Threads.cpp revision f777331418a86cd9fd709af898ef24a69967aeb4
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal mix buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalMixBufferSizeMs = 20; 109// maximum normal mix buffer size 110static const uint32_t kMaxNormalMixBufferSizeMs = 24; 111 112// Whether to use fast mixer 113static const enum { 114 FastMixer_Never, // never initialize or use: for debugging only 115 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 116 // normal mixer multiplier is 1 117 FastMixer_Static, // initialize if needed, then use all the time if initialized, 118 // multiplier is calculated based on min & max normal mixer buffer size 119 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 120 // multiplier is calculated based on min & max normal mixer buffer size 121 // FIXME for FastMixer_Dynamic: 122 // Supporting this option will require fixing HALs that can't handle large writes. 123 // For example, one HAL implementation returns an error from a large write, 124 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 125 // We could either fix the HAL implementations, or provide a wrapper that breaks 126 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 127} kUseFastMixer = FastMixer_Static; 128 129// Priorities for requestPriority 130static const int kPriorityAudioApp = 2; 131static const int kPriorityFastMixer = 3; 132 133// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 134// for the track. The client then sub-divides this into smaller buffers for its use. 135// Currently the client uses double-buffering by default, but doesn't tell us about that. 136// So for now we just assume that client is double-buffered. 137// FIXME It would be better for client to tell AudioFlinger whether it wants double-buffering or 138// N-buffering, so AudioFlinger could allocate the right amount of memory. 139// See the client's minBufCount and mNotificationFramesAct calculations for details. 140static const int kFastTrackMultiplier = 1; 141 142// ---------------------------------------------------------------------------- 143 144#ifdef ADD_BATTERY_DATA 145// To collect the amplifier usage 146static void addBatteryData(uint32_t params) { 147 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 148 if (service == NULL) { 149 // it already logged 150 return; 151 } 152 153 service->addBatteryData(params); 154} 155#endif 156 157 158// ---------------------------------------------------------------------------- 159// CPU Stats 160// ---------------------------------------------------------------------------- 161 162class CpuStats { 163public: 164 CpuStats(); 165 void sample(const String8 &title); 166#ifdef DEBUG_CPU_USAGE 167private: 168 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 169 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 170 171 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 172 173 int mCpuNum; // thread's current CPU number 174 int mCpukHz; // frequency of thread's current CPU in kHz 175#endif 176}; 177 178CpuStats::CpuStats() 179#ifdef DEBUG_CPU_USAGE 180 : mCpuNum(-1), mCpukHz(-1) 181#endif 182{ 183} 184 185void CpuStats::sample(const String8 &title) { 186#ifdef DEBUG_CPU_USAGE 187 // get current thread's delta CPU time in wall clock ns 188 double wcNs; 189 bool valid = mCpuUsage.sampleAndEnable(wcNs); 190 191 // record sample for wall clock statistics 192 if (valid) { 193 mWcStats.sample(wcNs); 194 } 195 196 // get the current CPU number 197 int cpuNum = sched_getcpu(); 198 199 // get the current CPU frequency in kHz 200 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 201 202 // check if either CPU number or frequency changed 203 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 204 mCpuNum = cpuNum; 205 mCpukHz = cpukHz; 206 // ignore sample for purposes of cycles 207 valid = false; 208 } 209 210 // if no change in CPU number or frequency, then record sample for cycle statistics 211 if (valid && mCpukHz > 0) { 212 double cycles = wcNs * cpukHz * 0.000001; 213 mHzStats.sample(cycles); 214 } 215 216 unsigned n = mWcStats.n(); 217 // mCpuUsage.elapsed() is expensive, so don't call it every loop 218 if ((n & 127) == 1) { 219 long long elapsed = mCpuUsage.elapsed(); 220 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 221 double perLoop = elapsed / (double) n; 222 double perLoop100 = perLoop * 0.01; 223 double perLoop1k = perLoop * 0.001; 224 double mean = mWcStats.mean(); 225 double stddev = mWcStats.stddev(); 226 double minimum = mWcStats.minimum(); 227 double maximum = mWcStats.maximum(); 228 double meanCycles = mHzStats.mean(); 229 double stddevCycles = mHzStats.stddev(); 230 double minCycles = mHzStats.minimum(); 231 double maxCycles = mHzStats.maximum(); 232 mCpuUsage.resetElapsed(); 233 mWcStats.reset(); 234 mHzStats.reset(); 235 ALOGD("CPU usage for %s over past %.1f secs\n" 236 " (%u mixer loops at %.1f mean ms per loop):\n" 237 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 238 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 239 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 240 title.string(), 241 elapsed * .000000001, n, perLoop * .000001, 242 mean * .001, 243 stddev * .001, 244 minimum * .001, 245 maximum * .001, 246 mean / perLoop100, 247 stddev / perLoop100, 248 minimum / perLoop100, 249 maximum / perLoop100, 250 meanCycles / perLoop1k, 251 stddevCycles / perLoop1k, 252 minCycles / perLoop1k, 253 maxCycles / perLoop1k); 254 255 } 256 } 257#endif 258}; 259 260// ---------------------------------------------------------------------------- 261// ThreadBase 262// ---------------------------------------------------------------------------- 263 264AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 265 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 266 : Thread(false /*canCallJava*/), 267 mType(type), 268 mAudioFlinger(audioFlinger), 269 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 270 // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() 271 mParamStatus(NO_ERROR), 272 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 273 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 274 // mName will be set by concrete (non-virtual) subclass 275 mDeathRecipient(new PMDeathRecipient(this)) 276{ 277} 278 279AudioFlinger::ThreadBase::~ThreadBase() 280{ 281 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 282 for (size_t i = 0; i < mConfigEvents.size(); i++) { 283 delete mConfigEvents[i]; 284 } 285 mConfigEvents.clear(); 286 287 mParamCond.broadcast(); 288 // do not lock the mutex in destructor 289 releaseWakeLock_l(); 290 if (mPowerManager != 0) { 291 sp<IBinder> binder = mPowerManager->asBinder(); 292 binder->unlinkToDeath(mDeathRecipient); 293 } 294} 295 296status_t AudioFlinger::ThreadBase::readyToRun() 297{ 298 status_t status = initCheck(); 299 if (status == NO_ERROR) { 300 ALOGI("AudioFlinger's thread %p ready to run", this); 301 } else { 302 ALOGE("No working audio driver found."); 303 } 304 return status; 305} 306 307void AudioFlinger::ThreadBase::exit() 308{ 309 ALOGV("ThreadBase::exit"); 310 // do any cleanup required for exit to succeed 311 preExit(); 312 { 313 // This lock prevents the following race in thread (uniprocessor for illustration): 314 // if (!exitPending()) { 315 // // context switch from here to exit() 316 // // exit() calls requestExit(), what exitPending() observes 317 // // exit() calls signal(), which is dropped since no waiters 318 // // context switch back from exit() to here 319 // mWaitWorkCV.wait(...); 320 // // now thread is hung 321 // } 322 AutoMutex lock(mLock); 323 requestExit(); 324 mWaitWorkCV.broadcast(); 325 } 326 // When Thread::requestExitAndWait is made virtual and this method is renamed to 327 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 328 requestExitAndWait(); 329} 330 331status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 332{ 333 status_t status; 334 335 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 336 Mutex::Autolock _l(mLock); 337 338 mNewParameters.add(keyValuePairs); 339 mWaitWorkCV.signal(); 340 // wait condition with timeout in case the thread loop has exited 341 // before the request could be processed 342 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 343 status = mParamStatus; 344 mWaitWorkCV.signal(); 345 } else { 346 status = TIMED_OUT; 347 } 348 return status; 349} 350 351void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 352{ 353 Mutex::Autolock _l(mLock); 354 sendIoConfigEvent_l(event, param); 355} 356 357// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 358void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 359{ 360 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 361 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 362 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 363 param); 364 mWaitWorkCV.signal(); 365} 366 367// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 368void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 369{ 370 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 371 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 372 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 373 mConfigEvents.size(), pid, tid, prio); 374 mWaitWorkCV.signal(); 375} 376 377void AudioFlinger::ThreadBase::processConfigEvents() 378{ 379 Mutex::Autolock _l(mLock); 380 processConfigEvents_l(); 381} 382 383void AudioFlinger::ThreadBase::processConfigEvents_l() 384{ 385 while (!mConfigEvents.isEmpty()) { 386 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 387 ConfigEvent *event = mConfigEvents[0]; 388 mConfigEvents.removeAt(0); 389 // release mLock before locking AudioFlinger mLock: lock order is always 390 // AudioFlinger then ThreadBase to avoid cross deadlock 391 mLock.unlock(); 392 switch (event->type()) { 393 case CFG_EVENT_PRIO: { 394 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 395 // FIXME Need to understand why this has be done asynchronously 396 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 397 true /*asynchronous*/); 398 if (err != 0) { 399 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 400 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 401 } 402 } break; 403 case CFG_EVENT_IO: { 404 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 405 mAudioFlinger->mLock.lock(); 406 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 407 mAudioFlinger->mLock.unlock(); 408 } break; 409 default: 410 ALOGE("processConfigEvents() unknown event type %d", event->type()); 411 break; 412 } 413 delete event; 414 mLock.lock(); 415 } 416} 417 418void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 419{ 420 const size_t SIZE = 256; 421 char buffer[SIZE]; 422 String8 result; 423 424 bool locked = AudioFlinger::dumpTryLock(mLock); 425 if (!locked) { 426 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 427 write(fd, buffer, strlen(buffer)); 428 } 429 430 snprintf(buffer, SIZE, "io handle: %d\n", mId); 431 result.append(buffer); 432 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 433 result.append(buffer); 434 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 435 result.append(buffer); 436 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 437 result.append(buffer); 438 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 439 result.append(buffer); 440 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 441 result.append(buffer); 442 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 443 result.append(buffer); 444 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 445 result.append(buffer); 446 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 447 result.append(buffer); 448 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 449 result.append(buffer); 450 451 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 452 result.append(buffer); 453 result.append(" Index Command"); 454 for (size_t i = 0; i < mNewParameters.size(); ++i) { 455 snprintf(buffer, SIZE, "\n %02d ", i); 456 result.append(buffer); 457 result.append(mNewParameters[i]); 458 } 459 460 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 461 result.append(buffer); 462 for (size_t i = 0; i < mConfigEvents.size(); i++) { 463 mConfigEvents[i]->dump(buffer, SIZE); 464 result.append(buffer); 465 } 466 result.append("\n"); 467 468 write(fd, result.string(), result.size()); 469 470 if (locked) { 471 mLock.unlock(); 472 } 473} 474 475void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 476{ 477 const size_t SIZE = 256; 478 char buffer[SIZE]; 479 String8 result; 480 481 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 482 write(fd, buffer, strlen(buffer)); 483 484 for (size_t i = 0; i < mEffectChains.size(); ++i) { 485 sp<EffectChain> chain = mEffectChains[i]; 486 if (chain != 0) { 487 chain->dump(fd, args); 488 } 489 } 490} 491 492void AudioFlinger::ThreadBase::acquireWakeLock() 493{ 494 Mutex::Autolock _l(mLock); 495 acquireWakeLock_l(); 496} 497 498void AudioFlinger::ThreadBase::acquireWakeLock_l() 499{ 500 if (mPowerManager == 0) { 501 // use checkService() to avoid blocking if power service is not up yet 502 sp<IBinder> binder = 503 defaultServiceManager()->checkService(String16("power")); 504 if (binder == 0) { 505 ALOGW("Thread %s cannot connect to the power manager service", mName); 506 } else { 507 mPowerManager = interface_cast<IPowerManager>(binder); 508 binder->linkToDeath(mDeathRecipient); 509 } 510 } 511 if (mPowerManager != 0) { 512 sp<IBinder> binder = new BBinder(); 513 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 514 binder, 515 String16(mName), 516 String16("media")); 517 if (status == NO_ERROR) { 518 mWakeLockToken = binder; 519 } 520 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 521 } 522} 523 524void AudioFlinger::ThreadBase::releaseWakeLock() 525{ 526 Mutex::Autolock _l(mLock); 527 releaseWakeLock_l(); 528} 529 530void AudioFlinger::ThreadBase::releaseWakeLock_l() 531{ 532 if (mWakeLockToken != 0) { 533 ALOGV("releaseWakeLock_l() %s", mName); 534 if (mPowerManager != 0) { 535 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 536 } 537 mWakeLockToken.clear(); 538 } 539} 540 541void AudioFlinger::ThreadBase::clearPowerManager() 542{ 543 Mutex::Autolock _l(mLock); 544 releaseWakeLock_l(); 545 mPowerManager.clear(); 546} 547 548void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 549{ 550 sp<ThreadBase> thread = mThread.promote(); 551 if (thread != 0) { 552 thread->clearPowerManager(); 553 } 554 ALOGW("power manager service died !!!"); 555} 556 557void AudioFlinger::ThreadBase::setEffectSuspended( 558 const effect_uuid_t *type, bool suspend, int sessionId) 559{ 560 Mutex::Autolock _l(mLock); 561 setEffectSuspended_l(type, suspend, sessionId); 562} 563 564void AudioFlinger::ThreadBase::setEffectSuspended_l( 565 const effect_uuid_t *type, bool suspend, int sessionId) 566{ 567 sp<EffectChain> chain = getEffectChain_l(sessionId); 568 if (chain != 0) { 569 if (type != NULL) { 570 chain->setEffectSuspended_l(type, suspend); 571 } else { 572 chain->setEffectSuspendedAll_l(suspend); 573 } 574 } 575 576 updateSuspendedSessions_l(type, suspend, sessionId); 577} 578 579void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 580{ 581 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 582 if (index < 0) { 583 return; 584 } 585 586 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 587 mSuspendedSessions.valueAt(index); 588 589 for (size_t i = 0; i < sessionEffects.size(); i++) { 590 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 591 for (int j = 0; j < desc->mRefCount; j++) { 592 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 593 chain->setEffectSuspendedAll_l(true); 594 } else { 595 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 596 desc->mType.timeLow); 597 chain->setEffectSuspended_l(&desc->mType, true); 598 } 599 } 600 } 601} 602 603void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 604 bool suspend, 605 int sessionId) 606{ 607 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 608 609 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 610 611 if (suspend) { 612 if (index >= 0) { 613 sessionEffects = mSuspendedSessions.valueAt(index); 614 } else { 615 mSuspendedSessions.add(sessionId, sessionEffects); 616 } 617 } else { 618 if (index < 0) { 619 return; 620 } 621 sessionEffects = mSuspendedSessions.valueAt(index); 622 } 623 624 625 int key = EffectChain::kKeyForSuspendAll; 626 if (type != NULL) { 627 key = type->timeLow; 628 } 629 index = sessionEffects.indexOfKey(key); 630 631 sp<SuspendedSessionDesc> desc; 632 if (suspend) { 633 if (index >= 0) { 634 desc = sessionEffects.valueAt(index); 635 } else { 636 desc = new SuspendedSessionDesc(); 637 if (type != NULL) { 638 desc->mType = *type; 639 } 640 sessionEffects.add(key, desc); 641 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 642 } 643 desc->mRefCount++; 644 } else { 645 if (index < 0) { 646 return; 647 } 648 desc = sessionEffects.valueAt(index); 649 if (--desc->mRefCount == 0) { 650 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 651 sessionEffects.removeItemsAt(index); 652 if (sessionEffects.isEmpty()) { 653 ALOGV("updateSuspendedSessions_l() restore removing session %d", 654 sessionId); 655 mSuspendedSessions.removeItem(sessionId); 656 } 657 } 658 } 659 if (!sessionEffects.isEmpty()) { 660 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 661 } 662} 663 664void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 665 bool enabled, 666 int sessionId) 667{ 668 Mutex::Autolock _l(mLock); 669 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 670} 671 672void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 673 bool enabled, 674 int sessionId) 675{ 676 if (mType != RECORD) { 677 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 678 // another session. This gives the priority to well behaved effect control panels 679 // and applications not using global effects. 