Threads.cpp revision 3a6c90aa0617666d9abc94c02b752d9eb3d64772
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_MIN; 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 switch (mType) { 1344 1345 case DIRECT: 1346 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1347 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1348 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1349 "for output %p with format %#x", 1350 sampleRate, format, channelMask, mOutput, mFormat); 1351 lStatus = BAD_VALUE; 1352 goto Exit; 1353 } 1354 } 1355 break; 1356 1357 case OFFLOAD: 1358 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1359 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1360 "for output %p with format %#x", 1361 sampleRate, format, channelMask, mOutput, mFormat); 1362 lStatus = BAD_VALUE; 1363 goto Exit; 1364 } 1365 break; 1366 1367 default: 1368 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1369 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1370 "for output %p with format %#x", 1371 format, mOutput, mFormat); 1372 lStatus = BAD_VALUE; 1373 goto Exit; 1374 } 1375 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1376 if (sampleRate > mSampleRate*2) { 1377 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1378 lStatus = BAD_VALUE; 1379 goto Exit; 1380 } 1381 break; 1382 1383 } 1384 1385 lStatus = initCheck(); 1386 if (lStatus != NO_ERROR) { 1387 ALOGE("createTrack_l() audio driver not initialized"); 1388 goto Exit; 1389 } 1390 1391 { // scope for mLock 1392 Mutex::Autolock _l(mLock); 1393 1394 // all tracks in same audio session must share the same routing strategy otherwise 1395 // conflicts will happen when tracks are moved from one output to another by audio policy 1396 // manager 1397 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1398 for (size_t i = 0; i < mTracks.size(); ++i) { 1399 sp<Track> t = mTracks[i]; 1400 if (t != 0 && !t->isOutputTrack()) { 1401 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1402 if (sessionId == t->sessionId() && strategy != actual) { 1403 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1404 strategy, actual); 1405 lStatus = BAD_VALUE; 1406 goto Exit; 1407 } 1408 } 1409 } 1410 1411 if (!isTimed) { 1412 track = new Track(this, client, streamType, sampleRate, format, 1413 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1414 } else { 1415 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1416 channelMask, frameCount, sharedBuffer, sessionId, uid); 1417 } 1418 1419 // new Track always returns non-NULL, 1420 // but TimedTrack::create() is a factory that could fail by returning NULL 1421 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1422 if (lStatus != NO_ERROR) { 1423 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1424 // track must be cleared from the caller as the caller has the AF lock 1425 goto Exit; 1426 } 1427 mTracks.add(track); 1428 1429 sp<EffectChain> chain = getEffectChain_l(sessionId); 1430 if (chain != 0) { 1431 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1432 track->setMainBuffer(chain->inBuffer()); 1433 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1434 chain->incTrackCnt(); 1435 } 1436 1437 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1438 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1439 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1440 // so ask activity manager to do this on our behalf 1441 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1442 } 1443 } 1444 1445 lStatus = NO_ERROR; 1446 1447Exit: 1448 *status = lStatus; 1449 return track; 1450} 1451 1452uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1453{ 1454 return latency; 1455} 1456 1457uint32_t AudioFlinger::PlaybackThread::latency() const 1458{ 1459 Mutex::Autolock _l(mLock); 1460 return latency_l(); 1461} 1462uint32_t AudioFlinger::PlaybackThread::latency_l() const 1463{ 1464 if (initCheck() == NO_ERROR) { 1465 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1466 } else { 1467 return 0; 1468 } 1469} 1470 1471void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1472{ 1473 Mutex::Autolock _l(mLock); 1474 // Don't apply master volume in SW if our HAL can do it for us. 1475 if (mOutput && mOutput->audioHwDev && 1476 mOutput->audioHwDev->canSetMasterVolume()) { 1477 mMasterVolume = 1.0; 1478 } else { 1479 mMasterVolume = value; 1480 } 1481} 1482 1483void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1484{ 1485 Mutex::Autolock _l(mLock); 1486 // Don't apply master mute in SW if our HAL can do it for us. 1487 if (mOutput && mOutput->audioHwDev && 1488 mOutput->audioHwDev->canSetMasterMute()) { 1489 mMasterMute = false; 1490 } else { 1491 mMasterMute = muted; 1492 } 1493} 1494 1495void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1496{ 1497 Mutex::Autolock _l(mLock); 1498 mStreamTypes[stream].volume = value; 1499 broadcast_l(); 1500} 1501 1502void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1503{ 1504 Mutex::Autolock _l(mLock); 1505 mStreamTypes[stream].mute = muted; 1506 broadcast_l(); 1507} 1508 1509float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1510{ 1511 Mutex::Autolock _l(mLock); 1512 return mStreamTypes[stream].volume; 1513} 1514 1515// addTrack_l() must be called with ThreadBase::mLock held 1516status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1517{ 1518 status_t status = ALREADY_EXISTS; 1519 1520 // set retry count for buffer fill 1521 track->mRetryCount = kMaxTrackStartupRetries; 1522 if (mActiveTracks.indexOf(track) < 0) { 1523 // the track is newly added, make sure it fills up all its 1524 // buffers before playing. This is to ensure the client will 1525 // effectively get the latency it requested. 1526 if (!track->isOutputTrack()) { 1527 TrackBase::track_state state = track->mState; 1528 mLock.unlock(); 1529 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1530 mLock.lock(); 1531 // abort track was stopped/paused while we released the lock 1532 if (state != track->mState) { 1533 if (status == NO_ERROR) { 1534 mLock.unlock(); 1535 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1536 mLock.lock(); 1537 } 1538 return INVALID_OPERATION; 1539 } 1540 // abort if start is rejected by audio policy manager 1541 if (status != NO_ERROR) { 1542 return PERMISSION_DENIED; 1543 } 1544#ifdef ADD_BATTERY_DATA 1545 // to track the speaker usage 1546 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1547#endif 1548 } 1549 1550 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1551 track->mResetDone = false; 1552 track->mPresentationCompleteFrames = 0; 1553 mActiveTracks.add(track); 1554 mWakeLockUids.add(track->uid()); 1555 mActiveTracksGeneration++; 1556 mLatestActiveTrack = track; 1557 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1558 if (chain != 0) { 1559 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1560 track->sessionId()); 1561 chain->incActiveTrackCnt(); 1562 } 1563 1564 status = NO_ERROR; 1565 } 1566 1567 onAddNewTrack_l(); 1568 return status; 1569} 1570 1571bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1572{ 1573 track->terminate(); 1574 // active tracks are removed by threadLoop() 1575 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1576 track->mState = TrackBase::STOPPED; 1577 if (!trackActive) { 1578 removeTrack_l(track); 1579 } else if (track->isFastTrack() || track->isOffloaded()) { 1580 track->mState = TrackBase::STOPPING_1; 1581 } 1582 1583 return trackActive; 1584} 1585 1586void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1587{ 1588 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1589 mTracks.remove(track); 1590 deleteTrackName_l(track->name()); 1591 // redundant as track is about to be destroyed, for dumpsys only 1592 track->mName = -1; 1593 if (track->isFastTrack()) { 1594 int index = track->mFastIndex; 1595 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1596 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1597 mFastTrackAvailMask |= 1 << index; 1598 // redundant as track is about to be destroyed, for dumpsys only 1599 track->mFastIndex = -1; 1600 } 1601 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1602 if (chain != 0) { 1603 chain->decTrackCnt(); 1604 } 1605} 1606 1607void AudioFlinger::PlaybackThread::broadcast_l() 1608{ 1609 // Thread could be blocked waiting for async 1610 // so signal it to handle state changes immediately 1611 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1612 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1613 mSignalPending = true; 1614 mWaitWorkCV.broadcast(); 1615} 1616 1617String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1618{ 1619 Mutex::Autolock _l(mLock); 1620 if (initCheck() != NO_ERROR) { 1621 return String8(); 1622 } 1623 1624 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1625 const String8 out_s8(s); 1626 free(s); 1627 return out_s8; 1628} 1629 1630// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1631void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1632 AudioSystem::OutputDescriptor desc; 1633 void *param2 = NULL; 1634 1635 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1636 param); 1637 1638 switch (event) { 1639 case AudioSystem::OUTPUT_OPENED: 1640 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1641 desc.channelMask = mChannelMask; 1642 desc.samplingRate = mSampleRate; 1643 desc.format = mFormat; 1644 desc.frameCount = mNormalFrameCount; // FIXME see 1645 // AudioFlinger::frameCount(audio_io_handle_t) 1646 desc.latency = latency(); 1647 param2 = &desc; 1648 break; 1649 1650 case AudioSystem::STREAM_CONFIG_CHANGED: 1651 param2 = ¶m; 1652 case AudioSystem::OUTPUT_CLOSED: 1653 default: 1654 break; 1655 } 1656 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1657} 1658 1659void AudioFlinger::PlaybackThread::writeCallback() 1660{ 1661 ALOG_ASSERT(mCallbackThread != 0); 1662 mCallbackThread->resetWriteBlocked(); 1663} 1664 1665void AudioFlinger::PlaybackThread::drainCallback() 1666{ 1667 ALOG_ASSERT(mCallbackThread != 0); 1668 mCallbackThread->resetDraining(); 1669} 1670 1671void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1672{ 1673 Mutex::Autolock _l(mLock); 1674 // reject out of sequence requests 1675 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1676 mWriteAckSequence &= ~1; 1677 mWaitWorkCV.signal(); 1678 } 1679} 1680 1681void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1682{ 1683 Mutex::Autolock _l(mLock); 1684 // reject out of sequence requests 1685 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1686 mDrainSequence &= ~1; 1687 mWaitWorkCV.signal(); 1688 } 1689} 1690 1691// static 1692int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1693 void *param __unused, 1694 void *cookie) 1695{ 1696 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1697 ALOGV("asyncCallback() event %d", event); 1698 switch (event) { 1699 case STREAM_CBK_EVENT_WRITE_READY: 1700 me->writeCallback(); 1701 break; 1702 case STREAM_CBK_EVENT_DRAIN_READY: 1703 me->drainCallback(); 1704 break; 1705 default: 1706 ALOGW("asyncCallback() unknown event %d", event); 1707 break; 1708 } 1709 return 0; 1710} 1711 1712void AudioFlinger::PlaybackThread::readOutputParameters_l() 1713{ 1714 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1715 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1716 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1717 if (!audio_is_output_channel(mChannelMask)) { 1718 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1719 } 1720 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1721 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1722 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1723 } 1724 mChannelCount = popcount(mChannelMask); 1725 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1726 if (!audio_is_valid_format(mFormat)) { 1727 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1728 } 1729 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1730 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1731 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1732 } 1733 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1734 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1735 mFrameCount = mBufferSize / mFrameSize; 1736 if (mFrameCount & 15) { 1737 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1738 mFrameCount); 1739 } 1740 1741 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1742 (mOutput->stream->set_callback != NULL)) { 1743 if (mOutput->stream->set_callback(mOutput->stream, 1744 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1745 mUseAsyncWrite = true; 1746 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1747 } 1748 } 1749 1750 // Calculate size of normal sink buffer relative to the HAL output buffer size 1751 double multiplier = 1.0; 1752 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1753 kUseFastMixer == FastMixer_Dynamic)) { 1754 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1755 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1756 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1757 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1758 maxNormalFrameCount = maxNormalFrameCount & ~15; 1759 if (maxNormalFrameCount < minNormalFrameCount) { 1760 maxNormalFrameCount = minNormalFrameCount; 1761 } 1762 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1763 if (multiplier <= 1.0) { 1764 multiplier = 1.0; 1765 } else if (multiplier <= 2.0) { 1766 if (2 * mFrameCount <= maxNormalFrameCount) { 1767 multiplier = 2.0; 1768 } else { 1769 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1770 } 1771 } else { 1772 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1773 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1774 // track, but we sometimes have to do this to satisfy the maximum frame count 1775 // constraint) 1776 // FIXME this rounding up should not be done if no HAL SRC 1777 uint32_t truncMult = (uint32_t) multiplier; 1778 if ((truncMult & 1)) { 1779 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1780 ++truncMult; 1781 } 1782 } 1783 multiplier = (double) truncMult; 1784 } 1785 } 1786 mNormalFrameCount = multiplier * mFrameCount; 1787 // round up to nearest 16 frames to satisfy AudioMixer 1788 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1789 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1790 mNormalFrameCount); 1791 1792 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1793 // Originally this was int16_t[] array, need to remove legacy implications. 1794 free(mSinkBuffer); 1795 mSinkBuffer = NULL; 1796 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1797 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1798 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1799 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1800 1801 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1802 // drives the output. 1803 free(mMixerBuffer); 1804 mMixerBuffer = NULL; 1805 if (mMixerBufferEnabled) { 1806 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1807 mMixerBufferSize = mNormalFrameCount * mChannelCount 1808 * audio_bytes_per_sample(mMixerBufferFormat); 1809 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1810 } 1811 free(mEffectBuffer); 1812 mEffectBuffer = NULL; 1813 if (mEffectBufferEnabled) { 1814 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1815 mEffectBufferSize = mNormalFrameCount * mChannelCount 1816 * audio_bytes_per_sample(mEffectBufferFormat); 1817 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1818 } 1819 1820 // force reconfiguration of effect chains and engines to take new buffer size and audio 1821 // parameters into account 1822 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1823 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1824 // matter. 