Threads.cpp revision 98ef978df4e928f486d244c4d7f7ad9f13111e98
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 delete[] 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 delete[] mSinkBuffer; 1786 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1787 // For historical reasons mSinkBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1788 mSinkBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1789 memset(mSinkBuffer, 0, normalBufferSize); 1790 1791 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1792 // drives the output. 1793 free(mMixerBuffer); 1794 mMixerBuffer = NULL; 1795 if (mMixerBufferEnabled) { 1796 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1797 mMixerBufferSize = mNormalFrameCount * mChannelCount 1798 * audio_bytes_per_sample(mMixerBufferFormat); 1799 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1800 } 1801 free(mEffectBuffer); 1802 mEffectBuffer = NULL; 1803 if (mEffectBufferEnabled) { 1804 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1805 mEffectBufferSize = mNormalFrameCount * mChannelCount 1806 * audio_bytes_per_sample(mEffectBufferFormat); 1807 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1808 } 1809 1810 // force reconfiguration of effect chains and engines to take new buffer size and audio 1811 // parameters into account 1812 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1813 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1814 // matter. 1815 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1816 Vector< sp<EffectChain> > effectChains = mEffectChains; 1817 for (size_t i = 0; i < effectChains.size(); i ++) { 1818 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1819 } 1820} 1821 1822 1823status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1824{ 1825 if (halFrames == NULL || dspFrames == NULL) { 1826 return BAD_VALUE; 1827 } 1828 Mutex::Autolock _l(mLock); 1829 if (initCheck() != NO_ERROR) { 1830 return INVALID_OPERATION; 1831 } 1832 size_t framesWritten = mBytesWritten / mFrameSize; 1833 *halFrames = framesWritten; 1834 1835 if (isSuspended()) { 1836 // return an estimation of rendered frames when the output is suspended 1837 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1838 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1839 return NO_ERROR; 1840 } else { 1841 status_t status; 1842 uint32_t frames; 1843 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1844 *dspFrames = (size_t)frames; 1845 return status; 1846 } 1847} 1848 1849uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1850{ 1851 Mutex::Autolock _l(mLock); 1852 uint32_t result = 0; 1853 if (getEffectChain_l(sessionId) != 0) { 1854 result = EFFECT_SESSION; 1855 } 1856 1857 for (size_t i = 0; i < mTracks.size(); ++i) { 1858 sp<Track> track = mTracks[i]; 1859 if (sessionId == track->sessionId() && !track->isInvalid()) { 1860 result |= TRACK_SESSION; 1861 break; 1862 } 1863 } 1864 1865 return result; 1866} 1867 1868uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1869{ 1870 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1871 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1872 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1873 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1874 } 1875 for (size_t i = 0; i < mTracks.size(); i++) { 1876 sp<Track> track = mTracks[i]; 1877 if (sessionId == track->sessionId() && !track->isInvalid()) { 1878 return AudioSystem::getStrategyForStream(track->streamType()); 1879 } 1880 } 1881 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1882} 1883 1884 1885AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1886{ 1887 Mutex::Autolock _l(mLock); 1888 return mOutput; 1889} 1890 1891AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1892{ 1893 Mutex::Autolock _l(mLock); 1894 AudioStreamOut *output = mOutput; 1895 mOutput = NULL; 1896 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1897 // must push a NULL and wait for ack 1898 mOutputSink.clear(); 1899 mPipeSink.clear(); 1900 mNormalSink.clear(); 1901 return output; 1902} 1903 1904// this method must always be called either with ThreadBase mLock held or inside the thread loop 1905audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1906{ 1907 if (mOutput == NULL) { 1908 return NULL; 1909 } 1910 return &mOutput->stream->common; 1911} 1912 1913uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1914{ 1915 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1916} 1917 1918status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1919{ 1920 if (!isValidSyncEvent(event)) { 1921 return BAD_VALUE; 1922 } 1923 1924 Mutex::Autolock _l(mLock); 1925 1926 for (size_t i = 0; i < mTracks.size(); ++i) { 1927 sp<Track> track = mTracks[i]; 1928 if (event->triggerSession() == track->sessionId()) { 1929 (void) track->setSyncEvent(event); 1930 return NO_ERROR; 1931 } 1932 } 1933 1934 return NAME_NOT_FOUND; 1935} 1936 1937bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1938{ 1939 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1940} 1941 1942void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1943 const Vector< sp<Track> >& tracksToRemove) 1944{ 1945 size_t count = tracksToRemove.size(); 1946 if (count > 0) { 1947 for (size_t i = 0 ; i < count ; i++) { 1948 const sp<Track>& track = tracksToRemove.itemAt(i); 1949 if (!track->isOutputTrack()) { 1950 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1951#ifdef ADD_BATTERY_DATA 1952 // to track the speaker usage 1953 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1954#endif 1955 if (track->isTerminated()) { 1956 AudioSystem::releaseOutput(mId); 1957 } 1958 } 1959 } 1960 } 1961} 1962 1963void AudioFlinger::PlaybackThread::checkSilentMode_l() 1964{ 1965 if (!mMasterMute) { 1966 char value[PROPERTY_VALUE_MAX]; 1967 if (property_get("ro.audio.silent", value, "0") > 0) { 1968 char *endptr; 1969 unsigned long ul = strtoul(value, &endptr, 0); 1970 if (*endptr == '\0' && ul != 0) { 1971 ALOGD("Silence is golden"); 1972 // The setprop command will not allow a property to be changed after 1973 // the first time it is set, so we don't have to worry about un-muting. 1974 setMasterMute_l(true); 1975 } 1976 } 1977 } 1978} 1979 1980// shared by MIXER and DIRECT, overridden by DUPLICATING 1981ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1982{ 1983 // FIXME rewrite to reduce number of system calls 1984 mLastWriteTime = systemTime(); 1985 mInWrite = true; 1986 ssize_t bytesWritten; 1987 1988 // If an NBAIO sink is present, use it to write the normal mixer's submix 1989 if (mNormalSink != 0) { 1990#define mBitShift 2 // FIXME 1991 size_t count = mBytesRemaining >> mBitShift; 1992 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1993 ATRACE_BEGIN("write"); 1994 // update the setpoint when AudioFlinger::mScreenState changes 1995 uint32_t screenState = AudioFlinger::mScreenState; 1996 if (screenState != mScreenState) { 1997 mScreenState = screenState; 1998 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1999 if (pipe != NULL) { 2000 pipe->setAvgFrames((mScreenState & 1) ? 2001 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2002 } 2003 } 2004 ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count); 2005 ATRACE_END(); 2006 if (framesWritten > 0) { 2007 bytesWritten = framesWritten << mBitShift; 2008 } else { 2009 bytesWritten = framesWritten; 2010 } 2011 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2012 if (status == NO_ERROR) { 2013 size_t totalFramesWritten = mNormalSink->framesWritten(); 2014 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2015 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2016 mLatchDValid = true; 2017 } 2018 } 2019 // otherwise use the HAL / AudioStreamOut directly 2020 } else { 2021 // Direct output and offload threads 2022 size_t offset = (mCurrentWriteLength - mBytesRemaining); 2023 if (mUseAsyncWrite) { 2024 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2025 mWriteAckSequence += 2; 2026 mWriteAckSequence |= 1; 2027 ALOG_ASSERT(mCallbackThread != 0); 2028 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2029 } 2030 // FIXME We should have an implementation of timestamps for direct output threads. 2031 // They are used e.g for multichannel PCM playback over HDMI. 2032 bytesWritten = mOutput->stream->write(mOutput->stream, 2033 (char *)mSinkBuffer + offset, mBytesRemaining); 2034 if (mUseAsyncWrite && 2035 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2036 // do not wait for async callback in case of error of full write 2037 mWriteAckSequence &= ~1; 2038 ALOG_ASSERT(mCallbackThread != 0); 2039 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2040 } 2041 } 2042 2043 mNumWrites++; 2044 mInWrite = false; 2045 mStandby = false; 2046 return bytesWritten; 2047} 2048 2049void AudioFlinger::PlaybackThread::threadLoop_drain() 2050{ 2051 if (mOutput->stream->drain) { 2052 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2053 if (mUseAsyncWrite) { 2054 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2055 mDrainSequence |= 1; 2056 ALOG_ASSERT(mCallbackThread != 0); 2057 mCallbackThread->setDraining(mDrainSequence); 2058 } 2059 mOutput->stream->drain(mOutput->stream, 2060 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2061 : AUDIO_DRAIN_ALL); 2062 } 2063} 2064 2065void AudioFlinger::PlaybackThread::threadLoop_exit() 2066{ 2067 // Default implementation has nothing to do 2068} 2069 2070/* 2071The derived values that are cached: 2072 - mSinkBufferSize from frame count * frame size 2073 - activeSleepTime from activeSleepTimeUs() 2074 - idleSleepTime from idleSleepTimeUs() 2075 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2076 - maxPeriod from frame count and sample rate (MIXER only) 2077 2078The parameters that affect these derived values are: 2079 - frame count 2080 - frame size 2081 - sample rate 2082 - device type: A2DP or not 2083 - device latency 2084 - format: PCM or not 2085 - active sleep time 2086 - idle sleep time 2087*/ 2088 2089void AudioFlinger::PlaybackThread::cacheParameters_l() 2090{ 2091 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2092 activeSleepTime = activeSleepTimeUs(); 2093 idleSleepTime = idleSleepTimeUs(); 2094} 2095 2096void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2097{ 2098 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2099 this, streamType, mTracks.size()); 2100 Mutex::Autolock _l(mLock); 2101 2102 size_t size = mTracks.size(); 2103 for (size_t i = 0; i < size; i++) { 2104 sp<Track> t = mTracks[i]; 2105 if (t->streamType() == streamType) { 2106 t->invalidate(); 2107 } 2108 } 2109} 2110 2111status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2112{ 2113 int session = chain->sessionId(); 2114 int16_t *buffer = mEffectBufferEnabled 2115 ? reinterpret_cast<int16_t*>(mEffectBuffer) : mSinkBuffer; 2116 bool ownsBuffer = false; 2117 2118 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2119 if (session > 0) { 2120 // Only one effect chain can be present in direct output thread and it uses 2121 // the sink buffer as input 2122 if (mType != DIRECT) { 2123 size_t numSamples = mNormalFrameCount * mChannelCount; 2124 buffer = new int16_t[numSamples]; 2125 memset(buffer, 0, numSamples * sizeof(int16_t)); 2126 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2127 ownsBuffer = true; 2128 } 2129 2130 // Attach all tracks with same session ID to this chain. 2131 for (size_t i = 0; i < mTracks.size(); ++i) { 2132 sp<Track> track = mTracks[i]; 2133 if (session == track->sessionId()) { 2134 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2135 buffer); 2136 track->setMainBuffer(buffer); 2137 chain->incTrackCnt(); 2138 } 2139 } 2140 2141 // indicate all active tracks in the chain 2142 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2143 sp<Track> track = mActiveTracks[i].promote(); 2144 if (track == 0) { 2145 continue; 2146 } 2147 if (session == track->sessionId()) { 2148 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2149 chain->incActiveTrackCnt(); 2150 } 2151 } 2152 } 2153 2154 chain->setInBuffer(buffer, ownsBuffer); 2155 chain->setOutBuffer(mEffectBufferEnabled 2156 ? reinterpret_cast<int16_t*>(mEffectBuffer) : mSinkBuffer); 2157 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2158 // chains list in order to be processed last as it contains output stage effects 2159 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2160 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2161 // after track specific effects and before output stage 2162 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2163 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2164 // Effect chain for other sessions are inserted at beginning of effect 2165 // chains list to be processed before output mix effects. Relative order between other 2166 // sessions is not important 2167 size_t size = mEffectChains.size(); 2168 size_t i = 0; 2169 for (i = 0; i < size; i++) { 2170 if (mEffectChains[i]->sessionId() < session) { 2171 break; 2172 } 2173 } 2174 mEffectChains.insertAt(chain, i); 2175 checkSuspendOnAddEffectChain_l(chain); 2176 2177 return NO_ERROR; 2178} 2179 2180size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2181{ 2182 int session = chain->sessionId(); 2183 2184 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2185 2186 for (size_t i = 0; i < mEffectChains.size(); i++) { 2187 if (chain == mEffectChains[i]) { 2188 mEffectChains.removeAt(i); 2189 // detach all active tracks from the chain 2190 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2191 sp<Track> track = mActiveTracks[i].promote(); 2192 if (track == 0) { 2193 continue; 2194 } 2195 if (session == track->sessionId()) { 2196 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2197 chain.get(), session); 2198 chain->decActiveTrackCnt(); 2199 } 2200 } 2201 2202 // detach all tracks with same session ID from this chain 2203 for (size_t i = 0; i < mTracks.size(); ++i) { 2204 sp<Track> track = mTracks[i]; 2205 if (session == track->sessionId()) { 2206 track->setMainBuffer(mSinkBuffer); 2207 chain->decTrackCnt(); 2208 } 2209 } 2210 break; 2211 } 2212 } 2213 return mEffectChains.size(); 2214} 2215 2216status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2217 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2218{ 2219 Mutex::Autolock _l(mLock); 2220 return attachAuxEffect_l(track, EffectId); 2221} 2222 2223status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2224 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2225{ 2226 status_t status = NO_ERROR; 2227 2228 if (EffectId == 0) { 2229 track->setAuxBuffer(0, NULL); 2230 } else { 2231 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2232 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2233 if (effect != 0) { 2234 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2235 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2236 } else { 2237 status = INVALID_OPERATION; 2238 } 2239 } else { 2240 status = BAD_VALUE; 2241 } 2242 } 2243 return status; 2244} 2245 2246void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2247{ 2248 for (size_t i = 0; i < mTracks.size(); ++i) { 2249 sp<Track> track = mTracks[i]; 2250 if (track->auxEffectId() == effectId) { 2251 attachAuxEffect_l(track, 0); 2252 } 2253 } 2254} 2255 2256bool AudioFlinger::PlaybackThread::threadLoop() 2257{ 2258 Vector< sp<Track> > tracksToRemove; 2259 2260 standbyTime = systemTime(); 2261 2262 // MIXER 2263 nsecs_t lastWarning = 0; 2264 2265 // DUPLICATING 2266 // FIXME could this be made local to while loop? 2267 writeFrames = 0; 2268 2269 int lastGeneration = 0; 2270 2271 cacheParameters_l(); 2272 sleepTime = idleSleepTime; 2273 2274 if (mType == MIXER) { 2275 sleepTimeShift = 0; 2276 } 2277 2278 CpuStats cpuStats; 2279 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2280 2281 acquireWakeLock(); 2282 2283 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2284 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2285 // and then that string will be logged at the next convenient opportunity. 