Threads.cpp revision 7f2f8042cf335ab1323dec3edbe9143a06109f4e
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include <math.h> 24#include <fcntl.h> 25#include <sys/stat.h> 26#include <cutils/properties.h> 27#include <cutils/compiler.h> 28#include <utils/Log.h> 29#include <utils/Trace.h> 30 31#include <private/media/AudioTrackShared.h> 32#include <hardware/audio.h> 33#include <audio_effects/effect_ns.h> 34#include <audio_effects/effect_aec.h> 35#include <audio_utils/primitives.h> 36 37// NBAIO implementations 38#include <media/nbaio/AudioStreamOutSink.h> 39#include <media/nbaio/MonoPipe.h> 40#include <media/nbaio/MonoPipeReader.h> 41#include <media/nbaio/Pipe.h> 42#include <media/nbaio/PipeReader.h> 43#include <media/nbaio/SourceAudioBufferProvider.h> 44 45#include <powermanager/PowerManager.h> 46 47#include <common_time/cc_helper.h> 48#include <common_time/local_clock.h> 49 50#include "AudioFlinger.h" 51#include "AudioMixer.h" 52#include "FastMixer.h" 53#include "ServiceUtilities.h" 54#include "SchedulingPolicyService.h" 55 56#undef ADD_BATTERY_DATA 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait for setParameters to complete 102static const nsecs_t kSetParametersTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal mix buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalMixBufferSizeMs = 20; 111// maximum normal mix buffer size 112static const uint32_t kMaxNormalMixBufferSizeMs = 24; 113 114// Whether to use fast mixer 115static const enum { 116 FastMixer_Never, // never initialize or use: for debugging only 117 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 118 // normal mixer multiplier is 1 119 FastMixer_Static, // initialize if needed, then use all the time if initialized, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 // FIXME for FastMixer_Dynamic: 124 // Supporting this option will require fixing HALs that can't handle large writes. 125 // For example, one HAL implementation returns an error from a large write, 126 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 127 // We could either fix the HAL implementations, or provide a wrapper that breaks 128 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 129} kUseFastMixer = FastMixer_Static; 130 131// Priorities for requestPriority 132static const int kPriorityAudioApp = 2; 133static const int kPriorityFastMixer = 3; 134 135// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 136// for the track. The client then sub-divides this into smaller buffers for its use. 137// Currently the client uses double-buffering by default, but doesn't tell us about that. 138// So for now we just assume that client is double-buffered. 139// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 140// N-buffering, so AudioFlinger could allocate the right amount of memory. 141// See the client's minBufCount and mNotificationFramesAct calculations for details. 142static const int kFastTrackMultiplier = 2; 143 144// ---------------------------------------------------------------------------- 145 146#ifdef ADD_BATTERY_DATA 147// To collect the amplifier usage 148static void addBatteryData(uint32_t params) { 149 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 150 if (service == NULL) { 151 // it already logged 152 return; 153 } 154 155 service->addBatteryData(params); 156} 157#endif 158 159 160// ---------------------------------------------------------------------------- 161// CPU Stats 162// ---------------------------------------------------------------------------- 163 164class CpuStats { 165public: 166 CpuStats(); 167 void sample(const String8 &title); 168#ifdef DEBUG_CPU_USAGE 169private: 170 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 171 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 172 173 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 174 175 int mCpuNum; // thread's current CPU number 176 int mCpukHz; // frequency of thread's current CPU in kHz 177#endif 178}; 179 180CpuStats::CpuStats() 181#ifdef DEBUG_CPU_USAGE 182 : mCpuNum(-1), mCpukHz(-1) 183#endif 184{ 185} 186 187void CpuStats::sample(const String8 &title) { 188#ifdef DEBUG_CPU_USAGE 189 // get current thread's delta CPU time in wall clock ns 190 double wcNs; 191 bool valid = mCpuUsage.sampleAndEnable(wcNs); 192 193 // record sample for wall clock statistics 194 if (valid) { 195 mWcStats.sample(wcNs); 196 } 197 198 // get the current CPU number 199 int cpuNum = sched_getcpu(); 200 201 // get the current CPU frequency in kHz 202 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 203 204 // check if either CPU number or frequency changed 205 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 206 mCpuNum = cpuNum; 207 mCpukHz = cpukHz; 208 // ignore sample for purposes of cycles 209 valid = false; 210 } 211 212 // if no change in CPU number or frequency, then record sample for cycle statistics 213 if (valid && mCpukHz > 0) { 214 double cycles = wcNs * cpukHz * 0.000001; 215 mHzStats.sample(cycles); 216 } 217 218 unsigned n = mWcStats.n(); 219 // mCpuUsage.elapsed() is expensive, so don't call it every loop 220 if ((n & 127) == 1) { 221 long long elapsed = mCpuUsage.elapsed(); 222 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 223 double perLoop = elapsed / (double) n; 224 double perLoop100 = perLoop * 0.01; 225 double perLoop1k = perLoop * 0.001; 226 double mean = mWcStats.mean(); 227 double stddev = mWcStats.stddev(); 228 double minimum = mWcStats.minimum(); 229 double maximum = mWcStats.maximum(); 230 double meanCycles = mHzStats.mean(); 231 double stddevCycles = mHzStats.stddev(); 232 double minCycles = mHzStats.minimum(); 233 double maxCycles = mHzStats.maximum(); 234 mCpuUsage.resetElapsed(); 235 mWcStats.reset(); 236 mHzStats.reset(); 237 ALOGD("CPU usage for %s over past %.1f secs\n" 238 " (%u mixer loops at %.1f mean ms per loop):\n" 239 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 240 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 241 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 242 title.string(), 243 elapsed * .000000001, n, perLoop * .000001, 244 mean * .001, 245 stddev * .001, 246 minimum * .001, 247 maximum * .001, 248 mean / perLoop100, 249 stddev / perLoop100, 250 minimum / perLoop100, 251 maximum / perLoop100, 252 meanCycles / perLoop1k, 253 stddevCycles / perLoop1k, 254 minCycles / perLoop1k, 255 maxCycles / perLoop1k); 256 257 } 258 } 259#endif 260}; 261 262// ---------------------------------------------------------------------------- 263// ThreadBase 264// ---------------------------------------------------------------------------- 265 266AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 267 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 268 : Thread(false /*canCallJava*/), 269 mType(type), 270 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 271 // mChannelMask 272 mChannelCount(0), 273 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 274 mParamStatus(NO_ERROR), 275 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 276 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 277 // mName will be set by concrete (non-virtual) subclass 278 mDeathRecipient(new PMDeathRecipient(this)) 279{ 280} 281 282AudioFlinger::ThreadBase::~ThreadBase() 283{ 284 mParamCond.broadcast(); 285 // do not lock the mutex in destructor 286 releaseWakeLock_l(); 287 if (mPowerManager != 0) { 288 sp<IBinder> binder = mPowerManager->asBinder(); 289 binder->unlinkToDeath(mDeathRecipient); 290 } 291} 292 293void AudioFlinger::ThreadBase::exit() 294{ 295 ALOGV("ThreadBase::exit"); 296 // do any cleanup required for exit to succeed 297 preExit(); 298 { 299 // This lock prevents the following race in thread (uniprocessor for illustration): 300 // if (!exitPending()) { 301 // // context switch from here to exit() 302 // // exit() calls requestExit(), what exitPending() observes 303 // // exit() calls signal(), which is dropped since no waiters 304 // // context switch back from exit() to here 305 // mWaitWorkCV.wait(...); 306 // // now thread is hung 307 // } 308 AutoMutex lock(mLock); 309 requestExit(); 310 mWaitWorkCV.broadcast(); 311 } 312 // When Thread::requestExitAndWait is made virtual and this method is renamed to 313 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 314 requestExitAndWait(); 315} 316 317status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 318{ 319 status_t status; 320 321 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 322 Mutex::Autolock _l(mLock); 323 324 mNewParameters.add(keyValuePairs); 325 mWaitWorkCV.signal(); 326 // wait condition with timeout in case the thread loop has exited 327 // before the request could be processed 328 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 329 status = mParamStatus; 330 mWaitWorkCV.signal(); 331 } else { 332 status = TIMED_OUT; 333 } 334 return status; 335} 336 337void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 338{ 339 Mutex::Autolock _l(mLock); 340 sendIoConfigEvent_l(event, param); 341} 342 343// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 344void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 345{ 346 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 347 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 348 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 349 param); 350 mWaitWorkCV.signal(); 351} 352 353// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 354void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 355{ 356 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 357 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 358 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 359 mConfigEvents.size(), pid, tid, prio); 360 mWaitWorkCV.signal(); 361} 362 363void AudioFlinger::ThreadBase::processConfigEvents() 364{ 365 mLock.lock(); 366 while (!mConfigEvents.isEmpty()) { 367 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 368 ConfigEvent *event = mConfigEvents[0]; 369 mConfigEvents.removeAt(0); 370 // release mLock before locking AudioFlinger mLock: lock order is always 371 // AudioFlinger then ThreadBase to avoid cross deadlock 372 mLock.unlock(); 373 switch(event->type()) { 374 case CFG_EVENT_PRIO: { 375 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 376 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio()); 377 if (err != 0) { 378 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 379 "error %d", 380 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 381 } 382 } break; 383 case CFG_EVENT_IO: { 384 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 385 mAudioFlinger->mLock.lock(); 386 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 387 mAudioFlinger->mLock.unlock(); 388 } break; 389 default: 390 ALOGE("processConfigEvents() unknown event type %d", event->type()); 391 break; 392 } 393 delete event; 394 mLock.lock(); 395 } 396 mLock.unlock(); 397} 398 399void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 400{ 401 const size_t SIZE = 256; 402 char buffer[SIZE]; 403 String8 result; 404 405 bool locked = AudioFlinger::dumpTryLock(mLock); 406 if (!locked) { 407 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 408 write(fd, buffer, strlen(buffer)); 409 } 410 411 snprintf(buffer, SIZE, "io handle: %d\n", mId); 412 result.append(buffer); 413 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 414 result.append(buffer); 415 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 416 result.append(buffer); 417 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 418 result.append(buffer); 419 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 420 result.append(buffer); 421 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 422 result.append(buffer); 423 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 424 result.append(buffer); 425 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 426 result.append(buffer); 427 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 430 result.append(buffer); 431 432 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 433 result.append(buffer); 434 result.append(" Index Command"); 435 for (size_t i = 0; i < mNewParameters.size(); ++i) { 436 snprintf(buffer, SIZE, "\n %02d ", i); 437 result.append(buffer); 438 result.append(mNewParameters[i]); 439 } 440 441 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 442 result.append(buffer); 443 for (size_t i = 0; i < mConfigEvents.size(); i++) { 444 mConfigEvents[i]->dump(buffer, SIZE); 445 result.append(buffer); 446 } 447 result.append("\n"); 448 449 write(fd, result.string(), result.size()); 450 451 if (locked) { 452 mLock.unlock(); 453 } 454} 455 456void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 457{ 458 const size_t SIZE = 256; 459 char buffer[SIZE]; 460 String8 result; 461 462 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 463 write(fd, buffer, strlen(buffer)); 464 465 for (size_t i = 0; i < mEffectChains.size(); ++i) { 466 sp<EffectChain> chain = mEffectChains[i]; 467 if (chain != 0) { 468 chain->dump(fd, args); 469 } 470 } 471} 472 473void AudioFlinger::ThreadBase::acquireWakeLock() 474{ 475 Mutex::Autolock _l(mLock); 476 acquireWakeLock_l(); 477} 478 479void AudioFlinger::ThreadBase::acquireWakeLock_l() 480{ 481 if (mPowerManager == 0) { 482 // use checkService() to avoid blocking if power service is not up yet 483 sp<IBinder> binder = 484 defaultServiceManager()->checkService(String16("power")); 485 if (binder == 0) { 486 ALOGW("Thread %s cannot connect to the power manager service", mName); 487 } else { 488 mPowerManager = interface_cast<IPowerManager>(binder); 489 binder->linkToDeath(mDeathRecipient); 490 } 491 } 492 if (mPowerManager != 0) { 493 sp<IBinder> binder = new BBinder(); 494 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 495 binder, 496 String16(mName)); 497 if (status == NO_ERROR) { 498 mWakeLockToken = binder; 499 } 500 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 501 } 502} 503 504void AudioFlinger::ThreadBase::releaseWakeLock() 505{ 506 Mutex::Autolock _l(mLock); 507 releaseWakeLock_l(); 508} 509 510void AudioFlinger::ThreadBase::releaseWakeLock_l() 511{ 512 if (mWakeLockToken != 0) { 513 ALOGV("releaseWakeLock_l() %s", mName); 514 if (mPowerManager != 0) { 515 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 516 } 517 mWakeLockToken.clear(); 518 } 519} 520 521void AudioFlinger::ThreadBase::clearPowerManager() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525 mPowerManager.clear(); 526} 527 528void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 529{ 530 sp<ThreadBase> thread = mThread.promote(); 531 if (thread != 0) { 532 thread->clearPowerManager(); 533 } 534 ALOGW("power manager service died !!!"); 535} 536 537void AudioFlinger::ThreadBase::setEffectSuspended( 538 const effect_uuid_t *type, bool suspend, int sessionId) 539{ 540 Mutex::Autolock _l(mLock); 541 setEffectSuspended_l(type, suspend, sessionId); 542} 543 544void AudioFlinger::ThreadBase::setEffectSuspended_l( 545 const effect_uuid_t *type, bool suspend, int sessionId) 546{ 547 sp<EffectChain> chain = getEffectChain_l(sessionId); 548 if (chain != 0) { 549 if (type != NULL) { 550 chain->setEffectSuspended_l(type, suspend); 551 } else { 552 chain->setEffectSuspendedAll_l(suspend); 553 } 554 } 555 556 updateSuspendedSessions_l(type, suspend, sessionId); 557} 558 559void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 560{ 561 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 562 if (index < 0) { 563 return; 564 } 565 566 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 567 mSuspendedSessions.valueAt(index); 568 569 for (size_t i = 0; i < sessionEffects.size(); i++) { 570 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 571 for (int j = 0; j < desc->mRefCount; j++) { 572 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 573 chain->setEffectSuspendedAll_l(true); 574 } else { 575 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 576 desc->mType.timeLow); 577 chain->setEffectSuspended_l(&desc->mType, true); 578 } 579 } 580 } 581} 582 583void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 584 bool suspend, 585 int sessionId) 586{ 587 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 588 589 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 590 591 if (suspend) { 592 if (index >= 0) { 593 sessionEffects = mSuspendedSessions.valueAt(index); 594 } else { 595 mSuspendedSessions.add(sessionId, sessionEffects); 596 } 597 } else { 598 if (index < 0) { 599 return; 600 } 601 sessionEffects = mSuspendedSessions.valueAt(index); 602 } 603 604 605 int key = EffectChain::kKeyForSuspendAll; 606 if (type != NULL) { 607 key = type->timeLow; 608 } 609 index = sessionEffects.indexOfKey(key); 610 611 sp<SuspendedSessionDesc> desc; 612 if (suspend) { 613 if (index >= 0) { 614 desc = sessionEffects.valueAt(index); 615 } else { 616 desc = new SuspendedSessionDesc(); 617 if (type != NULL) { 618 desc->mType = *type; 619 } 620 sessionEffects.add(key, desc); 621 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 622 } 623 desc->mRefCount++; 624 } else { 625 if (index < 0) { 626 return; 627 } 628 desc = sessionEffects.valueAt(index); 629 if (--desc->mRefCount == 0) { 630 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 631 sessionEffects.removeItemsAt(index); 632 if (sessionEffects.isEmpty()) { 633 ALOGV("updateSuspendedSessions_l() restore removing session %d", 634 sessionId); 635 mSuspendedSessions.removeItem(sessionId); 636 } 637 } 638 } 639 if (!sessionEffects.isEmpty()) { 640 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 641 } 642} 643 644void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 645 bool enabled, 646 int sessionId) 647{ 648 Mutex::Autolock _l(mLock); 649 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 650} 651 652void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 653 bool enabled, 654 int sessionId) 655{ 656 if (mType != RECORD) { 657 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 658 // another session. This gives the priority to well behaved effect control panels 659 // and applications not using global effects. 