Threads.cpp revision 9156ef3e11b68cc4b6d3cea77f1f63673855a6d1
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 mLock.lock(); 380 while (!mConfigEvents.isEmpty()) { 381 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 382 ConfigEvent *event = mConfigEvents[0]; 383 mConfigEvents.removeAt(0); 384 // release mLock before locking AudioFlinger mLock: lock order is always 385 // AudioFlinger then ThreadBase to avoid cross deadlock 386 mLock.unlock(); 387 switch(event->type()) { 388 case CFG_EVENT_PRIO: { 389 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 390 // FIXME Need to understand why this has be done asynchronously 391 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 392 true /*asynchronous*/); 393 if (err != 0) { 394 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; " 395 "error %d", 396 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 397 } 398 } break; 399 case CFG_EVENT_IO: { 400 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 401 mAudioFlinger->mLock.lock(); 402 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 403 mAudioFlinger->mLock.unlock(); 404 } break; 405 default: 406 ALOGE("processConfigEvents() unknown event type %d", event->type()); 407 break; 408 } 409 delete event; 410 mLock.lock(); 411 } 412 mLock.unlock(); 413} 414 415void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 416{ 417 const size_t SIZE = 256; 418 char buffer[SIZE]; 419 String8 result; 420 421 bool locked = AudioFlinger::dumpTryLock(mLock); 422 if (!locked) { 423 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 424 write(fd, buffer, strlen(buffer)); 425 } 426 427 snprintf(buffer, SIZE, "io handle: %d\n", mId); 428 result.append(buffer); 429 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 430 result.append(buffer); 431 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 432 result.append(buffer); 433 snprintf(buffer, SIZE, "Sample rate: %u\n", mSampleRate); 434 result.append(buffer); 435 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 436 result.append(buffer); 437 snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); 438 result.append(buffer); 439 snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); 440 result.append(buffer); 441 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 442 result.append(buffer); 443 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 444 result.append(buffer); 445 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 446 result.append(buffer); 447 448 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 449 result.append(buffer); 450 result.append(" Index Command"); 451 for (size_t i = 0; i < mNewParameters.size(); ++i) { 452 snprintf(buffer, SIZE, "\n %02d ", i); 453 result.append(buffer); 454 result.append(mNewParameters[i]); 455 } 456 457 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 458 result.append(buffer); 459 for (size_t i = 0; i < mConfigEvents.size(); i++) { 460 mConfigEvents[i]->dump(buffer, SIZE); 461 result.append(buffer); 462 } 463 result.append("\n"); 464 465 write(fd, result.string(), result.size()); 466 467 if (locked) { 468 mLock.unlock(); 469 } 470} 471 472void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 473{ 474 const size_t SIZE = 256; 475 char buffer[SIZE]; 476 String8 result; 477 478 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 479 write(fd, buffer, strlen(buffer)); 480 481 for (size_t i = 0; i < mEffectChains.size(); ++i) { 482 sp<EffectChain> chain = mEffectChains[i]; 483 if (chain != 0) { 484 chain->dump(fd, args); 485 } 486 } 487} 488 489void AudioFlinger::ThreadBase::acquireWakeLock() 490{ 491 Mutex::Autolock _l(mLock); 492 acquireWakeLock_l(); 493} 494 495void AudioFlinger::ThreadBase::acquireWakeLock_l() 496{ 497 if (mPowerManager == 0) { 498 // use checkService() to avoid blocking if power service is not up yet 499 sp<IBinder> binder = 500 defaultServiceManager()->checkService(String16("power")); 501 if (binder == 0) { 502 ALOGW("Thread %s cannot connect to the power manager service", mName); 503 } else { 504 mPowerManager = interface_cast<IPowerManager>(binder); 505 binder->linkToDeath(mDeathRecipient); 506 } 507 } 508 if (mPowerManager != 0) { 509 sp<IBinder> binder = new BBinder(); 510 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 511 binder, 512 String16(mName), 513 String16("media")); 514 if (status == NO_ERROR) { 515 mWakeLockToken = binder; 516 } 517 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 518 } 519} 520 521void AudioFlinger::ThreadBase::releaseWakeLock() 522{ 523 Mutex::Autolock _l(mLock); 524 releaseWakeLock_l(); 525} 526 527void AudioFlinger::ThreadBase::releaseWakeLock_l() 528{ 529 if (mWakeLockToken != 0) { 530 ALOGV("releaseWakeLock_l() %s", mName); 531 if (mPowerManager != 0) { 532 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 533 } 534 mWakeLockToken.clear(); 535 } 536} 537 538void AudioFlinger::ThreadBase::clearPowerManager() 539{ 540 Mutex::Autolock _l(mLock); 541 releaseWakeLock_l(); 542 mPowerManager.clear(); 543} 544 545void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 546{ 547 sp<ThreadBase> thread = mThread.promote(); 548 if (thread != 0) { 549 thread->clearPowerManager(); 550 } 551 ALOGW("power manager service died !!!"); 552} 553 554void AudioFlinger::ThreadBase::setEffectSuspended( 555 const effect_uuid_t *type, bool suspend, int sessionId) 556{ 557 Mutex::Autolock _l(mLock); 558 setEffectSuspended_l(type, suspend, sessionId); 559} 560 561void AudioFlinger::ThreadBase::setEffectSuspended_l( 562 const effect_uuid_t *type, bool suspend, int sessionId) 563{ 564 sp<EffectChain> chain = getEffectChain_l(sessionId); 565 if (chain != 0) { 566 if (type != NULL) { 567 chain->setEffectSuspended_l(type, suspend); 568 } else { 569 chain->setEffectSuspendedAll_l(suspend); 570 } 571 } 572 573 updateSuspendedSessions_l(type, suspend, sessionId); 574} 575 576void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 577{ 578 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 579 if (index < 0) { 580 return; 581 } 582 583 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 584 mSuspendedSessions.valueAt(index); 585 586 for (size_t i = 0; i < sessionEffects.size(); i++) { 587 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 588 for (int j = 0; j < desc->mRefCount; j++) { 589 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 590 chain->setEffectSuspendedAll_l(true); 591 } else { 592 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 593 desc->mType.timeLow); 594 chain->setEffectSuspended_l(&desc->mType, true); 595 } 596 } 597 } 598} 599 600void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 601 bool suspend, 602 int sessionId) 603{ 604 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 605 606 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 607 608 if (suspend) { 609 if (index >= 0) { 610 sessionEffects = mSuspendedSessions.valueAt(index); 611 } else { 612 mSuspendedSessions.add(sessionId, sessionEffects); 613 } 614 } else { 615 if (index < 0) { 616 return; 617 } 618 sessionEffects = mSuspendedSessions.valueAt(index); 619 } 620 621 622 int key = EffectChain::kKeyForSuspendAll; 623 if (type != NULL) { 624 key = type->timeLow; 625 } 626 index = sessionEffects.indexOfKey(key); 627 628 sp<SuspendedSessionDesc> desc; 629 if (suspend) { 630 if (index >= 0) { 631 desc = sessionEffects.valueAt(index); 632 } else { 633 desc = new SuspendedSessionDesc(); 634 if (type != NULL) { 635 desc->mType = *type; 636 } 637 sessionEffects.add(key, desc); 638 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 639 } 640 desc->mRefCount++; 641 } else { 642 if (index < 0) { 643 return; 644 } 645 desc = sessionEffects.valueAt(index); 646 if (--desc->mRefCount == 0) { 647 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 648 sessionEffects.removeItemsAt(index); 649 if (sessionEffects.isEmpty()) { 650 ALOGV("updateSuspendedSessions_l() restore removing session %d", 651 sessionId); 652 mSuspendedSessions.removeItem(sessionId); 653 } 654 } 655 } 656 if (!sessionEffects.isEmpty()) { 657 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 658 } 659} 660 661void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 662 bool enabled, 663 int sessionId) 664{ 665 Mutex::Autolock _l(mLock); 666 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 667} 668 669void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 670 bool enabled, 671 int sessionId) 672{ 673 if (mType != RECORD) { 674 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 675 // another session. This gives the priority to well behaved effect control panels 676 // and applications not using global effects. 677 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 678 // global effects 679 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 680 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 681 } 682 } 683 684 sp<EffectChain> chain = getEffectChain_l(sessionId); 685 if (chain != 0) { 686 chain->checkSuspendOnEffectEnabled(effect, enabled); 687 } 688} 689 690// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 691sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 692 const sp<AudioFlinger::Client>& client, 693 const sp<IEffectClient>& effectClient, 694 int32_t priority, 695 int sessionId, 696 effect_descriptor_t *desc, 697 int *enabled, 698 status_t *status) 699{ 700 sp<EffectModule> effect; 701 sp<EffectHandle> handle; 702 status_t lStatus; 703 sp<EffectChain> chain; 704 bool chainCreated = false; 705 bool effectCreated = false; 706 bool effectRegistered = false; 707 708 lStatus = initCheck(); 709 if (lStatus != NO_ERROR) { 710 ALOGW("createEffect_l() Audio driver not initialized."); 711 goto Exit; 712 } 713 714 // Do not allow effects with session ID 0 on direct output or duplicating threads 715 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 716 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 717 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 718 desc->name, sessionId); 719 lStatus = BAD_VALUE; 720 goto Exit; 721 } 722 // Only Pre processor effects are allowed on input threads and only on input threads 723 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 724 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 725 desc->name, desc->flags, mType); 726 lStatus = BAD_VALUE; 727 goto Exit; 728 } 729 730 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 731 732 { // scope for mLock 733 Mutex::Autolock _l(mLock); 734 735 // check for existing effect chain with the requested audio session 736 chain = getEffectChain_l(sessionId); 737 if (chain == 0) { 738 // create a new chain for this session 739 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 740 chain = new EffectChain(this, sessionId); 741 addEffectChain_l(chain); 742 chain->setStrategy(getStrategyForSession_l(sessionId)); 743 chainCreated = true; 744 } else { 745 effect = chain->getEffectFromDesc_l(desc); 746 } 747 748 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 749 750 if (effect == 0) { 751 int id = mAudioFlinger->nextUniqueId(); 752 // Check CPU and memory usage 753 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 754 if (lStatus != NO_ERROR) { 755 goto Exit; 756 } 757 effectRegistered = true; 758 // create a new effect module if none present in the chain 759 effect = new EffectModule(this, chain, desc, id, sessionId); 760 lStatus = effect->status(); 761 if (lStatus != NO_ERROR) { 762 goto Exit; 763 } 764 lStatus = chain->addEffect_l(effect); 765 if (lStatus != NO_ERROR) { 766 goto Exit; 767 } 768 effectCreated = true; 769 770 effect->setDevice(mOutDevice); 771 effect->setDevice(mInDevice); 772 effect->setMode(mAudioFlinger->getMode()); 773 effect->setAudioSource(mAudioSource); 774 } 775 // create effect handle and connect it to effect module 776 handle = new EffectHandle(effect, client, effectClient, priority); 777 lStatus = effect->addHandle(handle.get()); 778 if (enabled != NULL) { 779 *enabled = (int)effect->isEnabled(); 780 } 781 } 782 783Exit: 784 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 785 Mutex::Autolock _l(mLock); 786 if (effectCreated) { 787 chain->removeEffect_l(effect); 788 } 789 if (effectRegistered) { 790 AudioSystem::unregisterEffect(effect->id()); 791 } 792 if (chainCreated) { 793 removeEffectChain_l(chain); 794 } 795 handle.clear(); 796 } 797 798 *status = lStatus; 799 return handle; 800} 801 802sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 803{ 804 Mutex::Autolock _l(mLock); 805 return getEffect_l(sessionId, effectId); 806} 807 808sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 809{ 810 sp<EffectChain> chain = getEffectChain_l(sessionId); 811 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 812} 813 814// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 815// PlaybackThread::mLock held 816status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 817{ 818 // check for existing effect chain with the requested audio session 819 int sessionId = effect->sessionId(); 820 sp<EffectChain> chain = getEffectChain_l(sessionId); 821 bool chainCreated = false; 822 823 if (chain == 0) { 824 // create a new chain for this session 825 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 826 chain = new EffectChain(this, sessionId); 827 addEffectChain_l(chain); 828 chain->setStrategy(getStrategyForSession_l(sessionId)); 829 chainCreated = true; 830 } 831 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 832 833 if (chain->getEffectFromId_l(effect->id()) != 0) { 834 ALOGW("addEffect_l() %p effect %s already present in chain %p", 835 this, effect->desc().name, chain.get()); 836 return BAD_VALUE; 837 } 838 839 status_t status = chain->addEffect_l(effect); 840 if (status != NO_ERROR) { 841 if (chainCreated) { 842 removeEffectChain_l(chain); 843 } 844 return status; 845 } 846 847 effect->setDevice(mOutDevice); 848 effect->setDevice(mInDevice); 849 effect->setMode(mAudioFlinger->getMode()); 850 effect->setAudioSource(mAudioSource); 851 return NO_ERROR; 852} 853 854void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 855 856 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 857 effect_descriptor_t desc = effect->desc(); 858 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 859 detachAuxEffect_l(effect->id()); 860 } 861 862 sp<EffectChain> chain = effect->chain().promote(); 863 if (chain != 0) { 864 // remove effect chain if removing last effect 865 if (chain->removeEffect_l(effect) == 0) { 866 removeEffectChain_l(chain); 867 } 868 } else { 869 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 870 } 871} 872 873void AudioFlinger::ThreadBase::lockEffectChains_l( 874 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 875{ 876 effectChains = mEffectChains; 877 for (size_t i = 0; i < mEffectChains.size(); i++) { 878 mEffectChains[i]->lock(); 879 } 880} 881 882void AudioFlinger::ThreadBase::unlockEffectChains( 883 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 884{ 885 for (size_t i = 0; i < effectChains.size(); i++) { 886 effectChains[i]->unlock(); 887 } 888} 889 890sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 891{ 892 Mutex::Autolock _l(mLock); 893 return getEffectChain_l(sessionId); 894} 895 896sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 897{ 898 size_t size = mEffectChains.size(); 899 for (size_t i = 0; i < size; i++) { 900 if (mEffectChains[i]->sessionId() == sessionId) { 901 return mEffectChains[i]; 902 } 903 } 904 return 0; 905} 906 907void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 908{ 909 Mutex::Autolock _l(mLock); 910 size_t size = mEffectChains.size(); 911 for (size_t i = 0; i < size; i++) { 912 mEffectChains[i]->setMode_l(mode); 913 } 914} 915 916void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 917 EffectHandle *handle, 918 bool unpinIfLast) { 919 920 Mutex::Autolock _l(mLock); 921 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 922 // delete the effect module if removing last handle on it 923 if (effect->removeHandle(handle) == 0) { 924 if (!effect->isPinned() || unpinIfLast) { 925 removeEffect_l(effect); 926 AudioSystem::unregisterEffect(effect->id()); 927 } 928 } 929} 930 931// ---------------------------------------------------------------------------- 932// Playback 933// ---------------------------------------------------------------------------- 934 935AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 936 AudioStreamOut* output, 937 audio_io_handle_t id, 938 audio_devices_t device, 939 type_t type) 940 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 941 mNormalFrameCount(0), mMixBuffer(NULL), 942 mSuspended(0), mBytesWritten(0), 943 // mStreamTypes[] initialized in constructor body 944 mOutput(output), 945 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 946 mMixerStatus(MIXER_IDLE), 947 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 948 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 949 mBytesRemaining(0), 950 mCurrentWriteLength(0), 951 mUseAsyncWrite(false), 952 mWriteBlocked(false), 953 mDraining(false), 954 mScreenState(AudioFlinger::mScreenState), 955 // index 0 is reserved for normal mixer's submix 956 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 957{ 958 snprintf(mName, kNameLength, "AudioOut_%X", id); 959 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 960 961 // Assumes constructor is called by AudioFlinger with it's mLock held, but 962 // it would be safer to explicitly pass initial masterVolume/masterMute as 963 // parameter. 964 // 965 // If the HAL we are using has support for master volume or master mute, 966 // then do not attenuate or mute during mixing (just leave the volume at 1.0 967 // and the mute set to false). 