Threads.cpp revision e8a1ced4da17dc6c07803dc2af8060f62a8389c1
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 in sendConfigEvent_l() for a status to be received 101static const nsecs_t kConfigEventTimeoutNs = 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// See Thread::readOnlyHeap(). 147// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 148// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 149// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 150static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 151 152// ---------------------------------------------------------------------------- 153 154#ifdef ADD_BATTERY_DATA 155// To collect the amplifier usage 156static void addBatteryData(uint32_t params) { 157 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 158 if (service == NULL) { 159 // it already logged 160 return; 161 } 162 163 service->addBatteryData(params); 164} 165#endif 166 167 168// ---------------------------------------------------------------------------- 169// CPU Stats 170// ---------------------------------------------------------------------------- 171 172class CpuStats { 173public: 174 CpuStats(); 175 void sample(const String8 &title); 176#ifdef DEBUG_CPU_USAGE 177private: 178 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 179 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 180 181 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 182 183 int mCpuNum; // thread's current CPU number 184 int mCpukHz; // frequency of thread's current CPU in kHz 185#endif 186}; 187 188CpuStats::CpuStats() 189#ifdef DEBUG_CPU_USAGE 190 : mCpuNum(-1), mCpukHz(-1) 191#endif 192{ 193} 194 195void CpuStats::sample(const String8 &title 196#ifndef DEBUG_CPU_USAGE 197 __unused 198#endif 199 ) { 200#ifdef DEBUG_CPU_USAGE 201 // get current thread's delta CPU time in wall clock ns 202 double wcNs; 203 bool valid = mCpuUsage.sampleAndEnable(wcNs); 204 205 // record sample for wall clock statistics 206 if (valid) { 207 mWcStats.sample(wcNs); 208 } 209 210 // get the current CPU number 211 int cpuNum = sched_getcpu(); 212 213 // get the current CPU frequency in kHz 214 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 215 216 // check if either CPU number or frequency changed 217 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 218 mCpuNum = cpuNum; 219 mCpukHz = cpukHz; 220 // ignore sample for purposes of cycles 221 valid = false; 222 } 223 224 // if no change in CPU number or frequency, then record sample for cycle statistics 225 if (valid && mCpukHz > 0) { 226 double cycles = wcNs * cpukHz * 0.000001; 227 mHzStats.sample(cycles); 228 } 229 230 unsigned n = mWcStats.n(); 231 // mCpuUsage.elapsed() is expensive, so don't call it every loop 232 if ((n & 127) == 1) { 233 long long elapsed = mCpuUsage.elapsed(); 234 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 235 double perLoop = elapsed / (double) n; 236 double perLoop100 = perLoop * 0.01; 237 double perLoop1k = perLoop * 0.001; 238 double mean = mWcStats.mean(); 239 double stddev = mWcStats.stddev(); 240 double minimum = mWcStats.minimum(); 241 double maximum = mWcStats.maximum(); 242 double meanCycles = mHzStats.mean(); 243 double stddevCycles = mHzStats.stddev(); 244 double minCycles = mHzStats.minimum(); 245 double maxCycles = mHzStats.maximum(); 246 mCpuUsage.resetElapsed(); 247 mWcStats.reset(); 248 mHzStats.reset(); 249 ALOGD("CPU usage for %s over past %.1f secs\n" 250 " (%u mixer loops at %.1f mean ms per loop):\n" 251 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 252 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 253 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 254 title.string(), 255 elapsed * .000000001, n, perLoop * .000001, 256 mean * .001, 257 stddev * .001, 258 minimum * .001, 259 maximum * .001, 260 mean / perLoop100, 261 stddev / perLoop100, 262 minimum / perLoop100, 263 maximum / perLoop100, 264 meanCycles / perLoop1k, 265 stddevCycles / perLoop1k, 266 minCycles / perLoop1k, 267 maxCycles / perLoop1k); 268 269 } 270 } 271#endif 272}; 273 274// ---------------------------------------------------------------------------- 275// ThreadBase 276// ---------------------------------------------------------------------------- 277 278AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 279 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 280 : Thread(false /*canCallJava*/), 281 mType(type), 282 mAudioFlinger(audioFlinger), 283 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 284 // are set by PlaybackThread::readOutputParameters_l() or 285 // RecordThread::readInputParameters_l() 286 //FIXME: mStandby should be true here. Is this some kind of hack? 287 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 288 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 289 // mName will be set by concrete (non-virtual) subclass 290 mDeathRecipient(new PMDeathRecipient(this)) 291{ 292} 293 294AudioFlinger::ThreadBase::~ThreadBase() 295{ 296 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 297 mConfigEvents.clear(); 298 299 // do not lock the mutex in destructor 300 releaseWakeLock_l(); 301 if (mPowerManager != 0) { 302 sp<IBinder> binder = mPowerManager->asBinder(); 303 binder->unlinkToDeath(mDeathRecipient); 304 } 305} 306 307status_t AudioFlinger::ThreadBase::readyToRun() 308{ 309 status_t status = initCheck(); 310 if (status == NO_ERROR) { 311 ALOGI("AudioFlinger's thread %p ready to run", this); 312 } else { 313 ALOGE("No working audio driver found."); 314 } 315 return status; 316} 317 318void AudioFlinger::ThreadBase::exit() 319{ 320 ALOGV("ThreadBase::exit"); 321 // do any cleanup required for exit to succeed 322 preExit(); 323 { 324 // This lock prevents the following race in thread (uniprocessor for illustration): 325 // if (!exitPending()) { 326 // // context switch from here to exit() 327 // // exit() calls requestExit(), what exitPending() observes 328 // // exit() calls signal(), which is dropped since no waiters 329 // // context switch back from exit() to here 330 // mWaitWorkCV.wait(...); 331 // // now thread is hung 332 // } 333 AutoMutex lock(mLock); 334 requestExit(); 335 mWaitWorkCV.broadcast(); 336 } 337 // When Thread::requestExitAndWait is made virtual and this method is renamed to 338 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 339 requestExitAndWait(); 340} 341 342status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 343{ 344 status_t status; 345 346 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 347 Mutex::Autolock _l(mLock); 348 349 return sendSetParameterConfigEvent_l(keyValuePairs); 350} 351 352// sendConfigEvent_l() must be called with ThreadBase::mLock held 353// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 354status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 355{ 356 status_t status = NO_ERROR; 357 358 mConfigEvents.add(event); 359 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 360 mWaitWorkCV.signal(); 361 mLock.unlock(); 362 { 363 Mutex::Autolock _l(event->mLock); 364 while (event->mWaitStatus) { 365 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 366 event->mStatus = TIMED_OUT; 367 event->mWaitStatus = false; 368 } 369 } 370 status = event->mStatus; 371 } 372 mLock.lock(); 373 return status; 374} 375 376void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 377{ 378 Mutex::Autolock _l(mLock); 379 sendIoConfigEvent_l(event, param); 380} 381 382// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 383void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 384{ 385 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 386 sendConfigEvent_l(configEvent); 387} 388 389// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 390void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 391{ 392 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 393 sendConfigEvent_l(configEvent); 394} 395 396// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 397status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 398{ 399 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 400 return sendConfigEvent_l(configEvent); 401} 402 403// post condition: mConfigEvents.isEmpty() 404void AudioFlinger::ThreadBase::processConfigEvents_l() 405{ 406 bool configChanged = false; 407 408 while (!mConfigEvents.isEmpty()) { 409 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 410 sp<ConfigEvent> event = mConfigEvents[0]; 411 mConfigEvents.removeAt(0); 412 switch (event->mType) { 413 case CFG_EVENT_PRIO: { 414 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 415 // FIXME Need to understand why this has to be done asynchronously 416 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 417 true /*asynchronous*/); 418 if (err != 0) { 419 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 420 data->mPrio, data->mPid, data->mTid, err); 421 } 422 } break; 423 case CFG_EVENT_IO: { 424 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 425 audioConfigChanged(data->mEvent, data->mParam); 426 } break; 427 case CFG_EVENT_SET_PARAMETER: { 428 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 429 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 430 configChanged = true; 431 } 432 } break; 433 default: 434 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 435 break; 436 } 437 { 438 Mutex::Autolock _l(event->mLock); 439 if (event->mWaitStatus) { 440 event->mWaitStatus = false; 441 event->mCond.signal(); 442 } 443 } 444 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 445 } 446 447 if (configChanged) { 448 cacheParameters_l(); 449 } 450} 451 452String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 453 String8 s; 454 if (output) { 455 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 456 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 457 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 458 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 459 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 460 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 461 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 462 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 463 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 464 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 465 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 466 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 467 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 468 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 469 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 470 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 471 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 472 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 473 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 474 } else { 475 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 476 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 477 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 478 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 479 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 480 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 481 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 482 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 483 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 484 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 485 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 486 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 487 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 488 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 489 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 490 } 491 int len = s.length(); 492 if (s.length() > 2) { 493 char *str = s.lockBuffer(len); 494 s.unlockBuffer(len - 2); 495 } 496 return s; 497} 498 499void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 500{ 501 const size_t SIZE = 256; 502 char buffer[SIZE]; 503 String8 result; 504 505 bool locked = AudioFlinger::dumpTryLock(mLock); 506 if (!locked) { 507 fdprintf(fd, "thread %p maybe dead locked\n", this); 508 } 509 510 fdprintf(fd, " I/O handle: %d\n", mId); 511 fdprintf(fd, " TID: %d\n", getTid()); 512 fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 513 fdprintf(fd, " Sample rate: %u\n", mSampleRate); 514 fdprintf(fd, " HAL frame count: %zu\n", mFrameCount); 515 fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 516 fdprintf(fd, " Channel Count: %u\n", mChannelCount); 517 fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 518 channelMaskToString(mChannelMask, mType != RECORD).string()); 519 fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 520 fdprintf(fd, " Frame size: %zu\n", mFrameSize); 521 fdprintf(fd, " Pending config events:"); 522 size_t numConfig = mConfigEvents.size(); 523 if (numConfig) { 524 for (size_t i = 0; i < numConfig; i++) { 525 mConfigEvents[i]->dump(buffer, SIZE); 526 fdprintf(fd, "\n %s", buffer); 527 } 528 fdprintf(fd, "\n"); 529 } else { 530 fdprintf(fd, " none\n"); 531 } 532 533 if (locked) { 534 mLock.unlock(); 535 } 536} 537 538void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 539{ 540 const size_t SIZE = 256; 541 char buffer[SIZE]; 542 String8 result; 543 544 size_t numEffectChains = mEffectChains.size(); 545 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 546 write(fd, buffer, strlen(buffer)); 547 548 for (size_t i = 0; i < numEffectChains; ++i) { 549 sp<EffectChain> chain = mEffectChains[i]; 550 if (chain != 0) { 551 chain->dump(fd, args); 552 } 553 } 554} 555 556void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 557{ 558 Mutex::Autolock _l(mLock); 559 acquireWakeLock_l(uid); 560} 561 562String16 AudioFlinger::ThreadBase::getWakeLockTag() 563{ 564 switch (mType) { 565 case MIXER: 566 return String16("AudioMix"); 567 case DIRECT: 568 return String16("AudioDirectOut"); 569 case DUPLICATING: 570 return String16("AudioDup"); 571 case RECORD: 572 return String16("AudioIn"); 573 case OFFLOAD: 574 return String16("AudioOffload"); 575 default: 576 ALOG_ASSERT(false); 577 return String16("AudioUnknown"); 578 } 579} 580 581void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 582{ 583 getPowerManager_l(); 584 if (mPowerManager != 0) { 585 sp<IBinder> binder = new BBinder(); 586 status_t status; 587 if (uid >= 0) { 588 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 589 binder, 590 getWakeLockTag(), 591 String16("media"), 592 uid); 593 } else { 594 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 595 binder, 596 getWakeLockTag(), 597 String16("media")); 598 } 599 if (status == NO_ERROR) { 600 mWakeLockToken = binder; 601 } 602 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 603 } 604} 605 606void AudioFlinger::ThreadBase::releaseWakeLock() 607{ 608 Mutex::Autolock _l(mLock); 609 releaseWakeLock_l(); 610} 611 612void AudioFlinger::ThreadBase::releaseWakeLock_l() 613{ 614 if (mWakeLockToken != 0) { 615 ALOGV("releaseWakeLock_l() %s", mName); 616 if (mPowerManager != 0) { 617 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 618 } 619 mWakeLockToken.clear(); 620 } 621} 622 623void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 624 Mutex::Autolock _l(mLock); 625 updateWakeLockUids_l(uids); 626} 627 628void AudioFlinger::ThreadBase::getPowerManager_l() { 629 630 if (mPowerManager == 0) { 631 // use checkService() to avoid blocking if power service is not up yet 632 sp<IBinder> binder = 633 defaultServiceManager()->checkService(String16("power")); 634 if (binder == 0) { 635 ALOGW("Thread %s cannot connect to the power manager service", mName); 636 } else { 637 mPowerManager = interface_cast<IPowerManager>(binder); 638 binder->linkToDeath(mDeathRecipient); 639 } 640 } 641} 642 643void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 644 645 getPowerManager_l(); 646 if (mWakeLockToken == NULL) { 647 ALOGE("no wake lock to update!"); 648 return; 649 } 650 if (mPowerManager != 0) { 651 sp<IBinder> binder = new BBinder(); 652 status_t status; 653 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 654 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 655 } 656} 657 658void AudioFlinger::ThreadBase::clearPowerManager() 659{ 660 Mutex::Autolock _l(mLock); 661 releaseWakeLock_l(); 662 mPowerManager.clear(); 663} 664 665void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 666{ 667 sp<ThreadBase> thread = mThread.promote(); 668 if (thread != 0) { 669 thread->clearPowerManager(); 670 } 671 ALOGW("power manager service died !!!"); 672} 673 674void AudioFlinger::ThreadBase::setEffectSuspended( 675 const effect_uuid_t *type, bool suspend, int sessionId) 676{ 677 Mutex::Autolock _l(mLock); 678 setEffectSuspended_l(type, suspend, sessionId); 679} 680 681void AudioFlinger::ThreadBase::setEffectSuspended_l( 682 const effect_uuid_t *type, bool suspend, int sessionId) 683{ 684 sp<EffectChain> chain = getEffectChain_l(sessionId); 685 if (chain != 0) { 686 if (type != NULL) { 687 chain->setEffectSuspended_l(type, suspend); 688 } else { 689 chain->setEffectSuspendedAll_l(suspend); 690 } 691 } 692 693 updateSuspendedSessions_l(type, suspend, sessionId); 694} 695 696void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 697{ 698 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 699 if (index < 0) { 700 return; 701 } 702 703 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 704 mSuspendedSessions.valueAt(index); 705 706 for (size_t i = 0; i < sessionEffects.size(); i++) { 707 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 708 for (int j = 0; j < desc->mRefCount; j++) { 709 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 710 chain->setEffectSuspendedAll_l(true); 711 } else { 712 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 713 desc->mType.timeLow); 714 chain->setEffectSuspended_l(&desc->mType, true); 715 } 716 } 717 } 718} 719 720void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 721 bool suspend, 722 int sessionId) 723{ 724 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 725 726 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 727 728 if (suspend) { 729 if (index >= 0) { 730 sessionEffects = mSuspendedSessions.valueAt(index); 731 } else { 732 mSuspendedSessions.add(sessionId, sessionEffects); 733 } 734 } else { 735 if (index < 0) { 736 return; 737 } 738 sessionEffects = mSuspendedSessions.valueAt(index); 739 } 740 741 742 int key = EffectChain::kKeyForSuspendAll; 743 if (type != NULL) { 744 key = type->timeLow; 745 } 746 index = sessionEffects.indexOfKey(key); 747 748 sp<SuspendedSessionDesc> desc; 749 if (suspend) { 750 if (index >= 0) { 751 desc = sessionEffects.valueAt(index); 752 } else { 753 desc = new SuspendedSessionDesc(); 754 if (type != NULL) { 755 desc->mType = *type; 756 } 757 sessionEffects.add(key, desc); 758 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 759 } 760 desc->mRefCount++; 761 } else { 762 if (index < 0) { 763 return; 764 } 765 desc = sessionEffects.valueAt(index); 766 if (--desc->mRefCount == 0) { 767 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 768 sessionEffects.removeItemsAt(index); 769 if (sessionEffects.isEmpty()) { 770 ALOGV("updateSuspendedSessions_l() restore removing session %d", 771 sessionId); 772 mSuspendedSessions.removeItem(sessionId); 773 } 774 } 775 } 776 if (!sessionEffects.isEmpty()) { 777 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 778 } 779} 780 781void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 782 bool enabled, 783 int sessionId) 784{ 785 Mutex::Autolock _l(mLock); 786 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 787} 788 789void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 790 bool enabled, 791 int sessionId) 792{ 793 if (mType != RECORD) { 794 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 795 // another session. This gives the priority to well behaved effect control panels 796 // and applications not using global effects. 