Threads.cpp revision e7e676fd2866fa4898712c4effa9e624e969c182
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#include <audio_utils/format.h> 38 39// NBAIO implementations 40#include <media/nbaio/AudioStreamOutSink.h> 41#include <media/nbaio/MonoPipe.h> 42#include <media/nbaio/MonoPipeReader.h> 43#include <media/nbaio/Pipe.h> 44#include <media/nbaio/PipeReader.h> 45#include <media/nbaio/SourceAudioBufferProvider.h> 46 47#include <powermanager/PowerManager.h> 48 49#include <common_time/cc_helper.h> 50#include <common_time/local_clock.h> 51 52#include "AudioFlinger.h" 53#include "AudioMixer.h" 54#include "FastMixer.h" 55#include "ServiceUtilities.h" 56#include "SchedulingPolicyService.h" 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85// retry counts for buffer fill timeout 86// 50 * ~20msecs = 1 second 87static const int8_t kMaxTrackRetries = 50; 88static const int8_t kMaxTrackStartupRetries = 50; 89// allow less retry attempts on direct output thread. 90// direct outputs can be a scarce resource in audio hardware and should 91// be released as quickly as possible. 92static const int8_t kMaxTrackRetriesDirect = 2; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait for setParameters to complete 101static const nsecs_t kSetParametersTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108// minimum normal sink buffer size, expressed in milliseconds rather than frames 109static const uint32_t kMinNormalSinkBufferSizeMs = 20; 110// maximum normal sink buffer size 111static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 112 113// Offloaded output thread standby delay: allows track transition without going to standby 114static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 115 116// Whether to use fast mixer 117static const enum { 118 FastMixer_Never, // never initialize or use: for debugging only 119 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 120 // normal mixer multiplier is 1 121 FastMixer_Static, // initialize if needed, then use all the time if initialized, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 124 // multiplier is calculated based on min & max normal mixer buffer size 125 // FIXME for FastMixer_Dynamic: 126 // Supporting this option will require fixing HALs that can't handle large writes. 127 // For example, one HAL implementation returns an error from a large write, 128 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 129 // We could either fix the HAL implementations, or provide a wrapper that breaks 130 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 131} kUseFastMixer = FastMixer_Static; 132 133// Priorities for requestPriority 134static const int kPriorityAudioApp = 2; 135static const int kPriorityFastMixer = 3; 136 137// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 138// for the track. The client then sub-divides this into smaller buffers for its use. 139// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 140// So for now we just assume that client is double-buffered for fast tracks. 141// FIXME It would be better for client to tell AudioFlinger the value of N, 142// so AudioFlinger could allocate the right amount of memory. 143// See the client's minBufCount and mNotificationFramesAct calculations for details. 144static const int kFastTrackMultiplier = 2; 145 146// ---------------------------------------------------------------------------- 147 148#ifdef ADD_BATTERY_DATA 149// To collect the amplifier usage 150static void addBatteryData(uint32_t params) { 151 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 152 if (service == NULL) { 153 // it already logged 154 return; 155 } 156 157 service->addBatteryData(params); 158} 159#endif 160 161 162// ---------------------------------------------------------------------------- 163// CPU Stats 164// ---------------------------------------------------------------------------- 165 166class CpuStats { 167public: 168 CpuStats(); 169 void sample(const String8 &title); 170#ifdef DEBUG_CPU_USAGE 171private: 172 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 173 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 174 175 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 176 177 int mCpuNum; // thread's current CPU number 178 int mCpukHz; // frequency of thread's current CPU in kHz 179#endif 180}; 181 182CpuStats::CpuStats() 183#ifdef DEBUG_CPU_USAGE 184 : mCpuNum(-1), mCpukHz(-1) 185#endif 186{ 187} 188 189void CpuStats::sample(const String8 &title 190#ifndef DEBUG_CPU_USAGE 191 __unused 192#endif 193 ) { 194#ifdef DEBUG_CPU_USAGE 195 // get current thread's delta CPU time in wall clock ns 196 double wcNs; 197 bool valid = mCpuUsage.sampleAndEnable(wcNs); 198 199 // record sample for wall clock statistics 200 if (valid) { 201 mWcStats.sample(wcNs); 202 } 203 204 // get the current CPU number 205 int cpuNum = sched_getcpu(); 206 207 // get the current CPU frequency in kHz 208 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 209 210 // check if either CPU number or frequency changed 211 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 212 mCpuNum = cpuNum; 213 mCpukHz = cpukHz; 214 // ignore sample for purposes of cycles 215 valid = false; 216 } 217 218 // if no change in CPU number or frequency, then record sample for cycle statistics 219 if (valid && mCpukHz > 0) { 220 double cycles = wcNs * cpukHz * 0.000001; 221 mHzStats.sample(cycles); 222 } 223 224 unsigned n = mWcStats.n(); 225 // mCpuUsage.elapsed() is expensive, so don't call it every loop 226 if ((n & 127) == 1) { 227 long long elapsed = mCpuUsage.elapsed(); 228 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 229 double perLoop = elapsed / (double) n; 230 double perLoop100 = perLoop * 0.01; 231 double perLoop1k = perLoop * 0.001; 232 double mean = mWcStats.mean(); 233 double stddev = mWcStats.stddev(); 234 double minimum = mWcStats.minimum(); 235 double maximum = mWcStats.maximum(); 236 double meanCycles = mHzStats.mean(); 237 double stddevCycles = mHzStats.stddev(); 238 double minCycles = mHzStats.minimum(); 239 double maxCycles = mHzStats.maximum(); 240 mCpuUsage.resetElapsed(); 241 mWcStats.reset(); 242 mHzStats.reset(); 243 ALOGD("CPU usage for %s over past %.1f secs\n" 244 " (%u mixer loops at %.1f mean ms per loop):\n" 245 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 246 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 247 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 248 title.string(), 249 elapsed * .000000001, n, perLoop * .000001, 250 mean * .001, 251 stddev * .001, 252 minimum * .001, 253 maximum * .001, 254 mean / perLoop100, 255 stddev / perLoop100, 256 minimum / perLoop100, 257 maximum / perLoop100, 258 meanCycles / perLoop1k, 259 stddevCycles / perLoop1k, 260 minCycles / perLoop1k, 261 maxCycles / perLoop1k); 262 263 } 264 } 265#endif 266}; 267 268// ---------------------------------------------------------------------------- 269// ThreadBase 270// ---------------------------------------------------------------------------- 271 272AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 273 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 274 : Thread(false /*canCallJava*/), 275 mType(type), 276 mAudioFlinger(audioFlinger), 277 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 278 // are set by PlaybackThread::readOutputParameters_l() or 279 // RecordThread::readInputParameters_l() 280 mParamStatus(NO_ERROR), 281 //FIXME: mStandby should be true here. Is this some kind of hack? 282 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 283 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 284 // mName will be set by concrete (non-virtual) subclass 285 mDeathRecipient(new PMDeathRecipient(this)) 286{ 287} 288 289AudioFlinger::ThreadBase::~ThreadBase() 290{ 291 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 292 for (size_t i = 0; i < mConfigEvents.size(); i++) { 293 delete mConfigEvents[i]; 294 } 295 mConfigEvents.clear(); 296 297 mParamCond.broadcast(); 298 // do not lock the mutex in destructor 299 releaseWakeLock_l(); 300 if (mPowerManager != 0) { 301 sp<IBinder> binder = mPowerManager->asBinder(); 302 binder->unlinkToDeath(mDeathRecipient); 303 } 304} 305 306status_t AudioFlinger::ThreadBase::readyToRun() 307{ 308 status_t status = initCheck(); 309 if (status == NO_ERROR) { 310 ALOGI("AudioFlinger's thread %p ready to run", this); 311 } else { 312 ALOGE("No working audio driver found."); 313 } 314 return status; 315} 316 317void AudioFlinger::ThreadBase::exit() 318{ 319 ALOGV("ThreadBase::exit"); 320 // do any cleanup required for exit to succeed 321 preExit(); 322 { 323 // This lock prevents the following race in thread (uniprocessor for illustration): 324 // if (!exitPending()) { 325 // // context switch from here to exit() 326 // // exit() calls requestExit(), what exitPending() observes 327 // // exit() calls signal(), which is dropped since no waiters 328 // // context switch back from exit() to here 329 // mWaitWorkCV.wait(...); 330 // // now thread is hung 331 // } 332 AutoMutex lock(mLock); 333 requestExit(); 334 mWaitWorkCV.broadcast(); 335 } 336 // When Thread::requestExitAndWait is made virtual and this method is renamed to 337 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 338 requestExitAndWait(); 339} 340 341status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 342{ 343 status_t status; 344 345 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 346 Mutex::Autolock _l(mLock); 347 348 mNewParameters.add(keyValuePairs); 349 mWaitWorkCV.signal(); 350 // wait condition with timeout in case the thread loop has exited 351 // before the request could be processed 352 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 353 status = mParamStatus; 354 mWaitWorkCV.signal(); 355 } else { 356 status = TIMED_OUT; 357 } 358 return status; 359} 360 361void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 362{ 363 Mutex::Autolock _l(mLock); 364 sendIoConfigEvent_l(event, param); 365} 366 367// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 368void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 369{ 370 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 371 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 372 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 373 param); 374 mWaitWorkCV.signal(); 375} 376 377// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 378void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 379{ 380 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 381 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 382 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 383 mConfigEvents.size(), pid, tid, prio); 384 mWaitWorkCV.signal(); 385} 386 387void AudioFlinger::ThreadBase::processConfigEvents() 388{ 389 Mutex::Autolock _l(mLock); 390 processConfigEvents_l(); 391} 392 393// post condition: mConfigEvents.isEmpty() 394void AudioFlinger::ThreadBase::processConfigEvents_l() 395{ 396 while (!mConfigEvents.isEmpty()) { 397 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 398 ConfigEvent *event = mConfigEvents[0]; 399 mConfigEvents.removeAt(0); 400 // release mLock before locking AudioFlinger mLock: lock order is always 401 // AudioFlinger then ThreadBase to avoid cross deadlock 402 mLock.unlock(); 403 switch (event->type()) { 404 case CFG_EVENT_PRIO: { 405 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 406 // FIXME Need to understand why this has be done asynchronously 407 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 408 true /*asynchronous*/); 409 if (err != 0) { 410 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 411 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 412 } 413 } break; 414 case CFG_EVENT_IO: { 415 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 416 { 417 Mutex::Autolock _l(mAudioFlinger->mLock); 418 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 419 } 420 } break; 421 default: 422 ALOGE("processConfigEvents() unknown event type %d", event->type()); 423 break; 424 } 425 delete event; 426 mLock.lock(); 427 } 428} 429 430String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 431 String8 s; 432 if (output) { 433 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 434 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 435 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 436 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 437 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 438 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 439 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 440 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 441 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 442 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 443 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 444 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 445 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 446 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 447 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 448 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 449 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 450 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 451 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 452 } else { 453 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 454 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 455 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 456 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 457 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 458 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 459 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 460 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 461 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 462 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 463 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 464 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 465 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 466 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 467 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 468 } 469 int len = s.length(); 470 if (s.length() > 2) { 471 char *str = s.lockBuffer(len); 472 s.unlockBuffer(len - 2); 473 } 474 return s; 475} 476 477void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 478{ 479 const size_t SIZE = 256; 480 char buffer[SIZE]; 481 String8 result; 482 483 bool locked = AudioFlinger::dumpTryLock(mLock); 484 if (!locked) { 485 fdprintf(fd, "thread %p maybe dead locked\n", this); 486 } 487 488 fdprintf(fd, " I/O handle: %d\n", mId); 489 fdprintf(fd, " TID: %d\n", getTid()); 490 fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 491 fdprintf(fd, " Sample rate: %u\n", mSampleRate); 492 fdprintf(fd, " HAL frame count: %zu\n", mFrameCount); 493 fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 494 fdprintf(fd, " Channel Count: %u\n", mChannelCount); 495 fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 496 channelMaskToString(mChannelMask, mType != RECORD).string()); 497 fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 498 fdprintf(fd, " Frame size: %zu\n", mFrameSize); 499 fdprintf(fd, " Pending setParameters commands:"); 500 size_t numParams = mNewParameters.size(); 501 if (numParams) { 502 fdprintf(fd, "\n Index Command"); 503 for (size_t i = 0; i < numParams; ++i) { 504 fdprintf(fd, "\n %02zu ", i); 505 fdprintf(fd, mNewParameters[i]); 506 } 507 fdprintf(fd, "\n"); 508 } else { 509 fdprintf(fd, " none\n"); 510 } 511 fdprintf(fd, " Pending config events:"); 512 size_t numConfig = mConfigEvents.size(); 513 if (numConfig) { 514 for (size_t i = 0; i < numConfig; i++) { 515 mConfigEvents[i]->dump(buffer, SIZE); 516 fdprintf(fd, "\n %s", buffer); 517 } 518 fdprintf(fd, "\n"); 519 } else { 520 fdprintf(fd, " none\n"); 521 } 522 523 if (locked) { 524 mLock.unlock(); 525 } 526} 527 528void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 529{ 530 const size_t SIZE = 256; 531 char buffer[SIZE]; 532 String8 result; 533 534 size_t numEffectChains = mEffectChains.size(); 535 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 536 write(fd, buffer, strlen(buffer)); 537 538 for (size_t i = 0; i < numEffectChains; ++i) { 539 sp<EffectChain> chain = mEffectChains[i]; 540 if (chain != 0) { 541 chain->dump(fd, args); 542 } 543 } 544} 545 546void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 547{ 548 Mutex::Autolock _l(mLock); 549 acquireWakeLock_l(uid); 550} 551 552String16 AudioFlinger::ThreadBase::getWakeLockTag() 553{ 554 switch (mType) { 555 case MIXER: 556 return String16("AudioMix"); 557 case DIRECT: 558 return String16("AudioDirectOut"); 559 case DUPLICATING: 560 return String16("AudioDup"); 561 case RECORD: 562 return String16("AudioIn"); 563 case OFFLOAD: 564 return String16("AudioOffload"); 565 default: 566 ALOG_ASSERT(false); 567 return String16("AudioUnknown"); 568 } 569} 570 571void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 572{ 573 getPowerManager_l(); 574 if (mPowerManager != 0) { 575 sp<IBinder> binder = new BBinder(); 576 status_t status; 577 if (uid >= 0) { 578 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 579 binder, 580 getWakeLockTag(), 581 String16("media"), 582 uid); 583 } else { 584 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 585 binder, 586 getWakeLockTag(), 587 String16("media")); 588 } 589 if (status == NO_ERROR) { 590 mWakeLockToken = binder; 591 } 592 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 593 } 594} 595 596void AudioFlinger::ThreadBase::releaseWakeLock() 597{ 598 Mutex::Autolock _l(mLock); 599 releaseWakeLock_l(); 600} 601 602void AudioFlinger::ThreadBase::releaseWakeLock_l() 603{ 604 if (mWakeLockToken != 0) { 605 ALOGV("releaseWakeLock_l() %s", mName); 606 if (mPowerManager != 0) { 607 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 608 } 609 mWakeLockToken.clear(); 610 } 611} 612 613void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 614 Mutex::Autolock _l(mLock); 615 updateWakeLockUids_l(uids); 616} 617 618void AudioFlinger::ThreadBase::getPowerManager_l() { 619 620 if (mPowerManager == 0) { 621 // use checkService() to avoid blocking if power service is not up yet 622 sp<IBinder> binder = 623 defaultServiceManager()->checkService(String16("power")); 624 if (binder == 0) { 625 ALOGW("Thread %s cannot connect to the power manager service", mName); 626 } else { 627 mPowerManager = interface_cast<IPowerManager>(binder); 628 binder->linkToDeath(mDeathRecipient); 629 } 630 } 631} 632 633void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 634 635 getPowerManager_l(); 636 if (mWakeLockToken == NULL) { 637 ALOGE("no wake lock to update!"); 638 return; 639 } 640 if (mPowerManager != 0) { 641 sp<IBinder> binder = new BBinder(); 642 status_t status; 643 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 644 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 645 } 646} 647 648void AudioFlinger::ThreadBase::clearPowerManager() 649{ 650 Mutex::Autolock _l(mLock); 651 releaseWakeLock_l(); 652 mPowerManager.clear(); 653} 654 655void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 656{ 657 sp<ThreadBase> thread = mThread.promote(); 658 if (thread != 0) { 659 thread->clearPowerManager(); 660 } 661 ALOGW("power manager service died !!!"); 662} 663 664void AudioFlinger::ThreadBase::setEffectSuspended( 665 const effect_uuid_t *type, bool suspend, int sessionId) 666{ 667 Mutex::Autolock _l(mLock); 668 setEffectSuspended_l(type, suspend, sessionId); 669} 670 671void AudioFlinger::ThreadBase::setEffectSuspended_l( 672 const effect_uuid_t *type, bool suspend, int sessionId) 673{ 674 sp<EffectChain> chain = getEffectChain_l(sessionId); 675 if (chain != 0) { 676 if (type != NULL) { 677 chain->setEffectSuspended_l(type, suspend); 678 } else { 679 chain->setEffectSuspendedAll_l(suspend); 680 } 681 } 682 683 updateSuspendedSessions_l(type, suspend, sessionId); 684} 685 686void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 687{ 688 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 689 if (index < 0) { 690 return; 691 } 692 693 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 694 mSuspendedSessions.valueAt(index); 695 696 for (size_t i = 0; i < sessionEffects.size(); i++) { 697 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 698 for (int j = 0; j < desc->mRefCount; j++) { 699 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 700 chain->setEffectSuspendedAll_l(true); 701 } else { 702 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 703 desc->mType.timeLow); 704 chain->setEffectSuspended_l(&desc->mType, true); 705 } 706 } 707 } 708} 709 710void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 711 bool suspend, 712 int sessionId) 713{ 714 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 715 716 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 717 718 if (suspend) { 719 if (index >= 0) { 720 sessionEffects = mSuspendedSessions.valueAt(index); 721 } else { 722 mSuspendedSessions.add(sessionId, sessionEffects); 723 } 724 } else { 725 if (index < 0) { 726 return; 727 } 728 sessionEffects = mSuspendedSessions.valueAt(index); 729 } 730 731 732 int key = EffectChain::kKeyForSuspendAll; 733 if (type != NULL) { 734 key = type->timeLow; 735 } 736 index = sessionEffects.indexOfKey(key); 737 738 sp<SuspendedSessionDesc> desc; 739 if (suspend) { 740 if (index >= 0) { 741 desc = sessionEffects.valueAt(index); 742 } else { 743 desc = new SuspendedSessionDesc(); 744 if (type != NULL) { 745 desc->mType = *type; 746 } 747 sessionEffects.add(key, desc); 748 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 749 } 750 desc->mRefCount++; 751 } else { 752 if (index < 0) { 753 return; 754 } 755 desc = sessionEffects.valueAt(index); 756 if (--desc->mRefCount == 0) { 757 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 758 sessionEffects.removeItemsAt(index); 759 if (sessionEffects.isEmpty()) { 760 ALOGV("updateSuspendedSessions_l() restore removing session %d", 761 sessionId); 762 mSuspendedSessions.removeItem(sessionId); 763 } 764 } 765 } 766 if (!sessionEffects.isEmpty()) { 767 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 768 } 769} 770 771void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 772 bool enabled, 773 int sessionId) 774{ 775 Mutex::Autolock _l(mLock); 776 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 777} 778 779void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 780 bool enabled, 781 int sessionId) 782{ 783 if (mType != RECORD) { 784 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 785 // another session. This gives the priority to well behaved effect control panels 786 // and applications not using global effects. 