680 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 681 // global effects 682 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 683 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 684 } 685 } 686 687 sp<EffectChain> chain = getEffectChain_l(sessionId); 688 if (chain != 0) { 689 chain->checkSuspendOnEffectEnabled(effect, enabled); 690 } 691} 692 693// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 694sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 695 const sp<AudioFlinger::Client>& client, 696 const sp<IEffectClient>& effectClient, 697 int32_t priority, 698 int sessionId, 699 effect_descriptor_t *desc, 700 int *enabled, 701 status_t *status) 702{ 703 sp<EffectModule> effect; 704 sp<EffectHandle> handle; 705 status_t lStatus; 706 sp<EffectChain> chain; 707 bool chainCreated = false; 708 bool effectCreated = false; 709 bool effectRegistered = false; 710 711 lStatus = initCheck(); 712 if (lStatus != NO_ERROR) { 713 ALOGW("createEffect_l() Audio driver not initialized."); 714 goto Exit; 715 } 716 717 // Do not allow effects with session ID 0 on direct output or duplicating threads 718 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 719 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 720 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 721 desc->name, sessionId); 722 lStatus = BAD_VALUE; 723 goto Exit; 724 } 725 // Only Pre processor effects are allowed on input threads and only on input threads 726 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 727 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 728 desc->name, desc->flags, mType); 729 lStatus = BAD_VALUE; 730 goto Exit; 731 } 732 733 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 734 735 { // scope for mLock 736 Mutex::Autolock _l(mLock); 737 738 // check for existing effect chain with the requested audio session 739 chain = getEffectChain_l(sessionId); 740 if (chain == 0) { 741 // create a new chain for this session 742 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 743 chain = new EffectChain(this, sessionId); 744 addEffectChain_l(chain); 745 chain->setStrategy(getStrategyForSession_l(sessionId)); 746 chainCreated = true; 747 } else { 748 effect = chain->getEffectFromDesc_l(desc); 749 } 750 751 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 752 753 if (effect == 0) { 754 int id = mAudioFlinger->nextUniqueId(); 755 // Check CPU and memory usage 756 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 757 if (lStatus != NO_ERROR) { 758 goto Exit; 759 } 760 effectRegistered = true; 761 // create a new effect module if none present in the chain 762 effect = new EffectModule(this, chain, desc, id, sessionId); 763 lStatus = effect->status(); 764 if (lStatus != NO_ERROR) { 765 goto Exit; 766 } 767 lStatus = chain->addEffect_l(effect); 768 if (lStatus != NO_ERROR) { 769 goto Exit; 770 } 771 effectCreated = true; 772 773 effect->setDevice(mOutDevice); 774 effect->setDevice(mInDevice); 775 effect->setMode(mAudioFlinger->getMode()); 776 effect->setAudioSource(mAudioSource); 777 } 778 // create effect handle and connect it to effect module 779 handle = new EffectHandle(effect, client, effectClient, priority); 780 lStatus = effect->addHandle(handle.get()); 781 if (enabled != NULL) { 782 *enabled = (int)effect->isEnabled(); 783 } 784 } 785 786Exit: 787 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 788 Mutex::Autolock _l(mLock); 789 if (effectCreated) { 790 chain->removeEffect_l(effect); 791 } 792 if (effectRegistered) { 793 AudioSystem::unregisterEffect(effect->id()); 794 } 795 if (chainCreated) { 796 removeEffectChain_l(chain); 797 } 798 handle.clear(); 799 } 800 801 *status = lStatus; 802 return handle; 803} 804 805sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 806{ 807 Mutex::Autolock _l(mLock); 808 return getEffect_l(sessionId, effectId); 809} 810 811sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 812{ 813 sp<EffectChain> chain = getEffectChain_l(sessionId); 814 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 815} 816 817// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 818// PlaybackThread::mLock held 819status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 820{ 821 // check for existing effect chain with the requested audio session 822 int sessionId = effect->sessionId(); 823 sp<EffectChain> chain = getEffectChain_l(sessionId); 824 bool chainCreated = false; 825 826 if (chain == 0) { 827 // create a new chain for this session 828 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 829 chain = new EffectChain(this, sessionId); 830 addEffectChain_l(chain); 831 chain->setStrategy(getStrategyForSession_l(sessionId)); 832 chainCreated = true; 833 } 834 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 835 836 if (chain->getEffectFromId_l(effect->id()) != 0) { 837 ALOGW("addEffect_l() %p effect %s already present in chain %p", 838 this, effect->desc().name, chain.get()); 839 return BAD_VALUE; 840 } 841 842 status_t status = chain->addEffect_l(effect); 843 if (status != NO_ERROR) { 844 if (chainCreated) { 845 removeEffectChain_l(chain); 846 } 847 return status; 848 } 849 850 effect->setDevice(mOutDevice); 851 effect->setDevice(mInDevice); 852 effect->setMode(mAudioFlinger->getMode()); 853 effect->setAudioSource(mAudioSource); 854 return NO_ERROR; 855} 856 857void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 858 859 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 860 effect_descriptor_t desc = effect->desc(); 861 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 862 detachAuxEffect_l(effect->id()); 863 } 864 865 sp<EffectChain> chain = effect->chain().promote(); 866 if (chain != 0) { 867 // remove effect chain if removing last effect 868 if (chain->removeEffect_l(effect) == 0) { 869 removeEffectChain_l(chain); 870 } 871 } else { 872 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 873 } 874} 875 876void AudioFlinger::ThreadBase::lockEffectChains_l( 877 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 878{ 879 effectChains = mEffectChains; 880 for (size_t i = 0; i < mEffectChains.size(); i++) { 881 mEffectChains[i]->lock(); 882 } 883} 884 885void AudioFlinger::ThreadBase::unlockEffectChains( 886 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 887{ 888 for (size_t i = 0; i < effectChains.size(); i++) { 889 effectChains[i]->unlock(); 890 } 891} 892 893sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 894{ 895 Mutex::Autolock _l(mLock); 896 return getEffectChain_l(sessionId); 897} 898 899sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 900{ 901 size_t size = mEffectChains.size(); 902 for (size_t i = 0; i < size; i++) { 903 if (mEffectChains[i]->sessionId() == sessionId) { 904 return mEffectChains[i]; 905 } 906 } 907 return 0; 908} 909 910void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 911{ 912 Mutex::Autolock _l(mLock); 913 size_t size = mEffectChains.size(); 914 for (size_t i = 0; i < size; i++) { 915 mEffectChains[i]->setMode_l(mode); 916 } 917} 918 919void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 920 EffectHandle *handle, 921 bool unpinIfLast) { 922 923 Mutex::Autolock _l(mLock); 924 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 925 // delete the effect module if removing last handle on it 926 if (effect->removeHandle(handle) == 0) { 927 if (!effect->isPinned() || unpinIfLast) { 928 removeEffect_l(effect); 929 AudioSystem::unregisterEffect(effect->id()); 930 } 931 } 932} 933 934// ---------------------------------------------------------------------------- 935// Playback 936// ---------------------------------------------------------------------------- 937 938AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 939 AudioStreamOut* output, 940 audio_io_handle_t id, 941 audio_devices_t device, 942 type_t type) 943 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 944 mNormalFrameCount(0), mMixBuffer(NULL), 945 mSuspended(0), mBytesWritten(0), 946 // mStreamTypes[] initialized in constructor body 947 mOutput(output), 948 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 949 mMixerStatus(MIXER_IDLE), 950 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 951 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 952 mBytesRemaining(0), 953 mCurrentWriteLength(0), 954 mUseAsyncWrite(false), 955 mWriteBlocked(false), 956 mDraining(false), 957 mScreenState(AudioFlinger::mScreenState), 958 // index 0 is reserved for normal mixer's submix 959 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 960{ 961 snprintf(mName, kNameLength, "AudioOut_%X", id); 962 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 963 964 // Assumes constructor is called by AudioFlinger with it's mLock held, but 965 // it would be safer to explicitly pass initial masterVolume/masterMute as 966 // parameter. 967 // 968 // If the HAL we are using has support for master volume or master mute, 969 // then do not attenuate or mute during mixing (just leave the volume at 1.0 970 // and the mute set to false). 971 mMasterVolume = audioFlinger->masterVolume_l(); 972 mMasterMute = audioFlinger->masterMute_l(); 973 if (mOutput && mOutput->audioHwDev) { 974 if (mOutput->audioHwDev->canSetMasterVolume()) { 975 mMasterVolume = 1.0; 976 } 977 978 if (mOutput->audioHwDev->canSetMasterMute()) { 979 mMasterMute = false; 980 } 981 } 982 983 readOutputParameters(); 984 985 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 986 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 987 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 988 stream = (audio_stream_type_t) (stream + 1)) { 989 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 990 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 991 } 992 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 993 // because mAudioFlinger doesn't have one to copy from 994} 995 996AudioFlinger::PlaybackThread::~PlaybackThread() 997{ 998 mAudioFlinger->unregisterWriter(mNBLogWriter); 999 delete[] mMixBuffer; 1000} 1001 1002void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1003{ 1004 dumpInternals(fd, args); 1005 dumpTracks(fd, args); 1006 dumpEffectChains(fd, args); 1007} 1008 1009void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1010{ 1011 const size_t SIZE = 256; 1012 char buffer[SIZE]; 1013 String8 result; 1014 1015 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1016 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1017 const stream_type_t *st = &mStreamTypes[i]; 1018 if (i > 0) { 1019 result.appendFormat(", "); 1020 } 1021 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1022 if (st->mute) { 1023 result.append("M"); 1024 } 1025 } 1026 result.append("\n"); 1027 write(fd, result.string(), result.length()); 1028 result.clear(); 1029 1030 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1031 result.append(buffer); 1032 Track::appendDumpHeader(result); 1033 for (size_t i = 0; i < mTracks.size(); ++i) { 1034 sp<Track> track = mTracks[i]; 1035 if (track != 0) { 1036 track->dump(buffer, SIZE); 1037 result.append(buffer); 1038 } 1039 } 1040 1041 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1042 result.append(buffer); 1043 Track::appendDumpHeader(result); 1044 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1045 sp<Track> track = mActiveTracks[i].promote(); 1046 if (track != 0) { 1047 track->dump(buffer, SIZE); 1048 result.append(buffer); 1049 } 1050 } 1051 write(fd, result.string(), result.size()); 1052 1053 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1054 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1055 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1056 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1057} 1058 1059void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1060{ 1061 const size_t SIZE = 256; 1062 char buffer[SIZE]; 1063 String8 result; 1064 1065 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1066 result.append(buffer); 1067 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1068 result.append(buffer); 1069 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1070 ns2ms(systemTime() - mLastWriteTime)); 1071 result.append(buffer); 1072 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1073 result.append(buffer); 1074 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1079 result.append(buffer); 1080 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1081 result.append(buffer); 1082 write(fd, result.string(), result.size()); 1083 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1084 1085 dumpBase(fd, args); 1086} 1087 1088// Thread virtuals 1089 1090void AudioFlinger::PlaybackThread::onFirstRef() 1091{ 1092 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1093} 1094 1095// ThreadBase virtuals 1096void AudioFlinger::PlaybackThread::preExit() 1097{ 1098 ALOGV(" preExit()"); 1099 // FIXME this is using hard-coded strings but in the future, this functionality will be 1100 // converted to use audio HAL extensions required to support tunneling 1101 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1102} 1103 1104// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1105sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1106 const sp<AudioFlinger::Client>& client, 1107 audio_stream_type_t streamType, 1108 uint32_t sampleRate, 1109 audio_format_t format, 1110 audio_channel_mask_t channelMask, 1111 size_t frameCount, 1112 const sp<IMemory>& sharedBuffer, 1113 int sessionId, 1114 IAudioFlinger::track_flags_t *flags, 1115 pid_t tid, 1116 status_t *status) 1117{ 1118 sp<Track> track; 1119 status_t lStatus; 1120 1121 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1122 1123 // client expresses a preference for FAST, but we get the final say 1124 if (*flags & IAudioFlinger::TRACK_FAST) { 1125 if ( 1126 // not timed 1127 (!isTimed) && 1128 // either of these use cases: 1129 ( 1130 // use case 1: shared buffer with any frame count 1131 ( 1132 (sharedBuffer != 0) 1133 ) || 1134 // use case 2: callback handler and frame count is default or at least as large as HAL 1135 ( 1136 (tid != -1) && 1137 ((frameCount == 0) || 1138 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1139 ) 1140 ) && 1141 // PCM data 1142 audio_is_linear_pcm(format) && 1143 // mono or stereo 1144 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1145 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1146#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1147 // hardware sample rate 1148 (sampleRate == mSampleRate) && 1149#endif 1150 // normal mixer has an associated fast mixer 1151 hasFastMixer() && 1152 // there are sufficient fast track slots available 1153 (mFastTrackAvailMask != 0) 1154 // FIXME test that MixerThread for this fast track has a capable output HAL 1155 // FIXME add a permission test also? 1156 ) { 1157 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1158 if (frameCount == 0) { 1159 frameCount = mFrameCount * kFastTrackMultiplier; 1160 } 1161 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1162 frameCount, mFrameCount); 1163 } else { 1164 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1165 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1166 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1167 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1168 audio_is_linear_pcm(format), 1169 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1170 *flags &= ~IAudioFlinger::TRACK_FAST; 1171 // For compatibility with AudioTrack calculation, buffer depth is forced 1172 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1173 // This is probably too conservative, but legacy application code may depend on it. 1174 // If you change this calculation, also review the start threshold which is related. 1175 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1176 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1177 if (minBufCount < 2) { 1178 minBufCount = 2; 1179 } 1180 size_t minFrameCount = mNormalFrameCount * minBufCount; 1181 if (frameCount < minFrameCount) { 1182 frameCount = minFrameCount; 1183 } 1184 } 1185 } 1186 1187 if (mType == DIRECT) { 1188 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1189 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1190 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1191 "for output %p with format %d", 1192 sampleRate, format, channelMask, mOutput, mFormat); 1193 lStatus = BAD_VALUE; 1194 goto Exit; 1195 } 1196 } 1197 } else if (mType == OFFLOAD) { 1198 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1199 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1200 "for output %p with format %d", 1201 sampleRate, format, channelMask, mOutput, mFormat); 1202 lStatus = BAD_VALUE; 1203 goto Exit; 1204 } 1205 } else { 1206 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1207 ALOGE("createTrack_l() Bad parameter: format %d \"" 1208 "for output %p with format %d", 1209 format, mOutput, mFormat); 1210 lStatus = BAD_VALUE; 1211 goto Exit; 1212 } 1213 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1214 if (sampleRate > mSampleRate*2) { 1215 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1216 lStatus = BAD_VALUE; 1217 goto Exit; 1218 } 1219 } 1220 1221 lStatus = initCheck(); 1222 if (lStatus != NO_ERROR) { 1223 ALOGE("Audio driver not initialized."); 1224 goto Exit; 1225 } 1226 1227 { // scope for mLock 1228 Mutex::Autolock _l(mLock); 1229 1230 // all tracks in same audio session must share the same routing strategy otherwise 1231 // conflicts will happen when tracks are moved from one output to another by audio policy 1232 // manager 1233 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1234 for (size_t i = 0; i < mTracks.size(); ++i) { 1235 sp<Track> t = mTracks[i]; 1236 if (t != 0 && !t->isOutputTrack()) { 1237 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1238 if (sessionId == t->sessionId() && strategy != actual) { 1239 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1240 strategy, actual); 1241 lStatus = BAD_VALUE; 1242 goto Exit; 1243 } 1244 } 1245 } 1246 1247 if (!isTimed) { 1248 track = new Track(this, client, streamType, sampleRate, format, 1249 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1250 } else { 1251 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1252 channelMask, frameCount, sharedBuffer, sessionId); 1253 } 1254 1255 // new Track always returns non-NULL, 1256 // but TimedTrack::create() is a factory that could fail by returning NULL 1257 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1258 if (lStatus != NO_ERROR) { 1259 track.clear(); 1260 goto Exit; 1261 } 1262 1263 mTracks.add(track); 1264 1265 sp<EffectChain> chain = getEffectChain_l(sessionId); 1266 if (chain != 0) { 1267 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1268 track->setMainBuffer(chain->inBuffer()); 1269 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1270 chain->incTrackCnt(); 1271 } 1272 1273 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1274 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1275 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1276 // so ask activity manager to do this on our behalf 1277 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1278 } 1279 } 1280 1281 lStatus = NO_ERROR; 1282 1283Exit: 1284 *status = lStatus; 1285 return track; 1286} 1287 1288uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1289{ 1290 return latency; 1291} 1292 1293uint32_t AudioFlinger::PlaybackThread::latency() const 1294{ 1295 Mutex::Autolock _l(mLock); 1296 return latency_l(); 1297} 1298uint32_t AudioFlinger::PlaybackThread::latency_l() const 1299{ 1300 if (initCheck() == NO_ERROR) { 1301 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1302 } else { 1303 return 0; 1304 } 1305} 1306 1307void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1308{ 1309 Mutex::Autolock _l(mLock); 1310 // Don't apply master volume in SW if our HAL can do it for us. 1311 if (mOutput && mOutput->audioHwDev && 1312 mOutput->audioHwDev->canSetMasterVolume()) { 1313 mMasterVolume = 1.0; 1314 } else { 1315 mMasterVolume = value; 1316 } 1317} 1318 1319void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1320{ 1321 Mutex::Autolock _l(mLock); 1322 // Don't apply master mute in SW if our HAL can do it for us. 1323 if (mOutput && mOutput->audioHwDev && 1324 mOutput->audioHwDev->canSetMasterMute()) { 1325 mMasterMute = false; 1326 } else { 1327 mMasterMute = muted; 1328 } 1329} 1330 1331void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1332{ 1333 Mutex::Autolock _l(mLock); 1334 mStreamTypes[stream].volume = value; 1335 signal_l(); 1336} 1337 1338void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1339{ 1340 Mutex::Autolock _l(mLock); 1341 mStreamTypes[stream].mute = muted; 1342 signal_l(); 1343} 1344 1345float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1346{ 1347 Mutex::Autolock _l(mLock); 1348 return mStreamTypes[stream].volume; 1349} 1350 1351// addTrack_l() must be called with ThreadBase::mLock held 1352status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1353{ 1354 status_t status = ALREADY_EXISTS; 1355 1356 // set retry count for buffer fill 1357 track->mRetryCount = kMaxTrackStartupRetries; 1358 if (mActiveTracks.indexOf(track) < 0) { 1359 // the track is newly added, make sure it fills up all its 1360 // buffers before playing. This is to ensure the client will 1361 // effectively get the latency it requested. 