797 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 798 // global effects 799 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 800 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 801 } 802 } 803 804 sp<EffectChain> chain = getEffectChain_l(sessionId); 805 if (chain != 0) { 806 chain->checkSuspendOnEffectEnabled(effect, enabled); 807 } 808} 809 810// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 811sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 812 const sp<AudioFlinger::Client>& client, 813 const sp<IEffectClient>& effectClient, 814 int32_t priority, 815 int sessionId, 816 effect_descriptor_t *desc, 817 int *enabled, 818 status_t *status) 819{ 820 sp<EffectModule> effect; 821 sp<EffectHandle> handle; 822 status_t lStatus; 823 sp<EffectChain> chain; 824 bool chainCreated = false; 825 bool effectCreated = false; 826 bool effectRegistered = false; 827 828 lStatus = initCheck(); 829 if (lStatus != NO_ERROR) { 830 ALOGW("createEffect_l() Audio driver not initialized."); 831 goto Exit; 832 } 833 834 // Reject any effect on Direct output threads for now, since the format of 835 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 836 if (mType == DIRECT) { 837 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 838 desc->name, mName); 839 lStatus = BAD_VALUE; 840 goto Exit; 841 } 842 843 // Allow global effects only on offloaded and mixer threads 844 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 845 switch (mType) { 846 case MIXER: 847 case OFFLOAD: 848 break; 849 case DIRECT: 850 case DUPLICATING: 851 case RECORD: 852 default: 853 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 854 lStatus = BAD_VALUE; 855 goto Exit; 856 } 857 } 858 859 // Only Pre processor effects are allowed on input threads and only on input threads 860 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 861 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 862 desc->name, desc->flags, mType); 863 lStatus = BAD_VALUE; 864 goto Exit; 865 } 866 867 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 868 869 { // scope for mLock 870 Mutex::Autolock _l(mLock); 871 872 // check for existing effect chain with the requested audio session 873 chain = getEffectChain_l(sessionId); 874 if (chain == 0) { 875 // create a new chain for this session 876 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 877 chain = new EffectChain(this, sessionId); 878 addEffectChain_l(chain); 879 chain->setStrategy(getStrategyForSession_l(sessionId)); 880 chainCreated = true; 881 } else { 882 effect = chain->getEffectFromDesc_l(desc); 883 } 884 885 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 886 887 if (effect == 0) { 888 int id = mAudioFlinger->nextUniqueId(); 889 // Check CPU and memory usage 890 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 891 if (lStatus != NO_ERROR) { 892 goto Exit; 893 } 894 effectRegistered = true; 895 // create a new effect module if none present in the chain 896 effect = new EffectModule(this, chain, desc, id, sessionId); 897 lStatus = effect->status(); 898 if (lStatus != NO_ERROR) { 899 goto Exit; 900 } 901 effect->setOffloaded(mType == OFFLOAD, mId); 902 903 lStatus = chain->addEffect_l(effect); 904 if (lStatus != NO_ERROR) { 905 goto Exit; 906 } 907 effectCreated = true; 908 909 effect->setDevice(mOutDevice); 910 effect->setDevice(mInDevice); 911 effect->setMode(mAudioFlinger->getMode()); 912 effect->setAudioSource(mAudioSource); 913 } 914 // create effect handle and connect it to effect module 915 handle = new EffectHandle(effect, client, effectClient, priority); 916 lStatus = handle->initCheck(); 917 if (lStatus == OK) { 918 lStatus = effect->addHandle(handle.get()); 919 } 920 if (enabled != NULL) { 921 *enabled = (int)effect->isEnabled(); 922 } 923 } 924 925Exit: 926 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 927 Mutex::Autolock _l(mLock); 928 if (effectCreated) { 929 chain->removeEffect_l(effect); 930 } 931 if (effectRegistered) { 932 AudioSystem::unregisterEffect(effect->id()); 933 } 934 if (chainCreated) { 935 removeEffectChain_l(chain); 936 } 937 handle.clear(); 938 } 939 940 *status = lStatus; 941 return handle; 942} 943 944sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 945{ 946 Mutex::Autolock _l(mLock); 947 return getEffect_l(sessionId, effectId); 948} 949 950sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 951{ 952 sp<EffectChain> chain = getEffectChain_l(sessionId); 953 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 954} 955 956// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 957// PlaybackThread::mLock held 958status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 959{ 960 // check for existing effect chain with the requested audio session 961 int sessionId = effect->sessionId(); 962 sp<EffectChain> chain = getEffectChain_l(sessionId); 963 bool chainCreated = false; 964 965 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 966 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 967 this, effect->desc().name, effect->desc().flags); 968 969 if (chain == 0) { 970 // create a new chain for this session 971 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 972 chain = new EffectChain(this, sessionId); 973 addEffectChain_l(chain); 974 chain->setStrategy(getStrategyForSession_l(sessionId)); 975 chainCreated = true; 976 } 977 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 978 979 if (chain->getEffectFromId_l(effect->id()) != 0) { 980 ALOGW("addEffect_l() %p effect %s already present in chain %p", 981 this, effect->desc().name, chain.get()); 982 return BAD_VALUE; 983 } 984 985 effect->setOffloaded(mType == OFFLOAD, mId); 986 987 status_t status = chain->addEffect_l(effect); 988 if (status != NO_ERROR) { 989 if (chainCreated) { 990 removeEffectChain_l(chain); 991 } 992 return status; 993 } 994 995 effect->setDevice(mOutDevice); 996 effect->setDevice(mInDevice); 997 effect->setMode(mAudioFlinger->getMode()); 998 effect->setAudioSource(mAudioSource); 999 return NO_ERROR; 1000} 1001 1002void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1003 1004 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1005 effect_descriptor_t desc = effect->desc(); 1006 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1007 detachAuxEffect_l(effect->id()); 1008 } 1009 1010 sp<EffectChain> chain = effect->chain().promote(); 1011 if (chain != 0) { 1012 // remove effect chain if removing last effect 1013 if (chain->removeEffect_l(effect) == 0) { 1014 removeEffectChain_l(chain); 1015 } 1016 } else { 1017 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1018 } 1019} 1020 1021void AudioFlinger::ThreadBase::lockEffectChains_l( 1022 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1023{ 1024 effectChains = mEffectChains; 1025 for (size_t i = 0; i < mEffectChains.size(); i++) { 1026 mEffectChains[i]->lock(); 1027 } 1028} 1029 1030void AudioFlinger::ThreadBase::unlockEffectChains( 1031 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1032{ 1033 for (size_t i = 0; i < effectChains.size(); i++) { 1034 effectChains[i]->unlock(); 1035 } 1036} 1037 1038sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1039{ 1040 Mutex::Autolock _l(mLock); 1041 return getEffectChain_l(sessionId); 1042} 1043 1044sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1045{ 1046 size_t size = mEffectChains.size(); 1047 for (size_t i = 0; i < size; i++) { 1048 if (mEffectChains[i]->sessionId() == sessionId) { 1049 return mEffectChains[i]; 1050 } 1051 } 1052 return 0; 1053} 1054 1055void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1056{ 1057 Mutex::Autolock _l(mLock); 1058 size_t size = mEffectChains.size(); 1059 for (size_t i = 0; i < size; i++) { 1060 mEffectChains[i]->setMode_l(mode); 1061 } 1062} 1063 1064void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1065 EffectHandle *handle, 1066 bool unpinIfLast) { 1067 1068 Mutex::Autolock _l(mLock); 1069 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1070 // delete the effect module if removing last handle on it 1071 if (effect->removeHandle(handle) == 0) { 1072 if (!effect->isPinned() || unpinIfLast) { 1073 removeEffect_l(effect); 1074 AudioSystem::unregisterEffect(effect->id()); 1075 } 1076 } 1077} 1078 1079// ---------------------------------------------------------------------------- 1080// Playback 1081// ---------------------------------------------------------------------------- 1082 1083AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1084 AudioStreamOut* output, 1085 audio_io_handle_t id, 1086 audio_devices_t device, 1087 type_t type) 1088 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1089 mNormalFrameCount(0), mSinkBuffer(NULL), 1090 mMixerBufferEnabled(false), 1091 mMixerBuffer(NULL), 1092 mMixerBufferSize(0), 1093 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1094 mMixerBufferValid(false), 1095 mEffectBufferEnabled(false), 1096 mEffectBuffer(NULL), 1097 mEffectBufferSize(0), 1098 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1099 mEffectBufferValid(false), 1100 mSuspended(0), mBytesWritten(0), 1101 mActiveTracksGeneration(0), 1102 // mStreamTypes[] initialized in constructor body 1103 mOutput(output), 1104 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1105 mMixerStatus(MIXER_IDLE), 1106 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1107 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1108 mBytesRemaining(0), 1109 mCurrentWriteLength(0), 1110 mUseAsyncWrite(false), 1111 mWriteAckSequence(0), 1112 mDrainSequence(0), 1113 mSignalPending(false), 1114 mScreenState(AudioFlinger::mScreenState), 1115 // index 0 is reserved for normal mixer's submix 1116 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1117 // mLatchD, mLatchQ, 1118 mLatchDValid(false), mLatchQValid(false) 1119{ 1120 snprintf(mName, kNameLength, "AudioOut_%X", id); 1121 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1122 1123 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1124 // it would be safer to explicitly pass initial masterVolume/masterMute as 1125 // parameter. 1126 // 1127 // If the HAL we are using has support for master volume or master mute, 1128 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1129 // and the mute set to false). 1130 mMasterVolume = audioFlinger->masterVolume_l(); 1131 mMasterMute = audioFlinger->masterMute_l(); 1132 if (mOutput && mOutput->audioHwDev) { 1133 if (mOutput->audioHwDev->canSetMasterVolume()) { 1134 mMasterVolume = 1.0; 1135 } 1136 1137 if (mOutput->audioHwDev->canSetMasterMute()) { 1138 mMasterMute = false; 1139 } 1140 } 1141 1142 readOutputParameters_l(); 1143 1144 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1145 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1146 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1147 stream = (audio_stream_type_t) (stream + 1)) { 1148 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1149 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1150 } 1151 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1152 // because mAudioFlinger doesn't have one to copy from 1153} 1154 1155AudioFlinger::PlaybackThread::~PlaybackThread() 1156{ 1157 mAudioFlinger->unregisterWriter(mNBLogWriter); 1158 free(mSinkBuffer); 1159 free(mMixerBuffer); 1160 free(mEffectBuffer); 1161} 1162 1163void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1164{ 1165 dumpInternals(fd, args); 1166 dumpTracks(fd, args); 1167 dumpEffectChains(fd, args); 1168} 1169 1170void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1171{ 1172 const size_t SIZE = 256; 1173 char buffer[SIZE]; 1174 String8 result; 1175 1176 result.appendFormat(" Stream volumes in dB: "); 1177 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1178 const stream_type_t *st = &mStreamTypes[i]; 1179 if (i > 0) { 1180 result.appendFormat(", "); 1181 } 1182 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1183 if (st->mute) { 1184 result.append("M"); 1185 } 1186 } 1187 result.append("\n"); 1188 write(fd, result.string(), result.length()); 1189 result.clear(); 1190 1191 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1192 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1193 fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1194 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1195 1196 size_t numtracks = mTracks.size(); 1197 size_t numactive = mActiveTracks.size(); 1198 fdprintf(fd, " %d Tracks", numtracks); 1199 size_t numactiveseen = 0; 1200 if (numtracks) { 1201 fdprintf(fd, " of which %d are active\n", numactive); 1202 Track::appendDumpHeader(result); 1203 for (size_t i = 0; i < numtracks; ++i) { 1204 sp<Track> track = mTracks[i]; 1205 if (track != 0) { 1206 bool active = mActiveTracks.indexOf(track) >= 0; 1207 if (active) { 1208 numactiveseen++; 1209 } 1210 track->dump(buffer, SIZE, active); 1211 result.append(buffer); 1212 } 1213 } 1214 } else { 1215 result.append("\n"); 1216 } 1217 if (numactiveseen != numactive) { 1218 // some tracks in the active list were not in the tracks list 1219 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1220 " not in the track list\n"); 1221 result.append(buffer); 1222 Track::appendDumpHeader(result); 1223 for (size_t i = 0; i < numactive; ++i) { 1224 sp<Track> track = mActiveTracks[i].promote(); 1225 if (track != 0 && mTracks.indexOf(track) < 0) { 1226 track->dump(buffer, SIZE, true); 1227 result.append(buffer); 1228 } 1229 } 1230 } 1231 1232 write(fd, result.string(), result.size()); 1233 1234} 1235 1236void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1237{ 1238 fdprintf(fd, "\nOutput thread %p:\n", this); 1239 fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1240 fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1241 fdprintf(fd, " Total writes: %d\n", mNumWrites); 1242 fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1243 fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1244 fdprintf(fd, " Suspend count: %d\n", mSuspended); 1245 fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1246 fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1247 fdprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1248 fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1249 1250 dumpBase(fd, args); 1251} 1252 1253// Thread virtuals 1254 1255void AudioFlinger::PlaybackThread::onFirstRef() 1256{ 1257 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1258} 1259 1260// ThreadBase virtuals 1261void AudioFlinger::PlaybackThread::preExit() 1262{ 1263 ALOGV(" preExit()"); 1264 // FIXME this is using hard-coded strings but in the future, this functionality will be 1265 // converted to use audio HAL extensions required to support tunneling 1266 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1267} 1268 1269// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1270sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1271 const sp<AudioFlinger::Client>& client, 1272 audio_stream_type_t streamType, 1273 uint32_t sampleRate, 1274 audio_format_t format, 1275 audio_channel_mask_t channelMask, 1276 size_t *pFrameCount, 1277 const sp<IMemory>& sharedBuffer, 1278 int sessionId, 1279 IAudioFlinger::track_flags_t *flags, 1280 pid_t tid, 1281 int uid, 1282 status_t *status) 1283{ 1284 size_t frameCount = *pFrameCount; 1285 sp<Track> track; 1286 status_t lStatus; 1287 1288 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1289 1290 // client expresses a preference for FAST, but we get the final say 1291 if (*flags & IAudioFlinger::TRACK_FAST) { 1292 if ( 1293 // not timed 1294 (!isTimed) && 1295 // either of these use cases: 1296 ( 1297 // use case 1: shared buffer with any frame count 1298 ( 1299 (sharedBuffer != 0) 1300 ) || 1301 // use case 2: callback handler and frame count is default or at least as large as HAL 1302 ( 1303 (tid != -1) && 1304 ((frameCount == 0) || 1305 (frameCount >= mFrameCount)) 1306 ) 1307 ) && 1308 // PCM data 1309 audio_is_linear_pcm(format) && 1310 // mono or stereo 1311 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1312 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1313 // hardware sample rate 1314 (sampleRate == mSampleRate) && 1315 // normal mixer has an associated fast mixer 1316 hasFastMixer() && 1317 // there are sufficient fast track slots available 1318 (mFastTrackAvailMask != 0) 1319 // FIXME test that MixerThread for this fast track has a capable output HAL 1320 // FIXME add a permission test also? 1321 ) { 1322 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1323 if (frameCount == 0) { 1324 frameCount = mFrameCount * kFastTrackMultiplier; 1325 } 1326 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1327 frameCount, mFrameCount); 1328 } else { 1329 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1330 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1331 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1332 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1333 audio_is_linear_pcm(format), 1334 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1335 *flags &= ~IAudioFlinger::TRACK_FAST; 1336 // For compatibility with AudioTrack calculation, buffer depth is forced 1337 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1338 // This is probably too conservative, but legacy application code may depend on it. 1339 // If you change this calculation, also review the start threshold which is related. 1340 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1341 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1342 if (minBufCount < 2) { 1343 minBufCount = 2; 1344 } 1345 size_t minFrameCount = mNormalFrameCount * minBufCount; 1346 if (frameCount < minFrameCount) { 1347 frameCount = minFrameCount; 1348 } 1349 } 1350 } 1351 *pFrameCount = frameCount; 1352 1353 switch (mType) { 1354 1355 case DIRECT: 1356 if (audio_is_linear_pcm(format)) { 1357 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1358 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1359 "for output %p with format %#x", 1360 sampleRate, format, channelMask, mOutput, mFormat); 1361 lStatus = BAD_VALUE; 1362 goto Exit; 1363 } 1364 } 1365 break; 1366 1367 case OFFLOAD: 1368 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1369 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1370 "for output %p with format %#x", 1371 sampleRate, format, channelMask, mOutput, mFormat); 1372 lStatus = BAD_VALUE; 1373 goto Exit; 1374 } 1375 break; 1376 1377 default: 1378 if (!audio_is_linear_pcm(format)) { 1379 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1380 "for output %p with format %#x", 1381 format, mOutput, mFormat); 1382 lStatus = BAD_VALUE; 1383 goto Exit; 1384 } 1385 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1386 if (sampleRate > mSampleRate*2) { 1387 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1388 lStatus = BAD_VALUE; 1389 goto Exit; 1390 } 1391 break; 1392 1393 } 1394 1395 lStatus = initCheck(); 1396 if (lStatus != NO_ERROR) { 1397 ALOGE("createTrack_l() audio driver not initialized"); 1398 goto Exit; 1399 } 1400 1401 { // scope for mLock 1402 Mutex::Autolock _l(mLock); 1403 1404 // all tracks in same audio session must share the same routing strategy otherwise 1405 // conflicts will happen when tracks are moved from one output to another by audio policy 1406 // manager 1407 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1408 for (size_t i = 0; i < mTracks.size(); ++i) { 1409 sp<Track> t = mTracks[i]; 1410 if (t != 0 && !t->isOutputTrack()) { 1411 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1412 if (sessionId == t->sessionId() && strategy != actual) { 1413 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1414 strategy, actual); 1415 lStatus = BAD_VALUE; 1416 goto Exit; 1417 } 1418 } 1419 } 1420 1421 if (!isTimed) { 1422 track = new Track(this, client, streamType, sampleRate, format, 1423 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1424 } else { 1425 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1426 channelMask, frameCount, sharedBuffer, sessionId, uid); 1427 } 1428 1429 // new Track always returns non-NULL, 1430 // but TimedTrack::create() is a factory that could fail by returning NULL 1431 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1432 if (lStatus != NO_ERROR) { 1433 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1434 // track must be cleared from the caller as the caller has the AF lock 1435 goto Exit; 1436 } 1437 mTracks.add(track); 1438 1439 sp<EffectChain> chain = getEffectChain_l(sessionId); 1440 if (chain != 0) { 1441 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1442 track->setMainBuffer(chain->inBuffer()); 1443 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1444 chain->incTrackCnt(); 1445 } 1446 1447 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1448 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1449 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1450 // so ask activity manager to do this on our behalf 1451 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1452 } 1453 } 1454 1455 lStatus = NO_ERROR; 1456 1457Exit: 1458 *status = lStatus; 1459 return track; 1460} 1461 1462uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1463{ 1464 return latency; 1465} 1466 1467uint32_t AudioFlinger::PlaybackThread::latency() const 1468{ 1469 Mutex::Autolock _l(mLock); 1470 return latency_l(); 1471} 1472uint32_t AudioFlinger::PlaybackThread::latency_l() const 1473{ 1474 if (initCheck() == NO_ERROR) { 1475 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1476 } else { 1477 return 0; 1478 } 1479} 1480 1481void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1482{ 1483 Mutex::Autolock _l(mLock); 1484 // Don't apply master volume in SW if our HAL can do it for us. 1485 if (mOutput && mOutput->audioHwDev && 1486 mOutput->audioHwDev->canSetMasterVolume()) { 1487 mMasterVolume = 1.0; 1488 } else { 1489 mMasterVolume = value; 1490 } 1491} 1492 1493void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1494{ 1495 Mutex::Autolock _l(mLock); 1496 // Don't apply master mute in SW if our HAL can do it for us. 1497 if (mOutput && mOutput->audioHwDev && 1498 mOutput->audioHwDev->canSetMasterMute()) { 1499 mMasterMute = false; 1500 } else { 1501 mMasterMute = muted; 1502 } 1503} 1504 1505void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1506{ 1507 Mutex::Autolock _l(mLock); 1508 mStreamTypes[stream].volume = value; 1509 broadcast_l(); 1510} 1511 1512void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1513{ 1514 Mutex::Autolock _l(mLock); 1515 mStreamTypes[stream].mute = muted; 1516 broadcast_l(); 1517} 1518 1519float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1520{ 1521 Mutex::Autolock _l(mLock); 1522 return mStreamTypes[stream].volume; 1523} 1524 1525// addTrack_l() must be called with ThreadBase::mLock held 1526status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1527{ 1528 status_t status = ALREADY_EXISTS; 1529 1530 // set retry count for buffer fill 1531 track->mRetryCount = kMaxTrackStartupRetries; 1532 if (mActiveTracks.indexOf(track) < 0) { 1533 // the track is newly added, make sure it fills up all its 1534 // buffers before playing. This is to ensure the client will 1535 // effectively get the latency it requested. 