2286 const char *logString = NULL; 2287 2288 checkSilentMode_l(); 2289 2290 while (!exitPending()) 2291 { 2292 cpuStats.sample(myName); 2293 2294 Vector< sp<EffectChain> > effectChains; 2295 2296 processConfigEvents(); 2297 2298 { // scope for mLock 2299 2300 Mutex::Autolock _l(mLock); 2301 2302 if (logString != NULL) { 2303 mNBLogWriter->logTimestamp(); 2304 mNBLogWriter->log(logString); 2305 logString = NULL; 2306 } 2307 2308 if (mLatchDValid) { 2309 mLatchQ = mLatchD; 2310 mLatchDValid = false; 2311 mLatchQValid = true; 2312 } 2313 2314 if (checkForNewParameters_l()) { 2315 cacheParameters_l(); 2316 } 2317 2318 saveOutputTracks(); 2319 if (mSignalPending) { 2320 // A signal was raised while we were unlocked 2321 mSignalPending = false; 2322 } else if (waitingAsyncCallback_l()) { 2323 if (exitPending()) { 2324 break; 2325 } 2326 releaseWakeLock_l(); 2327 mWakeLockUids.clear(); 2328 mActiveTracksGeneration++; 2329 ALOGV("wait async completion"); 2330 mWaitWorkCV.wait(mLock); 2331 ALOGV("async completion/wake"); 2332 acquireWakeLock_l(); 2333 standbyTime = systemTime() + standbyDelay; 2334 sleepTime = 0; 2335 2336 continue; 2337 } 2338 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2339 isSuspended()) { 2340 // put audio hardware into standby after short delay 2341 if (shouldStandby_l()) { 2342 2343 threadLoop_standby(); 2344 2345 mStandby = true; 2346 } 2347 2348 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2349 // we're about to wait, flush the binder command buffer 2350 IPCThreadState::self()->flushCommands(); 2351 2352 clearOutputTracks(); 2353 2354 if (exitPending()) { 2355 break; 2356 } 2357 2358 releaseWakeLock_l(); 2359 mWakeLockUids.clear(); 2360 mActiveTracksGeneration++; 2361 // wait until we have something to do... 2362 ALOGV("%s going to sleep", myName.string()); 2363 mWaitWorkCV.wait(mLock); 2364 ALOGV("%s waking up", myName.string()); 2365 acquireWakeLock_l(); 2366 2367 mMixerStatus = MIXER_IDLE; 2368 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2369 mBytesWritten = 0; 2370 mBytesRemaining = 0; 2371 checkSilentMode_l(); 2372 2373 standbyTime = systemTime() + standbyDelay; 2374 sleepTime = idleSleepTime; 2375 if (mType == MIXER) { 2376 sleepTimeShift = 0; 2377 } 2378 2379 continue; 2380 } 2381 } 2382 // mMixerStatusIgnoringFastTracks is also updated internally 2383 mMixerStatus = prepareTracks_l(&tracksToRemove); 2384 2385 // compare with previously applied list 2386 if (lastGeneration != mActiveTracksGeneration) { 2387 // update wakelock 2388 updateWakeLockUids_l(mWakeLockUids); 2389 lastGeneration = mActiveTracksGeneration; 2390 } 2391 2392 // prevent any changes in effect chain list and in each effect chain 2393 // during mixing and effect process as the audio buffers could be deleted 2394 // or modified if an effect is created or deleted 2395 lockEffectChains_l(effectChains); 2396 } // mLock scope ends 2397 2398 if (mBytesRemaining == 0) { 2399 mCurrentWriteLength = 0; 2400 if (mMixerStatus == MIXER_TRACKS_READY) { 2401 // threadLoop_mix() sets mCurrentWriteLength 2402 threadLoop_mix(); 2403 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2404 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2405 // threadLoop_sleepTime sets sleepTime to 0 if data 2406 // must be written to HAL 2407 threadLoop_sleepTime(); 2408 if (sleepTime == 0) { 2409 mCurrentWriteLength = mSinkBufferSize; 2410 } 2411 } 2412 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2413 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2414 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2415 // or mSinkBuffer (if there are no effects). 2416 // 2417 // This is done pre-effects computation; if effects change to 2418 // support higher precision, this needs to move. 2419 // 2420 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2421 // TODO use sleepTime == 0 as an additional condition. 2422 if (mMixerBufferValid) { 2423 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2424 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2425 2426 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2427 mNormalFrameCount * mChannelCount); 2428 } 2429 2430 mBytesRemaining = mCurrentWriteLength; 2431 if (isSuspended()) { 2432 sleepTime = suspendSleepTimeUs(); 2433 // simulate write to HAL when suspended 2434 mBytesWritten += mSinkBufferSize; 2435 mBytesRemaining = 0; 2436 } 2437 2438 // only process effects if we're going to write 2439 if (sleepTime == 0 && mType != OFFLOAD) { 2440 for (size_t i = 0; i < effectChains.size(); i ++) { 2441 effectChains[i]->process_l(); 2442 } 2443 } 2444 } 2445 // Process effect chains for offloaded thread even if no audio 2446 // was read from audio track: process only updates effect state 2447 // and thus does have to be synchronized with audio writes but may have 2448 // to be called while waiting for async write callback 2449 if (mType == OFFLOAD) { 2450 for (size_t i = 0; i < effectChains.size(); i ++) { 2451 effectChains[i]->process_l(); 2452 } 2453 } 2454 2455 // Only if the Effects buffer is enabled and there is data in the 2456 // Effects buffer (buffer valid), we need to 2457 // copy into the sink buffer. 2458 // TODO use sleepTime == 0 as an additional condition. 2459 if (mEffectBufferValid) { 2460 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2461 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2462 mNormalFrameCount * mChannelCount); 2463 } 2464 2465 // enable changes in effect chain 2466 unlockEffectChains(effectChains); 2467 2468 if (!waitingAsyncCallback()) { 2469 // sleepTime == 0 means we must write to audio hardware 2470 if (sleepTime == 0) { 2471 if (mBytesRemaining) { 2472 ssize_t ret = threadLoop_write(); 2473 if (ret < 0) { 2474 mBytesRemaining = 0; 2475 } else { 2476 mBytesWritten += ret; 2477 mBytesRemaining -= ret; 2478 } 2479 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2480 (mMixerStatus == MIXER_DRAIN_ALL)) { 2481 threadLoop_drain(); 2482 } 2483 if (mType == MIXER) { 2484 // write blocked detection 2485 nsecs_t now = systemTime(); 2486 nsecs_t delta = now - mLastWriteTime; 2487 if (!mStandby && delta > maxPeriod) { 2488 mNumDelayedWrites++; 2489 if ((now - lastWarning) > kWarningThrottleNs) { 2490 ATRACE_NAME("underrun"); 2491 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2492 ns2ms(delta), mNumDelayedWrites, this); 2493 lastWarning = now; 2494 } 2495 } 2496 } 2497 2498 } else { 2499 usleep(sleepTime); 2500 } 2501 } 2502 2503 // Finally let go of removed track(s), without the lock held 2504 // since we can't guarantee the destructors won't acquire that 2505 // same lock. This will also mutate and push a new fast mixer state. 2506 threadLoop_removeTracks(tracksToRemove); 2507 tracksToRemove.clear(); 2508 2509 // FIXME I don't understand the need for this here; 2510 // it was in the original code but maybe the 2511 // assignment in saveOutputTracks() makes this unnecessary? 2512 clearOutputTracks(); 2513 2514 // Effect chains will be actually deleted here if they were removed from 2515 // mEffectChains list during mixing or effects processing 2516 effectChains.clear(); 2517 2518 // FIXME Note that the above .clear() is no longer necessary since effectChains 2519 // is now local to this block, but will keep it for now (at least until merge done). 2520 } 2521 2522 threadLoop_exit(); 2523 2524 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2525 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2526 // put output stream into standby mode 2527 if (!mStandby) { 2528 mOutput->stream->common.standby(&mOutput->stream->common); 2529 } 2530 } 2531 2532 releaseWakeLock(); 2533 mWakeLockUids.clear(); 2534 mActiveTracksGeneration++; 2535 2536 ALOGV("Thread %p type %d exiting", this, mType); 2537 return false; 2538} 2539 2540// removeTracks_l() must be called with ThreadBase::mLock held 2541void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2542{ 2543 size_t count = tracksToRemove.size(); 2544 if (count > 0) { 2545 for (size_t i=0 ; i<count ; i++) { 2546 const sp<Track>& track = tracksToRemove.itemAt(i); 2547 mActiveTracks.remove(track); 2548 mWakeLockUids.remove(track->uid()); 2549 mActiveTracksGeneration++; 2550 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2551 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2552 if (chain != 0) { 2553 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2554 track->sessionId()); 2555 chain->decActiveTrackCnt(); 2556 } 2557 if (track->isTerminated()) { 2558 removeTrack_l(track); 2559 } 2560 } 2561 } 2562 2563} 2564 2565status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2566{ 2567 if (mNormalSink != 0) { 2568 return mNormalSink->getTimestamp(timestamp); 2569 } 2570 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2571 uint64_t position64; 2572 int ret = mOutput->stream->get_presentation_position( 2573 mOutput->stream, &position64, ×tamp.mTime); 2574 if (ret == 0) { 2575 timestamp.mPosition = (uint32_t)position64; 2576 return NO_ERROR; 2577 } 2578 } 2579 return INVALID_OPERATION; 2580} 2581// ---------------------------------------------------------------------------- 2582 2583AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2584 audio_io_handle_t id, audio_devices_t device, type_t type) 2585 : PlaybackThread(audioFlinger, output, id, device, type), 2586 // mAudioMixer below 2587 // mFastMixer below 2588 mFastMixerFutex(0) 2589 // mOutputSink below 2590 // mPipeSink below 2591 // mNormalSink below 2592{ 2593 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2594 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2595 "mFrameCount=%d, mNormalFrameCount=%d", 2596 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2597 mNormalFrameCount); 2598 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2599 2600 // FIXME - Current mixer implementation only supports stereo output 2601 if (mChannelCount != FCC_2) { 2602 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2603 } 2604 2605 // create an NBAIO sink for the HAL output stream, and negotiate 2606 mOutputSink = new AudioStreamOutSink(output->stream); 2607 size_t numCounterOffers = 0; 2608 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2609 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2610 ALOG_ASSERT(index == 0); 2611 2612 // initialize fast mixer depending on configuration 2613 bool initFastMixer; 2614 switch (kUseFastMixer) { 2615 case FastMixer_Never: 2616 initFastMixer = false; 2617 break; 2618 case FastMixer_Always: 2619 initFastMixer = true; 2620 break; 2621 case FastMixer_Static: 2622 case FastMixer_Dynamic: 2623 initFastMixer = mFrameCount < mNormalFrameCount; 2624 break; 2625 } 2626 if (initFastMixer) { 2627 2628 // create a MonoPipe to connect our submix to FastMixer 2629 NBAIO_Format format = mOutputSink->format(); 2630 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2631 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2632 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2633 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2634 const NBAIO_Format offers[1] = {format}; 2635 size_t numCounterOffers = 0; 2636 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2637 ALOG_ASSERT(index == 0); 2638 monoPipe->setAvgFrames((mScreenState & 1) ? 2639 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2640 mPipeSink = monoPipe; 2641 2642#ifdef TEE_SINK 2643 if (mTeeSinkOutputEnabled) { 2644 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2645 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2646 numCounterOffers = 0; 2647 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2648 ALOG_ASSERT(index == 0); 2649 mTeeSink = teeSink; 2650 PipeReader *teeSource = new PipeReader(*teeSink); 2651 numCounterOffers = 0; 2652 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2653 ALOG_ASSERT(index == 0); 2654 mTeeSource = teeSource; 2655 } 2656#endif 2657 2658 // create fast mixer and configure it initially with just one fast track for our submix 2659 mFastMixer = new FastMixer(); 2660 FastMixerStateQueue *sq = mFastMixer->sq(); 2661#ifdef STATE_QUEUE_DUMP 2662 sq->setObserverDump(&mStateQueueObserverDump); 2663 sq->setMutatorDump(&mStateQueueMutatorDump); 2664#endif 2665 FastMixerState *state = sq->begin(); 2666 FastTrack *fastTrack = &state->mFastTracks[0]; 2667 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2668 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2669 fastTrack->mVolumeProvider = NULL; 2670 fastTrack->mGeneration++; 2671 state->mFastTracksGen++; 2672 state->mTrackMask = 1; 2673 // fast mixer will use the HAL output sink 2674 state->mOutputSink = mOutputSink.get(); 2675 state->mOutputSinkGen++; 2676 state->mFrameCount = mFrameCount; 2677 state->mCommand = FastMixerState::COLD_IDLE; 2678 // already done in constructor initialization list 2679 //mFastMixerFutex = 0; 2680 state->mColdFutexAddr = &mFastMixerFutex; 2681 state->mColdGen++; 2682 state->mDumpState = &mFastMixerDumpState; 2683#ifdef TEE_SINK 2684 state->mTeeSink = mTeeSink.get(); 2685#endif 2686 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2687 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2688 sq->end(); 2689 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2690 2691 // start the fast mixer 2692 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2693 pid_t tid = mFastMixer->getTid(); 2694 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2695 if (err != 0) { 2696 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2697 kPriorityFastMixer, getpid_cached, tid, err); 2698 } 2699 2700#ifdef AUDIO_WATCHDOG 2701 // create and start the watchdog 2702 mAudioWatchdog = new AudioWatchdog(); 2703 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2704 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2705 tid = mAudioWatchdog->getTid(); 2706 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2707 if (err != 0) { 2708 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2709 kPriorityFastMixer, getpid_cached, tid, err); 2710 } 2711#endif 2712 2713 } else { 2714 mFastMixer = NULL; 2715 } 2716 2717 switch (kUseFastMixer) { 2718 case FastMixer_Never: 2719 case FastMixer_Dynamic: 2720 mNormalSink = mOutputSink; 2721 break; 2722 case FastMixer_Always: 2723 mNormalSink = mPipeSink; 2724 break; 2725 case FastMixer_Static: 2726 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2727 break; 2728 } 2729} 2730 2731AudioFlinger::MixerThread::~MixerThread() 2732{ 2733 if (mFastMixer != NULL) { 2734 FastMixerStateQueue *sq = mFastMixer->sq(); 2735 FastMixerState *state = sq->begin(); 2736 if (state->mCommand == FastMixerState::COLD_IDLE) { 2737 int32_t old = android_atomic_inc(&mFastMixerFutex); 2738 if (old == -1) { 2739 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2740 } 2741 } 2742 state->mCommand = FastMixerState::EXIT; 2743 sq->end(); 2744 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2745 mFastMixer->join(); 2746 // Though the fast mixer thread has exited, it's state queue is still valid. 