660 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 661 // global effects 662 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 663 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 664 } 665 } 666 667 sp<EffectChain> chain = getEffectChain_l(sessionId); 668 if (chain != 0) { 669 chain->checkSuspendOnEffectEnabled(effect, enabled); 670 } 671} 672 673// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 674sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 675 const sp<AudioFlinger::Client>& client, 676 const sp<IEffectClient>& effectClient, 677 int32_t priority, 678 int sessionId, 679 effect_descriptor_t *desc, 680 int *enabled, 681 status_t *status 682 ) 683{ 684 sp<EffectModule> effect; 685 sp<EffectHandle> handle; 686 status_t lStatus; 687 sp<EffectChain> chain; 688 bool chainCreated = false; 689 bool effectCreated = false; 690 bool effectRegistered = false; 691 692 lStatus = initCheck(); 693 if (lStatus != NO_ERROR) { 694 ALOGW("createEffect_l() Audio driver not initialized."); 695 goto Exit; 696 } 697 698 // Do not allow effects with session ID 0 on direct output or duplicating threads 699 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 700 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 701 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 702 desc->name, sessionId); 703 lStatus = BAD_VALUE; 704 goto Exit; 705 } 706 // Only Pre processor effects are allowed on input threads and only on input threads 707 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 708 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 709 desc->name, desc->flags, mType); 710 lStatus = BAD_VALUE; 711 goto Exit; 712 } 713 714 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 715 716 { // scope for mLock 717 Mutex::Autolock _l(mLock); 718 719 // check for existing effect chain with the requested audio session 720 chain = getEffectChain_l(sessionId); 721 if (chain == 0) { 722 // create a new chain for this session 723 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 724 chain = new EffectChain(this, sessionId); 725 addEffectChain_l(chain); 726 chain->setStrategy(getStrategyForSession_l(sessionId)); 727 chainCreated = true; 728 } else { 729 effect = chain->getEffectFromDesc_l(desc); 730 } 731 732 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 733 734 if (effect == 0) { 735 int id = mAudioFlinger->nextUniqueId(); 736 // Check CPU and memory usage 737 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 738 if (lStatus != NO_ERROR) { 739 goto Exit; 740 } 741 effectRegistered = true; 742 // create a new effect module if none present in the chain 743 effect = new EffectModule(this, chain, desc, id, sessionId); 744 lStatus = effect->status(); 745 if (lStatus != NO_ERROR) { 746 goto Exit; 747 } 748 lStatus = chain->addEffect_l(effect); 749 if (lStatus != NO_ERROR) { 750 goto Exit; 751 } 752 effectCreated = true; 753 754 effect->setDevice(mOutDevice); 755 effect->setDevice(mInDevice); 756 effect->setMode(mAudioFlinger->getMode()); 757 effect->setAudioSource(mAudioSource); 758 } 759 // create effect handle and connect it to effect module 760 handle = new EffectHandle(effect, client, effectClient, priority); 761 lStatus = effect->addHandle(handle.get()); 762 if (enabled != NULL) { 763 *enabled = (int)effect->isEnabled(); 764 } 765 } 766 767Exit: 768 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 769 Mutex::Autolock _l(mLock); 770 if (effectCreated) { 771 chain->removeEffect_l(effect); 772 } 773 if (effectRegistered) { 774 AudioSystem::unregisterEffect(effect->id()); 775 } 776 if (chainCreated) { 777 removeEffectChain_l(chain); 778 } 779 handle.clear(); 780 } 781 782 if (status != NULL) { 783 *status = lStatus; 784 } 785 return handle; 786} 787 788sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 789{ 790 Mutex::Autolock _l(mLock); 791 return getEffect_l(sessionId, effectId); 792} 793 794sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 795{ 796 sp<EffectChain> chain = getEffectChain_l(sessionId); 797 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 798} 799 800// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 801// PlaybackThread::mLock held 802status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 803{ 804 // check for existing effect chain with the requested audio session 805 int sessionId = effect->sessionId(); 806 sp<EffectChain> chain = getEffectChain_l(sessionId); 807 bool chainCreated = false; 808 809 if (chain == 0) { 810 // create a new chain for this session 811 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 812 chain = new EffectChain(this, sessionId); 813 addEffectChain_l(chain); 814 chain->setStrategy(getStrategyForSession_l(sessionId)); 815 chainCreated = true; 816 } 817 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 818 819 if (chain->getEffectFromId_l(effect->id()) != 0) { 820 ALOGW("addEffect_l() %p effect %s already present in chain %p", 821 this, effect->desc().name, chain.get()); 822 return BAD_VALUE; 823 } 824 825 status_t status = chain->addEffect_l(effect); 826 if (status != NO_ERROR) { 827 if (chainCreated) { 828 removeEffectChain_l(chain); 829 } 830 return status; 831 } 832 833 effect->setDevice(mOutDevice); 834 effect->setDevice(mInDevice); 835 effect->setMode(mAudioFlinger->getMode()); 836 effect->setAudioSource(mAudioSource); 837 return NO_ERROR; 838} 839 840void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 841 842 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 843 effect_descriptor_t desc = effect->desc(); 844 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 845 detachAuxEffect_l(effect->id()); 846 } 847 848 sp<EffectChain> chain = effect->chain().promote(); 849 if (chain != 0) { 850 // remove effect chain if removing last effect 851 if (chain->removeEffect_l(effect) == 0) { 852 removeEffectChain_l(chain); 853 } 854 } else { 855 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 856 } 857} 858 859void AudioFlinger::ThreadBase::lockEffectChains_l( 860 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 861{ 862 effectChains = mEffectChains; 863 for (size_t i = 0; i < mEffectChains.size(); i++) { 864 mEffectChains[i]->lock(); 865 } 866} 867 868void AudioFlinger::ThreadBase::unlockEffectChains( 869 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 870{ 871 for (size_t i = 0; i < effectChains.size(); i++) { 872 effectChains[i]->unlock(); 873 } 874} 875 876sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 877{ 878 Mutex::Autolock _l(mLock); 879 return getEffectChain_l(sessionId); 880} 881 882sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 883{ 884 size_t size = mEffectChains.size(); 885 for (size_t i = 0; i < size; i++) { 886 if (mEffectChains[i]->sessionId() == sessionId) { 887 return mEffectChains[i]; 888 } 889 } 890 return 0; 891} 892 893void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 894{ 895 Mutex::Autolock _l(mLock); 896 size_t size = mEffectChains.size(); 897 for (size_t i = 0; i < size; i++) { 898 mEffectChains[i]->setMode_l(mode); 899 } 900} 901 902void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 903 EffectHandle *handle, 904 bool unpinIfLast) { 905 906 Mutex::Autolock _l(mLock); 907 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 908 // delete the effect module if removing last handle on it 909 if (effect->removeHandle(handle) == 0) { 910 if (!effect->isPinned() || unpinIfLast) { 911 removeEffect_l(effect); 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 } 915} 916 917// ---------------------------------------------------------------------------- 918// Playback 919// ---------------------------------------------------------------------------- 920 921AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 922 AudioStreamOut* output, 923 audio_io_handle_t id, 924 audio_devices_t device, 925 type_t type) 926 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 927 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 928 // mStreamTypes[] initialized in constructor body 929 mOutput(output), 930 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 931 mMixerStatus(MIXER_IDLE), 932 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 933 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 934 mScreenState(AudioFlinger::mScreenState), 935 // index 0 is reserved for normal mixer's submix 936 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 937{ 938 snprintf(mName, kNameLength, "AudioOut_%X", id); 939 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 940 941 // Assumes constructor is called by AudioFlinger with it's mLock held, but 942 // it would be safer to explicitly pass initial masterVolume/masterMute as 943 // parameter. 944 // 945 // If the HAL we are using has support for master volume or master mute, 946 // then do not attenuate or mute during mixing (just leave the volume at 1.0 947 // and the mute set to false). 948 mMasterVolume = audioFlinger->masterVolume_l(); 949 mMasterMute = audioFlinger->masterMute_l(); 950 if (mOutput && mOutput->audioHwDev) { 951 if (mOutput->audioHwDev->canSetMasterVolume()) { 952 mMasterVolume = 1.0; 953 } 954 955 if (mOutput->audioHwDev->canSetMasterMute()) { 956 mMasterMute = false; 957 } 958 } 959 960 readOutputParameters(); 961 962 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 963 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 964 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 965 stream = (audio_stream_type_t) (stream + 1)) { 966 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 967 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 968 } 969 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 970 // because mAudioFlinger doesn't have one to copy from 971} 972 973AudioFlinger::PlaybackThread::~PlaybackThread() 974{ 975 mAudioFlinger->unregisterWriter(mNBLogWriter); 976 delete [] mMixBuffer; 977} 978 979void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 980{ 981 dumpInternals(fd, args); 982 dumpTracks(fd, args); 983 dumpEffectChains(fd, args); 984} 985 986void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 987{ 988 const size_t SIZE = 256; 989 char buffer[SIZE]; 990 String8 result; 991 992 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 993 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 994 const stream_type_t *st = &mStreamTypes[i]; 995 if (i > 0) { 996 result.appendFormat(", "); 997 } 998 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 999 if (st->mute) { 1000 result.append("M"); 1001 } 1002 } 1003 result.append("\n"); 1004 write(fd, result.string(), result.length()); 1005 result.clear(); 1006 1007 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1008 result.append(buffer); 1009 Track::appendDumpHeader(result); 1010 for (size_t i = 0; i < mTracks.size(); ++i) { 1011 sp<Track> track = mTracks[i]; 1012 if (track != 0) { 1013 track->dump(buffer, SIZE); 1014 result.append(buffer); 1015 } 1016 } 1017 1018 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1019 result.append(buffer); 1020 Track::appendDumpHeader(result); 1021 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1022 sp<Track> track = mActiveTracks[i].promote(); 1023 if (track != 0) { 1024 track->dump(buffer, SIZE); 1025 result.append(buffer); 1026 } 1027 } 1028 write(fd, result.string(), result.size()); 1029 1030 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1031 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1032 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1033 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1034} 1035 1036void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1037{ 1038 const size_t SIZE = 256; 1039 char buffer[SIZE]; 1040 String8 result; 1041 1042 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1043 result.append(buffer); 1044 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1045 ns2ms(systemTime() - mLastWriteTime)); 1046 result.append(buffer); 1047 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1048 result.append(buffer); 1049 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1050 result.append(buffer); 1051 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1052 result.append(buffer); 1053 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1054 result.append(buffer); 1055 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1056 result.append(buffer); 1057 write(fd, result.string(), result.size()); 1058 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1059 1060 dumpBase(fd, args); 1061} 1062 1063// Thread virtuals 1064status_t AudioFlinger::PlaybackThread::readyToRun() 1065{ 1066 status_t status = initCheck(); 1067 if (status == NO_ERROR) { 1068 ALOGI("AudioFlinger's thread %p ready to run", this); 1069 } else { 1070 ALOGE("No working audio driver found."); 1071 } 1072 return status; 1073} 1074 1075void AudioFlinger::PlaybackThread::onFirstRef() 1076{ 1077 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1078} 1079 1080// ThreadBase virtuals 1081void AudioFlinger::PlaybackThread::preExit() 1082{ 1083 ALOGV(" preExit()"); 1084 // FIXME this is using hard-coded strings but in the future, this functionality will be 1085 // converted to use audio HAL extensions required to support tunneling 1086 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1087} 1088 1089// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1090sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1091 const sp<AudioFlinger::Client>& client, 1092 audio_stream_type_t streamType, 1093 uint32_t sampleRate, 1094 audio_format_t format, 1095 audio_channel_mask_t channelMask, 1096 size_t frameCount, 1097 const sp<IMemory>& sharedBuffer, 1098 int sessionId, 1099 IAudioFlinger::track_flags_t *flags, 1100 pid_t tid, 1101 status_t *status) 1102{ 1103 sp<Track> track; 1104 status_t lStatus; 1105 1106 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1107 1108 // client expresses a preference for FAST, but we get the final say 1109 if (*flags & IAudioFlinger::TRACK_FAST) { 1110 if ( 1111 // not timed 1112 (!isTimed) && 1113 // either of these use cases: 1114 ( 1115 // use case 1: shared buffer with any frame count 1116 ( 1117 (sharedBuffer != 0) 1118 ) || 1119 // use case 2: callback handler and frame count is default or at least as large as HAL 1120 ( 1121 (tid != -1) && 1122 ((frameCount == 0) || 1123 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1124 ) 1125 ) && 1126 // PCM data 1127 audio_is_linear_pcm(format) && 1128 // mono or stereo 1129 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1130 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1131#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1132 // hardware sample rate 1133 (sampleRate == mSampleRate) && 1134#endif 1135 // normal mixer has an associated fast mixer 1136 hasFastMixer() && 1137 // there are sufficient fast track slots available 1138 (mFastTrackAvailMask != 0) 1139 // FIXME test that MixerThread for this fast track has a capable output HAL 1140 // FIXME add a permission test also? 1141 ) { 1142 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1143 if (frameCount == 0) { 1144 frameCount = mFrameCount * kFastTrackMultiplier; 1145 } 1146 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1147 frameCount, mFrameCount); 1148 } else { 1149 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1150 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1151 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1152 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1153 audio_is_linear_pcm(format), 1154 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1155 *flags &= ~IAudioFlinger::TRACK_FAST; 1156 // For compatibility with AudioTrack calculation, buffer depth is forced 1157 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1158 // This is probably too conservative, but legacy application code may depend on it. 1159 // If you change this calculation, also review the start threshold which is related. 