1536 if (!track->isOutputTrack()) { 1537 TrackBase::track_state state = track->mState; 1538 mLock.unlock(); 1539 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1540 mLock.lock(); 1541 // abort track was stopped/paused while we released the lock 1542 if (state != track->mState) { 1543 if (status == NO_ERROR) { 1544 mLock.unlock(); 1545 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1546 mLock.lock(); 1547 } 1548 return INVALID_OPERATION; 1549 } 1550 // abort if start is rejected by audio policy manager 1551 if (status != NO_ERROR) { 1552 return PERMISSION_DENIED; 1553 } 1554#ifdef ADD_BATTERY_DATA 1555 // to track the speaker usage 1556 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1557#endif 1558 } 1559 1560 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1561 track->mResetDone = false; 1562 track->mPresentationCompleteFrames = 0; 1563 mActiveTracks.add(track); 1564 mWakeLockUids.add(track->uid()); 1565 mActiveTracksGeneration++; 1566 mLatestActiveTrack = track; 1567 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1568 if (chain != 0) { 1569 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1570 track->sessionId()); 1571 chain->incActiveTrackCnt(); 1572 } 1573 1574 status = NO_ERROR; 1575 } 1576 1577 onAddNewTrack_l(); 1578 return status; 1579} 1580 1581bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1582{ 1583 track->terminate(); 1584 // active tracks are removed by threadLoop() 1585 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1586 track->mState = TrackBase::STOPPED; 1587 if (!trackActive) { 1588 removeTrack_l(track); 1589 } else if (track->isFastTrack() || track->isOffloaded()) { 1590 track->mState = TrackBase::STOPPING_1; 1591 } 1592 1593 return trackActive; 1594} 1595 1596void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1597{ 1598 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1599 mTracks.remove(track); 1600 deleteTrackName_l(track->name()); 1601 // redundant as track is about to be destroyed, for dumpsys only 1602 track->mName = -1; 1603 if (track->isFastTrack()) { 1604 int index = track->mFastIndex; 1605 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1606 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1607 mFastTrackAvailMask |= 1 << index; 1608 // redundant as track is about to be destroyed, for dumpsys only 1609 track->mFastIndex = -1; 1610 } 1611 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1612 if (chain != 0) { 1613 chain->decTrackCnt(); 1614 } 1615} 1616 1617void AudioFlinger::PlaybackThread::broadcast_l() 1618{ 1619 // Thread could be blocked waiting for async 1620 // so signal it to handle state changes immediately 1621 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1622 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1623 mSignalPending = true; 1624 mWaitWorkCV.broadcast(); 1625} 1626 1627String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1628{ 1629 Mutex::Autolock _l(mLock); 1630 if (initCheck() != NO_ERROR) { 1631 return String8(); 1632 } 1633 1634 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1635 const String8 out_s8(s); 1636 free(s); 1637 return out_s8; 1638} 1639 1640void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1641 AudioSystem::OutputDescriptor desc; 1642 void *param2 = NULL; 1643 1644 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1645 param); 1646 1647 switch (event) { 1648 case AudioSystem::OUTPUT_OPENED: 1649 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1650 desc.channelMask = mChannelMask; 1651 desc.samplingRate = mSampleRate; 1652 desc.format = mFormat; 1653 desc.frameCount = mNormalFrameCount; // FIXME see 1654 // AudioFlinger::frameCount(audio_io_handle_t) 1655 desc.latency = latency_l(); 1656 param2 = &desc; 1657 break; 1658 1659 case AudioSystem::STREAM_CONFIG_CHANGED: 1660 param2 = ¶m; 1661 case AudioSystem::OUTPUT_CLOSED: 1662 default: 1663 break; 1664 } 1665 mAudioFlinger->audioConfigChanged(event, mId, param2); 1666} 1667 1668void AudioFlinger::PlaybackThread::writeCallback() 1669{ 1670 ALOG_ASSERT(mCallbackThread != 0); 1671 mCallbackThread->resetWriteBlocked(); 1672} 1673 1674void AudioFlinger::PlaybackThread::drainCallback() 1675{ 1676 ALOG_ASSERT(mCallbackThread != 0); 1677 mCallbackThread->resetDraining(); 1678} 1679 1680void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1681{ 1682 Mutex::Autolock _l(mLock); 1683 // reject out of sequence requests 1684 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1685 mWriteAckSequence &= ~1; 1686 mWaitWorkCV.signal(); 1687 } 1688} 1689 1690void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1691{ 1692 Mutex::Autolock _l(mLock); 1693 // reject out of sequence requests 1694 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1695 mDrainSequence &= ~1; 1696 mWaitWorkCV.signal(); 1697 } 1698} 1699 1700// static 1701int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1702 void *param __unused, 1703 void *cookie) 1704{ 1705 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1706 ALOGV("asyncCallback() event %d", event); 1707 switch (event) { 1708 case STREAM_CBK_EVENT_WRITE_READY: 1709 me->writeCallback(); 1710 break; 1711 case STREAM_CBK_EVENT_DRAIN_READY: 1712 me->drainCallback(); 1713 break; 1714 default: 1715 ALOGW("asyncCallback() unknown event %d", event); 1716 break; 1717 } 1718 return 0; 1719} 1720 1721void AudioFlinger::PlaybackThread::readOutputParameters_l() 1722{ 1723 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1724 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1725 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1726 if (!audio_is_output_channel(mChannelMask)) { 1727 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1728 } 1729 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1730 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1731 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1732 } 1733 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1734 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1735 if (!audio_is_valid_format(mFormat)) { 1736 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1737 } 1738 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1739 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1740 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1741 } 1742 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1743 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1744 mFrameCount = mBufferSize / mFrameSize; 1745 if (mFrameCount & 15) { 1746 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1747 mFrameCount); 1748 } 1749 1750 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1751 (mOutput->stream->set_callback != NULL)) { 1752 if (mOutput->stream->set_callback(mOutput->stream, 1753 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1754 mUseAsyncWrite = true; 1755 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1756 } 1757 } 1758 1759 // Calculate size of normal sink buffer relative to the HAL output buffer size 1760 double multiplier = 1.0; 1761 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1762 kUseFastMixer == FastMixer_Dynamic)) { 1763 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1764 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1765 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1766 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1767 maxNormalFrameCount = maxNormalFrameCount & ~15; 1768 if (maxNormalFrameCount < minNormalFrameCount) { 1769 maxNormalFrameCount = minNormalFrameCount; 1770 } 1771 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1772 if (multiplier <= 1.0) { 1773 multiplier = 1.0; 1774 } else if (multiplier <= 2.0) { 1775 if (2 * mFrameCount <= maxNormalFrameCount) { 1776 multiplier = 2.0; 1777 } else { 1778 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1779 } 1780 } else { 1781 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1782 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1783 // track, but we sometimes have to do this to satisfy the maximum frame count 1784 // constraint) 1785 // FIXME this rounding up should not be done if no HAL SRC 1786 uint32_t truncMult = (uint32_t) multiplier; 1787 if ((truncMult & 1)) { 1788 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1789 ++truncMult; 1790 } 1791 } 1792 multiplier = (double) truncMult; 1793 } 1794 } 1795 mNormalFrameCount = multiplier * mFrameCount; 1796 // round up to nearest 16 frames to satisfy AudioMixer 1797 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1798 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1799 mNormalFrameCount); 1800 1801 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1802 // Originally this was int16_t[] array, need to remove legacy implications. 1803 free(mSinkBuffer); 1804 mSinkBuffer = NULL; 1805 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1806 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1807 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1808 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1809 1810 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1811 // drives the output. 1812 free(mMixerBuffer); 1813 mMixerBuffer = NULL; 1814 if (mMixerBufferEnabled) { 1815 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1816 mMixerBufferSize = mNormalFrameCount * mChannelCount 1817 * audio_bytes_per_sample(mMixerBufferFormat); 1818 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1819 } 1820 free(mEffectBuffer); 1821 mEffectBuffer = NULL; 1822 if (mEffectBufferEnabled) { 1823 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1824 mEffectBufferSize = mNormalFrameCount * mChannelCount 1825 * audio_bytes_per_sample(mEffectBufferFormat); 1826 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1827 } 1828 1829 // force reconfiguration of effect chains and engines to take new buffer size and audio 1830 // parameters into account 1831 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1832 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1833 // matter. 1834 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1835 Vector< sp<EffectChain> > effectChains = mEffectChains; 1836 for (size_t i = 0; i < effectChains.size(); i ++) { 1837 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1838 } 1839} 1840 1841 1842status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1843{ 1844 if (halFrames == NULL || dspFrames == NULL) { 1845 return BAD_VALUE; 1846 } 1847 Mutex::Autolock _l(mLock); 1848 if (initCheck() != NO_ERROR) { 1849 return INVALID_OPERATION; 1850 } 1851 size_t framesWritten = mBytesWritten / mFrameSize; 1852 *halFrames = framesWritten; 1853 1854 if (isSuspended()) { 1855 // return an estimation of rendered frames when the output is suspended 1856 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1857 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1858 return NO_ERROR; 1859 } else { 1860 status_t status; 1861 uint32_t frames; 1862 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1863 *dspFrames = (size_t)frames; 1864 return status; 1865 } 1866} 1867 1868uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1869{ 1870 Mutex::Autolock _l(mLock); 1871 uint32_t result = 0; 1872 if (getEffectChain_l(sessionId) != 0) { 1873 result = EFFECT_SESSION; 1874 } 1875 1876 for (size_t i = 0; i < mTracks.size(); ++i) { 1877 sp<Track> track = mTracks[i]; 1878 if (sessionId == track->sessionId() && !track->isInvalid()) { 1879 result |= TRACK_SESSION; 1880 break; 1881 } 1882 } 1883 1884 return result; 1885} 1886 1887uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1888{ 1889 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1890 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1891 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1892 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1893 } 1894 for (size_t i = 0; i < mTracks.size(); i++) { 1895 sp<Track> track = mTracks[i]; 1896 if (sessionId == track->sessionId() && !track->isInvalid()) { 1897 return AudioSystem::getStrategyForStream(track->streamType()); 1898 } 1899 } 1900 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1901} 1902 1903 1904AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1905{ 1906 Mutex::Autolock _l(mLock); 1907 return mOutput; 1908} 1909 1910AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1911{ 1912 Mutex::Autolock _l(mLock); 1913 AudioStreamOut *output = mOutput; 1914 mOutput = NULL; 1915 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1916 // must push a NULL and wait for ack 1917 mOutputSink.clear(); 1918 mPipeSink.clear(); 1919 mNormalSink.clear(); 1920 return output; 1921} 1922 1923// this method must always be called either with ThreadBase mLock held or inside the thread loop 1924audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1925{ 1926 if (mOutput == NULL) { 1927 return NULL; 1928 } 1929 return &mOutput->stream->common; 1930} 1931 1932uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1933{ 1934 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1935} 1936 1937status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1938{ 1939 if (!isValidSyncEvent(event)) { 1940 return BAD_VALUE; 1941 } 1942 1943 Mutex::Autolock _l(mLock); 1944 1945 for (size_t i = 0; i < mTracks.size(); ++i) { 1946 sp<Track> track = mTracks[i]; 1947 if (event->triggerSession() == track->sessionId()) { 1948 (void) track->setSyncEvent(event); 1949 return NO_ERROR; 1950 } 1951 } 1952 1953 return NAME_NOT_FOUND; 1954} 1955 1956bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1957{ 1958 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1959} 1960 1961void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1962 const Vector< sp<Track> >& tracksToRemove) 1963{ 1964 size_t count = tracksToRemove.size(); 1965 if (count > 0) { 1966 for (size_t i = 0 ; i < count ; i++) { 1967 const sp<Track>& track = tracksToRemove.itemAt(i); 1968 if (!track->isOutputTrack()) { 1969 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1970#ifdef ADD_BATTERY_DATA 1971 // to track the speaker usage 1972 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1973#endif 1974 if (track->isTerminated()) { 1975 AudioSystem::releaseOutput(mId); 1976 } 1977 } 1978 } 1979 } 1980} 1981 1982void AudioFlinger::PlaybackThread::checkSilentMode_l() 1983{ 1984 if (!mMasterMute) { 1985 char value[PROPERTY_VALUE_MAX]; 1986 if (property_get("ro.audio.silent", value, "0") > 0) { 1987 char *endptr; 1988 unsigned long ul = strtoul(value, &endptr, 0); 1989 if (*endptr == '\0' && ul != 0) { 1990 ALOGD("Silence is golden"); 1991 // The setprop command will not allow a property to be changed after 1992 // the first time it is set, so we don't have to worry about un-muting. 1993 setMasterMute_l(true); 1994 } 1995 } 1996 } 1997} 1998 1999// shared by MIXER and DIRECT, overridden by DUPLICATING 2000ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2001{ 2002 // FIXME rewrite to reduce number of system calls 2003 mLastWriteTime = systemTime(); 2004 mInWrite = true; 2005 ssize_t bytesWritten; 2006 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2007 2008 // If an NBAIO sink is present, use it to write the normal mixer's submix 2009 if (mNormalSink != 0) { 2010 const size_t count = mBytesRemaining / mFrameSize; 2011 2012 ATRACE_BEGIN("write"); 2013 // update the setpoint when AudioFlinger::mScreenState changes 2014 uint32_t screenState = AudioFlinger::mScreenState; 2015 if (screenState != mScreenState) { 2016 mScreenState = screenState; 2017 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2018 if (pipe != NULL) { 2019 pipe->setAvgFrames((mScreenState & 1) ? 2020 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2021 } 2022 } 2023 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2024 ATRACE_END(); 2025 if (framesWritten > 0) { 2026 bytesWritten = framesWritten * mFrameSize; 2027 } else { 2028 bytesWritten = framesWritten; 2029 } 2030 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2031 if (status == NO_ERROR) { 2032 size_t totalFramesWritten = mNormalSink->framesWritten(); 2033 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2034 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2035 mLatchDValid = true; 2036 } 2037 } 2038 // otherwise use the HAL / AudioStreamOut directly 2039 } else { 2040 // Direct output and offload threads 2041 2042 if (mUseAsyncWrite) { 2043 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2044 mWriteAckSequence += 2; 2045 mWriteAckSequence |= 1; 2046 ALOG_ASSERT(mCallbackThread != 0); 2047 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2048 } 2049 // FIXME We should have an implementation of timestamps for direct output threads. 2050 // They are used e.g for multichannel PCM playback over HDMI. 2051 bytesWritten = mOutput->stream->write(mOutput->stream, 2052 (char *)mSinkBuffer + offset, mBytesRemaining); 2053 if (mUseAsyncWrite && 2054 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2055 // do not wait for async callback in case of error of full write 2056 mWriteAckSequence &= ~1; 2057 ALOG_ASSERT(mCallbackThread != 0); 2058 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2059 } 2060 } 2061 2062 mNumWrites++; 2063 mInWrite = false; 2064 mStandby = false; 2065 return bytesWritten; 2066} 2067 2068void AudioFlinger::PlaybackThread::threadLoop_drain() 2069{ 2070 if (mOutput->stream->drain) { 2071 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2072 if (mUseAsyncWrite) { 2073 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2074 mDrainSequence |= 1; 2075 ALOG_ASSERT(mCallbackThread != 0); 2076 mCallbackThread->setDraining(mDrainSequence); 2077 } 2078 mOutput->stream->drain(mOutput->stream, 2079 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2080 : AUDIO_DRAIN_ALL); 2081 } 2082} 2083 2084void AudioFlinger::PlaybackThread::threadLoop_exit() 2085{ 2086 // Default implementation has nothing to do 2087} 2088 2089/* 2090The derived values that are cached: 2091 - mSinkBufferSize from frame count * frame size 2092 - activeSleepTime from activeSleepTimeUs() 2093 - idleSleepTime from idleSleepTimeUs() 2094 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2095 - maxPeriod from frame count and sample rate (MIXER only) 2096 2097The parameters that affect these derived values are: 2098 - frame count 2099 - frame size 2100 - sample rate 2101 - device type: A2DP or not 2102 - device latency 2103 - format: PCM or not 2104 - active sleep time 2105 - idle sleep time 2106*/ 2107 2108void AudioFlinger::PlaybackThread::cacheParameters_l() 2109{ 2110 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2111 activeSleepTime = activeSleepTimeUs(); 2112 idleSleepTime = idleSleepTimeUs(); 2113} 2114 2115void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2116{ 2117 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2118 this, streamType, mTracks.size()); 2119 Mutex::Autolock _l(mLock); 2120 2121 size_t size = mTracks.size(); 2122 for (size_t i = 0; i < size; i++) { 2123 sp<Track> t = mTracks[i]; 2124 if (t->streamType() == streamType) { 2125 t->invalidate(); 2126 } 2127 } 2128} 2129 2130status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2131{ 2132 int session = chain->sessionId(); 2133 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2134 ? mEffectBuffer : mSinkBuffer); 2135 bool ownsBuffer = false; 2136 2137 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2138 if (session > 0) { 2139 // Only one effect chain can be present in direct output thread and it uses 2140 // the sink buffer as input 2141 if (mType != DIRECT) { 2142 size_t numSamples = mNormalFrameCount * mChannelCount; 2143 buffer = new int16_t[numSamples]; 2144 memset(buffer, 0, numSamples * sizeof(int16_t)); 2145 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2146 ownsBuffer = true; 2147 } 2148 2149 // Attach all tracks with same session ID to this chain. 2150 for (size_t i = 0; i < mTracks.size(); ++i) { 2151 sp<Track> track = mTracks[i]; 2152 if (session == track->sessionId()) { 2153 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2154 buffer); 2155 track->setMainBuffer(buffer); 2156 chain->incTrackCnt(); 2157 } 2158 } 2159 2160 // indicate all active tracks in the chain 2161 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2162 sp<Track> track = mActiveTracks[i].promote(); 2163 if (track == 0) { 2164 continue; 2165 } 2166 if (session == track->sessionId()) { 2167 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2168 chain->incActiveTrackCnt(); 2169 } 2170 } 2171 } 2172 2173 chain->setInBuffer(buffer, ownsBuffer); 2174 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2175 ? mEffectBuffer : mSinkBuffer)); 2176 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2177 // chains list in order to be processed last as it contains output stage effects 2178 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2179 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2180 // after track specific effects and before output stage 2181 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2182 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2183 // Effect chain for other sessions are inserted at beginning of effect 2184 // chains list to be processed before output mix effects. Relative order between other 2185 // sessions is not important 2186 size_t size = mEffectChains.size(); 2187 size_t i = 0; 2188 for (i = 0; i < size; i++) { 2189 if (mEffectChains[i]->sessionId() < session) { 2190 break; 2191 } 2192 } 2193 mEffectChains.insertAt(chain, i); 2194 checkSuspendOnAddEffectChain_l(chain); 2195 2196 return NO_ERROR; 2197} 2198 2199size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2200{ 2201 int session = chain->sessionId(); 2202 2203 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2204 2205 for (size_t i = 0; i < mEffectChains.size(); i++) { 2206 if (chain == mEffectChains[i]) { 2207 mEffectChains.removeAt(i); 2208 // detach all active tracks from the chain 2209 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2210 sp<Track> track = mActiveTracks[i].promote(); 2211 if (track == 0) { 2212 continue; 2213 } 2214 if (session == track->sessionId()) { 2215 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2216 chain.get(), session); 2217 chain->decActiveTrackCnt(); 2218 } 2219 } 2220 2221 // detach all tracks with same session ID from this chain 2222 for (size_t i = 0; i < mTracks.size(); ++i) { 2223 sp<Track> track = mTracks[i]; 2224 if (session == track->sessionId()) { 2225 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2226 chain->decTrackCnt(); 2227 } 2228 } 2229 break; 2230 } 2231 } 2232 return mEffectChains.size(); 2233} 2234 2235status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2236 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2237{ 2238 Mutex::Autolock _l(mLock); 2239 return attachAuxEffect_l(track, EffectId); 2240} 2241 2242status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2243 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2244{ 2245 status_t status = NO_ERROR; 2246 2247 if (EffectId == 0) { 2248 track->setAuxBuffer(0, NULL); 2249 } else { 2250 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2251 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2252 if (effect != 0) { 2253 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2254 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2255 } else { 2256 status = INVALID_OPERATION; 2257 } 2258 } else { 2259 status = BAD_VALUE; 2260 } 2261 } 2262 return status; 2263} 2264 2265void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2266{ 2267 for (size_t i = 0; i < mTracks.size(); ++i) { 2268 sp<Track> track = mTracks[i]; 2269 if (track->auxEffectId() == effectId) { 2270 attachAuxEffect_l(track, 0); 2271 } 2272 } 2273} 2274 2275bool AudioFlinger::PlaybackThread::threadLoop() 2276{ 2277 Vector< sp<Track> > tracksToRemove; 2278 2279 standbyTime = systemTime(); 2280 2281 // MIXER 2282 nsecs_t lastWarning = 0; 2283 2284 // DUPLICATING 2285 // FIXME could this be made local to while loop? 2286 writeFrames = 0; 2287 2288 int lastGeneration = 0; 2289 2290 cacheParameters_l(); 2291 sleepTime = idleSleepTime; 2292 2293 if (mType == MIXER) { 2294 sleepTimeShift = 0; 2295 } 2296 2297 CpuStats cpuStats; 2298 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2299 2300 acquireWakeLock(); 2301 2302 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2303 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2304 // and then that string will be logged at the next convenient opportunity. 