Threads.cpp revision e348c5b72ad889389c7c1c900c121f0fbee221b5
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#include <audio_utils/minifloat.h> 39 40// NBAIO implementations 41#include <media/nbaio/AudioStreamOutSink.h> 42#include <media/nbaio/MonoPipe.h> 43#include <media/nbaio/MonoPipeReader.h> 44#include <media/nbaio/Pipe.h> 45#include <media/nbaio/PipeReader.h> 46#include <media/nbaio/SourceAudioBufferProvider.h> 47 48#include <powermanager/PowerManager.h> 49 50#include <common_time/cc_helper.h> 51#include <common_time/local_clock.h> 52 53#include "AudioFlinger.h" 54#include "AudioMixer.h" 55#include "FastMixer.h" 56#include "ServiceUtilities.h" 57#include "SchedulingPolicyService.h" 58 59#ifdef ADD_BATTERY_DATA 60#include <media/IMediaPlayerService.h> 61#include <media/IMediaDeathNotifier.h> 62#endif 63 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait in sendConfigEvent_l() for a status to be received 102static const nsecs_t kConfigEventTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal sink buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalSinkBufferSizeMs = 20; 111// maximum normal sink buffer size 112static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 113 114// Offloaded output thread standby delay: allows track transition without going to standby 115static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 116 117// Whether to use fast mixer 118static const enum { 119 FastMixer_Never, // never initialize or use: for debugging only 120 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 121 // normal mixer multiplier is 1 122 FastMixer_Static, // initialize if needed, then use all the time if initialized, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 125 // multiplier is calculated based on min & max normal mixer buffer size 126 // FIXME for FastMixer_Dynamic: 127 // Supporting this option will require fixing HALs that can't handle large writes. 128 // For example, one HAL implementation returns an error from a large write, 129 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 130 // We could either fix the HAL implementations, or provide a wrapper that breaks 131 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 132} kUseFastMixer = FastMixer_Static; 133 134// Priorities for requestPriority 135static const int kPriorityAudioApp = 2; 136static const int kPriorityFastMixer = 3; 137 138// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 139// for the track. The client then sub-divides this into smaller buffers for its use. 140// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 141// So for now we just assume that client is double-buffered for fast tracks. 142// FIXME It would be better for client to tell AudioFlinger the value of N, 143// so AudioFlinger could allocate the right amount of memory. 144// See the client's minBufCount and mNotificationFramesAct calculations for details. 145static const int kFastTrackMultiplier = 2; 146 147// See Thread::readOnlyHeap(). 148// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 149// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 150// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 151static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 152 153// ---------------------------------------------------------------------------- 154 155#ifdef ADD_BATTERY_DATA 156// To collect the amplifier usage 157static void addBatteryData(uint32_t params) { 158 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 159 if (service == NULL) { 160 // it already logged 161 return; 162 } 163 164 service->addBatteryData(params); 165} 166#endif 167 168 169// ---------------------------------------------------------------------------- 170// CPU Stats 171// ---------------------------------------------------------------------------- 172 173class CpuStats { 174public: 175 CpuStats(); 176 void sample(const String8 &title); 177#ifdef DEBUG_CPU_USAGE 178private: 179 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 180 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 181 182 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 183 184 int mCpuNum; // thread's current CPU number 185 int mCpukHz; // frequency of thread's current CPU in kHz 186#endif 187}; 188 189CpuStats::CpuStats() 190#ifdef DEBUG_CPU_USAGE 191 : mCpuNum(-1), mCpukHz(-1) 192#endif 193{ 194} 195 196void CpuStats::sample(const String8 &title 197#ifndef DEBUG_CPU_USAGE 198 __unused 199#endif 200 ) { 201#ifdef DEBUG_CPU_USAGE 202 // get current thread's delta CPU time in wall clock ns 203 double wcNs; 204 bool valid = mCpuUsage.sampleAndEnable(wcNs); 205 206 // record sample for wall clock statistics 207 if (valid) { 208 mWcStats.sample(wcNs); 209 } 210 211 // get the current CPU number 212 int cpuNum = sched_getcpu(); 213 214 // get the current CPU frequency in kHz 215 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 216 217 // check if either CPU number or frequency changed 218 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 219 mCpuNum = cpuNum; 220 mCpukHz = cpukHz; 221 // ignore sample for purposes of cycles 222 valid = false; 223 } 224 225 // if no change in CPU number or frequency, then record sample for cycle statistics 226 if (valid && mCpukHz > 0) { 227 double cycles = wcNs * cpukHz * 0.000001; 228 mHzStats.sample(cycles); 229 } 230 231 unsigned n = mWcStats.n(); 232 // mCpuUsage.elapsed() is expensive, so don't call it every loop 233 if ((n & 127) == 1) { 234 long long elapsed = mCpuUsage.elapsed(); 235 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 236 double perLoop = elapsed / (double) n; 237 double perLoop100 = perLoop * 0.01; 238 double perLoop1k = perLoop * 0.001; 239 double mean = mWcStats.mean(); 240 double stddev = mWcStats.stddev(); 241 double minimum = mWcStats.minimum(); 242 double maximum = mWcStats.maximum(); 243 double meanCycles = mHzStats.mean(); 244 double stddevCycles = mHzStats.stddev(); 245 double minCycles = mHzStats.minimum(); 246 double maxCycles = mHzStats.maximum(); 247 mCpuUsage.resetElapsed(); 248 mWcStats.reset(); 249 mHzStats.reset(); 250 ALOGD("CPU usage for %s over past %.1f secs\n" 251 " (%u mixer loops at %.1f mean ms per loop):\n" 252 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 253 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 254 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 255 title.string(), 256 elapsed * .000000001, n, perLoop * .000001, 257 mean * .001, 258 stddev * .001, 259 minimum * .001, 260 maximum * .001, 261 mean / perLoop100, 262 stddev / perLoop100, 263 minimum / perLoop100, 264 maximum / perLoop100, 265 meanCycles / perLoop1k, 266 stddevCycles / perLoop1k, 267 minCycles / perLoop1k, 268 maxCycles / perLoop1k); 269 270 } 271 } 272#endif 273}; 274 275// ---------------------------------------------------------------------------- 276// ThreadBase 277// ---------------------------------------------------------------------------- 278 279AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 280 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 281 : Thread(false /*canCallJava*/), 282 mType(type), 283 mAudioFlinger(audioFlinger), 284 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 285 // are set by PlaybackThread::readOutputParameters_l() or 286 // RecordThread::readInputParameters_l() 287 //FIXME: mStandby should be true here. Is this some kind of hack? 288 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 289 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 290 // mName will be set by concrete (non-virtual) subclass 291 mDeathRecipient(new PMDeathRecipient(this)) 292{ 293} 294 295AudioFlinger::ThreadBase::~ThreadBase() 296{ 297 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 298 mConfigEvents.clear(); 299 300 // do not lock the mutex in destructor 301 releaseWakeLock_l(); 302 if (mPowerManager != 0) { 303 sp<IBinder> binder = mPowerManager->asBinder(); 304 binder->unlinkToDeath(mDeathRecipient); 305 } 306} 307 308status_t AudioFlinger::ThreadBase::readyToRun() 309{ 310 status_t status = initCheck(); 311 if (status == NO_ERROR) { 312 ALOGI("AudioFlinger's thread %p ready to run", this); 313 } else { 314 ALOGE("No working audio driver found."); 315 } 316 return status; 317} 318 319void AudioFlinger::ThreadBase::exit() 320{ 321 ALOGV("ThreadBase::exit"); 322 // do any cleanup required for exit to succeed 323 preExit(); 324 { 325 // This lock prevents the following race in thread (uniprocessor for illustration): 326 // if (!exitPending()) { 327 // // context switch from here to exit() 328 // // exit() calls requestExit(), what exitPending() observes 329 // // exit() calls signal(), which is dropped since no waiters 330 // // context switch back from exit() to here 331 // mWaitWorkCV.wait(...); 332 // // now thread is hung 333 // } 334 AutoMutex lock(mLock); 335 requestExit(); 336 mWaitWorkCV.broadcast(); 337 } 338 // When Thread::requestExitAndWait is made virtual and this method is renamed to 339 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 340 requestExitAndWait(); 341} 342 343status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 344{ 345 status_t status; 346 347 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 348 Mutex::Autolock _l(mLock); 349 350 return sendSetParameterConfigEvent_l(keyValuePairs); 351} 352 353// sendConfigEvent_l() must be called with ThreadBase::mLock held 354// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 355status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 356{ 357 status_t status = NO_ERROR; 358 359 mConfigEvents.add(event); 360 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 361 mWaitWorkCV.signal(); 362 mLock.unlock(); 363 { 364 Mutex::Autolock _l(event->mLock); 365 while (event->mWaitStatus) { 366 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 367 event->mStatus = TIMED_OUT; 368 event->mWaitStatus = false; 369 } 370 } 371 status = event->mStatus; 372 } 373 mLock.lock(); 374 return status; 375} 376 377void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 378{ 379 Mutex::Autolock _l(mLock); 380 sendIoConfigEvent_l(event, param); 381} 382 383// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 384void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 385{ 386 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 387 sendConfigEvent_l(configEvent); 388} 389 390// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 391void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 392{ 393 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 394 sendConfigEvent_l(configEvent); 395} 396 397// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 398status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 399{ 400 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 401 return sendConfigEvent_l(configEvent); 402} 403 404// post condition: mConfigEvents.isEmpty() 405void AudioFlinger::ThreadBase::processConfigEvents_l() 406{ 407 bool configChanged = false; 408 409 while (!mConfigEvents.isEmpty()) { 410 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 411 sp<ConfigEvent> event = mConfigEvents[0]; 412 mConfigEvents.removeAt(0); 413 switch (event->mType) { 414 case CFG_EVENT_PRIO: { 415 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 416 // FIXME Need to understand why this has to be done asynchronously 417 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 418 true /*asynchronous*/); 419 if (err != 0) { 420 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 421 data->mPrio, data->mPid, data->mTid, err); 422 } 423 } break; 424 case CFG_EVENT_IO: { 425 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 426 audioConfigChanged(data->mEvent, data->mParam); 427 } break; 428 case CFG_EVENT_SET_PARAMETER: { 429 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 430 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 431 configChanged = true; 432 } 433 } break; 434 default: 435 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 436 break; 437 } 438 { 439 Mutex::Autolock _l(event->mLock); 440 if (event->mWaitStatus) { 441 event->mWaitStatus = false; 442 event->mCond.signal(); 443 } 444 } 445 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 446 } 447 448 if (configChanged) { 449 cacheParameters_l(); 450 } 451} 452 453String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 454 String8 s; 455 if (output) { 456 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 457 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 458 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 459 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 460 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 461 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 462 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 463 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 464 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 465 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 466 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 467 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 468 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 469 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 470 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 471 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 472 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 473 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 474 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 475 } else { 476 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 477 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 478 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 479 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 480 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 481 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 482 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 483 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 484 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 485 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 486 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 487 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 488 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 489 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 490 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 491 } 492 int len = s.length(); 493 if (s.length() > 2) { 494 char *str = s.lockBuffer(len); 495 s.unlockBuffer(len - 2); 496 } 497 return s; 498} 499 500void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 501{ 502 const size_t SIZE = 256; 503 char buffer[SIZE]; 504 String8 result; 505 506 bool locked = AudioFlinger::dumpTryLock(mLock); 507 if (!locked) { 508 fdprintf(fd, "thread %p maybe dead locked\n", this); 509 } 510 511 fdprintf(fd, " I/O handle: %d\n", mId); 512 fdprintf(fd, " TID: %d\n", getTid()); 513 fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 514 fdprintf(fd, " Sample rate: %u\n", mSampleRate); 515 fdprintf(fd, " HAL frame count: %zu\n", mFrameCount); 516 fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 517 fdprintf(fd, " Channel Count: %u\n", mChannelCount); 518 fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 519 channelMaskToString(mChannelMask, mType != RECORD).string()); 520 fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 521 fdprintf(fd, " Frame size: %zu\n", mFrameSize); 522 fdprintf(fd, " Pending config events:"); 523 size_t numConfig = mConfigEvents.size(); 524 if (numConfig) { 525 for (size_t i = 0; i < numConfig; i++) { 526 mConfigEvents[i]->dump(buffer, SIZE); 527 fdprintf(fd, "\n %s", buffer); 528 } 529 fdprintf(fd, "\n"); 530 } else { 531 fdprintf(fd, " none\n"); 532 } 533 534 if (locked) { 535 mLock.unlock(); 536 } 537} 538 539void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 540{ 541 const size_t SIZE = 256; 542 char buffer[SIZE]; 543 String8 result; 544 545 size_t numEffectChains = mEffectChains.size(); 546 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 547 write(fd, buffer, strlen(buffer)); 548 549 for (size_t i = 0; i < numEffectChains; ++i) { 550 sp<EffectChain> chain = mEffectChains[i]; 551 if (chain != 0) { 552 chain->dump(fd, args); 553 } 554 } 555} 556 557void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 558{ 559 Mutex::Autolock _l(mLock); 560 acquireWakeLock_l(uid); 561} 562 563String16 AudioFlinger::ThreadBase::getWakeLockTag() 564{ 565 switch (mType) { 566 case MIXER: 567 return String16("AudioMix"); 568 case DIRECT: 569 return String16("AudioDirectOut"); 570 case DUPLICATING: 571 return String16("AudioDup"); 572 case RECORD: 573 return String16("AudioIn"); 574 case OFFLOAD: 575 return String16("AudioOffload"); 576 default: 577 ALOG_ASSERT(false); 578 return String16("AudioUnknown"); 579 } 580} 581 582void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 583{ 584 getPowerManager_l(); 585 if (mPowerManager != 0) { 586 sp<IBinder> binder = new BBinder(); 587 status_t status; 588 if (uid >= 0) { 589 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 590 binder, 591 getWakeLockTag(), 592 String16("media"), 593 uid); 594 } else { 595 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 596 binder, 597 getWakeLockTag(), 598 String16("media")); 599 } 600 if (status == NO_ERROR) { 601 mWakeLockToken = binder; 602 } 603 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 604 } 605} 606 607void AudioFlinger::ThreadBase::releaseWakeLock() 608{ 609 Mutex::Autolock _l(mLock); 610 releaseWakeLock_l(); 611} 612 613void AudioFlinger::ThreadBase::releaseWakeLock_l() 614{ 615 if (mWakeLockToken != 0) { 616 ALOGV("releaseWakeLock_l() %s", mName); 617 if (mPowerManager != 0) { 618 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 619 } 620 mWakeLockToken.clear(); 621 } 622} 623 624void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 625 Mutex::Autolock _l(mLock); 626 updateWakeLockUids_l(uids); 627} 628 629void AudioFlinger::ThreadBase::getPowerManager_l() { 630 631 if (mPowerManager == 0) { 632 // use checkService() to avoid blocking if power service is not up yet 633 sp<IBinder> binder = 634 defaultServiceManager()->checkService(String16("power")); 635 if (binder == 0) { 636 ALOGW("Thread %s cannot connect to the power manager service", mName); 637 } else { 638 mPowerManager = interface_cast<IPowerManager>(binder); 639 binder->linkToDeath(mDeathRecipient); 640 } 641 } 642} 643 644void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 645 646 getPowerManager_l(); 647 if (mWakeLockToken == NULL) { 648 ALOGE("no wake lock to update!"); 649 return; 650 } 651 if (mPowerManager != 0) { 652 sp<IBinder> binder = new BBinder(); 653 status_t status; 654 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 655 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 656 } 657} 658 659void AudioFlinger::ThreadBase::clearPowerManager() 660{ 661 Mutex::Autolock _l(mLock); 662 releaseWakeLock_l(); 663 mPowerManager.clear(); 664} 665 666void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 667{ 668 sp<ThreadBase> thread = mThread.promote(); 669 if (thread != 0) { 670 thread->clearPowerManager(); 671 } 672 ALOGW("power manager service died !!!"); 673} 674 675void AudioFlinger::ThreadBase::setEffectSuspended( 676 const effect_uuid_t *type, bool suspend, int sessionId) 677{ 678 Mutex::Autolock _l(mLock); 679 setEffectSuspended_l(type, suspend, sessionId); 680} 681 682void AudioFlinger::ThreadBase::setEffectSuspended_l( 683 const effect_uuid_t *type, bool suspend, int sessionId) 684{ 685 sp<EffectChain> chain = getEffectChain_l(sessionId); 686 if (chain != 0) { 687 if (type != NULL) { 688 chain->setEffectSuspended_l(type, suspend); 689 } else { 690 chain->setEffectSuspendedAll_l(suspend); 691 } 692 } 693 694 updateSuspendedSessions_l(type, suspend, sessionId); 695} 696 697void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 698{ 699 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 700 if (index < 0) { 701 return; 702 } 703 704 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 705 mSuspendedSessions.valueAt(index); 706 707 for (size_t i = 0; i < sessionEffects.size(); i++) { 708 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 709 for (int j = 0; j < desc->mRefCount; j++) { 710 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 711 chain->setEffectSuspendedAll_l(true); 712 } else { 713 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 714 desc->mType.timeLow); 715 chain->setEffectSuspended_l(&desc->mType, true); 716 } 717 } 718 } 719} 720 721void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 722 bool suspend, 723 int sessionId) 724{ 725 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 726 727 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 728 729 if (suspend) { 730 if (index >= 0) { 731 sessionEffects = mSuspendedSessions.valueAt(index); 732 } else { 733 mSuspendedSessions.add(sessionId, sessionEffects); 734 } 735 } else { 736 if (index < 0) { 737 return; 738 } 739 sessionEffects = mSuspendedSessions.valueAt(index); 740 } 741 742 743 int key = EffectChain::kKeyForSuspendAll; 744 if (type != NULL) { 745 key = type->timeLow; 746 } 747 index = sessionEffects.indexOfKey(key); 748 749 sp<SuspendedSessionDesc> desc; 750 if (suspend) { 751 if (index >= 0) { 752 desc = sessionEffects.valueAt(index); 753 } else { 754 desc = new SuspendedSessionDesc(); 755 if (type != NULL) { 756 desc->mType = *type; 757 } 758 sessionEffects.add(key, desc); 759 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 760 } 761 desc->mRefCount++; 762 } else { 763 if (index < 0) { 764 return; 765 } 766 desc = sessionEffects.valueAt(index); 767 if (--desc->mRefCount == 0) { 768 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 769 sessionEffects.removeItemsAt(index); 770 if (sessionEffects.isEmpty()) { 771 ALOGV("updateSuspendedSessions_l() restore removing session %d", 772 sessionId); 773 mSuspendedSessions.removeItem(sessionId); 774 } 775 } 776 } 777 if (!sessionEffects.isEmpty()) { 778 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 779 } 780} 781 782void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 783 bool enabled, 784 int sessionId) 785{ 786 Mutex::Autolock _l(mLock); 787 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 788} 789 790void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 791 bool enabled, 792 int sessionId) 793{ 794 if (mType != RECORD) { 795 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 796 // another session. This gives the priority to well behaved effect control panels 797 // and applications not using global effects. 