1362 if (!track->isOutputTrack()) { 1363 TrackBase::track_state state = track->mState; 1364 mLock.unlock(); 1365 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1366 mLock.lock(); 1367 // abort track was stopped/paused while we released the lock 1368 if (state != track->mState) { 1369 if (status == NO_ERROR) { 1370 mLock.unlock(); 1371 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1372 mLock.lock(); 1373 } 1374 return INVALID_OPERATION; 1375 } 1376 // abort if start is rejected by audio policy manager 1377 if (status != NO_ERROR) { 1378 return PERMISSION_DENIED; 1379 } 1380#ifdef ADD_BATTERY_DATA 1381 // to track the speaker usage 1382 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1383#endif 1384 } 1385 1386 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1387 track->mResetDone = false; 1388 track->mPresentationCompleteFrames = 0; 1389 mActiveTracks.add(track); 1390 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1391 if (chain != 0) { 1392 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1393 track->sessionId()); 1394 chain->incActiveTrackCnt(); 1395 } 1396 1397 status = NO_ERROR; 1398 } 1399 1400 ALOGV("mWaitWorkCV.broadcast"); 1401 mWaitWorkCV.broadcast(); 1402 1403 return status; 1404} 1405 1406bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1407{ 1408 track->terminate(); 1409 // active tracks are removed by threadLoop() 1410 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1411 track->mState = TrackBase::STOPPED; 1412 if (!trackActive) { 1413 removeTrack_l(track); 1414 } else if (track->isFastTrack() || track->isOffloaded()) { 1415 track->mState = TrackBase::STOPPING_1; 1416 } 1417 1418 return trackActive; 1419} 1420 1421void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1422{ 1423 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1424 mTracks.remove(track); 1425 deleteTrackName_l(track->name()); 1426 // redundant as track is about to be destroyed, for dumpsys only 1427 track->mName = -1; 1428 if (track->isFastTrack()) { 1429 int index = track->mFastIndex; 1430 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1431 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1432 mFastTrackAvailMask |= 1 << index; 1433 // redundant as track is about to be destroyed, for dumpsys only 1434 track->mFastIndex = -1; 1435 } 1436 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1437 if (chain != 0) { 1438 chain->decTrackCnt(); 1439 } 1440} 1441 1442void AudioFlinger::PlaybackThread::signal_l() 1443{ 1444 // Thread could be blocked waiting for async 1445 // so signal it to handle state changes immediately 1446 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1447 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1448 mSignalPending = true; 1449 mWaitWorkCV.signal(); 1450} 1451 1452String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1453{ 1454 Mutex::Autolock _l(mLock); 1455 if (initCheck() != NO_ERROR) { 1456 return String8(); 1457 } 1458 1459 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1460 const String8 out_s8(s); 1461 free(s); 1462 return out_s8; 1463} 1464 1465// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1466void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1467 AudioSystem::OutputDescriptor desc; 1468 void *param2 = NULL; 1469 1470 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1471 param); 1472 1473 switch (event) { 1474 case AudioSystem::OUTPUT_OPENED: 1475 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1476 desc.channelMask = mChannelMask; 1477 desc.samplingRate = mSampleRate; 1478 desc.format = mFormat; 1479 desc.frameCount = mNormalFrameCount; // FIXME see 1480 // AudioFlinger::frameCount(audio_io_handle_t) 1481 desc.latency = latency(); 1482 param2 = &desc; 1483 break; 1484 1485 case AudioSystem::STREAM_CONFIG_CHANGED: 1486 param2 = ¶m; 1487 case AudioSystem::OUTPUT_CLOSED: 1488 default: 1489 break; 1490 } 1491 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1492} 1493 1494void AudioFlinger::PlaybackThread::writeCallback() 1495{ 1496 ALOG_ASSERT(mCallbackThread != 0); 1497 mCallbackThread->setWriteBlocked(false); 1498} 1499 1500void AudioFlinger::PlaybackThread::drainCallback() 1501{ 1502 ALOG_ASSERT(mCallbackThread != 0); 1503 mCallbackThread->setDraining(false); 1504} 1505 1506void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1507{ 1508 Mutex::Autolock _l(mLock); 1509 mWriteBlocked = value; 1510 if (!value) { 1511 mWaitWorkCV.signal(); 1512 } 1513} 1514 1515void AudioFlinger::PlaybackThread::setDraining(bool value) 1516{ 1517 Mutex::Autolock _l(mLock); 1518 mDraining = value; 1519 if (!value) { 1520 mWaitWorkCV.signal(); 1521 } 1522} 1523 1524// static 1525int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1526 void *param, 1527 void *cookie) 1528{ 1529 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1530 ALOGV("asyncCallback() event %d", event); 1531 switch (event) { 1532 case STREAM_CBK_EVENT_WRITE_READY: 1533 me->writeCallback(); 1534 break; 1535 case STREAM_CBK_EVENT_DRAIN_READY: 1536 me->drainCallback(); 1537 break; 1538 default: 1539 ALOGW("asyncCallback() unknown event %d", event); 1540 break; 1541 } 1542 return 0; 1543} 1544 1545void AudioFlinger::PlaybackThread::readOutputParameters() 1546{ 1547 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1548 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1549 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1550 if (!audio_is_output_channel(mChannelMask)) { 1551 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1552 } 1553 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1554 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1555 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1556 } 1557 mChannelCount = popcount(mChannelMask); 1558 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1559 if (!audio_is_valid_format(mFormat)) { 1560 LOG_FATAL("HAL format %d not valid for output", mFormat); 1561 } 1562 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1563 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1564 mFormat); 1565 } 1566 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1567 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1568 mFrameCount = mBufferSize / mFrameSize; 1569 if (mFrameCount & 15) { 1570 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1571 mFrameCount); 1572 } 1573 1574 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1575 (mOutput->stream->set_callback != NULL)) { 1576 if (mOutput->stream->set_callback(mOutput->stream, 1577 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1578 mUseAsyncWrite = true; 1579 } 1580 } 1581 1582 // Calculate size of normal mix buffer relative to the HAL output buffer size 1583 double multiplier = 1.0; 1584 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1585 kUseFastMixer == FastMixer_Dynamic)) { 1586 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1587 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1588 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1589 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1590 maxNormalFrameCount = maxNormalFrameCount & ~15; 1591 if (maxNormalFrameCount < minNormalFrameCount) { 1592 maxNormalFrameCount = minNormalFrameCount; 1593 } 1594 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1595 if (multiplier <= 1.0) { 1596 multiplier = 1.0; 1597 } else if (multiplier <= 2.0) { 1598 if (2 * mFrameCount <= maxNormalFrameCount) { 1599 multiplier = 2.0; 1600 } else { 1601 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1602 } 1603 } else { 1604 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1605 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1606 // track, but we sometimes have to do this to satisfy the maximum frame count 1607 // constraint) 1608 // FIXME this rounding up should not be done if no HAL SRC 1609 uint32_t truncMult = (uint32_t) multiplier; 1610 if ((truncMult & 1)) { 1611 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1612 ++truncMult; 1613 } 1614 } 1615 multiplier = (double) truncMult; 1616 } 1617 } 1618 mNormalFrameCount = multiplier * mFrameCount; 1619 // round up to nearest 16 frames to satisfy AudioMixer 1620 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1621 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1622 mNormalFrameCount); 1623 1624 delete[] mMixBuffer; 1625 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1626 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1627 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1628 memset(mMixBuffer, 0, normalBufferSize); 1629 1630 // force reconfiguration of effect chains and engines to take new buffer size and audio 1631 // parameters into account 1632 // Note that mLock is not held when readOutputParameters() is called from the constructor 1633 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1634 // matter. 1635 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1636 Vector< sp<EffectChain> > effectChains = mEffectChains; 1637 for (size_t i = 0; i < effectChains.size(); i ++) { 1638 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1639 } 1640} 1641 1642 1643status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1644{ 1645 if (halFrames == NULL || dspFrames == NULL) { 1646 return BAD_VALUE; 1647 } 1648 Mutex::Autolock _l(mLock); 1649 if (initCheck() != NO_ERROR) { 1650 return INVALID_OPERATION; 1651 } 1652 size_t framesWritten = mBytesWritten / mFrameSize; 1653 *halFrames = framesWritten; 1654 1655 if (isSuspended()) { 1656 // return an estimation of rendered frames when the output is suspended 1657 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1658 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1659 return NO_ERROR; 1660 } else { 1661 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1662 } 1663} 1664 1665uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1666{ 1667 Mutex::Autolock _l(mLock); 1668 uint32_t result = 0; 1669 if (getEffectChain_l(sessionId) != 0) { 1670 result = EFFECT_SESSION; 1671 } 1672 1673 for (size_t i = 0; i < mTracks.size(); ++i) { 1674 sp<Track> track = mTracks[i]; 1675 if (sessionId == track->sessionId() && !track->isInvalid()) { 1676 result |= TRACK_SESSION; 1677 break; 1678 } 1679 } 1680 1681 return result; 1682} 1683 1684uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1685{ 1686 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1687 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1688 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1689 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1690 } 1691 for (size_t i = 0; i < mTracks.size(); i++) { 1692 sp<Track> track = mTracks[i]; 1693 if (sessionId == track->sessionId() && !track->isInvalid()) { 1694 return AudioSystem::getStrategyForStream(track->streamType()); 1695 } 1696 } 1697 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1698} 1699 1700 1701AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1702{ 1703 Mutex::Autolock _l(mLock); 1704 return mOutput; 1705} 1706 1707AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1708{ 1709 Mutex::Autolock _l(mLock); 1710 AudioStreamOut *output = mOutput; 1711 mOutput = NULL; 1712 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1713 // must push a NULL and wait for ack 1714 mOutputSink.clear(); 1715 mPipeSink.clear(); 1716 mNormalSink.clear(); 1717 return output; 1718} 1719 1720// this method must always be called either with ThreadBase mLock held or inside the thread loop 1721audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1722{ 1723 if (mOutput == NULL) { 1724 return NULL; 1725 } 1726 return &mOutput->stream->common; 1727} 1728 1729uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1730{ 1731 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1732} 1733 1734status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1735{ 1736 if (!isValidSyncEvent(event)) { 1737 return BAD_VALUE; 1738 } 1739 1740 Mutex::Autolock _l(mLock); 1741 1742 for (size_t i = 0; i < mTracks.size(); ++i) { 1743 sp<Track> track = mTracks[i]; 1744 if (event->triggerSession() == track->sessionId()) { 1745 (void) track->setSyncEvent(event); 1746 return NO_ERROR; 1747 } 1748 } 1749 1750 return NAME_NOT_FOUND; 1751} 1752 1753bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1754{ 1755 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1756} 1757 1758void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1759 const Vector< sp<Track> >& tracksToRemove) 1760{ 1761 size_t count = tracksToRemove.size(); 1762 if (count > 0) { 1763 for (size_t i = 0 ; i < count ; i++) { 1764 const sp<Track>& track = tracksToRemove.itemAt(i); 1765 if (!track->isOutputTrack()) { 1766 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1767#ifdef ADD_BATTERY_DATA 1768 // to track the speaker usage 1769 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1770#endif 1771 if (track->isTerminated()) { 1772 AudioSystem::releaseOutput(mId); 1773 } 1774 } 1775 } 1776 } 1777} 1778 1779void AudioFlinger::PlaybackThread::checkSilentMode_l() 1780{ 1781 if (!mMasterMute) { 1782 char value[PROPERTY_VALUE_MAX]; 1783 if (property_get("ro.audio.silent", value, "0") > 0) { 1784 char *endptr; 1785 unsigned long ul = strtoul(value, &endptr, 0); 1786 if (*endptr == '\0' && ul != 0) { 1787 ALOGD("Silence is golden"); 1788 // The setprop command will not allow a property to be changed after 1789 // the first time it is set, so we don't have to worry about un-muting. 1790 setMasterMute_l(true); 1791 } 1792 } 1793 } 1794} 1795 1796// shared by MIXER and DIRECT, overridden by DUPLICATING 1797ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1798{ 1799 // FIXME rewrite to reduce number of system calls 1800 mLastWriteTime = systemTime(); 1801 mInWrite = true; 1802 ssize_t bytesWritten; 1803 1804 // If an NBAIO sink is present, use it to write the normal mixer's submix 1805 if (mNormalSink != 0) { 1806#define mBitShift 2 // FIXME 1807 size_t count = mBytesRemaining >> mBitShift; 1808 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1809 ATRACE_BEGIN("write"); 1810 // update the setpoint when AudioFlinger::mScreenState changes 1811 uint32_t screenState = AudioFlinger::mScreenState; 1812 if (screenState != mScreenState) { 1813 mScreenState = screenState; 1814 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1815 if (pipe != NULL) { 1816 pipe->setAvgFrames((mScreenState & 1) ? 1817 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1818 } 1819 } 1820 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1821 ATRACE_END(); 1822 if (framesWritten > 0) { 1823 bytesWritten = framesWritten << mBitShift; 1824 } else { 1825 bytesWritten = framesWritten; 1826 } 1827 // otherwise use the HAL / AudioStreamOut directly 1828 } else { 1829 // Direct output and offload threads 1830 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1831 if (mUseAsyncWrite) { 1832 mWriteBlocked = true; 1833 ALOG_ASSERT(mCallbackThread != 0); 1834 mCallbackThread->setWriteBlocked(true); 1835 } 1836 bytesWritten = mOutput->stream->write(mOutput->stream, 1837 mMixBuffer + offset, mBytesRemaining); 1838 if (mUseAsyncWrite && 1839 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1840 // do not wait for async callback in case of error of full write 1841 mWriteBlocked = false; 1842 ALOG_ASSERT(mCallbackThread != 0); 1843 mCallbackThread->setWriteBlocked(false); 1844 } 1845 } 1846 1847 mNumWrites++; 1848 mInWrite = false; 1849 1850 return bytesWritten; 1851} 1852 1853void AudioFlinger::PlaybackThread::threadLoop_drain() 1854{ 1855 if (mOutput->stream->drain) { 1856 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1857 if (mUseAsyncWrite) { 1858 mDraining = true; 1859 ALOG_ASSERT(mCallbackThread != 0); 1860 mCallbackThread->setDraining(true); 1861 } 1862 mOutput->stream->drain(mOutput->stream, 1863 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1864 : AUDIO_DRAIN_ALL); 1865 } 1866} 1867 1868void AudioFlinger::PlaybackThread::threadLoop_exit() 1869{ 1870 // Default implementation has nothing to do 1871} 1872 1873/* 1874The derived values that are cached: 1875 - mixBufferSize from frame count * frame size 1876 - activeSleepTime from activeSleepTimeUs() 1877 - idleSleepTime from idleSleepTimeUs() 1878 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1879 - maxPeriod from frame count and sample rate (MIXER only) 1880 1881The parameters that affect these derived values are: 1882 - frame count 1883 - frame size 1884 - sample rate 1885 - device type: A2DP or not 1886 - device latency 1887 - format: PCM or not 1888 - active sleep time 1889 - idle sleep time 1890*/ 1891 1892void AudioFlinger::PlaybackThread::cacheParameters_l() 1893{ 1894 mixBufferSize = mNormalFrameCount * mFrameSize; 1895 activeSleepTime = activeSleepTimeUs(); 1896 idleSleepTime = idleSleepTimeUs(); 1897} 1898 1899void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1900{ 1901 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1902 this, streamType, mTracks.size()); 1903 Mutex::Autolock _l(mLock); 1904 1905 size_t size = mTracks.size(); 1906 for (size_t i = 0; i < size; i++) { 1907 sp<Track> t = mTracks[i]; 1908 if (t->streamType() == streamType) { 1909 t->invalidate(); 1910 } 1911 } 1912} 1913 1914status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1915{ 1916 int session = chain->sessionId(); 1917 int16_t *buffer = mMixBuffer; 1918 bool ownsBuffer = false; 1919 1920 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1921 if (session > 0) { 1922 // Only one effect chain can be present in direct output thread and it uses 1923 // the mix buffer as input 1924 if (mType != DIRECT) { 1925 size_t numSamples = mNormalFrameCount * mChannelCount; 1926 buffer = new int16_t[numSamples]; 1927 memset(buffer, 0, numSamples * sizeof(int16_t)); 1928 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1929 ownsBuffer = true; 1930 } 1931 1932 // Attach all tracks with same session ID to this chain. 1933 for (size_t i = 0; i < mTracks.size(); ++i) { 1934 sp<Track> track = mTracks[i]; 1935 if (session == track->sessionId()) { 1936 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1937 buffer); 1938 track->setMainBuffer(buffer); 1939 chain->incTrackCnt(); 1940 } 1941 } 1942 1943 // indicate all active tracks in the chain 1944 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1945 sp<Track> track = mActiveTracks[i].promote(); 1946 if (track == 0) { 1947 continue; 1948 } 1949 if (session == track->sessionId()) { 1950 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1951 chain->incActiveTrackCnt(); 1952 } 1953 } 1954 } 1955 1956 chain->setInBuffer(buffer, ownsBuffer); 1957 chain->setOutBuffer(mMixBuffer); 1958 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1959 // chains list in order to be processed last as it contains output stage effects 1960 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1961 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1962 // after track specific effects and before output stage 1963 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1964 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1965 // Effect chain for other sessions are inserted at beginning of effect 1966 // chains list to be processed before output mix effects. Relative order between other 1967 // sessions is not important 1968 size_t size = mEffectChains.size(); 1969 size_t i = 0; 1970 for (i = 0; i < size; i++) { 1971 if (mEffectChains[i]->sessionId() < session) { 1972 break; 1973 } 1974 } 1975 mEffectChains.insertAt(chain, i); 1976 checkSuspendOnAddEffectChain_l(chain); 1977 1978 return NO_ERROR; 1979} 1980 1981size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1982{ 1983 int session = chain->sessionId(); 1984 1985 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1986 1987 for (size_t i = 0; i < mEffectChains.size(); i++) { 1988 if (chain == mEffectChains[i]) { 1989 mEffectChains.removeAt(i); 1990 // detach all active tracks from the chain 1991 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1992 sp<Track> track = mActiveTracks[i].promote(); 1993 if (track == 0) { 1994 continue; 1995 } 1996 if (session == track->sessionId()) { 1997 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1998 chain.get(), session); 1999 chain->decActiveTrackCnt(); 2000 } 2001 } 2002 2003 // detach all tracks with same session ID from this chain 2004 for (size_t i = 0; i < mTracks.size(); ++i) { 2005 sp<Track> track = mTracks[i]; 2006 if (session == track->sessionId()) { 2007 track->setMainBuffer(mMixBuffer); 2008 chain->decTrackCnt(); 2009 } 2010 } 2011 break; 2012 } 2013 } 2014 return mEffectChains.size(); 2015} 2016 2017status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2018 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2019{ 2020 Mutex::Autolock _l(mLock); 2021 return attachAuxEffect_l(track, EffectId); 2022} 2023 2024status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2025 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2026{ 2027 status_t status = NO_ERROR; 2028 2029 if (EffectId == 0) { 2030 track->setAuxBuffer(0, NULL); 2031 } else { 2032 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2033 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2034 if (effect != 0) { 2035 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2036 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2037 } else { 2038 status = INVALID_OPERATION; 2039 } 2040 } else { 2041 status = BAD_VALUE; 2042 } 2043 } 2044 return status; 2045} 2046 2047void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2048{ 2049 for (size_t i = 0; i < mTracks.size(); ++i) { 2050 sp<Track> track = mTracks[i]; 2051 if (track->auxEffectId() == effectId) { 2052 attachAuxEffect_l(track, 0); 2053 } 2054 } 2055} 2056 2057bool AudioFlinger::PlaybackThread::threadLoop() 2058{ 2059 Vector< sp<Track> > tracksToRemove; 2060 2061 standbyTime = systemTime(); 2062 2063 // MIXER 2064 nsecs_t lastWarning = 0; 2065 2066 // DUPLICATING 2067 // FIXME could this be made local to while loop? 2068 writeFrames = 0; 2069 2070 cacheParameters_l(); 2071 sleepTime = idleSleepTime; 2072 2073 if (mType == MIXER) { 2074 sleepTimeShift = 0; 2075 } 2076 2077 CpuStats cpuStats; 2078 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2079 2080 acquireWakeLock(); 2081 2082 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2083 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2084 // and then that string will be logged at the next convenient opportunity. 