2747 // We'll use that extract the final state which contains one remaining fast track 2748 // corresponding to our sub-mix. 2749 state = sq->begin(); 2750 ALOG_ASSERT(state->mTrackMask == 1); 2751 FastTrack *fastTrack = &state->mFastTracks[0]; 2752 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2753 delete fastTrack->mBufferProvider; 2754 sq->end(false /*didModify*/); 2755 delete mFastMixer; 2756#ifdef AUDIO_WATCHDOG 2757 if (mAudioWatchdog != 0) { 2758 mAudioWatchdog->requestExit(); 2759 mAudioWatchdog->requestExitAndWait(); 2760 mAudioWatchdog.clear(); 2761 } 2762#endif 2763 } 2764 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2765 delete mAudioMixer; 2766} 2767 2768 2769uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2770{ 2771 if (mFastMixer != NULL) { 2772 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2773 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2774 } 2775 return latency; 2776} 2777 2778 2779void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2780{ 2781 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2782} 2783 2784ssize_t AudioFlinger::MixerThread::threadLoop_write() 2785{ 2786 // FIXME we should only do one push per cycle; confirm this is true 2787 // Start the fast mixer if it's not already running 2788 if (mFastMixer != NULL) { 2789 FastMixerStateQueue *sq = mFastMixer->sq(); 2790 FastMixerState *state = sq->begin(); 2791 if (state->mCommand != FastMixerState::MIX_WRITE && 2792 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2793 if (state->mCommand == FastMixerState::COLD_IDLE) { 2794 int32_t old = android_atomic_inc(&mFastMixerFutex); 2795 if (old == -1) { 2796 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2797 } 2798#ifdef AUDIO_WATCHDOG 2799 if (mAudioWatchdog != 0) { 2800 mAudioWatchdog->resume(); 2801 } 2802#endif 2803 } 2804 state->mCommand = FastMixerState::MIX_WRITE; 2805 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2806 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2807 sq->end(); 2808 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2809 if (kUseFastMixer == FastMixer_Dynamic) { 2810 mNormalSink = mPipeSink; 2811 } 2812 } else { 2813 sq->end(false /*didModify*/); 2814 } 2815 } 2816 return PlaybackThread::threadLoop_write(); 2817} 2818 2819void AudioFlinger::MixerThread::threadLoop_standby() 2820{ 2821 // Idle the fast mixer if it's currently running 2822 if (mFastMixer != NULL) { 2823 FastMixerStateQueue *sq = mFastMixer->sq(); 2824 FastMixerState *state = sq->begin(); 2825 if (!(state->mCommand & FastMixerState::IDLE)) { 2826 state->mCommand = FastMixerState::COLD_IDLE; 2827 state->mColdFutexAddr = &mFastMixerFutex; 2828 state->mColdGen++; 2829 mFastMixerFutex = 0; 2830 sq->end(); 2831 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2832 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2833 if (kUseFastMixer == FastMixer_Dynamic) { 2834 mNormalSink = mOutputSink; 2835 } 2836#ifdef AUDIO_WATCHDOG 2837 if (mAudioWatchdog != 0) { 2838 mAudioWatchdog->pause(); 2839 } 2840#endif 2841 } else { 2842 sq->end(false /*didModify*/); 2843 } 2844 } 2845 PlaybackThread::threadLoop_standby(); 2846} 2847 2848bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2849{ 2850 return false; 2851} 2852 2853bool AudioFlinger::PlaybackThread::shouldStandby_l() 2854{ 2855 return !mStandby; 2856} 2857 2858bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2859{ 2860 Mutex::Autolock _l(mLock); 2861 return waitingAsyncCallback_l(); 2862} 2863 2864// shared by MIXER and DIRECT, overridden by DUPLICATING 2865void AudioFlinger::PlaybackThread::threadLoop_standby() 2866{ 2867 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2868 mOutput->stream->common.standby(&mOutput->stream->common); 2869 if (mUseAsyncWrite != 0) { 2870 // discard any pending drain or write ack by incrementing sequence 2871 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2872 mDrainSequence = (mDrainSequence + 2) & ~1; 2873 ALOG_ASSERT(mCallbackThread != 0); 2874 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2875 mCallbackThread->setDraining(mDrainSequence); 2876 } 2877} 2878 2879void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2880{ 2881 ALOGV("signal playback thread"); 2882 broadcast_l(); 2883} 2884 2885void AudioFlinger::MixerThread::threadLoop_mix() 2886{ 2887 // obtain the presentation timestamp of the next output buffer 2888 int64_t pts; 2889 status_t status = INVALID_OPERATION; 2890 2891 if (mNormalSink != 0) { 2892 status = mNormalSink->getNextWriteTimestamp(&pts); 2893 } else { 2894 status = mOutputSink->getNextWriteTimestamp(&pts); 2895 } 2896 2897 if (status != NO_ERROR) { 2898 pts = AudioBufferProvider::kInvalidPTS; 2899 } 2900 2901 // mix buffers... 2902 mAudioMixer->process(pts); 2903 mCurrentWriteLength = mSinkBufferSize; 2904 // increase sleep time progressively when application underrun condition clears. 2905 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2906 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2907 // such that we would underrun the audio HAL. 2908 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2909 sleepTimeShift--; 2910 } 2911 sleepTime = 0; 2912 standbyTime = systemTime() + standbyDelay; 2913 //TODO: delay standby when effects have a tail 2914} 2915 2916void AudioFlinger::MixerThread::threadLoop_sleepTime() 2917{ 2918 // If no tracks are ready, sleep once for the duration of an output 2919 // buffer size, then write 0s to the output 2920 if (sleepTime == 0) { 2921 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2922 sleepTime = activeSleepTime >> sleepTimeShift; 2923 if (sleepTime < kMinThreadSleepTimeUs) { 2924 sleepTime = kMinThreadSleepTimeUs; 2925 } 2926 // reduce sleep time in case of consecutive application underruns to avoid 2927 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2928 // duration we would end up writing less data than needed by the audio HAL if 2929 // the condition persists. 2930 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2931 sleepTimeShift++; 2932 } 2933 } else { 2934 sleepTime = idleSleepTime; 2935 } 2936 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2937 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2938 // before effects processing or output. 2939 if (mMixerBufferValid) { 2940 memset(mMixerBuffer, 0, mMixerBufferSize); 2941 } else { 2942 memset(mSinkBuffer, 0, mSinkBufferSize); 2943 } 2944 sleepTime = 0; 2945 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2946 "anticipated start"); 2947 } 2948 // TODO add standby time extension fct of effect tail 2949} 2950 2951// prepareTracks_l() must be called with ThreadBase::mLock held 2952AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2953 Vector< sp<Track> > *tracksToRemove) 2954{ 2955 2956 mixer_state mixerStatus = MIXER_IDLE; 2957 // find out which tracks need to be processed 2958 size_t count = mActiveTracks.size(); 2959 size_t mixedTracks = 0; 2960 size_t tracksWithEffect = 0; 2961 // counts only _active_ fast tracks 2962 size_t fastTracks = 0; 2963 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2964 2965 float masterVolume = mMasterVolume; 2966 bool masterMute = mMasterMute; 2967 2968 if (masterMute) { 2969 masterVolume = 0; 2970 } 2971 // Delegate master volume control to effect in output mix effect chain if needed 2972 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2973 if (chain != 0) { 2974 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2975 chain->setVolume_l(&v, &v); 2976 masterVolume = (float)((v + (1 << 23)) >> 24); 2977 chain.clear(); 2978 } 2979 2980 // prepare a new state to push 2981 FastMixerStateQueue *sq = NULL; 2982 FastMixerState *state = NULL; 2983 bool didModify = false; 2984 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2985 if (mFastMixer != NULL) { 2986 sq = mFastMixer->sq(); 2987 state = sq->begin(); 2988 } 2989 2990 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 2991 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 2992 2993 for (size_t i=0 ; i<count ; i++) { 2994 const sp<Track> t = mActiveTracks[i].promote(); 2995 if (t == 0) { 2996 continue; 2997 } 2998 2999 // this const just means the local variable doesn't change 3000 Track* const track = t.get(); 3001 3002 // process fast tracks 3003 if (track->isFastTrack()) { 3004 3005 // It's theoretically possible (though unlikely) for a fast track to be created 3006 // and then removed within the same normal mix cycle. This is not a problem, as 3007 // the track never becomes active so it's fast mixer slot is never touched. 3008 // The converse, of removing an (active) track and then creating a new track 3009 // at the identical fast mixer slot within the same normal mix cycle, 3010 // is impossible because the slot isn't marked available until the end of each cycle. 3011 int j = track->mFastIndex; 3012 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3013 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3014 FastTrack *fastTrack = &state->mFastTracks[j]; 3015 3016 // Determine whether the track is currently in underrun condition, 3017 // and whether it had a recent underrun. 3018 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3019 FastTrackUnderruns underruns = ftDump->mUnderruns; 3020 uint32_t recentFull = (underruns.mBitFields.mFull - 3021 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3022 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3023 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3024 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3025 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3026 uint32_t recentUnderruns = recentPartial + recentEmpty; 3027 track->mObservedUnderruns = underruns; 3028 // don't count underruns that occur while stopping or pausing 3029 // or stopped which can occur when flush() is called while active 3030 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3031 recentUnderruns > 0) { 3032 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3033 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3034 } 3035 3036 // This is similar to the state machine for normal tracks, 3037 // with a few modifications for fast tracks. 3038 bool isActive = true; 3039 switch (track->mState) { 3040 case TrackBase::STOPPING_1: 3041 // track stays active in STOPPING_1 state until first underrun 3042 if (recentUnderruns > 0 || track->isTerminated()) { 3043 track->mState = TrackBase::STOPPING_2; 3044 } 3045 break; 3046 case TrackBase::PAUSING: 3047 // ramp down is not yet implemented 3048 track->setPaused(); 3049 break; 3050 case TrackBase::RESUMING: 3051 // ramp up is not yet implemented 3052 track->mState = TrackBase::ACTIVE; 3053 break; 3054 case TrackBase::ACTIVE: 3055 if (recentFull > 0 || recentPartial > 0) { 3056 // track has provided at least some frames recently: reset retry count 3057 track->mRetryCount = kMaxTrackRetries; 3058 } 3059 if (recentUnderruns == 0) { 3060 // no recent underruns: stay active 3061 break; 3062 } 3063 // there has recently been an underrun of some kind 3064 if (track->sharedBuffer() == 0) { 3065 // were any of the recent underruns "empty" (no frames available)? 3066 if (recentEmpty == 0) { 3067 // no, then ignore the partial underruns as they are allowed indefinitely 3068 break; 3069 } 3070 // there has recently been an "empty" underrun: decrement the retry counter 3071 if (--(track->mRetryCount) > 0) { 3072 break; 3073 } 3074 // indicate to client process that the track was disabled because of underrun; 3075 // it will then automatically call start() when data is available 3076 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3077 // remove from active list, but state remains ACTIVE [confusing but true] 3078 isActive = false; 3079 break; 3080 } 3081 // fall through 3082 case TrackBase::STOPPING_2: 3083 case TrackBase::PAUSED: 3084 case TrackBase::STOPPED: 3085 case TrackBase::FLUSHED: // flush() while active 3086 // Check for presentation complete if track is inactive 3087 // We have consumed all the buffers of this track. 3088 // This would be incomplete if we auto-paused on underrun 3089 { 3090 size_t audioHALFrames = 3091 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3092 size_t framesWritten = mBytesWritten / mFrameSize; 3093 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3094 // track stays in active list until presentation is complete 3095 break; 3096 } 3097 } 3098 if (track->isStopping_2()) { 3099 track->mState = TrackBase::STOPPED; 3100 } 3101 if (track->isStopped()) { 3102 // Can't reset directly, as fast mixer is still polling this track 3103 // track->reset(); 3104 // So instead mark this track as needing to be reset after push with ack 3105 resetMask |= 1 << i; 3106 } 3107 isActive = false; 3108 break; 3109 case TrackBase::IDLE: 3110 default: 3111 LOG_FATAL("unexpected track state %d", track->mState); 3112 } 3113 3114 if (isActive) { 3115 // was it previously inactive? 3116 if (!(state->mTrackMask & (1 << j))) { 3117 ExtendedAudioBufferProvider *eabp = track; 3118 VolumeProvider *vp = track; 3119 fastTrack->mBufferProvider = eabp; 3120 fastTrack->mVolumeProvider = vp; 3121 fastTrack->mChannelMask = track->mChannelMask; 3122 fastTrack->mGeneration++; 3123 state->mTrackMask |= 1 << j; 3124 didModify = true; 3125 // no acknowledgement required for newly active tracks 3126 } 3127 // cache the combined master volume and stream type volume for fast mixer; this 3128 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3129 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3130 ++fastTracks; 3131 } else { 3132 // was it previously active? 3133 if (state->mTrackMask & (1 << j)) { 3134 fastTrack->mBufferProvider = NULL; 3135 fastTrack->mGeneration++; 3136 state->mTrackMask &= ~(1 << j); 3137 didModify = true; 3138 // If any fast tracks were removed, we must wait for acknowledgement 3139 // because we're about to decrement the last sp<> on those tracks. 3140 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3141 } else { 3142 LOG_FATAL("fast track %d should have been active", j); 3143 } 3144 tracksToRemove->add(track); 3145 // Avoids a misleading display in dumpsys 3146 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3147 } 3148 continue; 3149 } 3150 3151 { // local variable scope to avoid goto warning 3152 3153 audio_track_cblk_t* cblk = track->cblk(); 3154 3155 // The first time a track is added we wait 3156 // for all its buffers to be filled before processing it 3157 int name = track->name(); 3158 // make sure that we have enough frames to mix one full buffer. 3159 // enforce this condition only once to enable draining the buffer in case the client 3160 // app does not call stop() and relies on underrun to stop: 3161 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3162 // during last round 3163 size_t desiredFrames; 3164 uint32_t sr = track->sampleRate(); 3165 if (sr == mSampleRate) { 3166 desiredFrames = mNormalFrameCount; 3167 } else { 3168 // +1 for rounding and +1 for additional sample needed for interpolation 3169 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3170 // add frames already consumed but not yet released by the resampler 3171 // because mAudioTrackServerProxy->framesReady() will include these frames 3172 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3173#if 0 3174 // the minimum track buffer size is normally twice the number of frames necessary 3175 // to fill one buffer and the resampler should not leave more than one buffer worth 3176 // of unreleased frames after each pass, but just in case... 3177 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3178#endif 3179 } 3180 uint32_t minFrames = 1; 3181 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3182 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3183 minFrames = desiredFrames; 3184 } 3185 3186 size_t framesReady = track->framesReady(); 3187 if ((framesReady >= minFrames) && track->isReady() && 3188 !track->isPaused() && !track->isTerminated()) 3189 { 3190 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3191 3192 mixedTracks++; 3193 3194 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3195 // there is an effect chain connected to the track 3196 chain.clear(); 3197 if (track->mainBuffer() != mSinkBuffer && 3198 track->mainBuffer() != mMixerBuffer) { 3199 if (mEffectBufferEnabled) { 3200 mEffectBufferValid = true; // Later can set directly. 