798 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 799 // global effects 800 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 801 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 802 } 803 } 804 805 sp<EffectChain> chain = getEffectChain_l(sessionId); 806 if (chain != 0) { 807 chain->checkSuspendOnEffectEnabled(effect, enabled); 808 } 809} 810 811// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 812sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 813 const sp<AudioFlinger::Client>& client, 814 const sp<IEffectClient>& effectClient, 815 int32_t priority, 816 int sessionId, 817 effect_descriptor_t *desc, 818 int *enabled, 819 status_t *status) 820{ 821 sp<EffectModule> effect; 822 sp<EffectHandle> handle; 823 status_t lStatus; 824 sp<EffectChain> chain; 825 bool chainCreated = false; 826 bool effectCreated = false; 827 bool effectRegistered = false; 828 829 lStatus = initCheck(); 830 if (lStatus != NO_ERROR) { 831 ALOGW("createEffect_l() Audio driver not initialized."); 832 goto Exit; 833 } 834 835 // Reject any effect on Direct output threads for now, since the format of 836 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 837 if (mType == DIRECT) { 838 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 839 desc->name, mName); 840 lStatus = BAD_VALUE; 841 goto Exit; 842 } 843 844 // Allow global effects only on offloaded and mixer threads 845 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 846 switch (mType) { 847 case MIXER: 848 case OFFLOAD: 849 break; 850 case DIRECT: 851 case DUPLICATING: 852 case RECORD: 853 default: 854 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 855 lStatus = BAD_VALUE; 856 goto Exit; 857 } 858 } 859 860 // Only Pre processor effects are allowed on input threads and only on input threads 861 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 862 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 863 desc->name, desc->flags, mType); 864 lStatus = BAD_VALUE; 865 goto Exit; 866 } 867 868 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 869 870 { // scope for mLock 871 Mutex::Autolock _l(mLock); 872 873 // check for existing effect chain with the requested audio session 874 chain = getEffectChain_l(sessionId); 875 if (chain == 0) { 876 // create a new chain for this session 877 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 878 chain = new EffectChain(this, sessionId); 879 addEffectChain_l(chain); 880 chain->setStrategy(getStrategyForSession_l(sessionId)); 881 chainCreated = true; 882 } else { 883 effect = chain->getEffectFromDesc_l(desc); 884 } 885 886 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 887 888 if (effect == 0) { 889 int id = mAudioFlinger->nextUniqueId(); 890 // Check CPU and memory usage 891 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 892 if (lStatus != NO_ERROR) { 893 goto Exit; 894 } 895 effectRegistered = true; 896 // create a new effect module if none present in the chain 897 effect = new EffectModule(this, chain, desc, id, sessionId); 898 lStatus = effect->status(); 899 if (lStatus != NO_ERROR) { 900 goto Exit; 901 } 902 effect->setOffloaded(mType == OFFLOAD, mId); 903 904 lStatus = chain->addEffect_l(effect); 905 if (lStatus != NO_ERROR) { 906 goto Exit; 907 } 908 effectCreated = true; 909 910 effect->setDevice(mOutDevice); 911 effect->setDevice(mInDevice); 912 effect->setMode(mAudioFlinger->getMode()); 913 effect->setAudioSource(mAudioSource); 914 } 915 // create effect handle and connect it to effect module 916 handle = new EffectHandle(effect, client, effectClient, priority); 917 lStatus = handle->initCheck(); 918 if (lStatus == OK) { 919 lStatus = effect->addHandle(handle.get()); 920 } 921 if (enabled != NULL) { 922 *enabled = (int)effect->isEnabled(); 923 } 924 } 925 926Exit: 927 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 928 Mutex::Autolock _l(mLock); 929 if (effectCreated) { 930 chain->removeEffect_l(effect); 931 } 932 if (effectRegistered) { 933 AudioSystem::unregisterEffect(effect->id()); 934 } 935 if (chainCreated) { 936 removeEffectChain_l(chain); 937 } 938 handle.clear(); 939 } 940 941 *status = lStatus; 942 return handle; 943} 944 945sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 946{ 947 Mutex::Autolock _l(mLock); 948 return getEffect_l(sessionId, effectId); 949} 950 951sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 952{ 953 sp<EffectChain> chain = getEffectChain_l(sessionId); 954 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 955} 956 957// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 958// PlaybackThread::mLock held 959status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 960{ 961 // check for existing effect chain with the requested audio session 962 int sessionId = effect->sessionId(); 963 sp<EffectChain> chain = getEffectChain_l(sessionId); 964 bool chainCreated = false; 965 966 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 967 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 968 this, effect->desc().name, effect->desc().flags); 969 970 if (chain == 0) { 971 // create a new chain for this session 972 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 973 chain = new EffectChain(this, sessionId); 974 addEffectChain_l(chain); 975 chain->setStrategy(getStrategyForSession_l(sessionId)); 976 chainCreated = true; 977 } 978 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 979 980 if (chain->getEffectFromId_l(effect->id()) != 0) { 981 ALOGW("addEffect_l() %p effect %s already present in chain %p", 982 this, effect->desc().name, chain.get()); 983 return BAD_VALUE; 984 } 985 986 effect->setOffloaded(mType == OFFLOAD, mId); 987 988 status_t status = chain->addEffect_l(effect); 989 if (status != NO_ERROR) { 990 if (chainCreated) { 991 removeEffectChain_l(chain); 992 } 993 return status; 994 } 995 996 effect->setDevice(mOutDevice); 997 effect->setDevice(mInDevice); 998 effect->setMode(mAudioFlinger->getMode()); 999 effect->setAudioSource(mAudioSource); 1000 return NO_ERROR; 1001} 1002 1003void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1004 1005 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1006 effect_descriptor_t desc = effect->desc(); 1007 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1008 detachAuxEffect_l(effect->id()); 1009 } 1010 1011 sp<EffectChain> chain = effect->chain().promote(); 1012 if (chain != 0) { 1013 // remove effect chain if removing last effect 1014 if (chain->removeEffect_l(effect) == 0) { 1015 removeEffectChain_l(chain); 1016 } 1017 } else { 1018 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1019 } 1020} 1021 1022void AudioFlinger::ThreadBase::lockEffectChains_l( 1023 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1024{ 1025 effectChains = mEffectChains; 1026 for (size_t i = 0; i < mEffectChains.size(); i++) { 1027 mEffectChains[i]->lock(); 1028 } 1029} 1030 1031void AudioFlinger::ThreadBase::unlockEffectChains( 1032 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1033{ 1034 for (size_t i = 0; i < effectChains.size(); i++) { 1035 effectChains[i]->unlock(); 1036 } 1037} 1038 1039sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1040{ 1041 Mutex::Autolock _l(mLock); 1042 return getEffectChain_l(sessionId); 1043} 1044 1045sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1046{ 1047 size_t size = mEffectChains.size(); 1048 for (size_t i = 0; i < size; i++) { 1049 if (mEffectChains[i]->sessionId() == sessionId) { 1050 return mEffectChains[i]; 1051 } 1052 } 1053 return 0; 1054} 1055 1056void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1057{ 1058 Mutex::Autolock _l(mLock); 1059 size_t size = mEffectChains.size(); 1060 for (size_t i = 0; i < size; i++) { 1061 mEffectChains[i]->setMode_l(mode); 1062 } 1063} 1064 1065void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1066 EffectHandle *handle, 1067 bool unpinIfLast) { 1068 1069 Mutex::Autolock _l(mLock); 1070 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1071 // delete the effect module if removing last handle on it 1072 if (effect->removeHandle(handle) == 0) { 1073 if (!effect->isPinned() || unpinIfLast) { 1074 removeEffect_l(effect); 1075 AudioSystem::unregisterEffect(effect->id()); 1076 } 1077 } 1078} 1079 1080// ---------------------------------------------------------------------------- 1081// Playback 1082// ---------------------------------------------------------------------------- 1083 1084AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1085 AudioStreamOut* output, 1086 audio_io_handle_t id, 1087 audio_devices_t device, 1088 type_t type) 1089 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1090 mNormalFrameCount(0), mSinkBuffer(NULL), 1091 mMixerBufferEnabled(false), 1092 mMixerBuffer(NULL), 1093 mMixerBufferSize(0), 1094 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1095 mMixerBufferValid(false), 1096 mEffectBufferEnabled(false), 1097 mEffectBuffer(NULL), 1098 mEffectBufferSize(0), 1099 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1100 mEffectBufferValid(false), 1101 mSuspended(0), mBytesWritten(0), 1102 mActiveTracksGeneration(0), 1103 // mStreamTypes[] initialized in constructor body 1104 mOutput(output), 1105 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1106 mMixerStatus(MIXER_IDLE), 1107 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1108 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1109 mBytesRemaining(0), 1110 mCurrentWriteLength(0), 1111 mUseAsyncWrite(false), 1112 mWriteAckSequence(0), 1113 mDrainSequence(0), 1114 mSignalPending(false), 1115 mScreenState(AudioFlinger::mScreenState), 1116 // index 0 is reserved for normal mixer's submix 1117 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1118 // mLatchD, mLatchQ, 1119 mLatchDValid(false), mLatchQValid(false) 1120{ 1121 snprintf(mName, kNameLength, "AudioOut_%X", id); 1122 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1123 1124 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1125 // it would be safer to explicitly pass initial masterVolume/masterMute as 1126 // parameter. 1127 // 1128 // If the HAL we are using has support for master volume or master mute, 1129 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1130 // and the mute set to false). 1131 mMasterVolume = audioFlinger->masterVolume_l(); 1132 mMasterMute = audioFlinger->masterMute_l(); 1133 if (mOutput && mOutput->audioHwDev) { 1134 if (mOutput->audioHwDev->canSetMasterVolume()) { 1135 mMasterVolume = 1.0; 1136 } 1137 1138 if (mOutput->audioHwDev->canSetMasterMute()) { 1139 mMasterMute = false; 1140 } 1141 } 1142 1143 readOutputParameters_l(); 1144 1145 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1146 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1147 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1148 stream = (audio_stream_type_t) (stream + 1)) { 1149 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1150 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1151 } 1152 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1153 // because mAudioFlinger doesn't have one to copy from 1154} 1155 1156AudioFlinger::PlaybackThread::~PlaybackThread() 1157{ 1158 mAudioFlinger->unregisterWriter(mNBLogWriter); 1159 free(mSinkBuffer); 1160 free(mMixerBuffer); 1161 free(mEffectBuffer); 1162} 1163 1164void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1165{ 1166 dumpInternals(fd, args); 1167 dumpTracks(fd, args); 1168 dumpEffectChains(fd, args); 1169} 1170 1171void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1172{ 1173 const size_t SIZE = 256; 1174 char buffer[SIZE]; 1175 String8 result; 1176 1177 result.appendFormat(" Stream volumes in dB: "); 1178 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1179 const stream_type_t *st = &mStreamTypes[i]; 1180 if (i > 0) { 1181 result.appendFormat(", "); 1182 } 1183 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1184 if (st->mute) { 1185 result.append("M"); 1186 } 1187 } 1188 result.append("\n"); 1189 write(fd, result.string(), result.length()); 1190 result.clear(); 1191 1192 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1193 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1194 fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1195 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1196 1197 size_t numtracks = mTracks.size(); 1198 size_t numactive = mActiveTracks.size(); 1199 fdprintf(fd, " %d Tracks", numtracks); 1200 size_t numactiveseen = 0; 1201 if (numtracks) { 1202 fdprintf(fd, " of which %d are active\n", numactive); 1203 Track::appendDumpHeader(result); 1204 for (size_t i = 0; i < numtracks; ++i) { 1205 sp<Track> track = mTracks[i]; 1206 if (track != 0) { 1207 bool active = mActiveTracks.indexOf(track) >= 0; 1208 if (active) { 1209 numactiveseen++; 1210 } 1211 track->dump(buffer, SIZE, active); 1212 result.append(buffer); 1213 } 1214 } 1215 } else { 1216 result.append("\n"); 1217 } 1218 if (numactiveseen != numactive) { 1219 // some tracks in the active list were not in the tracks list 1220 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1221 " not in the track list\n"); 1222 result.append(buffer); 1223 Track::appendDumpHeader(result); 1224 for (size_t i = 0; i < numactive; ++i) { 1225 sp<Track> track = mActiveTracks[i].promote(); 1226 if (track != 0 && mTracks.indexOf(track) < 0) { 1227 track->dump(buffer, SIZE, true); 1228 result.append(buffer); 1229 } 1230 } 1231 } 1232 1233 write(fd, result.string(), result.size()); 1234 1235} 1236 1237void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1238{ 1239 fdprintf(fd, "\nOutput thread %p:\n", this); 1240 fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1241 fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1242 fdprintf(fd, " Total writes: %d\n", mNumWrites); 1243 fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1244 fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1245 fdprintf(fd, " Suspend count: %d\n", mSuspended); 1246 fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1247 fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1248 fdprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1249 fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1250 1251 dumpBase(fd, args); 1252} 1253 1254// Thread virtuals 1255 1256void AudioFlinger::PlaybackThread::onFirstRef() 1257{ 1258 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1259} 1260 1261// ThreadBase virtuals 1262void AudioFlinger::PlaybackThread::preExit() 1263{ 1264 ALOGV(" preExit()"); 1265 // FIXME this is using hard-coded strings but in the future, this functionality will be 1266 // converted to use audio HAL extensions required to support tunneling 1267 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1268} 1269 1270// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1271sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1272 const sp<AudioFlinger::Client>& client, 1273 audio_stream_type_t streamType, 1274 uint32_t sampleRate, 1275 audio_format_t format, 1276 audio_channel_mask_t channelMask, 1277 size_t *pFrameCount, 1278 const sp<IMemory>& sharedBuffer, 1279 int sessionId, 1280 IAudioFlinger::track_flags_t *flags, 1281 pid_t tid, 1282 int uid, 1283 status_t *status) 1284{ 1285 size_t frameCount = *pFrameCount; 1286 sp<Track> track; 1287 status_t lStatus; 1288 1289 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1290 1291 // client expresses a preference for FAST, but we get the final say 1292 if (*flags & IAudioFlinger::TRACK_FAST) { 1293 if ( 1294 // not timed 1295 (!isTimed) && 1296 // either of these use cases: 1297 ( 1298 // use case 1: shared buffer with any frame count 1299 ( 1300 (sharedBuffer != 0) 1301 ) || 1302 // use case 2: callback handler and frame count is default or at least as large as HAL 1303 ( 1304 (tid != -1) && 1305 ((frameCount == 0) || 1306 (frameCount >= mFrameCount)) 1307 ) 1308 ) && 1309 // PCM data 1310 audio_is_linear_pcm(format) && 1311 // mono or stereo 1312 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1313 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1314 // hardware sample rate 1315 (sampleRate == mSampleRate) && 1316 // normal mixer has an associated fast mixer 1317 hasFastMixer() && 1318 // there are sufficient fast track slots available 1319 (mFastTrackAvailMask != 0) 1320 // FIXME test that MixerThread for this fast track has a capable output HAL 1321 // FIXME add a permission test also? 1322 ) { 1323 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1324 if (frameCount == 0) { 1325 frameCount = mFrameCount * kFastTrackMultiplier; 1326 } 1327 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1328 frameCount, mFrameCount); 1329 } else { 1330 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1331 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1332 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1333 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1334 audio_is_linear_pcm(format), 1335 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1336 *flags &= ~IAudioFlinger::TRACK_FAST; 1337 // For compatibility with AudioTrack calculation, buffer depth is forced 1338 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1339 // This is probably too conservative, but legacy application code may depend on it. 1340 // If you change this calculation, also review the start threshold which is related. 1341 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1342 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1343 if (minBufCount < 2) { 1344 minBufCount = 2; 1345 } 1346 size_t minFrameCount = mNormalFrameCount * minBufCount; 1347 if (frameCount < minFrameCount) { 1348 frameCount = minFrameCount; 1349 } 1350 } 1351 } 1352 *pFrameCount = frameCount; 1353 1354 switch (mType) { 1355 1356 case DIRECT: 1357 if (audio_is_linear_pcm(format)) { 1358 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1359 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1360 "for output %p with format %#x", 1361 sampleRate, format, channelMask, mOutput, mFormat); 1362 lStatus = BAD_VALUE; 1363 goto Exit; 1364 } 1365 } 1366 break; 1367 1368 case OFFLOAD: 1369 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1370 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1371 "for output %p with format %#x", 1372 sampleRate, format, channelMask, mOutput, mFormat); 1373 lStatus = BAD_VALUE; 1374 goto Exit; 1375 } 1376 break; 1377 1378 default: 1379 if (!audio_is_linear_pcm(format)) { 1380 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1381 "for output %p with format %#x", 1382 format, mOutput, mFormat); 1383 lStatus = BAD_VALUE; 1384 goto Exit; 1385 } 1386 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1387 if (sampleRate > mSampleRate*2) { 1388 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1389 lStatus = BAD_VALUE; 1390 goto Exit; 1391 } 1392 break; 1393 1394 } 1395 1396 lStatus = initCheck(); 1397 if (lStatus != NO_ERROR) { 1398 ALOGE("createTrack_l() audio driver not initialized"); 1399 goto Exit; 1400 } 1401 1402 { // scope for mLock 1403 Mutex::Autolock _l(mLock); 1404 1405 // all tracks in same audio session must share the same routing strategy otherwise 1406 // conflicts will happen when tracks are moved from one output to another by audio policy 1407 // manager 1408 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1409 for (size_t i = 0; i < mTracks.size(); ++i) { 1410 sp<Track> t = mTracks[i]; 1411 if (t != 0 && !t->isOutputTrack()) { 1412 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1413 if (sessionId == t->sessionId() && strategy != actual) { 1414 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1415 strategy, actual); 1416 lStatus = BAD_VALUE; 1417 goto Exit; 1418 } 1419 } 1420 } 1421 1422 if (!isTimed) { 1423 track = new Track(this, client, streamType, sampleRate, format, 1424 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1425 } else { 1426 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1427 channelMask, frameCount, sharedBuffer, sessionId, uid); 1428 } 1429 1430 // new Track always returns non-NULL, 1431 // but TimedTrack::create() is a factory that could fail by returning NULL 1432 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1433 if (lStatus != NO_ERROR) { 1434 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1435 // track must be cleared from the caller as the caller has the AF lock 1436 goto Exit; 1437 } 1438 mTracks.add(track); 1439 1440 sp<EffectChain> chain = getEffectChain_l(sessionId); 1441 if (chain != 0) { 1442 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1443 track->setMainBuffer(chain->inBuffer()); 1444 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1445 chain->incTrackCnt(); 1446 } 1447 1448 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1449 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1450 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1451 // so ask activity manager to do this on our behalf 1452 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1453 } 1454 } 1455 1456 lStatus = NO_ERROR; 1457 1458Exit: 1459 *status = lStatus; 1460 return track; 1461} 1462 1463uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1464{ 1465 return latency; 1466} 1467 1468uint32_t AudioFlinger::PlaybackThread::latency() const 1469{ 1470 Mutex::Autolock _l(mLock); 1471 return latency_l(); 1472} 1473uint32_t AudioFlinger::PlaybackThread::latency_l() const 1474{ 1475 if (initCheck() == NO_ERROR) { 1476 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1477 } else { 1478 return 0; 1479 } 1480} 1481 1482void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1483{ 1484 Mutex::Autolock _l(mLock); 1485 // Don't apply master volume in SW if our HAL can do it for us. 1486 if (mOutput && mOutput->audioHwDev && 1487 mOutput->audioHwDev->canSetMasterVolume()) { 1488 mMasterVolume = 1.0; 1489 } else { 1490 mMasterVolume = value; 1491 } 1492} 1493 1494void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1495{ 1496 Mutex::Autolock _l(mLock); 1497 // Don't apply master mute in SW if our HAL can do it for us. 1498 if (mOutput && mOutput->audioHwDev && 1499 mOutput->audioHwDev->canSetMasterMute()) { 1500 mMasterMute = false; 1501 } else { 1502 mMasterMute = muted; 1503 } 1504} 1505 1506void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1507{ 1508 Mutex::Autolock _l(mLock); 1509 mStreamTypes[stream].volume = value; 1510 broadcast_l(); 1511} 1512 1513void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1514{ 1515 Mutex::Autolock _l(mLock); 1516 mStreamTypes[stream].mute = muted; 1517 broadcast_l(); 1518} 1519 1520float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1521{ 1522 Mutex::Autolock _l(mLock); 1523 return mStreamTypes[stream].volume; 1524} 1525 1526// addTrack_l() must be called with ThreadBase::mLock held 1527status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1528{ 1529 status_t status = ALREADY_EXISTS; 1530 1531 // set retry count for buffer fill 1532 track->mRetryCount = kMaxTrackStartupRetries; 1533 if (mActiveTracks.indexOf(track) < 0) { 1534 // the track is newly added, make sure it fills up all its 1535 // buffers before playing. This is to ensure the client will 1536 // effectively get the latency it requested. 