Threads.cpp revision 755b0a611f539dfa49e88aac592a938427c7e1b8
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38 39// NBAIO implementations 40#include <media/nbaio/AudioStreamOutSink.h> 41#include <media/nbaio/MonoPipe.h> 42#include <media/nbaio/MonoPipeReader.h> 43#include <media/nbaio/Pipe.h> 44#include <media/nbaio/PipeReader.h> 45#include <media/nbaio/SourceAudioBufferProvider.h> 46 47#include <powermanager/PowerManager.h> 48 49#include <common_time/cc_helper.h> 50#include <common_time/local_clock.h> 51 52#include "AudioFlinger.h" 53#include "AudioMixer.h" 54#include "FastMixer.h" 55#include "ServiceUtilities.h" 56#include "SchedulingPolicyService.h" 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85// retry counts for buffer fill timeout 86// 50 * ~20msecs = 1 second 87static const int8_t kMaxTrackRetries = 50; 88static const int8_t kMaxTrackStartupRetries = 50; 89// allow less retry attempts on direct output thread. 90// direct outputs can be a scarce resource in audio hardware and should 91// be released as quickly as possible. 92static const int8_t kMaxTrackRetriesDirect = 2; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait in sendConfigEvent_l() for a status to be received 101static const nsecs_t kConfigEventTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108// minimum normal sink buffer size, expressed in milliseconds rather than frames 109static const uint32_t kMinNormalSinkBufferSizeMs = 20; 110// maximum normal sink buffer size 111static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 112 113// Offloaded output thread standby delay: allows track transition without going to standby 114static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 115 116// Whether to use fast mixer 117static const enum { 118 FastMixer_Never, // never initialize or use: for debugging only 119 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 120 // normal mixer multiplier is 1 121 FastMixer_Static, // initialize if needed, then use all the time if initialized, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 124 // multiplier is calculated based on min & max normal mixer buffer size 125 // FIXME for FastMixer_Dynamic: 126 // Supporting this option will require fixing HALs that can't handle large writes. 127 // For example, one HAL implementation returns an error from a large write, 128 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 129 // We could either fix the HAL implementations, or provide a wrapper that breaks 130 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 131} kUseFastMixer = FastMixer_Static; 132 133// Priorities for requestPriority 134static const int kPriorityAudioApp = 2; 135static const int kPriorityFastMixer = 3; 136 137// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 138// for the track. The client then sub-divides this into smaller buffers for its use. 139// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 140// So for now we just assume that client is double-buffered for fast tracks. 141// FIXME It would be better for client to tell AudioFlinger the value of N, 142// so AudioFlinger could allocate the right amount of memory. 143// See the client's minBufCount and mNotificationFramesAct calculations for details. 144static const int kFastTrackMultiplier = 2; 145 146// See Thread::readOnlyHeap(). 147// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 148// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 149// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 150static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 151 152// ---------------------------------------------------------------------------- 153 154#ifdef ADD_BATTERY_DATA 155// To collect the amplifier usage 156static void addBatteryData(uint32_t params) { 157 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 158 if (service == NULL) { 159 // it already logged 160 return; 161 } 162 163 service->addBatteryData(params); 164} 165#endif 166 167 168// ---------------------------------------------------------------------------- 169// CPU Stats 170// ---------------------------------------------------------------------------- 171 172class CpuStats { 173public: 174 CpuStats(); 175 void sample(const String8 &title); 176#ifdef DEBUG_CPU_USAGE 177private: 178 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 179 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 180 181 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 182 183 int mCpuNum; // thread's current CPU number 184 int mCpukHz; // frequency of thread's current CPU in kHz 185#endif 186}; 187 188CpuStats::CpuStats() 189#ifdef DEBUG_CPU_USAGE 190 : mCpuNum(-1), mCpukHz(-1) 191#endif 192{ 193} 194 195void CpuStats::sample(const String8 &title 196#ifndef DEBUG_CPU_USAGE 197 __unused 198#endif 199 ) { 200#ifdef DEBUG_CPU_USAGE 201 // get current thread's delta CPU time in wall clock ns 202 double wcNs; 203 bool valid = mCpuUsage.sampleAndEnable(wcNs); 204 205 // record sample for wall clock statistics 206 if (valid) { 207 mWcStats.sample(wcNs); 208 } 209 210 // get the current CPU number 211 int cpuNum = sched_getcpu(); 212 213 // get the current CPU frequency in kHz 214 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 215 216 // check if either CPU number or frequency changed 217 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 218 mCpuNum = cpuNum; 219 mCpukHz = cpukHz; 220 // ignore sample for purposes of cycles 221 valid = false; 222 } 223 224 // if no change in CPU number or frequency, then record sample for cycle statistics 225 if (valid && mCpukHz > 0) { 226 double cycles = wcNs * cpukHz * 0.000001; 227 mHzStats.sample(cycles); 228 } 229 230 unsigned n = mWcStats.n(); 231 // mCpuUsage.elapsed() is expensive, so don't call it every loop 232 if ((n & 127) == 1) { 233 long long elapsed = mCpuUsage.elapsed(); 234 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 235 double perLoop = elapsed / (double) n; 236 double perLoop100 = perLoop * 0.01; 237 double perLoop1k = perLoop * 0.001; 238 double mean = mWcStats.mean(); 239 double stddev = mWcStats.stddev(); 240 double minimum = mWcStats.minimum(); 241 double maximum = mWcStats.maximum(); 242 double meanCycles = mHzStats.mean(); 243 double stddevCycles = mHzStats.stddev(); 244 double minCycles = mHzStats.minimum(); 245 double maxCycles = mHzStats.maximum(); 246 mCpuUsage.resetElapsed(); 247 mWcStats.reset(); 248 mHzStats.reset(); 249 ALOGD("CPU usage for %s over past %.1f secs\n" 250 " (%u mixer loops at %.1f mean ms per loop):\n" 251 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 252 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 253 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 254 title.string(), 255 elapsed * .000000001, n, perLoop * .000001, 256 mean * .001, 257 stddev * .001, 258 minimum * .001, 259 maximum * .001, 260 mean / perLoop100, 261 stddev / perLoop100, 262 minimum / perLoop100, 263 maximum / perLoop100, 264 meanCycles / perLoop1k, 265 stddevCycles / perLoop1k, 266 minCycles / perLoop1k, 267 maxCycles / perLoop1k); 268 269 } 270 } 271#endif 272}; 273 274// ---------------------------------------------------------------------------- 275// ThreadBase 276// ---------------------------------------------------------------------------- 277 278AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 279 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 280 : Thread(false /*canCallJava*/), 281 mType(type), 282 mAudioFlinger(audioFlinger), 283 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 284 // are set by PlaybackThread::readOutputParameters_l() or 285 // RecordThread::readInputParameters_l() 286 //FIXME: mStandby should be true here. Is this some kind of hack? 287 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 288 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 289 // mName will be set by concrete (non-virtual) subclass 290 mDeathRecipient(new PMDeathRecipient(this)) 291{ 292} 293 294AudioFlinger::ThreadBase::~ThreadBase() 295{ 296 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 297 mConfigEvents.clear(); 298 299 // do not lock the mutex in destructor 300 releaseWakeLock_l(); 301 if (mPowerManager != 0) { 302 sp<IBinder> binder = mPowerManager->asBinder(); 303 binder->unlinkToDeath(mDeathRecipient); 304 } 305} 306 307status_t AudioFlinger::ThreadBase::readyToRun() 308{ 309 status_t status = initCheck(); 310 if (status == NO_ERROR) { 311 ALOGI("AudioFlinger's thread %p ready to run", this); 312 } else { 313 ALOGE("No working audio driver found."); 314 } 315 return status; 316} 317 318void AudioFlinger::ThreadBase::exit() 319{ 320 ALOGV("ThreadBase::exit"); 321 // do any cleanup required for exit to succeed 322 preExit(); 323 { 324 // This lock prevents the following race in thread (uniprocessor for illustration): 325 // if (!exitPending()) { 326 // // context switch from here to exit() 327 // // exit() calls requestExit(), what exitPending() observes 328 // // exit() calls signal(), which is dropped since no waiters 329 // // context switch back from exit() to here 330 // mWaitWorkCV.wait(...); 331 // // now thread is hung 332 // } 333 AutoMutex lock(mLock); 334 requestExit(); 335 mWaitWorkCV.broadcast(); 336 } 337 // When Thread::requestExitAndWait is made virtual and this method is renamed to 338 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 339 requestExitAndWait(); 340} 341 342status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 343{ 344 status_t status; 345 346 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 347 Mutex::Autolock _l(mLock); 348 349 return sendSetParameterConfigEvent_l(keyValuePairs); 350} 351 352// sendConfigEvent_l() must be called with ThreadBase::mLock held 353// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 354status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 355{ 356 status_t status = NO_ERROR; 357 358 mConfigEvents.add(event); 359 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 360 mWaitWorkCV.signal(); 361 mLock.unlock(); 362 { 363 Mutex::Autolock _l(event->mLock); 364 while (event->mWaitStatus) { 365 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 366 event->mStatus = TIMED_OUT; 367 event->mWaitStatus = false; 368 } 369 } 370 status = event->mStatus; 371 } 372 mLock.lock(); 373 return status; 374} 375 376void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 377{ 378 Mutex::Autolock _l(mLock); 379 sendIoConfigEvent_l(event, param); 380} 381 382// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 383void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 384{ 385 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 386 sendConfigEvent_l(configEvent); 387} 388 389// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 390void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 391{ 392 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 393 sendConfigEvent_l(configEvent); 394} 395 396// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 397status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 398{ 399 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 400 return sendConfigEvent_l(configEvent); 401} 402 403// post condition: mConfigEvents.isEmpty() 404void AudioFlinger::ThreadBase::processConfigEvents_l( 405 const DefaultKeyedVector< pid_t,sp<NotificationClient> >& notificationClients) 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_l(notificationClients, 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 1641// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1642void AudioFlinger::PlaybackThread::audioConfigChanged_l( 1643 const DefaultKeyedVector< pid_t,sp<NotificationClient> >& notificationClients, 1644 int event, 1645 int param) { 1646 AudioSystem::OutputDescriptor desc; 1647 void *param2 = NULL; 1648 1649 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1650 param); 1651 1652 switch (event) { 1653 case AudioSystem::OUTPUT_OPENED: 1654 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1655 desc.channelMask = mChannelMask; 1656 desc.samplingRate = mSampleRate; 1657 desc.format = mFormat; 1658 desc.frameCount = mNormalFrameCount; // FIXME see 1659 // AudioFlinger::frameCount(audio_io_handle_t) 1660 desc.latency = latency_l(); 1661 param2 = &desc; 1662 break; 1663 1664 case AudioSystem::STREAM_CONFIG_CHANGED: 1665 param2 = ¶m; 1666 case AudioSystem::OUTPUT_CLOSED: 1667 default: 1668 break; 1669 } 1670 mAudioFlinger->audioConfigChanged_l(notificationClients, event, mId, param2); 1671} 1672 1673void AudioFlinger::PlaybackThread::writeCallback() 1674{ 1675 ALOG_ASSERT(mCallbackThread != 0); 1676 mCallbackThread->resetWriteBlocked(); 1677} 1678 1679void AudioFlinger::PlaybackThread::drainCallback() 1680{ 1681 ALOG_ASSERT(mCallbackThread != 0); 1682 mCallbackThread->resetDraining(); 1683} 1684 1685void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1686{ 1687 Mutex::Autolock _l(mLock); 1688 // reject out of sequence requests 1689 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1690 mWriteAckSequence &= ~1; 1691 mWaitWorkCV.signal(); 1692 } 1693} 1694 1695void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 // reject out of sequence requests 1699 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1700 mDrainSequence &= ~1; 1701 mWaitWorkCV.signal(); 1702 } 1703} 1704 1705// static 1706int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1707 void *param __unused, 1708 void *cookie) 1709{ 1710 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1711 ALOGV("asyncCallback() event %d", event); 1712 switch (event) { 1713 case STREAM_CBK_EVENT_WRITE_READY: 1714 me->writeCallback(); 1715 break; 1716 case STREAM_CBK_EVENT_DRAIN_READY: 1717 me->drainCallback(); 1718 break; 1719 default: 1720 ALOGW("asyncCallback() unknown event %d", event); 1721 break; 1722 } 1723 return 0; 1724} 1725 1726void AudioFlinger::PlaybackThread::readOutputParameters_l() 1727{ 1728 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1729 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1730 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1731 if (!audio_is_output_channel(mChannelMask)) { 1732 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1733 } 1734 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1735 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1736 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1737 } 1738 mChannelCount = popcount(mChannelMask); 1739 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1740 if (!audio_is_valid_format(mFormat)) { 1741 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1742 } 1743 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1744 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1745 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1746 } 1747 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1748 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1749 mFrameCount = mBufferSize / mFrameSize; 1750 if (mFrameCount & 15) { 1751 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1752 mFrameCount); 1753 } 1754 1755 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1756 (mOutput->stream->set_callback != NULL)) { 1757 if (mOutput->stream->set_callback(mOutput->stream, 1758 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1759 mUseAsyncWrite = true; 1760 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1761 } 1762 } 1763 1764 // Calculate size of normal sink buffer relative to the HAL output buffer size 1765 double multiplier = 1.0; 1766 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1767 kUseFastMixer == FastMixer_Dynamic)) { 1768 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1769 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1770 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1771 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1772 maxNormalFrameCount = maxNormalFrameCount & ~15; 1773 if (maxNormalFrameCount < minNormalFrameCount) { 1774 maxNormalFrameCount = minNormalFrameCount; 1775 } 1776 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1777 if (multiplier <= 1.0) { 1778 multiplier = 1.0; 1779 } else if (multiplier <= 2.0) { 1780 if (2 * mFrameCount <= maxNormalFrameCount) { 1781 multiplier = 2.0; 1782 } else { 1783 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1784 } 1785 } else { 1786 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1787 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1788 // track, but we sometimes have to do this to satisfy the maximum frame count 1789 // constraint) 1790 // FIXME this rounding up should not be done if no HAL SRC 1791 uint32_t truncMult = (uint32_t) multiplier; 1792 if ((truncMult & 1)) { 1793 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1794 ++truncMult; 1795 } 1796 } 1797 multiplier = (double) truncMult; 1798 } 1799 } 1800 mNormalFrameCount = multiplier * mFrameCount; 1801 // round up to nearest 16 frames to satisfy AudioMixer 1802 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1803 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1804 mNormalFrameCount); 1805 1806 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1807 // Originally this was int16_t[] array, need to remove legacy implications. 1808 free(mSinkBuffer); 1809 mSinkBuffer = NULL; 1810 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1811 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1812 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1813 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1814 1815 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1816 // drives the output. 1817 free(mMixerBuffer); 1818 mMixerBuffer = NULL; 1819 if (mMixerBufferEnabled) { 1820 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1821 mMixerBufferSize = mNormalFrameCount * mChannelCount 1822 * audio_bytes_per_sample(mMixerBufferFormat); 1823 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1824 } 1825 free(mEffectBuffer); 1826 mEffectBuffer = NULL; 1827 if (mEffectBufferEnabled) { 1828 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1829 mEffectBufferSize = mNormalFrameCount * mChannelCount 1830 * audio_bytes_per_sample(mEffectBufferFormat); 1831 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1832 } 1833 1834 // force reconfiguration of effect chains and engines to take new buffer size and audio 1835 // parameters into account 1836 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1837 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1838 // matter. 