3201 } 3202 chain = getEffectChain_l(track->sessionId()); 3203 // Delegate volume control to effect in track effect chain if needed 3204 if (chain != 0) { 3205 tracksWithEffect++; 3206 } else { 3207 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3208 "session %d", 3209 name, track->sessionId()); 3210 } 3211 } 3212 3213 3214 int param = AudioMixer::VOLUME; 3215 if (track->mFillingUpStatus == Track::FS_FILLED) { 3216 // no ramp for the first volume setting 3217 track->mFillingUpStatus = Track::FS_ACTIVE; 3218 if (track->mState == TrackBase::RESUMING) { 3219 track->mState = TrackBase::ACTIVE; 3220 param = AudioMixer::RAMP_VOLUME; 3221 } 3222 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3223 // FIXME should not make a decision based on mServer 3224 } else if (cblk->mServer != 0) { 3225 // If the track is stopped before the first frame was mixed, 3226 // do not apply ramp 3227 param = AudioMixer::RAMP_VOLUME; 3228 } 3229 3230 // compute volume for this track 3231 uint32_t vl, vr, va; 3232 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3233 vl = vr = va = 0; 3234 if (track->isPausing()) { 3235 track->setPaused(); 3236 } 3237 } else { 3238 3239 // read original volumes with volume control 3240 float typeVolume = mStreamTypes[track->streamType()].volume; 3241 float v = masterVolume * typeVolume; 3242 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3243 uint32_t vlr = proxy->getVolumeLR(); 3244 vl = vlr & 0xFFFF; 3245 vr = vlr >> 16; 3246 // track volumes come from shared memory, so can't be trusted and must be clamped 3247 if (vl > MAX_GAIN_INT) { 3248 ALOGV("Track left volume out of range: %04X", vl); 3249 vl = MAX_GAIN_INT; 3250 } 3251 if (vr > MAX_GAIN_INT) { 3252 ALOGV("Track right volume out of range: %04X", vr); 3253 vr = MAX_GAIN_INT; 3254 } 3255 // now apply the master volume and stream type volume 3256 vl = (uint32_t)(v * vl) << 12; 3257 vr = (uint32_t)(v * vr) << 12; 3258 // assuming master volume and stream type volume each go up to 1.0, 3259 // vl and vr are now in 8.24 format 3260 3261 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3262 // send level comes from shared memory and so may be corrupt 3263 if (sendLevel > MAX_GAIN_INT) { 3264 ALOGV("Track send level out of range: %04X", sendLevel); 3265 sendLevel = MAX_GAIN_INT; 3266 } 3267 va = (uint32_t)(v * sendLevel); 3268 } 3269 3270 // Delegate volume control to effect in track effect chain if needed 3271 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3272 // Do not ramp volume if volume is controlled by effect 3273 param = AudioMixer::VOLUME; 3274 track->mHasVolumeController = true; 3275 } else { 3276 // force no volume ramp when volume controller was just disabled or removed 3277 // from effect chain to avoid volume spike 3278 if (track->mHasVolumeController) { 3279 param = AudioMixer::VOLUME; 3280 } 3281 track->mHasVolumeController = false; 3282 } 3283 3284 // Convert volumes from 8.24 to 4.12 format 3285 // This additional clamping is needed in case chain->setVolume_l() overshot 3286 vl = (vl + (1 << 11)) >> 12; 3287 if (vl > MAX_GAIN_INT) { 3288 vl = MAX_GAIN_INT; 3289 } 3290 vr = (vr + (1 << 11)) >> 12; 3291 if (vr > MAX_GAIN_INT) { 3292 vr = MAX_GAIN_INT; 3293 } 3294 3295 if (va > MAX_GAIN_INT) { 3296 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3297 } 3298 3299 // XXX: these things DON'T need to be done each time 3300 mAudioMixer->setBufferProvider(name, track); 3301 mAudioMixer->enable(name); 3302 3303 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3304 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3305 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3306 mAudioMixer->setParameter( 3307 name, 3308 AudioMixer::TRACK, 3309 AudioMixer::FORMAT, (void *)track->format()); 3310 mAudioMixer->setParameter( 3311 name, 3312 AudioMixer::TRACK, 3313 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3314 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3315 uint32_t maxSampleRate = mSampleRate * 2; 3316 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3317 if (reqSampleRate == 0) { 3318 reqSampleRate = mSampleRate; 3319 } else if (reqSampleRate > maxSampleRate) { 3320 reqSampleRate = maxSampleRate; 3321 } 3322 mAudioMixer->setParameter( 3323 name, 3324 AudioMixer::RESAMPLE, 3325 AudioMixer::SAMPLE_RATE, 3326 (void *)(uintptr_t)reqSampleRate); 3327 /* 3328 * Select the appropriate output buffer for the track. 3329 * 3330 * Tracks with effects go into their own effects chain buffer 3331 * and from there into either mEffectBuffer or mSinkBuffer. 3332 * 3333 * Other tracks can use mMixerBuffer for higher precision 3334 * channel accumulation. If this buffer is enabled 3335 * (mMixerBufferEnabled true), then selected tracks will accumulate 3336 * into it. 3337 * 3338 */ 3339 if (mMixerBufferEnabled 3340 && (track->mainBuffer() == mSinkBuffer 3341 || track->mainBuffer() == mMixerBuffer)) { 3342 mAudioMixer->setParameter( 3343 name, 3344 AudioMixer::TRACK, 3345 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3346 mAudioMixer->setParameter( 3347 name, 3348 AudioMixer::TRACK, 3349 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3350 // TODO: override track->mainBuffer()? 3351 mMixerBufferValid = true; 3352 } else { 3353 mAudioMixer->setParameter( 3354 name, 3355 AudioMixer::TRACK, 3356 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3357 mAudioMixer->setParameter( 3358 name, 3359 AudioMixer::TRACK, 3360 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3361 } 3362 mAudioMixer->setParameter( 3363 name, 3364 AudioMixer::TRACK, 3365 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3366 3367 // reset retry count 3368 track->mRetryCount = kMaxTrackRetries; 3369 3370 // If one track is ready, set the mixer ready if: 3371 // - the mixer was not ready during previous round OR 3372 // - no other track is not ready 3373 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3374 mixerStatus != MIXER_TRACKS_ENABLED) { 3375 mixerStatus = MIXER_TRACKS_READY; 3376 } 3377 } else { 3378 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3379 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3380 } 3381 // clear effect chain input buffer if an active track underruns to avoid sending 3382 // previous audio buffer again to effects 3383 chain = getEffectChain_l(track->sessionId()); 3384 if (chain != 0) { 3385 chain->clearInputBuffer(); 3386 } 3387 3388 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3389 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3390 track->isStopped() || track->isPaused()) { 3391 // We have consumed all the buffers of this track. 3392 // Remove it from the list of active tracks. 3393 // TODO: use actual buffer filling status instead of latency when available from 3394 // audio HAL 3395 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3396 size_t framesWritten = mBytesWritten / mFrameSize; 3397 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3398 if (track->isStopped()) { 3399 track->reset(); 3400 } 3401 tracksToRemove->add(track); 3402 } 3403 } else { 3404 // No buffers for this track. Give it a few chances to 3405 // fill a buffer, then remove it from active list. 3406 if (--(track->mRetryCount) <= 0) { 3407 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3408 tracksToRemove->add(track); 3409 // indicate to client process that the track was disabled because of underrun; 3410 // it will then automatically call start() when data is available 3411 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3412 // If one track is not ready, mark the mixer also not ready if: 3413 // - the mixer was ready during previous round OR 3414 // - no other track is ready 3415 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3416 mixerStatus != MIXER_TRACKS_READY) { 3417 mixerStatus = MIXER_TRACKS_ENABLED; 3418 } 3419 } 3420 mAudioMixer->disable(name); 3421 } 3422 3423 } // local variable scope to avoid goto warning 3424track_is_ready: ; 3425 3426 } 3427 3428 // Push the new FastMixer state if necessary 3429 bool pauseAudioWatchdog = false; 3430 if (didModify) { 3431 state->mFastTracksGen++; 3432 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3433 if (kUseFastMixer == FastMixer_Dynamic && 3434 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3435 state->mCommand = FastMixerState::COLD_IDLE; 3436 state->mColdFutexAddr = &mFastMixerFutex; 3437 state->mColdGen++; 3438 mFastMixerFutex = 0; 3439 if (kUseFastMixer == FastMixer_Dynamic) { 3440 mNormalSink = mOutputSink; 3441 } 3442 // If we go into cold idle, need to wait for acknowledgement 3443 // so that fast mixer stops doing I/O. 3444 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3445 pauseAudioWatchdog = true; 3446 } 3447 } 3448 if (sq != NULL) { 3449 sq->end(didModify); 3450 sq->push(block); 3451 } 3452#ifdef AUDIO_WATCHDOG 3453 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3454 mAudioWatchdog->pause(); 3455 } 3456#endif 3457 3458 // Now perform the deferred reset on fast tracks that have stopped 3459 while (resetMask != 0) { 3460 size_t i = __builtin_ctz(resetMask); 3461 ALOG_ASSERT(i < count); 3462 resetMask &= ~(1 << i); 3463 sp<Track> t = mActiveTracks[i].promote(); 3464 if (t == 0) { 3465 continue; 3466 } 3467 Track* track = t.get(); 3468 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3469 track->reset(); 3470 } 3471 3472 // remove all the tracks that need to be... 3473 removeTracks_l(*tracksToRemove); 3474 3475 // sink or mix buffer must be cleared if all tracks are connected to an 3476 // effect chain as in this case the mixer will not write to the sink or mix buffer 3477 // and track effects will accumulate into it 3478 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3479 (mixedTracks == 0 && fastTracks > 0))) { 3480 // FIXME as a performance optimization, should remember previous zero status 3481 if (mMixerBufferValid) { 3482 memset(mMixerBuffer, 0, mMixerBufferSize); 3483 // TODO: In testing, mSinkBuffer below need not be cleared because 3484 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3485 // after mixing. 3486 // 3487 // To enforce this guarantee: 3488 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3489 // (mixedTracks == 0 && fastTracks > 0)) 3490 // must imply MIXER_TRACKS_READY. 3491 // Later, we may clear buffers regardless, and skip much of this logic. 3492 } 3493 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3494 if (mEffectBufferValid) { 3495 memset(mEffectBuffer, 0, mEffectBufferSize); 3496 } 3497 // FIXME as a performance optimization, should remember previous zero status 3498 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3499 } 3500 3501 // if any fast tracks, then status is ready 3502 mMixerStatusIgnoringFastTracks = mixerStatus; 3503 if (fastTracks > 0) { 3504 mixerStatus = MIXER_TRACKS_READY; 3505 } 3506 return mixerStatus; 3507} 3508 3509// getTrackName_l() must be called with ThreadBase::mLock held 3510int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3511{ 3512 return mAudioMixer->getTrackName(channelMask, sessionId); 3513} 3514 3515// deleteTrackName_l() must be called with ThreadBase::mLock held 3516void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3517{ 3518 ALOGV("remove track (%d) and delete from mixer", name); 3519 mAudioMixer->deleteTrackName(name); 3520} 3521 3522// checkForNewParameters_l() must be called with ThreadBase::mLock held 3523bool AudioFlinger::MixerThread::checkForNewParameters_l() 3524{ 3525 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3526 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3527 bool reconfig = false; 3528 3529 while (!mNewParameters.isEmpty()) { 3530 3531 if (mFastMixer != NULL) { 3532 FastMixerStateQueue *sq = mFastMixer->sq(); 3533 FastMixerState *state = sq->begin(); 3534 if (!(state->mCommand & FastMixerState::IDLE)) { 3535 previousCommand = state->mCommand; 3536 state->mCommand = FastMixerState::HOT_IDLE; 3537 sq->end(); 3538 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3539 } else { 3540 sq->end(false /*didModify*/); 3541 } 3542 } 3543 3544 status_t status = NO_ERROR; 3545 String8 keyValuePair = mNewParameters[0]; 3546 AudioParameter param = AudioParameter(keyValuePair); 3547 int value; 3548 3549 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3550 reconfig = true; 3551 } 3552 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3553 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3554 status = BAD_VALUE; 3555 } else { 3556 // no need to save value, since it's constant 3557 reconfig = true; 3558 } 3559 } 3560 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3561 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3562 status = BAD_VALUE; 3563 } else { 3564 // no need to save value, since it's constant 3565 reconfig = true; 3566 } 3567 } 3568 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3569 // do not accept frame count changes if tracks are open as the track buffer 3570 // size depends on frame count and correct behavior would not be guaranteed 3571 // if frame count is changed after track creation 3572 if (!mTracks.isEmpty()) { 3573 status = INVALID_OPERATION; 3574 } else { 3575 reconfig = true; 3576 } 3577 } 3578 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3579#ifdef ADD_BATTERY_DATA 3580 // when changing the audio output device, call addBatteryData to notify 3581 // the change 3582 if (mOutDevice != value) { 3583 uint32_t params = 0; 3584 // check whether speaker is on 3585 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3586 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3587 } 3588 3589 audio_devices_t deviceWithoutSpeaker 3590 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3591 // check if any other device (except speaker) is on 3592 if (value & deviceWithoutSpeaker ) { 3593 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3594 } 3595 3596 if (params != 0) { 3597 addBatteryData(params); 3598 } 3599 } 3600#endif 3601 3602 // forward device change to effects that have requested to be 3603 // aware of attached audio device. 3604 if (value != AUDIO_DEVICE_NONE) { 3605 mOutDevice = value; 3606 for (size_t i = 0; i < mEffectChains.size(); i++) { 3607 mEffectChains[i]->setDevice_l(mOutDevice); 3608 } 3609 } 3610 } 3611 3612 if (status == NO_ERROR) { 3613 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3614 keyValuePair.string()); 3615 if (!mStandby && status == INVALID_OPERATION) { 3616 mOutput->stream->common.standby(&mOutput->stream->common); 3617 mStandby = true; 3618 mBytesWritten = 0; 3619 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3620 keyValuePair.string()); 3621 } 3622 if (status == NO_ERROR && reconfig) { 3623 readOutputParameters_l(); 3624 delete mAudioMixer; 3625 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3626 for (size_t i = 0; i < mTracks.size() ; i++) { 3627 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3628 if (name < 0) { 3629 break; 3630 } 3631 mTracks[i]->mName = name; 3632 } 3633 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3634 } 3635 } 3636 3637 mNewParameters.removeAt(0); 3638 3639 mParamStatus = status; 3640 mParamCond.signal(); 3641 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3642 // already timed out waiting for the status and will never signal the condition. 3643 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3644 } 3645 3646 if (!(previousCommand & FastMixerState::IDLE)) { 3647 ALOG_ASSERT(mFastMixer != NULL); 3648 FastMixerStateQueue *sq = mFastMixer->sq(); 3649 FastMixerState *state = sq->begin(); 3650 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3651 state->mCommand = previousCommand; 3652 sq->end(); 3653 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3654 } 3655 3656 return reconfig; 3657} 3658 3659 3660void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3661{ 3662 const size_t SIZE = 256; 3663 char buffer[SIZE]; 3664 String8 result; 3665 3666 PlaybackThread::dumpInternals(fd, args); 3667 3668 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3669 3670 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3671 const FastMixerDumpState copy(mFastMixerDumpState); 3672 copy.dump(fd); 3673 3674#ifdef STATE_QUEUE_DUMP 3675 // Similar for state queue 3676 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3677 observerCopy.dump(fd); 3678 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3679 mutatorCopy.dump(fd); 3680#endif 3681 3682#ifdef TEE_SINK 3683 // Write the tee output to a .wav file 3684 dumpTee(fd, mTeeSource, mId); 3685#endif 3686 3687#ifdef AUDIO_WATCHDOG 3688 if (mAudioWatchdog != 0) { 3689 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3690 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3691 wdCopy.dump(fd); 3692 } 3693#endif 3694} 3695 3696uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3697{ 3698 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3699} 3700 3701uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3702{ 3703 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3704} 3705 3706void AudioFlinger::MixerThread::cacheParameters_l() 3707{ 3708 PlaybackThread::cacheParameters_l(); 3709 3710 // FIXME: Relaxed timing because of a certain device that can't meet latency 3711 // Should be reduced to 2x after the vendor fixes the driver issue 3712 // increase threshold again due to low power audio mode. The way this warning 3713 // threshold is calculated and its usefulness should be reconsidered anyway. 