Threads.cpp revision 5b10a2037a835e790994b9ebec3c2e55052f1f3b
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38 39// NBAIO implementations 40#include <media/nbaio/AudioStreamOutSink.h> 41#include <media/nbaio/MonoPipe.h> 42#include <media/nbaio/MonoPipeReader.h> 43#include <media/nbaio/Pipe.h> 44#include <media/nbaio/PipeReader.h> 45#include <media/nbaio/SourceAudioBufferProvider.h> 46 47#include <powermanager/PowerManager.h> 48 49#include <common_time/cc_helper.h> 50#include <common_time/local_clock.h> 51 52#include "AudioFlinger.h" 53#include "AudioMixer.h" 54#include "FastMixer.h" 55#include "ServiceUtilities.h" 56#include "SchedulingPolicyService.h" 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85// retry counts for buffer fill timeout 86// 50 * ~20msecs = 1 second 87static const int8_t kMaxTrackRetries = 50; 88static const int8_t kMaxTrackStartupRetries = 50; 89// allow less retry attempts on direct output thread. 90// direct outputs can be a scarce resource in audio hardware and should 91// be released as quickly as possible. 92static const int8_t kMaxTrackRetriesDirect = 2; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait for setParameters to complete 101static const nsecs_t kSetParametersTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108// minimum normal sink buffer size, expressed in milliseconds rather than frames 109static const uint32_t kMinNormalSinkBufferSizeMs = 20; 110// maximum normal sink buffer size 111static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 112 113// Offloaded output thread standby delay: allows track transition without going to standby 114static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 115 116// Whether to use fast mixer 117static const enum { 118 FastMixer_Never, // never initialize or use: for debugging only 119 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 120 // normal mixer multiplier is 1 121 FastMixer_Static, // initialize if needed, then use all the time if initialized, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 124 // multiplier is calculated based on min & max normal mixer buffer size 125 // FIXME for FastMixer_Dynamic: 126 // Supporting this option will require fixing HALs that can't handle large writes. 127 // For example, one HAL implementation returns an error from a large write, 128 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 129 // We could either fix the HAL implementations, or provide a wrapper that breaks 130 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 131} kUseFastMixer = FastMixer_Static; 132 133// Priorities for requestPriority 134static const int kPriorityAudioApp = 2; 135static const int kPriorityFastMixer = 3; 136 137// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 138// for the track. The client then sub-divides this into smaller buffers for its use. 139// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 140// So for now we just assume that client is double-buffered for fast tracks. 141// FIXME It would be better for client to tell AudioFlinger the value of N, 142// so AudioFlinger could allocate the right amount of memory. 143// See the client's minBufCount and mNotificationFramesAct calculations for details. 144static const int kFastTrackMultiplier = 2; 145 146// ---------------------------------------------------------------------------- 147 148#ifdef ADD_BATTERY_DATA 149// To collect the amplifier usage 150static void addBatteryData(uint32_t params) { 151 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 152 if (service == NULL) { 153 // it already logged 154 return; 155 } 156 157 service->addBatteryData(params); 158} 159#endif 160 161 162// ---------------------------------------------------------------------------- 163// CPU Stats 164// ---------------------------------------------------------------------------- 165 166class CpuStats { 167public: 168 CpuStats(); 169 void sample(const String8 &title); 170#ifdef DEBUG_CPU_USAGE 171private: 172 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 173 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 174 175 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 176 177 int mCpuNum; // thread's current CPU number 178 int mCpukHz; // frequency of thread's current CPU in kHz 179#endif 180}; 181 182CpuStats::CpuStats() 183#ifdef DEBUG_CPU_USAGE 184 : mCpuNum(-1), mCpukHz(-1) 185#endif 186{ 187} 188 189void CpuStats::sample(const String8 &title 190#ifndef DEBUG_CPU_USAGE 191 __unused 192#endif 193 ) { 194#ifdef DEBUG_CPU_USAGE 195 // get current thread's delta CPU time in wall clock ns 196 double wcNs; 197 bool valid = mCpuUsage.sampleAndEnable(wcNs); 198 199 // record sample for wall clock statistics 200 if (valid) { 201 mWcStats.sample(wcNs); 202 } 203 204 // get the current CPU number 205 int cpuNum = sched_getcpu(); 206 207 // get the current CPU frequency in kHz 208 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 209 210 // check if either CPU number or frequency changed 211 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 212 mCpuNum = cpuNum; 213 mCpukHz = cpukHz; 214 // ignore sample for purposes of cycles 215 valid = false; 216 } 217 218 // if no change in CPU number or frequency, then record sample for cycle statistics 219 if (valid && mCpukHz > 0) { 220 double cycles = wcNs * cpukHz * 0.000001; 221 mHzStats.sample(cycles); 222 } 223 224 unsigned n = mWcStats.n(); 225 // mCpuUsage.elapsed() is expensive, so don't call it every loop 226 if ((n & 127) == 1) { 227 long long elapsed = mCpuUsage.elapsed(); 228 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 229 double perLoop = elapsed / (double) n; 230 double perLoop100 = perLoop * 0.01; 231 double perLoop1k = perLoop * 0.001; 232 double mean = mWcStats.mean(); 233 double stddev = mWcStats.stddev(); 234 double minimum = mWcStats.minimum(); 235 double maximum = mWcStats.maximum(); 236 double meanCycles = mHzStats.mean(); 237 double stddevCycles = mHzStats.stddev(); 238 double minCycles = mHzStats.minimum(); 239 double maxCycles = mHzStats.maximum(); 240 mCpuUsage.resetElapsed(); 241 mWcStats.reset(); 242 mHzStats.reset(); 243 ALOGD("CPU usage for %s over past %.1f secs\n" 244 " (%u mixer loops at %.1f mean ms per loop):\n" 245 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 246 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 247 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 248 title.string(), 249 elapsed * .000000001, n, perLoop * .000001, 250 mean * .001, 251 stddev * .001, 252 minimum * .001, 253 maximum * .001, 254 mean / perLoop100, 255 stddev / perLoop100, 256 minimum / perLoop100, 257 maximum / perLoop100, 258 meanCycles / perLoop1k, 259 stddevCycles / perLoop1k, 260 minCycles / perLoop1k, 261 maxCycles / perLoop1k); 262 263 } 264 } 265#endif 266}; 267 268// ---------------------------------------------------------------------------- 269// ThreadBase 270// ---------------------------------------------------------------------------- 271 272AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 273 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 274 : Thread(false /*canCallJava*/), 275 mType(type), 276 mAudioFlinger(audioFlinger), 277 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 278 // are set by PlaybackThread::readOutputParameters_l() or 279 // RecordThread::readInputParameters_l() 280 mParamStatus(NO_ERROR), 281 //FIXME: mStandby should be true here. Is this some kind of hack? 282 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 283 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 284 // mName will be set by concrete (non-virtual) subclass 285 mDeathRecipient(new PMDeathRecipient(this)) 286{ 287} 288 289AudioFlinger::ThreadBase::~ThreadBase() 290{ 291 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 292 for (size_t i = 0; i < mConfigEvents.size(); i++) { 293 delete mConfigEvents[i]; 294 } 295 mConfigEvents.clear(); 296 297 mParamCond.broadcast(); 298 // do not lock the mutex in destructor 299 releaseWakeLock_l(); 300 if (mPowerManager != 0) { 301 sp<IBinder> binder = mPowerManager->asBinder(); 302 binder->unlinkToDeath(mDeathRecipient); 303 } 304} 305 306status_t AudioFlinger::ThreadBase::readyToRun() 307{ 308 status_t status = initCheck(); 309 if (status == NO_ERROR) { 310 ALOGI("AudioFlinger's thread %p ready to run", this); 311 } else { 312 ALOGE("No working audio driver found."); 313 } 314 return status; 315} 316 317void AudioFlinger::ThreadBase::exit() 318{ 319 ALOGV("ThreadBase::exit"); 320 // do any cleanup required for exit to succeed 321 preExit(); 322 { 323 // This lock prevents the following race in thread (uniprocessor for illustration): 324 // if (!exitPending()) { 325 // // context switch from here to exit() 326 // // exit() calls requestExit(), what exitPending() observes 327 // // exit() calls signal(), which is dropped since no waiters 328 // // context switch back from exit() to here 329 // mWaitWorkCV.wait(...); 330 // // now thread is hung 331 // } 332 AutoMutex lock(mLock); 333 requestExit(); 334 mWaitWorkCV.broadcast(); 335 } 336 // When Thread::requestExitAndWait is made virtual and this method is renamed to 337 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 338 requestExitAndWait(); 339} 340 341status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 342{ 343 status_t status; 344 345 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 346 Mutex::Autolock _l(mLock); 347 348 mNewParameters.add(keyValuePairs); 349 mWaitWorkCV.signal(); 350 // wait condition with timeout in case the thread loop has exited 351 // before the request could be processed 352 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 353 status = mParamStatus; 354 mWaitWorkCV.signal(); 355 } else { 356 status = TIMED_OUT; 357 } 358 return status; 359} 360 361void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 362{ 363 Mutex::Autolock _l(mLock); 364 sendIoConfigEvent_l(event, param); 365} 366 367// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 368void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 369{ 370 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 371 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 372 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 373 param); 374 mWaitWorkCV.signal(); 375} 376 377// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 378void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 379{ 380 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 381 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 382 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 383 mConfigEvents.size(), pid, tid, prio); 384 mWaitWorkCV.signal(); 385} 386 387void AudioFlinger::ThreadBase::processConfigEvents() 388{ 389 Mutex::Autolock _l(mLock); 390 processConfigEvents_l(); 391} 392 393// post condition: mConfigEvents.isEmpty() 394void AudioFlinger::ThreadBase::processConfigEvents_l() 395{ 396 while (!mConfigEvents.isEmpty()) { 397 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 398 ConfigEvent *event = mConfigEvents[0]; 399 mConfigEvents.removeAt(0); 400 // release mLock before locking AudioFlinger mLock: lock order is always 401 // AudioFlinger then ThreadBase to avoid cross deadlock 402 mLock.unlock(); 403 switch (event->type()) { 404 case CFG_EVENT_PRIO: { 405 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 406 // FIXME Need to understand why this has be done asynchronously 407 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 408 true /*asynchronous*/); 409 if (err != 0) { 410 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 411 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 412 } 413 } break; 414 case CFG_EVENT_IO: { 415 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 416 { 417 Mutex::Autolock _l(mAudioFlinger->mLock); 418 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 419 } 420 } break; 421 default: 422 ALOGE("processConfigEvents() unknown event type %d", event->type()); 423 break; 424 } 425 delete event; 426 mLock.lock(); 427 } 428} 429 430String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 431 String8 s; 432 if (output) { 433 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 434 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 435 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 436 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 437 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 438 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 439 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 440 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 441 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 442 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 443 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 444 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 445 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 446 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 447 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 448 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 449 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 450 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 451 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 452 } else { 453 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 454 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 455 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 456 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 457 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 458 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 459 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 460 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 461 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 462 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 463 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 464 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 465 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 466 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 467 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 468 } 469 int len = s.length(); 470 if (s.length() > 2) { 471 char *str = s.lockBuffer(len); 472 s.unlockBuffer(len - 2); 473 } 474 return s; 475} 476 477void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 478{ 479 const size_t SIZE = 256; 480 char buffer[SIZE]; 481 String8 result; 482 483 bool locked = AudioFlinger::dumpTryLock(mLock); 484 if (!locked) { 485 fdprintf(fd, "thread %p maybe dead locked\n", this); 486 } 487 488 fdprintf(fd, " I/O handle: %d\n", mId); 489 fdprintf(fd, " TID: %d\n", getTid()); 490 fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 491 fdprintf(fd, " Sample rate: %u\n", mSampleRate); 492 fdprintf(fd, " HAL frame count: %zu\n", mFrameCount); 493 fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 494 fdprintf(fd, " Channel Count: %u\n", mChannelCount); 495 fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 496 channelMaskToString(mChannelMask, mType != RECORD).string()); 497 fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 498 fdprintf(fd, " Frame size: %zu\n", mFrameSize); 499 fdprintf(fd, " Pending setParameters commands:"); 500 size_t numParams = mNewParameters.size(); 501 if (numParams) { 502 fdprintf(fd, "\n Index Command"); 503 for (size_t i = 0; i < numParams; ++i) { 504 fdprintf(fd, "\n %02zu ", i); 505 fdprintf(fd, mNewParameters[i]); 506 } 507 fdprintf(fd, "\n"); 508 } else { 509 fdprintf(fd, " none\n"); 510 } 511 fdprintf(fd, " Pending config events:"); 512 size_t numConfig = mConfigEvents.size(); 513 if (numConfig) { 514 for (size_t i = 0; i < numConfig; i++) { 515 mConfigEvents[i]->dump(buffer, SIZE); 516 fdprintf(fd, "\n %s", buffer); 517 } 518 fdprintf(fd, "\n"); 519 } else { 520 fdprintf(fd, " none\n"); 521 } 522 523 if (locked) { 524 mLock.unlock(); 525 } 526} 527 528void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 529{ 530 const size_t SIZE = 256; 531 char buffer[SIZE]; 532 String8 result; 533 534 size_t numEffectChains = mEffectChains.size(); 535 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 536 write(fd, buffer, strlen(buffer)); 537 538 for (size_t i = 0; i < numEffectChains; ++i) { 539 sp<EffectChain> chain = mEffectChains[i]; 540 if (chain != 0) { 541 chain->dump(fd, args); 542 } 543 } 544} 545 546void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 547{ 548 Mutex::Autolock _l(mLock); 549 acquireWakeLock_l(uid); 550} 551 552String16 AudioFlinger::ThreadBase::getWakeLockTag() 553{ 554 switch (mType) { 555 case MIXER: 556 return String16("AudioMix"); 557 case DIRECT: 558 return String16("AudioDirectOut"); 559 case DUPLICATING: 560 return String16("AudioDup"); 561 case RECORD: 562 return String16("AudioIn"); 563 case OFFLOAD: 564 return String16("AudioOffload"); 565 default: 566 ALOG_ASSERT(false); 567 return String16("AudioUnknown"); 568 } 569} 570 571void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 572{ 573 getPowerManager_l(); 574 if (mPowerManager != 0) { 575 sp<IBinder> binder = new BBinder(); 576 status_t status; 577 if (uid >= 0) { 578 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 579 binder, 580 getWakeLockTag(), 581 String16("media"), 582 uid); 583 } else { 584 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 585 binder, 586 getWakeLockTag(), 587 String16("media")); 588 } 589 if (status == NO_ERROR) { 590 mWakeLockToken = binder; 591 } 592 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 593 } 594} 595 596void AudioFlinger::ThreadBase::releaseWakeLock() 597{ 598 Mutex::Autolock _l(mLock); 599 releaseWakeLock_l(); 600} 601 602void AudioFlinger::ThreadBase::releaseWakeLock_l() 603{ 604 if (mWakeLockToken != 0) { 605 ALOGV("releaseWakeLock_l() %s", mName); 606 if (mPowerManager != 0) { 607 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 608 } 609 mWakeLockToken.clear(); 610 } 611} 612 613void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 614 Mutex::Autolock _l(mLock); 615 updateWakeLockUids_l(uids); 616} 617 618void AudioFlinger::ThreadBase::getPowerManager_l() { 619 620 if (mPowerManager == 0) { 621 // use checkService() to avoid blocking if power service is not up yet 622 sp<IBinder> binder = 623 defaultServiceManager()->checkService(String16("power")); 624 if (binder == 0) { 625 ALOGW("Thread %s cannot connect to the power manager service", mName); 626 } else { 627 mPowerManager = interface_cast<IPowerManager>(binder); 628 binder->linkToDeath(mDeathRecipient); 629 } 630 } 631} 632 633void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 634 635 getPowerManager_l(); 636 if (mWakeLockToken == NULL) { 637 ALOGE("no wake lock to update!"); 638 return; 639 } 640 if (mPowerManager != 0) { 641 sp<IBinder> binder = new BBinder(); 642 status_t status; 643 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 644 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 645 } 646} 647 648void AudioFlinger::ThreadBase::clearPowerManager() 649{ 650 Mutex::Autolock _l(mLock); 651 releaseWakeLock_l(); 652 mPowerManager.clear(); 653} 654 655void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 656{ 657 sp<ThreadBase> thread = mThread.promote(); 658 if (thread != 0) { 659 thread->clearPowerManager(); 660 } 661 ALOGW("power manager service died !!!"); 662} 663 664void AudioFlinger::ThreadBase::setEffectSuspended( 665 const effect_uuid_t *type, bool suspend, int sessionId) 666{ 667 Mutex::Autolock _l(mLock); 668 setEffectSuspended_l(type, suspend, sessionId); 669} 670 671void AudioFlinger::ThreadBase::setEffectSuspended_l( 672 const effect_uuid_t *type, bool suspend, int sessionId) 673{ 674 sp<EffectChain> chain = getEffectChain_l(sessionId); 675 if (chain != 0) { 676 if (type != NULL) { 677 chain->setEffectSuspended_l(type, suspend); 678 } else { 679 chain->setEffectSuspendedAll_l(suspend); 680 } 681 } 682 683 updateSuspendedSessions_l(type, suspend, sessionId); 684} 685 686void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 687{ 688 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 689 if (index < 0) { 690 return; 691 } 692 693 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 694 mSuspendedSessions.valueAt(index); 695 696 for (size_t i = 0; i < sessionEffects.size(); i++) { 697 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 698 for (int j = 0; j < desc->mRefCount; j++) { 699 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 700 chain->setEffectSuspendedAll_l(true); 701 } else { 702 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 703 desc->mType.timeLow); 704 chain->setEffectSuspended_l(&desc->mType, true); 705 } 706 } 707 } 708} 709 710void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 711 bool suspend, 712 int sessionId) 713{ 714 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 715 716 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 717 718 if (suspend) { 719 if (index >= 0) { 720 sessionEffects = mSuspendedSessions.valueAt(index); 721 } else { 722 mSuspendedSessions.add(sessionId, sessionEffects); 723 } 724 } else { 725 if (index < 0) { 726 return; 727 } 728 sessionEffects = mSuspendedSessions.valueAt(index); 729 } 730 731 732 int key = EffectChain::kKeyForSuspendAll; 733 if (type != NULL) { 734 key = type->timeLow; 735 } 736 index = sessionEffects.indexOfKey(key); 737 738 sp<SuspendedSessionDesc> desc; 739 if (suspend) { 740 if (index >= 0) { 741 desc = sessionEffects.valueAt(index); 742 } else { 743 desc = new SuspendedSessionDesc(); 744 if (type != NULL) { 745 desc->mType = *type; 746 } 747 sessionEffects.add(key, desc); 748 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 749 } 750 desc->mRefCount++; 751 } else { 752 if (index < 0) { 753 return; 754 } 755 desc = sessionEffects.valueAt(index); 756 if (--desc->mRefCount == 0) { 757 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 758 sessionEffects.removeItemsAt(index); 759 if (sessionEffects.isEmpty()) { 760 ALOGV("updateSuspendedSessions_l() restore removing session %d", 761 sessionId); 762 mSuspendedSessions.removeItem(sessionId); 763 } 764 } 765 } 766 if (!sessionEffects.isEmpty()) { 767 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 768 } 769} 770 771void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 772 bool enabled, 773 int sessionId) 774{ 775 Mutex::Autolock _l(mLock); 776 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 777} 778 779void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 780 bool enabled, 781 int sessionId) 782{ 783 if (mType != RECORD) { 784 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 785 // another session. This gives the priority to well behaved effect control panels 786 // and applications not using global effects. 