1825 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1826 Vector< sp<EffectChain> > effectChains = mEffectChains; 1827 for (size_t i = 0; i < effectChains.size(); i ++) { 1828 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1829 } 1830} 1831 1832 1833status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1834{ 1835 if (halFrames == NULL || dspFrames == NULL) { 1836 return BAD_VALUE; 1837 } 1838 Mutex::Autolock _l(mLock); 1839 if (initCheck() != NO_ERROR) { 1840 return INVALID_OPERATION; 1841 } 1842 size_t framesWritten = mBytesWritten / mFrameSize; 1843 *halFrames = framesWritten; 1844 1845 if (isSuspended()) { 1846 // return an estimation of rendered frames when the output is suspended 1847 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1848 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1849 return NO_ERROR; 1850 } else { 1851 status_t status; 1852 uint32_t frames; 1853 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1854 *dspFrames = (size_t)frames; 1855 return status; 1856 } 1857} 1858 1859uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1860{ 1861 Mutex::Autolock _l(mLock); 1862 uint32_t result = 0; 1863 if (getEffectChain_l(sessionId) != 0) { 1864 result = EFFECT_SESSION; 1865 } 1866 1867 for (size_t i = 0; i < mTracks.size(); ++i) { 1868 sp<Track> track = mTracks[i]; 1869 if (sessionId == track->sessionId() && !track->isInvalid()) { 1870 result |= TRACK_SESSION; 1871 break; 1872 } 1873 } 1874 1875 return result; 1876} 1877 1878uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1879{ 1880 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1881 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1882 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1883 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1884 } 1885 for (size_t i = 0; i < mTracks.size(); i++) { 1886 sp<Track> track = mTracks[i]; 1887 if (sessionId == track->sessionId() && !track->isInvalid()) { 1888 return AudioSystem::getStrategyForStream(track->streamType()); 1889 } 1890 } 1891 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1892} 1893 1894 1895AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1896{ 1897 Mutex::Autolock _l(mLock); 1898 return mOutput; 1899} 1900 1901AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1902{ 1903 Mutex::Autolock _l(mLock); 1904 AudioStreamOut *output = mOutput; 1905 mOutput = NULL; 1906 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1907 // must push a NULL and wait for ack 1908 mOutputSink.clear(); 1909 mPipeSink.clear(); 1910 mNormalSink.clear(); 1911 return output; 1912} 1913 1914// this method must always be called either with ThreadBase mLock held or inside the thread loop 1915audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1916{ 1917 if (mOutput == NULL) { 1918 return NULL; 1919 } 1920 return &mOutput->stream->common; 1921} 1922 1923uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1924{ 1925 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1926} 1927 1928status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1929{ 1930 if (!isValidSyncEvent(event)) { 1931 return BAD_VALUE; 1932 } 1933 1934 Mutex::Autolock _l(mLock); 1935 1936 for (size_t i = 0; i < mTracks.size(); ++i) { 1937 sp<Track> track = mTracks[i]; 1938 if (event->triggerSession() == track->sessionId()) { 1939 (void) track->setSyncEvent(event); 1940 return NO_ERROR; 1941 } 1942 } 1943 1944 return NAME_NOT_FOUND; 1945} 1946 1947bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1948{ 1949 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1950} 1951 1952void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1953 const Vector< sp<Track> >& tracksToRemove) 1954{ 1955 size_t count = tracksToRemove.size(); 1956 if (count > 0) { 1957 for (size_t i = 0 ; i < count ; i++) { 1958 const sp<Track>& track = tracksToRemove.itemAt(i); 1959 if (!track->isOutputTrack()) { 1960 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1961#ifdef ADD_BATTERY_DATA 1962 // to track the speaker usage 1963 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1964#endif 1965 if (track->isTerminated()) { 1966 AudioSystem::releaseOutput(mId); 1967 } 1968 } 1969 } 1970 } 1971} 1972 1973void AudioFlinger::PlaybackThread::checkSilentMode_l() 1974{ 1975 if (!mMasterMute) { 1976 char value[PROPERTY_VALUE_MAX]; 1977 if (property_get("ro.audio.silent", value, "0") > 0) { 1978 char *endptr; 1979 unsigned long ul = strtoul(value, &endptr, 0); 1980 if (*endptr == '\0' && ul != 0) { 1981 ALOGD("Silence is golden"); 1982 // The setprop command will not allow a property to be changed after 1983 // the first time it is set, so we don't have to worry about un-muting. 1984 setMasterMute_l(true); 1985 } 1986 } 1987 } 1988} 1989 1990// shared by MIXER and DIRECT, overridden by DUPLICATING 1991ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1992{ 1993 // FIXME rewrite to reduce number of system calls 1994 mLastWriteTime = systemTime(); 1995 mInWrite = true; 1996 ssize_t bytesWritten; 1997 const size_t offset = mCurrentWriteLength - mBytesRemaining; 1998 1999 // If an NBAIO sink is present, use it to write the normal mixer's submix 2000 if (mNormalSink != 0) { 2001 const size_t count = mBytesRemaining / mFrameSize; 2002 2003 ATRACE_BEGIN("write"); 2004 // update the setpoint when AudioFlinger::mScreenState changes 2005 uint32_t screenState = AudioFlinger::mScreenState; 2006 if (screenState != mScreenState) { 2007 mScreenState = screenState; 2008 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2009 if (pipe != NULL) { 2010 pipe->setAvgFrames((mScreenState & 1) ? 2011 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2012 } 2013 } 2014 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2015 ATRACE_END(); 2016 if (framesWritten > 0) { 2017 bytesWritten = framesWritten * mFrameSize; 2018 } else { 2019 bytesWritten = framesWritten; 2020 } 2021 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2022 if (status == NO_ERROR) { 2023 size_t totalFramesWritten = mNormalSink->framesWritten(); 2024 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2025 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2026 mLatchDValid = true; 2027 } 2028 } 2029 // otherwise use the HAL / AudioStreamOut directly 2030 } else { 2031 // Direct output and offload threads 2032 2033 if (mUseAsyncWrite) { 2034 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2035 mWriteAckSequence += 2; 2036 mWriteAckSequence |= 1; 2037 ALOG_ASSERT(mCallbackThread != 0); 2038 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2039 } 2040 // FIXME We should have an implementation of timestamps for direct output threads. 2041 // They are used e.g for multichannel PCM playback over HDMI. 2042 bytesWritten = mOutput->stream->write(mOutput->stream, 2043 (char *)mSinkBuffer + offset, mBytesRemaining); 2044 if (mUseAsyncWrite && 2045 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2046 // do not wait for async callback in case of error of full write 2047 mWriteAckSequence &= ~1; 2048 ALOG_ASSERT(mCallbackThread != 0); 2049 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2050 } 2051 } 2052 2053 mNumWrites++; 2054 mInWrite = false; 2055 mStandby = false; 2056 return bytesWritten; 2057} 2058 2059void AudioFlinger::PlaybackThread::threadLoop_drain() 2060{ 2061 if (mOutput->stream->drain) { 2062 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2063 if (mUseAsyncWrite) { 2064 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2065 mDrainSequence |= 1; 2066 ALOG_ASSERT(mCallbackThread != 0); 2067 mCallbackThread->setDraining(mDrainSequence); 2068 } 2069 mOutput->stream->drain(mOutput->stream, 2070 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2071 : AUDIO_DRAIN_ALL); 2072 } 2073} 2074 2075void AudioFlinger::PlaybackThread::threadLoop_exit() 2076{ 2077 // Default implementation has nothing to do 2078} 2079 2080/* 2081The derived values that are cached: 2082 - mSinkBufferSize from frame count * frame size 2083 - activeSleepTime from activeSleepTimeUs() 2084 - idleSleepTime from idleSleepTimeUs() 2085 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2086 - maxPeriod from frame count and sample rate (MIXER only) 2087 2088The parameters that affect these derived values are: 2089 - frame count 2090 - frame size 2091 - sample rate 2092 - device type: A2DP or not 2093 - device latency 2094 - format: PCM or not 2095 - active sleep time 2096 - idle sleep time 2097*/ 2098 2099void AudioFlinger::PlaybackThread::cacheParameters_l() 2100{ 2101 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2102 activeSleepTime = activeSleepTimeUs(); 2103 idleSleepTime = idleSleepTimeUs(); 2104} 2105 2106void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2107{ 2108 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2109 this, streamType, mTracks.size()); 2110 Mutex::Autolock _l(mLock); 2111 2112 size_t size = mTracks.size(); 2113 for (size_t i = 0; i < size; i++) { 2114 sp<Track> t = mTracks[i]; 2115 if (t->streamType() == streamType) { 2116 t->invalidate(); 2117 } 2118 } 2119} 2120 2121status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2122{ 2123 int session = chain->sessionId(); 2124 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2125 ? mEffectBuffer : mSinkBuffer); 2126 bool ownsBuffer = false; 2127 2128 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2129 if (session > 0) { 2130 // Only one effect chain can be present in direct output thread and it uses 2131 // the sink buffer as input 2132 if (mType != DIRECT) { 2133 size_t numSamples = mNormalFrameCount * mChannelCount; 2134 buffer = new int16_t[numSamples]; 2135 memset(buffer, 0, numSamples * sizeof(int16_t)); 2136 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2137 ownsBuffer = true; 2138 } 2139 2140 // Attach all tracks with same session ID to this chain. 2141 for (size_t i = 0; i < mTracks.size(); ++i) { 2142 sp<Track> track = mTracks[i]; 2143 if (session == track->sessionId()) { 2144 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2145 buffer); 2146 track->setMainBuffer(buffer); 2147 chain->incTrackCnt(); 2148 } 2149 } 2150 2151 // indicate all active tracks in the chain 2152 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2153 sp<Track> track = mActiveTracks[i].promote(); 2154 if (track == 0) { 2155 continue; 2156 } 2157 if (session == track->sessionId()) { 2158 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2159 chain->incActiveTrackCnt(); 2160 } 2161 } 2162 } 2163 2164 chain->setInBuffer(buffer, ownsBuffer); 2165 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2166 ? mEffectBuffer : mSinkBuffer)); 2167 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2168 // chains list in order to be processed last as it contains output stage effects 2169 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2170 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2171 // after track specific effects and before output stage 2172 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2173 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2174 // Effect chain for other sessions are inserted at beginning of effect 2175 // chains list to be processed before output mix effects. Relative order between other 2176 // sessions is not important 2177 size_t size = mEffectChains.size(); 2178 size_t i = 0; 2179 for (i = 0; i < size; i++) { 2180 if (mEffectChains[i]->sessionId() < session) { 2181 break; 2182 } 2183 } 2184 mEffectChains.insertAt(chain, i); 2185 checkSuspendOnAddEffectChain_l(chain); 2186 2187 return NO_ERROR; 2188} 2189 2190size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2191{ 2192 int session = chain->sessionId(); 2193 2194 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2195 2196 for (size_t i = 0; i < mEffectChains.size(); i++) { 2197 if (chain == mEffectChains[i]) { 2198 mEffectChains.removeAt(i); 2199 // detach all active tracks from the chain 2200 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2201 sp<Track> track = mActiveTracks[i].promote(); 2202 if (track == 0) { 2203 continue; 2204 } 2205 if (session == track->sessionId()) { 2206 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2207 chain.get(), session); 2208 chain->decActiveTrackCnt(); 2209 } 2210 } 2211 2212 // detach all tracks with same session ID from this chain 2213 for (size_t i = 0; i < mTracks.size(); ++i) { 2214 sp<Track> track = mTracks[i]; 2215 if (session == track->sessionId()) { 2216 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2217 chain->decTrackCnt(); 2218 } 2219 } 2220 break; 2221 } 2222 } 2223 return mEffectChains.size(); 2224} 2225 2226status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2227 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2228{ 2229 Mutex::Autolock _l(mLock); 2230 return attachAuxEffect_l(track, EffectId); 2231} 2232 2233status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2234 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2235{ 2236 status_t status = NO_ERROR; 2237 2238 if (EffectId == 0) { 2239 track->setAuxBuffer(0, NULL); 2240 } else { 2241 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2242 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2243 if (effect != 0) { 2244 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2245 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2246 } else { 2247 status = INVALID_OPERATION; 2248 } 2249 } else { 2250 status = BAD_VALUE; 2251 } 2252 } 2253 return status; 2254} 2255 2256void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2257{ 2258 for (size_t i = 0; i < mTracks.size(); ++i) { 2259 sp<Track> track = mTracks[i]; 2260 if (track->auxEffectId() == effectId) { 2261 attachAuxEffect_l(track, 0); 2262 } 2263 } 2264} 2265 2266bool AudioFlinger::PlaybackThread::threadLoop() 2267{ 2268 Vector< sp<Track> > tracksToRemove; 2269 2270 standbyTime = systemTime(); 2271 2272 // MIXER 2273 nsecs_t lastWarning = 0; 2274 2275 // DUPLICATING 2276 // FIXME could this be made local to while loop? 2277 writeFrames = 0; 2278 2279 int lastGeneration = 0; 2280 2281 cacheParameters_l(); 2282 sleepTime = idleSleepTime; 2283 2284 if (mType == MIXER) { 2285 sleepTimeShift = 0; 2286 } 2287 2288 CpuStats cpuStats; 2289 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2290 2291 acquireWakeLock(); 2292 2293 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2294 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2295 // and then that string will be logged at the next convenient opportunity. 2296 const char *logString = NULL; 2297 2298 checkSilentMode_l(); 2299 2300 while (!exitPending()) 2301 { 2302 cpuStats.sample(myName); 2303 2304 Vector< sp<EffectChain> > effectChains; 2305 2306 processConfigEvents(); 2307 2308 { // scope for mLock 2309 2310 Mutex::Autolock _l(mLock); 2311 2312 if (logString != NULL) { 2313 mNBLogWriter->logTimestamp(); 2314 mNBLogWriter->log(logString); 2315 logString = NULL; 2316 } 2317 2318 if (mLatchDValid) { 2319 mLatchQ = mLatchD; 2320 mLatchDValid = false; 2321 mLatchQValid = true; 2322 } 2323 2324 if (checkForNewParameters_l()) { 2325 cacheParameters_l(); 2326 } 2327 2328 saveOutputTracks(); 2329 if (mSignalPending) { 2330 // A signal was raised while we were unlocked 2331 mSignalPending = false; 2332 } else if (waitingAsyncCallback_l()) { 2333 if (exitPending()) { 2334 break; 2335 } 2336 releaseWakeLock_l(); 2337 mWakeLockUids.clear(); 2338 mActiveTracksGeneration++; 2339 ALOGV("wait async completion"); 2340 mWaitWorkCV.wait(mLock); 2341 ALOGV("async completion/wake"); 2342 acquireWakeLock_l(); 2343 standbyTime = systemTime() + standbyDelay; 2344 sleepTime = 0; 2345 2346 continue; 2347 } 2348 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2349 isSuspended()) { 2350 // put audio hardware into standby after short delay 2351 if (shouldStandby_l()) { 2352 2353 threadLoop_standby(); 2354 2355 mStandby = true; 2356 } 2357 2358 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2359 // we're about to wait, flush the binder command buffer 2360 IPCThreadState::self()->flushCommands(); 2361 2362 clearOutputTracks(); 2363 2364 if (exitPending()) { 2365 break; 2366 } 2367 2368 releaseWakeLock_l(); 2369 mWakeLockUids.clear(); 2370 mActiveTracksGeneration++; 2371 // wait until we have something to do... 2372 ALOGV("%s going to sleep", myName.string()); 2373 mWaitWorkCV.wait(mLock); 2374 ALOGV("%s waking up", myName.string()); 2375 acquireWakeLock_l(); 2376 2377 mMixerStatus = MIXER_IDLE; 2378 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2379 mBytesWritten = 0; 2380 mBytesRemaining = 0; 2381 checkSilentMode_l(); 2382 2383 standbyTime = systemTime() + standbyDelay; 2384 sleepTime = idleSleepTime; 2385 if (mType == MIXER) { 2386 sleepTimeShift = 0; 2387 } 2388 2389 continue; 2390 } 2391 } 2392 // mMixerStatusIgnoringFastTracks is also updated internally 2393 mMixerStatus = prepareTracks_l(&tracksToRemove); 2394 2395 // compare with previously applied list 2396 if (lastGeneration != mActiveTracksGeneration) { 2397 // update wakelock 2398 updateWakeLockUids_l(mWakeLockUids); 2399 lastGeneration = mActiveTracksGeneration; 2400 } 2401 2402 // prevent any changes in effect chain list and in each effect chain 2403 // during mixing and effect process as the audio buffers could be deleted 2404 // or modified if an effect is created or deleted 2405 lockEffectChains_l(effectChains); 2406 } // mLock scope ends 2407 2408 if (mBytesRemaining == 0) { 2409 mCurrentWriteLength = 0; 2410 if (mMixerStatus == MIXER_TRACKS_READY) { 2411 // threadLoop_mix() sets mCurrentWriteLength 2412 threadLoop_mix(); 2413 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2414 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2415 // threadLoop_sleepTime sets sleepTime to 0 if data 2416 // must be written to HAL 2417 threadLoop_sleepTime(); 2418 if (sleepTime == 0) { 2419 mCurrentWriteLength = mSinkBufferSize; 2420 } 2421 } 2422 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2423 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2424 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2425 // or mSinkBuffer (if there are no effects). 2426 // 2427 // This is done pre-effects computation; if effects change to 2428 // support higher precision, this needs to move. 2429 // 2430 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2431 // TODO use sleepTime == 0 as an additional condition. 2432 if (mMixerBufferValid) { 2433 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2434 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2435 2436 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2437 mNormalFrameCount * mChannelCount); 2438 } 2439 2440 mBytesRemaining = mCurrentWriteLength; 2441 if (isSuspended()) { 2442 sleepTime = suspendSleepTimeUs(); 2443 // simulate write to HAL when suspended 2444 mBytesWritten += mSinkBufferSize; 2445 mBytesRemaining = 0; 2446 } 2447 2448 // only process effects if we're going to write 2449 if (sleepTime == 0 && mType != OFFLOAD) { 2450 for (size_t i = 0; i < effectChains.size(); i ++) { 2451 effectChains[i]->process_l(); 2452 } 2453 } 2454 } 2455 // Process effect chains for offloaded thread even if no audio 2456 // was read from audio track: process only updates effect state 2457 // and thus does have to be synchronized with audio writes but may have 2458 // to be called while waiting for async write callback 2459 if (mType == OFFLOAD) { 2460 for (size_t i = 0; i < effectChains.size(); i ++) { 2461 effectChains[i]->process_l(); 2462 } 2463 } 2464 2465 // Only if the Effects buffer is enabled and there is data in the 2466 // Effects buffer (buffer valid), we need to 2467 // copy into the sink buffer. 2468 // TODO use sleepTime == 0 as an additional condition. 