3714 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3715} 3716 3717// ---------------------------------------------------------------------------- 3718 3719AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3720 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3721 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3722 // mLeftVolFloat, mRightVolFloat 3723{ 3724} 3725 3726AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3727 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3728 ThreadBase::type_t type) 3729 : PlaybackThread(audioFlinger, output, id, device, type) 3730 // mLeftVolFloat, mRightVolFloat 3731{ 3732} 3733 3734AudioFlinger::DirectOutputThread::~DirectOutputThread() 3735{ 3736} 3737 3738void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3739{ 3740 audio_track_cblk_t* cblk = track->cblk(); 3741 float left, right; 3742 3743 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3744 left = right = 0; 3745 } else { 3746 float typeVolume = mStreamTypes[track->streamType()].volume; 3747 float v = mMasterVolume * typeVolume; 3748 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3749 uint32_t vlr = proxy->getVolumeLR(); 3750 float v_clamped = v * (vlr & 0xFFFF); 3751 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3752 left = v_clamped/MAX_GAIN; 3753 v_clamped = v * (vlr >> 16); 3754 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3755 right = v_clamped/MAX_GAIN; 3756 } 3757 3758 if (lastTrack) { 3759 if (left != mLeftVolFloat || right != mRightVolFloat) { 3760 mLeftVolFloat = left; 3761 mRightVolFloat = right; 3762 3763 // Convert volumes from float to 8.24 3764 uint32_t vl = (uint32_t)(left * (1 << 24)); 3765 uint32_t vr = (uint32_t)(right * (1 << 24)); 3766 3767 // Delegate volume control to effect in track effect chain if needed 3768 // only one effect chain can be present on DirectOutputThread, so if 3769 // there is one, the track is connected to it 3770 if (!mEffectChains.isEmpty()) { 3771 mEffectChains[0]->setVolume_l(&vl, &vr); 3772 left = (float)vl / (1 << 24); 3773 right = (float)vr / (1 << 24); 3774 } 3775 if (mOutput->stream->set_volume) { 3776 mOutput->stream->set_volume(mOutput->stream, left, right); 3777 } 3778 } 3779 } 3780} 3781 3782 3783AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3784 Vector< sp<Track> > *tracksToRemove 3785) 3786{ 3787 size_t count = mActiveTracks.size(); 3788 mixer_state mixerStatus = MIXER_IDLE; 3789 3790 // find out which tracks need to be processed 3791 for (size_t i = 0; i < count; i++) { 3792 sp<Track> t = mActiveTracks[i].promote(); 3793 // The track died recently 3794 if (t == 0) { 3795 continue; 3796 } 3797 3798 Track* const track = t.get(); 3799 audio_track_cblk_t* cblk = track->cblk(); 3800 // Only consider last track started for volume and mixer state control. 3801 // In theory an older track could underrun and restart after the new one starts 3802 // but as we only care about the transition phase between two tracks on a 3803 // direct output, it is not a problem to ignore the underrun case. 3804 sp<Track> l = mLatestActiveTrack.promote(); 3805 bool last = l.get() == track; 3806 3807 // The first time a track is added we wait 3808 // for all its buffers to be filled before processing it 3809 uint32_t minFrames; 3810 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3811 minFrames = mNormalFrameCount; 3812 } else { 3813 minFrames = 1; 3814 } 3815 3816 if ((track->framesReady() >= minFrames) && track->isReady() && 3817 !track->isPaused() && !track->isTerminated()) 3818 { 3819 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3820 3821 if (track->mFillingUpStatus == Track::FS_FILLED) { 3822 track->mFillingUpStatus = Track::FS_ACTIVE; 3823 // make sure processVolume_l() will apply new volume even if 0 3824 mLeftVolFloat = mRightVolFloat = -1.0; 3825 if (track->mState == TrackBase::RESUMING) { 3826 track->mState = TrackBase::ACTIVE; 3827 } 3828 } 3829 3830 // compute volume for this track 3831 processVolume_l(track, last); 3832 if (last) { 3833 // reset retry count 3834 track->mRetryCount = kMaxTrackRetriesDirect; 3835 mActiveTrack = t; 3836 mixerStatus = MIXER_TRACKS_READY; 3837 } 3838 } else { 3839 // clear effect chain input buffer if the last active track started underruns 3840 // to avoid sending previous audio buffer again to effects 3841 if (!mEffectChains.isEmpty() && last) { 3842 mEffectChains[0]->clearInputBuffer(); 3843 } 3844 3845 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3846 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3847 track->isStopped() || track->isPaused()) { 3848 // We have consumed all the buffers of this track. 3849 // Remove it from the list of active tracks. 3850 // TODO: implement behavior for compressed audio 3851 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3852 size_t framesWritten = mBytesWritten / mFrameSize; 3853 if (mStandby || !last || 3854 track->presentationComplete(framesWritten, audioHALFrames)) { 3855 if (track->isStopped()) { 3856 track->reset(); 3857 } 3858 tracksToRemove->add(track); 3859 } 3860 } else { 3861 // No buffers for this track. Give it a few chances to 3862 // fill a buffer, then remove it from active list. 3863 // Only consider last track started for mixer state control 3864 if (--(track->mRetryCount) <= 0) { 3865 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3866 tracksToRemove->add(track); 3867 // indicate to client process that the track was disabled because of underrun; 3868 // it will then automatically call start() when data is available 3869 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3870 } else if (last) { 3871 mixerStatus = MIXER_TRACKS_ENABLED; 3872 } 3873 } 3874 } 3875 } 3876 3877 // remove all the tracks that need to be... 3878 removeTracks_l(*tracksToRemove); 3879 3880 return mixerStatus; 3881} 3882 3883void AudioFlinger::DirectOutputThread::threadLoop_mix() 3884{ 3885 size_t frameCount = mFrameCount; 3886 int8_t *curBuf = (int8_t *)mSinkBuffer; 3887 // output audio to hardware 3888 while (frameCount) { 3889 AudioBufferProvider::Buffer buffer; 3890 buffer.frameCount = frameCount; 3891 mActiveTrack->getNextBuffer(&buffer); 3892 if (buffer.raw == NULL) { 3893 memset(curBuf, 0, frameCount * mFrameSize); 3894 break; 3895 } 3896 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3897 frameCount -= buffer.frameCount; 3898 curBuf += buffer.frameCount * mFrameSize; 3899 mActiveTrack->releaseBuffer(&buffer); 3900 } 3901 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3902 sleepTime = 0; 3903 standbyTime = systemTime() + standbyDelay; 3904 mActiveTrack.clear(); 3905} 3906 3907void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3908{ 3909 if (sleepTime == 0) { 3910 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3911 sleepTime = activeSleepTime; 3912 } else { 3913 sleepTime = idleSleepTime; 3914 } 3915 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3916 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3917 sleepTime = 0; 3918 } 3919} 3920 3921// getTrackName_l() must be called with ThreadBase::mLock held 3922int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3923 int sessionId __unused) 3924{ 3925 return 0; 3926} 3927 3928// deleteTrackName_l() must be called with ThreadBase::mLock held 3929void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3930{ 3931} 3932 3933// checkForNewParameters_l() must be called with ThreadBase::mLock held 3934bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3935{ 3936 bool reconfig = false; 3937 3938 while (!mNewParameters.isEmpty()) { 3939 status_t status = NO_ERROR; 3940 String8 keyValuePair = mNewParameters[0]; 3941 AudioParameter param = AudioParameter(keyValuePair); 3942 int value; 3943 3944 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3945 // do not accept frame count changes if tracks are open as the track buffer 3946 // size depends on frame count and correct behavior would not be garantied 3947 // if frame count is changed after track creation 3948 if (!mTracks.isEmpty()) { 3949 status = INVALID_OPERATION; 3950 } else { 3951 reconfig = true; 3952 } 3953 } 3954 if (status == NO_ERROR) { 3955 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3956 keyValuePair.string()); 3957 if (!mStandby && status == INVALID_OPERATION) { 3958 mOutput->stream->common.standby(&mOutput->stream->common); 3959 mStandby = true; 3960 mBytesWritten = 0; 3961 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3962 keyValuePair.string()); 3963 } 3964 if (status == NO_ERROR && reconfig) { 3965 readOutputParameters_l(); 3966 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3967 } 3968 } 3969 3970 mNewParameters.removeAt(0); 3971 3972 mParamStatus = status; 3973 mParamCond.signal(); 3974 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3975 // already timed out waiting for the status and will never signal the condition. 3976 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3977 } 3978 return reconfig; 3979} 3980 3981uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3982{ 3983 uint32_t time; 3984 if (audio_is_linear_pcm(mFormat)) { 3985 time = PlaybackThread::activeSleepTimeUs(); 3986 } else { 3987 time = 10000; 3988 } 3989 return time; 3990} 3991 3992uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3993{ 3994 uint32_t time; 3995 if (audio_is_linear_pcm(mFormat)) { 3996 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3997 } else { 3998 time = 10000; 3999 } 4000 return time; 4001} 4002 4003uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4004{ 4005 uint32_t time; 4006 if (audio_is_linear_pcm(mFormat)) { 4007 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4008 } else { 4009 time = 10000; 4010 } 4011 return time; 4012} 4013 4014void AudioFlinger::DirectOutputThread::cacheParameters_l() 4015{ 4016 PlaybackThread::cacheParameters_l(); 4017 4018 // use shorter standby delay as on normal output to release 4019 // hardware resources as soon as possible 4020 if (audio_is_linear_pcm(mFormat)) { 4021 standbyDelay = microseconds(activeSleepTime*2); 4022 } else { 4023 standbyDelay = kOffloadStandbyDelayNs; 4024 } 4025} 4026 4027// ---------------------------------------------------------------------------- 4028 4029AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4030 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4031 : Thread(false /*canCallJava*/), 4032 mPlaybackThread(playbackThread), 4033 mWriteAckSequence(0), 4034 mDrainSequence(0) 4035{ 4036} 4037 4038AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4039{ 4040} 4041 4042void AudioFlinger::AsyncCallbackThread::onFirstRef() 4043{ 4044 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4045} 4046 4047bool AudioFlinger::AsyncCallbackThread::threadLoop() 4048{ 4049 while (!exitPending()) { 4050 uint32_t writeAckSequence; 4051 uint32_t drainSequence; 4052 4053 { 4054 Mutex::Autolock _l(mLock); 4055 while (!((mWriteAckSequence & 1) || 4056 (mDrainSequence & 1) || 4057 exitPending())) { 4058 mWaitWorkCV.wait(mLock); 4059 } 4060 4061 if (exitPending()) { 4062 break; 4063 } 4064 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4065 mWriteAckSequence, mDrainSequence); 4066 writeAckSequence = mWriteAckSequence; 4067 mWriteAckSequence &= ~1; 4068 drainSequence = mDrainSequence; 4069 mDrainSequence &= ~1; 4070 } 4071 { 4072 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4073 if (playbackThread != 0) { 4074 if (writeAckSequence & 1) { 4075 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4076 } 4077 if (drainSequence & 1) { 4078 playbackThread->resetDraining(drainSequence >> 1); 4079 } 4080 } 4081 } 4082 } 4083 return false; 4084} 4085 4086void AudioFlinger::AsyncCallbackThread::exit() 4087{ 4088 ALOGV("AsyncCallbackThread::exit"); 4089 Mutex::Autolock _l(mLock); 4090 requestExit(); 4091 mWaitWorkCV.broadcast(); 4092} 4093 4094void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4095{ 4096 Mutex::Autolock _l(mLock); 4097 // bit 0 is cleared 4098 mWriteAckSequence = sequence << 1; 4099} 4100 4101void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4102{ 4103 Mutex::Autolock _l(mLock); 4104 // ignore unexpected callbacks 4105 if (mWriteAckSequence & 2) { 4106 mWriteAckSequence |= 1; 4107 mWaitWorkCV.signal(); 4108 } 4109} 4110 4111void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4112{ 4113 Mutex::Autolock _l(mLock); 4114 // bit 0 is cleared 4115 mDrainSequence = sequence << 1; 4116} 4117 4118void AudioFlinger::AsyncCallbackThread::resetDraining() 4119{ 4120 Mutex::Autolock _l(mLock); 4121 // ignore unexpected callbacks 4122 if (mDrainSequence & 2) { 4123 mDrainSequence |= 1; 4124 mWaitWorkCV.signal(); 4125 } 4126} 4127 4128 4129// ---------------------------------------------------------------------------- 4130AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4131 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4132 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4133 mHwPaused(false), 4134 mFlushPending(false), 4135 mPausedBytesRemaining(0) 4136{ 4137 //FIXME: mStandby should be set to true by ThreadBase constructor 4138 mStandby = true; 4139} 4140 4141void AudioFlinger::OffloadThread::threadLoop_exit() 4142{ 4143 if (mFlushPending || mHwPaused) { 4144 // If a flush is pending or track was paused, just discard buffered data 4145 flushHw_l(); 4146 } else { 4147 mMixerStatus = MIXER_DRAIN_ALL; 4148 threadLoop_drain(); 4149 } 4150 mCallbackThread->exit(); 4151 PlaybackThread::threadLoop_exit(); 4152} 4153 4154AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4155 Vector< sp<Track> > *tracksToRemove 4156) 4157{ 4158 size_t count = mActiveTracks.size(); 4159 4160 mixer_state mixerStatus = MIXER_IDLE; 4161 bool doHwPause = false; 4162 bool doHwResume = false; 4163 4164 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4165 4166 // find out which tracks need to be processed 4167 for (size_t i = 0; i < count; i++) { 4168 sp<Track> t = mActiveTracks[i].promote(); 4169 // The track died recently 4170 if (t == 0) { 4171 continue; 4172 } 4173 Track* const track = t.get(); 4174 audio_track_cblk_t* cblk = track->cblk(); 4175 // Only consider last track started for volume and mixer state control. 