1160 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1161 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1162 if (minBufCount < 2) { 1163 minBufCount = 2; 1164 } 1165 size_t minFrameCount = mNormalFrameCount * minBufCount; 1166 if (frameCount < minFrameCount) { 1167 frameCount = minFrameCount; 1168 } 1169 } 1170 } 1171 1172 if (mType == DIRECT) { 1173 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1174 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1175 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1176 "for output %p with format %d", 1177 sampleRate, format, channelMask, mOutput, mFormat); 1178 lStatus = BAD_VALUE; 1179 goto Exit; 1180 } 1181 } 1182 } else { 1183 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1184 if (sampleRate > mSampleRate*2) { 1185 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1186 lStatus = BAD_VALUE; 1187 goto Exit; 1188 } 1189 } 1190 1191 lStatus = initCheck(); 1192 if (lStatus != NO_ERROR) { 1193 ALOGE("Audio driver not initialized."); 1194 goto Exit; 1195 } 1196 1197 { // scope for mLock 1198 Mutex::Autolock _l(mLock); 1199 1200 // all tracks in same audio session must share the same routing strategy otherwise 1201 // conflicts will happen when tracks are moved from one output to another by audio policy 1202 // manager 1203 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1204 for (size_t i = 0; i < mTracks.size(); ++i) { 1205 sp<Track> t = mTracks[i]; 1206 if (t != 0 && !t->isOutputTrack()) { 1207 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1208 if (sessionId == t->sessionId() && strategy != actual) { 1209 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1210 strategy, actual); 1211 lStatus = BAD_VALUE; 1212 goto Exit; 1213 } 1214 } 1215 } 1216 1217 if (!isTimed) { 1218 track = new Track(this, client, streamType, sampleRate, format, 1219 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1220 } else { 1221 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1222 channelMask, frameCount, sharedBuffer, sessionId); 1223 } 1224 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1225 lStatus = NO_MEMORY; 1226 goto Exit; 1227 } 1228 mTracks.add(track); 1229 1230 sp<EffectChain> chain = getEffectChain_l(sessionId); 1231 if (chain != 0) { 1232 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1233 track->setMainBuffer(chain->inBuffer()); 1234 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1235 chain->incTrackCnt(); 1236 } 1237 1238 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1239 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1240 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1241 // so ask activity manager to do this on our behalf 1242 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1243 } 1244 } 1245 1246 lStatus = NO_ERROR; 1247 1248Exit: 1249 if (status) { 1250 *status = lStatus; 1251 } 1252 mNBLogWriter->logf("createTrack_l"); 1253 return track; 1254} 1255 1256uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1257{ 1258 return latency; 1259} 1260 1261uint32_t AudioFlinger::PlaybackThread::latency() const 1262{ 1263 Mutex::Autolock _l(mLock); 1264 return latency_l(); 1265} 1266uint32_t AudioFlinger::PlaybackThread::latency_l() const 1267{ 1268 if (initCheck() == NO_ERROR) { 1269 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1270 } else { 1271 return 0; 1272 } 1273} 1274 1275void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1276{ 1277 Mutex::Autolock _l(mLock); 1278 // Don't apply master volume in SW if our HAL can do it for us. 1279 if (mOutput && mOutput->audioHwDev && 1280 mOutput->audioHwDev->canSetMasterVolume()) { 1281 mMasterVolume = 1.0; 1282 } else { 1283 mMasterVolume = value; 1284 } 1285} 1286 1287void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1288{ 1289 Mutex::Autolock _l(mLock); 1290 // Don't apply master mute in SW if our HAL can do it for us. 1291 if (mOutput && mOutput->audioHwDev && 1292 mOutput->audioHwDev->canSetMasterMute()) { 1293 mMasterMute = false; 1294 } else { 1295 mMasterMute = muted; 1296 } 1297} 1298 1299void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1300{ 1301 Mutex::Autolock _l(mLock); 1302 mStreamTypes[stream].volume = value; 1303} 1304 1305void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1306{ 1307 Mutex::Autolock _l(mLock); 1308 mStreamTypes[stream].mute = muted; 1309} 1310 1311float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1312{ 1313 Mutex::Autolock _l(mLock); 1314 return mStreamTypes[stream].volume; 1315} 1316 1317// addTrack_l() must be called with ThreadBase::mLock held 1318status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1319{ 1320 mNBLogWriter->logf("addTrack_l mName=%d", track->mName); 1321 status_t status = ALREADY_EXISTS; 1322 1323 // set retry count for buffer fill 1324 track->mRetryCount = kMaxTrackStartupRetries; 1325 if (mActiveTracks.indexOf(track) < 0) { 1326 // the track is newly added, make sure it fills up all its 1327 // buffers before playing. This is to ensure the client will 1328 // effectively get the latency it requested. 1329 track->mFillingUpStatus = Track::FS_FILLING; 1330 track->mResetDone = false; 1331 track->mPresentationCompleteFrames = 0; 1332 mActiveTracks.add(track); 1333 if (track->mainBuffer() != mMixBuffer) { 1334 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1335 if (chain != 0) { 1336 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1337 track->sessionId()); 1338 chain->incActiveTrackCnt(); 1339 } 1340 } 1341 1342 status = NO_ERROR; 1343 } 1344 1345 ALOGV("mWaitWorkCV.broadcast"); 1346 mWaitWorkCV.broadcast(); 1347 1348 return status; 1349} 1350 1351// destroyTrack_l() must be called with ThreadBase::mLock held 1352void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1353{ 1354 mNBLogWriter->logf("destroyTrack_l mName=%d", track->mName); 1355 track->mState = TrackBase::TERMINATED; 1356 // active tracks are removed by threadLoop() 1357 if (mActiveTracks.indexOf(track) < 0) { 1358 removeTrack_l(track); 1359 } 1360} 1361 1362void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1363{ 1364 mNBLogWriter->logf("removeTrack_l mName=%d", track->mName); 1365 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1366 mTracks.remove(track); 1367 deleteTrackName_l(track->name()); 1368 // redundant as track is about to be destroyed, for dumpsys only 1369 track->mName = -1; 1370 if (track->isFastTrack()) { 1371 int index = track->mFastIndex; 1372 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1373 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1374 mFastTrackAvailMask |= 1 << index; 1375 // redundant as track is about to be destroyed, for dumpsys only 1376 track->mFastIndex = -1; 1377 } 1378 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1379 if (chain != 0) { 1380 chain->decTrackCnt(); 1381 } 1382} 1383 1384String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1385{ 1386 String8 out_s8 = String8(""); 1387 char *s; 1388 1389 Mutex::Autolock _l(mLock); 1390 if (initCheck() != NO_ERROR) { 1391 return out_s8; 1392 } 1393 1394 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1395 out_s8 = String8(s); 1396 free(s); 1397 return out_s8; 1398} 1399 1400// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1401void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1402 AudioSystem::OutputDescriptor desc; 1403 void *param2 = NULL; 1404 1405 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1406 param); 1407 1408 switch (event) { 1409 case AudioSystem::OUTPUT_OPENED: 1410 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1411 desc.channels = mChannelMask; 1412 desc.samplingRate = mSampleRate; 1413 desc.format = mFormat; 1414 desc.frameCount = mNormalFrameCount; // FIXME see 1415 // AudioFlinger::frameCount(audio_io_handle_t) 1416 desc.latency = latency(); 1417 param2 = &desc; 1418 break; 1419 1420 case AudioSystem::STREAM_CONFIG_CHANGED: 1421 param2 = ¶m; 1422 case AudioSystem::OUTPUT_CLOSED: 1423 default: 1424 break; 1425 } 1426 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1427} 1428 1429void AudioFlinger::PlaybackThread::readOutputParameters() 1430{ 1431 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1432 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1433 mChannelCount = (uint16_t)popcount(mChannelMask); 1434 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1435 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1436 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1437 if (mFrameCount & 15) { 1438 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1439 mFrameCount); 1440 } 1441 1442 // Calculate size of normal mix buffer relative to the HAL output buffer size 1443 double multiplier = 1.0; 1444 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1445 kUseFastMixer == FastMixer_Dynamic)) { 1446 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1447 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1448 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1449 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1450 maxNormalFrameCount = maxNormalFrameCount & ~15; 1451 if (maxNormalFrameCount < minNormalFrameCount) { 1452 maxNormalFrameCount = minNormalFrameCount; 1453 } 1454 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1455 if (multiplier <= 1.0) { 1456 multiplier = 1.0; 1457 } else if (multiplier <= 2.0) { 1458 if (2 * mFrameCount <= maxNormalFrameCount) { 1459 multiplier = 2.0; 1460 } else { 1461 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1462 } 1463 } else { 1464 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1465 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1466 // track, but we sometimes have to do this to satisfy the maximum frame count 1467 // constraint) 1468 // FIXME this rounding up should not be done if no HAL SRC 1469 uint32_t truncMult = (uint32_t) multiplier; 1470 if ((truncMult & 1)) { 1471 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1472 ++truncMult; 1473 } 1474 } 1475 multiplier = (double) truncMult; 1476 } 1477 } 1478 mNormalFrameCount = multiplier * mFrameCount; 1479 // round up to nearest 16 frames to satisfy AudioMixer 1480 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1481 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1482 mNormalFrameCount); 1483 1484 delete[] mMixBuffer; 1485 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 1486 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 1487 1488 // force reconfiguration of effect chains and engines to take new buffer size and audio 1489 // parameters into account 1490 // Note that mLock is not held when readOutputParameters() is called from the constructor 1491 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1492 // matter. 1493 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1494 Vector< sp<EffectChain> > effectChains = mEffectChains; 1495 for (size_t i = 0; i < effectChains.size(); i ++) { 1496 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1497 } 1498} 1499 1500 1501status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1502{ 1503 if (halFrames == NULL || dspFrames == NULL) { 1504 return BAD_VALUE; 1505 } 1506 Mutex::Autolock _l(mLock); 1507 if (initCheck() != NO_ERROR) { 1508 return INVALID_OPERATION; 1509 } 1510 size_t framesWritten = mBytesWritten / mFrameSize; 1511 *halFrames = framesWritten; 1512 1513 if (isSuspended()) { 1514 // return an estimation of rendered frames when the output is suspended 1515 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1516 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1517 return NO_ERROR; 1518 } else { 1519 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1520 } 1521} 1522 1523uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1524{ 1525 Mutex::Autolock _l(mLock); 1526 uint32_t result = 0; 1527 if (getEffectChain_l(sessionId) != 0) { 1528 result = EFFECT_SESSION; 1529 } 1530 1531 for (size_t i = 0; i < mTracks.size(); ++i) { 1532 sp<Track> track = mTracks[i]; 1533 if (sessionId == track->sessionId() && !track->isInvalid()) { 1534 result |= TRACK_SESSION; 1535 break; 1536 } 1537 } 1538 1539 return result; 1540} 1541 1542uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1543{ 1544 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1545 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1546 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1547 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1548 } 1549 for (size_t i = 0; i < mTracks.size(); i++) { 1550 sp<Track> track = mTracks[i]; 1551 if (sessionId == track->sessionId() && !track->isInvalid()) { 1552 return AudioSystem::getStrategyForStream(track->streamType()); 1553 } 1554 } 1555 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1556} 1557 1558 1559AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1560{ 1561 Mutex::Autolock _l(mLock); 1562 return mOutput; 1563} 1564 1565AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1566{ 1567 Mutex::Autolock _l(mLock); 1568 AudioStreamOut *output = mOutput; 1569 mOutput = NULL; 1570 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1571 // must push a NULL and wait for ack 1572 mOutputSink.clear(); 1573 mPipeSink.clear(); 1574 mNormalSink.clear(); 1575 return output; 1576} 1577 1578// this method must always be called either with ThreadBase mLock held or inside the thread loop 1579audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1580{ 1581 if (mOutput == NULL) { 1582 return NULL; 1583 } 1584 return &mOutput->stream->common; 1585} 1586 1587uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1588{ 1589 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1590} 1591 1592status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1593{ 1594 if (!isValidSyncEvent(event)) { 1595 return BAD_VALUE; 1596 } 1597 1598 Mutex::Autolock _l(mLock); 1599 1600 for (size_t i = 0; i < mTracks.size(); ++i) { 1601 sp<Track> track = mTracks[i]; 1602 if (event->triggerSession() == track->sessionId()) { 1603 (void) track->setSyncEvent(event); 1604 return NO_ERROR; 1605 } 1606 } 1607 1608 return NAME_NOT_FOUND; 1609} 1610 1611bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1612{ 1613 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1614} 1615 1616void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1617 const Vector< sp<Track> >& tracksToRemove) 1618{ 1619 size_t count = tracksToRemove.size(); 1620 if (CC_UNLIKELY(count)) { 1621 for (size_t i = 0 ; i < count ; i++) { 1622 const sp<Track>& track = tracksToRemove.itemAt(i); 1623 if ((track->sharedBuffer() != 0) && 1624 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 1625 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1626 } 1627 } 1628 } 1629 1630} 1631 1632void AudioFlinger::PlaybackThread::checkSilentMode_l() 1633{ 1634 if (!mMasterMute) { 1635 char value[PROPERTY_VALUE_MAX]; 1636 if (property_get("ro.audio.silent", value, "0") > 0) { 1637 char *endptr; 1638 unsigned long ul = strtoul(value, &endptr, 0); 1639 if (*endptr == '\0' && ul != 0) { 1640 ALOGD("Silence is golden"); 1641 // The setprop command will not allow a property to be changed after 1642 // the first time it is set, so we don't have to worry about un-muting. 1643 setMasterMute_l(true); 1644 } 1645 } 1646 } 1647} 1648 1649// shared by MIXER and DIRECT, overridden by DUPLICATING 1650void AudioFlinger::PlaybackThread::threadLoop_write() 1651{ 1652 // FIXME rewrite to reduce number of system calls 1653 mLastWriteTime = systemTime(); 1654 mInWrite = true; 1655 int bytesWritten; 1656 1657 // If an NBAIO sink is present, use it to write the normal mixer's submix 1658 if (mNormalSink != 0) { 1659#define mBitShift 2 // FIXME 1660 size_t count = mixBufferSize >> mBitShift; 1661 ATRACE_BEGIN("write"); 1662 // update the setpoint when AudioFlinger::mScreenState changes 1663 uint32_t screenState = AudioFlinger::mScreenState; 1664 if (screenState != mScreenState) { 1665 mScreenState = screenState; 1666 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1667 if (pipe != NULL) { 1668 pipe->setAvgFrames((mScreenState & 1) ? 1669 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1670 } 1671 } 1672 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 1673 ATRACE_END(); 1674 if (framesWritten > 0) { 1675 bytesWritten = framesWritten << mBitShift; 1676 } else { 1677 bytesWritten = framesWritten; 1678 } 1679 // otherwise use the HAL / AudioStreamOut directly 1680 } else { 1681 // Direct output thread. 1682 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1683 } 1684 1685 if (bytesWritten > 0) { 1686 mBytesWritten += mixBufferSize; 1687 } 1688 mNumWrites++; 1689 mInWrite = false; 1690} 1691 1692/* 1693The derived values that are cached: 1694 - mixBufferSize from frame count * frame size 1695 - activeSleepTime from activeSleepTimeUs() 1696 - idleSleepTime from idleSleepTimeUs() 1697 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1698 - maxPeriod from frame count and sample rate (MIXER only) 1699 1700The parameters that affect these derived values are: 1701 - frame count 1702 - frame size 1703 - sample rate 1704 - device type: A2DP or not 1705 - device latency 1706 - format: PCM or not 1707 - active sleep time 1708 - idle sleep time 1709*/ 1710 1711void AudioFlinger::PlaybackThread::cacheParameters_l() 1712{ 1713 mixBufferSize = mNormalFrameCount * mFrameSize; 1714 activeSleepTime = activeSleepTimeUs(); 1715 idleSleepTime = idleSleepTimeUs(); 1716} 1717 1718void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1719{ 1720 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1721 this, streamType, mTracks.size()); 1722 Mutex::Autolock _l(mLock); 1723 1724 size_t size = mTracks.size(); 1725 for (size_t i = 0; i < size; i++) { 1726 sp<Track> t = mTracks[i]; 1727 if (t->streamType() == streamType) { 1728 t->invalidate(); 1729 } 1730 } 1731} 1732 1733status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1734{ 1735 int session = chain->sessionId(); 1736 int16_t *buffer = mMixBuffer; 1737 bool ownsBuffer = false; 1738 1739 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1740 if (session > 0) { 1741 // Only one effect chain can be present in direct output thread and it uses 1742 // the mix buffer as input 1743 if (mType != DIRECT) { 1744 size_t numSamples = mNormalFrameCount * mChannelCount; 1745 buffer = new int16_t[numSamples]; 1746 memset(buffer, 0, numSamples * sizeof(int16_t)); 1747 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1748 ownsBuffer = true; 1749 } 1750 1751 // Attach all tracks with same session ID to this chain. 