1537 if (!track->isOutputTrack()) { 1538 TrackBase::track_state state = track->mState; 1539 mLock.unlock(); 1540 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1541 mLock.lock(); 1542 // abort track was stopped/paused while we released the lock 1543 if (state != track->mState) { 1544 if (status == NO_ERROR) { 1545 mLock.unlock(); 1546 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1547 mLock.lock(); 1548 } 1549 return INVALID_OPERATION; 1550 } 1551 // abort if start is rejected by audio policy manager 1552 if (status != NO_ERROR) { 1553 return PERMISSION_DENIED; 1554 } 1555#ifdef ADD_BATTERY_DATA 1556 // to track the speaker usage 1557 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1558#endif 1559 } 1560 1561 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1562 track->mResetDone = false; 1563 track->mPresentationCompleteFrames = 0; 1564 mActiveTracks.add(track); 1565 mWakeLockUids.add(track->uid()); 1566 mActiveTracksGeneration++; 1567 mLatestActiveTrack = track; 1568 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1569 if (chain != 0) { 1570 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1571 track->sessionId()); 1572 chain->incActiveTrackCnt(); 1573 } 1574 1575 status = NO_ERROR; 1576 } 1577 1578 onAddNewTrack_l(); 1579 return status; 1580} 1581 1582bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1583{ 1584 track->terminate(); 1585 // active tracks are removed by threadLoop() 1586 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1587 track->mState = TrackBase::STOPPED; 1588 if (!trackActive) { 1589 removeTrack_l(track); 1590 } else if (track->isFastTrack() || track->isOffloaded()) { 1591 track->mState = TrackBase::STOPPING_1; 1592 } 1593 1594 return trackActive; 1595} 1596 1597void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1598{ 1599 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1600 mTracks.remove(track); 1601 deleteTrackName_l(track->name()); 1602 // redundant as track is about to be destroyed, for dumpsys only 1603 track->mName = -1; 1604 if (track->isFastTrack()) { 1605 int index = track->mFastIndex; 1606 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1607 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1608 mFastTrackAvailMask |= 1 << index; 1609 // redundant as track is about to be destroyed, for dumpsys only 1610 track->mFastIndex = -1; 1611 } 1612 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1613 if (chain != 0) { 1614 chain->decTrackCnt(); 1615 } 1616} 1617 1618void AudioFlinger::PlaybackThread::broadcast_l() 1619{ 1620 // Thread could be blocked waiting for async 1621 // so signal it to handle state changes immediately 1622 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1623 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1624 mSignalPending = true; 1625 mWaitWorkCV.broadcast(); 1626} 1627 1628String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1629{ 1630 Mutex::Autolock _l(mLock); 1631 if (initCheck() != NO_ERROR) { 1632 return String8(); 1633 } 1634 1635 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1636 const String8 out_s8(s); 1637 free(s); 1638 return out_s8; 1639} 1640 1641void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1642 AudioSystem::OutputDescriptor desc; 1643 void *param2 = NULL; 1644 1645 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1646 param); 1647 1648 switch (event) { 1649 case AudioSystem::OUTPUT_OPENED: 1650 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1651 desc.channelMask = mChannelMask; 1652 desc.samplingRate = mSampleRate; 1653 desc.format = mFormat; 1654 desc.frameCount = mNormalFrameCount; // FIXME see 1655 // AudioFlinger::frameCount(audio_io_handle_t) 1656 desc.latency = latency_l(); 1657 param2 = &desc; 1658 break; 1659 1660 case AudioSystem::STREAM_CONFIG_CHANGED: 1661 param2 = ¶m; 1662 case AudioSystem::OUTPUT_CLOSED: 1663 default: 1664 break; 1665 } 1666 mAudioFlinger->audioConfigChanged(event, mId, param2); 1667} 1668 1669void AudioFlinger::PlaybackThread::writeCallback() 1670{ 1671 ALOG_ASSERT(mCallbackThread != 0); 1672 mCallbackThread->resetWriteBlocked(); 1673} 1674 1675void AudioFlinger::PlaybackThread::drainCallback() 1676{ 1677 ALOG_ASSERT(mCallbackThread != 0); 1678 mCallbackThread->resetDraining(); 1679} 1680 1681void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1682{ 1683 Mutex::Autolock _l(mLock); 1684 // reject out of sequence requests 1685 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1686 mWriteAckSequence &= ~1; 1687 mWaitWorkCV.signal(); 1688 } 1689} 1690 1691void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 // reject out of sequence requests 1695 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1696 mDrainSequence &= ~1; 1697 mWaitWorkCV.signal(); 1698 } 1699} 1700 1701// static 1702int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1703 void *param __unused, 1704 void *cookie) 1705{ 1706 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1707 ALOGV("asyncCallback() event %d", event); 1708 switch (event) { 1709 case STREAM_CBK_EVENT_WRITE_READY: 1710 me->writeCallback(); 1711 break; 1712 case STREAM_CBK_EVENT_DRAIN_READY: 1713 me->drainCallback(); 1714 break; 1715 default: 1716 ALOGW("asyncCallback() unknown event %d", event); 1717 break; 1718 } 1719 return 0; 1720} 1721 1722void AudioFlinger::PlaybackThread::readOutputParameters_l() 1723{ 1724 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1725 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1726 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1727 if (!audio_is_output_channel(mChannelMask)) { 1728 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1729 } 1730 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1731 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1732 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1733 } 1734 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1735 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1736 if (!audio_is_valid_format(mFormat)) { 1737 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1738 } 1739 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1740 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1741 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1742 } 1743 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1744 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1745 mFrameCount = mBufferSize / mFrameSize; 1746 if (mFrameCount & 15) { 1747 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1748 mFrameCount); 1749 } 1750 1751 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1752 (mOutput->stream->set_callback != NULL)) { 1753 if (mOutput->stream->set_callback(mOutput->stream, 1754 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1755 mUseAsyncWrite = true; 1756 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1757 } 1758 } 1759 1760 // Calculate size of normal sink buffer relative to the HAL output buffer size 1761 double multiplier = 1.0; 1762 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1763 kUseFastMixer == FastMixer_Dynamic)) { 1764 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1765 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1766 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1767 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1768 maxNormalFrameCount = maxNormalFrameCount & ~15; 1769 if (maxNormalFrameCount < minNormalFrameCount) { 1770 maxNormalFrameCount = minNormalFrameCount; 1771 } 1772 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1773 if (multiplier <= 1.0) { 1774 multiplier = 1.0; 1775 } else if (multiplier <= 2.0) { 1776 if (2 * mFrameCount <= maxNormalFrameCount) { 1777 multiplier = 2.0; 1778 } else { 1779 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1780 } 1781 } else { 1782 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1783 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1784 // track, but we sometimes have to do this to satisfy the maximum frame count 1785 // constraint) 1786 // FIXME this rounding up should not be done if no HAL SRC 1787 uint32_t truncMult = (uint32_t) multiplier; 1788 if ((truncMult & 1)) { 1789 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1790 ++truncMult; 1791 } 1792 } 1793 multiplier = (double) truncMult; 1794 } 1795 } 1796 mNormalFrameCount = multiplier * mFrameCount; 1797 // round up to nearest 16 frames to satisfy AudioMixer 1798 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1799 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1800 mNormalFrameCount); 1801 1802 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1803 // Originally this was int16_t[] array, need to remove legacy implications. 1804 free(mSinkBuffer); 1805 mSinkBuffer = NULL; 1806 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1807 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1808 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1809 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1810 1811 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1812 // drives the output. 1813 free(mMixerBuffer); 1814 mMixerBuffer = NULL; 1815 if (mMixerBufferEnabled) { 1816 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1817 mMixerBufferSize = mNormalFrameCount * mChannelCount 1818 * audio_bytes_per_sample(mMixerBufferFormat); 1819 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1820 } 1821 free(mEffectBuffer); 1822 mEffectBuffer = NULL; 1823 if (mEffectBufferEnabled) { 1824 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1825 mEffectBufferSize = mNormalFrameCount * mChannelCount 1826 * audio_bytes_per_sample(mEffectBufferFormat); 1827 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1828 } 1829 1830 // force reconfiguration of effect chains and engines to take new buffer size and audio 1831 // parameters into account 1832 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1833 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1834 // matter. 1835 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1836 Vector< sp<EffectChain> > effectChains = mEffectChains; 1837 for (size_t i = 0; i < effectChains.size(); i ++) { 1838 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1839 } 1840} 1841 1842 1843status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1844{ 1845 if (halFrames == NULL || dspFrames == NULL) { 1846 return BAD_VALUE; 1847 } 1848 Mutex::Autolock _l(mLock); 1849 if (initCheck() != NO_ERROR) { 1850 return INVALID_OPERATION; 1851 } 1852 size_t framesWritten = mBytesWritten / mFrameSize; 1853 *halFrames = framesWritten; 1854 1855 if (isSuspended()) { 1856 // return an estimation of rendered frames when the output is suspended 1857 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1858 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1859 return NO_ERROR; 1860 } else { 1861 status_t status; 1862 uint32_t frames; 1863 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1864 *dspFrames = (size_t)frames; 1865 return status; 1866 } 1867} 1868 1869uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1870{ 1871 Mutex::Autolock _l(mLock); 1872 uint32_t result = 0; 1873 if (getEffectChain_l(sessionId) != 0) { 1874 result = EFFECT_SESSION; 1875 } 1876 1877 for (size_t i = 0; i < mTracks.size(); ++i) { 1878 sp<Track> track = mTracks[i]; 1879 if (sessionId == track->sessionId() && !track->isInvalid()) { 1880 result |= TRACK_SESSION; 1881 break; 1882 } 1883 } 1884 1885 return result; 1886} 1887 1888uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1889{ 1890 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1891 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1892 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1893 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1894 } 1895 for (size_t i = 0; i < mTracks.size(); i++) { 1896 sp<Track> track = mTracks[i]; 1897 if (sessionId == track->sessionId() && !track->isInvalid()) { 1898 return AudioSystem::getStrategyForStream(track->streamType()); 1899 } 1900 } 1901 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1902} 1903 1904 1905AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1906{ 1907 Mutex::Autolock _l(mLock); 1908 return mOutput; 1909} 1910 1911AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1912{ 1913 Mutex::Autolock _l(mLock); 1914 AudioStreamOut *output = mOutput; 1915 mOutput = NULL; 1916 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1917 // must push a NULL and wait for ack 1918 mOutputSink.clear(); 1919 mPipeSink.clear(); 1920 mNormalSink.clear(); 1921 return output; 1922} 1923 1924// this method must always be called either with ThreadBase mLock held or inside the thread loop 1925audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1926{ 1927 if (mOutput == NULL) { 1928 return NULL; 1929 } 1930 return &mOutput->stream->common; 1931} 1932 1933uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1934{ 1935 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1936} 1937 1938status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1939{ 1940 if (!isValidSyncEvent(event)) { 1941 return BAD_VALUE; 1942 } 1943 1944 Mutex::Autolock _l(mLock); 1945 1946 for (size_t i = 0; i < mTracks.size(); ++i) { 1947 sp<Track> track = mTracks[i]; 1948 if (event->triggerSession() == track->sessionId()) { 1949 (void) track->setSyncEvent(event); 1950 return NO_ERROR; 1951 } 1952 } 1953 1954 return NAME_NOT_FOUND; 1955} 1956 1957bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1958{ 1959 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1960} 1961 1962void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1963 const Vector< sp<Track> >& tracksToRemove) 1964{ 1965 size_t count = tracksToRemove.size(); 1966 if (count > 0) { 1967 for (size_t i = 0 ; i < count ; i++) { 1968 const sp<Track>& track = tracksToRemove.itemAt(i); 1969 if (!track->isOutputTrack()) { 1970 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1971#ifdef ADD_BATTERY_DATA 1972 // to track the speaker usage 1973 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1974#endif 1975 if (track->isTerminated()) { 1976 AudioSystem::releaseOutput(mId); 1977 } 1978 } 1979 } 1980 } 1981} 1982 1983void AudioFlinger::PlaybackThread::checkSilentMode_l() 1984{ 1985 if (!mMasterMute) { 1986 char value[PROPERTY_VALUE_MAX]; 1987 if (property_get("ro.audio.silent", value, "0") > 0) { 1988 char *endptr; 1989 unsigned long ul = strtoul(value, &endptr, 0); 1990 if (*endptr == '\0' && ul != 0) { 1991 ALOGD("Silence is golden"); 1992 // The setprop command will not allow a property to be changed after 1993 // the first time it is set, so we don't have to worry about un-muting. 1994 setMasterMute_l(true); 1995 } 1996 } 1997 } 1998} 1999 2000// shared by MIXER and DIRECT, overridden by DUPLICATING 2001ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2002{ 2003 // FIXME rewrite to reduce number of system calls 2004 mLastWriteTime = systemTime(); 2005 mInWrite = true; 2006 ssize_t bytesWritten; 2007 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2008 2009 // If an NBAIO sink is present, use it to write the normal mixer's submix 2010 if (mNormalSink != 0) { 2011 const size_t count = mBytesRemaining / mFrameSize; 2012 2013 ATRACE_BEGIN("write"); 2014 // update the setpoint when AudioFlinger::mScreenState changes 2015 uint32_t screenState = AudioFlinger::mScreenState; 2016 if (screenState != mScreenState) { 2017 mScreenState = screenState; 2018 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2019 if (pipe != NULL) { 2020 pipe->setAvgFrames((mScreenState & 1) ? 2021 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2022 } 2023 } 2024 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2025 ATRACE_END(); 2026 if (framesWritten > 0) { 2027 bytesWritten = framesWritten * mFrameSize; 2028 } else { 2029 bytesWritten = framesWritten; 2030 } 2031 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2032 if (status == NO_ERROR) { 2033 size_t totalFramesWritten = mNormalSink->framesWritten(); 2034 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2035 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2036 mLatchDValid = true; 2037 } 2038 } 2039 // otherwise use the HAL / AudioStreamOut directly 2040 } else { 2041 // Direct output and offload threads 2042 2043 if (mUseAsyncWrite) { 2044 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2045 mWriteAckSequence += 2; 2046 mWriteAckSequence |= 1; 2047 ALOG_ASSERT(mCallbackThread != 0); 2048 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2049 } 2050 // FIXME We should have an implementation of timestamps for direct output threads. 2051 // They are used e.g for multichannel PCM playback over HDMI. 2052 bytesWritten = mOutput->stream->write(mOutput->stream, 2053 (char *)mSinkBuffer + offset, mBytesRemaining); 2054 if (mUseAsyncWrite && 2055 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2056 // do not wait for async callback in case of error of full write 2057 mWriteAckSequence &= ~1; 2058 ALOG_ASSERT(mCallbackThread != 0); 2059 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2060 } 2061 } 2062 2063 mNumWrites++; 2064 mInWrite = false; 2065 mStandby = false; 2066 return bytesWritten; 2067} 2068 2069void AudioFlinger::PlaybackThread::threadLoop_drain() 2070{ 2071 if (mOutput->stream->drain) { 2072 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2073 if (mUseAsyncWrite) { 2074 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2075 mDrainSequence |= 1; 2076 ALOG_ASSERT(mCallbackThread != 0); 2077 mCallbackThread->setDraining(mDrainSequence); 2078 } 2079 mOutput->stream->drain(mOutput->stream, 2080 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2081 : AUDIO_DRAIN_ALL); 2082 } 2083} 2084 2085void AudioFlinger::PlaybackThread::threadLoop_exit() 2086{ 2087 // Default implementation has nothing to do 2088} 2089 2090/* 2091The derived values that are cached: 2092 - mSinkBufferSize from frame count * frame size 2093 - activeSleepTime from activeSleepTimeUs() 2094 - idleSleepTime from idleSleepTimeUs() 2095 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2096 - maxPeriod from frame count and sample rate (MIXER only) 2097 2098The parameters that affect these derived values are: 2099 - frame count 2100 - frame size 2101 - sample rate 2102 - device type: A2DP or not 2103 - device latency 2104 - format: PCM or not 2105 - active sleep time 2106 - idle sleep time 2107*/ 2108 2109void AudioFlinger::PlaybackThread::cacheParameters_l() 2110{ 2111 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2112 activeSleepTime = activeSleepTimeUs(); 2113 idleSleepTime = idleSleepTimeUs(); 2114} 2115 2116void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2117{ 2118 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2119 this, streamType, mTracks.size()); 2120 Mutex::Autolock _l(mLock); 2121 2122 size_t size = mTracks.size(); 2123 for (size_t i = 0; i < size; i++) { 2124 sp<Track> t = mTracks[i]; 2125 if (t->streamType() == streamType) { 2126 t->invalidate(); 2127 } 2128 } 2129} 2130 2131status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2132{ 2133 int session = chain->sessionId(); 2134 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2135 ? mEffectBuffer : mSinkBuffer); 2136 bool ownsBuffer = false; 2137 2138 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2139 if (session > 0) { 2140 // Only one effect chain can be present in direct output thread and it uses 2141 // the sink buffer as input 2142 if (mType != DIRECT) { 2143 size_t numSamples = mNormalFrameCount * mChannelCount; 2144 buffer = new int16_t[numSamples]; 2145 memset(buffer, 0, numSamples * sizeof(int16_t)); 2146 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2147 ownsBuffer = true; 2148 } 2149 2150 // Attach all tracks with same session ID to this chain. 