787 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 788 // global effects 789 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 790 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 791 } 792 } 793 794 sp<EffectChain> chain = getEffectChain_l(sessionId); 795 if (chain != 0) { 796 chain->checkSuspendOnEffectEnabled(effect, enabled); 797 } 798} 799 800// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 801sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 802 const sp<AudioFlinger::Client>& client, 803 const sp<IEffectClient>& effectClient, 804 int32_t priority, 805 int sessionId, 806 effect_descriptor_t *desc, 807 int *enabled, 808 status_t *status) 809{ 810 sp<EffectModule> effect; 811 sp<EffectHandle> handle; 812 status_t lStatus; 813 sp<EffectChain> chain; 814 bool chainCreated = false; 815 bool effectCreated = false; 816 bool effectRegistered = false; 817 818 lStatus = initCheck(); 819 if (lStatus != NO_ERROR) { 820 ALOGW("createEffect_l() Audio driver not initialized."); 821 goto Exit; 822 } 823 824 // Reject any effect on Direct output threads for now, since the format of 825 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 826 if (mType == DIRECT) { 827 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 828 desc->name, mName); 829 lStatus = BAD_VALUE; 830 goto Exit; 831 } 832 833 // Allow global effects only on offloaded and mixer threads 834 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 835 switch (mType) { 836 case MIXER: 837 case OFFLOAD: 838 break; 839 case DIRECT: 840 case DUPLICATING: 841 case RECORD: 842 default: 843 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 844 lStatus = BAD_VALUE; 845 goto Exit; 846 } 847 } 848 849 // Only Pre processor effects are allowed on input threads and only on input threads 850 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 851 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 852 desc->name, desc->flags, mType); 853 lStatus = BAD_VALUE; 854 goto Exit; 855 } 856 857 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 858 859 { // scope for mLock 860 Mutex::Autolock _l(mLock); 861 862 // check for existing effect chain with the requested audio session 863 chain = getEffectChain_l(sessionId); 864 if (chain == 0) { 865 // create a new chain for this session 866 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 867 chain = new EffectChain(this, sessionId); 868 addEffectChain_l(chain); 869 chain->setStrategy(getStrategyForSession_l(sessionId)); 870 chainCreated = true; 871 } else { 872 effect = chain->getEffectFromDesc_l(desc); 873 } 874 875 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 876 877 if (effect == 0) { 878 int id = mAudioFlinger->nextUniqueId(); 879 // Check CPU and memory usage 880 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 881 if (lStatus != NO_ERROR) { 882 goto Exit; 883 } 884 effectRegistered = true; 885 // create a new effect module if none present in the chain 886 effect = new EffectModule(this, chain, desc, id, sessionId); 887 lStatus = effect->status(); 888 if (lStatus != NO_ERROR) { 889 goto Exit; 890 } 891 effect->setOffloaded(mType == OFFLOAD, mId); 892 893 lStatus = chain->addEffect_l(effect); 894 if (lStatus != NO_ERROR) { 895 goto Exit; 896 } 897 effectCreated = true; 898 899 effect->setDevice(mOutDevice); 900 effect->setDevice(mInDevice); 901 effect->setMode(mAudioFlinger->getMode()); 902 effect->setAudioSource(mAudioSource); 903 } 904 // create effect handle and connect it to effect module 905 handle = new EffectHandle(effect, client, effectClient, priority); 906 lStatus = handle->initCheck(); 907 if (lStatus == OK) { 908 lStatus = effect->addHandle(handle.get()); 909 } 910 if (enabled != NULL) { 911 *enabled = (int)effect->isEnabled(); 912 } 913 } 914 915Exit: 916 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 917 Mutex::Autolock _l(mLock); 918 if (effectCreated) { 919 chain->removeEffect_l(effect); 920 } 921 if (effectRegistered) { 922 AudioSystem::unregisterEffect(effect->id()); 923 } 924 if (chainCreated) { 925 removeEffectChain_l(chain); 926 } 927 handle.clear(); 928 } 929 930 *status = lStatus; 931 return handle; 932} 933 934sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 935{ 936 Mutex::Autolock _l(mLock); 937 return getEffect_l(sessionId, effectId); 938} 939 940sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 941{ 942 sp<EffectChain> chain = getEffectChain_l(sessionId); 943 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 944} 945 946// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 947// PlaybackThread::mLock held 948status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 949{ 950 // check for existing effect chain with the requested audio session 951 int sessionId = effect->sessionId(); 952 sp<EffectChain> chain = getEffectChain_l(sessionId); 953 bool chainCreated = false; 954 955 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 956 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 957 this, effect->desc().name, effect->desc().flags); 958 959 if (chain == 0) { 960 // create a new chain for this session 961 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 962 chain = new EffectChain(this, sessionId); 963 addEffectChain_l(chain); 964 chain->setStrategy(getStrategyForSession_l(sessionId)); 965 chainCreated = true; 966 } 967 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 968 969 if (chain->getEffectFromId_l(effect->id()) != 0) { 970 ALOGW("addEffect_l() %p effect %s already present in chain %p", 971 this, effect->desc().name, chain.get()); 972 return BAD_VALUE; 973 } 974 975 effect->setOffloaded(mType == OFFLOAD, mId); 976 977 status_t status = chain->addEffect_l(effect); 978 if (status != NO_ERROR) { 979 if (chainCreated) { 980 removeEffectChain_l(chain); 981 } 982 return status; 983 } 984 985 effect->setDevice(mOutDevice); 986 effect->setDevice(mInDevice); 987 effect->setMode(mAudioFlinger->getMode()); 988 effect->setAudioSource(mAudioSource); 989 return NO_ERROR; 990} 991 992void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 993 994 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 995 effect_descriptor_t desc = effect->desc(); 996 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 997 detachAuxEffect_l(effect->id()); 998 } 999 1000 sp<EffectChain> chain = effect->chain().promote(); 1001 if (chain != 0) { 1002 // remove effect chain if removing last effect 1003 if (chain->removeEffect_l(effect) == 0) { 1004 removeEffectChain_l(chain); 1005 } 1006 } else { 1007 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1008 } 1009} 1010 1011void AudioFlinger::ThreadBase::lockEffectChains_l( 1012 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1013{ 1014 effectChains = mEffectChains; 1015 for (size_t i = 0; i < mEffectChains.size(); i++) { 1016 mEffectChains[i]->lock(); 1017 } 1018} 1019 1020void AudioFlinger::ThreadBase::unlockEffectChains( 1021 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1022{ 1023 for (size_t i = 0; i < effectChains.size(); i++) { 1024 effectChains[i]->unlock(); 1025 } 1026} 1027 1028sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 return getEffectChain_l(sessionId); 1032} 1033 1034sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1035{ 1036 size_t size = mEffectChains.size(); 1037 for (size_t i = 0; i < size; i++) { 1038 if (mEffectChains[i]->sessionId() == sessionId) { 1039 return mEffectChains[i]; 1040 } 1041 } 1042 return 0; 1043} 1044 1045void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1046{ 1047 Mutex::Autolock _l(mLock); 1048 size_t size = mEffectChains.size(); 1049 for (size_t i = 0; i < size; i++) { 1050 mEffectChains[i]->setMode_l(mode); 1051 } 1052} 1053 1054void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1055 EffectHandle *handle, 1056 bool unpinIfLast) { 1057 1058 Mutex::Autolock _l(mLock); 1059 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1060 // delete the effect module if removing last handle on it 1061 if (effect->removeHandle(handle) == 0) { 1062 if (!effect->isPinned() || unpinIfLast) { 1063 removeEffect_l(effect); 1064 AudioSystem::unregisterEffect(effect->id()); 1065 } 1066 } 1067} 1068 1069// ---------------------------------------------------------------------------- 1070// Playback 1071// ---------------------------------------------------------------------------- 1072 1073AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1074 AudioStreamOut* output, 1075 audio_io_handle_t id, 1076 audio_devices_t device, 1077 type_t type) 1078 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1079 mNormalFrameCount(0), mSinkBuffer(NULL), 1080 mMixerBufferEnabled(false), 1081 mMixerBuffer(NULL), 1082 mMixerBufferSize(0), 1083 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1084 mMixerBufferValid(false), 1085 mEffectBufferEnabled(false), 1086 mEffectBuffer(NULL), 1087 mEffectBufferSize(0), 1088 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1089 mEffectBufferValid(false), 1090 mSuspended(0), mBytesWritten(0), 1091 mActiveTracksGeneration(0), 1092 // mStreamTypes[] initialized in constructor body 1093 mOutput(output), 1094 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1095 mMixerStatus(MIXER_IDLE), 1096 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1097 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1098 mBytesRemaining(0), 1099 mCurrentWriteLength(0), 1100 mUseAsyncWrite(false), 1101 mWriteAckSequence(0), 1102 mDrainSequence(0), 1103 mSignalPending(false), 1104 mScreenState(AudioFlinger::mScreenState), 1105 // index 0 is reserved for normal mixer's submix 1106 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1107 // mLatchD, mLatchQ, 1108 mLatchDValid(false), mLatchQValid(false) 1109{ 1110 snprintf(mName, kNameLength, "AudioOut_%X", id); 1111 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1112 1113 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1114 // it would be safer to explicitly pass initial masterVolume/masterMute as 1115 // parameter. 1116 // 1117 // If the HAL we are using has support for master volume or master mute, 1118 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1119 // and the mute set to false). 1120 mMasterVolume = audioFlinger->masterVolume_l(); 1121 mMasterMute = audioFlinger->masterMute_l(); 1122 if (mOutput && mOutput->audioHwDev) { 1123 if (mOutput->audioHwDev->canSetMasterVolume()) { 1124 mMasterVolume = 1.0; 1125 } 1126 1127 if (mOutput->audioHwDev->canSetMasterMute()) { 1128 mMasterMute = false; 1129 } 1130 } 1131 1132 readOutputParameters_l(); 1133 1134 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1135 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1136 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1137 stream = (audio_stream_type_t) (stream + 1)) { 1138 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1139 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1140 } 1141 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1142 // because mAudioFlinger doesn't have one to copy from 1143} 1144 1145AudioFlinger::PlaybackThread::~PlaybackThread() 1146{ 1147 mAudioFlinger->unregisterWriter(mNBLogWriter); 1148 free(mSinkBuffer); 1149 free(mMixerBuffer); 1150 free(mEffectBuffer); 1151} 1152 1153void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1154{ 1155 dumpInternals(fd, args); 1156 dumpTracks(fd, args); 1157 dumpEffectChains(fd, args); 1158} 1159 1160void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1161{ 1162 const size_t SIZE = 256; 1163 char buffer[SIZE]; 1164 String8 result; 1165 1166 result.appendFormat(" Stream volumes in dB: "); 1167 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1168 const stream_type_t *st = &mStreamTypes[i]; 1169 if (i > 0) { 1170 result.appendFormat(", "); 1171 } 1172 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1173 if (st->mute) { 1174 result.append("M"); 1175 } 1176 } 1177 result.append("\n"); 1178 write(fd, result.string(), result.length()); 1179 result.clear(); 1180 1181 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1182 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1183 fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1184 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1185 1186 size_t numtracks = mTracks.size(); 1187 size_t numactive = mActiveTracks.size(); 1188 fdprintf(fd, " %d Tracks", numtracks); 1189 size_t numactiveseen = 0; 1190 if (numtracks) { 1191 fdprintf(fd, " of which %d are active\n", numactive); 1192 Track::appendDumpHeader(result); 1193 for (size_t i = 0; i < numtracks; ++i) { 1194 sp<Track> track = mTracks[i]; 1195 if (track != 0) { 1196 bool active = mActiveTracks.indexOf(track) >= 0; 1197 if (active) { 1198 numactiveseen++; 1199 } 1200 track->dump(buffer, SIZE, active); 1201 result.append(buffer); 1202 } 1203 } 1204 } else { 1205 result.append("\n"); 1206 } 1207 if (numactiveseen != numactive) { 1208 // some tracks in the active list were not in the tracks list 1209 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1210 " not in the track list\n"); 1211 result.append(buffer); 1212 Track::appendDumpHeader(result); 1213 for (size_t i = 0; i < numactive; ++i) { 1214 sp<Track> track = mActiveTracks[i].promote(); 1215 if (track != 0 && mTracks.indexOf(track) < 0) { 1216 track->dump(buffer, SIZE, true); 1217 result.append(buffer); 1218 } 1219 } 1220 } 1221 1222 write(fd, result.string(), result.size()); 1223 1224} 1225 1226void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1227{ 1228 fdprintf(fd, "\nOutput thread %p:\n", this); 1229 fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1230 fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1231 fdprintf(fd, " Total writes: %d\n", mNumWrites); 1232 fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1233 fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1234 fdprintf(fd, " Suspend count: %d\n", mSuspended); 1235 fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1236 fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1237 fdprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1238 fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1239 1240 dumpBase(fd, args); 1241} 1242 1243// Thread virtuals 1244 1245void AudioFlinger::PlaybackThread::onFirstRef() 1246{ 1247 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1248} 1249 1250// ThreadBase virtuals 1251void AudioFlinger::PlaybackThread::preExit() 1252{ 1253 ALOGV(" preExit()"); 1254 // FIXME this is using hard-coded strings but in the future, this functionality will be 1255 // converted to use audio HAL extensions required to support tunneling 1256 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1257} 1258 1259// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1260sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1261 const sp<AudioFlinger::Client>& client, 1262 audio_stream_type_t streamType, 1263 uint32_t sampleRate, 1264 audio_format_t format, 1265 audio_channel_mask_t channelMask, 1266 size_t *pFrameCount, 1267 const sp<IMemory>& sharedBuffer, 1268 int sessionId, 1269 IAudioFlinger::track_flags_t *flags, 1270 pid_t tid, 1271 int uid, 1272 status_t *status) 1273{ 1274 size_t frameCount = *pFrameCount; 1275 sp<Track> track; 1276 status_t lStatus; 1277 1278 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1279 1280 // client expresses a preference for FAST, but we get the final say 1281 if (*flags & IAudioFlinger::TRACK_FAST) { 1282 if ( 1283 // not timed 1284 (!isTimed) && 1285 // either of these use cases: 1286 ( 1287 // use case 1: shared buffer with any frame count 1288 ( 1289 (sharedBuffer != 0) 1290 ) || 1291 // use case 2: callback handler and frame count is default or at least as large as HAL 1292 ( 1293 (tid != -1) && 1294 ((frameCount == 0) || 1295 (frameCount >= mFrameCount)) 1296 ) 1297 ) && 1298 // PCM data 1299 audio_is_linear_pcm(format) && 1300 // mono or stereo 1301 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1302 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1303 // hardware sample rate 1304 (sampleRate == mSampleRate) && 1305 // normal mixer has an associated fast mixer 1306 hasFastMixer() && 1307 // there are sufficient fast track slots available 1308 (mFastTrackAvailMask != 0) 1309 // FIXME test that MixerThread for this fast track has a capable output HAL 1310 // FIXME add a permission test also? 1311 ) { 1312 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1313 if (frameCount == 0) { 1314 frameCount = mFrameCount * kFastTrackMultiplier; 1315 } 1316 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1317 frameCount, mFrameCount); 1318 } else { 1319 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1320 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1321 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1322 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1323 audio_is_linear_pcm(format), 1324 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1325 *flags &= ~IAudioFlinger::TRACK_FAST; 1326 // For compatibility with AudioTrack calculation, buffer depth is forced 1327 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1328 // This is probably too conservative, but legacy application code may depend on it. 1329 // If you change this calculation, also review the start threshold which is related. 1330 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1331 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1332 if (minBufCount < 2) { 1333 minBufCount = 2; 1334 } 1335 size_t minFrameCount = mNormalFrameCount * minBufCount; 1336 if (frameCount < minFrameCount) { 1337 frameCount = minFrameCount; 1338 } 1339 } 1340 } 1341 *pFrameCount = frameCount; 1342 1343 if (mType == DIRECT) { 1344 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1345 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1346 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1347 "for output %p with format %#x", 1348 sampleRate, format, channelMask, mOutput, mFormat); 1349 lStatus = BAD_VALUE; 1350 goto Exit; 1351 } 1352 } 1353 } else if (mType == OFFLOAD) { 1354 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1355 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1356 "for output %p with format %#x", 1357 sampleRate, format, channelMask, mOutput, mFormat); 1358 lStatus = BAD_VALUE; 1359 goto Exit; 1360 } 1361 } else { 1362 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1363 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1364 "for output %p with format %#x", 1365 format, mOutput, mFormat); 1366 lStatus = BAD_VALUE; 1367 goto Exit; 1368 } 1369 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1370 if (sampleRate > mSampleRate*2) { 1371 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1372 lStatus = BAD_VALUE; 1373 goto Exit; 1374 } 1375 } 1376 1377 lStatus = initCheck(); 1378 if (lStatus != NO_ERROR) { 1379 ALOGE("Audio driver not initialized."); 1380 goto Exit; 1381 } 1382 1383 { // scope for mLock 1384 Mutex::Autolock _l(mLock); 1385 1386 // all tracks in same audio session must share the same routing strategy otherwise 1387 // conflicts will happen when tracks are moved from one output to another by audio policy 1388 // manager 1389 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1390 for (size_t i = 0; i < mTracks.size(); ++i) { 1391 sp<Track> t = mTracks[i]; 1392 if (t != 0 && !t->isOutputTrack()) { 1393 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1394 if (sessionId == t->sessionId() && strategy != actual) { 1395 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1396 strategy, actual); 1397 lStatus = BAD_VALUE; 1398 goto Exit; 1399 } 1400 } 1401 } 1402 1403 if (!isTimed) { 1404 track = new Track(this, client, streamType, sampleRate, format, 1405 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1406 } else { 1407 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1408 channelMask, frameCount, sharedBuffer, sessionId, uid); 1409 } 1410 1411 // new Track always returns non-NULL, 1412 // but TimedTrack::create() is a factory that could fail by returning NULL 1413 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1414 if (lStatus != NO_ERROR) { 1415 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1416 // track must be cleared from the caller as the caller has the AF lock 1417 goto Exit; 1418 } 1419 1420 mTracks.add(track); 1421 1422 sp<EffectChain> chain = getEffectChain_l(sessionId); 1423 if (chain != 0) { 1424 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1425 track->setMainBuffer(chain->inBuffer()); 1426 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1427 chain->incTrackCnt(); 1428 } 1429 1430 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1431 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1432 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1433 // so ask activity manager to do this on our behalf 1434 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1435 } 1436 } 1437 1438 lStatus = NO_ERROR; 1439 1440Exit: 1441 *status = lStatus; 1442 return track; 1443} 1444 1445uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1446{ 1447 return latency; 1448} 1449 1450uint32_t AudioFlinger::PlaybackThread::latency() const 1451{ 1452 Mutex::Autolock _l(mLock); 1453 return latency_l(); 1454} 1455uint32_t AudioFlinger::PlaybackThread::latency_l() const 1456{ 1457 if (initCheck() == NO_ERROR) { 1458 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1459 } else { 1460 return 0; 1461 } 1462} 1463 1464void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1465{ 1466 Mutex::Autolock _l(mLock); 1467 // Don't apply master volume in SW if our HAL can do it for us. 1468 if (mOutput && mOutput->audioHwDev && 1469 mOutput->audioHwDev->canSetMasterVolume()) { 1470 mMasterVolume = 1.0; 1471 } else { 1472 mMasterVolume = value; 1473 } 1474} 1475 1476void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1477{ 1478 Mutex::Autolock _l(mLock); 1479 // Don't apply master mute in SW if our HAL can do it for us. 1480 if (mOutput && mOutput->audioHwDev && 1481 mOutput->audioHwDev->canSetMasterMute()) { 1482 mMasterMute = false; 1483 } else { 1484 mMasterMute = muted; 1485 } 1486} 1487 1488void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1489{ 1490 Mutex::Autolock _l(mLock); 1491 mStreamTypes[stream].volume = value; 1492 broadcast_l(); 1493} 1494 1495void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1496{ 1497 Mutex::Autolock _l(mLock); 1498 mStreamTypes[stream].mute = muted; 1499 broadcast_l(); 1500} 1501 1502float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1503{ 1504 Mutex::Autolock _l(mLock); 1505 return mStreamTypes[stream].volume; 1506} 1507 1508// addTrack_l() must be called with ThreadBase::mLock held 1509status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1510{ 1511 status_t status = ALREADY_EXISTS; 1512 1513 // set retry count for buffer fill 1514 track->mRetryCount = kMaxTrackStartupRetries; 1515 if (mActiveTracks.indexOf(track) < 0) { 1516 // the track is newly added, make sure it fills up all its 1517 // buffers before playing. This is to ensure the client will 1518 // effectively get the latency it requested. 