Threads.cpp revision 021cf9634ab09c0753a40b7c9ef4ba603be5c3da
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38 39// NBAIO implementations 40#include <media/nbaio/AudioStreamOutSink.h> 41#include <media/nbaio/MonoPipe.h> 42#include <media/nbaio/MonoPipeReader.h> 43#include <media/nbaio/Pipe.h> 44#include <media/nbaio/PipeReader.h> 45#include <media/nbaio/SourceAudioBufferProvider.h> 46 47#include <powermanager/PowerManager.h> 48 49#include <common_time/cc_helper.h> 50#include <common_time/local_clock.h> 51 52#include "AudioFlinger.h" 53#include "AudioMixer.h" 54#include "FastMixer.h" 55#include "ServiceUtilities.h" 56#include "SchedulingPolicyService.h" 57 58#ifdef ADD_BATTERY_DATA 59#include <media/IMediaPlayerService.h> 60#include <media/IMediaDeathNotifier.h> 61#endif 62 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85// retry counts for buffer fill timeout 86// 50 * ~20msecs = 1 second 87static const int8_t kMaxTrackRetries = 50; 88static const int8_t kMaxTrackStartupRetries = 50; 89// allow less retry attempts on direct output thread. 90// direct outputs can be a scarce resource in audio hardware and should 91// be released as quickly as possible. 92static const int8_t kMaxTrackRetriesDirect = 2; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait in sendConfigEvent_l() for a status to be received 101static const nsecs_t kConfigEventTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108// minimum normal sink buffer size, expressed in milliseconds rather than frames 109static const uint32_t kMinNormalSinkBufferSizeMs = 20; 110// maximum normal sink buffer size 111static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 112 113// Offloaded output thread standby delay: allows track transition without going to standby 114static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 115 116// Whether to use fast mixer 117static const enum { 118 FastMixer_Never, // never initialize or use: for debugging only 119 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 120 // normal mixer multiplier is 1 121 FastMixer_Static, // initialize if needed, then use all the time if initialized, 122 // multiplier is calculated based on min & max normal mixer buffer size 123 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 124 // multiplier is calculated based on min & max normal mixer buffer size 125 // FIXME for FastMixer_Dynamic: 126 // Supporting this option will require fixing HALs that can't handle large writes. 127 // For example, one HAL implementation returns an error from a large write, 128 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 129 // We could either fix the HAL implementations, or provide a wrapper that breaks 130 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 131} kUseFastMixer = FastMixer_Static; 132 133// Priorities for requestPriority 134static const int kPriorityAudioApp = 2; 135static const int kPriorityFastMixer = 3; 136 137// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 138// for the track. The client then sub-divides this into smaller buffers for its use. 139// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 140// So for now we just assume that client is double-buffered for fast tracks. 141// FIXME It would be better for client to tell AudioFlinger the value of N, 142// so AudioFlinger could allocate the right amount of memory. 143// See the client's minBufCount and mNotificationFramesAct calculations for details. 144static const int kFastTrackMultiplier = 2; 145 146// See Thread::readOnlyHeap(). 147// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 148// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 149// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 150static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 151 152// ---------------------------------------------------------------------------- 153 154#ifdef ADD_BATTERY_DATA 155// To collect the amplifier usage 156static void addBatteryData(uint32_t params) { 157 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 158 if (service == NULL) { 159 // it already logged 160 return; 161 } 162 163 service->addBatteryData(params); 164} 165#endif 166 167 168// ---------------------------------------------------------------------------- 169// CPU Stats 170// ---------------------------------------------------------------------------- 171 172class CpuStats { 173public: 174 CpuStats(); 175 void sample(const String8 &title); 176#ifdef DEBUG_CPU_USAGE 177private: 178 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 179 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 180 181 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 182 183 int mCpuNum; // thread's current CPU number 184 int