968 mMasterVolume = audioFlinger->masterVolume_l(); 969 mMasterMute = audioFlinger->masterMute_l(); 970 if (mOutput && mOutput->audioHwDev) { 971 if (mOutput->audioHwDev->canSetMasterVolume()) { 972 mMasterVolume = 1.0; 973 } 974 975 if (mOutput->audioHwDev->canSetMasterMute()) { 976 mMasterMute = false; 977 } 978 } 979 980 readOutputParameters(); 981 982 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 983 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 984 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 985 stream = (audio_stream_type_t) (stream + 1)) { 986 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 987 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 988 } 989 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 990 // because mAudioFlinger doesn't have one to copy from 991} 992 993AudioFlinger::PlaybackThread::~PlaybackThread() 994{ 995 mAudioFlinger->unregisterWriter(mNBLogWriter); 996 delete[] mMixBuffer; 997} 998 999void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1000{ 1001 dumpInternals(fd, args); 1002 dumpTracks(fd, args); 1003 dumpEffectChains(fd, args); 1004} 1005 1006void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1007{ 1008 const size_t SIZE = 256; 1009 char buffer[SIZE]; 1010 String8 result; 1011 1012 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1013 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1014 const stream_type_t *st = &mStreamTypes[i]; 1015 if (i > 0) { 1016 result.appendFormat(", "); 1017 } 1018 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1019 if (st->mute) { 1020 result.append("M"); 1021 } 1022 } 1023 result.append("\n"); 1024 write(fd, result.string(), result.length()); 1025 result.clear(); 1026 1027 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1028 result.append(buffer); 1029 Track::appendDumpHeader(result); 1030 for (size_t i = 0; i < mTracks.size(); ++i) { 1031 sp<Track> track = mTracks[i]; 1032 if (track != 0) { 1033 track->dump(buffer, SIZE); 1034 result.append(buffer); 1035 } 1036 } 1037 1038 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1039 result.append(buffer); 1040 Track::appendDumpHeader(result); 1041 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1042 sp<Track> track = mActiveTracks[i].promote(); 1043 if (track != 0) { 1044 track->dump(buffer, SIZE); 1045 result.append(buffer); 1046 } 1047 } 1048 write(fd, result.string(), result.size()); 1049 1050 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1051 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1052 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1053 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1054} 1055 1056void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1057{ 1058 const size_t SIZE = 256; 1059 char buffer[SIZE]; 1060 String8 result; 1061 1062 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1063 result.append(buffer); 1064 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1065 result.append(buffer); 1066 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", 1067 ns2ms(systemTime() - mLastWriteTime)); 1068 result.append(buffer); 1069 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1070 result.append(buffer); 1071 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1072 result.append(buffer); 1073 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1074 result.append(buffer); 1075 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1076 result.append(buffer); 1077 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1078 result.append(buffer); 1079 write(fd, result.string(), result.size()); 1080 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1081 1082 dumpBase(fd, args); 1083} 1084 1085// Thread virtuals 1086 1087void AudioFlinger::PlaybackThread::onFirstRef() 1088{ 1089 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1090} 1091 1092// ThreadBase virtuals 1093void AudioFlinger::PlaybackThread::preExit() 1094{ 1095 ALOGV(" preExit()"); 1096 // FIXME this is using hard-coded strings but in the future, this functionality will be 1097 // converted to use audio HAL extensions required to support tunneling 1098 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1099} 1100 1101// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1102sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1103 const sp<AudioFlinger::Client>& client, 1104 audio_stream_type_t streamType, 1105 uint32_t sampleRate, 1106 audio_format_t format, 1107 audio_channel_mask_t channelMask, 1108 size_t frameCount, 1109 const sp<IMemory>& sharedBuffer, 1110 int sessionId, 1111 IAudioFlinger::track_flags_t *flags, 1112 pid_t tid, 1113 status_t *status) 1114{ 1115 sp<Track> track; 1116 status_t lStatus; 1117 1118 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1119 1120 // client expresses a preference for FAST, but we get the final say 1121 if (*flags & IAudioFlinger::TRACK_FAST) { 1122 if ( 1123 // not timed 1124 (!isTimed) && 1125 // either of these use cases: 1126 ( 1127 // use case 1: shared buffer with any frame count 1128 ( 1129 (sharedBuffer != 0) 1130 ) || 1131 // use case 2: callback handler and frame count is default or at least as large as HAL 1132 ( 1133 (tid != -1) && 1134 ((frameCount == 0) || 1135 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 1136 ) 1137 ) && 1138 // PCM data 1139 audio_is_linear_pcm(format) && 1140 // mono or stereo 1141 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1142 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1143#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1144 // hardware sample rate 1145 (sampleRate == mSampleRate) && 1146#endif 1147 // normal mixer has an associated fast mixer 1148 hasFastMixer() && 1149 // there are sufficient fast track slots available 1150 (mFastTrackAvailMask != 0) 1151 // FIXME test that MixerThread for this fast track has a capable output HAL 1152 // FIXME add a permission test also? 1153 ) { 1154 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1155 if (frameCount == 0) { 1156 frameCount = mFrameCount * kFastTrackMultiplier; 1157 } 1158 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1159 frameCount, mFrameCount); 1160 } else { 1161 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1162 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1163 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1164 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1165 audio_is_linear_pcm(format), 1166 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1167 *flags &= ~IAudioFlinger::TRACK_FAST; 1168 // For compatibility with AudioTrack calculation, buffer depth is forced 1169 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1170 // This is probably too conservative, but legacy application code may depend on it. 1171 // If you change this calculation, also review the start threshold which is related. 1172 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1173 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1174 if (minBufCount < 2) { 1175 minBufCount = 2; 1176 } 1177 size_t minFrameCount = mNormalFrameCount * minBufCount; 1178 if (frameCount < minFrameCount) { 1179 frameCount = minFrameCount; 1180 } 1181 } 1182 } 1183 1184 if (mType == DIRECT) { 1185 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1186 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1187 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %d, channelMask 0x%08x " 1188 "for output %p with format %d", 1189 sampleRate, format, channelMask, mOutput, mFormat); 1190 lStatus = BAD_VALUE; 1191 goto Exit; 1192 } 1193 } 1194 } else if (mType == OFFLOAD) { 1195 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1196 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1197 "for output %p with format %d", 1198 sampleRate, format, channelMask, mOutput, mFormat); 1199 lStatus = BAD_VALUE; 1200 goto Exit; 1201 } 1202 } else { 1203 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1204 ALOGE("createTrack_l() Bad parameter: format %d \"" 1205 "for output %p with format %d", 1206 format, mOutput, mFormat); 1207 lStatus = BAD_VALUE; 1208 goto Exit; 1209 } 1210 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1211 if (sampleRate > mSampleRate*2) { 1212 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1213 lStatus = BAD_VALUE; 1214 goto Exit; 1215 } 1216 } 1217 1218 lStatus = initCheck(); 1219 if (lStatus != NO_ERROR) { 1220 ALOGE("Audio driver not initialized."); 1221 goto Exit; 1222 } 1223 1224 { // scope for mLock 1225 Mutex::Autolock _l(mLock); 1226 1227 // all tracks in same audio session must share the same routing strategy otherwise 1228 // conflicts will happen when tracks are moved from one output to another by audio policy 1229 // manager 1230 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1231 for (size_t i = 0; i < mTracks.size(); ++i) { 1232 sp<Track> t = mTracks[i]; 1233 if (t != 0 && !t->isOutputTrack()) { 1234 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1235 if (sessionId == t->sessionId() && strategy != actual) { 1236 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1237 strategy, actual); 1238 lStatus = BAD_VALUE; 1239 goto Exit; 1240 } 1241 } 1242 } 1243 1244 if (!isTimed) { 1245 track = new Track(this, client, streamType, sampleRate, format, 1246 channelMask, frameCount, sharedBuffer, sessionId, *flags); 1247 } else { 1248 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1249 channelMask, frameCount, sharedBuffer, sessionId); 1250 } 1251 if (track == 0 || track->getCblk() == 0 || track->name() < 0) { 1252 lStatus = NO_MEMORY; 1253 goto Exit; 1254 } 1255 1256 mTracks.add(track); 1257 1258 sp<EffectChain> chain = getEffectChain_l(sessionId); 1259 if (chain != 0) { 1260 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1261 track->setMainBuffer(chain->inBuffer()); 1262 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1263 chain->incTrackCnt(); 1264 } 1265 1266 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1267 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1268 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1269 // so ask activity manager to do this on our behalf 1270 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1271 } 1272 } 1273 1274 lStatus = NO_ERROR; 1275 1276Exit: 1277 *status = lStatus; 1278 return track; 1279} 1280 1281uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1282{ 1283 return latency; 1284} 1285 1286uint32_t AudioFlinger::PlaybackThread::latency() const 1287{ 1288 Mutex::Autolock _l(mLock); 1289 return latency_l(); 1290} 1291uint32_t AudioFlinger::PlaybackThread::latency_l() const 1292{ 1293 if (initCheck() == NO_ERROR) { 1294 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1295 } else { 1296 return 0; 1297 } 1298} 1299 1300void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1301{ 1302 Mutex::Autolock _l(mLock); 1303 // Don't apply master volume in SW if our HAL can do it for us. 1304 if (mOutput && mOutput->audioHwDev && 1305 mOutput->audioHwDev->canSetMasterVolume()) { 1306 mMasterVolume = 1.0; 1307 } else { 1308 mMasterVolume = value; 1309 } 1310} 1311 1312void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1313{ 1314 Mutex::Autolock _l(mLock); 1315 // Don't apply master mute in SW if our HAL can do it for us. 1316 if (mOutput && mOutput->audioHwDev && 1317 mOutput->audioHwDev->canSetMasterMute()) { 1318 mMasterMute = false; 1319 } else { 1320 mMasterMute = muted; 1321 } 1322} 1323 1324void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1325{ 1326 Mutex::Autolock _l(mLock); 1327 mStreamTypes[stream].volume = value; 1328 signal_l(); 1329} 1330 1331void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1332{ 1333 Mutex::Autolock _l(mLock); 1334 mStreamTypes[stream].mute = muted; 1335 signal_l(); 1336} 1337 1338float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1339{ 1340 Mutex::Autolock _l(mLock); 1341 return mStreamTypes[stream].volume; 1342} 1343 1344// addTrack_l() must be called with ThreadBase::mLock held 1345status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1346{ 1347 status_t status = ALREADY_EXISTS; 1348 1349 // set retry count for buffer fill 1350 track->mRetryCount = kMaxTrackStartupRetries; 1351 if (mActiveTracks.indexOf(track) < 0) { 1352 // the track is newly added, make sure it fills up all its 1353 // buffers before playing. This is to ensure the client will 1354 // effectively get the latency it requested. 1355 if (!track->isOutputTrack()) { 1356 TrackBase::track_state state = track->mState; 1357 mLock.unlock(); 1358 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1359 mLock.lock(); 1360 // abort track was stopped/paused while we released the lock 1361 if (state != track->mState) { 1362 if (status == NO_ERROR) { 1363 mLock.unlock(); 1364 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1365 mLock.lock(); 1366 } 1367 return INVALID_OPERATION; 1368 } 1369 // abort if start is rejected by audio policy manager 1370 if (status != NO_ERROR) { 1371 return PERMISSION_DENIED; 1372 } 1373#ifdef ADD_BATTERY_DATA 1374 // to track the speaker usage 1375 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1376#endif 1377 } 1378 1379 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1380 track->mResetDone = false; 1381 track->mPresentationCompleteFrames = 0; 1382 mActiveTracks.add(track); 1383 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1384 if (chain != 0) { 1385 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1386 track->sessionId()); 1387 chain->incActiveTrackCnt(); 1388 } 1389 1390 status = NO_ERROR; 1391 } 1392 1393 ALOGV("mWaitWorkCV.broadcast"); 1394 mWaitWorkCV.broadcast(); 1395 1396 return status; 1397} 1398 1399bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1400{ 1401 track->terminate(); 1402 // active tracks are removed by threadLoop() 1403 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1404 track->mState = TrackBase::STOPPED; 1405 if (!trackActive) { 1406 removeTrack_l(track); 1407 } else if (track->isFastTrack() || track->isOffloaded()) { 1408 track->mState = TrackBase::STOPPING_1; 1409 } 1410 1411 return trackActive; 1412} 1413 1414void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1415{ 1416 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1417 mTracks.remove(track); 1418 deleteTrackName_l(track->name()); 1419 // redundant as track is about to be destroyed, for dumpsys only 1420 track->mName = -1; 1421 if (track->isFastTrack()) { 1422 int index = track->mFastIndex; 1423 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1424 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1425 mFastTrackAvailMask |= 1 << index; 1426 // redundant as track is about to be destroyed, for dumpsys only 1427 track->mFastIndex = -1; 1428 } 1429 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1430 if (chain != 0) { 1431 chain->decTrackCnt(); 1432 } 1433} 1434 1435void AudioFlinger::PlaybackThread::signal_l() 1436{ 1437 // Thread could be blocked waiting for async 1438 // so signal it to handle state changes immediately 1439 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1440 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1441 mSignalPending = true; 1442 mWaitWorkCV.signal(); 1443} 1444 1445String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1446{ 1447 Mutex::Autolock _l(mLock); 1448 if (initCheck() != NO_ERROR) { 1449 return String8(); 1450 } 1451 1452 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1453 const String8 out_s8(s); 1454 free(s); 1455 return out_s8; 1456} 1457 1458// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1459void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1460 AudioSystem::OutputDescriptor desc; 1461 void *param2 = NULL; 1462 1463 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1464 param); 1465 1466 switch (event) { 1467 case AudioSystem::OUTPUT_OPENED: 1468 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1469 desc.channelMask = mChannelMask; 1470 desc.samplingRate = mSampleRate; 1471 desc.format = mFormat; 1472 desc.frameCount = mNormalFrameCount; // FIXME see 1473 // AudioFlinger::frameCount(audio_io_handle_t) 1474 desc.latency = latency(); 1475 param2 = &desc; 1476 break; 1477 1478 case AudioSystem::STREAM_CONFIG_CHANGED: 1479 param2 = ¶m; 1480 case AudioSystem::OUTPUT_CLOSED: 1481 default: 1482 break; 1483 } 1484 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1485} 1486 1487void AudioFlinger::PlaybackThread::writeCallback() 1488{ 1489 ALOG_ASSERT(mCallbackThread != 0); 1490 mCallbackThread->setWriteBlocked(false); 1491} 1492 1493void AudioFlinger::PlaybackThread::drainCallback() 1494{ 1495 ALOG_ASSERT(mCallbackThread != 0); 1496 mCallbackThread->setDraining(false); 1497} 1498 1499void AudioFlinger::PlaybackThread::setWriteBlocked(bool value) 1500{ 1501 Mutex::Autolock _l(mLock); 1502 mWriteBlocked = value; 1503 if (!value) { 1504 mWaitWorkCV.signal(); 1505 } 1506} 1507 1508void AudioFlinger::PlaybackThread::setDraining(bool value) 1509{ 1510 Mutex::Autolock _l(mLock); 1511 mDraining = value; 1512 if (!value) { 1513 mWaitWorkCV.signal(); 1514 } 1515} 1516 1517// static 1518int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1519 void *param, 1520 void *cookie) 1521{ 1522 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1523 ALOGV("asyncCallback() event %d", event); 1524 switch (event) { 1525 case STREAM_CBK_EVENT_WRITE_READY: 1526 me->writeCallback(); 1527 break; 1528 case STREAM_CBK_EVENT_DRAIN_READY: 1529 me->drainCallback(); 1530 break; 1531 default: 1532 ALOGW("asyncCallback() unknown event %d", event); 1533 break; 1534 } 1535 return 0; 1536} 1537 1538void AudioFlinger::PlaybackThread::readOutputParameters() 1539{ 1540 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1541 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1542 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1543 if (!audio_is_output_channel(mChannelMask)) { 1544 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1545 } 1546 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1547 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1548 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1549 } 1550 mChannelCount = popcount(mChannelMask); 1551 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1552 if (!audio_is_valid_format(mFormat)) { 1553 LOG_FATAL("HAL format %d not valid for output", mFormat); 1554 } 1555 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1556 LOG_FATAL("HAL format %d not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1557 mFormat); 1558 } 1559 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1560 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1561 mFrameCount = mBufferSize / mFrameSize; 1562 if (mFrameCount & 15) { 1563 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1564 mFrameCount); 1565 } 1566 1567 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1568 (mOutput->stream->set_callback != NULL)) { 1569 if (mOutput->stream->set_callback(mOutput->stream, 1570 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1571 mUseAsyncWrite = true; 1572 } 1573 } 1574 1575 // Calculate size of normal mix buffer relative to the HAL output buffer size 1576 double multiplier = 1.0; 1577 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1578 kUseFastMixer == FastMixer_Dynamic)) { 1579 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1580 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1581 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1582 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1583 maxNormalFrameCount = maxNormalFrameCount & ~15; 1584 if (maxNormalFrameCount < minNormalFrameCount) { 1585 maxNormalFrameCount = minNormalFrameCount; 1586 } 1587 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1588 if (multiplier <= 1.0) { 1589 multiplier = 1.0; 1590 } else if (multiplier <= 2.0) { 1591 if (2 * mFrameCount <= maxNormalFrameCount) { 1592 multiplier = 2.0; 1593 } else { 1594 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1595 } 1596 } else { 1597 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1598 // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast 1599 // track, but we sometimes have to do this to satisfy the maximum frame count 1600 // constraint) 1601 // FIXME this rounding up should not be done if no HAL SRC 1602 uint32_t truncMult = (uint32_t) multiplier; 1603 if ((truncMult & 1)) { 1604 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1605 ++truncMult; 1606 } 1607 } 1608 multiplier = (double) truncMult; 1609 } 1610 } 1611 mNormalFrameCount = multiplier * mFrameCount; 1612 // round up to nearest 16 frames to satisfy AudioMixer 1613 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1614 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, 1615 mNormalFrameCount); 1616 1617 delete[] mMixBuffer; 1618 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1619 // For historical reasons mMixBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1620 mMixBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1621 memset(mMixBuffer, 0, normalBufferSize); 1622 1623 // force reconfiguration of effect chains and engines to take new buffer size and audio 1624 // parameters into account 1625 // Note that mLock is not held when readOutputParameters() is called from the constructor 1626 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1627 // matter. 