1839 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1840 Vector< sp<EffectChain> > effectChains = mEffectChains; 1841 for (size_t i = 0; i < effectChains.size(); i ++) { 1842 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1843 } 1844} 1845 1846 1847status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1848{ 1849 if (halFrames == NULL || dspFrames == NULL) { 1850 return BAD_VALUE; 1851 } 1852 Mutex::Autolock _l(mLock); 1853 if (initCheck() != NO_ERROR) { 1854 return INVALID_OPERATION; 1855 } 1856 size_t framesWritten = mBytesWritten / mFrameSize; 1857 *halFrames = framesWritten; 1858 1859 if (isSuspended()) { 1860 // return an estimation of rendered frames when the output is suspended 1861 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1862 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1863 return NO_ERROR; 1864 } else { 1865 status_t status; 1866 uint32_t frames; 1867 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1868 *dspFrames = (size_t)frames; 1869 return status; 1870 } 1871} 1872 1873uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1874{ 1875 Mutex::Autolock _l(mLock); 1876 uint32_t result = 0; 1877 if (getEffectChain_l(sessionId) != 0) { 1878 result = EFFECT_SESSION; 1879 } 1880 1881 for (size_t i = 0; i < mTracks.size(); ++i) { 1882 sp<Track> track = mTracks[i]; 1883 if (sessionId == track->sessionId() && !track->isInvalid()) { 1884 result |= TRACK_SESSION; 1885 break; 1886 } 1887 } 1888 1889 return result; 1890} 1891 1892uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1893{ 1894 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1895 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1896 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1897 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1898 } 1899 for (size_t i = 0; i < mTracks.size(); i++) { 1900 sp<Track> track = mTracks[i]; 1901 if (sessionId == track->sessionId() && !track->isInvalid()) { 1902 return AudioSystem::getStrategyForStream(track->streamType()); 1903 } 1904 } 1905 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1906} 1907 1908 1909AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1910{ 1911 Mutex::Autolock _l(mLock); 1912 return mOutput; 1913} 1914 1915AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1916{ 1917 Mutex::Autolock _l(mLock); 1918 AudioStreamOut *output = mOutput; 1919 mOutput = NULL; 1920 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1921 // must push a NULL and wait for ack 1922 mOutputSink.clear(); 1923 mPipeSink.clear(); 1924 mNormalSink.clear(); 1925 return output; 1926} 1927 1928// this method must always be called either with ThreadBase mLock held or inside the thread loop 1929audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1930{ 1931 if (mOutput == NULL) { 1932 return NULL; 1933 } 1934 return &mOutput->stream->common; 1935} 1936 1937uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1938{ 1939 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1940} 1941 1942status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1943{ 1944 if (!isValidSyncEvent(event)) { 1945 return BAD_VALUE; 1946 } 1947 1948 Mutex::Autolock _l(mLock); 1949 1950 for (size_t i = 0; i < mTracks.size(); ++i) { 1951 sp<Track> track = mTracks[i]; 1952 if (event->triggerSession() == track->sessionId()) { 1953 (void) track->setSyncEvent(event); 1954 return NO_ERROR; 1955 } 1956 } 1957 1958 return NAME_NOT_FOUND; 1959} 1960 1961bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1962{ 1963 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1964} 1965 1966void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1967 const Vector< sp<Track> >& tracksToRemove) 1968{ 1969 size_t count = tracksToRemove.size(); 1970 if (count > 0) { 1971 for (size_t i = 0 ; i < count ; i++) { 1972 const sp<Track>& track = tracksToRemove.itemAt(i); 1973 if (!track->isOutputTrack()) { 1974 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1975#ifdef ADD_BATTERY_DATA 1976 // to track the speaker usage 1977 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1978#endif 1979 if (track->isTerminated()) { 1980 AudioSystem::releaseOutput(mId); 1981 } 1982 } 1983 } 1984 } 1985} 1986 1987void AudioFlinger::PlaybackThread::checkSilentMode_l() 1988{ 1989 if (!mMasterMute) { 1990 char value[PROPERTY_VALUE_MAX]; 1991 if (property_get("ro.audio.silent", value, "0") > 0) { 1992 char *endptr; 1993 unsigned long ul = strtoul(value, &endptr, 0); 1994 if (*endptr == '\0' && ul != 0) { 1995 ALOGD("Silence is golden"); 1996 // The setprop command will not allow a property to be changed after 1997 // the first time it is set, so we don't have to worry about un-muting. 1998 setMasterMute_l(true); 1999 } 2000 } 2001 } 2002} 2003 2004// shared by MIXER and DIRECT, overridden by DUPLICATING 2005ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2006{ 2007 // FIXME rewrite to reduce number of system calls 2008 mLastWriteTime = systemTime(); 2009 mInWrite = true; 2010 ssize_t bytesWritten; 2011 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2012 2013 // If an NBAIO sink is present, use it to write the normal mixer's submix 2014 if (mNormalSink != 0) { 2015 const size_t count = mBytesRemaining / mFrameSize; 2016 2017 ATRACE_BEGIN("write"); 2018 // update the setpoint when AudioFlinger::mScreenState changes 2019 uint32_t screenState = AudioFlinger::mScreenState; 2020 if (screenState != mScreenState) { 2021 mScreenState = screenState; 2022 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2023 if (pipe != NULL) { 2024 pipe->setAvgFrames((mScreenState & 1) ? 2025 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2026 } 2027 } 2028 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2029 ATRACE_END(); 2030 if (framesWritten > 0) { 2031 bytesWritten = framesWritten * mFrameSize; 2032 } else { 2033 bytesWritten = framesWritten; 2034 } 2035 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2036 if (status == NO_ERROR) { 2037 size_t totalFramesWritten = mNormalSink->framesWritten(); 2038 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2039 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2040 mLatchDValid = true; 2041 } 2042 } 2043 // otherwise use the HAL / AudioStreamOut directly 2044 } else { 2045 // Direct output and offload threads 2046 2047 if (mUseAsyncWrite) { 2048 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2049 mWriteAckSequence += 2; 2050 mWriteAckSequence |= 1; 2051 ALOG_ASSERT(mCallbackThread != 0); 2052 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2053 } 2054 // FIXME We should have an implementation of timestamps for direct output threads. 2055 // They are used e.g for multichannel PCM playback over HDMI. 2056 bytesWritten = mOutput->stream->write(mOutput->stream, 2057 (char *)mSinkBuffer + offset, mBytesRemaining); 2058 if (mUseAsyncWrite && 2059 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2060 // do not wait for async callback in case of error of full write 2061 mWriteAckSequence &= ~1; 2062 ALOG_ASSERT(mCallbackThread != 0); 2063 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2064 } 2065 } 2066 2067 mNumWrites++; 2068 mInWrite = false; 2069 mStandby = false; 2070 return bytesWritten; 2071} 2072 2073void AudioFlinger::PlaybackThread::threadLoop_drain() 2074{ 2075 if (mOutput->stream->drain) { 2076 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2077 if (mUseAsyncWrite) { 2078 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2079 mDrainSequence |= 1; 2080 ALOG_ASSERT(mCallbackThread != 0); 2081 mCallbackThread->setDraining(mDrainSequence); 2082 } 2083 mOutput->stream->drain(mOutput->stream, 2084 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2085 : AUDIO_DRAIN_ALL); 2086 } 2087} 2088 2089void AudioFlinger::PlaybackThread::threadLoop_exit() 2090{ 2091 // Default implementation has nothing to do 2092} 2093 2094/* 2095The derived values that are cached: 2096 - mSinkBufferSize from frame count * frame size 2097 - activeSleepTime from activeSleepTimeUs() 2098 - idleSleepTime from idleSleepTimeUs() 2099 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2100 - maxPeriod from frame count and sample rate (MIXER only) 2101 2102The parameters that affect these derived values are: 2103 - frame count 2104 - frame size 2105 - sample rate 2106 - device type: A2DP or not 2107 - device latency 2108 - format: PCM or not 2109 - active sleep time 2110 - idle sleep time 2111*/ 2112 2113void AudioFlinger::PlaybackThread::cacheParameters_l() 2114{ 2115 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2116 activeSleepTime = activeSleepTimeUs(); 2117 idleSleepTime = idleSleepTimeUs(); 2118} 2119 2120void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2121{ 2122 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2123 this, streamType, mTracks.size()); 2124 Mutex::Autolock _l(mLock); 2125 2126 size_t size = mTracks.size(); 2127 for (size_t i = 0; i < size; i++) { 2128 sp<Track> t = mTracks[i]; 2129 if (t->streamType() == streamType) { 2130 t->invalidate(); 2131 } 2132 } 2133} 2134 2135status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2136{ 2137 int session = chain->sessionId(); 2138 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2139 ? mEffectBuffer : mSinkBuffer); 2140 bool ownsBuffer = false; 2141 2142 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2143 if (session > 0) { 2144 // Only one effect chain can be present in direct output thread and it uses 2145 // the sink buffer as input 2146 if (mType != DIRECT) { 2147 size_t numSamples = mNormalFrameCount * mChannelCount; 2148 buffer = new int16_t[numSamples]; 2149 memset(buffer, 0, numSamples * sizeof(int16_t)); 2150 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2151 ownsBuffer = true; 2152 } 2153 2154 // Attach all tracks with same session ID to this chain. 2155 for (size_t i = 0; i < mTracks.size(); ++i) { 2156 sp<Track> track = mTracks[i]; 2157 if (session == track->sessionId()) { 2158 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2159 buffer); 2160 track->setMainBuffer(buffer); 2161 chain->incTrackCnt(); 2162 } 2163 } 2164 2165 // indicate all active tracks in the chain 2166 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2167 sp<Track> track = mActiveTracks[i].promote(); 2168 if (track == 0) { 2169 continue; 2170 } 2171 if (session == track->sessionId()) { 2172 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2173 chain->incActiveTrackCnt(); 2174 } 2175 } 2176 } 2177 2178 chain->setInBuffer(buffer, ownsBuffer); 2179 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2180 ? mEffectBuffer : mSinkBuffer)); 2181 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2182 // chains list in order to be processed last as it contains output stage effects 2183 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2184 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2185 // after track specific effects and before output stage 2186 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2187 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2188 // Effect chain for other sessions are inserted at beginning of effect 2189 // chains list to be processed before output mix effects. Relative order between other 2190 // sessions is not important 2191 size_t size = mEffectChains.size(); 2192 size_t i = 0; 2193 for (i = 0; i < size; i++) { 2194 if (mEffectChains[i]->sessionId() < session) { 2195 break; 2196 } 2197 } 2198 mEffectChains.insertAt(chain, i); 2199 checkSuspendOnAddEffectChain_l(chain); 2200 2201 return NO_ERROR; 2202} 2203 2204size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2205{ 2206 int session = chain->sessionId(); 2207 2208 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2209 2210 for (size_t i = 0; i < mEffectChains.size(); i++) { 2211 if (chain == mEffectChains[i]) { 2212 mEffectChains.removeAt(i); 2213 // detach all active tracks from the chain 2214 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2215 sp<Track> track = mActiveTracks[i].promote(); 2216 if (track == 0) { 2217 continue; 2218 } 2219 if (session == track->sessionId()) { 2220 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2221 chain.get(), session); 2222 chain->decActiveTrackCnt(); 2223 } 2224 } 2225 2226 // detach all tracks with same session ID from this chain 2227 for (size_t i = 0; i < mTracks.size(); ++i) { 2228 sp<Track> track = mTracks[i]; 2229 if (session == track->sessionId()) { 2230 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2231 chain->decTrackCnt(); 2232 } 2233 } 2234 break; 2235 } 2236 } 2237 return mEffectChains.size(); 2238} 2239 2240status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2241 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2242{ 2243 Mutex::Autolock _l(mLock); 2244 return attachAuxEffect_l(track, EffectId); 2245} 2246 2247status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2248 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2249{ 2250 status_t status = NO_ERROR; 2251 2252 if (EffectId == 0) { 2253 track->setAuxBuffer(0, NULL); 2254 } else { 2255 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2256 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2257 if (effect != 0) { 2258 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2259 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2260 } else { 2261 status = INVALID_OPERATION; 2262 } 2263 } else { 2264 status = BAD_VALUE; 2265 } 2266 } 2267 return status; 2268} 2269 2270void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2271{ 2272 for (size_t i = 0; i < mTracks.size(); ++i) { 2273 sp<Track> track = mTracks[i]; 2274 if (track->auxEffectId() == effectId) { 2275 attachAuxEffect_l(track, 0); 2276 } 2277 } 2278} 2279 2280bool AudioFlinger::PlaybackThread::threadLoop() 2281{ 2282 Vector< sp<Track> > tracksToRemove; 2283 2284 standbyTime = systemTime(); 2285 2286 // MIXER 2287 nsecs_t lastWarning = 0; 2288 2289 // DUPLICATING 2290 // FIXME could this be made local to while loop? 2291 writeFrames = 0; 2292 2293 int lastGeneration = 0; 2294 2295 cacheParameters_l(); 2296 sleepTime = idleSleepTime; 2297 2298 if (mType == MIXER) { 2299 sleepTimeShift = 0; 2300 } 2301 2302 CpuStats cpuStats; 2303 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2304 2305 acquireWakeLock(); 2306 2307 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2308 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2309 // and then that string will be logged at the next convenient opportunity. 2310 const char *logString = NULL; 2311 2312 checkSilentMode_l(); 2313 2314 while (!exitPending()) 2315 { 2316 cpuStats.sample(myName); 2317 2318 Vector< sp<EffectChain> > effectChains; 2319 2320 DefaultKeyedVector< pid_t,sp<NotificationClient> > notificationClients = 2321 mAudioFlinger->notificationClients(); 2322 2323 { // scope for mLock 2324 2325 Mutex::Autolock _l(mLock); 2326 2327 processConfigEvents_l(notificationClients); 2328 notificationClients.clear(); 2329 2330 if (logString != NULL) { 2331 mNBLogWriter->logTimestamp(); 2332 mNBLogWriter->log(logString); 2333 logString = NULL; 2334 } 2335 2336 if (mLatchDValid) { 2337 mLatchQ = mLatchD; 2338 mLatchDValid = false; 2339 mLatchQValid = true; 2340 } 2341 2342 saveOutputTracks(); 2343 if (mSignalPending) { 2344 // A signal was raised while we were unlocked 2345 mSignalPending = false; 2346 } else if (waitingAsyncCallback_l()) { 2347 if (exitPending()) { 2348 break; 2349 } 2350 releaseWakeLock_l(); 2351 mWakeLockUids.clear(); 2352 mActiveTracksGeneration++; 2353 ALOGV("wait async completion"); 2354 mWaitWorkCV.wait(mLock); 2355 ALOGV("async completion/wake"); 2356 acquireWakeLock_l(); 2357 standbyTime = systemTime() + standbyDelay; 2358 sleepTime = 0; 2359 2360 continue; 2361 } 2362 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2363 isSuspended()) { 2364 // put audio hardware into standby after short delay 2365 if (shouldStandby_l()) { 2366 2367 threadLoop_standby(); 2368 2369 mStandby = true; 2370 } 2371 2372 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2373 // we're about to wait, flush the binder command buffer 2374 IPCThreadState::self()->flushCommands(); 2375 2376 clearOutputTracks(); 2377 2378 if (exitPending()) { 2379 break; 2380 } 2381 2382 releaseWakeLock_l(); 2383 mWakeLockUids.clear(); 2384 mActiveTracksGeneration++; 2385 // wait until we have something to do... 2386 ALOGV("%s going to sleep", myName.string()); 2387 mWaitWorkCV.wait(mLock); 2388 ALOGV("%s waking up", myName.string()); 2389 acquireWakeLock_l(); 2390 2391 mMixerStatus = MIXER_IDLE; 2392 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2393 mBytesWritten = 0; 2394 mBytesRemaining = 0; 2395 checkSilentMode_l(); 2396 2397 standbyTime = systemTime() + standbyDelay; 2398 sleepTime = idleSleepTime; 2399 if (mType == MIXER) { 2400 sleepTimeShift = 0; 2401 } 2402 2403 continue; 2404 } 2405 } 2406 // mMixerStatusIgnoringFastTracks is also updated internally 2407 mMixerStatus = prepareTracks_l(&tracksToRemove); 2408 2409 // compare with previously applied list 2410 if (lastGeneration != mActiveTracksGeneration) { 2411 // update wakelock 2412 updateWakeLockUids_l(mWakeLockUids); 2413 lastGeneration = mActiveTracksGeneration; 2414 } 2415 2416 // prevent any changes in effect chain list and in each effect chain 2417 // during mixing and effect process as the audio buffers could be deleted 2418 // or modified if an effect is created or deleted 2419 lockEffectChains_l(effectChains); 2420 } // mLock scope ends 2421 2422 if (mBytesRemaining == 0) { 2423 mCurrentWriteLength = 0; 2424 if (mMixerStatus == MIXER_TRACKS_READY) { 2425 // threadLoop_mix() sets mCurrentWriteLength 2426 threadLoop_mix(); 2427 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2428 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2429 // threadLoop_sleepTime sets sleepTime to 0 if data 2430 // must be written to HAL 2431 threadLoop_sleepTime(); 2432 if (sleepTime == 0) { 2433 mCurrentWriteLength = mSinkBufferSize; 2434 } 2435 } 2436 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2437 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2438 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2439 // or mSinkBuffer (if there are no effects). 