2151 for (size_t i = 0; i < mTracks.size(); ++i) { 2152 sp<Track> track = mTracks[i]; 2153 if (session == track->sessionId()) { 2154 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2155 buffer); 2156 track->setMainBuffer(buffer); 2157 chain->incTrackCnt(); 2158 } 2159 } 2160 2161 // indicate all active tracks in the chain 2162 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2163 sp<Track> track = mActiveTracks[i].promote(); 2164 if (track == 0) { 2165 continue; 2166 } 2167 if (session == track->sessionId()) { 2168 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2169 chain->incActiveTrackCnt(); 2170 } 2171 } 2172 } 2173 2174 chain->setInBuffer(buffer, ownsBuffer); 2175 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2176 ? mEffectBuffer : mSinkBuffer)); 2177 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2178 // chains list in order to be processed last as it contains output stage effects 2179 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2180 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2181 // after track specific effects and before output stage 2182 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2183 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2184 // Effect chain for other sessions are inserted at beginning of effect 2185 // chains list to be processed before output mix effects. Relative order between other 2186 // sessions is not important 2187 size_t size = mEffectChains.size(); 2188 size_t i = 0; 2189 for (i = 0; i < size; i++) { 2190 if (mEffectChains[i]->sessionId() < session) { 2191 break; 2192 } 2193 } 2194 mEffectChains.insertAt(chain, i); 2195 checkSuspendOnAddEffectChain_l(chain); 2196 2197 return NO_ERROR; 2198} 2199 2200size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2201{ 2202 int session = chain->sessionId(); 2203 2204 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2205 2206 for (size_t i = 0; i < mEffectChains.size(); i++) { 2207 if (chain == mEffectChains[i]) { 2208 mEffectChains.removeAt(i); 2209 // detach all active tracks from the chain 2210 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2211 sp<Track> track = mActiveTracks[i].promote(); 2212 if (track == 0) { 2213 continue; 2214 } 2215 if (session == track->sessionId()) { 2216 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2217 chain.get(), session); 2218 chain->decActiveTrackCnt(); 2219 } 2220 } 2221 2222 // detach all tracks with same session ID from this chain 2223 for (size_t i = 0; i < mTracks.size(); ++i) { 2224 sp<Track> track = mTracks[i]; 2225 if (session == track->sessionId()) { 2226 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2227 chain->decTrackCnt(); 2228 } 2229 } 2230 break; 2231 } 2232 } 2233 return mEffectChains.size(); 2234} 2235 2236status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2237 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2238{ 2239 Mutex::Autolock _l(mLock); 2240 return attachAuxEffect_l(track, EffectId); 2241} 2242 2243status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2244 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2245{ 2246 status_t status = NO_ERROR; 2247 2248 if (EffectId == 0) { 2249 track->setAuxBuffer(0, NULL); 2250 } else { 2251 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2252 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2253 if (effect != 0) { 2254 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2255 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2256 } else { 2257 status = INVALID_OPERATION; 2258 } 2259 } else { 2260 status = BAD_VALUE; 2261 } 2262 } 2263 return status; 2264} 2265 2266void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2267{ 2268 for (size_t i = 0; i < mTracks.size(); ++i) { 2269 sp<Track> track = mTracks[i]; 2270 if (track->auxEffectId() == effectId) { 2271 attachAuxEffect_l(track, 0); 2272 } 2273 } 2274} 2275 2276bool AudioFlinger::PlaybackThread::threadLoop() 2277{ 2278 Vector< sp<Track> > tracksToRemove; 2279 2280 standbyTime = systemTime(); 2281 2282 // MIXER 2283 nsecs_t lastWarning = 0; 2284 2285 // DUPLICATING 2286 // FIXME could this be made local to while loop? 2287 writeFrames = 0; 2288 2289 int lastGeneration = 0; 2290 2291 cacheParameters_l(); 2292 sleepTime = idleSleepTime; 2293 2294 if (mType == MIXER) { 2295 sleepTimeShift = 0; 2296 } 2297 2298 CpuStats cpuStats; 2299 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2300 2301 acquireWakeLock(); 2302 2303 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2304 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2305 // and then that string will be logged at the next convenient opportunity. 2306 const char *logString = NULL; 2307 2308 checkSilentMode_l(); 2309 2310 while (!exitPending()) 2311 { 2312 cpuStats.sample(myName); 2313 2314 Vector< sp<EffectChain> > effectChains; 2315 2316 { // scope for mLock 2317 2318 Mutex::Autolock _l(mLock); 2319 2320 processConfigEvents_l(); 2321 2322 if (logString != NULL) { 2323 mNBLogWriter->logTimestamp(); 2324 mNBLogWriter->log(logString); 2325 logString = NULL; 2326 } 2327 2328 if (mLatchDValid) { 2329 mLatchQ = mLatchD; 2330 mLatchDValid = false; 2331 mLatchQValid = true; 2332 } 2333 2334 saveOutputTracks(); 2335 if (mSignalPending) { 2336 // A signal was raised while we were unlocked 2337 mSignalPending = false; 2338 } else if (waitingAsyncCallback_l()) { 2339 if (exitPending()) { 2340 break; 2341 } 2342 releaseWakeLock_l(); 2343 mWakeLockUids.clear(); 2344 mActiveTracksGeneration++; 2345 ALOGV("wait async completion"); 2346 mWaitWorkCV.wait(mLock); 2347 ALOGV("async completion/wake"); 2348 acquireWakeLock_l(); 2349 standbyTime = systemTime() + standbyDelay; 2350 sleepTime = 0; 2351 2352 continue; 2353 } 2354 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2355 isSuspended()) { 2356 // put audio hardware into standby after short delay 2357 if (shouldStandby_l()) { 2358 2359 threadLoop_standby(); 2360 2361 mStandby = true; 2362 } 2363 2364 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2365 // we're about to wait, flush the binder command buffer 2366 IPCThreadState::self()->flushCommands(); 2367 2368 clearOutputTracks(); 2369 2370 if (exitPending()) { 2371 break; 2372 } 2373 2374 releaseWakeLock_l(); 2375 mWakeLockUids.clear(); 2376 mActiveTracksGeneration++; 2377 // wait until we have something to do... 2378 ALOGV("%s going to sleep", myName.string()); 2379 mWaitWorkCV.wait(mLock); 2380 ALOGV("%s waking up", myName.string()); 2381 acquireWakeLock_l(); 2382 2383 mMixerStatus = MIXER_IDLE; 2384 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2385 mBytesWritten = 0; 2386 mBytesRemaining = 0; 2387 checkSilentMode_l(); 2388 2389 standbyTime = systemTime() + standbyDelay; 2390 sleepTime = idleSleepTime; 2391 if (mType == MIXER) { 2392 sleepTimeShift = 0; 2393 } 2394 2395 continue; 2396 } 2397 } 2398 // mMixerStatusIgnoringFastTracks is also updated internally 2399 mMixerStatus = prepareTracks_l(&tracksToRemove); 2400 2401 // compare with previously applied list 2402 if (lastGeneration != mActiveTracksGeneration) { 2403 // update wakelock 2404 updateWakeLockUids_l(mWakeLockUids); 2405 lastGeneration = mActiveTracksGeneration; 2406 } 2407 2408 // prevent any changes in effect chain list and in each effect chain 2409 // during mixing and effect process as the audio buffers could be deleted 2410 // or modified if an effect is created or deleted 2411 lockEffectChains_l(effectChains); 2412 } // mLock scope ends 2413 2414 if (mBytesRemaining == 0) { 2415 mCurrentWriteLength = 0; 2416 if (mMixerStatus == MIXER_TRACKS_READY) { 2417 // threadLoop_mix() sets mCurrentWriteLength 2418 threadLoop_mix(); 2419 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2420 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2421 // threadLoop_sleepTime sets sleepTime to 0 if data 2422 // must be written to HAL 2423 threadLoop_sleepTime(); 2424 if (sleepTime == 0) { 2425 mCurrentWriteLength = mSinkBufferSize; 2426 } 2427 } 2428 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2429 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2430 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2431 // or mSinkBuffer (if there are no effects). 2432 // 2433 // This is done pre-effects computation; if effects change to 2434 // support higher precision, this needs to move. 2435 // 2436 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2437 // TODO use sleepTime == 0 as an additional condition. 2438 if (mMixerBufferValid) { 2439 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2440 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2441 2442 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2443 mNormalFrameCount * mChannelCount); 2444 } 2445 2446 mBytesRemaining = mCurrentWriteLength; 2447 if (isSuspended()) { 2448 sleepTime = suspendSleepTimeUs(); 2449 // simulate write to HAL when suspended 2450 mBytesWritten += mSinkBufferSize; 2451 mBytesRemaining = 0; 2452 } 2453 2454 // only process effects if we're going to write 2455 if (sleepTime == 0 && mType != OFFLOAD) { 2456 for (size_t i = 0; i < effectChains.size(); i ++) { 2457 effectChains[i]->process_l(); 2458 } 2459 } 2460 } 2461 // Process effect chains for offloaded thread even if no audio 2462 // was read from audio track: process only updates effect state 2463 // and thus does have to be synchronized with audio writes but may have 2464 // to be called while waiting for async write callback 2465 if (mType == OFFLOAD) { 2466 for (size_t i = 0; i < effectChains.size(); i ++) { 2467 effectChains[i]->process_l(); 2468 } 2469 } 2470 2471 // Only if the Effects buffer is enabled and there is data in the 2472 // Effects buffer (buffer valid), we need to 2473 // copy into the sink buffer. 2474 // TODO use sleepTime == 0 as an additional condition. 2475 if (mEffectBufferValid) { 2476 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2477 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2478 mNormalFrameCount * mChannelCount); 2479 } 2480 2481 // enable changes in effect chain 2482 unlockEffectChains(effectChains); 2483 2484 if (!waitingAsyncCallback()) { 2485 // sleepTime == 0 means we must write to audio hardware 2486 if (sleepTime == 0) { 2487 if (mBytesRemaining) { 2488 ssize_t ret = threadLoop_write(); 2489 if (ret < 0) { 2490 mBytesRemaining = 0; 2491 } else { 2492 mBytesWritten += ret; 2493 mBytesRemaining -= ret; 2494 } 2495 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2496 (mMixerStatus == MIXER_DRAIN_ALL)) { 2497 threadLoop_drain(); 2498 } 2499 if (mType == MIXER) { 2500 // write blocked detection 2501 nsecs_t now = systemTime(); 2502 nsecs_t delta = now - mLastWriteTime; 2503 if (!mStandby && delta > maxPeriod) { 2504 mNumDelayedWrites++; 2505 if ((now - lastWarning) > kWarningThrottleNs) { 2506 ATRACE_NAME("underrun"); 2507 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2508 ns2ms(delta), mNumDelayedWrites, this); 2509 lastWarning = now; 2510 } 2511 } 2512 } 2513 2514 } else { 2515 usleep(sleepTime); 2516 } 2517 } 2518 2519 // Finally let go of removed track(s), without the lock held 2520 // since we can't guarantee the destructors won't acquire that 2521 // same lock. This will also mutate and push a new fast mixer state. 2522 threadLoop_removeTracks(tracksToRemove); 2523 tracksToRemove.clear(); 2524 2525 // FIXME I don't understand the need for this here; 2526 // it was in the original code but maybe the 2527 // assignment in saveOutputTracks() makes this unnecessary? 2528 clearOutputTracks(); 2529 2530 // Effect chains will be actually deleted here if they were removed from 2531 // mEffectChains list during mixing or effects processing 2532 effectChains.clear(); 2533 2534 // FIXME Note that the above .clear() is no longer necessary since effectChains 2535 // is now local to this block, but will keep it for now (at least until merge done). 2536 } 2537 2538 threadLoop_exit(); 2539 2540 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2541 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2542 // put output stream into standby mode 2543 if (!mStandby) { 2544 mOutput->stream->common.standby(&mOutput->stream->common); 2545 } 2546 } 2547 2548 releaseWakeLock(); 2549 mWakeLockUids.clear(); 2550 mActiveTracksGeneration++; 2551 2552 ALOGV("Thread %p type %d exiting", this, mType); 2553 return false; 2554} 2555 2556// removeTracks_l() must be called with ThreadBase::mLock held 2557void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2558{ 2559 size_t count = tracksToRemove.size(); 2560 if (count > 0) { 2561 for (size_t i=0 ; i<count ; i++) { 2562 const sp<Track>& track = tracksToRemove.itemAt(i); 2563 mActiveTracks.remove(track); 2564 mWakeLockUids.remove(track->uid()); 2565 mActiveTracksGeneration++; 2566 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2567 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2568 if (chain != 0) { 2569 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2570 track->sessionId()); 2571 chain->decActiveTrackCnt(); 2572 } 2573 if (track->isTerminated()) { 2574 removeTrack_l(track); 2575 } 2576 } 2577 } 2578 2579} 2580 2581status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2582{ 2583 if (mNormalSink != 0) { 2584 return mNormalSink->getTimestamp(timestamp); 2585 } 2586 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2587 uint64_t position64; 2588 int ret = mOutput->stream->get_presentation_position( 2589 mOutput->stream, &position64, ×tamp.mTime); 2590 if (ret == 0) { 2591 timestamp.mPosition = (uint32_t)position64; 2592 return NO_ERROR; 2593 } 2594 } 2595 return INVALID_OPERATION; 2596} 2597// ---------------------------------------------------------------------------- 2598 2599AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2600 audio_io_handle_t id, audio_devices_t device, type_t type) 2601 : PlaybackThread(audioFlinger, output, id, device, type), 2602 // mAudioMixer below 2603 // mFastMixer below 2604 mFastMixerFutex(0) 2605 // mOutputSink below 2606 // mPipeSink below 2607 // mNormalSink below 2608{ 2609 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2610 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2611 "mFrameCount=%d, mNormalFrameCount=%d", 2612 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2613 mNormalFrameCount); 2614 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2615 2616 // FIXME - Current mixer implementation only supports stereo output 2617 if (mChannelCount != FCC_2) { 2618 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2619 } 2620 2621 // create an NBAIO sink for the HAL output stream, and negotiate 2622 mOutputSink = new AudioStreamOutSink(output->stream); 2623 size_t numCounterOffers = 0; 2624 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2625 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2626 ALOG_ASSERT(index == 0); 2627 2628 // initialize fast mixer depending on configuration 2629 bool initFastMixer; 2630 switch (kUseFastMixer) { 2631 case FastMixer_Never: 2632 initFastMixer = false; 2633 break; 2634 case FastMixer_Always: 2635 initFastMixer = true; 2636 break; 2637 case FastMixer_Static: 2638 case FastMixer_Dynamic: 2639 initFastMixer = mFrameCount < mNormalFrameCount; 2640 break; 2641 } 2642 if (initFastMixer) { 2643 2644 // create a MonoPipe to connect our submix to FastMixer 2645 NBAIO_Format format = mOutputSink->format(); 2646 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2647 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2648 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2649 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2650 const NBAIO_Format offers[1] = {format}; 2651 size_t numCounterOffers = 0; 2652 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2653 ALOG_ASSERT(index == 0); 2654 monoPipe->setAvgFrames((mScreenState & 1) ? 2655 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2656 mPipeSink = monoPipe; 2657 2658#ifdef TEE_SINK 2659 if (mTeeSinkOutputEnabled) { 2660 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2661 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2662 numCounterOffers = 0; 2663 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2664 ALOG_ASSERT(index == 0); 2665 mTeeSink = teeSink; 2666 PipeReader *teeSource = new PipeReader(*teeSink); 2667 numCounterOffers = 0; 2668 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2669 ALOG_ASSERT(index == 0); 2670 mTeeSource = teeSource; 2671 } 2672#endif 2673 2674 // create fast mixer and configure it initially with just one fast track for our submix 2675 mFastMixer = new FastMixer(); 2676 FastMixerStateQueue *sq = mFastMixer->sq(); 2677#ifdef STATE_QUEUE_DUMP 2678 sq->setObserverDump(&mStateQueueObserverDump); 2679 sq->setMutatorDump(&mStateQueueMutatorDump); 2680#endif 2681 FastMixerState *state = sq->begin(); 2682 FastTrack *fastTrack = &state->mFastTracks[0]; 2683 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2684 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2685 fastTrack->mVolumeProvider = NULL; 2686 fastTrack->mGeneration++; 2687 state->mFastTracksGen++; 2688 state->mTrackMask = 1; 2689 // fast mixer will use the HAL output sink 2690 state->mOutputSink = mOutputSink.get(); 2691 state->mOutputSinkGen++; 2692 state->mFrameCount = mFrameCount; 2693 state->mCommand = FastMixerState::COLD_IDLE; 2694 // already done in constructor initialization list 2695 //mFastMixerFutex = 0; 2696 state->mColdFutexAddr = &mFastMixerFutex; 2697 state->mColdGen++; 2698 state->mDumpState = &mFastMixerDumpState; 2699#ifdef TEE_SINK 2700 state->mTeeSink = mTeeSink.get(); 2701#endif 2702 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2703 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2704 sq->end(); 2705 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2706 2707 // start the fast mixer 2708 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2709 pid_t tid = mFastMixer->getTid(); 2710 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2711 if (err != 0) { 2712 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2713 kPriorityFastMixer, getpid_cached, tid, err); 2714 } 2715 2716#ifdef AUDIO_WATCHDOG 2717 // create and start the watchdog 2718 mAudioWatchdog = new AudioWatchdog(); 2719 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2720 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2721 tid = mAudioWatchdog->getTid(); 2722 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2723 if (err != 0) { 2724 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2725 kPriorityFastMixer, getpid_cached, tid, err); 2726 } 2727#endif 2728 2729 } else { 2730 mFastMixer = NULL; 2731 } 2732 2733 switch (kUseFastMixer) { 2734 case FastMixer_Never: 2735 case FastMixer_Dynamic: 2736 mNormalSink = mOutputSink; 2737 break; 2738 case FastMixer_Always: 2739 mNormalSink = mPipeSink; 2740 break; 2741 case FastMixer_Static: 2742 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2743 break; 2744 } 2745} 2746 2747AudioFlinger::MixerThread::~MixerThread() 2748{ 2749 if (mFastMixer != NULL) { 2750 FastMixerStateQueue *sq = mFastMixer->sq(); 2751 FastMixerState *state = sq->begin(); 2752 if (state->mCommand == FastMixerState::COLD_IDLE) { 2753 int32_t old = android_atomic_inc(&mFastMixerFutex); 2754 if (old == -1) { 2755 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2756 } 2757 } 2758 state->mCommand = FastMixerState::EXIT; 2759 sq->end(); 2760 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2761 mFastMixer->join(); 2762 // Though the fast mixer thread has exited, it's state queue is still valid. 2763 // We'll use that extract the final state which contains one remaining fast track 2764 // corresponding to our sub-mix. 2765 state = sq->begin(); 2766 ALOG_ASSERT(state->mTrackMask == 1); 2767 FastTrack *fastTrack = &state->mFastTracks[0]; 2768 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2769 delete fastTrack->mBufferProvider; 2770 sq->end(false /*didModify*/); 2771 delete mFastMixer; 2772#ifdef AUDIO_WATCHDOG 2773 if (mAudioWatchdog != 0) { 2774 mAudioWatchdog->requestExit(); 2775 mAudioWatchdog->requestExitAndWait(); 2776 mAudioWatchdog.clear(); 2777 } 2778#endif 2779 } 2780 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2781 delete mAudioMixer; 2782} 2783 2784 2785uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2786{ 2787 if (mFastMixer != NULL) { 2788 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2789 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2790 } 2791 return latency; 2792} 2793 2794 2795void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2796{ 2797 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2798} 2799 2800ssize_t AudioFlinger::MixerThread::threadLoop_write() 2801{ 2802 // FIXME we should only do one push per cycle; confirm this is true 2803 // Start the fast mixer if it's not already running 2804 if (mFastMixer != NULL) { 2805 FastMixerStateQueue *sq = mFastMixer->sq(); 2806 FastMixerState *state = sq->begin(); 2807 if (state->mCommand != FastMixerState::MIX_WRITE && 2808 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2809 if (state->mCommand == FastMixerState::COLD_IDLE) { 2810 int32_t old = android_atomic_inc(&mFastMixerFutex); 2811 if (old == -1) { 2812 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2813 } 2814#ifdef AUDIO_WATCHDOG 2815 if (mAudioWatchdog != 0) { 2816 mAudioWatchdog->resume(); 2817 } 2818#endif 2819 } 2820 state->mCommand = FastMixerState::MIX_WRITE; 2821 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2822 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2823 sq->end(); 2824 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2825 if (kUseFastMixer == FastMixer_Dynamic) { 2826 mNormalSink = mPipeSink; 2827 } 2828 } else { 2829 sq->end(false /*didModify*/); 2830 } 2831 } 2832 return PlaybackThread::threadLoop_write(); 2833} 2834 2835void AudioFlinger::MixerThread::threadLoop_standby() 2836{ 2837 // Idle the fast mixer if it's currently running 2838 if (mFastMixer != NULL) { 2839 FastMixerStateQueue *sq = mFastMixer->sq(); 2840 FastMixerState *state = sq->begin(); 2841 if (!(state->mCommand & FastMixerState::IDLE)) { 2842 state->mCommand = FastMixerState::COLD_IDLE; 2843 state->mColdFutexAddr = &mFastMixerFutex; 2844 state->mColdGen++; 2845 mFastMixerFutex = 0; 2846 sq->end(); 2847 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2848 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2849 if (kUseFastMixer == FastMixer_Dynamic) { 2850 mNormalSink = mOutputSink; 2851 } 2852#ifdef AUDIO_WATCHDOG 2853 if (mAudioWatchdog != 0) { 2854 mAudioWatchdog->pause(); 2855 } 2856#endif 2857 } else { 2858 sq->end(false /*didModify*/); 2859 } 2860 } 2861 PlaybackThread::threadLoop_standby(); 2862} 2863 2864bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2865{ 2866 return false; 2867} 2868 2869bool AudioFlinger::PlaybackThread::shouldStandby_l() 2870{ 2871 return !mStandby; 2872} 2873 2874bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2875{ 2876 Mutex::Autolock _l(mLock); 2877 return waitingAsyncCallback_l(); 2878} 2879 2880// shared by MIXER and DIRECT, overridden by DUPLICATING 2881void AudioFlinger::PlaybackThread::threadLoop_standby() 2882{ 2883 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2884 mOutput->stream->common.standby(&mOutput->stream->common); 2885 if (mUseAsyncWrite != 0) { 2886 // discard any pending drain or write ack by incrementing sequence 2887 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2888 mDrainSequence = (mDrainSequence + 2) & ~1; 2889 ALOG_ASSERT(mCallbackThread != 0); 2890 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2891 mCallbackThread->setDraining(mDrainSequence); 2892 } 2893} 2894 2895void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2896{ 2897 ALOGV("signal playback thread"); 2898 broadcast_l(); 2899} 2900 2901void AudioFlinger::MixerThread::threadLoop_mix() 2902{ 2903 // obtain the presentation timestamp of the next output buffer 2904 int64_t pts; 2905 status_t status = INVALID_OPERATION; 2906 2907 if (mNormalSink != 0) { 2908 status = mNormalSink->getNextWriteTimestamp(&pts); 2909 } else { 2910 status = mOutputSink->getNextWriteTimestamp(&pts); 2911 } 2912 2913 if (status != NO_ERROR) { 2914 pts = AudioBufferProvider::kInvalidPTS; 2915 } 2916 2917 // mix buffers... 