2085 const char *logString = NULL; 2086 2087 while (!exitPending()) 2088 { 2089 cpuStats.sample(myName); 2090 2091 Vector< sp<EffectChain> > effectChains; 2092 2093 processConfigEvents(); 2094 2095 { // scope for mLock 2096 2097 Mutex::Autolock _l(mLock); 2098 2099 if (logString != NULL) { 2100 mNBLogWriter->logTimestamp(); 2101 mNBLogWriter->log(logString); 2102 logString = NULL; 2103 } 2104 2105 if (checkForNewParameters_l()) { 2106 cacheParameters_l(); 2107 } 2108 2109 saveOutputTracks(); 2110 2111 if (mSignalPending) { 2112 // A signal was raised while we were unlocked 2113 mSignalPending = false; 2114 } else if (waitingAsyncCallback_l()) { 2115 if (exitPending()) { 2116 break; 2117 } 2118 releaseWakeLock_l(); 2119 ALOGV("wait async completion"); 2120 mWaitWorkCV.wait(mLock); 2121 ALOGV("async completion/wake"); 2122 acquireWakeLock_l(); 2123 if (exitPending()) { 2124 break; 2125 } 2126 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2127 continue; 2128 } 2129 sleepTime = 0; 2130 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2131 isSuspended()) { 2132 // put audio hardware into standby after short delay 2133 if (shouldStandby_l()) { 2134 2135 threadLoop_standby(); 2136 2137 mStandby = true; 2138 } 2139 2140 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2141 // we're about to wait, flush the binder command buffer 2142 IPCThreadState::self()->flushCommands(); 2143 2144 clearOutputTracks(); 2145 2146 if (exitPending()) { 2147 break; 2148 } 2149 2150 releaseWakeLock_l(); 2151 // wait until we have something to do... 2152 ALOGV("%s going to sleep", myName.string()); 2153 mWaitWorkCV.wait(mLock); 2154 ALOGV("%s waking up", myName.string()); 2155 acquireWakeLock_l(); 2156 2157 mMixerStatus = MIXER_IDLE; 2158 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2159 mBytesWritten = 0; 2160 mBytesRemaining = 0; 2161 checkSilentMode_l(); 2162 2163 standbyTime = systemTime() + standbyDelay; 2164 sleepTime = idleSleepTime; 2165 if (mType == MIXER) { 2166 sleepTimeShift = 0; 2167 } 2168 2169 continue; 2170 } 2171 } 2172 2173 // mMixerStatusIgnoringFastTracks is also updated internally 2174 mMixerStatus = prepareTracks_l(&tracksToRemove); 2175 2176 // prevent any changes in effect chain list and in each effect chain 2177 // during mixing and effect process as the audio buffers could be deleted 2178 // or modified if an effect is created or deleted 2179 lockEffectChains_l(effectChains); 2180 } 2181 2182 if (mBytesRemaining == 0) { 2183 mCurrentWriteLength = 0; 2184 if (mMixerStatus == MIXER_TRACKS_READY) { 2185 // threadLoop_mix() sets mCurrentWriteLength 2186 threadLoop_mix(); 2187 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2188 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2189 // threadLoop_sleepTime sets sleepTime to 0 if data 2190 // must be written to HAL 2191 threadLoop_sleepTime(); 2192 if (sleepTime == 0) { 2193 mCurrentWriteLength = mixBufferSize; 2194 } 2195 } 2196 mBytesRemaining = mCurrentWriteLength; 2197 if (isSuspended()) { 2198 sleepTime = suspendSleepTimeUs(); 2199 // simulate write to HAL when suspended 2200 mBytesWritten += mixBufferSize; 2201 mBytesRemaining = 0; 2202 } 2203 2204 // only process effects if we're going to write 2205 if (sleepTime == 0) { 2206 for (size_t i = 0; i < effectChains.size(); i ++) { 2207 effectChains[i]->process_l(); 2208 } 2209 } 2210 } 2211 2212 // enable changes in effect chain 2213 unlockEffectChains(effectChains); 2214 2215 if (!waitingAsyncCallback()) { 2216 // sleepTime == 0 means we must write to audio hardware 2217 if (sleepTime == 0) { 2218 if (mBytesRemaining) { 2219 ssize_t ret = threadLoop_write(); 2220 if (ret < 0) { 2221 mBytesRemaining = 0; 2222 } else { 2223 mBytesWritten += ret; 2224 mBytesRemaining -= ret; 2225 } 2226 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2227 (mMixerStatus == MIXER_DRAIN_ALL)) { 2228 threadLoop_drain(); 2229 } 2230if (mType == MIXER) { 2231 // write blocked detection 2232 nsecs_t now = systemTime(); 2233 nsecs_t delta = now - mLastWriteTime; 2234 if (!mStandby && delta > maxPeriod) { 2235 mNumDelayedWrites++; 2236 if ((now - lastWarning) > kWarningThrottleNs) { 2237 ATRACE_NAME("underrun"); 2238 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2239 ns2ms(delta), mNumDelayedWrites, this); 2240 lastWarning = now; 2241 } 2242 } 2243} 2244 2245 mStandby = false; 2246 } else { 2247 usleep(sleepTime); 2248 } 2249 } 2250 2251 // Finally let go of removed track(s), without the lock held 2252 // since we can't guarantee the destructors won't acquire that 2253 // same lock. This will also mutate and push a new fast mixer state. 2254 threadLoop_removeTracks(tracksToRemove); 2255 tracksToRemove.clear(); 2256 2257 // FIXME I don't understand the need for this here; 2258 // it was in the original code but maybe the 2259 // assignment in saveOutputTracks() makes this unnecessary? 2260 clearOutputTracks(); 2261 2262 // Effect chains will be actually deleted here if they were removed from 2263 // mEffectChains list during mixing or effects processing 2264 effectChains.clear(); 2265 2266 // FIXME Note that the above .clear() is no longer necessary since effectChains 2267 // is now local to this block, but will keep it for now (at least until merge done). 2268 } 2269 2270 threadLoop_exit(); 2271 2272 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2273 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2274 // put output stream into standby mode 2275 if (!mStandby) { 2276 mOutput->stream->common.standby(&mOutput->stream->common); 2277 } 2278 } 2279 2280 releaseWakeLock(); 2281 2282 ALOGV("Thread %p type %d exiting", this, mType); 2283 return false; 2284} 2285 2286// removeTracks_l() must be called with ThreadBase::mLock held 2287void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2288{ 2289 size_t count = tracksToRemove.size(); 2290 if (count > 0) { 2291 for (size_t i=0 ; i<count ; i++) { 2292 const sp<Track>& track = tracksToRemove.itemAt(i); 2293 mActiveTracks.remove(track); 2294 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2295 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2296 if (chain != 0) { 2297 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2298 track->sessionId()); 2299 chain->decActiveTrackCnt(); 2300 } 2301 if (track->isTerminated()) { 2302 removeTrack_l(track); 2303 } 2304 } 2305 } 2306 2307} 2308 2309// ---------------------------------------------------------------------------- 2310 2311AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2312 audio_io_handle_t id, audio_devices_t device, type_t type) 2313 : PlaybackThread(audioFlinger, output, id, device, type), 2314 // mAudioMixer below 2315 // mFastMixer below 2316 mFastMixerFutex(0) 2317 // mOutputSink below 2318 // mPipeSink below 2319 // mNormalSink below 2320{ 2321 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2322 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2323 "mFrameCount=%d, mNormalFrameCount=%d", 2324 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2325 mNormalFrameCount); 2326 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2327 2328 // FIXME - Current mixer implementation only supports stereo output 2329 if (mChannelCount != FCC_2) { 2330 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2331 } 2332 2333 // create an NBAIO sink for the HAL output stream, and negotiate 2334 mOutputSink = new AudioStreamOutSink(output->stream); 2335 size_t numCounterOffers = 0; 2336 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2337 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2338 ALOG_ASSERT(index == 0); 2339 2340 // initialize fast mixer depending on configuration 2341 bool initFastMixer; 2342 switch (kUseFastMixer) { 2343 case FastMixer_Never: 2344 initFastMixer = false; 2345 break; 2346 case FastMixer_Always: 2347 initFastMixer = true; 2348 break; 2349 case FastMixer_Static: 2350 case FastMixer_Dynamic: 2351 initFastMixer = mFrameCount < mNormalFrameCount; 2352 break; 2353 } 2354 if (initFastMixer) { 2355 2356 // create a MonoPipe to connect our submix to FastMixer 2357 NBAIO_Format format = mOutputSink->format(); 2358 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2359 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2360 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2361 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2362 const NBAIO_Format offers[1] = {format}; 2363 size_t numCounterOffers = 0; 2364 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2365 ALOG_ASSERT(index == 0); 2366 monoPipe->setAvgFrames((mScreenState & 1) ? 2367 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2368 mPipeSink = monoPipe; 2369 2370#ifdef TEE_SINK 2371 if (mTeeSinkOutputEnabled) { 2372 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2373 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2374 numCounterOffers = 0; 2375 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2376 ALOG_ASSERT(index == 0); 2377 mTeeSink = teeSink; 2378 PipeReader *teeSource = new PipeReader(*teeSink); 2379 numCounterOffers = 0; 2380 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2381 ALOG_ASSERT(index == 0); 2382 mTeeSource = teeSource; 2383 } 2384#endif 2385 2386 // create fast mixer and configure it initially with just one fast track for our submix 2387 mFastMixer = new FastMixer(); 2388 FastMixerStateQueue *sq = mFastMixer->sq(); 2389#ifdef STATE_QUEUE_DUMP 2390 sq->setObserverDump(&mStateQueueObserverDump); 2391 sq->setMutatorDump(&mStateQueueMutatorDump); 2392#endif 2393 FastMixerState *state = sq->begin(); 2394 FastTrack *fastTrack = &state->mFastTracks[0]; 2395 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2396 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2397 fastTrack->mVolumeProvider = NULL; 2398 fastTrack->mGeneration++; 2399 state->mFastTracksGen++; 2400 state->mTrackMask = 1; 2401 // fast mixer will use the HAL output sink 2402 state->mOutputSink = mOutputSink.get(); 2403 state->mOutputSinkGen++; 2404 state->mFrameCount = mFrameCount; 2405 state->mCommand = FastMixerState::COLD_IDLE; 2406 // already done in constructor initialization list 2407 //mFastMixerFutex = 0; 2408 state->mColdFutexAddr = &mFastMixerFutex; 2409 state->mColdGen++; 2410 state->mDumpState = &mFastMixerDumpState; 2411#ifdef TEE_SINK 2412 state->mTeeSink = mTeeSink.get(); 2413#endif 2414 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2415 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2416 sq->end(); 2417 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2418 2419 // start the fast mixer 2420 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2421 pid_t tid = mFastMixer->getTid(); 2422 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2423 if (err != 0) { 2424 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2425 kPriorityFastMixer, getpid_cached, tid, err); 2426 } 2427 2428#ifdef AUDIO_WATCHDOG 2429 // create and start the watchdog 2430 mAudioWatchdog = new AudioWatchdog(); 2431 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2432 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2433 tid = mAudioWatchdog->getTid(); 2434 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2435 if (err != 0) { 2436 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2437 kPriorityFastMixer, getpid_cached, tid, err); 2438 } 2439#endif 2440 2441 } else { 2442 mFastMixer = NULL; 2443 } 2444 2445 switch (kUseFastMixer) { 2446 case FastMixer_Never: 2447 case FastMixer_Dynamic: 2448 mNormalSink = mOutputSink; 2449 break; 2450 case FastMixer_Always: 2451 mNormalSink = mPipeSink; 2452 break; 2453 case FastMixer_Static: 2454 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2455 break; 2456 } 2457} 2458 2459AudioFlinger::MixerThread::~MixerThread() 2460{ 2461 if (mFastMixer != NULL) { 2462 FastMixerStateQueue *sq = mFastMixer->sq(); 2463 FastMixerState *state = sq->begin(); 2464 if (state->mCommand == FastMixerState::COLD_IDLE) { 2465 int32_t old = android_atomic_inc(&mFastMixerFutex); 2466 if (old == -1) { 2467 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2468 } 2469 } 2470 state->mCommand = FastMixerState::EXIT; 2471 sq->end(); 2472 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2473 mFastMixer->join(); 2474 // Though the fast mixer thread has exited, it's state queue is still valid. 2475 // We'll use that extract the final state which contains one remaining fast track 2476 // corresponding to our sub-mix. 2477 state = sq->begin(); 2478 ALOG_ASSERT(state->mTrackMask == 1); 2479 FastTrack *fastTrack = &state->mFastTracks[0]; 2480 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2481 delete fastTrack->mBufferProvider; 2482 sq->end(false /*didModify*/); 2483 delete mFastMixer; 2484#ifdef AUDIO_WATCHDOG 2485 if (mAudioWatchdog != 0) { 2486 mAudioWatchdog->requestExit(); 2487 mAudioWatchdog->requestExitAndWait(); 2488 mAudioWatchdog.clear(); 2489 } 2490#endif 2491 } 2492 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2493 delete mAudioMixer; 2494} 2495 2496 2497uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2498{ 2499 if (mFastMixer != NULL) { 2500 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2501 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2502 } 2503 return latency; 2504} 2505 2506 2507void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2508{ 2509 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2510} 2511 2512ssize_t AudioFlinger::MixerThread::threadLoop_write() 2513{ 2514 // FIXME we should only do one push per cycle; confirm this is true 2515 // Start the fast mixer if it's not already running 2516 if (mFastMixer != NULL) { 2517 FastMixerStateQueue *sq = mFastMixer->sq(); 2518 FastMixerState *state = sq->begin(); 2519 if (state->mCommand != FastMixerState::MIX_WRITE && 2520 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2521 if (state->mCommand == FastMixerState::COLD_IDLE) { 2522 int32_t old = android_atomic_inc(&mFastMixerFutex); 2523 if (old == -1) { 2524 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2525 } 2526#ifdef AUDIO_WATCHDOG 2527 if (mAudioWatchdog != 0) { 2528 mAudioWatchdog->resume(); 2529 } 2530#endif 2531 } 2532 state->mCommand = FastMixerState::MIX_WRITE; 2533 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2534 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2535 sq->end(); 2536 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2537 if (kUseFastMixer == FastMixer_Dynamic) { 2538 mNormalSink = mPipeSink; 2539 } 2540 } else { 2541 sq->end(false /*didModify*/); 2542 } 2543 } 2544 return PlaybackThread::threadLoop_write(); 2545} 2546 2547void AudioFlinger::MixerThread::threadLoop_standby() 2548{ 2549 // Idle the fast mixer if it's currently running 2550 if (mFastMixer != NULL) { 2551 FastMixerStateQueue *sq = mFastMixer->sq(); 2552 FastMixerState *state = sq->begin(); 2553 if (!(state->mCommand & FastMixerState::IDLE)) { 2554 state->mCommand = FastMixerState::COLD_IDLE; 2555 state->mColdFutexAddr = &mFastMixerFutex; 2556 state->mColdGen++; 2557 mFastMixerFutex = 0; 2558 sq->end(); 2559 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2560 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2561 if (kUseFastMixer == FastMixer_Dynamic) { 2562 mNormalSink = mOutputSink; 2563 } 2564#ifdef AUDIO_WATCHDOG 2565 if (mAudioWatchdog != 0) { 2566 mAudioWatchdog->pause(); 2567 } 2568#endif 2569 } else { 2570 sq->end(false /*didModify*/); 2571 } 2572 } 2573 PlaybackThread::threadLoop_standby(); 2574} 2575 2576// Empty implementation for standard mixer 2577// Overridden for offloaded playback 2578void AudioFlinger::PlaybackThread::flushOutput_l() 2579{ 2580} 2581 2582bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2583{ 2584 return false; 2585} 2586 2587bool AudioFlinger::PlaybackThread::shouldStandby_l() 2588{ 2589 return !mStandby; 2590} 2591 2592bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2593{ 2594 Mutex::Autolock _l(mLock); 2595 return waitingAsyncCallback_l(); 2596} 2597 2598// shared by MIXER and DIRECT, overridden by DUPLICATING 2599void AudioFlinger::PlaybackThread::threadLoop_standby() 2600{ 2601 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2602 mOutput->stream->common.standby(&mOutput->stream->common); 2603 if (mUseAsyncWrite != 0) { 2604 mWriteBlocked = false; 2605 mDraining = false; 2606 ALOG_ASSERT(mCallbackThread != 0); 2607 mCallbackThread->setWriteBlocked(false); 2608 mCallbackThread->setDraining(false); 2609 } 2610} 2611 2612void AudioFlinger::MixerThread::threadLoop_mix() 2613{ 2614 // obtain the presentation timestamp of the next output buffer 2615 int64_t pts; 2616 status_t status = INVALID_OPERATION; 2617 2618 if (mNormalSink != 0) { 2619 status = mNormalSink->getNextWriteTimestamp(&pts); 2620 } else { 2621 status = mOutputSink->getNextWriteTimestamp(&pts); 2622 } 2623 2624 if (status != NO_ERROR) { 2625 pts = AudioBufferProvider::kInvalidPTS; 2626 } 2627 2628 // mix buffers... 2629 mAudioMixer->process(pts); 2630 mCurrentWriteLength = mixBufferSize; 2631 // increase sleep time progressively when application underrun condition clears. 2632 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2633 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2634 // such that we would underrun the audio HAL. 2635 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2636 sleepTimeShift--; 2637 } 2638 sleepTime = 0; 2639 standbyTime = systemTime() + standbyDelay; 2640 //TODO: delay standby when effects have a tail 2641} 2642 2643void AudioFlinger::MixerThread::threadLoop_sleepTime() 2644{ 2645 // If no tracks are ready, sleep once for the duration of an output 2646 // buffer size, then write 0s to the output 2647 if (sleepTime == 0) { 2648 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2649 sleepTime = activeSleepTime >> sleepTimeShift; 2650 if (sleepTime < kMinThreadSleepTimeUs) { 2651 sleepTime = kMinThreadSleepTimeUs; 2652 } 2653 // reduce sleep time in case of consecutive application underruns to avoid 2654 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2655 // duration we would end up writing less data than needed by the audio HAL if 2656 // the condition persists. 