2305 const char *logString = NULL; 2306 2307 checkSilentMode_l(); 2308 2309 while (!exitPending()) 2310 { 2311 cpuStats.sample(myName); 2312 2313 Vector< sp<EffectChain> > effectChains; 2314 2315 { // scope for mLock 2316 2317 Mutex::Autolock _l(mLock); 2318 2319 processConfigEvents_l(); 2320 2321 if (logString != NULL) { 2322 mNBLogWriter->logTimestamp(); 2323 mNBLogWriter->log(logString); 2324 logString = NULL; 2325 } 2326 2327 if (mLatchDValid) { 2328 mLatchQ = mLatchD; 2329 mLatchDValid = false; 2330 mLatchQValid = true; 2331 } 2332 2333 saveOutputTracks(); 2334 if (mSignalPending) { 2335 // A signal was raised while we were unlocked 2336 mSignalPending = false; 2337 } else if (waitingAsyncCallback_l()) { 2338 if (exitPending()) { 2339 break; 2340 } 2341 releaseWakeLock_l(); 2342 mWakeLockUids.clear(); 2343 mActiveTracksGeneration++; 2344 ALOGV("wait async completion"); 2345 mWaitWorkCV.wait(mLock); 2346 ALOGV("async completion/wake"); 2347 acquireWakeLock_l(); 2348 standbyTime = systemTime() + standbyDelay; 2349 sleepTime = 0; 2350 2351 continue; 2352 } 2353 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2354 isSuspended()) { 2355 // put audio hardware into standby after short delay 2356 if (shouldStandby_l()) { 2357 2358 threadLoop_standby(); 2359 2360 mStandby = true; 2361 } 2362 2363 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2364 // we're about to wait, flush the binder command buffer 2365 IPCThreadState::self()->flushCommands(); 2366 2367 clearOutputTracks(); 2368 2369 if (exitPending()) { 2370 break; 2371 } 2372 2373 releaseWakeLock_l(); 2374 mWakeLockUids.clear(); 2375 mActiveTracksGeneration++; 2376 // wait until we have something to do... 2377 ALOGV("%s going to sleep", myName.string()); 2378 mWaitWorkCV.wait(mLock); 2379 ALOGV("%s waking up", myName.string()); 2380 acquireWakeLock_l(); 2381 2382 mMixerStatus = MIXER_IDLE; 2383 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2384 mBytesWritten = 0; 2385 mBytesRemaining = 0; 2386 checkSilentMode_l(); 2387 2388 standbyTime = systemTime() + standbyDelay; 2389 sleepTime = idleSleepTime; 2390 if (mType == MIXER) { 2391 sleepTimeShift = 0; 2392 } 2393 2394 continue; 2395 } 2396 } 2397 // mMixerStatusIgnoringFastTracks is also updated internally 2398 mMixerStatus = prepareTracks_l(&tracksToRemove); 2399 2400 // compare with previously applied list 2401 if (lastGeneration != mActiveTracksGeneration) { 2402 // update wakelock 2403 updateWakeLockUids_l(mWakeLockUids); 2404 lastGeneration = mActiveTracksGeneration; 2405 } 2406 2407 // prevent any changes in effect chain list and in each effect chain 2408 // during mixing and effect process as the audio buffers could be deleted 2409 // or modified if an effect is created or deleted 2410 lockEffectChains_l(effectChains); 2411 } // mLock scope ends 2412 2413 if (mBytesRemaining == 0) { 2414 mCurrentWriteLength = 0; 2415 if (mMixerStatus == MIXER_TRACKS_READY) { 2416 // threadLoop_mix() sets mCurrentWriteLength 2417 threadLoop_mix(); 2418 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2419 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2420 // threadLoop_sleepTime sets sleepTime to 0 if data 2421 // must be written to HAL 2422 threadLoop_sleepTime(); 2423 if (sleepTime == 0) { 2424 mCurrentWriteLength = mSinkBufferSize; 2425 } 2426 } 2427 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2428 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2429 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2430 // or mSinkBuffer (if there are no effects). 2431 // 2432 // This is done pre-effects computation; if effects change to 2433 // support higher precision, this needs to move. 2434 // 2435 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2436 // TODO use sleepTime == 0 as an additional condition. 2437 if (mMixerBufferValid) { 2438 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2439 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2440 2441 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2442 mNormalFrameCount * mChannelCount); 2443 } 2444 2445 mBytesRemaining = mCurrentWriteLength; 2446 if (isSuspended()) { 2447 sleepTime = suspendSleepTimeUs(); 2448 // simulate write to HAL when suspended 2449 mBytesWritten += mSinkBufferSize; 2450 mBytesRemaining = 0; 2451 } 2452 2453 // only process effects if we're going to write 2454 if (sleepTime == 0 && mType != OFFLOAD) { 2455 for (size_t i = 0; i < effectChains.size(); i ++) { 2456 effectChains[i]->process_l(); 2457 } 2458 } 2459 } 2460 // Process effect chains for offloaded thread even if no audio 2461 // was read from audio track: process only updates effect state 2462 // and thus does have to be synchronized with audio writes but may have 2463 // to be called while waiting for async write callback 2464 if (mType == OFFLOAD) { 2465 for (size_t i = 0; i < effectChains.size(); i ++) { 2466 effectChains[i]->process_l(); 2467 } 2468 } 2469 2470 // Only if the Effects buffer is enabled and there is data in the 2471 // Effects buffer (buffer valid), we need to 2472 // copy into the sink buffer. 2473 // TODO use sleepTime == 0 as an additional condition. 2474 if (mEffectBufferValid) { 2475 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2476 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2477 mNormalFrameCount * mChannelCount); 2478 } 2479 2480 // enable changes in effect chain 2481 unlockEffectChains(effectChains); 2482 2483 if (!waitingAsyncCallback()) { 2484 // sleepTime == 0 means we must write to audio hardware 2485 if (sleepTime == 0) { 2486 if (mBytesRemaining) { 2487 ssize_t ret = threadLoop_write(); 2488 if (ret < 0) { 2489 mBytesRemaining = 0; 2490 } else { 2491 mBytesWritten += ret; 2492 mBytesRemaining -= ret; 2493 } 2494 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2495 (mMixerStatus == MIXER_DRAIN_ALL)) { 2496 threadLoop_drain(); 2497 } 2498 if (mType == MIXER) { 2499 // write blocked detection 2500 nsecs_t now = systemTime(); 2501 nsecs_t delta = now - mLastWriteTime; 2502 if (!mStandby && delta > maxPeriod) { 2503 mNumDelayedWrites++; 2504 if ((now - lastWarning) > kWarningThrottleNs) { 2505 ATRACE_NAME("underrun"); 2506 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2507 ns2ms(delta), mNumDelayedWrites, this); 2508 lastWarning = now; 2509 } 2510 } 2511 } 2512 2513 } else { 2514 usleep(sleepTime); 2515 } 2516 } 2517 2518 // Finally let go of removed track(s), without the lock held 2519 // since we can't guarantee the destructors won't acquire that 2520 // same lock. This will also mutate and push a new fast mixer state. 2521 threadLoop_removeTracks(tracksToRemove); 2522 tracksToRemove.clear(); 2523 2524 // FIXME I don't understand the need for this here; 2525 // it was in the original code but maybe the 2526 // assignment in saveOutputTracks() makes this unnecessary? 2527 clearOutputTracks(); 2528 2529 // Effect chains will be actually deleted here if they were removed from 2530 // mEffectChains list during mixing or effects processing 2531 effectChains.clear(); 2532 2533 // FIXME Note that the above .clear() is no longer necessary since effectChains 2534 // is now local to this block, but will keep it for now (at least until merge done). 2535 } 2536 2537 threadLoop_exit(); 2538 2539 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2540 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2541 // put output stream into standby mode 2542 if (!mStandby) { 2543 mOutput->stream->common.standby(&mOutput->stream->common); 2544 } 2545 } 2546 2547 releaseWakeLock(); 2548 mWakeLockUids.clear(); 2549 mActiveTracksGeneration++; 2550 2551 ALOGV("Thread %p type %d exiting", this, mType); 2552 return false; 2553} 2554 2555// removeTracks_l() must be called with ThreadBase::mLock held 2556void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2557{ 2558 size_t count = tracksToRemove.size(); 2559 if (count > 0) { 2560 for (size_t i=0 ; i<count ; i++) { 2561 const sp<Track>& track = tracksToRemove.itemAt(i); 2562 mActiveTracks.remove(track); 2563 mWakeLockUids.remove(track->uid()); 2564 mActiveTracksGeneration++; 2565 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2566 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2567 if (chain != 0) { 2568 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2569 track->sessionId()); 2570 chain->decActiveTrackCnt(); 2571 } 2572 if (track->isTerminated()) { 2573 removeTrack_l(track); 2574 } 2575 } 2576 } 2577 2578} 2579 2580status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2581{ 2582 if (mNormalSink != 0) { 2583 return mNormalSink->getTimestamp(timestamp); 2584 } 2585 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2586 uint64_t position64; 2587 int ret = mOutput->stream->get_presentation_position( 2588 mOutput->stream, &position64, ×tamp.mTime); 2589 if (ret == 0) { 2590 timestamp.mPosition = (uint32_t)position64; 2591 return NO_ERROR; 2592 } 2593 } 2594 return INVALID_OPERATION; 2595} 2596// ---------------------------------------------------------------------------- 2597 2598AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2599 audio_io_handle_t id, audio_devices_t device, type_t type) 2600 : PlaybackThread(audioFlinger, output, id, device, type), 2601 // mAudioMixer below 2602 // mFastMixer below 2603 mFastMixerFutex(0) 2604 // mOutputSink below 2605 // mPipeSink below 2606 // mNormalSink below 2607{ 2608 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2609 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2610 "mFrameCount=%d, mNormalFrameCount=%d", 2611 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2612 mNormalFrameCount); 2613 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2614 2615 // FIXME - Current mixer implementation only supports stereo output 2616 if (mChannelCount != FCC_2) { 2617 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2618 } 2619 2620 // create an NBAIO sink for the HAL output stream, and negotiate 2621 mOutputSink = new AudioStreamOutSink(output->stream); 2622 size_t numCounterOffers = 0; 2623 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2624 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2625 ALOG_ASSERT(index == 0); 2626 2627 // initialize fast mixer depending on configuration 2628 bool initFastMixer; 2629 switch (kUseFastMixer) { 2630 case FastMixer_Never: 2631 initFastMixer = false; 2632 break; 2633 case FastMixer_Always: 2634 initFastMixer = true; 2635 break; 2636 case FastMixer_Static: 2637 case FastMixer_Dynamic: 2638 initFastMixer = mFrameCount < mNormalFrameCount; 2639 break; 2640 } 2641 if (initFastMixer) { 2642 2643 // create a MonoPipe to connect our submix to FastMixer 2644 NBAIO_Format format = mOutputSink->format(); 2645 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2646 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2647 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2648 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2649 const NBAIO_Format offers[1] = {format}; 2650 size_t numCounterOffers = 0; 2651 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2652 ALOG_ASSERT(index == 0); 2653 monoPipe->setAvgFrames((mScreenState & 1) ? 2654 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2655 mPipeSink = monoPipe; 2656 2657#ifdef TEE_SINK 2658 if (mTeeSinkOutputEnabled) { 2659 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2660 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2661 numCounterOffers = 0; 2662 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2663 ALOG_ASSERT(index == 0); 2664 mTeeSink = teeSink; 2665 PipeReader *teeSource = new PipeReader(*teeSink); 2666 numCounterOffers = 0; 2667 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2668 ALOG_ASSERT(index == 0); 2669 mTeeSource = teeSource; 2670 } 2671#endif 2672 2673 // create fast mixer and configure it initially with just one fast track for our submix 2674 mFastMixer = new FastMixer(); 2675 FastMixerStateQueue *sq = mFastMixer->sq(); 2676#ifdef STATE_QUEUE_DUMP 2677 sq->setObserverDump(&mStateQueueObserverDump); 2678 sq->setMutatorDump(&mStateQueueMutatorDump); 2679#endif 2680 FastMixerState *state = sq->begin(); 2681 FastTrack *fastTrack = &state->mFastTracks[0]; 2682 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2683 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2684 fastTrack->mVolumeProvider = NULL; 2685 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2686 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2687 fastTrack->mGeneration++; 2688 state->mFastTracksGen++; 2689 state->mTrackMask = 1; 2690 // fast mixer will use the HAL output sink 2691 state->mOutputSink = mOutputSink.get(); 2692 state->mOutputSinkGen++; 2693 state->mFrameCount = mFrameCount; 2694 state->mCommand = FastMixerState::COLD_IDLE; 2695 // already done in constructor initialization list 2696 //mFastMixerFutex = 0; 2697 state->mColdFutexAddr = &mFastMixerFutex; 2698 state->mColdGen++; 2699 state->mDumpState = &mFastMixerDumpState; 2700#ifdef TEE_SINK 2701 state->mTeeSink = mTeeSink.get(); 2702#endif 2703 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2704 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2705 sq->end(); 2706 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2707 2708 // start the fast mixer 2709 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2710 pid_t tid = mFastMixer->getTid(); 2711 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2712 if (err != 0) { 2713 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2714 kPriorityFastMixer, getpid_cached, tid, err); 2715 } 2716 2717#ifdef AUDIO_WATCHDOG 2718 // create and start the watchdog 2719 mAudioWatchdog = new AudioWatchdog(); 2720 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2721 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2722 tid = mAudioWatchdog->getTid(); 2723 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2724 if (err != 0) { 2725 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2726 kPriorityFastMixer, getpid_cached, tid, err); 2727 } 2728#endif 2729 2730 } else { 2731 mFastMixer = NULL; 2732 } 2733 2734 switch (kUseFastMixer) { 2735 case FastMixer_Never: 2736 case FastMixer_Dynamic: 2737 mNormalSink = mOutputSink; 2738 break; 2739 case FastMixer_Always: 2740 mNormalSink = mPipeSink; 2741 break; 2742 case FastMixer_Static: 2743 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2744 break; 2745 } 2746} 2747 2748AudioFlinger::MixerThread::~MixerThread() 2749{ 2750 if (mFastMixer != NULL) { 2751 FastMixerStateQueue *sq = mFastMixer->sq(); 2752 FastMixerState *state = sq->begin(); 2753 if (state->mCommand == FastMixerState::COLD_IDLE) { 2754 int32_t old = android_atomic_inc(&mFastMixerFutex); 2755 if (old == -1) { 2756 (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2757 } 2758 } 2759 state->mCommand = FastMixerState::EXIT; 2760 sq->end(); 2761 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2762 mFastMixer->join(); 2763 // Though the fast mixer thread has exited, it's state queue is still valid. 2764 // We'll use that extract the final state which contains one remaining fast track 2765 // corresponding to our sub-mix. 2766 state = sq->begin(); 2767 ALOG_ASSERT(state->mTrackMask == 1); 2768 FastTrack *fastTrack = &state->mFastTracks[0]; 2769 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2770 delete fastTrack->mBufferProvider; 2771 sq->end(false /*didModify*/); 2772 delete mFastMixer; 2773#ifdef AUDIO_WATCHDOG 2774 if (mAudioWatchdog != 0) { 2775 mAudioWatchdog->requestExit(); 2776 mAudioWatchdog->requestExitAndWait(); 2777 mAudioWatchdog.clear(); 2778 } 2779#endif 2780 } 2781 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2782 delete mAudioMixer; 2783} 2784 2785 2786uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2787{ 2788 if (mFastMixer != NULL) { 2789 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2790 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2791 } 2792 return latency; 2793} 2794 2795 2796void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2797{ 2798 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2799} 2800 2801ssize_t AudioFlinger::MixerThread::threadLoop_write() 2802{ 2803 // FIXME we should only do one push per cycle; confirm this is true 2804 // Start the fast mixer if it's not already running 2805 if (mFastMixer != NULL) { 2806 FastMixerStateQueue *sq = mFastMixer->sq(); 2807 FastMixerState *state = sq->begin(); 2808 if (state->mCommand != FastMixerState::MIX_WRITE && 2809 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2810 if (state->mCommand == FastMixerState::COLD_IDLE) { 2811 int32_t old = android_atomic_inc(&mFastMixerFutex); 2812 if (old == -1) { 2813 (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2814 } 2815#ifdef AUDIO_WATCHDOG 2816 if (mAudioWatchdog != 0) { 2817 mAudioWatchdog->resume(); 2818 } 2819#endif 2820 } 2821 state->mCommand = FastMixerState::MIX_WRITE; 2822 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2823 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2824 sq->end(); 2825 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2826 if (kUseFastMixer == FastMixer_Dynamic) { 2827 mNormalSink = mPipeSink; 2828 } 2829 } else { 2830 sq->end(false /*didModify*/); 2831 } 2832 } 2833 return PlaybackThread::threadLoop_write(); 2834} 2835 2836void AudioFlinger::MixerThread::threadLoop_standby() 2837{ 2838 // Idle the fast mixer if it's currently running 2839 if (mFastMixer != NULL) { 2840 FastMixerStateQueue *sq = mFastMixer->sq(); 2841 FastMixerState *state = sq->begin(); 2842 if (!(state->mCommand & FastMixerState::IDLE)) { 2843 state->mCommand = FastMixerState::COLD_IDLE; 2844 state->mColdFutexAddr = &mFastMixerFutex; 2845 state->mColdGen++; 2846 mFastMixerFutex = 0; 2847 sq->end(); 2848 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2849 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2850 if (kUseFastMixer == FastMixer_Dynamic) { 2851 mNormalSink = mOutputSink; 2852 } 2853#ifdef AUDIO_WATCHDOG 2854 if (mAudioWatchdog != 0) { 2855 mAudioWatchdog->pause(); 2856 } 2857#endif 2858 } else { 2859 sq->end(false /*didModify*/); 2860 } 2861 } 2862 PlaybackThread::threadLoop_standby(); 2863} 2864 2865bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2866{ 2867 return false; 2868} 2869 2870bool AudioFlinger::PlaybackThread::shouldStandby_l() 2871{ 2872 return !mStandby; 2873} 2874 2875bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2876{ 2877 Mutex::Autolock _l(mLock); 2878 return waitingAsyncCallback_l(); 2879} 2880 2881// shared by MIXER and DIRECT, overridden by DUPLICATING 2882void AudioFlinger::PlaybackThread::threadLoop_standby() 2883{ 2884 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2885 mOutput->stream->common.standby(&mOutput->stream->common); 2886 if (mUseAsyncWrite != 0) { 2887 // discard any pending drain or write ack by incrementing sequence 2888 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2889 mDrainSequence = (mDrainSequence + 2) & ~1; 2890 ALOG_ASSERT(mCallbackThread != 0); 2891 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2892 mCallbackThread->setDraining(mDrainSequence); 2893 } 2894} 2895 2896void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2897{ 2898 ALOGV("signal playback thread"); 2899 broadcast_l(); 2900} 2901 2902void AudioFlinger::MixerThread::threadLoop_mix() 2903{ 2904 // obtain the presentation timestamp of the next output buffer 2905 int64_t pts; 2906 status_t status = INVALID_OPERATION; 2907 2908 if (mNormalSink != 0) { 2909 status = mNormalSink->getNextWriteTimestamp(&pts); 2910 } else { 2911 status = mOutputSink->getNextWriteTimestamp(&pts); 2912 } 2913 2914 if (status != NO_ERROR) { 2915 pts = AudioBufferProvider::kInvalidPTS; 2916 } 2917 2918 // mix buffers... 2919 mAudioMixer->process(pts); 2920 mCurrentWriteLength = mSinkBufferSize; 2921 // increase sleep time progressively when application underrun condition clears. 2922 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2923 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2924 // such that we would underrun the audio HAL. 2925 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2926 sleepTimeShift--; 2927 } 2928 sleepTime = 0; 2929 standbyTime = systemTime() + standbyDelay; 2930 //TODO: delay standby when effects have a tail 2931} 2932 2933void AudioFlinger::MixerThread::threadLoop_sleepTime() 2934{ 2935 // If no tracks are ready, sleep once for the duration of an output 2936 // buffer size, then write 0s to the output 2937 if (sleepTime == 0) { 2938 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2939 sleepTime = activeSleepTime >> sleepTimeShift; 2940 if (sleepTime < kMinThreadSleepTimeUs) { 2941 sleepTime = kMinThreadSleepTimeUs; 2942 } 2943 // reduce sleep time in case of consecutive application underruns to avoid 2944 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2945 // duration we would end up writing less data than needed by the audio HAL if 2946 // the condition persists. 2947 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2948 sleepTimeShift++; 2949 } 2950 } else { 2951 sleepTime = idleSleepTime; 2952 } 2953 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2954 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2955 // before effects processing or output. 2956 if (mMixerBufferValid) { 2957 memset(mMixerBuffer, 0, mMixerBufferSize); 2958 } else { 2959 memset(mSinkBuffer, 0, mSinkBufferSize); 2960 } 2961 sleepTime = 0; 2962 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2963 "anticipated start"); 2964 } 2965 // TODO add standby time extension fct of effect tail 2966} 2967 2968// prepareTracks_l() must be called with ThreadBase::mLock held 2969AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2970 Vector< sp<Track> > *tracksToRemove) 2971{ 2972 2973 mixer_state mixerStatus = MIXER_IDLE; 2974 // find out which tracks need to be processed 2975 size_t count = mActiveTracks.size(); 2976 size_t mixedTracks = 0; 2977 size_t tracksWithEffect = 0; 2978 // counts only _active_ fast tracks 2979 size_t fastTracks = 0; 2980 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2981 2982 float masterVolume = mMasterVolume; 2983 bool masterMute = mMasterMute; 2984 2985 if (masterMute) { 2986 masterVolume = 0; 2987 } 2988 // Delegate master volume control to effect in output mix effect chain if needed 2989 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2990 if (chain != 0) { 2991 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2992 chain->setVolume_l(&v, &v); 2993 masterVolume = (float)((v + (1 << 23)) >> 24); 2994 chain.clear(); 2995 } 2996 2997 // prepare a new state to push 2998 FastMixerStateQueue *sq = NULL; 2999 FastMixerState *state = NULL; 3000 bool didModify = false; 3001 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3002 if (mFastMixer != NULL) { 3003 sq = mFastMixer->sq(); 3004 state = sq->begin(); 3005 } 3006 3007 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3008 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3009 3010 for (size_t i=0 ; i<count ; i++) { 3011 const sp<Track> t = mActiveTracks[i].promote(); 3012 if (t == 0) { 3013 continue; 3014 } 3015 3016 // this const just means the local variable doesn't change 3017 Track* const track = t.get(); 3018 3019 // process fast tracks 3020 if (track->isFastTrack()) { 3021 3022 // It's theoretically possible (though unlikely) for a fast track to be created 3023 // and then removed within the same normal mix cycle. This is not a problem, as 3024 // the track never becomes active so it's fast mixer slot is never touched. 