2440 // 2441 // This is done pre-effects computation; if effects change to 2442 // support higher precision, this needs to move. 2443 // 2444 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2445 // TODO use sleepTime == 0 as an additional condition. 2446 if (mMixerBufferValid) { 2447 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2448 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2449 2450 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2451 mNormalFrameCount * mChannelCount); 2452 } 2453 2454 mBytesRemaining = mCurrentWriteLength; 2455 if (isSuspended()) { 2456 sleepTime = suspendSleepTimeUs(); 2457 // simulate write to HAL when suspended 2458 mBytesWritten += mSinkBufferSize; 2459 mBytesRemaining = 0; 2460 } 2461 2462 // only process effects if we're going to write 2463 if (sleepTime == 0 && mType != OFFLOAD) { 2464 for (size_t i = 0; i < effectChains.size(); i ++) { 2465 effectChains[i]->process_l(); 2466 } 2467 } 2468 } 2469 // Process effect chains for offloaded thread even if no audio 2470 // was read from audio track: process only updates effect state 2471 // and thus does have to be synchronized with audio writes but may have 2472 // to be called while waiting for async write callback 2473 if (mType == OFFLOAD) { 2474 for (size_t i = 0; i < effectChains.size(); i ++) { 2475 effectChains[i]->process_l(); 2476 } 2477 } 2478 2479 // Only if the Effects buffer is enabled and there is data in the 2480 // Effects buffer (buffer valid), we need to 2481 // copy into the sink buffer. 2482 // TODO use sleepTime == 0 as an additional condition. 2483 if (mEffectBufferValid) { 2484 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2485 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2486 mNormalFrameCount * mChannelCount); 2487 } 2488 2489 // enable changes in effect chain 2490 unlockEffectChains(effectChains); 2491 2492 if (!waitingAsyncCallback()) { 2493 // sleepTime == 0 means we must write to audio hardware 2494 if (sleepTime == 0) { 2495 if (mBytesRemaining) { 2496 ssize_t ret = threadLoop_write(); 2497 if (ret < 0) { 2498 mBytesRemaining = 0; 2499 } else { 2500 mBytesWritten += ret; 2501 mBytesRemaining -= ret; 2502 } 2503 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2504 (mMixerStatus == MIXER_DRAIN_ALL)) { 2505 threadLoop_drain(); 2506 } 2507 if (mType == MIXER) { 2508 // write blocked detection 2509 nsecs_t now = systemTime(); 2510 nsecs_t delta = now - mLastWriteTime; 2511 if (!mStandby && delta > maxPeriod) { 2512 mNumDelayedWrites++; 2513 if ((now - lastWarning) > kWarningThrottleNs) { 2514 ATRACE_NAME("underrun"); 2515 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2516 ns2ms(delta), mNumDelayedWrites, this); 2517 lastWarning = now; 2518 } 2519 } 2520 } 2521 2522 } else { 2523 usleep(sleepTime); 2524 } 2525 } 2526 2527 // Finally let go of removed track(s), without the lock held 2528 // since we can't guarantee the destructors won't acquire that 2529 // same lock. This will also mutate and push a new fast mixer state. 2530 threadLoop_removeTracks(tracksToRemove); 2531 tracksToRemove.clear(); 2532 2533 // FIXME I don't understand the need for this here; 2534 // it was in the original code but maybe the 2535 // assignment in saveOutputTracks() makes this unnecessary? 2536 clearOutputTracks(); 2537 2538 // Effect chains will be actually deleted here if they were removed from 2539 // mEffectChains list during mixing or effects processing 2540 effectChains.clear(); 2541 2542 // FIXME Note that the above .clear() is no longer necessary since effectChains 2543 // is now local to this block, but will keep it for now (at least until merge done). 2544 } 2545 2546 threadLoop_exit(); 2547 2548 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2549 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2550 // put output stream into standby mode 2551 if (!mStandby) { 2552 mOutput->stream->common.standby(&mOutput->stream->common); 2553 } 2554 } 2555 2556 releaseWakeLock(); 2557 mWakeLockUids.clear(); 2558 mActiveTracksGeneration++; 2559 2560 ALOGV("Thread %p type %d exiting", this, mType); 2561 return false; 2562} 2563 2564// removeTracks_l() must be called with ThreadBase::mLock held 2565void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2566{ 2567 size_t count = tracksToRemove.size(); 2568 if (count > 0) { 2569 for (size_t i=0 ; i<count ; i++) { 2570 const sp<Track>& track = tracksToRemove.itemAt(i); 2571 mActiveTracks.remove(track); 2572 mWakeLockUids.remove(track->uid()); 2573 mActiveTracksGeneration++; 2574 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2575 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2576 if (chain != 0) { 2577 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2578 track->sessionId()); 2579 chain->decActiveTrackCnt(); 2580 } 2581 if (track->isTerminated()) { 2582 removeTrack_l(track); 2583 } 2584 } 2585 } 2586 2587} 2588 2589status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2590{ 2591 if (mNormalSink != 0) { 2592 return mNormalSink->getTimestamp(timestamp); 2593 } 2594 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2595 uint64_t position64; 2596 int ret = mOutput->stream->get_presentation_position( 2597 mOutput->stream, &position64, ×tamp.mTime); 2598 if (ret == 0) { 2599 timestamp.mPosition = (uint32_t)position64; 2600 return NO_ERROR; 2601 } 2602 } 2603 return INVALID_OPERATION; 2604} 2605// ---------------------------------------------------------------------------- 2606 2607AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2608 audio_io_handle_t id, audio_devices_t device, type_t type) 2609 : PlaybackThread(audioFlinger, output, id, device, type), 2610 // mAudioMixer below 2611 // mFastMixer below 2612 mFastMixerFutex(0) 2613 // mOutputSink below 2614 // mPipeSink below 2615 // mNormalSink below 2616{ 2617 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2618 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2619 "mFrameCount=%d, mNormalFrameCount=%d", 2620 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2621 mNormalFrameCount); 2622 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2623 2624 // FIXME - Current mixer implementation only supports stereo output 2625 if (mChannelCount != FCC_2) { 2626 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2627 } 2628 2629 // create an NBAIO sink for the HAL output stream, and negotiate 2630 mOutputSink = new AudioStreamOutSink(output->stream); 2631 size_t numCounterOffers = 0; 2632 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2633 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2634 ALOG_ASSERT(index == 0); 2635 2636 // initialize fast mixer depending on configuration 2637 bool initFastMixer; 2638 switch (kUseFastMixer) { 2639 case FastMixer_Never: 2640 initFastMixer = false; 2641 break; 2642 case FastMixer_Always: 2643 initFastMixer = true; 2644 break; 2645 case FastMixer_Static: 2646 case FastMixer_Dynamic: 2647 initFastMixer = mFrameCount < mNormalFrameCount; 2648 break; 2649 } 2650 if (initFastMixer) { 2651 2652 // create a MonoPipe to connect our submix to FastMixer 2653 NBAIO_Format format = mOutputSink->format(); 2654 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2655 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2656 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2657 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2658 const NBAIO_Format offers[1] = {format}; 2659 size_t numCounterOffers = 0; 2660 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2661 ALOG_ASSERT(index == 0); 2662 monoPipe->setAvgFrames((mScreenState & 1) ? 2663 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2664 mPipeSink = monoPipe; 2665 2666#ifdef TEE_SINK 2667 if (mTeeSinkOutputEnabled) { 2668 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2669 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2670 numCounterOffers = 0; 2671 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2672 ALOG_ASSERT(index == 0); 2673 mTeeSink = teeSink; 2674 PipeReader *teeSource = new PipeReader(*teeSink); 2675 numCounterOffers = 0; 2676 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2677 ALOG_ASSERT(index == 0); 2678 mTeeSource = teeSource; 2679 } 2680#endif 2681 2682 // create fast mixer and configure it initially with just one fast track for our submix 2683 mFastMixer = new FastMixer(); 2684 FastMixerStateQueue *sq = mFastMixer->sq(); 2685#ifdef STATE_QUEUE_DUMP 2686 sq->setObserverDump(&mStateQueueObserverDump); 2687 sq->setMutatorDump(&mStateQueueMutatorDump); 2688#endif 2689 FastMixerState *state = sq->begin(); 2690 FastTrack *fastTrack = &state->mFastTracks[0]; 2691 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2692 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2693 fastTrack->mVolumeProvider = NULL; 2694 fastTrack->mGeneration++; 2695 state->mFastTracksGen++; 2696 state->mTrackMask = 1; 2697 // fast mixer will use the HAL output sink 2698 state->mOutputSink = mOutputSink.get(); 2699 state->mOutputSinkGen++; 2700 state->mFrameCount = mFrameCount; 2701 state->mCommand = FastMixerState::COLD_IDLE; 2702 // already done in constructor initialization list 2703 //mFastMixerFutex = 0; 2704 state->mColdFutexAddr = &mFastMixerFutex; 2705 state->mColdGen++; 2706 state->mDumpState = &mFastMixerDumpState; 2707#ifdef TEE_SINK 2708 state->mTeeSink = mTeeSink.get(); 2709#endif 2710 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2711 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2712 sq->end(); 2713 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2714 2715 // start the fast mixer 2716 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2717 pid_t tid = mFastMixer->getTid(); 2718 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2719 if (err != 0) { 2720 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2721 kPriorityFastMixer, getpid_cached, tid, err); 2722 } 2723 2724#ifdef AUDIO_WATCHDOG 2725 // create and start the watchdog 2726 mAudioWatchdog = new AudioWatchdog(); 2727 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2728 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2729 tid = mAudioWatchdog->getTid(); 2730 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2731 if (err != 0) { 2732 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2733 kPriorityFastMixer, getpid_cached, tid, err); 2734 } 2735#endif 2736 2737 } else { 2738 mFastMixer = NULL; 2739 } 2740 2741 switch (kUseFastMixer) { 2742 case FastMixer_Never: 2743 case FastMixer_Dynamic: 2744 mNormalSink = mOutputSink; 2745 break; 2746 case FastMixer_Always: 2747 mNormalSink = mPipeSink; 2748 break; 2749 case FastMixer_Static: 2750 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2751 break; 2752 } 2753} 2754 2755AudioFlinger::MixerThread::~MixerThread() 2756{ 2757 if (mFastMixer != NULL) { 2758 FastMixerStateQueue *sq = mFastMixer->sq(); 2759 FastMixerState *state = sq->begin(); 2760 if (state->mCommand == FastMixerState::COLD_IDLE) { 2761 int32_t old = android_atomic_inc(&mFastMixerFutex); 2762 if (old == -1) { 2763 (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2764 } 2765 } 2766 state->mCommand = FastMixerState::EXIT; 2767 sq->end(); 2768 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2769 mFastMixer->join(); 2770 // Though the fast mixer thread has exited, it's state queue is still valid. 2771 // We'll use that extract the final state which contains one remaining fast track 2772 // corresponding to our sub-mix. 2773 state = sq->begin(); 2774 ALOG_ASSERT(state->mTrackMask == 1); 2775 FastTrack *fastTrack = &state->mFastTracks[0]; 2776 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2777 delete fastTrack->mBufferProvider; 2778 sq->end(false /*didModify*/); 2779 delete mFastMixer; 2780#ifdef AUDIO_WATCHDOG 2781 if (mAudioWatchdog != 0) { 2782 mAudioWatchdog->requestExit(); 2783 mAudioWatchdog->requestExitAndWait(); 2784 mAudioWatchdog.clear(); 2785 } 2786#endif 2787 } 2788 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2789 delete mAudioMixer; 2790} 2791 2792 2793uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2794{ 2795 if (mFastMixer != NULL) { 2796 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2797 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2798 } 2799 return latency; 2800} 2801 2802 2803void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2804{ 2805 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2806} 2807 2808ssize_t AudioFlinger::MixerThread::threadLoop_write() 2809{ 2810 // FIXME we should only do one push per cycle; confirm this is true 2811 // Start the fast mixer if it's not already running 2812 if (mFastMixer != NULL) { 2813 FastMixerStateQueue *sq = mFastMixer->sq(); 2814 FastMixerState *state = sq->begin(); 2815 if (state->mCommand != FastMixerState::MIX_WRITE && 2816 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2817 if (state->mCommand == FastMixerState::COLD_IDLE) { 2818 int32_t old = android_atomic_inc(&mFastMixerFutex); 2819 if (old == -1) { 2820 (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2821 } 2822#ifdef AUDIO_WATCHDOG 2823 if (mAudioWatchdog != 0) { 2824 mAudioWatchdog->resume(); 2825 } 2826#endif 2827 } 2828 state->mCommand = FastMixerState::MIX_WRITE; 2829 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2830 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2831 sq->end(); 2832 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2833 if (kUseFastMixer == FastMixer_Dynamic) { 2834 mNormalSink = mPipeSink; 2835 } 2836 } else { 2837 sq->end(false /*didModify*/); 2838 } 2839 } 2840 return PlaybackThread::threadLoop_write(); 2841} 2842 2843void AudioFlinger::MixerThread::threadLoop_standby() 2844{ 2845 // Idle the fast mixer if it's currently running 2846 if (mFastMixer != NULL) { 2847 FastMixerStateQueue *sq = mFastMixer->sq(); 2848 FastMixerState *state = sq->begin(); 2849 if (!(state->mCommand & FastMixerState::IDLE)) { 2850 state->mCommand = FastMixerState::COLD_IDLE; 2851 state->mColdFutexAddr = &mFastMixerFutex; 2852 state->mColdGen++; 2853 mFastMixerFutex = 0; 2854 sq->end(); 2855 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2856 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2857 if (kUseFastMixer == FastMixer_Dynamic) { 2858 mNormalSink = mOutputSink; 2859 } 2860#ifdef AUDIO_WATCHDOG 2861 if (mAudioWatchdog != 0) { 2862 mAudioWatchdog->pause(); 2863 } 2864#endif 2865 } else { 2866 sq->end(false /*didModify*/); 2867 } 2868 } 2869 PlaybackThread::threadLoop_standby(); 2870} 2871 2872bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2873{ 2874 return false; 2875} 2876 2877bool AudioFlinger::PlaybackThread::shouldStandby_l() 2878{ 2879 return !mStandby; 2880} 2881 2882bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2883{ 2884 Mutex::Autolock _l(mLock); 2885 return waitingAsyncCallback_l(); 2886} 2887 2888// shared by MIXER and DIRECT, overridden by DUPLICATING 2889void AudioFlinger::PlaybackThread::threadLoop_standby() 2890{ 2891 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2892 mOutput->stream->common.standby(&mOutput->stream->common); 2893 if (mUseAsyncWrite != 0) { 2894 // discard any pending drain or write ack by incrementing sequence 2895 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2896 mDrainSequence = (mDrainSequence + 2) & ~1; 2897 ALOG_ASSERT(mCallbackThread != 0); 2898 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2899 mCallbackThread->setDraining(mDrainSequence); 2900 } 2901} 2902 2903void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2904{ 2905 ALOGV("signal playback thread"); 2906 broadcast_l(); 2907} 2908 2909void AudioFlinger::MixerThread::threadLoop_mix() 2910{ 2911 // obtain the presentation timestamp of the next output buffer 2912 int64_t pts; 2913 status_t status = INVALID_OPERATION; 2914 2915 if (mNormalSink != 0) { 2916 status = mNormalSink->getNextWriteTimestamp(&pts); 2917 } else { 2918 status = mOutputSink->getNextWriteTimestamp(&pts); 2919 } 2920 2921 if (status != NO_ERROR) { 2922 pts = AudioBufferProvider::kInvalidPTS; 2923 } 2924 2925 // mix buffers... 2926 mAudioMixer->process(pts); 2927 mCurrentWriteLength = mSinkBufferSize; 2928 // increase sleep time progressively when application underrun condition clears. 2929 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2930 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2931 // such that we would underrun the audio HAL. 2932 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2933 sleepTimeShift--; 2934 } 2935 sleepTime = 0; 2936 standbyTime = systemTime() + standbyDelay; 2937 //TODO: delay standby when effects have a tail 2938} 2939 2940void AudioFlinger::MixerThread::threadLoop_sleepTime() 2941{ 2942 // If no tracks are ready, sleep once for the duration of an output 2943 // buffer size, then write 0s to the output 2944 if (sleepTime == 0) { 2945 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2946 sleepTime = activeSleepTime >> sleepTimeShift; 2947 if (sleepTime < kMinThreadSleepTimeUs) { 2948 sleepTime = kMinThreadSleepTimeUs; 2949 } 2950 // reduce sleep time in case of consecutive application underruns to avoid 2951 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2952 // duration we would end up writing less data than needed by the audio HAL if 2953 // the condition persists. 2954 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2955 sleepTimeShift++; 2956 } 2957 } else { 2958 sleepTime = idleSleepTime; 2959 } 2960 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2961 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2962 // before effects processing or output. 