1519 if (!track->isOutputTrack()) { 1520 TrackBase::track_state state = track->mState; 1521 mLock.unlock(); 1522 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1523 mLock.lock(); 1524 // abort track was stopped/paused while we released the lock 1525 if (state != track->mState) { 1526 if (status == NO_ERROR) { 1527 mLock.unlock(); 1528 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1529 mLock.lock(); 1530 } 1531 return INVALID_OPERATION; 1532 } 1533 // abort if start is rejected by audio policy manager 1534 if (status != NO_ERROR) { 1535 return PERMISSION_DENIED; 1536 } 1537#ifdef ADD_BATTERY_DATA 1538 // to track the speaker usage 1539 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1540#endif 1541 } 1542 1543 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1544 track->mResetDone = false; 1545 track->mPresentationCompleteFrames = 0; 1546 mActiveTracks.add(track); 1547 mWakeLockUids.add(track->uid()); 1548 mActiveTracksGeneration++; 1549 mLatestActiveTrack = track; 1550 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1551 if (chain != 0) { 1552 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1553 track->sessionId()); 1554 chain->incActiveTrackCnt(); 1555 } 1556 1557 status = NO_ERROR; 1558 } 1559 1560 onAddNewTrack_l(); 1561 return status; 1562} 1563 1564bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1565{ 1566 track->terminate(); 1567 // active tracks are removed by threadLoop() 1568 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1569 track->mState = TrackBase::STOPPED; 1570 if (!trackActive) { 1571 removeTrack_l(track); 1572 } else if (track->isFastTrack() || track->isOffloaded()) { 1573 track->mState = TrackBase::STOPPING_1; 1574 } 1575 1576 return trackActive; 1577} 1578 1579void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1580{ 1581 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1582 mTracks.remove(track); 1583 deleteTrackName_l(track->name()); 1584 // redundant as track is about to be destroyed, for dumpsys only 1585 track->mName = -1; 1586 if (track->isFastTrack()) { 1587 int index = track->mFastIndex; 1588 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1589 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1590 mFastTrackAvailMask |= 1 << index; 1591 // redundant as track is about to be destroyed, for dumpsys only 1592 track->mFastIndex = -1; 1593 } 1594 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1595 if (chain != 0) { 1596 chain->decTrackCnt(); 1597 } 1598} 1599 1600void AudioFlinger::PlaybackThread::broadcast_l() 1601{ 1602 // Thread could be blocked waiting for async 1603 // so signal it to handle state changes immediately 1604 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1605 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1606 mSignalPending = true; 1607 mWaitWorkCV.broadcast(); 1608} 1609 1610String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1611{ 1612 Mutex::Autolock _l(mLock); 1613 if (initCheck() != NO_ERROR) { 1614 return String8(); 1615 } 1616 1617 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1618 const String8 out_s8(s); 1619 free(s); 1620 return out_s8; 1621} 1622 1623// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1624void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1625 AudioSystem::OutputDescriptor desc; 1626 void *param2 = NULL; 1627 1628 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1629 param); 1630 1631 switch (event) { 1632 case AudioSystem::OUTPUT_OPENED: 1633 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1634 desc.channelMask = mChannelMask; 1635 desc.samplingRate = mSampleRate; 1636 desc.format = mFormat; 1637 desc.frameCount = mNormalFrameCount; // FIXME see 1638 // AudioFlinger::frameCount(audio_io_handle_t) 1639 desc.latency = latency(); 1640 param2 = &desc; 1641 break; 1642 1643 case AudioSystem::STREAM_CONFIG_CHANGED: 1644 param2 = ¶m; 1645 case AudioSystem::OUTPUT_CLOSED: 1646 default: 1647 break; 1648 } 1649 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1650} 1651 1652void AudioFlinger::PlaybackThread::writeCallback() 1653{ 1654 ALOG_ASSERT(mCallbackThread != 0); 1655 mCallbackThread->resetWriteBlocked(); 1656} 1657 1658void AudioFlinger::PlaybackThread::drainCallback() 1659{ 1660 ALOG_ASSERT(mCallbackThread != 0); 1661 mCallbackThread->resetDraining(); 1662} 1663 1664void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1665{ 1666 Mutex::Autolock _l(mLock); 1667 // reject out of sequence requests 1668 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1669 mWriteAckSequence &= ~1; 1670 mWaitWorkCV.signal(); 1671 } 1672} 1673 1674void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1675{ 1676 Mutex::Autolock _l(mLock); 1677 // reject out of sequence requests 1678 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1679 mDrainSequence &= ~1; 1680 mWaitWorkCV.signal(); 1681 } 1682} 1683 1684// static 1685int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1686 void *param __unused, 1687 void *cookie) 1688{ 1689 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1690 ALOGV("asyncCallback() event %d", event); 1691 switch (event) { 1692 case STREAM_CBK_EVENT_WRITE_READY: 1693 me->writeCallback(); 1694 break; 1695 case STREAM_CBK_EVENT_DRAIN_READY: 1696 me->drainCallback(); 1697 break; 1698 default: 1699 ALOGW("asyncCallback() unknown event %d", event); 1700 break; 1701 } 1702 return 0; 1703} 1704 1705void AudioFlinger::PlaybackThread::readOutputParameters_l() 1706{ 1707 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1708 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1709 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1710 if (!audio_is_output_channel(mChannelMask)) { 1711 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1712 } 1713 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1714 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1715 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1716 } 1717 mChannelCount = popcount(mChannelMask); 1718 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1719 if (!audio_is_valid_format(mFormat)) { 1720 LOG_FATAL("HAL format %#x not valid for output", mFormat); 1721 } 1722 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1723 LOG_FATAL("HAL format %#x not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1724 mFormat); 1725 } 1726 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1727 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1728 mFrameCount = mBufferSize / mFrameSize; 1729 if (mFrameCount & 15) { 1730 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1731 mFrameCount); 1732 } 1733 1734 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1735 (mOutput->stream->set_callback != NULL)) { 1736 if (mOutput->stream->set_callback(mOutput->stream, 1737 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1738 mUseAsyncWrite = true; 1739 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1740 } 1741 } 1742 1743 // Calculate size of normal sink buffer relative to the HAL output buffer size 1744 double multiplier = 1.0; 1745 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1746 kUseFastMixer == FastMixer_Dynamic)) { 1747 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1748 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1749 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1750 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1751 maxNormalFrameCount = maxNormalFrameCount & ~15; 1752 if (maxNormalFrameCount < minNormalFrameCount) { 1753 maxNormalFrameCount = minNormalFrameCount; 1754 } 1755 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1756 if (multiplier <= 1.0) { 1757 multiplier = 1.0; 1758 } else if (multiplier <= 2.0) { 1759 if (2 * mFrameCount <= maxNormalFrameCount) { 1760 multiplier = 2.0; 1761 } else { 1762 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1763 } 1764 } else { 1765 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1766 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1767 // track, but we sometimes have to do this to satisfy the maximum frame count 1768 // constraint) 1769 // FIXME this rounding up should not be done if no HAL SRC 1770 uint32_t truncMult = (uint32_t) multiplier; 1771 if ((truncMult & 1)) { 1772 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1773 ++truncMult; 1774 } 1775 } 1776 multiplier = (double) truncMult; 1777 } 1778 } 1779 mNormalFrameCount = multiplier * mFrameCount; 1780 // round up to nearest 16 frames to satisfy AudioMixer 1781 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1782 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1783 mNormalFrameCount); 1784 1785 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1786 // Originally this was int16_t[] array, need to remove legacy implications. 1787 free(mSinkBuffer); 1788 mSinkBuffer = NULL; 1789 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1790 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1791 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1792 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1793 1794 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1795 // drives the output. 1796 free(mMixerBuffer); 1797 mMixerBuffer = NULL; 1798 if (mMixerBufferEnabled) { 1799 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1800 mMixerBufferSize = mNormalFrameCount * mChannelCount 1801 * audio_bytes_per_sample(mMixerBufferFormat); 1802 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1803 } 1804 free(mEffectBuffer); 1805 mEffectBuffer = NULL; 1806 if (mEffectBufferEnabled) { 1807 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1808 mEffectBufferSize = mNormalFrameCount * mChannelCount 1809 * audio_bytes_per_sample(mEffectBufferFormat); 1810 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1811 } 1812 1813 // force reconfiguration of effect chains and engines to take new buffer size and audio 1814 // parameters into account 1815 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1816 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1817 // matter. 1818 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1819 Vector< sp<EffectChain> > effectChains = mEffectChains; 1820 for (size_t i = 0; i < effectChains.size(); i ++) { 1821 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1822 } 1823} 1824 1825 1826status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1827{ 1828 if (halFrames == NULL || dspFrames == NULL) { 1829 return BAD_VALUE; 1830 } 1831 Mutex::Autolock _l(mLock); 1832 if (initCheck() != NO_ERROR) { 1833 return INVALID_OPERATION; 1834 } 1835 size_t framesWritten = mBytesWritten / mFrameSize; 1836 *halFrames = framesWritten; 1837 1838 if (isSuspended()) { 1839 // return an estimation of rendered frames when the output is suspended 1840 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1841 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1842 return NO_ERROR; 1843 } else { 1844 status_t status; 1845 uint32_t frames; 1846 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1847 *dspFrames = (size_t)frames; 1848 return status; 1849 } 1850} 1851 1852uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1853{ 1854 Mutex::Autolock _l(mLock); 1855 uint32_t result = 0; 1856 if (getEffectChain_l(sessionId) != 0) { 1857 result = EFFECT_SESSION; 1858 } 1859 1860 for (size_t i = 0; i < mTracks.size(); ++i) { 1861 sp<Track> track = mTracks[i]; 1862 if (sessionId == track->sessionId() && !track->isInvalid()) { 1863 result |= TRACK_SESSION; 1864 break; 1865 } 1866 } 1867 1868 return result; 1869} 1870 1871uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1872{ 1873 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1874 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1875 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1876 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1877 } 1878 for (size_t i = 0; i < mTracks.size(); i++) { 1879 sp<Track> track = mTracks[i]; 1880 if (sessionId == track->sessionId() && !track->isInvalid()) { 1881 return AudioSystem::getStrategyForStream(track->streamType()); 1882 } 1883 } 1884 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1885} 1886 1887 1888AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1889{ 1890 Mutex::Autolock _l(mLock); 1891 return mOutput; 1892} 1893 1894AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1895{ 1896 Mutex::Autolock _l(mLock); 1897 AudioStreamOut *output = mOutput; 1898 mOutput = NULL; 1899 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1900 // must push a NULL and wait for ack 1901 mOutputSink.clear(); 1902 mPipeSink.clear(); 1903 mNormalSink.clear(); 1904 return output; 1905} 1906 1907// this method must always be called either with ThreadBase mLock held or inside the thread loop 1908audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1909{ 1910 if (mOutput == NULL) { 1911 return NULL; 1912 } 1913 return &mOutput->stream->common; 1914} 1915 1916uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1917{ 1918 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1919} 1920 1921status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1922{ 1923 if (!isValidSyncEvent(event)) { 1924 return BAD_VALUE; 1925 } 1926 1927 Mutex::Autolock _l(mLock); 1928 1929 for (size_t i = 0; i < mTracks.size(); ++i) { 1930 sp<Track> track = mTracks[i]; 1931 if (event->triggerSession() == track->sessionId()) { 1932 (void) track->setSyncEvent(event); 1933 return NO_ERROR; 1934 } 1935 } 1936 1937 return NAME_NOT_FOUND; 1938} 1939 1940bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1941{ 1942 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1943} 1944 1945void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1946 const Vector< sp<Track> >& tracksToRemove) 1947{ 1948 size_t count = tracksToRemove.size(); 1949 if (count > 0) { 1950 for (size_t i = 0 ; i < count ; i++) { 1951 const sp<Track>& track = tracksToRemove.itemAt(i); 1952 if (!track->isOutputTrack()) { 1953 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1954#ifdef ADD_BATTERY_DATA 1955 // to track the speaker usage 1956 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1957#endif 1958 if (track->isTerminated()) { 1959 AudioSystem::releaseOutput(mId); 1960 } 1961 } 1962 } 1963 } 1964} 1965 1966void AudioFlinger::PlaybackThread::checkSilentMode_l() 1967{ 1968 if (!mMasterMute) { 1969 char value[PROPERTY_VALUE_MAX]; 1970 if (property_get("ro.audio.silent", value, "0") > 0) { 1971 char *endptr; 1972 unsigned long ul = strtoul(value, &endptr, 0); 1973 if (*endptr == '\0' && ul != 0) { 1974 ALOGD("Silence is golden"); 1975 // The setprop command will not allow a property to be changed after 1976 // the first time it is set, so we don't have to worry about un-muting. 1977 setMasterMute_l(true); 1978 } 1979 } 1980 } 1981} 1982 1983// shared by MIXER and DIRECT, overridden by DUPLICATING 1984ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1985{ 1986 // FIXME rewrite to reduce number of system calls 1987 mLastWriteTime = systemTime(); 1988 mInWrite = true; 1989 ssize_t bytesWritten; 1990 const size_t offset = mCurrentWriteLength - mBytesRemaining; 1991 1992 // If an NBAIO sink is present, use it to write the normal mixer's submix 1993 if (mNormalSink != 0) { 1994 const size_t count = mBytesRemaining / mFrameSize; 1995 1996 ATRACE_BEGIN("write"); 1997 // update the setpoint when AudioFlinger::mScreenState changes 1998 uint32_t screenState = AudioFlinger::mScreenState; 1999 if (screenState != mScreenState) { 2000 mScreenState = screenState; 2001 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2002 if (pipe != NULL) { 2003 pipe->setAvgFrames((mScreenState & 1) ? 2004 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2005 } 2006 } 2007 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2008 ATRACE_END(); 2009 if (framesWritten > 0) { 2010 bytesWritten = framesWritten * mFrameSize; 2011 } else { 2012 bytesWritten = framesWritten; 2013 } 2014 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2015 if (status == NO_ERROR) { 2016 size_t totalFramesWritten = mNormalSink->framesWritten(); 2017 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2018 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2019 mLatchDValid = true; 2020 } 2021 } 2022 // otherwise use the HAL / AudioStreamOut directly 2023 } else { 2024 // Direct output and offload threads 2025 2026 if (mUseAsyncWrite) { 2027 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2028 mWriteAckSequence += 2; 2029 mWriteAckSequence |= 1; 2030 ALOG_ASSERT(mCallbackThread != 0); 2031 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2032 } 2033 // FIXME We should have an implementation of timestamps for direct output threads. 2034 // They are used e.g for multichannel PCM playback over HDMI. 2035 bytesWritten = mOutput->stream->write(mOutput->stream, 2036 (char *)mSinkBuffer + offset, mBytesRemaining); 2037 if (mUseAsyncWrite && 2038 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2039 // do not wait for async callback in case of error of full write 2040 mWriteAckSequence &= ~1; 2041 ALOG_ASSERT(mCallbackThread != 0); 2042 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2043 } 2044 } 2045 2046 mNumWrites++; 2047 mInWrite = false; 2048 mStandby = false; 2049 return bytesWritten; 2050} 2051 2052void AudioFlinger::PlaybackThread::threadLoop_drain() 2053{ 2054 if (mOutput->stream->drain) { 2055 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2056 if (mUseAsyncWrite) { 2057 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2058 mDrainSequence |= 1; 2059 ALOG_ASSERT(mCallbackThread != 0); 2060 mCallbackThread->setDraining(mDrainSequence); 2061 } 2062 mOutput->stream->drain(mOutput->stream, 2063 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2064 : AUDIO_DRAIN_ALL); 2065 } 2066} 2067 2068void AudioFlinger::PlaybackThread::threadLoop_exit() 2069{ 2070 // Default implementation has nothing to do 2071} 2072 2073/* 2074The derived values that are cached: 2075 - mSinkBufferSize from frame count * frame size 2076 - activeSleepTime from activeSleepTimeUs() 2077 - idleSleepTime from idleSleepTimeUs() 2078 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2079 - maxPeriod from frame count and sample rate (MIXER only) 2080 2081The parameters that affect these derived values are: 2082 - frame count 2083 - frame size 2084 - sample rate 2085 - device type: A2DP or not 2086 - device latency 2087 - format: PCM or not 2088 - active sleep time 2089 - idle sleep time 2090*/ 2091 2092void AudioFlinger::PlaybackThread::cacheParameters_l() 2093{ 2094 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2095 activeSleepTime = activeSleepTimeUs(); 2096 idleSleepTime = idleSleepTimeUs(); 2097} 2098 2099void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2100{ 2101 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2102 this, streamType, mTracks.size()); 2103 Mutex::Autolock _l(mLock); 2104 2105 size_t size = mTracks.size(); 2106 for (size_t i = 0; i < size; i++) { 2107 sp<Track> t = mTracks[i]; 2108 if (t->streamType() == streamType) { 2109 t->invalidate(); 2110 } 2111 } 2112} 2113 2114status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2115{ 2116 int session = chain->sessionId(); 2117 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2118 ? mEffectBuffer : mSinkBuffer); 2119 bool ownsBuffer = false; 2120 2121 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2122 if (session > 0) { 2123 // Only one effect chain can be present in direct output thread and it uses 2124 // the sink buffer as input 2125 if (mType != DIRECT) { 2126 size_t numSamples = mNormalFrameCount * mChannelCount; 2127 buffer = new int16_t[numSamples]; 2128 memset(buffer, 0, numSamples * sizeof(int16_t)); 2129 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2130 ownsBuffer = true; 2131 } 2132 2133 // Attach all tracks with same session ID to this chain. 2134 for (size_t i = 0; i < mTracks.size(); ++i) { 2135 sp<Track> track = mTracks[i]; 2136 if (session == track->sessionId()) { 2137 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2138 buffer); 2139 track->setMainBuffer(buffer); 2140 chain->incTrackCnt(); 2141 } 2142 } 2143 2144 // indicate all active tracks in the chain 2145 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2146 sp<Track> track = mActiveTracks[i].promote(); 2147 if (track == 0) { 2148 continue; 2149 } 2150 if (session == track->sessionId()) { 2151 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2152 chain->incActiveTrackCnt(); 2153 } 2154 } 2155 } 2156 2157 chain->setInBuffer(buffer, ownsBuffer); 2158 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2159 ? mEffectBuffer : mSinkBuffer)); 2160 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2161 // chains list in order to be processed last as it contains output stage effects 2162 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2163 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2164 // after track specific effects and before output stage 2165 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2166 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2167 // Effect chain for other sessions are inserted at beginning of effect 2168 // chains list to be processed before output mix effects. Relative order between other 2169 // sessions is not important 2170 size_t size = mEffectChains.size(); 2171 size_t i = 0; 2172 for (i = 0; i < size; i++) { 2173 if (mEffectChains[i]->sessionId() < session) { 2174 break; 2175 } 2176 } 2177 mEffectChains.insertAt(chain, i); 2178 checkSuspendOnAddEffectChain_l(chain); 2179 2180 return NO_ERROR; 2181} 2182 2183size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2184{ 2185 int session = chain->sessionId(); 2186 2187 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2188 2189 for (size_t i = 0; i < mEffectChains.size(); i++) { 2190 if (chain == mEffectChains[i]) { 2191 mEffectChains.removeAt(i); 2192 // detach all active tracks from the chain 2193 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2194 sp<Track> track = mActiveTracks[i].promote(); 2195 if (track == 0) { 2196 continue; 2197 } 2198 if (session == track->sessionId()) { 2199 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2200 chain.get(), session); 2201 chain->decActiveTrackCnt(); 2202 } 2203 } 2204 2205 // detach all tracks with same session ID from this chain 2206 for (size_t i = 0; i < mTracks.size(); ++i) { 2207 sp<Track> track = mTracks[i]; 2208 if (session == track->sessionId()) { 2209 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2210 chain->decTrackCnt(); 2211 } 2212 } 2213 break; 2214 } 2215 } 2216 return mEffectChains.size(); 2217} 2218 2219status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2220 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2221{ 2222 Mutex::Autolock _l(mLock); 2223 return attachAuxEffect_l(track, EffectId); 2224} 2225 2226status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2227 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2228{ 2229 status_t status = NO_ERROR; 2230 2231 if (EffectId == 0) { 2232 track->setAuxBuffer(0, NULL); 2233 } else { 2234 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2235 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2236 if (effect != 0) { 2237 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2238 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2239 } else { 2240 status = INVALID_OPERATION; 2241 } 2242 } else { 2243 status = BAD_VALUE; 2244 } 2245 } 2246 return status; 2247} 2248 2249void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2250{ 2251 for (size_t i = 0; i < mTracks.size(); ++i) { 2252 sp<Track> track = mTracks[i]; 2253 if (track->auxEffectId() == effectId) { 2254 attachAuxEffect_l(track, 0); 2255 } 2256 } 2257} 2258 2259bool AudioFlinger::PlaybackThread::threadLoop() 2260{ 2261 Vector< sp<Track> > tracksToRemove; 2262 2263 standbyTime = systemTime(); 2264 2265 // MIXER 2266 nsecs_t lastWarning = 0; 2267 2268 // DUPLICATING 2269 // FIXME could this be made local to while loop? 2270 writeFrames = 0; 2271 2272 int lastGeneration = 0; 2273 2274 cacheParameters_l(); 2275 sleepTime = idleSleepTime; 2276 2277 if (mType == MIXER) { 2278 sleepTimeShift = 0; 2279 } 2280 2281 CpuStats cpuStats; 2282 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2283 2284 acquireWakeLock(); 2285 2286 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2287 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2288 // and then that string will be logged at the next convenient opportunity. 