787 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 788 // global effects 789 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 790 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 791 } 792 } 793 794 sp<EffectChain> chain = getEffectChain_l(sessionId); 795 if (chain != 0) { 796 chain->checkSuspendOnEffectEnabled(effect, enabled); 797 } 798} 799 800// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 801sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 802 const sp<AudioFlinger::Client>& client, 803 const sp<IEffectClient>& effectClient, 804 int32_t priority, 805 int sessionId, 806 effect_descriptor_t *desc, 807 int *enabled, 808 status_t *status) 809{ 810 sp<EffectModule> effect; 811 sp<EffectHandle> handle; 812 status_t lStatus; 813 sp<EffectChain> chain; 814 bool chainCreated = false; 815 bool effectCreated = false; 816 bool effectRegistered = false; 817 818 lStatus = initCheck(); 819 if (lStatus != NO_ERROR) { 820 ALOGW("createEffect_l() Audio driver not initialized."); 821 goto Exit; 822 } 823 824 // Reject any effect on Direct output threads for now, since the format of 825 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 826 if (mType == DIRECT) { 827 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 828 desc->name, mName); 829 lStatus = BAD_VALUE; 830 goto Exit; 831 } 832 833 // Allow global effects only on offloaded and mixer threads 834 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 835 switch (mType) { 836 case MIXER: 837 case OFFLOAD: 838 break; 839 case DIRECT: 840 case DUPLICATING: 841 case RECORD: 842 default: 843 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 844 lStatus = BAD_VALUE; 845 goto Exit; 846 } 847 } 848 849 // Only Pre processor effects are allowed on input threads and only on input threads 850 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 851 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 852 desc->name, desc->flags, mType); 853 lStatus = BAD_VALUE; 854 goto Exit; 855 } 856 857 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 858 859 { // scope for mLock 860 Mutex::Autolock _l(mLock); 861 862 // check for existing effect chain with the requested audio session 863 chain = getEffectChain_l(sessionId); 864 if (chain == 0) { 865 // create a new chain for this session 866 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 867 chain = new EffectChain(this, sessionId); 868 addEffectChain_l(chain); 869 chain->setStrategy(getStrategyForSession_l(sessionId)); 870 chainCreated = true; 871 } else { 872 effect = chain->getEffectFromDesc_l(desc); 873 } 874 875 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 876 877 if (effect == 0) { 878 int id = mAudioFlinger->nextUniqueId(); 879 // Check CPU and memory usage 880 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 881 if (lStatus != NO_ERROR) { 882 goto Exit; 883 } 884 effectRegistered = true; 885 // create a new effect module if none present in the chain 886 effect = new EffectModule(this, chain, desc, id, sessionId); 887 lStatus = effect->status(); 888 if (lStatus != NO_ERROR) { 889 goto Exit; 890 } 891 effect->setOffloaded(mType == OFFLOAD, mId); 892 893 lStatus = chain->addEffect_l(effect); 894 if (lStatus != NO_ERROR) { 895 goto Exit; 896 } 897 effectCreated = true; 898 899 effect->setDevice(mOutDevice); 900 effect->setDevice(mInDevice); 901 effect->setMode(mAudioFlinger->getMode()); 902 effect->setAudioSource(mAudioSource); 903 } 904 // create effect handle and connect it to effect module 905 handle = new EffectHandle(effect, client, effectClient, priority); 906 lStatus = handle->initCheck(); 907 if (lStatus == OK) { 908 lStatus = effect->addHandle(handle.get()); 909 } 910 if (enabled != NULL) { 911 *enabled = (int)effect->isEnabled(); 912 } 913 } 914 915Exit: 916 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 917 Mutex::Autolock _l(mLock); 918 if (effectCreated) { 919 chain->removeEffect_l(effect); 920 } 921 if (effectRegistered) { 922 AudioSystem::unregisterEffect(effect->id()); 923 } 924 if (chainCreated) { 925 removeEffectChain_l(chain); 926 } 927 handle.clear(); 928 } 929 930 *status = lStatus; 931 return handle; 932} 933 934sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 935{ 936 Mutex::Autolock _l(mLock); 937 return getEffect_l(sessionId, effectId); 938} 939 940sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 941{ 942 sp<EffectChain> chain = getEffectChain_l(sessionId); 943 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 944} 945 946// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 947// PlaybackThread::mLock held 948status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 949{ 950 // check for existing effect chain with the requested audio session 951 int sessionId = effect->sessionId(); 952 sp<EffectChain> chain = getEffectChain_l(sessionId); 953 bool chainCreated = false; 954 955 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 956 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 957 this, effect->desc().name, effect->desc().flags); 958 959 if (chain == 0) { 960 // create a new chain for this session 961 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 962 chain = new EffectChain(this, sessionId); 963 addEffectChain_l(chain); 964 chain->setStrategy(getStrategyForSession_l(sessionId)); 965 chainCreated = true; 966 } 967 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 968 969 if (chain->getEffectFromId_l(effect->id()) != 0) { 970 ALOGW("addEffect_l() %p effect %s already present in chain %p", 971 this, effect->desc().name, chain.get()); 972 return BAD_VALUE; 973 } 974 975 effect->setOffloaded(mType == OFFLOAD, mId); 976 977 status_t status = chain->addEffect_l(effect); 978 if (status != NO_ERROR) { 979 if (chainCreated) { 980 removeEffectChain_l(chain); 981 } 982 return status; 983 } 984 985 effect->setDevice(mOutDevice); 986 effect->setDevice(mInDevice); 987 effect->setMode(mAudioFlinger->getMode()); 988 effect->setAudioSource(mAudioSource); 989 return NO_ERROR; 990} 991 992void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 993 994 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 995 effect_descriptor_t desc = effect->desc(); 996 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 997 detachAuxEffect_l(effect->id()); 998 } 999 1000 sp<EffectChain> chain = effect->chain().promote(); 1001 if (chain != 0) { 1002 // remove effect chain if removing last effect 1003 if (chain->removeEffect_l(effect) == 0) { 1004 removeEffectChain_l(chain); 1005 } 1006 } else { 1007 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1008 } 1009} 1010 1011void AudioFlinger::ThreadBase::lockEffectChains_l( 1012 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1013{ 1014 effectChains = mEffectChains; 1015 for (size_t i = 0; i < mEffectChains.size(); i++) { 1016 mEffectChains[i]->lock(); 1017 } 1018} 1019 1020void AudioFlinger::ThreadBase::unlockEffectChains( 1021 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1022{ 1023 for (size_t i = 0; i < effectChains.size(); i++) { 1024 effectChains[i]->unlock(); 1025 } 1026} 1027 1028sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 return getEffectChain_l(sessionId); 1032} 1033 1034sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1035{ 1036 size_t size = mEffectChains.size(); 1037 for (size_t i = 0; i < size; i++) { 1038 if (mEffectChains[i]->sessionId() == sessionId) { 1039 return mEffectChains[i]; 1040 } 1041 } 1042 return 0; 1043} 1044 1045void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1046{ 1047 Mutex::Autolock _l(mLock); 1048 size_t size = mEffectChains.size(); 1049 for (size_t i = 0; i < size; i++) { 1050 mEffectChains[i]->setMode_l(mode); 1051 } 1052} 1053 1054void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1055 EffectHandle *handle, 1056 bool unpinIfLast) { 1057 1058 Mutex::Autolock _l(mLock); 1059 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1060 // delete the effect module if removing last handle on it 1061 if (effect->removeHandle(handle) == 0) { 1062 if (!effect->isPinned() || unpinIfLast) { 1063 removeEffect_l(effect); 1064 AudioSystem::unregisterEffect(effect->id()); 1065 } 1066 } 1067} 1068 1069// ---------------------------------------------------------------------------- 1070// Playback 1071// ---------------------------------------------------------------------------- 1072 1073AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1074 AudioStreamOut* output, 1075 audio_io_handle_t id, 1076 audio_devices_t device, 1077 type_t type) 1078 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1079 mNormalFrameCount(0), mSinkBuffer(NULL), 1080 mMixerBufferEnabled(false), 1081 mMixerBuffer(NULL), 1082 mMixerBufferSize(0), 1083 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1084 mMixerBufferValid(false), 1085 mEffectBufferEnabled(false), 1086 mEffectBuffer(NULL), 1087 mEffectBufferSize(0), 1088 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1089 mEffectBufferValid(false), 1090 mSuspended(0), mBytesWritten(0), 1091 mActiveTracksGeneration(0), 1092 // mStreamTypes[] initialized in constructor body 1093 mOutput(output), 1094 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1095 mMixerStatus(MIXER_IDLE), 1096 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1097 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1098 mBytesRemaining(0), 1099 mCurrentWriteLength(0), 1100 mUseAsyncWrite(false), 1101 mWriteAckSequence(0), 1102 mDrainSequence(0), 1103 mSignalPending(false), 1104 mScreenState(AudioFlinger::mScreenState), 1105 // index 0 is reserved for normal mixer's submix 1106 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1107 // mLatchD, mLatchQ, 1108 mLatchDValid(false), mLatchQValid(false) 1109{ 1110 snprintf(mName, kNameLength, "AudioOut_%X", id); 1111 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1112 1113 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1114 // it would be safer to explicitly pass initial masterVolume/masterMute as 1115 // parameter. 1116 // 1117 // If the HAL we are using has support for master volume or master mute, 1118 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1119 // and the mute set to false). 1120 mMasterVolume = audioFlinger->masterVolume_l(); 1121 mMasterMute = audioFlinger->masterMute_l(); 1122 if (mOutput && mOutput->audioHwDev) { 1123 if (mOutput->audioHwDev->canSetMasterVolume()) { 1124 mMasterVolume = 1.0; 1125 } 1126 1127 if (mOutput->audioHwDev->canSetMasterMute()) { 1128 mMasterMute = false; 1129 } 1130 } 1131 1132 readOutputParameters_l(); 1133 1134 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1135 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1136 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1137 stream = (audio_stream_type_t) (stream + 1)) { 1138 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1139 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1140 } 1141 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1142 // because mAudioFlinger doesn't have one to copy from 1143} 1144 1145AudioFlinger::PlaybackThread::~PlaybackThread() 1146{ 1147 mAudioFlinger->unregisterWriter(mNBLogWriter); 1148 free(mSinkBuffer); 1149 free(mMixerBuffer); 1150 free(mEffectBuffer); 1151} 1152 1153void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1154{ 1155 dumpInternals(fd, args); 1156 dumpTracks(fd, args); 1157 dumpEffectChains(fd, args); 1158} 1159 1160void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1161{ 1162 const size_t SIZE = 256; 1163 char buffer[SIZE]; 1164 String8 result; 1165 1166 result.appendFormat(" Stream volumes in dB: "); 1167 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1168 const stream_type_t *st = &mStreamTypes[i]; 1169 if (i > 0) { 1170 result.appendFormat(", "); 1171 } 1172 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1173 if (st->mute) { 1174 result.append("M"); 1175 } 1176 } 1177 result.append("\n"); 1178 write(fd, result.string(), result.length()); 1179 result.clear(); 1180 1181 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1182 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1183 fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1184 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1185 1186 size_t numtracks = mTracks.size(); 1187 size_t numactive = mActiveTracks.size(); 1188 fdprintf(fd, " %d Tracks", numtracks); 1189 size_t numactiveseen = 0; 1190 if (numtracks) { 1191 fdprintf(fd, " of which %d are active\n", numactive); 1192 Track::appendDumpHeader(result); 1193 for (size_t i = 0; i < numtracks; ++i) { 1194 sp<Track> track = mTracks[i]; 1195 if (track != 0) { 1196 bool active = mActiveTracks.indexOf(track) >= 0; 1197 if (active) { 1198 numactiveseen++; 1199 } 1200 track->dump(buffer, SIZE, active); 1201 result.append(buffer); 1202 } 1203 } 1204 } else { 1205 result.append("\n"); 1206 } 1207 if (numactiveseen != numactive) { 1208 // some tracks in the active list were not in the tracks list 1209 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1210 " not in the track list\n"); 1211 result.append(buffer); 1212 Track::appendDumpHeader(result); 1213 for (size_t i = 0; i < numactive; ++i) { 1214 sp<Track> track = mActiveTracks[i].promote(); 1215 if (track != 0 && mTracks.indexOf(track) < 0) { 1216 track->dump(buffer, SIZE, true); 1217 result.append(buffer); 1218 } 1219 } 1220 } 1221 1222 write(fd, result.string(), result.size()); 1223 1224} 1225 1226void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1227{ 1228 fdprintf(fd, "\nOutput thread %p:\n", this); 1229 fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1230 fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1231 fdprintf(fd, " Total writes: %d\n", mNumWrites); 1232 fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1233 fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1234 fdprintf(fd, " Suspend count: %d\n", mSuspended); 1235 fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1236 fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1237 fdprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1238 fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1239 1240 dumpBase(fd, args); 1241} 1242 1243// Thread virtuals 1244 1245void AudioFlinger::PlaybackThread::onFirstRef() 1246{ 1247 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1248} 1249 1250// ThreadBase virtuals 1251void AudioFlinger::PlaybackThread::preExit() 1252{ 1253 ALOGV(" preExit()"); 1254 // FIXME this is using hard-coded strings but in the future, this functionality will be 1255 // converted to use audio HAL extensions required to support tunneling 1256 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1257} 1258 1259// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1260sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1261 const sp<AudioFlinger::Client>& client, 1262 audio_stream_type_t streamType, 1263 uint32_t sampleRate, 1264 audio_format_t format, 1265 audio_channel_mask_t channelMask, 1266 size_t *pFrameCount, 1267 const sp<IMemory>& sharedBuffer, 1268 int sessionId, 1269 IAudioFlinger::track_flags_t *flags, 1270 pid_t tid, 1271 int uid, 1272 status_t *status) 1273{ 1274 size_t frameCount = *pFrameCount; 1275 sp<Track> track; 1276 status_t lStatus; 1277 1278 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1279 1280 // client expresses a preference for FAST, but we get the final say 1281 if (*flags & IAudioFlinger::TRACK_FAST) { 1282 if ( 1283 // not timed 1284 (!isTimed) && 1285 // either of these use cases: 1286 ( 1287 // use case 1: shared buffer with any frame count 1288 ( 1289 (sharedBuffer != 0) 1290 ) || 1291 // use case 2: callback handler and frame count is default or at least as large as HAL 1292 ( 1293 (tid != -1) && 1294 ((frameCount == 0) || 1295 (frameCount >= mFrameCount)) 1296 ) 1297 ) && 1298 // PCM data 1299 audio_is_linear_pcm(format) && 1300 // mono or stereo 1301 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1302 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1303 // hardware sample rate 1304 (sampleRate == mSampleRate) && 1305 // normal mixer has an associated fast mixer 1306 hasFastMixer() && 1307 // there are sufficient fast track slots available 1308 (mFastTrackAvailMask != 0) 1309 // FIXME test that MixerThread for this fast track has a capable output HAL 1310 // FIXME add a permission test also? 1311 ) { 1312 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1313 if (frameCount == 0) { 1314 frameCount = mFrameCount * kFastTrackMultiplier; 1315 } 1316 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1317 frameCount, mFrameCount); 1318 } else { 1319 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1320 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1321 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1322 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1323 audio_is_linear_pcm(format), 1324 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1325 *flags &= ~IAudioFlinger::TRACK_FAST; 1326 // For compatibility with AudioTrack calculation, buffer depth is forced 1327 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1328 // This is probably too conservative, but legacy application code may depend on it. 1329 // If you change this calculation, also review the start threshold which is related. 1330 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1331 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1332 if (minBufCount < 2) { 1333 minBufCount = 2; 1334 } 1335 size_t minFrameCount = mNormalFrameCount * minBufCount; 1336 if (frameCount < minFrameCount) { 1337 frameCount = minFrameCount; 1338 } 1339 } 1340 } 1341 *pFrameCount = frameCount; 1342 1343 if (mType == DIRECT) { 1344 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1345 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1346 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1347 "for output %p with format %#x", 1348 sampleRate, format, channelMask, mOutput, mFormat); 1349 lStatus = BAD_VALUE; 1350 goto Exit; 1351 } 1352 } 1353 } else if (mType == OFFLOAD) { 1354 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1355 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1356 "for output %p with format %#x", 1357 sampleRate, format, channelMask, mOutput, mFormat); 1358 lStatus = BAD_VALUE; 1359 goto Exit; 1360 } 1361 } else { 1362 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1363 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1364 "for output %p with format %#x", 1365 format, mOutput, mFormat); 1366 lStatus = BAD_VALUE; 1367 goto Exit; 1368 } 1369 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1370 if (sampleRate > mSampleRate*2) { 1371 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1372 lStatus = BAD_VALUE; 1373 goto Exit; 1374 } 1375 } 1376 1377 lStatus = initCheck(); 1378 if (lStatus != NO_ERROR) { 1379 ALOGE("Audio driver not initialized."); 1380 goto Exit; 1381 } 1382 1383 { // scope for mLock 1384 Mutex::Autolock _l(mLock); 1385 1386 // all tracks in same audio session must share the same routing strategy otherwise 1387 // conflicts will happen when tracks are moved from one output to another by audio policy 1388 // manager 1389 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1390 for (size_t i = 0; i < mTracks.size(); ++i) { 1391 sp<Track> t = mTracks[i]; 1392 if (t != 0 && !t->isOutputTrack()) { 1393 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1394 if (sessionId == t->sessionId() && strategy != actual) { 1395 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1396 strategy, actual); 1397 lStatus = BAD_VALUE; 1398 goto Exit; 1399 } 1400 } 1401 } 1402 1403 if (!isTimed) { 1404 track = new Track(this, client, streamType, sampleRate, format, 1405 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1406 } else { 1407 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1408 channelMask, frameCount, sharedBuffer, sessionId, uid); 1409 } 1410 1411 // new Track always returns non-NULL, 1412 // but TimedTrack::create() is a factory that could fail by returning NULL 1413 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1414 if (lStatus != NO_ERROR) { 1415 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1416 // track must be cleared from the caller as the caller has the AF lock 1417 goto Exit; 1418 } 1419 1420 mTracks.add(track); 1421 1422 sp<EffectChain> chain = getEffectChain_l(sessionId); 1423 if (chain != 0) { 1424 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1425 track->setMainBuffer(chain->inBuffer()); 1426 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1427 chain->incTrackCnt(); 1428 } 1429 1430 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1431 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1432 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1433 // so ask activity manager to do this on our behalf 1434 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1435 } 1436 } 1437 1438 lStatus = NO_ERROR; 1439 1440Exit: 1441 *status = lStatus; 1442 return track; 1443} 1444 1445uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1446{ 1447 return latency; 1448} 1449 1450uint32_t AudioFlinger::PlaybackThread::latency() const 1451{ 1452 Mutex::Autolock _l(mLock); 1453 return latency_l(); 1454} 1455uint32_t AudioFlinger::PlaybackThread::latency_l() const 1456{ 1457 if (initCheck() == NO_ERROR) { 1458 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1459 } else { 1460 return 0; 1461 } 1462} 1463 1464void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1465{ 1466 Mutex::Autolock _l(mLock); 1467 // Don't apply master volume in SW if our HAL can do it for us. 1468 if (mOutput && mOutput->audioHwDev && 1469 mOutput->audioHwDev->canSetMasterVolume()) { 1470 mMasterVolume = 1.0; 1471 } else { 1472 mMasterVolume = value; 1473 } 1474} 1475 1476void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1477{ 1478 Mutex::Autolock _l(mLock); 1479 // Don't apply master mute in SW if our HAL can do it for us. 1480 if (mOutput && mOutput->audioHwDev && 1481 mOutput->audioHwDev->canSetMasterMute()) { 1482 mMasterMute = false; 1483 } else { 1484 mMasterMute = muted; 1485 } 1486} 1487 1488void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1489{ 1490 Mutex::Autolock _l(mLock); 1491 mStreamTypes[stream].volume = value; 1492 broadcast_l(); 1493} 1494 1495void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1496{ 1497 Mutex::Autolock _l(mLock); 1498 mStreamTypes[stream].mute = muted; 1499 broadcast_l(); 1500} 1501 1502float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1503{ 1504 Mutex::Autolock _l(mLock); 1505 return mStreamTypes[stream].volume; 1506} 1507 1508// addTrack_l() must be called with ThreadBase::mLock held 1509status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1510{ 1511 status_t status = ALREADY_EXISTS; 1512 1513 // set retry count for buffer fill 1514 track->mRetryCount = kMaxTrackStartupRetries; 1515 if (mActiveTracks.indexOf(track) < 0) { 1516 // the track is newly added, make sure it fills up all its 1517 // buffers before playing. This is to ensure the client will 1518 // effectively get the latency it requested. 