2469 if (mEffectBufferValid) { 2470 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2471 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2472 mNormalFrameCount * mChannelCount); 2473 } 2474 2475 // enable changes in effect chain 2476 unlockEffectChains(effectChains); 2477 2478 if (!waitingAsyncCallback()) { 2479 // sleepTime == 0 means we must write to audio hardware 2480 if (sleepTime == 0) { 2481 if (mBytesRemaining) { 2482 ssize_t ret = threadLoop_write(); 2483 if (ret < 0) { 2484 mBytesRemaining = 0; 2485 } else { 2486 mBytesWritten += ret; 2487 mBytesRemaining -= ret; 2488 } 2489 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2490 (mMixerStatus == MIXER_DRAIN_ALL)) { 2491 threadLoop_drain(); 2492 } 2493 if (mType == MIXER) { 2494 // write blocked detection 2495 nsecs_t now = systemTime(); 2496 nsecs_t delta = now - mLastWriteTime; 2497 if (!mStandby && delta > maxPeriod) { 2498 mNumDelayedWrites++; 2499 if ((now - lastWarning) > kWarningThrottleNs) { 2500 ATRACE_NAME("underrun"); 2501 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2502 ns2ms(delta), mNumDelayedWrites, this); 2503 lastWarning = now; 2504 } 2505 } 2506 } 2507 2508 } else { 2509 usleep(sleepTime); 2510 } 2511 } 2512 2513 // Finally let go of removed track(s), without the lock held 2514 // since we can't guarantee the destructors won't acquire that 2515 // same lock. This will also mutate and push a new fast mixer state. 2516 threadLoop_removeTracks(tracksToRemove); 2517 tracksToRemove.clear(); 2518 2519 // FIXME I don't understand the need for this here; 2520 // it was in the original code but maybe the 2521 // assignment in saveOutputTracks() makes this unnecessary? 2522 clearOutputTracks(); 2523 2524 // Effect chains will be actually deleted here if they were removed from 2525 // mEffectChains list during mixing or effects processing 2526 effectChains.clear(); 2527 2528 // FIXME Note that the above .clear() is no longer necessary since effectChains 2529 // is now local to this block, but will keep it for now (at least until merge done). 2530 } 2531 2532 threadLoop_exit(); 2533 2534 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2535 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2536 // put output stream into standby mode 2537 if (!mStandby) { 2538 mOutput->stream->common.standby(&mOutput->stream->common); 2539 } 2540 } 2541 2542 releaseWakeLock(); 2543 mWakeLockUids.clear(); 2544 mActiveTracksGeneration++; 2545 2546 ALOGV("Thread %p type %d exiting", this, mType); 2547 return false; 2548} 2549 2550// removeTracks_l() must be called with ThreadBase::mLock held 2551void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2552{ 2553 size_t count = tracksToRemove.size(); 2554 if (count > 0) { 2555 for (size_t i=0 ; i<count ; i++) { 2556 const sp<Track>& track = tracksToRemove.itemAt(i); 2557 mActiveTracks.remove(track); 2558 mWakeLockUids.remove(track->uid()); 2559 mActiveTracksGeneration++; 2560 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2561 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2562 if (chain != 0) { 2563 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2564 track->sessionId()); 2565 chain->decActiveTrackCnt(); 2566 } 2567 if (track->isTerminated()) { 2568 removeTrack_l(track); 2569 } 2570 } 2571 } 2572 2573} 2574 2575status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2576{ 2577 if (mNormalSink != 0) { 2578 return mNormalSink->getTimestamp(timestamp); 2579 } 2580 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2581 uint64_t position64; 2582 int ret = mOutput->stream->get_presentation_position( 2583 mOutput->stream, &position64, ×tamp.mTime); 2584 if (ret == 0) { 2585 timestamp.mPosition = (uint32_t)position64; 2586 return NO_ERROR; 2587 } 2588 } 2589 return INVALID_OPERATION; 2590} 2591// ---------------------------------------------------------------------------- 2592 2593AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2594 audio_io_handle_t id, audio_devices_t device, type_t type) 2595 : PlaybackThread(audioFlinger, output, id, device, type), 2596 // mAudioMixer below 2597 // mFastMixer below 2598 mFastMixerFutex(0) 2599 // mOutputSink below 2600 // mPipeSink below 2601 // mNormalSink below 2602{ 2603 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2604 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2605 "mFrameCount=%d, mNormalFrameCount=%d", 2606 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2607 mNormalFrameCount); 2608 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2609 2610 // FIXME - Current mixer implementation only supports stereo output 2611 if (mChannelCount != FCC_2) { 2612 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2613 } 2614 2615 // create an NBAIO sink for the HAL output stream, and negotiate 2616 mOutputSink = new AudioStreamOutSink(output->stream); 2617 size_t numCounterOffers = 0; 2618 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2619 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2620 ALOG_ASSERT(index == 0); 2621 2622 // initialize fast mixer depending on configuration 2623 bool initFastMixer; 2624 switch (kUseFastMixer) { 2625 case FastMixer_Never: 2626 initFastMixer = false; 2627 break; 2628 case FastMixer_Always: 2629 initFastMixer = true; 2630 break; 2631 case FastMixer_Static: 2632 case FastMixer_Dynamic: 2633 initFastMixer = mFrameCount < mNormalFrameCount; 2634 break; 2635 } 2636 if (initFastMixer) { 2637 2638 // create a MonoPipe to connect our submix to FastMixer 2639 NBAIO_Format format = mOutputSink->format(); 2640 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2641 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2642 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2643 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2644 const NBAIO_Format offers[1] = {format}; 2645 size_t numCounterOffers = 0; 2646 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2647 ALOG_ASSERT(index == 0); 2648 monoPipe->setAvgFrames((mScreenState & 1) ? 2649 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2650 mPipeSink = monoPipe; 2651 2652#ifdef TEE_SINK 2653 if (mTeeSinkOutputEnabled) { 2654 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2655 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2656 numCounterOffers = 0; 2657 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2658 ALOG_ASSERT(index == 0); 2659 mTeeSink = teeSink; 2660 PipeReader *teeSource = new PipeReader(*teeSink); 2661 numCounterOffers = 0; 2662 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2663 ALOG_ASSERT(index == 0); 2664 mTeeSource = teeSource; 2665 } 2666#endif 2667 2668 // create fast mixer and configure it initially with just one fast track for our submix 2669 mFastMixer = new FastMixer(); 2670 FastMixerStateQueue *sq = mFastMixer->sq(); 2671#ifdef STATE_QUEUE_DUMP 2672 sq->setObserverDump(&mStateQueueObserverDump); 2673 sq->setMutatorDump(&mStateQueueMutatorDump); 2674#endif 2675 FastMixerState *state = sq->begin(); 2676 FastTrack *fastTrack = &state->mFastTracks[0]; 2677 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2678 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2679 fastTrack->mVolumeProvider = NULL; 2680 fastTrack->mGeneration++; 2681 state->mFastTracksGen++; 2682 state->mTrackMask = 1; 2683 // fast mixer will use the HAL output sink 2684 state->mOutputSink = mOutputSink.get(); 2685 state->mOutputSinkGen++; 2686 state->mFrameCount = mFrameCount; 2687 state->mCommand = FastMixerState::COLD_IDLE; 2688 // already done in constructor initialization list 2689 //mFastMixerFutex = 0; 2690 state->mColdFutexAddr = &mFastMixerFutex; 2691 state->mColdGen++; 2692 state->mDumpState = &mFastMixerDumpState; 2693#ifdef TEE_SINK 2694 state->mTeeSink = mTeeSink.get(); 2695#endif 2696 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2697 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2698 sq->end(); 2699 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2700 2701 // start the fast mixer 2702 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2703 pid_t tid = mFastMixer->getTid(); 2704 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2705 if (err != 0) { 2706 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2707 kPriorityFastMixer, getpid_cached, tid, err); 2708 } 2709 2710#ifdef AUDIO_WATCHDOG 2711 // create and start the watchdog 2712 mAudioWatchdog = new AudioWatchdog(); 2713 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2714 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2715 tid = mAudioWatchdog->getTid(); 2716 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2717 if (err != 0) { 2718 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2719 kPriorityFastMixer, getpid_cached, tid, err); 2720 } 2721#endif 2722 2723 } else { 2724 mFastMixer = NULL; 2725 } 2726 2727 switch (kUseFastMixer) { 2728 case FastMixer_Never: 2729 case FastMixer_Dynamic: 2730 mNormalSink = mOutputSink; 2731 break; 2732 case FastMixer_Always: 2733 mNormalSink = mPipeSink; 2734 break; 2735 case FastMixer_Static: 2736 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2737 break; 2738 } 2739} 2740 2741AudioFlinger::MixerThread::~MixerThread() 2742{ 2743 if (mFastMixer != NULL) { 2744 FastMixerStateQueue *sq = mFastMixer->sq(); 2745 FastMixerState *state = sq->begin(); 2746 if (state->mCommand == FastMixerState::COLD_IDLE) { 2747 int32_t old = android_atomic_inc(&mFastMixerFutex); 2748 if (old == -1) { 2749 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2750 } 2751 } 2752 state->mCommand = FastMixerState::EXIT; 2753 sq->end(); 2754 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2755 mFastMixer->join(); 2756 // Though the fast mixer thread has exited, it's state queue is still valid. 2757 // We'll use that extract the final state which contains one remaining fast track 2758 // corresponding to our sub-mix. 2759 state = sq->begin(); 2760 ALOG_ASSERT(state->mTrackMask == 1); 2761 FastTrack *fastTrack = &state->mFastTracks[0]; 2762 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2763 delete fastTrack->mBufferProvider; 2764 sq->end(false /*didModify*/); 2765 delete mFastMixer; 2766#ifdef AUDIO_WATCHDOG 2767 if (mAudioWatchdog != 0) { 2768 mAudioWatchdog->requestExit(); 2769 mAudioWatchdog->requestExitAndWait(); 2770 mAudioWatchdog.clear(); 2771 } 2772#endif 2773 } 2774 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2775 delete mAudioMixer; 2776} 2777 2778 2779uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2780{ 2781 if (mFastMixer != NULL) { 2782 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2783 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2784 } 2785 return latency; 2786} 2787 2788 2789void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2790{ 2791 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2792} 2793 2794ssize_t AudioFlinger::MixerThread::threadLoop_write() 2795{ 2796 // FIXME we should only do one push per cycle; confirm this is true 2797 // Start the fast mixer if it's not already running 2798 if (mFastMixer != NULL) { 2799 FastMixerStateQueue *sq = mFastMixer->sq(); 2800 FastMixerState *state = sq->begin(); 2801 if (state->mCommand != FastMixerState::MIX_WRITE && 2802 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2803 if (state->mCommand == FastMixerState::COLD_IDLE) { 2804 int32_t old = android_atomic_inc(&mFastMixerFutex); 2805 if (old == -1) { 2806 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2807 } 2808#ifdef AUDIO_WATCHDOG 2809 if (mAudioWatchdog != 0) { 2810 mAudioWatchdog->resume(); 2811 } 2812#endif 2813 } 2814 state->mCommand = FastMixerState::MIX_WRITE; 2815 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2816 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2817 sq->end(); 2818 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2819 if (kUseFastMixer == FastMixer_Dynamic) { 2820 mNormalSink = mPipeSink; 2821 } 2822 } else { 2823 sq->end(false /*didModify*/); 2824 } 2825 } 2826 return PlaybackThread::threadLoop_write(); 2827} 2828 2829void AudioFlinger::MixerThread::threadLoop_standby() 2830{ 2831 // Idle the fast mixer if it's currently running 2832 if (mFastMixer != NULL) { 2833 FastMixerStateQueue *sq = mFastMixer->sq(); 2834 FastMixerState *state = sq->begin(); 2835 if (!(state->mCommand & FastMixerState::IDLE)) { 2836 state->mCommand = FastMixerState::COLD_IDLE; 2837 state->mColdFutexAddr = &mFastMixerFutex; 2838 state->mColdGen++; 2839 mFastMixerFutex = 0; 2840 sq->end(); 2841 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2842 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2843 if (kUseFastMixer == FastMixer_Dynamic) { 2844 mNormalSink = mOutputSink; 2845 } 2846#ifdef AUDIO_WATCHDOG 2847 if (mAudioWatchdog != 0) { 2848 mAudioWatchdog->pause(); 2849 } 2850#endif 2851 } else { 2852 sq->end(false /*didModify*/); 2853 } 2854 } 2855 PlaybackThread::threadLoop_standby(); 2856} 2857 2858bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2859{ 2860 return false; 2861} 2862 2863bool AudioFlinger::PlaybackThread::shouldStandby_l() 2864{ 2865 return !mStandby; 2866} 2867 2868bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2869{ 2870 Mutex::Autolock _l(mLock); 2871 return waitingAsyncCallback_l(); 2872} 2873 2874// shared by MIXER and DIRECT, overridden by DUPLICATING 2875void AudioFlinger::PlaybackThread::threadLoop_standby() 2876{ 2877 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2878 mOutput->stream->common.standby(&mOutput->stream->common); 2879 if (mUseAsyncWrite != 0) { 2880 // discard any pending drain or write ack by incrementing sequence 2881 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2882 mDrainSequence = (mDrainSequence + 2) & ~1; 2883 ALOG_ASSERT(mCallbackThread != 0); 2884 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2885 mCallbackThread->setDraining(mDrainSequence); 2886 } 2887} 2888 2889void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2890{ 2891 ALOGV("signal playback thread"); 2892 broadcast_l(); 2893} 2894 2895void AudioFlinger::MixerThread::threadLoop_mix() 2896{ 2897 // obtain the presentation timestamp of the next output buffer 2898 int64_t pts; 2899 status_t status = INVALID_OPERATION; 2900 2901 if (mNormalSink != 0) { 2902 status = mNormalSink->getNextWriteTimestamp(&pts); 2903 } else { 2904 status = mOutputSink->getNextWriteTimestamp(&pts); 2905 } 2906 2907 if (status != NO_ERROR) { 2908 pts = AudioBufferProvider::kInvalidPTS; 2909 } 2910 2911 // mix buffers... 2912 mAudioMixer->process(pts); 2913 mCurrentWriteLength = mSinkBufferSize; 2914 // increase sleep time progressively when application underrun condition clears. 2915 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2916 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2917 // such that we would underrun the audio HAL. 2918 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2919 sleepTimeShift--; 2920 } 2921 sleepTime = 0; 2922 standbyTime = systemTime() + standbyDelay; 2923 //TODO: delay standby when effects have a tail 2924} 2925 2926void AudioFlinger::MixerThread::threadLoop_sleepTime() 2927{ 2928 // If no tracks are ready, sleep once for the duration of an output 2929 // buffer size, then write 0s to the output 2930 if (sleepTime == 0) { 2931 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2932 sleepTime = activeSleepTime >> sleepTimeShift; 2933 if (sleepTime < kMinThreadSleepTimeUs) { 2934 sleepTime = kMinThreadSleepTimeUs; 2935 } 2936 // reduce sleep time in case of consecutive application underruns to avoid 2937 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2938 // duration we would end up writing less data than needed by the audio HAL if 2939 // the condition persists. 2940 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2941 sleepTimeShift++; 2942 } 2943 } else { 2944 sleepTime = idleSleepTime; 2945 } 2946 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2947 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2948 // before effects processing or output. 2949 if (mMixerBufferValid) { 2950 memset(mMixerBuffer, 0, mMixerBufferSize); 2951 } else { 2952 memset(mSinkBuffer, 0, mSinkBufferSize); 2953 } 2954 sleepTime = 0; 2955 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2956 "anticipated start"); 2957 } 2958 // TODO add standby time extension fct of effect tail 2959} 2960 2961// prepareTracks_l() must be called with ThreadBase::mLock held 2962AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2963 Vector< sp<Track> > *tracksToRemove) 2964{ 2965 2966 mixer_state mixerStatus = MIXER_IDLE; 2967 // find out which tracks need to be processed 2968 size_t count = mActiveTracks.size(); 2969 size_t mixedTracks = 0; 2970 size_t tracksWithEffect = 0; 2971 // counts only _active_ fast tracks 2972 size_t fastTracks = 0; 2973 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2974 2975 float masterVolume = mMasterVolume; 2976 bool masterMute = mMasterMute; 2977 2978 if (masterMute) { 2979 masterVolume = 0; 2980 } 2981 // Delegate master volume control to effect in output mix effect chain if needed 2982 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2983 if (chain != 0) { 2984 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2985 chain->setVolume_l(&v, &v); 2986 masterVolume = (float)((v + (1 << 23)) >> 24); 2987 chain.clear(); 2988 } 2989 2990 // prepare a new state to push 2991 FastMixerStateQueue *sq = NULL; 2992 FastMixerState *state = NULL; 2993 bool didModify = false; 2994 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2995 if (mFastMixer != NULL) { 2996 sq = mFastMixer->sq(); 2997 state = sq->begin(); 2998 } 2999 3000 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3001 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3002 3003 for (size_t i=0 ; i<count ; i++) { 3004 const sp<Track> t = mActiveTracks[i].promote(); 3005 if (t == 0) { 3006 continue; 3007 } 3008 3009 // this const just means the local variable doesn't change 3010 Track* const track = t.get(); 3011 3012 // process fast tracks 3013 if (track->isFastTrack()) { 3014 3015 // It's theoretically possible (though unlikely) for a fast track to be created 3016 // and then removed within the same normal mix cycle. This is not a problem, as 3017 // the track never becomes active so it's fast mixer slot is never touched. 