mCpukHz; // frequency of thread's current CPU in kHz 185#endif 186}; 187 188CpuStats::CpuStats() 189#ifdef DEBUG_CPU_USAGE 190 : mCpuNum(-1), mCpukHz(-1) 191#endif 192{ 193} 194 195void CpuStats::sample(const String8 &title 196#ifndef DEBUG_CPU_USAGE 197 __unused 198#endif 199 ) { 200#ifdef DEBUG_CPU_USAGE 201 // get current thread's delta CPU time in wall clock ns 202 double wcNs; 203 bool valid = mCpuUsage.sampleAndEnable(wcNs); 204 205 // record sample for wall clock statistics 206 if (valid) { 207 mWcStats.sample(wcNs); 208 } 209 210 // get the current CPU number 211 int cpuNum = sched_getcpu(); 212 213 // get the current CPU frequency in kHz 214 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 215 216 // check if either CPU number or frequency changed 217 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 218 mCpuNum = cpuNum; 219 mCpukHz = cpukHz; 220 // ignore sample for purposes of cycles 221 valid = false; 222 } 223 224 // if no change in CPU number or frequency, then record sample for cycle statistics 225 if (valid && mCpukHz > 0) { 226 double cycles = wcNs * cpukHz * 0.000001; 227 mHzStats.sample(cycles); 228 } 229 230 unsigned n = mWcStats.n(); 231 // mCpuUsage.elapsed() is expensive, so don't call it every loop 232 if ((n & 127) == 1) { 233 long long elapsed = mCpuUsage.elapsed(); 234 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 235 double perLoop = elapsed / (double) n; 236 double perLoop100 = perLoop * 0.01; 237 double perLoop1k = perLoop * 0.001; 238 double mean = mWcStats.mean(); 239 double stddev = mWcStats.stddev(); 240 double minimum = mWcStats.minimum(); 241 double maximum = mWcStats.maximum(); 242 double meanCycles = mHzStats.mean(); 243 double stddevCycles = mHzStats.stddev(); 244 double minCycles = mHzStats.minimum(); 245 double maxCycles = mHzStats.maximum(); 246 mCpuUsage.resetElapsed(); 247 mWcStats.reset(); 248 mHzStats.reset(); 249 ALOGD("CPU usage for %s over past %.1f secs\n" 250 " (%u mixer loops at %.1f mean ms per loop):\n" 251 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 252 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 253 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 254 title.string(), 255 elapsed * .000000001, n, perLoop * .000001, 256 mean * .001, 257 stddev * .001, 258 minimum * .001, 259 maximum * .001, 260 mean / perLoop100, 261 stddev / perLoop100, 262 minimum / perLoop100, 263 maximum / perLoop100, 264 meanCycles / perLoop1k, 265 stddevCycles / perLoop1k, 266 minCycles / perLoop1k, 267 maxCycles / perLoop1k); 268 269 } 270 } 271#endif 272}; 273 274// ---------------------------------------------------------------------------- 275// ThreadBase 276// ---------------------------------------------------------------------------- 277 278AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 279 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 280 : Thread(false /*canCallJava*/), 281 mType(type), 282 mAudioFlinger(audioFlinger), 283 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 284 // are set by PlaybackThread::readOutputParameters_l() or 285 // RecordThread::readInputParameters_l() 286 //FIXME: mStandby should be true here. Is this some kind of hack? 287 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 288 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 289 // mName will be set by concrete (non-virtual) subclass 290 mDeathRecipient(new PMDeathRecipient(this)) 291{ 292} 293 294AudioFlinger::ThreadBase::~ThreadBase() 295{ 296 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 297 mConfigEvents.clear(); 298 299 // do not lock the mutex in destructor 300 releaseWakeLock_l(); 301 if (mPowerManager != 0) { 302 sp<IBinder> binder = mPowerManager->asBinder(); 303 binder->unlinkToDeath(mDeathRecipient); 304 } 305} 306 307status_t AudioFlinger::ThreadBase::readyToRun() 308{ 309 status_t status = initCheck(); 310 if (status == NO_ERROR) { 311 ALOGI("AudioFlinger's thread %p ready to run", this); 312 } else { 313 ALOGE("No working audio driver found."); 314 } 315 return status; 316} 317 318void AudioFlinger::ThreadBase::exit() 319{ 320 ALOGV("ThreadBase::exit"); 321 // do any cleanup required for exit to succeed 322 preExit(); 323 { 324 // This lock prevents the following race in thread (uniprocessor for illustration): 325 // if (!exitPending()) { 326 // // context switch from here to exit() 327 // // exit() calls requestExit(), what exitPending() observes 328 // // exit() calls signal(), which is dropped since no waiters 329 // // context switch back from exit() to here 330 // mWaitWorkCV.wait(...); 331 // // now thread is hung 332 // } 333 AutoMutex lock(mLock); 334 requestExit(); 335 mWaitWorkCV.broadcast(); 336 } 337 // When