2963 if (mMixerBufferValid) { 2964 memset(mMixerBuffer, 0, mMixerBufferSize); 2965 } else { 2966 memset(mSinkBuffer, 0, mSinkBufferSize); 2967 } 2968 sleepTime = 0; 2969 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2970 "anticipated start"); 2971 } 2972 // TODO add standby time extension fct of effect tail 2973} 2974 2975// prepareTracks_l() must be called with ThreadBase::mLock held 2976AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2977 Vector< sp<Track> > *tracksToRemove) 2978{ 2979 2980 mixer_state mixerStatus = MIXER_IDLE; 2981 // find out which tracks need to be processed 2982 size_t count = mActiveTracks.size(); 2983 size_t mixedTracks = 0; 2984 size_t tracksWithEffect = 0; 2985 // counts only _active_ fast tracks 2986 size_t fastTracks = 0; 2987 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2988 2989 float masterVolume = mMasterVolume; 2990 bool masterMute = mMasterMute; 2991 2992 if (masterMute) { 2993 masterVolume = 0; 2994 } 2995 // Delegate master volume control to effect in output mix effect chain if needed 2996 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2997 if (chain != 0) { 2998 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2999 chain->setVolume_l(&v, &v); 3000 masterVolume = (float)((v + (1 << 23)) >> 24); 3001 chain.clear(); 3002 } 3003 3004 // prepare a new state to push 3005 FastMixerStateQueue *sq = NULL; 3006 FastMixerState *state = NULL; 3007 bool didModify = false; 3008 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3009 if (mFastMixer != NULL) { 3010 sq = mFastMixer->sq(); 3011 state = sq->begin(); 3012 } 3013 3014 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3015 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3016 3017 for (size_t i=0 ; i<count ; i++) { 3018 const sp<Track> t = mActiveTracks[i].promote(); 3019 if (t == 0) { 3020 continue; 3021 } 3022 3023 // this const just means the local variable doesn't change 3024 Track* const track = t.get(); 3025 3026 // process fast tracks 3027 if (track->isFastTrack()) { 3028 3029 // It's theoretically possible (though unlikely) for a fast track to be created 3030 // and then removed within the same normal mix cycle. This is not a problem, as 3031 // the track never becomes active so it's fast mixer slot is never touched. 3032 // The converse, of removing an (active) track and then creating a new track 3033 // at the identical fast mixer slot within the same normal mix cycle, 3034 // is impossible because the slot isn't marked available until the end of each cycle. 3035 int j = track->mFastIndex; 3036 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3037 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3038 FastTrack *fastTrack = &state->mFastTracks[j]; 3039 3040 // Determine whether the track is currently in underrun condition, 3041 // and whether it had a recent underrun. 3042 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3043 FastTrackUnderruns underruns = ftDump->mUnderruns; 3044 uint32_t recentFull = (underruns.mBitFields.mFull - 3045 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3046 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3047 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3048 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3049 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3050 uint32_t recentUnderruns = recentPartial + recentEmpty; 3051 track->mObservedUnderruns = underruns; 3052 // don't count underruns that occur while stopping or pausing 3053 // or stopped which can occur when flush() is called while active 3054 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3055 recentUnderruns > 0) { 3056 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3057 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3058 } 3059 3060 // This is similar to the state machine for normal tracks, 3061 // with a few modifications for fast tracks. 3062 bool isActive = true; 3063 switch (track->mState) { 3064 case TrackBase::STOPPING_1: 3065 // track stays active in STOPPING_1 state until first underrun 3066 if (recentUnderruns > 0 || track->isTerminated()) { 3067 track->mState = TrackBase::STOPPING_2; 3068 } 3069 break; 3070 case TrackBase::PAUSING: 3071 // ramp down is not yet implemented 3072 track->setPaused(); 3073 break; 3074 case TrackBase::RESUMING: 3075 // ramp up is not yet implemented 3076 track->mState = TrackBase::ACTIVE; 3077 break; 3078 case TrackBase::ACTIVE: 3079 if (recentFull > 0 || recentPartial > 0) { 3080 // track has provided at least some frames recently: reset retry count 3081 track->mRetryCount = kMaxTrackRetries; 3082 } 3083 if (recentUnderruns == 0) { 3084 // no recent underruns: stay active 3085 break; 3086 } 3087 // there has recently been an underrun of some kind 3088 if (track->sharedBuffer() == 0) { 3089 // were any of the recent underruns "empty" (no frames available)? 3090 if (recentEmpty == 0) { 3091 // no, then ignore the partial underruns as they are allowed indefinitely 3092 break; 3093 } 3094 // there has recently been an "empty" underrun: decrement the retry counter 3095 if (--(track->mRetryCount) > 0) { 3096 break; 3097 } 3098 // indicate to client process that the track was disabled because of underrun; 3099 // it will then automatically call start() when data is available 3100 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3101 // remove from active list, but state remains ACTIVE [confusing but true] 3102 isActive = false; 3103 break; 3104 } 3105 // fall through 3106 case TrackBase::STOPPING_2: 3107 case TrackBase::PAUSED: 3108 case TrackBase::STOPPED: 3109 case TrackBase::FLUSHED: // flush() while active 3110 // Check for presentation complete if track is inactive 3111 // We have consumed all the buffers of this track. 3112 // This would be incomplete if we auto-paused on underrun 3113 { 3114 size_t audioHALFrames = 3115 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3116 size_t framesWritten = mBytesWritten / mFrameSize; 3117 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3118 // track stays in active list until presentation is complete 3119 break; 3120 } 3121 } 3122 if (track->isStopping_2()) { 3123 track->mState = TrackBase::STOPPED; 3124 } 3125 if (track->isStopped()) { 3126 // Can't reset directly, as fast mixer is still polling this track 3127 // track->reset(); 3128 // So instead mark this track as needing to be reset after push with ack 3129 resetMask |= 1 << i; 3130 } 3131 isActive = false; 3132 break; 3133 case TrackBase::IDLE: 3134 default: 3135 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3136 } 3137 3138 if (isActive) { 3139 // was it previously inactive? 3140 if (!(state->mTrackMask & (1 << j))) { 3141 ExtendedAudioBufferProvider *eabp = track; 3142 VolumeProvider *vp = track; 3143 fastTrack->mBufferProvider = eabp; 3144 fastTrack->mVolumeProvider = vp; 3145 fastTrack->mChannelMask = track->mChannelMask; 3146 fastTrack->mGeneration++; 3147 state->mTrackMask |= 1 << j; 3148 didModify = true; 3149 // no acknowledgement required for newly active tracks 3150 } 3151 // cache the combined master volume and stream type volume for fast mixer; this 3152 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3153 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3154 ++fastTracks; 3155 } else { 3156 // was it previously active? 3157 if (state->mTrackMask & (1 << j)) { 3158 fastTrack->mBufferProvider = NULL; 3159 fastTrack->mGeneration++; 3160 state->mTrackMask &= ~(1 << j); 3161 didModify = true; 3162 // If any fast tracks were removed, we must wait for acknowledgement 3163 // because we're about to decrement the last sp<> on those tracks. 3164 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3165 } else { 3166 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3167 } 3168 tracksToRemove->add(track); 3169 // Avoids a misleading display in dumpsys 3170 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3171 } 3172 continue; 3173 } 3174 3175 { // local variable scope to avoid goto warning 3176 3177 audio_track_cblk_t* cblk = track->cblk(); 3178 3179 // The first time a track is added we wait 3180 // for all its buffers to be filled before processing it 3181 int name = track->name(); 3182 // make sure that we have enough frames to mix one full buffer. 3183 // enforce this condition only once to enable draining the buffer in case the client 3184 // app does not call stop() and relies on underrun to stop: 3185 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3186 // during last round 3187 size_t desiredFrames; 3188 uint32_t sr = track->sampleRate(); 3189 if (sr == mSampleRate) { 3190 desiredFrames = mNormalFrameCount; 3191 } else { 3192 // +1 for rounding and +1 for additional sample needed for interpolation 3193 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3194 // add frames already consumed but not yet released by the resampler 3195 // because mAudioTrackServerProxy->framesReady() will include these frames 3196 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3197#if 0 3198 // the minimum track buffer size is normally twice the number of frames necessary 3199 // to fill one buffer and the resampler should not leave more than one buffer worth 3200 // of unreleased frames after each pass, but just in case... 3201 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3202#endif 3203 } 3204 uint32_t minFrames = 1; 3205 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3206 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3207 minFrames = desiredFrames; 3208 } 3209 3210 size_t framesReady = track->framesReady(); 3211 if ((framesReady >= minFrames) && track->isReady() && 3212 !track->isPaused() && !track->isTerminated()) 3213 { 3214 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3215 3216 mixedTracks++; 3217 3218 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3219 // there is an effect chain connected to the track 3220 chain.clear(); 3221 if (track->mainBuffer() != mSinkBuffer && 3222 track->mainBuffer() != mMixerBuffer) { 3223 if (mEffectBufferEnabled) { 3224 mEffectBufferValid = true; // Later can set directly. 3225 } 3226 chain = getEffectChain_l(track->sessionId()); 3227 // Delegate volume control to effect in track effect chain if needed 3228 if (chain != 0) { 3229 tracksWithEffect++; 3230 } else { 3231 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3232 "session %d", 3233 name, track->sessionId()); 3234 } 3235 } 3236 3237 3238 int param = AudioMixer::VOLUME; 3239 if (track->mFillingUpStatus == Track::FS_FILLED) { 3240 // no ramp for the first volume setting 3241 track->mFillingUpStatus = Track::FS_ACTIVE; 3242 if (track->mState == TrackBase::RESUMING) { 3243 track->mState = TrackBase::ACTIVE; 3244 param = AudioMixer::RAMP_VOLUME; 3245 } 3246 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3247 // FIXME should not make a decision based on mServer 3248 } else if (cblk->mServer != 0) { 3249 // If the track is stopped before the first frame was mixed, 3250 // do not apply ramp 3251 param = AudioMixer::RAMP_VOLUME; 3252 } 3253 3254 // compute volume for this track 3255 uint32_t vl, vr, va; 3256 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3257 vl = vr = va = 0; 3258 if (track->isPausing()) { 3259 track->setPaused(); 3260 } 3261 } else { 3262 3263 // read original volumes with volume control 3264 float typeVolume = mStreamTypes[track->streamType()].volume; 3265 float v = masterVolume * typeVolume; 3266 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3267 uint32_t vlr = proxy->getVolumeLR(); 3268 vl = vlr & 0xFFFF; 3269 vr = vlr >> 16; 3270 // track volumes come from shared memory, so can't be trusted and must be clamped 3271 if (vl > MAX_GAIN_INT) { 3272 ALOGV("Track left volume out of range: %04X", vl); 3273 vl = MAX_GAIN_INT; 3274 } 3275 if (vr > MAX_GAIN_INT) { 3276 ALOGV("Track right volume out of range: %04X", vr); 3277 vr = MAX_GAIN_INT; 3278 } 3279 // now apply the master volume and stream type volume 3280 vl = (uint32_t)(v * vl) << 12; 3281 vr = (uint32_t)(v * vr) << 12; 3282 // assuming master volume and stream type volume each go up to 1.0, 3283 // vl and vr are now in 8.24 format 3284 3285 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3286 // send level comes from shared memory and so may be corrupt 3287 if (sendLevel > MAX_GAIN_INT) { 3288 ALOGV("Track send level out of range: %04X", sendLevel); 3289 sendLevel = MAX_GAIN_INT; 3290 } 3291 va = (uint32_t)(v * sendLevel); 3292 } 3293 3294 // Delegate volume control to effect in track effect chain if needed 3295 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3296 // Do not ramp volume if volume is controlled by effect 3297 param = AudioMixer::VOLUME; 3298 track->mHasVolumeController = true; 3299 } else { 3300 // force no volume ramp when volume controller was just disabled or removed 3301 // from effect chain to avoid volume spike 3302 if (track->mHasVolumeController) { 3303 param = AudioMixer::VOLUME; 3304 } 3305 track->mHasVolumeController = false; 3306 } 3307 3308 // Convert volumes from 8.24 to 4.12 format 3309 // This additional clamping is needed in case chain->setVolume_l() overshot 3310 vl = (vl + (1 << 11)) >> 12; 3311 if (vl > MAX_GAIN_INT) { 3312 vl = MAX_GAIN_INT; 3313 } 3314 vr = (vr + (1 << 11)) >> 12; 3315 if (vr > MAX_GAIN_INT) { 3316 vr = MAX_GAIN_INT; 3317 } 3318 3319 if (va > MAX_GAIN_INT) { 3320 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3321 } 3322 3323 // XXX: these things DON'T need to be done each time 3324 mAudioMixer->setBufferProvider(name, track); 3325 mAudioMixer->enable(name); 3326 3327 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3328 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3329 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3330 mAudioMixer->setParameter( 3331 name, 3332 AudioMixer::TRACK, 3333 AudioMixer::FORMAT, (void *)track->format()); 3334 mAudioMixer->setParameter( 3335 name, 3336 AudioMixer::TRACK, 3337 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3338 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3339 uint32_t maxSampleRate = mSampleRate * 2; 3340 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3341 if (reqSampleRate == 0) { 3342 reqSampleRate = mSampleRate; 3343 } else if (reqSampleRate > maxSampleRate) { 3344 reqSampleRate = maxSampleRate; 3345 } 3346 mAudioMixer->setParameter( 3347 name, 3348 AudioMixer::RESAMPLE, 3349 AudioMixer::SAMPLE_RATE, 3350 (void *)(uintptr_t)reqSampleRate); 3351 /* 3352 * Select the appropriate output buffer for the track. 3353 * 3354 * Tracks with effects go into their own effects chain buffer 3355 * and from there into either mEffectBuffer or mSinkBuffer. 3356 * 3357 * Other tracks can use mMixerBuffer for higher precision 3358 * channel accumulation. If this buffer is enabled 3359 * (mMixerBufferEnabled true), then selected tracks will accumulate 3360 * into it. 3361 * 3362 */ 3363 if (mMixerBufferEnabled 3364 && (track->mainBuffer() == mSinkBuffer 3365 || track->mainBuffer() == mMixerBuffer)) { 3366 mAudioMixer->setParameter( 3367 name, 3368 AudioMixer::TRACK, 3369 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3370 mAudioMixer->setParameter( 3371 name, 3372 AudioMixer::TRACK, 3373 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3374 // TODO: override track->mainBuffer()? 3375 mMixerBufferValid = true; 3376 } else { 3377 mAudioMixer->setParameter( 3378 name, 3379 AudioMixer::TRACK, 3380 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3381 mAudioMixer->setParameter( 3382 name, 3383 AudioMixer::TRACK, 3384 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3385 } 3386 mAudioMixer->setParameter( 3387 name, 3388 AudioMixer::TRACK, 3389 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3390 3391 // reset retry count 3392 track->mRetryCount = kMaxTrackRetries; 3393 3394 // If one track is ready, set the mixer ready if: 3395 // - the mixer was not ready during previous round OR 3396 // - no other track is not ready 3397 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3398 mixerStatus != MIXER_TRACKS_ENABLED) { 3399 mixerStatus = MIXER_TRACKS_READY; 3400 } 3401 } else { 3402 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3403 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3404 } 3405 // clear effect chain input buffer if an active track underruns to avoid sending 3406 // previous audio buffer again to effects 3407 chain = getEffectChain_l(track->sessionId()); 3408 if (chain != 0) { 3409 chain->clearInputBuffer(); 3410 } 3411 3412 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3413 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3414 track->isStopped() || track->isPaused()) { 3415 // We have consumed all the buffers of this track. 3416 // Remove it from the list of active tracks. 3417 // TODO: use actual buffer filling status instead of latency when available from 3418 // audio HAL 3419 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3420 size_t framesWritten = mBytesWritten / mFrameSize; 3421 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3422 if (track->isStopped()) { 3423 track->reset(); 3424 } 3425 tracksToRemove->add(track); 3426 } 3427 } else { 3428 // No buffers for this track. Give it a few chances to 3429 // fill a buffer, then remove it from active list. 3430 if (--(track->mRetryCount) <= 0) { 3431 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3432 tracksToRemove->add(track); 3433 // indicate to client process that the track was disabled because of underrun; 3434 // it will then automatically call start() when data is available 3435 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3436 // If one track is not ready, mark the mixer also not ready if: 3437 // - the mixer was ready during previous round OR 3438 // - no other track is ready 3439 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3440 mixerStatus != MIXER_TRACKS_READY) { 3441 mixerStatus = MIXER_TRACKS_ENABLED; 3442 } 3443 } 3444 mAudioMixer->disable(name); 3445 } 3446 3447 } // local variable scope to avoid goto warning 3448track_is_ready: ; 3449 3450 } 3451 3452 // Push the new FastMixer state if necessary 3453 bool pauseAudioWatchdog = false; 3454 if (didModify) { 3455 state->mFastTracksGen++; 3456 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3457 if (kUseFastMixer == FastMixer_Dynamic && 3458 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3459 state->mCommand = FastMixerState::COLD_IDLE; 3460 state->mColdFutexAddr = &mFastMixerFutex; 3461 state->mColdGen++; 3462 mFastMixerFutex = 0; 3463 if (kUseFastMixer == FastMixer_Dynamic) { 3464 mNormalSink = mOutputSink; 3465 } 3466 // If we go into cold idle, need to wait for acknowledgement 3467 // so that fast mixer stops doing I/O. 3468 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3469 pauseAudioWatchdog = true; 3470 } 3471 } 3472 if (sq != NULL) { 3473 sq->end(didModify); 3474 sq->push(block); 3475 } 3476#ifdef AUDIO_WATCHDOG 3477 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3478 mAudioWatchdog->pause(); 3479 } 3480#endif 3481 3482 // Now perform the deferred reset on fast tracks that have stopped 3483 while (resetMask != 0) { 3484 size_t i = __builtin_ctz(resetMask); 3485 ALOG_ASSERT(i < count); 3486 resetMask &= ~(1 << i); 3487 sp<Track> t = mActiveTracks[i].promote(); 3488 if (t == 0) { 3489 continue; 3490 } 3491 Track* track = t.get(); 3492 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3493 track->reset(); 3494 } 3495 3496 // remove all the tracks that need to be... 3497 removeTracks_l(*tracksToRemove); 3498 3499 // sink or mix buffer must be cleared if all tracks are connected to an 3500 // effect chain as in this case the mixer will not write to the sink or mix buffer 3501 // and track effects will accumulate into it 3502 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3503 (mixedTracks == 0 && fastTracks > 0))) { 3504 // FIXME as a performance optimization, should remember previous zero status 3505 if (mMixerBufferValid) { 3506 memset(mMixerBuffer, 0, mMixerBufferSize); 3507 // TODO: In testing, mSinkBuffer below need not be cleared because 3508 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3509 // after mixing. 