1752 for (size_t i = 0; i < mTracks.size(); ++i) { 1753 sp<Track> track = mTracks[i]; 1754 if (session == track->sessionId()) { 1755 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1756 buffer); 1757 track->setMainBuffer(buffer); 1758 chain->incTrackCnt(); 1759 } 1760 } 1761 1762 // indicate all active tracks in the chain 1763 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1764 sp<Track> track = mActiveTracks[i].promote(); 1765 if (track == 0) { 1766 continue; 1767 } 1768 if (session == track->sessionId()) { 1769 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1770 chain->incActiveTrackCnt(); 1771 } 1772 } 1773 } 1774 1775 chain->setInBuffer(buffer, ownsBuffer); 1776 chain->setOutBuffer(mMixBuffer); 1777 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1778 // chains list in order to be processed last as it contains output stage effects 1779 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1780 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1781 // after track specific effects and before output stage 1782 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1783 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1784 // Effect chain for other sessions are inserted at beginning of effect 1785 // chains list to be processed before output mix effects. Relative order between other 1786 // sessions is not important 1787 size_t size = mEffectChains.size(); 1788 size_t i = 0; 1789 for (i = 0; i < size; i++) { 1790 if (mEffectChains[i]->sessionId() < session) { 1791 break; 1792 } 1793 } 1794 mEffectChains.insertAt(chain, i); 1795 checkSuspendOnAddEffectChain_l(chain); 1796 1797 return NO_ERROR; 1798} 1799 1800size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1801{ 1802 int session = chain->sessionId(); 1803 1804 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1805 1806 for (size_t i = 0; i < mEffectChains.size(); i++) { 1807 if (chain == mEffectChains[i]) { 1808 mEffectChains.removeAt(i); 1809 // detach all active tracks from the chain 1810 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1811 sp<Track> track = mActiveTracks[i].promote(); 1812 if (track == 0) { 1813 continue; 1814 } 1815 if (session == track->sessionId()) { 1816 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1817 chain.get(), session); 1818 chain->decActiveTrackCnt(); 1819 } 1820 } 1821 1822 // detach all tracks with same session ID from this chain 1823 for (size_t i = 0; i < mTracks.size(); ++i) { 1824 sp<Track> track = mTracks[i]; 1825 if (session == track->sessionId()) { 1826 track->setMainBuffer(mMixBuffer); 1827 chain->decTrackCnt(); 1828 } 1829 } 1830 break; 1831 } 1832 } 1833 return mEffectChains.size(); 1834} 1835 1836status_t AudioFlinger::PlaybackThread::attachAuxEffect( 1837 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1838{ 1839 Mutex::Autolock _l(mLock); 1840 return attachAuxEffect_l(track, EffectId); 1841} 1842 1843status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 1844 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 1845{ 1846 status_t status = NO_ERROR; 1847 1848 if (EffectId == 0) { 1849 track->setAuxBuffer(0, NULL); 1850 } else { 1851 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 1852 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 1853 if (effect != 0) { 1854 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1855 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 1856 } else { 1857 status = INVALID_OPERATION; 1858 } 1859 } else { 1860 status = BAD_VALUE; 1861 } 1862 } 1863 return status; 1864} 1865 1866void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 1867{ 1868 for (size_t i = 0; i < mTracks.size(); ++i) { 1869 sp<Track> track = mTracks[i]; 1870 if (track->auxEffectId() == effectId) { 1871 attachAuxEffect_l(track, 0); 1872 } 1873 } 1874} 1875 1876bool AudioFlinger::PlaybackThread::threadLoop() 1877{ 1878 Vector< sp<Track> > tracksToRemove; 1879 1880 standbyTime = systemTime(); 1881 1882 // MIXER 1883 nsecs_t lastWarning = 0; 1884 1885 // DUPLICATING 1886 // FIXME could this be made local to while loop? 1887 writeFrames = 0; 1888 1889 cacheParameters_l(); 1890 sleepTime = idleSleepTime; 1891 1892 if (mType == MIXER) { 1893 sleepTimeShift = 0; 1894 } 1895 1896 CpuStats cpuStats; 1897 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 1898 1899 acquireWakeLock(); 1900 1901 // mNBLogWriter->log can only be called while thread mutex mLock is held. 1902 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 1903 // and then that string will be logged at the next convenient opportunity. 1904 const char *logString = NULL; 1905 1906 while (!exitPending()) 1907 { 1908 cpuStats.sample(myName); 1909 1910 Vector< sp<EffectChain> > effectChains; 1911 1912 processConfigEvents(); 1913 1914 { // scope for mLock 1915 1916 Mutex::Autolock _l(mLock); 1917 1918 if (logString != NULL) { 1919 mNBLogWriter->logTimestamp(); 1920 mNBLogWriter->log(logString); 1921 logString = NULL; 1922 } 1923 1924 if (checkForNewParameters_l()) { 1925 cacheParameters_l(); 1926 } 1927 1928 saveOutputTracks(); 1929 1930 // put audio hardware into standby after short delay 1931 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 1932 isSuspended())) { 1933 if (!mStandby) { 1934 1935 threadLoop_standby(); 1936 1937 mNBLogWriter->log("standby"); 1938 mStandby = true; 1939 } 1940 1941 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 1942 // we're about to wait, flush the binder command buffer 1943 IPCThreadState::self()->flushCommands(); 1944 1945 clearOutputTracks(); 1946 1947 if (exitPending()) { 1948 break; 1949 } 1950 1951 releaseWakeLock_l(); 1952 // wait until we have something to do... 1953 ALOGV("%s going to sleep", myName.string()); 1954 mWaitWorkCV.wait(mLock); 1955 ALOGV("%s waking up", myName.string()); 1956 acquireWakeLock_l(); 1957 1958 mMixerStatus = MIXER_IDLE; 1959 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 1960 mBytesWritten = 0; 1961 1962 checkSilentMode_l(); 1963 1964 standbyTime = systemTime() + standbyDelay; 1965 sleepTime = idleSleepTime; 1966 if (mType == MIXER) { 1967 sleepTimeShift = 0; 1968 } 1969 1970 continue; 1971 } 1972 } 1973 1974 // mMixerStatusIgnoringFastTracks is also updated internally 1975 mMixerStatus = prepareTracks_l(&tracksToRemove); 1976 1977 // prevent any changes in effect chain list and in each effect chain 1978 // during mixing and effect process as the audio buffers could be deleted 1979 // or modified if an effect is created or deleted 1980 lockEffectChains_l(effectChains); 1981 } 1982 1983 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 1984 threadLoop_mix(); 1985 } else { 1986 threadLoop_sleepTime(); 1987 } 1988 1989 if (isSuspended()) { 1990 sleepTime = suspendSleepTimeUs(); 1991 mBytesWritten += mixBufferSize; 1992 } 1993 1994 // only process effects if we're going to write 1995 if (sleepTime == 0) { 1996 for (size_t i = 0; i < effectChains.size(); i ++) { 1997 effectChains[i]->process_l(); 1998 } 1999 } 2000 2001 // enable changes in effect chain 2002 unlockEffectChains(effectChains); 2003 2004 // sleepTime == 0 means we must write to audio hardware 2005 if (sleepTime == 0) { 2006 2007 threadLoop_write(); 2008 2009if (mType == MIXER) { 2010 // write blocked detection 2011 nsecs_t now = systemTime(); 2012 nsecs_t delta = now - mLastWriteTime; 2013 if (!mStandby && delta > maxPeriod) { 2014 mNumDelayedWrites++; 2015 if ((now - lastWarning) > kWarningThrottleNs) { 2016 ATRACE_NAME("underrun"); 2017 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2018 ns2ms(delta), mNumDelayedWrites, this); 2019 lastWarning = now; 2020 } 2021 } 2022} 2023 2024 mStandby = false; 2025 } else { 2026 usleep(sleepTime); 2027 } 2028 2029 // Finally let go of removed track(s), without the lock held 2030 // since we can't guarantee the destructors won't acquire that 2031 // same lock. This will also mutate and push a new fast mixer state. 2032 threadLoop_removeTracks(tracksToRemove); 2033 if (tracksToRemove.size() > 0) { 2034 logString = "remove"; 2035 } 2036 tracksToRemove.clear(); 2037 2038 // FIXME I don't understand the need for this here; 2039 // it was in the original code but maybe the 2040 // assignment in saveOutputTracks() makes this unnecessary? 2041 clearOutputTracks(); 2042 2043 // Effect chains will be actually deleted here if they were removed from 2044 // mEffectChains list during mixing or effects processing 2045 effectChains.clear(); 2046 2047 // FIXME Note that the above .clear() is no longer necessary since effectChains 2048 // is now local to this block, but will keep it for now (at least until merge done). 2049 } 2050 2051 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2052 if (mType == MIXER || mType == DIRECT) { 2053 // put output stream into standby mode 2054 if (!mStandby) { 2055 mOutput->stream->common.standby(&mOutput->stream->common); 2056 } 2057 } 2058 2059 releaseWakeLock(); 2060 2061 ALOGV("Thread %p type %d exiting", this, mType); 2062 return false; 2063} 2064 2065 2066// ---------------------------------------------------------------------------- 2067 2068AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2069 audio_io_handle_t id, audio_devices_t device, type_t type) 2070 : PlaybackThread(audioFlinger, output, id, device, type), 2071 // mAudioMixer below 2072 // mFastMixer below 2073 mFastMixerFutex(0) 2074 // mOutputSink below 2075 // mPipeSink below 2076 // mNormalSink below 2077{ 2078 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2079 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%u, " 2080 "mFrameCount=%d, mNormalFrameCount=%d", 2081 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2082 mNormalFrameCount); 2083 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2084 2085 // FIXME - Current mixer implementation only supports stereo output 2086 if (mChannelCount != FCC_2) { 2087 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2088 } 2089 2090 // create an NBAIO sink for the HAL output stream, and negotiate 2091 mOutputSink = new AudioStreamOutSink(output->stream); 2092 size_t numCounterOffers = 0; 2093 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2094 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2095 ALOG_ASSERT(index == 0); 2096 2097 // initialize fast mixer depending on configuration 2098 bool initFastMixer; 2099 switch (kUseFastMixer) { 2100 case FastMixer_Never: 2101 initFastMixer = false; 2102 break; 2103 case FastMixer_Always: 2104 initFastMixer = true; 2105 break; 2106 case FastMixer_Static: 2107 case FastMixer_Dynamic: 2108 initFastMixer = mFrameCount < mNormalFrameCount; 2109 break; 2110 } 2111 if (initFastMixer) { 2112 2113 // create a MonoPipe to connect our submix to FastMixer 2114 NBAIO_Format format = mOutputSink->format(); 2115 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2116 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2117 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2118 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2119 const NBAIO_Format offers[1] = {format}; 2120 size_t numCounterOffers = 0; 2121 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2122 ALOG_ASSERT(index == 0); 2123 monoPipe->setAvgFrames((mScreenState & 1) ? 2124 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2125 mPipeSink = monoPipe; 2126 2127#ifdef TEE_SINK_FRAMES 2128 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2129 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2130 numCounterOffers = 0; 2131 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2132 ALOG_ASSERT(index == 0); 2133 mTeeSink = teeSink; 2134 PipeReader *teeSource = new PipeReader(*teeSink); 2135 numCounterOffers = 0; 2136 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2137 ALOG_ASSERT(index == 0); 2138 mTeeSource = teeSource; 2139#endif 2140 2141 // create fast mixer and configure it initially with just one fast track for our submix 2142 mFastMixer = new FastMixer(); 2143 FastMixerStateQueue *sq = mFastMixer->sq(); 2144#ifdef STATE_QUEUE_DUMP 2145 sq->setObserverDump(&mStateQueueObserverDump); 2146 sq->setMutatorDump(&mStateQueueMutatorDump); 2147#endif 2148 FastMixerState *state = sq->begin(); 2149 FastTrack *fastTrack = &state->mFastTracks[0]; 2150 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2151 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2152 fastTrack->mVolumeProvider = NULL; 2153 fastTrack->mGeneration++; 2154 state->mFastTracksGen++; 2155 state->mTrackMask = 1; 2156 // fast mixer will use the HAL output sink 2157 state->mOutputSink = mOutputSink.get(); 2158 state->mOutputSinkGen++; 2159 state->mFrameCount = mFrameCount; 2160 state->mCommand = FastMixerState::COLD_IDLE; 2161 // already done in constructor initialization list 2162 //mFastMixerFutex = 0; 2163 state->mColdFutexAddr = &mFastMixerFutex; 2164 state->mColdGen++; 2165 state->mDumpState = &mFastMixerDumpState; 2166 state->mTeeSink = mTeeSink.get(); 2167 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2168 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2169 sq->end(); 2170 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2171 2172 // start the fast mixer 2173 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2174 pid_t tid = mFastMixer->getTid(); 2175 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2176 if (err != 0) { 2177 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2178 kPriorityFastMixer, getpid_cached, tid, err); 2179 } 2180 2181#ifdef AUDIO_WATCHDOG 2182 // create and start the watchdog 2183 mAudioWatchdog = new AudioWatchdog(); 2184 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2185 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2186 tid = mAudioWatchdog->getTid(); 2187 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2188 if (err != 0) { 2189 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2190 kPriorityFastMixer, getpid_cached, tid, err); 2191 } 2192#endif 2193 2194 } else { 2195 mFastMixer = NULL; 2196 } 2197 2198 switch (kUseFastMixer) { 2199 case FastMixer_Never: 2200 case FastMixer_Dynamic: 2201 mNormalSink = mOutputSink; 2202 break; 2203 case FastMixer_Always: 2204 mNormalSink = mPipeSink; 2205 break; 2206 case FastMixer_Static: 2207 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2208 break; 2209 } 2210} 2211 2212AudioFlinger::MixerThread::~MixerThread() 2213{ 2214 if (mFastMixer != NULL) { 2215 FastMixerStateQueue *sq = mFastMixer->sq(); 2216 FastMixerState *state = sq->begin(); 2217 if (state->mCommand == FastMixerState::COLD_IDLE) { 2218 int32_t old = android_atomic_inc(&mFastMixerFutex); 2219 if (old == -1) { 2220 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2221 } 2222 } 2223 state->mCommand = FastMixerState::EXIT; 2224 sq->end(); 2225 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2226 mFastMixer->join(); 2227 // Though the fast mixer thread has exited, it's state queue is still valid. 2228 // We'll use that extract the final state which contains one remaining fast track 2229 // corresponding to our sub-mix. 2230 state = sq->begin(); 2231 ALOG_ASSERT(state->mTrackMask == 1); 2232 FastTrack *fastTrack = &state->mFastTracks[0]; 2233 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2234 delete fastTrack->mBufferProvider; 2235 sq->end(false /*didModify*/); 2236 delete mFastMixer; 2237#ifdef AUDIO_WATCHDOG 2238 if (mAudioWatchdog != 0) { 2239 mAudioWatchdog->requestExit(); 2240 mAudioWatchdog->requestExitAndWait(); 2241 mAudioWatchdog.clear(); 2242 } 2243#endif 2244 } 2245 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2246 delete mAudioMixer; 2247} 2248 2249 2250uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2251{ 2252 if (mFastMixer != NULL) { 2253 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2254 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2255 } 2256 return latency; 2257} 2258 2259 2260void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2261{ 2262 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2263} 2264 2265void AudioFlinger::MixerThread::threadLoop_write() 2266{ 2267 // FIXME we should only do one push per cycle; confirm this is true 2268 // Start the fast mixer if it's not already running 2269 if (mFastMixer != NULL) { 2270 FastMixerStateQueue *sq = mFastMixer->sq(); 2271 FastMixerState *state = sq->begin(); 2272 if (state->mCommand != FastMixerState::MIX_WRITE && 2273 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2274 if (state->mCommand == FastMixerState::COLD_IDLE) { 2275 int32_t old = android_atomic_inc(&mFastMixerFutex); 2276 if (old == -1) { 2277 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2278 } 2279#ifdef AUDIO_WATCHDOG 2280 if (mAudioWatchdog != 0) { 2281 mAudioWatchdog->resume(); 2282 } 2283#endif 2284 } 2285 state->mCommand = FastMixerState::MIX_WRITE; 2286 sq->end(); 2287 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2288 if (kUseFastMixer == FastMixer_Dynamic) { 2289 mNormalSink = mPipeSink; 2290 } 2291 } else { 2292 sq->end(false /*didModify*/); 2293 } 2294 } 2295 PlaybackThread::threadLoop_write(); 2296} 2297 2298void AudioFlinger::MixerThread::threadLoop_standby() 2299{ 2300 // Idle the fast mixer if it's currently running 2301 if (mFastMixer != NULL) { 2302 FastMixerStateQueue *sq = mFastMixer->sq(); 2303 FastMixerState *state = sq->begin(); 2304 if (!