Thread::requestExitAndWait is made virtual and this method is renamed to 338 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 339 requestExitAndWait(); 340} 341 342status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 343{ 344 status_t status; 345 346 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 347 Mutex::Autolock _l(mLock); 348 349 return sendSetParameterConfigEvent_l(keyValuePairs); 350} 351 352// sendConfigEvent_l() must be called with ThreadBase::mLock held 353// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 354status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 355{ 356 status_t status = NO_ERROR; 357 358 mConfigEvents.add(event); 359 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 360 mWaitWorkCV.signal(); 361 mLock.unlock(); 362 { 363 Mutex::Autolock _l(event->mLock); 364 while (event->mWaitStatus) { 365 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 366 event->mStatus = TIMED_OUT; 367 event->mWaitStatus = false; 368 } 369 } 370 status = event->mStatus; 371 } 372 mLock.lock(); 373 return status; 374} 375 376void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 377{ 378 Mutex::Autolock _l(mLock); 379 sendIoConfigEvent_l(event, param); 380} 381 382// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 383void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 384{ 385 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 386 sendConfigEvent_l(configEvent); 387} 388 389// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 390void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 391{ 392 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 393 sendConfigEvent_l(configEvent); 394} 395 396// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 397status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 398{ 399 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 400 return sendConfigEvent_l(configEvent); 401} 402 403// post condition: mConfigEvents.isEmpty() 404void AudioFlinger::ThreadBase::processConfigEvents_l() 405{ 406 bool configChanged = false; 407 408 while (!mConfigEvents.isEmpty()) { 409 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 410 sp<ConfigEvent> event = mConfigEvents[0]; 411 mConfigEvents.removeAt(0); 412 switch (event->mType) { 413 case CFG_EVENT_PRIO: { 414 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 415 // FIXME Need to understand why this has to be done asynchronously 416 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 417 true /*asynchronous*/); 418 if (err != 0) { 419 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 420 data->mPrio, data->mPid, data->mTid, err); 421 } 422 } break; 423 case CFG_EVENT_IO: { 424 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 425 audioConfigChanged(data->mEvent, data->mParam); 426 } break; 427 case CFG_EVENT_SET_PARAMETER: { 428 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 429 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 430 configChanged = true; 431 } 432 } break; 433 default: 434 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 435 break; 436 } 437 { 438 Mutex::Autolock _l(event->mLock); 439 if (event->mWaitStatus) { 440 event->mWaitStatus = false; 441 event->mCond.signal(); 442 } 443 } 444 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 445 } 446 447 if (configChanged) { 448 cacheParameters_l(); 449 } 450} 451 452String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 453 String8 s; 454 if (output) { 455 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 456 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 457 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 458 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 459 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 460 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 461 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 462 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 463 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 464 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 465 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 466 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 467 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 468 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 469 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 470 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 471 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 472 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 473 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 474 } else { 475 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 476 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 477 