1628 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1629 Vector< sp<EffectChain> > effectChains = mEffectChains; 1630 for (size_t i = 0; i < effectChains.size(); i ++) { 1631 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1632 } 1633} 1634 1635 1636status_t AudioFlinger::PlaybackThread::getRenderPosition(size_t *halFrames, size_t *dspFrames) 1637{ 1638 if (halFrames == NULL || dspFrames == NULL) { 1639 return BAD_VALUE; 1640 } 1641 Mutex::Autolock _l(mLock); 1642 if (initCheck() != NO_ERROR) { 1643 return INVALID_OPERATION; 1644 } 1645 size_t framesWritten = mBytesWritten / mFrameSize; 1646 *halFrames = framesWritten; 1647 1648 if (isSuspended()) { 1649 // return an estimation of rendered frames when the output is suspended 1650 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1651 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1652 return NO_ERROR; 1653 } else { 1654 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1655 } 1656} 1657 1658uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1659{ 1660 Mutex::Autolock _l(mLock); 1661 uint32_t result = 0; 1662 if (getEffectChain_l(sessionId) != 0) { 1663 result = EFFECT_SESSION; 1664 } 1665 1666 for (size_t i = 0; i < mTracks.size(); ++i) { 1667 sp<Track> track = mTracks[i]; 1668 if (sessionId == track->sessionId() && !track->isInvalid()) { 1669 result |= TRACK_SESSION; 1670 break; 1671 } 1672 } 1673 1674 return result; 1675} 1676 1677uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1678{ 1679 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1680 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1681 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1682 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1683 } 1684 for (size_t i = 0; i < mTracks.size(); i++) { 1685 sp<Track> track = mTracks[i]; 1686 if (sessionId == track->sessionId() && !track->isInvalid()) { 1687 return AudioSystem::getStrategyForStream(track->streamType()); 1688 } 1689 } 1690 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1691} 1692 1693 1694AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1695{ 1696 Mutex::Autolock _l(mLock); 1697 return mOutput; 1698} 1699 1700AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1701{ 1702 Mutex::Autolock _l(mLock); 1703 AudioStreamOut *output = mOutput; 1704 mOutput = NULL; 1705 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1706 // must push a NULL and wait for ack 1707 mOutputSink.clear(); 1708 mPipeSink.clear(); 1709 mNormalSink.clear(); 1710 return output; 1711} 1712 1713// this method must always be called either with ThreadBase mLock held or inside the thread loop 1714audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1715{ 1716 if (mOutput == NULL) { 1717 return NULL; 1718 } 1719 return &mOutput->stream->common; 1720} 1721 1722uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1723{ 1724 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1725} 1726 1727status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1728{ 1729 if (!isValidSyncEvent(event)) { 1730 return BAD_VALUE; 1731 } 1732 1733 Mutex::Autolock _l(mLock); 1734 1735 for (size_t i = 0; i < mTracks.size(); ++i) { 1736 sp<Track> track = mTracks[i]; 1737 if (event->triggerSession() == track->sessionId()) { 1738 (void) track->setSyncEvent(event); 1739 return NO_ERROR; 1740 } 1741 } 1742 1743 return NAME_NOT_FOUND; 1744} 1745 1746bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1747{ 1748 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1749} 1750 1751void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1752 const Vector< sp<Track> >& tracksToRemove) 1753{ 1754 size_t count = tracksToRemove.size(); 1755 if (count) { 1756 for (size_t i = 0 ; i < count ; i++) { 1757 const sp<Track>& track = tracksToRemove.itemAt(i); 1758 if (!track->isOutputTrack()) { 1759 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1760#ifdef ADD_BATTERY_DATA 1761 // to track the speaker usage 1762 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1763#endif 1764 if (track->isTerminated()) { 1765 AudioSystem::releaseOutput(mId); 1766 } 1767 } 1768 } 1769 } 1770} 1771 1772void AudioFlinger::PlaybackThread::checkSilentMode_l() 1773{ 1774 if (!mMasterMute) { 1775 char value[PROPERTY_VALUE_MAX]; 1776 if (property_get("ro.audio.silent", value, "0") > 0) { 1777 char *endptr; 1778 unsigned long ul = strtoul(value, &endptr, 0); 1779 if (*endptr == '\0' && ul != 0) { 1780 ALOGD("Silence is golden"); 1781 // The setprop command will not allow a property to be changed after 1782 // the first time it is set, so we don't have to worry about un-muting. 1783 setMasterMute_l(true); 1784 } 1785 } 1786 } 1787} 1788 1789// shared by MIXER and DIRECT, overridden by DUPLICATING 1790ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1791{ 1792 // FIXME rewrite to reduce number of system calls 1793 mLastWriteTime = systemTime(); 1794 mInWrite = true; 1795 ssize_t bytesWritten; 1796 1797 // If an NBAIO sink is present, use it to write the normal mixer's submix 1798 if (mNormalSink != 0) { 1799#define mBitShift 2 // FIXME 1800 size_t count = mBytesRemaining >> mBitShift; 1801 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1802 ATRACE_BEGIN("write"); 1803 // update the setpoint when AudioFlinger::mScreenState changes 1804 uint32_t screenState = AudioFlinger::mScreenState; 1805 if (screenState != mScreenState) { 1806 mScreenState = screenState; 1807 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1808 if (pipe != NULL) { 1809 pipe->setAvgFrames((mScreenState & 1) ? 1810 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1811 } 1812 } 1813 ssize_t framesWritten = mNormalSink->write(mMixBuffer + offset, count); 1814 ATRACE_END(); 1815 if (framesWritten > 0) { 1816 bytesWritten = framesWritten << mBitShift; 1817 } else { 1818 bytesWritten = framesWritten; 1819 } 1820 // otherwise use the HAL / AudioStreamOut directly 1821 } else { 1822 // Direct output and offload threads 1823 size_t offset = (mCurrentWriteLength - mBytesRemaining) / sizeof(int16_t); 1824 if (mUseAsyncWrite) { 1825 mWriteBlocked = true; 1826 ALOG_ASSERT(mCallbackThread != 0); 1827 mCallbackThread->setWriteBlocked(true); 1828 } 1829 bytesWritten = mOutput->stream->write(mOutput->stream, 1830 mMixBuffer + offset, mBytesRemaining); 1831 if (mUseAsyncWrite && 1832 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 1833 // do not wait for async callback in case of error of full write 1834 mWriteBlocked = false; 1835 ALOG_ASSERT(mCallbackThread != 0); 1836 mCallbackThread->setWriteBlocked(false); 1837 } 1838 } 1839 1840 mNumWrites++; 1841 mInWrite = false; 1842 1843 return bytesWritten; 1844} 1845 1846void AudioFlinger::PlaybackThread::threadLoop_drain() 1847{ 1848 if (mOutput->stream->drain) { 1849 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 1850 if (mUseAsyncWrite) { 1851 mDraining = true; 1852 ALOG_ASSERT(mCallbackThread != 0); 1853 mCallbackThread->setDraining(true); 1854 } 1855 mOutput->stream->drain(mOutput->stream, 1856 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 1857 : AUDIO_DRAIN_ALL); 1858 } 1859} 1860 1861void AudioFlinger::PlaybackThread::threadLoop_exit() 1862{ 1863 // Default implementation has nothing to do 1864} 1865 1866/* 1867The derived values that are cached: 1868 - mixBufferSize from frame count * frame size 1869 - activeSleepTime from activeSleepTimeUs() 1870 - idleSleepTime from idleSleepTimeUs() 1871 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 1872 - maxPeriod from frame count and sample rate (MIXER only) 1873 1874The parameters that affect these derived values are: 1875 - frame count 1876 - frame size 1877 - sample rate 1878 - device type: A2DP or not 1879 - device latency 1880 - format: PCM or not 1881 - active sleep time 1882 - idle sleep time 1883*/ 1884 1885void AudioFlinger::PlaybackThread::cacheParameters_l() 1886{ 1887 mixBufferSize = mNormalFrameCount * mFrameSize; 1888 activeSleepTime = activeSleepTimeUs(); 1889 idleSleepTime = idleSleepTimeUs(); 1890} 1891 1892void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 1893{ 1894 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1895 this, streamType, mTracks.size()); 1896 Mutex::Autolock _l(mLock); 1897 1898 size_t size = mTracks.size(); 1899 for (size_t i = 0; i < size; i++) { 1900 sp<Track> t = mTracks[i]; 1901 if (t->streamType() == streamType) { 1902 t->invalidate(); 1903 } 1904 } 1905} 1906 1907status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 1908{ 1909 int session = chain->sessionId(); 1910 int16_t *buffer = mMixBuffer; 1911 bool ownsBuffer = false; 1912 1913 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 1914 if (session > 0) { 1915 // Only one effect chain can be present in direct output thread and it uses 1916 // the mix buffer as input 1917 if (mType != DIRECT) { 1918 size_t numSamples = mNormalFrameCount * mChannelCount; 1919 buffer = new int16_t[numSamples]; 1920 memset(buffer, 0, numSamples * sizeof(int16_t)); 1921 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 1922 ownsBuffer = true; 1923 } 1924 1925 // Attach all tracks with same session ID to this chain. 1926 for (size_t i = 0; i < mTracks.size(); ++i) { 1927 sp<Track> track = mTracks[i]; 1928 if (session == track->sessionId()) { 1929 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 1930 buffer); 1931 track->setMainBuffer(buffer); 1932 chain->incTrackCnt(); 1933 } 1934 } 1935 1936 // indicate all active tracks in the chain 1937 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1938 sp<Track> track = mActiveTracks[i].promote(); 1939 if (track == 0) { 1940 continue; 1941 } 1942 if (session == track->sessionId()) { 1943 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 1944 chain->incActiveTrackCnt(); 1945 } 1946 } 1947 } 1948 1949 chain->setInBuffer(buffer, ownsBuffer); 1950 chain->setOutBuffer(mMixBuffer); 1951 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 1952 // chains list in order to be processed last as it contains output stage effects 1953 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 1954 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 1955 // after track specific effects and before output stage 1956 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 1957 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 1958 // Effect chain for other sessions are inserted at beginning of effect 1959 // chains list to be processed before output mix effects. Relative order between other 1960 // sessions is not important 1961 size_t size = mEffectChains.size(); 1962 size_t i = 0; 1963 for (i = 0; i < size; i++) { 1964 if (mEffectChains[i]->sessionId() < session) { 1965 break; 1966 } 1967 } 1968 mEffectChains.insertAt(chain, i); 1969 checkSuspendOnAddEffectChain_l(chain); 1970 1971 return NO_ERROR; 1972} 1973 1974size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 1975{ 1976 int session = chain->sessionId(); 1977 1978 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 1979 1980 for (size_t i = 0; i < mEffectChains.size(); i++) { 1981 if (chain == mEffectChains[i]) { 1982 mEffectChains.removeAt(i); 1983 // detach all active tracks from the chain 1984 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 1985 sp<Track> track = mActiveTracks[i].promote(); 1986 if (track == 0) { 1987 continue; 1988 } 1989 if (session == track->sessionId()) { 1990 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 1991 chain.get(), session); 1992 chain->decActiveTrackCnt(); 1993 } 1994 } 1995 1996 // detach all tracks with same session ID from this chain 1997 for (size_t i = 0; i < mTracks.size(); ++i) { 1998 sp<Track> track = mTracks[i]; 1999 if (session == track->sessionId()) { 2000 track->setMainBuffer(mMixBuffer); 2001 chain->decTrackCnt(); 2002 } 2003 } 2004 break; 2005 } 2006 } 2007 return mEffectChains.size(); 2008} 2009 2010status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2011 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2012{ 2013 Mutex::Autolock _l(mLock); 2014 return attachAuxEffect_l(track, EffectId); 2015} 2016 2017status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2018 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2019{ 2020 status_t status = NO_ERROR; 2021 2022 if (EffectId == 0) { 2023 track->setAuxBuffer(0, NULL); 2024 } else { 2025 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2026 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2027 if (effect != 0) { 2028 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2029 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2030 } else { 2031 status = INVALID_OPERATION; 2032 } 2033 } else { 2034 status = BAD_VALUE; 2035 } 2036 } 2037 return status; 2038} 2039 2040void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2041{ 2042 for (size_t i = 0; i < mTracks.size(); ++i) { 2043 sp<Track> track = mTracks[i]; 2044 if (track->auxEffectId() == effectId) { 2045 attachAuxEffect_l(track, 0); 2046 } 2047 } 2048} 2049 2050bool AudioFlinger::PlaybackThread::threadLoop() 2051{ 2052 Vector< sp<Track> > tracksToRemove; 2053 2054 standbyTime = systemTime(); 2055 2056 // MIXER 2057 nsecs_t lastWarning = 0; 2058 2059 // DUPLICATING 2060 // FIXME could this be made local to while loop? 2061 writeFrames = 0; 2062 2063 cacheParameters_l(); 2064 sleepTime = idleSleepTime; 2065 2066 if (mType == MIXER) { 2067 sleepTimeShift = 0; 2068 } 2069 2070 CpuStats cpuStats; 2071 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2072 2073 acquireWakeLock(); 2074 2075 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2076 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2077 // and then that string will be logged at the next convenient opportunity. 2078 const char *logString = NULL; 2079 2080 while (!exitPending()) 2081 { 2082 cpuStats.sample(myName); 2083 2084 Vector< sp<EffectChain> > effectChains; 2085 2086 processConfigEvents(); 2087 2088 { // scope for mLock 2089 2090 Mutex::Autolock _l(mLock); 2091 2092 if (logString != NULL) { 2093 mNBLogWriter->logTimestamp(); 2094 mNBLogWriter->log(logString); 2095 logString = NULL; 2096 } 2097 2098 if (checkForNewParameters_l()) { 2099 cacheParameters_l(); 2100 } 2101 2102 saveOutputTracks(); 2103 2104 if (mSignalPending) { 2105 // A signal was raised while we were unlocked 2106 mSignalPending = false; 2107 } else if (waitingAsyncCallback_l()) { 2108 if (exitPending()) { 2109 break; 2110 } 2111 releaseWakeLock_l(); 2112 ALOGV("wait async completion"); 2113 mWaitWorkCV.wait(mLock); 2114 ALOGV("async completion/wake"); 2115 acquireWakeLock_l(); 2116 if (exitPending()) { 2117 break; 2118 } 2119 if (!mActiveTracks.size() && (systemTime() > standbyTime)) { 2120 continue; 2121 } 2122 sleepTime = 0; 2123 } else if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2124 isSuspended()) { 2125 // put audio hardware into standby after short delay 2126 if (shouldStandby_l()) { 2127 2128 threadLoop_standby(); 2129 2130 mStandby = true; 2131 } 2132 2133 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2134 // we're about to wait, flush the binder command buffer 2135 IPCThreadState::self()->flushCommands(); 2136 2137 clearOutputTracks(); 2138 2139 if (exitPending()) { 2140 break; 2141 } 2142 2143 releaseWakeLock_l(); 2144 // wait until we have something to do... 2145 ALOGV("%s going to sleep", myName.string()); 2146 mWaitWorkCV.wait(mLock); 2147 ALOGV("%s waking up", myName.string()); 2148 acquireWakeLock_l(); 2149 2150 mMixerStatus = MIXER_IDLE; 2151 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2152 mBytesWritten = 0; 2153 mBytesRemaining = 0; 2154 checkSilentMode_l(); 2155 2156 standbyTime = systemTime() + standbyDelay; 2157 sleepTime = idleSleepTime; 2158 if (mType == MIXER) { 2159 sleepTimeShift = 0; 2160 } 2161 2162 continue; 2163 } 2164 } 2165 2166 // mMixerStatusIgnoringFastTracks is also updated internally 2167 mMixerStatus = prepareTracks_l(&tracksToRemove); 2168 2169 // prevent any changes in effect chain list and in each effect chain 2170 // during mixing and effect process as the audio buffers could be deleted 2171 // or modified if an effect is created or deleted 2172 lockEffectChains_l(effectChains); 2173 } 2174 2175 if (mBytesRemaining == 0) { 2176 mCurrentWriteLength = 0; 2177 if (mMixerStatus == MIXER_TRACKS_READY) { 2178 // threadLoop_mix() sets mCurrentWriteLength 2179 threadLoop_mix(); 2180 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2181 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2182 // threadLoop_sleepTime sets sleepTime to 0 if data 2183 // must be written to HAL 2184 threadLoop_sleepTime(); 2185 if (sleepTime == 0) { 2186 mCurrentWriteLength = mixBufferSize; 2187 } 2188 } 2189 mBytesRemaining = mCurrentWriteLength; 2190 if (isSuspended()) { 2191 sleepTime = suspendSleepTimeUs(); 2192 // simulate write to HAL when suspended 2193 mBytesWritten += mixBufferSize; 2194 mBytesRemaining = 0; 2195 } 2196 2197 // only process effects if we're going to write 2198 if (sleepTime == 0) { 2199 for (size_t i = 0; i < effectChains.size(); i ++) { 2200 effectChains[i]->process_l(); 2201 } 2202 } 2203 } 2204 2205 // enable changes in effect chain 2206 unlockEffectChains(effectChains); 2207 2208 if (!waitingAsyncCallback()) { 2209 // sleepTime == 0 means we must write to audio hardware 2210 if (sleepTime == 0) { 2211 if (mBytesRemaining) { 2212 ssize_t ret = threadLoop_write(); 2213 if (ret < 0) { 2214 mBytesRemaining = 0; 2215 } else { 2216 mBytesWritten += ret; 2217 mBytesRemaining -= ret; 2218 } 2219 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2220 (mMixerStatus == MIXER_DRAIN_ALL)) { 2221 threadLoop_drain(); 2222 } 2223if (mType == MIXER) { 2224 // write blocked detection 2225 nsecs_t now = systemTime(); 2226 nsecs_t delta = now - mLastWriteTime; 2227 if (!mStandby && delta > maxPeriod) { 2228 mNumDelayedWrites++; 2229 if ((now - lastWarning) > kWarningThrottleNs) { 2230 ATRACE_NAME("underrun"); 2231 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2232 ns2ms(delta), mNumDelayedWrites, this); 2233 lastWarning = now; 2234 } 2235 } 2236} 2237 2238 mStandby = false; 2239 } else { 2240 usleep(sleepTime); 2241 } 2242 } 2243 2244 // Finally let go of removed track(s), without the lock held 2245 // since we can't guarantee the destructors won't acquire that 2246 // same lock. This will also mutate and push a new fast mixer state. 