3025 // The converse, of removing an (active) track and then creating a new track 3026 // at the identical fast mixer slot within the same normal mix cycle, 3027 // is impossible because the slot isn't marked available until the end of each cycle. 3028 int j = track->mFastIndex; 3029 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3030 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3031 FastTrack *fastTrack = &state->mFastTracks[j]; 3032 3033 // Determine whether the track is currently in underrun condition, 3034 // and whether it had a recent underrun. 3035 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3036 FastTrackUnderruns underruns = ftDump->mUnderruns; 3037 uint32_t recentFull = (underruns.mBitFields.mFull - 3038 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3039 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3040 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3041 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3042 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3043 uint32_t recentUnderruns = recentPartial + recentEmpty; 3044 track->mObservedUnderruns = underruns; 3045 // don't count underruns that occur while stopping or pausing 3046 // or stopped which can occur when flush() is called while active 3047 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3048 recentUnderruns > 0) { 3049 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3050 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3051 } 3052 3053 // This is similar to the state machine for normal tracks, 3054 // with a few modifications for fast tracks. 3055 bool isActive = true; 3056 switch (track->mState) { 3057 case TrackBase::STOPPING_1: 3058 // track stays active in STOPPING_1 state until first underrun 3059 if (recentUnderruns > 0 || track->isTerminated()) { 3060 track->mState = TrackBase::STOPPING_2; 3061 } 3062 break; 3063 case TrackBase::PAUSING: 3064 // ramp down is not yet implemented 3065 track->setPaused(); 3066 break; 3067 case TrackBase::RESUMING: 3068 // ramp up is not yet implemented 3069 track->mState = TrackBase::ACTIVE; 3070 break; 3071 case TrackBase::ACTIVE: 3072 if (recentFull > 0 || recentPartial > 0) { 3073 // track has provided at least some frames recently: reset retry count 3074 track->mRetryCount = kMaxTrackRetries; 3075 } 3076 if (recentUnderruns == 0) { 3077 // no recent underruns: stay active 3078 break; 3079 } 3080 // there has recently been an underrun of some kind 3081 if (track->sharedBuffer() == 0) { 3082 // were any of the recent underruns "empty" (no frames available)? 3083 if (recentEmpty == 0) { 3084 // no, then ignore the partial underruns as they are allowed indefinitely 3085 break; 3086 } 3087 // there has recently been an "empty" underrun: decrement the retry counter 3088 if (--(track->mRetryCount) > 0) { 3089 break; 3090 } 3091 // indicate to client process that the track was disabled because of underrun; 3092 // it will then automatically call start() when data is available 3093 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3094 // remove from active list, but state remains ACTIVE [confusing but true] 3095 isActive = false; 3096 break; 3097 } 3098 // fall through 3099 case TrackBase::STOPPING_2: 3100 case TrackBase::PAUSED: 3101 case TrackBase::STOPPED: 3102 case TrackBase::FLUSHED: // flush() while active 3103 // Check for presentation complete if track is inactive 3104 // We have consumed all the buffers of this track. 3105 // This would be incomplete if we auto-paused on underrun 3106 { 3107 size_t audioHALFrames = 3108 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3109 size_t framesWritten = mBytesWritten / mFrameSize; 3110 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3111 // track stays in active list until presentation is complete 3112 break; 3113 } 3114 } 3115 if (track->isStopping_2()) { 3116 track->mState = TrackBase::STOPPED; 3117 } 3118 if (track->isStopped()) { 3119 // Can't reset directly, as fast mixer is still polling this track 3120 // track->reset(); 3121 // So instead mark this track as needing to be reset after push with ack 3122 resetMask |= 1 << i; 3123 } 3124 isActive = false; 3125 break; 3126 case TrackBase::IDLE: 3127 default: 3128 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3129 } 3130 3131 if (isActive) { 3132 // was it previously inactive? 3133 if (!(state->mTrackMask & (1 << j))) { 3134 ExtendedAudioBufferProvider *eabp = track; 3135 VolumeProvider *vp = track; 3136 fastTrack->mBufferProvider = eabp; 3137 fastTrack->mVolumeProvider = vp; 3138 fastTrack->mChannelMask = track->mChannelMask; 3139 fastTrack->mFormat = track->mFormat; 3140 fastTrack->mGeneration++; 3141 state->mTrackMask |= 1 << j; 3142 didModify = true; 3143 // no acknowledgement required for newly active tracks 3144 } 3145 // cache the combined master volume and stream type volume for fast mixer; this 3146 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3147 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3148 ++fastTracks; 3149 } else { 3150 // was it previously active? 3151 if (state->mTrackMask & (1 << j)) { 3152 fastTrack->mBufferProvider = NULL; 3153 fastTrack->mGeneration++; 3154 state->mTrackMask &= ~(1 << j); 3155 didModify = true; 3156 // If any fast tracks were removed, we must wait for acknowledgement 3157 // because we're about to decrement the last sp<> on those tracks. 3158 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3159 } else { 3160 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3161 } 3162 tracksToRemove->add(track); 3163 // Avoids a misleading display in dumpsys 3164 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3165 } 3166 continue; 3167 } 3168 3169 { // local variable scope to avoid goto warning 3170 3171 audio_track_cblk_t* cblk = track->cblk(); 3172 3173 // The first time a track is added we wait 3174 // for all its buffers to be filled before processing it 3175 int name = track->name(); 3176 // make sure that we have enough frames to mix one full buffer. 3177 // enforce this condition only once to enable draining the buffer in case the client 3178 // app does not call stop() and relies on underrun to stop: 3179 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3180 // during last round 3181 size_t desiredFrames; 3182 uint32_t sr = track->sampleRate(); 3183 if (sr == mSampleRate) { 3184 desiredFrames = mNormalFrameCount; 3185 } else { 3186 // +1 for rounding and +1 for additional sample needed for interpolation 3187 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3188 // add frames already consumed but not yet released by the resampler 3189 // because mAudioTrackServerProxy->framesReady() will include these frames 3190 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3191#if 0 3192 // the minimum track buffer size is normally twice the number of frames necessary 3193 // to fill one buffer and the resampler should not leave more than one buffer worth 3194 // of unreleased frames after each pass, but just in case... 3195 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3196#endif 3197 } 3198 uint32_t minFrames = 1; 3199 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3200 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3201 minFrames = desiredFrames; 3202 } 3203 3204 size_t framesReady = track->framesReady(); 3205 if ((framesReady >= minFrames) && track->isReady() && 3206 !track->isPaused() && !track->isTerminated()) 3207 { 3208 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3209 3210 mixedTracks++; 3211 3212 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3213 // there is an effect chain connected to the track 3214 chain.clear(); 3215 if (track->mainBuffer() != mSinkBuffer && 3216 track->mainBuffer() != mMixerBuffer) { 3217 if (mEffectBufferEnabled) { 3218 mEffectBufferValid = true; // Later can set directly. 3219 } 3220 chain = getEffectChain_l(track->sessionId()); 3221 // Delegate volume control to effect in track effect chain if needed 3222 if (chain != 0) { 3223 tracksWithEffect++; 3224 } else { 3225 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3226 "session %d", 3227 name, track->sessionId()); 3228 } 3229 } 3230 3231 3232 int param = AudioMixer::VOLUME; 3233 if (track->mFillingUpStatus == Track::FS_FILLED) { 3234 // no ramp for the first volume setting 3235 track->mFillingUpStatus = Track::FS_ACTIVE; 3236 if (track->mState == TrackBase::RESUMING) { 3237 track->mState = TrackBase::ACTIVE; 3238 param = AudioMixer::RAMP_VOLUME; 3239 } 3240 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3241 // FIXME should not make a decision based on mServer 3242 } else if (cblk->mServer != 0) { 3243 // If the track is stopped before the first frame was mixed, 3244 // do not apply ramp 3245 param = AudioMixer::RAMP_VOLUME; 3246 } 3247 3248 // compute volume for this track 3249 uint32_t vl, vr, va; 3250 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3251 vl = vr = va = 0; 3252 if (track->isPausing()) { 3253 track->setPaused(); 3254 } 3255 } else { 3256 3257 // read original volumes with volume control 3258 float typeVolume = mStreamTypes[track->streamType()].volume; 3259 float v = masterVolume * typeVolume; 3260 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3261 uint32_t vlr = proxy->getVolumeLR(); 3262 vl = vlr & 0xFFFF; 3263 vr = vlr >> 16; 3264 // track volumes come from shared memory, so can't be trusted and must be clamped 3265 if (vl > MAX_GAIN_INT) { 3266 ALOGV("Track left volume out of range: %04X", vl); 3267 vl = MAX_GAIN_INT; 3268 } 3269 if (vr > MAX_GAIN_INT) { 3270 ALOGV("Track right volume out of range: %04X", vr); 3271 vr = MAX_GAIN_INT; 3272 } 3273 // now apply the master volume and stream type volume 3274 vl = (uint32_t)(v * vl) << 12; 3275 vr = (uint32_t)(v * vr) << 12; 3276 // assuming master volume and stream type volume each go up to 1.0, 3277 // vl and vr are now in 8.24 format 3278 3279 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3280 // send level comes from shared memory and so may be corrupt 3281 if (sendLevel > MAX_GAIN_INT) { 3282 ALOGV("Track send level out of range: %04X", sendLevel); 3283 sendLevel = MAX_GAIN_INT; 3284 } 3285 va = (uint32_t)(v * sendLevel); 3286 } 3287 3288 // Delegate volume control to effect in track effect chain if needed 3289 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3290 // Do not ramp volume if volume is controlled by effect 3291 param = AudioMixer::VOLUME; 3292 track->mHasVolumeController = true; 3293 } else { 3294 // force no volume ramp when volume controller was just disabled or removed 3295 // from effect chain to avoid volume spike 3296 if (track->mHasVolumeController) { 3297 param = AudioMixer::VOLUME; 3298 } 3299 track->mHasVolumeController = false; 3300 } 3301 3302 // Convert volumes from 8.24 to 4.12 format 3303 // This additional clamping is needed in case chain->setVolume_l() overshot 3304 vl = (vl + (1 << 11)) >> 12; 3305 if (vl > MAX_GAIN_INT) { 3306 vl = MAX_GAIN_INT; 3307 } 3308 vr = (vr + (1 << 11)) >> 12; 3309 if (vr > MAX_GAIN_INT) { 3310 vr = MAX_GAIN_INT; 3311 } 3312 3313 if (va > MAX_GAIN_INT) { 3314 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3315 } 3316 3317 // XXX: these things DON'T need to be done each time 3318 mAudioMixer->setBufferProvider(name, track); 3319 mAudioMixer->enable(name); 3320 3321 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3322 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3323 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3324 mAudioMixer->setParameter( 3325 name, 3326 AudioMixer::TRACK, 3327 AudioMixer::FORMAT, (void *)track->format()); 3328 mAudioMixer->setParameter( 3329 name, 3330 AudioMixer::TRACK, 3331 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3332 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3333 uint32_t maxSampleRate = mSampleRate * 2; 3334 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3335 if (reqSampleRate == 0) { 3336 reqSampleRate = mSampleRate; 3337 } else if (reqSampleRate > maxSampleRate) { 3338 reqSampleRate = maxSampleRate; 3339 } 3340 mAudioMixer->setParameter( 3341 name, 3342 AudioMixer::RESAMPLE, 3343 AudioMixer::SAMPLE_RATE, 3344 (void *)(uintptr_t)reqSampleRate); 3345 /* 3346 * Select the appropriate output buffer for the track. 3347 * 3348 * Tracks with effects go into their own effects chain buffer 3349 * and from there into either mEffectBuffer or mSinkBuffer. 3350 * 3351 * Other tracks can use mMixerBuffer for higher precision 3352 * channel accumulation. If this buffer is enabled 3353 * (mMixerBufferEnabled true), then selected tracks will accumulate 3354 * into it. 3355 * 3356 */ 3357 if (mMixerBufferEnabled 3358 && (track->mainBuffer() == mSinkBuffer 3359 || track->mainBuffer() == mMixerBuffer)) { 3360 mAudioMixer->setParameter( 3361 name, 3362 AudioMixer::TRACK, 3363 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3364 mAudioMixer->setParameter( 3365 name, 3366 AudioMixer::TRACK, 3367 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3368 // TODO: override track->mainBuffer()? 3369 mMixerBufferValid = true; 3370 } else { 3371 mAudioMixer->setParameter( 3372 name, 3373 AudioMixer::TRACK, 3374 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3375 mAudioMixer->setParameter( 3376 name, 3377 AudioMixer::TRACK, 3378 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3379 } 3380 mAudioMixer->setParameter( 3381 name, 3382 AudioMixer::TRACK, 3383 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3384 3385 // reset retry count 3386 track->mRetryCount = kMaxTrackRetries; 3387 3388 // If one track is ready, set the mixer ready if: 3389 // - the mixer was not ready during previous round OR 3390 // - no other track is not ready 3391 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3392 mixerStatus != MIXER_TRACKS_ENABLED) { 3393 mixerStatus = MIXER_TRACKS_READY; 3394 } 3395 } else { 3396 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3397 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3398 } 3399 // clear effect chain input buffer if an active track underruns to avoid sending 3400 // previous audio buffer again to effects 3401 chain = getEffectChain_l(track->sessionId()); 3402 if (chain != 0) { 3403 chain->clearInputBuffer(); 3404 } 3405 3406 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3407 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3408 track->isStopped() || track->isPaused()) { 3409 // We have consumed all the buffers of this track. 3410 // Remove it from the list of active tracks. 3411 // TODO: use actual buffer filling status instead of latency when available from 3412 // audio HAL 3413 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3414 size_t framesWritten = mBytesWritten / mFrameSize; 3415 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3416 if (track->isStopped()) { 3417 track->reset(); 3418 } 3419 tracksToRemove->add(track); 3420 } 3421 } else { 3422 // No buffers for this track. Give it a few chances to 3423 // fill a buffer, then remove it from active list. 3424 if (--(track->mRetryCount) <= 0) { 3425 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3426 tracksToRemove->add(track); 3427 // indicate to client process that the track was disabled because of underrun; 3428 // it will then automatically call start() when data is available 3429 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3430 // If one track is not ready, mark the mixer also not ready if: 3431 // - the mixer was ready during previous round OR 3432 // - no other track is ready 3433 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3434 mixerStatus != MIXER_TRACKS_READY) { 3435 mixerStatus = MIXER_TRACKS_ENABLED; 3436 } 3437 } 3438 mAudioMixer->disable(name); 3439 } 3440 3441 } // local variable scope to avoid goto warning 3442track_is_ready: ; 3443 3444 } 3445 3446 // Push the new FastMixer state if necessary 3447 bool pauseAudioWatchdog = false; 3448 if (didModify) { 3449 state->mFastTracksGen++; 3450 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3451 if (kUseFastMixer == FastMixer_Dynamic && 3452 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3453 state->mCommand = FastMixerState::COLD_IDLE; 3454 state->mColdFutexAddr = &mFastMixerFutex; 3455 state->mColdGen++; 3456 mFastMixerFutex = 0; 3457 if (kUseFastMixer == FastMixer_Dynamic) { 3458 mNormalSink = mOutputSink; 3459 } 3460 // If we go into cold idle, need to wait for acknowledgement 3461 // so that fast mixer stops doing I/O. 3462 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3463 pauseAudioWatchdog = true; 3464 } 3465 } 3466 if (sq != NULL) { 3467 sq->end(didModify); 3468 sq->push(block); 3469 } 3470#ifdef AUDIO_WATCHDOG 3471 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3472 mAudioWatchdog->pause(); 3473 } 3474#endif 3475 3476 // Now perform the deferred reset on fast tracks that have stopped 3477 while (resetMask != 0) { 3478 size_t i = __builtin_ctz(resetMask); 3479 ALOG_ASSERT(i < count); 3480 resetMask &= ~(1 << i); 3481 sp<Track> t = mActiveTracks[i].promote(); 3482 if (t == 0) { 3483 continue; 3484 } 3485 Track* track = t.get(); 3486 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3487 track->reset(); 3488 } 3489 3490 // remove all the tracks that need to be... 3491 removeTracks_l(*tracksToRemove); 3492 3493 // sink or mix buffer must be cleared if all tracks are connected to an 3494 // effect chain as in this case the mixer will not write to the sink or mix buffer 3495 // and track effects will accumulate into it 3496 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3497 (mixedTracks == 0 && fastTracks > 0))) { 3498 // FIXME as a performance optimization, should remember previous zero status 3499 if (mMixerBufferValid) { 3500 memset(mMixerBuffer, 0, mMixerBufferSize); 3501 // TODO: In testing, mSinkBuffer below need not be cleared because 3502 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3503 // after mixing. 3504 // 3505 // To enforce this guarantee: 3506 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3507 // (mixedTracks == 0 && fastTracks > 0)) 3508 // must imply MIXER_TRACKS_READY. 3509 // Later, we may clear buffers regardless, and skip much of this logic. 3510 } 3511 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3512 if (mEffectBufferValid) { 3513 memset(mEffectBuffer, 0, mEffectBufferSize); 3514 } 3515 // FIXME as a performance optimization, should remember previous zero status 3516 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3517 } 3518 3519 // if any fast tracks, then status is ready 3520 mMixerStatusIgnoringFastTracks = mixerStatus; 3521 if (fastTracks > 0) { 3522 mixerStatus = MIXER_TRACKS_READY; 3523 } 3524 return mixerStatus; 3525} 3526 3527// getTrackName_l() must be called with ThreadBase::mLock held 3528int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3529 audio_format_t format, int sessionId) 3530{ 3531 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3532} 3533 3534// deleteTrackName_l() must be called with ThreadBase::mLock held 3535void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3536{ 3537 ALOGV("remove track (%d) and delete from mixer", name); 3538 mAudioMixer->deleteTrackName(name); 3539} 3540 3541// checkForNewParameter_l() must be called with ThreadBase::mLock held 3542bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3543 status_t& status) 3544{ 3545 bool reconfig = false; 3546 3547 status = NO_ERROR; 3548 3549 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3550 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3551 if (mFastMixer != NULL) { 3552 FastMixerStateQueue *sq = mFastMixer->sq(); 3553 FastMixerState *state = sq->begin(); 3554 if (!(state->mCommand & FastMixerState::IDLE)) { 3555 previousCommand = state->mCommand; 3556 state->mCommand = FastMixerState::HOT_IDLE; 3557 sq->end(); 3558 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3559 } else { 3560 sq->end(false /*didModify*/); 3561 } 3562 } 3563 3564 AudioParameter param = AudioParameter(keyValuePair); 3565 int value; 3566 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3567 reconfig = true; 3568 } 3569 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3570 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3571 status = BAD_VALUE; 3572 } else { 3573 // no need to save value, since it's constant 3574 reconfig = true; 3575 } 3576 } 3577 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3578 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3579 status = BAD_VALUE; 3580 } else { 3581 // no need to save value, since it's constant 3582 reconfig = true; 3583 } 3584 } 3585 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3586 // do not accept frame count changes if tracks are open as the track buffer 3587 // size depends on frame count and correct behavior would not be guaranteed 3588 // if frame count is changed after track creation 3589 if (!mTracks.isEmpty()) { 3590 status = INVALID_OPERATION; 3591 } else { 3592 reconfig = true; 3593 } 3594 } 3595 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3596#ifdef ADD_BATTERY_DATA 3597 // when changing the audio output device, call addBatteryData to notify 3598 // the change 3599 if (mOutDevice != value) { 3600 uint32_t params = 0; 3601 // check whether speaker is on 3602 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3603 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3604 } 3605 3606 audio_devices_t deviceWithoutSpeaker 3607 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3608 // check if any other device (except speaker) is on 3609 if (value & deviceWithoutSpeaker ) { 3610 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3611 } 3612 3613 if (params != 0) { 3614 addBatteryData(params); 3615 } 3616 } 3617#endif 3618 3619 // forward device change to effects that have requested to be 3620 // aware of attached audio device. 