2289 const char *logString = NULL; 2290 2291 checkSilentMode_l(); 2292 2293 while (!exitPending()) 2294 { 2295 cpuStats.sample(myName); 2296 2297 Vector< sp<EffectChain> > effectChains; 2298 2299 processConfigEvents(); 2300 2301 { // scope for mLock 2302 2303 Mutex::Autolock _l(mLock); 2304 2305 if (logString != NULL) { 2306 mNBLogWriter->logTimestamp(); 2307 mNBLogWriter->log(logString); 2308 logString = NULL; 2309 } 2310 2311 if (mLatchDValid) { 2312 mLatchQ = mLatchD; 2313 mLatchDValid = false; 2314 mLatchQValid = true; 2315 } 2316 2317 if (checkForNewParameters_l()) { 2318 cacheParameters_l(); 2319 } 2320 2321 saveOutputTracks(); 2322 if (mSignalPending) { 2323 // A signal was raised while we were unlocked 2324 mSignalPending = false; 2325 } else if (waitingAsyncCallback_l()) { 2326 if (exitPending()) { 2327 break; 2328 } 2329 releaseWakeLock_l(); 2330 mWakeLockUids.clear(); 2331 mActiveTracksGeneration++; 2332 ALOGV("wait async completion"); 2333 mWaitWorkCV.wait(mLock); 2334 ALOGV("async completion/wake"); 2335 acquireWakeLock_l(); 2336 standbyTime = systemTime() + standbyDelay; 2337 sleepTime = 0; 2338 2339 continue; 2340 } 2341 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2342 isSuspended()) { 2343 // put audio hardware into standby after short delay 2344 if (shouldStandby_l()) { 2345 2346 threadLoop_standby(); 2347 2348 mStandby = true; 2349 } 2350 2351 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2352 // we're about to wait, flush the binder command buffer 2353 IPCThreadState::self()->flushCommands(); 2354 2355 clearOutputTracks(); 2356 2357 if (exitPending()) { 2358 break; 2359 } 2360 2361 releaseWakeLock_l(); 2362 mWakeLockUids.clear(); 2363 mActiveTracksGeneration++; 2364 // wait until we have something to do... 2365 ALOGV("%s going to sleep", myName.string()); 2366 mWaitWorkCV.wait(mLock); 2367 ALOGV("%s waking up", myName.string()); 2368 acquireWakeLock_l(); 2369 2370 mMixerStatus = MIXER_IDLE; 2371 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2372 mBytesWritten = 0; 2373 mBytesRemaining = 0; 2374 checkSilentMode_l(); 2375 2376 standbyTime = systemTime() + standbyDelay; 2377 sleepTime = idleSleepTime; 2378 if (mType == MIXER) { 2379 sleepTimeShift = 0; 2380 } 2381 2382 continue; 2383 } 2384 } 2385 // mMixerStatusIgnoringFastTracks is also updated internally 2386 mMixerStatus = prepareTracks_l(&tracksToRemove); 2387 2388 // compare with previously applied list 2389 if (lastGeneration != mActiveTracksGeneration) { 2390 // update wakelock 2391 updateWakeLockUids_l(mWakeLockUids); 2392 lastGeneration = mActiveTracksGeneration; 2393 } 2394 2395 // prevent any changes in effect chain list and in each effect chain 2396 // during mixing and effect process as the audio buffers could be deleted 2397 // or modified if an effect is created or deleted 2398 lockEffectChains_l(effectChains); 2399 } // mLock scope ends 2400 2401 if (mBytesRemaining == 0) { 2402 mCurrentWriteLength = 0; 2403 if (mMixerStatus == MIXER_TRACKS_READY) { 2404 // threadLoop_mix() sets mCurrentWriteLength 2405 threadLoop_mix(); 2406 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2407 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2408 // threadLoop_sleepTime sets sleepTime to 0 if data 2409 // must be written to HAL 2410 threadLoop_sleepTime(); 2411 if (sleepTime == 0) { 2412 mCurrentWriteLength = mSinkBufferSize; 2413 } 2414 } 2415 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2416 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2417 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2418 // or mSinkBuffer (if there are no effects). 2419 // 2420 // This is done pre-effects computation; if effects change to 2421 // support higher precision, this needs to move. 2422 // 2423 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2424 // TODO use sleepTime == 0 as an additional condition. 2425 if (mMixerBufferValid) { 2426 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2427 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2428 2429 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2430 mNormalFrameCount * mChannelCount); 2431 } 2432 2433 mBytesRemaining = mCurrentWriteLength; 2434 if (isSuspended()) { 2435 sleepTime = suspendSleepTimeUs(); 2436 // simulate write to HAL when suspended 2437 mBytesWritten += mSinkBufferSize; 2438 mBytesRemaining = 0; 2439 } 2440 2441 // only process effects if we're going to write 2442 if (sleepTime == 0 && mType != OFFLOAD) { 2443 for (size_t i = 0; i < effectChains.size(); i ++) { 2444 effectChains[i]->process_l(); 2445 } 2446 } 2447 } 2448 // Process effect chains for offloaded thread even if no audio 2449 // was read from audio track: process only updates effect state 2450 // and thus does have to be synchronized with audio writes but may have 2451 // to be called while waiting for async write callback 2452 if (mType == OFFLOAD) { 2453 for (size_t i = 0; i < effectChains.size(); i ++) { 2454 effectChains[i]->process_l(); 2455 } 2456 } 2457 2458 // Only if the Effects buffer is enabled and there is data in the 2459 // Effects buffer (buffer valid), we need to 2460 // copy into the sink buffer. 2461 // TODO use sleepTime == 0 as an additional condition. 2462 if (mEffectBufferValid) { 2463 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2464 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2465 mNormalFrameCount * mChannelCount); 2466 } 2467 2468 // enable changes in effect chain 2469 unlockEffectChains(effectChains); 2470 2471 if (!waitingAsyncCallback()) { 2472 // sleepTime == 0 means we must write to audio hardware 2473 if (sleepTime == 0) { 2474 if (mBytesRemaining) { 2475 ssize_t ret = threadLoop_write(); 2476 if (ret < 0) { 2477 mBytesRemaining = 0; 2478 } else { 2479 mBytesWritten += ret; 2480 mBytesRemaining -= ret; 2481 } 2482 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2483 (mMixerStatus == MIXER_DRAIN_ALL)) { 2484 threadLoop_drain(); 2485 } 2486 if (mType == MIXER) { 2487 // write blocked detection 2488 nsecs_t now = systemTime(); 2489 nsecs_t delta = now - mLastWriteTime; 2490 if (!mStandby && delta > maxPeriod) { 2491 mNumDelayedWrites++; 2492 if ((now - lastWarning) > kWarningThrottleNs) { 2493 ATRACE_NAME("underrun"); 2494 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2495 ns2ms(delta), mNumDelayedWrites, this); 2496 lastWarning = now; 2497 } 2498 } 2499 } 2500 2501 } else { 2502 usleep(sleepTime); 2503 } 2504 } 2505 2506 // Finally let go of removed track(s), without the lock held 2507 // since we can't guarantee the destructors won't acquire that 2508 // same lock. This will also mutate and push a new fast mixer state. 2509 threadLoop_removeTracks(tracksToRemove); 2510 tracksToRemove.clear(); 2511 2512 // FIXME I don't understand the need for this here; 2513 // it was in the original code but maybe the 2514 // assignment in saveOutputTracks() makes this unnecessary? 2515 clearOutputTracks(); 2516 2517 // Effect chains will be actually deleted here if they were removed from 2518 // mEffectChains list during mixing or effects processing 2519 effectChains.clear(); 2520 2521 // FIXME Note that the above .clear() is no longer necessary since effectChains 2522 // is now local to this block, but will keep it for now (at least until merge done). 2523 } 2524 2525 threadLoop_exit(); 2526 2527 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2528 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2529 // put output stream into standby mode 2530 if (!mStandby) { 2531 mOutput->stream->common.standby(&mOutput->stream->common); 2532 } 2533 } 2534 2535 releaseWakeLock(); 2536 mWakeLockUids.clear(); 2537 mActiveTracksGeneration++; 2538 2539 ALOGV("Thread %p type %d exiting", this, mType); 2540 return false; 2541} 2542 2543// removeTracks_l() must be called with ThreadBase::mLock held 2544void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2545{ 2546 size_t count = tracksToRemove.size(); 2547 if (count > 0) { 2548 for (size_t i=0 ; i<count ; i++) { 2549 const sp<Track>& track = tracksToRemove.itemAt(i); 2550 mActiveTracks.remove(track); 2551 mWakeLockUids.remove(track->uid()); 2552 mActiveTracksGeneration++; 2553 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2554 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2555 if (chain != 0) { 2556 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2557 track->sessionId()); 2558 chain->decActiveTrackCnt(); 2559 } 2560 if (track->isTerminated()) { 2561 removeTrack_l(track); 2562 } 2563 } 2564 } 2565 2566} 2567 2568status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2569{ 2570 if (mNormalSink != 0) { 2571 return mNormalSink->getTimestamp(timestamp); 2572 } 2573 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2574 uint64_t position64; 2575 int ret = mOutput->stream->get_presentation_position( 2576 mOutput->stream, &position64, ×tamp.mTime); 2577 if (ret == 0) { 2578 timestamp.mPosition = (uint32_t)position64; 2579 return NO_ERROR; 2580 } 2581 } 2582 return INVALID_OPERATION; 2583} 2584// ---------------------------------------------------------------------------- 2585 2586AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2587 audio_io_handle_t id, audio_devices_t device, type_t type) 2588 : PlaybackThread(audioFlinger, output, id, device, type), 2589 // mAudioMixer below 2590 // mFastMixer below 2591 mFastMixerFutex(0) 2592 // mOutputSink below 2593 // mPipeSink below 2594 // mNormalSink below 2595{ 2596 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2597 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2598 "mFrameCount=%d, mNormalFrameCount=%d", 2599 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2600 mNormalFrameCount); 2601 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2602 2603 // FIXME - Current mixer implementation only supports stereo output 2604 if (mChannelCount != FCC_2) { 2605 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2606 } 2607 2608 // create an NBAIO sink for the HAL output stream, and negotiate 2609 mOutputSink = new AudioStreamOutSink(output->stream); 2610 size_t numCounterOffers = 0; 2611 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2612 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2613 ALOG_ASSERT(index == 0); 2614 2615 // initialize fast mixer depending on configuration 2616 bool initFastMixer; 2617 switch (kUseFastMixer) { 2618 case FastMixer_Never: 2619 initFastMixer = false; 2620 break; 2621 case FastMixer_Always: 2622 initFastMixer = true; 2623 break; 2624 case FastMixer_Static: 2625 case FastMixer_Dynamic: 2626 initFastMixer = mFrameCount < mNormalFrameCount; 2627 break; 2628 } 2629 if (initFastMixer) { 2630 2631 // create a MonoPipe to connect our submix to FastMixer 2632 NBAIO_Format format = mOutputSink->format(); 2633 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2634 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2635 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2636 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2637 const NBAIO_Format offers[1] = {format}; 2638 size_t numCounterOffers = 0; 2639 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2640 ALOG_ASSERT(index == 0); 2641 monoPipe->setAvgFrames((mScreenState & 1) ? 2642 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2643 mPipeSink = monoPipe; 2644 2645#ifdef TEE_SINK 2646 if (mTeeSinkOutputEnabled) { 2647 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2648 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2649 numCounterOffers = 0; 2650 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2651 ALOG_ASSERT(index == 0); 2652 mTeeSink = teeSink; 2653 PipeReader *teeSource = new PipeReader(*teeSink); 2654 numCounterOffers = 0; 2655 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2656 ALOG_ASSERT(index == 0); 2657 mTeeSource = teeSource; 2658 } 2659#endif 2660 2661 // create fast mixer and configure it initially with just one fast track for our submix 2662 mFastMixer = new FastMixer(); 2663 FastMixerStateQueue *sq = mFastMixer->sq(); 2664#ifdef STATE_QUEUE_DUMP 2665 sq->setObserverDump(&mStateQueueObserverDump); 2666 sq->setMutatorDump(&mStateQueueMutatorDump); 2667#endif 2668 FastMixerState *state = sq->begin(); 2669 FastTrack *fastTrack = &state->mFastTracks[0]; 2670 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2671 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2672 fastTrack->mVolumeProvider = NULL; 2673 fastTrack->mGeneration++; 2674 state->mFastTracksGen++; 2675 state->mTrackMask = 1; 2676 // fast mixer will use the HAL output sink 2677 state->mOutputSink = mOutputSink.get(); 2678 state->mOutputSinkGen++; 2679 state->mFrameCount = mFrameCount; 2680 state->mCommand = FastMixerState::COLD_IDLE; 2681 // already done in constructor initialization list 2682 //mFastMixerFutex = 0; 2683 state->mColdFutexAddr = &mFastMixerFutex; 2684 state->mColdGen++; 2685 state->mDumpState = &mFastMixerDumpState; 2686#ifdef TEE_SINK 2687 state->mTeeSink = mTeeSink.get(); 2688#endif 2689 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2690 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2691 sq->end(); 2692 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2693 2694 // start the fast mixer 2695 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2696 pid_t tid = mFastMixer->getTid(); 2697 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2698 if (err != 0) { 2699 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2700 kPriorityFastMixer, getpid_cached, tid, err); 2701 } 2702 2703#ifdef AUDIO_WATCHDOG 2704 // create and start the watchdog 2705 mAudioWatchdog = new AudioWatchdog(); 2706 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2707 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2708 tid = mAudioWatchdog->getTid(); 2709 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2710 if (err != 0) { 2711 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2712 kPriorityFastMixer, getpid_cached, tid, err); 2713 } 2714#endif 2715 2716 } else { 2717 mFastMixer = NULL; 2718 } 2719 2720 switch (kUseFastMixer) { 2721 case FastMixer_Never: 2722 case FastMixer_Dynamic: 2723 mNormalSink = mOutputSink; 2724 break; 2725 case FastMixer_Always: 2726 mNormalSink = mPipeSink; 2727 break; 2728 case FastMixer_Static: 2729 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2730 break; 2731 } 2732} 2733 2734AudioFlinger::MixerThread::~MixerThread() 2735{ 2736 if (mFastMixer != NULL) { 2737 FastMixerStateQueue *sq = mFastMixer->sq(); 2738 FastMixerState *state = sq->begin(); 2739 if (state->mCommand == FastMixerState::COLD_IDLE) { 2740 int32_t old = android_atomic_inc(&mFastMixerFutex); 2741 if (old == -1) { 2742 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2743 } 2744 } 2745 state->mCommand = FastMixerState::EXIT; 2746 sq->end(); 2747 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2748 mFastMixer->join(); 2749 // Though the fast mixer thread has exited, it's state queue is still valid. 2750 // We'll use that extract the final state which contains one remaining fast track 2751 // corresponding to our sub-mix. 2752 state = sq->begin(); 2753 ALOG_ASSERT(state->mTrackMask == 1); 2754 FastTrack *fastTrack = &state->mFastTracks[0]; 2755 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2756 delete fastTrack->mBufferProvider; 2757 sq->end(false /*didModify*/); 2758 delete mFastMixer; 2759#ifdef AUDIO_WATCHDOG 2760 if (mAudioWatchdog != 0) { 2761 mAudioWatchdog->requestExit(); 2762 mAudioWatchdog->requestExitAndWait(); 2763 mAudioWatchdog.clear(); 2764 } 2765#endif 2766 } 2767 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2768 delete mAudioMixer; 2769} 2770 2771 2772uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2773{ 2774 if (mFastMixer != NULL) { 2775 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2776 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2777 } 2778 return latency; 2779} 2780 2781 2782void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2783{ 2784 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2785} 2786 2787ssize_t AudioFlinger::MixerThread::threadLoop_write() 2788{ 2789 // FIXME we should only do one push per cycle; confirm this is true 2790 // Start the fast mixer if it's not already running 2791 if (mFastMixer != NULL) { 2792 FastMixerStateQueue *sq = mFastMixer->sq(); 2793 FastMixerState *state = sq->begin(); 2794 if (state->mCommand != FastMixerState::MIX_WRITE && 2795 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2796 if (state->mCommand == FastMixerState::COLD_IDLE) { 2797 int32_t old = android_atomic_inc(&mFastMixerFutex); 2798 if (old == -1) { 2799 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2800 } 2801#ifdef AUDIO_WATCHDOG 2802 if (mAudioWatchdog != 0) { 2803 mAudioWatchdog->resume(); 2804 } 2805#endif 2806 } 2807 state->mCommand = FastMixerState::MIX_WRITE; 2808 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2809 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2810 sq->end(); 2811 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2812 if (kUseFastMixer == FastMixer_Dynamic) { 2813 mNormalSink = mPipeSink; 2814 } 2815 } else { 2816 sq->end(false /*didModify*/); 2817 } 2818 } 2819 return PlaybackThread::threadLoop_write(); 2820} 2821 2822void AudioFlinger::MixerThread::threadLoop_standby() 2823{ 2824 // Idle the fast mixer if it's currently running 2825 if (mFastMixer != NULL) { 2826 FastMixerStateQueue *sq = mFastMixer->sq(); 2827 FastMixerState *state = sq->begin(); 2828 if (!(state->mCommand & FastMixerState::IDLE)) { 2829 state->mCommand = FastMixerState::COLD_IDLE; 2830 state->mColdFutexAddr = &mFastMixerFutex; 2831 state->mColdGen++; 2832 mFastMixerFutex = 0; 2833 sq->end(); 2834 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2835 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2836 if (kUseFastMixer == FastMixer_Dynamic) { 2837 mNormalSink = mOutputSink; 2838 } 2839#ifdef AUDIO_WATCHDOG 2840 if (mAudioWatchdog != 0) { 2841 mAudioWatchdog->pause(); 2842 } 2843#endif 2844 } else { 2845 sq->end(false /*didModify*/); 2846 } 2847 } 2848 PlaybackThread::threadLoop_standby(); 2849} 2850 2851bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2852{ 2853 return false; 2854} 2855 2856bool AudioFlinger::PlaybackThread::shouldStandby_l() 2857{ 2858 return !mStandby; 2859} 2860 2861bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2862{ 2863 Mutex::Autolock _l(mLock); 2864 return waitingAsyncCallback_l(); 2865} 2866 2867// shared by MIXER and DIRECT, overridden by DUPLICATING 2868void AudioFlinger::PlaybackThread::threadLoop_standby() 2869{ 2870 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2871 mOutput->stream->common.standby(&mOutput->stream->common); 2872 if (mUseAsyncWrite != 0) { 2873 // discard any pending drain or write ack by incrementing sequence 2874 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2875 mDrainSequence = (mDrainSequence + 2) & ~1; 2876 ALOG_ASSERT(mCallbackThread != 0); 2877 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2878 mCallbackThread->setDraining(mDrainSequence); 2879 } 2880} 2881 2882void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2883{ 2884 ALOGV("signal playback thread"); 2885 broadcast_l(); 2886} 2887 2888void AudioFlinger::MixerThread::threadLoop_mix() 2889{ 2890 // obtain the presentation timestamp of the next output buffer 2891 int64_t pts; 2892 status_t status = INVALID_OPERATION; 2893 2894 if (mNormalSink != 0) { 2895 status = mNormalSink->getNextWriteTimestamp(&pts); 2896 } else { 2897 status = mOutputSink->getNextWriteTimestamp(&pts); 2898 } 2899 2900 if (status != NO_ERROR) { 2901 pts = AudioBufferProvider::kInvalidPTS; 2902 } 2903 2904 // mix buffers... 2905 mAudioMixer->process(pts); 2906 mCurrentWriteLength = mSinkBufferSize; 2907 // increase sleep time progressively when application underrun condition clears. 2908 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2909 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2910 // such that we would underrun the audio HAL. 2911 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2912 sleepTimeShift--; 2913 } 2914 sleepTime = 0; 2915 standbyTime = systemTime() + standbyDelay; 2916 //TODO: delay standby when effects have a tail 2917} 2918 2919void AudioFlinger::MixerThread::threadLoop_sleepTime() 2920{ 2921 // If no tracks are ready, sleep once for the duration of an output 2922 // buffer size, then write 0s to the output 2923 if (sleepTime == 0) { 2924 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2925 sleepTime = activeSleepTime >> sleepTimeShift; 2926 if (sleepTime < kMinThreadSleepTimeUs) { 2927 sleepTime = kMinThreadSleepTimeUs; 2928 } 2929 // reduce sleep time in case of consecutive application underruns to avoid 2930 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2931 // duration we would end up writing less data than needed by the audio HAL if 2932 // the condition persists. 