2247 threadLoop_removeTracks(tracksToRemove); 2248 tracksToRemove.clear(); 2249 2250 // FIXME I don't understand the need for this here; 2251 // it was in the original code but maybe the 2252 // assignment in saveOutputTracks() makes this unnecessary? 2253 clearOutputTracks(); 2254 2255 // Effect chains will be actually deleted here if they were removed from 2256 // mEffectChains list during mixing or effects processing 2257 effectChains.clear(); 2258 2259 // FIXME Note that the above .clear() is no longer necessary since effectChains 2260 // is now local to this block, but will keep it for now (at least until merge done). 2261 } 2262 2263 threadLoop_exit(); 2264 2265 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2266 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2267 // put output stream into standby mode 2268 if (!mStandby) { 2269 mOutput->stream->common.standby(&mOutput->stream->common); 2270 } 2271 } 2272 2273 releaseWakeLock(); 2274 2275 ALOGV("Thread %p type %d exiting", this, mType); 2276 return false; 2277} 2278 2279// removeTracks_l() must be called with ThreadBase::mLock held 2280void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2281{ 2282 size_t count = tracksToRemove.size(); 2283 if (count) { 2284 for (size_t i=0 ; i<count ; i++) { 2285 const sp<Track>& track = tracksToRemove.itemAt(i); 2286 mActiveTracks.remove(track); 2287 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2288 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2289 if (chain != 0) { 2290 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2291 track->sessionId()); 2292 chain->decActiveTrackCnt(); 2293 } 2294 if (track->isTerminated()) { 2295 removeTrack_l(track); 2296 } 2297 } 2298 } 2299 2300} 2301 2302// ---------------------------------------------------------------------------- 2303 2304AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2305 audio_io_handle_t id, audio_devices_t device, type_t type) 2306 : PlaybackThread(audioFlinger, output, id, device, type), 2307 // mAudioMixer below 2308 // mFastMixer below 2309 mFastMixerFutex(0) 2310 // mOutputSink below 2311 // mPipeSink below 2312 // mNormalSink below 2313{ 2314 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2315 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2316 "mFrameCount=%d, mNormalFrameCount=%d", 2317 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2318 mNormalFrameCount); 2319 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2320 2321 // FIXME - Current mixer implementation only supports stereo output 2322 if (mChannelCount != FCC_2) { 2323 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2324 } 2325 2326 // create an NBAIO sink for the HAL output stream, and negotiate 2327 mOutputSink = new AudioStreamOutSink(output->stream); 2328 size_t numCounterOffers = 0; 2329 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2330 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2331 ALOG_ASSERT(index == 0); 2332 2333 // initialize fast mixer depending on configuration 2334 bool initFastMixer; 2335 switch (kUseFastMixer) { 2336 case FastMixer_Never: 2337 initFastMixer = false; 2338 break; 2339 case FastMixer_Always: 2340 initFastMixer = true; 2341 break; 2342 case FastMixer_Static: 2343 case FastMixer_Dynamic: 2344 initFastMixer = mFrameCount < mNormalFrameCount; 2345 break; 2346 } 2347 if (initFastMixer) { 2348 2349 // create a MonoPipe to connect our submix to FastMixer 2350 NBAIO_Format format = mOutputSink->format(); 2351 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2352 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2353 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2354 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2355 const NBAIO_Format offers[1] = {format}; 2356 size_t numCounterOffers = 0; 2357 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2358 ALOG_ASSERT(index == 0); 2359 monoPipe->setAvgFrames((mScreenState & 1) ? 2360 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2361 mPipeSink = monoPipe; 2362 2363#ifdef TEE_SINK 2364 if (mTeeSinkOutputEnabled) { 2365 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2366 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2367 numCounterOffers = 0; 2368 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2369 ALOG_ASSERT(index == 0); 2370 mTeeSink = teeSink; 2371 PipeReader *teeSource = new PipeReader(*teeSink); 2372 numCounterOffers = 0; 2373 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2374 ALOG_ASSERT(index == 0); 2375 mTeeSource = teeSource; 2376 } 2377#endif 2378 2379 // create fast mixer and configure it initially with just one fast track for our submix 2380 mFastMixer = new FastMixer(); 2381 FastMixerStateQueue *sq = mFastMixer->sq(); 2382#ifdef STATE_QUEUE_DUMP 2383 sq->setObserverDump(&mStateQueueObserverDump); 2384 sq->setMutatorDump(&mStateQueueMutatorDump); 2385#endif 2386 FastMixerState *state = sq->begin(); 2387 FastTrack *fastTrack = &state->mFastTracks[0]; 2388 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2389 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2390 fastTrack->mVolumeProvider = NULL; 2391 fastTrack->mGeneration++; 2392 state->mFastTracksGen++; 2393 state->mTrackMask = 1; 2394 // fast mixer will use the HAL output sink 2395 state->mOutputSink = mOutputSink.get(); 2396 state->mOutputSinkGen++; 2397 state->mFrameCount = mFrameCount; 2398 state->mCommand = FastMixerState::COLD_IDLE; 2399 // already done in constructor initialization list 2400 //mFastMixerFutex = 0; 2401 state->mColdFutexAddr = &mFastMixerFutex; 2402 state->mColdGen++; 2403 state->mDumpState = &mFastMixerDumpState; 2404#ifdef TEE_SINK 2405 state->mTeeSink = mTeeSink.get(); 2406#endif 2407 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2408 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2409 sq->end(); 2410 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2411 2412 // start the fast mixer 2413 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2414 pid_t tid = mFastMixer->getTid(); 2415 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2416 if (err != 0) { 2417 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2418 kPriorityFastMixer, getpid_cached, tid, err); 2419 } 2420 2421#ifdef AUDIO_WATCHDOG 2422 // create and start the watchdog 2423 mAudioWatchdog = new AudioWatchdog(); 2424 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2425 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2426 tid = mAudioWatchdog->getTid(); 2427 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2428 if (err != 0) { 2429 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2430 kPriorityFastMixer, getpid_cached, tid, err); 2431 } 2432#endif 2433 2434 } else { 2435 mFastMixer = NULL; 2436 } 2437 2438 switch (kUseFastMixer) { 2439 case FastMixer_Never: 2440 case FastMixer_Dynamic: 2441 mNormalSink = mOutputSink; 2442 break; 2443 case FastMixer_Always: 2444 mNormalSink = mPipeSink; 2445 break; 2446 case FastMixer_Static: 2447 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2448 break; 2449 } 2450} 2451 2452AudioFlinger::MixerThread::~MixerThread() 2453{ 2454 if (mFastMixer != NULL) { 2455 FastMixerStateQueue *sq = mFastMixer->sq(); 2456 FastMixerState *state = sq->begin(); 2457 if (state->mCommand == FastMixerState::COLD_IDLE) { 2458 int32_t old = android_atomic_inc(&mFastMixerFutex); 2459 if (old == -1) { 2460 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2461 } 2462 } 2463 state->mCommand = FastMixerState::EXIT; 2464 sq->end(); 2465 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2466 mFastMixer->join(); 2467 // Though the fast mixer thread has exited, it's state queue is still valid. 2468 // We'll use that extract the final state which contains one remaining fast track 2469 // corresponding to our sub-mix. 2470 state = sq->begin(); 2471 ALOG_ASSERT(state->mTrackMask == 1); 2472 FastTrack *fastTrack = &state->mFastTracks[0]; 2473 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2474 delete fastTrack->mBufferProvider; 2475 sq->end(false /*didModify*/); 2476 delete mFastMixer; 2477#ifdef AUDIO_WATCHDOG 2478 if (mAudioWatchdog != 0) { 2479 mAudioWatchdog->requestExit(); 2480 mAudioWatchdog->requestExitAndWait(); 2481 mAudioWatchdog.clear(); 2482 } 2483#endif 2484 } 2485 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2486 delete mAudioMixer; 2487} 2488 2489 2490uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2491{ 2492 if (mFastMixer != NULL) { 2493 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2494 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2495 } 2496 return latency; 2497} 2498 2499 2500void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2501{ 2502 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2503} 2504 2505ssize_t AudioFlinger::MixerThread::threadLoop_write() 2506{ 2507 // FIXME we should only do one push per cycle; confirm this is true 2508 // Start the fast mixer if it's not already running 2509 if (mFastMixer != NULL) { 2510 FastMixerStateQueue *sq = mFastMixer->sq(); 2511 FastMixerState *state = sq->begin(); 2512 if (state->mCommand != FastMixerState::MIX_WRITE && 2513 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2514 if (state->mCommand == FastMixerState::COLD_IDLE) { 2515 int32_t old = android_atomic_inc(&mFastMixerFutex); 2516 if (old == -1) { 2517 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2518 } 2519#ifdef AUDIO_WATCHDOG 2520 if (mAudioWatchdog != 0) { 2521 mAudioWatchdog->resume(); 2522 } 2523#endif 2524 } 2525 state->mCommand = FastMixerState::MIX_WRITE; 2526 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2527 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2528 sq->end(); 2529 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2530 if (kUseFastMixer == FastMixer_Dynamic) { 2531 mNormalSink = mPipeSink; 2532 } 2533 } else { 2534 sq->end(false /*didModify*/); 2535 } 2536 } 2537 return PlaybackThread::threadLoop_write(); 2538} 2539 2540void AudioFlinger::MixerThread::threadLoop_standby() 2541{ 2542 // Idle the fast mixer if it's currently running 2543 if (mFastMixer != NULL) { 2544 FastMixerStateQueue *sq = mFastMixer->sq(); 2545 FastMixerState *state = sq->begin(); 2546 if (!(state->mCommand & FastMixerState::IDLE)) { 2547 state->mCommand = FastMixerState::COLD_IDLE; 2548 state->mColdFutexAddr = &mFastMixerFutex; 2549 state->mColdGen++; 2550 mFastMixerFutex = 0; 2551 sq->end(); 2552 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2553 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2554 if (kUseFastMixer == FastMixer_Dynamic) { 2555 mNormalSink = mOutputSink; 2556 } 2557#ifdef AUDIO_WATCHDOG 2558 if (mAudioWatchdog != 0) { 2559 mAudioWatchdog->pause(); 2560 } 2561#endif 2562 } else { 2563 sq->end(false /*didModify*/); 2564 } 2565 } 2566 PlaybackThread::threadLoop_standby(); 2567} 2568 2569// Empty implementation for standard mixer 2570// Overridden for offloaded playback 2571void AudioFlinger::PlaybackThread::flushOutput_l() 2572{ 2573} 2574 2575bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2576{ 2577 return false; 2578} 2579 2580bool AudioFlinger::PlaybackThread::shouldStandby_l() 2581{ 2582 return !mStandby; 2583} 2584 2585bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2586{ 2587 Mutex::Autolock _l(mLock); 2588 return waitingAsyncCallback_l(); 2589} 2590 2591// shared by MIXER and DIRECT, overridden by DUPLICATING 2592void AudioFlinger::PlaybackThread::threadLoop_standby() 2593{ 2594 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2595 mOutput->stream->common.standby(&mOutput->stream->common); 2596 if (mUseAsyncWrite != 0) { 2597 mWriteBlocked = false; 2598 mDraining = false; 2599 ALOG_ASSERT(mCallbackThread != 0); 2600 mCallbackThread->setWriteBlocked(false); 2601 mCallbackThread->setDraining(false); 2602 } 2603} 2604 2605void AudioFlinger::MixerThread::threadLoop_mix() 2606{ 2607 // obtain the presentation timestamp of the next output buffer 2608 int64_t pts; 2609 status_t status = INVALID_OPERATION; 2610 2611 if (mNormalSink != 0) { 2612 status = mNormalSink->getNextWriteTimestamp(&pts); 2613 } else { 2614 status = mOutputSink->getNextWriteTimestamp(&pts); 2615 } 2616 2617 if (status != NO_ERROR) { 2618 pts = AudioBufferProvider::kInvalidPTS; 2619 } 2620 2621 // mix buffers... 2622 mAudioMixer->process(pts); 2623 mCurrentWriteLength = mixBufferSize; 2624 // increase sleep time progressively when application underrun condition clears. 2625 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2626 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2627 // such that we would underrun the audio HAL. 2628 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2629 sleepTimeShift--; 2630 } 2631 sleepTime = 0; 2632 standbyTime = systemTime() + standbyDelay; 2633 //TODO: delay standby when effects have a tail 2634} 2635 2636void AudioFlinger::MixerThread::threadLoop_sleepTime() 2637{ 2638 // If no tracks are ready, sleep once for the duration of an output 2639 // buffer size, then write 0s to the output 2640 if (sleepTime == 0) { 2641 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2642 sleepTime = activeSleepTime >> sleepTimeShift; 2643 if (sleepTime < kMinThreadSleepTimeUs) { 2644 sleepTime = kMinThreadSleepTimeUs; 2645 } 2646 // reduce sleep time in case of consecutive application underruns to avoid 2647 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2648 // duration we would end up writing less data than needed by the audio HAL if 2649 // the condition persists. 2650 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2651 sleepTimeShift++; 2652 } 2653 } else { 2654 sleepTime = idleSleepTime; 2655 } 2656 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2657 memset (mMixBuffer, 0, mixBufferSize); 2658 sleepTime = 0; 2659 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2660 "anticipated start"); 2661 } 2662 // TODO add standby time extension fct of effect tail 2663} 2664 2665// prepareTracks_l() must be called with ThreadBase::mLock held 2666AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2667 Vector< sp<Track> > *tracksToRemove) 2668{ 2669 2670 mixer_state mixerStatus = MIXER_IDLE; 2671 // find out which tracks need to be processed 2672 size_t count = mActiveTracks.size(); 2673 size_t mixedTracks = 0; 2674 size_t tracksWithEffect = 0; 2675 // counts only _active_ fast tracks 2676 size_t fastTracks = 0; 2677 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2678 2679 float masterVolume = mMasterVolume; 2680 bool masterMute = mMasterMute; 2681 2682 if (masterMute) { 2683 masterVolume = 0; 2684 } 2685 // Delegate master volume control to effect in output mix effect chain if needed 2686 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2687 if (chain != 0) { 2688 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2689 chain->setVolume_l(&v, &v); 2690 masterVolume = (float)((v + (1 << 23)) >> 24); 2691 chain.clear(); 2692 } 2693 2694 // prepare a new state to push 2695 FastMixerStateQueue *sq = NULL; 2696 FastMixerState *state = NULL; 2697 bool didModify = false; 2698 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2699 if (mFastMixer != NULL) { 2700 sq = mFastMixer->sq(); 2701 state = sq->begin(); 2702 } 2703 2704 for (size_t i=0 ; i<count ; i++) { 2705 const sp<Track> t = mActiveTracks[i].promote(); 2706 if (t == 0) { 2707 continue; 2708 } 2709 2710 // this const just means the local variable doesn't change 2711 Track* const track = t.get(); 2712 2713 // process fast tracks 2714 if (track->isFastTrack()) { 2715 2716 // It's theoretically possible (though unlikely) for a fast track to be created 2717 // and then removed within the same normal mix cycle. This is not a problem, as 2718 // the track never becomes active so it's fast mixer slot is never touched. 2719 // The converse, of removing an (active) track and then creating a new track 2720 // at the identical fast mixer slot within the same normal mix cycle, 2721 // is impossible because the slot isn't marked available until the end of each cycle. 2722 int j = track->mFastIndex; 2723 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2724 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2725 FastTrack *fastTrack = &state->mFastTracks[j]; 2726 2727 // Determine whether the track is currently in underrun condition, 2728 // and whether it had a recent underrun. 2729 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2730 FastTrackUnderruns underruns = ftDump->mUnderruns; 2731 uint32_t recentFull = (underruns.mBitFields.mFull - 2732 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2733 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2734 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2735 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2736 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2737 uint32_t recentUnderruns = recentPartial + recentEmpty; 2738 track->mObservedUnderruns = underruns; 2739 // don't count underruns that occur while stopping or pausing 2740 // or stopped which can occur when flush() is called while active 2741 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2742 recentUnderruns > 0) { 2743 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2744 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2745 } 2746 2747 // This is similar to the state machine for normal tracks, 2748 // with a few modifications for fast tracks. 