3621 if (value != AUDIO_DEVICE_NONE) { 3622 mOutDevice = value; 3623 for (size_t i = 0; i < mEffectChains.size(); i++) { 3624 mEffectChains[i]->setDevice_l(mOutDevice); 3625 } 3626 } 3627 } 3628 3629 if (status == NO_ERROR) { 3630 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3631 keyValuePair.string()); 3632 if (!mStandby && status == INVALID_OPERATION) { 3633 mOutput->stream->common.standby(&mOutput->stream->common); 3634 mStandby = true; 3635 mBytesWritten = 0; 3636 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3637 keyValuePair.string()); 3638 } 3639 if (status == NO_ERROR && reconfig) { 3640 readOutputParameters_l(); 3641 delete mAudioMixer; 3642 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3643 for (size_t i = 0; i < mTracks.size() ; i++) { 3644 int name = getTrackName_l(mTracks[i]->mChannelMask, 3645 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3646 if (name < 0) { 3647 break; 3648 } 3649 mTracks[i]->mName = name; 3650 } 3651 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3652 } 3653 } 3654 3655 if (!(previousCommand & FastMixerState::IDLE)) { 3656 ALOG_ASSERT(mFastMixer != NULL); 3657 FastMixerStateQueue *sq = mFastMixer->sq(); 3658 FastMixerState *state = sq->begin(); 3659 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3660 state->mCommand = previousCommand; 3661 sq->end(); 3662 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3663 } 3664 3665 return reconfig; 3666} 3667 3668 3669void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3670{ 3671 const size_t SIZE = 256; 3672 char buffer[SIZE]; 3673 String8 result; 3674 3675 PlaybackThread::dumpInternals(fd, args); 3676 3677 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3678 3679 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3680 const FastMixerDumpState copy(mFastMixerDumpState); 3681 copy.dump(fd); 3682 3683#ifdef STATE_QUEUE_DUMP 3684 // Similar for state queue 3685 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3686 observerCopy.dump(fd); 3687 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3688 mutatorCopy.dump(fd); 3689#endif 3690 3691#ifdef TEE_SINK 3692 // Write the tee output to a .wav file 3693 dumpTee(fd, mTeeSource, mId); 3694#endif 3695 3696#ifdef AUDIO_WATCHDOG 3697 if (mAudioWatchdog != 0) { 3698 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3699 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3700 wdCopy.dump(fd); 3701 } 3702#endif 3703} 3704 3705uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3706{ 3707 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3708} 3709 3710uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3711{ 3712 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3713} 3714 3715void AudioFlinger::MixerThread::cacheParameters_l() 3716{ 3717 PlaybackThread::cacheParameters_l(); 3718 3719 // FIXME: Relaxed timing because of a certain device that can't meet latency 3720 // Should be reduced to 2x after the vendor fixes the driver issue 3721 // increase threshold again due to low power audio mode. The way this warning 3722 // threshold is calculated and its usefulness should be reconsidered anyway. 3723 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3724} 3725 3726// ---------------------------------------------------------------------------- 3727 3728AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3729 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3730 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3731 // mLeftVolFloat, mRightVolFloat 3732{ 3733} 3734 3735AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3736 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3737 ThreadBase::type_t type) 3738 : PlaybackThread(audioFlinger, output, id, device, type) 3739 // mLeftVolFloat, mRightVolFloat 3740{ 3741} 3742 3743AudioFlinger::DirectOutputThread::~DirectOutputThread() 3744{ 3745} 3746 3747void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3748{ 3749 audio_track_cblk_t* cblk = track->cblk(); 3750 float left, right; 3751 3752 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3753 left = right = 0; 3754 } else { 3755 float typeVolume = mStreamTypes[track->streamType()].volume; 3756 float v = mMasterVolume * typeVolume; 3757 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3758 uint32_t vlr = proxy->getVolumeLR(); 3759 float v_clamped = v * (vlr & 0xFFFF); 3760 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3761 left = v_clamped/MAX_GAIN; 3762 v_clamped = v * (vlr >> 16); 3763 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3764 right = v_clamped/MAX_GAIN; 3765 } 3766 3767 if (lastTrack) { 3768 if (left != mLeftVolFloat || right != mRightVolFloat) { 3769 mLeftVolFloat = left; 3770 mRightVolFloat = right; 3771 3772 // Convert volumes from float to 8.24 3773 uint32_t vl = (uint32_t)(left * (1 << 24)); 3774 uint32_t vr = (uint32_t)(right * (1 << 24)); 3775 3776 // Delegate volume control to effect in track effect chain if needed 3777 // only one effect chain can be present on DirectOutputThread, so if 3778 // there is one, the track is connected to it 3779 if (!mEffectChains.isEmpty()) { 3780 mEffectChains[0]->setVolume_l(&vl, &vr); 3781 left = (float)vl / (1 << 24); 3782 right = (float)vr / (1 << 24); 3783 } 3784 if (mOutput->stream->set_volume) { 3785 mOutput->stream->set_volume(mOutput->stream, left, right); 3786 } 3787 } 3788 } 3789} 3790 3791 3792AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3793 Vector< sp<Track> > *tracksToRemove 3794) 3795{ 3796 size_t count = mActiveTracks.size(); 3797 mixer_state mixerStatus = MIXER_IDLE; 3798 3799 // find out which tracks need to be processed 3800 for (size_t i = 0; i < count; i++) { 3801 sp<Track> t = mActiveTracks[i].promote(); 3802 // The track died recently 3803 if (t == 0) { 3804 continue; 3805 } 3806 3807 Track* const track = t.get(); 3808 audio_track_cblk_t* cblk = track->cblk(); 3809 // Only consider last track started for volume and mixer state control. 3810 // In theory an older track could underrun and restart after the new one starts 3811 // but as we only care about the transition phase between two tracks on a 3812 // direct output, it is not a problem to ignore the underrun case. 3813 sp<Track> l = mLatestActiveTrack.promote(); 3814 bool last = l.get() == track; 3815 3816 // The first time a track is added we wait 3817 // for all its buffers to be filled before processing it 3818 uint32_t minFrames; 3819 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3820 minFrames = mNormalFrameCount; 3821 } else { 3822 minFrames = 1; 3823 } 3824 3825 if ((track->framesReady() >= minFrames) && track->isReady() && 3826 !track->isPaused() && !track->isTerminated()) 3827 { 3828 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3829 3830 if (track->mFillingUpStatus == Track::FS_FILLED) { 3831 track->mFillingUpStatus = Track::FS_ACTIVE; 3832 // make sure processVolume_l() will apply new volume even if 0 3833 mLeftVolFloat = mRightVolFloat = -1.0; 3834 if (track->mState == TrackBase::RESUMING) { 3835 track->mState = TrackBase::ACTIVE; 3836 } 3837 } 3838 3839 // compute volume for this track 3840 processVolume_l(track, last); 3841 if (last) { 3842 // reset retry count 3843 track->mRetryCount = kMaxTrackRetriesDirect; 3844 mActiveTrack = t; 3845 mixerStatus = MIXER_TRACKS_READY; 3846 } 3847 } else { 3848 // clear effect chain input buffer if the last active track started underruns 3849 // to avoid sending previous audio buffer again to effects 3850 if (!mEffectChains.isEmpty() && last) { 3851 mEffectChains[0]->clearInputBuffer(); 3852 } 3853 3854 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3855 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3856 track->isStopped() || track->isPaused()) { 3857 // We have consumed all the buffers of this track. 3858 // Remove it from the list of active tracks. 3859 // TODO: implement behavior for compressed audio 3860 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3861 size_t framesWritten = mBytesWritten / mFrameSize; 3862 if (mStandby || !last || 3863 track->presentationComplete(framesWritten, audioHALFrames)) { 3864 if (track->isStopped()) { 3865 track->reset(); 3866 } 3867 tracksToRemove->add(track); 3868 } 3869 } else { 3870 // No buffers for this track. Give it a few chances to 3871 // fill a buffer, then remove it from active list. 3872 // Only consider last track started for mixer state control 3873 if (--(track->mRetryCount) <= 0) { 3874 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3875 tracksToRemove->add(track); 3876 // indicate to client process that the track was disabled because of underrun; 3877 // it will then automatically call start() when data is available 3878 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3879 } else if (last) { 3880 mixerStatus = MIXER_TRACKS_ENABLED; 3881 } 3882 } 3883 } 3884 } 3885 3886 // remove all the tracks that need to be... 3887 removeTracks_l(*tracksToRemove); 3888 3889 return mixerStatus; 3890} 3891 3892void AudioFlinger::DirectOutputThread::threadLoop_mix() 3893{ 3894 size_t frameCount = mFrameCount; 3895 int8_t *curBuf = (int8_t *)mSinkBuffer; 3896 // output audio to hardware 3897 while (frameCount) { 3898 AudioBufferProvider::Buffer buffer; 3899 buffer.frameCount = frameCount; 3900 mActiveTrack->getNextBuffer(&buffer); 3901 if (buffer.raw == NULL) { 3902 memset(curBuf, 0, frameCount * mFrameSize); 3903 break; 3904 } 3905 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3906 frameCount -= buffer.frameCount; 3907 curBuf += buffer.frameCount * mFrameSize; 3908 mActiveTrack->releaseBuffer(&buffer); 3909 } 3910 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3911 sleepTime = 0; 3912 standbyTime = systemTime() + standbyDelay; 3913 mActiveTrack.clear(); 3914} 3915 3916void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3917{ 3918 if (sleepTime == 0) { 3919 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3920 sleepTime = activeSleepTime; 3921 } else { 3922 sleepTime = idleSleepTime; 3923 } 3924 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3925 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3926 sleepTime = 0; 3927 } 3928} 3929 3930// getTrackName_l() must be called with ThreadBase::mLock held 3931int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3932 audio_format_t format __unused, int sessionId __unused) 3933{ 3934 return 0; 3935} 3936 3937// deleteTrackName_l() must be called with ThreadBase::mLock held 3938void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3939{ 3940} 3941 3942// checkForNewParameter_l() must be called with ThreadBase::mLock held 3943bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 3944 status_t& status) 3945{ 3946 bool reconfig = false; 3947 3948 status = NO_ERROR; 3949 3950 AudioParameter param = AudioParameter(keyValuePair); 3951 int value; 3952 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3953 // forward device change to effects that have requested to be 3954 // aware of attached audio device. 3955 if (value != AUDIO_DEVICE_NONE) { 3956 mOutDevice = value; 3957 for (size_t i = 0; i < mEffectChains.size(); i++) { 3958 mEffectChains[i]->setDevice_l(mOutDevice); 3959 } 3960 } 3961 } 3962 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3963 // do not accept frame count changes if tracks are open as the track buffer 3964 // size depends on frame count and correct behavior would not be garantied 3965 // if frame count is changed after track creation 3966 if (!mTracks.isEmpty()) { 3967 status = INVALID_OPERATION; 3968 } else { 3969 reconfig = true; 3970 } 3971 } 3972 if (status == NO_ERROR) { 3973 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3974 keyValuePair.string()); 3975 if (!mStandby && status == INVALID_OPERATION) { 3976 mOutput->stream->common.standby(&mOutput->stream->common); 3977 mStandby = true; 3978 mBytesWritten = 0; 3979 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3980 keyValuePair.string()); 3981 } 3982 if (status == NO_ERROR && reconfig) { 3983 readOutputParameters_l(); 3984 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3985 } 3986 } 3987 3988 return reconfig; 3989} 3990 3991uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3992{ 3993 uint32_t time; 3994 if (audio_is_linear_pcm(mFormat)) { 3995 time = PlaybackThread::activeSleepTimeUs(); 3996 } else { 3997 time = 10000; 3998 } 3999 return time; 4000} 4001 4002uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4003{ 4004 uint32_t time; 4005 if (audio_is_linear_pcm(mFormat)) { 4006 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4007 } else { 4008 time = 10000; 4009 } 4010 return time; 4011} 4012 4013uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4014{ 4015 uint32_t time; 4016 if (audio_is_linear_pcm(mFormat)) { 4017 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4018 } else { 4019 time = 10000; 4020 } 4021 return time; 4022} 4023 4024void AudioFlinger::DirectOutputThread::cacheParameters_l() 4025{ 4026 PlaybackThread::cacheParameters_l(); 4027 4028 // use shorter standby delay as on normal output to release 4029 // hardware resources as soon as possible 4030 if (audio_is_linear_pcm(mFormat)) { 4031 standbyDelay = microseconds(activeSleepTime*2); 4032 } else { 4033 standbyDelay = kOffloadStandbyDelayNs; 4034 } 4035} 4036 4037// ---------------------------------------------------------------------------- 4038 4039AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4040 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4041 : Thread(false /*canCallJava*/), 4042 mPlaybackThread(playbackThread), 4043 mWriteAckSequence(0), 4044 mDrainSequence(0) 4045{ 4046} 4047 4048AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4049{ 4050} 4051 4052void AudioFlinger::AsyncCallbackThread::onFirstRef() 4053{ 4054 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4055} 4056 4057bool AudioFlinger::AsyncCallbackThread::threadLoop() 4058{ 4059 while (!exitPending()) { 4060 uint32_t writeAckSequence; 4061 uint32_t drainSequence; 4062 4063 { 4064 Mutex::Autolock _l(mLock); 4065 while (!((mWriteAckSequence & 1) || 4066 (mDrainSequence & 1) || 4067 exitPending())) { 4068 mWaitWorkCV.wait(mLock); 4069 } 4070 4071 if (exitPending()) { 4072 break; 4073 } 4074 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4075 mWriteAckSequence, mDrainSequence); 4076 writeAckSequence = mWriteAckSequence; 4077 mWriteAckSequence &= ~1; 4078 drainSequence = mDrainSequence; 4079 mDrainSequence &= ~1; 4080 } 4081 { 4082 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4083 if (playbackThread != 0) { 4084 if (writeAckSequence & 1) { 4085 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4086 } 4087 if (drainSequence & 1) { 4088 playbackThread->resetDraining(drainSequence >> 1); 4089 } 4090 } 4091 } 4092 } 4093 return false; 4094} 4095 4096void AudioFlinger::AsyncCallbackThread::exit() 4097{ 4098 ALOGV("AsyncCallbackThread::exit"); 4099 Mutex::Autolock _l(mLock); 4100 requestExit(); 4101 mWaitWorkCV.broadcast(); 4102} 4103 4104void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4105{ 4106 Mutex::Autolock _l(mLock); 4107 // bit 0 is cleared 4108 mWriteAckSequence = sequence << 1; 4109} 4110 4111void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4112{ 4113 Mutex::Autolock _l(mLock); 4114 // ignore unexpected callbacks 4115 if (mWriteAckSequence & 2) { 4116 mWriteAckSequence |= 1; 4117 mWaitWorkCV.signal(); 4118 } 4119} 4120 4121void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4122{ 4123 Mutex::Autolock _l(mLock); 4124 // bit 0 is cleared 4125 mDrainSequence = sequence << 1; 4126} 4127 4128void AudioFlinger::AsyncCallbackThread::resetDraining() 4129{ 4130 Mutex::Autolock _l(mLock); 4131 // ignore unexpected callbacks 4132 if (mDrainSequence & 2) { 4133 mDrainSequence |= 1; 4134 mWaitWorkCV.signal(); 4135 } 4136} 4137 4138 4139// ---------------------------------------------------------------------------- 4140AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4141 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4142 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4143 mHwPaused(false), 4144 mFlushPending(false), 4145 mPausedBytesRemaining(0) 4146{ 4147 //FIXME: mStandby should be set to true by ThreadBase constructor 4148 mStandby = true; 4149} 4150 4151void AudioFlinger::OffloadThread::threadLoop_exit() 4152{ 4153 if (mFlushPending || mHwPaused) { 4154 // If a flush is pending or track was paused, just discard buffered data 4155 flushHw_l(); 4156 } else { 4157 mMixerStatus = MIXER_DRAIN_ALL; 4158 threadLoop_drain(); 4159 } 4160 mCallbackThread->exit(); 4161 PlaybackThread::threadLoop_exit(); 4162} 4163 4164AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4165 Vector< sp<Track> > *tracksToRemove 4166) 4167{ 4168 size_t count = mActiveTracks.size(); 4169 4170 mixer_state mixerStatus = MIXER_IDLE; 4171 bool doHwPause = false; 4172 bool doHwResume = false; 4173 4174 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4175 4176 // find out which tracks need to be processed 4177 for (size_t i = 0; i < count; i++) { 4178 sp<Track> t = mActiveTracks[i].promote(); 4179 // The track died recently 4180 if (t == 0) { 4181 continue; 4182 } 4183 Track* const track = t.get(); 4184 audio_track_cblk_t* cblk = track->cblk(); 4185 // Only consider last track started for volume and mixer state control. 4186 // In theory an older track could underrun and restart after the new one starts 4187 // but as we only care about the transition phase between two tracks on a 4188 // direct output, it is not a problem to ignore the underrun case. 4189 sp<Track> l = mLatestActiveTrack.promote(); 4190 bool last = l.get() == track; 4191 4192 if (track->isInvalid()) { 4193 ALOGW("An invalidated track shouldn't be in active list"); 4194 tracksToRemove->add(track); 4195 continue; 4196 } 4197 4198 if (track->mState == TrackBase::IDLE) { 4199 ALOGW("An idle track shouldn't be in active list"); 4200 continue; 4201 } 4202 4203 if (track->isPausing()) { 4204 track->setPaused(); 4205 if (last) { 4206 if (!mHwPaused) { 4207 doHwPause = true; 4208 mHwPaused = true; 4209 } 4210 // If we were part way through writing the mixbuffer to 4211 // the HAL we must save this until we resume 4212 // BUG - this will be wrong if a different track is made active, 4213 // in that case we want to discard the pending data in the 4214 // mixbuffer and tell the client to present it again when the 4215 // track is resumed 4216 mPausedWriteLength = mCurrentWriteLength; 4217 mPausedBytesRemaining = mBytesRemaining; 4218 mBytesRemaining = 0; // stop writing 4219 } 4220 tracksToRemove->add(track); 4221 } else if (track->isFlushPending()) { 4222 track->flushAck(); 4223 if (last) { 4224 mFlushPending = true; 4225 } 4226 } else if (track->isResumePending()){ 4227 track->resumeAck(); 4228 if (last) { 4229 if (mPausedBytesRemaining) { 4230 // Need to continue write that was interrupted 4231 mCurrentWriteLength = mPausedWriteLength; 4232 mBytesRemaining = mPausedBytesRemaining; 4233 mPausedBytesRemaining = 0; 4234 } 4235 if (mHwPaused) { 4236 doHwResume = true; 4237 mHwPaused = false; 4238 // threadLoop_mix() will handle the case that we need to 4239 // resume an interrupted write 4240 } 4241 // enable write to audio HAL 4242 sleepTime = 0; 4243 4244 // Do not handle new data in this iteration even if track->framesReady() 4245 mixerStatus = MIXER_TRACKS_ENABLED; 4246 } 4247 } else if (track->framesReady() && track->isReady() && 4248 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4249 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4250 if (track->mFillingUpStatus == Track::FS_FILLED) { 4251 track->mFillingUpStatus = Track::FS_ACTIVE; 4252 // make sure processVolume_l() will apply new volume even if 0 4253 mLeftVolFloat = mRightVolFloat = -1.0; 4254 } 4255 4256 if (last) { 4257 sp<Track> previousTrack = mPreviousTrack.promote(); 4258 if (previousTrack != 0) { 4259 if (track != previousTrack.get()) { 4260 // Flush any data still being written from last track 4261 mBytesRemaining = 0; 4262 if (mPausedBytesRemaining) { 4263 // Last track was paused so we also need to flush saved 4264 // mixbuffer state and invalidate track so that it will 4265 // re-submit that unwritten data when it is next resumed 4266 mPausedBytesRemaining = 0; 4267 // Invalidate is a bit drastic - would be more efficient 4268 // to have a flag to tell client that some of the 4269 // previously written data was lost 4270 previousTrack->invalidate(); 4271 } 4272 // flush data already sent to the DSP if changing audio session as audio 4273 // comes from a different source. Also invalidate previous track to force a 4274 // seek when resuming. 4275 if (previousTrack->sessionId() != track->sessionId()) { 4276 previousTrack->invalidate(); 4277 } 4278 } 4279 } 4280 mPreviousTrack = track; 4281 // reset retry count 4282 track->mRetryCount = kMaxTrackRetriesOffload; 4283 mActiveTrack = t; 4284 mixerStatus = MIXER_TRACKS_READY; 4285 } 4286 } else { 4287 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4288 if (track->isStopping_1()) { 4289 // Hardware buffer can hold a large amount of audio so we must 4290 // wait for all current track's data to drain before we say 4291 // that the track is stopped. 4292 if (mBytesRemaining == 0) { 4293 // Only start draining when all data in mixbuffer 4294 // has been written 4295 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4296 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4297 // do not drain if no data was ever sent to HAL (mStandby == true) 4298 if (last && !mStandby) { 4299 // do not modify drain sequence if we are already draining. This happens 4300 // when resuming from pause after drain. 4301 if ((mDrainSequence & 1) == 0) { 4302 sleepTime = 0; 4303 standbyTime = systemTime() + standbyDelay; 4304 mixerStatus = MIXER_DRAIN_TRACK; 4305 mDrainSequence += 2; 4306 } 4307 if (mHwPaused) { 4308 // It is possible to move from PAUSED to STOPPING_1 without 4309 // a resume so we must ensure hardware is running 4310 doHwResume = true; 4311 mHwPaused = false; 4312 } 4313 } 4314 } 4315 } else if (track->isStopping_2()) { 4316 // Drain has completed or we are in standby, signal presentation complete 4317 if (!