2933 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2934 sleepTimeShift++; 2935 } 2936 } else { 2937 sleepTime = idleSleepTime; 2938 } 2939 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2940 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2941 // before effects processing or output. 2942 if (mMixerBufferValid) { 2943 memset(mMixerBuffer, 0, mMixerBufferSize); 2944 } else { 2945 memset(mSinkBuffer, 0, mSinkBufferSize); 2946 } 2947 sleepTime = 0; 2948 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2949 "anticipated start"); 2950 } 2951 // TODO add standby time extension fct of effect tail 2952} 2953 2954// prepareTracks_l() must be called with ThreadBase::mLock held 2955AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2956 Vector< sp<Track> > *tracksToRemove) 2957{ 2958 2959 mixer_state mixerStatus = MIXER_IDLE; 2960 // find out which tracks need to be processed 2961 size_t count = mActiveTracks.size(); 2962 size_t mixedTracks = 0; 2963 size_t tracksWithEffect = 0; 2964 // counts only _active_ fast tracks 2965 size_t fastTracks = 0; 2966 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2967 2968 float masterVolume = mMasterVolume; 2969 bool masterMute = mMasterMute; 2970 2971 if (masterMute) { 2972 masterVolume = 0; 2973 } 2974 // Delegate master volume control to effect in output mix effect chain if needed 2975 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2976 if (chain != 0) { 2977 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2978 chain->setVolume_l(&v, &v); 2979 masterVolume = (float)((v + (1 << 23)) >> 24); 2980 chain.clear(); 2981 } 2982 2983 // prepare a new state to push 2984 FastMixerStateQueue *sq = NULL; 2985 FastMixerState *state = NULL; 2986 bool didModify = false; 2987 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2988 if (mFastMixer != NULL) { 2989 sq = mFastMixer->sq(); 2990 state = sq->begin(); 2991 } 2992 2993 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 2994 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 2995 2996 for (size_t i=0 ; i<count ; i++) { 2997 const sp<Track> t = mActiveTracks[i].promote(); 2998 if (t == 0) { 2999 continue; 3000 } 3001 3002 // this const just means the local variable doesn't change 3003 Track* const track = t.get(); 3004 3005 // process fast tracks 3006 if (track->isFastTrack()) { 3007 3008 // It's theoretically possible (though unlikely) for a fast track to be created 3009 // and then removed within the same normal mix cycle. This is not a problem, as 3010 // the track never becomes active so it's fast mixer slot is never touched. 3011 // The converse, of removing an (active) track and then creating a new track 3012 // at the identical fast mixer slot within the same normal mix cycle, 3013 // is impossible because the slot isn't marked available until the end of each cycle. 3014 int j = track->mFastIndex; 3015 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3016 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3017 FastTrack *fastTrack = &state->mFastTracks[j]; 3018 3019 // Determine whether the track is currently in underrun condition, 3020 // and whether it had a recent underrun. 3021 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3022 FastTrackUnderruns underruns = ftDump->mUnderruns; 3023 uint32_t recentFull = (underruns.mBitFields.mFull - 3024 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3025 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3026 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3027 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3028 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3029 uint32_t recentUnderruns = recentPartial + recentEmpty; 3030 track->mObservedUnderruns = underruns; 3031 // don't count underruns that occur while stopping or pausing 3032 // or stopped which can occur when flush() is called while active 3033 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3034 recentUnderruns > 0) { 3035 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3036 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3037 } 3038 3039 // This is similar to the state machine for normal tracks, 3040 // with a few modifications for fast tracks. 3041 bool isActive = true; 3042 switch (track->mState) { 3043 case TrackBase::STOPPING_1: 3044 // track stays active in STOPPING_1 state until first underrun 3045 if (recentUnderruns > 0 || track->isTerminated()) { 3046 track->mState = TrackBase::STOPPING_2; 3047 } 3048 break; 3049 case TrackBase::PAUSING: 3050 // ramp down is not yet implemented 3051 track->setPaused(); 3052 break; 3053 case TrackBase::RESUMING: 3054 // ramp up is not yet implemented 3055 track->mState = TrackBase::ACTIVE; 3056 break; 3057 case TrackBase::ACTIVE: 3058 if (recentFull > 0 || recentPartial > 0) { 3059 // track has provided at least some frames recently: reset retry count 3060 track->mRetryCount = kMaxTrackRetries; 3061 } 3062 if (recentUnderruns == 0) { 3063 // no recent underruns: stay active 3064 break; 3065 } 3066 // there has recently been an underrun of some kind 3067 if (track->sharedBuffer() == 0) { 3068 // were any of the recent underruns "empty" (no frames available)? 3069 if (recentEmpty == 0) { 3070 // no, then ignore the partial underruns as they are allowed indefinitely 3071 break; 3072 } 3073 // there has recently been an "empty" underrun: decrement the retry counter 3074 if (--(track->mRetryCount) > 0) { 3075 break; 3076 } 3077 // indicate to client process that the track was disabled because of underrun; 3078 // it will then automatically call start() when data is available 3079 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3080 // remove from active list, but state remains ACTIVE [confusing but true] 3081 isActive = false; 3082 break; 3083 } 3084 // fall through 3085 case TrackBase::STOPPING_2: 3086 case TrackBase::PAUSED: 3087 case TrackBase::STOPPED: 3088 case TrackBase::FLUSHED: // flush() while active 3089 // Check for presentation complete if track is inactive 3090 // We have consumed all the buffers of this track. 3091 // This would be incomplete if we auto-paused on underrun 3092 { 3093 size_t audioHALFrames = 3094 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3095 size_t framesWritten = mBytesWritten / mFrameSize; 3096 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3097 // track stays in active list until presentation is complete 3098 break; 3099 } 3100 } 3101 if (track->isStopping_2()) { 3102 track->mState = TrackBase::STOPPED; 3103 } 3104 if (track->isStopped()) { 3105 // Can't reset directly, as fast mixer is still polling this track 3106 // track->reset(); 3107 // So instead mark this track as needing to be reset after push with ack 3108 resetMask |= 1 << i; 3109 } 3110 isActive = false; 3111 break; 3112 case TrackBase::IDLE: 3113 default: 3114 LOG_FATAL("unexpected track state %d", track->mState); 3115 } 3116 3117 if (isActive) { 3118 // was it previously inactive? 3119 if (!(state->mTrackMask & (1 << j))) { 3120 ExtendedAudioBufferProvider *eabp = track; 3121 VolumeProvider *vp = track; 3122 fastTrack->mBufferProvider = eabp; 3123 fastTrack->mVolumeProvider = vp; 3124 fastTrack->mChannelMask = track->mChannelMask; 3125 fastTrack->mGeneration++; 3126 state->mTrackMask |= 1 << j; 3127 didModify = true; 3128 // no acknowledgement required for newly active tracks 3129 } 3130 // cache the combined master volume and stream type volume for fast mixer; this 3131 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3132 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3133 ++fastTracks; 3134 } else { 3135 // was it previously active? 3136 if (state->mTrackMask & (1 << j)) { 3137 fastTrack->mBufferProvider = NULL; 3138 fastTrack->mGeneration++; 3139 state->mTrackMask &= ~(1 << j); 3140 didModify = true; 3141 // If any fast tracks were removed, we must wait for acknowledgement 3142 // because we're about to decrement the last sp<> on those tracks. 3143 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3144 } else { 3145 LOG_FATAL("fast track %d should have been active", j); 3146 } 3147 tracksToRemove->add(track); 3148 // Avoids a misleading display in dumpsys 3149 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3150 } 3151 continue; 3152 } 3153 3154 { // local variable scope to avoid goto warning 3155 3156 audio_track_cblk_t* cblk = track->cblk(); 3157 3158 // The first time a track is added we wait 3159 // for all its buffers to be filled before processing it 3160 int name = track->name(); 3161 // make sure that we have enough frames to mix one full buffer. 3162 // enforce this condition only once to enable draining the buffer in case the client 3163 // app does not call stop() and relies on underrun to stop: 3164 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3165 // during last round 3166 size_t desiredFrames; 3167 uint32_t sr = track->sampleRate(); 3168 if (sr == mSampleRate) { 3169 desiredFrames = mNormalFrameCount; 3170 } else { 3171 // +1 for rounding and +1 for additional sample needed for interpolation 3172 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3173 // add frames already consumed but not yet released by the resampler 3174 // because mAudioTrackServerProxy->framesReady() will include these frames 3175 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3176#if 0 3177 // the minimum track buffer size is normally twice the number of frames necessary 3178 // to fill one buffer and the resampler should not leave more than one buffer worth 3179 // of unreleased frames after each pass, but just in case... 3180 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3181#endif 3182 } 3183 uint32_t minFrames = 1; 3184 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3185 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3186 minFrames = desiredFrames; 3187 } 3188 3189 size_t framesReady = track->framesReady(); 3190 if ((framesReady >= minFrames) && track->isReady() && 3191 !track->isPaused() && !track->isTerminated()) 3192 { 3193 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3194 3195 mixedTracks++; 3196 3197 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3198 // there is an effect chain connected to the track 3199 chain.clear(); 3200 if (track->mainBuffer() != mSinkBuffer && 3201 track->mainBuffer() != mMixerBuffer) { 3202 if (mEffectBufferEnabled) { 3203 mEffectBufferValid = true; // Later can set directly. 3204 } 3205 chain = getEffectChain_l(track->sessionId()); 3206 // Delegate volume control to effect in track effect chain if needed 3207 if (chain != 0) { 3208 tracksWithEffect++; 3209 } else { 3210 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3211 "session %d", 3212 name, track->sessionId()); 3213 } 3214 } 3215 3216 3217 int param = AudioMixer::VOLUME; 3218 if (track->mFillingUpStatus == Track::FS_FILLED) { 3219 // no ramp for the first volume setting 3220 track->mFillingUpStatus = Track::FS_ACTIVE; 3221 if (track->mState == TrackBase::RESUMING) { 3222 track->mState = TrackBase::ACTIVE; 3223 param = AudioMixer::RAMP_VOLUME; 3224 } 3225 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3226 // FIXME should not make a decision based on mServer 3227 } else if (cblk->mServer != 0) { 3228 // If the track is stopped before the first frame was mixed, 3229 // do not apply ramp 3230 param = AudioMixer::RAMP_VOLUME; 3231 } 3232 3233 // compute volume for this track 3234 uint32_t vl, vr, va; 3235 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3236 vl = vr = va = 0; 3237 if (track->isPausing()) { 3238 track->setPaused(); 3239 } 3240 } else { 3241 3242 // read original volumes with volume control 3243 float typeVolume = mStreamTypes[track->streamType()].volume; 3244 float v = masterVolume * typeVolume; 3245 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3246 uint32_t vlr = proxy->getVolumeLR(); 3247 vl = vlr & 0xFFFF; 3248 vr = vlr >> 16; 3249 // track volumes come from shared memory, so can't be trusted and must be clamped 3250 if (vl > MAX_GAIN_INT) { 3251 ALOGV("Track left volume out of range: %04X", vl); 3252 vl = MAX_GAIN_INT; 3253 } 3254 if (vr > MAX_GAIN_INT) { 3255 ALOGV("Track right volume out of range: %04X", vr); 3256 vr = MAX_GAIN_INT; 3257 } 3258 // now apply the master volume and stream type volume 3259 vl = (uint32_t)(v * vl) << 12; 3260 vr = (uint32_t)(v * vr) << 12; 3261 // assuming master volume and stream type volume each go up to 1.0, 3262 // vl and vr are now in 8.24 format 3263 3264 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3265 // send level comes from shared memory and so may be corrupt 3266 if (sendLevel > MAX_GAIN_INT) { 3267 ALOGV("Track send level out of range: %04X", sendLevel); 3268 sendLevel = MAX_GAIN_INT; 3269 } 3270 va = (uint32_t)(v * sendLevel); 3271 } 3272 3273 // Delegate volume control to effect in track effect chain if needed 3274 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3275 // Do not ramp volume if volume is controlled by effect 3276 param = AudioMixer::VOLUME; 3277 track->mHasVolumeController = true; 3278 } else { 3279 // force no volume ramp when volume controller was just disabled or removed 3280 // from effect chain to avoid volume spike 3281 if (track->mHasVolumeController) { 3282 param = AudioMixer::VOLUME; 3283 } 3284 track->mHasVolumeController = false; 3285 } 3286 3287 // Convert volumes from 8.24 to 4.12 format 3288 // This additional clamping is needed in case chain->setVolume_l() overshot 3289 vl = (vl + (1 << 11)) >> 12; 3290 if (vl > MAX_GAIN_INT) { 3291 vl = MAX_GAIN_INT; 3292 } 3293 vr = (vr + (1 << 11)) >> 12; 3294 if (vr > MAX_GAIN_INT) { 3295 vr = MAX_GAIN_INT; 3296 } 3297 3298 if (va > MAX_GAIN_INT) { 3299 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3300 } 3301 3302 // XXX: these things DON'T need to be done each time 3303 mAudioMixer->setBufferProvider(name, track); 3304 mAudioMixer->enable(name); 3305 3306 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3307 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3308 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3309 mAudioMixer->setParameter( 3310 name, 3311 AudioMixer::TRACK, 3312 AudioMixer::FORMAT, (void *)track->format()); 3313 mAudioMixer->setParameter( 3314 name, 3315 AudioMixer::TRACK, 3316 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3317 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3318 uint32_t maxSampleRate = mSampleRate * 2; 3319 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3320 if (reqSampleRate == 0) { 3321 reqSampleRate = mSampleRate; 3322 } else if (reqSampleRate > maxSampleRate) { 3323 reqSampleRate = maxSampleRate; 3324 } 3325 mAudioMixer->setParameter( 3326 name, 3327 AudioMixer::RESAMPLE, 3328 AudioMixer::SAMPLE_RATE, 3329 (void *)(uintptr_t)reqSampleRate); 3330 /* 3331 * Select the appropriate output buffer for the track. 3332 * 3333 * Tracks with effects go into their own effects chain buffer 3334 * and from there into either mEffectBuffer or mSinkBuffer. 3335 * 3336 * Other tracks can use mMixerBuffer for higher precision 3337 * channel accumulation. If this buffer is enabled 3338 * (mMixerBufferEnabled true), then selected tracks will accumulate 3339 * into it. 3340 * 3341 */ 3342 if (mMixerBufferEnabled 3343 && (track->mainBuffer() == mSinkBuffer 3344 || track->mainBuffer() == mMixerBuffer)) { 3345 mAudioMixer->setParameter( 3346 name, 3347 AudioMixer::TRACK, 3348 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3349 mAudioMixer->setParameter( 3350 name, 3351 AudioMixer::TRACK, 3352 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3353 // TODO: override track->mainBuffer()? 3354 mMixerBufferValid = true; 3355 } else { 3356 mAudioMixer->setParameter( 3357 name, 3358 AudioMixer::TRACK, 3359 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3360 mAudioMixer->setParameter( 3361 name, 3362 AudioMixer::TRACK, 3363 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3364 } 3365 mAudioMixer->setParameter( 3366 name, 3367 AudioMixer::TRACK, 3368 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3369 3370 // reset retry count 3371 track->mRetryCount = kMaxTrackRetries; 3372 3373 // If one track is ready, set the mixer ready if: 3374 // - the mixer was not ready during previous round OR 3375 // - no other track is not ready 3376 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3377 mixerStatus != MIXER_TRACKS_ENABLED) { 3378 mixerStatus = MIXER_TRACKS_READY; 3379 } 3380 } else { 3381 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3382 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3383 } 3384 // clear effect chain input buffer if an active track underruns to avoid sending 3385 // previous audio buffer again to effects 3386 chain = getEffectChain_l(track->sessionId()); 3387 if (chain != 0) { 3388 chain->clearInputBuffer(); 3389 } 3390 3391 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3392 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3393 track->isStopped() || track->isPaused()) { 3394 // We have consumed all the buffers of this track. 3395 // Remove it from the list of active tracks. 3396 // TODO: use actual buffer filling status instead of latency when available from 3397 // audio HAL 3398 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3399 size_t framesWritten = mBytesWritten / mFrameSize; 3400 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3401 if (track->isStopped()) { 3402 track->reset(); 3403 } 3404 tracksToRemove->add(track); 3405 } 3406 } else { 3407 // No buffers for this track. Give it a few chances to 3408 // fill a buffer, then remove it from active list. 3409 if (--(track->mRetryCount) <= 0) { 3410 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3411 tracksToRemove->add(track); 3412 // indicate to client process that the track was disabled because of underrun; 3413 // it will then automatically call start() when data is available 3414 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3415 // If one track is not ready, mark the mixer also not ready if: 3416 // - the mixer was ready during previous round OR 3417 // - no other track is ready 3418 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3419 mixerStatus != MIXER_TRACKS_READY) { 3420 mixerStatus = MIXER_TRACKS_ENABLED; 3421 } 3422 } 3423 mAudioMixer->disable(name); 3424 } 3425 3426 } // local variable scope to avoid goto warning 3427track_is_ready: ; 3428 3429 } 3430 3431 // Push the new FastMixer state if necessary 3432 bool pauseAudioWatchdog = false; 3433 if (didModify) { 3434 state->mFastTracksGen++; 3435 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3436 if (kUseFastMixer == FastMixer_Dynamic && 3437 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3438 state->mCommand = FastMixerState::COLD_IDLE; 3439 state->mColdFutexAddr = &mFastMixerFutex; 3440 state->mColdGen++; 3441 mFastMixerFutex = 0; 3442 if (kUseFastMixer == FastMixer_Dynamic) { 3443 mNormalSink = mOutputSink; 3444 } 3445 // If we go into cold idle, need to wait for acknowledgement 3446 // so that fast mixer stops doing I/O. 3447 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3448 pauseAudioWatchdog = true; 3449 } 3450 } 3451 if (sq != NULL) { 3452 sq->end(didModify); 3453 sq->push(block); 3454 } 3455#ifdef AUDIO_WATCHDOG 3456 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3457 mAudioWatchdog->pause(); 3458 } 3459#endif 3460 3461 // Now perform the deferred reset on fast tracks that have stopped 3462 while (resetMask != 0) { 3463 size_t i = __builtin_ctz(resetMask); 3464 ALOG_ASSERT(i < count); 3465 resetMask &= ~(1 << i); 3466 sp<Track> t = mActiveTracks[i].promote(); 3467 if (t == 0) { 3468 continue; 3469 } 3470 Track* track = t.get(); 3471 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3472 track->reset(); 3473 } 3474 3475 // remove all the tracks that need to be... 3476 removeTracks_l(*tracksToRemove); 3477 3478 // sink or mix buffer must be cleared if all tracks are connected to an 3479 // effect chain as in this case the mixer will not write to the sink or mix buffer 3480 // and track effects will accumulate into it 3481 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3482 (mixedTracks == 0 && fastTracks > 0))) { 3483 // FIXME as a performance optimization, should remember previous zero status 3484 if (mMixerBufferValid) { 3485 memset(mMixerBuffer, 0, mMixerBufferSize); 3486 // TODO: In testing, mSinkBuffer below need not be cleared because 3487 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3488 // after mixing. 3489 // 3490 // To enforce this guarantee: 3491 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3492 // (mixedTracks == 0 && fastTracks > 0)) 3493 // must imply MIXER_TRACKS_READY. 3494 // Later, we may clear buffers regardless, and skip much of this logic. 3495 } 3496 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3497 if (mEffectBufferValid) { 3498 memset(mEffectBuffer, 0, mEffectBufferSize); 3499 } 3500 // FIXME as a performance optimization, should remember previous zero status 3501 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3502 } 3503 3504 // if any fast tracks, then status is ready 3505 mMixerStatusIgnoringFastTracks = mixerStatus; 3506 if (fastTracks > 0) { 3507 mixerStatus = MIXER_TRACKS_READY; 3508 } 3509 return mixerStatus; 3510} 3511 3512// getTrackName_l() must be called with ThreadBase::mLock held 3513int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3514{ 3515 return mAudioMixer->getTrackName(channelMask, sessionId); 3516} 3517 3518// deleteTrackName_l() must be called with ThreadBase::mLock held 3519void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3520{ 3521 ALOGV("remove track (%d) and delete from mixer", name); 3522 mAudioMixer->deleteTrackName(name); 3523} 3524 3525// checkForNewParameters_l() must be called with ThreadBase::mLock held 3526bool AudioFlinger::MixerThread::checkForNewParameters_l() 3527{ 3528 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3529 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3530 bool reconfig = false; 3531 3532 while (!mNewParameters.isEmpty()) { 3533 3534 if (mFastMixer != NULL) { 3535 FastMixerStateQueue *sq = mFastMixer->sq(); 3536 FastMixerState *state = sq->begin(); 3537 if (!