3018 // The converse, of removing an (active) track and then creating a new track 3019 // at the identical fast mixer slot within the same normal mix cycle, 3020 // is impossible because the slot isn't marked available until the end of each cycle. 3021 int j = track->mFastIndex; 3022 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3023 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3024 FastTrack *fastTrack = &state->mFastTracks[j]; 3025 3026 // Determine whether the track is currently in underrun condition, 3027 // and whether it had a recent underrun. 3028 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3029 FastTrackUnderruns underruns = ftDump->mUnderruns; 3030 uint32_t recentFull = (underruns.mBitFields.mFull - 3031 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3032 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3033 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3034 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3035 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3036 uint32_t recentUnderruns = recentPartial + recentEmpty; 3037 track->mObservedUnderruns = underruns; 3038 // don't count underruns that occur while stopping or pausing 3039 // or stopped which can occur when flush() is called while active 3040 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3041 recentUnderruns > 0) { 3042 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3043 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3044 } 3045 3046 // This is similar to the state machine for normal tracks, 3047 // with a few modifications for fast tracks. 3048 bool isActive = true; 3049 switch (track->mState) { 3050 case TrackBase::STOPPING_1: 3051 // track stays active in STOPPING_1 state until first underrun 3052 if (recentUnderruns > 0 || track->isTerminated()) { 3053 track->mState = TrackBase::STOPPING_2; 3054 } 3055 break; 3056 case TrackBase::PAUSING: 3057 // ramp down is not yet implemented 3058 track->setPaused(); 3059 break; 3060 case TrackBase::RESUMING: 3061 // ramp up is not yet implemented 3062 track->mState = TrackBase::ACTIVE; 3063 break; 3064 case TrackBase::ACTIVE: 3065 if (recentFull > 0 || recentPartial > 0) { 3066 // track has provided at least some frames recently: reset retry count 3067 track->mRetryCount = kMaxTrackRetries; 3068 } 3069 if (recentUnderruns == 0) { 3070 // no recent underruns: stay active 3071 break; 3072 } 3073 // there has recently been an underrun of some kind 3074 if (track->sharedBuffer() == 0) { 3075 // were any of the recent underruns "empty" (no frames available)? 3076 if (recentEmpty == 0) { 3077 // no, then ignore the partial underruns as they are allowed indefinitely 3078 break; 3079 } 3080 // there has recently been an "empty" underrun: decrement the retry counter 3081 if (--(track->mRetryCount) > 0) { 3082 break; 3083 } 3084 // indicate to client process that the track was disabled because of underrun; 3085 // it will then automatically call start() when data is available 3086 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3087 // remove from active list, but state remains ACTIVE [confusing but true] 3088 isActive = false; 3089 break; 3090 } 3091 // fall through 3092 case TrackBase::STOPPING_2: 3093 case TrackBase::PAUSED: 3094 case TrackBase::STOPPED: 3095 case TrackBase::FLUSHED: // flush() while active 3096 // Check for presentation complete if track is inactive 3097 // We have consumed all the buffers of this track. 3098 // This would be incomplete if we auto-paused on underrun 3099 { 3100 size_t audioHALFrames = 3101 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3102 size_t framesWritten = mBytesWritten / mFrameSize; 3103 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3104 // track stays in active list until presentation is complete 3105 break; 3106 } 3107 } 3108 if (track->isStopping_2()) { 3109 track->mState = TrackBase::STOPPED; 3110 } 3111 if (track->isStopped()) { 3112 // Can't reset directly, as fast mixer is still polling this track 3113 // track->reset(); 3114 // So instead mark this track as needing to be reset after push with ack 3115 resetMask |= 1 << i; 3116 } 3117 isActive = false; 3118 break; 3119 case TrackBase::IDLE: 3120 default: 3121 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3122 } 3123 3124 if (isActive) { 3125 // was it previously inactive? 3126 if (!(state->mTrackMask & (1 << j))) { 3127 ExtendedAudioBufferProvider *eabp = track; 3128 VolumeProvider *vp = track; 3129 fastTrack->mBufferProvider = eabp; 3130 fastTrack->mVolumeProvider = vp; 3131 fastTrack->mChannelMask = track->mChannelMask; 3132 fastTrack->mGeneration++; 3133 state->mTrackMask |= 1 << j; 3134 didModify = true; 3135 // no acknowledgement required for newly active tracks 3136 } 3137 // cache the combined master volume and stream type volume for fast mixer; this 3138 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3139 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3140 ++fastTracks; 3141 } else { 3142 // was it previously active? 3143 if (state->mTrackMask & (1 << j)) { 3144 fastTrack->mBufferProvider = NULL; 3145 fastTrack->mGeneration++; 3146 state->mTrackMask &= ~(1 << j); 3147 didModify = true; 3148 // If any fast tracks were removed, we must wait for acknowledgement 3149 // because we're about to decrement the last sp<> on those tracks. 3150 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3151 } else { 3152 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3153 } 3154 tracksToRemove->add(track); 3155 // Avoids a misleading display in dumpsys 3156 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3157 } 3158 continue; 3159 } 3160 3161 { // local variable scope to avoid goto warning 3162 3163 audio_track_cblk_t* cblk = track->cblk(); 3164 3165 // The first time a track is added we wait 3166 // for all its buffers to be filled before processing it 3167 int name = track->name(); 3168 // make sure that we have enough frames to mix one full buffer. 3169 // enforce this condition only once to enable draining the buffer in case the client 3170 // app does not call stop() and relies on underrun to stop: 3171 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3172 // during last round 3173 size_t desiredFrames; 3174 uint32_t sr = track->sampleRate(); 3175 if (sr == mSampleRate) { 3176 desiredFrames = mNormalFrameCount; 3177 } else { 3178 // +1 for rounding and +1 for additional sample needed for interpolation 3179 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3180 // add frames already consumed but not yet released by the resampler 3181 // because mAudioTrackServerProxy->framesReady() will include these frames 3182 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3183#if 0 3184 // the minimum track buffer size is normally twice the number of frames necessary 3185 // to fill one buffer and the resampler should not leave more than one buffer worth 3186 // of unreleased frames after each pass, but just in case... 3187 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3188#endif 3189 } 3190 uint32_t minFrames = 1; 3191 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3192 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3193 minFrames = desiredFrames; 3194 } 3195 3196 size_t framesReady = track->framesReady(); 3197 if ((framesReady >= minFrames) && track->isReady() && 3198 !track->isPaused() && !track->isTerminated()) 3199 { 3200 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3201 3202 mixedTracks++; 3203 3204 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3205 // there is an effect chain connected to the track 3206 chain.clear(); 3207 if (track->mainBuffer() != mSinkBuffer && 3208 track->mainBuffer() != mMixerBuffer) { 3209 if (mEffectBufferEnabled) { 3210 mEffectBufferValid = true; // Later can set directly. 3211 } 3212 chain = getEffectChain_l(track->sessionId()); 3213 // Delegate volume control to effect in track effect chain if needed 3214 if (chain != 0) { 3215 tracksWithEffect++; 3216 } else { 3217 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3218 "session %d", 3219 name, track->sessionId()); 3220 } 3221 } 3222 3223 3224 int param = AudioMixer::VOLUME; 3225 if (track->mFillingUpStatus == Track::FS_FILLED) { 3226 // no ramp for the first volume setting 3227 track->mFillingUpStatus = Track::FS_ACTIVE; 3228 if (track->mState == TrackBase::RESUMING) { 3229 track->mState = TrackBase::ACTIVE; 3230 param = AudioMixer::RAMP_VOLUME; 3231 } 3232 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3233 // FIXME should not make a decision based on mServer 3234 } else if (cblk->mServer != 0) { 3235 // If the track is stopped before the first frame was mixed, 3236 // do not apply ramp 3237 param = AudioMixer::RAMP_VOLUME; 3238 } 3239 3240 // compute volume for this track 3241 uint32_t vl, vr, va; 3242 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3243 vl = vr = va = 0; 3244 if (track->isPausing()) { 3245 track->setPaused(); 3246 } 3247 } else { 3248 3249 // read original volumes with volume control 3250 float typeVolume = mStreamTypes[track->streamType()].volume; 3251 float v = masterVolume * typeVolume; 3252 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3253 uint32_t vlr = proxy->getVolumeLR(); 3254 vl = vlr & 0xFFFF; 3255 vr = vlr >> 16; 3256 // track volumes come from shared memory, so can't be trusted and must be clamped 3257 if (vl > MAX_GAIN_INT) { 3258 ALOGV("Track left volume out of range: %04X", vl); 3259 vl = MAX_GAIN_INT; 3260 } 3261 if (vr > MAX_GAIN_INT) { 3262 ALOGV("Track right volume out of range: %04X", vr); 3263 vr = MAX_GAIN_INT; 3264 } 3265 // now apply the master volume and stream type volume 3266 vl = (uint32_t)(v * vl) << 12; 3267 vr = (uint32_t)(v * vr) << 12; 3268 // assuming master volume and stream type volume each go up to 1.0, 3269 // vl and vr are now in 8.24 format 3270 3271 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3272 // send level comes from shared memory and so may be corrupt 3273 if (sendLevel > MAX_GAIN_INT) { 3274 ALOGV("Track send level out of range: %04X", sendLevel); 3275 sendLevel = MAX_GAIN_INT; 3276 } 3277 va = (uint32_t)(v * sendLevel); 3278 } 3279 3280 // Delegate volume control to effect in track effect chain if needed 3281 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3282 // Do not ramp volume if volume is controlled by effect 3283 param = AudioMixer::VOLUME; 3284 track->mHasVolumeController = true; 3285 } else { 3286 // force no volume ramp when volume controller was just disabled or removed 3287 // from effect chain to avoid volume spike 3288 if (track->mHasVolumeController) { 3289 param = AudioMixer::VOLUME; 3290 } 3291 track->mHasVolumeController = false; 3292 } 3293 3294 // Convert volumes from 8.24 to 4.12 format 3295 // This additional clamping is needed in case chain->setVolume_l() overshot 3296 vl = (vl + (1 << 11)) >> 12; 3297 if (vl > MAX_GAIN_INT) { 3298 vl = MAX_GAIN_INT; 3299 } 3300 vr = (vr + (1 << 11)) >> 12; 3301 if (vr > MAX_GAIN_INT) { 3302 vr = MAX_GAIN_INT; 3303 } 3304 3305 if (va > MAX_GAIN_INT) { 3306 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3307 } 3308 3309 // XXX: these things DON'T need to be done each time 3310 mAudioMixer->setBufferProvider(name, track); 3311 mAudioMixer->enable(name); 3312 3313 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3314 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3315 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3316 mAudioMixer->setParameter( 3317 name, 3318 AudioMixer::TRACK, 3319 AudioMixer::FORMAT, (void *)track->format()); 3320 mAudioMixer->setParameter( 3321 name, 3322 AudioMixer::TRACK, 3323 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3324 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3325 uint32_t maxSampleRate = mSampleRate * 2; 3326 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3327 if (reqSampleRate == 0) { 3328 reqSampleRate = mSampleRate; 3329 } else if (reqSampleRate > maxSampleRate) { 3330 reqSampleRate = maxSampleRate; 3331 } 3332 mAudioMixer->setParameter( 3333 name, 3334 AudioMixer::RESAMPLE, 3335 AudioMixer::SAMPLE_RATE, 3336 (void *)(uintptr_t)reqSampleRate); 3337 /* 3338 * Select the appropriate output buffer for the track. 3339 * 3340 * Tracks with effects go into their own effects chain buffer 3341 * and from there into either mEffectBuffer or mSinkBuffer. 3342 * 3343 * Other tracks can use mMixerBuffer for higher precision 3344 * channel accumulation. If this buffer is enabled 3345 * (mMixerBufferEnabled true), then selected tracks will accumulate 3346 * into it. 3347 * 3348 */ 3349 if (mMixerBufferEnabled 3350 && (track->mainBuffer() == mSinkBuffer 3351 || track->mainBuffer() == mMixerBuffer)) { 3352 mAudioMixer->setParameter( 3353 name, 3354 AudioMixer::TRACK, 3355 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3356 mAudioMixer->setParameter( 3357 name, 3358 AudioMixer::TRACK, 3359 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3360 // TODO: override track->mainBuffer()? 3361 mMixerBufferValid = true; 3362 } else { 3363 mAudioMixer->setParameter( 3364 name, 3365 AudioMixer::TRACK, 3366 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3367 mAudioMixer->setParameter( 3368 name, 3369 AudioMixer::TRACK, 3370 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3371 } 3372 mAudioMixer->setParameter( 3373 name, 3374 AudioMixer::TRACK, 3375 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3376 3377 // reset retry count 3378 track->mRetryCount = kMaxTrackRetries; 3379 3380 // If one track is ready, set the mixer ready if: 3381 // - the mixer was not ready during previous round OR 3382 // - no other track is not ready 3383 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3384 mixerStatus != MIXER_TRACKS_ENABLED) { 3385 mixerStatus = MIXER_TRACKS_READY; 3386 } 3387 } else { 3388 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3389 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3390 } 3391 // clear effect chain input buffer if an active track underruns to avoid sending 3392 // previous audio buffer again to effects 3393 chain = getEffectChain_l(track->sessionId()); 3394 if (chain != 0) { 3395 chain->clearInputBuffer(); 3396 } 3397 3398 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3399 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3400 track->isStopped() || track->isPaused()) { 3401 // We have consumed all the buffers of this track. 3402 // Remove it from the list of active tracks. 3403 // TODO: use actual buffer filling status instead of latency when available from 3404 // audio HAL 3405 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3406 size_t framesWritten = mBytesWritten / mFrameSize; 3407 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3408 if (track->isStopped()) { 3409 track->reset(); 3410 } 3411 tracksToRemove->add(track); 3412 } 3413 } else { 3414 // No buffers for this track. Give it a few chances to 3415 // fill a buffer, then remove it from active list. 3416 if (--(track->mRetryCount) <= 0) { 3417 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3418 tracksToRemove->add(track); 3419 // indicate to client process that the track was disabled because of underrun; 3420 // it will then automatically call start() when data is available 3421 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3422 // If one track is not ready, mark the mixer also not ready if: 3423 // - the mixer was ready during previous round OR 3424 // - no other track is ready 3425 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3426 mixerStatus != MIXER_TRACKS_READY) { 3427 mixerStatus = MIXER_TRACKS_ENABLED; 3428 } 3429 } 3430 mAudioMixer->disable(name); 3431 } 3432 3433 } // local variable scope to avoid goto warning 3434track_is_ready: ; 3435 3436 } 3437 3438 // Push the new FastMixer state if necessary 3439 bool pauseAudioWatchdog = false; 3440 if (didModify) { 3441 state->mFastTracksGen++; 3442 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3443 if (kUseFastMixer == FastMixer_Dynamic && 3444 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3445 state->mCommand = FastMixerState::COLD_IDLE; 3446 state->mColdFutexAddr = &mFastMixerFutex; 3447 state->mColdGen++; 3448 mFastMixerFutex = 0; 3449 if (kUseFastMixer == FastMixer_Dynamic) { 3450 mNormalSink = mOutputSink; 3451 } 3452 // If we go into cold idle, need to wait for acknowledgement 3453 // so that fast mixer stops doing I/O. 3454 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3455 pauseAudioWatchdog = true; 3456 } 3457 } 3458 if (sq != NULL) { 3459 sq->end(didModify); 3460 sq->push(block); 3461 } 3462#ifdef AUDIO_WATCHDOG 3463 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3464 mAudioWatchdog->pause(); 3465 } 3466#endif 3467 3468 // Now perform the deferred reset on fast tracks that have stopped 3469 while (resetMask != 0) { 3470 size_t i = __builtin_ctz(resetMask); 3471 ALOG_ASSERT(i < count); 3472 resetMask &= ~(1 << i); 3473 sp<Track> t = mActiveTracks[i].promote(); 3474 if (t == 0) { 3475 continue; 3476 } 3477 Track* track = t.get(); 3478 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3479 track->reset(); 3480 } 3481 3482 // remove all the tracks that need to be... 3483 removeTracks_l(*tracksToRemove); 3484 3485 // sink or mix buffer must be cleared if all tracks are connected to an 3486 // effect chain as in this case the mixer will not write to the sink or mix buffer 3487 // and track effects will accumulate into it 3488 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3489 (mixedTracks == 0 && fastTracks > 0))) { 3490 // FIXME as a performance optimization, should remember previous zero status 3491 if (mMixerBufferValid) { 3492 memset(mMixerBuffer, 0, mMixerBufferSize); 3493 // TODO: In testing, mSinkBuffer below need not be cleared because 3494 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3495 // after mixing. 