3510 // 3511 // To enforce this guarantee: 3512 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3513 // (mixedTracks == 0 && fastTracks > 0)) 3514 // must imply MIXER_TRACKS_READY. 3515 // Later, we may clear buffers regardless, and skip much of this logic. 3516 } 3517 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3518 if (mEffectBufferValid) { 3519 memset(mEffectBuffer, 0, mEffectBufferSize); 3520 } 3521 // FIXME as a performance optimization, should remember previous zero status 3522 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3523 } 3524 3525 // if any fast tracks, then status is ready 3526 mMixerStatusIgnoringFastTracks = mixerStatus; 3527 if (fastTracks > 0) { 3528 mixerStatus = MIXER_TRACKS_READY; 3529 } 3530 return mixerStatus; 3531} 3532 3533// getTrackName_l() must be called with ThreadBase::mLock held 3534int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3535{ 3536 return mAudioMixer->getTrackName(channelMask, sessionId); 3537} 3538 3539// deleteTrackName_l() must be called with ThreadBase::mLock held 3540void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3541{ 3542 ALOGV("remove track (%d) and delete from mixer", name); 3543 mAudioMixer->deleteTrackName(name); 3544} 3545 3546// checkForNewParameter_l() must be called with ThreadBase::mLock held 3547bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3548 status_t& status) 3549{ 3550 bool reconfig = false; 3551 3552 status = NO_ERROR; 3553 3554 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3555 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3556 if (mFastMixer != NULL) { 3557 FastMixerStateQueue *sq = mFastMixer->sq(); 3558 FastMixerState *state = sq->begin(); 3559 if (!(state->mCommand & FastMixerState::IDLE)) { 3560 previousCommand = state->mCommand; 3561 state->mCommand = FastMixerState::HOT_IDLE; 3562 sq->end(); 3563 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3564 } else { 3565 sq->end(false /*didModify*/); 3566 } 3567 } 3568 3569 AudioParameter param = AudioParameter(keyValuePair); 3570 int value; 3571 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3572 reconfig = true; 3573 } 3574 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3575 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3576 status = BAD_VALUE; 3577 } else { 3578 // no need to save value, since it's constant 3579 reconfig = true; 3580 } 3581 } 3582 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3583 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3584 status = BAD_VALUE; 3585 } else { 3586 // no need to save value, since it's constant 3587 reconfig = true; 3588 } 3589 } 3590 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3591 // do not accept frame count changes if tracks are open as the track buffer 3592 // size depends on frame count and correct behavior would not be guaranteed 3593 // if frame count is changed after track creation 3594 if (!mTracks.isEmpty()) { 3595 status = INVALID_OPERATION; 3596 } else { 3597 reconfig = true; 3598 } 3599 } 3600 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3601#ifdef ADD_BATTERY_DATA 3602 // when changing the audio output device, call addBatteryData to notify 3603 // the change 3604 if (mOutDevice != value) { 3605 uint32_t params = 0; 3606 // check whether speaker is on 3607 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3608 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3609 } 3610 3611 audio_devices_t deviceWithoutSpeaker 3612 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3613 // check if any other device (except speaker) is on 3614 if (value & deviceWithoutSpeaker ) { 3615 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3616 } 3617 3618 if (params != 0) { 3619 addBatteryData(params); 3620 } 3621 } 3622#endif 3623 3624 // forward device change to effects that have requested to be 3625 // aware of attached audio device. 3626 if (value != AUDIO_DEVICE_NONE) { 3627 mOutDevice = value; 3628 for (size_t i = 0; i < mEffectChains.size(); i++) { 3629 mEffectChains[i]->setDevice_l(mOutDevice); 3630 } 3631 } 3632 } 3633 3634 if (status == NO_ERROR) { 3635 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3636 keyValuePair.string()); 3637 if (!mStandby && status == INVALID_OPERATION) { 3638 mOutput->stream->common.standby(&mOutput->stream->common); 3639 mStandby = true; 3640 mBytesWritten = 0; 3641 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3642 keyValuePair.string()); 3643 } 3644 if (status == NO_ERROR && reconfig) { 3645 readOutputParameters_l(); 3646 delete mAudioMixer; 3647 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3648 for (size_t i = 0; i < mTracks.size() ; i++) { 3649 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3650 if (name < 0) { 3651 break; 3652 } 3653 mTracks[i]->mName = name; 3654 } 3655 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3656 } 3657 } 3658 3659 if (!(previousCommand & FastMixerState::IDLE)) { 3660 ALOG_ASSERT(mFastMixer != NULL); 3661 FastMixerStateQueue *sq = mFastMixer->sq(); 3662 FastMixerState *state = sq->begin(); 3663 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3664 state->mCommand = previousCommand; 3665 sq->end(); 3666 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3667 } 3668 3669 return reconfig; 3670} 3671 3672 3673void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3674{ 3675 const size_t SIZE = 256; 3676 char buffer[SIZE]; 3677 String8 result; 3678 3679 PlaybackThread::dumpInternals(fd, args); 3680 3681 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3682 3683 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3684 const FastMixerDumpState copy(mFastMixerDumpState); 3685 copy.dump(fd); 3686 3687#ifdef STATE_QUEUE_DUMP 3688 // Similar for state queue 3689 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3690 observerCopy.dump(fd); 3691 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3692 mutatorCopy.dump(fd); 3693#endif 3694 3695#ifdef TEE_SINK 3696 // Write the tee output to a .wav file 3697 dumpTee(fd, mTeeSource, mId); 3698#endif 3699 3700#ifdef AUDIO_WATCHDOG 3701 if (mAudioWatchdog != 0) { 3702 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3703 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3704 wdCopy.dump(fd); 3705 } 3706#endif 3707} 3708 3709uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3710{ 3711 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3712} 3713 3714uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3715{ 3716 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3717} 3718 3719void AudioFlinger::MixerThread::cacheParameters_l() 3720{ 3721 PlaybackThread::cacheParameters_l(); 3722 3723 // FIXME: Relaxed timing because of a certain device that can't meet latency 3724 // Should be reduced to 2x after the vendor fixes the driver issue 3725 // increase threshold again due to low power audio mode. The way this warning 3726 // threshold is calculated and its usefulness should be reconsidered anyway. 3727 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3728} 3729 3730// ---------------------------------------------------------------------------- 3731 3732AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3733 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3734 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3735 // mLeftVolFloat, mRightVolFloat 3736{ 3737} 3738 3739AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3740 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3741 ThreadBase::type_t type) 3742 : PlaybackThread(audioFlinger, output, id, device, type) 3743 // mLeftVolFloat, mRightVolFloat 3744{ 3745} 3746 3747AudioFlinger::DirectOutputThread::~DirectOutputThread() 3748{ 3749} 3750 3751void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3752{ 3753 audio_track_cblk_t* cblk = track->cblk(); 3754 float left, right; 3755 3756 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3757 left = right = 0; 3758 } else { 3759 float typeVolume = mStreamTypes[track->streamType()].volume; 3760 float v = mMasterVolume * typeVolume; 3761 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3762 uint32_t vlr = proxy->getVolumeLR(); 3763 float v_clamped = v * (vlr & 0xFFFF); 3764 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3765 left = v_clamped/MAX_GAIN; 3766 v_clamped = v * (vlr >> 16); 3767 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3768 right = v_clamped/MAX_GAIN; 3769 } 3770 3771 if (lastTrack) { 3772 if (left != mLeftVolFloat || right != mRightVolFloat) { 3773 mLeftVolFloat = left; 3774 mRightVolFloat = right; 3775 3776 // Convert volumes from float to 8.24 3777 uint32_t vl = (uint32_t)(left * (1 << 24)); 3778 uint32_t vr = (uint32_t)(right * (1 << 24)); 3779 3780 // Delegate volume control to effect in track effect chain if needed 3781 // only one effect chain can be present on DirectOutputThread, so if 3782 // there is one, the track is connected to it 3783 if (!mEffectChains.isEmpty()) { 3784 mEffectChains[0]->setVolume_l(&vl, &vr); 3785 left = (float)vl / (1 << 24); 3786 right = (float)vr / (1 << 24); 3787 } 3788 if (mOutput->stream->set_volume) { 3789 mOutput->stream->set_volume(mOutput->stream, left, right); 3790 } 3791 } 3792 } 3793} 3794 3795 3796AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3797 Vector< sp<Track> > *tracksToRemove 3798) 3799{ 3800 size_t count = mActiveTracks.size(); 3801 mixer_state mixerStatus = MIXER_IDLE; 3802 3803 // find out which tracks need to be processed 3804 for (size_t i = 0; i < count; i++) { 3805 sp<Track> t = mActiveTracks[i].promote(); 3806 // The track died recently 3807 if (t == 0) { 3808 continue; 3809 } 3810 3811 Track* const track = t.get(); 3812 audio_track_cblk_t* cblk = track->cblk(); 3813 // Only consider last track started for volume and mixer state control. 3814 // In theory an older track could underrun and restart after the new one starts 3815 // but as we only care about the transition phase between two tracks on a 3816 // direct output, it is not a problem to ignore the underrun case. 3817 sp<Track> l = mLatestActiveTrack.promote(); 3818 bool last = l.get() == track; 3819 3820 // The first time a track is added we wait 3821 // for all its buffers to be filled before processing it 3822 uint32_t minFrames; 3823 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3824 minFrames = mNormalFrameCount; 3825 } else { 3826 minFrames = 1; 3827 } 3828 3829 if ((track->framesReady() >= minFrames) && track->isReady() && 3830 !track->isPaused() && !track->isTerminated()) 3831 { 3832 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3833 3834 if (track->mFillingUpStatus == Track::FS_FILLED) { 3835 track->mFillingUpStatus = Track::FS_ACTIVE; 3836 // make sure processVolume_l() will apply new volume even if 0 3837 mLeftVolFloat = mRightVolFloat = -1.0; 3838 if (track->mState == TrackBase::RESUMING) { 3839 track->mState = TrackBase::ACTIVE; 3840 } 3841 } 3842 3843 // compute volume for this track 3844 processVolume_l(track, last); 3845 if (last) { 3846 // reset retry count 3847 track->mRetryCount = kMaxTrackRetriesDirect; 3848 mActiveTrack = t; 3849 mixerStatus = MIXER_TRACKS_READY; 3850 } 3851 } else { 3852 // clear effect chain input buffer if the last active track started underruns 3853 // to avoid sending previous audio buffer again to effects 3854 if (!mEffectChains.isEmpty() && last) { 3855 mEffectChains[0]->clearInputBuffer(); 3856 } 3857 3858 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3859 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3860 track->isStopped() || track->isPaused()) { 3861 // We have consumed all the buffers of this track. 3862 // Remove it from the list of active tracks. 3863 // TODO: implement behavior for compressed audio 3864 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3865 size_t framesWritten = mBytesWritten / mFrameSize; 3866 if (mStandby || !last || 3867 track->presentationComplete(framesWritten, audioHALFrames)) { 3868 if (track->isStopped()) { 3869 track->reset(); 3870 } 3871 tracksToRemove->add(track); 3872 } 3873 } else { 3874 // No buffers for this track. Give it a few chances to 3875 // fill a buffer, then remove it from active list. 3876 // Only consider last track started for mixer state control 3877 if (--(track->mRetryCount) <= 0) { 3878 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3879 tracksToRemove->add(track); 3880 // indicate to client process that the track was disabled because of underrun; 3881 // it will then automatically call start() when data is available 3882 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3883 } else if (last) { 3884 mixerStatus = MIXER_TRACKS_ENABLED; 3885 } 3886 } 3887 } 3888 } 3889 3890 // remove all the tracks that need to be... 3891 removeTracks_l(*tracksToRemove); 3892 3893 return mixerStatus; 3894} 3895 3896void AudioFlinger::DirectOutputThread::threadLoop_mix() 3897{ 3898 size_t frameCount = mFrameCount; 3899 int8_t *curBuf = (int8_t *)mSinkBuffer; 3900 // output audio to hardware 3901 while (frameCount) { 3902 AudioBufferProvider::Buffer buffer; 3903 buffer.frameCount = frameCount; 3904 mActiveTrack->getNextBuffer(&buffer); 3905 if (buffer.raw == NULL) { 3906 memset(curBuf, 0, frameCount * mFrameSize); 3907 break; 3908 } 3909 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3910 frameCount -= buffer.frameCount; 3911 curBuf += buffer.frameCount * mFrameSize; 3912 mActiveTrack->releaseBuffer(&buffer); 3913 } 3914 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3915 sleepTime = 0; 3916 standbyTime = systemTime() + standbyDelay; 3917 mActiveTrack.clear(); 3918} 3919 3920void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3921{ 3922 if (sleepTime == 0) { 3923 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3924 sleepTime = activeSleepTime; 3925 } else { 3926 sleepTime = idleSleepTime; 3927 } 3928 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3929 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3930 sleepTime = 0; 3931 } 3932} 3933 3934// getTrackName_l() must be called with ThreadBase::mLock held 3935int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3936 int sessionId __unused) 3937{ 3938 return 0; 3939} 3940 3941// deleteTrackName_l() must be called with ThreadBase::mLock held 3942void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3943{ 3944} 3945 3946// checkForNewParameter_l() must be called with ThreadBase::mLock held 3947bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 3948 status_t& status) 3949{ 3950 bool reconfig = false; 3951 3952 status = NO_ERROR; 3953 3954 AudioParameter param = AudioParameter(keyValuePair); 3955 int value; 3956 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3957 // forward device change to effects that have requested to be 3958 // aware of attached audio device. 3959 if (value != AUDIO_DEVICE_NONE) { 3960 mOutDevice = value; 3961 for (size_t i = 0; i < mEffectChains.size(); i++) { 3962 mEffectChains[i]->setDevice_l(mOutDevice); 3963 } 3964 } 3965 } 3966 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3967 // do not accept frame count changes if tracks are open as the track buffer 3968 // size depends on frame count and correct behavior would not be garantied 3969 // if frame count is changed after track creation 3970 if (!mTracks.isEmpty()) { 3971 status = INVALID_OPERATION; 3972 } else { 3973 reconfig = true; 3974 } 3975 } 3976 if (status == NO_ERROR) { 3977 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3978 keyValuePair.string()); 3979 if (!mStandby && status == INVALID_OPERATION) { 3980 mOutput->stream->common.standby(&mOutput->stream->common); 3981 mStandby = true; 3982 mBytesWritten = 0; 3983 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3984 keyValuePair.string()); 3985 } 3986 if (status == NO_ERROR && reconfig) { 3987 readOutputParameters_l(); 3988 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3989 } 3990 } 3991 3992 return reconfig; 3993} 3994 3995uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3996{ 3997 uint32_t time; 3998 if (audio_is_linear_pcm(mFormat)) { 3999 time = PlaybackThread::activeSleepTimeUs(); 4000 } else { 4001 time = 10000; 4002 } 4003 return time; 4004} 4005 4006uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4007{ 4008 uint32_t time; 4009 if (audio_is_linear_pcm(mFormat)) { 4010 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4011 } else { 4012 time = 10000; 4013 } 4014 return time; 4015} 4016 4017uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4018{ 4019 uint32_t time; 4020 if (audio_is_linear_pcm(mFormat)) { 4021 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4022 } else { 4023 time = 10000; 4024 } 4025 return time; 4026} 4027 4028void AudioFlinger::DirectOutputThread::cacheParameters_l() 4029{ 4030 PlaybackThread::cacheParameters_l(); 4031 4032 // use shorter standby delay as on normal output to release 4033 // hardware resources as soon as possible 4034 if (audio_is_linear_pcm(mFormat)) { 4035 standbyDelay = microseconds(activeSleepTime*2); 4036 } else { 4037 standbyDelay = kOffloadStandbyDelayNs; 4038 } 4039} 4040 4041// ---------------------------------------------------------------------------- 4042 4043AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4044 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4045 : Thread(false /*canCallJava*/), 4046 mPlaybackThread(playbackThread), 4047 mWriteAckSequence(0), 4048 mDrainSequence(0) 4049{ 4050} 4051 4052AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4053{ 4054} 4055 4056void AudioFlinger::AsyncCallbackThread::onFirstRef() 4057{ 4058 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4059} 4060 4061bool AudioFlinger::AsyncCallbackThread::threadLoop() 4062{ 4063 while (!exitPending()) { 4064 uint32_t writeAckSequence; 4065 uint32_t drainSequence; 4066 4067 { 4068 Mutex::Autolock _l(mLock); 4069 while (!