2657 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2658 sleepTimeShift++; 2659 } 2660 } else { 2661 sleepTime = idleSleepTime; 2662 } 2663 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2664 memset(mMixBuffer, 0, mixBufferSize); 2665 sleepTime = 0; 2666 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2667 "anticipated start"); 2668 } 2669 // TODO add standby time extension fct of effect tail 2670} 2671 2672// prepareTracks_l() must be called with ThreadBase::mLock held 2673AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2674 Vector< sp<Track> > *tracksToRemove) 2675{ 2676 2677 mixer_state mixerStatus = MIXER_IDLE; 2678 // find out which tracks need to be processed 2679 size_t count = mActiveTracks.size(); 2680 size_t mixedTracks = 0; 2681 size_t tracksWithEffect = 0; 2682 // counts only _active_ fast tracks 2683 size_t fastTracks = 0; 2684 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2685 2686 float masterVolume = mMasterVolume; 2687 bool masterMute = mMasterMute; 2688 2689 if (masterMute) { 2690 masterVolume = 0; 2691 } 2692 // Delegate master volume control to effect in output mix effect chain if needed 2693 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2694 if (chain != 0) { 2695 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2696 chain->setVolume_l(&v, &v); 2697 masterVolume = (float)((v + (1 << 23)) >> 24); 2698 chain.clear(); 2699 } 2700 2701 // prepare a new state to push 2702 FastMixerStateQueue *sq = NULL; 2703 FastMixerState *state = NULL; 2704 bool didModify = false; 2705 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2706 if (mFastMixer != NULL) { 2707 sq = mFastMixer->sq(); 2708 state = sq->begin(); 2709 } 2710 2711 for (size_t i=0 ; i<count ; i++) { 2712 const sp<Track> t = mActiveTracks[i].promote(); 2713 if (t == 0) { 2714 continue; 2715 } 2716 2717 // this const just means the local variable doesn't change 2718 Track* const track = t.get(); 2719 2720 // process fast tracks 2721 if (track->isFastTrack()) { 2722 2723 // It's theoretically possible (though unlikely) for a fast track to be created 2724 // and then removed within the same normal mix cycle. This is not a problem, as 2725 // the track never becomes active so it's fast mixer slot is never touched. 2726 // The converse, of removing an (active) track and then creating a new track 2727 // at the identical fast mixer slot within the same normal mix cycle, 2728 // is impossible because the slot isn't marked available until the end of each cycle. 2729 int j = track->mFastIndex; 2730 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2731 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2732 FastTrack *fastTrack = &state->mFastTracks[j]; 2733 2734 // Determine whether the track is currently in underrun condition, 2735 // and whether it had a recent underrun. 2736 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2737 FastTrackUnderruns underruns = ftDump->mUnderruns; 2738 uint32_t recentFull = (underruns.mBitFields.mFull - 2739 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2740 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2741 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2742 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2743 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2744 uint32_t recentUnderruns = recentPartial + recentEmpty; 2745 track->mObservedUnderruns = underruns; 2746 // don't count underruns that occur while stopping or pausing 2747 // or stopped which can occur when flush() is called while active 2748 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2749 recentUnderruns > 0) { 2750 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2751 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2752 } 2753 2754 // This is similar to the state machine for normal tracks, 2755 // with a few modifications for fast tracks. 2756 bool isActive = true; 2757 switch (track->mState) { 2758 case TrackBase::STOPPING_1: 2759 // track stays active in STOPPING_1 state until first underrun 2760 if (recentUnderruns > 0 || track->isTerminated()) { 2761 track->mState = TrackBase::STOPPING_2; 2762 } 2763 break; 2764 case TrackBase::PAUSING: 2765 // ramp down is not yet implemented 2766 track->setPaused(); 2767 break; 2768 case TrackBase::RESUMING: 2769 // ramp up is not yet implemented 2770 track->mState = TrackBase::ACTIVE; 2771 break; 2772 case TrackBase::ACTIVE: 2773 if (recentFull > 0 || recentPartial > 0) { 2774 // track has provided at least some frames recently: reset retry count 2775 track->mRetryCount = kMaxTrackRetries; 2776 } 2777 if (recentUnderruns == 0) { 2778 // no recent underruns: stay active 2779 break; 2780 } 2781 // there has recently been an underrun of some kind 2782 if (track->sharedBuffer() == 0) { 2783 // were any of the recent underruns "empty" (no frames available)? 2784 if (recentEmpty == 0) { 2785 // no, then ignore the partial underruns as they are allowed indefinitely 2786 break; 2787 } 2788 // there has recently been an "empty" underrun: decrement the retry counter 2789 if (--(track->mRetryCount) > 0) { 2790 break; 2791 } 2792 // indicate to client process that the track was disabled because of underrun; 2793 // it will then automatically call start() when data is available 2794 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2795 // remove from active list, but state remains ACTIVE [confusing but true] 2796 isActive = false; 2797 break; 2798 } 2799 // fall through 2800 case TrackBase::STOPPING_2: 2801 case TrackBase::PAUSED: 2802 case TrackBase::STOPPED: 2803 case TrackBase::FLUSHED: // flush() while active 2804 // Check for presentation complete if track is inactive 2805 // We have consumed all the buffers of this track. 2806 // This would be incomplete if we auto-paused on underrun 2807 { 2808 size_t audioHALFrames = 2809 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2810 size_t framesWritten = mBytesWritten / mFrameSize; 2811 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2812 // track stays in active list until presentation is complete 2813 break; 2814 } 2815 } 2816 if (track->isStopping_2()) { 2817 track->mState = TrackBase::STOPPED; 2818 } 2819 if (track->isStopped()) { 2820 // Can't reset directly, as fast mixer is still polling this track 2821 // track->reset(); 2822 // So instead mark this track as needing to be reset after push with ack 2823 resetMask |= 1 << i; 2824 } 2825 isActive = false; 2826 break; 2827 case TrackBase::IDLE: 2828 default: 2829 LOG_FATAL("unexpected track state %d", track->mState); 2830 } 2831 2832 if (isActive) { 2833 // was it previously inactive? 2834 if (!(state->mTrackMask & (1 << j))) { 2835 ExtendedAudioBufferProvider *eabp = track; 2836 VolumeProvider *vp = track; 2837 fastTrack->mBufferProvider = eabp; 2838 fastTrack->mVolumeProvider = vp; 2839 fastTrack->mSampleRate = track->mSampleRate; 2840 fastTrack->mChannelMask = track->mChannelMask; 2841 fastTrack->mGeneration++; 2842 state->mTrackMask |= 1 << j; 2843 didModify = true; 2844 // no acknowledgement required for newly active tracks 2845 } 2846 // cache the combined master volume and stream type volume for fast mixer; this 2847 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2848 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2849 ++fastTracks; 2850 } else { 2851 // was it previously active? 2852 if (state->mTrackMask & (1 << j)) { 2853 fastTrack->mBufferProvider = NULL; 2854 fastTrack->mGeneration++; 2855 state->mTrackMask &= ~(1 << j); 2856 didModify = true; 2857 // If any fast tracks were removed, we must wait for acknowledgement 2858 // because we're about to decrement the last sp<> on those tracks. 2859 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2860 } else { 2861 LOG_FATAL("fast track %d should have been active", j); 2862 } 2863 tracksToRemove->add(track); 2864 // Avoids a misleading display in dumpsys 2865 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2866 } 2867 continue; 2868 } 2869 2870 { // local variable scope to avoid goto warning 2871 2872 audio_track_cblk_t* cblk = track->cblk(); 2873 2874 // The first time a track is added we wait 2875 // for all its buffers to be filled before processing it 2876 int name = track->name(); 2877 // make sure that we have enough frames to mix one full buffer. 2878 // enforce this condition only once to enable draining the buffer in case the client 2879 // app does not call stop() and relies on underrun to stop: 2880 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2881 // during last round 2882 size_t desiredFrames; 2883 uint32_t sr = track->sampleRate(); 2884 if (sr == mSampleRate) { 2885 desiredFrames = mNormalFrameCount; 2886 } else { 2887 // +1 for rounding and +1 for additional sample needed for interpolation 2888 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2889 // add frames already consumed but not yet released by the resampler 2890 // because mAudioTrackServerProxy->framesReady() will include these frames 2891 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2892 // the minimum track buffer size is normally twice the number of frames necessary 2893 // to fill one buffer and the resampler should not leave more than one buffer worth 2894 // of unreleased frames after each pass, but just in case... 2895 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2896 } 2897 uint32_t minFrames = 1; 2898 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2899 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2900 minFrames = desiredFrames; 2901 } 2902 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2903 size_t framesReady; 2904 if (track->sharedBuffer() == 0) { 2905 framesReady = track->framesReady(); 2906 } else if (track->isStopped()) { 2907 framesReady = 0; 2908 } else { 2909 framesReady = 1; 2910 } 2911 if ((framesReady >= minFrames) && track->isReady() && 2912 !track->isPaused() && !track->isTerminated()) 2913 { 2914 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2915 2916 mixedTracks++; 2917 2918 // track->mainBuffer() != mMixBuffer means there is an effect chain 2919 // connected to the track 2920 chain.clear(); 2921 if (track->mainBuffer() != mMixBuffer) { 2922 chain = getEffectChain_l(track->sessionId()); 2923 // Delegate volume control to effect in track effect chain if needed 2924 if (chain != 0) { 2925 tracksWithEffect++; 2926 } else { 2927 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2928 "session %d", 2929 name, track->sessionId()); 2930 } 2931 } 2932 2933 2934 int param = AudioMixer::VOLUME; 2935 if (track->mFillingUpStatus == Track::FS_FILLED) { 2936 // no ramp for the first volume setting 2937 track->mFillingUpStatus = Track::FS_ACTIVE; 2938 if (track->mState == TrackBase::RESUMING) { 2939 track->mState = TrackBase::ACTIVE; 2940 param = AudioMixer::RAMP_VOLUME; 2941 } 2942 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2943 // FIXME should not make a decision based on mServer 2944 } else if (cblk->mServer != 0) { 2945 // If the track is stopped before the first frame was mixed, 2946 // do not apply ramp 2947 param = AudioMixer::RAMP_VOLUME; 2948 } 2949 2950 // compute volume for this track 2951 uint32_t vl, vr, va; 2952 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2953 vl = vr = va = 0; 2954 if (track->isPausing()) { 2955 track->setPaused(); 2956 } 2957 } else { 2958 2959 // read original volumes with volume control 2960 float typeVolume = mStreamTypes[track->streamType()].volume; 2961 float v = masterVolume * typeVolume; 2962 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2963 uint32_t vlr = proxy->getVolumeLR(); 2964 vl = vlr & 0xFFFF; 2965 vr = vlr >> 16; 2966 // track volumes come from shared memory, so can't be trusted and must be clamped 2967 if (vl > MAX_GAIN_INT) { 2968 ALOGV("Track left volume out of range: %04X", vl); 2969 vl = MAX_GAIN_INT; 2970 } 2971 if (vr > MAX_GAIN_INT) { 2972 ALOGV("Track right volume out of range: %04X", vr); 2973 vr = MAX_GAIN_INT; 2974 } 2975 // now apply the master volume and stream type volume 2976 vl = (uint32_t)(v * vl) << 12; 2977 vr = (uint32_t)(v * vr) << 12; 2978 // assuming master volume and stream type volume each go up to 1.0, 2979 // vl and vr are now in 8.24 format 2980 2981 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2982 // send level comes from shared memory and so may be corrupt 2983 if (sendLevel > MAX_GAIN_INT) { 2984 ALOGV("Track send level out of range: %04X", sendLevel); 2985 sendLevel = MAX_GAIN_INT; 2986 } 2987 va = (uint32_t)(v * sendLevel); 2988 } 2989 2990 // Delegate volume control to effect in track effect chain if needed 2991 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2992 // Do not ramp volume if volume is controlled by effect 2993 param = AudioMixer::VOLUME; 2994 track->mHasVolumeController = true; 2995 } else { 2996 // force no volume ramp when volume controller was just disabled or removed 2997 // from effect chain to avoid volume spike 2998 if (track->mHasVolumeController) { 2999 param = AudioMixer::VOLUME; 3000 } 3001 track->mHasVolumeController = false; 3002 } 3003 3004 // Convert volumes from 8.24 to 4.12 format 3005 // This additional clamping is needed in case chain->setVolume_l() overshot 3006 vl = (vl + (1 << 11)) >> 12; 3007 if (vl > MAX_GAIN_INT) { 3008 vl = MAX_GAIN_INT; 3009 } 3010 vr = (vr + (1 << 11)) >> 12; 3011 if (vr > MAX_GAIN_INT) { 3012 vr = MAX_GAIN_INT; 3013 } 3014 3015 if (va > MAX_GAIN_INT) { 3016 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3017 } 3018 3019 // XXX: these things DON'T need to be done each time 3020 mAudioMixer->setBufferProvider(name, track); 3021 mAudioMixer->enable(name); 3022 3023 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3024 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3025 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3026 mAudioMixer->setParameter( 3027 name, 3028 AudioMixer::TRACK, 3029 AudioMixer::FORMAT, (void *)track->format()); 3030 mAudioMixer->setParameter( 3031 name, 3032 AudioMixer::TRACK, 3033 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3034 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3035 uint32_t maxSampleRate = mSampleRate * 2; 3036 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3037 if (reqSampleRate == 0) { 3038 reqSampleRate = mSampleRate; 3039 } else if (reqSampleRate > maxSampleRate) { 3040 reqSampleRate = maxSampleRate; 3041 } 3042 mAudioMixer->setParameter( 3043 name, 3044 AudioMixer::RESAMPLE, 3045 AudioMixer::SAMPLE_RATE, 3046 (void *)reqSampleRate); 3047 mAudioMixer->setParameter( 3048 name, 3049 AudioMixer::TRACK, 3050 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3051 mAudioMixer->setParameter( 3052 name, 3053 AudioMixer::TRACK, 3054 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3055 3056 // reset retry count 3057 track->mRetryCount = kMaxTrackRetries; 3058 3059 // If one track is ready, set the mixer ready if: 3060 // - the mixer was not ready during previous round OR 3061 // - no other track is not ready 3062 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3063 mixerStatus != MIXER_TRACKS_ENABLED) { 3064 mixerStatus = MIXER_TRACKS_READY; 3065 } 3066 } else { 3067 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3068 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3069 } 3070 // clear effect chain input buffer if an active track underruns to avoid sending 3071 // previous audio buffer again to effects 3072 chain = getEffectChain_l(track->sessionId()); 3073 if (chain != 0) { 3074 chain->clearInputBuffer(); 3075 } 3076 3077 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3078 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3079 track->isStopped() || track->isPaused()) { 3080 // We have consumed all the buffers of this track. 3081 // Remove it from the list of active tracks. 3082 // TODO: use actual buffer filling status instead of latency when available from 3083 // audio HAL 3084 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3085 size_t framesWritten = mBytesWritten / mFrameSize; 3086 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3087 if (track->isStopped()) { 3088 track->reset(); 3089 } 3090 tracksToRemove->add(track); 3091 } 3092 } else { 3093 // No buffers for this track. Give it a few chances to 3094 // fill a buffer, then remove it from active list. 3095 if (--(track->mRetryCount) <= 0) { 3096 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3097 tracksToRemove->add(track); 3098 // indicate to client process that the track was disabled because of underrun; 3099 // it will then automatically call start() when data is available 3100 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3101 // If one track is not ready, mark the mixer also not ready if: 3102 // - the mixer was ready during previous round OR 3103 // - no other track is ready 3104 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3105 mixerStatus != MIXER_TRACKS_READY) { 3106 mixerStatus = MIXER_TRACKS_ENABLED; 3107 } 3108 } 3109 mAudioMixer->disable(name); 3110 } 3111 3112 } // local variable scope to avoid goto warning 3113track_is_ready: ; 3114 3115 } 3116 3117 // Push the new FastMixer state if necessary 3118 bool pauseAudioWatchdog = false; 3119 if (didModify) { 3120 state->mFastTracksGen++; 3121 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3122 if (kUseFastMixer == FastMixer_Dynamic && 3123 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3124 state->mCommand = FastMixerState::COLD_IDLE; 3125 state->mColdFutexAddr = &mFastMixerFutex; 3126 state->mColdGen++; 3127 mFastMixerFutex = 0; 3128 if (kUseFastMixer == FastMixer_Dynamic) { 3129 mNormalSink = mOutputSink; 3130 } 3131 // If we go into cold idle, need to wait for acknowledgement 3132 // so that fast mixer stops doing I/O. 3133 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3134 pauseAudioWatchdog = true; 3135 } 3136 } 3137 if (sq != NULL) { 3138 sq->end(didModify); 3139 sq->push(block); 3140 } 3141#ifdef AUDIO_WATCHDOG 3142 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3143 mAudioWatchdog->pause(); 3144 } 3145#endif 3146 3147 // Now perform the deferred reset on fast tracks that have stopped 3148 while (resetMask != 0) { 3149 size_t i = __builtin_ctz(resetMask); 3150 ALOG_ASSERT(i < count); 3151 resetMask &= ~(1 << i); 3152 sp<Track> t = mActiveTracks[i].promote(); 3153 if (t == 0) { 3154 continue; 3155 } 3156 Track* track = t.get(); 3157 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3158 track->reset(); 3159 } 3160 3161 // remove all the tracks that need to be... 3162 removeTracks_l(*tracksToRemove); 3163 3164 // mix buffer must be cleared if all tracks are connected to an 3165 // effect chain as in this case the mixer will not write to 3166 // mix buffer and track effects will accumulate into it 3167 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3168 (mixedTracks == 0 && fastTracks > 0))) { 3169 // FIXME as a performance optimization, should remember previous zero status 3170 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3171 } 3172 3173 // if any fast tracks, then status is ready 3174 mMixerStatusIgnoringFastTracks = mixerStatus; 3175 if (fastTracks > 0) { 3176 mixerStatus = MIXER_TRACKS_READY; 3177 } 3178 return mixerStatus; 3179} 3180 3181// getTrackName_l() must be called with ThreadBase::mLock held 3182int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3183{ 3184 return mAudioMixer->getTrackName(channelMask, sessionId); 3185} 3186 3187// deleteTrackName_l() must be called with ThreadBase::mLock held 3188void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3189{ 3190 ALOGV("remove track (%d) and delete from mixer", name); 3191 mAudioMixer->deleteTrackName(name); 3192} 3193 3194// checkForNewParameters_l() must be called with ThreadBase::mLock held 3195bool AudioFlinger::MixerThread::checkForNewParameters_l() 3196{ 3197 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3198 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3199 bool reconfig = false; 3200 3201 while (!mNewParameters.isEmpty()) { 3202 3203 if (mFastMixer != NULL) { 3204 FastMixerStateQueue *sq = mFastMixer->sq(); 3205 FastMixerState *state = sq->begin(); 3206 if (!