(mDrainSequence & 1) || !last || mStandby) { 4318 track->mState = TrackBase::STOPPED; 4319 size_t audioHALFrames = 4320 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4321 size_t framesWritten = 4322 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4323 track->presentationComplete(framesWritten, audioHALFrames); 4324 track->reset(); 4325 tracksToRemove->add(track); 4326 } 4327 } else { 4328 // No buffers for this track. Give it a few chances to 4329 // fill a buffer, then remove it from active list. 4330 if (--(track->mRetryCount) <= 0) { 4331 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4332 track->name()); 4333 tracksToRemove->add(track); 4334 // indicate to client process that the track was disabled because of underrun; 4335 // it will then automatically call start() when data is available 4336 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4337 } else if (last){ 4338 mixerStatus = MIXER_TRACKS_ENABLED; 4339 } 4340 } 4341 } 4342 // compute volume for this track 4343 processVolume_l(track, last); 4344 } 4345 4346 // make sure the pause/flush/resume sequence is executed in the right order. 4347 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4348 // before flush and then resume HW. This can happen in case of pause/flush/resume 4349 // if resume is received before pause is executed. 4350 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4351 mOutput->stream->pause(mOutput->stream); 4352 } 4353 if (mFlushPending) { 4354 flushHw_l(); 4355 mFlushPending = false; 4356 } 4357 if (!mStandby && doHwResume) { 4358 mOutput->stream->resume(mOutput->stream); 4359 } 4360 4361 // remove all the tracks that need to be... 4362 removeTracks_l(*tracksToRemove); 4363 4364 return mixerStatus; 4365} 4366 4367// must be called with thread mutex locked 4368bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4369{ 4370 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4371 mWriteAckSequence, mDrainSequence); 4372 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4373 return true; 4374 } 4375 return false; 4376} 4377 4378// must be called with thread mutex locked 4379bool AudioFlinger::OffloadThread::shouldStandby_l() 4380{ 4381 bool trackPaused = false; 4382 4383 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4384 // after a timeout and we will enter standby then. 4385 if (mTracks.size() > 0) { 4386 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4387 } 4388 4389 return !mStandby && !trackPaused; 4390} 4391 4392 4393bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4394{ 4395 Mutex::Autolock _l(mLock); 4396 return waitingAsyncCallback_l(); 4397} 4398 4399void AudioFlinger::OffloadThread::flushHw_l() 4400{ 4401 mOutput->stream->flush(mOutput->stream); 4402 // Flush anything still waiting in the mixbuffer 4403 mCurrentWriteLength = 0; 4404 mBytesRemaining = 0; 4405 mPausedWriteLength = 0; 4406 mPausedBytesRemaining = 0; 4407 mHwPaused = false; 4408 4409 if (mUseAsyncWrite) { 4410 // discard any pending drain or write ack by incrementing sequence 4411 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4412 mDrainSequence = (mDrainSequence + 2) & ~1; 4413 ALOG_ASSERT(mCallbackThread != 0); 4414 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4415 mCallbackThread->setDraining(mDrainSequence); 4416 } 4417} 4418 4419void AudioFlinger::OffloadThread::onAddNewTrack_l() 4420{ 4421 sp<Track> previousTrack = mPreviousTrack.promote(); 4422 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4423 4424 if (previousTrack != 0 && latestTrack != 0 && 4425 (previousTrack->sessionId() != latestTrack->sessionId())) { 4426 mFlushPending = true; 4427 } 4428 PlaybackThread::onAddNewTrack_l(); 4429} 4430 4431// ---------------------------------------------------------------------------- 4432 4433AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4434 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4435 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4436 DUPLICATING), 4437 mWaitTimeMs(UINT_MAX) 4438{ 4439 addOutputTrack(mainThread); 4440} 4441 4442AudioFlinger::DuplicatingThread::~DuplicatingThread() 4443{ 4444 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4445 mOutputTracks[i]->destroy(); 4446 } 4447} 4448 4449void AudioFlinger::DuplicatingThread::threadLoop_mix() 4450{ 4451 // mix buffers... 4452 if (outputsReady(outputTracks)) { 4453 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4454 } else { 4455 memset(mSinkBuffer, 0, mSinkBufferSize); 4456 } 4457 sleepTime = 0; 4458 writeFrames = mNormalFrameCount; 4459 mCurrentWriteLength = mSinkBufferSize; 4460 standbyTime = systemTime() + standbyDelay; 4461} 4462 4463void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4464{ 4465 if (sleepTime == 0) { 4466 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4467 sleepTime = activeSleepTime; 4468 } else { 4469 sleepTime = idleSleepTime; 4470 } 4471 } else if (mBytesWritten != 0) { 4472 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4473 writeFrames = mNormalFrameCount; 4474 memset(mSinkBuffer, 0, mSinkBufferSize); 4475 } else { 4476 // flush remaining overflow buffers in output tracks 4477 writeFrames = 0; 4478 } 4479 sleepTime = 0; 4480 } 4481} 4482 4483ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4484{ 4485 for (size_t i = 0; i < outputTracks.size(); i++) { 4486 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4487 // for delivery downstream as needed. This in-place conversion is safe as 4488 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4489 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4490 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4491 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4492 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4493 } 4494 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4495 } 4496 mStandby = false; 4497 return (ssize_t)mSinkBufferSize; 4498} 4499 4500void AudioFlinger::DuplicatingThread::threadLoop_standby() 4501{ 4502 // DuplicatingThread implements standby by stopping all tracks 4503 for (size_t i = 0; i < outputTracks.size(); i++) { 4504 outputTracks[i]->stop(); 4505 } 4506} 4507 4508void AudioFlinger::DuplicatingThread::saveOutputTracks() 4509{ 4510 outputTracks = mOutputTracks; 4511} 4512 4513void AudioFlinger::DuplicatingThread::clearOutputTracks() 4514{ 4515 outputTracks.clear(); 4516} 4517 4518void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4519{ 4520 Mutex::Autolock _l(mLock); 4521 // FIXME explain this formula 4522 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4523 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4524 // due to current usage case and restrictions on the AudioBufferProvider. 4525 // Actual buffer conversion is done in threadLoop_write(). 4526 // 4527 // TODO: This may change in the future, depending on multichannel 4528 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4529 OutputTrack *outputTrack = new OutputTrack(thread, 4530 this, 4531 mSampleRate, 4532 AUDIO_FORMAT_PCM_16_BIT, 4533 mChannelMask, 4534 frameCount, 4535 IPCThreadState::self()->getCallingUid()); 4536 if (outputTrack->cblk() != NULL) { 4537 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4538 mOutputTracks.add(outputTrack); 4539 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4540 updateWaitTime_l(); 4541 } 4542} 4543 4544void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4545{ 4546 Mutex::Autolock _l(mLock); 4547 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4548 if (mOutputTracks[i]->thread() == thread) { 4549 mOutputTracks[i]->destroy(); 4550 mOutputTracks.removeAt(i); 4551 updateWaitTime_l(); 4552 return; 4553 } 4554 } 4555 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4556} 4557 4558// caller must hold mLock 4559void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4560{ 4561 mWaitTimeMs = UINT_MAX; 4562 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4563 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4564 if (strong != 0) { 4565 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4566 if (waitTimeMs < mWaitTimeMs) { 4567 mWaitTimeMs = waitTimeMs; 4568 } 4569 } 4570 } 4571} 4572 4573 4574bool AudioFlinger::DuplicatingThread::outputsReady( 4575 const SortedVector< sp<OutputTrack> > &outputTracks) 4576{ 4577 for (size_t i = 0; i < outputTracks.size(); i++) { 4578 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4579 if (thread == 0) { 4580 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4581 outputTracks[i].get()); 4582 return false; 4583 } 4584 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4585 // see note at standby() declaration 4586 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4587 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4588 thread.get()); 4589 return false; 4590 } 4591 } 4592 return true; 4593} 4594 4595uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4596{ 4597 return (mWaitTimeMs * 1000) / 2; 4598} 4599 4600void AudioFlinger::DuplicatingThread::cacheParameters_l() 4601{ 4602 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4603 updateWaitTime_l(); 4604 4605 MixerThread::cacheParameters_l(); 4606} 4607 4608// ---------------------------------------------------------------------------- 4609// Record 4610// ---------------------------------------------------------------------------- 4611 4612AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4613 AudioStreamIn *input, 4614 audio_io_handle_t id, 4615 audio_devices_t outDevice, 4616 audio_devices_t inDevice 4617#ifdef TEE_SINK 4618 , const sp<NBAIO_Sink>& teeSink 4619#endif 4620 ) : 4621 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4622 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4623 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4624 mRsmpInRear(0) 4625#ifdef TEE_SINK 4626 , mTeeSink(teeSink) 4627#endif 4628 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4629 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4630{ 4631 snprintf(mName, kNameLength, "AudioIn_%X", id); 4632 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4633 4634 readInputParameters_l(); 4635} 4636 4637 4638AudioFlinger::RecordThread::~RecordThread() 4639{ 4640 mAudioFlinger->unregisterWriter(mNBLogWriter); 4641 delete[] mRsmpInBuffer; 4642} 4643 4644void AudioFlinger::RecordThread::onFirstRef() 4645{ 4646 run(mName, PRIORITY_URGENT_AUDIO); 4647} 4648 4649bool AudioFlinger::RecordThread::threadLoop() 4650{ 4651 nsecs_t lastWarning = 0; 4652 4653 inputStandBy(); 4654 4655reacquire_wakelock: 4656 sp<RecordTrack> activeTrack; 4657 int activeTracksGen; 4658 { 4659 Mutex::Autolock _l(mLock); 4660 size_t size = mActiveTracks.size(); 4661 activeTracksGen = mActiveTracksGen; 4662 if (size > 0) { 4663 // FIXME an arbitrary choice 4664 activeTrack = mActiveTracks[0]; 4665 acquireWakeLock_l(activeTrack->uid()); 4666 if (size > 1) { 4667 SortedVector<int> tmp; 4668 for (size_t i = 0; i < size; i++) { 4669 tmp.add(mActiveTracks[i]->uid()); 4670 } 4671 updateWakeLockUids_l(tmp); 4672 } 4673 } else { 4674 acquireWakeLock_l(-1); 4675 } 4676 } 4677 4678 // used to request a deferred sleep, to be executed later while mutex is unlocked 4679 uint32_t sleepUs = 0; 4680 4681 // loop while there is work to do 4682 for (;;) { 4683 Vector< sp<EffectChain> > effectChains; 4684 4685 // sleep with mutex unlocked 4686 if (sleepUs > 0) { 4687 usleep(sleepUs); 4688 sleepUs = 0; 4689 } 4690 4691 // activeTracks accumulates a copy of a subset of mActiveTracks 4692 Vector< sp<RecordTrack> > activeTracks; 4693 4694 4695 { // scope for mLock 4696 Mutex::Autolock _l(mLock); 4697 4698 processConfigEvents_l(); 4699 4700 // check exitPending here because checkForNewParameters_l() and 4701 // checkForNewParameters_l() can temporarily release mLock 4702 if (exitPending()) { 4703 break; 4704 } 4705 4706 // if no active track(s), then standby and release wakelock 4707 size_t size = mActiveTracks.size(); 4708 if (size == 0) { 4709 standbyIfNotAlreadyInStandby(); 4710 // exitPending() can't become true here 4711 releaseWakeLock_l(); 4712 ALOGV("RecordThread: loop stopping"); 4713 // go to sleep 4714 mWaitWorkCV.wait(mLock); 4715 ALOGV("RecordThread: loop starting"); 4716 goto reacquire_wakelock; 4717 } 4718 4719 if (mActiveTracksGen != activeTracksGen) { 4720 activeTracksGen = mActiveTracksGen; 4721 SortedVector<int> tmp; 4722 for (size_t i = 0; i < size; i++) { 4723 tmp.add(mActiveTracks[i]->uid()); 4724 } 4725 updateWakeLockUids_l(tmp); 4726 } 4727 4728 bool doBroadcast = false; 4729 for (size_t i = 0; i < size; ) { 4730 4731 activeTrack = mActiveTracks[i]; 4732 if (activeTrack->isTerminated()) { 4733 removeTrack_l(activeTrack); 4734 mActiveTracks.remove(activeTrack); 4735 mActiveTracksGen++; 4736 size--; 4737 continue; 4738 } 4739 4740 TrackBase::track_state activeTrackState = activeTrack->mState; 4741 switch (activeTrackState) { 4742 4743 case TrackBase::PAUSING: 4744 mActiveTracks.remove(activeTrack); 4745 mActiveTracksGen++; 4746 doBroadcast = true; 4747 size--; 4748 continue; 4749 4750 case TrackBase::STARTING_1: 4751 sleepUs = 10000; 4752 i++; 4753 continue; 4754 4755 case TrackBase::STARTING_2: 4756 doBroadcast = true; 4757 mStandby = false; 4758 activeTrack->mState = TrackBase::ACTIVE; 4759 break; 4760 4761 case TrackBase::ACTIVE: 4762 break; 4763 4764 case TrackBase::IDLE: 4765 i++; 4766 continue; 4767 4768 default: 4769 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4770 } 4771 4772 activeTracks.add(activeTrack); 4773 i++; 4774 4775 } 4776 if (doBroadcast) { 4777 mStartStopCond.broadcast(); 4778 } 4779 4780 // sleep if there are no active tracks to process 4781 if (activeTracks.size() == 0) { 4782 if (sleepUs == 0) { 4783 sleepUs = kRecordThreadSleepUs; 4784 } 4785 continue; 4786 } 4787 sleepUs = 0; 4788 4789 lockEffectChains_l(effectChains); 4790 } 4791 4792 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4793 4794 size_t size = effectChains.size(); 4795 for (size_t i = 0; i < size; i++) { 4796 // thread mutex is not locked, but effect chain is locked 4797 effectChains[i]->process_l(); 4798 } 4799 4800 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4801 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4802 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4803 // If destination is non-contiguous, first read past the nominal end of buffer, then 4804 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4805 4806 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4807 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4808 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4809 if (bytesRead <= 0) { 4810 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4811 // Force input into standby so that it tries to recover at next read attempt 4812 inputStandBy(); 4813 sleepUs = kRecordThreadSleepUs; 4814 continue; 4815 } 4816 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4817 size_t framesRead = bytesRead / mFrameSize; 4818 ALOG_ASSERT(framesRead > 0); 4819 if (mTeeSink != 0) { 4820 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4821 } 4822 // If destination is non-contiguous, we now correct for reading past end of buffer. 4823 size_t part1 = mRsmpInFramesP2 - rear; 4824 if (framesRead > part1) { 4825 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4826 (framesRead - part1) * mFrameSize); 4827 } 4828 rear = mRsmpInRear += framesRead; 4829 4830 size = activeTracks.size(); 4831 // loop over each active track 4832 for (size_t i = 0; i < size; i++) { 4833 activeTrack = activeTracks[i]; 4834 4835 enum { 4836 OVERRUN_UNKNOWN, 4837 OVERRUN_TRUE, 4838 OVERRUN_FALSE 4839 } overrun = OVERRUN_UNKNOWN; 4840 4841 // loop over getNextBuffer to handle circular sink 4842 for (;;) { 4843 4844 activeTrack->mSink.frameCount = ~0; 4845 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4846 size_t framesOut = activeTrack->mSink.frameCount; 4847 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4848 4849 int32_t front = activeTrack->mRsmpInFront; 4850 ssize_t filled = rear - front; 4851 size_t framesIn; 4852 4853 if (filled < 0) { 4854 // should not happen, but treat like a massive overrun and re-sync 4855 framesIn = 0; 4856 activeTrack->mRsmpInFront = rear; 4857 overrun = OVERRUN_TRUE; 4858 } else if ((size_t) filled <= mRsmpInFrames) { 4859 framesIn = (size_t) filled; 4860 } else { 4861 // client is not keeping up with server, but give it latest data 4862 framesIn = mRsmpInFrames; 4863 activeTrack->mRsmpInFront = front = rear - framesIn; 4864 overrun = OVERRUN_TRUE; 4865 } 4866 4867 if (framesOut == 0 || framesIn == 0) { 4868 break; 4869 } 4870 4871 if (activeTrack->mResampler == NULL) { 4872 // no resampling 4873 if (framesIn > framesOut) { 4874 framesIn = framesOut; 4875 } else { 4876 framesOut = framesIn; 4877 } 4878 int8_t *dst = activeTrack->mSink.i8; 4879 while (framesIn > 0) { 4880 front &= mRsmpInFramesP2 - 1; 4881 size_t part1 = mRsmpInFramesP2 - front; 4882 if (part1 > framesIn) { 4883 part1 = framesIn; 4884 } 4885 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4886 if (mChannelCount == activeTrack->mChannelCount) { 4887 memcpy(dst, src, part1 * mFrameSize); 4888 } else if (mChannelCount == 1) { 4889 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4890 part1); 4891 } else { 4892 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4893 part1); 4894 } 4895 dst += part1 * activeTrack->mFrameSize; 4896 front += part1; 4897 framesIn -= part1; 4898 } 4899 activeTrack->mRsmpInFront += framesOut; 4900 4901 } else { 4902 // resampling 4903 // FIXME framesInNeeded should really be part of resampler API, and should 4904 // depend on the SRC ratio 4905 // to keep mRsmpInBuffer full so resampler always has sufficient input 4906 size_t framesInNeeded; 4907 // FIXME only re-calculate when it changes, and optimize for common ratios 4908 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4909 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4910 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4911 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4912 framesInNeeded, framesOut, inOverOut); 4913 // Although we theoretically have framesIn in circular buffer, some of those are 4914 // unreleased frames, and thus must be discounted for purpose of budgeting. 4915 size_t unreleased = activeTrack->mRsmpInUnrel; 4916 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4917 if (framesIn < framesInNeeded) { 4918 ALOGV("not enough to resample: have %u frames in but need %u in to " 4919 "produce %u out given in/out ratio of %.4g", 4920 framesIn, framesInNeeded, framesOut, inOverOut); 4921 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4922 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4923 if (newFramesOut == 0) { 4924 break; 4925 } 4926 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4927 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4928 framesInNeeded, newFramesOut, outOverIn); 4929 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4930 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4931 "given in/out ratio of %.4g", 4932 framesIn, framesInNeeded, newFramesOut, inOverOut); 4933 framesOut = newFramesOut; 4934 } else { 4935 ALOGV("success 1: have %u in and need %u in to produce %u out " 4936 "given in/out ratio of %.4g", 4937 framesIn, framesInNeeded, framesOut, inOverOut); 4938 } 4939 4940 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4941 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4942 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4943 delete[] activeTrack->mRsmpOutBuffer; 4944 // resampler always outputs stereo 4945 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4946 activeTrack->mRsmpOutFrameCount = framesOut; 4947 } 4948 4949 // resampler accumulates, but we only have one source track 4950 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4951 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4952 // FIXME how about having activeTrack implement this interface itself? 4953 activeTrack->mResamplerBufferProvider 4954 /*this*/ /* AudioBufferProvider* */); 4955 // ditherAndClamp() works as long as all buffers returned by 4956 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4957 if (activeTrack->mChannelCount == 1) { 4958 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 4959 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4960 framesOut); 4961 // the resampler always outputs stereo samples: 4962 // do post stereo to mono conversion 4963 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4964 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4965 } else { 4966 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4967 activeTrack->mRsmpOutBuffer, framesOut); 4968 } 4969 // now done with mRsmpOutBuffer 4970 4971 } 4972 4973 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4974 overrun = OVERRUN_FALSE; 4975 } 4976 4977 if (activeTrack->mFramesToDrop == 0) { 4978 if (framesOut > 0) { 4979 activeTrack->mSink.frameCount = framesOut; 4980 activeTrack->releaseBuffer(&activeTrack->mSink); 4981 } 4982 } else { 4983 // FIXME could do a partial drop of framesOut 4984 if (activeTrack->mFramesToDrop > 0) { 4985 activeTrack->mFramesToDrop -= framesOut; 4986 if (activeTrack->mFramesToDrop <= 0) { 4987 activeTrack->clearSyncStartEvent(); 4988 } 4989 } else { 4990 activeTrack->mFramesToDrop += framesOut; 4991 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4992 activeTrack->mSyncStartEvent->isCancelled()) { 4993 ALOGW("Synced record %s, session %d, trigger session %d", 4994 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 4995 activeTrack->sessionId(), 4996 (activeTrack->mSyncStartEvent != 0) ? 4997 activeTrack->mSyncStartEvent->triggerSession() : 0); 4998 activeTrack->clearSyncStartEvent(); 4999 } 5000 } 5001 } 5002 5003 if (framesOut == 0) { 5004 break; 5005 } 5006 } 5007 5008 switch (overrun) { 5009 case OVERRUN_TRUE: 5010 // client isn't retrieving buffers fast enough 5011 if (!activeTrack->setOverflow()) { 5012 nsecs_t now = systemTime(); 5013 // FIXME should lastWarning per track? 