((mWriteAckSequence & 1) || 4070 (mDrainSequence & 1) || 4071 exitPending())) { 4072 mWaitWorkCV.wait(mLock); 4073 } 4074 4075 if (exitPending()) { 4076 break; 4077 } 4078 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4079 mWriteAckSequence, mDrainSequence); 4080 writeAckSequence = mWriteAckSequence; 4081 mWriteAckSequence &= ~1; 4082 drainSequence = mDrainSequence; 4083 mDrainSequence &= ~1; 4084 } 4085 { 4086 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4087 if (playbackThread != 0) { 4088 if (writeAckSequence & 1) { 4089 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4090 } 4091 if (drainSequence & 1) { 4092 playbackThread->resetDraining(drainSequence >> 1); 4093 } 4094 } 4095 } 4096 } 4097 return false; 4098} 4099 4100void AudioFlinger::AsyncCallbackThread::exit() 4101{ 4102 ALOGV("AsyncCallbackThread::exit"); 4103 Mutex::Autolock _l(mLock); 4104 requestExit(); 4105 mWaitWorkCV.broadcast(); 4106} 4107 4108void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4109{ 4110 Mutex::Autolock _l(mLock); 4111 // bit 0 is cleared 4112 mWriteAckSequence = sequence << 1; 4113} 4114 4115void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4116{ 4117 Mutex::Autolock _l(mLock); 4118 // ignore unexpected callbacks 4119 if (mWriteAckSequence & 2) { 4120 mWriteAckSequence |= 1; 4121 mWaitWorkCV.signal(); 4122 } 4123} 4124 4125void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4126{ 4127 Mutex::Autolock _l(mLock); 4128 // bit 0 is cleared 4129 mDrainSequence = sequence << 1; 4130} 4131 4132void AudioFlinger::AsyncCallbackThread::resetDraining() 4133{ 4134 Mutex::Autolock _l(mLock); 4135 // ignore unexpected callbacks 4136 if (mDrainSequence & 2) { 4137 mDrainSequence |= 1; 4138 mWaitWorkCV.signal(); 4139 } 4140} 4141 4142 4143// ---------------------------------------------------------------------------- 4144AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4145 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4146 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4147 mHwPaused(false), 4148 mFlushPending(false), 4149 mPausedBytesRemaining(0) 4150{ 4151 //FIXME: mStandby should be set to true by ThreadBase constructor 4152 mStandby = true; 4153} 4154 4155void AudioFlinger::OffloadThread::threadLoop_exit() 4156{ 4157 if (mFlushPending || mHwPaused) { 4158 // If a flush is pending or track was paused, just discard buffered data 4159 flushHw_l(); 4160 } else { 4161 mMixerStatus = MIXER_DRAIN_ALL; 4162 threadLoop_drain(); 4163 } 4164 mCallbackThread->exit(); 4165 PlaybackThread::threadLoop_exit(); 4166} 4167 4168AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4169 Vector< sp<Track> > *tracksToRemove 4170) 4171{ 4172 size_t count = mActiveTracks.size(); 4173 4174 mixer_state mixerStatus = MIXER_IDLE; 4175 bool doHwPause = false; 4176 bool doHwResume = false; 4177 4178 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4179 4180 // find out which tracks need to be processed 4181 for (size_t i = 0; i < count; i++) { 4182 sp<Track> t = mActiveTracks[i].promote(); 4183 // The track died recently 4184 if (t == 0) { 4185 continue; 4186 } 4187 Track* const track = t.get(); 4188 audio_track_cblk_t* cblk = track->cblk(); 4189 // Only consider last track started for volume and mixer state control. 4190 // In theory an older track could underrun and restart after the new one starts 4191 // but as we only care about the transition phase between two tracks on a 4192 // direct output, it is not a problem to ignore the underrun case. 4193 sp<Track> l = mLatestActiveTrack.promote(); 4194 bool last = l.get() == track; 4195 4196 if (track->isInvalid()) { 4197 ALOGW("An invalidated track shouldn't be in active list"); 4198 tracksToRemove->add(track); 4199 continue; 4200 } 4201 4202 if (track->mState == TrackBase::IDLE) { 4203 ALOGW("An idle track shouldn't be in active list"); 4204 continue; 4205 } 4206 4207 if (track->isPausing()) { 4208 track->setPaused(); 4209 if (last) { 4210 if (!mHwPaused) { 4211 doHwPause = true; 4212 mHwPaused = true; 4213 } 4214 // If we were part way through writing the mixbuffer to 4215 // the HAL we must save this until we resume 4216 // BUG - this will be wrong if a different track is made active, 4217 // in that case we want to discard the pending data in the 4218 // mixbuffer and tell the client to present it again when the 4219 // track is resumed 4220 mPausedWriteLength = mCurrentWriteLength; 4221 mPausedBytesRemaining = mBytesRemaining; 4222 mBytesRemaining = 0; // stop writing 4223 } 4224 tracksToRemove->add(track); 4225 } else if (track->isFlushPending()) { 4226 track->flushAck(); 4227 if (last) { 4228 mFlushPending = true; 4229 } 4230 } else if (track->isResumePending()){ 4231 track->resumeAck(); 4232 if (last) { 4233 if (mPausedBytesRemaining) { 4234 // Need to continue write that was interrupted 4235 mCurrentWriteLength = mPausedWriteLength; 4236 mBytesRemaining = mPausedBytesRemaining; 4237 mPausedBytesRemaining = 0; 4238 } 4239 if (mHwPaused) { 4240 doHwResume = true; 4241 mHwPaused = false; 4242 // threadLoop_mix() will handle the case that we need to 4243 // resume an interrupted write 4244 } 4245 // enable write to audio HAL 4246 sleepTime = 0; 4247 4248 // Do not handle new data in this iteration even if track->framesReady() 4249 mixerStatus = MIXER_TRACKS_ENABLED; 4250 } 4251 } else if (track->framesReady() && track->isReady() && 4252 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4253 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4254 if (track->mFillingUpStatus == Track::FS_FILLED) { 4255 track->mFillingUpStatus = Track::FS_ACTIVE; 4256 // make sure processVolume_l() will apply new volume even if 0 4257 mLeftVolFloat = mRightVolFloat = -1.0; 4258 } 4259 4260 if (last) { 4261 sp<Track> previousTrack = mPreviousTrack.promote(); 4262 if (previousTrack != 0) { 4263 if (track != previousTrack.get()) { 4264 // Flush any data still being written from last track 4265 mBytesRemaining = 0; 4266 if (mPausedBytesRemaining) { 4267 // Last track was paused so we also need to flush saved 4268 // mixbuffer state and invalidate track so that it will 4269 // re-submit that unwritten data when it is next resumed 4270 mPausedBytesRemaining = 0; 4271 // Invalidate is a bit drastic - would be more efficient 4272 // to have a flag to tell client that some of the 4273 // previously written data was lost 4274 previousTrack->invalidate(); 4275 } 4276 // flush data already sent to the DSP if changing audio session as audio 4277 // comes from a different source. Also invalidate previous track to force a 4278 // seek when resuming. 4279 if (previousTrack->sessionId() != track->sessionId()) { 4280 previousTrack->invalidate(); 4281 } 4282 } 4283 } 4284 mPreviousTrack = track; 4285 // reset retry count 4286 track->mRetryCount = kMaxTrackRetriesOffload; 4287 mActiveTrack = t; 4288 mixerStatus = MIXER_TRACKS_READY; 4289 } 4290 } else { 4291 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4292 if (track->isStopping_1()) { 4293 // Hardware buffer can hold a large amount of audio so we must 4294 // wait for all current track's data to drain before we say 4295 // that the track is stopped. 4296 if (mBytesRemaining == 0) { 4297 // Only start draining when all data in mixbuffer 4298 // has been written 4299 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4300 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4301 // do not drain if no data was ever sent to HAL (mStandby == true) 4302 if (last && !mStandby) { 4303 // do not modify drain sequence if we are already draining. This happens 4304 // when resuming from pause after drain. 4305 if ((mDrainSequence & 1) == 0) { 4306 sleepTime = 0; 4307 standbyTime = systemTime() + standbyDelay; 4308 mixerStatus = MIXER_DRAIN_TRACK; 4309 mDrainSequence += 2; 4310 } 4311 if (mHwPaused) { 4312 // It is possible to move from PAUSED to STOPPING_1 without 4313 // a resume so we must ensure hardware is running 4314 doHwResume = true; 4315 mHwPaused = false; 4316 } 4317 } 4318 } 4319 } else if (track->isStopping_2()) { 4320 // Drain has completed or we are in standby, signal presentation complete 4321 if (!(mDrainSequence & 1) || !last || mStandby) { 4322 track->mState = TrackBase::STOPPED; 4323 size_t audioHALFrames = 4324 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4325 size_t framesWritten = 4326 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4327 track->presentationComplete(framesWritten, audioHALFrames); 4328 track->reset(); 4329 tracksToRemove->add(track); 4330 } 4331 } else { 4332 // No buffers for this track. Give it a few chances to 4333 // fill a buffer, then remove it from active list. 4334 if (--(track->mRetryCount) <= 0) { 4335 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4336 track->name()); 4337 tracksToRemove->add(track); 4338 // indicate to client process that the track was disabled because of underrun; 4339 // it will then automatically call start() when data is available 4340 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4341 } else if (last){ 4342 mixerStatus = MIXER_TRACKS_ENABLED; 4343 } 4344 } 4345 } 4346 // compute volume for this track 4347 processVolume_l(track, last); 4348 } 4349 4350 // make sure the pause/flush/resume sequence is executed in the right order. 4351 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4352 // before flush and then resume HW. This can happen in case of pause/flush/resume 4353 // if resume is received before pause is executed. 4354 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4355 mOutput->stream->pause(mOutput->stream); 4356 } 4357 if (mFlushPending) { 4358 flushHw_l(); 4359 mFlushPending = false; 4360 } 4361 if (!mStandby && doHwResume) { 4362 mOutput->stream->resume(mOutput->stream); 4363 } 4364 4365 // remove all the tracks that need to be... 4366 removeTracks_l(*tracksToRemove); 4367 4368 return mixerStatus; 4369} 4370 4371// must be called with thread mutex locked 4372bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4373{ 4374 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4375 mWriteAckSequence, mDrainSequence); 4376 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4377 return true; 4378 } 4379 return false; 4380} 4381 4382// must be called with thread mutex locked 4383bool AudioFlinger::OffloadThread::shouldStandby_l() 4384{ 4385 bool trackPaused = false; 4386 4387 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4388 // after a timeout and we will enter standby then. 4389 if (mTracks.size() > 0) { 4390 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4391 } 4392 4393 return !mStandby && !trackPaused; 4394} 4395 4396 4397bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4398{ 4399 Mutex::Autolock _l(mLock); 4400 return waitingAsyncCallback_l(); 4401} 4402 4403void AudioFlinger::OffloadThread::flushHw_l() 4404{ 4405 mOutput->stream->flush(mOutput->stream); 4406 // Flush anything still waiting in the mixbuffer 4407 mCurrentWriteLength = 0; 4408 mBytesRemaining = 0; 4409 mPausedWriteLength = 0; 4410 mPausedBytesRemaining = 0; 4411 mHwPaused = false; 4412 4413 if (mUseAsyncWrite) { 4414 // discard any pending drain or write ack by incrementing sequence 4415 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4416 mDrainSequence = (mDrainSequence + 2) & ~1; 4417 ALOG_ASSERT(mCallbackThread != 0); 4418 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4419 mCallbackThread->setDraining(mDrainSequence); 4420 } 4421} 4422 4423void AudioFlinger::OffloadThread::onAddNewTrack_l() 4424{ 4425 sp<Track> previousTrack = mPreviousTrack.promote(); 4426 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4427 4428 if (previousTrack != 0 && latestTrack != 0 && 4429 (previousTrack->sessionId() != latestTrack->sessionId())) { 4430 mFlushPending = true; 4431 } 4432 PlaybackThread::onAddNewTrack_l(); 4433} 4434 4435// ---------------------------------------------------------------------------- 4436 4437AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4438 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4439 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4440 DUPLICATING), 4441 mWaitTimeMs(UINT_MAX) 4442{ 4443 addOutputTrack(mainThread); 4444} 4445 4446AudioFlinger::DuplicatingThread::~DuplicatingThread() 4447{ 4448 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4449 mOutputTracks[i]->destroy(); 4450 } 4451} 4452 4453void AudioFlinger::DuplicatingThread::threadLoop_mix() 4454{ 4455 // mix buffers... 4456 if (outputsReady(outputTracks)) { 4457 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4458 } else { 4459 memset(mSinkBuffer, 0, mSinkBufferSize); 4460 } 4461 sleepTime = 0; 4462 writeFrames = mNormalFrameCount; 4463 mCurrentWriteLength = mSinkBufferSize; 4464 standbyTime = systemTime() + standbyDelay; 4465} 4466 4467void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4468{ 4469 if (sleepTime == 0) { 4470 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4471 sleepTime = activeSleepTime; 4472 } else { 4473 sleepTime = idleSleepTime; 4474 } 4475 } else if (mBytesWritten != 0) { 4476 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4477 writeFrames = mNormalFrameCount; 4478 memset(mSinkBuffer, 0, mSinkBufferSize); 4479 } else { 4480 // flush remaining overflow buffers in output tracks 4481 writeFrames = 0; 4482 } 4483 sleepTime = 0; 4484 } 4485} 4486 4487ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4488{ 4489 for (size_t i = 0; i < outputTracks.size(); i++) { 4490 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4491 // for delivery downstream as needed. This in-place conversion is safe as 4492 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4493 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4494 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4495 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4496 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4497 } 4498 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4499 } 4500 mStandby = false; 4501 return (ssize_t)mSinkBufferSize; 4502} 4503 4504void AudioFlinger::DuplicatingThread::threadLoop_standby() 4505{ 4506 // DuplicatingThread implements standby by stopping all tracks 4507 for (size_t i = 0; i < outputTracks.size(); i++) { 4508 outputTracks[i]->stop(); 4509 } 4510} 4511 4512void AudioFlinger::DuplicatingThread::saveOutputTracks() 4513{ 4514 outputTracks = mOutputTracks; 4515} 4516 4517void AudioFlinger::DuplicatingThread::clearOutputTracks() 4518{ 4519 outputTracks.clear(); 4520} 4521 4522void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4523{ 4524 Mutex::Autolock _l(mLock); 4525 // FIXME explain this formula 4526 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4527 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4528 // due to current usage case and restrictions on the AudioBufferProvider. 4529 // Actual buffer conversion is done in threadLoop_write(). 