2749 bool isActive = true; 2750 switch (track->mState) { 2751 case TrackBase::STOPPING_1: 2752 // track stays active in STOPPING_1 state until first underrun 2753 if (recentUnderruns > 0 || track->isTerminated()) { 2754 track->mState = TrackBase::STOPPING_2; 2755 } 2756 break; 2757 case TrackBase::PAUSING: 2758 // ramp down is not yet implemented 2759 track->setPaused(); 2760 break; 2761 case TrackBase::RESUMING: 2762 // ramp up is not yet implemented 2763 track->mState = TrackBase::ACTIVE; 2764 break; 2765 case TrackBase::ACTIVE: 2766 if (recentFull > 0 || recentPartial > 0) { 2767 // track has provided at least some frames recently: reset retry count 2768 track->mRetryCount = kMaxTrackRetries; 2769 } 2770 if (recentUnderruns == 0) { 2771 // no recent underruns: stay active 2772 break; 2773 } 2774 // there has recently been an underrun of some kind 2775 if (track->sharedBuffer() == 0) { 2776 // were any of the recent underruns "empty" (no frames available)? 2777 if (recentEmpty == 0) { 2778 // no, then ignore the partial underruns as they are allowed indefinitely 2779 break; 2780 } 2781 // there has recently been an "empty" underrun: decrement the retry counter 2782 if (--(track->mRetryCount) > 0) { 2783 break; 2784 } 2785 // indicate to client process that the track was disabled because of underrun; 2786 // it will then automatically call start() when data is available 2787 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 2788 // remove from active list, but state remains ACTIVE [confusing but true] 2789 isActive = false; 2790 break; 2791 } 2792 // fall through 2793 case TrackBase::STOPPING_2: 2794 case TrackBase::PAUSED: 2795 case TrackBase::STOPPED: 2796 case TrackBase::FLUSHED: // flush() while active 2797 // Check for presentation complete if track is inactive 2798 // We have consumed all the buffers of this track. 2799 // This would be incomplete if we auto-paused on underrun 2800 { 2801 size_t audioHALFrames = 2802 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2803 size_t framesWritten = mBytesWritten / mFrameSize; 2804 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 2805 // track stays in active list until presentation is complete 2806 break; 2807 } 2808 } 2809 if (track->isStopping_2()) { 2810 track->mState = TrackBase::STOPPED; 2811 } 2812 if (track->isStopped()) { 2813 // Can't reset directly, as fast mixer is still polling this track 2814 // track->reset(); 2815 // So instead mark this track as needing to be reset after push with ack 2816 resetMask |= 1 << i; 2817 } 2818 isActive = false; 2819 break; 2820 case TrackBase::IDLE: 2821 default: 2822 LOG_FATAL("unexpected track state %d", track->mState); 2823 } 2824 2825 if (isActive) { 2826 // was it previously inactive? 2827 if (!(state->mTrackMask & (1 << j))) { 2828 ExtendedAudioBufferProvider *eabp = track; 2829 VolumeProvider *vp = track; 2830 fastTrack->mBufferProvider = eabp; 2831 fastTrack->mVolumeProvider = vp; 2832 fastTrack->mSampleRate = track->mSampleRate; 2833 fastTrack->mChannelMask = track->mChannelMask; 2834 fastTrack->mGeneration++; 2835 state->mTrackMask |= 1 << j; 2836 didModify = true; 2837 // no acknowledgement required for newly active tracks 2838 } 2839 // cache the combined master volume and stream type volume for fast mixer; this 2840 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2841 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 2842 ++fastTracks; 2843 } else { 2844 // was it previously active? 2845 if (state->mTrackMask & (1 << j)) { 2846 fastTrack->mBufferProvider = NULL; 2847 fastTrack->mGeneration++; 2848 state->mTrackMask &= ~(1 << j); 2849 didModify = true; 2850 // If any fast tracks were removed, we must wait for acknowledgement 2851 // because we're about to decrement the last sp<> on those tracks. 2852 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2853 } else { 2854 LOG_FATAL("fast track %d should have been active", j); 2855 } 2856 tracksToRemove->add(track); 2857 // Avoids a misleading display in dumpsys 2858 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2859 } 2860 continue; 2861 } 2862 2863 { // local variable scope to avoid goto warning 2864 2865 audio_track_cblk_t* cblk = track->cblk(); 2866 2867 // The first time a track is added we wait 2868 // for all its buffers to be filled before processing it 2869 int name = track->name(); 2870 // make sure that we have enough frames to mix one full buffer. 2871 // enforce this condition only once to enable draining the buffer in case the client 2872 // app does not call stop() and relies on underrun to stop: 2873 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2874 // during last round 2875 size_t desiredFrames; 2876 uint32_t sr = track->sampleRate(); 2877 if (sr == mSampleRate) { 2878 desiredFrames = mNormalFrameCount; 2879 } else { 2880 // +1 for rounding and +1 for additional sample needed for interpolation 2881 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 2882 // add frames already consumed but not yet released by the resampler 2883 // because cblk->framesReady() will include these frames 2884 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2885 // the minimum track buffer size is normally twice the number of frames necessary 2886 // to fill one buffer and the resampler should not leave more than one buffer worth 2887 // of unreleased frames after each pass, but just in case... 2888 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 2889 } 2890 uint32_t minFrames = 1; 2891 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2892 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2893 minFrames = desiredFrames; 2894 } 2895 // It's not safe to call framesReady() for a static buffer track, so assume it's ready 2896 size_t framesReady; 2897 if (track->sharedBuffer() == 0) { 2898 framesReady = track->framesReady(); 2899 } else if (track->isStopped()) { 2900 framesReady = 0; 2901 } else { 2902 framesReady = 1; 2903 } 2904 if ((framesReady >= minFrames) && track->isReady() && 2905 !track->isPaused() && !track->isTerminated()) 2906 { 2907 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 2908 2909 mixedTracks++; 2910 2911 // track->mainBuffer() != mMixBuffer means there is an effect chain 2912 // connected to the track 2913 chain.clear(); 2914 if (track->mainBuffer() != mMixBuffer) { 2915 chain = getEffectChain_l(track->sessionId()); 2916 // Delegate volume control to effect in track effect chain if needed 2917 if (chain != 0) { 2918 tracksWithEffect++; 2919 } else { 2920 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 2921 "session %d", 2922 name, track->sessionId()); 2923 } 2924 } 2925 2926 2927 int param = AudioMixer::VOLUME; 2928 if (track->mFillingUpStatus == Track::FS_FILLED) { 2929 // no ramp for the first volume setting 2930 track->mFillingUpStatus = Track::FS_ACTIVE; 2931 if (track->mState == TrackBase::RESUMING) { 2932 track->mState = TrackBase::ACTIVE; 2933 param = AudioMixer::RAMP_VOLUME; 2934 } 2935 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2936 // FIXME should not make a decision based on mServer 2937 } else if (cblk->mServer != 0) { 2938 // If the track is stopped before the first frame was mixed, 2939 // do not apply ramp 2940 param = AudioMixer::RAMP_VOLUME; 2941 } 2942 2943 // compute volume for this track 2944 uint32_t vl, vr, va; 2945 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 2946 vl = vr = va = 0; 2947 if (track->isPausing()) { 2948 track->setPaused(); 2949 } 2950 } else { 2951 2952 // read original volumes with volume control 2953 float typeVolume = mStreamTypes[track->streamType()].volume; 2954 float v = masterVolume * typeVolume; 2955 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 2956 uint32_t vlr = proxy->getVolumeLR(); 2957 vl = vlr & 0xFFFF; 2958 vr = vlr >> 16; 2959 // track volumes come from shared memory, so can't be trusted and must be clamped 2960 if (vl > MAX_GAIN_INT) { 2961 ALOGV("Track left volume out of range: %04X", vl); 2962 vl = MAX_GAIN_INT; 2963 } 2964 if (vr > MAX_GAIN_INT) { 2965 ALOGV("Track right volume out of range: %04X", vr); 2966 vr = MAX_GAIN_INT; 2967 } 2968 // now apply the master volume and stream type volume 2969 vl = (uint32_t)(v * vl) << 12; 2970 vr = (uint32_t)(v * vr) << 12; 2971 // assuming master volume and stream type volume each go up to 1.0, 2972 // vl and vr are now in 8.24 format 2973 2974 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 2975 // send level comes from shared memory and so may be corrupt 2976 if (sendLevel > MAX_GAIN_INT) { 2977 ALOGV("Track send level out of range: %04X", sendLevel); 2978 sendLevel = MAX_GAIN_INT; 2979 } 2980 va = (uint32_t)(v * sendLevel); 2981 } 2982 2983 // Delegate volume control to effect in track effect chain if needed 2984 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2985 // Do not ramp volume if volume is controlled by effect 2986 param = AudioMixer::VOLUME; 2987 track->mHasVolumeController = true; 2988 } else { 2989 // force no volume ramp when volume controller was just disabled or removed 2990 // from effect chain to avoid volume spike 2991 if (track->mHasVolumeController) { 2992 param = AudioMixer::VOLUME; 2993 } 2994 track->mHasVolumeController = false; 2995 } 2996 2997 // Convert volumes from 8.24 to 4.12 format 2998 // This additional clamping is needed in case chain->setVolume_l() overshot 2999 vl = (vl + (1 << 11)) >> 12; 3000 if (vl > MAX_GAIN_INT) { 3001 vl = MAX_GAIN_INT; 3002 } 3003 vr = (vr + (1 << 11)) >> 12; 3004 if (vr > MAX_GAIN_INT) { 3005 vr = MAX_GAIN_INT; 3006 } 3007 3008 if (va > MAX_GAIN_INT) { 3009 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3010 } 3011 3012 // XXX: these things DON'T need to be done each time 3013 mAudioMixer->setBufferProvider(name, track); 3014 mAudioMixer->enable(name); 3015 3016 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3017 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3018 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3019 mAudioMixer->setParameter( 3020 name, 3021 AudioMixer::TRACK, 3022 AudioMixer::FORMAT, (void *)track->format()); 3023 mAudioMixer->setParameter( 3024 name, 3025 AudioMixer::TRACK, 3026 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3027 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3028 uint32_t maxSampleRate = mSampleRate * 2; 3029 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3030 if (reqSampleRate == 0) { 3031 reqSampleRate = mSampleRate; 3032 } else if (reqSampleRate > maxSampleRate) { 3033 reqSampleRate = maxSampleRate; 3034 } 3035 mAudioMixer->setParameter( 3036 name, 3037 AudioMixer::RESAMPLE, 3038 AudioMixer::SAMPLE_RATE, 3039 (void *)reqSampleRate); 3040 mAudioMixer->setParameter( 3041 name, 3042 AudioMixer::TRACK, 3043 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3044 mAudioMixer->setParameter( 3045 name, 3046 AudioMixer::TRACK, 3047 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3048 3049 // reset retry count 3050 track->mRetryCount = kMaxTrackRetries; 3051 3052 // If one track is ready, set the mixer ready if: 3053 // - the mixer was not ready during previous round OR 3054 // - no other track is not ready 3055 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3056 mixerStatus != MIXER_TRACKS_ENABLED) { 3057 mixerStatus = MIXER_TRACKS_READY; 3058 } 3059 } else { 3060 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3061 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3062 } 3063 // clear effect chain input buffer if an active track underruns to avoid sending 3064 // previous audio buffer again to effects 3065 chain = getEffectChain_l(track->sessionId()); 3066 if (chain != 0) { 3067 chain->clearInputBuffer(); 3068 } 3069 3070 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3071 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3072 track->isStopped() || track->isPaused()) { 3073 // We have consumed all the buffers of this track. 3074 // Remove it from the list of active tracks. 3075 // TODO: use actual buffer filling status instead of latency when available from 3076 // audio HAL 3077 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3078 size_t framesWritten = mBytesWritten / mFrameSize; 3079 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3080 if (track->isStopped()) { 3081 track->reset(); 3082 } 3083 tracksToRemove->add(track); 3084 } 3085 } else { 3086 // No buffers for this track. Give it a few chances to 3087 // fill a buffer, then remove it from active list. 3088 if (--(track->mRetryCount) <= 0) { 3089 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3090 tracksToRemove->add(track); 3091 // indicate to client process that the track was disabled because of underrun; 3092 // it will then automatically call start() when data is available 3093 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3094 // If one track is not ready, mark the mixer also not ready if: 3095 // - the mixer was ready during previous round OR 3096 // - no other track is ready 3097 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3098 mixerStatus != MIXER_TRACKS_READY) { 3099 mixerStatus = MIXER_TRACKS_ENABLED; 3100 } 3101 } 3102 mAudioMixer->disable(name); 3103 } 3104 3105 } // local variable scope to avoid goto warning 3106track_is_ready: ; 3107 3108 } 3109 3110 // Push the new FastMixer state if necessary 3111 bool pauseAudioWatchdog = false; 3112 if (didModify) { 3113 state->mFastTracksGen++; 3114 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3115 if (kUseFastMixer == FastMixer_Dynamic && 3116 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3117 state->mCommand = FastMixerState::COLD_IDLE; 3118 state->mColdFutexAddr = &mFastMixerFutex; 3119 state->mColdGen++; 3120 mFastMixerFutex = 0; 3121 if (kUseFastMixer == FastMixer_Dynamic) { 3122 mNormalSink = mOutputSink; 3123 } 3124 // If we go into cold idle, need to wait for acknowledgement 3125 // so that fast mixer stops doing I/O. 3126 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3127 pauseAudioWatchdog = true; 3128 } 3129 } 3130 if (sq != NULL) { 3131 sq->end(didModify); 3132 sq->push(block); 3133 } 3134#ifdef AUDIO_WATCHDOG 3135 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3136 mAudioWatchdog->pause(); 3137 } 3138#endif 3139 3140 // Now perform the deferred reset on fast tracks that have stopped 3141 while (resetMask != 0) { 3142 size_t i = __builtin_ctz(resetMask); 3143 ALOG_ASSERT(i < count); 3144 resetMask &= ~(1 << i); 3145 sp<Track> t = mActiveTracks[i].promote(); 3146 if (t == 0) { 3147 continue; 3148 } 3149 Track* track = t.get(); 3150 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3151 track->reset(); 3152 } 3153 3154 // remove all the tracks that need to be... 3155 removeTracks_l(*tracksToRemove); 3156 3157 // mix buffer must be cleared if all tracks are connected to an 3158 // effect chain as in this case the mixer will not write to 3159 // mix buffer and track effects will accumulate into it 3160 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3161 (mixedTracks == 0 && fastTracks > 0))) { 3162 // FIXME as a performance optimization, should remember previous zero status 3163 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3164 } 3165 3166 // if any fast tracks, then status is ready 3167 mMixerStatusIgnoringFastTracks = mixerStatus; 3168 if (fastTracks > 0) { 3169 mixerStatus = MIXER_TRACKS_READY; 3170 } 3171 return mixerStatus; 3172} 3173 3174// getTrackName_l() must be called with ThreadBase::mLock held 3175int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3176{ 3177 return mAudioMixer->getTrackName(channelMask, sessionId); 3178} 3179 3180// deleteTrackName_l() must be called with ThreadBase::mLock held 3181void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3182{ 3183 ALOGV("remove track (%d) and delete from mixer", name); 3184 mAudioMixer->deleteTrackName(name); 3185} 3186 3187// checkForNewParameters_l() must be called with ThreadBase::mLock held 3188bool AudioFlinger::MixerThread::checkForNewParameters_l() 3189{ 3190 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3191 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3192 bool reconfig = false; 3193 3194 while (!mNewParameters.isEmpty()) { 3195 3196 if (mFastMixer != NULL) { 3197 FastMixerStateQueue *sq = mFastMixer->sq(); 3198 FastMixerState *state = sq->begin(); 3199 if (!(state->mCommand & FastMixerState::IDLE)) { 3200 previousCommand = state->mCommand; 3201 state->mCommand = FastMixerState::HOT_IDLE; 3202 sq->end(); 3203 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3204 } else { 3205 sq->end(false /*didModify*/); 3206 } 3207 } 3208 3209 status_t status = NO_ERROR; 3210 String8 keyValuePair = mNewParameters[0]; 3211 AudioParameter param = AudioParameter(keyValuePair); 3212 int value; 3213 3214 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3215 reconfig = true; 3216 } 3217 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3218 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3219 status = BAD_VALUE; 3220 } else { 3221 reconfig = true; 3222 } 3223 } 3224 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3225 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3226 status = BAD_VALUE; 3227 } else { 3228 reconfig = true; 3229 } 3230 } 3231 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3232 // do not accept frame count changes if tracks are open as the track buffer 3233 // size depends on frame count and correct behavior would not be guaranteed 3234 // if frame count is changed after track creation 3235 if (!mTracks.isEmpty()) { 3236 status = INVALID_OPERATION; 3237 } else { 3238 reconfig = true; 3239 } 3240 } 3241 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3242#ifdef ADD_BATTERY_DATA 3243 // when changing the audio output device, call addBatteryData to notify 3244 // the change 3245 if (mOutDevice != value) { 3246 uint32_t params = 0; 3247 // check whether speaker is on 3248 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3249 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3250 } 3251 3252 audio_devices_t deviceWithoutSpeaker 3253 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3254 // check if any other device (except speaker) is on 3255 if (value & deviceWithoutSpeaker ) { 3256 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3257 } 3258 3259 if (params != 0) { 3260 addBatteryData(params); 3261 } 3262 } 3263#endif 3264 3265 // forward device change to effects that have requested to be 3266 // aware of attached audio device. 