(state->mCommand & FastMixerState::IDLE)) { 3207 previousCommand = state->mCommand; 3208 state->mCommand = FastMixerState::HOT_IDLE; 3209 sq->end(); 3210 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3211 } else { 3212 sq->end(false /*didModify*/); 3213 } 3214 } 3215 3216 status_t status = NO_ERROR; 3217 String8 keyValuePair = mNewParameters[0]; 3218 AudioParameter param = AudioParameter(keyValuePair); 3219 int value; 3220 3221 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3222 reconfig = true; 3223 } 3224 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3225 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3226 status = BAD_VALUE; 3227 } else { 3228 // no need to save value, since it's constant 3229 reconfig = true; 3230 } 3231 } 3232 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3233 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3234 status = BAD_VALUE; 3235 } else { 3236 // no need to save value, since it's constant 3237 reconfig = true; 3238 } 3239 } 3240 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3241 // do not accept frame count changes if tracks are open as the track buffer 3242 // size depends on frame count and correct behavior would not be guaranteed 3243 // if frame count is changed after track creation 3244 if (!mTracks.isEmpty()) { 3245 status = INVALID_OPERATION; 3246 } else { 3247 reconfig = true; 3248 } 3249 } 3250 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3251#ifdef ADD_BATTERY_DATA 3252 // when changing the audio output device, call addBatteryData to notify 3253 // the change 3254 if (mOutDevice != value) { 3255 uint32_t params = 0; 3256 // check whether speaker is on 3257 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3258 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3259 } 3260 3261 audio_devices_t deviceWithoutSpeaker 3262 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3263 // check if any other device (except speaker) is on 3264 if (value & deviceWithoutSpeaker ) { 3265 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3266 } 3267 3268 if (params != 0) { 3269 addBatteryData(params); 3270 } 3271 } 3272#endif 3273 3274 // forward device change to effects that have requested to be 3275 // aware of attached audio device. 3276 if (value != AUDIO_DEVICE_NONE) { 3277 mOutDevice = value; 3278 for (size_t i = 0; i < mEffectChains.size(); i++) { 3279 mEffectChains[i]->setDevice_l(mOutDevice); 3280 } 3281 } 3282 } 3283 3284 if (status == NO_ERROR) { 3285 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3286 keyValuePair.string()); 3287 if (!mStandby && status == INVALID_OPERATION) { 3288 mOutput->stream->common.standby(&mOutput->stream->common); 3289 mStandby = true; 3290 mBytesWritten = 0; 3291 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3292 keyValuePair.string()); 3293 } 3294 if (status == NO_ERROR && reconfig) { 3295 readOutputParameters(); 3296 delete mAudioMixer; 3297 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3298 for (size_t i = 0; i < mTracks.size() ; i++) { 3299 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3300 if (name < 0) { 3301 break; 3302 } 3303 mTracks[i]->mName = name; 3304 } 3305 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3306 } 3307 } 3308 3309 mNewParameters.removeAt(0); 3310 3311 mParamStatus = status; 3312 mParamCond.signal(); 3313 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3314 // already timed out waiting for the status and will never signal the condition. 3315 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3316 } 3317 3318 if (!(previousCommand & FastMixerState::IDLE)) { 3319 ALOG_ASSERT(mFastMixer != NULL); 3320 FastMixerStateQueue *sq = mFastMixer->sq(); 3321 FastMixerState *state = sq->begin(); 3322 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3323 state->mCommand = previousCommand; 3324 sq->end(); 3325 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3326 } 3327 3328 return reconfig; 3329} 3330 3331 3332void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3333{ 3334 const size_t SIZE = 256; 3335 char buffer[SIZE]; 3336 String8 result; 3337 3338 PlaybackThread::dumpInternals(fd, args); 3339 3340 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3341 result.append(buffer); 3342 write(fd, result.string(), result.size()); 3343 3344 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3345 const FastMixerDumpState copy(mFastMixerDumpState); 3346 copy.dump(fd); 3347 3348#ifdef STATE_QUEUE_DUMP 3349 // Similar for state queue 3350 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3351 observerCopy.dump(fd); 3352 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3353 mutatorCopy.dump(fd); 3354#endif 3355 3356#ifdef TEE_SINK 3357 // Write the tee output to a .wav file 3358 dumpTee(fd, mTeeSource, mId); 3359#endif 3360 3361#ifdef AUDIO_WATCHDOG 3362 if (mAudioWatchdog != 0) { 3363 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3364 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3365 wdCopy.dump(fd); 3366 } 3367#endif 3368} 3369 3370uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3371{ 3372 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3373} 3374 3375uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3376{ 3377 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3378} 3379 3380void AudioFlinger::MixerThread::cacheParameters_l() 3381{ 3382 PlaybackThread::cacheParameters_l(); 3383 3384 // FIXME: Relaxed timing because of a certain device that can't meet latency 3385 // Should be reduced to 2x after the vendor fixes the driver issue 3386 // increase threshold again due to low power audio mode. The way this warning 3387 // threshold is calculated and its usefulness should be reconsidered anyway. 3388 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3389} 3390 3391// ---------------------------------------------------------------------------- 3392 3393AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3394 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3395 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3396 // mLeftVolFloat, mRightVolFloat 3397{ 3398} 3399 3400AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3401 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3402 ThreadBase::type_t type) 3403 : PlaybackThread(audioFlinger, output, id, device, type) 3404 // mLeftVolFloat, mRightVolFloat 3405{ 3406} 3407 3408AudioFlinger::DirectOutputThread::~DirectOutputThread() 3409{ 3410} 3411 3412void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3413{ 3414 audio_track_cblk_t* cblk = track->cblk(); 3415 float left, right; 3416 3417 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3418 left = right = 0; 3419 } else { 3420 float typeVolume = mStreamTypes[track->streamType()].volume; 3421 float v = mMasterVolume * typeVolume; 3422 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3423 uint32_t vlr = proxy->getVolumeLR(); 3424 float v_clamped = v * (vlr & 0xFFFF); 3425 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3426 left = v_clamped/MAX_GAIN; 3427 v_clamped = v * (vlr >> 16); 3428 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3429 right = v_clamped/MAX_GAIN; 3430 } 3431 3432 if (lastTrack) { 3433 if (left != mLeftVolFloat || right != mRightVolFloat) { 3434 mLeftVolFloat = left; 3435 mRightVolFloat = right; 3436 3437 // Convert volumes from float to 8.24 3438 uint32_t vl = (uint32_t)(left * (1 << 24)); 3439 uint32_t vr = (uint32_t)(right * (1 << 24)); 3440 3441 // Delegate volume control to effect in track effect chain if needed 3442 // only one effect chain can be present on DirectOutputThread, so if 3443 // there is one, the track is connected to it 3444 if (!mEffectChains.isEmpty()) { 3445 mEffectChains[0]->setVolume_l(&vl, &vr); 3446 left = (float)vl / (1 << 24); 3447 right = (float)vr / (1 << 24); 3448 } 3449 if (mOutput->stream->set_volume) { 3450 mOutput->stream->set_volume(mOutput->stream, left, right); 3451 } 3452 } 3453 } 3454} 3455 3456 3457AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3458 Vector< sp<Track> > *tracksToRemove 3459) 3460{ 3461 size_t count = mActiveTracks.size(); 3462 mixer_state mixerStatus = MIXER_IDLE; 3463 3464 // find out which tracks need to be processed 3465 for (size_t i = 0; i < count; i++) { 3466 sp<Track> t = mActiveTracks[i].promote(); 3467 // The track died recently 3468 if (t == 0) { 3469 continue; 3470 } 3471 3472 Track* const track = t.get(); 3473 audio_track_cblk_t* cblk = track->cblk(); 3474 3475 // The first time a track is added we wait 3476 // for all its buffers to be filled before processing it 3477 uint32_t minFrames; 3478 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3479 minFrames = mNormalFrameCount; 3480 } else { 3481 minFrames = 1; 3482 } 3483 // Only consider last track started for volume and mixer state control. 3484 // This is the last entry in mActiveTracks unless a track underruns. 3485 // As we only care about the transition phase between two tracks on a 3486 // direct output, it is not a problem to ignore the underrun case. 3487 bool last = (i == (count - 1)); 3488 3489 if ((track->framesReady() >= minFrames) && track->isReady() && 3490 !track->isPaused() && !track->isTerminated()) 3491 { 3492 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3493 3494 if (track->mFillingUpStatus == Track::FS_FILLED) { 3495 track->mFillingUpStatus = Track::FS_ACTIVE; 3496 mLeftVolFloat = mRightVolFloat = 0; 3497 if (track->mState == TrackBase::RESUMING) { 3498 track->mState = TrackBase::ACTIVE; 3499 } 3500 } 3501 3502 // compute volume for this track 3503 processVolume_l(track, last); 3504 if (last) { 3505 // reset retry count 3506 track->mRetryCount = kMaxTrackRetriesDirect; 3507 mActiveTrack = t; 3508 mixerStatus = MIXER_TRACKS_READY; 3509 } 3510 } else { 3511 // clear effect chain input buffer if the last active track started underruns 3512 // to avoid sending previous audio buffer again to effects 3513 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3514 mEffectChains[0]->clearInputBuffer(); 3515 } 3516 3517 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3518 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3519 track->isStopped() || track->isPaused()) { 3520 // We have consumed all the buffers of this track. 3521 // Remove it from the list of active tracks. 3522 // TODO: implement behavior for compressed audio 3523 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3524 size_t framesWritten = mBytesWritten / mFrameSize; 3525 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3526 if (track->isStopped()) { 3527 track->reset(); 3528 } 3529 tracksToRemove->add(track); 3530 } 3531 } else { 3532 // No buffers for this track. Give it a few chances to 3533 // fill a buffer, then remove it from active list. 3534 // Only consider last track started for mixer state control 3535 if (--(track->mRetryCount) <= 0) { 3536 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3537 tracksToRemove->add(track); 3538 } else if (last) { 3539 mixerStatus = MIXER_TRACKS_ENABLED; 3540 } 3541 } 3542 } 3543 } 3544 3545 // remove all the tracks that need to be... 3546 removeTracks_l(*tracksToRemove); 3547 3548 return mixerStatus; 3549} 3550 3551void AudioFlinger::DirectOutputThread::threadLoop_mix() 3552{ 3553 size_t frameCount = mFrameCount; 3554 int8_t *curBuf = (int8_t *)mMixBuffer; 3555 // output audio to hardware 3556 while (frameCount) { 3557 AudioBufferProvider::Buffer buffer; 3558 buffer.frameCount = frameCount; 3559 mActiveTrack->getNextBuffer(&buffer); 3560 if (buffer.raw == NULL) { 3561 memset(curBuf, 0, frameCount * mFrameSize); 3562 break; 3563 } 3564 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3565 frameCount -= buffer.frameCount; 3566 curBuf += buffer.frameCount * mFrameSize; 3567 mActiveTrack->releaseBuffer(&buffer); 3568 } 3569 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3570 sleepTime = 0; 3571 standbyTime = systemTime() + standbyDelay; 3572 mActiveTrack.clear(); 3573} 3574 3575void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3576{ 3577 if (sleepTime == 0) { 3578 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3579 sleepTime = activeSleepTime; 3580 } else { 3581 sleepTime = idleSleepTime; 3582 } 3583 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3584 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3585 sleepTime = 0; 3586 } 3587} 3588 3589// getTrackName_l() must be called with ThreadBase::mLock held 3590int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3591 int sessionId) 3592{ 3593 return 0; 3594} 3595 3596// deleteTrackName_l() must be called with ThreadBase::mLock held 3597void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3598{ 3599} 3600 3601// checkForNewParameters_l() must be called with ThreadBase::mLock held 3602bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3603{ 3604 bool reconfig = false; 3605 3606 while (!mNewParameters.isEmpty()) { 3607 status_t status = NO_ERROR; 3608 String8 keyValuePair = mNewParameters[0]; 3609 AudioParameter param = AudioParameter(keyValuePair); 3610 int value; 3611 3612 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3613 // do not accept frame count changes if tracks are open as the track buffer 3614 // size depends on frame count and correct behavior would not be garantied 3615 // if frame count is changed after track creation 3616 if (!mTracks.isEmpty()) { 3617 status = INVALID_OPERATION; 3618 } else { 3619 reconfig = true; 3620 } 3621 } 3622 if (status == NO_ERROR) { 3623 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3624 keyValuePair.string()); 3625 if (!mStandby && status == INVALID_OPERATION) { 3626 mOutput->stream->common.standby(&mOutput->stream->common); 3627 mStandby = true; 3628 mBytesWritten = 0; 3629 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3630 keyValuePair.string()); 3631 } 3632 if (status == NO_ERROR && reconfig) { 3633 readOutputParameters(); 3634 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3635 } 3636 } 3637 3638 mNewParameters.removeAt(0); 3639 3640 mParamStatus = status; 3641 mParamCond.signal(); 3642 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3643 // already timed out waiting for the status and will never signal the condition. 3644 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3645 } 3646 return reconfig; 3647} 3648 3649uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3650{ 3651 uint32_t time; 3652 if (audio_is_linear_pcm(mFormat)) { 3653 time = PlaybackThread::activeSleepTimeUs(); 3654 } else { 3655 time = 10000; 3656 } 3657 return time; 3658} 3659 3660uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3661{ 3662 uint32_t time; 3663 if (audio_is_linear_pcm(mFormat)) { 3664 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3665 } else { 3666 time = 10000; 3667 } 3668 return time; 3669} 3670 3671uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3672{ 3673 uint32_t time; 3674 if (audio_is_linear_pcm(mFormat)) { 3675 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3676 } else { 3677 time = 10000; 3678 } 3679 return time; 3680} 3681 3682void AudioFlinger::DirectOutputThread::cacheParameters_l() 3683{ 3684 PlaybackThread::cacheParameters_l(); 3685 3686 // use shorter standby delay as on normal output to release 3687 // hardware resources as soon as possible 3688 standbyDelay = microseconds(activeSleepTime*2); 3689} 3690 3691// ---------------------------------------------------------------------------- 3692 3693AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3694 const sp<AudioFlinger::OffloadThread>& offloadThread) 3695 : Thread(false /*canCallJava*/), 3696 mOffloadThread(offloadThread), 3697 mWriteBlocked(false), 3698 mDraining(false) 3699{ 3700} 3701 3702AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3703{ 3704} 3705 3706void AudioFlinger::AsyncCallbackThread::onFirstRef() 3707{ 3708 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3709} 3710 3711bool AudioFlinger::AsyncCallbackThread::threadLoop() 3712{ 3713 while (!exitPending()) { 3714 bool writeBlocked; 3715 bool draining; 3716 3717 { 3718 Mutex::Autolock _l(mLock); 3719 mWaitWorkCV.wait(mLock); 3720 if (exitPending()) { 3721 break; 3722 } 3723 writeBlocked = mWriteBlocked; 3724 draining = mDraining; 3725 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3726 } 3727 { 3728 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3729 if (offloadThread != 0) { 3730 if (writeBlocked == false) { 3731 offloadThread->setWriteBlocked(false); 3732 } 3733 if (draining == false) { 3734 offloadThread->setDraining(false); 3735 } 3736 } 3737 } 3738 } 3739 return false; 3740} 3741 3742void AudioFlinger::AsyncCallbackThread::exit() 3743{ 3744 ALOGV("AsyncCallbackThread::exit"); 3745 Mutex::Autolock _l(mLock); 3746 requestExit(); 3747 mWaitWorkCV.broadcast(); 3748} 3749 3750void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3751{ 3752 Mutex::Autolock _l(mLock); 3753 mWriteBlocked = value; 3754 if (!value) { 3755 mWaitWorkCV.signal(); 3756 } 3757} 3758 3759void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3760{ 3761 Mutex::Autolock _l(mLock); 3762 mDraining = value; 3763 if (!value) { 3764 mWaitWorkCV.signal(); 3765 } 3766} 3767 3768 3769// ---------------------------------------------------------------------------- 3770AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3771 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3772 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3773 mHwPaused(false), 3774 mPausedBytesRemaining(0) 3775{ 3776 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3777} 3778 3779AudioFlinger::OffloadThread::~OffloadThread() 3780{ 3781 mPreviousTrack.clear(); 3782} 3783 3784void AudioFlinger::OffloadThread::threadLoop_exit() 3785{ 3786 if (mFlushPending || mHwPaused) { 3787 // If a flush is pending or track was paused, just discard buffered data 3788 flushHw_l(); 3789 } else { 3790 mMixerStatus = MIXER_DRAIN_ALL; 3791 threadLoop_drain(); 3792 } 3793 mCallbackThread->exit(); 3794 PlaybackThread::threadLoop_exit(); 3795} 3796 3797AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3798 Vector< sp<Track> > *tracksToRemove 3799) 3800{ 3801 ALOGV("OffloadThread::prepareTracks_l"); 3802 size_t count = mActiveTracks.size(); 3803 3804 mixer_state mixerStatus = MIXER_IDLE; 3805 if (mFlushPending) { 3806 flushHw_l(); 3807 mFlushPending = false; 3808 } 3809 // find out which tracks need to be processed 3810 for (size_t i = 0; i < count; i++) { 3811 sp<Track> t = mActiveTracks[i].promote(); 3812 // The track died recently 3813 if (t == 0) { 3814 continue; 3815 } 3816 Track* const track = t.get(); 3817 audio_track_cblk_t* cblk = track->cblk(); 3818 if (mPreviousTrack != NULL) { 3819 if (t != mPreviousTrack) { 3820 // Flush any data still being written from last track 3821 mBytesRemaining = 0; 3822 if (mPausedBytesRemaining) { 3823 // Last track was paused so we also need to flush saved 3824 // mixbuffer state and invalidate track so that it will 3825 // re-submit that unwritten data when it is next resumed 3826 mPausedBytesRemaining = 0; 3827 // Invalidate is a bit drastic - would be more efficient 3828 // to have a flag to tell client that some of the 3829 // previously written data was lost 3830 mPreviousTrack->invalidate(); 3831 } 3832 } 3833 } 3834 mPreviousTrack = t; 3835 bool last = (i == (count - 1)); 3836 if (track->isPausing()) { 3837 track->setPaused(); 3838 if (last) { 3839 if (!mHwPaused) { 3840 mOutput->stream->pause(mOutput->stream); 3841 mHwPaused = true; 3842 } 3843 // If we were part way through writing the mixbuffer to 3844 // the HAL we must save this until we resume 3845 // BUG - this will be wrong if a different track is made active, 3846 // in that case we want to discard the pending data in the 3847 // mixbuffer and tell the client to present it again when the 3848 // track is resumed 3849 mPausedWriteLength = mCurrentWriteLength; 3850 mPausedBytesRemaining = mBytesRemaining; 3851 mBytesRemaining = 0; // stop writing 3852 } 3853 tracksToRemove->add(track); 3854 } else if (track->framesReady() && track->isReady() && 3855 !track->isPaused() && !track->isTerminated()) { 3856 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3857 if (track->mFillingUpStatus == Track::FS_FILLED) { 3858 track->mFillingUpStatus = Track::FS_ACTIVE; 3859 mLeftVolFloat = mRightVolFloat = 0; 3860 if (track->mState == TrackBase::RESUMING) { 3861 if (mPausedBytesRemaining) { 3862 // Need to continue write that was interrupted 3863 mCurrentWriteLength = mPausedWriteLength; 3864 mBytesRemaining = mPausedBytesRemaining; 3865 mPausedBytesRemaining = 0; 3866 } 3867 track->mState = TrackBase::ACTIVE; 3868 } 3869 } 3870 3871 if (last) { 3872 if (mHwPaused) { 3873 mOutput->stream->resume(mOutput->stream); 3874 mHwPaused = false; 3875 // threadLoop_mix() will handle the case that we need to 3876 // resume an interrupted write 3877 } 3878 // reset retry count 3879 track->mRetryCount = kMaxTrackRetriesOffload; 3880 mActiveTrack = t; 3881 mixerStatus = MIXER_TRACKS_READY; 3882 } 3883 } else { 3884 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3885 if (track->isStopping_1()) { 3886 // Hardware buffer can hold a large amount of audio so we must 3887 // wait for all current track's data to drain before we say 3888 // that the track is stopped. 