(state->mCommand & FastMixerState::IDLE)) { 2305 state->mCommand = FastMixerState::COLD_IDLE; 2306 state->mColdFutexAddr = &mFastMixerFutex; 2307 state->mColdGen++; 2308 mFastMixerFutex = 0; 2309 sq->end(); 2310 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2311 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2312 if (kUseFastMixer == FastMixer_Dynamic) { 2313 mNormalSink = mOutputSink; 2314 } 2315#ifdef AUDIO_WATCHDOG 2316 if (mAudioWatchdog != 0) { 2317 mAudioWatchdog->pause(); 2318 } 2319#endif 2320 } else { 2321 sq->end(false /*didModify*/); 2322 } 2323 } 2324 PlaybackThread::threadLoop_standby(); 2325} 2326 2327// shared by MIXER and DIRECT, overridden by DUPLICATING 2328void AudioFlinger::PlaybackThread::threadLoop_standby() 2329{ 2330 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2331 mOutput->stream->common.standby(&mOutput->stream->common); 2332} 2333 2334void AudioFlinger::MixerThread::threadLoop_mix() 2335{ 2336 // obtain the presentation timestamp of the next output buffer 2337 int64_t pts; 2338 status_t status = INVALID_OPERATION; 2339 2340 if (mNormalSink != 0) { 2341 status = mNormalSink->getNextWriteTimestamp(&pts); 2342 } else { 2343 status = mOutputSink->getNextWriteTimestamp(&pts); 2344 } 2345 2346 if (status != NO_ERROR) { 2347 pts = AudioBufferProvider::kInvalidPTS; 2348 } 2349 2350 // mix buffers... 2351 mAudioMixer->process(pts); 2352 // increase sleep time progressively when application underrun condition clears. 2353 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2354 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2355 // such that we would underrun the audio HAL. 2356 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2357 sleepTimeShift--; 2358 } 2359 sleepTime = 0; 2360 standbyTime = systemTime() + standbyDelay; 2361 //TODO: delay standby when effects have a tail 2362} 2363 2364void AudioFlinger::MixerThread::threadLoop_sleepTime() 2365{ 2366 // If no tracks are ready, sleep once for the duration of an output 2367 // buffer size, then write 0s to the output 2368 if (sleepTime == 0) { 2369 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2370 sleepTime = activeSleepTime >> sleepTimeShift; 2371 if (sleepTime < kMinThreadSleepTimeUs) { 2372 sleepTime = kMinThreadSleepTimeUs; 2373 } 2374 // reduce sleep time in case of consecutive application underruns to avoid 2375 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2376 // duration we would end up writing less data than needed by the audio HAL if 2377 // the condition persists. 2378 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2379 sleepTimeShift++; 2380 } 2381 } else { 2382 sleepTime = idleSleepTime; 2383 } 2384 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2385 memset (mMixBuffer, 0, mixBufferSize); 2386 sleepTime = 0; 2387 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2388 "anticipated start"); 2389 } 2390 // TODO add standby time extension fct of effect tail 2391} 2392 2393// prepareTracks_l() must be called with ThreadBase::mLock held 2394AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2395 Vector< sp<Track> > *tracksToRemove) 2396{ 2397 2398 mixer_state mixerStatus = MIXER_IDLE; 2399 // find out which tracks need to be processed 2400 size_t count = mActiveTracks.size(); 2401 size_t mixedTracks = 0; 2402 size_t tracksWithEffect = 0; 2403 // counts only _active_ fast tracks 2404 size_t fastTracks = 0; 2405 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2406 2407 float masterVolume = mMasterVolume; 2408 bool masterMute = mMasterMute; 2409 2410 if (masterMute) { 2411 masterVolume = 0; 2412 } 2413 // Delegate master volume control to effect in output mix effect chain if needed 2414 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2415 if (chain != 0) { 2416 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2417 chain->setVolume_l(&v, &v); 2418 masterVolume = (float)((v + (1 << 23)) >> 24); 2419 chain.clear(); 2420 } 2421 2422 // prepare a new state to push 2423 FastMixerStateQueue *sq = NULL; 2424 FastMixerState *state = NULL; 2425 bool didModify = false; 2426 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2427 if (mFastMixer != NULL) { 2428 sq = mFastMixer->sq(); 2429 state = sq->begin(); 2430 } 2431 2432 for (size_t i=0 ; i<count ; i++) { 2433 sp<Track> t = mActiveTracks[i].promote(); 2434 if (t == 0) { 2435 continue; 2436 } 2437 2438 // this const just means the local variable doesn't change 2439 Track* const track = t.get(); 2440 2441 // process fast tracks 2442 if (track->isFastTrack()) { 2443 2444 // It's theoretically possible (though unlikely) for a fast track to be created 2445 // and then removed within the same normal mix cycle. This is not a problem, as 2446 // the track never becomes active so it's fast mixer slot is never touched. 2447 // The converse, of removing an (active) track and then creating a new track 2448 // at the identical fast mixer slot within the same normal mix cycle, 2449 // is impossible because the slot isn't marked available until the end of each cycle. 2450 int j = track->mFastIndex; 2451 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2452 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2453 FastTrack *fastTrack = &state->mFastTracks[j]; 2454 2455 // Determine whether the track is currently in underrun condition, 2456 // and whether it had a recent underrun. 2457 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2458 FastTrackUnderruns underruns = ftDump->mUnderruns; 2459 uint32_t recentFull = (underruns.mBitFields.mFull - 2460 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2461 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2462 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2463 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2464 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2465 uint32_t recentUnderruns = recentPartial + recentEmpty; 2466 track->mObservedUnderruns = underruns; 2467 // don't count underruns that occur while stopping or pausing 2468 // or stopped which can occur when flush() is called while active 2469 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2470 track->mUnderrunCount += recentUnderruns; 2471 } 2472 2473 // This is similar to the state machine for normal tracks, 2474 // with a few modifications for fast tracks. 2475 bool isActive = true; 2476 switch (track->mState) { 2477 case TrackBase::STOPPING_1: 2478 // track stays active in STOPPING_1 state until first underrun 2479 if (recentUnderruns > 0) { 2480 track->mState = TrackBase::STOPPING_2; 2481 } 2482 break; 2483 case TrackBase::PAUSING: 2484 // ramp down is not yet implemented 2485 track->setPaused(); 2486 break; 2487 case TrackBase::RESUMING: 2488 // ramp up is not yet implemented 2489 track->mState = TrackBase::ACTIVE; 2490 break; 2491 case TrackBase::ACTIVE: 2492 if (recentFull > 0 || recentPartial > 0) { 2493 // track has provided at least some frames recently: reset retry count 2494 track->mRetryCount = kMaxTrackRetries; 2495 } 2496 if (recentUnderruns == 0) { 2497 // no recent underruns: stay active 2498 break; 2499 } 2500 // there has recently been an underrun of some kind 2501 if (track->sharedBuffer() == 0) { 2502 // were any of the recent underruns "empty" (no frames available)? 2503 if (recentEmpty == 0) { 2504 // no, then ignore the partial underruns as they are allowed indefinitely 2505 break; 2506 } 2507 // there has recently been an "empty" underrun: decrement the retry counter 2508 if (--(track->mRetryCount) > 0) { 2509 break; 2510 } 2511 // indicate to client process that the track was disabled because of underrun; 2512 // it will then automatically call start() when data is available 2513 android_atomic_or(CBLK_DISABLED, &track->mCblk->flags); 2514 // remove from active list, but state remains ACTIVE [confusing but true] 2515 isActive = false; 2516 break; 2517 } 2518 // fall through 2519 case TrackBase::STOPPING_2: 2520 case TrackBase::PAUSED: 2521 case TrackBase::TERMINATED: 2522 case TrackBase::STOPPED: 2523 case TrackBase::FLUSHED: // flush() while active 2524 // Check for presentation complete if track is inactive 2525 // We have consumed all the buffers of this track. 2526 // This would be incomplete if we auto-paused on underrun 2527 { 2528 size_t audioHALFrames = 2529 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2530 size_t framesWritten = mBytesWritten / mFrameSize; 2531 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2532 // track stays in active list until presentation is complete 2533 break; 2534 } 2535 } 2536 if (track->isStopping_2()) { 2537 track->mState = TrackBase::STOPPED; 2538 } 2539 if (track->isStopped()) { 2540 // Can't reset directly, as fast mixer is still polling this track 2541 // track->reset(); 2542 // So instead mark this track as needing to be reset after push with ack 2543 resetMask |= 1 << i; 2544 } 2545 isActive = false; 2546 break; 2547 case TrackBase::IDLE: 2548 default: 2549 LOG_FATAL("unexpected track state %d", track->mState); 2550 } 2551 2552 if (isActive) { 2553 // was it previously inactive? 2554 if (!(state->mTrackMask & (1 << j))) { 2555 ExtendedAudioBufferProvider *eabp = track; 2556 VolumeProvider *vp = track; 2557 fastTrack->mBufferProvider = eabp; 2558 fastTrack->mVolumeProvider = vp; 2559 fastTrack->mSampleRate = track->mSampleRate; 2560 fastTrack->mChannelMask = track->mChannelMask; 2561 fastTrack->mGeneration++; 2562 state->mTrackMask |= 1 << j; 2563 didModify = true; 2564 // no acknowledgement required for newly active tracks 2565 } 2566 // cache the combined master volume and stream type volume for fast mixer; this 2567 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2568 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2569 ++fastTracks; 2570 } else { 2571 // was it previously active? 2572 if (state->mTrackMask & (1 << j)) { 2573 fastTrack->mBufferProvider = NULL; 2574 fastTrack->mGeneration++; 2575 state->mTrackMask &= ~(1 << j); 2576 didModify = true; 2577 // If any fast tracks were removed, we must wait for acknowledgement 2578 // because we're about to decrement the last sp<> on those tracks. 2579 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2580 } else { 2581 LOG_FATAL("fast track %d should have been active", j); 2582 } 2583 tracksToRemove->add(track); 2584 // Avoids a misleading display in dumpsys 2585 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2586 } 2587 continue; 2588 } 2589 2590 { // local variable scope to avoid goto warning 2591 2592 audio_track_cblk_t* cblk = track->cblk(); 2593 2594 // The first time a track is added we wait 2595 // for all its buffers to be filled before processing it 2596 int name = track->name(); 2597 // make sure that we have enough frames to mix one full buffer. 2598 // enforce this condition only once to enable draining the buffer in case the client 2599 // app does not call stop() and relies on underrun to stop: 2600 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2601 // during last round 2602 uint32_t minFrames = 1; 2603 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2604 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2605 if (t->sampleRate() == mSampleRate) { 2606 minFrames = mNormalFrameCount; 2607 } else { 2608 // +1 for rounding and +1 for additional sample needed for interpolation 2609 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2610 // add frames already consumed but not yet released by the resampler 2611 // because cblk->framesReady() will include these frames 2612 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2613 // the minimum track buffer size is normally twice the number of frames necessary 2614 // to fill one buffer and the resampler should not leave more than one buffer worth 2615 // of unreleased frames after each pass, but just in case... 2616 ALOG_ASSERT(minFrames <= cblk->frameCount_); 2617 } 2618 } 2619 if ((track->framesReady() >= minFrames) && track->isReady() && 2620 !track->isPaused() && !track->isTerminated()) 2621 { 2622 ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, 2623 this); 2624 2625 mixedTracks++; 2626 2627 // track->mainBuffer() != mMixBuffer means there is an effect chain 2628 // connected to the track 2629 chain.clear(); 2630 if (track->mainBuffer() != mMixBuffer) { 2631 chain = getEffectChain_l(track->sessionId()); 2632 // Delegate volume control to effect in track effect chain if needed 2633 if (chain != 0) { 2634 tracksWithEffect++; 2635 } else { 2636 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2637 "session %d", 2638 name, track->sessionId()); 2639 } 2640 } 2641 2642 2643 int param = AudioMixer::VOLUME; 2644 if (track->mFillingUpStatus == Track::FS_FILLED) { 2645 // no ramp for the first volume setting 2646 track->mFillingUpStatus = Track::FS_ACTIVE; 2647 if (track->mState == TrackBase::RESUMING) { 2648 track->mState = TrackBase::ACTIVE; 2649 param = AudioMixer::RAMP_VOLUME; 2650 } 2651 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2652 } else if (cblk->server != 0) { 2653 // If the track is stopped before the first frame was mixed, 2654 // do not apply ramp 2655 param = AudioMixer::RAMP_VOLUME; 2656 } 2657 2658 // compute volume for this track 2659 uint32_t vl, vr, va; 2660 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2661 vl = vr = va = 0; 2662 if (track->isPausing()) { 2663 track->setPaused(); 2664 } 2665 } else { 2666 2667 // read original volumes with volume control 2668 float typeVolume = mStreamTypes[track->streamType()].volume; 2669 float v = masterVolume * typeVolume; 2670 ServerProxy *proxy = track->mServerProxy; 2671 uint32_t vlr = proxy->getVolumeLR(); 2672 vl = vlr & 0xFFFF; 2673 vr = vlr >> 16; 2674 // track volumes come from shared memory, so can't be trusted and must be clamped 2675 if (vl > MAX_GAIN_INT) { 2676 ALOGV("Track left volume out of range: %04X", vl); 2677 vl = MAX_GAIN_INT; 2678 } 2679 if (vr > MAX_GAIN_INT) { 2680 ALOGV("Track right volume out of range: %04X", vr); 2681 vr = MAX_GAIN_INT; 2682 } 2683 // now apply the master volume and stream type volume 2684 vl = (uint32_t)(v * vl) << 12; 2685 vr = (uint32_t)(v * vr) << 12; 2686 // assuming master volume and stream type volume each go up to 1.0, 2687 // vl and vr are now in 8.24 format 2688 2689 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2690 // send level comes from shared memory and so may be corrupt 2691 if (sendLevel > MAX_GAIN_INT) { 2692 ALOGV("Track send level out of range: %04X", sendLevel); 2693 sendLevel = MAX_GAIN_INT; 2694 } 2695 va = (uint32_t)(v * sendLevel); 2696 } 2697 // Delegate volume control to effect in track effect chain if needed 2698 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2699 // Do not ramp volume if volume is controlled by effect 2700 param = AudioMixer::VOLUME; 2701 track->mHasVolumeController = true; 2702 } else { 2703 // force no volume ramp when volume controller was just disabled or removed 2704 // from effect chain to avoid volume spike 2705 if (track->mHasVolumeController) { 2706 param = AudioMixer::VOLUME; 2707 } 2708 track->mHasVolumeController = false; 2709 } 2710 2711 // Convert volumes from 8.24 to 4.12 format 2712 // This additional clamping is needed in case chain->setVolume_l() overshot 2713 vl = (vl + (1 << 11)) >> 12; 2714 if (vl > MAX_GAIN_INT) { 2715 vl = MAX_GAIN_INT; 2716 } 2717 vr = (vr + (1 << 11)) >> 12; 2718 if (vr > MAX_GAIN_INT) { 2719 vr = MAX_GAIN_INT; 2720 } 2721 2722 if (va > MAX_GAIN_INT) { 2723 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2724 } 2725 2726 // XXX: these things DON'T need to be done each time 2727 mAudioMixer->setBufferProvider(name, track); 2728 mAudioMixer->enable(name); 2729 2730 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2731 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2732 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2733 mAudioMixer->setParameter( 2734 name, 2735 AudioMixer::TRACK, 2736 AudioMixer::FORMAT, (void *)track->format()); 2737 mAudioMixer->setParameter( 2738 name, 2739 AudioMixer::TRACK, 2740 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2741 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 2742 uint32_t maxSampleRate = mSampleRate * 2; 2743 uint32_t reqSampleRate = track->mServerProxy->getSampleRate(); 2744 if (reqSampleRate == 0) { 2745 reqSampleRate = mSampleRate; 2746 } else if (reqSampleRate > maxSampleRate) { 2747 reqSampleRate = maxSampleRate; 2748 } 2749 mAudioMixer->setParameter( 2750 name, 2751 AudioMixer::RESAMPLE, 2752 AudioMixer::SAMPLE_RATE, 2753 (void *)reqSampleRate); 2754 mAudioMixer->setParameter( 2755 name, 2756 AudioMixer::TRACK, 2757 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2758 mAudioMixer->setParameter( 2759 name, 2760 AudioMixer::TRACK, 2761 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2762 2763 // reset retry count 2764 track->mRetryCount = kMaxTrackRetries; 2765 2766 // If one track is ready, set the mixer ready if: 2767 // - the mixer was not ready during previous round OR 2768 // - no other track is not ready 2769 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 2770 mixerStatus != MIXER_TRACKS_ENABLED) { 2771 mixerStatus = MIXER_TRACKS_READY; 2772 } 2773 } else { 2774 // clear effect chain input buffer if an active track underruns to avoid sending 2775 // previous audio buffer again to effects 2776 chain = getEffectChain_l(track->sessionId()); 2777 if (chain != 0) { 2778 chain->clearInputBuffer(); 2779 } 2780 2781 ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, 2782 cblk->server, this); 2783 if ((track->sharedBuffer() != 0) || track->isTerminated() || 2784 track->isStopped() || track->isPaused()) { 2785 // We have consumed all the buffers of this track. 