2918 mAudioMixer->process(pts); 2919 mCurrentWriteLength = mSinkBufferSize; 2920 // increase sleep time progressively when application underrun condition clears. 2921 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2922 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2923 // such that we would underrun the audio HAL. 2924 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2925 sleepTimeShift--; 2926 } 2927 sleepTime = 0; 2928 standbyTime = systemTime() + standbyDelay; 2929 //TODO: delay standby when effects have a tail 2930} 2931 2932void AudioFlinger::MixerThread::threadLoop_sleepTime() 2933{ 2934 // If no tracks are ready, sleep once for the duration of an output 2935 // buffer size, then write 0s to the output 2936 if (sleepTime == 0) { 2937 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2938 sleepTime = activeSleepTime >> sleepTimeShift; 2939 if (sleepTime < kMinThreadSleepTimeUs) { 2940 sleepTime = kMinThreadSleepTimeUs; 2941 } 2942 // reduce sleep time in case of consecutive application underruns to avoid 2943 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2944 // duration we would end up writing less data than needed by the audio HAL if 2945 // the condition persists. 2946 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2947 sleepTimeShift++; 2948 } 2949 } else { 2950 sleepTime = idleSleepTime; 2951 } 2952 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2953 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2954 // before effects processing or output. 2955 if (mMixerBufferValid) { 2956 memset(mMixerBuffer, 0, mMixerBufferSize); 2957 } else { 2958 memset(mSinkBuffer, 0, mSinkBufferSize); 2959 } 2960 sleepTime = 0; 2961 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2962 "anticipated start"); 2963 } 2964 // TODO add standby time extension fct of effect tail 2965} 2966 2967// prepareTracks_l() must be called with ThreadBase::mLock held 2968AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2969 Vector< sp<Track> > *tracksToRemove) 2970{ 2971 2972 mixer_state mixerStatus = MIXER_IDLE; 2973 // find out which tracks need to be processed 2974 size_t count = mActiveTracks.size(); 2975 size_t mixedTracks = 0; 2976 size_t tracksWithEffect = 0; 2977 // counts only _active_ fast tracks 2978 size_t fastTracks = 0; 2979 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2980 2981 float masterVolume = mMasterVolume; 2982 bool masterMute = mMasterMute; 2983 2984 if (masterMute) { 2985 masterVolume = 0; 2986 } 2987 // Delegate master volume control to effect in output mix effect chain if needed 2988 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2989 if (chain != 0) { 2990 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2991 chain->setVolume_l(&v, &v); 2992 masterVolume = (float)((v + (1 << 23)) >> 24); 2993 chain.clear(); 2994 } 2995 2996 // prepare a new state to push 2997 FastMixerStateQueue *sq = NULL; 2998 FastMixerState *state = NULL; 2999 bool didModify = false; 3000 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3001 if (mFastMixer != NULL) { 3002 sq = mFastMixer->sq(); 3003 state = sq->begin(); 3004 } 3005 3006 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3007 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3008 3009 for (size_t i=0 ; i<count ; i++) { 3010 const sp<Track> t = mActiveTracks[i].promote(); 3011 if (t == 0) { 3012 continue; 3013 } 3014 3015 // this const just means the local variable doesn't change 3016 Track* const track = t.get(); 3017 3018 // process fast tracks 3019 if (track->isFastTrack()) { 3020 3021 // It's theoretically possible (though unlikely) for a fast track to be created 3022 // and then removed within the same normal mix cycle. This is not a problem, as 3023 // the track never becomes active so it's fast mixer slot is never touched. 3024 // The converse, of removing an (active) track and then creating a new track 3025 // at the identical fast mixer slot within the same normal mix cycle, 3026 // is impossible because the slot isn't marked available until the end of each cycle. 3027 int j = track->mFastIndex; 3028 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3029 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3030 FastTrack *fastTrack = &state->mFastTracks[j]; 3031 3032 // Determine whether the track is currently in underrun condition, 3033 // and whether it had a recent underrun. 3034 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3035 FastTrackUnderruns underruns = ftDump->mUnderruns; 3036 uint32_t recentFull = (underruns.mBitFields.mFull - 3037 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3038 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3039 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3040 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3041 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3042 uint32_t recentUnderruns = recentPartial + recentEmpty; 3043 track->mObservedUnderruns = underruns; 3044 // don't count underruns that occur while stopping or pausing 3045 // or stopped which can occur when flush() is called while active 3046 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3047 recentUnderruns > 0) { 3048 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3049 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3050 } 3051 3052 // This is similar to the state machine for normal tracks, 3053 // with a few modifications for fast tracks. 3054 bool isActive = true; 3055 switch (track->mState) { 3056 case TrackBase::STOPPING_1: 3057 // track stays active in STOPPING_1 state until first underrun 3058 if (recentUnderruns > 0 || track->isTerminated()) { 3059 track->mState = TrackBase::STOPPING_2; 3060 } 3061 break; 3062 case TrackBase::PAUSING: 3063 // ramp down is not yet implemented 3064 track->setPaused(); 3065 break; 3066 case TrackBase::RESUMING: 3067 // ramp up is not yet implemented 3068 track->mState = TrackBase::ACTIVE; 3069 break; 3070 case TrackBase::ACTIVE: 3071 if (recentFull > 0 || recentPartial > 0) { 3072 // track has provided at least some frames recently: reset retry count 3073 track->mRetryCount = kMaxTrackRetries; 3074 } 3075 if (recentUnderruns == 0) { 3076 // no recent underruns: stay active 3077 break; 3078 } 3079 // there has recently been an underrun of some kind 3080 if (track->sharedBuffer() == 0) { 3081 // were any of the recent underruns "empty" (no frames available)? 3082 if (recentEmpty == 0) { 3083 // no, then ignore the partial underruns as they are allowed indefinitely 3084 break; 3085 } 3086 // there has recently been an "empty" underrun: decrement the retry counter 3087 if (--(track->mRetryCount) > 0) { 3088 break; 3089 } 3090 // indicate to client process that the track was disabled because of underrun; 3091 // it will then automatically call start() when data is available 3092 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3093 // remove from active list, but state remains ACTIVE [confusing but true] 3094 isActive = false; 3095 break; 3096 } 3097 // fall through 3098 case TrackBase::STOPPING_2: 3099 case TrackBase::PAUSED: 3100 case TrackBase::STOPPED: 3101 case TrackBase::FLUSHED: // flush() while active 3102 // Check for presentation complete if track is inactive 3103 // We have consumed all the buffers of this track. 3104 // This would be incomplete if we auto-paused on underrun 3105 { 3106 size_t audioHALFrames = 3107 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3108 size_t framesWritten = mBytesWritten / mFrameSize; 3109 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3110 // track stays in active list until presentation is complete 3111 break; 3112 } 3113 } 3114 if (track->isStopping_2()) { 3115 track->mState = TrackBase::STOPPED; 3116 } 3117 if (track->isStopped()) { 3118 // Can't reset directly, as fast mixer is still polling this track 3119 // track->reset(); 3120 // So instead mark this track as needing to be reset after push with ack 3121 resetMask |= 1 << i; 3122 } 3123 isActive = false; 3124 break; 3125 case TrackBase::IDLE: 3126 default: 3127 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3128 } 3129 3130 if (isActive) { 3131 // was it previously inactive? 3132 if (!(state->mTrackMask & (1 << j))) { 3133 ExtendedAudioBufferProvider *eabp = track; 3134 VolumeProvider *vp = track; 3135 fastTrack->mBufferProvider = eabp; 3136 fastTrack->mVolumeProvider = vp; 3137 fastTrack->mChannelMask = track->mChannelMask; 3138 fastTrack->mGeneration++; 3139 state->mTrackMask |= 1 << j; 3140 didModify = true; 3141 // no acknowledgement required for newly active tracks 3142 } 3143 // cache the combined master volume and stream type volume for fast mixer; this 3144 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3145 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3146 ++fastTracks; 3147 } else { 3148 // was it previously active? 3149 if (state->mTrackMask & (1 << j)) { 3150 fastTrack->mBufferProvider = NULL; 3151 fastTrack->mGeneration++; 3152 state->mTrackMask &= ~(1 << j); 3153 didModify = true; 3154 // If any fast tracks were removed, we must wait for acknowledgement 3155 // because we're about to decrement the last sp<> on those tracks. 3156 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3157 } else { 3158 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3159 } 3160 tracksToRemove->add(track); 3161 // Avoids a misleading display in dumpsys 3162 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3163 } 3164 continue; 3165 } 3166 3167 { // local variable scope to avoid goto warning 3168 3169 audio_track_cblk_t* cblk = track->cblk(); 3170 3171 // The first time a track is added we wait 3172 // for all its buffers to be filled before processing it 3173 int name = track->name(); 3174 // make sure that we have enough frames to mix one full buffer. 3175 // enforce this condition only once to enable draining the buffer in case the client 3176 // app does not call stop() and relies on underrun to stop: 3177 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3178 // during last round 3179 size_t desiredFrames; 3180 uint32_t sr = track->sampleRate(); 3181 if (sr == mSampleRate) { 3182 desiredFrames = mNormalFrameCount; 3183 } else { 3184 // +1 for rounding and +1 for additional sample needed for interpolation 3185 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3186 // add frames already consumed but not yet released by the resampler 3187 // because mAudioTrackServerProxy->framesReady() will include these frames 3188 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3189#if 0 3190 // the minimum track buffer size is normally twice the number of frames necessary 3191 // to fill one buffer and the resampler should not leave more than one buffer worth 3192 // of unreleased frames after each pass, but just in case... 3193 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3194#endif 3195 } 3196 uint32_t minFrames = 1; 3197 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3198 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3199 minFrames = desiredFrames; 3200 } 3201 3202 size_t framesReady = track->framesReady(); 3203 if ((framesReady >= minFrames) && track->isReady() && 3204 !track->isPaused() && !track->isTerminated()) 3205 { 3206 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3207 3208 mixedTracks++; 3209 3210 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3211 // there is an effect chain connected to the track 3212 chain.clear(); 3213 if (track->mainBuffer() != mSinkBuffer && 3214 track->mainBuffer() != mMixerBuffer) { 3215 if (mEffectBufferEnabled) { 3216 mEffectBufferValid = true; // Later can set directly. 3217 } 3218 chain = getEffectChain_l(track->sessionId()); 3219 // Delegate volume control to effect in track effect chain if needed 3220 if (chain != 0) { 3221 tracksWithEffect++; 3222 } else { 3223 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3224 "session %d", 3225 name, track->sessionId()); 3226 } 3227 } 3228 3229 3230 int param = AudioMixer::VOLUME; 3231 if (track->mFillingUpStatus == Track::FS_FILLED) { 3232 // no ramp for the first volume setting 3233 track->mFillingUpStatus = Track::FS_ACTIVE; 3234 if (track->mState == TrackBase::RESUMING) { 3235 track->mState = TrackBase::ACTIVE; 3236 param = AudioMixer::RAMP_VOLUME; 3237 } 3238 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3239 // FIXME should not make a decision based on mServer 3240 } else if (cblk->mServer != 0) { 3241 // If the track is stopped before the first frame was mixed, 3242 // do not apply ramp 3243 param = AudioMixer::RAMP_VOLUME; 3244 } 3245 3246 // compute volume for this track 3247 uint32_t vl, vr, va; 3248 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3249 vl = vr = va = 0; 3250 if (track->isPausing()) { 3251 track->setPaused(); 3252 } 3253 } else { 3254 3255 // read original volumes with volume control 3256 float typeVolume = mStreamTypes[track->streamType()].volume; 3257 float v = masterVolume * typeVolume; 3258 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3259 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3260 float vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3261 float vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3262 // track volumes come from shared memory, so can't be trusted and must be clamped 3263 if (vlf > GAIN_FLOAT_UNITY) { 3264 ALOGV("Track left volume out of range: %.3g", vlf); 3265 vlf = GAIN_FLOAT_UNITY; 3266 } 3267 if (vrf > GAIN_FLOAT_UNITY) { 3268 ALOGV("Track right volume out of range: %.3g", vrf); 3269 vrf = GAIN_FLOAT_UNITY; 3270 } 3271 // now apply the master volume and stream type volume 3272 // FIXME we're losing the wonderful dynamic range in the minifloat representation 3273 float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT); 3274 vl = (uint32_t) (v8_24 * vlf); 3275 vr = (uint32_t) (v8_24 * vrf); 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 // FIXME Use float 3303 // Convert volumes from 8.24 to 4.12 format 3304 // This additional clamping is needed in case chain->setVolume_l() overshot 3305 vl = (vl + (1 << 11)) >> 12; 3306 if (vl > MAX_GAIN_INT) { 3307 vl = MAX_GAIN_INT; 3308 } 3309 vr = (vr + (1 << 11)) >> 12; 3310 if (vr > MAX_GAIN_INT) { 3311 vr = MAX_GAIN_INT; 3312 } 3313 3314 if (va > MAX_GAIN_INT) { 3315 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3316 } 3317 3318 // XXX: these things DON'T need to be done each time 3319 mAudioMixer->setBufferProvider(name, track); 3320 mAudioMixer->enable(name); 3321 3322 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3323 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3324 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3325 mAudioMixer->setParameter( 3326 name, 3327 AudioMixer::TRACK, 3328 AudioMixer::FORMAT, (void *)track->format()); 3329 mAudioMixer->setParameter( 3330 name, 3331 AudioMixer::TRACK, 3332 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3333 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3334 uint32_t maxSampleRate = mSampleRate * 2; 3335 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3336 if (reqSampleRate == 0) { 3337 reqSampleRate = mSampleRate; 3338 } else if (reqSampleRate > maxSampleRate) { 3339 reqSampleRate = maxSampleRate; 3340 } 3341 mAudioMixer->setParameter( 3342 name, 3343 AudioMixer::RESAMPLE, 3344 AudioMixer::SAMPLE_RATE, 3345 (void *)(uintptr_t)reqSampleRate); 3346 /* 3347 * Select the appropriate output buffer for the track. 3348 * 3349 * Tracks with effects go into their own effects chain buffer 3350 * and from there into either mEffectBuffer or mSinkBuffer. 3351 * 3352 * Other tracks can use mMixerBuffer for higher precision 3353 * channel accumulation. If this buffer is enabled 3354 * (mMixerBufferEnabled true), then selected tracks will accumulate 3355 * into it. 3356 * 3357 */ 3358 if (mMixerBufferEnabled 3359 && (track->mainBuffer() == mSinkBuffer 3360 || track->mainBuffer() == mMixerBuffer)) { 3361 mAudioMixer->setParameter( 3362 name, 3363 AudioMixer::TRACK, 3364 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3365 mAudioMixer->setParameter( 3366 name, 3367 AudioMixer::TRACK, 3368 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3369 // TODO: override track->mainBuffer()? 3370 mMixerBufferValid = true; 3371 } else { 3372 mAudioMixer->setParameter( 3373 name, 3374 AudioMixer::TRACK, 3375 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3376 mAudioMixer->setParameter( 3377 name, 3378 AudioMixer::TRACK, 3379 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3380 } 3381 mAudioMixer->setParameter( 3382 name, 3383 AudioMixer::TRACK, 3384 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3385 3386 // reset retry count 3387 track->mRetryCount = kMaxTrackRetries; 3388 3389 // If one track is ready, set the mixer ready if: 3390 // - the mixer was not ready during previous round OR 3391 // - no other track is not ready 3392 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3393 mixerStatus != MIXER_TRACKS_ENABLED) { 3394 mixerStatus = MIXER_TRACKS_READY; 3395 } 3396 } else { 3397 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3398 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3399 } 3400 // clear effect chain input buffer if an active track underruns to avoid sending 3401 // previous audio buffer again to effects 3402 chain = getEffectChain_l(track->sessionId()); 3403 if (chain != 0) { 3404 chain->clearInputBuffer(); 3405 } 3406 3407 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3408 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3409 track->isStopped() || track->isPaused()) { 3410 // We have consumed all the buffers of this track. 3411 // Remove it from the list of active tracks. 3412 // TODO: use actual buffer filling status instead of latency when available from 3413 // audio HAL 3414 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3415 size_t framesWritten = mBytesWritten / mFrameSize; 3416 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3417 if (track->isStopped()) { 3418 track->reset(); 3419 } 3420 tracksToRemove->add(track); 3421 } 3422 } else { 3423 // No buffers for this track. Give it a few chances to 3424 // fill a buffer, then remove it from active list. 3425 if (--(track->mRetryCount) <= 0) { 3426 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3427 tracksToRemove->add(track); 3428 // indicate to client process that the track was disabled because of underrun; 3429 // it will then automatically call start() when data is available 3430 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3431 // If one track is not ready, mark the mixer also not ready if: 3432 // - the mixer was ready during previous round OR 3433 // - no other track is ready 3434 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3435 mixerStatus != MIXER_TRACKS_READY) { 3436 mixerStatus = MIXER_TRACKS_ENABLED; 3437 } 3438 } 3439 mAudioMixer->disable(name); 3440 } 3441 3442 } // local variable scope to avoid goto warning 3443track_is_ready: ; 3444 3445 } 3446 3447 // Push the new FastMixer state if necessary 3448 bool pauseAudioWatchdog = false; 3449 if (didModify) { 3450 state->mFastTracksGen++; 3451 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3452 if (kUseFastMixer == FastMixer_Dynamic && 3453 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3454 state->mCommand = FastMixerState::COLD_IDLE; 3455 state->mColdFutexAddr = &mFastMixerFutex; 3456 state->mColdGen++; 3457 mFastMixerFutex = 0; 3458 if (kUseFastMixer == FastMixer_Dynamic) { 3459 mNormalSink = mOutputSink; 3460 } 3461 // If we go into cold idle, need to wait for acknowledgement 3462 // so that fast mixer stops doing I/O. 3463 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3464 pauseAudioWatchdog = true; 3465 } 3466 } 3467 if (sq != NULL) { 3468 sq->end(didModify); 3469 sq->push(block); 3470 } 3471#ifdef AUDIO_WATCHDOG 3472 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3473 mAudioWatchdog->pause(); 3474 } 3475#endif 3476 3477 // Now perform the deferred reset on fast tracks that have stopped 3478 while (resetMask != 0) { 3479 size_t i = __builtin_ctz(resetMask); 3480 ALOG_ASSERT(i < count); 3481 resetMask &= ~(1 << i); 3482 sp<Track> t = mActiveTracks[i].promote(); 3483 if (t == 0) { 3484 continue; 3485 } 3486 Track* track = t.get(); 3487 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3488 track->reset(); 3489 } 3490 3491 // remove all the tracks that need to be... 3492 removeTracks_l(*tracksToRemove); 3493 3494 // sink or mix buffer must be cleared if all tracks are connected to an 3495 // effect chain as in this case the mixer will not write to the sink or mix buffer 3496 // and track effects will accumulate into it 3497 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3498 (mixedTracks == 0 && fastTracks > 0))) { 3499 // FIXME as a performance optimization, should remember previous zero status 3500 if (mMixerBufferValid) { 3501 memset(mMixerBuffer, 0, mMixerBufferSize); 3502 // TODO: In testing, mSinkBuffer below need not be cleared because 3503 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3504 // after mixing. 3505 // 3506 // To enforce this guarantee: 3507 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3508 // (mixedTracks == 0 && fastTracks > 0)) 3509 // must imply MIXER_TRACKS_READY. 3510 // Later, we may clear buffers regardless, and skip much of this logic. 