3889 if (mBytesRemaining == 0) { 3890 // Only start draining when all data in mixbuffer 3891 // has been written 3892 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3893 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3894 sleepTime = 0; 3895 standbyTime = systemTime() + standbyDelay; 3896 if (last) { 3897 mixerStatus = MIXER_DRAIN_TRACK; 3898 if (mHwPaused) { 3899 // It is possible to move from PAUSED to STOPPING_1 without 3900 // a resume so we must ensure hardware is running 3901 mOutput->stream->resume(mOutput->stream); 3902 mHwPaused = false; 3903 } 3904 } 3905 } 3906 } else if (track->isStopping_2()) { 3907 // Drain has completed, signal presentation complete 3908 if (!mDraining || !last) { 3909 track->mState = TrackBase::STOPPED; 3910 size_t audioHALFrames = 3911 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3912 size_t framesWritten = 3913 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3914 track->presentationComplete(framesWritten, audioHALFrames); 3915 track->reset(); 3916 tracksToRemove->add(track); 3917 } 3918 } else { 3919 // No buffers for this track. Give it a few chances to 3920 // fill a buffer, then remove it from active list. 3921 if (--(track->mRetryCount) <= 0) { 3922 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3923 track->name()); 3924 tracksToRemove->add(track); 3925 } else if (last){ 3926 mixerStatus = MIXER_TRACKS_ENABLED; 3927 } 3928 } 3929 } 3930 // compute volume for this track 3931 processVolume_l(track, last); 3932 } 3933 // remove all the tracks that need to be... 3934 removeTracks_l(*tracksToRemove); 3935 3936 return mixerStatus; 3937} 3938 3939void AudioFlinger::OffloadThread::flushOutput_l() 3940{ 3941 mFlushPending = true; 3942} 3943 3944// must be called with thread mutex locked 3945bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3946{ 3947 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3948 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3949 return true; 3950 } 3951 return false; 3952} 3953 3954// must be called with thread mutex locked 3955bool AudioFlinger::OffloadThread::shouldStandby_l() 3956{ 3957 bool TrackPaused = false; 3958 3959 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3960 // after a timeout and we will enter standby then. 3961 if (mTracks.size() > 0) { 3962 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3963 } 3964 3965 return !mStandby && !TrackPaused; 3966} 3967 3968 3969bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3970{ 3971 Mutex::Autolock _l(mLock); 3972 return waitingAsyncCallback_l(); 3973} 3974 3975void AudioFlinger::OffloadThread::flushHw_l() 3976{ 3977 mOutput->stream->flush(mOutput->stream); 3978 // Flush anything still waiting in the mixbuffer 3979 mCurrentWriteLength = 0; 3980 mBytesRemaining = 0; 3981 mPausedWriteLength = 0; 3982 mPausedBytesRemaining = 0; 3983 if (mUseAsyncWrite) { 3984 mWriteBlocked = false; 3985 mDraining = false; 3986 ALOG_ASSERT(mCallbackThread != 0); 3987 mCallbackThread->setWriteBlocked(false); 3988 mCallbackThread->setDraining(false); 3989 } 3990} 3991 3992// ---------------------------------------------------------------------------- 3993 3994AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3995 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3996 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3997 DUPLICATING), 3998 mWaitTimeMs(UINT_MAX) 3999{ 4000 addOutputTrack(mainThread); 4001} 4002 4003AudioFlinger::DuplicatingThread::~DuplicatingThread() 4004{ 4005 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4006 mOutputTracks[i]->destroy(); 4007 } 4008} 4009 4010void AudioFlinger::DuplicatingThread::threadLoop_mix() 4011{ 4012 // mix buffers... 4013 if (outputsReady(outputTracks)) { 4014 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4015 } else { 4016 memset(mMixBuffer, 0, mixBufferSize); 4017 } 4018 sleepTime = 0; 4019 writeFrames = mNormalFrameCount; 4020 mCurrentWriteLength = mixBufferSize; 4021 standbyTime = systemTime() + standbyDelay; 4022} 4023 4024void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4025{ 4026 if (sleepTime == 0) { 4027 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4028 sleepTime = activeSleepTime; 4029 } else { 4030 sleepTime = idleSleepTime; 4031 } 4032 } else if (mBytesWritten != 0) { 4033 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4034 writeFrames = mNormalFrameCount; 4035 memset(mMixBuffer, 0, mixBufferSize); 4036 } else { 4037 // flush remaining overflow buffers in output tracks 4038 writeFrames = 0; 4039 } 4040 sleepTime = 0; 4041 } 4042} 4043 4044ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4045{ 4046 for (size_t i = 0; i < outputTracks.size(); i++) { 4047 outputTracks[i]->write(mMixBuffer, writeFrames); 4048 } 4049 return (ssize_t)mixBufferSize; 4050} 4051 4052void AudioFlinger::DuplicatingThread::threadLoop_standby() 4053{ 4054 // DuplicatingThread implements standby by stopping all tracks 4055 for (size_t i = 0; i < outputTracks.size(); i++) { 4056 outputTracks[i]->stop(); 4057 } 4058} 4059 4060void AudioFlinger::DuplicatingThread::saveOutputTracks() 4061{ 4062 outputTracks = mOutputTracks; 4063} 4064 4065void AudioFlinger::DuplicatingThread::clearOutputTracks() 4066{ 4067 outputTracks.clear(); 4068} 4069 4070void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4071{ 4072 Mutex::Autolock _l(mLock); 4073 // FIXME explain this formula 4074 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4075 OutputTrack *outputTrack = new OutputTrack(thread, 4076 this, 4077 mSampleRate, 4078 mFormat, 4079 mChannelMask, 4080 frameCount); 4081 if (outputTrack->cblk() != NULL) { 4082 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4083 mOutputTracks.add(outputTrack); 4084 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4085 updateWaitTime_l(); 4086 } 4087} 4088 4089void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4090{ 4091 Mutex::Autolock _l(mLock); 4092 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4093 if (mOutputTracks[i]->thread() == thread) { 4094 mOutputTracks[i]->destroy(); 4095 mOutputTracks.removeAt(i); 4096 updateWaitTime_l(); 4097 return; 4098 } 4099 } 4100 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4101} 4102 4103// caller must hold mLock 4104void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4105{ 4106 mWaitTimeMs = UINT_MAX; 4107 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4108 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4109 if (strong != 0) { 4110 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4111 if (waitTimeMs < mWaitTimeMs) { 4112 mWaitTimeMs = waitTimeMs; 4113 } 4114 } 4115 } 4116} 4117 4118 4119bool AudioFlinger::DuplicatingThread::outputsReady( 4120 const SortedVector< sp<OutputTrack> > &outputTracks) 4121{ 4122 for (size_t i = 0; i < outputTracks.size(); i++) { 4123 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4124 if (thread == 0) { 4125 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4126 outputTracks[i].get()); 4127 return false; 4128 } 4129 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4130 // see note at standby() declaration 4131 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4132 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4133 thread.get()); 4134 return false; 4135 } 4136 } 4137 return true; 4138} 4139 4140uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4141{ 4142 return (mWaitTimeMs * 1000) / 2; 4143} 4144 4145void AudioFlinger::DuplicatingThread::cacheParameters_l() 4146{ 4147 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4148 updateWaitTime_l(); 4149 4150 MixerThread::cacheParameters_l(); 4151} 4152 4153// ---------------------------------------------------------------------------- 4154// Record 4155// ---------------------------------------------------------------------------- 4156 4157AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4158 AudioStreamIn *input, 4159 uint32_t sampleRate, 4160 audio_channel_mask_t channelMask, 4161 audio_io_handle_t id, 4162 audio_devices_t outDevice, 4163 audio_devices_t inDevice 4164#ifdef TEE_SINK 4165 , const sp<NBAIO_Sink>& teeSink 4166#endif 4167 ) : 4168 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4169 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4170 // mRsmpInIndex set by readInputParameters() 4171 mReqChannelCount(popcount(channelMask)), 4172 mReqSampleRate(sampleRate) 4173 // mBytesRead is only meaningful while active, and so is cleared in start() 4174 // (but might be better to also clear here for dump?) 4175#ifdef TEE_SINK 4176 , mTeeSink(teeSink) 4177#endif 4178{ 4179 snprintf(mName, kNameLength, "AudioIn_%X", id); 4180 4181 readInputParameters(); 4182 4183} 4184 4185 4186AudioFlinger::RecordThread::~RecordThread() 4187{ 4188 delete[] mRsmpInBuffer; 4189 delete mResampler; 4190 delete[] mRsmpOutBuffer; 4191} 4192 4193void AudioFlinger::RecordThread::onFirstRef() 4194{ 4195 run(mName, PRIORITY_URGENT_AUDIO); 4196} 4197 4198bool AudioFlinger::RecordThread::threadLoop() 4199{ 4200 AudioBufferProvider::Buffer buffer; 4201 sp<RecordTrack> activeTrack; 4202 4203 nsecs_t lastWarning = 0; 4204 4205 inputStandBy(); 4206 acquireWakeLock(); 4207 4208 // used to verify we've read at least once before evaluating how many bytes were read 4209 bool readOnce = false; 4210 4211 // start recording 4212 // FIXME Race here: exitPending could become true immediately after testing. 4213 // It is only set to true while mLock held, but we don't hold mLock yet. 4214 // Probably a benign race, but it would be safer to check exitPending with mLock held. 4215 while (!exitPending()) { 4216 4217 processConfigEvents(); 4218 4219 Vector< sp<EffectChain> > effectChains; 4220 { // scope for mLock 4221 Mutex::Autolock _l(mLock); 4222 checkForNewParameters_l(); 4223 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4224 standby(); 4225 4226 if (exitPending()) { 4227 break; 4228 } 4229 4230 releaseWakeLock_l(); 4231 ALOGV("RecordThread: loop stopping"); 4232 // go to sleep 4233 mWaitWorkCV.wait(mLock); 4234 ALOGV("RecordThread: loop starting"); 4235 acquireWakeLock_l(); 4236 continue; 4237 } 4238 if (mActiveTrack != 0) { 4239 if (mActiveTrack->isTerminated()) { 4240 removeTrack_l(mActiveTrack); 4241 mActiveTrack.clear(); 4242 } else { 4243 switch (mActiveTrack->mState) { 4244 case TrackBase::PAUSING: 4245 standby(); 4246 mActiveTrack.clear(); 4247 mStartStopCond.broadcast(); 4248 break; 4249 4250 case TrackBase::RESUMING: 4251 if (mReqChannelCount != mActiveTrack->channelCount()) { 4252 mActiveTrack.clear(); 4253 mStartStopCond.broadcast(); 4254 } else if (readOnce) { 4255 // record start succeeds only if first read from audio input 4256 // succeeds 4257 if (mBytesRead >= 0) { 4258 mActiveTrack->mState = TrackBase::ACTIVE; 4259 } else { 4260 mActiveTrack.clear(); 4261 } 4262 mStartStopCond.broadcast(); 4263 } 4264 mStandby = false; 4265 break; 4266 4267 case TrackBase::ACTIVE: 4268 break; 4269 4270 case TrackBase::IDLE: 4271 break; 4272 4273 default: 4274 LOG_FATAL("Unexpected mActiveTrack->mState %d", mActiveTrack->mState); 4275 } 4276 4277 } 4278 } 4279 lockEffectChains_l(effectChains); 4280 } 4281 4282 // thread mutex is now unlocked 4283 // FIXME RecordThread::start assigns to mActiveTrack under lock, but we read without lock 4284 if (mActiveTrack != 0) { 4285 // FIXME RecordThread::stop assigns to mState under lock, but we read without lock 4286 if (mActiveTrack->mState != TrackBase::ACTIVE && 4287 mActiveTrack->mState != TrackBase::RESUMING) { 4288 unlockEffectChains(effectChains); 4289 usleep(kRecordThreadSleepUs); 4290 continue; 4291 } 4292 for (size_t i = 0; i < effectChains.size(); i ++) { 4293 // thread mutex is not locked, but effect chain is locked 4294 effectChains[i]->process_l(); 4295 } 4296 4297 buffer.frameCount = mFrameCount; 4298 status_t status = mActiveTrack->getNextBuffer(&buffer); 4299 if (status == NO_ERROR) { 4300 readOnce = true; 4301 size_t framesOut = buffer.frameCount; 4302 if (mResampler == NULL) { 4303 // no resampling 4304 while (framesOut) { 4305 size_t framesIn = mFrameCount - mRsmpInIndex; 4306 if (framesIn > 0) { 4307 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4308 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4309 mActiveTrack->mFrameSize; 4310 if (framesIn > framesOut) { 4311 framesIn = framesOut; 4312 } 4313 mRsmpInIndex += framesIn; 4314 framesOut -= framesIn; 4315 if (mChannelCount == mReqChannelCount) { 4316 memcpy(dst, src, framesIn * mFrameSize); 4317 } else { 4318 if (mChannelCount == 1) { 4319 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4320 (int16_t *)src, framesIn); 4321 } else { 4322 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4323 (int16_t *)src, framesIn); 4324 } 4325 } 4326 } 4327 if (framesOut > 0 && mFrameCount == mRsmpInIndex) { 4328 void *readInto; 4329 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4330 readInto = buffer.raw; 4331 framesOut = 0; 4332 } else { 4333 readInto = mRsmpInBuffer; 4334 mRsmpInIndex = 0; 4335 } 4336 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4337 mBufferSize); 4338 if (mBytesRead <= 0) { 4339 // FIXME read mState without lock 4340 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4341 { 4342 ALOGE("Error reading audio input"); 4343 // Force input into standby so that it tries to 4344 // recover at next read attempt 4345 inputStandBy(); 4346 // FIXME sleep with effect chains locked 4347 usleep(kRecordThreadSleepUs); 4348 } 4349 mRsmpInIndex = mFrameCount; 4350 framesOut = 0; 4351 buffer.frameCount = 0; 4352 } 4353#ifdef TEE_SINK 4354 else if (mTeeSink != 0) { 4355 (void) mTeeSink->write(readInto, 4356 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4357 } 4358#endif 4359 } 4360 } 4361 } else { 4362 // resampling 4363 4364 // resampler accumulates, but we only have one source track 4365 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4366 // alter output frame count as if we were expecting stereo samples 4367 if (mChannelCount == 1 && mReqChannelCount == 1) { 4368 framesOut >>= 1; 4369 } 4370 mResampler->resample(mRsmpOutBuffer, framesOut, 4371 this /* AudioBufferProvider* */); 4372 // ditherAndClamp() works as long as all buffers returned by 4373 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4374 if (mChannelCount == 2 && mReqChannelCount == 1) { 4375 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4376 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4377 // the resampler always outputs stereo samples: 4378 // do post stereo to mono conversion 4379 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4380 framesOut); 4381 } else { 4382 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4383 } 4384 // now done with mRsmpOutBuffer 4385 4386 } 4387 if (mFramestoDrop == 0) { 4388 mActiveTrack->releaseBuffer(&buffer); 4389 } else { 4390 if (mFramestoDrop > 0) { 4391 mFramestoDrop -= buffer.frameCount; 4392 if (mFramestoDrop <= 0) { 4393 clearSyncStartEvent(); 4394 } 4395 } else { 4396 mFramestoDrop += buffer.frameCount; 4397 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4398 mSyncStartEvent->isCancelled()) { 4399 ALOGW("Synced record %s, session %d, trigger session %d", 4400 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4401 mActiveTrack->sessionId(), 4402 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4403 clearSyncStartEvent(); 4404 } 4405 } 4406 } 4407 mActiveTrack->clearOverflow(); 4408 } 4409 // client isn't retrieving buffers fast enough 4410 else { 4411 if (!mActiveTrack->setOverflow()) { 4412 nsecs_t now = systemTime(); 4413 if ((now - lastWarning) > kWarningThrottleNs) { 4414 ALOGW("RecordThread: buffer overflow"); 4415 lastWarning = now; 4416 } 4417 } 4418 // Release the processor for a while before asking for a new buffer. 4419 // This will give the application more chance to read from the buffer and 4420 // clear the overflow. 4421 // FIXME sleep with effect chains locked 4422 usleep(kRecordThreadSleepUs); 4423 } 4424 } 4425 // enable changes in effect chain 4426 unlockEffectChains(effectChains); 4427 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4428 } 4429 4430 standby(); 4431 4432 { 4433 Mutex::Autolock _l(mLock); 4434 mActiveTrack.clear(); 4435 mStartStopCond.broadcast(); 4436 } 4437 4438 releaseWakeLock(); 4439 4440 ALOGV("RecordThread %p exiting", this); 4441 return false; 4442} 4443 4444void AudioFlinger::RecordThread::standby() 4445{ 4446 if (!mStandby) { 4447 inputStandBy(); 4448 mStandby = true; 4449 } 4450} 4451 4452void AudioFlinger::RecordThread::inputStandBy() 4453{ 4454 mInput->stream->common.standby(&mInput->stream->common); 4455} 4456 4457sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4458 const sp<AudioFlinger::Client>& client, 4459 uint32_t sampleRate, 4460 audio_format_t format, 4461 audio_channel_mask_t channelMask, 4462 size_t frameCount, 4463 int sessionId, 4464 IAudioFlinger::track_flags_t *flags, 4465 pid_t tid, 4466 status_t *status) 4467{ 4468 sp<RecordTrack> track; 4469 status_t lStatus; 4470 4471 lStatus = initCheck(); 4472 if (lStatus != NO_ERROR) { 4473 ALOGE("Audio driver not initialized."); 4474 goto Exit; 4475 } 4476 4477 // client expresses a preference for FAST, but we get the final say 4478 if (*flags & IAudioFlinger::TRACK_FAST) { 4479 if ( 4480 // use case: callback handler and frame count is default or at least as large as HAL 4481 ( 4482 (tid != -1) && 4483 ((frameCount == 0) || 4484 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4485 ) && 4486 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4487 // mono or stereo 4488 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4489 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4490 // hardware sample rate 4491 (sampleRate == mSampleRate) && 4492 // record thread has an associated fast recorder 4493 hasFastRecorder() 4494 // FIXME test that RecordThread for this fast track has a capable output HAL 4495 // FIXME add a permission test also? 4496 ) { 4497 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4498 if (frameCount == 0) { 4499 frameCount = mFrameCount * kFastTrackMultiplier; 4500 } 4501 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4502 frameCount, mFrameCount); 4503 } else { 4504 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4505 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4506 "hasFastRecorder=%d tid=%d", 4507 frameCount, mFrameCount, format, 4508 audio_is_linear_pcm(format), 4509 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4510 *flags &= ~IAudioFlinger::TRACK_FAST; 4511 // For compatibility with AudioRecord calculation, buffer depth is forced 4512 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4513 // This is probably too conservative, but legacy application code may depend on it. 4514 // If you change this calculation, also review the start threshold which is related. 