4176 // In theory an older track could underrun and restart after the new one starts 4177 // but as we only care about the transition phase between two tracks on a 4178 // direct output, it is not a problem to ignore the underrun case. 4179 sp<Track> l = mLatestActiveTrack.promote(); 4180 bool last = l.get() == track; 4181 4182 if (track->isInvalid()) { 4183 ALOGW("An invalidated track shouldn't be in active list"); 4184 tracksToRemove->add(track); 4185 continue; 4186 } 4187 4188 if (track->mState == TrackBase::IDLE) { 4189 ALOGW("An idle track shouldn't be in active list"); 4190 continue; 4191 } 4192 4193 if (track->isPausing()) { 4194 track->setPaused(); 4195 if (last) { 4196 if (!mHwPaused) { 4197 doHwPause = true; 4198 mHwPaused = true; 4199 } 4200 // If we were part way through writing the mixbuffer to 4201 // the HAL we must save this until we resume 4202 // BUG - this will be wrong if a different track is made active, 4203 // in that case we want to discard the pending data in the 4204 // mixbuffer and tell the client to present it again when the 4205 // track is resumed 4206 mPausedWriteLength = mCurrentWriteLength; 4207 mPausedBytesRemaining = mBytesRemaining; 4208 mBytesRemaining = 0; // stop writing 4209 } 4210 tracksToRemove->add(track); 4211 } else if (track->isFlushPending()) { 4212 track->flushAck(); 4213 if (last) { 4214 mFlushPending = true; 4215 } 4216 } else if (track->framesReady() && track->isReady() && 4217 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4218 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4219 if (track->mFillingUpStatus == Track::FS_FILLED) { 4220 track->mFillingUpStatus = Track::FS_ACTIVE; 4221 // make sure processVolume_l() will apply new volume even if 0 4222 mLeftVolFloat = mRightVolFloat = -1.0; 4223 if (track->mState == TrackBase::RESUMING) { 4224 track->mState = TrackBase::ACTIVE; 4225 if (last) { 4226 if (mPausedBytesRemaining) { 4227 // Need to continue write that was interrupted 4228 mCurrentWriteLength = mPausedWriteLength; 4229 mBytesRemaining = mPausedBytesRemaining; 4230 mPausedBytesRemaining = 0; 4231 } 4232 if (mHwPaused) { 4233 doHwResume = true; 4234 mHwPaused = false; 4235 // threadLoop_mix() will handle the case that we need to 4236 // resume an interrupted write 4237 } 4238 // enable write to audio HAL 4239 sleepTime = 0; 4240 } 4241 } 4242 } 4243 4244 if (last) { 4245 sp<Track> previousTrack = mPreviousTrack.promote(); 4246 if (previousTrack != 0) { 4247 if (track != previousTrack.get()) { 4248 // Flush any data still being written from last track 4249 mBytesRemaining = 0; 4250 if (mPausedBytesRemaining) { 4251 // Last track was paused so we also need to flush saved 4252 // mixbuffer state and invalidate track so that it will 4253 // re-submit that unwritten data when it is next resumed 4254 mPausedBytesRemaining = 0; 4255 // Invalidate is a bit drastic - would be more efficient 4256 // to have a flag to tell client that some of the 4257 // previously written data was lost 4258 previousTrack->invalidate(); 4259 } 4260 // flush data already sent to the DSP if changing audio session as audio 4261 // comes from a different source. Also invalidate previous track to force a 4262 // seek when resuming. 4263 if (previousTrack->sessionId() != track->sessionId()) { 4264 previousTrack->invalidate(); 4265 } 4266 } 4267 } 4268 mPreviousTrack = track; 4269 // reset retry count 4270 track->mRetryCount = kMaxTrackRetriesOffload; 4271 mActiveTrack = t; 4272 mixerStatus = MIXER_TRACKS_READY; 4273 } 4274 } else { 4275 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4276 if (track->isStopping_1()) { 4277 // Hardware buffer can hold a large amount of audio so we must 4278 // wait for all current track's data to drain before we say 4279 // that the track is stopped. 4280 if (mBytesRemaining == 0) { 4281 // Only start draining when all data in mixbuffer 4282 // has been written 4283 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4284 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4285 // do not drain if no data was ever sent to HAL (mStandby == true) 4286 if (last && !mStandby) { 4287 // do not modify drain sequence if we are already draining. This happens 4288 // when resuming from pause after drain. 4289 if ((mDrainSequence & 1) == 0) { 4290 sleepTime = 0; 4291 standbyTime = systemTime() + standbyDelay; 4292 mixerStatus = MIXER_DRAIN_TRACK; 4293 mDrainSequence += 2; 4294 } 4295 if (mHwPaused) { 4296 // It is possible to move from PAUSED to STOPPING_1 without 4297 // a resume so we must ensure hardware is running 4298 doHwResume = true; 4299 mHwPaused = false; 4300 } 4301 } 4302 } 4303 } else if (track->isStopping_2()) { 4304 // Drain has completed or we are in standby, signal presentation complete 4305 if (!(mDrainSequence & 1) || !last || mStandby) { 4306 track->mState = TrackBase::STOPPED; 4307 size_t audioHALFrames = 4308 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4309 size_t framesWritten = 4310 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4311 track->presentationComplete(framesWritten, audioHALFrames); 4312 track->reset(); 4313 tracksToRemove->add(track); 4314 } 4315 } else { 4316 // No buffers for this track. Give it a few chances to 4317 // fill a buffer, then remove it from active list. 4318 if (--(track->mRetryCount) <= 0) { 4319 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4320 track->name()); 4321 tracksToRemove->add(track); 4322 // indicate to client process that the track was disabled because of underrun; 4323 // it will then automatically call start() when data is available 4324 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4325 } else if (last){ 4326 mixerStatus = MIXER_TRACKS_ENABLED; 4327 } 4328 } 4329 } 4330 // compute volume for this track 4331 processVolume_l(track, last); 4332 } 4333 4334 // make sure the pause/flush/resume sequence is executed in the right order. 4335 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4336 // before flush and then resume HW. This can happen in case of pause/flush/resume 4337 // if resume is received before pause is executed. 4338 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4339 mOutput->stream->pause(mOutput->stream); 4340 } 4341 if (mFlushPending) { 4342 flushHw_l(); 4343 mFlushPending = false; 4344 } 4345 if (!mStandby && doHwResume) { 4346 mOutput->stream->resume(mOutput->stream); 4347 } 4348 4349 // remove all the tracks that need to be... 4350 removeTracks_l(*tracksToRemove); 4351 4352 return mixerStatus; 4353} 4354 4355// must be called with thread mutex locked 4356bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4357{ 4358 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4359 mWriteAckSequence, mDrainSequence); 4360 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4361 return true; 4362 } 4363 return false; 4364} 4365 4366// must be called with thread mutex locked 4367bool AudioFlinger::OffloadThread::shouldStandby_l() 4368{ 4369 bool trackPaused = false; 4370 4371 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4372 // after a timeout and we will enter standby then. 4373 if (mTracks.size() > 0) { 4374 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4375 } 4376 4377 return !mStandby && !trackPaused; 4378} 4379 4380 4381bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4382{ 4383 Mutex::Autolock _l(mLock); 4384 return waitingAsyncCallback_l(); 4385} 4386 4387void AudioFlinger::OffloadThread::flushHw_l() 4388{ 4389 mOutput->stream->flush(mOutput->stream); 4390 // Flush anything still waiting in the mixbuffer 4391 mCurrentWriteLength = 0; 4392 mBytesRemaining = 0; 4393 mPausedWriteLength = 0; 4394 mPausedBytesRemaining = 0; 4395 mHwPaused = false; 4396 4397 if (mUseAsyncWrite) { 4398 // discard any pending drain or write ack by incrementing sequence 4399 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4400 mDrainSequence = (mDrainSequence + 2) & ~1; 4401 ALOG_ASSERT(mCallbackThread != 0); 4402 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4403 mCallbackThread->setDraining(mDrainSequence); 4404 } 4405} 4406 4407void AudioFlinger::OffloadThread::onAddNewTrack_l() 4408{ 4409 sp<Track> previousTrack = mPreviousTrack.promote(); 4410 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4411 4412 if (previousTrack != 0 && latestTrack != 0 && 4413 (previousTrack->sessionId() != latestTrack->sessionId())) { 4414 mFlushPending = true; 4415 } 4416 PlaybackThread::onAddNewTrack_l(); 4417} 4418 4419// ---------------------------------------------------------------------------- 4420 4421AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4422 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4423 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4424 DUPLICATING), 4425 mWaitTimeMs(UINT_MAX) 4426{ 4427 addOutputTrack(mainThread); 4428} 4429 4430AudioFlinger::DuplicatingThread::~DuplicatingThread() 4431{ 4432 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4433 mOutputTracks[i]->destroy(); 4434 } 4435} 4436 4437void AudioFlinger::DuplicatingThread::threadLoop_mix() 4438{ 4439 // mix buffers... 4440 if (outputsReady(outputTracks)) { 4441 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4442 } else { 4443 memset(mSinkBuffer, 0, mSinkBufferSize); 4444 } 4445 sleepTime = 0; 4446 writeFrames = mNormalFrameCount; 4447 mCurrentWriteLength = mSinkBufferSize; 4448 standbyTime = systemTime() + standbyDelay; 4449} 4450 4451void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4452{ 4453 if (sleepTime == 0) { 4454 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4455 sleepTime = activeSleepTime; 4456 } else { 4457 sleepTime = idleSleepTime; 4458 } 4459 } else if (mBytesWritten != 0) { 4460 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4461 writeFrames = mNormalFrameCount; 4462 memset(mSinkBuffer, 0, mSinkBufferSize); 4463 } else { 4464 // flush remaining overflow buffers in output tracks 4465 writeFrames = 0; 4466 } 4467 sleepTime = 0; 4468 } 4469} 4470 4471ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4472{ 4473 for (size_t i = 0; i < outputTracks.size(); i++) { 4474 outputTracks[i]->write(mSinkBuffer, writeFrames); 4475 } 4476 mStandby = false; 4477 return (ssize_t)mSinkBufferSize; 4478} 4479 4480void AudioFlinger::DuplicatingThread::threadLoop_standby() 4481{ 4482 // DuplicatingThread implements standby by stopping all tracks 4483 for (size_t i = 0; i < outputTracks.size(); i++) { 4484 outputTracks[i]->stop(); 4485 } 4486} 4487 4488void AudioFlinger::DuplicatingThread::saveOutputTracks() 4489{ 4490 outputTracks = mOutputTracks; 4491} 4492 4493void AudioFlinger::DuplicatingThread::clearOutputTracks() 4494{ 4495 outputTracks.clear(); 4496} 4497 4498void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4499{ 4500 Mutex::Autolock _l(mLock); 4501 // FIXME explain this formula 4502 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4503 OutputTrack *outputTrack = new OutputTrack(thread, 4504 this, 4505 mSampleRate, 4506 mFormat, 4507 mChannelMask, 4508 frameCount, 4509 IPCThreadState::self()->getCallingUid()); 4510 if (outputTrack->cblk() != NULL) { 4511 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4512 mOutputTracks.add(outputTrack); 4513 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4514 updateWaitTime_l(); 4515 } 4516} 4517 4518void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4519{ 4520 Mutex::Autolock _l(mLock); 4521 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4522 if (mOutputTracks[i]->thread() == thread) { 4523 mOutputTracks[i]->destroy(); 4524 mOutputTracks.removeAt(i); 4525 updateWaitTime_l(); 4526 return; 4527 } 4528 } 4529 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4530} 4531 4532// caller must hold mLock 4533void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4534{ 4535 mWaitTimeMs = UINT_MAX; 4536 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4537 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4538 if (strong != 0) { 4539 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4540 if (waitTimeMs < mWaitTimeMs) { 4541 mWaitTimeMs = waitTimeMs; 4542 } 4543 } 4544 } 4545} 4546 4547 4548bool AudioFlinger::DuplicatingThread::outputsReady( 4549 const SortedVector< sp<OutputTrack> > &outputTracks) 4550{ 4551 for (size_t i = 0; i < outputTracks.size(); i++) { 4552 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4553 if (thread == 0) { 4554 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4555 outputTracks[i].get()); 4556 return false; 4557 } 4558 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4559 // see note at standby() declaration 4560 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4561 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4562 thread.get()); 4563 return false; 4564 } 4565 } 4566 return true; 4567} 4568 4569uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4570{ 4571 return (mWaitTimeMs * 1000) / 2; 4572} 4573 4574void AudioFlinger::DuplicatingThread::cacheParameters_l() 4575{ 4576 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4577 updateWaitTime_l(); 4578 4579 MixerThread::cacheParameters_l(); 4580} 4581 4582// ---------------------------------------------------------------------------- 4583// Record 4584// ---------------------------------------------------------------------------- 4585 4586AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4587 AudioStreamIn *input, 4588 audio_io_handle_t id, 4589 audio_devices_t outDevice, 4590 audio_devices_t inDevice 4591#ifdef TEE_SINK 4592 , const sp<NBAIO_Sink>& teeSink 4593#endif 4594 ) : 4595 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4596 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4597 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4598 mRsmpInRear(0) 4599#ifdef TEE_SINK 4600 , mTeeSink(teeSink) 4601#endif 4602{ 4603 snprintf(mName, kNameLength, "AudioIn_%X", id); 4604 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4605 4606 readInputParameters_l(); 4607} 4608 4609 4610AudioFlinger::RecordThread::~RecordThread() 4611{ 4612 mAudioFlinger->unregisterWriter(mNBLogWriter); 4613 delete[] mRsmpInBuffer; 4614} 4615 4616void AudioFlinger::RecordThread::onFirstRef() 4617{ 4618 run(mName, PRIORITY_URGENT_AUDIO); 4619} 4620 4621bool AudioFlinger::RecordThread::threadLoop() 4622{ 4623 nsecs_t lastWarning = 0; 4624 4625 inputStandBy(); 4626 4627reacquire_wakelock: 4628 sp<RecordTrack> activeTrack; 4629 int activeTracksGen; 4630 { 4631 Mutex::Autolock _l(mLock); 4632 size_t size = mActiveTracks.size(); 4633 activeTracksGen = mActiveTracksGen; 4634 if (size > 0) { 4635 // FIXME an arbitrary choice 4636 activeTrack = mActiveTracks[0]; 4637 acquireWakeLock_l(activeTrack->uid()); 4638 if (size > 1) { 4639 SortedVector<int> tmp; 4640 for (size_t i = 0; i < size; i++) { 4641 tmp.add(mActiveTracks[i]->uid()); 4642 } 4643 updateWakeLockUids_l(tmp); 4644 } 4645 } else { 4646 acquireWakeLock_l(-1); 4647 } 4648 } 4649 4650 // used to request a deferred sleep, to be executed later while mutex is unlocked 4651 uint32_t sleepUs = 0; 4652 4653 // loop while there is work to do 4654 for (;;) { 4655 Vector< sp<EffectChain> > effectChains; 4656 4657 // sleep with mutex unlocked 4658 if (sleepUs > 0) { 4659 usleep(sleepUs); 4660 sleepUs = 0; 4661 } 4662 4663 // activeTracks accumulates a copy of a subset of mActiveTracks 4664 Vector< sp<RecordTrack> > activeTracks; 4665 4666 { // scope for mLock 4667 Mutex::Autolock _l(mLock); 4668 4669 processConfigEvents_l(); 4670 // return value 'reconfig' is currently unused 4671 bool reconfig = checkForNewParameters_l(); 4672 4673 // check exitPending here because checkForNewParameters_l() and 4674 // checkForNewParameters_l() can temporarily release mLock 4675 if (exitPending()) { 4676 break; 4677 } 4678 4679 // if no active track(s), then standby and release wakelock 4680 size_t size = mActiveTracks.size(); 4681 if (size == 0) { 4682 standbyIfNotAlreadyInStandby(); 4683 // exitPending() can't become true here 4684 releaseWakeLock_l(); 4685 ALOGV("RecordThread: loop stopping"); 4686 // go to sleep 4687 mWaitWorkCV.wait(mLock); 4688 ALOGV("RecordThread: loop starting"); 4689 goto reacquire_wakelock; 4690 } 4691 4692 if (mActiveTracksGen != activeTracksGen) { 4693 activeTracksGen = mActiveTracksGen; 4694 SortedVector<int> tmp; 4695 for (size_t i = 0; i < size; i++) { 4696 tmp.add(mActiveTracks[i]->uid()); 4697 } 4698 updateWakeLockUids_l(tmp); 4699 } 4700 4701 bool doBroadcast = false; 4702 for (size_t i = 0; i < size; ) { 4703 4704 activeTrack = mActiveTracks[i]; 4705 if (activeTrack->isTerminated()) { 4706 removeTrack_l(activeTrack); 4707 mActiveTracks.remove(activeTrack); 4708 mActiveTracksGen++; 4709 size--; 4710 continue; 4711 } 4712 4713 TrackBase::track_state activeTrackState = activeTrack->mState; 4714 switch (activeTrackState) { 4715 4716 case TrackBase::PAUSING: 4717 mActiveTracks.remove(activeTrack); 4718 mActiveTracksGen++; 4719 doBroadcast = true; 4720 size--; 4721 continue; 4722 4723 case TrackBase::STARTING_1: 4724 sleepUs = 10000; 4725 i++; 4726 continue; 4727 4728 case TrackBase::STARTING_2: 4729 doBroadcast = true; 4730 mStandby = false; 4731 activeTrack->mState = TrackBase::ACTIVE; 4732 break; 4733 4734 case TrackBase::ACTIVE: 4735 break; 4736 4737 case TrackBase::IDLE: 4738 i++; 4739 continue; 4740 4741 default: 4742 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4743 } 4744 4745 activeTracks.add(activeTrack); 4746 i++; 4747 4748 } 4749 if (doBroadcast) { 4750 mStartStopCond.broadcast(); 4751 } 4752 4753 // sleep if there are no active tracks to process 4754 if (activeTracks.size() == 0) { 4755 if (sleepUs == 0) { 4756 sleepUs = kRecordThreadSleepUs; 4757 } 4758 continue; 4759 } 4760 sleepUs = 0; 4761 4762 lockEffectChains_l(effectChains); 4763 } 4764 4765 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4766 4767 size_t size = effectChains.size(); 4768 for (size_t i = 0; i < size; i++) { 4769 // thread mutex is not locked, but effect chain is locked 4770 effectChains[i]->process_l(); 4771 } 4772 4773 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4774 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4775 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4776 // If destination is non-contiguous, first read past the nominal end of buffer, then 4777 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4778 4779 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4780 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4781 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4782 if (bytesRead <= 0) { 4783 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4784 // Force input into standby so that it tries to recover at next read attempt 4785 inputStandBy(); 4786 sleepUs = kRecordThreadSleepUs; 4787 continue; 4788 } 4789 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4790 size_t framesRead = bytesRead / mFrameSize; 4791 ALOG_ASSERT(framesRead > 0); 4792 if (mTeeSink != 0) { 4793 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4794 } 4795 // If destination is non-contiguous, we now correct for reading past end of buffer. 