(state->mCommand & FastMixerState::IDLE)) { 3538 previousCommand = state->mCommand; 3539 state->mCommand = FastMixerState::HOT_IDLE; 3540 sq->end(); 3541 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3542 } else { 3543 sq->end(false /*didModify*/); 3544 } 3545 } 3546 3547 status_t status = NO_ERROR; 3548 String8 keyValuePair = mNewParameters[0]; 3549 AudioParameter param = AudioParameter(keyValuePair); 3550 int value; 3551 3552 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3553 reconfig = true; 3554 } 3555 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3556 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3557 status = BAD_VALUE; 3558 } else { 3559 // no need to save value, since it's constant 3560 reconfig = true; 3561 } 3562 } 3563 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3564 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3565 status = BAD_VALUE; 3566 } else { 3567 // no need to save value, since it's constant 3568 reconfig = true; 3569 } 3570 } 3571 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3572 // do not accept frame count changes if tracks are open as the track buffer 3573 // size depends on frame count and correct behavior would not be guaranteed 3574 // if frame count is changed after track creation 3575 if (!mTracks.isEmpty()) { 3576 status = INVALID_OPERATION; 3577 } else { 3578 reconfig = true; 3579 } 3580 } 3581 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3582#ifdef ADD_BATTERY_DATA 3583 // when changing the audio output device, call addBatteryData to notify 3584 // the change 3585 if (mOutDevice != value) { 3586 uint32_t params = 0; 3587 // check whether speaker is on 3588 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3589 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3590 } 3591 3592 audio_devices_t deviceWithoutSpeaker 3593 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3594 // check if any other device (except speaker) is on 3595 if (value & deviceWithoutSpeaker ) { 3596 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3597 } 3598 3599 if (params != 0) { 3600 addBatteryData(params); 3601 } 3602 } 3603#endif 3604 3605 // forward device change to effects that have requested to be 3606 // aware of attached audio device. 3607 if (value != AUDIO_DEVICE_NONE) { 3608 mOutDevice = value; 3609 for (size_t i = 0; i < mEffectChains.size(); i++) { 3610 mEffectChains[i]->setDevice_l(mOutDevice); 3611 } 3612 } 3613 } 3614 3615 if (status == NO_ERROR) { 3616 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3617 keyValuePair.string()); 3618 if (!mStandby && status == INVALID_OPERATION) { 3619 mOutput->stream->common.standby(&mOutput->stream->common); 3620 mStandby = true; 3621 mBytesWritten = 0; 3622 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3623 keyValuePair.string()); 3624 } 3625 if (status == NO_ERROR && reconfig) { 3626 readOutputParameters_l(); 3627 delete mAudioMixer; 3628 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3629 for (size_t i = 0; i < mTracks.size() ; i++) { 3630 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3631 if (name < 0) { 3632 break; 3633 } 3634 mTracks[i]->mName = name; 3635 } 3636 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3637 } 3638 } 3639 3640 mNewParameters.removeAt(0); 3641 3642 mParamStatus = status; 3643 mParamCond.signal(); 3644 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3645 // already timed out waiting for the status and will never signal the condition. 3646 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3647 } 3648 3649 if (!(previousCommand & FastMixerState::IDLE)) { 3650 ALOG_ASSERT(mFastMixer != NULL); 3651 FastMixerStateQueue *sq = mFastMixer->sq(); 3652 FastMixerState *state = sq->begin(); 3653 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3654 state->mCommand = previousCommand; 3655 sq->end(); 3656 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3657 } 3658 3659 return reconfig; 3660} 3661 3662 3663void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3664{ 3665 const size_t SIZE = 256; 3666 char buffer[SIZE]; 3667 String8 result; 3668 3669 PlaybackThread::dumpInternals(fd, args); 3670 3671 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3672 3673 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3674 const FastMixerDumpState copy(mFastMixerDumpState); 3675 copy.dump(fd); 3676 3677#ifdef STATE_QUEUE_DUMP 3678 // Similar for state queue 3679 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3680 observerCopy.dump(fd); 3681 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3682 mutatorCopy.dump(fd); 3683#endif 3684 3685#ifdef TEE_SINK 3686 // Write the tee output to a .wav file 3687 dumpTee(fd, mTeeSource, mId); 3688#endif 3689 3690#ifdef AUDIO_WATCHDOG 3691 if (mAudioWatchdog != 0) { 3692 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3693 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3694 wdCopy.dump(fd); 3695 } 3696#endif 3697} 3698 3699uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3700{ 3701 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3702} 3703 3704uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3705{ 3706 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3707} 3708 3709void AudioFlinger::MixerThread::cacheParameters_l() 3710{ 3711 PlaybackThread::cacheParameters_l(); 3712 3713 // FIXME: Relaxed timing because of a certain device that can't meet latency 3714 // Should be reduced to 2x after the vendor fixes the driver issue 3715 // increase threshold again due to low power audio mode. The way this warning 3716 // threshold is calculated and its usefulness should be reconsidered anyway. 3717 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3718} 3719 3720// ---------------------------------------------------------------------------- 3721 3722AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3723 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3724 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3725 // mLeftVolFloat, mRightVolFloat 3726{ 3727} 3728 3729AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3730 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3731 ThreadBase::type_t type) 3732 : PlaybackThread(audioFlinger, output, id, device, type) 3733 // mLeftVolFloat, mRightVolFloat 3734{ 3735} 3736 3737AudioFlinger::DirectOutputThread::~DirectOutputThread() 3738{ 3739} 3740 3741void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3742{ 3743 audio_track_cblk_t* cblk = track->cblk(); 3744 float left, right; 3745 3746 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3747 left = right = 0; 3748 } else { 3749 float typeVolume = mStreamTypes[track->streamType()].volume; 3750 float v = mMasterVolume * typeVolume; 3751 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3752 uint32_t vlr = proxy->getVolumeLR(); 3753 float v_clamped = v * (vlr & 0xFFFF); 3754 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3755 left = v_clamped/MAX_GAIN; 3756 v_clamped = v * (vlr >> 16); 3757 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3758 right = v_clamped/MAX_GAIN; 3759 } 3760 3761 if (lastTrack) { 3762 if (left != mLeftVolFloat || right != mRightVolFloat) { 3763 mLeftVolFloat = left; 3764 mRightVolFloat = right; 3765 3766 // Convert volumes from float to 8.24 3767 uint32_t vl = (uint32_t)(left * (1 << 24)); 3768 uint32_t vr = (uint32_t)(right * (1 << 24)); 3769 3770 // Delegate volume control to effect in track effect chain if needed 3771 // only one effect chain can be present on DirectOutputThread, so if 3772 // there is one, the track is connected to it 3773 if (!mEffectChains.isEmpty()) { 3774 mEffectChains[0]->setVolume_l(&vl, &vr); 3775 left = (float)vl / (1 << 24); 3776 right = (float)vr / (1 << 24); 3777 } 3778 if (mOutput->stream->set_volume) { 3779 mOutput->stream->set_volume(mOutput->stream, left, right); 3780 } 3781 } 3782 } 3783} 3784 3785 3786AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3787 Vector< sp<Track> > *tracksToRemove 3788) 3789{ 3790 size_t count = mActiveTracks.size(); 3791 mixer_state mixerStatus = MIXER_IDLE; 3792 3793 // find out which tracks need to be processed 3794 for (size_t i = 0; i < count; i++) { 3795 sp<Track> t = mActiveTracks[i].promote(); 3796 // The track died recently 3797 if (t == 0) { 3798 continue; 3799 } 3800 3801 Track* const track = t.get(); 3802 audio_track_cblk_t* cblk = track->cblk(); 3803 // Only consider last track started for volume and mixer state control. 3804 // In theory an older track could underrun and restart after the new one starts 3805 // but as we only care about the transition phase between two tracks on a 3806 // direct output, it is not a problem to ignore the underrun case. 3807 sp<Track> l = mLatestActiveTrack.promote(); 3808 bool last = l.get() == track; 3809 3810 // The first time a track is added we wait 3811 // for all its buffers to be filled before processing it 3812 uint32_t minFrames; 3813 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3814 minFrames = mNormalFrameCount; 3815 } else { 3816 minFrames = 1; 3817 } 3818 3819 if ((track->framesReady() >= minFrames) && track->isReady() && 3820 !track->isPaused() && !track->isTerminated()) 3821 { 3822 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3823 3824 if (track->mFillingUpStatus == Track::FS_FILLED) { 3825 track->mFillingUpStatus = Track::FS_ACTIVE; 3826 // make sure processVolume_l() will apply new volume even if 0 3827 mLeftVolFloat = mRightVolFloat = -1.0; 3828 if (track->mState == TrackBase::RESUMING) { 3829 track->mState = TrackBase::ACTIVE; 3830 } 3831 } 3832 3833 // compute volume for this track 3834 processVolume_l(track, last); 3835 if (last) { 3836 // reset retry count 3837 track->mRetryCount = kMaxTrackRetriesDirect; 3838 mActiveTrack = t; 3839 mixerStatus = MIXER_TRACKS_READY; 3840 } 3841 } else { 3842 // clear effect chain input buffer if the last active track started underruns 3843 // to avoid sending previous audio buffer again to effects 3844 if (!mEffectChains.isEmpty() && last) { 3845 mEffectChains[0]->clearInputBuffer(); 3846 } 3847 3848 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3849 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3850 track->isStopped() || track->isPaused()) { 3851 // We have consumed all the buffers of this track. 3852 // Remove it from the list of active tracks. 3853 // TODO: implement behavior for compressed audio 3854 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3855 size_t framesWritten = mBytesWritten / mFrameSize; 3856 if (mStandby || !last || 3857 track->presentationComplete(framesWritten, audioHALFrames)) { 3858 if (track->isStopped()) { 3859 track->reset(); 3860 } 3861 tracksToRemove->add(track); 3862 } 3863 } else { 3864 // No buffers for this track. Give it a few chances to 3865 // fill a buffer, then remove it from active list. 3866 // Only consider last track started for mixer state control 3867 if (--(track->mRetryCount) <= 0) { 3868 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3869 tracksToRemove->add(track); 3870 // indicate to client process that the track was disabled because of underrun; 3871 // it will then automatically call start() when data is available 3872 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3873 } else if (last) { 3874 mixerStatus = MIXER_TRACKS_ENABLED; 3875 } 3876 } 3877 } 3878 } 3879 3880 // remove all the tracks that need to be... 3881 removeTracks_l(*tracksToRemove); 3882 3883 return mixerStatus; 3884} 3885 3886void AudioFlinger::DirectOutputThread::threadLoop_mix() 3887{ 3888 size_t frameCount = mFrameCount; 3889 int8_t *curBuf = (int8_t *)mSinkBuffer; 3890 // output audio to hardware 3891 while (frameCount) { 3892 AudioBufferProvider::Buffer buffer; 3893 buffer.frameCount = frameCount; 3894 mActiveTrack->getNextBuffer(&buffer); 3895 if (buffer.raw == NULL) { 3896 memset(curBuf, 0, frameCount * mFrameSize); 3897 break; 3898 } 3899 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3900 frameCount -= buffer.frameCount; 3901 curBuf += buffer.frameCount * mFrameSize; 3902 mActiveTrack->releaseBuffer(&buffer); 3903 } 3904 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3905 sleepTime = 0; 3906 standbyTime = systemTime() + standbyDelay; 3907 mActiveTrack.clear(); 3908} 3909 3910void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3911{ 3912 if (sleepTime == 0) { 3913 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3914 sleepTime = activeSleepTime; 3915 } else { 3916 sleepTime = idleSleepTime; 3917 } 3918 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3919 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3920 sleepTime = 0; 3921 } 3922} 3923 3924// getTrackName_l() must be called with ThreadBase::mLock held 3925int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3926 int sessionId __unused) 3927{ 3928 return 0; 3929} 3930 3931// deleteTrackName_l() must be called with ThreadBase::mLock held 3932void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3933{ 3934} 3935 3936// checkForNewParameters_l() must be called with ThreadBase::mLock held 3937bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3938{ 3939 bool reconfig = false; 3940 3941 while (!mNewParameters.isEmpty()) { 3942 status_t status = NO_ERROR; 3943 String8 keyValuePair = mNewParameters[0]; 3944 AudioParameter param = AudioParameter(keyValuePair); 3945 int value; 3946 3947 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3948 // do not accept frame count changes if tracks are open as the track buffer 3949 // size depends on frame count and correct behavior would not be garantied 3950 // if frame count is changed after track creation 3951 if (!mTracks.isEmpty()) { 3952 status = INVALID_OPERATION; 3953 } else { 3954 reconfig = true; 3955 } 3956 } 3957 if (status == NO_ERROR) { 3958 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3959 keyValuePair.string()); 3960 if (!mStandby && status == INVALID_OPERATION) { 3961 mOutput->stream->common.standby(&mOutput->stream->common); 3962 mStandby = true; 3963 mBytesWritten = 0; 3964 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3965 keyValuePair.string()); 3966 } 3967 if (status == NO_ERROR && reconfig) { 3968 readOutputParameters_l(); 3969 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3970 } 3971 } 3972 3973 mNewParameters.removeAt(0); 3974 3975 mParamStatus = status; 3976 mParamCond.signal(); 3977 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3978 // already timed out waiting for the status and will never signal the condition. 3979 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3980 } 3981 return reconfig; 3982} 3983 3984uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3985{ 3986 uint32_t time; 3987 if (audio_is_linear_pcm(mFormat)) { 3988 time = PlaybackThread::activeSleepTimeUs(); 3989 } else { 3990 time = 10000; 3991 } 3992 return time; 3993} 3994 3995uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3996{ 3997 uint32_t time; 3998 if (audio_is_linear_pcm(mFormat)) { 3999 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4000 } else { 4001 time = 10000; 4002 } 4003 return time; 4004} 4005 4006uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4007{ 4008 uint32_t time; 4009 if (audio_is_linear_pcm(mFormat)) { 4010 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4011 } else { 4012 time = 10000; 4013 } 4014 return time; 4015} 4016 4017void AudioFlinger::DirectOutputThread::cacheParameters_l() 4018{ 4019 PlaybackThread::cacheParameters_l(); 4020 4021 // use shorter standby delay as on normal output to release 4022 // hardware resources as soon as possible 4023 if (audio_is_linear_pcm(mFormat)) { 4024 standbyDelay = microseconds(activeSleepTime*2); 4025 } else { 4026 standbyDelay = kOffloadStandbyDelayNs; 4027 } 4028} 4029 4030// ---------------------------------------------------------------------------- 4031 4032AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4033 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4034 : Thread(false /*canCallJava*/), 4035 mPlaybackThread(playbackThread), 4036 mWriteAckSequence(0), 4037 mDrainSequence(0) 4038{ 4039} 4040 4041AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4042{ 4043} 4044 4045void AudioFlinger::AsyncCallbackThread::onFirstRef() 4046{ 4047 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4048} 4049 4050bool AudioFlinger::AsyncCallbackThread::threadLoop() 4051{ 4052 while (!exitPending()) { 4053 uint32_t writeAckSequence; 4054 uint32_t drainSequence; 4055 4056 { 4057 Mutex::Autolock _l(mLock); 4058 while (!((mWriteAckSequence & 1) || 4059 (mDrainSequence & 1) || 4060 exitPending())) { 4061 mWaitWorkCV.wait(mLock); 4062 } 4063 4064 if (exitPending()) { 4065 break; 4066 } 4067 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4068 mWriteAckSequence, mDrainSequence); 4069 writeAckSequence = mWriteAckSequence; 4070 mWriteAckSequence &= ~1; 4071 drainSequence = mDrainSequence; 4072 mDrainSequence &= ~1; 4073 } 4074 { 4075 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4076 if (playbackThread != 0) { 4077 if (writeAckSequence & 1) { 4078 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4079 } 4080 if (drainSequence & 1) { 4081 playbackThread->resetDraining(drainSequence >> 1); 4082 } 4083 } 4084 } 4085 } 4086 return false; 4087} 4088 4089void AudioFlinger::AsyncCallbackThread::exit() 4090{ 4091 ALOGV("AsyncCallbackThread::exit"); 4092 Mutex::Autolock _l(mLock); 4093 requestExit(); 4094 mWaitWorkCV.broadcast(); 4095} 4096 4097void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4098{ 4099 Mutex::Autolock _l(mLock); 4100 // bit 0 is cleared 4101 mWriteAckSequence = sequence << 1; 4102} 4103 4104void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4105{ 4106 Mutex::Autolock _l(mLock); 4107 // ignore unexpected callbacks 4108 if (mWriteAckSequence & 2) { 4109 mWriteAckSequence |= 1; 4110 mWaitWorkCV.signal(); 4111 } 4112} 4113 4114void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4115{ 4116 Mutex::Autolock _l(mLock); 4117 // bit 0 is cleared 4118 mDrainSequence = sequence << 1; 4119} 4120 4121void AudioFlinger::AsyncCallbackThread::resetDraining() 4122{ 4123 Mutex::Autolock _l(mLock); 4124 // ignore unexpected callbacks 4125 if (mDrainSequence & 2) { 4126 mDrainSequence |= 1; 4127 mWaitWorkCV.signal(); 4128 } 4129} 4130 4131 4132// ---------------------------------------------------------------------------- 4133AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4134 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4135 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4136 mHwPaused(false), 4137 mFlushPending(false), 4138 mPausedBytesRemaining(0) 4139{ 4140 //FIXME: mStandby should be set to true by ThreadBase constructor 4141 mStandby = true; 4142} 4143 4144void AudioFlinger::OffloadThread::threadLoop_exit() 4145{ 4146 if (mFlushPending || mHwPaused) { 4147 // If a flush is pending or track was paused, just discard buffered data 4148 flushHw_l(); 4149 } else { 4150 mMixerStatus = MIXER_DRAIN_ALL; 4151 threadLoop_drain(); 4152 } 4153 mCallbackThread->exit(); 4154 PlaybackThread::threadLoop_exit(); 4155} 4156 4157AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4158 Vector< sp<Track> > *tracksToRemove 4159) 4160{ 4161 size_t count = mActiveTracks.size(); 4162 4163 mixer_state mixerStatus = MIXER_IDLE; 4164 bool doHwPause = false; 4165 bool doHwResume = false; 4166 4167 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4168 4169 // find out which tracks need to be processed 4170 for (size_t i = 0; i < count; i++) { 4171 sp<Track> t = mActiveTracks[i].promote(); 4172 // The track died recently 4173 if (t == 0) { 4174 continue; 4175 } 4176 Track* const track = t.get(); 4177 audio_track_cblk_t* cblk = track->cblk(); 4178 // Only consider last track started for volume and mixer state control. 4179 // In theory an older track could underrun and restart after the new one starts 4180 // but as we only care about the transition phase between two tracks on a 4181 // direct output, it is not a problem to ignore the underrun case. 4182 sp<Track> l = mLatestActiveTrack.promote(); 4183 bool last = l.get() == track; 4184 4185 if (track->isInvalid()) { 4186 ALOGW("An invalidated track shouldn't be in active list"); 4187 tracksToRemove->add(track); 4188 continue; 4189 } 4190 4191 if (track->mState == TrackBase::IDLE) { 4192 ALOGW("An idle track shouldn't be in active list"); 4193 continue; 4194 } 4195 4196 if (track->isPausing()) { 4197 track->setPaused(); 4198 if (last) { 4199 if (!mHwPaused) { 4200 doHwPause = true; 4201 mHwPaused = true; 4202 } 4203 // If we were part way through writing the mixbuffer to 4204 // the HAL we must save this until we resume 4205 // BUG - this will be wrong if a different track is made active, 4206 // in that case we want to discard the pending data in the 4207 // mixbuffer and tell the client to present it again when the 4208 // track is resumed 4209 mPausedWriteLength = mCurrentWriteLength; 4210 mPausedBytesRemaining = mBytesRemaining; 4211 mBytesRemaining = 0; // stop writing 4212 } 4213 tracksToRemove->add(track); 4214 } else if (track->isFlushPending()) { 4215 track->flushAck(); 4216 if (last) { 4217 mFlushPending = true; 4218 } 4219 } else if (track->framesReady() && track->isReady() && 4220 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4221 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4222 if (track->mFillingUpStatus == Track::FS_FILLED) { 4223 track->mFillingUpStatus = Track::FS_ACTIVE; 4224 // make sure processVolume_l() will apply new volume even if 0 4225 mLeftVolFloat = mRightVolFloat = -1.0; 4226 if (track->mState == TrackBase::RESUMING) { 4227 track->mState = TrackBase::ACTIVE; 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 } 4245 } 4246 4247 if (last) { 4248 sp<Track> previousTrack = mPreviousTrack.promote(); 4249 if (previousTrack != 0) { 4250 if (track != previousTrack.get()) { 4251 // Flush any data still being written from last track 4252 mBytesRemaining = 0; 4253 if (mPausedBytesRemaining) { 4254 // Last track was paused so we also need to flush saved 4255 // mixbuffer state and invalidate track so that it will 4256 // re-submit that unwritten data when it is next resumed 4257 mPausedBytesRemaining = 0; 4258 // Invalidate is a bit drastic - would be more efficient 4259 // to have a flag to tell client that some of the 4260 // previously written data was lost 4261 previousTrack->invalidate(); 4262 } 4263 // flush data already sent to the DSP if changing audio session as audio 4264 // comes from a different source. Also invalidate previous track to force a 4265 // seek when resuming. 4266 if (previousTrack->sessionId() != track->sessionId()) { 4267 previousTrack->invalidate(); 4268 } 4269 } 4270 } 4271 mPreviousTrack = track; 4272 // reset retry count 4273 track->mRetryCount = kMaxTrackRetriesOffload; 4274 mActiveTrack = t; 4275 mixerStatus = MIXER_TRACKS_READY; 4276 } 4277 } else { 4278 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4279 if (track->isStopping_1()) { 4280 // Hardware buffer can hold a large amount of audio so we must 4281 // wait for all current track's data to drain before we say 4282 // that the track is stopped. 4283 if (mBytesRemaining == 0) { 4284 // Only start draining when all data in mixbuffer 4285 // has been written 4286 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4287 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4288 // do not drain if no data was ever sent to HAL (mStandby == true) 4289 if (last && !mStandby) { 4290 // do not modify drain sequence if we are already draining. This happens 4291 // when resuming from pause after drain. 