4515 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4516 size_t mNormalFrameCount = 2048; // FIXME 4517 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4518 if (minBufCount < 2) { 4519 minBufCount = 2; 4520 } 4521 size_t minFrameCount = mNormalFrameCount * minBufCount; 4522 if (frameCount < minFrameCount) { 4523 frameCount = minFrameCount; 4524 } 4525 } 4526 } 4527 4528 // FIXME use flags and tid similar to createTrack_l() 4529 4530 { // scope for mLock 4531 Mutex::Autolock _l(mLock); 4532 4533 track = new RecordTrack(this, client, sampleRate, 4534 format, channelMask, frameCount, sessionId); 4535 4536 lStatus = track->initCheck(); 4537 if (lStatus != NO_ERROR) { 4538 track.clear(); 4539 goto Exit; 4540 } 4541 mTracks.add(track); 4542 4543 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4544 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4545 mAudioFlinger->btNrecIsOff(); 4546 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4547 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4548 4549 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4550 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4551 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4552 // so ask activity manager to do this on our behalf 4553 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4554 } 4555 } 4556 lStatus = NO_ERROR; 4557 4558Exit: 4559 *status = lStatus; 4560 return track; 4561} 4562 4563status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4564 AudioSystem::sync_event_t event, 4565 int triggerSession) 4566{ 4567 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4568 sp<ThreadBase> strongMe = this; 4569 status_t status = NO_ERROR; 4570 4571 if (event == AudioSystem::SYNC_EVENT_NONE) { 4572 clearSyncStartEvent(); 4573 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4574 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4575 triggerSession, 4576 recordTrack->sessionId(), 4577 syncStartEventCallback, 4578 this); 4579 // Sync event can be cancelled by the trigger session if the track is not in a 4580 // compatible state in which case we start record immediately 4581 if (mSyncStartEvent->isCancelled()) { 4582 clearSyncStartEvent(); 4583 } else { 4584 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4585 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4586 } 4587 } 4588 4589 { 4590 // This section is a rendezvous between binder thread executing start() and RecordThread 4591 AutoMutex lock(mLock); 4592 if (mActiveTrack != 0) { 4593 if (recordTrack != mActiveTrack.get()) { 4594 status = -EBUSY; 4595 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4596 mActiveTrack->mState = TrackBase::ACTIVE; 4597 } 4598 return status; 4599 } 4600 4601 // FIXME why? already set in constructor, 'STARTING_1' would be more accurate 4602 recordTrack->mState = TrackBase::IDLE; 4603 mActiveTrack = recordTrack; 4604 mLock.unlock(); 4605 status_t status = AudioSystem::startInput(mId); 4606 mLock.lock(); 4607 // FIXME should verify that mActiveTrack is still == recordTrack 4608 if (status != NO_ERROR) { 4609 mActiveTrack.clear(); 4610 clearSyncStartEvent(); 4611 return status; 4612 } 4613 mRsmpInIndex = mFrameCount; 4614 mBytesRead = 0; 4615 if (mResampler != NULL) { 4616 mResampler->reset(); 4617 } 4618 // FIXME hijacking a playback track state name which was intended for start after pause; 4619 // here 'STARTING_2' would be more accurate 4620 mActiveTrack->mState = TrackBase::RESUMING; 4621 // signal thread to start 4622 ALOGV("Signal record thread"); 4623 mWaitWorkCV.broadcast(); 4624 // do not wait for mStartStopCond if exiting 4625 if (exitPending()) { 4626 mActiveTrack.clear(); 4627 status = INVALID_OPERATION; 4628 goto startError; 4629 } 4630 // FIXME incorrect usage of wait: no explicit predicate or loop 4631 mStartStopCond.wait(mLock); 4632 if (mActiveTrack == 0) { 4633 ALOGV("Record failed to start"); 4634 status = BAD_VALUE; 4635 goto startError; 4636 } 4637 ALOGV("Record started OK"); 4638 return status; 4639 } 4640 4641startError: 4642 AudioSystem::stopInput(mId); 4643 clearSyncStartEvent(); 4644 return status; 4645} 4646 4647void AudioFlinger::RecordThread::clearSyncStartEvent() 4648{ 4649 if (mSyncStartEvent != 0) { 4650 mSyncStartEvent->cancel(); 4651 } 4652 mSyncStartEvent.clear(); 4653 mFramestoDrop = 0; 4654} 4655 4656void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4657{ 4658 sp<SyncEvent> strongEvent = event.promote(); 4659 4660 if (strongEvent != 0) { 4661 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4662 me->handleSyncStartEvent(strongEvent); 4663 } 4664} 4665 4666void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4667{ 4668 if (event == mSyncStartEvent) { 4669 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4670 // from audio HAL 4671 mFramestoDrop = mFrameCount * 2; 4672 } 4673} 4674 4675bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4676 ALOGV("RecordThread::stop"); 4677 AutoMutex _l(mLock); 4678 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4679 return false; 4680 } 4681 // note that threadLoop may still be processing the track at this point [without lock] 4682 recordTrack->mState = TrackBase::PAUSING; 4683 // do not wait for mStartStopCond if exiting 4684 if (exitPending()) { 4685 return true; 4686 } 4687 // FIXME incorrect usage of wait: no explicit predicate or loop 4688 mStartStopCond.wait(mLock); 4689 // if we have been restarted, recordTrack == mActiveTrack.get() here 4690 if (exitPending() || recordTrack != mActiveTrack.get()) { 4691 ALOGV("Record stopped OK"); 4692 return true; 4693 } 4694 return false; 4695} 4696 4697bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4698{ 4699 return false; 4700} 4701 4702status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4703{ 4704#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4705 if (!isValidSyncEvent(event)) { 4706 return BAD_VALUE; 4707 } 4708 4709 int eventSession = event->triggerSession(); 4710 status_t ret = NAME_NOT_FOUND; 4711 4712 Mutex::Autolock _l(mLock); 4713 4714 for (size_t i = 0; i < mTracks.size(); i++) { 4715 sp<RecordTrack> track = mTracks[i]; 4716 if (eventSession == track->sessionId()) { 4717 (void) track->setSyncEvent(event); 4718 ret = NO_ERROR; 4719 } 4720 } 4721 return ret; 4722#else 4723 return BAD_VALUE; 4724#endif 4725} 4726 4727// destroyTrack_l() must be called with ThreadBase::mLock held 4728void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4729{ 4730 track->terminate(); 4731 track->mState = TrackBase::STOPPED; 4732 // active tracks are removed by threadLoop() 4733 if (mActiveTrack != track) { 4734 removeTrack_l(track); 4735 } 4736} 4737 4738void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4739{ 4740 mTracks.remove(track); 4741 // need anything related to effects here? 4742} 4743 4744void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4745{ 4746 dumpInternals(fd, args); 4747 dumpTracks(fd, args); 4748 dumpEffectChains(fd, args); 4749} 4750 4751void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4752{ 4753 const size_t SIZE = 256; 4754 char buffer[SIZE]; 4755 String8 result; 4756 4757 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4758 result.append(buffer); 4759 4760 if (mActiveTrack != 0) { 4761 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4762 result.append(buffer); 4763 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4764 result.append(buffer); 4765 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4766 result.append(buffer); 4767 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4768 result.append(buffer); 4769 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4770 result.append(buffer); 4771 } else { 4772 result.append("No active record client\n"); 4773 } 4774 4775 write(fd, result.string(), result.size()); 4776 4777 dumpBase(fd, args); 4778} 4779 4780void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4781{ 4782 const size_t SIZE = 256; 4783 char buffer[SIZE]; 4784 String8 result; 4785 4786 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4787 result.append(buffer); 4788 RecordTrack::appendDumpHeader(result); 4789 for (size_t i = 0; i < mTracks.size(); ++i) { 4790 sp<RecordTrack> track = mTracks[i]; 4791 if (track != 0) { 4792 track->dump(buffer, SIZE); 4793 result.append(buffer); 4794 } 4795 } 4796 4797 if (mActiveTrack != 0) { 4798 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4799 result.append(buffer); 4800 RecordTrack::appendDumpHeader(result); 4801 mActiveTrack->dump(buffer, SIZE); 4802 result.append(buffer); 4803 4804 } 4805 write(fd, result.string(), result.size()); 4806} 4807 4808// AudioBufferProvider interface 4809status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4810{ 4811 size_t framesReq = buffer->frameCount; 4812 size_t framesReady = mFrameCount - mRsmpInIndex; 4813 int channelCount; 4814 4815 if (framesReady == 0) { 4816 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4817 if (mBytesRead <= 0) { 4818 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4819 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4820 // Force input into standby so that it tries to 4821 // recover at next read attempt 4822 inputStandBy(); 4823 usleep(kRecordThreadSleepUs); 4824 } 4825 buffer->raw = NULL; 4826 buffer->frameCount = 0; 4827 return NOT_ENOUGH_DATA; 4828 } 4829 mRsmpInIndex = 0; 4830 framesReady = mFrameCount; 4831 } 4832 4833 if (framesReq > framesReady) { 4834 framesReq = framesReady; 4835 } 4836 4837 if (mChannelCount == 1 && mReqChannelCount == 2) { 4838 channelCount = 1; 4839 } else { 4840 channelCount = 2; 4841 } 4842 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4843 buffer->frameCount = framesReq; 4844 return NO_ERROR; 4845} 4846 4847// AudioBufferProvider interface 4848void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4849{ 4850 mRsmpInIndex += buffer->frameCount; 4851 buffer->frameCount = 0; 4852} 4853 4854bool AudioFlinger::RecordThread::checkForNewParameters_l() 4855{ 4856 bool reconfig = false; 4857 4858 while (!mNewParameters.isEmpty()) { 4859 status_t status = NO_ERROR; 4860 String8 keyValuePair = mNewParameters[0]; 4861 AudioParameter param = AudioParameter(keyValuePair); 4862 int value; 4863 audio_format_t reqFormat = mFormat; 4864 uint32_t reqSamplingRate = mReqSampleRate; 4865 audio_channel_mask_t reqChannelMask = audio_channel_in_mask_from_count(mReqChannelCount); 4866 4867 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4868 reqSamplingRate = value; 4869 reconfig = true; 4870 } 4871 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4872 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4873 status = BAD_VALUE; 4874 } else { 4875 reqFormat = (audio_format_t) value; 4876 reconfig = true; 4877 } 4878 } 4879 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4880 audio_channel_mask_t mask = (audio_channel_mask_t) value; 4881 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 4882 status = BAD_VALUE; 4883 } else { 4884 reqChannelMask = mask; 4885 reconfig = true; 4886 } 4887 } 4888 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4889 // do not accept frame count changes if tracks are open as the track buffer 4890 // size depends on frame count and correct behavior would not be guaranteed 4891 // if frame count is changed after track creation 4892 if (mActiveTrack != 0) { 4893 status = INVALID_OPERATION; 4894 } else { 4895 reconfig = true; 4896 } 4897 } 4898 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4899 // forward device change to effects that have requested to be 4900 // aware of attached audio device. 4901 for (size_t i = 0; i < mEffectChains.size(); i++) { 4902 mEffectChains[i]->setDevice_l(value); 4903 } 4904 4905 // store input device and output device but do not forward output device to audio HAL. 4906 // Note that status is ignored by the caller for output device 4907 // (see AudioFlinger::setParameters() 4908 if (audio_is_output_devices(value)) { 4909 mOutDevice = value; 4910 status = BAD_VALUE; 4911 } else { 4912 mInDevice = value; 4913 // disable AEC and NS if the device is a BT SCO headset supporting those 4914 // pre processings 4915 if (mTracks.size() > 0) { 4916 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4917 mAudioFlinger->btNrecIsOff(); 4918 for (size_t i = 0; i < mTracks.size(); i++) { 4919 sp<RecordTrack> track = mTracks[i]; 4920 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4921 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4922 } 4923 } 4924 } 4925 } 4926 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4927 mAudioSource != (audio_source_t)value) { 4928 // forward device change to effects that have requested to be 4929 // aware of attached audio device. 4930 for (size_t i = 0; i < mEffectChains.size(); i++) { 4931 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4932 } 4933 mAudioSource = (audio_source_t)value; 4934 } 4935 4936 if (status == NO_ERROR) { 4937 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4938 keyValuePair.string()); 4939 if (status == INVALID_OPERATION) { 4940 inputStandBy(); 4941 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4942 keyValuePair.string()); 4943 } 4944 if (reconfig) { 4945 if (status == BAD_VALUE && 4946 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4947 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4948 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4949 <= (2 * reqSamplingRate)) && 4950 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4951 <= FCC_2 && 4952 (reqChannelMask == AUDIO_CHANNEL_IN_MONO || 4953 reqChannelMask == AUDIO_CHANNEL_IN_STEREO)) { 4954 status = NO_ERROR; 4955 } 4956 if (status == NO_ERROR) { 4957 readInputParameters(); 4958 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4959 } 4960 } 4961 } 4962 4963 mNewParameters.removeAt(0); 4964 4965 mParamStatus = status; 4966 mParamCond.signal(); 4967 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4968 // already timed out waiting for the status and will never signal the condition. 4969 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4970 } 4971 return reconfig; 4972} 4973 4974String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4975{ 4976 Mutex::Autolock _l(mLock); 4977 if (initCheck() != NO_ERROR) { 4978 return String8(); 4979 } 4980 4981 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4982 const String8 out_s8(s); 4983 free(s); 4984 return out_s8; 4985} 4986 4987void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4988 AudioSystem::OutputDescriptor desc; 4989 void *param2 = NULL; 4990 4991 switch (event) { 4992 case AudioSystem::INPUT_OPENED: 4993 case AudioSystem::INPUT_CONFIG_CHANGED: 4994 desc.channelMask = mChannelMask; 4995 desc.samplingRate = mSampleRate; 4996 desc.format = mFormat; 4997 desc.frameCount = mFrameCount; 4998 desc.latency = 0; 4999 param2 = &desc; 5000 break; 5001 5002 case AudioSystem::INPUT_CLOSED: 5003 default: 5004 break; 5005 } 5006 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5007} 5008 5009void AudioFlinger::RecordThread::readInputParameters() 5010{ 5011 delete[] mRsmpInBuffer; 5012 // mRsmpInBuffer is always assigned a new[] below 5013 delete[] mRsmpOutBuffer; 5014 mRsmpOutBuffer = NULL; 5015 delete mResampler; 5016 mResampler = NULL; 5017 5018 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5019 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5020 mChannelCount = popcount(mChannelMask); 5021 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5022 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5023 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5024 } 5025 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5026 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5027 mFrameCount = mBufferSize / mFrameSize; 5028 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5029 5030 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) { 5031 int channelCount; 5032 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5033 // stereo to mono post process as the resampler always outputs stereo. 5034 if (mChannelCount == 1 && mReqChannelCount == 2) { 5035 channelCount = 1; 5036 } else { 5037 channelCount = 2; 5038 } 5039 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5040 mResampler->setSampleRate(mSampleRate); 5041 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5042 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 5043 5044 // optmization: if mono to mono, alter input frame count as if we were inputing 5045 // stereo samples 5046 if (mChannelCount == 1 && mReqChannelCount == 1) { 5047 mFrameCount >>= 1; 5048 } 5049 5050 } 5051 mRsmpInIndex = mFrameCount; 5052} 5053 5054unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5055{ 5056 Mutex::Autolock _l(mLock); 5057 if (initCheck() != NO_ERROR) { 5058 return 0; 5059 } 5060 5061 return mInput->stream->get_input_frames_lost(mInput->stream); 5062} 5063 5064uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5065{ 5066 Mutex::Autolock _l(mLock); 5067 uint32_t result = 0; 5068 if (getEffectChain_l(sessionId) != 0) { 5069 result = EFFECT_SESSION; 5070 } 5071 5072 for (size_t i = 0; i < mTracks.size(); ++i) { 5073 if (sessionId == mTracks[i]->sessionId()) { 5074 result |= TRACK_SESSION; 5075 break; 5076 } 5077 } 5078 5079 return result; 5080} 5081 5082KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5083{ 5084 KeyedVector<int, bool> ids; 5085 Mutex::Autolock _l(mLock); 5086 for (size_t j = 0; j < mTracks.size(); ++j) { 5087 sp<RecordThread::RecordTrack> track = mTracks[j]; 5088 int sessionId = track->sessionId(); 5089 if (ids.indexOfKey(sessionId) < 0) { 5090 ids.add(sessionId, true); 5091 } 5092 } 5093 return ids; 5094} 5095 5096AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5097{ 5098 Mutex::Autolock _l(mLock); 5099 AudioStreamIn *input = mInput; 5100 mInput = NULL; 5101 return input; 5102} 5103 5104// this method must always be called either with ThreadBase mLock held or inside the thread loop 5105audio_stream_t* AudioFlinger::RecordThread::stream() const 5106{ 5107 if (mInput == NULL) { 5108 return NULL; 5109 } 5110 return &mInput->stream->common; 5111} 5112 5113status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5114{ 5115 // only one chain per input thread 5116 if (mEffectChains.size() != 0) { 5117 return INVALID_OPERATION; 5118 } 5119 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5120 5121 chain->setInBuffer(NULL); 5122 chain->setOutBuffer(NULL); 5123 5124 checkSuspendOnAddEffectChain_l(chain); 5125 5126 mEffectChains.add(chain); 5127 5128 return NO_ERROR; 5129} 5130 5131size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5132{ 5133 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5134 ALOGW_IF(mEffectChains.size() != 1, 5135 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5136 chain.get(), mEffectChains.size(), this); 5137 if (mEffectChains.size() == 1) { 5138 mEffectChains.removeAt(0); 5139 } 5140 return 0; 5141} 5142 5143}; // namespace android 5144