3511 } 3512 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3513 if (mEffectBufferValid) { 3514 memset(mEffectBuffer, 0, mEffectBufferSize); 3515 } 3516 // FIXME as a performance optimization, should remember previous zero status 3517 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3518 } 3519 3520 // if any fast tracks, then status is ready 3521 mMixerStatusIgnoringFastTracks = mixerStatus; 3522 if (fastTracks > 0) { 3523 mixerStatus = MIXER_TRACKS_READY; 3524 } 3525 return mixerStatus; 3526} 3527 3528// getTrackName_l() must be called with ThreadBase::mLock held 3529int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3530{ 3531 return mAudioMixer->getTrackName(channelMask, 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, mTracks[i]->mSessionId); 3645 if (name < 0) { 3646 break; 3647 } 3648 mTracks[i]->mName = name; 3649 } 3650 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3651 } 3652 } 3653 3654 if (!(previousCommand & FastMixerState::IDLE)) { 3655 ALOG_ASSERT(mFastMixer != NULL); 3656 FastMixerStateQueue *sq = mFastMixer->sq(); 3657 FastMixerState *state = sq->begin(); 3658 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3659 state->mCommand = previousCommand; 3660 sq->end(); 3661 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3662 } 3663 3664 return reconfig; 3665} 3666 3667 3668void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3669{ 3670 const size_t SIZE = 256; 3671 char buffer[SIZE]; 3672 String8 result; 3673 3674 PlaybackThread::dumpInternals(fd, args); 3675 3676 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3677 3678 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3679 const FastMixerDumpState copy(mFastMixerDumpState); 3680 copy.dump(fd); 3681 3682#ifdef STATE_QUEUE_DUMP 3683 // Similar for state queue 3684 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3685 observerCopy.dump(fd); 3686 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3687 mutatorCopy.dump(fd); 3688#endif 3689 3690#ifdef TEE_SINK 3691 // Write the tee output to a .wav file 3692 dumpTee(fd, mTeeSource, mId); 3693#endif 3694 3695#ifdef AUDIO_WATCHDOG 3696 if (mAudioWatchdog != 0) { 3697 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3698 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3699 wdCopy.dump(fd); 3700 } 3701#endif 3702} 3703 3704uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3705{ 3706 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3707} 3708 3709uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3710{ 3711 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3712} 3713 3714void AudioFlinger::MixerThread::cacheParameters_l() 3715{ 3716 PlaybackThread::cacheParameters_l(); 3717 3718 // FIXME: Relaxed timing because of a certain device that can't meet latency 3719 // Should be reduced to 2x after the vendor fixes the driver issue 3720 // increase threshold again due to low power audio mode. The way this warning 3721 // threshold is calculated and its usefulness should be reconsidered anyway. 3722 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3723} 3724 3725// ---------------------------------------------------------------------------- 3726 3727AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3728 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3729 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3730 // mLeftVolFloat, mRightVolFloat 3731{ 3732} 3733 3734AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3735 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3736 ThreadBase::type_t type) 3737 : PlaybackThread(audioFlinger, output, id, device, type) 3738 // mLeftVolFloat, mRightVolFloat 3739{ 3740} 3741 3742AudioFlinger::DirectOutputThread::~DirectOutputThread() 3743{ 3744} 3745 3746void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3747{ 3748 audio_track_cblk_t* cblk = track->cblk(); 3749 float left, right; 3750 3751 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3752 left = right = 0; 3753 } else { 3754 float typeVolume = mStreamTypes[track->streamType()].volume; 3755 float v = mMasterVolume * typeVolume; 3756 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3757 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3758 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3759 if (left > GAIN_FLOAT_UNITY) { 3760 left = GAIN_FLOAT_UNITY; 3761 } 3762 left *= v; 3763 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3764 if (right > GAIN_FLOAT_UNITY) { 3765 right = GAIN_FLOAT_UNITY; 3766 } 3767 right *= v; 3768 } 3769 3770 if (lastTrack) { 3771 if (left != mLeftVolFloat || right != mRightVolFloat) { 3772 mLeftVolFloat = left; 3773 mRightVolFloat = right; 3774 3775 // Convert volumes from float to 8.24 3776 uint32_t vl = (uint32_t)(left * (1 << 24)); 3777 uint32_t vr = (uint32_t)(right * (1 << 24)); 3778 3779 // Delegate volume control to effect in track effect chain if needed 3780 // only one effect chain can be present on DirectOutputThread, so if 3781 // there is one, the track is connected to it 3782 if (!mEffectChains.isEmpty()) { 3783 mEffectChains[0]->setVolume_l(&vl, &vr); 3784 left = (float)vl / (1 << 24); 3785 right = (float)vr / (1 << 24); 3786 } 3787 if (mOutput->stream->set_volume) { 3788 mOutput->stream->set_volume(mOutput->stream, left, right); 3789 } 3790 } 3791 } 3792} 3793 3794 3795AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3796 Vector< sp<Track> > *tracksToRemove 3797) 3798{ 3799 size_t count = mActiveTracks.size(); 3800 mixer_state mixerStatus = MIXER_IDLE; 3801 3802 // find out which tracks need to be processed 3803 for (size_t i = 0; i < count; i++) { 3804 sp<Track> t = mActiveTracks[i].promote(); 3805 // The track died recently 3806 if (t == 0) { 3807 continue; 3808 } 3809 3810 Track* const track = t.get(); 3811 audio_track_cblk_t* cblk = track->cblk(); 3812 // Only consider last track started for volume and mixer state control. 3813 // In theory an older track could underrun and restart after the new one starts 3814 // but as we only care about the transition phase between two tracks on a 3815 // direct output, it is not a problem to ignore the underrun case. 3816 sp<Track> l = mLatestActiveTrack.promote(); 3817 bool last = l.get() == track; 3818 3819 // The first time a track is added we wait 3820 // for all its buffers to be filled before processing it 3821 uint32_t minFrames; 3822 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3823 minFrames = mNormalFrameCount; 3824 } else { 3825 minFrames = 1; 3826 } 3827 3828 if ((track->framesReady() >= minFrames) && track->isReady() && 3829 !track->isPaused() && !track->isTerminated()) 3830 { 3831 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3832 3833 if (track->mFillingUpStatus == Track::FS_FILLED) { 3834 track->mFillingUpStatus = Track::FS_ACTIVE; 3835 // make sure processVolume_l() will apply new volume even if 0 3836 mLeftVolFloat = mRightVolFloat = -1.0; 3837 if (track->mState == TrackBase::RESUMING) { 3838 track->mState = TrackBase::ACTIVE; 3839 } 3840 } 3841 3842 // compute volume for this track 3843 processVolume_l(track, last); 3844 if (last) { 3845 // reset retry count 3846 track->mRetryCount = kMaxTrackRetriesDirect; 3847 mActiveTrack = t; 3848 mixerStatus = MIXER_TRACKS_READY; 3849 } 3850 } else { 3851 // clear effect chain input buffer if the last active track started underruns 3852 // to avoid sending previous audio buffer again to effects 3853 if (!mEffectChains.isEmpty() && last) { 3854 mEffectChains[0]->clearInputBuffer(); 3855 } 3856 3857 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3858 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3859 track->isStopped() || track->isPaused()) { 3860 // We have consumed all the buffers of this track. 3861 // Remove it from the list of active tracks. 3862 // TODO: implement behavior for compressed audio 3863 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3864 size_t framesWritten = mBytesWritten / mFrameSize; 3865 if (mStandby || !last || 3866 track->presentationComplete(framesWritten, audioHALFrames)) { 3867 if (track->isStopped()) { 3868 track->reset(); 3869 } 3870 tracksToRemove->add(track); 3871 } 3872 } else { 3873 // No buffers for this track. Give it a few chances to 3874 // fill a buffer, then remove it from active list. 3875 // Only consider last track started for mixer state control 3876 if (--(track->mRetryCount) <= 0) { 3877 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3878 tracksToRemove->add(track); 3879 // indicate to client process that the track was disabled because of underrun; 3880 // it will then automatically call start() when data is available 3881 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3882 } else if (last) { 3883 mixerStatus = MIXER_TRACKS_ENABLED; 3884 } 3885 } 3886 } 3887 } 3888 3889 // remove all the tracks that need to be... 3890 removeTracks_l(*tracksToRemove); 3891 3892 return mixerStatus; 3893} 3894 3895void AudioFlinger::DirectOutputThread::threadLoop_mix() 3896{ 3897 size_t frameCount = mFrameCount; 3898 int8_t *curBuf = (int8_t *)mSinkBuffer; 3899 // output audio to hardware 3900 while (frameCount) { 3901 AudioBufferProvider::Buffer buffer; 3902 buffer.frameCount = frameCount; 3903 mActiveTrack->getNextBuffer(&buffer); 3904 if (buffer.raw == NULL) { 3905 memset(curBuf, 0, frameCount * mFrameSize); 3906 break; 3907 } 3908 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3909 frameCount -= buffer.frameCount; 3910 curBuf += buffer.frameCount * mFrameSize; 3911 mActiveTrack->releaseBuffer(&buffer); 3912 } 3913 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3914 sleepTime = 0; 3915 standbyTime = systemTime() + standbyDelay; 3916 mActiveTrack.clear(); 3917} 3918 3919void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3920{ 3921 if (sleepTime == 0) { 3922 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3923 sleepTime = activeSleepTime; 3924 } else { 3925 sleepTime = idleSleepTime; 3926 } 3927 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3928 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3929 sleepTime = 0; 3930 } 3931} 3932 3933// getTrackName_l() must be called with ThreadBase::mLock held 3934int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3935 int sessionId __unused) 3936{ 3937 return 0; 3938} 3939 3940// deleteTrackName_l() must be called with ThreadBase::mLock held 3941void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3942{ 3943} 3944 3945// checkForNewParameter_l() must be called with ThreadBase::mLock held 3946bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 3947 status_t& status) 3948{ 3949 bool reconfig = false; 3950 3951 status = NO_ERROR; 3952 3953 AudioParameter param = AudioParameter(keyValuePair); 3954 int value; 3955 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3956 // forward device change to effects that have requested to be 3957 // aware of attached audio device. 3958 if (value != AUDIO_DEVICE_NONE) { 3959 mOutDevice = value; 3960 for (size_t i = 0; i < mEffectChains.size(); i++) { 3961 mEffectChains[i]->setDevice_l(mOutDevice); 3962 } 3963 } 3964 } 3965 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3966 // do not accept frame count changes if tracks are open as the track buffer 3967 // size depends on frame count and correct behavior would not be garantied 3968 // if frame count is changed after track creation 3969 if (!mTracks.isEmpty()) { 3970 status = INVALID_OPERATION; 3971 } else { 3972 reconfig = true; 3973 } 3974 } 3975 if (status == NO_ERROR) { 3976 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3977 keyValuePair.string()); 3978 if (!mStandby && status == INVALID_OPERATION) { 3979 mOutput->stream->common.standby(&mOutput->stream->common); 3980 mStandby = true; 3981 mBytesWritten = 0; 3982 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3983 keyValuePair.string()); 3984 } 3985 if (status == NO_ERROR && reconfig) { 3986 readOutputParameters_l(); 3987 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3988 } 3989 } 3990 3991 return reconfig; 3992} 3993 3994uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3995{ 3996 uint32_t time; 3997 if (audio_is_linear_pcm(mFormat)) { 3998 time = PlaybackThread::activeSleepTimeUs(); 3999 } else { 4000 time = 10000; 4001 } 4002 return time; 4003} 4004 4005uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4006{ 4007 uint32_t time; 4008 if (audio_is_linear_pcm(mFormat)) { 4009 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4010 } else { 4011 time = 10000; 4012 } 4013 return time; 4014} 4015 4016uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4017{ 4018 uint32_t time; 4019 if (audio_is_linear_pcm(mFormat)) { 4020 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4021 } else { 4022 time = 10000; 4023 } 4024 return time; 4025} 4026 4027void AudioFlinger::DirectOutputThread::cacheParameters_l() 4028{ 4029 PlaybackThread::cacheParameters_l(); 4030 4031 // use shorter standby delay as on normal output to release 4032 // hardware resources as soon as possible 4033 if (audio_is_linear_pcm(mFormat)) { 4034 standbyDelay = microseconds(activeSleepTime*2); 4035 } else { 4036 standbyDelay = kOffloadStandbyDelayNs; 4037 } 4038} 4039 4040// ---------------------------------------------------------------------------- 4041 4042AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4043 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4044 : Thread(false /*canCallJava*/), 4045 mPlaybackThread(playbackThread), 4046 mWriteAckSequence(0), 4047 mDrainSequence(0) 4048{ 4049} 4050 4051AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4052{ 4053} 4054 4055void AudioFlinger::AsyncCallbackThread::onFirstRef() 4056{ 4057 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4058} 4059 4060bool AudioFlinger::AsyncCallbackThread::threadLoop() 4061{ 4062 while (!exitPending()) { 4063 uint32_t writeAckSequence; 4064 uint32_t drainSequence; 4065 4066 { 4067 Mutex::Autolock _l(mLock); 4068 while (!((mWriteAckSequence & 1) || 4069 (mDrainSequence & 1) || 4070 exitPending())) { 4071 mWaitWorkCV.wait(mLock); 4072 } 4073 4074 if (exitPending()) { 4075 break; 4076 } 4077 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4078 mWriteAckSequence, mDrainSequence); 4079 writeAckSequence = mWriteAckSequence; 4080 mWriteAckSequence &= ~1; 4081 drainSequence = mDrainSequence; 4082 mDrainSequence &= ~1; 4083 } 4084 { 4085 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4086 if (playbackThread != 0) { 4087 if (writeAckSequence & 1) { 4088 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4089 } 4090 if (drainSequence & 1) { 4091 playbackThread->resetDraining(drainSequence >> 1); 4092 } 4093 } 4094 } 4095 } 4096 return false; 4097} 4098 4099void AudioFlinger::AsyncCallbackThread::exit() 4100{ 4101 ALOGV("AsyncCallbackThread::exit"); 4102 Mutex::Autolock _l(mLock); 4103 requestExit(); 4104 mWaitWorkCV.broadcast(); 4105} 4106 4107void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4108{ 4109 Mutex::Autolock _l(mLock); 4110 // bit 0 is cleared 4111 mWriteAckSequence = sequence << 1; 4112} 4113 4114void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4115{ 4116 Mutex::Autolock _l(mLock); 4117 // ignore unexpected callbacks 4118 if (mWriteAckSequence & 2) { 4119 mWriteAckSequence |= 1; 4120 mWaitWorkCV.signal(); 4121 } 4122} 4123 4124void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4125{ 4126 Mutex::Autolock _l(mLock); 4127 // bit 0 is cleared 4128 mDrainSequence = sequence << 1; 4129} 4130 4131void AudioFlinger::AsyncCallbackThread::resetDraining() 4132{ 4133 Mutex::Autolock _l(mLock); 4134 // ignore unexpected callbacks 4135 if (mDrainSequence & 2) { 4136 mDrainSequence |= 1; 4137 mWaitWorkCV.signal(); 4138 } 4139} 4140 4141 4142// ---------------------------------------------------------------------------- 4143AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4144 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4145 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4146 mHwPaused(false), 4147 mFlushPending(false), 4148 mPausedBytesRemaining(0) 4149{ 4150 //FIXME: mStandby should be set to true by ThreadBase constructor 4151 mStandby = true; 4152} 4153 4154void AudioFlinger::OffloadThread::threadLoop_exit() 4155{ 4156 if (mFlushPending || mHwPaused) { 4157 // If a flush is pending or track was paused, just discard buffered data 4158 flushHw_l(); 4159 } else { 4160 mMixerStatus = MIXER_DRAIN_ALL; 4161 threadLoop_drain(); 4162 } 4163 if (mUseAsyncWrite) { 4164 ALOG_ASSERT(mCallbackThread != 0); 4165 mCallbackThread->exit(); 4166 } 4167 PlaybackThread::threadLoop_exit(); 4168} 4169 4170AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4171 Vector< sp<Track> > *tracksToRemove 4172) 4173{ 4174 size_t count = mActiveTracks.size(); 4175 4176 mixer_state mixerStatus = MIXER_IDLE; 4177 bool doHwPause = false; 4178 bool doHwResume = false; 4179 4180 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4181 4182 // find out which tracks need to be processed 4183 for (size_t i = 0; i < count; i++) { 4184 sp<Track> t = mActiveTracks[i].promote(); 4185 // The track died recently 4186 if (t == 0) { 4187 continue; 4188 } 4189 Track* const track = t.get(); 4190 audio_track_cblk_t* cblk = track->cblk(); 4191 // Only consider last track started for volume and mixer state control. 4192 // In theory an older track could underrun and restart after the new one starts 4193 // but as we only care about the transition phase between two tracks on a 4194 // direct output, it is not a problem to ignore the underrun case. 4195 sp<Track> l = mLatestActiveTrack.promote(); 4196 bool last = l.get() == track; 4197 4198 if (track->isInvalid()) { 4199 ALOGW("An invalidated track shouldn't be in active list"); 4200 tracksToRemove->add(track); 4201 continue; 4202 } 4203 4204 if (track->mState == TrackBase::IDLE) { 4205 ALOGW("An idle track shouldn't be in active list"); 4206 continue; 4207 } 4208 4209 if (track->isPausing()) { 4210 track->setPaused(); 4211 if (last) { 4212 if (!mHwPaused) { 4213 doHwPause = true; 4214 mHwPaused = true; 4215 } 4216 // If we were part way through writing the mixbuffer to 4217 // the HAL we must save this until we resume 4218 // BUG - this will be wrong if a different track is made active, 4219 // in that case we want to discard the pending data in the 4220 // mixbuffer and tell the client to present it again when the 4221 // track is resumed 4222 mPausedWriteLength = mCurrentWriteLength; 4223 mPausedBytesRemaining = mBytesRemaining; 4224 mBytesRemaining = 0; // stop writing 4225 } 4226 tracksToRemove->add(track); 4227 } else if (track->isFlushPending()) { 4228 track->flushAck(); 4229 if (last) { 4230 mFlushPending = true; 4231 } 4232 } else if (track->isResumePending()){ 4233 track->resumeAck(); 4234 if (last) { 4235 if (mPausedBytesRemaining) { 4236 // Need to continue write that was interrupted 4237 mCurrentWriteLength = mPausedWriteLength; 4238 mBytesRemaining = mPausedBytesRemaining; 4239 mPausedBytesRemaining = 0; 4240 } 4241 if (mHwPaused) { 4242 doHwResume = true; 4243 mHwPaused = false; 4244 // threadLoop_mix() will handle the case that we need to 4245 // resume an interrupted write 4246 } 4247 // enable write to audio HAL 4248 sleepTime = 0; 4249 4250 // Do not handle new data in this iteration even if track->framesReady() 4251 mixerStatus = MIXER_TRACKS_ENABLED; 4252 } 4253 } else if (track->framesReady() && track->isReady() && 4254 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4255 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4256 if (track->mFillingUpStatus == Track::FS_FILLED) { 4257 track->mFillingUpStatus = Track::FS_ACTIVE; 4258 // make sure processVolume_l() will apply new volume even if 0 4259 mLeftVolFloat = mRightVolFloat = -1.0; 4260 } 4261 4262 if (last) { 4263 sp<Track> previousTrack = mPreviousTrack.promote(); 4264 if (previousTrack != 0) { 4265 if (track != previousTrack.get()) { 4266 // Flush any data still being written from last track 4267 mBytesRemaining = 0; 4268 if (mPausedBytesRemaining) { 4269 // Last track was paused so we also need to flush saved 4270 // mixbuffer state and invalidate track so that it will 4271 // re-submit that unwritten data when it is next resumed 4272 mPausedBytesRemaining = 0; 4273 // Invalidate is a bit drastic - would be more efficient 4274 // to have a flag to tell client that some of the 4275 // previously written data was lost 4276 previousTrack->invalidate(); 4277 } 4278 // flush data already sent to the DSP if changing audio session as audio 4279 // comes from a different source. Also invalidate previous track to force a 4280 // seek when resuming. 4281 if (previousTrack->sessionId() != track->sessionId()) { 4282 previousTrack->invalidate(); 4283 } 4284 } 4285 } 4286 mPreviousTrack = track; 4287 // reset retry count 4288 track->mRetryCount = kMaxTrackRetriesOffload; 4289 mActiveTrack = t; 4290 mixerStatus = MIXER_TRACKS_READY; 4291 } 4292 } else { 4293 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4294 if (track->isStopping_1()) { 4295 // Hardware buffer can hold a large amount of audio so we must 4296 // wait for all current track's data to drain before we say 4297 // that the track is stopped. 4298 if (mBytesRemaining == 0) { 4299 // Only start draining when all data in mixbuffer 4300 // has been written 4301 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4302 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4303 // do not drain if no data was ever sent to HAL (mStandby == true) 4304 if (last && !mStandby) { 4305 // do not modify drain sequence if we are already draining. This happens 4306 // when resuming from pause after drain. 4307 if ((mDrainSequence & 1) == 0) { 4308 sleepTime = 0; 4309 standbyTime = systemTime() + standbyDelay; 4310 mixerStatus = MIXER_DRAIN_TRACK; 4311 mDrainSequence += 2; 4312 } 4313 if (mHwPaused) { 4314 // It is possible to move from PAUSED to STOPPING_1 without 4315 // a resume so we must ensure hardware is running 4316 doHwResume = true; 4317 mHwPaused = false; 4318 } 4319 } 4320 } 4321 } else if (track->isStopping_2()) { 4322 // Drain has completed or we are in standby, signal presentation complete 4323 if (!