3496 // 3497 // To enforce this guarantee: 3498 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3499 // (mixedTracks == 0 && fastTracks > 0)) 3500 // must imply MIXER_TRACKS_READY. 3501 // Later, we may clear buffers regardless, and skip much of this logic. 3502 } 3503 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3504 if (mEffectBufferValid) { 3505 memset(mEffectBuffer, 0, mEffectBufferSize); 3506 } 3507 // FIXME as a performance optimization, should remember previous zero status 3508 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3509 } 3510 3511 // if any fast tracks, then status is ready 3512 mMixerStatusIgnoringFastTracks = mixerStatus; 3513 if (fastTracks > 0) { 3514 mixerStatus = MIXER_TRACKS_READY; 3515 } 3516 return mixerStatus; 3517} 3518 3519// getTrackName_l() must be called with ThreadBase::mLock held 3520int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3521{ 3522 return mAudioMixer->getTrackName(channelMask, sessionId); 3523} 3524 3525// deleteTrackName_l() must be called with ThreadBase::mLock held 3526void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3527{ 3528 ALOGV("remove track (%d) and delete from mixer", name); 3529 mAudioMixer->deleteTrackName(name); 3530} 3531 3532// checkForNewParameters_l() must be called with ThreadBase::mLock held 3533bool AudioFlinger::MixerThread::checkForNewParameters_l() 3534{ 3535 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3536 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3537 bool reconfig = false; 3538 3539 while (!mNewParameters.isEmpty()) { 3540 3541 if (mFastMixer != NULL) { 3542 FastMixerStateQueue *sq = mFastMixer->sq(); 3543 FastMixerState *state = sq->begin(); 3544 if (!(state->mCommand & FastMixerState::IDLE)) { 3545 previousCommand = state->mCommand; 3546 state->mCommand = FastMixerState::HOT_IDLE; 3547 sq->end(); 3548 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3549 } else { 3550 sq->end(false /*didModify*/); 3551 } 3552 } 3553 3554 status_t status = NO_ERROR; 3555 String8 keyValuePair = mNewParameters[0]; 3556 AudioParameter param = AudioParameter(keyValuePair); 3557 int value; 3558 3559 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3560 reconfig = true; 3561 } 3562 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3563 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 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::keyChannels), value) == NO_ERROR) { 3571 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3572 status = BAD_VALUE; 3573 } else { 3574 // no need to save value, since it's constant 3575 reconfig = true; 3576 } 3577 } 3578 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3579 // do not accept frame count changes if tracks are open as the track buffer 3580 // size depends on frame count and correct behavior would not be guaranteed 3581 // if frame count is changed after track creation 3582 if (!mTracks.isEmpty()) { 3583 status = INVALID_OPERATION; 3584 } else { 3585 reconfig = true; 3586 } 3587 } 3588 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3589#ifdef ADD_BATTERY_DATA 3590 // when changing the audio output device, call addBatteryData to notify 3591 // the change 3592 if (mOutDevice != value) { 3593 uint32_t params = 0; 3594 // check whether speaker is on 3595 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3596 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3597 } 3598 3599 audio_devices_t deviceWithoutSpeaker 3600 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3601 // check if any other device (except speaker) is on 3602 if (value & deviceWithoutSpeaker ) { 3603 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3604 } 3605 3606 if (params != 0) { 3607 addBatteryData(params); 3608 } 3609 } 3610#endif 3611 3612 // forward device change to effects that have requested to be 3613 // aware of attached audio device. 3614 if (value != AUDIO_DEVICE_NONE) { 3615 mOutDevice = value; 3616 for (size_t i = 0; i < mEffectChains.size(); i++) { 3617 mEffectChains[i]->setDevice_l(mOutDevice); 3618 } 3619 } 3620 } 3621 3622 if (status == NO_ERROR) { 3623 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3624 keyValuePair.string()); 3625 if (!mStandby && status == INVALID_OPERATION) { 3626 mOutput->stream->common.standby(&mOutput->stream->common); 3627 mStandby = true; 3628 mBytesWritten = 0; 3629 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3630 keyValuePair.string()); 3631 } 3632 if (status == NO_ERROR && reconfig) { 3633 readOutputParameters_l(); 3634 delete mAudioMixer; 3635 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3636 for (size_t i = 0; i < mTracks.size() ; i++) { 3637 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3638 if (name < 0) { 3639 break; 3640 } 3641 mTracks[i]->mName = name; 3642 } 3643 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3644 } 3645 } 3646 3647 mNewParameters.removeAt(0); 3648 3649 mParamStatus = status; 3650 mParamCond.signal(); 3651 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3652 // already timed out waiting for the status and will never signal the condition. 3653 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3654 } 3655 3656 if (!(previousCommand & FastMixerState::IDLE)) { 3657 ALOG_ASSERT(mFastMixer != NULL); 3658 FastMixerStateQueue *sq = mFastMixer->sq(); 3659 FastMixerState *state = sq->begin(); 3660 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3661 state->mCommand = previousCommand; 3662 sq->end(); 3663 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3664 } 3665 3666 return reconfig; 3667} 3668 3669 3670void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3671{ 3672 const size_t SIZE = 256; 3673 char buffer[SIZE]; 3674 String8 result; 3675 3676 PlaybackThread::dumpInternals(fd, args); 3677 3678 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3679 3680 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3681 const FastMixerDumpState copy(mFastMixerDumpState); 3682 copy.dump(fd); 3683 3684#ifdef STATE_QUEUE_DUMP 3685 // Similar for state queue 3686 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3687 observerCopy.dump(fd); 3688 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3689 mutatorCopy.dump(fd); 3690#endif 3691 3692#ifdef TEE_SINK 3693 // Write the tee output to a .wav file 3694 dumpTee(fd, mTeeSource, mId); 3695#endif 3696 3697#ifdef AUDIO_WATCHDOG 3698 if (mAudioWatchdog != 0) { 3699 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3700 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3701 wdCopy.dump(fd); 3702 } 3703#endif 3704} 3705 3706uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3707{ 3708 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3709} 3710 3711uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3712{ 3713 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3714} 3715 3716void AudioFlinger::MixerThread::cacheParameters_l() 3717{ 3718 PlaybackThread::cacheParameters_l(); 3719 3720 // FIXME: Relaxed timing because of a certain device that can't meet latency 3721 // Should be reduced to 2x after the vendor fixes the driver issue 3722 // increase threshold again due to low power audio mode. The way this warning 3723 // threshold is calculated and its usefulness should be reconsidered anyway. 3724 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3725} 3726 3727// ---------------------------------------------------------------------------- 3728 3729AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3730 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3731 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3732 // mLeftVolFloat, mRightVolFloat 3733{ 3734} 3735 3736AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3737 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3738 ThreadBase::type_t type) 3739 : PlaybackThread(audioFlinger, output, id, device, type) 3740 // mLeftVolFloat, mRightVolFloat 3741{ 3742} 3743 3744AudioFlinger::DirectOutputThread::~DirectOutputThread() 3745{ 3746} 3747 3748void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3749{ 3750 audio_track_cblk_t* cblk = track->cblk(); 3751 float left, right; 3752 3753 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3754 left = right = 0; 3755 } else { 3756 float typeVolume = mStreamTypes[track->streamType()].volume; 3757 float v = mMasterVolume * typeVolume; 3758 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3759 uint32_t vlr = proxy->getVolumeLR(); 3760 float v_clamped = v * (vlr & 0xFFFF); 3761 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3762 left = v_clamped/MAX_GAIN; 3763 v_clamped = v * (vlr >> 16); 3764 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3765 right = v_clamped/MAX_GAIN; 3766 } 3767 3768 if (lastTrack) { 3769 if (left != mLeftVolFloat || right != mRightVolFloat) { 3770 mLeftVolFloat = left; 3771 mRightVolFloat = right; 3772 3773 // Convert volumes from float to 8.24 3774 uint32_t vl = (uint32_t)(left * (1 << 24)); 3775 uint32_t vr = (uint32_t)(right * (1 << 24)); 3776 3777 // Delegate volume control to effect in track effect chain if needed 3778 // only one effect chain can be present on DirectOutputThread, so if 3779 // there is one, the track is connected to it 3780 if (!mEffectChains.isEmpty()) { 3781 mEffectChains[0]->setVolume_l(&vl, &vr); 3782 left = (float)vl / (1 << 24); 3783 right = (float)vr / (1 << 24); 3784 } 3785 if (mOutput->stream->set_volume) { 3786 mOutput->stream->set_volume(mOutput->stream, left, right); 3787 } 3788 } 3789 } 3790} 3791 3792 3793AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3794 Vector< sp<Track> > *tracksToRemove 3795) 3796{ 3797 size_t count = mActiveTracks.size(); 3798 mixer_state mixerStatus = MIXER_IDLE; 3799 3800 // find out which tracks need to be processed 3801 for (size_t i = 0; i < count; i++) { 3802 sp<Track> t = mActiveTracks[i].promote(); 3803 // The track died recently 3804 if (t == 0) { 3805 continue; 3806 } 3807 3808 Track* const track = t.get(); 3809 audio_track_cblk_t* cblk = track->cblk(); 3810 // Only consider last track started for volume and mixer state control. 3811 // In theory an older track could underrun and restart after the new one starts 3812 // but as we only care about the transition phase between two tracks on a 3813 // direct output, it is not a problem to ignore the underrun case. 3814 sp<Track> l = mLatestActiveTrack.promote(); 3815 bool last = l.get() == track; 3816 3817 // The first time a track is added we wait 3818 // for all its buffers to be filled before processing it 3819 uint32_t minFrames; 3820 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3821 minFrames = mNormalFrameCount; 3822 } else { 3823 minFrames = 1; 3824 } 3825 3826 if ((track->framesReady() >= minFrames) && track->isReady() && 3827 !track->isPaused() && !track->isTerminated()) 3828 { 3829 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3830 3831 if (track->mFillingUpStatus == Track::FS_FILLED) { 3832 track->mFillingUpStatus = Track::FS_ACTIVE; 3833 // make sure processVolume_l() will apply new volume even if 0 3834 mLeftVolFloat = mRightVolFloat = -1.0; 3835 if (track->mState == TrackBase::RESUMING) { 3836 track->mState = TrackBase::ACTIVE; 3837 } 3838 } 3839 3840 // compute volume for this track 3841 processVolume_l(track, last); 3842 if (last) { 3843 // reset retry count 3844 track->mRetryCount = kMaxTrackRetriesDirect; 3845 mActiveTrack = t; 3846 mixerStatus = MIXER_TRACKS_READY; 3847 } 3848 } else { 3849 // clear effect chain input buffer if the last active track started underruns 3850 // to avoid sending previous audio buffer again to effects 3851 if (!mEffectChains.isEmpty() && last) { 3852 mEffectChains[0]->clearInputBuffer(); 3853 } 3854 3855 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3856 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3857 track->isStopped() || track->isPaused()) { 3858 // We have consumed all the buffers of this track. 3859 // Remove it from the list of active tracks. 3860 // TODO: implement behavior for compressed audio 3861 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3862 size_t framesWritten = mBytesWritten / mFrameSize; 3863 if (mStandby || !last || 3864 track->presentationComplete(framesWritten, audioHALFrames)) { 3865 if (track->isStopped()) { 3866 track->reset(); 3867 } 3868 tracksToRemove->add(track); 3869 } 3870 } else { 3871 // No buffers for this track. Give it a few chances to 3872 // fill a buffer, then remove it from active list. 3873 // Only consider last track started for mixer state control 3874 if (--(track->mRetryCount) <= 0) { 3875 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3876 tracksToRemove->add(track); 3877 // indicate to client process that the track was disabled because of underrun; 3878 // it will then automatically call start() when data is available 3879 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3880 } else if (last) { 3881 mixerStatus = MIXER_TRACKS_ENABLED; 3882 } 3883 } 3884 } 3885 } 3886 3887 // remove all the tracks that need to be... 3888 removeTracks_l(*tracksToRemove); 3889 3890 return mixerStatus; 3891} 3892 3893void AudioFlinger::DirectOutputThread::threadLoop_mix() 3894{ 3895 size_t frameCount = mFrameCount; 3896 int8_t *curBuf = (int8_t *)mSinkBuffer; 3897 // output audio to hardware 3898 while (frameCount) { 3899 AudioBufferProvider::Buffer buffer; 3900 buffer.frameCount = frameCount; 3901 mActiveTrack->getNextBuffer(&buffer); 3902 if (buffer.raw == NULL) { 3903 memset(curBuf, 0, frameCount * mFrameSize); 3904 break; 3905 } 3906 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3907 frameCount -= buffer.frameCount; 3908 curBuf += buffer.frameCount * mFrameSize; 3909 mActiveTrack->releaseBuffer(&buffer); 3910 } 3911 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3912 sleepTime = 0; 3913 standbyTime = systemTime() + standbyDelay; 3914 mActiveTrack.clear(); 3915} 3916 3917void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3918{ 3919 if (sleepTime == 0) { 3920 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3921 sleepTime = activeSleepTime; 3922 } else { 3923 sleepTime = idleSleepTime; 3924 } 3925 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3926 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3927 sleepTime = 0; 3928 } 3929} 3930 3931// getTrackName_l() must be called with ThreadBase::mLock held 3932int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3933 int sessionId __unused) 3934{ 3935 return 0; 3936} 3937 3938// deleteTrackName_l() must be called with ThreadBase::mLock held 3939void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3940{ 3941} 3942 3943// checkForNewParameters_l() must be called with ThreadBase::mLock held 3944bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3945{ 3946 bool reconfig = false; 3947 3948 while (!mNewParameters.isEmpty()) { 3949 status_t status = NO_ERROR; 3950 String8 keyValuePair = mNewParameters[0]; 3951 AudioParameter param = AudioParameter(keyValuePair); 3952 int value; 3953 3954 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3955 // do not accept frame count changes if tracks are open as the track buffer 3956 // size depends on frame count and correct behavior would not be garantied 3957 // if frame count is changed after track creation 3958 if (!mTracks.isEmpty()) { 3959 status = INVALID_OPERATION; 3960 } else { 3961 reconfig = true; 3962 } 3963 } 3964 if (status == NO_ERROR) { 3965 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3966 keyValuePair.string()); 3967 if (!mStandby && status == INVALID_OPERATION) { 3968 mOutput->stream->common.standby(&mOutput->stream->common); 3969 mStandby = true; 3970 mBytesWritten = 0; 3971 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3972 keyValuePair.string()); 3973 } 3974 if (status == NO_ERROR && reconfig) { 3975 readOutputParameters_l(); 3976 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3977 } 3978 } 3979 3980 mNewParameters.removeAt(0); 3981 3982 mParamStatus = status; 3983 mParamCond.signal(); 3984 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3985 // already timed out waiting for the status and will never signal the condition. 3986 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3987 } 3988 return reconfig; 3989} 3990 3991uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3992{ 3993 uint32_t time; 3994 if (audio_is_linear_pcm(mFormat)) { 3995 time = PlaybackThread::activeSleepTimeUs(); 3996 } else { 3997 time = 10000; 3998 } 3999 return time; 4000} 4001 4002uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4003{ 4004 uint32_t time; 4005 if (audio_is_linear_pcm(mFormat)) { 4006 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4007 } else { 4008 time = 10000; 4009 } 4010 return time; 4011} 4012 4013uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4014{ 4015 uint32_t time; 4016 if (audio_is_linear_pcm(mFormat)) { 4017 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4018 } else { 4019 time = 10000; 4020 } 4021 return time; 4022} 4023 4024void AudioFlinger::DirectOutputThread::cacheParameters_l() 4025{ 4026 PlaybackThread::cacheParameters_l(); 4027 4028 // use shorter standby delay as on normal output to release 4029 // hardware resources as soon as possible 4030 if (audio_is_linear_pcm(mFormat)) { 4031 standbyDelay = microseconds(activeSleepTime*2); 4032 } else { 4033 standbyDelay = kOffloadStandbyDelayNs; 4034 } 4035} 4036 4037// ---------------------------------------------------------------------------- 4038 4039AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4040 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4041 : Thread(false /*canCallJava*/), 4042 mPlaybackThread(playbackThread), 4043 mWriteAckSequence(0), 4044 mDrainSequence(0) 4045{ 4046} 4047 4048AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4049{ 4050} 4051 4052void AudioFlinger::AsyncCallbackThread::onFirstRef() 4053{ 4054 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4055} 4056 4057bool AudioFlinger::AsyncCallbackThread::threadLoop() 4058{ 4059 while (!exitPending()) { 4060 uint32_t writeAckSequence; 4061 uint32_t drainSequence; 4062 4063 { 4064 Mutex::Autolock _l(mLock); 4065 while (!