3267 if (value != AUDIO_DEVICE_NONE) { 3268 mOutDevice = value; 3269 for (size_t i = 0; i < mEffectChains.size(); i++) { 3270 mEffectChains[i]->setDevice_l(mOutDevice); 3271 } 3272 } 3273 } 3274 3275 if (status == NO_ERROR) { 3276 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3277 keyValuePair.string()); 3278 if (!mStandby && status == INVALID_OPERATION) { 3279 mOutput->stream->common.standby(&mOutput->stream->common); 3280 mStandby = true; 3281 mBytesWritten = 0; 3282 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3283 keyValuePair.string()); 3284 } 3285 if (status == NO_ERROR && reconfig) { 3286 readOutputParameters(); 3287 delete mAudioMixer; 3288 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3289 for (size_t i = 0; i < mTracks.size() ; i++) { 3290 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3291 if (name < 0) { 3292 break; 3293 } 3294 mTracks[i]->mName = name; 3295 } 3296 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3297 } 3298 } 3299 3300 mNewParameters.removeAt(0); 3301 3302 mParamStatus = status; 3303 mParamCond.signal(); 3304 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3305 // already timed out waiting for the status and will never signal the condition. 3306 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3307 } 3308 3309 if (!(previousCommand & FastMixerState::IDLE)) { 3310 ALOG_ASSERT(mFastMixer != NULL); 3311 FastMixerStateQueue *sq = mFastMixer->sq(); 3312 FastMixerState *state = sq->begin(); 3313 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3314 state->mCommand = previousCommand; 3315 sq->end(); 3316 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3317 } 3318 3319 return reconfig; 3320} 3321 3322 3323void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3324{ 3325 const size_t SIZE = 256; 3326 char buffer[SIZE]; 3327 String8 result; 3328 3329 PlaybackThread::dumpInternals(fd, args); 3330 3331 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3332 result.append(buffer); 3333 write(fd, result.string(), result.size()); 3334 3335 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3336 const FastMixerDumpState copy(mFastMixerDumpState); 3337 copy.dump(fd); 3338 3339#ifdef STATE_QUEUE_DUMP 3340 // Similar for state queue 3341 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3342 observerCopy.dump(fd); 3343 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3344 mutatorCopy.dump(fd); 3345#endif 3346 3347#ifdef TEE_SINK 3348 // Write the tee output to a .wav file 3349 dumpTee(fd, mTeeSource, mId); 3350#endif 3351 3352#ifdef AUDIO_WATCHDOG 3353 if (mAudioWatchdog != 0) { 3354 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3355 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3356 wdCopy.dump(fd); 3357 } 3358#endif 3359} 3360 3361uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3362{ 3363 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3364} 3365 3366uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3367{ 3368 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3369} 3370 3371void AudioFlinger::MixerThread::cacheParameters_l() 3372{ 3373 PlaybackThread::cacheParameters_l(); 3374 3375 // FIXME: Relaxed timing because of a certain device that can't meet latency 3376 // Should be reduced to 2x after the vendor fixes the driver issue 3377 // increase threshold again due to low power audio mode. The way this warning 3378 // threshold is calculated and its usefulness should be reconsidered anyway. 3379 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3380} 3381 3382// ---------------------------------------------------------------------------- 3383 3384AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3385 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3386 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3387 // mLeftVolFloat, mRightVolFloat 3388{ 3389} 3390 3391AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3392 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3393 ThreadBase::type_t type) 3394 : PlaybackThread(audioFlinger, output, id, device, type) 3395 // mLeftVolFloat, mRightVolFloat 3396{ 3397} 3398 3399AudioFlinger::DirectOutputThread::~DirectOutputThread() 3400{ 3401} 3402 3403void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3404{ 3405 audio_track_cblk_t* cblk = track->cblk(); 3406 float left, right; 3407 3408 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3409 left = right = 0; 3410 } else { 3411 float typeVolume = mStreamTypes[track->streamType()].volume; 3412 float v = mMasterVolume * typeVolume; 3413 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3414 uint32_t vlr = proxy->getVolumeLR(); 3415 float v_clamped = v * (vlr & 0xFFFF); 3416 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3417 left = v_clamped/MAX_GAIN; 3418 v_clamped = v * (vlr >> 16); 3419 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3420 right = v_clamped/MAX_GAIN; 3421 } 3422 3423 if (lastTrack) { 3424 if (left != mLeftVolFloat || right != mRightVolFloat) { 3425 mLeftVolFloat = left; 3426 mRightVolFloat = right; 3427 3428 // Convert volumes from float to 8.24 3429 uint32_t vl = (uint32_t)(left * (1 << 24)); 3430 uint32_t vr = (uint32_t)(right * (1 << 24)); 3431 3432 // Delegate volume control to effect in track effect chain if needed 3433 // only one effect chain can be present on DirectOutputThread, so if 3434 // there is one, the track is connected to it 3435 if (!mEffectChains.isEmpty()) { 3436 mEffectChains[0]->setVolume_l(&vl, &vr); 3437 left = (float)vl / (1 << 24); 3438 right = (float)vr / (1 << 24); 3439 } 3440 if (mOutput->stream->set_volume) { 3441 mOutput->stream->set_volume(mOutput->stream, left, right); 3442 } 3443 } 3444 } 3445} 3446 3447 3448AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3449 Vector< sp<Track> > *tracksToRemove 3450) 3451{ 3452 size_t count = mActiveTracks.size(); 3453 mixer_state mixerStatus = MIXER_IDLE; 3454 3455 // find out which tracks need to be processed 3456 for (size_t i = 0; i < count; i++) { 3457 sp<Track> t = mActiveTracks[i].promote(); 3458 // The track died recently 3459 if (t == 0) { 3460 continue; 3461 } 3462 3463 Track* const track = t.get(); 3464 audio_track_cblk_t* cblk = track->cblk(); 3465 3466 // The first time a track is added we wait 3467 // for all its buffers to be filled before processing it 3468 uint32_t minFrames; 3469 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3470 minFrames = mNormalFrameCount; 3471 } else { 3472 minFrames = 1; 3473 } 3474 // Only consider last track started for volume and mixer state control. 3475 // This is the last entry in mActiveTracks unless a track underruns. 3476 // As we only care about the transition phase between two tracks on a 3477 // direct output, it is not a problem to ignore the underrun case. 3478 bool last = (i == (count - 1)); 3479 3480 if ((track->framesReady() >= minFrames) && track->isReady() && 3481 !track->isPaused() && !track->isTerminated()) 3482 { 3483 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3484 3485 if (track->mFillingUpStatus == Track::FS_FILLED) { 3486 track->mFillingUpStatus = Track::FS_ACTIVE; 3487 mLeftVolFloat = mRightVolFloat = 0; 3488 if (track->mState == TrackBase::RESUMING) { 3489 track->mState = TrackBase::ACTIVE; 3490 } 3491 } 3492 3493 // compute volume for this track 3494 processVolume_l(track, last); 3495 if (last) { 3496 // reset retry count 3497 track->mRetryCount = kMaxTrackRetriesDirect; 3498 mActiveTrack = t; 3499 mixerStatus = MIXER_TRACKS_READY; 3500 } 3501 } else { 3502 // clear effect chain input buffer if the last active track started underruns 3503 // to avoid sending previous audio buffer again to effects 3504 if (!mEffectChains.isEmpty() && (i == (count -1))) { 3505 mEffectChains[0]->clearInputBuffer(); 3506 } 3507 3508 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3509 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3510 track->isStopped() || track->isPaused()) { 3511 // We have consumed all the buffers of this track. 3512 // Remove it from the list of active tracks. 3513 // TODO: implement behavior for compressed audio 3514 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3515 size_t framesWritten = mBytesWritten / mFrameSize; 3516 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3517 if (track->isStopped()) { 3518 track->reset(); 3519 } 3520 tracksToRemove->add(track); 3521 } 3522 } else { 3523 // No buffers for this track. Give it a few chances to 3524 // fill a buffer, then remove it from active list. 3525 // Only consider last track started for mixer state control 3526 if (--(track->mRetryCount) <= 0) { 3527 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3528 tracksToRemove->add(track); 3529 } else if (last) { 3530 mixerStatus = MIXER_TRACKS_ENABLED; 3531 } 3532 } 3533 } 3534 } 3535 3536 // remove all the tracks that need to be... 3537 removeTracks_l(*tracksToRemove); 3538 3539 return mixerStatus; 3540} 3541 3542void AudioFlinger::DirectOutputThread::threadLoop_mix() 3543{ 3544 size_t frameCount = mFrameCount; 3545 int8_t *curBuf = (int8_t *)mMixBuffer; 3546 // output audio to hardware 3547 while (frameCount) { 3548 AudioBufferProvider::Buffer buffer; 3549 buffer.frameCount = frameCount; 3550 mActiveTrack->getNextBuffer(&buffer); 3551 if (buffer.raw == NULL) { 3552 memset(curBuf, 0, frameCount * mFrameSize); 3553 break; 3554 } 3555 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3556 frameCount -= buffer.frameCount; 3557 curBuf += buffer.frameCount * mFrameSize; 3558 mActiveTrack->releaseBuffer(&buffer); 3559 } 3560 mCurrentWriteLength = curBuf - (int8_t *)mMixBuffer; 3561 sleepTime = 0; 3562 standbyTime = systemTime() + standbyDelay; 3563 mActiveTrack.clear(); 3564} 3565 3566void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3567{ 3568 if (sleepTime == 0) { 3569 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3570 sleepTime = activeSleepTime; 3571 } else { 3572 sleepTime = idleSleepTime; 3573 } 3574 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3575 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3576 sleepTime = 0; 3577 } 3578} 3579 3580// getTrackName_l() must be called with ThreadBase::mLock held 3581int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask, 3582 int sessionId) 3583{ 3584 return 0; 3585} 3586 3587// deleteTrackName_l() must be called with ThreadBase::mLock held 3588void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3589{ 3590} 3591 3592// checkForNewParameters_l() must be called with ThreadBase::mLock held 3593bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3594{ 3595 bool reconfig = false; 3596 3597 while (!mNewParameters.isEmpty()) { 3598 status_t status = NO_ERROR; 3599 String8 keyValuePair = mNewParameters[0]; 3600 AudioParameter param = AudioParameter(keyValuePair); 3601 int value; 3602 3603 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3604 // do not accept frame count changes if tracks are open as the track buffer 3605 // size depends on frame count and correct behavior would not be garantied 3606 // if frame count is changed after track creation 3607 if (!mTracks.isEmpty()) { 3608 status = INVALID_OPERATION; 3609 } else { 3610 reconfig = true; 3611 } 3612 } 3613 if (status == NO_ERROR) { 3614 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3615 keyValuePair.string()); 3616 if (!mStandby && status == INVALID_OPERATION) { 3617 mOutput->stream->common.standby(&mOutput->stream->common); 3618 mStandby = true; 3619 mBytesWritten = 0; 3620 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3621 keyValuePair.string()); 3622 } 3623 if (status == NO_ERROR && reconfig) { 3624 readOutputParameters(); 3625 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3626 } 3627 } 3628 3629 mNewParameters.removeAt(0); 3630 3631 mParamStatus = status; 3632 mParamCond.signal(); 3633 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3634 // already timed out waiting for the status and will never signal the condition. 3635 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3636 } 3637 return reconfig; 3638} 3639 3640uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3641{ 3642 uint32_t time; 3643 if (audio_is_linear_pcm(mFormat)) { 3644 time = PlaybackThread::activeSleepTimeUs(); 3645 } else { 3646 time = 10000; 3647 } 3648 return time; 3649} 3650 3651uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3652{ 3653 uint32_t time; 3654 if (audio_is_linear_pcm(mFormat)) { 3655 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3656 } else { 3657 time = 10000; 3658 } 3659 return time; 3660} 3661 3662uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3663{ 3664 uint32_t time; 3665 if (audio_is_linear_pcm(mFormat)) { 3666 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3667 } else { 3668 time = 10000; 3669 } 3670 return time; 3671} 3672 3673void AudioFlinger::DirectOutputThread::cacheParameters_l() 3674{ 3675 PlaybackThread::cacheParameters_l(); 3676 3677 // use shorter standby delay as on normal output to release 3678 // hardware resources as soon as possible 3679 standbyDelay = microseconds(activeSleepTime*2); 3680} 3681 3682// ---------------------------------------------------------------------------- 3683 3684AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3685 const sp<AudioFlinger::OffloadThread>& offloadThread) 3686 : Thread(false /*canCallJava*/), 3687 mOffloadThread(offloadThread), 3688 mWriteBlocked(false), 3689 mDraining(false) 3690{ 3691} 3692 3693AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3694{ 3695} 3696 3697void AudioFlinger::AsyncCallbackThread::onFirstRef() 3698{ 3699 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3700} 3701 3702bool AudioFlinger::AsyncCallbackThread::threadLoop() 3703{ 3704 while (!exitPending()) { 3705 bool writeBlocked; 3706 bool draining; 3707 3708 { 3709 Mutex::Autolock _l(mLock); 3710 mWaitWorkCV.wait(mLock); 3711 if (exitPending()) { 3712 break; 3713 } 3714 writeBlocked = mWriteBlocked; 3715 draining = mDraining; 3716 ALOGV("AsyncCallbackThread mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3717 } 3718 { 3719 sp<AudioFlinger::OffloadThread> offloadThread = mOffloadThread.promote(); 3720 if (offloadThread != 0) { 3721 if (writeBlocked == false) { 3722 offloadThread->setWriteBlocked(false); 3723 } 3724 if (draining == false) { 3725 offloadThread->setDraining(false); 3726 } 3727 } 3728 } 3729 } 3730 return false; 3731} 3732 3733void AudioFlinger::AsyncCallbackThread::exit() 3734{ 3735 ALOGV("AsyncCallbackThread::exit"); 3736 Mutex::Autolock _l(mLock); 3737 requestExit(); 3738 mWaitWorkCV.broadcast(); 3739} 3740 3741void AudioFlinger::AsyncCallbackThread::setWriteBlocked(bool value) 3742{ 3743 Mutex::Autolock _l(mLock); 3744 mWriteBlocked = value; 3745 if (!value) { 3746 mWaitWorkCV.signal(); 3747 } 3748} 3749 3750void AudioFlinger::AsyncCallbackThread::setDraining(bool value) 3751{ 3752 Mutex::Autolock _l(mLock); 3753 mDraining = value; 3754 if (!value) { 3755 mWaitWorkCV.signal(); 3756 } 3757} 3758 3759 3760// ---------------------------------------------------------------------------- 3761AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 3762 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3763 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 3764 mHwPaused(false), 3765 mPausedBytesRemaining(0) 3766{ 3767 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 3768} 3769 3770AudioFlinger::OffloadThread::~OffloadThread() 3771{ 3772 mPreviousTrack.clear(); 3773} 3774 3775void AudioFlinger::OffloadThread::threadLoop_exit() 3776{ 3777 if (mFlushPending || mHwPaused) { 3778 // If a flush is pending or track was paused, just discard buffered data 3779 flushHw_l(); 3780 } else { 3781 mMixerStatus = MIXER_DRAIN_ALL; 3782 threadLoop_drain(); 3783 } 3784 mCallbackThread->exit(); 3785 PlaybackThread::threadLoop_exit(); 3786} 3787 3788AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 3789 Vector< sp<Track> > *tracksToRemove 3790) 3791{ 3792 ALOGV("OffloadThread::prepareTracks_l"); 3793 size_t count = mActiveTracks.size(); 3794 3795 mixer_state mixerStatus = MIXER_IDLE; 3796 if (mFlushPending) { 3797 flushHw_l(); 3798 mFlushPending = false; 3799 } 3800 // find out which tracks need to be processed 3801 for (size_t i = 0; i < count; i++) { 3802 sp<Track> t = mActiveTracks[i].promote(); 3803 // The track died recently 3804 if (t == 0) { 3805 continue; 3806 } 3807 Track* const track = t.get(); 3808 audio_track_cblk_t* cblk = track->cblk(); 3809 if (mPreviousTrack != NULL) { 3810 if (t != mPreviousTrack) { 3811 // Flush any data still being written from last track 3812 mBytesRemaining = 0; 3813 if (mPausedBytesRemaining) { 3814 // Last track was paused so we also need to flush saved 3815 // mixbuffer state and invalidate track so that it will 3816 // re-submit that unwritten data when it is next resumed 3817 mPausedBytesRemaining = 0; 3818 // Invalidate is a bit drastic - would be more efficient 3819 // to have a flag to tell client that some of the 3820 // previously written data was lost 3821 mPreviousTrack->invalidate(); 3822 } 3823 } 3824 } 3825 mPreviousTrack = t; 3826 bool last = (i == (count - 1)); 3827 if (track->isPausing()) { 3828 track->setPaused(); 3829 if (last) { 3830 if (!mHwPaused) { 3831 mOutput->stream->pause(mOutput->stream); 3832 mHwPaused = true; 3833 } 3834 // If we were part way through writing the mixbuffer to 3835 // the HAL we must save this until we resume 3836 // BUG - this will be wrong if a different track is made active, 3837 // in that case we want to discard the pending data in the 3838 // mixbuffer and tell the client to present it again when the 3839 // track is resumed 3840 mPausedWriteLength = mCurrentWriteLength; 3841 mPausedBytesRemaining = mBytesRemaining; 3842 mBytesRemaining = 0; // stop writing 3843 } 3844 tracksToRemove->add(track); 3845 } else if (track->framesReady() && track->isReady() && 3846 !track->isPaused() && !track->isTerminated()) { 3847 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 3848 if (track->mFillingUpStatus == Track::FS_FILLED) { 3849 track->mFillingUpStatus = Track::FS_ACTIVE; 3850 mLeftVolFloat = mRightVolFloat = 0; 3851 if (track->mState == TrackBase::RESUMING) { 3852 if (mPausedBytesRemaining) { 3853 // Need to continue write that was interrupted 3854 mCurrentWriteLength = mPausedWriteLength; 3855 mBytesRemaining = mPausedBytesRemaining; 3856 mPausedBytesRemaining = 0; 3857 } 3858 track->mState = TrackBase::ACTIVE; 3859 } 3860 } 3861 3862 if (last) { 3863 if (mHwPaused) { 3864 mOutput->stream->resume(mOutput->stream); 3865 mHwPaused = false; 3866 // threadLoop_mix() will handle the case that we need to 3867 // resume an interrupted write 3868 } 3869 // reset retry count 3870 track->mRetryCount = kMaxTrackRetriesOffload; 3871 mActiveTrack = t; 3872 mixerStatus = MIXER_TRACKS_READY; 3873 } 3874 } else { 3875 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3876 if (track->isStopping_1()) { 3877 // Hardware buffer can hold a large amount of audio so we must 3878 // wait for all current track's data to drain before we say 3879 // that the track is stopped. 