(mDrainSequence & 1) || !last || mStandby) { 4324 track->mState = TrackBase::STOPPED; 4325 size_t audioHALFrames = 4326 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4327 size_t framesWritten = 4328 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4329 track->presentationComplete(framesWritten, audioHALFrames); 4330 track->reset(); 4331 tracksToRemove->add(track); 4332 } 4333 } else { 4334 // No buffers for this track. Give it a few chances to 4335 // fill a buffer, then remove it from active list. 4336 if (--(track->mRetryCount) <= 0) { 4337 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4338 track->name()); 4339 tracksToRemove->add(track); 4340 // indicate to client process that the track was disabled because of underrun; 4341 // it will then automatically call start() when data is available 4342 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4343 } else if (last){ 4344 mixerStatus = MIXER_TRACKS_ENABLED; 4345 } 4346 } 4347 } 4348 // compute volume for this track 4349 processVolume_l(track, last); 4350 } 4351 4352 // make sure the pause/flush/resume sequence is executed in the right order. 4353 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4354 // before flush and then resume HW. This can happen in case of pause/flush/resume 4355 // if resume is received before pause is executed. 4356 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4357 mOutput->stream->pause(mOutput->stream); 4358 } 4359 if (mFlushPending) { 4360 flushHw_l(); 4361 mFlushPending = false; 4362 } 4363 if (!mStandby && doHwResume) { 4364 mOutput->stream->resume(mOutput->stream); 4365 } 4366 4367 // remove all the tracks that need to be... 4368 removeTracks_l(*tracksToRemove); 4369 4370 return mixerStatus; 4371} 4372 4373// must be called with thread mutex locked 4374bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4375{ 4376 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4377 mWriteAckSequence, mDrainSequence); 4378 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4379 return true; 4380 } 4381 return false; 4382} 4383 4384// must be called with thread mutex locked 4385bool AudioFlinger::OffloadThread::shouldStandby_l() 4386{ 4387 bool trackPaused = false; 4388 4389 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4390 // after a timeout and we will enter standby then. 4391 if (mTracks.size() > 0) { 4392 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4393 } 4394 4395 return !mStandby && !trackPaused; 4396} 4397 4398 4399bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4400{ 4401 Mutex::Autolock _l(mLock); 4402 return waitingAsyncCallback_l(); 4403} 4404 4405void AudioFlinger::OffloadThread::flushHw_l() 4406{ 4407 mOutput->stream->flush(mOutput->stream); 4408 // Flush anything still waiting in the mixbuffer 4409 mCurrentWriteLength = 0; 4410 mBytesRemaining = 0; 4411 mPausedWriteLength = 0; 4412 mPausedBytesRemaining = 0; 4413 mHwPaused = false; 4414 4415 if (mUseAsyncWrite) { 4416 // discard any pending drain or write ack by incrementing sequence 4417 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4418 mDrainSequence = (mDrainSequence + 2) & ~1; 4419 ALOG_ASSERT(mCallbackThread != 0); 4420 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4421 mCallbackThread->setDraining(mDrainSequence); 4422 } 4423} 4424 4425void AudioFlinger::OffloadThread::onAddNewTrack_l() 4426{ 4427 sp<Track> previousTrack = mPreviousTrack.promote(); 4428 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4429 4430 if (previousTrack != 0 && latestTrack != 0 && 4431 (previousTrack->sessionId() != latestTrack->sessionId())) { 4432 mFlushPending = true; 4433 } 4434 PlaybackThread::onAddNewTrack_l(); 4435} 4436 4437// ---------------------------------------------------------------------------- 4438 4439AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4440 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4441 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4442 DUPLICATING), 4443 mWaitTimeMs(UINT_MAX) 4444{ 4445 addOutputTrack(mainThread); 4446} 4447 4448AudioFlinger::DuplicatingThread::~DuplicatingThread() 4449{ 4450 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4451 mOutputTracks[i]->destroy(); 4452 } 4453} 4454 4455void AudioFlinger::DuplicatingThread::threadLoop_mix() 4456{ 4457 // mix buffers... 4458 if (outputsReady(outputTracks)) { 4459 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4460 } else { 4461 memset(mSinkBuffer, 0, mSinkBufferSize); 4462 } 4463 sleepTime = 0; 4464 writeFrames = mNormalFrameCount; 4465 mCurrentWriteLength = mSinkBufferSize; 4466 standbyTime = systemTime() + standbyDelay; 4467} 4468 4469void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4470{ 4471 if (sleepTime == 0) { 4472 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4473 sleepTime = activeSleepTime; 4474 } else { 4475 sleepTime = idleSleepTime; 4476 } 4477 } else if (mBytesWritten != 0) { 4478 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4479 writeFrames = mNormalFrameCount; 4480 memset(mSinkBuffer, 0, mSinkBufferSize); 4481 } else { 4482 // flush remaining overflow buffers in output tracks 4483 writeFrames = 0; 4484 } 4485 sleepTime = 0; 4486 } 4487} 4488 4489ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4490{ 4491 for (size_t i = 0; i < outputTracks.size(); i++) { 4492 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4493 // for delivery downstream as needed. This in-place conversion is safe as 4494 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4495 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4496 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4497 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4498 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4499 } 4500 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4501 } 4502 mStandby = false; 4503 return (ssize_t)mSinkBufferSize; 4504} 4505 4506void AudioFlinger::DuplicatingThread::threadLoop_standby() 4507{ 4508 // DuplicatingThread implements standby by stopping all tracks 4509 for (size_t i = 0; i < outputTracks.size(); i++) { 4510 outputTracks[i]->stop(); 4511 } 4512} 4513 4514void AudioFlinger::DuplicatingThread::saveOutputTracks() 4515{ 4516 outputTracks = mOutputTracks; 4517} 4518 4519void AudioFlinger::DuplicatingThread::clearOutputTracks() 4520{ 4521 outputTracks.clear(); 4522} 4523 4524void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4525{ 4526 Mutex::Autolock _l(mLock); 4527 // FIXME explain this formula 4528 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4529 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4530 // due to current usage case and restrictions on the AudioBufferProvider. 4531 // Actual buffer conversion is done in threadLoop_write(). 4532 // 4533 // TODO: This may change in the future, depending on multichannel 4534 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4535 OutputTrack *outputTrack = new OutputTrack(thread, 4536 this, 4537 mSampleRate, 4538 AUDIO_FORMAT_PCM_16_BIT, 4539 mChannelMask, 4540 frameCount, 4541 IPCThreadState::self()->getCallingUid()); 4542 if (outputTrack->cblk() != NULL) { 4543 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4544 mOutputTracks.add(outputTrack); 4545 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4546 updateWaitTime_l(); 4547 } 4548} 4549 4550void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4551{ 4552 Mutex::Autolock _l(mLock); 4553 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4554 if (mOutputTracks[i]->thread() == thread) { 4555 mOutputTracks[i]->destroy(); 4556 mOutputTracks.removeAt(i); 4557 updateWaitTime_l(); 4558 return; 4559 } 4560 } 4561 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4562} 4563 4564// caller must hold mLock 4565void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4566{ 4567 mWaitTimeMs = UINT_MAX; 4568 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4569 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4570 if (strong != 0) { 4571 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4572 if (waitTimeMs < mWaitTimeMs) { 4573 mWaitTimeMs = waitTimeMs; 4574 } 4575 } 4576 } 4577} 4578 4579 4580bool AudioFlinger::DuplicatingThread::outputsReady( 4581 const SortedVector< sp<OutputTrack> > &outputTracks) 4582{ 4583 for (size_t i = 0; i < outputTracks.size(); i++) { 4584 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4585 if (thread == 0) { 4586 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4587 outputTracks[i].get()); 4588 return false; 4589 } 4590 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4591 // see note at standby() declaration 4592 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4593 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4594 thread.get()); 4595 return false; 4596 } 4597 } 4598 return true; 4599} 4600 4601uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4602{ 4603 return (mWaitTimeMs * 1000) / 2; 4604} 4605 4606void AudioFlinger::DuplicatingThread::cacheParameters_l() 4607{ 4608 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4609 updateWaitTime_l(); 4610 4611 MixerThread::cacheParameters_l(); 4612} 4613 4614// ---------------------------------------------------------------------------- 4615// Record 4616// ---------------------------------------------------------------------------- 4617 4618AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4619 AudioStreamIn *input, 4620 audio_io_handle_t id, 4621 audio_devices_t outDevice, 4622 audio_devices_t inDevice 4623#ifdef TEE_SINK 4624 , const sp<NBAIO_Sink>& teeSink 4625#endif 4626 ) : 4627 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4628 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4629 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4630 mRsmpInRear(0) 4631#ifdef TEE_SINK 4632 , mTeeSink(teeSink) 4633#endif 4634 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4635 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4636{ 4637 snprintf(mName, kNameLength, "AudioIn_%X", id); 4638 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4639 4640 readInputParameters_l(); 4641} 4642 4643 4644AudioFlinger::RecordThread::~RecordThread() 4645{ 4646 mAudioFlinger->unregisterWriter(mNBLogWriter); 4647 delete[] mRsmpInBuffer; 4648} 4649 4650void AudioFlinger::RecordThread::onFirstRef() 4651{ 4652 run(mName, PRIORITY_URGENT_AUDIO); 4653} 4654 4655bool AudioFlinger::RecordThread::threadLoop() 4656{ 4657 nsecs_t lastWarning = 0; 4658 4659 inputStandBy(); 4660 4661reacquire_wakelock: 4662 sp<RecordTrack> activeTrack; 4663 int activeTracksGen; 4664 { 4665 Mutex::Autolock _l(mLock); 4666 size_t size = mActiveTracks.size(); 4667 activeTracksGen = mActiveTracksGen; 4668 if (size > 0) { 4669 // FIXME an arbitrary choice 4670 activeTrack = mActiveTracks[0]; 4671 acquireWakeLock_l(activeTrack->uid()); 4672 if (size > 1) { 4673 SortedVector<int> tmp; 4674 for (size_t i = 0; i < size; i++) { 4675 tmp.add(mActiveTracks[i]->uid()); 4676 } 4677 updateWakeLockUids_l(tmp); 4678 } 4679 } else { 4680 acquireWakeLock_l(-1); 4681 } 4682 } 4683 4684 // used to request a deferred sleep, to be executed later while mutex is unlocked 4685 uint32_t sleepUs = 0; 4686 4687 // loop while there is work to do 4688 for (;;) { 4689 Vector< sp<EffectChain> > effectChains; 4690 4691 // sleep with mutex unlocked 4692 if (sleepUs > 0) { 4693 usleep(sleepUs); 4694 sleepUs = 0; 4695 } 4696 4697 // activeTracks accumulates a copy of a subset of mActiveTracks 4698 Vector< sp<RecordTrack> > activeTracks; 4699 4700 4701 { // scope for mLock 4702 Mutex::Autolock _l(mLock); 4703 4704 processConfigEvents_l(); 4705 4706 // check exitPending here because checkForNewParameters_l() and 4707 // checkForNewParameters_l() can temporarily release mLock 4708 if (exitPending()) { 4709 break; 4710 } 4711 4712 // if no active track(s), then standby and release wakelock 4713 size_t size = mActiveTracks.size(); 4714 if (size == 0) { 4715 standbyIfNotAlreadyInStandby(); 4716 // exitPending() can't become true here 4717 releaseWakeLock_l(); 4718 ALOGV("RecordThread: loop stopping"); 4719 // go to sleep 4720 mWaitWorkCV.wait(mLock); 4721 ALOGV("RecordThread: loop starting"); 4722 goto reacquire_wakelock; 4723 } 4724 4725 if (mActiveTracksGen != activeTracksGen) { 4726 activeTracksGen = mActiveTracksGen; 4727 SortedVector<int> tmp; 4728 for (size_t i = 0; i < size; i++) { 4729 tmp.add(mActiveTracks[i]->uid()); 4730 } 4731 updateWakeLockUids_l(tmp); 4732 } 4733 4734 bool doBroadcast = false; 4735 for (size_t i = 0; i < size; ) { 4736 4737 activeTrack = mActiveTracks[i]; 4738 if (activeTrack->isTerminated()) { 4739 removeTrack_l(activeTrack); 4740 mActiveTracks.remove(activeTrack); 4741 mActiveTracksGen++; 4742 size--; 4743 continue; 4744 } 4745 4746 TrackBase::track_state activeTrackState = activeTrack->mState; 4747 switch (activeTrackState) { 4748 4749 case TrackBase::PAUSING: 4750 mActiveTracks.remove(activeTrack); 4751 mActiveTracksGen++; 4752 doBroadcast = true; 4753 size--; 4754 continue; 4755 4756 case TrackBase::STARTING_1: 4757 sleepUs = 10000; 4758 i++; 4759 continue; 4760 4761 case TrackBase::STARTING_2: 4762 doBroadcast = true; 4763 mStandby = false; 4764 activeTrack->mState = TrackBase::ACTIVE; 4765 break; 4766 4767 case TrackBase::ACTIVE: 4768 break; 4769 4770 case TrackBase::IDLE: 4771 i++; 4772 continue; 4773 4774 default: 4775 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4776 } 4777 4778 activeTracks.add(activeTrack); 4779 i++; 4780 4781 } 4782 if (doBroadcast) { 4783 mStartStopCond.broadcast(); 4784 } 4785 4786 // sleep if there are no active tracks to process 4787 if (activeTracks.size() == 0) { 4788 if (sleepUs == 0) { 4789 sleepUs = kRecordThreadSleepUs; 4790 } 4791 continue; 4792 } 4793 sleepUs = 0; 4794 4795 lockEffectChains_l(effectChains); 4796 } 4797 4798 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4799 4800 size_t size = effectChains.size(); 4801 for (size_t i = 0; i < size; i++) { 4802 // thread mutex is not locked, but effect chain is locked 4803 effectChains[i]->process_l(); 4804 } 4805 4806 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4807 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4808 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4809 // If destination is non-contiguous, first read past the nominal end of buffer, then 4810 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4811 4812 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4813 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4814 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4815 if (bytesRead <= 0) { 4816 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4817 // Force input into standby so that it tries to recover at next read attempt 4818 inputStandBy(); 4819 sleepUs = kRecordThreadSleepUs; 4820 continue; 4821 } 4822 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4823 size_t framesRead = bytesRead / mFrameSize; 4824 ALOG_ASSERT(framesRead > 0); 4825 if (mTeeSink != 0) { 4826 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4827 } 4828 // If destination is non-contiguous, we now correct for reading past end of buffer. 4829 size_t part1 = mRsmpInFramesP2 - rear; 4830 if (framesRead > part1) { 4831 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4832 (framesRead - part1) * mFrameSize); 4833 } 4834 rear = mRsmpInRear += framesRead; 4835 4836 size = activeTracks.size(); 4837 // loop over each active track 4838 for (size_t i = 0; i < size; i++) { 4839 activeTrack = activeTracks[i]; 4840 4841 enum { 4842 OVERRUN_UNKNOWN, 4843 OVERRUN_TRUE, 4844 OVERRUN_FALSE 4845 } overrun = OVERRUN_UNKNOWN; 4846 4847 // loop over getNextBuffer to handle circular sink 4848 for (;;) { 4849 4850 activeTrack->mSink.frameCount = ~0; 4851 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4852 size_t framesOut = activeTrack->mSink.frameCount; 4853 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4854 4855 int32_t front = activeTrack->mRsmpInFront; 4856 ssize_t filled = rear - front; 4857 size_t framesIn; 4858 4859 if (filled < 0) { 4860 // should not happen, but treat like a massive overrun and re-sync 4861 framesIn = 0; 4862 activeTrack->mRsmpInFront = rear; 4863 overrun = OVERRUN_TRUE; 4864 } else if ((size_t) filled <= mRsmpInFrames) { 4865 framesIn = (size_t) filled; 4866 } else { 4867 // client is not keeping up with server, but give it latest data 4868 framesIn = mRsmpInFrames; 4869 activeTrack->mRsmpInFront = front = rear - framesIn; 4870 overrun = OVERRUN_TRUE; 4871 } 4872 4873 if (framesOut == 0 || framesIn == 0) { 4874 break; 4875 } 4876 4877 if (activeTrack->mResampler == NULL) { 4878 // no resampling 4879 if (framesIn > framesOut) { 4880 framesIn = framesOut; 4881 } else { 4882 framesOut = framesIn; 4883 } 4884 int8_t *dst = activeTrack->mSink.i8; 4885 while (framesIn > 0) { 4886 front &= mRsmpInFramesP2 - 1; 4887 size_t part1 = mRsmpInFramesP2 - front; 4888 if (part1 > framesIn) { 4889 part1 = framesIn; 4890 } 4891 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4892 if (mChannelCount == activeTrack->mChannelCount) { 4893 memcpy(dst, src, part1 * mFrameSize); 4894 } else if (mChannelCount == 1) { 4895 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4896 part1); 4897 } else { 4898 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4899 part1); 4900 } 4901 dst += part1 * activeTrack->mFrameSize; 4902 front += part1; 4903 framesIn -= part1; 4904 } 4905 activeTrack->mRsmpInFront += framesOut; 4906 4907 } else { 4908 // resampling 4909 // FIXME framesInNeeded should really be part of resampler API, and should 4910 // depend on the SRC ratio 4911 // to keep mRsmpInBuffer full so resampler always has sufficient input 4912 size_t framesInNeeded; 4913 // FIXME only re-calculate when it changes, and optimize for common ratios 4914 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4915 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4916 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4917 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4918 framesInNeeded, framesOut, inOverOut); 4919 // Although we theoretically have framesIn in circular buffer, some of those are 4920 // unreleased frames, and thus must be discounted for purpose of budgeting. 4921 size_t unreleased = activeTrack->mRsmpInUnrel; 4922 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4923 if (framesIn < framesInNeeded) { 4924 ALOGV("not enough to resample: have %u frames in but need %u in to " 4925 "produce %u out given in/out ratio of %.4g", 4926 framesIn, framesInNeeded, framesOut, inOverOut); 4927 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4928 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4929 if (newFramesOut == 0) { 4930 break; 4931 } 4932 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4933 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4934 framesInNeeded, newFramesOut, outOverIn); 4935 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4936 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4937 "given in/out ratio of %.4g", 4938 framesIn, framesInNeeded, newFramesOut, inOverOut); 4939 framesOut = newFramesOut; 4940 } else { 4941 ALOGV("success 1: have %u in and need %u in to produce %u out " 4942 "given in/out ratio of %.4g", 4943 framesIn, framesInNeeded, framesOut, inOverOut); 4944 } 4945 4946 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4947 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4948 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4949 delete[] activeTrack->mRsmpOutBuffer; 4950 // resampler always outputs stereo 4951 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4952 activeTrack->mRsmpOutFrameCount = framesOut; 4953 } 4954 4955 // resampler accumulates, but we only have one source track 4956 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4957 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4958 // FIXME how about having activeTrack implement this interface itself? 4959 activeTrack->mResamplerBufferProvider 4960 /*this*/ /* AudioBufferProvider* */); 4961 // ditherAndClamp() works as long as all buffers returned by 4962 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4963 if (activeTrack->mChannelCount == 1) { 4964 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 4965 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4966 framesOut); 4967 // the resampler always outputs stereo samples: 4968 // do post stereo to mono conversion 4969 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4970 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4971 } else { 4972 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4973 activeTrack->mRsmpOutBuffer, framesOut); 4974 } 4975 // now done with mRsmpOutBuffer 4976 4977 } 4978 4979 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4980 overrun = OVERRUN_FALSE; 4981 } 4982 4983 if (activeTrack->mFramesToDrop == 0) { 4984 if (framesOut > 0) { 4985 activeTrack->mSink.frameCount = framesOut; 4986 activeTrack->releaseBuffer(&activeTrack->mSink); 4987 } 4988 } else { 4989 // FIXME could do a partial drop of framesOut 4990 if (activeTrack->mFramesToDrop > 0) { 4991 activeTrack->mFramesToDrop -= framesOut; 4992 if (activeTrack->mFramesToDrop <= 0) { 4993 activeTrack->clearSyncStartEvent(); 4994 } 4995 } else { 4996 activeTrack->mFramesToDrop += framesOut; 4997 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4998 activeTrack->mSyncStartEvent->isCancelled()) { 4999 ALOGW("Synced record %s, session %d, trigger session %d", 5000 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5001 activeTrack->sessionId(), 5002 (activeTrack->mSyncStartEvent != 0) ? 5003 activeTrack->mSyncStartEvent->triggerSession() : 0); 5004 activeTrack->clearSyncStartEvent(); 5005 } 5006 } 5007 } 5008 5009 if (framesOut == 0) { 5010 break; 5011 } 5012 } 5013 5014 switch (overrun) { 5015 case OVERRUN_TRUE: 5016 // client isn't retrieving buffers fast enough 5017 if (!activeTrack->setOverflow()) { 5018 nsecs_t now = systemTime(); 5019 // FIXME should lastWarning per track? 