4292 if ((mDrainSequence & 1) == 0) { 4293 sleepTime = 0; 4294 standbyTime = systemTime() + standbyDelay; 4295 mixerStatus = MIXER_DRAIN_TRACK; 4296 mDrainSequence += 2; 4297 } 4298 if (mHwPaused) { 4299 // It is possible to move from PAUSED to STOPPING_1 without 4300 // a resume so we must ensure hardware is running 4301 doHwResume = true; 4302 mHwPaused = false; 4303 } 4304 } 4305 } 4306 } else if (track->isStopping_2()) { 4307 // Drain has completed or we are in standby, signal presentation complete 4308 if (!(mDrainSequence & 1) || !last || mStandby) { 4309 track->mState = TrackBase::STOPPED; 4310 size_t audioHALFrames = 4311 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4312 size_t framesWritten = 4313 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4314 track->presentationComplete(framesWritten, audioHALFrames); 4315 track->reset(); 4316 tracksToRemove->add(track); 4317 } 4318 } else { 4319 // No buffers for this track. Give it a few chances to 4320 // fill a buffer, then remove it from active list. 4321 if (--(track->mRetryCount) <= 0) { 4322 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4323 track->name()); 4324 tracksToRemove->add(track); 4325 // indicate to client process that the track was disabled because of underrun; 4326 // it will then automatically call start() when data is available 4327 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4328 } else if (last){ 4329 mixerStatus = MIXER_TRACKS_ENABLED; 4330 } 4331 } 4332 } 4333 // compute volume for this track 4334 processVolume_l(track, last); 4335 } 4336 4337 // make sure the pause/flush/resume sequence is executed in the right order. 4338 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4339 // before flush and then resume HW. This can happen in case of pause/flush/resume 4340 // if resume is received before pause is executed. 4341 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4342 mOutput->stream->pause(mOutput->stream); 4343 } 4344 if (mFlushPending) { 4345 flushHw_l(); 4346 mFlushPending = false; 4347 } 4348 if (!mStandby && doHwResume) { 4349 mOutput->stream->resume(mOutput->stream); 4350 } 4351 4352 // remove all the tracks that need to be... 4353 removeTracks_l(*tracksToRemove); 4354 4355 return mixerStatus; 4356} 4357 4358// must be called with thread mutex locked 4359bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4360{ 4361 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4362 mWriteAckSequence, mDrainSequence); 4363 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4364 return true; 4365 } 4366 return false; 4367} 4368 4369// must be called with thread mutex locked 4370bool AudioFlinger::OffloadThread::shouldStandby_l() 4371{ 4372 bool trackPaused = false; 4373 4374 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4375 // after a timeout and we will enter standby then. 4376 if (mTracks.size() > 0) { 4377 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4378 } 4379 4380 return !mStandby && !trackPaused; 4381} 4382 4383 4384bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4385{ 4386 Mutex::Autolock _l(mLock); 4387 return waitingAsyncCallback_l(); 4388} 4389 4390void AudioFlinger::OffloadThread::flushHw_l() 4391{ 4392 mOutput->stream->flush(mOutput->stream); 4393 // Flush anything still waiting in the mixbuffer 4394 mCurrentWriteLength = 0; 4395 mBytesRemaining = 0; 4396 mPausedWriteLength = 0; 4397 mPausedBytesRemaining = 0; 4398 mHwPaused = false; 4399 4400 if (mUseAsyncWrite) { 4401 // discard any pending drain or write ack by incrementing sequence 4402 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4403 mDrainSequence = (mDrainSequence + 2) & ~1; 4404 ALOG_ASSERT(mCallbackThread != 0); 4405 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4406 mCallbackThread->setDraining(mDrainSequence); 4407 } 4408} 4409 4410void AudioFlinger::OffloadThread::onAddNewTrack_l() 4411{ 4412 sp<Track> previousTrack = mPreviousTrack.promote(); 4413 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4414 4415 if (previousTrack != 0 && latestTrack != 0 && 4416 (previousTrack->sessionId() != latestTrack->sessionId())) { 4417 mFlushPending = true; 4418 } 4419 PlaybackThread::onAddNewTrack_l(); 4420} 4421 4422// ---------------------------------------------------------------------------- 4423 4424AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4425 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4426 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4427 DUPLICATING), 4428 mWaitTimeMs(UINT_MAX) 4429{ 4430 addOutputTrack(mainThread); 4431} 4432 4433AudioFlinger::DuplicatingThread::~DuplicatingThread() 4434{ 4435 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4436 mOutputTracks[i]->destroy(); 4437 } 4438} 4439 4440void AudioFlinger::DuplicatingThread::threadLoop_mix() 4441{ 4442 // mix buffers... 4443 if (outputsReady(outputTracks)) { 4444 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4445 } else { 4446 memset(mSinkBuffer, 0, mSinkBufferSize); 4447 } 4448 sleepTime = 0; 4449 writeFrames = mNormalFrameCount; 4450 mCurrentWriteLength = mSinkBufferSize; 4451 standbyTime = systemTime() + standbyDelay; 4452} 4453 4454void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4455{ 4456 if (sleepTime == 0) { 4457 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4458 sleepTime = activeSleepTime; 4459 } else { 4460 sleepTime = idleSleepTime; 4461 } 4462 } else if (mBytesWritten != 0) { 4463 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4464 writeFrames = mNormalFrameCount; 4465 memset(mSinkBuffer, 0, mSinkBufferSize); 4466 } else { 4467 // flush remaining overflow buffers in output tracks 4468 writeFrames = 0; 4469 } 4470 sleepTime = 0; 4471 } 4472} 4473 4474ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4475{ 4476 for (size_t i = 0; i < outputTracks.size(); i++) { 4477 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4478 // for delivery downstream as needed. This in-place conversion is safe as 4479 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4480 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4481 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4482 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4483 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4484 } 4485 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4486 } 4487 mStandby = false; 4488 return (ssize_t)mSinkBufferSize; 4489} 4490 4491void AudioFlinger::DuplicatingThread::threadLoop_standby() 4492{ 4493 // DuplicatingThread implements standby by stopping all tracks 4494 for (size_t i = 0; i < outputTracks.size(); i++) { 4495 outputTracks[i]->stop(); 4496 } 4497} 4498 4499void AudioFlinger::DuplicatingThread::saveOutputTracks() 4500{ 4501 outputTracks = mOutputTracks; 4502} 4503 4504void AudioFlinger::DuplicatingThread::clearOutputTracks() 4505{ 4506 outputTracks.clear(); 4507} 4508 4509void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4510{ 4511 Mutex::Autolock _l(mLock); 4512 // FIXME explain this formula 4513 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4514 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4515 // due to current usage case and restrictions on the AudioBufferProvider. 4516 // Actual buffer conversion is done in threadLoop_write(). 4517 // 4518 // TODO: This may change in the future, depending on multichannel 4519 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4520 OutputTrack *outputTrack = new OutputTrack(thread, 4521 this, 4522 mSampleRate, 4523 AUDIO_FORMAT_PCM_16_BIT, 4524 mChannelMask, 4525 frameCount, 4526 IPCThreadState::self()->getCallingUid()); 4527 if (outputTrack->cblk() != NULL) { 4528 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4529 mOutputTracks.add(outputTrack); 4530 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4531 updateWaitTime_l(); 4532 } 4533} 4534 4535void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4536{ 4537 Mutex::Autolock _l(mLock); 4538 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4539 if (mOutputTracks[i]->thread() == thread) { 4540 mOutputTracks[i]->destroy(); 4541 mOutputTracks.removeAt(i); 4542 updateWaitTime_l(); 4543 return; 4544 } 4545 } 4546 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4547} 4548 4549// caller must hold mLock 4550void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4551{ 4552 mWaitTimeMs = UINT_MAX; 4553 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4554 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4555 if (strong != 0) { 4556 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4557 if (waitTimeMs < mWaitTimeMs) { 4558 mWaitTimeMs = waitTimeMs; 4559 } 4560 } 4561 } 4562} 4563 4564 4565bool AudioFlinger::DuplicatingThread::outputsReady( 4566 const SortedVector< sp<OutputTrack> > &outputTracks) 4567{ 4568 for (size_t i = 0; i < outputTracks.size(); i++) { 4569 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4570 if (thread == 0) { 4571 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4572 outputTracks[i].get()); 4573 return false; 4574 } 4575 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4576 // see note at standby() declaration 4577 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4578 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4579 thread.get()); 4580 return false; 4581 } 4582 } 4583 return true; 4584} 4585 4586uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4587{ 4588 return (mWaitTimeMs * 1000) / 2; 4589} 4590 4591void AudioFlinger::DuplicatingThread::cacheParameters_l() 4592{ 4593 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4594 updateWaitTime_l(); 4595 4596 MixerThread::cacheParameters_l(); 4597} 4598 4599// ---------------------------------------------------------------------------- 4600// Record 4601// ---------------------------------------------------------------------------- 4602 4603AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4604 AudioStreamIn *input, 4605 audio_io_handle_t id, 4606 audio_devices_t outDevice, 4607 audio_devices_t inDevice 4608#ifdef TEE_SINK 4609 , const sp<NBAIO_Sink>& teeSink 4610#endif 4611 ) : 4612 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4613 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4614 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4615 mRsmpInRear(0) 4616#ifdef TEE_SINK 4617 , mTeeSink(teeSink) 4618#endif 4619{ 4620 snprintf(mName, kNameLength, "AudioIn_%X", id); 4621 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4622 4623 readInputParameters_l(); 4624} 4625 4626 4627AudioFlinger::RecordThread::~RecordThread() 4628{ 4629 mAudioFlinger->unregisterWriter(mNBLogWriter); 4630 delete[] mRsmpInBuffer; 4631} 4632 4633void AudioFlinger::RecordThread::onFirstRef() 4634{ 4635 run(mName, PRIORITY_URGENT_AUDIO); 4636} 4637 4638bool AudioFlinger::RecordThread::threadLoop() 4639{ 4640 nsecs_t lastWarning = 0; 4641 4642 inputStandBy(); 4643 4644reacquire_wakelock: 4645 sp<RecordTrack> activeTrack; 4646 int activeTracksGen; 4647 { 4648 Mutex::Autolock _l(mLock); 4649 size_t size = mActiveTracks.size(); 4650 activeTracksGen = mActiveTracksGen; 4651 if (size > 0) { 4652 // FIXME an arbitrary choice 4653 activeTrack = mActiveTracks[0]; 4654 acquireWakeLock_l(activeTrack->uid()); 4655 if (size > 1) { 4656 SortedVector<int> tmp; 4657 for (size_t i = 0; i < size; i++) { 4658 tmp.add(mActiveTracks[i]->uid()); 4659 } 4660 updateWakeLockUids_l(tmp); 4661 } 4662 } else { 4663 acquireWakeLock_l(-1); 4664 } 4665 } 4666 4667 // used to request a deferred sleep, to be executed later while mutex is unlocked 4668 uint32_t sleepUs = 0; 4669 4670 // loop while there is work to do 4671 for (;;) { 4672 Vector< sp<EffectChain> > effectChains; 4673 4674 // sleep with mutex unlocked 4675 if (sleepUs > 0) { 4676 usleep(sleepUs); 4677 sleepUs = 0; 4678 } 4679 4680 // activeTracks accumulates a copy of a subset of mActiveTracks 4681 Vector< sp<RecordTrack> > activeTracks; 4682 4683 { // scope for mLock 4684 Mutex::Autolock _l(mLock); 4685 4686 processConfigEvents_l(); 4687 // return value 'reconfig' is currently unused 4688 bool reconfig = checkForNewParameters_l(); 4689 4690 // check exitPending here because checkForNewParameters_l() and 4691 // checkForNewParameters_l() can temporarily release mLock 4692 if (exitPending()) { 4693 break; 4694 } 4695 4696 // if no active track(s), then standby and release wakelock 4697 size_t size = mActiveTracks.size(); 4698 if (size == 0) { 4699 standbyIfNotAlreadyInStandby(); 4700 // exitPending() can't become true here 4701 releaseWakeLock_l(); 4702 ALOGV("RecordThread: loop stopping"); 4703 // go to sleep 4704 mWaitWorkCV.wait(mLock); 4705 ALOGV("RecordThread: loop starting"); 4706 goto reacquire_wakelock; 4707 } 4708 4709 if (mActiveTracksGen != activeTracksGen) { 4710 activeTracksGen = mActiveTracksGen; 4711 SortedVector<int> tmp; 4712 for (size_t i = 0; i < size; i++) { 4713 tmp.add(mActiveTracks[i]->uid()); 4714 } 4715 updateWakeLockUids_l(tmp); 4716 } 4717 4718 bool doBroadcast = false; 4719 for (size_t i = 0; i < size; ) { 4720 4721 activeTrack = mActiveTracks[i]; 4722 if (activeTrack->isTerminated()) { 4723 removeTrack_l(activeTrack); 4724 mActiveTracks.remove(activeTrack); 4725 mActiveTracksGen++; 4726 size--; 4727 continue; 4728 } 4729 4730 TrackBase::track_state activeTrackState = activeTrack->mState; 4731 switch (activeTrackState) { 4732 4733 case TrackBase::PAUSING: 4734 mActiveTracks.remove(activeTrack); 4735 mActiveTracksGen++; 4736 doBroadcast = true; 4737 size--; 4738 continue; 4739 4740 case TrackBase::STARTING_1: 4741 sleepUs = 10000; 4742 i++; 4743 continue; 4744 4745 case TrackBase::STARTING_2: 4746 doBroadcast = true; 4747 mStandby = false; 4748 activeTrack->mState = TrackBase::ACTIVE; 4749 break; 4750 4751 case TrackBase::ACTIVE: 4752 break; 4753 4754 case TrackBase::IDLE: 4755 i++; 4756 continue; 4757 4758 default: 4759 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4760 } 4761 4762 activeTracks.add(activeTrack); 4763 i++; 4764 4765 } 4766 if (doBroadcast) { 4767 mStartStopCond.broadcast(); 4768 } 4769 4770 // sleep if there are no active tracks to process 4771 if (activeTracks.size() == 0) { 4772 if (sleepUs == 0) { 4773 sleepUs = kRecordThreadSleepUs; 4774 } 4775 continue; 4776 } 4777 sleepUs = 0; 4778 4779 lockEffectChains_l(effectChains); 4780 } 4781 4782 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4783 4784 size_t size = effectChains.size(); 4785 for (size_t i = 0; i < size; i++) { 4786 // thread mutex is not locked, but effect chain is locked 4787 effectChains[i]->process_l(); 4788 } 4789 4790 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4791 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4792 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4793 // If destination is non-contiguous, first read past the nominal end of buffer, then 4794 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4795 4796 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4797 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4798 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4799 if (bytesRead <= 0) { 4800 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4801 // Force input into standby so that it tries to recover at next read attempt 4802 inputStandBy(); 4803 sleepUs = kRecordThreadSleepUs; 4804 continue; 4805 } 4806 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4807 size_t framesRead = bytesRead / mFrameSize; 4808 ALOG_ASSERT(framesRead > 0); 4809 if (mTeeSink != 0) { 4810 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4811 } 4812 // If destination is non-contiguous, we now correct for reading past end of buffer. 4813 size_t part1 = mRsmpInFramesP2 - rear; 4814 if (framesRead > part1) { 4815 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4816 (framesRead - part1) * mFrameSize); 4817 } 4818 rear = mRsmpInRear += framesRead; 4819 4820 size = activeTracks.size(); 4821 // loop over each active track 4822 for (size_t i = 0; i < size; i++) { 4823 activeTrack = activeTracks[i]; 4824 4825 enum { 4826 OVERRUN_UNKNOWN, 4827 OVERRUN_TRUE, 4828 OVERRUN_FALSE 4829 } overrun = OVERRUN_UNKNOWN; 4830 4831 // loop over getNextBuffer to handle circular sink 4832 for (;;) { 4833 4834 activeTrack->mSink.frameCount = ~0; 4835 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4836 size_t framesOut = activeTrack->mSink.frameCount; 4837 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4838 4839 int32_t front = activeTrack->mRsmpInFront; 4840 ssize_t filled = rear - front; 4841 size_t framesIn; 4842 4843 if (filled < 0) { 4844 // should not happen, but treat like a massive overrun and re-sync 4845 framesIn = 0; 4846 activeTrack->mRsmpInFront = rear; 4847 overrun = OVERRUN_TRUE; 4848 } else if ((size_t) filled <= mRsmpInFrames) { 4849 framesIn = (size_t) filled; 4850 } else { 4851 // client is not keeping up with server, but give it latest data 4852 framesIn = mRsmpInFrames; 4853 activeTrack->mRsmpInFront = front = rear - framesIn; 4854 overrun = OVERRUN_TRUE; 4855 } 4856 4857 if (framesOut == 0 || framesIn == 0) { 4858 break; 4859 } 4860 4861 if (activeTrack->mResampler == NULL) { 4862 // no resampling 4863 if (framesIn > framesOut) { 4864 framesIn = framesOut; 4865 } else { 4866 framesOut = framesIn; 4867 } 4868 int8_t *dst = activeTrack->mSink.i8; 4869 while (framesIn > 0) { 4870 front &= mRsmpInFramesP2 - 1; 4871 size_t part1 = mRsmpInFramesP2 - front; 4872 if (part1 > framesIn) { 4873 part1 = framesIn; 4874 } 4875 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4876 if (mChannelCount == activeTrack->mChannelCount) { 4877 memcpy(dst, src, part1 * mFrameSize); 4878 } else if (mChannelCount == 1) { 4879 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4880 part1); 4881 } else { 4882 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4883 part1); 4884 } 4885 dst += part1 * activeTrack->mFrameSize; 4886 front += part1; 4887 framesIn -= part1; 4888 } 4889 activeTrack->mRsmpInFront += framesOut; 4890 4891 } else { 4892 // resampling 4893 // FIXME framesInNeeded should really be part of resampler API, and should 4894 // depend on the SRC ratio 4895 // to keep mRsmpInBuffer full so resampler always has sufficient input 4896 size_t framesInNeeded; 4897 // FIXME only re-calculate when it changes, and optimize for common ratios 4898 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4899 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4900 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4901 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4902 framesInNeeded, framesOut, inOverOut); 4903 // Although we theoretically have framesIn in circular buffer, some of those are 4904 // unreleased frames, and thus must be discounted for purpose of budgeting. 4905 size_t unreleased = activeTrack->mRsmpInUnrel; 4906 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4907 if (framesIn < framesInNeeded) { 4908 ALOGV("not enough to resample: have %u frames in but need %u in to " 4909 "produce %u out given in/out ratio of %.4g", 4910 framesIn, framesInNeeded, framesOut, inOverOut); 4911 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4912 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4913 if (newFramesOut == 0) { 4914 break; 4915 } 4916 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4917 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4918 framesInNeeded, newFramesOut, outOverIn); 4919 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4920 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4921 "given in/out ratio of %.4g", 4922 framesIn, framesInNeeded, newFramesOut, inOverOut); 4923 framesOut = newFramesOut; 4924 } else { 4925 ALOGV("success 1: have %u in and need %u in to produce %u out " 4926 "given in/out ratio of %.4g", 4927 framesIn, framesInNeeded, framesOut, inOverOut); 4928 } 4929 4930 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4931 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4932 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4933 delete[] activeTrack->mRsmpOutBuffer; 4934 // resampler always outputs stereo 4935 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4936 activeTrack->mRsmpOutFrameCount = framesOut; 4937 } 4938 4939 // resampler accumulates, but we only have one source track 4940 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4941 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4942 // FIXME how about having activeTrack implement this interface itself? 4943 activeTrack->mResamplerBufferProvider 4944 /*this*/ /* AudioBufferProvider* */); 4945 // ditherAndClamp() works as long as all buffers returned by 4946 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4947 if (activeTrack->mChannelCount == 1) { 4948 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4949 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4950 framesOut); 4951 // the resampler always outputs stereo samples: 4952 // do post stereo to mono conversion 4953 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4954 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4955 } else { 4956 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4957 activeTrack->mRsmpOutBuffer, framesOut); 4958 } 4959 // now done with mRsmpOutBuffer 4960 4961 } 4962 4963 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4964 overrun = OVERRUN_FALSE; 4965 } 4966 4967 if (activeTrack->mFramesToDrop == 0) { 4968 if (framesOut > 0) { 4969 activeTrack->mSink.frameCount = framesOut; 4970 activeTrack->releaseBuffer(&activeTrack->mSink); 4971 } 4972 } else { 4973 // FIXME could do a partial drop of framesOut 4974 if (activeTrack->mFramesToDrop > 0) { 4975 activeTrack->mFramesToDrop -= framesOut; 4976 if (activeTrack->mFramesToDrop <= 0) { 4977 activeTrack->clearSyncStartEvent(); 4978 } 4979 } else { 4980 activeTrack->mFramesToDrop += framesOut; 4981 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4982 activeTrack->mSyncStartEvent->isCancelled()) { 4983 ALOGW("Synced record %s, session %d, trigger session %d", 4984 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 4985 activeTrack->sessionId(), 4986 (activeTrack->mSyncStartEvent != 0) ? 4987 activeTrack->mSyncStartEvent->triggerSession() : 0); 4988 activeTrack->clearSyncStartEvent(); 4989 } 4990 } 4991 } 4992 4993 if (framesOut == 0) { 4994 break; 4995 } 4996 } 4997 4998 switch (overrun) { 4999 case OVERRUN_TRUE: 5000 // client isn't retrieving buffers fast enough 5001 if (!activeTrack->setOverflow()) { 5002 nsecs_t now = systemTime(); 5003 // FIXME should lastWarning per track? 