleRate; 4280 desc.format = mFormat; 4281 desc.frameCount = mFrameCount; 4282 desc.latency = 0; 4283 param2 = &desc; 4284 break; 4285 4286 case AudioSystem::INPUT_CLOSED: 4287 default: 4288 break; 4289 } 4290 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4291} 4292 4293void AudioFlinger::RecordThread::readInputParameters() 4294{ 4295 delete mRsmpInBuffer; 4296 // mRsmpInBuffer is always assigned a new[] below 4297 delete mRsmpOutBuffer; 4298 mRsmpOutBuffer = NULL; 4299 delete mResampler; 4300 mResampler = NULL; 4301 4302 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4303 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4304 mChannelCount = (uint16_t)popcount(mChannelMask); 4305 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4306 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4307 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4308 mFrameCount = mInputBytes / mFrameSize; 4309 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4310 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4311 4312 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4313 { 4314 int channelCount; 4315 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4316 // stereo to mono post process as the resampler always outputs stereo. 4317 if (mChannelCount == 1 && mReqChannelCount == 2) { 4318 channelCount = 1; 4319 } else { 4320 channelCount = 2; 4321 } 4322 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4323 mResampler->setSampleRate(mSampleRate); 4324 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4325 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4326 4327 // optmization: if mono to mono, alter input frame count as if we were inputing 4328 // stereo samples 4329 if (mChannelCount == 1 && mReqChannelCount == 1) { 4330 mFrameCount >>= 1; 4331 } 4332 4333 } 4334 mRsmpInIndex = mFrameCount; 4335} 4336 4337unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4338{ 4339 Mutex::Autolock _l(mLock); 4340 if (initCheck() != NO_ERROR) { 4341 return 0; 4342 } 4343 4344 return mInput->stream->get_input_frames_lost(mInput->stream); 4345} 4346 4347uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4348{ 4349 Mutex::Autolock _l(mLock); 4350 uint32_t result = 0; 4351 if (getEffectChain_l(sessionId) != 0) { 4352 result = EFFECT_SESSION; 4353 } 4354 4355 for (size_t i = 0; i < mTracks.size(); ++i) { 4356 if (sessionId == mTracks[i]->sessionId()) { 4357 result |= TRACK_SESSION; 4358 break; 4359 } 4360 } 4361 4362 return result; 4363} 4364 4365KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4366{ 4367 KeyedVector<int, bool> ids; 4368 Mutex::Autolock _l(mLock); 4369 for (size_t j = 0; j < mTracks.size(); ++j) { 4370 sp<RecordThread::RecordTrack> track = mTracks[j]; 4371 int sessionId = track->sessionId(); 4372 if (ids.indexOfKey(sessionId) < 0) { 4373 ids.add(sessionId, true); 4374 } 4375 } 4376 return ids; 4377} 4378 4379AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4380{ 4381 Mutex::Autolock _l(mLock); 4382 AudioStreamIn *input = mInput; 4383 mInput = NULL; 4384 return input; 4385} 4386 4387// this method must always be called either with ThreadBase mLock held or inside the thread loop 4388audio_stream_t* AudioFlinger::RecordThread::stream() const 4389{ 4390 if (mInput == NULL) { 4391 return NULL; 4392 } 4393 return &mInput->stream->common; 4394} 4395 4396status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4397{ 4398 // only one chain per input thread 4399 if (mEffectChains.size() != 0) { 4400 return INVALID_OPERATION; 4401 } 4402 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4403 4404 chain->setInBuffer(NULL); 4405 chain->setOutBuffer(NULL); 4406 4407 checkSuspendOnAddEffectChain_l(chain); 4408 4409 mEffectChains.add(chain); 4410 4411 return NO_ERROR; 4412} 4413 4414size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4415{ 4416 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4417 ALOGW_IF(mEffectChains.size() != 1, 4418 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4419 chain.get(), mEffectChains.size(), this); 4420 if (mEffectChains.size() == 1) { 4421 mEffectChains.removeAt(0); 4422 } 4423 return 0; 4424} 4425 4426}; // namespace android 4427