2786 // Remove it from the list of active tracks. 2787 // TODO: use actual buffer filling status instead of latency when available from 2788 // audio HAL 2789 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 2790 size_t framesWritten = mBytesWritten / mFrameSize; 2791 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 2792 if (track->isStopped()) { 2793 track->reset(); 2794 } 2795 tracksToRemove->add(track); 2796 } 2797 } else { 2798 track->mUnderrunCount++; 2799 // No buffers for this track. Give it a few chances to 2800 // fill a buffer, then remove it from active list. 2801 if (--(track->mRetryCount) <= 0) { 2802 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2803 tracksToRemove->add(track); 2804 // indicate to client process that the track was disabled because of underrun; 2805 // it will then automatically call start() when data is available 2806 android_atomic_or(CBLK_DISABLED, &cblk->flags); 2807 // If one track is not ready, mark the mixer also not ready if: 2808 // - the mixer was ready during previous round OR 2809 // - no other track is ready 2810 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 2811 mixerStatus != MIXER_TRACKS_READY) { 2812 mixerStatus = MIXER_TRACKS_ENABLED; 2813 } 2814 } 2815 mAudioMixer->disable(name); 2816 } 2817 2818 } // local variable scope to avoid goto warning 2819track_is_ready: ; 2820 2821 } 2822 2823 // Push the new FastMixer state if necessary 2824 bool pauseAudioWatchdog = false; 2825 if (didModify) { 2826 state->mFastTracksGen++; 2827 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 2828 if (kUseFastMixer == FastMixer_Dynamic && 2829 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 2830 state->mCommand = FastMixerState::COLD_IDLE; 2831 state->mColdFutexAddr = &mFastMixerFutex; 2832 state->mColdGen++; 2833 mFastMixerFutex = 0; 2834 if (kUseFastMixer == FastMixer_Dynamic) { 2835 mNormalSink = mOutputSink; 2836 } 2837 // If we go into cold idle, need to wait for acknowledgement 2838 // so that fast mixer stops doing I/O. 2839 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2840 pauseAudioWatchdog = true; 2841 } 2842 sq->end(); 2843 } 2844 if (sq != NULL) { 2845 sq->end(didModify); 2846 sq->push(block); 2847 } 2848#ifdef AUDIO_WATCHDOG 2849 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 2850 mAudioWatchdog->pause(); 2851 } 2852#endif 2853 2854 // Now perform the deferred reset on fast tracks that have stopped 2855 while (resetMask != 0) { 2856 size_t i = __builtin_ctz(resetMask); 2857 ALOG_ASSERT(i < count); 2858 resetMask &= ~(1 << i); 2859 sp<Track> t = mActiveTracks[i].promote(); 2860 if (t == 0) { 2861 continue; 2862 } 2863 Track* track = t.get(); 2864 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 2865 track->reset(); 2866 } 2867 2868 // remove all the tracks that need to be... 2869 count = tracksToRemove->size(); 2870 if (CC_UNLIKELY(count)) { 2871 for (size_t i=0 ; i<count ; i++) { 2872 const sp<Track>& track = tracksToRemove->itemAt(i); 2873 mNBLogWriter->logf("prepareTracks_l remove name=%u", track->name()); 2874 mActiveTracks.remove(track); 2875 if (track->mainBuffer() != mMixBuffer) { 2876 chain = getEffectChain_l(track->sessionId()); 2877 if (chain != 0) { 2878 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2879 track->sessionId()); 2880 chain->decActiveTrackCnt(); 2881 } 2882 } 2883 if (track->isTerminated()) { 2884 removeTrack_l(track); 2885 } 2886 } 2887 } 2888 2889 // mix buffer must be cleared if all tracks are connected to an 2890 // effect chain as in this case the mixer will not write to 2891 // mix buffer and track effects will accumulate into it 2892 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 2893 (mixedTracks == 0 && fastTracks > 0)) { 2894 // FIXME as a performance optimization, should remember previous zero status 2895 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2896 } 2897 2898 // if any fast tracks, then status is ready 2899 mMixerStatusIgnoringFastTracks = mixerStatus; 2900 if (fastTracks > 0) { 2901 mixerStatus = MIXER_TRACKS_READY; 2902 } 2903 return mixerStatus; 2904} 2905 2906// getTrackName_l() must be called with ThreadBase::mLock held 2907int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 2908{ 2909 return mAudioMixer->getTrackName(channelMask, sessionId); 2910} 2911 2912// deleteTrackName_l() must be called with ThreadBase::mLock held 2913void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2914{ 2915 ALOGV("remove track (%d) and delete from mixer", name); 2916 mAudioMixer->deleteTrackName(name); 2917} 2918 2919// checkForNewParameters_l() must be called with ThreadBase::mLock held 2920bool AudioFlinger::MixerThread::checkForNewParameters_l() 2921{ 2922 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 2923 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 2924 bool reconfig = false; 2925 2926 while (!mNewParameters.isEmpty()) { 2927 2928 if (mFastMixer != NULL) { 2929 FastMixerStateQueue *sq = mFastMixer->sq(); 2930 FastMixerState *state = sq->begin(); 2931 if (!(state->mCommand & FastMixerState::IDLE)) { 2932 previousCommand = state->mCommand; 2933 state->mCommand = FastMixerState::HOT_IDLE; 2934 sq->end(); 2935 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2936 } else { 2937 sq->end(false /*didModify*/); 2938 } 2939 } 2940 2941 status_t status = NO_ERROR; 2942 String8 keyValuePair = mNewParameters[0]; 2943 AudioParameter param = AudioParameter(keyValuePair); 2944 int value; 2945 2946 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2947 reconfig = true; 2948 } 2949 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2950 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2951 status = BAD_VALUE; 2952 } else { 2953 reconfig = true; 2954 } 2955 } 2956 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2957 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2958 status = BAD_VALUE; 2959 } else { 2960 reconfig = true; 2961 } 2962 } 2963 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2964 // do not accept frame count changes if tracks are open as the track buffer 2965 // size depends on frame count and correct behavior would not be guaranteed 2966 // if frame count is changed after track creation 2967 if (!mTracks.isEmpty()) { 2968 status = INVALID_OPERATION; 2969 } else { 2970 reconfig = true; 2971 } 2972 } 2973 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2974#ifdef ADD_BATTERY_DATA 2975 // when changing the audio output device, call addBatteryData to notify 2976 // the change 2977 if (mOutDevice != value) { 2978 uint32_t params = 0; 2979 // check whether speaker is on 2980 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2981 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2982 } 2983 2984 audio_devices_t deviceWithoutSpeaker 2985 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2986 // check if any other device (except speaker) is on 2987 if (value & deviceWithoutSpeaker ) { 2988 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2989 } 2990 2991 if (params != 0) { 2992 addBatteryData(params); 2993 } 2994 } 2995#endif 2996 2997 // forward device change to effects that have requested to be 2998 // aware of attached audio device. 2999 mOutDevice = value; 3000 for (size_t i = 0; i < mEffectChains.size(); i++) { 3001 mEffectChains[i]->setDevice_l(mOutDevice); 3002 } 3003 } 3004 3005 if (status == NO_ERROR) { 3006 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3007 keyValuePair.string()); 3008 if (!mStandby && status == INVALID_OPERATION) { 3009 mOutput->stream->common.standby(&mOutput->stream->common); 3010 mStandby = true; 3011 mBytesWritten = 0; 3012 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3013 keyValuePair.string()); 3014 } 3015 if (status == NO_ERROR && reconfig) { 3016 delete mAudioMixer; 3017 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3018 mAudioMixer = NULL; 3019 readOutputParameters(); 3020 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3021 for (size_t i = 0; i < mTracks.size() ; i++) { 3022 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3023 if (name < 0) { 3024 break; 3025 } 3026 mTracks[i]->mName = name; 3027 } 3028 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3029 } 3030 } 3031 3032 mNewParameters.removeAt(0); 3033 3034 mParamStatus = status; 3035 mParamCond.signal(); 3036 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3037 // already timed out waiting for the status and will never signal the condition. 3038 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3039 } 3040 3041 if (!(previousCommand & FastMixerState::IDLE)) { 3042 ALOG_ASSERT(mFastMixer != NULL); 3043 FastMixerStateQueue *sq = mFastMixer->sq(); 3044 FastMixerState *state = sq->begin(); 3045 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3046 state->mCommand = previousCommand; 3047 sq->end(); 3048 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3049 } 3050 3051 return reconfig; 3052} 3053 3054 3055void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3056{ 3057 const size_t SIZE = 256; 3058 char buffer[SIZE]; 3059 String8 result; 3060 3061 PlaybackThread::dumpInternals(fd, args); 3062 3063 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3064 result.append(buffer); 3065 write(fd, result.string(), result.size()); 3066 3067 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3068 FastMixerDumpState copy = mFastMixerDumpState; 3069 copy.dump(fd); 3070 3071#ifdef STATE_QUEUE_DUMP 3072 // Similar for state queue 3073 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3074 observerCopy.dump(fd); 3075 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3076 mutatorCopy.dump(fd); 3077#endif 3078 3079 // Write the tee output to a .wav file 3080 dumpTee(fd, mTeeSource, mId); 3081 3082#ifdef AUDIO_WATCHDOG 3083 if (mAudioWatchdog != 0) { 3084 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3085 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3086 wdCopy.dump(fd); 3087 } 3088#endif 3089} 3090 3091uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3092{ 3093 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3094} 3095 3096uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3097{ 3098 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3099} 3100 3101void AudioFlinger::MixerThread::cacheParameters_l() 3102{ 3103 PlaybackThread::cacheParameters_l(); 3104 3105 // FIXME: Relaxed timing because of a certain device that can't meet latency 3106 // Should be reduced to 2x after the vendor fixes the driver issue 3107 // increase threshold again due to low power audio mode. The way this warning 3108 // threshold is calculated and its usefulness should be reconsidered anyway. 3109 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3110} 3111 3112// ---------------------------------------------------------------------------- 3113 3114AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3115 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3116 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3117 // mLeftVolFloat, mRightVolFloat 3118{ 3119} 3120 3121AudioFlinger::DirectOutputThread::~DirectOutputThread() 3122{ 3123} 3124 3125AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3126 Vector< sp<Track> > *tracksToRemove 3127) 3128{ 3129 sp<Track> trackToRemove; 3130 3131 mixer_state mixerStatus = MIXER_IDLE; 3132 3133 // find out which tracks need to be processed 3134 if (mActiveTracks.size() != 0) { 3135 sp<Track> t = mActiveTracks[0].promote(); 3136 // The track died recently 3137 if (t == 0) { 3138 return MIXER_IDLE; 3139 } 3140 3141 Track* const track = t.get(); 3142 audio_track_cblk_t* cblk = track->cblk(); 3143 3144 // The first time a track is added we wait 3145 // for all its buffers to be filled before processing it 3146 uint32_t minFrames; 3147 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3148 minFrames = mNormalFrameCount; 3149 } else { 3150 minFrames = 1; 3151 } 3152 if ((track->framesReady() >= minFrames) && track->isReady() && 3153 !track->isPaused() && !track->isTerminated()) 3154 { 3155 ALOGVV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3156 3157 if (track->mFillingUpStatus == Track::FS_FILLED) { 3158 track->mFillingUpStatus = Track::FS_ACTIVE; 3159 mLeftVolFloat = mRightVolFloat = 0; 3160 if (track->mState == TrackBase::RESUMING) { 3161 track->mState = TrackBase::ACTIVE; 3162 } 3163 } 3164 3165 // compute volume for this track 3166 float left, right; 3167 if (mMasterMute || track->isPausing() || mStreamTypes[track->streamType()].mute) { 3168 left = right = 0; 3169 if (track->isPausing()) { 3170 track->setPaused(); 3171 } 3172 } else { 3173 float typeVolume = mStreamTypes[track->streamType()].volume; 3174 float v = mMasterVolume * typeVolume; 3175 uint32_t vlr = track->mServerProxy->getVolumeLR(); 3176 float v_clamped = v * (vlr & 0xFFFF); 3177 if (v_clamped > MAX_GAIN) { 3178 v_clamped = MAX_GAIN; 3179 } 3180 left = v_clamped/MAX_GAIN; 3181 v_clamped = v * (vlr >> 16); 3182 if (v_clamped > MAX_GAIN) { 3183 v_clamped = MAX_GAIN; 3184 } 3185 right = v_clamped/MAX_GAIN; 3186 } 3187 3188 if (left != mLeftVolFloat || right != mRightVolFloat) { 3189 mLeftVolFloat = left; 3190 mRightVolFloat = right; 3191 3192 // Convert volumes from float to 8.24 3193 uint32_t vl = (uint32_t)(left * (1 << 24)); 3194 uint32_t vr = (uint32_t)(right * (1 << 24)); 3195 3196 // Delegate volume control to effect in track effect chain if needed 3197 // only one effect chain can be present on DirectOutputThread, so if 3198 // there is one, the track is connected to it 3199 if (!mEffectChains.isEmpty()) { 3200 // Do not ramp volume if volume is controlled by effect 3201 mEffectChains[0]->setVolume_l(&vl, &vr); 3202 left = (float)vl / (1 << 24); 3203 right = (float)vr / (1 << 24); 3204 } 3205 mOutput->stream->set_volume(mOutput->stream, left, right); 3206 } 3207 3208 // reset retry count 3209 track->mRetryCount = kMaxTrackRetriesDirect; 3210 mActiveTrack = t; 3211 mixerStatus = MIXER_TRACKS_READY; 3212 } else { 3213 // clear effect chain input buffer if an active track underruns to avoid sending 3214 // previous audio buffer again to effects 3215 if (!mEffectChains.isEmpty()) { 3216 mEffectChains[0]->clearInputBuffer(); 3217 } 3218 3219 ALOGVV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3220 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3221 track->isStopped() || track->isPaused()) { 3222 // We have consumed all the buffers of this track. 3223 // Remove it from the list of active tracks. 