5004 if ((now - lastWarning) > kWarningThrottleNs) { 5005 ALOGW("RecordThread: buffer overflow"); 5006 lastWarning = now; 5007 } 5008 } 5009 break; 5010 case OVERRUN_FALSE: 5011 activeTrack->clearOverflow(); 5012 break; 5013 case OVERRUN_UNKNOWN: 5014 break; 5015 } 5016 5017 } 5018 5019 // enable changes in effect chain 5020 unlockEffectChains(effectChains); 5021 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5022 } 5023 5024 standbyIfNotAlreadyInStandby(); 5025 5026 { 5027 Mutex::Autolock _l(mLock); 5028 for (size_t i = 0; i < mTracks.size(); i++) { 5029 sp<RecordTrack> track = mTracks[i]; 5030 track->invalidate(); 5031 } 5032 mActiveTracks.clear(); 5033 mActiveTracksGen++; 5034 mStartStopCond.broadcast(); 5035 } 5036 5037 releaseWakeLock(); 5038 5039 ALOGV("RecordThread %p exiting", this); 5040 return false; 5041} 5042 5043void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5044{ 5045 if (!mStandby) { 5046 inputStandBy(); 5047 mStandby = true; 5048 } 5049} 5050 5051void AudioFlinger::RecordThread::inputStandBy() 5052{ 5053 mInput->stream->common.standby(&mInput->stream->common); 5054} 5055 5056sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5057 const sp<AudioFlinger::Client>& client, 5058 uint32_t sampleRate, 5059 audio_format_t format, 5060 audio_channel_mask_t channelMask, 5061 size_t *pFrameCount, 5062 int sessionId, 5063 int uid, 5064 IAudioFlinger::track_flags_t *flags, 5065 pid_t tid, 5066 status_t *status) 5067{ 5068 size_t frameCount = *pFrameCount; 5069 sp<RecordTrack> track; 5070 status_t lStatus; 5071 5072 lStatus = initCheck(); 5073 if (lStatus != NO_ERROR) { 5074 ALOGE("createRecordTrack_l() audio driver not initialized"); 5075 goto Exit; 5076 } 5077 5078 // client expresses a preference for FAST, but we get the final say 5079 if (*flags & IAudioFlinger::TRACK_FAST) { 5080 if ( 5081 // use case: callback handler and frame count is default or at least as large as HAL 5082 ( 5083 (tid != -1) && 5084 ((frameCount == 0) || 5085 (frameCount >= mFrameCount)) 5086 ) && 5087 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 5088 // mono or stereo 5089 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 5090 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 5091 // hardware sample rate 5092 (sampleRate == mSampleRate) && 5093 // record thread has an associated fast recorder 5094 hasFastRecorder() 5095 // FIXME test that RecordThread for this fast track has a capable output HAL 5096 // FIXME add a permission test also? 5097 ) { 5098 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 5099 if (frameCount == 0) { 5100 frameCount = mFrameCount * kFastTrackMultiplier; 5101 } 5102 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5103 frameCount, mFrameCount); 5104 } else { 5105 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5106 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5107 "hasFastRecorder=%d tid=%d", 5108 frameCount, mFrameCount, format, 5109 audio_is_linear_pcm(format), 5110 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 5111 *flags &= ~IAudioFlinger::TRACK_FAST; 5112 // For compatibility with AudioRecord calculation, buffer depth is forced 5113 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5114 // This is probably too conservative, but legacy application code may depend on it. 5115 // If you change this calculation, also review the start threshold which is related. 5116 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5117 size_t mNormalFrameCount = 2048; // FIXME 5118 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5119 if (minBufCount < 2) { 5120 minBufCount = 2; 5121 } 5122 size_t minFrameCount = mNormalFrameCount * minBufCount; 5123 if (frameCount < minFrameCount) { 5124 frameCount = minFrameCount; 5125 } 5126 } 5127 } 5128 *pFrameCount = frameCount; 5129 5130 // FIXME use flags and tid similar to createTrack_l() 5131 5132 { // scope for mLock 5133 Mutex::Autolock _l(mLock); 5134 5135 track = new RecordTrack(this, client, sampleRate, 5136 format, channelMask, frameCount, sessionId, uid); 5137 5138 lStatus = track->initCheck(); 5139 if (lStatus != NO_ERROR) { 5140 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5141 // track must be cleared from the caller as the caller has the AF lock 5142 goto Exit; 5143 } 5144 mTracks.add(track); 5145 5146 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5147 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5148 mAudioFlinger->btNrecIsOff(); 5149 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5150 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5151 5152 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5153 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5154 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5155 // so ask activity manager to do this on our behalf 5156 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5157 } 5158 } 5159 lStatus = NO_ERROR; 5160 5161Exit: 5162 *status = lStatus; 5163 return track; 5164} 5165 5166status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5167 AudioSystem::sync_event_t event, 5168 int triggerSession) 5169{ 5170 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5171 sp<ThreadBase> strongMe = this; 5172 status_t status = NO_ERROR; 5173 5174 if (event == AudioSystem::SYNC_EVENT_NONE) { 5175 recordTrack->clearSyncStartEvent(); 5176 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5177 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5178 triggerSession, 5179 recordTrack->sessionId(), 5180 syncStartEventCallback, 5181 recordTrack); 5182 // Sync event can be cancelled by the trigger session if the track is not in a 5183 // compatible state in which case we start record immediately 5184 if (recordTrack->mSyncStartEvent->isCancelled()) { 5185 recordTrack->clearSyncStartEvent(); 5186 } else { 5187 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5188 recordTrack->mFramesToDrop = - 5189 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5190 } 5191 } 5192 5193 { 5194 // This section is a rendezvous between binder thread executing start() and RecordThread 5195 AutoMutex lock(mLock); 5196 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5197 if (recordTrack->mState == TrackBase::PAUSING) { 5198 ALOGV("active record track PAUSING -> ACTIVE"); 5199 recordTrack->mState = TrackBase::ACTIVE; 5200 } else { 5201 ALOGV("active record track state %d", recordTrack->mState); 5202 } 5203 return status; 5204 } 5205 5206 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5207 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5208 // or using a separate command thread 5209 recordTrack->mState = TrackBase::STARTING_1; 5210 mActiveTracks.add(recordTrack); 5211 mActiveTracksGen++; 5212 mLock.unlock(); 5213 status_t status = AudioSystem::startInput(mId); 5214 mLock.lock(); 5215 // FIXME should verify that recordTrack is still in mActiveTracks 5216 if (status != NO_ERROR) { 5217 mActiveTracks.remove(recordTrack); 5218 mActiveTracksGen++; 5219 recordTrack->clearSyncStartEvent(); 5220 return status; 5221 } 5222 // Catch up with current buffer indices if thread is already running. 5223 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5224 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5225 // see previously buffered data before it called start(), but with greater risk of overrun. 5226 5227 recordTrack->mRsmpInFront = mRsmpInRear; 5228 recordTrack->mRsmpInUnrel = 0; 5229 // FIXME why reset? 5230 if (recordTrack->mResampler != NULL) { 5231 recordTrack->mResampler->reset(); 5232 } 5233 recordTrack->mState = TrackBase::STARTING_2; 5234 // signal thread to start 5235 mWaitWorkCV.broadcast(); 5236 if (mActiveTracks.indexOf(recordTrack) < 0) { 5237 ALOGV("Record failed to start"); 5238 status = BAD_VALUE; 5239 goto startError; 5240 } 5241 return status; 5242 } 5243 5244startError: 5245 AudioSystem::stopInput(mId); 5246 recordTrack->clearSyncStartEvent(); 5247 // FIXME I wonder why we do not reset the state here? 5248 return status; 5249} 5250 5251void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5252{ 5253 sp<SyncEvent> strongEvent = event.promote(); 5254 5255 if (strongEvent != 0) { 5256 sp<RefBase> ptr = strongEvent->cookie().promote(); 5257 if (ptr != 0) { 5258 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5259 recordTrack->handleSyncStartEvent(strongEvent); 5260 } 5261 } 5262} 5263 5264bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5265 ALOGV("RecordThread::stop"); 5266 AutoMutex _l(mLock); 5267 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5268 return false; 5269 } 5270 // note that threadLoop may still be processing the track at this point [without lock] 5271 recordTrack->mState = TrackBase::PAUSING; 5272 // do not wait for mStartStopCond if exiting 5273 if (exitPending()) { 5274 return true; 5275 } 5276 // FIXME incorrect usage of wait: no explicit predicate or loop 5277 mStartStopCond.wait(mLock); 5278 // if we have been restarted, recordTrack is in mActiveTracks here 5279 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5280 ALOGV("Record stopped OK"); 5281 return true; 5282 } 5283 return false; 5284} 5285 5286bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5287{ 5288 return false; 5289} 5290 5291status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5292{ 5293#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5294 if (!isValidSyncEvent(event)) { 5295 return BAD_VALUE; 5296 } 5297 5298 int eventSession = event->triggerSession(); 5299 status_t ret = NAME_NOT_FOUND; 5300 5301 Mutex::Autolock _l(mLock); 5302 5303 for (size_t i = 0; i < mTracks.size(); i++) { 5304 sp<RecordTrack> track = mTracks[i]; 5305 if (eventSession == track->sessionId()) { 5306 (void) track->setSyncEvent(event); 5307 ret = NO_ERROR; 5308 } 5309 } 5310 return ret; 5311#else 5312 return BAD_VALUE; 5313#endif 5314} 5315 5316// destroyTrack_l() must be called with ThreadBase::mLock held 5317void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5318{ 5319 track->terminate(); 5320 track->mState = TrackBase::STOPPED; 5321 // active tracks are removed by threadLoop() 5322 if (mActiveTracks.indexOf(track) < 0) { 5323 removeTrack_l(track); 5324 } 5325} 5326 5327void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5328{ 5329 mTracks.remove(track); 5330 // need anything related to effects here? 5331} 5332 5333void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5334{ 5335 dumpInternals(fd, args); 5336 dumpTracks(fd, args); 5337 dumpEffectChains(fd, args); 5338} 5339 5340void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5341{ 5342 fdprintf(fd, "\nInput thread %p:\n", this); 5343 5344 if (mActiveTracks.size() > 0) { 5345 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5346 } else { 5347 fdprintf(fd, " No active record clients\n"); 5348 } 5349 5350 dumpBase(fd, args); 5351} 5352 5353void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5354{ 5355 const size_t SIZE = 256; 5356 char buffer[SIZE]; 5357 String8 result; 5358 5359 size_t numtracks = mTracks.size(); 5360 size_t numactive = mActiveTracks.size(); 5361 size_t numactiveseen = 0; 5362 fdprintf(fd, " %d Tracks", numtracks); 5363 if (numtracks) { 5364 fdprintf(fd, " of which %d are active\n", numactive); 5365 RecordTrack::appendDumpHeader(result); 5366 for (size_t i = 0; i < numtracks ; ++i) { 5367 sp<RecordTrack> track = mTracks[i]; 5368 if (track != 0) { 5369 bool active = mActiveTracks.indexOf(track) >= 0; 5370 if (active) { 5371 numactiveseen++; 5372 } 5373 track->dump(buffer, SIZE, active); 5374 result.append(buffer); 5375 } 5376 } 5377 } else { 5378 fdprintf(fd, "\n"); 5379 } 5380 5381 if (numactiveseen != numactive) { 5382 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5383 " not in the track list\n"); 5384 result.append(buffer); 5385 RecordTrack::appendDumpHeader(result); 5386 for (size_t i = 0; i < numactive; ++i) { 5387 sp<RecordTrack> track = mActiveTracks[i]; 5388 if (mTracks.indexOf(track) < 0) { 5389 track->dump(buffer, SIZE, true); 5390 result.append(buffer); 5391 } 5392 } 5393 5394 } 5395 write(fd, result.string(), result.size()); 5396} 5397 5398// AudioBufferProvider interface 5399status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5400 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5401{ 5402 RecordTrack *activeTrack = mRecordTrack; 5403 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5404 if (threadBase == 0) { 5405 buffer->frameCount = 0; 5406 buffer->raw = NULL; 5407 return NOT_ENOUGH_DATA; 5408 } 5409 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5410 int32_t rear = recordThread->mRsmpInRear; 5411 int32_t front = activeTrack->mRsmpInFront; 5412 ssize_t filled = rear - front; 5413 // FIXME should not be P2 (don't want to increase latency) 5414 // FIXME if client not keeping up, discard 5415 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5416 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5417 front &= recordThread->mRsmpInFramesP2 - 1; 5418 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5419 if (part1 > (size_t) filled) { 5420 part1 = filled; 5421 } 5422 size_t ask = buffer->frameCount; 5423 ALOG_ASSERT(ask > 0); 5424 if (part1 > ask) { 5425 part1 = ask; 5426 } 5427 if (part1 == 0) { 5428 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5429 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5430 buffer->raw = NULL; 5431 buffer->frameCount = 0; 5432 activeTrack->mRsmpInUnrel = 0; 5433 return NOT_ENOUGH_DATA; 5434 } 5435 5436 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5437 buffer->frameCount = part1; 5438 activeTrack->mRsmpInUnrel = part1; 5439 return NO_ERROR; 5440} 5441 5442// AudioBufferProvider interface 5443void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5444 AudioBufferProvider::Buffer* buffer) 5445{ 5446 RecordTrack *activeTrack = mRecordTrack; 5447 size_t stepCount = buffer->frameCount; 5448 if (stepCount == 0) { 5449 return; 5450 } 5451 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5452 activeTrack->mRsmpInUnrel -= stepCount; 5453 activeTrack->mRsmpInFront += stepCount; 5454 buffer->raw = NULL; 5455 buffer->frameCount = 0; 5456} 5457 5458bool AudioFlinger::RecordThread::checkForNewParameters_l() 5459{ 5460 bool reconfig = false; 5461 5462 while (!mNewParameters.isEmpty()) { 5463 status_t status = NO_ERROR; 5464 String8 keyValuePair = mNewParameters[0]; 5465 AudioParameter param = AudioParameter(keyValuePair); 5466 int value; 5467 audio_format_t reqFormat = mFormat; 5468 uint32_t samplingRate = mSampleRate; 5469 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5470 5471 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5472 // channel count change can be requested. Do we mandate the first client defines the 5473 // HAL sampling rate and channel count or do we allow changes on the fly? 5474 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5475 samplingRate = value; 5476 reconfig = true; 5477 } 5478 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5479 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5480 status = BAD_VALUE; 5481 } else { 5482 reqFormat = (audio_format_t) value; 5483 reconfig = true; 5484 } 5485 } 5486 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5487 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5488 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5489 status = BAD_VALUE; 5490 } else { 5491 channelMask = mask; 5492 reconfig = true; 5493 } 5494 } 5495 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5496 // do not accept frame count changes if tracks are open as the track buffer 5497 // size depends on frame count and correct behavior would not be guaranteed 5498 // if frame count is changed after track creation 5499 if (mActiveTracks.size() > 0) { 5500 status = INVALID_OPERATION; 5501 } else { 5502 reconfig = true; 5503 } 5504 } 5505 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5506 // forward device change to effects that have requested to be 5507 // aware of attached audio device. 5508 for (size_t i = 0; i < mEffectChains.size(); i++) { 5509 mEffectChains[i]->setDevice_l(value); 5510 } 5511 5512 // store input device and output device but do not forward output device to audio HAL. 5513 // Note that status is ignored by the caller for output device 5514 // (see AudioFlinger::setParameters() 5515 if (audio_is_output_devices(value)) { 5516 mOutDevice = value; 5517 status = BAD_VALUE; 5518 } else { 5519 mInDevice = value; 5520 // disable AEC and NS if the device is a BT SCO headset supporting those 5521 // pre processings 5522 if (mTracks.size() > 0) { 5523 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5524 mAudioFlinger->btNrecIsOff(); 5525 for (size_t i = 0; i < mTracks.size(); i++) { 5526 sp<RecordTrack> track = mTracks[i]; 5527 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5528 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5529 } 5530 } 5531 } 5532 } 5533 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5534 mAudioSource != (audio_source_t)value) { 5535 // forward device change to effects that have requested to be 5536 // aware of attached audio device. 5537 for (size_t i = 0; i < mEffectChains.size(); i++) { 5538 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5539 } 5540 mAudioSource = (audio_source_t)value; 5541 } 5542 5543 if (status == NO_ERROR) { 5544 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5545 keyValuePair.string()); 5546 if (status == INVALID_OPERATION) { 5547 inputStandBy(); 5548 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5549 keyValuePair.string()); 5550 } 5551 if (reconfig) { 5552 if (status == BAD_VALUE && 5553 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5554 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5555 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5556 <= (2 * samplingRate)) && 5557 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5558 <= FCC_2 && 5559 (channelMask == AUDIO_CHANNEL_IN_MONO || 5560 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5561 status = NO_ERROR; 5562 } 5563 if (status == NO_ERROR) { 5564 readInputParameters_l(); 5565 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5566 } 5567 } 5568 } 5569 5570 mNewParameters.removeAt(0); 5571 5572 mParamStatus = status; 5573 mParamCond.signal(); 5574 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5575 // already timed out waiting for the status and will never signal the condition. 5576 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5577 } 5578 return reconfig; 5579} 5580 5581String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5582{ 5583 Mutex::Autolock _l(mLock); 5584 if (initCheck() != NO_ERROR) { 5585 return String8(); 5586 } 5587 5588 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5589 const String8 out_s8(s); 5590 free(s); 5591 return out_s8; 5592} 5593 5594void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5595 AudioSystem::OutputDescriptor desc; 5596 const void *param2 = NULL; 5597 5598 switch (event) { 5599 case AudioSystem::INPUT_OPENED: 5600 case AudioSystem::INPUT_CONFIG_CHANGED: 5601 desc.channelMask = mChannelMask; 5602 desc.samplingRate = mSampleRate; 5603 desc.format = mFormat; 5604 desc.frameCount = mFrameCount; 5605 desc.latency = 0; 5606 param2 = &desc; 5607 break; 5608 5609 case AudioSystem::INPUT_CLOSED: 5610 default: 5611 break; 5612 } 5613 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5614} 5615 5616void AudioFlinger::RecordThread::readInputParameters_l() 5617{ 5618 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5619 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5620 mChannelCount = popcount(mChannelMask); 5621 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5622 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5623 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5624 } 5625 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5626 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5627 mFrameCount = mBufferSize / mFrameSize; 5628 // This is the formula for calculating the temporary buffer size. 5629 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5630 // 1 full output buffer, regardless of the alignment of the available input. 5631 // The value is somewhat arbitrary, and could probably be even larger. 5632 // A larger value should allow more old data to be read after a track calls start(), 5633 // without increasing latency. 5634 mRsmpInFrames = mFrameCount * 7; 5635 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5636 delete[] mRsmpInBuffer; 5637 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5638 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5639 5640 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5641 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5642} 5643 5644uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5645{ 5646 Mutex::Autolock _l(mLock); 5647 if (initCheck() != NO_ERROR) { 5648 return 0; 5649 } 5650 5651 return mInput->stream->get_input_frames_lost(mInput->stream); 5652} 5653 5654uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5655{ 5656 Mutex::Autolock _l(mLock); 5657 uint32_t result = 0; 5658 if (getEffectChain_l(sessionId) != 0) { 5659 result = EFFECT_SESSION; 5660 } 5661 5662 for (size_t i = 0; i < mTracks.size(); ++i) { 5663 if (sessionId == mTracks[i]->sessionId()) { 5664 result |= TRACK_SESSION; 5665 break; 5666 } 5667 } 5668 5669 return result; 5670} 5671 5672KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5673{ 5674 KeyedVector<int, bool> ids; 5675 Mutex::Autolock _l(mLock); 5676 for (size_t j = 0; j < mTracks.size(); ++j) { 5677 sp<RecordThread::RecordTrack> track = mTracks[j]; 5678 int sessionId = track->sessionId(); 5679 if (ids.indexOfKey(sessionId) < 0) { 5680 ids.add(sessionId, true); 5681 } 5682 } 5683 return ids; 5684} 5685 5686AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5687{ 5688 Mutex::Autolock _l(mLock); 5689 AudioStreamIn *input = mInput; 5690 mInput = NULL; 5691 return input; 5692} 5693 5694// this method must always be called either with ThreadBase mLock held or inside the thread loop 5695audio_stream_t* AudioFlinger::RecordThread::stream() const 5696{ 5697 if (mInput == NULL) { 5698 return NULL; 5699 } 5700 return &mInput->stream->common; 5701} 5702 5703status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5704{ 5705 // only one chain per input thread 5706 if (mEffectChains.size() != 0) { 5707 return INVALID_OPERATION; 5708 } 5709 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5710 5711 chain->setInBuffer(NULL); 5712 chain->setOutBuffer(NULL); 5713 5714 checkSuspendOnAddEffectChain_l(chain); 5715 5716 mEffectChains.add(chain); 5717 5718 return NO_ERROR; 5719} 5720 5721size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5722{ 5723 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5724 ALOGW_IF(mEffectChains.size() != 1, 5725 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5726 chain.get(), mEffectChains.size(), this); 5727 if (mEffectChains.size() == 1) { 5728 mEffectChains.removeAt(0); 5729 } 5730 return 0; 5731} 5732 5733}; // namespace android 5734