3880 if (mBytesRemaining == 0) { 3881 // Only start draining when all data in mixbuffer 3882 // has been written 3883 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 3884 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 3885 sleepTime = 0; 3886 standbyTime = systemTime() + standbyDelay; 3887 if (last) { 3888 mixerStatus = MIXER_DRAIN_TRACK; 3889 if (mHwPaused) { 3890 // It is possible to move from PAUSED to STOPPING_1 without 3891 // a resume so we must ensure hardware is running 3892 mOutput->stream->resume(mOutput->stream); 3893 mHwPaused = false; 3894 } 3895 } 3896 } 3897 } else if (track->isStopping_2()) { 3898 // Drain has completed, signal presentation complete 3899 if (!mDraining || !last) { 3900 track->mState = TrackBase::STOPPED; 3901 size_t audioHALFrames = 3902 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3903 size_t framesWritten = 3904 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3905 track->presentationComplete(framesWritten, audioHALFrames); 3906 track->reset(); 3907 tracksToRemove->add(track); 3908 } 3909 } else { 3910 // No buffers for this track. Give it a few chances to 3911 // fill a buffer, then remove it from active list. 3912 if (--(track->mRetryCount) <= 0) { 3913 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 3914 track->name()); 3915 tracksToRemove->add(track); 3916 } else if (last){ 3917 mixerStatus = MIXER_TRACKS_ENABLED; 3918 } 3919 } 3920 } 3921 // compute volume for this track 3922 processVolume_l(track, last); 3923 } 3924 // remove all the tracks that need to be... 3925 removeTracks_l(*tracksToRemove); 3926 3927 return mixerStatus; 3928} 3929 3930void AudioFlinger::OffloadThread::flushOutput_l() 3931{ 3932 mFlushPending = true; 3933} 3934 3935// must be called with thread mutex locked 3936bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 3937{ 3938 ALOGV("waitingAsyncCallback_l mWriteBlocked %d mDraining %d", mWriteBlocked, mDraining); 3939 if (mUseAsyncWrite && (mWriteBlocked || mDraining)) { 3940 return true; 3941 } 3942 return false; 3943} 3944 3945// must be called with thread mutex locked 3946bool AudioFlinger::OffloadThread::shouldStandby_l() 3947{ 3948 bool TrackPaused = false; 3949 3950 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 3951 // after a timeout and we will enter standby then. 3952 if (mTracks.size() > 0) { 3953 TrackPaused = mTracks[mTracks.size() - 1]->isPaused(); 3954 } 3955 3956 return !mStandby && !TrackPaused; 3957} 3958 3959 3960bool AudioFlinger::OffloadThread::waitingAsyncCallback() 3961{ 3962 Mutex::Autolock _l(mLock); 3963 return waitingAsyncCallback_l(); 3964} 3965 3966void AudioFlinger::OffloadThread::flushHw_l() 3967{ 3968 mOutput->stream->flush(mOutput->stream); 3969 // Flush anything still waiting in the mixbuffer 3970 mCurrentWriteLength = 0; 3971 mBytesRemaining = 0; 3972 mPausedWriteLength = 0; 3973 mPausedBytesRemaining = 0; 3974 if (mUseAsyncWrite) { 3975 mWriteBlocked = false; 3976 mDraining = false; 3977 ALOG_ASSERT(mCallbackThread != 0); 3978 mCallbackThread->setWriteBlocked(false); 3979 mCallbackThread->setDraining(false); 3980 } 3981} 3982 3983// ---------------------------------------------------------------------------- 3984 3985AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3986 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3987 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 3988 DUPLICATING), 3989 mWaitTimeMs(UINT_MAX) 3990{ 3991 addOutputTrack(mainThread); 3992} 3993 3994AudioFlinger::DuplicatingThread::~DuplicatingThread() 3995{ 3996 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3997 mOutputTracks[i]->destroy(); 3998 } 3999} 4000 4001void AudioFlinger::DuplicatingThread::threadLoop_mix() 4002{ 4003 // mix buffers... 4004 if (outputsReady(outputTracks)) { 4005 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4006 } else { 4007 memset(mMixBuffer, 0, mixBufferSize); 4008 } 4009 sleepTime = 0; 4010 writeFrames = mNormalFrameCount; 4011 mCurrentWriteLength = mixBufferSize; 4012 standbyTime = systemTime() + standbyDelay; 4013} 4014 4015void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4016{ 4017 if (sleepTime == 0) { 4018 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4019 sleepTime = activeSleepTime; 4020 } else { 4021 sleepTime = idleSleepTime; 4022 } 4023 } else if (mBytesWritten != 0) { 4024 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4025 writeFrames = mNormalFrameCount; 4026 memset(mMixBuffer, 0, mixBufferSize); 4027 } else { 4028 // flush remaining overflow buffers in output tracks 4029 writeFrames = 0; 4030 } 4031 sleepTime = 0; 4032 } 4033} 4034 4035ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4036{ 4037 for (size_t i = 0; i < outputTracks.size(); i++) { 4038 outputTracks[i]->write(mMixBuffer, writeFrames); 4039 } 4040 return (ssize_t)mixBufferSize; 4041} 4042 4043void AudioFlinger::DuplicatingThread::threadLoop_standby() 4044{ 4045 // DuplicatingThread implements standby by stopping all tracks 4046 for (size_t i = 0; i < outputTracks.size(); i++) { 4047 outputTracks[i]->stop(); 4048 } 4049} 4050 4051void AudioFlinger::DuplicatingThread::saveOutputTracks() 4052{ 4053 outputTracks = mOutputTracks; 4054} 4055 4056void AudioFlinger::DuplicatingThread::clearOutputTracks() 4057{ 4058 outputTracks.clear(); 4059} 4060 4061void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4062{ 4063 Mutex::Autolock _l(mLock); 4064 // FIXME explain this formula 4065 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4066 OutputTrack *outputTrack = new OutputTrack(thread, 4067 this, 4068 mSampleRate, 4069 mFormat, 4070 mChannelMask, 4071 frameCount); 4072 if (outputTrack->cblk() != NULL) { 4073 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4074 mOutputTracks.add(outputTrack); 4075 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4076 updateWaitTime_l(); 4077 } 4078} 4079 4080void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4081{ 4082 Mutex::Autolock _l(mLock); 4083 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4084 if (mOutputTracks[i]->thread() == thread) { 4085 mOutputTracks[i]->destroy(); 4086 mOutputTracks.removeAt(i); 4087 updateWaitTime_l(); 4088 return; 4089 } 4090 } 4091 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4092} 4093 4094// caller must hold mLock 4095void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4096{ 4097 mWaitTimeMs = UINT_MAX; 4098 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4099 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4100 if (strong != 0) { 4101 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4102 if (waitTimeMs < mWaitTimeMs) { 4103 mWaitTimeMs = waitTimeMs; 4104 } 4105 } 4106 } 4107} 4108 4109 4110bool AudioFlinger::DuplicatingThread::outputsReady( 4111 const SortedVector< sp<OutputTrack> > &outputTracks) 4112{ 4113 for (size_t i = 0; i < outputTracks.size(); i++) { 4114 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4115 if (thread == 0) { 4116 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4117 outputTracks[i].get()); 4118 return false; 4119 } 4120 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4121 // see note at standby() declaration 4122 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4123 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4124 thread.get()); 4125 return false; 4126 } 4127 } 4128 return true; 4129} 4130 4131uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4132{ 4133 return (mWaitTimeMs * 1000) / 2; 4134} 4135 4136void AudioFlinger::DuplicatingThread::cacheParameters_l() 4137{ 4138 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4139 updateWaitTime_l(); 4140 4141 MixerThread::cacheParameters_l(); 4142} 4143 4144// ---------------------------------------------------------------------------- 4145// Record 4146// ---------------------------------------------------------------------------- 4147 4148AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4149 AudioStreamIn *input, 4150 uint32_t sampleRate, 4151 audio_channel_mask_t channelMask, 4152 audio_io_handle_t id, 4153 audio_devices_t outDevice, 4154 audio_devices_t inDevice 4155#ifdef TEE_SINK 4156 , const sp<NBAIO_Sink>& teeSink 4157#endif 4158 ) : 4159 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4160 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4161 // mRsmpInIndex set by readInputParameters() 4162 mReqChannelCount(popcount(channelMask)), 4163 mReqSampleRate(sampleRate) 4164 // mBytesRead is only meaningful while active, and so is cleared in start() 4165 // (but might be better to also clear here for dump?) 4166#ifdef TEE_SINK 4167 , mTeeSink(teeSink) 4168#endif 4169{ 4170 snprintf(mName, kNameLength, "AudioIn_%X", id); 4171 4172 readInputParameters(); 4173 4174} 4175 4176 4177AudioFlinger::RecordThread::~RecordThread() 4178{ 4179 delete[] mRsmpInBuffer; 4180 delete mResampler; 4181 delete[] mRsmpOutBuffer; 4182} 4183 4184void AudioFlinger::RecordThread::onFirstRef() 4185{ 4186 run(mName, PRIORITY_URGENT_AUDIO); 4187} 4188 4189bool AudioFlinger::RecordThread::threadLoop() 4190{ 4191 AudioBufferProvider::Buffer buffer; 4192 sp<RecordTrack> activeTrack; 4193 Vector< sp<EffectChain> > effectChains; 4194 4195 nsecs_t lastWarning = 0; 4196 4197 inputStandBy(); 4198 acquireWakeLock(); 4199 4200 // used to verify we've read at least once before evaluating how many bytes were read 4201 bool readOnce = false; 4202 4203 // start recording 4204 while (!exitPending()) { 4205 4206 processConfigEvents(); 4207 4208 { // scope for mLock 4209 Mutex::Autolock _l(mLock); 4210 checkForNewParameters_l(); 4211 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4212 standby(); 4213 4214 if (exitPending()) { 4215 break; 4216 } 4217 4218 releaseWakeLock_l(); 4219 ALOGV("RecordThread: loop stopping"); 4220 // go to sleep 4221 mWaitWorkCV.wait(mLock); 4222 ALOGV("RecordThread: loop starting"); 4223 acquireWakeLock_l(); 4224 continue; 4225 } 4226 if (mActiveTrack != 0) { 4227 if (mActiveTrack->isTerminated()) { 4228 removeTrack_l(mActiveTrack); 4229 mActiveTrack.clear(); 4230 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4231 standby(); 4232 mActiveTrack.clear(); 4233 mStartStopCond.broadcast(); 4234 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4235 if (mReqChannelCount != mActiveTrack->channelCount()) { 4236 mActiveTrack.clear(); 4237 mStartStopCond.broadcast(); 4238 } else if (readOnce) { 4239 // record start succeeds only if first read from audio input 4240 // succeeds 4241 if (mBytesRead >= 0) { 4242 mActiveTrack->mState = TrackBase::ACTIVE; 4243 } else { 4244 mActiveTrack.clear(); 4245 } 4246 mStartStopCond.broadcast(); 4247 } 4248 mStandby = false; 4249 } 4250 } 4251 lockEffectChains_l(effectChains); 4252 } 4253 4254 if (mActiveTrack != 0) { 4255 if (mActiveTrack->mState != TrackBase::ACTIVE && 4256 mActiveTrack->mState != TrackBase::RESUMING) { 4257 unlockEffectChains(effectChains); 4258 usleep(kRecordThreadSleepUs); 4259 continue; 4260 } 4261 for (size_t i = 0; i < effectChains.size(); i ++) { 4262 effectChains[i]->process_l(); 4263 } 4264 4265 buffer.frameCount = mFrameCount; 4266 status_t status = mActiveTrack->getNextBuffer(&buffer); 4267 if (status == NO_ERROR) { 4268 readOnce = true; 4269 size_t framesOut = buffer.frameCount; 4270 if (mResampler == NULL) { 4271 // no resampling 4272 while (framesOut) { 4273 size_t framesIn = mFrameCount - mRsmpInIndex; 4274 if (framesIn) { 4275 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4276 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * 4277 mActiveTrack->mFrameSize; 4278 if (framesIn > framesOut) 4279 framesIn = framesOut; 4280 mRsmpInIndex += framesIn; 4281 framesOut -= framesIn; 4282 if (mChannelCount == mReqChannelCount) { 4283 memcpy(dst, src, framesIn * mFrameSize); 4284 } else { 4285 if (mChannelCount == 1) { 4286 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 4287 (int16_t *)src, framesIn); 4288 } else { 4289 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 4290 (int16_t *)src, framesIn); 4291 } 4292 } 4293 } 4294 if (framesOut && mFrameCount == mRsmpInIndex) { 4295 void *readInto; 4296 if (framesOut == mFrameCount && mChannelCount == mReqChannelCount) { 4297 readInto = buffer.raw; 4298 framesOut = 0; 4299 } else { 4300 readInto = mRsmpInBuffer; 4301 mRsmpInIndex = 0; 4302 } 4303 mBytesRead = mInput->stream->read(mInput->stream, readInto, 4304 mBufferSize); 4305 if (mBytesRead <= 0) { 4306 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) 4307 { 4308 ALOGE("Error reading audio input"); 4309 // Force input into standby so that it tries to 4310 // recover at next read attempt 4311 inputStandBy(); 4312 usleep(kRecordThreadSleepUs); 4313 } 4314 mRsmpInIndex = mFrameCount; 4315 framesOut = 0; 4316 buffer.frameCount = 0; 4317 } 4318#ifdef TEE_SINK 4319 else if (mTeeSink != 0) { 4320 (void) mTeeSink->write(readInto, 4321 mBytesRead >> Format_frameBitShift(mTeeSink->format())); 4322 } 4323#endif 4324 } 4325 } 4326 } else { 4327 // resampling 4328 4329 // resampler accumulates, but we only have one source track 4330 memset(mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4331 // alter output frame count as if we were expecting stereo samples 4332 if (mChannelCount == 1 && mReqChannelCount == 1) { 4333 framesOut >>= 1; 4334 } 4335 mResampler->resample(mRsmpOutBuffer, framesOut, 4336 this /* AudioBufferProvider* */); 4337 // ditherAndClamp() works as long as all buffers returned by 4338 // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true. 4339 if (mChannelCount == 2 && mReqChannelCount == 1) { 4340 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4341 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4342 // the resampler always outputs stereo samples: 4343 // do post stereo to mono conversion 4344 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 4345 framesOut); 4346 } else { 4347 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4348 } 4349 // now done with mRsmpOutBuffer 4350 4351 } 4352 if (mFramestoDrop == 0) { 4353 mActiveTrack->releaseBuffer(&buffer); 4354 } else { 4355 if (mFramestoDrop > 0) { 4356 mFramestoDrop -= buffer.frameCount; 4357 if (mFramestoDrop <= 0) { 4358 clearSyncStartEvent(); 4359 } 4360 } else { 4361 mFramestoDrop += buffer.frameCount; 4362 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 4363 mSyncStartEvent->isCancelled()) { 4364 ALOGW("Synced record %s, session %d, trigger session %d", 4365 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 4366 mActiveTrack->sessionId(), 4367 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 4368 clearSyncStartEvent(); 4369 } 4370 } 4371 } 4372 mActiveTrack->clearOverflow(); 4373 } 4374 // client isn't retrieving buffers fast enough 4375 else { 4376 if (!mActiveTrack->setOverflow()) { 4377 nsecs_t now = systemTime(); 4378 if ((now - lastWarning) > kWarningThrottleNs) { 4379 ALOGW("RecordThread: buffer overflow"); 4380 lastWarning = now; 4381 } 4382 } 4383 // Release the processor for a while before asking for a new buffer. 4384 // This will give the application more chance to read from the buffer and 4385 // clear the overflow. 4386 usleep(kRecordThreadSleepUs); 4387 } 4388 } 4389 // enable changes in effect chain 4390 unlockEffectChains(effectChains); 4391 effectChains.clear(); 4392 } 4393 4394 standby(); 4395 4396 { 4397 Mutex::Autolock _l(mLock); 4398 mActiveTrack.clear(); 4399 mStartStopCond.broadcast(); 4400 } 4401 4402 releaseWakeLock(); 4403 4404 ALOGV("RecordThread %p exiting", this); 4405 return false; 4406} 4407 4408void AudioFlinger::RecordThread::standby() 4409{ 4410 if (!mStandby) { 4411 inputStandBy(); 4412 mStandby = true; 4413 } 4414} 4415 4416void AudioFlinger::RecordThread::inputStandBy() 4417{ 4418 mInput->stream->common.standby(&mInput->stream->common); 4419} 4420 4421sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4422 const sp<AudioFlinger::Client>& client, 4423 uint32_t sampleRate, 4424 audio_format_t format, 4425 audio_channel_mask_t channelMask, 4426 size_t frameCount, 4427 int sessionId, 4428 IAudioFlinger::track_flags_t *flags, 4429 pid_t tid, 4430 status_t *status) 4431{ 4432 sp<RecordTrack> track; 4433 status_t lStatus; 4434 4435 lStatus = initCheck(); 4436 if (lStatus != NO_ERROR) { 4437 ALOGE("Audio driver not initialized."); 4438 goto Exit; 4439 } 4440 4441 // client expresses a preference for FAST, but we get the final say 4442 if (*flags & IAudioFlinger::TRACK_FAST) { 4443 if ( 4444 // use case: callback handler and frame count is default or at least as large as HAL 4445 ( 4446 (tid != -1) && 4447 ((frameCount == 0) || 4448 (frameCount >= (mFrameCount * kFastTrackMultiplier))) 4449 ) && 4450 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 4451 // mono or stereo 4452 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 4453 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 4454 // hardware sample rate 4455 (sampleRate == mSampleRate) && 4456 // record thread has an associated fast recorder 4457 hasFastRecorder() 4458 // FIXME test that RecordThread for this fast track has a capable output HAL 4459 // FIXME add a permission test also? 4460 ) { 4461 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 4462 if (frameCount == 0) { 4463 frameCount = mFrameCount * kFastTrackMultiplier; 4464 } 4465 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 4466 frameCount, mFrameCount); 4467 } else { 4468 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 4469 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 4470 "hasFastRecorder=%d tid=%d", 4471 frameCount, mFrameCount, format, 4472 audio_is_linear_pcm(format), 4473 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 4474 *flags &= ~IAudioFlinger::TRACK_FAST; 4475 // For compatibility with AudioRecord calculation, buffer depth is forced 4476 // to be at least 2 x the record thread frame count and cover audio hardware latency. 