1519 if (!track->isOutputTrack()) { 1520 TrackBase::track_state state = track->mState; 1521 mLock.unlock(); 1522 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1523 mLock.lock(); 1524 // abort track was stopped/paused while we released the lock 1525 if (state != track->mState) { 1526 if (status == NO_ERROR) { 1527 mLock.unlock(); 1528 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1529 mLock.lock(); 1530 } 1531 return INVALID_OPERATION; 1532 } 1533 // abort if start is rejected by audio policy manager 1534 if (status != NO_ERROR) { 1535 return PERMISSION_DENIED; 1536 } 1537#ifdef ADD_BATTERY_DATA 1538 // to track the speaker usage 1539 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1540#endif 1541 } 1542 1543 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1544 track->mResetDone = false; 1545 track->mPresentationCompleteFrames = 0; 1546 mActiveTracks.add(track); 1547 mWakeLockUids.add(track->uid()); 1548 mActiveTracksGeneration++; 1549 mLatestActiveTrack = track; 1550 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1551 if (chain != 0) { 1552 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1553 track->sessionId()); 1554 chain->incActiveTrackCnt(); 1555 } 1556 1557 status = NO_ERROR; 1558 } 1559 1560 onAddNewTrack_l(); 1561 return status; 1562} 1563 1564bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1565{ 1566 track->terminate(); 1567 // active tracks are removed by threadLoop() 1568 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1569 track->mState = TrackBase::STOPPED; 1570 if (!trackActive) { 1571 removeTrack_l(track); 1572 } else if (track->isFastTrack() || track->isOffloaded()) { 1573 track->mState = TrackBase::STOPPING_1; 1574 } 1575 1576 return trackActive; 1577} 1578 1579void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1580{ 1581 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1582 mTracks.remove(track); 1583 deleteTrackName_l(track->name()); 1584 // redundant as track is about to be destroyed, for dumpsys only 1585 track->mName = -1; 1586 if (track->isFastTrack()) { 1587 int index = track->mFastIndex; 1588 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1589 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1590 mFastTrackAvailMask |= 1 << index; 1591 // redundant as track is about to be destroyed, for dumpsys only 1592 track->mFastIndex = -1; 1593 } 1594 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1595 if (chain != 0) { 1596 chain->decTrackCnt(); 1597 } 1598} 1599 1600void AudioFlinger::PlaybackThread::broadcast_l() 1601{ 1602 // Thread could be blocked waiting for async 1603 // so signal it to handle state changes immediately 1604 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1605 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1606 mSignalPending = true; 1607 mWaitWorkCV.broadcast(); 1608} 1609 1610String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1611{ 1612 Mutex::Autolock _l(mLock); 1613 if (initCheck() != NO_ERROR) { 1614 return String8(); 1615 } 1616 1617 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1618 const String8 out_s8(s); 1619 free(s); 1620 return out_s8; 1621} 1622 1623// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1624void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1625 AudioSystem::OutputDescriptor desc; 1626 void *param2 = NULL; 1627 1628 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1629 param); 1630 1631 switch (event) { 1632 case AudioSystem::OUTPUT_OPENED: 1633 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1634 desc.channelMask = mChannelMask; 1635 desc.samplingRate = mSampleRate; 1636 desc.format = mFormat; 1637 desc.frameCount = mNormalFrameCount; // FIXME see 1638 // AudioFlinger::frameCount(audio_io_handle_t) 1639 desc.latency = latency(); 1640 param2 = &desc; 1641 break; 1642 1643 case AudioSystem::STREAM_CONFIG_CHANGED: 1644 param2 = ¶m; 1645 case AudioSystem::OUTPUT_CLOSED: 1646 default: 1647 break; 1648 } 1649 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1650} 1651 1652void AudioFlinger::PlaybackThread::writeCallback() 1653{ 1654 ALOG_ASSERT(mCallbackThread != 0); 1655 mCallbackThread->resetWriteBlocked(); 1656} 1657 1658void AudioFlinger::PlaybackThread::drainCallback() 1659{ 1660 ALOG_ASSERT(mCallbackThread != 0); 1661 mCallbackThread->resetDraining(); 1662} 1663 1664void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1665{ 1666 Mutex::Autolock _l(mLock); 1667 // reject out of sequence requests 1668 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1669 mWriteAckSequence &= ~1; 1670 mWaitWorkCV.signal(); 1671 } 1672} 1673 1674void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1675{ 1676 Mutex::Autolock _l(mLock); 1677 // reject out of sequence requests 1678 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1679 mDrainSequence &= ~1; 1680 mWaitWorkCV.signal(); 1681 } 1682} 1683 1684// static 1685int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1686 void *param __unused, 1687 void *cookie) 1688{ 1689 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1690 ALOGV("asyncCallback() event %d", event); 1691 switch (event) { 1692 case STREAM_CBK_EVENT_WRITE_READY: 1693 me->writeCallback(); 1694 break; 1695 case STREAM_CBK_EVENT_DRAIN_READY: 1696 me->drainCallback(); 1697 break; 1698 default: 1699 ALOGW("asyncCallback() unknown event %d", event); 1700 break; 1701 } 1702 return 0; 1703} 1704 1705void AudioFlinger::PlaybackThread::readOutputParameters_l() 1706{ 1707 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1708 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1709 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1710 if (!audio_is_output_channel(mChannelMask)) { 1711 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1712 } 1713 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1714 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1715 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1716 } 1717 mChannelCount = popcount(mChannelMask); 1718 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1719 if (!audio_is_valid_format(mFormat)) { 1720 LOG_FATAL("HAL format %#x not valid for output", mFormat); 1721 } 1722 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1723 LOG_FATAL("HAL format %#x not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1724 mFormat); 1725 } 1726 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1727 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1728 mFrameCount = mBufferSize / mFrameSize; 1729 if (mFrameCount & 15) { 1730 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1731 mFrameCount); 1732 } 1733 1734 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1735 (mOutput->stream->set_callback != NULL)) { 1736 if (mOutput->stream->set_callback(mOutput->stream, 1737 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1738 mUseAsyncWrite = true; 1739 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1740 } 1741 } 1742 1743 // Calculate size of normal sink buffer relative to the HAL output buffer size 1744 double multiplier = 1.0; 1745 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1746 kUseFastMixer == FastMixer_Dynamic)) { 1747 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1748 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1749 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1750 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1751 maxNormalFrameCount = maxNormalFrameCount & ~15; 1752 if (maxNormalFrameCount < minNormalFrameCount) { 1753 maxNormalFrameCount = minNormalFrameCount; 1754 } 1755 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1756 if (multiplier <= 1.0) { 1757 multiplier = 1.0; 1758 } else if (multiplier <= 2.0) { 1759 if (2 * mFrameCount <= maxNormalFrameCount) { 1760 multiplier = 2.0; 1761 } else { 1762 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1763 } 1764 } else { 1765 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1766 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1767 // track, but we sometimes have to do this to satisfy the maximum frame count 1768 // constraint) 1769 // FIXME this rounding up should not be done if no HAL SRC 1770 uint32_t truncMult = (uint32_t) multiplier; 1771 if ((truncMult & 1)) { 1772 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1773 ++truncMult; 1774 } 1775 } 1776 multiplier = (double) truncMult; 1777 } 1778 } 1779 mNormalFrameCount = multiplier * mFrameCount; 1780 // round up to nearest 16 frames to satisfy AudioMixer 1781 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1782 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1783 mNormalFrameCount); 1784 1785 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1786 // Originally this was int16_t[] array, need to remove legacy implications. 1787 free(mSinkBuffer); 1788 mSinkBuffer = NULL; 1789 const size_t sinkBufferSize = mNormalFrameCount * mChannelCount 1790 * audio_bytes_per_sample(mFormat); 1791 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1792 1793 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1794 // drives the output. 1795 free(mMixerBuffer); 1796 mMixerBuffer = NULL; 1797 if (mMixerBufferEnabled) { 1798 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1799 mMixerBufferSize = mNormalFrameCount * mChannelCount 1800 * audio_bytes_per_sample(mMixerBufferFormat); 1801 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1802 } 1803 free(mEffectBuffer); 1804 mEffectBuffer = NULL; 1805 if (mEffectBufferEnabled) { 1806 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1807 mEffectBufferSize = mNormalFrameCount * mChannelCount 1808 * audio_bytes_per_sample(mEffectBufferFormat); 1809 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1810 } 1811 1812 // force reconfiguration of effect chains and engines to take new buffer size and audio 1813 // parameters into account 1814 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1815 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1816 // matter. 1817 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1818 Vector< sp<EffectChain> > effectChains = mEffectChains; 1819 for (size_t i = 0; i < effectChains.size(); i ++) { 1820 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1821 } 1822} 1823 1824 1825status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1826{ 1827 if (halFrames == NULL || dspFrames == NULL) { 1828 return BAD_VALUE; 1829 } 1830 Mutex::Autolock _l(mLock); 1831 if (initCheck() != NO_ERROR) { 1832 return INVALID_OPERATION; 1833 } 1834 size_t framesWritten = mBytesWritten / mFrameSize; 1835 *halFrames = framesWritten; 1836 1837 if (isSuspended()) { 1838 // return an estimation of rendered frames when the output is suspended 1839 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1840 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1841 return NO_ERROR; 1842 } else { 1843 status_t status; 1844 uint32_t frames; 1845 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1846 *dspFrames = (size_t)frames; 1847 return status; 1848 } 1849} 1850 1851uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1852{ 1853 Mutex::Autolock _l(mLock); 1854 uint32_t result = 0; 1855 if (getEffectChain_l(sessionId) != 0) { 1856 result = EFFECT_SESSION; 1857 } 1858 1859 for (size_t i = 0; i < mTracks.size(); ++i) { 1860 sp<Track> track = mTracks[i]; 1861 if (sessionId == track->sessionId() && !track->isInvalid()) { 1862 result |= TRACK_SESSION; 1863 break; 1864 } 1865 } 1866 1867 return result; 1868} 1869 1870uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1871{ 1872 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1873 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1874 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1875 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1876 } 1877 for (size_t i = 0; i < mTracks.size(); i++) { 1878 sp<Track> track = mTracks[i]; 1879 if (sessionId == track->sessionId() && !track->isInvalid()) { 1880 return AudioSystem::getStrategyForStream(track->streamType()); 1881 } 1882 } 1883 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1884} 1885 1886 1887AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1888{ 1889 Mutex::Autolock _l(mLock); 1890 return mOutput; 1891} 1892 1893AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1894{ 1895 Mutex::Autolock _l(mLock); 1896 AudioStreamOut *output = mOutput; 1897 mOutput = NULL; 1898 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1899 // must push a NULL and wait for ack 1900 mOutputSink.clear(); 1901 mPipeSink.clear(); 1902 mNormalSink.clear(); 1903 return output; 1904} 1905 1906// this method must always be called either with ThreadBase mLock held or inside the thread loop 1907audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1908{ 1909 if (mOutput == NULL) { 1910 return NULL; 1911 } 1912 return &mOutput->stream->common; 1913} 1914 1915uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1916{ 1917 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1918} 1919 1920status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1921{ 1922 if (!isValidSyncEvent(event)) { 1923 return BAD_VALUE; 1924 } 1925 1926 Mutex::Autolock _l(mLock); 1927 1928 for (size_t i = 0; i < mTracks.size(); ++i) { 1929 sp<Track> track = mTracks[i]; 1930 if (event->triggerSession() == track->sessionId()) { 1931 (void) track->setSyncEvent(event); 1932 return NO_ERROR; 1933 } 1934 } 1935 1936 return NAME_NOT_FOUND; 1937} 1938 1939bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1940{ 1941 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1942} 1943 1944void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1945 const Vector< sp<Track> >& tracksToRemove) 1946{ 1947 size_t count = tracksToRemove.size(); 1948 if (count > 0) { 1949 for (size_t i = 0 ; i < count ; i++) { 1950 const sp<Track>& track = tracksToRemove.itemAt(i); 1951 if (!track->isOutputTrack()) { 1952 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1953#ifdef ADD_BATTERY_DATA 1954 // to track the speaker usage 1955 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1956#endif 1957 if (track->isTerminated()) { 1958 AudioSystem::releaseOutput(mId); 1959 } 1960 } 1961 } 1962 } 1963} 1964 1965void AudioFlinger::PlaybackThread::checkSilentMode_l() 1966{ 1967 if (!mMasterMute) { 1968 char value[PROPERTY_VALUE_MAX]; 1969 if (property_get("ro.audio.silent", value, "0") > 0) { 1970 char *endptr; 1971 unsigned long ul = strtoul(value, &endptr, 0); 1972 if (*endptr == '\0' && ul != 0) { 1973 ALOGD("Silence is golden"); 1974 // The setprop command will not allow a property to be changed after 1975 // the first time it is set, so we don't have to worry about un-muting. 1976 setMasterMute_l(true); 1977 } 1978 } 1979 } 1980} 1981 1982// shared by MIXER and DIRECT, overridden by DUPLICATING 1983ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1984{ 1985 // FIXME rewrite to reduce number of system calls 1986 mLastWriteTime = systemTime(); 1987 mInWrite = true; 1988 ssize_t bytesWritten; 1989 const size_t offset = mCurrentWriteLength - mBytesRemaining; 1990 1991 // If an NBAIO sink is present, use it to write the normal mixer's submix 1992 if (mNormalSink != 0) { 1993 const size_t count = mBytesRemaining / mFrameSize; 1994 1995 ATRACE_BEGIN("write"); 1996 // update the setpoint when AudioFlinger::mScreenState changes 1997 uint32_t screenState = AudioFlinger::mScreenState; 1998 if (screenState != mScreenState) { 1999 mScreenState = screenState; 2000 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2001 if (pipe != NULL) { 2002 pipe->setAvgFrames((mScreenState & 1) ? 2003 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2004 } 2005 } 2006 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2007 ATRACE_END(); 2008 if (framesWritten > 0) { 2009 bytesWritten = framesWritten * mFrameSize; 2010 } else { 2011 bytesWritten = framesWritten; 2012 } 2013 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2014 if (status == NO_ERROR) { 2015 size_t totalFramesWritten = mNormalSink->framesWritten(); 2016 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2017 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2018 mLatchDValid = true; 2019 } 2020 } 2021 // otherwise use the HAL / AudioStreamOut directly 2022 } else { 2023 // Direct output and offload threads 2024 2025 if (mUseAsyncWrite) { 2026 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2027 mWriteAckSequence += 2; 2028 mWriteAckSequence |= 1; 2029 ALOG_ASSERT(mCallbackThread != 0); 2030 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2031 } 2032 // FIXME We should have an implementation of timestamps for direct output threads. 2033 // They are used e.g for multichannel PCM playback over HDMI. 2034 bytesWritten = mOutput->stream->write(mOutput->stream, 2035 (char *)mSinkBuffer + offset, mBytesRemaining); 2036 if (mUseAsyncWrite && 2037 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2038 // do not wait for async callback in case of error of full write 2039 mWriteAckSequence &= ~1; 2040 ALOG_ASSERT(mCallbackThread != 0); 2041 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2042 } 2043 } 2044 2045 mNumWrites++; 2046 mInWrite = false; 2047 mStandby = false; 2048 return bytesWritten; 2049} 2050 2051void AudioFlinger::PlaybackThread::threadLoop_drain() 2052{ 2053 if (mOutput->stream->drain) { 2054 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2055 if (mUseAsyncWrite) { 2056 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2057 mDrainSequence |= 1; 2058 ALOG_ASSERT(mCallbackThread != 0); 2059 mCallbackThread->setDraining(mDrainSequence); 2060 } 2061 mOutput->stream->drain(mOutput->stream, 2062 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2063 : AUDIO_DRAIN_ALL); 2064 } 2065} 2066 2067void AudioFlinger::PlaybackThread::threadLoop_exit() 2068{ 2069 // Default implementation has nothing to do 2070} 2071 2072/* 2073The derived values that are cached: 2074 - mSinkBufferSize from frame count * frame size 2075 - activeSleepTime from activeSleepTimeUs() 2076 - idleSleepTime from idleSleepTimeUs() 2077 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2078 - maxPeriod from frame count and sample rate (MIXER only) 2079 2080The parameters that affect these derived values are: 2081 - frame count 2082 - frame size 2083 - sample rate 2084 - device type: A2DP or not 2085 - device latency 2086 - format: PCM or not 2087 - active sleep time 2088 - idle sleep time 2089*/ 2090 2091void AudioFlinger::PlaybackThread::cacheParameters_l() 2092{ 2093 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2094 activeSleepTime = activeSleepTimeUs(); 2095 idleSleepTime = idleSleepTimeUs(); 2096} 2097 2098void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2099{ 2100 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2101 this, streamType, mTracks.size()); 2102 Mutex::Autolock _l(mLock); 2103 2104 size_t size = mTracks.size(); 2105 for (size_t i = 0; i < size; i++) { 2106 sp<Track> t = mTracks[i]; 2107 if (t->streamType() == streamType) { 2108 t->invalidate(); 2109 } 2110 } 2111} 2112 2113status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2114{ 2115 int session = chain->sessionId(); 2116 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2117 ? mEffectBuffer : mSinkBuffer); 2118 bool ownsBuffer = false; 2119 2120 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2121 if (session > 0) { 2122 // Only one effect chain can be present in direct output thread and it uses 2123 // the sink buffer as input 2124 if (mType != DIRECT) { 2125 size_t numSamples = mNormalFrameCount * mChannelCount; 2126 buffer = new int16_t[numSamples]; 2127 memset(buffer, 0, numSamples * sizeof(int16_t)); 2128 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2129 ownsBuffer = true; 2130 } 2131 2132 // Attach all tracks with same session ID to this chain. 2133 for (size_t i = 0; i < mTracks.size(); ++i) { 2134 sp<Track> track = mTracks[i]; 2135 if (session == track->sessionId()) { 2136 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2137 buffer); 2138 track->setMainBuffer(buffer); 2139 chain->incTrackCnt(); 2140 } 2141 } 2142 2143 // indicate all active tracks in the chain 2144 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2145 sp<Track> track = mActiveTracks[i].promote(); 2146 if (track == 0) { 2147 continue; 2148 } 2149 if (session == track->sessionId()) { 2150 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2151 chain->incActiveTrackCnt(); 2152 } 2153 } 2154 } 2155 2156 chain->setInBuffer(buffer, ownsBuffer); 2157 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2158 ? mEffectBuffer : mSinkBuffer)); 2159 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2160 // chains list in order to be processed last as it contains output stage effects 2161 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2162 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2163 // after track specific effects and before output stage 2164 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2165 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2166 // Effect chain for other sessions are inserted at beginning of effect 2167 // chains list to be processed before output mix effects. Relative order between other 2168 // sessions is not important 2169 size_t size = mEffectChains.size(); 2170 size_t i = 0; 2171 for (i = 0; i < size; i++) { 2172 if (mEffectChains[i]->sessionId() < session) { 2173 break; 2174 } 2175 } 2176 mEffectChains.insertAt(chain, i); 2177 checkSuspendOnAddEffectChain_l(chain); 2178 2179 return NO_ERROR; 2180} 2181 2182size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2183{ 2184 int session = chain->sessionId(); 2185 2186 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2187 2188 for (size_t i = 0; i < mEffectChains.size(); i++) { 2189 if (chain == mEffectChains[i]) { 2190 mEffectChains.removeAt(i); 2191 // detach all active tracks from the chain 2192 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2193 sp<Track> track = mActiveTracks[i].promote(); 2194 if (track == 0) { 2195 continue; 2196 } 2197 if (session == track->sessionId()) { 2198 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2199 chain.get(), session); 2200 chain->decActiveTrackCnt(); 2201 } 2202 } 2203 2204 // detach all tracks with same session ID from this chain 2205 for (size_t i = 0; i < mTracks.size(); ++i) { 2206 sp<Track> track = mTracks[i]; 2207 if (session == track->sessionId()) { 2208 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2209 chain->decTrackCnt(); 2210 } 2211 } 2212 break; 2213 } 2214 } 2215 return mEffectChains.size(); 2216} 2217 2218status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2219 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2220{ 2221 Mutex::Autolock _l(mLock); 2222 return attachAuxEffect_l(track, EffectId); 2223} 2224 2225status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2226 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2227{ 2228 status_t status = NO_ERROR; 2229 2230 if (EffectId == 0) { 2231 track->setAuxBuffer(0, NULL); 2232 } else { 2233 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2234 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2235 if (effect != 0) { 2236 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2237 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2238 } else { 2239 status = INVALID_OPERATION; 2240 } 2241 } else { 2242 status = BAD_VALUE; 2243 } 2244 } 2245 return status; 2246} 2247 2248void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2249{ 2250 for (size_t i = 0; i < mTracks.size(); ++i) { 2251 sp<Track> track = mTracks[i]; 2252 if (track->auxEffectId() == effectId) { 2253 attachAuxEffect_l(track, 0); 2254 } 2255 } 2256} 2257 2258bool AudioFlinger::PlaybackThread::threadLoop() 2259{ 2260 Vector< sp<Track> > tracksToRemove; 2261 2262 standbyTime = systemTime(); 2263 2264 // MIXER 2265 nsecs_t lastWarning = 0; 2266 2267 // DUPLICATING 2268 // FIXME could this be made local to while loop? 2269 writeFrames = 0; 2270 2271 int lastGeneration = 0; 2272 2273 cacheParameters_l(); 2274 sleepTime = idleSleepTime; 2275 2276 if (mType == MIXER) { 2277 sleepTimeShift = 0; 2278 } 2279 2280 CpuStats cpuStats; 2281 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2282 2283 acquireWakeLock(); 2284 2285 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2286 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2287 // and then that string will be logged at the next convenient opportunity. 