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 478 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 479 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 480 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 481 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 482 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 483 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 484 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 485 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 486 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 487 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 488 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 489 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 490 } 491 int len = s.length(); 492 if (s.length() > 2) { 493 char *str = s.lockBuffer(len); 494 s.unlockBuffer(len - 2); 495 } 496 return s; 497} 498 499void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 500{ 501 const size_t SIZE = 256; 502 char buffer[SIZE]; 503 String8 result; 504 505 bool locked = AudioFlinger::dumpTryLock(mLock); 506 if (!locked) { 507 fdprintf(fd, "thread %p maybe dead locked\n", this); 508 } 509 510 fdprintf(fd, " I/O handle: %d\n", mId); 511 fdprintf(fd, " TID: %d\n", getTid()); 512 fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 513 fdprintf(fd, " Sample rate: %u\n", mSampleRate); 514 fdprintf(fd, " HAL frame count: %zu\n", mFrameCount); 515 fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 516 fdprintf(fd, " Channel Count: %u\n", mChannelCount); 517 fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 518 channelMaskToString(mChannelMask, mType != RECORD).string()); 519 fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 520 fdprintf(fd, " Frame size: %zu\n", mFrameSize); 521 fdprintf(fd, " Pending config events:"); 522 size_t numConfig = mConfigEvents.size(); 523 if (numConfig) { 524 for (size_t i = 0; i < numConfig; i++) { 525 mConfigEvents[i]->dump(buffer, SIZE); 526 fdprintf(fd, "\n %s", buffer); 527 } 528 fdprintf(fd, "\n"); 529 } else { 530 fdprintf(fd, " none\n"); 531 } 532 533 if (locked) { 534 mLock.unlock(); 535 } 536} 537 538void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 539{ 540 const size_t SIZE = 256; 541 char buffer[SIZE]; 542 String8 result; 543 544 size_t numEffectChains = mEffectChains.size(); 545 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 546 write(fd, buffer, strlen(buffer)); 547 548 for (size_t i = 0; i < numEffectChains; ++i) { 549 sp<EffectChain> chain = mEffectChains[i]; 550 if (chain != 0) { 551 chain->dump(fd, args); 552 } 553 } 554} 555 556void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 557{ 558 Mutex::Autolock _l(mLock); 559 acquireWakeLock_l(uid); 560} 561 562String16 AudioFlinger::ThreadBase::getWakeLockTag() 563{ 564 switch (mType) { 565 case MIXER: 566 return String16("AudioMix"); 567 case DIRECT: 568 return String16("AudioDirectOut"); 569 case DUPLICATING: 570 return String16("AudioDup"); 571 case RECORD: 572 return String16("AudioIn"); 573 case OFFLOAD: 574 return String16("AudioOffload"); 575 default: 576 ALOG_ASSERT(false); 577 return String16("AudioUnknown"); 578 } 579} 580 581void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 582{ 583 getPowerManager_l(); 584 if (mPowerManager != 0) { 585 sp<IBinder> binder = new BBinder(); 586 status_t status; 587 if (uid >= 0) { 588 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 589 binder, 590 getWakeLockTag(), 591 String16("media"), 592 uid); 593 } else { 594 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 595 binder, 596 getWakeLockTag(), 597 String16("media")); 598 } 599 if (status == NO_ERROR) { 600 mWakeLockToken = binder; 601 } 602 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 603 } 604} 605 606void AudioFlinger::ThreadBase::releaseWakeLock() 607{ 608 Mutex::Autolock _l(mLock); 609 releaseWakeLock_l(); 610} 611 612void AudioFlinger::ThreadBase::releaseWakeLock_l() 613{ 614 if (mWakeLockToken != 0) { 615 ALOGV("releaseWakeLock_l() %s", mName); 616 if (mPowerManager != 0) { 617 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 618 } 619 mWakeLockToken.clear(); 620 } 621} 622 623void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 624 Mutex::Autolock _l(mLock); 625 updateWakeLockUids_l(uids); 626} 627 628void AudioFlinger::ThreadBase::getPowerManager_l() { 629 630 if (mPowerManager == 0) { 631 // use checkService() to avoid blocking if power service is not up yet 632 sp<IBinder> binder = 633 defaultServiceManager()->checkService(String16("power")); 634 if (binder == 0) { 635 ALOGW("Thread %s cannot connect to the power manager service", mName); 636 } else { 637 mPowerManager = interface_cast<IPowerManager>(binder); 638 