((mWriteAckSequence & 1) || 4066 (mDrainSequence & 1) || 4067 exitPending())) { 4068 mWaitWorkCV.wait(mLock); 4069 } 4070 4071 if (exitPending()) { 4072 break; 4073 } 4074 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4075 mWriteAckSequence, mDrainSequence); 4076 writeAckSequence = mWriteAckSequence; 4077 mWriteAckSequence &= ~1; 4078 drainSequence = mDrainSequence; 4079 mDrainSequence &= ~1; 4080 } 4081 { 4082 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4083 if (playbackThread != 0) { 4084 if (writeAckSequence & 1) { 4085 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4086 } 4087 if (drainSequence & 1) { 4088 playbackThread->resetDraining(drainSequence >> 1); 4089 } 4090 } 4091 } 4092 } 4093 return false; 4094} 4095 4096void AudioFlinger::AsyncCallbackThread::exit() 4097{ 4098 ALOGV("AsyncCallbackThread::exit"); 4099 Mutex::Autolock _l(mLock); 4100 requestExit(); 4101 mWaitWorkCV.broadcast(); 4102} 4103 4104void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4105{ 4106 Mutex::Autolock _l(mLock); 4107 // bit 0 is cleared 4108 mWriteAckSequence = sequence << 1; 4109} 4110 4111void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4112{ 4113 Mutex::Autolock _l(mLock); 4114 // ignore unexpected callbacks 4115 if (mWriteAckSequence & 2) { 4116 mWriteAckSequence |= 1; 4117 mWaitWorkCV.signal(); 4118 } 4119} 4120 4121void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4122{ 4123 Mutex::Autolock _l(mLock); 4124 // bit 0 is cleared 4125 mDrainSequence = sequence << 1; 4126} 4127 4128void AudioFlinger::AsyncCallbackThread::resetDraining() 4129{ 4130 Mutex::Autolock _l(mLock); 4131 // ignore unexpected callbacks 4132 if (mDrainSequence & 2) { 4133 mDrainSequence |= 1; 4134 mWaitWorkCV.signal(); 4135 } 4136} 4137 4138 4139// ---------------------------------------------------------------------------- 4140AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4141 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4142 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4143 mHwPaused(false), 4144 mFlushPending(false), 4145 mPausedBytesRemaining(0) 4146{ 4147 //FIXME: mStandby should be set to true by ThreadBase constructor 4148 mStandby = true; 4149} 4150 4151void AudioFlinger::OffloadThread::threadLoop_exit() 4152{ 4153 if (mFlushPending || mHwPaused) { 4154 // If a flush is pending or track was paused, just discard buffered data 4155 flushHw_l(); 4156 } else { 4157 mMixerStatus = MIXER_DRAIN_ALL; 4158 threadLoop_drain(); 4159 } 4160 mCallbackThread->exit(); 4161 PlaybackThread::threadLoop_exit(); 4162} 4163 4164AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4165 Vector< sp<Track> > *tracksToRemove 4166) 4167{ 4168 size_t count = mActiveTracks.size(); 4169 4170 mixer_state mixerStatus = MIXER_IDLE; 4171 bool doHwPause = false; 4172 bool doHwResume = false; 4173 4174 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4175 4176 // find out which tracks need to be processed 4177 for (size_t i = 0; i < count; i++) { 4178 sp<Track> t = mActiveTracks[i].promote(); 4179 // The track died recently 4180 if (t == 0) { 4181 continue; 4182 } 4183 Track* const track = t.get(); 4184 audio_track_cblk_t* cblk = track->cblk(); 4185 // Only consider last track started for volume and mixer state control. 4186 // In theory an older track could underrun and restart after the new one starts 4187 // but as we only care about the transition phase between two tracks on a 4188 // direct output, it is not a problem to ignore the underrun case. 4189 sp<Track> l = mLatestActiveTrack.promote(); 4190 bool last = l.get() == track; 4191 4192 if (track->isInvalid()) { 4193 ALOGW("An invalidated track shouldn't be in active list"); 4194 tracksToRemove->add(track); 4195 continue; 4196 } 4197 4198 if (track->mState == TrackBase::IDLE) { 4199 ALOGW("An idle track shouldn't be in active list"); 4200 continue; 4201 } 4202 4203 if (track->isPausing()) { 4204 track->setPaused(); 4205 if (last) { 4206 if (!mHwPaused) { 4207 doHwPause = true; 4208 mHwPaused = true; 4209 } 4210 // If we were part way through writing the mixbuffer to 4211 // the HAL we must save this until we resume 4212 // BUG - this will be wrong if a different track is made active, 4213 // in that case we want to discard the pending data in the 4214 // mixbuffer and tell the client to present it again when the 4215 // track is resumed 4216 mPausedWriteLength = mCurrentWriteLength; 4217 mPausedBytesRemaining = mBytesRemaining; 4218 mBytesRemaining = 0; // stop writing 4219 } 4220 tracksToRemove->add(track); 4221 } else if (track->isFlushPending()) { 4222 track->flushAck(); 4223 if (last) { 4224 mFlushPending = true; 4225 } 4226 } else if (track->isResumePending()){ 4227 track->resumeAck(); 4228 if (last) { 4229 if (mPausedBytesRemaining) { 4230 // Need to continue write that was interrupted 4231 mCurrentWriteLength = mPausedWriteLength; 4232 mBytesRemaining = mPausedBytesRemaining; 4233 mPausedBytesRemaining = 0; 4234 } 4235 if (mHwPaused) { 4236 doHwResume = true; 4237 mHwPaused = false; 4238 // threadLoop_mix() will handle the case that we need to 4239 // resume an interrupted write 4240 } 4241 // enable write to audio HAL 4242 sleepTime = 0; 4243 4244 // Do not handle new data in this iteration even if track->framesReady() 4245 mixerStatus = MIXER_TRACKS_ENABLED; 4246 } 4247 } else if (track->framesReady() && track->isReady() && 4248 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4249 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4250 if (track->mFillingUpStatus == Track::FS_FILLED) { 4251 track->mFillingUpStatus = Track::FS_ACTIVE; 4252 // make sure processVolume_l() will apply new volume even if 0 4253 mLeftVolFloat = mRightVolFloat = -1.0; 4254 } 4255 4256 if (last) { 4257 sp<Track> previousTrack = mPreviousTrack.promote(); 4258 if (previousTrack != 0) { 4259 if (track != previousTrack.get()) { 4260 // Flush any data still being written from last track 4261 mBytesRemaining = 0; 4262 if (mPausedBytesRemaining) { 4263 // Last track was paused so we also need to flush saved 4264 // mixbuffer state and invalidate track so that it will 4265 // re-submit that unwritten data when it is next resumed 4266 mPausedBytesRemaining = 0; 4267 // Invalidate is a bit drastic - would be more efficient 4268 // to have a flag to tell client that some of the 4269 // previously written data was lost 4270 previousTrack->invalidate(); 4271 } 4272 // flush data already sent to the DSP if changing audio session as audio 4273 // comes from a different source. Also invalidate previous track to force a 4274 // seek when resuming. 4275 if (previousTrack->sessionId() != track->sessionId()) { 4276 previousTrack->invalidate(); 4277 } 4278 } 4279 } 4280 mPreviousTrack = track; 4281 // reset retry count 4282 track->mRetryCount = kMaxTrackRetriesOffload; 4283 mActiveTrack = t; 4284 mixerStatus = MIXER_TRACKS_READY; 4285 } 4286 } else { 4287 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4288 if (track->isStopping_1()) { 4289 // Hardware buffer can hold a large amount of audio so we must 4290 // wait for all current track's data to drain before we say 4291 // that the track is stopped. 4292 if (mBytesRemaining == 0) { 4293 // Only start draining when all data in mixbuffer 4294 // has been written 4295 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4296 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4297 // do not drain if no data was ever sent to HAL (mStandby == true) 4298 if (last && !mStandby) { 4299 // do not modify drain sequence if we are already draining. This happens 4300 // when resuming from pause after drain. 4301 if ((mDrainSequence & 1) == 0) { 4302 sleepTime = 0; 4303 standbyTime = systemTime() + standbyDelay; 4304 mixerStatus = MIXER_DRAIN_TRACK; 4305 mDrainSequence += 2; 4306 } 4307 if (mHwPaused) { 4308 // It is possible to move from PAUSED to STOPPING_1 without 4309 // a resume so we must ensure hardware is running 4310 doHwResume = true; 4311 mHwPaused = false; 4312 } 4313 } 4314 } 4315 } else if (track->isStopping_2()) { 4316 // Drain has completed or we are in standby, signal presentation complete 4317 if (!(mDrainSequence & 1) || !last || mStandby) { 4318 track->mState = TrackBase::STOPPED; 4319 size_t audioHALFrames = 4320 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4321 size_t framesWritten = 4322 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4323 track->presentationComplete(framesWritten, audioHALFrames); 4324 track->reset(); 4325 tracksToRemove->add(track); 4326 } 4327 } else { 4328 // No buffers for this track. Give it a few chances to 4329 // fill a buffer, then remove it from active list. 4330 if (--(track->mRetryCount) <= 0) { 4331 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4332 track->name()); 4333 tracksToRemove->add(track); 4334 // indicate to client process that the track was disabled because of underrun; 4335 // it will then automatically call start() when data is available 4336 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4337 } else if (last){ 4338 mixerStatus = MIXER_TRACKS_ENABLED; 4339 } 4340 } 4341 } 4342 // compute volume for this track 4343 processVolume_l(track, last); 4344 } 4345 4346 // make sure the pause/flush/resume sequence is executed in the right order. 4347 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4348 // before flush and then resume HW. This can happen in case of pause/flush/resume 4349 // if resume is received before pause is executed. 4350 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4351 mOutput->stream->pause(mOutput->stream); 4352 } 4353 if (mFlushPending) { 4354 flushHw_l(); 4355 mFlushPending = false; 4356 } 4357 if (!mStandby && doHwResume) { 4358 mOutput->stream->resume(mOutput->stream); 4359 } 4360 4361 // remove all the tracks that need to be... 4362 removeTracks_l(*tracksToRemove); 4363 4364 return mixerStatus; 4365} 4366 4367// must be called with thread mutex locked 4368bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4369{ 4370 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4371 mWriteAckSequence, mDrainSequence); 4372 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4373 return true; 4374 } 4375 return false; 4376} 4377 4378// must be called with thread mutex locked 4379bool AudioFlinger::OffloadThread::shouldStandby_l() 4380{ 4381 bool trackPaused = false; 4382 4383 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4384 // after a timeout and we will enter standby then. 4385 if (mTracks.size() > 0) { 4386 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4387 } 4388 4389 return !mStandby && !trackPaused; 4390} 4391 4392 4393bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4394{ 4395 Mutex::Autolock _l(mLock); 4396 return waitingAsyncCallback_l(); 4397} 4398 4399void AudioFlinger::OffloadThread::flushHw_l() 4400{ 4401 mOutput->stream->flush(mOutput->stream); 4402 // Flush anything still waiting in the mixbuffer 4403 mCurrentWriteLength = 0; 4404 mBytesRemaining = 0; 4405 mPausedWriteLength = 0; 4406 mPausedBytesRemaining = 0; 4407 mHwPaused = false; 4408 4409 if (mUseAsyncWrite) { 4410 // discard any pending drain or write ack by incrementing sequence 4411 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4412 mDrainSequence = (mDrainSequence + 2) & ~1; 4413 ALOG_ASSERT(mCallbackThread != 0); 4414 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4415 mCallbackThread->setDraining(mDrainSequence); 4416 } 4417} 4418 4419void AudioFlinger::OffloadThread::onAddNewTrack_l() 4420{ 4421 sp<Track> previousTrack = mPreviousTrack.promote(); 4422 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4423 4424 if (previousTrack != 0 && latestTrack != 0 && 4425 (previousTrack->sessionId() != latestTrack->sessionId())) { 4426 mFlushPending = true; 4427 } 4428 PlaybackThread::onAddNewTrack_l(); 4429} 4430 4431// ---------------------------------------------------------------------------- 4432 4433AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4434 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4435 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4436 DUPLICATING), 4437 mWaitTimeMs(UINT_MAX) 4438{ 4439 addOutputTrack(mainThread); 4440} 4441 4442AudioFlinger::DuplicatingThread::~DuplicatingThread() 4443{ 4444 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4445 mOutputTracks[i]->destroy(); 4446 } 4447} 4448 4449void AudioFlinger::DuplicatingThread::threadLoop_mix() 4450{ 4451 // mix buffers... 4452 if (outputsReady(outputTracks)) { 4453 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4454 } else { 4455 memset(mSinkBuffer, 0, mSinkBufferSize); 4456 } 4457 sleepTime = 0; 4458 writeFrames = mNormalFrameCount; 4459 mCurrentWriteLength = mSinkBufferSize; 4460 standbyTime = systemTime() + standbyDelay; 4461} 4462 4463void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4464{ 4465 if (sleepTime == 0) { 4466 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4467 sleepTime = activeSleepTime; 4468 } else { 4469 sleepTime = idleSleepTime; 4470 } 4471 } else if (mBytesWritten != 0) { 4472 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4473 writeFrames = mNormalFrameCount; 4474 memset(mSinkBuffer, 0, mSinkBufferSize); 4475 } else { 4476 // flush remaining overflow buffers in output tracks 4477 writeFrames = 0; 4478 } 4479 sleepTime = 0; 4480 } 4481} 4482 4483ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4484{ 4485 for (size_t i = 0; i < outputTracks.size(); i++) { 4486 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4487 // for delivery downstream as needed. This in-place conversion is safe as 4488 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4489 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4490 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4491 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4492 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4493 } 4494 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4495 } 4496 mStandby = false; 4497 return (ssize_t)mSinkBufferSize; 4498} 4499 4500void AudioFlinger::DuplicatingThread::threadLoop_standby() 4501{ 4502 // DuplicatingThread implements standby by stopping all tracks 4503 for (size_t i = 0; i < outputTracks.size(); i++) { 4504 outputTracks[i]->stop(); 4505 } 4506} 4507 4508void AudioFlinger::DuplicatingThread::saveOutputTracks() 4509{ 4510 outputTracks = mOutputTracks; 4511} 4512 4513void AudioFlinger::DuplicatingThread::clearOutputTracks() 4514{ 4515 outputTracks.clear(); 4516} 4517 4518void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4519{ 4520 Mutex::Autolock _l(mLock); 4521 // FIXME explain this formula 4522 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4523 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4524 // due to current usage case and restrictions on the AudioBufferProvider. 4525 // Actual buffer conversion is done in threadLoop_write(). 4526 // 4527 // TODO: This may change in the future, depending on multichannel 4528 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4529 OutputTrack *outputTrack = new OutputTrack(thread, 4530 this, 4531 mSampleRate, 4532 AUDIO_FORMAT_PCM_16_BIT, 4533 mChannelMask, 4534 frameCount, 4535 IPCThreadState::self()->getCallingUid()); 4536 if (outputTrack->cblk() != NULL) { 4537 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4538 mOutputTracks.add(outputTrack); 4539 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4540 updateWaitTime_l(); 4541 } 4542} 4543 4544void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4545{ 4546 Mutex::Autolock _l(mLock); 4547 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4548 if (mOutputTracks[i]->thread() == thread) { 4549 mOutputTracks[i]->destroy(); 4550 mOutputTracks.removeAt(i); 4551 updateWaitTime_l(); 4552 return; 4553 } 4554 } 4555 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4556} 4557 4558// caller must hold mLock 4559void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4560{ 4561 mWaitTimeMs = UINT_MAX; 4562 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4563 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4564 if (strong != 0) { 4565 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4566 if (waitTimeMs < mWaitTimeMs) { 4567 mWaitTimeMs = waitTimeMs; 4568 } 4569 } 4570 } 4571} 4572 4573 4574bool AudioFlinger::DuplicatingThread::outputsReady( 4575 const SortedVector< sp<OutputTrack> > &outputTracks) 4576{ 4577 for (size_t i = 0; i < outputTracks.size(); i++) { 4578 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4579 if (thread == 0) { 4580 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4581 outputTracks[i].get()); 4582 return false; 4583 } 4584 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4585 // see note at standby() declaration 4586 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4587 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4588 thread.get()); 4589 return false; 4590 } 4591 } 4592 return true; 4593} 4594 4595uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4596{ 4597 return (mWaitTimeMs * 1000) / 2; 4598} 4599 4600void AudioFlinger::DuplicatingThread::cacheParameters_l() 4601{ 4602 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4603 updateWaitTime_l(); 4604 4605 MixerThread::cacheParameters_l(); 4606} 4607 4608// ---------------------------------------------------------------------------- 4609// Record 4610// ---------------------------------------------------------------------------- 4611 4612AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4613 AudioStreamIn *input, 4614 audio_io_handle_t id, 