2288 const char *logString = NULL; 2289 2290 checkSilentMode_l(); 2291 2292 while (!exitPending()) 2293 { 2294 cpuStats.sample(myName); 2295 2296 Vector< sp<EffectChain> > effectChains; 2297 2298 processConfigEvents(); 2299 2300 { // scope for mLock 2301 2302 Mutex::Autolock _l(mLock); 2303 2304 if (logString != NULL) { 2305 mNBLogWriter->logTimestamp(); 2306 mNBLogWriter->log(logString); 2307 logString = NULL; 2308 } 2309 2310 if (mLatchDValid) { 2311 mLatchQ = mLatchD; 2312 mLatchDValid = false; 2313 mLatchQValid = true; 2314 } 2315 2316 if (checkForNewParameters_l()) { 2317 cacheParameters_l(); 2318 } 2319 2320 saveOutputTracks(); 2321 if (mSignalPending) { 2322 // A signal was raised while we were unlocked 2323 mSignalPending = false; 2324 } else if (waitingAsyncCallback_l()) { 2325 if (exitPending()) { 2326 break; 2327 } 2328 releaseWakeLock_l(); 2329 mWakeLockUids.clear(); 2330 mActiveTracksGeneration++; 2331 ALOGV("wait async completion"); 2332 mWaitWorkCV.wait(mLock); 2333 ALOGV("async completion/wake"); 2334 acquireWakeLock_l(); 2335 standbyTime = systemTime() + standbyDelay; 2336 sleepTime = 0; 2337 2338 continue; 2339 } 2340 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2341 isSuspended()) { 2342 // put audio hardware into standby after short delay 2343 if (shouldStandby_l()) { 2344 2345 threadLoop_standby(); 2346 2347 mStandby = true; 2348 } 2349 2350 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2351 // we're about to wait, flush the binder command buffer 2352 IPCThreadState::self()->flushCommands(); 2353 2354 clearOutputTracks(); 2355 2356 if (exitPending()) { 2357 break; 2358 } 2359 2360 releaseWakeLock_l(); 2361 mWakeLockUids.clear(); 2362 mActiveTracksGeneration++; 2363 // wait until we have something to do... 2364 ALOGV("%s going to sleep", myName.string()); 2365 mWaitWorkCV.wait(mLock); 2366 ALOGV("%s waking up", myName.string()); 2367 acquireWakeLock_l(); 2368 2369 mMixerStatus = MIXER_IDLE; 2370 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2371 mBytesWritten = 0; 2372 mBytesRemaining = 0; 2373 checkSilentMode_l(); 2374 2375 standbyTime = systemTime() + standbyDelay; 2376 sleepTime = idleSleepTime; 2377 if (mType == MIXER) { 2378 sleepTimeShift = 0; 2379 } 2380 2381 continue; 2382 } 2383 } 2384 // mMixerStatusIgnoringFastTracks is also updated internally 2385 mMixerStatus = prepareTracks_l(&tracksToRemove); 2386 2387 // compare with previously applied list 2388 if (lastGeneration != mActiveTracksGeneration) { 2389 // update wakelock 2390 updateWakeLockUids_l(mWakeLockUids); 2391 lastGeneration = mActiveTracksGeneration; 2392 } 2393 2394 // prevent any changes in effect chain list and in each effect chain 2395 // during mixing and effect process as the audio buffers could be deleted 2396 // or modified if an effect is created or deleted 2397 lockEffectChains_l(effectChains); 2398 } // mLock scope ends 2399 2400 if (mBytesRemaining == 0) { 2401 mCurrentWriteLength = 0; 2402 if (mMixerStatus == MIXER_TRACKS_READY) { 2403 // threadLoop_mix() sets mCurrentWriteLength 2404 threadLoop_mix(); 2405 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2406 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2407 // threadLoop_sleepTime sets sleepTime to 0 if data 2408 // must be written to HAL 2409 threadLoop_sleepTime(); 2410 if (sleepTime == 0) { 2411 mCurrentWriteLength = mSinkBufferSize; 2412 } 2413 } 2414 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2415 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2416 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2417 // or mSinkBuffer (if there are no effects). 2418 // 2419 // This is done pre-effects computation; if effects change to 2420 // support higher precision, this needs to move. 2421 // 2422 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2423 // TODO use sleepTime == 0 as an additional condition. 2424 if (mMixerBufferValid) { 2425 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2426 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2427 2428 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2429 mNormalFrameCount * mChannelCount); 2430 } 2431 2432 mBytesRemaining = mCurrentWriteLength; 2433 if (isSuspended()) { 2434 sleepTime = suspendSleepTimeUs(); 2435 // simulate write to HAL when suspended 2436 mBytesWritten += mSinkBufferSize; 2437 mBytesRemaining = 0; 2438 } 2439 2440 // only process effects if we're going to write 2441 if (sleepTime == 0 && mType != OFFLOAD) { 2442 for (size_t i = 0; i < effectChains.size(); i ++) { 2443 effectChains[i]->process_l(); 2444 } 2445 } 2446 } 2447 // Process effect chains for offloaded thread even if no audio 2448 // was read from audio track: process only updates effect state 2449 // and thus does have to be synchronized with audio writes but may have 2450 // to be called while waiting for async write callback 2451 if (mType == OFFLOAD) { 2452 for (size_t i = 0; i < effectChains.size(); i ++) { 2453 effectChains[i]->process_l(); 2454 } 2455 } 2456 2457 // Only if the Effects buffer is enabled and there is data in the 2458 // Effects buffer (buffer valid), we need to 2459 // copy into the sink buffer. 2460 // TODO use sleepTime == 0 as an additional condition. 2461 if (mEffectBufferValid) { 2462 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2463 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2464 mNormalFrameCount * mChannelCount); 2465 } 2466 2467 // enable changes in effect chain 2468 unlockEffectChains(effectChains); 2469 2470 if (!waitingAsyncCallback()) { 2471 // sleepTime == 0 means we must write to audio hardware 2472 if (sleepTime == 0) { 2473 if (mBytesRemaining) { 2474 ssize_t ret = threadLoop_write(); 2475 if (ret < 0) { 2476 mBytesRemaining = 0; 2477 } else { 2478 mBytesWritten += ret; 2479 mBytesRemaining -= ret; 2480 } 2481 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2482 (mMixerStatus == MIXER_DRAIN_ALL)) { 2483 threadLoop_drain(); 2484 } 2485 if (mType == MIXER) { 2486 // write blocked detection 2487 nsecs_t now = systemTime(); 2488 nsecs_t delta = now - mLastWriteTime; 2489 if (!mStandby && delta > maxPeriod) { 2490 mNumDelayedWrites++; 2491 if ((now - lastWarning) > kWarningThrottleNs) { 2492 ATRACE_NAME("underrun"); 2493 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2494 ns2ms(delta), mNumDelayedWrites, this); 2495 lastWarning = now; 2496 } 2497 } 2498 } 2499 2500 } else { 2501 usleep(sleepTime); 2502 } 2503 } 2504 2505 // Finally let go of removed track(s), without the lock held 2506 // since we can't guarantee the destructors won't acquire that 2507 // same lock. This will also mutate and push a new fast mixer state. 2508 threadLoop_removeTracks(tracksToRemove); 2509 tracksToRemove.clear(); 2510 2511 // FIXME I don't understand the need for this here; 2512 // it was in the original code but maybe the 2513 // assignment in saveOutputTracks() makes this unnecessary? 2514 clearOutputTracks(); 2515 2516 // Effect chains will be actually deleted here if they were removed from 2517 // mEffectChains list during mixing or effects processing 2518 effectChains.clear(); 2519 2520 // FIXME Note that the above .clear() is no longer necessary since effectChains 2521 // is now local to this block, but will keep it for now (at least until merge done). 2522 } 2523 2524 threadLoop_exit(); 2525 2526 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2527 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2528 // put output stream into standby mode 2529 if (!mStandby) { 2530 mOutput->stream->common.standby(&mOutput->stream->common); 2531 } 2532 } 2533 2534 releaseWakeLock(); 2535 mWakeLockUids.clear(); 2536 mActiveTracksGeneration++; 2537 2538 ALOGV("Thread %p type %d exiting", this, mType); 2539 return false; 2540} 2541 2542// removeTracks_l() must be called with ThreadBase::mLock held 2543void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2544{ 2545 size_t count = tracksToRemove.size(); 2546 if (count > 0) { 2547 for (size_t i=0 ; i<count ; i++) { 2548 const sp<Track>& track = tracksToRemove.itemAt(i); 2549 mActiveTracks.remove(track); 2550 mWakeLockUids.remove(track->uid()); 2551 mActiveTracksGeneration++; 2552 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2553 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2554 if (chain != 0) { 2555 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2556 track->sessionId()); 2557 chain->decActiveTrackCnt(); 2558 } 2559 if (track->isTerminated()) { 2560 removeTrack_l(track); 2561 } 2562 } 2563 } 2564 2565} 2566 2567status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2568{ 2569 if (mNormalSink != 0) { 2570 return mNormalSink->getTimestamp(timestamp); 2571 } 2572 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2573 uint64_t position64; 2574 int ret = mOutput->stream->get_presentation_position( 2575 mOutput->stream, &position64, ×tamp.mTime); 2576 if (ret == 0) { 2577 timestamp.mPosition = (uint32_t)position64; 2578 return NO_ERROR; 2579 } 2580 } 2581 return INVALID_OPERATION; 2582} 2583// ---------------------------------------------------------------------------- 2584 2585AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2586 audio_io_handle_t id, audio_devices_t device, type_t type) 2587 : PlaybackThread(audioFlinger, output, id, device, type), 2588 // mAudioMixer below 2589 // mFastMixer below 2590 mFastMixerFutex(0) 2591 // mOutputSink below 2592 // mPipeSink below 2593 // mNormalSink below 2594{ 2595 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2596 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2597 "mFrameCount=%d, mNormalFrameCount=%d", 2598 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2599 mNormalFrameCount); 2600 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2601 2602 // FIXME - Current mixer implementation only supports stereo output 2603 if (mChannelCount != FCC_2) { 2604 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2605 } 2606 2607 // create an NBAIO sink for the HAL output stream, and negotiate 2608 mOutputSink = new AudioStreamOutSink(output->stream); 2609 size_t numCounterOffers = 0; 2610 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2611 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2612 ALOG_ASSERT(index == 0); 2613 2614 // initialize fast mixer depending on configuration 2615 bool initFastMixer; 2616 switch (kUseFastMixer) { 2617 case FastMixer_Never: 2618 initFastMixer = false; 2619 break; 2620 case FastMixer_Always: 2621 initFastMixer = true; 2622 break; 2623 case FastMixer_Static: 2624 case FastMixer_Dynamic: 2625 initFastMixer = mFrameCount < mNormalFrameCount; 2626 break; 2627 } 2628 if (initFastMixer) { 2629 2630 // create a MonoPipe to connect our submix to FastMixer 2631 NBAIO_Format format = mOutputSink->format(); 2632 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2633 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2634 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2635 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2636 const NBAIO_Format offers[1] = {format}; 2637 size_t numCounterOffers = 0; 2638 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2639 ALOG_ASSERT(index == 0); 2640 monoPipe->setAvgFrames((mScreenState & 1) ? 2641 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2642 mPipeSink = monoPipe; 2643 2644#ifdef TEE_SINK 2645 if (mTeeSinkOutputEnabled) { 2646 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2647 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2648 numCounterOffers = 0; 2649 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2650 ALOG_ASSERT(index == 0); 2651 mTeeSink = teeSink; 2652 PipeReader *teeSource = new PipeReader(*teeSink); 2653 numCounterOffers = 0; 2654 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2655 ALOG_ASSERT(index == 0); 2656 mTeeSource = teeSource; 2657 } 2658#endif 2659 2660 // create fast mixer and configure it initially with just one fast track for our submix 2661 mFastMixer = new FastMixer(); 2662 FastMixerStateQueue *sq = mFastMixer->sq(); 2663#ifdef STATE_QUEUE_DUMP 2664 sq->setObserverDump(&mStateQueueObserverDump); 2665 sq->setMutatorDump(&mStateQueueMutatorDump); 2666#endif 2667 FastMixerState *state = sq->begin(); 2668 FastTrack *fastTrack = &state->mFastTracks[0]; 2669 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2670 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2671 fastTrack->mVolumeProvider = NULL; 2672 fastTrack->mGeneration++; 2673 state->mFastTracksGen++; 2674 state->mTrackMask = 1; 2675 // fast mixer will use the HAL output sink 2676 state->mOutputSink = mOutputSink.get(); 2677 state->mOutputSinkGen++; 2678 state->mFrameCount = mFrameCount; 2679 state->mCommand = FastMixerState::COLD_IDLE; 2680 // already done in constructor initialization list 2681 //mFastMixerFutex = 0; 2682 state->mColdFutexAddr = &mFastMixerFutex; 2683 state->mColdGen++; 2684 state->mDumpState = &mFastMixerDumpState; 2685#ifdef TEE_SINK 2686 state->mTeeSink = mTeeSink.get(); 2687#endif 2688 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2689 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2690 sq->end(); 2691 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2692 2693 // start the fast mixer 2694 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2695 pid_t tid = mFastMixer->getTid(); 2696 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2697 if (err != 0) { 2698 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2699 kPriorityFastMixer, getpid_cached, tid, err); 2700 } 2701 2702#ifdef AUDIO_WATCHDOG 2703 // create and start the watchdog 2704 mAudioWatchdog = new AudioWatchdog(); 2705 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2706 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2707 tid = mAudioWatchdog->getTid(); 2708 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2709 if (err != 0) { 2710 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2711 kPriorityFastMixer, getpid_cached, tid, err); 2712 } 2713#endif 2714 2715 } else { 2716 mFastMixer = NULL; 2717 } 2718 2719 switch (kUseFastMixer) { 2720 case FastMixer_Never: 2721 case FastMixer_Dynamic: 2722 mNormalSink = mOutputSink; 2723 break; 2724 case FastMixer_Always: 2725 mNormalSink = mPipeSink; 2726 break; 2727 case FastMixer_Static: 2728 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2729 break; 2730 } 2731} 2732 2733AudioFlinger::MixerThread::~MixerThread() 2734{ 2735 if (mFastMixer != NULL) { 2736 FastMixerStateQueue *sq = mFastMixer->sq(); 2737 FastMixerState *state = sq->begin(); 2738 if (state->mCommand == FastMixerState::COLD_IDLE) { 2739 int32_t old = android_atomic_inc(&mFastMixerFutex); 2740 if (old == -1) { 2741 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2742 } 2743 } 2744 state->mCommand = FastMixerState::EXIT; 2745 sq->end(); 2746 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2747 mFastMixer->join(); 2748 // Though the fast mixer thread has exited, it's state queue is still valid. 2749 // We'll use that extract the final state which contains one remaining fast track 2750 // corresponding to our sub-mix. 2751 state = sq->begin(); 2752 ALOG_ASSERT(state->mTrackMask == 1); 2753 FastTrack *fastTrack = &state->mFastTracks[0]; 2754 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2755 delete fastTrack->mBufferProvider; 2756 sq->end(false /*didModify*/); 2757 delete mFastMixer; 2758#ifdef AUDIO_WATCHDOG 2759 if (mAudioWatchdog != 0) { 2760 mAudioWatchdog->requestExit(); 2761 mAudioWatchdog->requestExitAndWait(); 2762 mAudioWatchdog.clear(); 2763 } 2764#endif 2765 } 2766 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2767 delete mAudioMixer; 2768} 2769 2770 2771uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2772{ 2773 if (mFastMixer != NULL) { 2774 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2775 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2776 } 2777 return latency; 2778} 2779 2780 2781void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2782{ 2783 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2784} 2785 2786ssize_t AudioFlinger::MixerThread::threadLoop_write() 2787{ 2788 // FIXME we should only do one push per cycle; confirm this is true 2789 // Start the fast mixer if it's not already running 2790 if (mFastMixer != NULL) { 2791 FastMixerStateQueue *sq = mFastMixer->sq(); 2792 FastMixerState *state = sq->begin(); 2793 if (state->mCommand != FastMixerState::MIX_WRITE && 2794 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2795 if (state->mCommand == FastMixerState::COLD_IDLE) { 2796 int32_t old = android_atomic_inc(&mFastMixerFutex); 2797 if (old == -1) { 2798 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2799 } 2800#ifdef AUDIO_WATCHDOG 2801 if (mAudioWatchdog != 0) { 2802 mAudioWatchdog->resume(); 2803 } 2804#endif 2805 } 2806 state->mCommand = FastMixerState::MIX_WRITE; 2807 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2808 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2809 sq->end(); 2810 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2811 if (kUseFastMixer == FastMixer_Dynamic) { 2812 mNormalSink = mPipeSink; 2813 } 2814 } else { 2815 sq->end(false /*didModify*/); 2816 } 2817 } 2818 return PlaybackThread::threadLoop_write(); 2819} 2820 2821void AudioFlinger::MixerThread::threadLoop_standby() 2822{ 2823 // Idle the fast mixer if it's currently running 2824 if (mFastMixer != NULL) { 2825 FastMixerStateQueue *sq = mFastMixer->sq(); 2826 FastMixerState *state = sq->begin(); 2827 if (!(state->mCommand & FastMixerState::IDLE)) { 2828 state->mCommand = FastMixerState::COLD_IDLE; 2829 state->mColdFutexAddr = &mFastMixerFutex; 2830 state->mColdGen++; 2831 mFastMixerFutex = 0; 2832 sq->end(); 2833 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2834 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2835 if (kUseFastMixer == FastMixer_Dynamic) { 2836 mNormalSink = mOutputSink; 2837 } 2838#ifdef AUDIO_WATCHDOG 2839 if (mAudioWatchdog != 0) { 2840 mAudioWatchdog->pause(); 2841 } 2842#endif 2843 } else { 2844 sq->end(false /*didModify*/); 2845 } 2846 } 2847 PlaybackThread::threadLoop_standby(); 2848} 2849 2850bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2851{ 2852 return false; 2853} 2854 2855bool AudioFlinger::PlaybackThread::shouldStandby_l() 2856{ 2857 return !mStandby; 2858} 2859 2860bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2861{ 2862 Mutex::Autolock _l(mLock); 2863 return waitingAsyncCallback_l(); 2864} 2865 2866// shared by MIXER and DIRECT, overridden by DUPLICATING 2867void AudioFlinger::PlaybackThread::threadLoop_standby() 2868{ 2869 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2870 mOutput->stream->common.standby(&mOutput->stream->common); 2871 if (mUseAsyncWrite != 0) { 2872 // discard any pending drain or write ack by incrementing sequence 2873 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2874 mDrainSequence = (mDrainSequence + 2) & ~1; 2875 ALOG_ASSERT(mCallbackThread != 0); 2876 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2877 mCallbackThread->setDraining(mDrainSequence); 2878 } 2879} 2880 2881void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2882{ 2883 ALOGV("signal playback thread"); 2884 broadcast_l(); 2885} 2886 2887void AudioFlinger::MixerThread::threadLoop_mix() 2888{ 2889 // obtain the presentation timestamp of the next output buffer 2890 int64_t pts; 2891 status_t status = INVALID_OPERATION; 2892 2893 if (mNormalSink != 0) { 2894 status = mNormalSink->getNextWriteTimestamp(&pts); 2895 } else { 2896 status = mOutputSink->getNextWriteTimestamp(&pts); 2897 } 2898 2899 if (status != NO_ERROR) { 2900 pts = AudioBufferProvider::kInvalidPTS; 2901 } 2902 2903 // mix buffers... 2904 mAudioMixer->process(pts); 2905 mCurrentWriteLength = mSinkBufferSize; 2906 // increase sleep time progressively when application underrun condition clears. 2907 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2908 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2909 // such that we would underrun the audio HAL. 2910 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2911 sleepTimeShift--; 2912 } 2913 sleepTime = 0; 2914 standbyTime = systemTime() + standbyDelay; 2915 //TODO: delay standby when effects have a tail 2916} 2917 2918void AudioFlinger::MixerThread::threadLoop_sleepTime() 2919{ 2920 // If no tracks are ready, sleep once for the duration of an output 2921 // buffer size, then write 0s to the output 2922 if (sleepTime == 0) { 2923 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2924 sleepTime = activeSleepTime >> sleepTimeShift; 2925 if (sleepTime < kMinThreadSleepTimeUs) { 2926 sleepTime = kMinThreadSleepTimeUs; 2927 } 2928 // reduce sleep time in case of consecutive application underruns to avoid 2929 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2930 // duration we would end up writing less data than needed by the audio HAL if 2931 // the condition persists. 2932 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2933 sleepTimeShift++; 2934 } 2935 } else { 2936 sleepTime = idleSleepTime; 2937 } 2938 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2939 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2940 // before effects processing or output. 