4796 size_t part1 = mRsmpInFramesP2 - rear; 4797 if (framesRead > part1) { 4798 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4799 (framesRead - part1) * mFrameSize); 4800 } 4801 rear = mRsmpInRear += framesRead; 4802 4803 size = activeTracks.size(); 4804 // loop over each active track 4805 for (size_t i = 0; i < size; i++) { 4806 activeTrack = activeTracks[i]; 4807 4808 enum { 4809 OVERRUN_UNKNOWN, 4810 OVERRUN_TRUE, 4811 OVERRUN_FALSE 4812 } overrun = OVERRUN_UNKNOWN; 4813 4814 // loop over getNextBuffer to handle circular sink 4815 for (;;) { 4816 4817 activeTrack->mSink.frameCount = ~0; 4818 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4819 size_t framesOut = activeTrack->mSink.frameCount; 4820 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4821 4822 int32_t front = activeTrack->mRsmpInFront; 4823 ssize_t filled = rear - front; 4824 size_t framesIn; 4825 4826 if (filled < 0) { 4827 // should not happen, but treat like a massive overrun and re-sync 4828 framesIn = 0; 4829 activeTrack->mRsmpInFront = rear; 4830 overrun = OVERRUN_TRUE; 4831 } else if ((size_t) filled <= mRsmpInFrames) { 4832 framesIn = (size_t) filled; 4833 } else { 4834 // client is not keeping up with server, but give it latest data 4835 framesIn = mRsmpInFrames; 4836 activeTrack->mRsmpInFront = front = rear - framesIn; 4837 overrun = OVERRUN_TRUE; 4838 } 4839 4840 if (framesOut == 0 || framesIn == 0) { 4841 break; 4842 } 4843 4844 if (activeTrack->mResampler == NULL) { 4845 // no resampling 4846 if (framesIn > framesOut) { 4847 framesIn = framesOut; 4848 } else { 4849 framesOut = framesIn; 4850 } 4851 int8_t *dst = activeTrack->mSink.i8; 4852 while (framesIn > 0) { 4853 front &= mRsmpInFramesP2 - 1; 4854 size_t part1 = mRsmpInFramesP2 - front; 4855 if (part1 > framesIn) { 4856 part1 = framesIn; 4857 } 4858 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4859 if (mChannelCount == activeTrack->mChannelCount) { 4860 memcpy(dst, src, part1 * mFrameSize); 4861 } else if (mChannelCount == 1) { 4862 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4863 part1); 4864 } else { 4865 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4866 part1); 4867 } 4868 dst += part1 * activeTrack->mFrameSize; 4869 front += part1; 4870 framesIn -= part1; 4871 } 4872 activeTrack->mRsmpInFront += framesOut; 4873 4874 } else { 4875 // resampling 4876 // FIXME framesInNeeded should really be part of resampler API, and should 4877 // depend on the SRC ratio 4878 // to keep mRsmpInBuffer full so resampler always has sufficient input 4879 size_t framesInNeeded; 4880 // FIXME only re-calculate when it changes, and optimize for common ratios 4881 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4882 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4883 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4884 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4885 framesInNeeded, framesOut, inOverOut); 4886 // Although we theoretically have framesIn in circular buffer, some of those are 4887 // unreleased frames, and thus must be discounted for purpose of budgeting. 4888 size_t unreleased = activeTrack->mRsmpInUnrel; 4889 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4890 if (framesIn < framesInNeeded) { 4891 ALOGV("not enough to resample: have %u frames in but need %u in to " 4892 "produce %u out given in/out ratio of %.4g", 4893 framesIn, framesInNeeded, framesOut, inOverOut); 4894 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4895 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4896 if (newFramesOut == 0) { 4897 break; 4898 } 4899 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4900 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4901 framesInNeeded, newFramesOut, outOverIn); 4902 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4903 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4904 "given in/out ratio of %.4g", 4905 framesIn, framesInNeeded, newFramesOut, inOverOut); 4906 framesOut = newFramesOut; 4907 } else { 4908 ALOGV("success 1: have %u in and need %u in to produce %u out " 4909 "given in/out ratio of %.4g", 4910 framesIn, framesInNeeded, framesOut, inOverOut); 4911 } 4912 4913 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4914 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4915 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4916 delete[] activeTrack->mRsmpOutBuffer; 4917 // resampler always outputs stereo 4918 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4919 activeTrack->mRsmpOutFrameCount = framesOut; 4920 } 4921 4922 // resampler accumulates, but we only have one source track 4923 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4924 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4925 // FIXME how about having activeTrack implement this interface itself? 4926 activeTrack->mResamplerBufferProvider 4927 /*this*/ /* AudioBufferProvider* */); 4928 // ditherAndClamp() works as long as all buffers returned by 4929 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4930 if (activeTrack->mChannelCount == 1) { 4931 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4932 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4933 framesOut); 4934 // the resampler always outputs stereo samples: 4935 // do post stereo to mono conversion 4936 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4937 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4938 } else { 4939 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4940 activeTrack->mRsmpOutBuffer, framesOut); 4941 } 4942 // now done with mRsmpOutBuffer 4943 4944 } 4945 4946 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4947 overrun = OVERRUN_FALSE; 4948 } 4949 4950 if (activeTrack->mFramesToDrop == 0) { 4951 if (framesOut > 0) { 4952 activeTrack->mSink.frameCount = framesOut; 4953 activeTrack->releaseBuffer(&activeTrack->mSink); 4954 } 4955 } else { 4956 // FIXME could do a partial drop of framesOut 4957 if (activeTrack->mFramesToDrop > 0) { 4958 activeTrack->mFramesToDrop -= framesOut; 4959 if (activeTrack->mFramesToDrop <= 0) { 4960 activeTrack->clearSyncStartEvent(); 4961 } 4962 } else { 4963 activeTrack->mFramesToDrop += framesOut; 4964 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4965 activeTrack->mSyncStartEvent->isCancelled()) { 4966 ALOGW("Synced record %s, session %d, trigger session %d", 4967 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 4968 activeTrack->sessionId(), 4969 (activeTrack->mSyncStartEvent != 0) ? 4970 activeTrack->mSyncStartEvent->triggerSession() : 0); 4971 activeTrack->clearSyncStartEvent(); 4972 } 4973 } 4974 } 4975 4976 if (framesOut == 0) { 4977 break; 4978 } 4979 } 4980 4981 switch (overrun) { 4982 case OVERRUN_TRUE: 4983 // client isn't retrieving buffers fast enough 4984 if (!activeTrack->setOverflow()) { 4985 nsecs_t now = systemTime(); 4986 // FIXME should lastWarning per track? 4987 if ((now - lastWarning) > kWarningThrottleNs) { 4988 ALOGW("RecordThread: buffer overflow"); 4989 lastWarning = now; 4990 } 4991 } 4992 break; 4993 case OVERRUN_FALSE: 4994 activeTrack->clearOverflow(); 4995 break; 4996 case OVERRUN_UNKNOWN: 4997 break; 4998 } 4999 5000 } 5001 5002 // enable changes in effect chain 5003 unlockEffectChains(effectChains); 5004 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5005 } 5006 5007 standbyIfNotAlreadyInStandby(); 5008 5009 { 5010 Mutex::Autolock _l(mLock); 5011 for (size_t i = 0; i < mTracks.size(); i++) { 5012 sp<RecordTrack> track = mTracks[i]; 5013 track->invalidate(); 5014 } 5015 mActiveTracks.clear(); 5016 mActiveTracksGen++; 5017 mStartStopCond.broadcast(); 5018 } 5019 5020 releaseWakeLock(); 5021 5022 ALOGV("RecordThread %p exiting", this); 5023 return false; 5024} 5025 5026void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5027{ 5028 if (!mStandby) { 5029 inputStandBy(); 5030 mStandby = true; 5031 } 5032} 5033 5034void AudioFlinger::RecordThread::inputStandBy() 5035{ 5036 mInput->stream->common.standby(&mInput->stream->common); 5037} 5038 5039sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5040 const sp<AudioFlinger::Client>& client, 5041 uint32_t sampleRate, 5042 audio_format_t format, 5043 audio_channel_mask_t channelMask, 5044 size_t *pFrameCount, 5045 int sessionId, 5046 int uid, 5047 IAudioFlinger::track_flags_t *flags, 5048 pid_t tid, 5049 status_t *status) 5050{ 5051 size_t frameCount = *pFrameCount; 5052 sp<RecordTrack> track; 5053 status_t lStatus; 5054 5055 lStatus = initCheck(); 5056 if (lStatus != NO_ERROR) { 5057 ALOGE("createRecordTrack_l() audio driver not initialized"); 5058 goto Exit; 5059 } 5060 5061 // client expresses a preference for FAST, but we get the final say 5062 if (*flags & IAudioFlinger::TRACK_FAST) { 5063 if ( 5064 // use case: callback handler and frame count is default or at least as large as HAL 5065 ( 5066 (tid != -1) && 5067 ((frameCount == 0) || 5068 (frameCount >= mFrameCount)) 5069 ) && 5070 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 5071 // mono or stereo 5072 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 5073 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 5074 // hardware sample rate 5075 (sampleRate == mSampleRate) && 5076 // record thread has an associated fast recorder 5077 hasFastRecorder() 5078 // FIXME test that RecordThread for this fast track has a capable output HAL 5079 // FIXME add a permission test also? 5080 ) { 5081 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 5082 if (frameCount == 0) { 5083 frameCount = mFrameCount * kFastTrackMultiplier; 5084 } 5085 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5086 frameCount, mFrameCount); 5087 } else { 5088 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5089 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5090 "hasFastRecorder=%d tid=%d", 5091 frameCount, mFrameCount, format, 5092 audio_is_linear_pcm(format), 5093 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 5094 *flags &= ~IAudioFlinger::TRACK_FAST; 5095 // For compatibility with AudioRecord calculation, buffer depth is forced 5096 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5097 // This is probably too conservative, but legacy application code may depend on it. 5098 // If you change this calculation, also review the start threshold which is related. 5099 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5100 size_t mNormalFrameCount = 2048; // FIXME 5101 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5102 if (minBufCount < 2) { 5103 minBufCount = 2; 5104 } 5105 size_t minFrameCount = mNormalFrameCount * minBufCount; 5106 if (frameCount < minFrameCount) { 5107 frameCount = minFrameCount; 5108 } 5109 } 5110 } 5111 *pFrameCount = frameCount; 5112 5113 // FIXME use flags and tid similar to createTrack_l() 5114 5115 { // scope for mLock 5116 Mutex::Autolock _l(mLock); 5117 5118 track = new RecordTrack(this, client, sampleRate, 5119 format, channelMask, frameCount, sessionId, uid); 5120 5121 lStatus = track->initCheck(); 5122 if (lStatus != NO_ERROR) { 5123 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5124 // track must be cleared from the caller as the caller has the AF lock 5125 goto Exit; 5126 } 5127 mTracks.add(track); 5128 5129 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5130 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5131 mAudioFlinger->btNrecIsOff(); 5132 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5133 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5134 5135 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5136 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5137 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5138 // so ask activity manager to do this on our behalf 5139 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5140 } 5141 } 5142 lStatus = NO_ERROR; 5143 5144Exit: 5145 *status = lStatus; 5146 return track; 5147} 5148 5149status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5150 AudioSystem::sync_event_t event, 5151 int triggerSession) 5152{ 5153 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5154 sp<ThreadBase> strongMe = this; 5155 status_t status = NO_ERROR; 5156 5157 if (event == AudioSystem::SYNC_EVENT_NONE) { 5158 recordTrack->clearSyncStartEvent(); 5159 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5160 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5161 triggerSession, 5162 recordTrack->sessionId(), 5163 syncStartEventCallback, 5164 recordTrack); 5165 // Sync event can be cancelled by the trigger session if the track is not in a 5166 // compatible state in which case we start record immediately 5167 if (recordTrack->mSyncStartEvent->isCancelled()) { 5168 recordTrack->clearSyncStartEvent(); 5169 } else { 5170 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5171 recordTrack->mFramesToDrop = - 5172 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5173 } 5174 } 5175 5176 { 5177 // This section is a rendezvous between binder thread executing start() and RecordThread 5178 AutoMutex lock(mLock); 5179 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5180 if (recordTrack->mState == TrackBase::PAUSING) { 5181 ALOGV("active record track PAUSING -> ACTIVE"); 5182 recordTrack->mState = TrackBase::ACTIVE; 5183 } else { 5184 ALOGV("active record track state %d", recordTrack->mState); 5185 } 5186 return status; 5187 } 5188 5189 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5190 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5191 // or using a separate command thread 5192 recordTrack->mState = TrackBase::STARTING_1; 5193 mActiveTracks.add(recordTrack); 5194 mActiveTracksGen++; 5195 mLock.unlock(); 5196 status_t status = AudioSystem::startInput(mId); 5197 mLock.lock(); 5198 // FIXME should verify that recordTrack is still in mActiveTracks 5199 if (status != NO_ERROR) { 5200 mActiveTracks.remove(recordTrack); 5201 mActiveTracksGen++; 5202 recordTrack->clearSyncStartEvent(); 5203 return status; 5204 } 5205 // Catch up with current buffer indices if thread is already running. 