4615 audio_devices_t outDevice, 4616 audio_devices_t inDevice 4617#ifdef TEE_SINK 4618 , const sp<NBAIO_Sink>& teeSink 4619#endif 4620 ) : 4621 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4622 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4623 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4624 mRsmpInRear(0) 4625#ifdef TEE_SINK 4626 , mTeeSink(teeSink) 4627#endif 4628{ 4629 snprintf(mName, kNameLength, "AudioIn_%X", id); 4630 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4631 4632 readInputParameters_l(); 4633} 4634 4635 4636AudioFlinger::RecordThread::~RecordThread() 4637{ 4638 mAudioFlinger->unregisterWriter(mNBLogWriter); 4639 delete[] mRsmpInBuffer; 4640} 4641 4642void AudioFlinger::RecordThread::onFirstRef() 4643{ 4644 run(mName, PRIORITY_URGENT_AUDIO); 4645} 4646 4647bool AudioFlinger::RecordThread::threadLoop() 4648{ 4649 nsecs_t lastWarning = 0; 4650 4651 inputStandBy(); 4652 4653reacquire_wakelock: 4654 sp<RecordTrack> activeTrack; 4655 int activeTracksGen; 4656 { 4657 Mutex::Autolock _l(mLock); 4658 size_t size = mActiveTracks.size(); 4659 activeTracksGen = mActiveTracksGen; 4660 if (size > 0) { 4661 // FIXME an arbitrary choice 4662 activeTrack = mActiveTracks[0]; 4663 acquireWakeLock_l(activeTrack->uid()); 4664 if (size > 1) { 4665 SortedVector<int> tmp; 4666 for (size_t i = 0; i < size; i++) { 4667 tmp.add(mActiveTracks[i]->uid()); 4668 } 4669 updateWakeLockUids_l(tmp); 4670 } 4671 } else { 4672 acquireWakeLock_l(-1); 4673 } 4674 } 4675 4676 // used to request a deferred sleep, to be executed later while mutex is unlocked 4677 uint32_t sleepUs = 0; 4678 4679 // loop while there is work to do 4680 for (;;) { 4681 Vector< sp<EffectChain> > effectChains; 4682 4683 // sleep with mutex unlocked 4684 if (sleepUs > 0) { 4685 usleep(sleepUs); 4686 sleepUs = 0; 4687 } 4688 4689 // activeTracks accumulates a copy of a subset of mActiveTracks 4690 Vector< sp<RecordTrack> > activeTracks; 4691 4692 { // scope for mLock 4693 Mutex::Autolock _l(mLock); 4694 4695 processConfigEvents_l(); 4696 // return value 'reconfig' is currently unused 4697 bool reconfig = checkForNewParameters_l(); 4698 4699 // check exitPending here because checkForNewParameters_l() and 4700 // checkForNewParameters_l() can temporarily release mLock 4701 if (exitPending()) { 4702 break; 4703 } 4704 4705 // if no active track(s), then standby and release wakelock 4706 size_t size = mActiveTracks.size(); 4707 if (size == 0) { 4708 standbyIfNotAlreadyInStandby(); 4709 // exitPending() can't become true here 4710 releaseWakeLock_l(); 4711 ALOGV("RecordThread: loop stopping"); 4712 // go to sleep 4713 mWaitWorkCV.wait(mLock); 4714 ALOGV("RecordThread: loop starting"); 4715 goto reacquire_wakelock; 4716 } 4717 4718 if (mActiveTracksGen != activeTracksGen) { 4719 activeTracksGen = mActiveTracksGen; 4720 SortedVector<int> tmp; 4721 for (size_t i = 0; i < size; i++) { 4722 tmp.add(mActiveTracks[i]->uid()); 4723 } 4724 updateWakeLockUids_l(tmp); 4725 } 4726 4727 bool doBroadcast = false; 4728 for (size_t i = 0; i < size; ) { 4729 4730 activeTrack = mActiveTracks[i]; 4731 if (activeTrack->isTerminated()) { 4732 removeTrack_l(activeTrack); 4733 mActiveTracks.remove(activeTrack); 4734 mActiveTracksGen++; 4735 size--; 4736 continue; 4737 } 4738 4739 TrackBase::track_state activeTrackState = activeTrack->mState; 4740 switch (activeTrackState) { 4741 4742 case TrackBase::PAUSING: 4743 mActiveTracks.remove(activeTrack); 4744 mActiveTracksGen++; 4745 doBroadcast = true; 4746 size--; 4747 continue; 4748 4749 case TrackBase::STARTING_1: 4750 sleepUs = 10000; 4751 i++; 4752 continue; 4753 4754 case TrackBase::STARTING_2: 4755 doBroadcast = true; 4756 mStandby = false; 4757 activeTrack->mState = TrackBase::ACTIVE; 4758 break; 4759 4760 case TrackBase::ACTIVE: 4761 break; 4762 4763 case TrackBase::IDLE: 4764 i++; 4765 continue; 4766 4767 default: 4768 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4769 } 4770 4771 activeTracks.add(activeTrack); 4772 i++; 4773 4774 } 4775 if (doBroadcast) { 4776 mStartStopCond.broadcast(); 4777 } 4778 4779 // sleep if there are no active tracks to process 4780 if (activeTracks.size() == 0) { 4781 if (sleepUs == 0) { 4782 sleepUs = kRecordThreadSleepUs; 4783 } 4784 continue; 4785 } 4786 sleepUs = 0; 4787 4788 lockEffectChains_l(effectChains); 4789 } 4790 4791 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4792 4793 size_t size = effectChains.size(); 4794 for (size_t i = 0; i < size; i++) { 4795 // thread mutex is not locked, but effect chain is locked 4796 effectChains[i]->process_l(); 4797 } 4798 4799 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4800 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4801 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4802 // If destination is non-contiguous, first read past the nominal end of buffer, then 4803 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4804 4805 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4806 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4807 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4808 if (bytesRead <= 0) { 4809 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4810 // Force input into standby so that it tries to recover at next read attempt 4811 inputStandBy(); 4812 sleepUs = kRecordThreadSleepUs; 4813 continue; 4814 } 4815 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4816 size_t framesRead = bytesRead / mFrameSize; 4817 ALOG_ASSERT(framesRead > 0); 4818 if (mTeeSink != 0) { 4819 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4820 } 4821 // If destination is non-contiguous, we now correct for reading past end of buffer. 4822 size_t part1 = mRsmpInFramesP2 - rear; 4823 if (framesRead > part1) { 4824 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4825 (framesRead - part1) * mFrameSize); 4826 } 4827 rear = mRsmpInRear += framesRead; 4828 4829 size = activeTracks.size(); 4830 // loop over each active track 4831 for (size_t i = 0; i < size; i++) { 4832 activeTrack = activeTracks[i]; 4833 4834 enum { 4835 OVERRUN_UNKNOWN, 4836 OVERRUN_TRUE, 4837 OVERRUN_FALSE 4838 } overrun = OVERRUN_UNKNOWN; 4839 4840 // loop over getNextBuffer to handle circular sink 4841 for (;;) { 4842 4843 activeTrack->mSink.frameCount = ~0; 4844 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4845 size_t framesOut = activeTrack->mSink.frameCount; 4846 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4847 4848 int32_t front = activeTrack->mRsmpInFront; 4849 ssize_t filled = rear - front; 4850 size_t framesIn; 4851 4852 if (filled < 0) { 4853 // should not happen, but treat like a massive overrun and re-sync 4854 framesIn = 0; 4855 activeTrack->mRsmpInFront = rear; 4856 overrun = OVERRUN_TRUE; 4857 } else if ((size_t) filled <= mRsmpInFrames) { 4858 framesIn = (size_t) filled; 4859 } else { 4860 // client is not keeping up with server, but give it latest data 4861 framesIn = mRsmpInFrames; 4862 activeTrack->mRsmpInFront = front = rear - framesIn; 4863 overrun = OVERRUN_TRUE; 4864 } 4865 4866 if (framesOut == 0 || framesIn == 0) { 4867 break; 4868 } 4869 4870 if (activeTrack->mResampler == NULL) { 4871 // no resampling 4872 if (framesIn > framesOut) { 4873 framesIn = framesOut; 4874 } else { 4875 framesOut = framesIn; 4876 } 4877 int8_t *dst = activeTrack->mSink.i8; 4878 while (framesIn > 0) { 4879 front &= mRsmpInFramesP2 - 1; 4880 size_t part1 = mRsmpInFramesP2 - front; 4881 if (part1 > framesIn) { 4882 part1 = framesIn; 4883 } 4884 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4885 if (mChannelCount == activeTrack->mChannelCount) { 4886 memcpy(dst, src, part1 * mFrameSize); 4887 } else if (mChannelCount == 1) { 4888 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4889 part1); 4890 } else { 4891 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4892 part1); 4893 } 4894 dst += part1 * activeTrack->mFrameSize; 4895 front += part1; 4896 framesIn -= part1; 4897 } 4898 activeTrack->mRsmpInFront += framesOut; 4899 4900 } else { 4901 // resampling 4902 // FIXME framesInNeeded should really be part of resampler API, and should 4903 // depend on the SRC ratio 4904 // to keep mRsmpInBuffer full so resampler always has sufficient input 4905 size_t framesInNeeded; 4906 // FIXME only re-calculate when it changes, and optimize for common ratios 4907 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4908 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4909 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4910 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4911 framesInNeeded, framesOut, inOverOut); 4912 // Although we theoretically have framesIn in circular buffer, some of those are 4913 // unreleased frames, and thus must be discounted for purpose of budgeting. 4914 size_t unreleased = activeTrack->mRsmpInUnrel; 4915 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4916 if (framesIn < framesInNeeded) { 4917 ALOGV("not enough to resample: have %u frames in but need %u in to " 4918 "produce %u out given in/out ratio of %.4g", 4919 framesIn, framesInNeeded, framesOut, inOverOut); 4920 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4921 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4922 if (newFramesOut == 0) { 4923 break; 4924 } 4925 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4926 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4927 framesInNeeded, newFramesOut, outOverIn); 4928 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4929 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4930 "given in/out ratio of %.4g", 4931 framesIn, framesInNeeded, newFramesOut, inOverOut); 4932 framesOut = newFramesOut; 4933 } else { 4934 ALOGV("success 1: have %u in and need %u in to produce %u out " 4935 "given in/out ratio of %.4g", 4936 framesIn, framesInNeeded, framesOut, inOverOut); 4937 } 4938 4939 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4940 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4941 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4942 delete[] activeTrack->mRsmpOutBuffer; 4943 // resampler always outputs stereo 4944 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4945 activeTrack->mRsmpOutFrameCount = framesOut; 4946 } 4947 4948 // resampler accumulates, but we only have one source track 4949 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4950 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4951 // FIXME how about having activeTrack implement this interface itself? 4952 activeTrack->mResamplerBufferProvider 4953 /*this*/ /* AudioBufferProvider* */); 4954 // ditherAndClamp() works as long as all buffers returned by 4955 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4956 if (activeTrack->mChannelCount == 1) { 4957 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4958 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4959 framesOut); 4960 // the resampler always outputs stereo samples: 4961 // do post stereo to mono conversion 4962 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4963 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4964 } else { 4965 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4966 activeTrack->mRsmpOutBuffer, framesOut); 4967 } 4968 // now done with mRsmpOutBuffer 4969 4970 } 4971 4972 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4973 overrun = OVERRUN_FALSE; 4974 } 4975 4976 if (activeTrack->mFramesToDrop == 0) { 4977 if (framesOut > 0) { 4978 activeTrack->mSink.frameCount = framesOut; 4979 activeTrack->releaseBuffer(&activeTrack->mSink); 4980 } 4981 } else { 4982 // FIXME could do a partial drop of framesOut 4983 if (activeTrack->mFramesToDrop > 0) { 4984 activeTrack->mFramesToDrop -= framesOut; 4985 if (activeTrack->mFramesToDrop <= 0) { 4986 activeTrack->clearSyncStartEvent(); 4987 } 4988 } else { 4989 activeTrack->mFramesToDrop += framesOut; 4990 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4991 activeTrack->mSyncStartEvent->isCancelled()) { 4992 ALOGW("Synced record %s, session %d, trigger session %d", 4993 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 4994 activeTrack->sessionId(), 4995 (activeTrack->mSyncStartEvent != 0) ? 4996 activeTrack->mSyncStartEvent->triggerSession() : 0); 4997 activeTrack->clearSyncStartEvent(); 4998 } 4999 } 5000 } 5001 5002 if (framesOut == 0) { 5003 break; 5004 } 5005 } 5006 5007 switch (overrun) { 5008 case OVERRUN_TRUE: 5009 // client isn't retrieving buffers fast enough 5010 if (!activeTrack->setOverflow()) { 5011 nsecs_t now = systemTime(); 5012 // FIXME should lastWarning per track? 5013 if ((now - lastWarning) > kWarningThrottleNs) { 5014 ALOGW("RecordThread: buffer overflow"); 5015 lastWarning = now; 5016 } 5017 } 5018 break; 5019 case OVERRUN_FALSE: 5020 activeTrack->clearOverflow(); 5021 break; 5022 case OVERRUN_UNKNOWN: 5023 break; 5024 } 5025 5026 } 5027 5028 // enable changes in effect chain 5029 unlockEffectChains(effectChains); 5030 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5031 } 5032 5033 standbyIfNotAlreadyInStandby(); 5034 5035 { 5036 Mutex::Autolock _l(mLock); 5037 for (size_t i = 0; i < mTracks.size(); i++) { 5038 sp<RecordTrack> track = mTracks[i]; 5039 track->invalidate(); 5040 } 5041 mActiveTracks.clear(); 5042 mActiveTracksGen++; 5043 mStartStopCond.broadcast(); 5044 } 5045 5046 releaseWakeLock(); 5047 5048 ALOGV("RecordThread %p exiting", this); 5049 return false; 5050} 5051 5052void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5053{ 5054 if (!mStandby) { 5055 inputStandBy(); 5056 mStandby = true; 5057 } 5058} 5059 5060void AudioFlinger::RecordThread::inputStandBy() 5061{ 5062 mInput->stream->common.standby(&mInput->stream->common); 5063} 5064 5065// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5066sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5067 const sp<AudioFlinger::Client>& client, 5068 uint32_t sampleRate, 5069 audio_format_t format, 5070 audio_channel_mask_t channelMask, 5071 size_t *pFrameCount, 5072 int sessionId, 5073 int uid, 5074 IAudioFlinger::track_flags_t *flags, 5075 pid_t tid, 5076 status_t *status) 5077{ 5078 size_t frameCount = *pFrameCount; 5079 sp<RecordTrack> track; 5080 status_t lStatus; 5081 5082 // client expresses a preference for FAST, but we get the final say 5083 if (*flags & IAudioFlinger::TRACK_FAST) { 5084 if ( 5085 // use case: callback handler and frame count is default or at least as large as HAL 5086 ( 5087 (tid != -1) && 5088 ((frameCount == 0) || 5089 // FIXME not necessarily true, should be native frame count for native SR! 5090 (frameCount >= mFrameCount)) 5091 ) && 5092 // PCM data 5093 audio_is_linear_pcm(format) && 5094 // mono or stereo 5095 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 5096 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 5097 // hardware sample rate 5098 // FIXME actually the native hardware sample rate 5099 (sampleRate == mSampleRate) && 5100 // record thread has an associated fast capture 5101 hasFastCapture() 5102 // fast capture does not require slots 5103 ) { 5104 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5105 if (frameCount == 0) { 5106 // FIXME wrong mFrameCount 5107 frameCount = mFrameCount * kFastTrackMultiplier; 5108 } 5109 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5110 frameCount, mFrameCount); 5111 } else { 5112 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5113 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5114 "hasFastCapture=%d tid=%d", 5115 frameCount, mFrameCount, format, 5116 audio_is_linear_pcm(format), 5117 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5118 *flags &= ~IAudioFlinger::TRACK_FAST; 5119 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5120 // For compatibility with AudioRecord calculation, buffer depth is forced 5121 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5122 // This is probably too conservative, but legacy application code may depend on it. 5123 // If you change this calculation, also review the start threshold which is related. 