2941 if (mMixerBufferValid) { 2942 memset(mMixerBuffer, 0, mMixerBufferSize); 2943 } else { 2944 memset(mSinkBuffer, 0, mSinkBufferSize); 2945 } 2946 sleepTime = 0; 2947 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2948 "anticipated start"); 2949 } 2950 // TODO add standby time extension fct of effect tail 2951} 2952 2953// prepareTracks_l() must be called with ThreadBase::mLock held 2954AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2955 Vector< sp<Track> > *tracksToRemove) 2956{ 2957 2958 mixer_state mixerStatus = MIXER_IDLE; 2959 // find out which tracks need to be processed 2960 size_t count = mActiveTracks.size(); 2961 size_t mixedTracks = 0; 2962 size_t tracksWithEffect = 0; 2963 // counts only _active_ fast tracks 2964 size_t fastTracks = 0; 2965 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2966 2967 float masterVolume = mMasterVolume; 2968 bool masterMute = mMasterMute; 2969 2970 if (masterMute) { 2971 masterVolume = 0; 2972 } 2973 // Delegate master volume control to effect in output mix effect chain if needed 2974 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2975 if (chain != 0) { 2976 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2977 chain->setVolume_l(&v, &v); 2978 masterVolume = (float)((v + (1 << 23)) >> 24); 2979 chain.clear(); 2980 } 2981 2982 // prepare a new state to push 2983 FastMixerStateQueue *sq = NULL; 2984 FastMixerState *state = NULL; 2985 bool didModify = false; 2986 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2987 if (mFastMixer != NULL) { 2988 sq = mFastMixer->sq(); 2989 state = sq->begin(); 2990 } 2991 2992 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 2993 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 2994 2995 for (size_t i=0 ; i<count ; i++) { 2996 const sp<Track> t = mActiveTracks[i].promote(); 2997 if (t == 0) { 2998 continue; 2999 } 3000 3001 // this const just means the local variable doesn't change 3002 Track* const track = t.get(); 3003 3004 // process fast tracks 3005 if (track->isFastTrack()) { 3006 3007 // It's theoretically possible (though unlikely) for a fast track to be created 3008 // and then removed within the same normal mix cycle. This is not a problem, as 3009 // the track never becomes active so it's fast mixer slot is never touched. 3010 // The converse, of removing an (active) track and then creating a new track 3011 // at the identical fast mixer slot within the same normal mix cycle, 3012 // is impossible because the slot isn't marked available until the end of each cycle. 3013 int j = track->mFastIndex; 3014 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3015 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3016 FastTrack *fastTrack = &state->mFastTracks[j]; 3017 3018 // Determine whether the track is currently in underrun condition, 3019 // and whether it had a recent underrun. 3020 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3021 FastTrackUnderruns underruns = ftDump->mUnderruns; 3022 uint32_t recentFull = (underruns.mBitFields.mFull - 3023 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3024 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3025 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3026 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3027 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3028 uint32_t recentUnderruns = recentPartial + recentEmpty; 3029 track->mObservedUnderruns = underruns; 3030 // don't count underruns that occur while stopping or pausing 3031 // or stopped which can occur when flush() is called while active 3032 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3033 recentUnderruns > 0) { 3034 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3035 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3036 } 3037 3038 // This is similar to the state machine for normal tracks, 3039 // with a few modifications for fast tracks. 3040 bool isActive = true; 3041 switch (track->mState) { 3042 case TrackBase::STOPPING_1: 3043 // track stays active in STOPPING_1 state until first underrun 3044 if (recentUnderruns > 0 || track->isTerminated()) { 3045 track->mState = TrackBase::STOPPING_2; 3046 } 3047 break; 3048 case TrackBase::PAUSING: 3049 // ramp down is not yet implemented 3050 track->setPaused(); 3051 break; 3052 case TrackBase::RESUMING: 3053 // ramp up is not yet implemented 3054 track->mState = TrackBase::ACTIVE; 3055 break; 3056 case TrackBase::ACTIVE: 3057 if (recentFull > 0 || recentPartial > 0) { 3058 // track has provided at least some frames recently: reset retry count 3059 track->mRetryCount = kMaxTrackRetries; 3060 } 3061 if (recentUnderruns == 0) { 3062 // no recent underruns: stay active 3063 break; 3064 } 3065 // there has recently been an underrun of some kind 3066 if (track->sharedBuffer() == 0) { 3067 // were any of the recent underruns "empty" (no frames available)? 3068 if (recentEmpty == 0) { 3069 // no, then ignore the partial underruns as they are allowed indefinitely 3070 break; 3071 } 3072 // there has recently been an "empty" underrun: decrement the retry counter 3073 if (--(track->mRetryCount) > 0) { 3074 break; 3075 } 3076 // indicate to client process that the track was disabled because of underrun; 3077 // it will then automatically call start() when data is available 3078 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3079 // remove from active list, but state remains ACTIVE [confusing but true] 3080 isActive = false; 3081 break; 3082 } 3083 // fall through 3084 case TrackBase::STOPPING_2: 3085 case TrackBase::PAUSED: 3086 case TrackBase::STOPPED: 3087 case TrackBase::FLUSHED: // flush() while active 3088 // Check for presentation complete if track is inactive 3089 // We have consumed all the buffers of this track. 3090 // This would be incomplete if we auto-paused on underrun 3091 { 3092 size_t audioHALFrames = 3093 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3094 size_t framesWritten = mBytesWritten / mFrameSize; 3095 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3096 // track stays in active list until presentation is complete 3097 break; 3098 } 3099 } 3100 if (track->isStopping_2()) { 3101 track->mState = TrackBase::STOPPED; 3102 } 3103 if (track->isStopped()) { 3104 // Can't reset directly, as fast mixer is still polling this track 3105 // track->reset(); 3106 // So instead mark this track as needing to be reset after push with ack 3107 resetMask |= 1 << i; 3108 } 3109 isActive = false; 3110 break; 3111 case TrackBase::IDLE: 3112 default: 3113 LOG_FATAL("unexpected track state %d", track->mState); 3114 } 3115 3116 if (isActive) { 3117 // was it previously inactive? 3118 if (!(state->mTrackMask & (1 << j))) { 3119 ExtendedAudioBufferProvider *eabp = track; 3120 VolumeProvider *vp = track; 3121 fastTrack->mBufferProvider = eabp; 3122 fastTrack->mVolumeProvider = vp; 3123 fastTrack->mChannelMask = track->mChannelMask; 3124 fastTrack->mGeneration++; 3125 state->mTrackMask |= 1 << j; 3126 didModify = true; 3127 // no acknowledgement required for newly active tracks 3128 } 3129 // cache the combined master volume and stream type volume for fast mixer; this 3130 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3131 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3132 ++fastTracks; 3133 } else { 3134 // was it previously active? 3135 if (state->mTrackMask & (1 << j)) { 3136 fastTrack->mBufferProvider = NULL; 3137 fastTrack->mGeneration++; 3138 state->mTrackMask &= ~(1 << j); 3139 didModify = true; 3140 // If any fast tracks were removed, we must wait for acknowledgement 3141 // because we're about to decrement the last sp<> on those tracks. 3142 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3143 } else { 3144 LOG_FATAL("fast track %d should have been active", j); 3145 } 3146 tracksToRemove->add(track); 3147 // Avoids a misleading display in dumpsys 3148 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3149 } 3150 continue; 3151 } 3152 3153 { // local variable scope to avoid goto warning 3154 3155 audio_track_cblk_t* cblk = track->cblk(); 3156 3157 // The first time a track is added we wait 3158 // for all its buffers to be filled before processing it 3159 int name = track->name(); 3160 // make sure that we have enough frames to mix one full buffer. 3161 // enforce this condition only once to enable draining the buffer in case the client 3162 // app does not call stop() and relies on underrun to stop: 3163 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3164 // during last round 3165 size_t desiredFrames; 3166 uint32_t sr = track->sampleRate(); 3167 if (sr == mSampleRate) { 3168 desiredFrames = mNormalFrameCount; 3169 } else { 3170 // +1 for rounding and +1 for additional sample needed for interpolation 3171 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3172 // add frames already consumed but not yet released by the resampler 3173 // because mAudioTrackServerProxy->framesReady() will include these frames 3174 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3175#if 0 3176 // the minimum track buffer size is normally twice the number of frames necessary 3177 // to fill one buffer and the resampler should not leave more than one buffer worth 3178 // of unreleased frames after each pass, but just in case... 3179 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3180#endif 3181 } 3182 uint32_t minFrames = 1; 3183 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3184 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3185 minFrames = desiredFrames; 3186 } 3187 3188 size_t framesReady = track->framesReady(); 3189 if ((framesReady >= minFrames) && track->isReady() && 3190 !track->isPaused() && !track->isTerminated()) 3191 { 3192 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3193 3194 mixedTracks++; 3195 3196 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3197 // there is an effect chain connected to the track 3198 chain.clear(); 3199 if (track->mainBuffer() != mSinkBuffer && 3200 track->mainBuffer() != mMixerBuffer) { 3201 if (mEffectBufferEnabled) { 3202 mEffectBufferValid = true; // Later can set directly. 3203 } 3204 chain = getEffectChain_l(track->sessionId()); 3205 // Delegate volume control to effect in track effect chain if needed 3206 if (chain != 0) { 3207 tracksWithEffect++; 3208 } else { 3209 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3210 "session %d", 3211 name, track->sessionId()); 3212 } 3213 } 3214 3215 3216 int param = AudioMixer::VOLUME; 3217 if (track->mFillingUpStatus == Track::FS_FILLED) { 3218 // no ramp for the first volume setting 3219 track->mFillingUpStatus = Track::FS_ACTIVE; 3220 if (track->mState == TrackBase::RESUMING) { 3221 track->mState = TrackBase::ACTIVE; 3222 param = AudioMixer::RAMP_VOLUME; 3223 } 3224 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3225 // FIXME should not make a decision based on mServer 3226 } else if (cblk->mServer != 0) { 3227 // If the track is stopped before the first frame was mixed, 3228 // do not apply ramp 3229 param = AudioMixer::RAMP_VOLUME; 3230 } 3231 3232 // compute volume for this track 3233 uint32_t vl, vr, va; 3234 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3235 vl = vr = va = 0; 3236 if (track->isPausing()) { 3237 track->setPaused(); 3238 } 3239 } else { 3240 3241 // read original volumes with volume control 3242 float typeVolume = mStreamTypes[track->streamType()].volume; 3243 float v = masterVolume * typeVolume; 3244 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3245 uint32_t vlr = proxy->getVolumeLR(); 3246 vl = vlr & 0xFFFF; 3247 vr = vlr >> 16; 3248 // track volumes come from shared memory, so can't be trusted and must be clamped 3249 if (vl > MAX_GAIN_INT) { 3250 ALOGV("Track left volume out of range: %04X", vl); 3251 vl = MAX_GAIN_INT; 3252 } 3253 if (vr > MAX_GAIN_INT) { 3254 ALOGV("Track right volume out of range: %04X", vr); 3255 vr = MAX_GAIN_INT; 3256 } 3257 // now apply the master volume and stream type volume 3258 vl = (uint32_t)(v * vl) << 12; 3259 vr = (uint32_t)(v * vr) << 12; 3260 // assuming master volume and stream type volume each go up to 1.0, 3261 // vl and vr are now in 8.24 format 3262 3263 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3264 // send level comes from shared memory and so may be corrupt 3265 if (sendLevel > MAX_GAIN_INT) { 3266 ALOGV("Track send level out of range: %04X", sendLevel); 3267 sendLevel = MAX_GAIN_INT; 3268 } 3269 va = (uint32_t)(v * sendLevel); 3270 } 3271 3272 // Delegate volume control to effect in track effect chain if needed 3273 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3274 // Do not ramp volume if volume is controlled by effect 3275 param = AudioMixer::VOLUME; 3276 track->mHasVolumeController = true; 3277 } else { 3278 // force no volume ramp when volume controller was just disabled or removed 3279 // from effect chain to avoid volume spike 3280 if (track->mHasVolumeController) { 3281 param = AudioMixer::VOLUME; 3282 } 3283 track->mHasVolumeController = false; 3284 } 3285 3286 // Convert volumes from 8.24 to 4.12 format 3287 // This additional clamping is needed in case chain->setVolume_l() overshot 3288 vl = (vl + (1 << 11)) >> 12; 3289 if (vl > MAX_GAIN_INT) { 3290 vl = MAX_GAIN_INT; 3291 } 3292 vr = (vr + (1 << 11)) >> 12; 3293 if (vr > MAX_GAIN_INT) { 3294 vr = MAX_GAIN_INT; 3295 } 3296 3297 if (va > MAX_GAIN_INT) { 3298 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3299 } 3300 3301 // XXX: these things DON'T need to be done each time 3302 mAudioMixer->setBufferProvider(name, track); 3303 mAudioMixer->enable(name); 3304 3305 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3306 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3307 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3308 mAudioMixer->setParameter( 3309 name, 3310 AudioMixer::TRACK, 3311 AudioMixer::FORMAT, (void *)track->format()); 3312 mAudioMixer->setParameter( 3313 name, 3314 AudioMixer::TRACK, 3315 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3316 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3317 uint32_t maxSampleRate = mSampleRate * 2; 3318 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3319 if (reqSampleRate == 0) { 3320 reqSampleRate = mSampleRate; 3321 } else if (reqSampleRate > maxSampleRate) { 3322 reqSampleRate = maxSampleRate; 3323 } 3324 mAudioMixer->setParameter( 3325 name, 3326 AudioMixer::RESAMPLE, 3327 AudioMixer::SAMPLE_RATE, 3328 (void *)(uintptr_t)reqSampleRate); 3329 /* 3330 * Select the appropriate output buffer for the track. 3331 * 3332 * Tracks with effects go into their own effects chain buffer 3333 * and from there into either mEffectBuffer or mSinkBuffer. 3334 * 3335 * Other tracks can use mMixerBuffer for higher precision 3336 * channel accumulation. If this buffer is enabled 3337 * (mMixerBufferEnabled true), then selected tracks will accumulate 3338 * into it. 3339 * 3340 */ 3341 if (mMixerBufferEnabled 3342 && (track->mainBuffer() == mSinkBuffer 3343 || track->mainBuffer() == mMixerBuffer)) { 3344 mAudioMixer->setParameter( 3345 name, 3346 AudioMixer::TRACK, 3347 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3348 mAudioMixer->setParameter( 3349 name, 3350 AudioMixer::TRACK, 3351 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3352 // TODO: override track->mainBuffer()? 3353 mMixerBufferValid = true; 3354 } else { 3355 mAudioMixer->setParameter( 3356 name, 3357 AudioMixer::TRACK, 3358 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3359 mAudioMixer->setParameter( 3360 name, 3361 AudioMixer::TRACK, 3362 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3363 } 3364 mAudioMixer->setParameter( 3365 name, 3366 AudioMixer::TRACK, 3367 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3368 3369 // reset retry count 3370 track->mRetryCount = kMaxTrackRetries; 3371 3372 // If one track is ready, set the mixer ready if: 3373 // - the mixer was not ready during previous round OR 3374 // - no other track is not ready 3375 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3376 mixerStatus != MIXER_TRACKS_ENABLED) { 3377 mixerStatus = MIXER_TRACKS_READY; 3378 } 3379 } else { 3380 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3381 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3382 } 3383 // clear effect chain input buffer if an active track underruns to avoid sending 3384 // previous audio buffer again to effects 3385 chain = getEffectChain_l(track->sessionId()); 3386 if (chain != 0) { 3387 chain->clearInputBuffer(); 3388 } 3389 3390 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3391 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3392 track->isStopped() || track->isPaused()) { 3393 // We have consumed all the buffers of this track. 3394 // Remove it from the list of active tracks. 3395 // TODO: use actual buffer filling status instead of latency when available from 3396 // audio HAL 3397 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3398 size_t framesWritten = mBytesWritten / mFrameSize; 3399 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3400 if (track->isStopped()) { 3401 track->reset(); 3402 } 3403 tracksToRemove->add(track); 3404 } 3405 } else { 3406 // No buffers for this track. Give it a few chances to 3407 // fill a buffer, then remove it from active list. 3408 if (--(track->mRetryCount) <= 0) { 3409 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3410 tracksToRemove->add(track); 3411 // indicate to client process that the track was disabled because of underrun; 3412 // it will then automatically call start() when data is available 3413 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3414 // If one track is not ready, mark the mixer also not ready if: 3415 // - the mixer was ready during previous round OR 3416 // - no other track is ready 3417 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3418 mixerStatus != MIXER_TRACKS_READY) { 3419 mixerStatus = MIXER_TRACKS_ENABLED; 3420 } 3421 } 3422 mAudioMixer->disable(name); 3423 } 3424 3425 } // local variable scope to avoid goto warning 3426track_is_ready: ; 3427 3428 } 3429 3430 // Push the new FastMixer state if necessary 3431 bool pauseAudioWatchdog = false; 3432 if (didModify) { 3433 state->mFastTracksGen++; 3434 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3435 if (kUseFastMixer == FastMixer_Dynamic && 3436 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3437 state->mCommand = FastMixerState::COLD_IDLE; 3438 state->mColdFutexAddr = &mFastMixerFutex; 3439 state->mColdGen++; 3440 mFastMixerFutex = 0; 3441 if (kUseFastMixer == FastMixer_Dynamic) { 3442 mNormalSink = mOutputSink; 3443 } 3444 // If we go into cold idle, need to wait for acknowledgement 3445 // so that fast mixer stops doing I/O. 3446 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3447 pauseAudioWatchdog = true; 3448 } 3449 } 3450 if (sq != NULL) { 3451 sq->end(didModify); 3452 sq->push(block); 3453 } 3454#ifdef AUDIO_WATCHDOG 3455 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3456 mAudioWatchdog->pause(); 3457 } 3458#endif 3459 3460 // Now perform the deferred reset on fast tracks that have stopped 3461 while (resetMask != 0) { 3462 size_t i = __builtin_ctz(resetMask); 3463 ALOG_ASSERT(i < count); 3464 resetMask &= ~(1 << i); 3465 sp<Track> t = mActiveTracks[i].promote(); 3466 if (t == 0) { 3467 continue; 3468 } 3469 Track* track = t.get(); 3470 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3471 track->reset(); 3472 } 3473 3474 // remove all the tracks that need to be... 3475 removeTracks_l(*tracksToRemove); 3476 3477 // sink or mix buffer must be cleared if all tracks are connected to an 3478 // effect chain as in this case the mixer will not write to the sink or mix buffer 3479 // and track effects will accumulate into it 3480 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3481 (mixedTracks == 0 && fastTracks > 0))) { 3482 // FIXME as a performance optimization, should remember previous zero status 3483 if (mMixerBufferValid) { 3484 memset(mMixerBuffer, 0, mMixerBufferSize); 3485 // TODO: In testing, mSinkBuffer below need not be cleared because 3486 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3487 // after mixing. 3488 // 3489 // To enforce this guarantee: 3490 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3491 // (mixedTracks == 0 && fastTracks > 0)) 3492 // must imply MIXER_TRACKS_READY. 3493 // Later, we may clear buffers regardless, and skip much of this logic. 3494 } 3495 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3496 if (mEffectBufferValid) { 3497 memset(mEffectBuffer, 0, mEffectBufferSize); 3498 } 3499 // FIXME as a performance optimization, should remember previous zero status 3500 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3501 } 3502 3503 // if any fast tracks, then status is ready 3504 mMixerStatusIgnoringFastTracks = mixerStatus; 3505 if (fastTracks > 0) { 3506 mixerStatus = MIXER_TRACKS_READY; 3507 } 3508 return mixerStatus; 3509} 3510 3511// getTrackName_l() must be called with ThreadBase::mLock held 3512int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3513{ 3514 return mAudioMixer->getTrackName(channelMask, sessionId); 3515} 3516 3517// deleteTrackName_l() must be called with ThreadBase::mLock held 3518void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3519{ 3520 ALOGV("remove track (%d) and delete from mixer", name); 3521 mAudioMixer->deleteTrackName(name); 3522} 3523 3524// checkForNewParameters_l() must be called with ThreadBase::mLock held 3525bool AudioFlinger::MixerThread::checkForNewParameters_l() 3526{ 3527 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3528 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3529 bool reconfig = false; 3530 3531 while (!mNewParameters.isEmpty()) { 3532 3533 if (mFastMixer != NULL) { 3534 FastMixerStateQueue *sq = mFastMixer->sq(); 3535 FastMixerState *state = sq->begin(); 3536 if (!