4530 // 4531 // TODO: This may change in the future, depending on multichannel 4532 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4533 OutputTrack *outputTrack = new OutputTrack(thread, 4534 this, 4535 mSampleRate, 4536 AUDIO_FORMAT_PCM_16_BIT, 4537 mChannelMask, 4538 frameCount, 4539 IPCThreadState::self()->getCallingUid()); 4540 if (outputTrack->cblk() != NULL) { 4541 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4542 mOutputTracks.add(outputTrack); 4543 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4544 updateWaitTime_l(); 4545 } 4546} 4547 4548void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4549{ 4550 Mutex::Autolock _l(mLock); 4551 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4552 if (mOutputTracks[i]->thread() == thread) { 4553 mOutputTracks[i]->destroy(); 4554 mOutputTracks.removeAt(i); 4555 updateWaitTime_l(); 4556 return; 4557 } 4558 } 4559 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4560} 4561 4562// caller must hold mLock 4563void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4564{ 4565 mWaitTimeMs = UINT_MAX; 4566 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4567 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4568 if (strong != 0) { 4569 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4570 if (waitTimeMs < mWaitTimeMs) { 4571 mWaitTimeMs = waitTimeMs; 4572 } 4573 } 4574 } 4575} 4576 4577 4578bool AudioFlinger::DuplicatingThread::outputsReady( 4579 const SortedVector< sp<OutputTrack> > &outputTracks) 4580{ 4581 for (size_t i = 0; i < outputTracks.size(); i++) { 4582 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4583 if (thread == 0) { 4584 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4585 outputTracks[i].get()); 4586 return false; 4587 } 4588 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4589 // see note at standby() declaration 4590 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4591 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4592 thread.get()); 4593 return false; 4594 } 4595 } 4596 return true; 4597} 4598 4599uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4600{ 4601 return (mWaitTimeMs * 1000) / 2; 4602} 4603 4604void AudioFlinger::DuplicatingThread::cacheParameters_l() 4605{ 4606 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4607 updateWaitTime_l(); 4608 4609 MixerThread::cacheParameters_l(); 4610} 4611 4612// ---------------------------------------------------------------------------- 4613// Record 4614// ---------------------------------------------------------------------------- 4615 4616AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4617 AudioStreamIn *input, 4618 audio_io_handle_t id, 4619 audio_devices_t outDevice, 4620 audio_devices_t inDevice 4621#ifdef TEE_SINK 4622 , const sp<NBAIO_Sink>& teeSink 4623#endif 4624 ) : 4625 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4626 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4627 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4628 mRsmpInRear(0) 4629#ifdef TEE_SINK 4630 , mTeeSink(teeSink) 4631#endif 4632 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4633 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4634{ 4635 snprintf(mName, kNameLength, "AudioIn_%X", id); 4636 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4637 4638 readInputParameters_l(); 4639} 4640 4641 4642AudioFlinger::RecordThread::~RecordThread() 4643{ 4644 mAudioFlinger->unregisterWriter(mNBLogWriter); 4645 delete[] mRsmpInBuffer; 4646} 4647 4648void AudioFlinger::RecordThread::onFirstRef() 4649{ 4650 run(mName, PRIORITY_URGENT_AUDIO); 4651} 4652 4653bool AudioFlinger::RecordThread::threadLoop() 4654{ 4655 nsecs_t lastWarning = 0; 4656 4657 inputStandBy(); 4658 4659reacquire_wakelock: 4660 sp<RecordTrack> activeTrack; 4661 int activeTracksGen; 4662 { 4663 Mutex::Autolock _l(mLock); 4664 size_t size = mActiveTracks.size(); 4665 activeTracksGen = mActiveTracksGen; 4666 if (size > 0) { 4667 // FIXME an arbitrary choice 4668 activeTrack = mActiveTracks[0]; 4669 acquireWakeLock_l(activeTrack->uid()); 4670 if (size > 1) { 4671 SortedVector<int> tmp; 4672 for (size_t i = 0; i < size; i++) { 4673 tmp.add(mActiveTracks[i]->uid()); 4674 } 4675 updateWakeLockUids_l(tmp); 4676 } 4677 } else { 4678 acquireWakeLock_l(-1); 4679 } 4680 } 4681 4682 // used to request a deferred sleep, to be executed later while mutex is unlocked 4683 uint32_t sleepUs = 0; 4684 4685 // loop while there is work to do 4686 for (;;) { 4687 Vector< sp<EffectChain> > effectChains; 4688 4689 // sleep with mutex unlocked 4690 if (sleepUs > 0) { 4691 usleep(sleepUs); 4692 sleepUs = 0; 4693 } 4694 4695 // activeTracks accumulates a copy of a subset of mActiveTracks 4696 Vector< sp<RecordTrack> > activeTracks; 4697 4698 DefaultKeyedVector< pid_t,sp<NotificationClient> > notificationClients = 4699 mAudioFlinger->notificationClients(); 4700 4701 { // scope for mLock 4702 Mutex::Autolock _l(mLock); 4703 4704 processConfigEvents_l(notificationClients); 4705 notificationClients.clear(); 4706 4707 // check exitPending here because checkForNewParameters_l() and 4708 // checkForNewParameters_l() can temporarily release mLock 4709 if (exitPending()) { 4710 break; 4711 } 4712 4713 // if no active track(s), then standby and release wakelock 4714 size_t size = mActiveTracks.size(); 4715 if (size == 0) { 4716 standbyIfNotAlreadyInStandby(); 4717 // exitPending() can't become true here 4718 releaseWakeLock_l(); 4719 ALOGV("RecordThread: loop stopping"); 4720 // go to sleep 4721 mWaitWorkCV.wait(mLock); 4722 ALOGV("RecordThread: loop starting"); 4723 goto reacquire_wakelock; 4724 } 4725 4726 if (mActiveTracksGen != activeTracksGen) { 4727 activeTracksGen = mActiveTracksGen; 4728 SortedVector<int> tmp; 4729 for (size_t i = 0; i < size; i++) { 4730 tmp.add(mActiveTracks[i]->uid()); 4731 } 4732 updateWakeLockUids_l(tmp); 4733 } 4734 4735 bool doBroadcast = false; 4736 for (size_t i = 0; i < size; ) { 4737 4738 activeTrack = mActiveTracks[i]; 4739 if (activeTrack->isTerminated()) { 4740 removeTrack_l(activeTrack); 4741 mActiveTracks.remove(activeTrack); 4742 mActiveTracksGen++; 4743 size--; 4744 continue; 4745 } 4746 4747 TrackBase::track_state activeTrackState = activeTrack->mState; 4748 switch (activeTrackState) { 4749 4750 case TrackBase::PAUSING: 4751 mActiveTracks.remove(activeTrack); 4752 mActiveTracksGen++; 4753 doBroadcast = true; 4754 size--; 4755 continue; 4756 4757 case TrackBase::STARTING_1: 4758 sleepUs = 10000; 4759 i++; 4760 continue; 4761 4762 case TrackBase::STARTING_2: 4763 doBroadcast = true; 4764 mStandby = false; 4765 activeTrack->mState = TrackBase::ACTIVE; 4766 break; 4767 4768 case TrackBase::ACTIVE: 4769 break; 4770 4771 case TrackBase::IDLE: 4772 i++; 4773 continue; 4774 4775 default: 4776 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4777 } 4778 4779 activeTracks.add(activeTrack); 4780 i++; 4781 4782 } 4783 if (doBroadcast) { 4784 mStartStopCond.broadcast(); 4785 } 4786 4787 // sleep if there are no active tracks to process 4788 if (activeTracks.size() == 0) { 4789 if (sleepUs == 0) { 4790 sleepUs = kRecordThreadSleepUs; 4791 } 4792 continue; 4793 } 4794 sleepUs = 0; 4795 4796 lockEffectChains_l(effectChains); 4797 } 4798 4799 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4800 4801 size_t size = effectChains.size(); 4802 for (size_t i = 0; i < size; i++) { 4803 // thread mutex is not locked, but effect chain is locked 4804 effectChains[i]->process_l(); 4805 } 4806 4807 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4808 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4809 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4810 // If destination is non-contiguous, first read past the nominal end of buffer, then 4811 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4812 4813 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4814 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4815 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4816 if (bytesRead <= 0) { 4817 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4818 // Force input into standby so that it tries to recover at next read attempt 4819 inputStandBy(); 4820 sleepUs = kRecordThreadSleepUs; 4821 continue; 4822 } 4823 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4824 size_t framesRead = bytesRead / mFrameSize; 4825 ALOG_ASSERT(framesRead > 0); 4826 if (mTeeSink != 0) { 4827 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4828 } 4829 // If destination is non-contiguous, we now correct for reading past end of buffer. 4830 size_t part1 = mRsmpInFramesP2 - rear; 4831 if (framesRead > part1) { 4832 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4833 (framesRead - part1) * mFrameSize); 4834 } 4835 rear = mRsmpInRear += framesRead; 4836 4837 size = activeTracks.size(); 4838 // loop over each active track 4839 for (size_t i = 0; i < size; i++) { 4840 activeTrack = activeTracks[i]; 4841 4842 enum { 4843 OVERRUN_UNKNOWN, 4844 OVERRUN_TRUE, 4845 OVERRUN_FALSE 4846 } overrun = OVERRUN_UNKNOWN; 4847 4848 // loop over getNextBuffer to handle circular sink 4849 for (;;) { 4850 4851 activeTrack->mSink.frameCount = ~0; 4852 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4853 size_t framesOut = activeTrack->mSink.frameCount; 4854 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4855 4856 int32_t front = activeTrack->mRsmpInFront; 4857 ssize_t filled = rear - front; 4858 size_t framesIn; 4859 4860 if (filled < 0) { 4861 // should not happen, but treat like a massive overrun and re-sync 4862 framesIn = 0; 4863 activeTrack->mRsmpInFront = rear; 4864 overrun = OVERRUN_TRUE; 4865 } else if ((size_t) filled <= mRsmpInFrames) { 4866 framesIn = (size_t) filled; 4867 } else { 4868 // client is not keeping up with server, but give it latest data 4869 framesIn = mRsmpInFrames; 4870 activeTrack->mRsmpInFront = front = rear - framesIn; 4871 overrun = OVERRUN_TRUE; 4872 } 4873 4874 if (framesOut == 0 || framesIn == 0) { 4875 break; 4876 } 4877 4878 if (activeTrack->mResampler == NULL) { 4879 // no resampling 4880 if (framesIn > framesOut) { 4881 framesIn = framesOut; 4882 } else { 4883 framesOut = framesIn; 4884 } 4885 int8_t *dst = activeTrack->mSink.i8; 4886 while (framesIn > 0) { 4887 front &= mRsmpInFramesP2 - 1; 4888 size_t part1 = mRsmpInFramesP2 - front; 4889 if (part1 > framesIn) { 4890 part1 = framesIn; 4891 } 4892 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4893 if (mChannelCount == activeTrack->mChannelCount) { 4894 memcpy(dst, src, part1 * mFrameSize); 4895 } else if (mChannelCount == 1) { 4896 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4897 part1); 4898 } else { 4899 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4900 part1); 4901 } 4902 dst += part1 * activeTrack->mFrameSize; 4903 front += part1; 4904 framesIn -= part1; 4905 } 4906 activeTrack->mRsmpInFront += framesOut; 4907 4908 } else { 4909 // resampling 4910 // FIXME framesInNeeded should really be part of resampler API, and should 4911 // depend on the SRC ratio 4912 // to keep mRsmpInBuffer full so resampler always has sufficient input 4913 size_t framesInNeeded; 4914 // FIXME only re-calculate when it changes, and optimize for common ratios 4915 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4916 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4917 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4918 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4919 framesInNeeded, framesOut, inOverOut); 4920 // Although we theoretically have framesIn in circular buffer, some of those are 4921 // unreleased frames, and thus must be discounted for purpose of budgeting. 4922 size_t unreleased = activeTrack->mRsmpInUnrel; 4923 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4924 if (framesIn < framesInNeeded) { 4925 ALOGV("not enough to resample: have %u frames in but need %u in to " 4926 "produce %u out given in/out ratio of %.4g", 4927 framesIn, framesInNeeded, framesOut, inOverOut); 4928 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4929 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4930 if (newFramesOut == 0) { 4931 break; 4932 } 4933 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4934 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4935 framesInNeeded, newFramesOut, outOverIn); 4936 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4937 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4938 "given in/out ratio of %.4g", 4939 framesIn, framesInNeeded, newFramesOut, inOverOut); 4940 framesOut = newFramesOut; 4941 } else { 4942 ALOGV("success 1: have %u in and need %u in to produce %u out " 4943 "given in/out ratio of %.4g", 4944 framesIn, framesInNeeded, framesOut, inOverOut); 4945 } 4946 4947 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4948 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4949 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4950 delete[] activeTrack->mRsmpOutBuffer; 4951 // resampler always outputs stereo 4952 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4953 activeTrack->mRsmpOutFrameCount = framesOut; 4954 } 4955 4956 // resampler accumulates, but we only have one source track 4957 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4958 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4959 // FIXME how about having activeTrack implement this interface itself? 4960 activeTrack->mResamplerBufferProvider 4961 /*this*/ /* AudioBufferProvider* */); 4962 // ditherAndClamp() works as long as all buffers returned by 4963 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4964 if (activeTrack->mChannelCount == 1) { 4965 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 4966 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4967 framesOut); 4968 // the resampler always outputs stereo samples: 4969 // do post stereo to mono conversion 4970 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4971 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4972 } else { 4973 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4974 activeTrack->mRsmpOutBuffer, framesOut); 4975 } 4976 // now done with mRsmpOutBuffer 4977 4978 } 4979 4980 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4981 overrun = OVERRUN_FALSE; 4982 } 4983 4984 if (activeTrack->mFramesToDrop == 0) { 4985 if (framesOut > 0) { 4986 activeTrack->mSink.frameCount = framesOut; 4987 activeTrack->releaseBuffer(&activeTrack->mSink); 4988 } 4989 } else { 4990 // FIXME could do a partial drop of framesOut 4991 if (activeTrack->mFramesToDrop > 0) { 4992 activeTrack->mFramesToDrop -= framesOut; 4993 if (activeTrack->mFramesToDrop <= 0) { 4994 activeTrack->clearSyncStartEvent(); 4995 } 4996 } else { 4997 activeTrack->mFramesToDrop += framesOut; 4998 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4999 activeTrack->mSyncStartEvent->isCancelled()) { 5000 ALOGW("Synced record %s, session %d, trigger session %d", 5001 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5002 activeTrack->sessionId(), 5003 (activeTrack->mSyncStartEvent != 0) ? 5004 activeTrack->mSyncStartEvent->triggerSession() : 0); 5005 activeTrack->clearSyncStartEvent(); 5006 } 5007 } 5008 } 5009 5010 if (framesOut == 0) { 5011 break; 5012 } 5013 } 5014 5015 switch (overrun) { 5016 case OVERRUN_TRUE: 5017 // client isn't retrieving buffers fast enough 5018 if (!activeTrack->setOverflow()) { 5019 nsecs_t now = systemTime(); 5020 // FIXME should lastWarning per track? 5021 if ((now - lastWarning) > kWarningThrottleNs) { 5022 ALOGW("RecordThread: buffer overflow"); 5023 lastWarning = now; 5024 } 5025 } 5026 break; 5027 case OVERRUN_FALSE: 5028 activeTrack->clearOverflow(); 5029 break; 5030 case OVERRUN_UNKNOWN: 5031 break; 5032 } 5033 5034 } 5035 5036 // enable changes in effect chain 5037 unlockEffectChains(effectChains); 5038 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5039 } 5040 5041 standbyIfNotAlreadyInStandby(); 5042 5043 { 5044 Mutex::Autolock _l(mLock); 5045 for (size_t i = 0; i < mTracks.size(); i++) { 5046 sp<RecordTrack> track = mTracks[i]; 5047 track->invalidate(); 5048 } 5049 mActiveTracks.clear(); 5050 mActiveTracksGen++; 5051 mStartStopCond.broadcast(); 5052 } 5053 5054 releaseWakeLock(); 5055 5056 ALOGV("RecordThread %p exiting", this); 5057 return false; 5058} 5059 5060void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5061{ 5062 if (!mStandby) { 5063 inputStandBy(); 5064 mStandby = true; 5065 } 5066} 5067 5068void AudioFlinger::RecordThread::inputStandBy() 5069{ 5070 mInput->stream->common.standby(&mInput->stream->common); 5071} 5072 5073// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5074sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5075 const sp<AudioFlinger::Client>& client, 5076 uint32_t sampleRate, 5077 audio_format_t format, 5078 audio_channel_mask_t channelMask, 5079 size_t *pFrameCount, 5080 int sessionId, 5081 int uid, 5082 IAudioFlinger::track_flags_t *flags, 5083 pid_t tid, 5084 status_t *status) 5085{ 5086 size_t frameCount = *pFrameCount; 5087 sp<RecordTrack> track; 5088 status_t lStatus; 5089 5090 // client expresses a preference for FAST, but we get the final say 5091 if (*flags & IAudioFlinger::TRACK_FAST) { 5092 if ( 5093 // use case: callback handler and frame count is default or at least as large as HAL 5094 ( 5095 (tid != -1) && 5096 ((frameCount == 0) || 5097 // FIXME not necessarily true, should be native frame count for native SR! 5098 (frameCount >= mFrameCount)) 5099 ) && 5100 // PCM data 5101 audio_is_linear_pcm(format) && 5102 // mono or stereo 5103 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5104 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5105 // hardware sample rate 5106 // FIXME actually the native hardware sample rate 5107 (sampleRate == mSampleRate) && 5108 // record thread has an associated fast capture 5109 hasFastCapture() 5110 // fast capture does not require slots 5111 ) { 5112 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5113 if (frameCount == 0) { 5114 // FIXME wrong mFrameCount 5115 frameCount = mFrameCount * kFastTrackMultiplier; 5116 } 5117 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5118 frameCount, mFrameCount); 5119 } else { 5120 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5121 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5122 "hasFastCapture=%d tid=%d", 5123 frameCount, mFrameCount, format, 5124 audio_is_linear_pcm(format), 5125 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5126 *flags &= ~IAudioFlinger::TRACK_FAST; 5127 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5128 // For compatibility with AudioRecord calculation, buffer depth is forced 5129 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5130 // This is probably too conservative, but legacy application code may depend on it. 5131 // If you change this calculation, also review the start threshold which is related. 