5206 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5207 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5208 // see previously buffered data before it called start(), but with greater risk of overrun. 5209 5210 recordTrack->mRsmpInFront = mRsmpInRear; 5211 recordTrack->mRsmpInUnrel = 0; 5212 // FIXME why reset? 5213 if (recordTrack->mResampler != NULL) { 5214 recordTrack->mResampler->reset(); 5215 } 5216 recordTrack->mState = TrackBase::STARTING_2; 5217 // signal thread to start 5218 mWaitWorkCV.broadcast(); 5219 if (mActiveTracks.indexOf(recordTrack) < 0) { 5220 ALOGV("Record failed to start"); 5221 status = BAD_VALUE; 5222 goto startError; 5223 } 5224 return status; 5225 } 5226 5227startError: 5228 AudioSystem::stopInput(mId); 5229 recordTrack->clearSyncStartEvent(); 5230 // FIXME I wonder why we do not reset the state here? 5231 return status; 5232} 5233 5234void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5235{ 5236 sp<SyncEvent> strongEvent = event.promote(); 5237 5238 if (strongEvent != 0) { 5239 sp<RefBase> ptr = strongEvent->cookie().promote(); 5240 if (ptr != 0) { 5241 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5242 recordTrack->handleSyncStartEvent(strongEvent); 5243 } 5244 } 5245} 5246 5247bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5248 ALOGV("RecordThread::stop"); 5249 AutoMutex _l(mLock); 5250 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5251 return false; 5252 } 5253 // note that threadLoop may still be processing the track at this point [without lock] 5254 recordTrack->mState = TrackBase::PAUSING; 5255 // do not wait for mStartStopCond if exiting 5256 if (exitPending()) { 5257 return true; 5258 } 5259 // FIXME incorrect usage of wait: no explicit predicate or loop 5260 mStartStopCond.wait(mLock); 5261 // if we have been restarted, recordTrack is in mActiveTracks here 5262 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5263 ALOGV("Record stopped OK"); 5264 return true; 5265 } 5266 return false; 5267} 5268 5269bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5270{ 5271 return false; 5272} 5273 5274status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5275{ 5276#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5277 if (!isValidSyncEvent(event)) { 5278 return BAD_VALUE; 5279 } 5280 5281 int eventSession = event->triggerSession(); 5282 status_t ret = NAME_NOT_FOUND; 5283 5284 Mutex::Autolock _l(mLock); 5285 5286 for (size_t i = 0; i < mTracks.size(); i++) { 5287 sp<RecordTrack> track = mTracks[i]; 5288 if (eventSession == track->sessionId()) { 5289 (void) track->setSyncEvent(event); 5290 ret = NO_ERROR; 5291 } 5292 } 5293 return ret; 5294#else 5295 return BAD_VALUE; 5296#endif 5297} 5298 5299// destroyTrack_l() must be called with ThreadBase::mLock held 5300void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5301{ 5302 track->terminate(); 5303 track->mState = TrackBase::STOPPED; 5304 // active tracks are removed by threadLoop() 5305 if (mActiveTracks.indexOf(track) < 0) { 5306 removeTrack_l(track); 5307 } 5308} 5309 5310void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5311{ 5312 mTracks.remove(track); 5313 // need anything related to effects here? 5314} 5315 5316void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5317{ 5318 dumpInternals(fd, args); 5319 dumpTracks(fd, args); 5320 dumpEffectChains(fd, args); 5321} 5322 5323void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5324{ 5325 fdprintf(fd, "\nInput thread %p:\n", this); 5326 5327 if (mActiveTracks.size() > 0) { 5328 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5329 } else { 5330 fdprintf(fd, " No active record clients\n"); 5331 } 5332 5333 dumpBase(fd, args); 5334} 5335 5336void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5337{ 5338 const size_t SIZE = 256; 5339 char buffer[SIZE]; 5340 String8 result; 5341 5342 size_t numtracks = mTracks.size(); 5343 size_t numactive = mActiveTracks.size(); 5344 size_t numactiveseen = 0; 5345 fdprintf(fd, " %d Tracks", numtracks); 5346 if (numtracks) { 5347 fdprintf(fd, " of which %d are active\n", numactive); 5348 RecordTrack::appendDumpHeader(result); 5349 for (size_t i = 0; i < numtracks ; ++i) { 5350 sp<RecordTrack> track = mTracks[i]; 5351 if (track != 0) { 5352 bool active = mActiveTracks.indexOf(track) >= 0; 5353 if (active) { 5354 numactiveseen++; 5355 } 5356 track->dump(buffer, SIZE, active); 5357 result.append(buffer); 5358 } 5359 } 5360 } else { 5361 fdprintf(fd, "\n"); 5362 } 5363 5364 if (numactiveseen != numactive) { 5365 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5366 " not in the track list\n"); 5367 result.append(buffer); 5368 RecordTrack::appendDumpHeader(result); 5369 for (size_t i = 0; i < numactive; ++i) { 5370 sp<RecordTrack> track = mActiveTracks[i]; 5371 if (mTracks.indexOf(track) < 0) { 5372 track->dump(buffer, SIZE, true); 5373 result.append(buffer); 5374 } 5375 } 5376 5377 } 5378 write(fd, result.string(), result.size()); 5379} 5380 5381// AudioBufferProvider interface 5382status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5383 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5384{ 5385 RecordTrack *activeTrack = mRecordTrack; 5386 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5387 if (threadBase == 0) { 5388 buffer->frameCount = 0; 5389 buffer->raw = NULL; 5390 return NOT_ENOUGH_DATA; 5391 } 5392 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5393 int32_t rear = recordThread->mRsmpInRear; 5394 int32_t front = activeTrack->mRsmpInFront; 5395 ssize_t filled = rear - front; 5396 // FIXME should not be P2 (don't want to increase latency) 5397 // FIXME if client not keeping up, discard 5398 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5399 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5400 front &= recordThread->mRsmpInFramesP2 - 1; 5401 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5402 if (part1 > (size_t) filled) { 5403 part1 = filled; 5404 } 5405 size_t ask = buffer->frameCount; 5406 ALOG_ASSERT(ask > 0); 5407 if (part1 > ask) { 5408 part1 = ask; 5409 } 5410 if (part1 == 0) { 5411 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5412 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5413 buffer->raw = NULL; 5414 buffer->frameCount = 0; 5415 activeTrack->mRsmpInUnrel = 0; 5416 return NOT_ENOUGH_DATA; 5417 } 5418 5419 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5420 buffer->frameCount = part1; 5421 activeTrack->mRsmpInUnrel = part1; 5422 return NO_ERROR; 5423} 5424 5425// AudioBufferProvider interface 5426void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5427 AudioBufferProvider::Buffer* buffer) 5428{ 5429 RecordTrack *activeTrack = mRecordTrack; 5430 size_t stepCount = buffer->frameCount; 5431 if (stepCount == 0) { 5432 return; 5433 } 5434 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5435 activeTrack->mRsmpInUnrel -= stepCount; 5436 activeTrack->mRsmpInFront += stepCount; 5437 buffer->raw = NULL; 5438 buffer->frameCount = 0; 5439} 5440 5441bool AudioFlinger::RecordThread::checkForNewParameters_l() 5442{ 5443 bool reconfig = false; 5444 5445 while (!mNewParameters.isEmpty()) { 5446 status_t status = NO_ERROR; 5447 String8 keyValuePair = mNewParameters[0]; 5448 AudioParameter param = AudioParameter(keyValuePair); 5449 int value; 5450 audio_format_t reqFormat = mFormat; 5451 uint32_t samplingRate = mSampleRate; 5452 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5453 5454 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5455 // channel count change can be requested. Do we mandate the first client defines the 5456 // HAL sampling rate and channel count or do we allow changes on the fly? 5457 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5458 samplingRate = value; 5459 reconfig = true; 5460 } 5461 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5462 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5463 status = BAD_VALUE; 5464 } else { 5465 reqFormat = (audio_format_t) value; 5466 reconfig = true; 5467 } 5468 } 5469 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5470 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5471 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5472 status = BAD_VALUE; 5473 } else { 5474 channelMask = mask; 5475 reconfig = true; 5476 } 5477 } 5478 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5479 // do not accept frame count changes if tracks are open as the track buffer 5480 // size depends on frame count and correct behavior would not be guaranteed 5481 // if frame count is changed after track creation 5482 if (mActiveTracks.size() > 0) { 5483 status = INVALID_OPERATION; 5484 } else { 5485 reconfig = true; 5486 } 5487 } 5488 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5489 // forward device change to effects that have requested to be 5490 // aware of attached audio device. 5491 for (size_t i = 0; i < mEffectChains.size(); i++) { 5492 mEffectChains[i]->setDevice_l(value); 5493 } 5494 5495 // store input device and output device but do not forward output device to audio HAL. 5496 // Note that status is ignored by the caller for output device 5497 // (see AudioFlinger::setParameters() 5498 if (audio_is_output_devices(value)) { 5499 mOutDevice = value; 5500 status = BAD_VALUE; 5501 } else { 5502 mInDevice = value; 5503 // disable AEC and NS if the device is a BT SCO headset supporting those 5504 // pre processings 5505 if (mTracks.size() > 0) { 5506 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5507 mAudioFlinger->btNrecIsOff(); 5508 for (size_t i = 0; i < mTracks.size(); i++) { 5509 sp<RecordTrack> track = mTracks[i]; 5510 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5511 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5512 } 5513 } 5514 } 5515 } 5516 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5517 mAudioSource != (audio_source_t)value) { 5518 // forward device change to effects that have requested to be 5519 // aware of attached audio device. 5520 for (size_t i = 0; i < mEffectChains.size(); i++) { 5521 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5522 } 5523 mAudioSource = (audio_source_t)value; 5524 } 5525 5526 if (status == NO_ERROR) { 5527 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5528 keyValuePair.string()); 5529 if (status == INVALID_OPERATION) { 5530 inputStandBy(); 5531 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5532 keyValuePair.string()); 5533 } 5534 if (reconfig) { 5535 if (status == BAD_VALUE && 5536 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5537 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5538 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5539 <= (2 * samplingRate)) && 5540 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5541 <= FCC_2 && 5542 (channelMask == AUDIO_CHANNEL_IN_MONO || 5543 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5544 status = NO_ERROR; 5545 } 5546 if (status == NO_ERROR) { 5547 readInputParameters_l(); 5548 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5549 } 5550 } 5551 } 5552 5553 mNewParameters.removeAt(0); 5554 5555 mParamStatus = status; 5556 mParamCond.signal(); 5557 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5558 // already timed out waiting for the status and will never signal the condition. 5559 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5560 } 5561 return reconfig; 5562} 5563 5564String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5565{ 5566 Mutex::Autolock _l(mLock); 5567 if (initCheck() != NO_ERROR) { 5568 return String8(); 5569 } 5570 5571 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5572 const String8 out_s8(s); 5573 free(s); 5574 return out_s8; 5575} 5576 5577void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5578 AudioSystem::OutputDescriptor desc; 5579 const void *param2 = NULL; 5580 5581 switch (event) { 5582 case AudioSystem::INPUT_OPENED: 5583 case AudioSystem::INPUT_CONFIG_CHANGED: 5584 desc.channelMask = mChannelMask; 5585 desc.samplingRate = mSampleRate; 5586 desc.format = mFormat; 5587 desc.frameCount = mFrameCount; 5588 desc.latency = 0; 5589 param2 = &desc; 5590 break; 5591 5592 case AudioSystem::INPUT_CLOSED: 5593 default: 5594 break; 5595 } 5596 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5597} 5598 5599void AudioFlinger::RecordThread::readInputParameters_l() 5600{ 5601 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5602 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5603 mChannelCount = popcount(mChannelMask); 5604 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5605 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5606 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5607 } 5608 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5609 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5610 mFrameCount = mBufferSize / mFrameSize; 5611 // This is the formula for calculating the temporary buffer size. 5612 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5613 // 1 full output buffer, regardless of the alignment of the available input. 5614 // The value is somewhat arbitrary, and could probably be even larger. 5615 // A larger value should allow more old data to be read after a track calls start(), 5616 // without increasing latency. 5617 mRsmpInFrames = mFrameCount * 7; 5618 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5619 delete[] mRsmpInBuffer; 5620 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5621 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5622 5623 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5624 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5625} 5626 5627uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5628{ 5629 Mutex::Autolock _l(mLock); 5630 if (initCheck() != NO_ERROR) { 5631 return 0; 5632 } 5633 5634 return mInput->stream->get_input_frames_lost(mInput->stream); 5635} 5636 5637uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5638{ 5639 Mutex::Autolock _l(mLock); 5640 uint32_t result = 0; 5641 if (getEffectChain_l(sessionId) != 0) { 5642 result = EFFECT_SESSION; 5643 } 5644 5645 for (size_t i = 0; i < mTracks.size(); ++i) { 5646 if (sessionId == mTracks[i]->sessionId()) { 5647 result |= TRACK_SESSION; 5648 break; 5649 } 5650 } 5651 5652 return result; 5653} 5654 5655KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5656{ 5657 KeyedVector<int, bool> ids; 5658 Mutex::Autolock _l(mLock); 5659 for (size_t j = 0; j < mTracks.size(); ++j) { 5660 sp<RecordThread::RecordTrack> track = mTracks[j]; 5661 int sessionId = track->sessionId(); 5662 if (ids.indexOfKey(sessionId) < 0) { 5663 ids.add(sessionId, true); 5664 } 5665 } 5666 return ids; 5667} 5668 5669AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5670{ 5671 Mutex::Autolock _l(mLock); 5672 AudioStreamIn *input = mInput; 5673 mInput = NULL; 5674 return input; 5675} 5676 5677// this method must always be called either with ThreadBase mLock held or inside the thread loop 5678audio_stream_t* AudioFlinger::RecordThread::stream() const 5679{ 5680 if (mInput == NULL) { 5681 return NULL; 5682 } 5683 return &mInput->stream->common; 5684} 5685 5686status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5687{ 5688 // only one chain per input thread 5689 if (mEffectChains.size() != 0) { 5690 return INVALID_OPERATION; 5691 } 5692 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5693 5694 chain->setInBuffer(NULL); 5695 chain->setOutBuffer(NULL); 5696 5697 checkSuspendOnAddEffectChain_l(chain); 5698 5699 mEffectChains.add(chain); 5700 5701 return NO_ERROR; 5702} 5703 5704size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5705{ 5706 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5707 ALOGW_IF(mEffectChains.size() != 1, 5708 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5709 chain.get(), mEffectChains.size(), this); 5710 if (mEffectChains.size() == 1) { 5711 mEffectChains.removeAt(0); 5712 } 5713 return 0; 5714} 5715 5716}; // namespace android 5717