(state->mCommand & FastMixerState::IDLE)) { 3537 previousCommand = state->mCommand; 3538 state->mCommand = FastMixerState::HOT_IDLE; 3539 sq->end(); 3540 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3541 } else { 3542 sq->end(false /*didModify*/); 3543 } 3544 } 3545 3546 status_t status = NO_ERROR; 3547 String8 keyValuePair = mNewParameters[0]; 3548 AudioParameter param = AudioParameter(keyValuePair); 3549 int value; 3550 3551 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3552 reconfig = true; 3553 } 3554 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3555 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3556 status = BAD_VALUE; 3557 } else { 3558 // no need to save value, since it's constant 3559 reconfig = true; 3560 } 3561 } 3562 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3563 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3564 status = BAD_VALUE; 3565 } else { 3566 // no need to save value, since it's constant 3567 reconfig = true; 3568 } 3569 } 3570 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3571 // do not accept frame count changes if tracks are open as the track buffer 3572 // size depends on frame count and correct behavior would not be guaranteed 3573 // if frame count is changed after track creation 3574 if (!mTracks.isEmpty()) { 3575 status = INVALID_OPERATION; 3576 } else { 3577 reconfig = true; 3578 } 3579 } 3580 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3581#ifdef ADD_BATTERY_DATA 3582 // when changing the audio output device, call addBatteryData to notify 3583 // the change 3584 if (mOutDevice != value) { 3585 uint32_t params = 0; 3586 // check whether speaker is on 3587 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3588 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3589 } 3590 3591 audio_devices_t deviceWithoutSpeaker 3592 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3593 // check if any other device (except speaker) is on 3594 if (value & deviceWithoutSpeaker ) { 3595 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3596 } 3597 3598 if (params != 0) { 3599 addBatteryData(params); 3600 } 3601 } 3602#endif 3603 3604 // forward device change to effects that have requested to be 3605 // aware of attached audio device. 3606 if (value != AUDIO_DEVICE_NONE) { 3607 mOutDevice = value; 3608 for (size_t i = 0; i < mEffectChains.size(); i++) { 3609 mEffectChains[i]->setDevice_l(mOutDevice); 3610 } 3611 } 3612 } 3613 3614 if (status == NO_ERROR) { 3615 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3616 keyValuePair.string()); 3617 if (!mStandby && status == INVALID_OPERATION) { 3618 mOutput->stream->common.standby(&mOutput->stream->common); 3619 mStandby = true; 3620 mBytesWritten = 0; 3621 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3622 keyValuePair.string()); 3623 } 3624 if (status == NO_ERROR && reconfig) { 3625 readOutputParameters_l(); 3626 delete mAudioMixer; 3627 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3628 for (size_t i = 0; i < mTracks.size() ; i++) { 3629 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3630 if (name < 0) { 3631 break; 3632 } 3633 mTracks[i]->mName = name; 3634 } 3635 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3636 } 3637 } 3638 3639 mNewParameters.removeAt(0); 3640 3641 mParamStatus = status; 3642 mParamCond.signal(); 3643 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3644 // already timed out waiting for the status and will never signal the condition. 3645 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3646 } 3647 3648 if (!(previousCommand & FastMixerState::IDLE)) { 3649 ALOG_ASSERT(mFastMixer != NULL); 3650 FastMixerStateQueue *sq = mFastMixer->sq(); 3651 FastMixerState *state = sq->begin(); 3652 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3653 state->mCommand = previousCommand; 3654 sq->end(); 3655 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3656 } 3657 3658 return reconfig; 3659} 3660 3661 3662void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3663{ 3664 const size_t SIZE = 256; 3665 char buffer[SIZE]; 3666 String8 result; 3667 3668 PlaybackThread::dumpInternals(fd, args); 3669 3670 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3671 3672 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3673 const FastMixerDumpState copy(mFastMixerDumpState); 3674 copy.dump(fd); 3675 3676#ifdef STATE_QUEUE_DUMP 3677 // Similar for state queue 3678 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3679 observerCopy.dump(fd); 3680 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3681 mutatorCopy.dump(fd); 3682#endif 3683 3684#ifdef TEE_SINK 3685 // Write the tee output to a .wav file 3686 dumpTee(fd, mTeeSource, mId); 3687#endif 3688 3689#ifdef AUDIO_WATCHDOG 3690 if (mAudioWatchdog != 0) { 3691 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3692 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3693 wdCopy.dump(fd); 3694 } 3695#endif 3696} 3697 3698uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3699{ 3700 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3701} 3702 3703uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3704{ 3705 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3706} 3707 3708void AudioFlinger::MixerThread::cacheParameters_l() 3709{ 3710 PlaybackThread::cacheParameters_l(); 3711 3712 // FIXME: Relaxed timing because of a certain device that can't meet latency 3713 // Should be reduced to 2x after the vendor fixes the driver issue 3714 // increase threshold again due to low power audio mode. The way this warning 3715 // threshold is calculated and its usefulness should be reconsidered anyway. 3716 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3717} 3718 3719// ---------------------------------------------------------------------------- 3720 3721AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3722 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3723 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3724 // mLeftVolFloat, mRightVolFloat 3725{ 3726} 3727 3728AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3729 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3730 ThreadBase::type_t type) 3731 : PlaybackThread(audioFlinger, output, id, device, type) 3732 // mLeftVolFloat, mRightVolFloat 3733{ 3734} 3735 3736AudioFlinger::DirectOutputThread::~DirectOutputThread() 3737{ 3738} 3739 3740void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3741{ 3742 audio_track_cblk_t* cblk = track->cblk(); 3743 float left, right; 3744 3745 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3746 left = right = 0; 3747 } else { 3748 float typeVolume = mStreamTypes[track->streamType()].volume; 3749 float v = mMasterVolume * typeVolume; 3750 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3751 uint32_t vlr = proxy->getVolumeLR(); 3752 float v_clamped = v * (vlr & 0xFFFF); 3753 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3754 left = v_clamped/MAX_GAIN; 3755 v_clamped = v * (vlr >> 16); 3756 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3757 right = v_clamped/MAX_GAIN; 3758 } 3759 3760 if (lastTrack) { 3761 if (left != mLeftVolFloat || right != mRightVolFloat) { 3762 mLeftVolFloat = left; 3763 mRightVolFloat = right; 3764 3765 // Convert volumes from float to 8.24 3766 uint32_t vl = (uint32_t)(left * (1 << 24)); 3767 uint32_t vr = (uint32_t)(right * (1 << 24)); 3768 3769 // Delegate volume control to effect in track effect chain if needed 3770 // only one effect chain can be present on DirectOutputThread, so if 3771 // there is one, the track is connected to it 3772 if (!mEffectChains.isEmpty()) { 3773 mEffectChains[0]->setVolume_l(&vl, &vr); 3774 left = (float)vl / (1 << 24); 3775 right = (float)vr / (1 << 24); 3776 } 3777 if (mOutput->stream->set_volume) { 3778 mOutput->stream->set_volume(mOutput->stream, left, right); 3779 } 3780 } 3781 } 3782} 3783 3784 3785AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3786 Vector< sp<Track> > *tracksToRemove 3787) 3788{ 3789 size_t count = mActiveTracks.size(); 3790 mixer_state mixerStatus = MIXER_IDLE; 3791 3792 // find out which tracks need to be processed 3793 for (size_t i = 0; i < count; i++) { 3794 sp<Track> t = mActiveTracks[i].promote(); 3795 // The track died recently 3796 if (t == 0) { 3797 continue; 3798 } 3799 3800 Track* const track = t.get(); 3801 audio_track_cblk_t* cblk = track->cblk(); 3802 // Only consider last track started for volume and mixer state control. 3803 // In theory an older track could underrun and restart after the new one starts 3804 // but as we only care about the transition phase between two tracks on a 3805 // direct output, it is not a problem to ignore the underrun case. 3806 sp<Track> l = mLatestActiveTrack.promote(); 3807 bool last = l.get() == track; 3808 3809 // The first time a track is added we wait 3810 // for all its buffers to be filled before processing it 3811 uint32_t minFrames; 3812 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3813 minFrames = mNormalFrameCount; 3814 } else { 3815 minFrames = 1; 3816 } 3817 3818 if ((track->framesReady() >= minFrames) && track->isReady() && 3819 !track->isPaused() && !track->isTerminated()) 3820 { 3821 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3822 3823 if (track->mFillingUpStatus == Track::FS_FILLED) { 3824 track->mFillingUpStatus = Track::FS_ACTIVE; 3825 // make sure processVolume_l() will apply new volume even if 0 3826 mLeftVolFloat = mRightVolFloat = -1.0; 3827 if (track->mState == TrackBase::RESUMING) { 3828 track->mState = TrackBase::ACTIVE; 3829 } 3830 } 3831 3832 // compute volume for this track 3833 processVolume_l(track, last); 3834 if (last) { 3835 // reset retry count 3836 track->mRetryCount = kMaxTrackRetriesDirect; 3837 mActiveTrack = t; 3838 mixerStatus = MIXER_TRACKS_READY; 3839 } 3840 } else { 3841 // clear effect chain input buffer if the last active track started underruns 3842 // to avoid sending previous audio buffer again to effects 3843 if (!mEffectChains.isEmpty() && last) { 3844 mEffectChains[0]->clearInputBuffer(); 3845 } 3846 3847 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3848 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3849 track->isStopped() || track->isPaused()) { 3850 // We have consumed all the buffers of this track. 3851 // Remove it from the list of active tracks. 3852 // TODO: implement behavior for compressed audio 3853 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3854 size_t framesWritten = mBytesWritten / mFrameSize; 3855 if (mStandby || !last || 3856 track->presentationComplete(framesWritten, audioHALFrames)) { 3857 if (track->isStopped()) { 3858 track->reset(); 3859 } 3860 tracksToRemove->add(track); 3861 } 3862 } else { 3863 // No buffers for this track. Give it a few chances to 3864 // fill a buffer, then remove it from active list. 3865 // Only consider last track started for mixer state control 3866 if (--(track->mRetryCount) <= 0) { 3867 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3868 tracksToRemove->add(track); 3869 // indicate to client process that the track was disabled because of underrun; 3870 // it will then automatically call start() when data is available 3871 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3872 } else if (last) { 3873 mixerStatus = MIXER_TRACKS_ENABLED; 3874 } 3875 } 3876 } 3877 } 3878 3879 // remove all the tracks that need to be... 3880 removeTracks_l(*tracksToRemove); 3881 3882 return mixerStatus; 3883} 3884 3885void AudioFlinger::DirectOutputThread::threadLoop_mix() 3886{ 3887 size_t frameCount = mFrameCount; 3888 int8_t *curBuf = (int8_t *)mSinkBuffer; 3889 // output audio to hardware 3890 while (frameCount) { 3891 AudioBufferProvider::Buffer buffer; 3892 buffer.frameCount = frameCount; 3893 mActiveTrack->getNextBuffer(&buffer); 3894 if (buffer.raw == NULL) { 3895 memset(curBuf, 0, frameCount * mFrameSize); 3896 break; 3897 } 3898 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3899 frameCount -= buffer.frameCount; 3900 curBuf += buffer.frameCount * mFrameSize; 3901 mActiveTrack->releaseBuffer(&buffer); 3902 } 3903 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3904 sleepTime = 0; 3905 standbyTime = systemTime() + standbyDelay; 3906 mActiveTrack.clear(); 3907} 3908 3909void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3910{ 3911 if (sleepTime == 0) { 3912 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3913 sleepTime = activeSleepTime; 3914 } else { 3915 sleepTime = idleSleepTime; 3916 } 3917 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3918 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3919 sleepTime = 0; 3920 } 3921} 3922 3923// getTrackName_l() must be called with ThreadBase::mLock held 3924int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3925 int sessionId __unused) 3926{ 3927 return 0; 3928} 3929 3930// deleteTrackName_l() must be called with ThreadBase::mLock held 3931void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3932{ 3933} 3934 3935// checkForNewParameters_l() must be called with ThreadBase::mLock held 3936bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3937{ 3938 bool reconfig = false; 3939 3940 while (!mNewParameters.isEmpty()) { 3941 status_t status = NO_ERROR; 3942 String8 keyValuePair = mNewParameters[0]; 3943 AudioParameter param = AudioParameter(keyValuePair); 3944 int value; 3945 3946 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3947 // do not accept frame count changes if tracks are open as the track buffer 3948 // size depends on frame count and correct behavior would not be garantied 3949 // if frame count is changed after track creation 3950 if (!mTracks.isEmpty()) { 3951 status = INVALID_OPERATION; 3952 } else { 3953 reconfig = true; 3954 } 3955 } 3956 if (status == NO_ERROR) { 3957 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3958 keyValuePair.string()); 3959 if (!mStandby && status == INVALID_OPERATION) { 3960 mOutput->stream->common.standby(&mOutput->stream->common); 3961 mStandby = true; 3962 mBytesWritten = 0; 3963 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3964 keyValuePair.string()); 3965 } 3966 if (status == NO_ERROR && reconfig) { 3967 readOutputParameters_l(); 3968 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3969 } 3970 } 3971 3972 mNewParameters.removeAt(0); 3973 3974 mParamStatus = status; 3975 mParamCond.signal(); 3976 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3977 // already timed out waiting for the status and will never signal the condition. 3978 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3979 } 3980 return reconfig; 3981} 3982 3983uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3984{ 3985 uint32_t time; 3986 if (audio_is_linear_pcm(mFormat)) { 3987 time = PlaybackThread::activeSleepTimeUs(); 3988 } else { 3989 time = 10000; 3990 } 3991 return time; 3992} 3993 3994uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3995{ 3996 uint32_t time; 3997 if (audio_is_linear_pcm(mFormat)) { 3998 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3999 } else { 4000 time = 10000; 4001 } 4002 return time; 4003} 4004 4005uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4006{ 4007 uint32_t time; 4008 if (audio_is_linear_pcm(mFormat)) { 4009 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4010 } else { 4011 time = 10000; 4012 } 4013 return time; 4014} 4015 4016void AudioFlinger::DirectOutputThread::cacheParameters_l() 4017{ 4018 PlaybackThread::cacheParameters_l(); 4019 4020 // use shorter standby delay as on normal output to release 4021 // hardware resources as soon as possible 4022 if (audio_is_linear_pcm(mFormat)) { 4023 standbyDelay = microseconds(activeSleepTime*2); 4024 } else { 4025 standbyDelay = kOffloadStandbyDelayNs; 4026 } 4027} 4028 4029// ---------------------------------------------------------------------------- 4030 4031AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4032 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4033 : Thread(false /*canCallJava*/), 4034 mPlaybackThread(playbackThread), 4035 mWriteAckSequence(0), 4036 mDrainSequence(0) 4037{ 4038} 4039 4040AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4041{ 4042} 4043 4044void AudioFlinger::AsyncCallbackThread::onFirstRef() 4045{ 4046 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4047} 4048 4049bool AudioFlinger::AsyncCallbackThread::threadLoop() 4050{ 4051 while (!exitPending()) { 4052 uint32_t writeAckSequence; 4053 uint32_t drainSequence; 4054 4055 { 4056 Mutex::Autolock _l(mLock); 4057 while (!((mWriteAckSequence & 1) || 4058 (mDrainSequence & 1) || 4059 exitPending())) { 4060 mWaitWorkCV.wait(mLock); 4061 } 4062 4063 if (exitPending()) { 4064 break; 4065 } 4066 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4067 mWriteAckSequence, mDrainSequence); 4068 writeAckSequence = mWriteAckSequence; 4069 mWriteAckSequence &= ~1; 4070 drainSequence = mDrainSequence; 4071 mDrainSequence &= ~1; 4072 } 4073 { 4074 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4075 if (playbackThread != 0) { 4076 if (writeAckSequence & 1) { 4077 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4078 } 4079 if (drainSequence & 1) { 4080 playbackThread->resetDraining(drainSequence >> 1); 4081 } 4082 } 4083 } 4084 } 4085 return false; 4086} 4087 4088void AudioFlinger::AsyncCallbackThread::exit() 4089{ 4090 ALOGV("AsyncCallbackThread::exit"); 4091 Mutex::Autolock _l(mLock); 4092 requestExit(); 4093 mWaitWorkCV.broadcast(); 4094} 4095 4096void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4097{ 4098 Mutex::Autolock _l(mLock); 4099 // bit 0 is cleared 4100 mWriteAckSequence = sequence << 1; 4101} 4102 4103void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4104{ 4105 Mutex::Autolock _l(mLock); 4106 // ignore unexpected callbacks 4107 if (mWriteAckSequence & 2) { 4108 mWriteAckSequence |= 1; 4109 mWaitWorkCV.signal(); 4110 } 4111} 4112 4113void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4114{ 4115 Mutex::Autolock _l(mLock); 4116 // bit 0 is cleared 4117 mDrainSequence = sequence << 1; 4118} 4119 4120void AudioFlinger::AsyncCallbackThread::resetDraining() 4121{ 4122 Mutex::Autolock _l(mLock); 4123 // ignore unexpected callbacks 4124 if (mDrainSequence & 2) { 4125 mDrainSequence |= 1; 4126 mWaitWorkCV.signal(); 4127 } 4128} 4129 4130 4131// ---------------------------------------------------------------------------- 4132AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4133 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4134 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4135 mHwPaused(false), 4136 mFlushPending(false), 4137 mPausedBytesRemaining(0) 4138{ 4139 //FIXME: mStandby should be set to true by ThreadBase constructor 4140 mStandby = true; 4141} 4142 4143void AudioFlinger::OffloadThread::threadLoop_exit() 4144{ 4145 if (mFlushPending || mHwPaused) { 4146 // If a flush is pending or track was paused, just discard buffered data 4147 flushHw_l(); 4148 } else { 4149 mMixerStatus = MIXER_DRAIN_ALL; 4150 threadLoop_drain(); 4151 } 4152 mCallbackThread->exit(); 4153 PlaybackThread::threadLoop_exit(); 4154} 4155 4156AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4157 Vector< sp<Track> > *tracksToRemove 4158) 4159{ 4160 size_t count = mActiveTracks.size(); 4161 4162 mixer_state mixerStatus = MIXER_IDLE; 4163 bool doHwPause = false; 4164 bool doHwResume = false; 4165 4166 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4167 4168 // find out which tracks need to be processed 4169 for (size_t i = 0; i < count; i++) { 4170 sp<Track> t = mActiveTracks[i].promote(); 4171 // The track died recently 4172 if (t == 0) { 4173 continue; 4174 } 4175 Track* const track = t.get(); 4176 audio_track_cblk_t* cblk = track->cblk(); 4177 // Only consider last track started for volume and mixer state control. 4178 // In theory an older track could underrun and restart after the new one starts 4179 // but as we only care about the transition phase between two tracks on a 4180 // direct output, it is not a problem to ignore the underrun case. 4181 sp<Track> l = mLatestActiveTrack.promote(); 4182 bool last = l.get() == track; 4183 4184 if (track->isInvalid()) { 4185 ALOGW("An invalidated track shouldn't be in active list"); 4186 tracksToRemove->add(track); 4187 continue; 4188 } 4189 4190 if (track->mState == TrackBase::IDLE) { 4191 ALOGW("An idle track shouldn't be in active list"); 4192 continue; 4193 } 4194 4195 if (track->isPausing()) { 4196 track->setPaused(); 4197 if (last) { 4198 if (!mHwPaused) { 4199 doHwPause = true; 4200 mHwPaused = true; 4201 } 4202 // If we were part way through writing the mixbuffer to 4203 // the HAL we must save this until we resume 4204 // BUG - this will be wrong if a different track is made active, 4205 // in that case we want to discard the pending data in the 4206 // mixbuffer and tell the client to present it again when the 4207 // track is resumed 4208 mPausedWriteLength = mCurrentWriteLength; 4209 mPausedBytesRemaining = mBytesRemaining; 4210 mBytesRemaining = 0; // stop writing 4211 } 4212 tracksToRemove->add(track); 4213 } else if (track->isFlushPending()) { 4214 track->flushAck(); 4215 if (last) { 4216 mFlushPending = true; 4217 } 4218 } else if (track->framesReady() && track->isReady() && 4219 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4220 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4221 if (track->mFillingUpStatus == Track::FS_FILLED) { 4222 track->mFillingUpStatus = Track::FS_ACTIVE; 4223 // make sure processVolume_l() will apply new volume even if 0 4224 mLeftVolFloat = mRightVolFloat = -1.0; 4225 if (track->mState == TrackBase::RESUMING) { 4226 track->mState = TrackBase::ACTIVE; 4227 if (last) { 4228 if (mPausedBytesRemaining) { 4229 // Need to continue write that was interrupted 4230 mCurrentWriteLength = mPausedWriteLength; 4231 mBytesRemaining = mPausedBytesRemaining; 4232 mPausedBytesRemaining = 0; 4233 } 4234 if (mHwPaused) { 4235 doHwResume = true; 4236 mHwPaused = false; 4237 // threadLoop_mix() will handle the case that we need to 4238 // resume an interrupted write 4239 } 4240 // enable write to audio HAL 4241 sleepTime = 0; 4242 } 4243 } 4244 } 4245 4246 if (last) { 4247 sp<Track> previousTrack = mPreviousTrack.promote(); 4248 if (previousTrack != 0) { 4249 if (track != previousTrack.get()) { 4250 // Flush any data still being written from last track 4251 mBytesRemaining = 0; 4252 if (mPausedBytesRemaining) { 4253 // Last track was paused so we also need to flush saved 4254 // mixbuffer state and invalidate track so that it will 4255 // re-submit that unwritten data when it is next resumed 4256 mPausedBytesRemaining = 0; 4257 // Invalidate is a bit drastic - would be more efficient 4258 // to have a flag to tell client that some of the 4259 // previously written data was lost 4260 previousTrack->invalidate(); 4261 } 4262 // flush data already sent to the DSP if changing audio session as audio 4263 // comes from a different source. Also invalidate previous track to force a 4264 // seek when resuming. 4265 if (previousTrack->sessionId() != track->sessionId()) { 4266 previousTrack->invalidate(); 4267 } 4268 } 4269 } 4270 mPreviousTrack = track; 4271 // reset retry count 4272 track->mRetryCount = kMaxTrackRetriesOffload; 4273 mActiveTrack = t; 4274 mixerStatus = MIXER_TRACKS_READY; 4275 } 4276 } else { 4277 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4278 if (track->isStopping_1()) { 4279 // Hardware buffer can hold a large amount of audio so we must 4280 // wait for all current track's data to drain before we say 4281 // that the track is stopped. 4282 if (mBytesRemaining == 0) { 4283 // Only start draining when all data in mixbuffer 4284 // has been written 4285 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4286 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4287 // do not drain if no data was ever sent to HAL (mStandby == true) 4288 if (last && !mStandby) { 4289 // do not modify drain sequence if we are already draining. This happens 4290 // when resuming from pause after drain. 