3224 // TODO: implement behavior for compressed audio 3225 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3226 size_t framesWritten = mBytesWritten / mFrameSize; 3227 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3228 if (track->isStopped()) { 3229 track->reset(); 3230 } 3231 trackToRemove = track; 3232 } 3233 } else { 3234 // No buffers for this track. Give it a few chances to 3235 // fill a buffer, then remove it from active list. 3236 if (--(track->mRetryCount) <= 0) { 3237 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3238 trackToRemove = track; 3239 } else { 3240 mixerStatus = MIXER_TRACKS_ENABLED; 3241 } 3242 } 3243 } 3244 } 3245 3246 // FIXME merge this with similar code for removing multiple tracks 3247 // remove all the tracks that need to be... 3248 if (CC_UNLIKELY(trackToRemove != 0)) { 3249 tracksToRemove->add(trackToRemove); 3250#if 0 3251 mNBLogWriter->logf("prepareTracks_l remove name=%u", trackToRemove->name()); 3252#endif 3253 mActiveTracks.remove(trackToRemove); 3254 if (!mEffectChains.isEmpty()) { 3255 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3256 trackToRemove->sessionId()); 3257 mEffectChains[0]->decActiveTrackCnt(); 3258 } 3259 if (trackToRemove->isTerminated()) { 3260 removeTrack_l(trackToRemove); 3261 } 3262 } 3263 3264 return mixerStatus; 3265} 3266 3267void AudioFlinger::DirectOutputThread::threadLoop_mix() 3268{ 3269 AudioBufferProvider::Buffer buffer; 3270 size_t frameCount = mFrameCount; 3271 int8_t *curBuf = (int8_t *)mMixBuffer; 3272 // output audio to hardware 3273 while (frameCount) { 3274 buffer.frameCount = frameCount; 3275 mActiveTrack->getNextBuffer(&buffer); 3276 if (CC_UNLIKELY(buffer.raw == NULL)) { 3277 memset(curBuf, 0, frameCount * mFrameSize); 3278 break; 3279 } 3280 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3281 frameCount -= buffer.frameCount; 3282 curBuf += buffer.frameCount * mFrameSize; 3283 mActiveTrack->releaseBuffer(&buffer); 3284 } 3285 sleepTime = 0; 3286 standbyTime = systemTime() + standbyDelay; 3287 mActiveTrack.clear(); 3288 3289} 3290 3291void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3292{ 3293 if (sleepTime == 0) { 3294 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3295 sleepTime = activeSleepTime; 3296 } else { 3297 sleepTime = idleSleepTime; 3298 } 3299 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3300 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3301 sleepTime = 0; 3302 } 3303} 3304 3305// getTrackName_l() must be called with ThreadBase::mLock held 3306int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3307 int sessionId) 3308{ 3309 return 0; 3310} 3311 3312// deleteTrackName_l() must be called with ThreadBase::mLock held 3313void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3314{ 3315} 3316 3317// checkForNewParameters_l() must be called with ThreadBase::mLock held 3318bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3319{ 3320 bool reconfig = false; 3321 3322 while (!mNewParameters.isEmpty()) { 3323 status_t status = NO_ERROR; 3324 String8 keyValuePair = mNewParameters[0]; 3325 AudioParameter param = AudioParameter(keyValuePair); 3326 int value; 3327 3328 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3329 // do not accept frame count changes if tracks are open as the track buffer 3330 // size depends on frame count and correct behavior would not be garantied 3331 // if frame count is changed after track creation 3332 if (!mTracks.isEmpty()) { 3333 status = INVALID_OPERATION; 3334 } else { 3335 reconfig = true; 3336 } 3337 } 3338 if (status == NO_ERROR) { 3339 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3340 keyValuePair.string()); 3341 if (!mStandby && status == INVALID_OPERATION) { 3342 mOutput->stream->common.standby(&mOutput->stream->common); 3343 mStandby = true; 3344 mBytesWritten = 0; 3345 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3346 keyValuePair.string()); 3347 } 3348 if (status == NO_ERROR && reconfig) { 3349 readOutputParameters(); 3350 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3351 } 3352 } 3353 3354 mNewParameters.removeAt(0); 3355 3356 mParamStatus = status; 3357 mParamCond.signal(); 3358 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3359 // already timed out waiting for the status and will never signal the condition. 3360 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3361 } 3362 return reconfig; 3363} 3364 3365uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3366{ 3367 uint32_t time; 3368 if (audio_is_linear_pcm(mFormat)) { 3369 time = PlaybackThread::activeSleepTimeUs(); 3370 } else { 3371 time = 10000; 3372 } 3373 return time; 3374} 3375 3376uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3377{ 3378 uint32_t time; 3379 if (audio_is_linear_pcm(mFormat)) { 3380 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3381 } else { 3382 time = 10000; 3383 } 3384 return time; 3385} 3386 3387uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3388{ 3389 uint32_t time; 3390 if (audio_is_linear_pcm(mFormat)) { 3391 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3392 } else { 3393 time = 10000; 3394 } 3395 return time; 3396} 3397 3398void AudioFlinger::DirectOutputThread::cacheParameters_l() 3399{ 3400 PlaybackThread::cacheParameters_l(); 3401 3402 // use shorter standby delay as on normal output to release 3403 // hardware resources as soon as possible 3404 standbyDelay = microseconds(activeSleepTime*2); 3405} 3406 3407// ---------------------------------------------------------------------------- 3408 3409AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3410 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3411 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3412 DUPLICATING), 3413 mWaitTimeMs(UINT_MAX) 3414{ 3415 addOutputTrack(mainThread); 3416} 3417 3418AudioFlinger::DuplicatingThread::~DuplicatingThread() 3419{ 3420 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3421 mOutputTracks[i]->destroy(); 3422 } 3423} 3424 3425void AudioFlinger::DuplicatingThread::threadLoop_mix() 3426{ 3427 // mix buffers... 3428 if (outputsReady(outputTracks)) { 3429 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3430 } else { 3431 memset(mMixBuffer, 0, mixBufferSize); 3432 } 3433 sleepTime = 0; 3434 writeFrames = mNormalFrameCount; 3435 standbyTime = systemTime() + standbyDelay; 3436} 3437 3438void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3439{ 3440 if (sleepTime == 0) { 3441 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3442 sleepTime = activeSleepTime; 3443 } else { 3444 sleepTime = idleSleepTime; 3445 } 3446 } else if (mBytesWritten != 0) { 3447 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3448 writeFrames = mNormalFrameCount; 3449 memset(mMixBuffer, 0, mixBufferSize); 3450 } else { 3451 // flush remaining overflow buffers in output tracks 3452 writeFrames = 0; 3453 } 3454 sleepTime = 0; 3455 } 3456} 3457 3458void AudioFlinger::DuplicatingThread::threadLoop_write() 3459{ 3460 for (size_t i = 0; i < outputTracks.size(); i++) { 3461 outputTracks[i]->write(mMixBuffer, writeFrames); 3462 } 3463 mBytesWritten += mixBufferSize; 3464} 3465 3466void AudioFlinger::DuplicatingThread::threadLoop_standby() 3467{ 3468 // DuplicatingThread implements standby by stopping all tracks 3469 for (size_t i = 0; i < outputTracks.size(); i++) { 3470 outputTracks[i]->stop(); 3471 } 3472} 3473 3474void AudioFlinger::DuplicatingThread::saveOutputTracks() 3475{ 3476 outputTracks = mOutputTracks; 3477} 3478 3479void AudioFlinger::DuplicatingThread::clearOutputTracks() 3480{ 3481 outputTracks.clear(); 3482} 3483 3484void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3485{ 3486 Mutex::Autolock _l(mLock); 3487 // FIXME explain this formula 3488 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3489 OutputTrack *outputTrack = new OutputTrack(thread, 3490 this, 3491 mSampleRate, 3492 mFormat, 3493 mChannelMask, 3494 frameCount); 3495 if (outputTrack->cblk() != NULL) { 3496 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3497 mOutputTracks.add(outputTrack); 3498 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3499 updateWaitTime_l(); 3500 } 3501} 3502 3503void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3504{ 3505 Mutex::Autolock _l(mLock); 3506 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3507 if (mOutputTracks[i]->thread() == thread) { 3508 mOutputTracks[i]->destroy(); 3509 mOutputTracks.removeAt(i); 3510 updateWaitTime_l(); 3511 return; 3512 } 3513 } 3514 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3515} 3516 3517// caller must hold mLock 3518void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3519{ 3520 mWaitTimeMs = UINT_MAX; 3521 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3522 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3523 if (strong != 0) { 3524 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3525 if (waitTimeMs < mWaitTimeMs) { 3526 mWaitTimeMs = waitTimeMs; 3527 } 3528 } 3529 } 3530} 3531 3532 3533bool AudioFlinger::DuplicatingThread::outputsReady( 3534 const SortedVector< sp<OutputTrack> > &outputTracks) 3535{ 3536 for (size_t i = 0; i < outputTracks.size(); i++) { 3537 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3538 if (thread == 0) { 3539 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 3540 outputTracks[i].get()); 3541 return false; 3542 } 3543 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3544 // see note at standby() declaration 3545 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3546 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 3547 thread.get()); 3548 return false; 3549 } 3550 } 3551 return true; 3552} 3553 3554uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3555{ 3556 return (mWaitTimeMs * 1000) / 2; 3557} 3558 3559void AudioFlinger::DuplicatingThread::cacheParameters_l() 3560{ 3561 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3562 updateWaitTime_l(); 3563 3564 MixerThread::cacheParameters_l(); 3565} 3566 3567// ---------------------------------------------------------------------------- 3568// Record 3569// ---------------------------------------------------------------------------- 3570 3571AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3572 AudioStreamIn *input, 3573 uint32_t sampleRate, 3574 audio_channel_mask_t channelMask, 3575 audio_io_handle_t id, 3576 audio_devices_t outDevice, 3577 audio_devices_t inDevice, 3578 const sp<NBAIO_Sink>& teeSink) : 3579 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 3580 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 3581 // mRsmpInIndex and mInputBytes set by readInputParameters() 3582 mReqChannelCount(popcount(channelMask)), 3583 mReqSampleRate(sampleRate), 3584 // mBytesRead is only meaningful while active, and so is cleared in start() 3585 // (but might be better to also clear here for dump?) 3586 mTeeSink(teeSink) 3587{ 3588 snprintf(mName, kNameLength, "AudioIn_%X", id); 3589 3590 readInputParameters(); 3591 3592} 3593 3594 3595AudioFlinger::RecordThread::~RecordThread() 3596{ 3597 delete[] mRsmpInBuffer; 3598 delete mResampler; 3599 delete[] mRsmpOutBuffer; 3600} 3601 3602void AudioFlinger::RecordThread::onFirstRef() 3603{ 3604 run(mName, PRIORITY_URGENT_AUDIO); 3605} 3606 3607status_t AudioFlinger::RecordThread::readyToRun() 3608{ 3609 status_t status = initCheck(); 3610 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 3611 return status; 3612} 3613 3614bool AudioFlinger::RecordThread::threadLoop() 3615{ 3616 AudioBufferProvider::Buffer buffer; 3617 sp<RecordTrack> activeTrack; 3618 Vector< sp<EffectChain> > effectChains; 3619 3620 nsecs_t lastWarning = 0; 3621 3622 inputStandBy(); 3623 acquireWakeLock(); 3624 3625 // used to verify we've read at least once before evaluating how many bytes were read 3626 bool readOnce = false; 3627 3628 // start recording 3629 while (!exitPending()) { 3630 3631 processConfigEvents(); 3632 3633 { // scope for mLock 3634 Mutex::Autolock _l(mLock); 3635 checkForNewParameters_l(); 3636 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3637 standby(); 3638 3639 if (exitPending()) { 3640 break; 3641 } 3642 3643 releaseWakeLock_l(); 3644 ALOGV("RecordThread: loop stopping"); 3645 // go to sleep 3646 mWaitWorkCV.wait(mLock); 3647 ALOGV("RecordThread: loop starting"); 3648 acquireWakeLock_l(); 3649 continue; 3650 } 3651 if (mActiveTrack != 0) { 3652 if (mActiveTrack->mState == TrackBase::PAUSING) { 3653 standby(); 3654 mActiveTrack.clear(); 3655 mStartStopCond.broadcast(); 3656 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3657 if (mReqChannelCount != mActiveTrack->channelCount()) { 3658 mActiveTrack.clear(); 3659 mStartStopCond.broadcast(); 3660 } else if (readOnce) { 3661 // record start succeeds only if first read from audio input 3662 // succeeds 3663 if (mBytesRead >= 0) { 3664 mActiveTrack->mState = TrackBase::ACTIVE; 3665 } else { 3666 mActiveTrack.clear(); 3667 } 3668 mStartStopCond.broadcast(); 3669 } 3670 mStandby = false; 3671 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 3672 removeTrack_l(mActiveTrack); 3673 mActiveTrack.clear(); 3674 } 3675 } 3676 lockEffectChains_l(effectChains); 3677 } 3678 3679 if (mActiveTrack != 0) { 3680 if (mActiveTrack->mState != TrackBase::ACTIVE && 3681 mActiveTrack->mState != TrackBase::RESUMING) { 3682 unlockEffectChains(effectChains); 3683 usleep(kRecordThreadSleepUs); 3684 continue; 3685 } 3686 for (size_t i = 0; i < effectChains.size(); i ++) { 3687 effectChains[i]->process_l(); 3688 } 3689 3690 buffer.frameCount = mFrameCount; 3691 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3692 readOnce = true; 3693 size_t framesOut = buffer.frameCount; 3694 if (mResampler == NULL) { 3695 // no resampling 3696 while (framesOut) { 3697 size_t framesIn = mFrameCount - mRsmpInIndex; 3698 if (framesIn) { 3699 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3700 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 3701 mActiveTrack->mFrameSize; 3702 if (framesIn > framesOut) 3703 framesIn = framesOut; 3704 mRsmpInIndex += framesIn; 3705 framesOut -= framesIn; 3706 if (mChannelCount == mReqChannelCount || 3707 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3708 memcpy(dst, src, framesIn * mFrameSize); 3709 } else { 3710 if (mChannelCount == 1) { 3711 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 3712 (int16_t *)src, framesIn); 3713 } else { 3714 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 3715 (int16_t *)src, framesIn); 3716 } 3717 } 3718 } 3719 if (framesOut && mFrameCount == mRsmpInIndex) { 3720 void *readInto; 3721 if (framesOut == mFrameCount && 3722 (mChannelCount == mReqChannelCount || 3723 mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3724 readInto = buffer.raw; 3725 framesOut = 0; 3726 } else { 3727 readInto = mRsmpInBuffer; 3728 mRsmpInIndex = 0; 3729 } 3730 mBytesRead = mInput->stream->read(mInput->stream, readInto, mInputBytes); 3731 if (mBytesRead <= 0) { 3732 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 3733 { 3734 ALOGE("Error reading audio input"); 3735 // Force input into standby so that it tries to 3736 // recover at next read attempt 3737 inputStandBy(); 3738 usleep(kRecordThreadSleepUs); 3739 } 3740 mRsmpInIndex = mFrameCount; 3741 framesOut = 0; 3742 buffer.frameCount = 0; 3743 } else if (mTeeSink != 0) { 3744 (void) mTeeSink->write(readInto, 3745 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 3746 } 3747 } 3748 } 3749 } else { 3750 // resampling 3751 3752 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3753 // alter output frame count as if we were expecting stereo samples 3754 if (mChannelCount == 1 && mReqChannelCount == 1) { 3755 framesOut >>= 1; 3756 } 3757 mResampler->resample(mRsmpOutBuffer, framesOut, 3758 this /* AudioBufferProvider* */); 3759 // ditherAndClamp() works as long as all buffers returned by 3760 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 3761 if (mChannelCount == 2 && mReqChannelCount == 1) { 3762 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3763 // the resampler always outputs stereo samples: 3764 // do post stereo to mono conversion 3765 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 3766 framesOut); 3767 } else { 3768 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3769 } 3770 3771 } 3772 if (mFramestoDrop == 0) { 3773 mActiveTrack->releaseBuffer(&buffer); 3774 } else { 3775 if (mFramestoDrop > 0) { 3776 mFramestoDrop -= buffer.frameCount; 3777 if (mFramestoDrop <= 0) { 3778 clearSyncStartEvent(); 3779 } 3780 } else { 3781 mFramestoDrop += buffer.frameCount; 3782 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 3783 mSyncStartEvent->isCancelled()) { 3784 ALOGW("Synced record %s, session %d, trigger session %d", 3785 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 3786 mActiveTrack->sessionId(), 3787 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 3788 clearSyncStartEvent(); 3789 } 3790 } 3791 } 3792 mActiveTrack->clearOverflow(); 3793 } 3794 // client isn't retrieving buffers fast enough 3795 else { 3796 if (!mActiveTrack->setOverflow()) { 3797 nsecs_t now = systemTime(); 3798 if ((now - lastWarning) > kWarningThrottleNs) { 3799 ALOGW("RecordThread: buffer overflow"); 3800 lastWarning = now; 3801 } 3802 } 3803 // Release the processor for a while before asking for a new buffer. 3804 // This will give the application more chance to read from the buffer and 3805 // clear the overflow. 