5132 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5133 size_t mNormalFrameCount = 2048; // FIXME 5134 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5135 if (minBufCount < 2) { 5136 minBufCount = 2; 5137 } 5138 size_t minFrameCount = mNormalFrameCount * minBufCount; 5139 if (frameCount < minFrameCount) { 5140 frameCount = minFrameCount; 5141 } 5142 } 5143 } 5144 *pFrameCount = frameCount; 5145 5146 lStatus = initCheck(); 5147 if (lStatus != NO_ERROR) { 5148 ALOGE("createRecordTrack_l() audio driver not initialized"); 5149 goto Exit; 5150 } 5151 5152 { // scope for mLock 5153 Mutex::Autolock _l(mLock); 5154 5155 track = new RecordTrack(this, client, sampleRate, 5156 format, channelMask, frameCount, sessionId, uid, 5157 *flags); 5158 5159 lStatus = track->initCheck(); 5160 if (lStatus != NO_ERROR) { 5161 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5162 // track must be cleared from the caller as the caller has the AF lock 5163 goto Exit; 5164 } 5165 mTracks.add(track); 5166 5167 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5168 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5169 mAudioFlinger->btNrecIsOff(); 5170 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5171 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5172 5173 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5174 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5175 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5176 // so ask activity manager to do this on our behalf 5177 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5178 } 5179 } 5180 5181 lStatus = NO_ERROR; 5182 5183Exit: 5184 *status = lStatus; 5185 return track; 5186} 5187 5188status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5189 AudioSystem::sync_event_t event, 5190 int triggerSession) 5191{ 5192 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5193 sp<ThreadBase> strongMe = this; 5194 status_t status = NO_ERROR; 5195 5196 if (event == AudioSystem::SYNC_EVENT_NONE) { 5197 recordTrack->clearSyncStartEvent(); 5198 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5199 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5200 triggerSession, 5201 recordTrack->sessionId(), 5202 syncStartEventCallback, 5203 recordTrack); 5204 // Sync event can be cancelled by the trigger session if the track is not in a 5205 // compatible state in which case we start record immediately 5206 if (recordTrack->mSyncStartEvent->isCancelled()) { 5207 recordTrack->clearSyncStartEvent(); 5208 } else { 5209 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5210 recordTrack->mFramesToDrop = - 5211 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5212 } 5213 } 5214 5215 { 5216 // This section is a rendezvous between binder thread executing start() and RecordThread 5217 AutoMutex lock(mLock); 5218 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5219 if (recordTrack->mState == TrackBase::PAUSING) { 5220 ALOGV("active record track PAUSING -> ACTIVE"); 5221 recordTrack->mState = TrackBase::ACTIVE; 5222 } else { 5223 ALOGV("active record track state %d", recordTrack->mState); 5224 } 5225 return status; 5226 } 5227 5228 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5229 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5230 // or using a separate command thread 5231 recordTrack->mState = TrackBase::STARTING_1; 5232 mActiveTracks.add(recordTrack); 5233 mActiveTracksGen++; 5234 mLock.unlock(); 5235 status_t status = AudioSystem::startInput(mId); 5236 mLock.lock(); 5237 // FIXME should verify that recordTrack is still in mActiveTracks 5238 if (status != NO_ERROR) { 5239 mActiveTracks.remove(recordTrack); 5240 mActiveTracksGen++; 5241 recordTrack->clearSyncStartEvent(); 5242 return status; 5243 } 5244 // Catch up with current buffer indices if thread is already running. 5245 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5246 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5247 // see previously buffered data before it called start(), but with greater risk of overrun. 5248 5249 recordTrack->mRsmpInFront = mRsmpInRear; 5250 recordTrack->mRsmpInUnrel = 0; 5251 // FIXME why reset? 5252 if (recordTrack->mResampler != NULL) { 5253 recordTrack->mResampler->reset(); 5254 } 5255 recordTrack->mState = TrackBase::STARTING_2; 5256 // signal thread to start 5257 mWaitWorkCV.broadcast(); 5258 if (mActiveTracks.indexOf(recordTrack) < 0) { 5259 ALOGV("Record failed to start"); 5260 status = BAD_VALUE; 5261 goto startError; 5262 } 5263 return status; 5264 } 5265 5266startError: 5267 AudioSystem::stopInput(mId); 5268 recordTrack->clearSyncStartEvent(); 5269 // FIXME I wonder why we do not reset the state here? 5270 return status; 5271} 5272 5273void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5274{ 5275 sp<SyncEvent> strongEvent = event.promote(); 5276 5277 if (strongEvent != 0) { 5278 sp<RefBase> ptr = strongEvent->cookie().promote(); 5279 if (ptr != 0) { 5280 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5281 recordTrack->handleSyncStartEvent(strongEvent); 5282 } 5283 } 5284} 5285 5286bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5287 ALOGV("RecordThread::stop"); 5288 AutoMutex _l(mLock); 5289 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5290 return false; 5291 } 5292 // note that threadLoop may still be processing the track at this point [without lock] 5293 recordTrack->mState = TrackBase::PAUSING; 5294 // do not wait for mStartStopCond if exiting 5295 if (exitPending()) { 5296 return true; 5297 } 5298 // FIXME incorrect usage of wait: no explicit predicate or loop 5299 mStartStopCond.wait(mLock); 5300 // if we have been restarted, recordTrack is in mActiveTracks here 5301 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5302 ALOGV("Record stopped OK"); 5303 return true; 5304 } 5305 return false; 5306} 5307 5308bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5309{ 5310 return false; 5311} 5312 5313status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5314{ 5315#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5316 if (!isValidSyncEvent(event)) { 5317 return BAD_VALUE; 5318 } 5319 5320 int eventSession = event->triggerSession(); 5321 status_t ret = NAME_NOT_FOUND; 5322 5323 Mutex::Autolock _l(mLock); 5324 5325 for (size_t i = 0; i < mTracks.size(); i++) { 5326 sp<RecordTrack> track = mTracks[i]; 5327 if (eventSession == track->sessionId()) { 5328 (void) track->setSyncEvent(event); 5329 ret = NO_ERROR; 5330 } 5331 } 5332 return ret; 5333#else 5334 return BAD_VALUE; 5335#endif 5336} 5337 5338// destroyTrack_l() must be called with ThreadBase::mLock held 5339void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5340{ 5341 track->terminate(); 5342 track->mState = TrackBase::STOPPED; 5343 // active tracks are removed by threadLoop() 5344 if (mActiveTracks.indexOf(track) < 0) { 5345 removeTrack_l(track); 5346 } 5347} 5348 5349void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5350{ 5351 mTracks.remove(track); 5352 // need anything related to effects here? 5353} 5354 5355void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5356{ 5357 dumpInternals(fd, args); 5358 dumpTracks(fd, args); 5359 dumpEffectChains(fd, args); 5360} 5361 5362void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5363{ 5364 fdprintf(fd, "\nInput thread %p:\n", this); 5365 5366 if (mActiveTracks.size() > 0) { 5367 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5368 } else { 5369 fdprintf(fd, " No active record clients\n"); 5370 } 5371 5372 dumpBase(fd, args); 5373} 5374 5375void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5376{ 5377 const size_t SIZE = 256; 5378 char buffer[SIZE]; 5379 String8 result; 5380 5381 size_t numtracks = mTracks.size(); 5382 size_t numactive = mActiveTracks.size(); 5383 size_t numactiveseen = 0; 5384 fdprintf(fd, " %d Tracks", numtracks); 5385 if (numtracks) { 5386 fdprintf(fd, " of which %d are active\n", numactive); 5387 RecordTrack::appendDumpHeader(result); 5388 for (size_t i = 0; i < numtracks ; ++i) { 5389 sp<RecordTrack> track = mTracks[i]; 5390 if (track != 0) { 5391 bool active = mActiveTracks.indexOf(track) >= 0; 5392 if (active) { 5393 numactiveseen++; 5394 } 5395 track->dump(buffer, SIZE, active); 5396 result.append(buffer); 5397 } 5398 } 5399 } else { 5400 fdprintf(fd, "\n"); 5401 } 5402 5403 if (numactiveseen != numactive) { 5404 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5405 " not in the track list\n"); 5406 result.append(buffer); 5407 RecordTrack::appendDumpHeader(result); 5408 for (size_t i = 0; i < numactive; ++i) { 5409 sp<RecordTrack> track = mActiveTracks[i]; 5410 if (mTracks.indexOf(track) < 0) { 5411 track->dump(buffer, SIZE, true); 5412 result.append(buffer); 5413 } 5414 } 5415 5416 } 5417 write(fd, result.string(), result.size()); 5418} 5419 5420// AudioBufferProvider interface 5421status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5422 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5423{ 5424 RecordTrack *activeTrack = mRecordTrack; 5425 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5426 if (threadBase == 0) { 5427 buffer->frameCount = 0; 5428 buffer->raw = NULL; 5429 return NOT_ENOUGH_DATA; 5430 } 5431 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5432 int32_t rear = recordThread->mRsmpInRear; 5433 int32_t front = activeTrack->mRsmpInFront; 5434 ssize_t filled = rear - front; 5435 // FIXME should not be P2 (don't want to increase latency) 5436 // FIXME if client not keeping up, discard 5437 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5438 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5439 front &= recordThread->mRsmpInFramesP2 - 1; 5440 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5441 if (part1 > (size_t) filled) { 5442 part1 = filled; 5443 } 5444 size_t ask = buffer->frameCount; 5445 ALOG_ASSERT(ask > 0); 5446 if (part1 > ask) { 5447 part1 = ask; 5448 } 5449 if (part1 == 0) { 5450 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5451 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5452 buffer->raw = NULL; 5453 buffer->frameCount = 0; 5454 activeTrack->mRsmpInUnrel = 0; 5455 return NOT_ENOUGH_DATA; 5456 } 5457 5458 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5459 buffer->frameCount = part1; 5460 activeTrack->mRsmpInUnrel = part1; 5461 return NO_ERROR; 5462} 5463 5464// AudioBufferProvider interface 5465void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5466 AudioBufferProvider::Buffer* buffer) 5467{ 5468 RecordTrack *activeTrack = mRecordTrack; 5469 size_t stepCount = buffer->frameCount; 5470 if (stepCount == 0) { 5471 return; 5472 } 5473 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5474 activeTrack->mRsmpInUnrel -= stepCount; 5475 activeTrack->mRsmpInFront += stepCount; 5476 buffer->raw = NULL; 5477 buffer->frameCount = 0; 5478} 5479 5480bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5481 status_t& status) 5482{ 5483 bool reconfig = false; 5484 5485 status = NO_ERROR; 5486 5487 audio_format_t reqFormat = mFormat; 5488 uint32_t samplingRate = mSampleRate; 5489 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5490 5491 AudioParameter param = AudioParameter(keyValuePair); 5492 int value; 5493 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5494 // channel count change can be requested. Do we mandate the first client defines the 5495 // HAL sampling rate and channel count or do we allow changes on the fly? 5496 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5497 samplingRate = value; 5498 reconfig = true; 5499 } 5500 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5501 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5502 status = BAD_VALUE; 5503 } else { 5504 reqFormat = (audio_format_t) value; 5505 reconfig = true; 5506 } 5507 } 5508 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5509 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5510 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5511 status = BAD_VALUE; 5512 } else { 5513 channelMask = mask; 5514 reconfig = true; 5515 } 5516 } 5517 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5518 // do not accept frame count changes if tracks are open as the track buffer 5519 // size depends on frame count and correct behavior would not be guaranteed 5520 // if frame count is changed after track creation 5521 if (mActiveTracks.size() > 0) { 5522 status = INVALID_OPERATION; 5523 } else { 5524 reconfig = true; 5525 } 5526 } 5527 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5528 // forward device change to effects that have requested to be 5529 // aware of attached audio device. 5530 for (size_t i = 0; i < mEffectChains.size(); i++) { 5531 mEffectChains[i]->setDevice_l(value); 5532 } 5533 5534 // store input device and output device but do not forward output device to audio HAL. 5535 // Note that status is ignored by the caller for output device 5536 // (see AudioFlinger::setParameters() 5537 if (audio_is_output_devices(value)) { 5538 mOutDevice = value; 5539 status = BAD_VALUE; 5540 } else { 5541 mInDevice = value; 5542 // disable AEC and NS if the device is a BT SCO headset supporting those 5543 // pre processings 5544 if (mTracks.size() > 0) { 5545 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5546 mAudioFlinger->btNrecIsOff(); 5547 for (size_t i = 0; i < mTracks.size(); i++) { 5548 sp<RecordTrack> track = mTracks[i]; 5549 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5550 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5551 } 5552 } 5553 } 5554 } 5555 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5556 mAudioSource != (audio_source_t)value) { 5557 // forward device change to effects that have requested to be 5558 // aware of attached audio device. 5559 for (size_t i = 0; i < mEffectChains.size(); i++) { 5560 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5561 } 5562 mAudioSource = (audio_source_t)value; 5563 } 5564 5565 if (status == NO_ERROR) { 5566 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5567 keyValuePair.string()); 5568 if (status == INVALID_OPERATION) { 5569 inputStandBy(); 5570 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5571 keyValuePair.string()); 5572 } 5573 if (reconfig) { 5574 if (status == BAD_VALUE && 5575 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5576 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5577 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5578 <= (2 * samplingRate)) && 5579 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5580 <= FCC_2 && 5581 (channelMask == AUDIO_CHANNEL_IN_MONO || 5582 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5583 status = NO_ERROR; 5584 } 5585 if (status == NO_ERROR) { 5586 readInputParameters_l(); 5587 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5588 } 5589 } 5590 } 5591 5592 return reconfig; 5593} 5594 5595String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5596{ 5597 Mutex::Autolock _l(mLock); 5598 if (initCheck() != NO_ERROR) { 5599 return String8(); 5600 } 5601 5602 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5603 const String8 out_s8(s); 5604 free(s); 5605 return out_s8; 5606} 5607 5608void AudioFlinger::RecordThread::audioConfigChanged_l( 5609 const DefaultKeyedVector< pid_t,sp<NotificationClient> >& notificationClients, 5610 int event, 5611 int param __unused) { 5612 AudioSystem::OutputDescriptor desc; 5613 const void *param2 = NULL; 5614 5615 switch (event) { 5616 case AudioSystem::INPUT_OPENED: 5617 case AudioSystem::INPUT_CONFIG_CHANGED: 5618 desc.channelMask = mChannelMask; 5619 desc.samplingRate = mSampleRate; 5620 desc.format = mFormat; 5621 desc.frameCount = mFrameCount; 5622 desc.latency = 0; 5623 param2 = &desc; 5624 break; 5625 5626 case AudioSystem::INPUT_CLOSED: 5627 default: 5628 break; 5629 } 5630 mAudioFlinger->audioConfigChanged_l(notificationClients, event, mId, param2); 5631} 5632 5633void AudioFlinger::RecordThread::readInputParameters_l() 5634{ 5635 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5636 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5637 mChannelCount = popcount(mChannelMask); 5638 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5639 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5640 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5641 } 5642 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5643 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5644 mFrameCount = mBufferSize / mFrameSize; 5645 // This is the formula for calculating the temporary buffer size. 5646 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5647 // 1 full output buffer, regardless of the alignment of the available input. 5648 // The value is somewhat arbitrary, and could probably be even larger. 5649 // A larger value should allow more old data to be read after a track calls start(), 5650 // without increasing latency. 5651 mRsmpInFrames = mFrameCount * 7; 5652 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5653 delete[] mRsmpInBuffer; 5654 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5655 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5656 5657 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5658 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5659} 5660 5661uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5662{ 5663 Mutex::Autolock _l(mLock); 5664 if (initCheck() != NO_ERROR) { 5665 return 0; 5666 } 5667 5668 return mInput->stream->get_input_frames_lost(mInput->stream); 5669} 5670 5671uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5672{ 5673 Mutex::Autolock _l(mLock); 5674 uint32_t result = 0; 5675 if (getEffectChain_l(sessionId) != 0) { 5676 result = EFFECT_SESSION; 5677 } 5678 5679 for (size_t i = 0; i < mTracks.size(); ++i) { 5680 if (sessionId == mTracks[i]->sessionId()) { 5681 result |= TRACK_SESSION; 5682 break; 5683 } 5684 } 5685 5686 return result; 5687} 5688 5689KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5690{ 5691 KeyedVector<int, bool> ids; 5692 Mutex::Autolock _l(mLock); 5693 for (size_t j = 0; j < mTracks.size(); ++j) { 5694 sp<RecordThread::RecordTrack> track = mTracks[j]; 5695 int sessionId = track->sessionId(); 5696 if (ids.indexOfKey(sessionId) < 0) { 5697 ids.add(sessionId, true); 5698 } 5699 } 5700 return ids; 5701} 5702 5703AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5704{ 5705 Mutex::Autolock _l(mLock); 5706 AudioStreamIn *input = mInput; 5707 mInput = NULL; 5708 return input; 5709} 5710 5711// this method must always be called either with ThreadBase mLock held or inside the thread loop 5712audio_stream_t* AudioFlinger::RecordThread::stream() const 5713{ 5714 if (mInput == NULL) { 5715 return NULL; 5716 } 5717 return &mInput->stream->common; 5718} 5719 5720status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5721{ 5722 // only one chain per input thread 5723 if (mEffectChains.size() != 0) { 5724 return INVALID_OPERATION; 5725 } 5726 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5727 5728 chain->setInBuffer(NULL); 5729 chain->setOutBuffer(NULL); 5730 5731 checkSuspendOnAddEffectChain_l(chain); 5732 5733 mEffectChains.add(chain); 5734 5735 return NO_ERROR; 5736} 5737 5738size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5739{ 5740 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5741 ALOGW_IF(mEffectChains.size() != 1, 5742 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5743 chain.get(), mEffectChains.size(), this); 5744 if (mEffectChains.size() == 1) { 5745 mEffectChains.removeAt(0); 5746 } 5747 return 0; 5748} 5749 5750}; // namespace android 5751