4291 if ((mDrainSequence & 1) == 0) { 4292 sleepTime = 0; 4293 standbyTime = systemTime() + standbyDelay; 4294 mixerStatus = MIXER_DRAIN_TRACK; 4295 mDrainSequence += 2; 4296 } 4297 if (mHwPaused) { 4298 // It is possible to move from PAUSED to STOPPING_1 without 4299 // a resume so we must ensure hardware is running 4300 doHwResume = true; 4301 mHwPaused = false; 4302 } 4303 } 4304 } 4305 } else if (track->isStopping_2()) { 4306 // Drain has completed or we are in standby, signal presentation complete 4307 if (!(mDrainSequence & 1) || !last || mStandby) { 4308 track->mState = TrackBase::STOPPED; 4309 size_t audioHALFrames = 4310 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4311 size_t framesWritten = 4312 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4313 track->presentationComplete(framesWritten, audioHALFrames); 4314 track->reset(); 4315 tracksToRemove->add(track); 4316 } 4317 } else { 4318 // No buffers for this track. Give it a few chances to 4319 // fill a buffer, then remove it from active list. 4320 if (--(track->mRetryCount) <= 0) { 4321 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4322 track->name()); 4323 tracksToRemove->add(track); 4324 // indicate to client process that the track was disabled because of underrun; 4325 // it will then automatically call start() when data is available 4326 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4327 } else if (last){ 4328 mixerStatus = MIXER_TRACKS_ENABLED; 4329 } 4330 } 4331 } 4332 // compute volume for this track 4333 processVolume_l(track, last); 4334 } 4335 4336 // make sure the pause/flush/resume sequence is executed in the right order. 4337 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4338 // before flush and then resume HW. This can happen in case of pause/flush/resume 4339 // if resume is received before pause is executed. 4340 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4341 mOutput->stream->pause(mOutput->stream); 4342 } 4343 if (mFlushPending) { 4344 flushHw_l(); 4345 mFlushPending = false; 4346 } 4347 if (!mStandby && doHwResume) { 4348 mOutput->stream->resume(mOutput->stream); 4349 } 4350 4351 // remove all the tracks that need to be... 4352 removeTracks_l(*tracksToRemove); 4353 4354 return mixerStatus; 4355} 4356 4357// must be called with thread mutex locked 4358bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4359{ 4360 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4361 mWriteAckSequence, mDrainSequence); 4362 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4363 return true; 4364 } 4365 return false; 4366} 4367 4368// must be called with thread mutex locked 4369bool AudioFlinger::OffloadThread::shouldStandby_l() 4370{ 4371 bool trackPaused = false; 4372 4373 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4374 // after a timeout and we will enter standby then. 4375 if (mTracks.size() > 0) { 4376 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4377 } 4378 4379 return !mStandby && !trackPaused; 4380} 4381 4382 4383bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4384{ 4385 Mutex::Autolock _l(mLock); 4386 return waitingAsyncCallback_l(); 4387} 4388 4389void AudioFlinger::OffloadThread::flushHw_l() 4390{ 4391 mOutput->stream->flush(mOutput->stream); 4392 // Flush anything still waiting in the mixbuffer 4393 mCurrentWriteLength = 0; 4394 mBytesRemaining = 0; 4395 mPausedWriteLength = 0; 4396 mPausedBytesRemaining = 0; 4397 mHwPaused = false; 4398 4399 if (mUseAsyncWrite) { 4400 // discard any pending drain or write ack by incrementing sequence 4401 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4402 mDrainSequence = (mDrainSequence + 2) & ~1; 4403 ALOG_ASSERT(mCallbackThread != 0); 4404 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4405 mCallbackThread->setDraining(mDrainSequence); 4406 } 4407} 4408 4409void AudioFlinger::OffloadThread::onAddNewTrack_l() 4410{ 4411 sp<Track> previousTrack = mPreviousTrack.promote(); 4412 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4413 4414 if (previousTrack != 0 && latestTrack != 0 && 4415 (previousTrack->sessionId() != latestTrack->sessionId())) { 4416 mFlushPending = true; 4417 } 4418 PlaybackThread::onAddNewTrack_l(); 4419} 4420 4421// ---------------------------------------------------------------------------- 4422 4423AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4424 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4425 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4426 DUPLICATING), 4427 mWaitTimeMs(UINT_MAX) 4428{ 4429 addOutputTrack(mainThread); 4430} 4431 4432AudioFlinger::DuplicatingThread::~DuplicatingThread() 4433{ 4434 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4435 mOutputTracks[i]->destroy(); 4436 } 4437} 4438 4439void AudioFlinger::DuplicatingThread::threadLoop_mix() 4440{ 4441 // mix buffers... 4442 if (outputsReady(outputTracks)) { 4443 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4444 } else { 4445 memset(mSinkBuffer, 0, mSinkBufferSize); 4446 } 4447 sleepTime = 0; 4448 writeFrames = mNormalFrameCount; 4449 mCurrentWriteLength = mSinkBufferSize; 4450 standbyTime = systemTime() + standbyDelay; 4451} 4452 4453void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4454{ 4455 if (sleepTime == 0) { 4456 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4457 sleepTime = activeSleepTime; 4458 } else { 4459 sleepTime = idleSleepTime; 4460 } 4461 } else if (mBytesWritten != 0) { 4462 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4463 writeFrames = mNormalFrameCount; 4464 memset(mSinkBuffer, 0, mSinkBufferSize); 4465 } else { 4466 // flush remaining overflow buffers in output tracks 4467 writeFrames = 0; 4468 } 4469 sleepTime = 0; 4470 } 4471} 4472 4473ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4474{ 4475 for (size_t i = 0; i < outputTracks.size(); i++) { 4476 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4477 // for delivery downstream as needed. This in-place conversion is safe as 4478 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4479 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4480 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4481 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4482 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4483 } 4484 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4485 } 4486 mStandby = false; 4487 return (ssize_t)mSinkBufferSize; 4488} 4489 4490void AudioFlinger::DuplicatingThread::threadLoop_standby() 4491{ 4492 // DuplicatingThread implements standby by stopping all tracks 4493 for (size_t i = 0; i < outputTracks.size(); i++) { 4494 outputTracks[i]->stop(); 4495 } 4496} 4497 4498void AudioFlinger::DuplicatingThread::saveOutputTracks() 4499{ 4500 outputTracks = mOutputTracks; 4501} 4502 4503void AudioFlinger::DuplicatingThread::clearOutputTracks() 4504{ 4505 outputTracks.clear(); 4506} 4507 4508void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4509{ 4510 Mutex::Autolock _l(mLock); 4511 // FIXME explain this formula 4512 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4513 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4514 // due to current usage case and restrictions on the AudioBufferProvider. 4515 // Actual buffer conversion is done in threadLoop_write(). 4516 // 4517 // TODO: This may change in the future, depending on multichannel 4518 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4519 OutputTrack *outputTrack = new OutputTrack(thread, 4520 this, 4521 mSampleRate, 4522 AUDIO_FORMAT_PCM_16_BIT, 4523 mChannelMask, 4524 frameCount, 4525 IPCThreadState::self()->getCallingUid()); 4526 if (outputTrack->cblk() != NULL) { 4527 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4528 mOutputTracks.add(outputTrack); 4529 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4530 updateWaitTime_l(); 4531 } 4532} 4533 4534void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4535{ 4536 Mutex::Autolock _l(mLock); 4537 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4538 if (mOutputTracks[i]->thread() == thread) { 4539 mOutputTracks[i]->destroy(); 4540 mOutputTracks.removeAt(i); 4541 updateWaitTime_l(); 4542 return; 4543 } 4544 } 4545 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4546} 4547 4548// caller must hold mLock 4549void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4550{ 4551 mWaitTimeMs = UINT_MAX; 4552 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4553 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4554 if (strong != 0) { 4555 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4556 if (waitTimeMs < mWaitTimeMs) { 4557 mWaitTimeMs = waitTimeMs; 4558 } 4559 } 4560 } 4561} 4562 4563 4564bool AudioFlinger::DuplicatingThread::outputsReady( 4565 const SortedVector< sp<OutputTrack> > &outputTracks) 4566{ 4567 for (size_t i = 0; i < outputTracks.size(); i++) { 4568 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4569 if (thread == 0) { 4570 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4571 outputTracks[i].get()); 4572 return false; 4573 } 4574 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4575 // see note at standby() declaration 4576 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4577 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4578 thread.get()); 4579 return false; 4580 } 4581 } 4582 return true; 4583} 4584 4585uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4586{ 4587 return (mWaitTimeMs * 1000) / 2; 4588} 4589 4590void AudioFlinger::DuplicatingThread::cacheParameters_l() 4591{ 4592 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4593 updateWaitTime_l(); 4594 4595 MixerThread::cacheParameters_l(); 4596} 4597 4598// ---------------------------------------------------------------------------- 4599// Record 4600// ---------------------------------------------------------------------------- 4601 4602AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4603 AudioStreamIn *input, 4604 audio_io_handle_t id, 4605 audio_devices_t outDevice, 4606 audio_devices_t inDevice 4607#ifdef TEE_SINK 4608 , const sp<NBAIO_Sink>& teeSink 4609#endif 4610 ) : 4611 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4612 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4613 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4614 mRsmpInRear(0) 4615#ifdef TEE_SINK 4616 , mTeeSink(teeSink) 4617#endif 4618{ 4619 snprintf(mName, kNameLength, "AudioIn_%X", id); 4620 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4621 4622 readInputParameters_l(); 4623} 4624 4625 4626AudioFlinger::RecordThread::~RecordThread() 4627{ 4628 mAudioFlinger->unregisterWriter(mNBLogWriter); 4629 delete[] mRsmpInBuffer; 4630} 4631 4632void AudioFlinger::RecordThread::onFirstRef() 4633{ 4634 run(mName, PRIORITY_URGENT_AUDIO); 4635} 4636 4637bool AudioFlinger::RecordThread::threadLoop() 4638{ 4639 nsecs_t lastWarning = 0; 4640 4641 inputStandBy(); 4642 4643reacquire_wakelock: 4644 sp<RecordTrack> activeTrack; 4645 int activeTracksGen; 4646 { 4647 Mutex::Autolock _l(mLock); 4648 size_t size = mActiveTracks.size(); 4649 activeTracksGen = mActiveTracksGen; 4650 if (size > 0) { 4651 // FIXME an arbitrary choice 4652 activeTrack = mActiveTracks[0]; 4653 acquireWakeLock_l(activeTrack->uid()); 4654 if (size > 1) { 4655 SortedVector<int> tmp; 4656 for (size_t i = 0; i < size; i++) { 4657 tmp.add(mActiveTracks[i]->uid()); 4658 } 4659 updateWakeLockUids_l(tmp); 4660 } 4661 } else { 4662 acquireWakeLock_l(-1); 4663 } 4664 } 4665 4666 // used to request a deferred sleep, to be executed later while mutex is unlocked 4667 uint32_t sleepUs = 0; 4668 4669 // loop while there is work to do 4670 for (;;) { 4671 Vector< sp<EffectChain> > effectChains; 4672 4673 // sleep with mutex unlocked 4674 if (sleepUs > 0) { 4675 usleep(sleepUs); 4676 sleepUs = 0; 4677 } 4678 4679 // activeTracks accumulates a copy of a subset of mActiveTracks 4680 Vector< sp<RecordTrack> > activeTracks; 4681 4682 { // scope for mLock 4683 Mutex::Autolock _l(mLock); 4684 4685 processConfigEvents_l(); 4686 // return value 'reconfig' is currently unused 4687 bool reconfig = checkForNewParameters_l(); 4688 4689 // check exitPending here because checkForNewParameters_l() and 4690 // checkForNewParameters_l() can temporarily release mLock 4691 if (exitPending()) { 4692 break; 4693 } 4694 4695 // if no active track(s), then standby and release wakelock 4696 size_t size = mActiveTracks.size(); 4697 if (size == 0) { 4698 standbyIfNotAlreadyInStandby(); 4699 // exitPending() can't become true here 4700 releaseWakeLock_l(); 4701 ALOGV("RecordThread: loop stopping"); 4702 // go to sleep 4703 mWaitWorkCV.wait(mLock); 4704 ALOGV("RecordThread: loop starting"); 4705 goto reacquire_wakelock; 4706 } 4707 4708 if (mActiveTracksGen != activeTracksGen) { 4709 activeTracksGen = mActiveTracksGen; 4710 SortedVector<int> tmp; 4711 for (size_t i = 0; i < size; i++) { 4712 tmp.add(mActiveTracks[i]->uid()); 4713 } 4714 updateWakeLockUids_l(tmp); 4715 } 4716 4717 bool doBroadcast = false; 4718 for (size_t i = 0; i < size; ) { 4719 4720 activeTrack = mActiveTracks[i]; 4721 if (activeTrack->isTerminated()) { 4722 removeTrack_l(activeTrack); 4723 mActiveTracks.remove(activeTrack); 4724 mActiveTracksGen++; 4725 size--; 4726 continue; 4727 } 4728 4729 TrackBase::track_state activeTrackState = activeTrack->mState; 4730 switch (activeTrackState) { 4731 4732 case TrackBase::PAUSING: 4733 mActiveTracks.remove(activeTrack); 4734 mActiveTracksGen++; 4735 doBroadcast = true; 4736 size--; 4737 continue; 4738 4739 case TrackBase::STARTING_1: 4740 sleepUs = 10000; 4741 i++; 4742 continue; 4743 4744 case TrackBase::STARTING_2: 4745 doBroadcast = true; 4746 mStandby = false; 4747 activeTrack->mState = TrackBase::ACTIVE; 4748 break; 4749 4750 case TrackBase::ACTIVE: 4751 break; 4752 4753 case TrackBase::IDLE: 4754 i++; 4755 continue; 4756 4757 default: 4758 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4759 } 4760 4761 activeTracks.add(activeTrack); 4762 i++; 4763 4764 } 4765 if (doBroadcast) { 4766 mStartStopCond.broadcast(); 4767 } 4768 4769 // sleep if there are no active tracks to process 4770 if (activeTracks.size() == 0) { 4771 if (sleepUs == 0) { 4772 sleepUs = kRecordThreadSleepUs; 4773 } 4774 continue; 4775 } 4776 sleepUs = 0; 4777 4778 lockEffectChains_l(effectChains); 4779 } 4780 4781 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4782 4783 size_t size = effectChains.size(); 4784 for (size_t i = 0; i < size; i++) { 4785 // thread mutex is not locked, but effect chain is locked 4786 effectChains[i]->process_l(); 4787 } 4788 4789 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4790 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4791 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4792 // If destination is non-contiguous, first read past the nominal end of buffer, then 4793 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4794 4795 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4796 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4797 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4798 if (bytesRead <= 0) { 4799 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4800 // Force input into standby so that it tries to recover at next read attempt 4801 inputStandBy(); 4802 sleepUs = kRecordThreadSleepUs; 4803 continue; 4804 } 4805 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4806 size_t framesRead = bytesRead / mFrameSize; 4807 ALOG_ASSERT(framesRead > 0); 4808 if (mTeeSink != 0) { 4809 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4810 } 4811 // If destination is non-contiguous, we now correct for reading past end of buffer. 4812 size_t part1 = mRsmpInFramesP2 - rear; 4813 if (framesRead > part1) { 4814 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4815 (framesRead - part1) * mFrameSize); 4816 } 4817 rear = mRsmpInRear += framesRead; 4818 4819 size = activeTracks.size(); 4820 // loop over each active track 4821 for (size_t i = 0; i < size; i++) { 4822 activeTrack = activeTracks[i]; 4823 4824 enum { 4825 OVERRUN_UNKNOWN, 4826 OVERRUN_TRUE, 4827 OVERRUN_FALSE 4828 } overrun = OVERRUN_UNKNOWN; 4829 4830 // loop over getNextBuffer to handle circular sink 4831 for (;;) { 4832 4833 activeTrack->mSink.frameCount = ~0; 4834 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4835 size_t framesOut = activeTrack->mSink.frameCount; 4836 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4837 4838 int32_t front = activeTrack->mRsmpInFront; 4839 ssize_t filled = rear - front; 4840 size_t framesIn; 4841 4842 if (filled < 0) { 4843 // should not happen, but treat like a massive overrun and re-sync 4844 framesIn = 0; 4845 activeTrack->mRsmpInFront = rear; 4846 overrun = OVERRUN_TRUE; 4847 } else if ((size_t) filled <= mRsmpInFrames) { 4848 framesIn = (size_t) filled; 4849 } else { 4850 // client is not keeping up with server, but give it latest data 4851 framesIn = mRsmpInFrames; 4852 activeTrack->mRsmpInFront = front = rear - framesIn; 4853 overrun = OVERRUN_TRUE; 4854 } 4855 4856 if (framesOut == 0 || framesIn == 0) { 4857 break; 4858 } 4859 4860 if (activeTrack->mResampler == NULL) { 4861 // no resampling 4862 if (framesIn > framesOut) { 4863 framesIn = framesOut; 4864 } else { 4865 framesOut = framesIn; 4866 } 4867 int8_t *dst = activeTrack->mSink.i8; 4868 while (framesIn > 0) { 4869 front &= mRsmpInFramesP2 - 1; 4870 size_t part1 = mRsmpInFramesP2 - front; 4871 if (part1 > framesIn) { 4872 part1 = framesIn; 4873 } 4874 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4875 if (mChannelCount == activeTrack->mChannelCount) { 4876 memcpy(dst, src, part1 * mFrameSize); 4877 } else if (mChannelCount == 1) { 4878 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4879 part1); 4880 } else { 4881 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4882 part1); 4883 } 4884 dst += part1 * activeTrack->mFrameSize; 4885 front += part1; 4886 framesIn -= part1; 4887 } 4888 activeTrack->mRsmpInFront += framesOut; 4889 4890 } else { 4891 // resampling 4892 // FIXME framesInNeeded should really be part of resampler API, and should 4893 // depend on the SRC ratio 4894 // to keep mRsmpInBuffer full so resampler always has sufficient input 4895 size_t framesInNeeded; 4896 // FIXME only re-calculate when it changes, and optimize for common ratios 4897 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4898 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4899 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4900 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4901 framesInNeeded, framesOut, inOverOut); 4902 // Although we theoretically have framesIn in circular buffer, some of those are 4903 // unreleased frames, and thus must be discounted for purpose of budgeting. 4904 size_t unreleased = activeTrack->mRsmpInUnrel; 4905 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4906 if (framesIn < framesInNeeded) { 4907 ALOGV("not enough to resample: have %u frames in but need %u in to " 4908 "produce %u out given in/out ratio of %.4g", 4909 framesIn, framesInNeeded, framesOut, inOverOut); 4910 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4911 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4912 if (newFramesOut == 0) { 4913 break; 4914 } 4915 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4916 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4917 framesInNeeded, newFramesOut, outOverIn); 4918 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4919 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4920 "given in/out ratio of %.4g", 4921 framesIn, framesInNeeded, newFramesOut, inOverOut); 4922 framesOut = newFramesOut; 4923 } else { 4924 ALOGV("success 1: have %u in and need %u in to produce %u out " 4925 "given in/out ratio of %.4g", 4926 framesIn, framesInNeeded, framesOut, inOverOut); 4927 } 4928 4929 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4930 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4931 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4932 delete[] activeTrack->mRsmpOutBuffer; 4933 // resampler always outputs stereo 4934 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4935 activeTrack->mRsmpOutFrameCount = framesOut; 4936 } 4937 4938 // resampler accumulates, but we only have one source track 4939 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4940 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4941 // FIXME how about having activeTrack implement this interface itself? 4942 activeTrack->mResamplerBufferProvider 4943 /*this*/ /* AudioBufferProvider* */); 4944 // ditherAndClamp() works as long as all buffers returned by 4945 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4946 if (activeTrack->mChannelCount == 1) { 4947 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4948 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4949 framesOut); 4950 // the resampler always outputs stereo samples: 4951 // do post stereo to mono conversion 4952 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4953 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4954 } else { 4955 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4956 activeTrack->mRsmpOutBuffer, framesOut); 4957 } 4958 // now done with mRsmpOutBuffer 4959 4960 } 4961 4962 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4963 overrun = OVERRUN_FALSE; 4964 } 4965 4966 if (activeTrack->mFramesToDrop == 0) { 4967 if (framesOut > 0) { 4968 activeTrack->mSink.frameCount = framesOut; 4969 activeTrack->releaseBuffer(&activeTrack->mSink); 4970 } 4971 } else { 4972 // FIXME could do a partial drop of framesOut 4973 if (activeTrack->mFramesToDrop > 0) { 4974 activeTrack->mFramesToDrop -= framesOut; 4975 if (activeTrack->mFramesToDrop <= 0) { 4976 activeTrack->clearSyncStartEvent(); 4977 } 4978 } else { 4979 activeTrack->mFramesToDrop += framesOut; 4980 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4981 activeTrack->mSyncStartEvent->isCancelled()) { 4982 ALOGW("Synced record %s, session %d, trigger session %d", 4983 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 4984 activeTrack->sessionId(), 4985 (activeTrack->mSyncStartEvent != 0) ? 4986 activeTrack->mSyncStartEvent->triggerSession() : 0); 4987 activeTrack->clearSyncStartEvent(); 4988 } 4989 } 4990 } 4991 4992 if (framesOut == 0) { 4993 break; 4994 } 4995 } 4996 4997 switch (overrun) { 4998 case OVERRUN_TRUE: 4999 // client isn't retrieving buffers fast enough 5000 if (!activeTrack->setOverflow()) { 5001 nsecs_t now = systemTime(); 5002 // FIXME should lastWarning per track? 