5020 if ((now - lastWarning) > kWarningThrottleNs) { 5021 ALOGW("RecordThread: buffer overflow"); 5022 lastWarning = now; 5023 } 5024 } 5025 break; 5026 case OVERRUN_FALSE: 5027 activeTrack->clearOverflow(); 5028 break; 5029 case OVERRUN_UNKNOWN: 5030 break; 5031 } 5032 5033 } 5034 5035 // enable changes in effect chain 5036 unlockEffectChains(effectChains); 5037 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5038 } 5039 5040 standbyIfNotAlreadyInStandby(); 5041 5042 { 5043 Mutex::Autolock _l(mLock); 5044 for (size_t i = 0; i < mTracks.size(); i++) { 5045 sp<RecordTrack> track = mTracks[i]; 5046 track->invalidate(); 5047 } 5048 mActiveTracks.clear(); 5049 mActiveTracksGen++; 5050 mStartStopCond.broadcast(); 5051 } 5052 5053 releaseWakeLock(); 5054 5055 ALOGV("RecordThread %p exiting", this); 5056 return false; 5057} 5058 5059void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5060{ 5061 if (!mStandby) { 5062 inputStandBy(); 5063 mStandby = true; 5064 } 5065} 5066 5067void AudioFlinger::RecordThread::inputStandBy() 5068{ 5069 mInput->stream->common.standby(&mInput->stream->common); 5070} 5071 5072// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5073sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5074 const sp<AudioFlinger::Client>& client, 5075 uint32_t sampleRate, 5076 audio_format_t format, 5077 audio_channel_mask_t channelMask, 5078 size_t *pFrameCount, 5079 int sessionId, 5080 int uid, 5081 IAudioFlinger::track_flags_t *flags, 5082 pid_t tid, 5083 status_t *status) 5084{ 5085 size_t frameCount = *pFrameCount; 5086 sp<RecordTrack> track; 5087 status_t lStatus; 5088 5089 // client expresses a preference for FAST, but we get the final say 5090 if (*flags & IAudioFlinger::TRACK_FAST) { 5091 if ( 5092 // use case: callback handler and frame count is default or at least as large as HAL 5093 ( 5094 (tid != -1) && 5095 ((frameCount == 0) || 5096 // FIXME not necessarily true, should be native frame count for native SR! 5097 (frameCount >= mFrameCount)) 5098 ) && 5099 // PCM data 5100 audio_is_linear_pcm(format) && 5101 // mono or stereo 5102 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5103 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5104 // hardware sample rate 5105 // FIXME actually the native hardware sample rate 5106 (sampleRate == mSampleRate) && 5107 // record thread has an associated fast capture 5108 hasFastCapture() 5109 // fast capture does not require slots 5110 ) { 5111 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5112 if (frameCount == 0) { 5113 // FIXME wrong mFrameCount 5114 frameCount = mFrameCount * kFastTrackMultiplier; 5115 } 5116 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5117 frameCount, mFrameCount); 5118 } else { 5119 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5120 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5121 "hasFastCapture=%d tid=%d", 5122 frameCount, mFrameCount, format, 5123 audio_is_linear_pcm(format), 5124 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5125 *flags &= ~IAudioFlinger::TRACK_FAST; 5126 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5127 // For compatibility with AudioRecord calculation, buffer depth is forced 5128 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5129 // This is probably too conservative, but legacy application code may depend on it. 5130 // If you change this calculation, also review the start threshold which is related. 5131 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5132 size_t mNormalFrameCount = 2048; // FIXME 5133 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5134 if (minBufCount < 2) { 5135 minBufCount = 2; 5136 } 5137 size_t minFrameCount = mNormalFrameCount * minBufCount; 5138 if (frameCount < minFrameCount) { 5139 frameCount = minFrameCount; 5140 } 5141 } 5142 } 5143 *pFrameCount = frameCount; 5144 5145 lStatus = initCheck(); 5146 if (lStatus != NO_ERROR) { 5147 ALOGE("createRecordTrack_l() audio driver not initialized"); 5148 goto Exit; 5149 } 5150 5151 { // scope for mLock 5152 Mutex::Autolock _l(mLock); 5153 5154 track = new RecordTrack(this, client, sampleRate, 5155 format, channelMask, frameCount, sessionId, uid, 5156 *flags); 5157 5158 lStatus = track->initCheck(); 5159 if (lStatus != NO_ERROR) { 5160 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5161 // track must be cleared from the caller as the caller has the AF lock 5162 goto Exit; 5163 } 5164 mTracks.add(track); 5165 5166 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5167 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5168 mAudioFlinger->btNrecIsOff(); 5169 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5170 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5171 5172 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5173 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5174 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5175 // so ask activity manager to do this on our behalf 5176 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5177 } 5178 } 5179 5180 lStatus = NO_ERROR; 5181 5182Exit: 5183 *status = lStatus; 5184 return track; 5185} 5186 5187status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5188 AudioSystem::sync_event_t event, 5189 int triggerSession) 5190{ 5191 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5192 sp<ThreadBase> strongMe = this; 5193 status_t status = NO_ERROR; 5194 5195 if (event == AudioSystem::SYNC_EVENT_NONE) { 5196 recordTrack->clearSyncStartEvent(); 5197 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5198 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5199 triggerSession, 5200 recordTrack->sessionId(), 5201 syncStartEventCallback, 5202 recordTrack); 5203 // Sync event can be cancelled by the trigger session if the track is not in a 5204 // compatible state in which case we start record immediately 5205 if (recordTrack->mSyncStartEvent->isCancelled()) { 5206 recordTrack->clearSyncStartEvent(); 5207 } else { 5208 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5209 recordTrack->mFramesToDrop = - 5210 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5211 } 5212 } 5213 5214 { 5215 // This section is a rendezvous between binder thread executing start() and RecordThread 5216 AutoMutex lock(mLock); 5217 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5218 if (recordTrack->mState == TrackBase::PAUSING) { 5219 ALOGV("active record track PAUSING -> ACTIVE"); 5220 recordTrack->mState = TrackBase::ACTIVE; 5221 } else { 5222 ALOGV("active record track state %d", recordTrack->mState); 5223 } 5224 return status; 5225 } 5226 5227 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5228 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5229 // or using a separate command thread 5230 recordTrack->mState = TrackBase::STARTING_1; 5231 mActiveTracks.add(recordTrack); 5232 mActiveTracksGen++; 5233 mLock.unlock(); 5234 status_t status = AudioSystem::startInput(mId); 5235 mLock.lock(); 5236 // FIXME should verify that recordTrack is still in mActiveTracks 5237 if (status != NO_ERROR) { 5238 mActiveTracks.remove(recordTrack); 5239 mActiveTracksGen++; 5240 recordTrack->clearSyncStartEvent(); 5241 return status; 5242 } 5243 // Catch up with current buffer indices if thread is already running. 5244 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5245 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5246 // see previously buffered data before it called start(), but with greater risk of overrun. 5247 5248 recordTrack->mRsmpInFront = mRsmpInRear; 5249 recordTrack->mRsmpInUnrel = 0; 5250 // FIXME why reset? 5251 if (recordTrack->mResampler != NULL) { 5252 recordTrack->mResampler->reset(); 5253 } 5254 recordTrack->mState = TrackBase::STARTING_2; 5255 // signal thread to start 5256 mWaitWorkCV.broadcast(); 5257 if (mActiveTracks.indexOf(recordTrack) < 0) { 5258 ALOGV("Record failed to start"); 5259 status = BAD_VALUE; 5260 goto startError; 5261 } 5262 return status; 5263 } 5264 5265startError: 5266 AudioSystem::stopInput(mId); 5267 recordTrack->clearSyncStartEvent(); 5268 // FIXME I wonder why we do not reset the state here? 5269 return status; 5270} 5271 5272void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5273{ 5274 sp<SyncEvent> strongEvent = event.promote(); 5275 5276 if (strongEvent != 0) { 5277 sp<RefBase> ptr = strongEvent->cookie().promote(); 5278 if (ptr != 0) { 5279 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5280 recordTrack->handleSyncStartEvent(strongEvent); 5281 } 5282 } 5283} 5284 5285bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5286 ALOGV("RecordThread::stop"); 5287 AutoMutex _l(mLock); 5288 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5289 return false; 5290 } 5291 // note that threadLoop may still be processing the track at this point [without lock] 5292 recordTrack->mState = TrackBase::PAUSING; 5293 // do not wait for mStartStopCond if exiting 5294 if (exitPending()) { 5295 return true; 5296 } 5297 // FIXME incorrect usage of wait: no explicit predicate or loop 5298 mStartStopCond.wait(mLock); 5299 // if we have been restarted, recordTrack is in mActiveTracks here 5300 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5301 ALOGV("Record stopped OK"); 5302 return true; 5303 } 5304 return false; 5305} 5306 5307bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5308{ 5309 return false; 5310} 5311 5312status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5313{ 5314#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5315 if (!isValidSyncEvent(event)) { 5316 return BAD_VALUE; 5317 } 5318 5319 int eventSession = event->triggerSession(); 5320 status_t ret = NAME_NOT_FOUND; 5321 5322 Mutex::Autolock _l(mLock); 5323 5324 for (size_t i = 0; i < mTracks.size(); i++) { 5325 sp<RecordTrack> track = mTracks[i]; 5326 if (eventSession == track->sessionId()) { 5327 (void) track->setSyncEvent(event); 5328 ret = NO_ERROR; 5329 } 5330 } 5331 return ret; 5332#else 5333 return BAD_VALUE; 5334#endif 5335} 5336 5337// destroyTrack_l() must be called with ThreadBase::mLock held 5338void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5339{ 5340 track->terminate(); 5341 track->mState = TrackBase::STOPPED; 5342 // active tracks are removed by threadLoop() 5343 if (mActiveTracks.indexOf(track) < 0) { 5344 removeTrack_l(track); 5345 } 5346} 5347 5348void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5349{ 5350 mTracks.remove(track); 5351 // need anything related to effects here? 5352} 5353 5354void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5355{ 5356 dumpInternals(fd, args); 5357 dumpTracks(fd, args); 5358 dumpEffectChains(fd, args); 5359} 5360 5361void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5362{ 5363 fdprintf(fd, "\nInput thread %p:\n", this); 5364 5365 if (mActiveTracks.size() > 0) { 5366 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5367 } else { 5368 fdprintf(fd, " No active record clients\n"); 5369 } 5370 5371 dumpBase(fd, args); 5372} 5373 5374void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5375{ 5376 const size_t SIZE = 256; 5377 char buffer[SIZE]; 5378 String8 result; 5379 5380 size_t numtracks = mTracks.size(); 5381 size_t numactive = mActiveTracks.size(); 5382 size_t numactiveseen = 0; 5383 fdprintf(fd, " %d Tracks", numtracks); 5384 if (numtracks) { 5385 fdprintf(fd, " of which %d are active\n", numactive); 5386 RecordTrack::appendDumpHeader(result); 5387 for (size_t i = 0; i < numtracks ; ++i) { 5388 sp<RecordTrack> track = mTracks[i]; 5389 if (track != 0) { 5390 bool active = mActiveTracks.indexOf(track) >= 0; 5391 if (active) { 5392 numactiveseen++; 5393 } 5394 track->dump(buffer, SIZE, active); 5395 result.append(buffer); 5396 } 5397 } 5398 } else { 5399 fdprintf(fd, "\n"); 5400 } 5401 5402 if (numactiveseen != numactive) { 5403 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5404 " not in the track list\n"); 5405 result.append(buffer); 5406 RecordTrack::appendDumpHeader(result); 5407 for (size_t i = 0; i < numactive; ++i) { 5408 sp<RecordTrack> track = mActiveTracks[i]; 5409 if (mTracks.indexOf(track) < 0) { 5410 track->dump(buffer, SIZE, true); 5411 result.append(buffer); 5412 } 5413 } 5414 5415 } 5416 write(fd, result.string(), result.size()); 5417} 5418 5419// AudioBufferProvider interface 5420status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5421 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5422{ 5423 RecordTrack *activeTrack = mRecordTrack; 5424 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5425 if (threadBase == 0) { 5426 buffer->frameCount = 0; 5427 buffer->raw = NULL; 5428 return NOT_ENOUGH_DATA; 5429 } 5430 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5431 int32_t rear = recordThread->mRsmpInRear; 5432 int32_t front = activeTrack->mRsmpInFront; 5433 ssize_t filled = rear - front; 5434 // FIXME should not be P2 (don't want to increase latency) 5435 // FIXME if client not keeping up, discard 5436 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5437 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5438 front &= recordThread->mRsmpInFramesP2 - 1; 5439 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5440 if (part1 > (size_t) filled) { 5441 part1 = filled; 5442 } 5443 size_t ask = buffer->frameCount; 5444 ALOG_ASSERT(ask > 0); 5445 if (part1 > ask) { 5446 part1 = ask; 5447 } 5448 if (part1 == 0) { 5449 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5450 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5451 buffer->raw = NULL; 5452 buffer->frameCount = 0; 5453 activeTrack->mRsmpInUnrel = 0; 5454 return NOT_ENOUGH_DATA; 5455 } 5456 5457 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5458 buffer->frameCount = part1; 5459 activeTrack->mRsmpInUnrel = part1; 5460 return NO_ERROR; 5461} 5462 5463// AudioBufferProvider interface 5464void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5465 AudioBufferProvider::Buffer* buffer) 5466{ 5467 RecordTrack *activeTrack = mRecordTrack; 5468 size_t stepCount = buffer->frameCount; 5469 if (stepCount == 0) { 5470 return; 5471 } 5472 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5473 activeTrack->mRsmpInUnrel -= stepCount; 5474 activeTrack->mRsmpInFront += stepCount; 5475 buffer->raw = NULL; 5476 buffer->frameCount = 0; 5477} 5478 5479bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5480 status_t& status) 5481{ 5482 bool reconfig = false; 5483 5484 status = NO_ERROR; 5485 5486 audio_format_t reqFormat = mFormat; 5487 uint32_t samplingRate = mSampleRate; 5488 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5489 5490 AudioParameter param = AudioParameter(keyValuePair); 5491 int value; 5492 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5493 // channel count change can be requested. Do we mandate the first client defines the 5494 // HAL sampling rate and channel count or do we allow changes on the fly? 5495 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5496 samplingRate = value; 5497 reconfig = true; 5498 } 5499 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5500 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5501 status = BAD_VALUE; 5502 } else { 5503 reqFormat = (audio_format_t) value; 5504 reconfig = true; 5505 } 5506 } 5507 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5508 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5509 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5510 status = BAD_VALUE; 5511 } else { 5512 channelMask = mask; 5513 reconfig = true; 5514 } 5515 } 5516 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5517 // do not accept frame count changes if tracks are open as the track buffer 5518 // size depends on frame count and correct behavior would not be guaranteed 5519 // if frame count is changed after track creation 5520 if (mActiveTracks.size() > 0) { 5521 status = INVALID_OPERATION; 5522 } else { 5523 reconfig = true; 5524 } 5525 } 5526 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5527 // forward device change to effects that have requested to be 5528 // aware of attached audio device. 5529 for (size_t i = 0; i < mEffectChains.size(); i++) { 5530 mEffectChains[i]->setDevice_l(value); 5531 } 5532 5533 // store input device and output device but do not forward output device to audio HAL. 5534 // Note that status is ignored by the caller for output device 5535 // (see AudioFlinger::setParameters() 5536 if (audio_is_output_devices(value)) { 5537 mOutDevice = value; 5538 status = BAD_VALUE; 5539 } else { 5540 mInDevice = value; 5541 // disable AEC and NS if the device is a BT SCO headset supporting those 5542 // pre processings 5543 if (mTracks.size() > 0) { 5544 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5545 mAudioFlinger->btNrecIsOff(); 5546 for (size_t i = 0; i < mTracks.size(); i++) { 5547 sp<RecordTrack> track = mTracks[i]; 5548 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5549 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5550 } 5551 } 5552 } 5553 } 5554 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5555 mAudioSource != (audio_source_t)value) { 5556 // forward device change to effects that have requested to be 5557 // aware of attached audio device. 5558 for (size_t i = 0; i < mEffectChains.size(); i++) { 5559 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5560 } 5561 mAudioSource = (audio_source_t)value; 5562 } 5563 5564 if (status == NO_ERROR) { 5565 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5566 keyValuePair.string()); 5567 if (status == INVALID_OPERATION) { 5568 inputStandBy(); 5569 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5570 keyValuePair.string()); 5571 } 5572 if (reconfig) { 5573 if (status == BAD_VALUE && 5574 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5575 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5576 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5577 <= (2 * samplingRate)) && 5578 audio_channel_count_from_in_mask( 5579 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5580 (channelMask == AUDIO_CHANNEL_IN_MONO || 5581 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5582 status = NO_ERROR; 5583 } 5584 if (status == NO_ERROR) { 5585 readInputParameters_l(); 5586 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5587 } 5588 } 5589 } 5590 5591 return reconfig; 5592} 5593 5594String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5595{ 5596 Mutex::Autolock _l(mLock); 5597 if (initCheck() != NO_ERROR) { 5598 return String8(); 5599 } 5600 5601 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5602 const String8 out_s8(s); 5603 free(s); 5604 return out_s8; 5605} 5606 5607void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5608 AudioSystem::OutputDescriptor desc; 5609 const void *param2 = NULL; 5610 5611 switch (event) { 5612 case AudioSystem::INPUT_OPENED: 5613 case AudioSystem::INPUT_CONFIG_CHANGED: 5614 desc.channelMask = mChannelMask; 5615 desc.samplingRate = mSampleRate; 5616 desc.format = mFormat; 5617 desc.frameCount = mFrameCount; 5618 desc.latency = 0; 5619 param2 = &desc; 5620 break; 5621 5622 case AudioSystem::INPUT_CLOSED: 5623 default: 5624 break; 5625 } 5626 mAudioFlinger->audioConfigChanged(event, mId, param2); 5627} 5628 5629void AudioFlinger::RecordThread::readInputParameters_l() 5630{ 5631 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5632 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5633 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5634 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5635 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5636 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5637 } 5638 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5639 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5640 mFrameCount = mBufferSize / mFrameSize; 5641 // This is the formula for calculating the temporary buffer size. 5642 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5643 // 1 full output buffer, regardless of the alignment of the available input. 5644 // The value is somewhat arbitrary, and could probably be even larger. 5645 // A larger value should allow more old data to be read after a track calls start(), 5646 // without increasing latency. 5647 mRsmpInFrames = mFrameCount * 7; 5648 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5649 delete[] mRsmpInBuffer; 5650 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5651 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5652 5653 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5654 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5655} 5656 5657uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5658{ 5659 Mutex::Autolock _l(mLock); 5660 if (initCheck() != NO_ERROR) { 5661 return 0; 5662 } 5663 5664 return mInput->stream->get_input_frames_lost(mInput->stream); 5665} 5666 5667uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5668{ 5669 Mutex::Autolock _l(mLock); 5670 uint32_t result = 0; 5671 if (getEffectChain_l(sessionId) != 0) { 5672 result = EFFECT_SESSION; 5673 } 5674 5675 for (size_t i = 0; i < mTracks.size(); ++i) { 5676 if (sessionId == mTracks[i]->sessionId()) { 5677 result |= TRACK_SESSION; 5678 break; 5679 } 5680 } 5681 5682 return result; 5683} 5684 5685KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5686{ 5687 KeyedVector<int, bool> ids; 5688 Mutex::Autolock _l(mLock); 5689 for (size_t j = 0; j < mTracks.size(); ++j) { 5690 sp<RecordThread::RecordTrack> track = mTracks[j]; 5691 int sessionId = track->sessionId(); 5692 if (ids.indexOfKey(sessionId) < 0) { 5693 ids.add(sessionId, true); 5694 } 5695 } 5696 return ids; 5697} 5698 5699AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5700{ 5701 Mutex::Autolock _l(mLock); 5702 AudioStreamIn *input = mInput; 5703 mInput = NULL; 5704 return input; 5705} 5706 5707// this method must always be called either with ThreadBase mLock held or inside the thread loop 5708audio_stream_t* AudioFlinger::RecordThread::stream() const 5709{ 5710 if (mInput == NULL) { 5711 return NULL; 5712 } 5713 return &mInput->stream->common; 5714} 5715 5716status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5717{ 5718 // only one chain per input thread 5719 if (mEffectChains.size() != 0) { 5720 return INVALID_OPERATION; 5721 } 5722 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5723 5724 chain->setInBuffer(NULL); 5725 chain->setOutBuffer(NULL); 5726 5727 checkSuspendOnAddEffectChain_l(chain); 5728 5729 mEffectChains.add(chain); 5730 5731 return NO_ERROR; 5732} 5733 5734size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5735{ 5736 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5737 ALOGW_IF(mEffectChains.size() != 1, 5738 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5739 chain.get(), mEffectChains.size(), this); 5740 if (mEffectChains.size() == 1) { 5741 mEffectChains.removeAt(0); 5742 } 5743 return 0; 5744} 5745 5746}; // namespace android 5747