3806 usleep(kRecordThreadSleepUs); 3807 } 3808 } 3809 // enable changes in effect chain 3810 unlockEffectChains(effectChains); 3811 effectChains.clear(); 3812 } 3813 3814 standby(); 3815 3816 { 3817 Mutex::Autolock _l(mLock); 3818 mActiveTrack.clear(); 3819 mStartStopCond.broadcast(); 3820 } 3821 3822 releaseWakeLock(); 3823 3824 ALOGV("RecordThread %p exiting", this); 3825 return false; 3826} 3827 3828void AudioFlinger::RecordThread::standby() 3829{ 3830 if (!mStandby) { 3831 inputStandBy(); 3832 mStandby = true; 3833 } 3834} 3835 3836void AudioFlinger::RecordThread::inputStandBy() 3837{ 3838 mInput->stream->common.standby(&mInput->stream->common); 3839} 3840 3841sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 3842 const sp<AudioFlinger::Client>& client, 3843 uint32_t sampleRate, 3844 audio_format_t format, 3845 audio_channel_mask_t channelMask, 3846 size_t frameCount, 3847 int sessionId, 3848 IAudioFlinger::track_flags_t flags, 3849 pid_t tid, 3850 status_t *status) 3851{ 3852 sp<RecordTrack> track; 3853 status_t lStatus; 3854 3855 lStatus = initCheck(); 3856 if (lStatus != NO_ERROR) { 3857 ALOGE("Audio driver not initialized."); 3858 goto Exit; 3859 } 3860 3861 // FIXME use flags and tid similar to createTrack_l() 3862 3863 { // scope for mLock 3864 Mutex::Autolock _l(mLock); 3865 3866 track = new RecordTrack(this, client, sampleRate, 3867 format, channelMask, frameCount, sessionId); 3868 3869 if (track->getCblk() == 0) { 3870 lStatus = NO_MEMORY; 3871 goto Exit; 3872 } 3873 mTracks.add(track); 3874 3875 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 3876 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 3877 mAudioFlinger->btNrecIsOff(); 3878 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 3879 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 3880 } 3881 lStatus = NO_ERROR; 3882 3883Exit: 3884 if (status) { 3885 *status = lStatus; 3886 } 3887 return track; 3888} 3889 3890status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 3891 AudioSystem::sync_event_t event, 3892 int triggerSession) 3893{ 3894 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 3895 sp<ThreadBase> strongMe = this; 3896 status_t status = NO_ERROR; 3897 3898 if (event == AudioSystem::SYNC_EVENT_NONE) { 3899 clearSyncStartEvent(); 3900 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 3901 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 3902 triggerSession, 3903 recordTrack->sessionId(), 3904 syncStartEventCallback, 3905 this); 3906 // Sync event can be cancelled by the trigger session if the track is not in a 3907 // compatible state in which case we start record immediately 3908 if (mSyncStartEvent->isCancelled()) { 3909 clearSyncStartEvent(); 3910 } else { 3911 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 3912 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 3913 } 3914 } 3915 3916 { 3917 AutoMutex lock(mLock); 3918 if (mActiveTrack != 0) { 3919 if (recordTrack != mActiveTrack.get()) { 3920 status = -EBUSY; 3921 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3922 mActiveTrack->mState = TrackBase::ACTIVE; 3923 } 3924 return status; 3925 } 3926 3927 recordTrack->mState = TrackBase::IDLE; 3928 mActiveTrack = recordTrack; 3929 mLock.unlock(); 3930 status_t status = AudioSystem::startInput(mId); 3931 mLock.lock(); 3932 if (status != NO_ERROR) { 3933 mActiveTrack.clear(); 3934 clearSyncStartEvent(); 3935 return status; 3936 } 3937 mRsmpInIndex = mFrameCount; 3938 mBytesRead = 0; 3939 if (mResampler != NULL) { 3940 mResampler->reset(); 3941 } 3942 mActiveTrack->mState = TrackBase::RESUMING; 3943 // signal thread to start 3944 ALOGV("Signal record thread"); 3945 mWaitWorkCV.broadcast(); 3946 // do not wait for mStartStopCond if exiting 3947 if (exitPending()) { 3948 mActiveTrack.clear(); 3949 status = INVALID_OPERATION; 3950 goto startError; 3951 } 3952 mStartStopCond.wait(mLock); 3953 if (mActiveTrack == 0) { 3954 ALOGV("Record failed to start"); 3955 status = BAD_VALUE; 3956 goto startError; 3957 } 3958 ALOGV("Record started OK"); 3959 return status; 3960 } 3961startError: 3962 AudioSystem::stopInput(mId); 3963 clearSyncStartEvent(); 3964 return status; 3965} 3966 3967void AudioFlinger::RecordThread::clearSyncStartEvent() 3968{ 3969 if (mSyncStartEvent != 0) { 3970 mSyncStartEvent->cancel(); 3971 } 3972 mSyncStartEvent.clear(); 3973 mFramestoDrop = 0; 3974} 3975 3976void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 3977{ 3978 sp<SyncEvent> strongEvent = event.promote(); 3979 3980 if (strongEvent != 0) { 3981 RecordThread *me = (RecordThread *)strongEvent->cookie(); 3982 me->handleSyncStartEvent(strongEvent); 3983 } 3984} 3985 3986void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 3987{ 3988 if (event == mSyncStartEvent) { 3989 // TODO: use actual buffer filling status instead of 2 buffers when info is available 3990 // from audio HAL 3991 mFramestoDrop = mFrameCount * 2; 3992 } 3993} 3994 3995bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 3996 ALOGV("RecordThread::stop"); 3997 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 3998 return false; 3999 } 4000 recordTrack->mState = TrackBase::PAUSING; 4001 // do not wait for mStartStopCond if exiting 4002 if (exitPending()) { 4003 return true; 4004 } 4005 mStartStopCond.wait(mLock); 4006 // if we have been restarted, recordTrack == mActiveTrack.get() here 4007 if (exitPending() || recordTrack != mActiveTrack.get()) { 4008 ALOGV("Record stopped OK"); 4009 return true; 4010 } 4011 return false; 4012} 4013 4014bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4015{ 4016 return false; 4017} 4018 4019status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4020{ 4021#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4022 if (!isValidSyncEvent(event)) { 4023 return BAD_VALUE; 4024 } 4025 4026 int eventSession = event->triggerSession(); 4027 status_t ret = NAME_NOT_FOUND; 4028 4029 Mutex::Autolock _l(mLock); 4030 4031 for (size_t i = 0; i < mTracks.size(); i++) { 4032 sp<RecordTrack> track = mTracks[i]; 4033 if (eventSession == track->sessionId()) { 4034 (void) track->setSyncEvent(event); 4035 ret = NO_ERROR; 4036 } 4037 } 4038 return ret; 4039#else 4040 return BAD_VALUE; 4041#endif 4042} 4043 4044// destroyTrack_l() must be called with ThreadBase::mLock held 4045void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4046{ 4047 track->mState = TrackBase::TERMINATED; 4048 // active tracks are removed by threadLoop() 4049 if (mActiveTrack != track) { 4050 removeTrack_l(track); 4051 } 4052} 4053 4054void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4055{ 4056 mTracks.remove(track); 4057 // need anything related to effects here? 4058} 4059 4060void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4061{ 4062 dumpInternals(fd, args); 4063 dumpTracks(fd, args); 4064 dumpEffectChains(fd, args); 4065} 4066 4067void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4068{ 4069 const size_t SIZE = 256; 4070 char buffer[SIZE]; 4071 String8 result; 4072 4073 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4074 result.append(buffer); 4075 4076 if (mActiveTrack != 0) { 4077 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4078 result.append(buffer); 4079 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4080 result.append(buffer); 4081 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4082 result.append(buffer); 4083 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4084 result.append(buffer); 4085 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4086 result.append(buffer); 4087 } else { 4088 result.append("No active record client\n"); 4089 } 4090 4091 write(fd, result.string(), result.size()); 4092 4093 dumpBase(fd, args); 4094} 4095 4096void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4097{ 4098 const size_t SIZE = 256; 4099 char buffer[SIZE]; 4100 String8 result; 4101 4102 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4103 result.append(buffer); 4104 RecordTrack::appendDumpHeader(result); 4105 for (size_t i = 0; i < mTracks.size(); ++i) { 4106 sp<RecordTrack> track = mTracks[i]; 4107 if (track != 0) { 4108 track->dump(buffer, SIZE); 4109 result.append(buffer); 4110 } 4111 } 4112 4113 if (mActiveTrack != 0) { 4114 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4115 result.append(buffer); 4116 RecordTrack::appendDumpHeader(result); 4117 mActiveTrack->dump(buffer, SIZE); 4118 result.append(buffer); 4119 4120 } 4121 write(fd, result.string(), result.size()); 4122} 4123 4124// AudioBufferProvider interface 4125status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4126{ 4127 size_t framesReq = buffer->frameCount; 4128 size_t framesReady = mFrameCount - mRsmpInIndex; 4129 int channelCount; 4130 4131 if (framesReady == 0) { 4132 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4133 if (mBytesRead <= 0) { 4134 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4135 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4136 // Force input into standby so that it tries to 4137 // recover at next read attempt 4138 inputStandBy(); 4139 usleep(kRecordThreadSleepUs); 4140 } 4141 buffer->raw = NULL; 4142 buffer->frameCount = 0; 4143 return NOT_ENOUGH_DATA; 4144 } 4145 mRsmpInIndex = 0; 4146 framesReady = mFrameCount; 4147 } 4148 4149 if (framesReq > framesReady) { 4150 framesReq = framesReady; 4151 } 4152 4153 if (mChannelCount == 1 && mReqChannelCount == 2) { 4154 channelCount = 1; 4155 } else { 4156 channelCount = 2; 4157 } 4158 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4159 buffer->frameCount = framesReq; 4160 return NO_ERROR; 4161} 4162 4163// AudioBufferProvider interface 4164void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4165{ 4166 mRsmpInIndex += buffer->frameCount; 4167 buffer->frameCount = 0; 4168} 4169 4170bool AudioFlinger::RecordThread::checkForNewParameters_l() 4171{ 4172 bool reconfig = false; 4173 4174 while (!mNewParameters.isEmpty()) { 4175 status_t status = NO_ERROR; 4176 String8 keyValuePair = mNewParameters[0]; 4177 AudioParameter param = AudioParameter(keyValuePair); 4178 int value; 4179 audio_format_t reqFormat = mFormat; 4180 uint32_t reqSamplingRate = mReqSampleRate; 4181 uint32_t reqChannelCount = mReqChannelCount; 4182 4183 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4184 reqSamplingRate = value; 4185 reconfig = true; 4186 } 4187 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4188 reqFormat = (audio_format_t) value; 4189 reconfig = true; 4190 } 4191 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4192 reqChannelCount = popcount(value); 4193 reconfig = true; 4194 } 4195 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4196 // do not accept frame count changes if tracks are open as the track buffer 4197 // size depends on frame count and correct behavior would not be guaranteed 4198 // if frame count is changed after track creation 4199 if (mActiveTrack != 0) { 4200 status = INVALID_OPERATION; 4201 } else { 4202 reconfig = true; 4203 } 4204 } 4205 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4206 // forward device change to effects that have requested to be 4207 // aware of attached audio device. 4208 for (size_t i = 0; i < mEffectChains.size(); i++) { 4209 mEffectChains[i]->setDevice_l(value); 4210 } 4211 4212 // store input device and output device but do not forward output device to audio HAL. 4213 // Note that status is ignored by the caller for output device 4214 // (see AudioFlinger::setParameters() 4215 if (audio_is_output_devices(value)) { 4216 mOutDevice = value; 4217 status = BAD_VALUE; 4218 } else { 4219 mInDevice = value; 4220 // disable AEC and NS if the device is a BT SCO headset supporting those 4221 // pre processings 4222 if (mTracks.size() > 0) { 4223 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4224 mAudioFlinger->btNrecIsOff(); 4225 for (size_t i = 0; i < mTracks.size(); i++) { 4226 sp<RecordTrack> track = mTracks[i]; 4227 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4228 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4229 } 4230 } 4231 } 4232 } 4233 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4234 mAudioSource != (audio_source_t)value) { 4235 // forward device change to effects that have requested to be 4236 // aware of attached audio device. 4237 for (size_t i = 0; i < mEffectChains.size(); i++) { 4238 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4239 } 4240 mAudioSource = (audio_source_t)value; 4241 } 4242 if (status == NO_ERROR) { 4243 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4244 keyValuePair.string()); 4245 if (status == INVALID_OPERATION) { 4246 inputStandBy(); 4247 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4248 keyValuePair.string()); 4249 } 4250 if (reconfig) { 4251 if (status == BAD_VALUE && 4252 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4253 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4254 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4255 <= (2 * reqSamplingRate)) && 4256 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4257 <= FCC_2 && 4258 (reqChannelCount <= FCC_2)) { 4259 status = NO_ERROR; 4260 } 4261 if (status == NO_ERROR) { 4262 readInputParameters(); 4263 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4264 } 4265 } 4266 } 4267 4268 mNewParameters.removeAt(0); 4269 4270 mParamStatus = status; 4271 mParamCond.signal(); 4272 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4273 // already timed out waiting for the status and will never signal the condition. 4274 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4275 } 4276 return reconfig; 4277} 4278 4279String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4280{ 4281 char *s; 4282 String8 out_s8 = String8(); 4283 4284 Mutex::Autolock _l(mLock); 4285 if (initCheck() != NO_ERROR) { 4286 return out_s8; 4287 } 4288 4289 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4290 out_s8 = String8(s); 4291 free(s); 4292 return out_s8; 4293} 4294 4295void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4296 AudioSystem::OutputDescriptor desc; 4297 void *param2 = NULL; 4298 4299 switch (event) { 4300 case AudioSystem::INPUT_OPENED: 4301 case AudioSystem::INPUT_CONFIG_CHANGED: 4302 desc.channels = mChannelMask; 4303 desc.samplingRate = mSampleRate; 4304 desc.format = mFormat; 4305 desc.frameCount = mFrameCount; 4306 desc.latency = 0; 4307 param2 = &desc; 4308 break; 4309 4310 case AudioSystem::INPUT_CLOSED: 4311 default: 4312 break; 4313 } 4314 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4315} 4316 4317void AudioFlinger::RecordThread::readInputParameters() 4318{ 4319 delete mRsmpInBuffer; 4320 // mRsmpInBuffer is always assigned a new[] below 4321 delete mRsmpOutBuffer; 4322 mRsmpOutBuffer = NULL; 4323 delete mResampler; 4324 mResampler = NULL; 4325 4326 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4327 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4328 mChannelCount = (uint16_t)popcount(mChannelMask); 4329 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4330 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4331 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4332 mFrameCount = mInputBytes / mFrameSize; 4333 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 4334 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4335 4336 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4337 { 4338 int channelCount; 4339 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4340 // stereo to mono post process as the resampler always outputs stereo. 4341 if (mChannelCount == 1 && mReqChannelCount == 2) { 4342 channelCount = 1; 4343 } else { 4344 channelCount = 2; 4345 } 4346 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4347 mResampler->setSampleRate(mSampleRate); 4348 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4349 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4350 4351 // optmization: if mono to mono, alter input frame count as if we were inputing 4352 // stereo samples 4353 if (mChannelCount == 1 && mReqChannelCount == 1) { 4354 mFrameCount >>= 1; 4355 } 4356 4357 } 4358 mRsmpInIndex = mFrameCount; 4359} 4360 4361unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4362{ 4363 Mutex::Autolock _l(mLock); 4364 if (initCheck() != NO_ERROR) { 4365 return 0; 4366 } 4367 4368 return mInput->stream->get_input_frames_lost(mInput->stream); 4369} 4370 4371uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 4372{ 4373 Mutex::Autolock _l(mLock); 4374 uint32_t result = 0; 4375 if (getEffectChain_l(sessionId) != 0) { 4376 result = EFFECT_SESSION; 4377 } 4378 4379 for (size_t i = 0; i < mTracks.size(); ++i) { 4380 if (sessionId == mTracks[i]->sessionId()) { 4381 result |= TRACK_SESSION; 4382 break; 4383 } 4384 } 4385 4386 return result; 4387} 4388 4389KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 4390{ 4391 KeyedVector<int, bool> ids; 4392 Mutex::Autolock _l(mLock); 4393 for (size_t j = 0; j < mTracks.size(); ++j) { 4394 sp<RecordThread::RecordTrack> track = mTracks[j]; 4395 int sessionId = track->sessionId(); 4396 if (ids.indexOfKey(sessionId) < 0) { 4397 ids.add(sessionId, true); 4398 } 4399 } 4400 return ids; 4401} 4402 4403AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4404{ 4405 Mutex::Autolock _l(mLock); 4406 AudioStreamIn *input = mInput; 4407 mInput = NULL; 4408 return input; 4409} 4410 4411// this method must always be called either with ThreadBase mLock held or inside the thread loop 4412audio_stream_t* AudioFlinger::RecordThread::stream() const 4413{ 4414 if (mInput == NULL) { 4415 return NULL; 4416 } 4417 return &mInput->stream->common; 4418} 4419 4420status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 4421{ 4422 // only one chain per input thread 4423 if (mEffectChains.size() != 0) { 4424 return INVALID_OPERATION; 4425 } 4426 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 4427 4428 chain->setInBuffer(NULL); 4429 chain->setOutBuffer(NULL); 4430 4431 checkSuspendOnAddEffectChain_l(chain); 4432 4433 mEffectChains.add(chain); 4434 4435 return NO_ERROR; 4436} 4437 4438size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 4439{ 4440 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 4441 ALOGW_IF(mEffectChains.size() != 1, 4442 "removeEffectChain_l() %p invalid chain size %d on thread %p", 4443 chain.get(), mEffectChains.size(), this); 4444 if (mEffectChains.size() == 1) { 4445 mEffectChains.removeAt(0); 4446 } 4447 return 0; 4448} 4449 4450}; // namespace android 4451