5003 if ((now - lastWarning) > kWarningThrottleNs) { 5004 ALOGW("RecordThread: buffer overflow"); 5005 lastWarning = now; 5006 } 5007 } 5008 break; 5009 case OVERRUN_FALSE: 5010 activeTrack->clearOverflow(); 5011 break; 5012 case OVERRUN_UNKNOWN: 5013 break; 5014 } 5015 5016 } 5017 5018 // enable changes in effect chain 5019 unlockEffectChains(effectChains); 5020 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5021 } 5022 5023 standbyIfNotAlreadyInStandby(); 5024 5025 { 5026 Mutex::Autolock _l(mLock); 5027 for (size_t i = 0; i < mTracks.size(); i++) { 5028 sp<RecordTrack> track = mTracks[i]; 5029 track->invalidate(); 5030 } 5031 mActiveTracks.clear(); 5032 mActiveTracksGen++; 5033 mStartStopCond.broadcast(); 5034 } 5035 5036 releaseWakeLock(); 5037 5038 ALOGV("RecordThread %p exiting", this); 5039 return false; 5040} 5041 5042void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5043{ 5044 if (!mStandby) { 5045 inputStandBy(); 5046 mStandby = true; 5047 } 5048} 5049 5050void AudioFlinger::RecordThread::inputStandBy() 5051{ 5052 mInput->stream->common.standby(&mInput->stream->common); 5053} 5054 5055sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5056 const sp<AudioFlinger::Client>& client, 5057 uint32_t sampleRate, 5058 audio_format_t format, 5059 audio_channel_mask_t channelMask, 5060 size_t *pFrameCount, 5061 int sessionId, 5062 int uid, 5063 IAudioFlinger::track_flags_t *flags, 5064 pid_t tid, 5065 status_t *status) 5066{ 5067 size_t frameCount = *pFrameCount; 5068 sp<RecordTrack> track; 5069 status_t lStatus; 5070 5071 lStatus = initCheck(); 5072 if (lStatus != NO_ERROR) { 5073 ALOGE("createRecordTrack_l() audio driver not initialized"); 5074 goto Exit; 5075 } 5076 5077 // client expresses a preference for FAST, but we get the final say 5078 if (*flags & IAudioFlinger::TRACK_FAST) { 5079 if ( 5080 // use case: callback handler and frame count is default or at least as large as HAL 5081 ( 5082 (tid != -1) && 5083 ((frameCount == 0) || 5084 (frameCount >= mFrameCount)) 5085 ) && 5086 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 5087 // mono or stereo 5088 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 5089 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 5090 // hardware sample rate 5091 (sampleRate == mSampleRate) && 5092 // record thread has an associated fast recorder 5093 hasFastRecorder() 5094 // FIXME test that RecordThread for this fast track has a capable output HAL 5095 // FIXME add a permission test also? 5096 ) { 5097 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 5098 if (frameCount == 0) { 5099 frameCount = mFrameCount * kFastTrackMultiplier; 5100 } 5101 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5102 frameCount, mFrameCount); 5103 } else { 5104 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5105 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5106 "hasFastRecorder=%d tid=%d", 5107 frameCount, mFrameCount, format, 5108 audio_is_linear_pcm(format), 5109 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 5110 *flags &= ~IAudioFlinger::TRACK_FAST; 5111 // For compatibility with AudioRecord calculation, buffer depth is forced 5112 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5113 // This is probably too conservative, but legacy application code may depend on it. 5114 // If you change this calculation, also review the start threshold which is related. 5115 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5116 size_t mNormalFrameCount = 2048; // FIXME 5117 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5118 if (minBufCount < 2) { 5119 minBufCount = 2; 5120 } 5121 size_t minFrameCount = mNormalFrameCount * minBufCount; 5122 if (frameCount < minFrameCount) { 5123 frameCount = minFrameCount; 5124 } 5125 } 5126 } 5127 *pFrameCount = frameCount; 5128 5129 // FIXME use flags and tid similar to createTrack_l() 5130 5131 { // scope for mLock 5132 Mutex::Autolock _l(mLock); 5133 5134 track = new RecordTrack(this, client, sampleRate, 5135 format, channelMask, frameCount, sessionId, uid); 5136 5137 lStatus = track->initCheck(); 5138 if (lStatus != NO_ERROR) { 5139 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5140 // track must be cleared from the caller as the caller has the AF lock 5141 goto Exit; 5142 } 5143 mTracks.add(track); 5144 5145 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5146 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5147 mAudioFlinger->btNrecIsOff(); 5148 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5149 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5150 5151 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5152 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5153 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5154 // so ask activity manager to do this on our behalf 5155 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5156 } 5157 } 5158 lStatus = NO_ERROR; 5159 5160Exit: 5161 *status = lStatus; 5162 return track; 5163} 5164 5165status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5166 AudioSystem::sync_event_t event, 5167 int triggerSession) 5168{ 5169 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5170 sp<ThreadBase> strongMe = this; 5171 status_t status = NO_ERROR; 5172 5173 if (event == AudioSystem::SYNC_EVENT_NONE) { 5174 recordTrack->clearSyncStartEvent(); 5175 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5176 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5177 triggerSession, 5178 recordTrack->sessionId(), 5179 syncStartEventCallback, 5180 recordTrack); 5181 // Sync event can be cancelled by the trigger session if the track is not in a 5182 // compatible state in which case we start record immediately 5183 if (recordTrack->mSyncStartEvent->isCancelled()) { 5184 recordTrack->clearSyncStartEvent(); 5185 } else { 5186 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5187 recordTrack->mFramesToDrop = - 5188 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5189 } 5190 } 5191 5192 { 5193 // This section is a rendezvous between binder thread executing start() and RecordThread 5194 AutoMutex lock(mLock); 5195 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5196 if (recordTrack->mState == TrackBase::PAUSING) { 5197 ALOGV("active record track PAUSING -> ACTIVE"); 5198 recordTrack->mState = TrackBase::ACTIVE; 5199 } else { 5200 ALOGV("active record track state %d", recordTrack->mState); 5201 } 5202 return status; 5203 } 5204 5205 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5206 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5207 // or using a separate command thread 5208 recordTrack->mState = TrackBase::STARTING_1; 5209 mActiveTracks.add(recordTrack); 5210 mActiveTracksGen++; 5211 mLock.unlock(); 5212 status_t status = AudioSystem::startInput(mId); 5213 mLock.lock(); 5214 // FIXME should verify that recordTrack is still in mActiveTracks 5215 if (status != NO_ERROR) { 5216 mActiveTracks.remove(recordTrack); 5217 mActiveTracksGen++; 5218 recordTrack->clearSyncStartEvent(); 5219 return status; 5220 } 5221 // Catch up with current buffer indices if thread is already running. 5222 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5223 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5224 // see previously buffered data before it called start(), but with greater risk of overrun. 5225 5226 recordTrack->mRsmpInFront = mRsmpInRear; 5227 recordTrack->mRsmpInUnrel = 0; 5228 // FIXME why reset? 5229 if (recordTrack->mResampler != NULL) { 5230 recordTrack->mResampler->reset(); 5231 } 5232 recordTrack->mState = TrackBase::STARTING_2; 5233 // signal thread to start 5234 mWaitWorkCV.broadcast(); 5235 if (mActiveTracks.indexOf(recordTrack) < 0) { 5236 ALOGV("Record failed to start"); 5237 status = BAD_VALUE; 5238 goto startError; 5239 } 5240 return status; 5241 } 5242 5243startError: 5244 AudioSystem::stopInput(mId); 5245 recordTrack->clearSyncStartEvent(); 5246 // FIXME I wonder why we do not reset the state here? 5247 return status; 5248} 5249 5250void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5251{ 5252 sp<SyncEvent> strongEvent = event.promote(); 5253 5254 if (strongEvent != 0) { 5255 sp<RefBase> ptr = strongEvent->cookie().promote(); 5256 if (ptr != 0) { 5257 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5258 recordTrack->handleSyncStartEvent(strongEvent); 5259 } 5260 } 5261} 5262 5263bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5264 ALOGV("RecordThread::stop"); 5265 AutoMutex _l(mLock); 5266 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5267 return false; 5268 } 5269 // note that threadLoop may still be processing the track at this point [without lock] 5270 recordTrack->mState = TrackBase::PAUSING; 5271 // do not wait for mStartStopCond if exiting 5272 if (exitPending()) { 5273 return true; 5274 } 5275 // FIXME incorrect usage of wait: no explicit predicate or loop 5276 mStartStopCond.wait(mLock); 5277 // if we have been restarted, recordTrack is in mActiveTracks here 5278 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5279 ALOGV("Record stopped OK"); 5280 return true; 5281 } 5282 return false; 5283} 5284 5285bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5286{ 5287 return false; 5288} 5289 5290status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5291{ 5292#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5293 if (!isValidSyncEvent(event)) { 5294 return BAD_VALUE; 5295 } 5296 5297 int eventSession = event->triggerSession(); 5298 status_t ret = NAME_NOT_FOUND; 5299 5300 Mutex::Autolock _l(mLock); 5301 5302 for (size_t i = 0; i < mTracks.size(); i++) { 5303 sp<RecordTrack> track = mTracks[i]; 5304 if (eventSession == track->sessionId()) { 5305 (void) track->setSyncEvent(event); 5306 ret = NO_ERROR; 5307 } 5308 } 5309 return ret; 5310#else 5311 return BAD_VALUE; 5312#endif 5313} 5314 5315// destroyTrack_l() must be called with ThreadBase::mLock held 5316void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5317{ 5318 track->terminate(); 5319 track->mState = TrackBase::STOPPED; 5320 // active tracks are removed by threadLoop() 5321 if (mActiveTracks.indexOf(track) < 0) { 5322 removeTrack_l(track); 5323 } 5324} 5325 5326void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5327{ 5328 mTracks.remove(track); 5329 // need anything related to effects here? 5330} 5331 5332void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5333{ 5334 dumpInternals(fd, args); 5335 dumpTracks(fd, args); 5336 dumpEffectChains(fd, args); 5337} 5338 5339void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5340{ 5341 fdprintf(fd, "\nInput thread %p:\n", this); 5342 5343 if (mActiveTracks.size() > 0) { 5344 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5345 } else { 5346 fdprintf(fd, " No active record clients\n"); 5347 } 5348 5349 dumpBase(fd, args); 5350} 5351 5352void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5353{ 5354 const size_t SIZE = 256; 5355 char buffer[SIZE]; 5356 String8 result; 5357 5358 size_t numtracks = mTracks.size(); 5359 size_t numactive = mActiveTracks.size(); 5360 size_t numactiveseen = 0; 5361 fdprintf(fd, " %d Tracks", numtracks); 5362 if (numtracks) { 5363 fdprintf(fd, " of which %d are active\n", numactive); 5364 RecordTrack::appendDumpHeader(result); 5365 for (size_t i = 0; i < numtracks ; ++i) { 5366 sp<RecordTrack> track = mTracks[i]; 5367 if (track != 0) { 5368 bool active = mActiveTracks.indexOf(track) >= 0; 5369 if (active) { 5370 numactiveseen++; 5371 } 5372 track->dump(buffer, SIZE, active); 5373 result.append(buffer); 5374 } 5375 } 5376 } else { 5377 fdprintf(fd, "\n"); 5378 } 5379 5380 if (numactiveseen != numactive) { 5381 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5382 " not in the track list\n"); 5383 result.append(buffer); 5384 RecordTrack::appendDumpHeader(result); 5385 for (size_t i = 0; i < numactive; ++i) { 5386 sp<RecordTrack> track = mActiveTracks[i]; 5387 if (mTracks.indexOf(track) < 0) { 5388 track->dump(buffer, SIZE, true); 5389 result.append(buffer); 5390 } 5391 } 5392 5393 } 5394 write(fd, result.string(), result.size()); 5395} 5396 5397// AudioBufferProvider interface 5398status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5399 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5400{ 5401 RecordTrack *activeTrack = mRecordTrack; 5402 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5403 if (threadBase == 0) { 5404 buffer->frameCount = 0; 5405 buffer->raw = NULL; 5406 return NOT_ENOUGH_DATA; 5407 } 5408 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5409 int32_t rear = recordThread->mRsmpInRear; 5410 int32_t front = activeTrack->mRsmpInFront; 5411 ssize_t filled = rear - front; 5412 // FIXME should not be P2 (don't want to increase latency) 5413 // FIXME if client not keeping up, discard 5414 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5415 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5416 front &= recordThread->mRsmpInFramesP2 - 1; 5417 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5418 if (part1 > (size_t) filled) { 5419 part1 = filled; 5420 } 5421 size_t ask = buffer->frameCount; 5422 ALOG_ASSERT(ask > 0); 5423 if (part1 > ask) { 5424 part1 = ask; 5425 } 5426 if (part1 == 0) { 5427 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5428 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5429 buffer->raw = NULL; 5430 buffer->frameCount = 0; 5431 activeTrack->mRsmpInUnrel = 0; 5432 return NOT_ENOUGH_DATA; 5433 } 5434 5435 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5436 buffer->frameCount = part1; 5437 activeTrack->mRsmpInUnrel = part1; 5438 return NO_ERROR; 5439} 5440 5441// AudioBufferProvider interface 5442void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5443 AudioBufferProvider::Buffer* buffer) 5444{ 5445 RecordTrack *activeTrack = mRecordTrack; 5446 size_t stepCount = buffer->frameCount; 5447 if (stepCount == 0) { 5448 return; 5449 } 5450 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5451 activeTrack->mRsmpInUnrel -= stepCount; 5452 activeTrack->mRsmpInFront += stepCount; 5453 buffer->raw = NULL; 5454 buffer->frameCount = 0; 5455} 5456 5457bool AudioFlinger::RecordThread::checkForNewParameters_l() 5458{ 5459 bool reconfig = false; 5460 5461 while (!mNewParameters.isEmpty()) { 5462 status_t status = NO_ERROR; 5463 String8 keyValuePair = mNewParameters[0]; 5464 AudioParameter param = AudioParameter(keyValuePair); 5465 int value; 5466 audio_format_t reqFormat = mFormat; 5467 uint32_t samplingRate = mSampleRate; 5468 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5469 5470 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5471 // channel count change can be requested. Do we mandate the first client defines the 5472 // HAL sampling rate and channel count or do we allow changes on the fly? 5473 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5474 samplingRate = value; 5475 reconfig = true; 5476 } 5477 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5478 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5479 status = BAD_VALUE; 5480 } else { 5481 reqFormat = (audio_format_t) value; 5482 reconfig = true; 5483 } 5484 } 5485 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5486 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5487 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5488 status = BAD_VALUE; 5489 } else { 5490 channelMask = mask; 5491 reconfig = true; 5492 } 5493 } 5494 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5495 // do not accept frame count changes if tracks are open as the track buffer 5496 // size depends on frame count and correct behavior would not be guaranteed 5497 // if frame count is changed after track creation 5498 if (mActiveTracks.size() > 0) { 5499 status = INVALID_OPERATION; 5500 } else { 5501 reconfig = true; 5502 } 5503 } 5504 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5505 // forward device change to effects that have requested to be 5506 // aware of attached audio device. 5507 for (size_t i = 0; i < mEffectChains.size(); i++) { 5508 mEffectChains[i]->setDevice_l(value); 5509 } 5510 5511 // store input device and output device but do not forward output device to audio HAL. 5512 // Note that status is ignored by the caller for output device 5513 // (see AudioFlinger::setParameters() 5514 if (audio_is_output_devices(value)) { 5515 mOutDevice = value; 5516 status = BAD_VALUE; 5517 } else { 5518 mInDevice = value; 5519 // disable AEC and NS if the device is a BT SCO headset supporting those 5520 // pre processings 5521 if (mTracks.size() > 0) { 5522 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5523 mAudioFlinger->btNrecIsOff(); 5524 for (size_t i = 0; i < mTracks.size(); i++) { 5525 sp<RecordTrack> track = mTracks[i]; 5526 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5527 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5528 } 5529 } 5530 } 5531 } 5532 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5533 mAudioSource != (audio_source_t)value) { 5534 // forward device change to effects that have requested to be 5535 // aware of attached audio device. 5536 for (size_t i = 0; i < mEffectChains.size(); i++) { 5537 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5538 } 5539 mAudioSource = (audio_source_t)value; 5540 } 5541 5542 if (status == NO_ERROR) { 5543 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5544 keyValuePair.string()); 5545 if (status == INVALID_OPERATION) { 5546 inputStandBy(); 5547 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5548 keyValuePair.string()); 5549 } 5550 if (reconfig) { 5551 if (status == BAD_VALUE && 5552 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5553 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5554 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5555 <= (2 * samplingRate)) && 5556 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5557 <= FCC_2 && 5558 (channelMask == AUDIO_CHANNEL_IN_MONO || 5559 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5560 status = NO_ERROR; 5561 } 5562 if (status == NO_ERROR) { 5563 readInputParameters_l(); 5564 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5565 } 5566 } 5567 } 5568 5569 mNewParameters.removeAt(0); 5570 5571 mParamStatus = status; 5572 mParamCond.signal(); 5573 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5574 // already timed out waiting for the status and will never signal the condition. 5575 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5576 } 5577 return reconfig; 5578} 5579 5580String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5581{ 5582 Mutex::Autolock _l(mLock); 5583 if (initCheck() != NO_ERROR) { 5584 return String8(); 5585 } 5586 5587 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5588 const String8 out_s8(s); 5589 free(s); 5590 return out_s8; 5591} 5592 5593void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5594 AudioSystem::OutputDescriptor desc; 5595 const void *param2 = NULL; 5596 5597 switch (event) { 5598 case AudioSystem::INPUT_OPENED: 5599 case AudioSystem::INPUT_CONFIG_CHANGED: 5600 desc.channelMask = mChannelMask; 5601 desc.samplingRate = mSampleRate; 5602 desc.format = mFormat; 5603 desc.frameCount = mFrameCount; 5604 desc.latency = 0; 5605 param2 = &desc; 5606 break; 5607 5608 case AudioSystem::INPUT_CLOSED: 5609 default: 5610 break; 5611 } 5612 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5613} 5614 5615void AudioFlinger::RecordThread::readInputParameters_l() 5616{ 5617 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5618 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5619 mChannelCount = popcount(mChannelMask); 5620 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5621 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5622 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5623 } 5624 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5625 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5626 mFrameCount = mBufferSize / mFrameSize; 5627 // This is the formula for calculating the temporary buffer size. 5628 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5629 // 1 full output buffer, regardless of the alignment of the available input. 5630 // The value is somewhat arbitrary, and could probably be even larger. 5631 // A larger value should allow more old data to be read after a track calls start(), 5632 // without increasing latency. 5633 mRsmpInFrames = mFrameCount * 7; 5634 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5635 delete[] mRsmpInBuffer; 5636 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5637 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5638 5639 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5640 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5641} 5642 5643uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5644{ 5645 Mutex::Autolock _l(mLock); 5646 if (initCheck() != NO_ERROR) { 5647 return 0; 5648 } 5649 5650 return mInput->stream->get_input_frames_lost(mInput->stream); 5651} 5652 5653uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5654{ 5655 Mutex::Autolock _l(mLock); 5656 uint32_t result = 0; 5657 if (getEffectChain_l(sessionId) != 0) { 5658 result = EFFECT_SESSION; 5659 } 5660 5661 for (size_t i = 0; i < mTracks.size(); ++i) { 5662 if (sessionId == mTracks[i]->sessionId()) { 5663 result |= TRACK_SESSION; 5664 break; 5665 } 5666 } 5667 5668 return result; 5669} 5670 5671KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5672{ 5673 KeyedVector<int, bool> ids; 5674 Mutex::Autolock _l(mLock); 5675 for (size_t j = 0; j < mTracks.size(); ++j) { 5676 sp<RecordThread::RecordTrack> track = mTracks[j]; 5677 int sessionId = track->sessionId(); 5678 if (ids.indexOfKey(sessionId) < 0) { 5679 ids.add(sessionId, true); 5680 } 5681 } 5682 return ids; 5683} 5684 5685AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5686{ 5687 Mutex::Autolock _l(mLock); 5688 AudioStreamIn *input = mInput; 5689 mInput = NULL; 5690 return input; 5691} 5692 5693// this method must always be called either with ThreadBase mLock held or inside the thread loop 5694audio_stream_t* AudioFlinger::RecordThread::stream() const 5695{ 5696 if (mInput == NULL) { 5697 return NULL; 5698 } 5699 return &mInput->stream->common; 5700} 5701 5702status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5703{ 5704 // only one chain per input thread 5705 if (mEffectChains.size() != 0) { 5706 return INVALID_OPERATION; 5707 } 5708 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5709 5710 chain->setInBuffer(NULL); 5711 chain->setOutBuffer(NULL); 5712 5713 checkSuspendOnAddEffectChain_l(chain); 5714 5715 mEffectChains.add(chain); 5716 5717 return NO_ERROR; 5718} 5719 5720size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5721{ 5722 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5723 ALOGW_IF(mEffectChains.size() != 1, 5724 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5725 chain.get(), mEffectChains.size(), this); 5726 if (mEffectChains.size() == 1) { 5727 mEffectChains.removeAt(0); 5728 } 5729 return 0; 5730} 5731 5732}; // namespace android 5733