4477 // This is probably too conservative, but legacy application code may depend on it. 4478 // If you change this calculation, also review the start threshold which is related. 4479 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 4480 size_t mNormalFrameCount = 2048; // FIXME 4481 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 4482 if (minBufCount < 2) { 4483 minBufCount = 2; 4484 } 4485 size_t minFrameCount = mNormalFrameCount * minBufCount; 4486 if (frameCount < minFrameCount) { 4487 frameCount = minFrameCount; 4488 } 4489 } 4490 } 4491 4492 // FIXME use flags and tid similar to createTrack_l() 4493 4494 { // scope for mLock 4495 Mutex::Autolock _l(mLock); 4496 4497 track = new RecordTrack(this, client, sampleRate, 4498 format, channelMask, frameCount, sessionId); 4499 4500 if (track->getCblk() == 0) { 4501 lStatus = NO_MEMORY; 4502 goto Exit; 4503 } 4504 mTracks.add(track); 4505 4506 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4507 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4508 mAudioFlinger->btNrecIsOff(); 4509 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4510 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4511 4512 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 4513 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 4514 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 4515 // so ask activity manager to do this on our behalf 4516 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 4517 } 4518 } 4519 lStatus = NO_ERROR; 4520 4521Exit: 4522 *status = lStatus; 4523 return track; 4524} 4525 4526status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 4527 AudioSystem::sync_event_t event, 4528 int triggerSession) 4529{ 4530 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 4531 sp<ThreadBase> strongMe = this; 4532 status_t status = NO_ERROR; 4533 4534 if (event == AudioSystem::SYNC_EVENT_NONE) { 4535 clearSyncStartEvent(); 4536 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 4537 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 4538 triggerSession, 4539 recordTrack->sessionId(), 4540 syncStartEventCallback, 4541 this); 4542 // Sync event can be cancelled by the trigger session if the track is not in a 4543 // compatible state in which case we start record immediately 4544 if (mSyncStartEvent->isCancelled()) { 4545 clearSyncStartEvent(); 4546 } else { 4547 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 4548 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 4549 } 4550 } 4551 4552 { 4553 AutoMutex lock(mLock); 4554 if (mActiveTrack != 0) { 4555 if (recordTrack != mActiveTrack.get()) { 4556 status = -EBUSY; 4557 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4558 mActiveTrack->mState = TrackBase::ACTIVE; 4559 } 4560 return status; 4561 } 4562 4563 recordTrack->mState = TrackBase::IDLE; 4564 mActiveTrack = recordTrack; 4565 mLock.unlock(); 4566 status_t status = AudioSystem::startInput(mId); 4567 mLock.lock(); 4568 if (status != NO_ERROR) { 4569 mActiveTrack.clear(); 4570 clearSyncStartEvent(); 4571 return status; 4572 } 4573 mRsmpInIndex = mFrameCount; 4574 mBytesRead = 0; 4575 if (mResampler != NULL) { 4576 mResampler->reset(); 4577 } 4578 mActiveTrack->mState = TrackBase::RESUMING; 4579 // signal thread to start 4580 ALOGV("Signal record thread"); 4581 mWaitWorkCV.broadcast(); 4582 // do not wait for mStartStopCond if exiting 4583 if (exitPending()) { 4584 mActiveTrack.clear(); 4585 status = INVALID_OPERATION; 4586 goto startError; 4587 } 4588 mStartStopCond.wait(mLock); 4589 if (mActiveTrack == 0) { 4590 ALOGV("Record failed to start"); 4591 status = BAD_VALUE; 4592 goto startError; 4593 } 4594 ALOGV("Record started OK"); 4595 return status; 4596 } 4597 4598startError: 4599 AudioSystem::stopInput(mId); 4600 clearSyncStartEvent(); 4601 return status; 4602} 4603 4604void AudioFlinger::RecordThread::clearSyncStartEvent() 4605{ 4606 if (mSyncStartEvent != 0) { 4607 mSyncStartEvent->cancel(); 4608 } 4609 mSyncStartEvent.clear(); 4610 mFramestoDrop = 0; 4611} 4612 4613void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 4614{ 4615 sp<SyncEvent> strongEvent = event.promote(); 4616 4617 if (strongEvent != 0) { 4618 RecordThread *me = (RecordThread *)strongEvent->cookie(); 4619 me->handleSyncStartEvent(strongEvent); 4620 } 4621} 4622 4623void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 4624{ 4625 if (event == mSyncStartEvent) { 4626 // TODO: use actual buffer filling status instead of 2 buffers when info is available 4627 // from audio HAL 4628 mFramestoDrop = mFrameCount * 2; 4629 } 4630} 4631 4632bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4633 ALOGV("RecordThread::stop"); 4634 AutoMutex _l(mLock); 4635 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 4636 return false; 4637 } 4638 recordTrack->mState = TrackBase::PAUSING; 4639 // do not wait for mStartStopCond if exiting 4640 if (exitPending()) { 4641 return true; 4642 } 4643 mStartStopCond.wait(mLock); 4644 // if we have been restarted, recordTrack == mActiveTrack.get() here 4645 if (exitPending() || recordTrack != mActiveTrack.get()) { 4646 ALOGV("Record stopped OK"); 4647 return true; 4648 } 4649 return false; 4650} 4651 4652bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) const 4653{ 4654 return false; 4655} 4656 4657status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 4658{ 4659#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 4660 if (!isValidSyncEvent(event)) { 4661 return BAD_VALUE; 4662 } 4663 4664 int eventSession = event->triggerSession(); 4665 status_t ret = NAME_NOT_FOUND; 4666 4667 Mutex::Autolock _l(mLock); 4668 4669 for (size_t i = 0; i < mTracks.size(); i++) { 4670 sp<RecordTrack> track = mTracks[i]; 4671 if (eventSession == track->sessionId()) { 4672 (void) track->setSyncEvent(event); 4673 ret = NO_ERROR; 4674 } 4675 } 4676 return ret; 4677#else 4678 return BAD_VALUE; 4679#endif 4680} 4681 4682// destroyTrack_l() must be called with ThreadBase::mLock held 4683void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 4684{ 4685 track->terminate(); 4686 track->mState = TrackBase::STOPPED; 4687 // active tracks are removed by threadLoop() 4688 if (mActiveTrack != track) { 4689 removeTrack_l(track); 4690 } 4691} 4692 4693void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 4694{ 4695 mTracks.remove(track); 4696 // need anything related to effects here? 4697} 4698 4699void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4700{ 4701 dumpInternals(fd, args); 4702 dumpTracks(fd, args); 4703 dumpEffectChains(fd, args); 4704} 4705 4706void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 4707{ 4708 const size_t SIZE = 256; 4709 char buffer[SIZE]; 4710 String8 result; 4711 4712 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4713 result.append(buffer); 4714 4715 if (mActiveTrack != 0) { 4716 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4717 result.append(buffer); 4718 snprintf(buffer, SIZE, "Buffer size: %u bytes\n", mBufferSize); 4719 result.append(buffer); 4720 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4721 result.append(buffer); 4722 snprintf(buffer, SIZE, "Out channel count: %u\n", mReqChannelCount); 4723 result.append(buffer); 4724 snprintf(buffer, SIZE, "Out sample rate: %u\n", mReqSampleRate); 4725 result.append(buffer); 4726 } else { 4727 result.append("No active record client\n"); 4728 } 4729 4730 write(fd, result.string(), result.size()); 4731 4732 dumpBase(fd, args); 4733} 4734 4735void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 4736{ 4737 const size_t SIZE = 256; 4738 char buffer[SIZE]; 4739 String8 result; 4740 4741 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 4742 result.append(buffer); 4743 RecordTrack::appendDumpHeader(result); 4744 for (size_t i = 0; i < mTracks.size(); ++i) { 4745 sp<RecordTrack> track = mTracks[i]; 4746 if (track != 0) { 4747 track->dump(buffer, SIZE); 4748 result.append(buffer); 4749 } 4750 } 4751 4752 if (mActiveTrack != 0) { 4753 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 4754 result.append(buffer); 4755 RecordTrack::appendDumpHeader(result); 4756 mActiveTrack->dump(buffer, SIZE); 4757 result.append(buffer); 4758 4759 } 4760 write(fd, result.string(), result.size()); 4761} 4762 4763// AudioBufferProvider interface 4764status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4765{ 4766 size_t framesReq = buffer->frameCount; 4767 size_t framesReady = mFrameCount - mRsmpInIndex; 4768 int channelCount; 4769 4770 if (framesReady == 0) { 4771 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mBufferSize); 4772 if (mBytesRead <= 0) { 4773 if ((mBytesRead < 0) && (mActiveTrack->mState == TrackBase::ACTIVE)) { 4774 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4775 // Force input into standby so that it tries to 4776 // recover at next read attempt 4777 inputStandBy(); 4778 usleep(kRecordThreadSleepUs); 4779 } 4780 buffer->raw = NULL; 4781 buffer->frameCount = 0; 4782 return NOT_ENOUGH_DATA; 4783 } 4784 mRsmpInIndex = 0; 4785 framesReady = mFrameCount; 4786 } 4787 4788 if (framesReq > framesReady) { 4789 framesReq = framesReady; 4790 } 4791 4792 if (mChannelCount == 1 && mReqChannelCount == 2) { 4793 channelCount = 1; 4794 } else { 4795 channelCount = 2; 4796 } 4797 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4798 buffer->frameCount = framesReq; 4799 return NO_ERROR; 4800} 4801 4802// AudioBufferProvider interface 4803void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4804{ 4805 mRsmpInIndex += buffer->frameCount; 4806 buffer->frameCount = 0; 4807} 4808 4809bool AudioFlinger::RecordThread::checkForNewParameters_l() 4810{ 4811 bool reconfig = false; 4812 4813 while (!mNewParameters.isEmpty()) { 4814 status_t status = NO_ERROR; 4815 String8 keyValuePair = mNewParameters[0]; 4816 AudioParameter param = AudioParameter(keyValuePair); 4817 int value; 4818 audio_format_t reqFormat = mFormat; 4819 uint32_t reqSamplingRate = mReqSampleRate; 4820 uint32_t reqChannelCount = mReqChannelCount; 4821 4822 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4823 reqSamplingRate = value; 4824 reconfig = true; 4825 } 4826 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4827 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 4828 status = BAD_VALUE; 4829 } else { 4830 reqFormat = (audio_format_t) value; 4831 reconfig = true; 4832 } 4833 } 4834 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4835 reqChannelCount = popcount(value); 4836 reconfig = true; 4837 } 4838 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4839 // do not accept frame count changes if tracks are open as the track buffer 4840 // size depends on frame count and correct behavior would not be guaranteed 4841 // if frame count is changed after track creation 4842 if (mActiveTrack != 0) { 4843 status = INVALID_OPERATION; 4844 } else { 4845 reconfig = true; 4846 } 4847 } 4848 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4849 // forward device change to effects that have requested to be 4850 // aware of attached audio device. 4851 for (size_t i = 0; i < mEffectChains.size(); i++) { 4852 mEffectChains[i]->setDevice_l(value); 4853 } 4854 4855 // store input device and output device but do not forward output device to audio HAL. 4856 // Note that status is ignored by the caller for output device 4857 // (see AudioFlinger::setParameters() 4858 if (audio_is_output_devices(value)) { 4859 mOutDevice = value; 4860 status = BAD_VALUE; 4861 } else { 4862 mInDevice = value; 4863 // disable AEC and NS if the device is a BT SCO headset supporting those 4864 // pre processings 4865 if (mTracks.size() > 0) { 4866 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 4867 mAudioFlinger->btNrecIsOff(); 4868 for (size_t i = 0; i < mTracks.size(); i++) { 4869 sp<RecordTrack> track = mTracks[i]; 4870 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 4871 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 4872 } 4873 } 4874 } 4875 } 4876 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 4877 mAudioSource != (audio_source_t)value) { 4878 // forward device change to effects that have requested to be 4879 // aware of attached audio device. 4880 for (size_t i = 0; i < mEffectChains.size(); i++) { 4881 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 4882 } 4883 mAudioSource = (audio_source_t)value; 4884 } 4885 if (status == NO_ERROR) { 4886 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4887 keyValuePair.string()); 4888 if (status == INVALID_OPERATION) { 4889 inputStandBy(); 4890 status = mInput->stream->common.set_parameters(&mInput->stream->common, 4891 keyValuePair.string()); 4892 } 4893 if (reconfig) { 4894 if (status == BAD_VALUE && 4895 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4896 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4897 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 4898 <= (2 * reqSamplingRate)) && 4899 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 4900 <= FCC_2 && 4901 (reqChannelCount <= FCC_2)) { 4902 status = NO_ERROR; 4903 } 4904 if (status == NO_ERROR) { 4905 readInputParameters(); 4906 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4907 } 4908 } 4909 } 4910 4911 mNewParameters.removeAt(0); 4912 4913 mParamStatus = status; 4914 mParamCond.signal(); 4915 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4916 // already timed out waiting for the status and will never signal the condition. 4917 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4918 } 4919 return reconfig; 4920} 4921 4922String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4923{ 4924 Mutex::Autolock _l(mLock); 4925 if (initCheck() != NO_ERROR) { 4926 return String8(); 4927 } 4928 4929 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4930 const String8 out_s8(s); 4931 free(s); 4932 return out_s8; 4933} 4934 4935void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4936 AudioSystem::OutputDescriptor desc; 4937 void *param2 = NULL; 4938 4939 switch (event) { 4940 case AudioSystem::INPUT_OPENED: 4941 case AudioSystem::INPUT_CONFIG_CHANGED: 4942 desc.channelMask = mChannelMask; 4943 desc.samplingRate = mSampleRate; 4944 desc.format = mFormat; 4945 desc.frameCount = mFrameCount; 4946 desc.latency = 0; 4947 param2 = &desc; 4948 break; 4949 4950 case AudioSystem::INPUT_CLOSED: 4951 default: 4952 break; 4953 } 4954 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4955} 4956 4957void AudioFlinger::RecordThread::readInputParameters() 4958{ 4959 delete[] mRsmpInBuffer; 4960 // mRsmpInBuffer is always assigned a new[] below 4961 delete[] mRsmpOutBuffer; 4962 mRsmpOutBuffer = NULL; 4963 delete mResampler; 4964 mResampler = NULL; 4965 4966 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4967 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4968 mChannelCount = popcount(mChannelMask); 4969 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4970 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4971 ALOGE("HAL format %d not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 4972 } 4973 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4974 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4975 mFrameCount = mBufferSize / mFrameSize; 4976 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4977 4978 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 4979 { 4980 int channelCount; 4981 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4982 // stereo to mono post process as the resampler always outputs stereo. 4983 if (mChannelCount == 1 && mReqChannelCount == 2) { 4984 channelCount = 1; 4985 } else { 4986 channelCount = 2; 4987 } 4988 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4989 mResampler->setSampleRate(mSampleRate); 4990 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4991 mRsmpOutBuffer = new int32_t[mFrameCount * FCC_2]; 4992 4993 // optmization: if mono to mono, alter input frame count as if we were inputing 4994 // stereo samples 4995 if (mChannelCount == 1 && mReqChannelCount == 1) { 4996 mFrameCount >>= 1; 4997 } 4998 4999 } 5000 mRsmpInIndex = mFrameCount; 5001} 5002 5003unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5004{ 5005 Mutex::Autolock _l(mLock); 5006 if (initCheck() != NO_ERROR) { 5007 return 0; 5008 } 5009 5010 return mInput->stream->get_input_frames_lost(mInput->stream); 5011} 5012 5013uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5014{ 5015 Mutex::Autolock _l(mLock); 5016 uint32_t result = 0; 5017 if (getEffectChain_l(sessionId) != 0) { 5018 result = EFFECT_SESSION; 5019 } 5020 5021 for (size_t i = 0; i < mTracks.size(); ++i) { 5022 if (sessionId == mTracks[i]->sessionId()) { 5023 result |= TRACK_SESSION; 5024 break; 5025 } 5026 } 5027 5028 return result; 5029} 5030 5031KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5032{ 5033 KeyedVector<int, bool> ids; 5034 Mutex::Autolock _l(mLock); 5035 for (size_t j = 0; j < mTracks.size(); ++j) { 5036 sp<RecordThread::RecordTrack> track = mTracks[j]; 5037 int sessionId = track->sessionId(); 5038 if (ids.indexOfKey(sessionId) < 0) { 5039 ids.add(sessionId, true); 5040 } 5041 } 5042 return ids; 5043} 5044 5045AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5046{ 5047 Mutex::Autolock _l(mLock); 5048 AudioStreamIn *input = mInput; 5049 mInput = NULL; 5050 return input; 5051} 5052 5053// this method must always be called either with ThreadBase mLock held or inside the thread loop 5054audio_stream_t* AudioFlinger::RecordThread::stream() const 5055{ 5056 if (mInput == NULL) { 5057 return NULL; 5058 } 5059 return &mInput->stream->common; 5060} 5061 5062status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5063{ 5064 // only one chain per input thread 5065 if (mEffectChains.size() != 0) { 5066 return INVALID_OPERATION; 5067 } 5068 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5069 5070 chain->setInBuffer(NULL); 5071 chain->setOutBuffer(NULL); 5072 5073 checkSuspendOnAddEffectChain_l(chain); 5074 5075 mEffectChains.add(chain); 5076 5077 return NO_ERROR; 5078} 5079 5080size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5081{ 5082 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5083 ALOGW_IF(mEffectChains.size() != 1, 5084 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5085 chain.get(), mEffectChains.size(), this); 5086 if (mEffectChains.size() == 1) { 5087 mEffectChains.removeAt(0); 5088 } 5089 return 0; 5090} 5091 5092}; // namespace android 5093