5014 if ((now - lastWarning) > kWarningThrottleNs) { 5015 ALOGW("RecordThread: buffer overflow"); 5016 lastWarning = now; 5017 } 5018 } 5019 break; 5020 case OVERRUN_FALSE: 5021 activeTrack->clearOverflow(); 5022 break; 5023 case OVERRUN_UNKNOWN: 5024 break; 5025 } 5026 5027 } 5028 5029 // enable changes in effect chain 5030 unlockEffectChains(effectChains); 5031 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5032 } 5033 5034 standbyIfNotAlreadyInStandby(); 5035 5036 { 5037 Mutex::Autolock _l(mLock); 5038 for (size_t i = 0; i < mTracks.size(); i++) { 5039 sp<RecordTrack> track = mTracks[i]; 5040 track->invalidate(); 5041 } 5042 mActiveTracks.clear(); 5043 mActiveTracksGen++; 5044 mStartStopCond.broadcast(); 5045 } 5046 5047 releaseWakeLock(); 5048 5049 ALOGV("RecordThread %p exiting", this); 5050 return false; 5051} 5052 5053void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5054{ 5055 if (!mStandby) { 5056 inputStandBy(); 5057 mStandby = true; 5058 } 5059} 5060 5061void AudioFlinger::RecordThread::inputStandBy() 5062{ 5063 mInput->stream->common.standby(&mInput->stream->common); 5064} 5065 5066// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5067sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5068 const sp<AudioFlinger::Client>& client, 5069 uint32_t sampleRate, 5070 audio_format_t format, 5071 audio_channel_mask_t channelMask, 5072 size_t *pFrameCount, 5073 int sessionId, 5074 int uid, 5075 IAudioFlinger::track_flags_t *flags, 5076 pid_t tid, 5077 status_t *status) 5078{ 5079 size_t frameCount = *pFrameCount; 5080 sp<RecordTrack> track; 5081 status_t lStatus; 5082 5083 // client expresses a preference for FAST, but we get the final say 5084 if (*flags & IAudioFlinger::TRACK_FAST) { 5085 if ( 5086 // use case: callback handler and frame count is default or at least as large as HAL 5087 ( 5088 (tid != -1) && 5089 ((frameCount == 0) || 5090 // FIXME not necessarily true, should be native frame count for native SR! 5091 (frameCount >= mFrameCount)) 5092 ) && 5093 // PCM data 5094 audio_is_linear_pcm(format) && 5095 // mono or stereo 5096 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5097 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5098 // hardware sample rate 5099 // FIXME actually the native hardware sample rate 5100 (sampleRate == mSampleRate) && 5101 // record thread has an associated fast capture 5102 hasFastCapture() 5103 // fast capture does not require slots 5104 ) { 5105 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5106 if (frameCount == 0) { 5107 // FIXME wrong mFrameCount 5108 frameCount = mFrameCount * kFastTrackMultiplier; 5109 } 5110 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5111 frameCount, mFrameCount); 5112 } else { 5113 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5114 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5115 "hasFastCapture=%d tid=%d", 5116 frameCount, mFrameCount, format, 5117 audio_is_linear_pcm(format), 5118 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5119 *flags &= ~IAudioFlinger::TRACK_FAST; 5120 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5121 // For compatibility with AudioRecord calculation, buffer depth is forced 5122 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5123 // This is probably too conservative, but legacy application code may depend on it. 5124 // If you change this calculation, also review the start threshold which is related. 5125 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5126 size_t mNormalFrameCount = 2048; // FIXME 5127 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5128 if (minBufCount < 2) { 5129 minBufCount = 2; 5130 } 5131 size_t minFrameCount = mNormalFrameCount * minBufCount; 5132 if (frameCount < minFrameCount) { 5133 frameCount = minFrameCount; 5134 } 5135 } 5136 } 5137 *pFrameCount = frameCount; 5138 5139 lStatus = initCheck(); 5140 if (lStatus != NO_ERROR) { 5141 ALOGE("createRecordTrack_l() audio driver not initialized"); 5142 goto Exit; 5143 } 5144 5145 { // scope for mLock 5146 Mutex::Autolock _l(mLock); 5147 5148 track = new RecordTrack(this, client, sampleRate, 5149 format, channelMask, frameCount, sessionId, uid, 5150 *flags); 5151 5152 lStatus = track->initCheck(); 5153 if (lStatus != NO_ERROR) { 5154 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5155 // track must be cleared from the caller as the caller has the AF lock 5156 goto Exit; 5157 } 5158 mTracks.add(track); 5159 5160 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5161 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5162 mAudioFlinger->btNrecIsOff(); 5163 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5164 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5165 5166 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5167 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5168 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5169 // so ask activity manager to do this on our behalf 5170 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5171 } 5172 } 5173 5174 lStatus = NO_ERROR; 5175 5176Exit: 5177 *status = lStatus; 5178 return track; 5179} 5180 5181status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5182 AudioSystem::sync_event_t event, 5183 int triggerSession) 5184{ 5185 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5186 sp<ThreadBase> strongMe = this; 5187 status_t status = NO_ERROR; 5188 5189 if (event == AudioSystem::SYNC_EVENT_NONE) { 5190 recordTrack->clearSyncStartEvent(); 5191 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5192 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5193 triggerSession, 5194 recordTrack->sessionId(), 5195 syncStartEventCallback, 5196 recordTrack); 5197 // Sync event can be cancelled by the trigger session if the track is not in a 5198 // compatible state in which case we start record immediately 5199 if (recordTrack->mSyncStartEvent->isCancelled()) { 5200 recordTrack->clearSyncStartEvent(); 5201 } else { 5202 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5203 recordTrack->mFramesToDrop = - 5204 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5205 } 5206 } 5207 5208 { 5209 // This section is a rendezvous between binder thread executing start() and RecordThread 5210 AutoMutex lock(mLock); 5211 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5212 if (recordTrack->mState == TrackBase::PAUSING) { 5213 ALOGV("active record track PAUSING -> ACTIVE"); 5214 recordTrack->mState = TrackBase::ACTIVE; 5215 } else { 5216 ALOGV("active record track state %d", recordTrack->mState); 5217 } 5218 return status; 5219 } 5220 5221 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5222 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5223 // or using a separate command thread 5224 recordTrack->mState = TrackBase::STARTING_1; 5225 mActiveTracks.add(recordTrack); 5226 mActiveTracksGen++; 5227 mLock.unlock(); 5228 status_t status = AudioSystem::startInput(mId); 5229 mLock.lock(); 5230 // FIXME should verify that recordTrack is still in mActiveTracks 5231 if (status != NO_ERROR) { 5232 mActiveTracks.remove(recordTrack); 5233 mActiveTracksGen++; 5234 recordTrack->clearSyncStartEvent(); 5235 return status; 5236 } 5237 // Catch up with current buffer indices if thread is already running. 5238 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5239 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5240 // see previously buffered data before it called start(), but with greater risk of overrun. 5241 5242 recordTrack->mRsmpInFront = mRsmpInRear; 5243 recordTrack->mRsmpInUnrel = 0; 5244 // FIXME why reset? 5245 if (recordTrack->mResampler != NULL) { 5246 recordTrack->mResampler->reset(); 5247 } 5248 recordTrack->mState = TrackBase::STARTING_2; 5249 // signal thread to start 5250 mWaitWorkCV.broadcast(); 5251 if (mActiveTracks.indexOf(recordTrack) < 0) { 5252 ALOGV("Record failed to start"); 5253 status = BAD_VALUE; 5254 goto startError; 5255 } 5256 return status; 5257 } 5258 5259startError: 5260 AudioSystem::stopInput(mId); 5261 recordTrack->clearSyncStartEvent(); 5262 // FIXME I wonder why we do not reset the state here? 5263 return status; 5264} 5265 5266void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5267{ 5268 sp<SyncEvent> strongEvent = event.promote(); 5269 5270 if (strongEvent != 0) { 5271 sp<RefBase> ptr = strongEvent->cookie().promote(); 5272 if (ptr != 0) { 5273 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5274 recordTrack->handleSyncStartEvent(strongEvent); 5275 } 5276 } 5277} 5278 5279bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5280 ALOGV("RecordThread::stop"); 5281 AutoMutex _l(mLock); 5282 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5283 return false; 5284 } 5285 // note that threadLoop may still be processing the track at this point [without lock] 5286 recordTrack->mState = TrackBase::PAUSING; 5287 // do not wait for mStartStopCond if exiting 5288 if (exitPending()) { 5289 return true; 5290 } 5291 // FIXME incorrect usage of wait: no explicit predicate or loop 5292 mStartStopCond.wait(mLock); 5293 // if we have been restarted, recordTrack is in mActiveTracks here 5294 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5295 ALOGV("Record stopped OK"); 5296 return true; 5297 } 5298 return false; 5299} 5300 5301bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5302{ 5303 return false; 5304} 5305 5306status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5307{ 5308#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5309 if (!isValidSyncEvent(event)) { 5310 return BAD_VALUE; 5311 } 5312 5313 int eventSession = event->triggerSession(); 5314 status_t ret = NAME_NOT_FOUND; 5315 5316 Mutex::Autolock _l(mLock); 5317 5318 for (size_t i = 0; i < mTracks.size(); i++) { 5319 sp<RecordTrack> track = mTracks[i]; 5320 if (eventSession == track->sessionId()) { 5321 (void) track->setSyncEvent(event); 5322 ret = NO_ERROR; 5323 } 5324 } 5325 return ret; 5326#else 5327 return BAD_VALUE; 5328#endif 5329} 5330 5331// destroyTrack_l() must be called with ThreadBase::mLock held 5332void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5333{ 5334 track->terminate(); 5335 track->mState = TrackBase::STOPPED; 5336 // active tracks are removed by threadLoop() 5337 if (mActiveTracks.indexOf(track) < 0) { 5338 removeTrack_l(track); 5339 } 5340} 5341 5342void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5343{ 5344 mTracks.remove(track); 5345 // need anything related to effects here? 5346} 5347 5348void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5349{ 5350 dumpInternals(fd, args); 5351 dumpTracks(fd, args); 5352 dumpEffectChains(fd, args); 5353} 5354 5355void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5356{ 5357 fdprintf(fd, "\nInput thread %p:\n", this); 5358 5359 if (mActiveTracks.size() > 0) { 5360 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5361 } else { 5362 fdprintf(fd, " No active record clients\n"); 5363 } 5364 5365 dumpBase(fd, args); 5366} 5367 5368void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5369{ 5370 const size_t SIZE = 256; 5371 char buffer[SIZE]; 5372 String8 result; 5373 5374 size_t numtracks = mTracks.size(); 5375 size_t numactive = mActiveTracks.size(); 5376 size_t numactiveseen = 0; 5377 fdprintf(fd, " %d Tracks", numtracks); 5378 if (numtracks) { 5379 fdprintf(fd, " of which %d are active\n", numactive); 5380 RecordTrack::appendDumpHeader(result); 5381 for (size_t i = 0; i < numtracks ; ++i) { 5382 sp<RecordTrack> track = mTracks[i]; 5383 if (track != 0) { 5384 bool active = mActiveTracks.indexOf(track) >= 0; 5385 if (active) { 5386 numactiveseen++; 5387 } 5388 track->dump(buffer, SIZE, active); 5389 result.append(buffer); 5390 } 5391 } 5392 } else { 5393 fdprintf(fd, "\n"); 5394 } 5395 5396 if (numactiveseen != numactive) { 5397 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5398 " not in the track list\n"); 5399 result.append(buffer); 5400 RecordTrack::appendDumpHeader(result); 5401 for (size_t i = 0; i < numactive; ++i) { 5402 sp<RecordTrack> track = mActiveTracks[i]; 5403 if (mTracks.indexOf(track) < 0) { 5404 track->dump(buffer, SIZE, true); 5405 result.append(buffer); 5406 } 5407 } 5408 5409 } 5410 write(fd, result.string(), result.size()); 5411} 5412 5413// AudioBufferProvider interface 5414status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5415 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5416{ 5417 RecordTrack *activeTrack = mRecordTrack; 5418 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5419 if (threadBase == 0) { 5420 buffer->frameCount = 0; 5421 buffer->raw = NULL; 5422 return NOT_ENOUGH_DATA; 5423 } 5424 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5425 int32_t rear = recordThread->mRsmpInRear; 5426 int32_t front = activeTrack->mRsmpInFront; 5427 ssize_t filled = rear - front; 5428 // FIXME should not be P2 (don't want to increase latency) 5429 // FIXME if client not keeping up, discard 5430 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5431 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5432 front &= recordThread->mRsmpInFramesP2 - 1; 5433 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5434 if (part1 > (size_t) filled) { 5435 part1 = filled; 5436 } 5437 size_t ask = buffer->frameCount; 5438 ALOG_ASSERT(ask > 0); 5439 if (part1 > ask) { 5440 part1 = ask; 5441 } 5442 if (part1 == 0) { 5443 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5444 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5445 buffer->raw = NULL; 5446 buffer->frameCount = 0; 5447 activeTrack->mRsmpInUnrel = 0; 5448 return NOT_ENOUGH_DATA; 5449 } 5450 5451 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5452 buffer->frameCount = part1; 5453 activeTrack->mRsmpInUnrel = part1; 5454 return NO_ERROR; 5455} 5456 5457// AudioBufferProvider interface 5458void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5459 AudioBufferProvider::Buffer* buffer) 5460{ 5461 RecordTrack *activeTrack = mRecordTrack; 5462 size_t stepCount = buffer->frameCount; 5463 if (stepCount == 0) { 5464 return; 5465 } 5466 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5467 activeTrack->mRsmpInUnrel -= stepCount; 5468 activeTrack->mRsmpInFront += stepCount; 5469 buffer->raw = NULL; 5470 buffer->frameCount = 0; 5471} 5472 5473bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5474 status_t& status) 5475{ 5476 bool reconfig = false; 5477 5478 status = NO_ERROR; 5479 5480 audio_format_t reqFormat = mFormat; 5481 uint32_t samplingRate = mSampleRate; 5482 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5483 5484 AudioParameter param = AudioParameter(keyValuePair); 5485 int value; 5486 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5487 // channel count change can be requested. Do we mandate the first client defines the 5488 // HAL sampling rate and channel count or do we allow changes on the fly? 5489 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5490 samplingRate = value; 5491 reconfig = true; 5492 } 5493 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5494 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5495 status = BAD_VALUE; 5496 } else { 5497 reqFormat = (audio_format_t) value; 5498 reconfig = true; 5499 } 5500 } 5501 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5502 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5503 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5504 status = BAD_VALUE; 5505 } else { 5506 channelMask = mask; 5507 reconfig = true; 5508 } 5509 } 5510 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5511 // do not accept frame count changes if tracks are open as the track buffer 5512 // size depends on frame count and correct behavior would not be guaranteed 5513 // if frame count is changed after track creation 5514 if (mActiveTracks.size() > 0) { 5515 status = INVALID_OPERATION; 5516 } else { 5517 reconfig = true; 5518 } 5519 } 5520 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5521 // forward device change to effects that have requested to be 5522 // aware of attached audio device. 5523 for (size_t i = 0; i < mEffectChains.size(); i++) { 5524 mEffectChains[i]->setDevice_l(value); 5525 } 5526 5527 // store input device and output device but do not forward output device to audio HAL. 5528 // Note that status is ignored by the caller for output device 5529 // (see AudioFlinger::setParameters() 5530 if (audio_is_output_devices(value)) { 5531 mOutDevice = value; 5532 status = BAD_VALUE; 5533 } else { 5534 mInDevice = value; 5535 // disable AEC and NS if the device is a BT SCO headset supporting those 5536 // pre processings 5537 if (mTracks.size() > 0) { 5538 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5539 mAudioFlinger->btNrecIsOff(); 5540 for (size_t i = 0; i < mTracks.size(); i++) { 5541 sp<RecordTrack> track = mTracks[i]; 5542 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5543 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5544 } 5545 } 5546 } 5547 } 5548 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5549 mAudioSource != (audio_source_t)value) { 5550 // forward device change to effects that have requested to be 5551 // aware of attached audio device. 5552 for (size_t i = 0; i < mEffectChains.size(); i++) { 5553 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5554 } 5555 mAudioSource = (audio_source_t)value; 5556 } 5557 5558 if (status == NO_ERROR) { 5559 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5560 keyValuePair.string()); 5561 if (status == INVALID_OPERATION) { 5562 inputStandBy(); 5563 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5564 keyValuePair.string()); 5565 } 5566 if (reconfig) { 5567 if (status == BAD_VALUE && 5568 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5569 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5570 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5571 <= (2 * samplingRate)) && 5572 audio_channel_count_from_in_mask( 5573 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5574 (channelMask == AUDIO_CHANNEL_IN_MONO || 5575 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5576 status = NO_ERROR; 5577 } 5578 if (status == NO_ERROR) { 5579 readInputParameters_l(); 5580 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5581 } 5582 } 5583 } 5584 5585 return reconfig; 5586} 5587 5588String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5589{ 5590 Mutex::Autolock _l(mLock); 5591 if (initCheck() != NO_ERROR) { 5592 return String8(); 5593 } 5594 5595 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5596 const String8 out_s8(s); 5597 free(s); 5598 return out_s8; 5599} 5600 5601void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5602 AudioSystem::OutputDescriptor desc; 5603 const void *param2 = NULL; 5604 5605 switch (event) { 5606 case AudioSystem::INPUT_OPENED: 5607 case AudioSystem::INPUT_CONFIG_CHANGED: 5608 desc.channelMask = mChannelMask; 5609 desc.samplingRate = mSampleRate; 5610 desc.format = mFormat; 5611 desc.frameCount = mFrameCount; 5612 desc.latency = 0; 5613 param2 = &desc; 5614 break; 5615 5616 case AudioSystem::INPUT_CLOSED: 5617 default: 5618 break; 5619 } 5620 mAudioFlinger->audioConfigChanged(event, mId, param2); 5621} 5622 5623void AudioFlinger::RecordThread::readInputParameters_l() 5624{ 5625 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5626 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5627 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5628 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5629 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5630 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5631 } 5632 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5633 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5634 mFrameCount = mBufferSize / mFrameSize; 5635 // This is the formula for calculating the temporary buffer size. 5636 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5637 // 1 full output buffer, regardless of the alignment of the available input. 5638 // The value is somewhat arbitrary, and could probably be even larger. 5639 // A larger value should allow more old data to be read after a track calls start(), 5640 // without increasing latency. 5641 mRsmpInFrames = mFrameCount * 7; 5642 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5643 delete[] mRsmpInBuffer; 5644 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5645 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5646 5647 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5648 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5649} 5650 5651uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5652{ 5653 Mutex::Autolock _l(mLock); 5654 if (initCheck() != NO_ERROR) { 5655 return 0; 5656 } 5657 5658 return mInput->stream->get_input_frames_lost(mInput->stream); 5659} 5660 5661uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5662{ 5663 Mutex::Autolock _l(mLock); 5664 uint32_t result = 0; 5665 if (getEffectChain_l(sessionId) != 0) { 5666 result = EFFECT_SESSION; 5667 } 5668 5669 for (size_t i = 0; i < mTracks.size(); ++i) { 5670 if (sessionId == mTracks[i]->sessionId()) { 5671 result |= TRACK_SESSION; 5672 break; 5673 } 5674 } 5675 5676 return result; 5677} 5678 5679KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5680{ 5681 KeyedVector<int, bool> ids; 5682 Mutex::Autolock _l(mLock); 5683 for (size_t j = 0; j < mTracks.size(); ++j) { 5684 sp<RecordThread::RecordTrack> track = mTracks[j]; 5685 int sessionId = track->sessionId(); 5686 if (ids.indexOfKey(sessionId) < 0) { 5687 ids.add(sessionId, true); 5688 } 5689 } 5690 return ids; 5691} 5692 5693AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5694{ 5695 Mutex::Autolock _l(mLock); 5696 AudioStreamIn *input = mInput; 5697 mInput = NULL; 5698 return input; 5699} 5700 5701// this method must always be called either with ThreadBase mLock held or inside the thread loop 5702audio_stream_t* AudioFlinger::RecordThread::stream() const 5703{ 5704 if (mInput == NULL) { 5705 return NULL; 5706 } 5707 return &mInput->stream->common; 5708} 5709 5710status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5711{ 5712 // only one chain per input thread 5713 if (mEffectChains.size() != 0) { 5714 return INVALID_OPERATION; 5715 } 5716 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5717 5718 chain->setInBuffer(NULL); 5719 chain->setOutBuffer(NULL); 5720 5721 checkSuspendOnAddEffectChain_l(chain); 5722 5723 mEffectChains.add(chain); 5724 5725 return NO_ERROR; 5726} 5727 5728size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5729{ 5730 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5731 ALOGW_IF(mEffectChains.size() != 1, 5732 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5733 chain.get(), mEffectChains.size(), this); 5734 if (mEffectChains.size() == 1) { 5735 mEffectChains.removeAt(0); 5736 } 5737 return 0; 5738} 5739 5740}; // namespace android 5741