5124 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5125 size_t mNormalFrameCount = 2048; // FIXME 5126 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5127 if (minBufCount < 2) { 5128 minBufCount = 2; 5129 } 5130 size_t minFrameCount = mNormalFrameCount * minBufCount; 5131 if (frameCount < minFrameCount) { 5132 frameCount = minFrameCount; 5133 } 5134 } 5135 } 5136 *pFrameCount = frameCount; 5137 5138 lStatus = initCheck(); 5139 if (lStatus != NO_ERROR) { 5140 ALOGE("createRecordTrack_l() audio driver not initialized"); 5141 goto Exit; 5142 } 5143 5144 { // scope for mLock 5145 Mutex::Autolock _l(mLock); 5146 5147 track = new RecordTrack(this, client, sampleRate, 5148 format, channelMask, frameCount, sessionId, uid); 5149 5150 lStatus = track->initCheck(); 5151 if (lStatus != NO_ERROR) { 5152 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5153 // track must be cleared from the caller as the caller has the AF lock 5154 goto Exit; 5155 } 5156 mTracks.add(track); 5157 5158 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5159 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5160 mAudioFlinger->btNrecIsOff(); 5161 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5162 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5163 5164 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5165 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5166 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5167 // so ask activity manager to do this on our behalf 5168 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5169 } 5170 } 5171 5172 lStatus = NO_ERROR; 5173 5174Exit: 5175 *status = lStatus; 5176 return track; 5177} 5178 5179status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5180 AudioSystem::sync_event_t event, 5181 int triggerSession) 5182{ 5183 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5184 sp<ThreadBase> strongMe = this; 5185 status_t status = NO_ERROR; 5186 5187 if (event == AudioSystem::SYNC_EVENT_NONE) { 5188 recordTrack->clearSyncStartEvent(); 5189 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5190 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5191 triggerSession, 5192 recordTrack->sessionId(), 5193 syncStartEventCallback, 5194 recordTrack); 5195 // Sync event can be cancelled by the trigger session if the track is not in a 5196 // compatible state in which case we start record immediately 5197 if (recordTrack->mSyncStartEvent->isCancelled()) { 5198 recordTrack->clearSyncStartEvent(); 5199 } else { 5200 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5201 recordTrack->mFramesToDrop = - 5202 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5203 } 5204 } 5205 5206 { 5207 // This section is a rendezvous between binder thread executing start() and RecordThread 5208 AutoMutex lock(mLock); 5209 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5210 if (recordTrack->mState == TrackBase::PAUSING) { 5211 ALOGV("active record track PAUSING -> ACTIVE"); 5212 recordTrack->mState = TrackBase::ACTIVE; 5213 } else { 5214 ALOGV("active record track state %d", recordTrack->mState); 5215 } 5216 return status; 5217 } 5218 5219 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5220 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5221 // or using a separate command thread 5222 recordTrack->mState = TrackBase::STARTING_1; 5223 mActiveTracks.add(recordTrack); 5224 mActiveTracksGen++; 5225 mLock.unlock(); 5226 status_t status = AudioSystem::startInput(mId); 5227 mLock.lock(); 5228 // FIXME should verify that recordTrack is still in mActiveTracks 5229 if (status != NO_ERROR) { 5230 mActiveTracks.remove(recordTrack); 5231 mActiveTracksGen++; 5232 recordTrack->clearSyncStartEvent(); 5233 return status; 5234 } 5235 // Catch up with current buffer indices if thread is already running. 5236 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5237 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5238 // see previously buffered data before it called start(), but with greater risk of overrun. 5239 5240 recordTrack->mRsmpInFront = mRsmpInRear; 5241 recordTrack->mRsmpInUnrel = 0; 5242 // FIXME why reset? 5243 if (recordTrack->mResampler != NULL) { 5244 recordTrack->mResampler->reset(); 5245 } 5246 recordTrack->mState = TrackBase::STARTING_2; 5247 // signal thread to start 5248 mWaitWorkCV.broadcast(); 5249 if (mActiveTracks.indexOf(recordTrack) < 0) { 5250 ALOGV("Record failed to start"); 5251 status = BAD_VALUE; 5252 goto startError; 5253 } 5254 return status; 5255 } 5256 5257startError: 5258 AudioSystem::stopInput(mId); 5259 recordTrack->clearSyncStartEvent(); 5260 // FIXME I wonder why we do not reset the state here? 5261 return status; 5262} 5263 5264void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5265{ 5266 sp<SyncEvent> strongEvent = event.promote(); 5267 5268 if (strongEvent != 0) { 5269 sp<RefBase> ptr = strongEvent->cookie().promote(); 5270 if (ptr != 0) { 5271 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5272 recordTrack->handleSyncStartEvent(strongEvent); 5273 } 5274 } 5275} 5276 5277bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5278 ALOGV("RecordThread::stop"); 5279 AutoMutex _l(mLock); 5280 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5281 return false; 5282 } 5283 // note that threadLoop may still be processing the track at this point [without lock] 5284 recordTrack->mState = TrackBase::PAUSING; 5285 // do not wait for mStartStopCond if exiting 5286 if (exitPending()) { 5287 return true; 5288 } 5289 // FIXME incorrect usage of wait: no explicit predicate or loop 5290 mStartStopCond.wait(mLock); 5291 // if we have been restarted, recordTrack is in mActiveTracks here 5292 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5293 ALOGV("Record stopped OK"); 5294 return true; 5295 } 5296 return false; 5297} 5298 5299bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5300{ 5301 return false; 5302} 5303 5304status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5305{ 5306#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5307 if (!isValidSyncEvent(event)) { 5308 return BAD_VALUE; 5309 } 5310 5311 int eventSession = event->triggerSession(); 5312 status_t ret = NAME_NOT_FOUND; 5313 5314 Mutex::Autolock _l(mLock); 5315 5316 for (size_t i = 0; i < mTracks.size(); i++) { 5317 sp<RecordTrack> track = mTracks[i]; 5318 if (eventSession == track->sessionId()) { 5319 (void) track->setSyncEvent(event); 5320 ret = NO_ERROR; 5321 } 5322 } 5323 return ret; 5324#else 5325 return BAD_VALUE; 5326#endif 5327} 5328 5329// destroyTrack_l() must be called with ThreadBase::mLock held 5330void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5331{ 5332 track->terminate(); 5333 track->mState = TrackBase::STOPPED; 5334 // active tracks are removed by threadLoop() 5335 if (mActiveTracks.indexOf(track) < 0) { 5336 removeTrack_l(track); 5337 } 5338} 5339 5340void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5341{ 5342 mTracks.remove(track); 5343 // need anything related to effects here? 5344} 5345 5346void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5347{ 5348 dumpInternals(fd, args); 5349 dumpTracks(fd, args); 5350 dumpEffectChains(fd, args); 5351} 5352 5353void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5354{ 5355 fdprintf(fd, "\nInput thread %p:\n", this); 5356 5357 if (mActiveTracks.size() > 0) { 5358 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5359 } else { 5360 fdprintf(fd, " No active record clients\n"); 5361 } 5362 5363 dumpBase(fd, args); 5364} 5365 5366void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5367{ 5368 const size_t SIZE = 256; 5369 char buffer[SIZE]; 5370 String8 result; 5371 5372 size_t numtracks = mTracks.size(); 5373 size_t numactive = mActiveTracks.size(); 5374 size_t numactiveseen = 0; 5375 fdprintf(fd, " %d Tracks", numtracks); 5376 if (numtracks) { 5377 fdprintf(fd, " of which %d are active\n", numactive); 5378 RecordTrack::appendDumpHeader(result); 5379 for (size_t i = 0; i < numtracks ; ++i) { 5380 sp<RecordTrack> track = mTracks[i]; 5381 if (track != 0) { 5382 bool active = mActiveTracks.indexOf(track) >= 0; 5383 if (active) { 5384 numactiveseen++; 5385 } 5386 track->dump(buffer, SIZE, active); 5387 result.append(buffer); 5388 } 5389 } 5390 } else { 5391 fdprintf(fd, "\n"); 5392 } 5393 5394 if (numactiveseen != numactive) { 5395 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5396 " not in the track list\n"); 5397 result.append(buffer); 5398 RecordTrack::appendDumpHeader(result); 5399 for (size_t i = 0; i < numactive; ++i) { 5400 sp<RecordTrack> track = mActiveTracks[i]; 5401 if (mTracks.indexOf(track) < 0) { 5402 track->dump(buffer, SIZE, true); 5403 result.append(buffer); 5404 } 5405 } 5406 5407 } 5408 write(fd, result.string(), result.size()); 5409} 5410 5411// AudioBufferProvider interface 5412status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5413 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5414{ 5415 RecordTrack *activeTrack = mRecordTrack; 5416 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5417 if (threadBase == 0) { 5418 buffer->frameCount = 0; 5419 buffer->raw = NULL; 5420 return NOT_ENOUGH_DATA; 5421 } 5422 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5423 int32_t rear = recordThread->mRsmpInRear; 5424 int32_t front = activeTrack->mRsmpInFront; 5425 ssize_t filled = rear - front; 5426 // FIXME should not be P2 (don't want to increase latency) 5427 // FIXME if client not keeping up, discard 5428 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5429 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5430 front &= recordThread->mRsmpInFramesP2 - 1; 5431 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5432 if (part1 > (size_t) filled) { 5433 part1 = filled; 5434 } 5435 size_t ask = buffer->frameCount; 5436 ALOG_ASSERT(ask > 0); 5437 if (part1 > ask) { 5438 part1 = ask; 5439 } 5440 if (part1 == 0) { 5441 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5442 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5443 buffer->raw = NULL; 5444 buffer->frameCount = 0; 5445 activeTrack->mRsmpInUnrel = 0; 5446 return NOT_ENOUGH_DATA; 5447 } 5448 5449 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5450 buffer->frameCount = part1; 5451 activeTrack->mRsmpInUnrel = part1; 5452 return NO_ERROR; 5453} 5454 5455// AudioBufferProvider interface 5456void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5457 AudioBufferProvider::Buffer* buffer) 5458{ 5459 RecordTrack *activeTrack = mRecordTrack; 5460 size_t stepCount = buffer->frameCount; 5461 if (stepCount == 0) { 5462 return; 5463 } 5464 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5465 activeTrack->mRsmpInUnrel -= stepCount; 5466 activeTrack->mRsmpInFront += stepCount; 5467 buffer->raw = NULL; 5468 buffer->frameCount = 0; 5469} 5470 5471bool AudioFlinger::RecordThread::checkForNewParameters_l() 5472{ 5473 bool reconfig = false; 5474 5475 while (!mNewParameters.isEmpty()) { 5476 status_t status = NO_ERROR; 5477 String8 keyValuePair = mNewParameters[0]; 5478 AudioParameter param = AudioParameter(keyValuePair); 5479 int value; 5480 audio_format_t reqFormat = mFormat; 5481 uint32_t samplingRate = mSampleRate; 5482 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5483 5484 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5485 // channel count change can be requested. Do we mandate the first client defines the 5486 // HAL sampling rate and channel count or do we allow changes on the fly? 5487 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5488 samplingRate = value; 5489 reconfig = true; 5490 } 5491 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5492 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5493 status = BAD_VALUE; 5494 } else { 5495 reqFormat = (audio_format_t) value; 5496 reconfig = true; 5497 } 5498 } 5499 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5500 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5501 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5502 status = BAD_VALUE; 5503 } else { 5504 channelMask = mask; 5505 reconfig = true; 5506 } 5507 } 5508 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5509 // do not accept frame count changes if tracks are open as the track buffer 5510 // size depends on frame count and correct behavior would not be guaranteed 5511 // if frame count is changed after track creation 5512 if (mActiveTracks.size() > 0) { 5513 status = INVALID_OPERATION; 5514 } else { 5515 reconfig = true; 5516 } 5517 } 5518 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5519 // forward device change to effects that have requested to be 5520 // aware of attached audio device. 5521 for (size_t i = 0; i < mEffectChains.size(); i++) { 5522 mEffectChains[i]->setDevice_l(value); 5523 } 5524 5525 // store input device and output device but do not forward output device to audio HAL. 5526 // Note that status is ignored by the caller for output device 5527 // (see AudioFlinger::setParameters() 5528 if (audio_is_output_devices(value)) { 5529 mOutDevice = value; 5530 status = BAD_VALUE; 5531 } else { 5532 mInDevice = value; 5533 // disable AEC and NS if the device is a BT SCO headset supporting those 5534 // pre processings 5535 if (mTracks.size() > 0) { 5536 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5537 mAudioFlinger->btNrecIsOff(); 5538 for (size_t i = 0; i < mTracks.size(); i++) { 5539 sp<RecordTrack> track = mTracks[i]; 5540 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5541 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5542 } 5543 } 5544 } 5545 } 5546 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5547 mAudioSource != (audio_source_t)value) { 5548 // forward device change to effects that have requested to be 5549 // aware of attached audio device. 5550 for (size_t i = 0; i < mEffectChains.size(); i++) { 5551 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5552 } 5553 mAudioSource = (audio_source_t)value; 5554 } 5555 5556 if (status == NO_ERROR) { 5557 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5558 keyValuePair.string()); 5559 if (status == INVALID_OPERATION) { 5560 inputStandBy(); 5561 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5562 keyValuePair.string()); 5563 } 5564 if (reconfig) { 5565 if (status == BAD_VALUE && 5566 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5567 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5568 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5569 <= (2 * samplingRate)) && 5570 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5571 <= FCC_2 && 5572 (channelMask == AUDIO_CHANNEL_IN_MONO || 5573 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5574 status = NO_ERROR; 5575 } 5576 if (status == NO_ERROR) { 5577 readInputParameters_l(); 5578 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5579 } 5580 } 5581 } 5582 5583 mNewParameters.removeAt(0); 5584 5585 mParamStatus = status; 5586 mParamCond.signal(); 5587 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5588 // already timed out waiting for the status and will never signal the condition. 5589 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5590 } 5591 return reconfig; 5592} 5593 5594String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5595{ 5596 Mutex::Autolock _l(mLock); 5597 if (initCheck() != NO_ERROR) { 5598 return String8(); 5599 } 5600 5601 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5602 const String8 out_s8(s); 5603 free(s); 5604 return out_s8; 5605} 5606 5607void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5608 AudioSystem::OutputDescriptor desc; 5609 const void *param2 = NULL; 5610 5611 switch (event) { 5612 case AudioSystem::INPUT_OPENED: 5613 case AudioSystem::INPUT_CONFIG_CHANGED: 5614 desc.channelMask = mChannelMask; 5615 desc.samplingRate = mSampleRate; 5616 desc.format = mFormat; 5617 desc.frameCount = mFrameCount; 5618 desc.latency = 0; 5619 param2 = &desc; 5620 break; 5621 5622 case AudioSystem::INPUT_CLOSED: 5623 default: 5624 break; 5625 } 5626 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5627} 5628 5629void AudioFlinger::RecordThread::readInputParameters_l() 5630{ 5631 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5632 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5633 mChannelCount = popcount(mChannelMask); 5634 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5635 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5636 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5637 } 5638 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5639 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5640 mFrameCount = mBufferSize / mFrameSize; 5641 // This is the formula for calculating the temporary buffer size. 5642 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5643 // 1 full output buffer, regardless of the alignment of the available input. 5644 // The value is somewhat arbitrary, and could probably be even larger. 5645 // A larger value should allow more old data to be read after a track calls start(), 5646 // without increasing latency. 5647 mRsmpInFrames = mFrameCount * 7; 5648 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5649 delete[] mRsmpInBuffer; 5650 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5651 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5652 5653 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5654 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5655} 5656 5657uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5658{ 5659 Mutex::Autolock _l(mLock); 5660 if (initCheck() != NO_ERROR) { 5661 return 0; 5662 } 5663 5664 return mInput->stream->get_input_frames_lost(mInput->stream); 5665} 5666 5667uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5668{ 5669 Mutex::Autolock _l(mLock); 5670 uint32_t result = 0; 5671 if (getEffectChain_l(sessionId) != 0) { 5672 result = EFFECT_SESSION; 5673 } 5674 5675 for (size_t i = 0; i < mTracks.size(); ++i) { 5676 if (sessionId == mTracks[i]->sessionId()) { 5677 result |= TRACK_SESSION; 5678 break; 5679 } 5680 } 5681 5682 return result; 5683} 5684 5685KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5686{ 5687 KeyedVector<int, bool> ids; 5688 Mutex::Autolock _l(mLock); 5689 for (size_t j = 0; j < mTracks.size(); ++j) { 5690 sp<RecordThread::RecordTrack> track = mTracks[j]; 5691 int sessionId = track->sessionId(); 5692 if (ids.indexOfKey(sessionId) < 0) { 5693 ids.add(sessionId, true); 5694 } 5695 } 5696 return ids; 5697} 5698 5699AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5700{ 5701 Mutex::Autolock _l(mLock); 5702 AudioStreamIn *input = mInput; 5703 mInput = NULL; 5704 return input; 5705} 5706 5707// this method must always be called either with ThreadBase mLock held or inside the thread loop 5708audio_stream_t* AudioFlinger::RecordThread::stream() const 5709{ 5710 if (mInput == NULL) { 5711 return NULL; 5712 } 5713 return &mInput->stream->common; 5714} 5715 5716status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5717{ 5718 // only one chain per input thread 5719 if (mEffectChains.size() != 0) { 5720 return INVALID_OPERATION; 5721 } 5722 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5723 5724 chain->setInBuffer(NULL); 5725 chain->setOutBuffer(NULL); 5726 5727 checkSuspendOnAddEffectChain_l(chain); 5728 5729 mEffectChains.add(chain); 5730 5731 return NO_ERROR; 5732} 5733 5734size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5735{ 5736 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5737 ALOGW_IF(mEffectChains.size() != 1, 5738 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5739 chain.get(), mEffectChains.size(), this); 5740 if (mEffectChains.size() == 1) { 5741 mEffectChains.removeAt(0); 5742 } 5743 return 0; 5744} 5745 5746}; // namespace android 5747