binder->linkToDeath(mDeathRecipient); 639 } 640 } 641} 642 643void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 644 645 getPowerManager_l(); 646 if (mWakeLockToken == NULL) { 647 ALOGE("no wake lock to update!"); 648 return; 649 } 650 if (mPowerManager != 0) { 651 sp<IBinder> binder = new BBinder(); 652 status_t status; 653 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 654 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 655 } 656} 657 658void AudioFlinger::ThreadBase::clearPowerManager() 659{ 660 Mutex::Autolock _l(mLock); 661 releaseWakeLock_l(); 662 mPowerManager.clear(); 663} 664 665void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 666{ 667 sp<ThreadBase> thread = mThread.promote(); 668 if (thread != 0) { 669 thread->clearPowerManager(); 670 } 671 ALOGW("power manager service died !!!"); 672} 673 674void AudioFlinger::ThreadBase::setEffectSuspended( 675 const effect_uuid_t *type, bool suspend, int sessionId) 676{ 677 Mutex::Autolock _l(mLock); 678 setEffectSuspended_l(type, suspend, sessionId); 679} 680 681void AudioFlinger::ThreadBase::setEffectSuspended_l( 682 const effect_uuid_t *type, bool suspend, int sessionId) 683{ 684 sp<EffectChain> chain = getEffectChain_l(sessionId); 685 if (chain != 0) { 686 if (type != NULL) { 687 chain->setEffectSuspended_l(type, suspend); 688 } else { 689 chain->setEffectSuspendedAll_l(suspend); 690 } 691 } 692 693 updateSuspendedSessions_l(type, suspend, sessionId); 694} 695 696void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 697{ 698 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 699 if (index < 0) { 700 return; 701 } 702 703 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 704 mSuspendedSessions.valueAt(index); 705 706 for (size_t i = 0; i < sessionEffects.size(); i++) { 707 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 708 for (int j = 0; j < desc->mRefCount; j++) { 709 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 710 chain->setEffectSuspendedAll_l(true); 711 } else { 712 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 713 desc->mType.timeLow); 714 chain->setEffectSuspended_l(&desc->mType, true); 715 } 716 } 717 } 718} 719 720void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 721 bool suspend, 722 int sessionId) 723{ 724 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 725 726 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 727 728 if (suspend) { 729 if (index >= 0) { 730 sessionEffects = mSuspendedSessions.valueAt(index); 731 } else { 732 mSuspendedSessions.add(sessionId, sessionEffects); 733 } 734 } else { 735 if (index < 0) { 736 return; 737 } 738 sessionEffects = mSuspendedSessions.valueAt(index); 739 } 740 741 742 int key = EffectChain::kKeyForSuspendAll; 743 if (type != NULL) { 744 key = type->timeLow; 745 } 746 index = sessionEffects.indexOfKey(key); 747 748 sp<SuspendedSessionDesc> desc; 749 if (suspend) { 750 if (index >= 0) { 751 desc = sessionEffects.valueAt(index); 752 } else { 753 desc = new SuspendedSessionDesc(); 754 if (type != NULL) { 755 desc->mType = *type; 756 } 757 sessionEffects.add(key, desc); 758 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 759 } 760 desc->mRefCount++; 761 } else { 762 if (index < 0) { 763 return; 764 } 765 desc = sessionEffects.valueAt(index); 766 if (--desc->mRefCount == 0) { 767 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 768 sessionEffects.removeItemsAt(index); 769 if (sessionEffects.isEmpty()) { 770 ALOGV("updateSuspendedSessions_l() restore removing session %d", 771 sessionId); 772 mSuspendedSessions.removeItem(sessionId); 773 } 774 } 775 } 776 if (!sessionEffects.isEmpty()) { 777 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 778 } 779} 780 781void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 782 bool enabled, 783 int sessionId) 784{ 785 Mutex::Autolock _l(mLock); 786 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 787} 788 789void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 790 bool enabled, 791 int sessionId) 792{ 793 if (mType != RECORD) { 794 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 795 // another session. This gives the priority to well behaved effect control panels 796 // and applications not using global effects. 797 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 798 // global effects 799 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 800 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 801 } 802 } 803 804 sp<EffectChain> chain = getEffectChain_l(sessionId); 805 if (chain != 0) { 806 chain->checkSuspendOnEffectEnabled(effect, enabled); 807 } 808} 809 810// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 811sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 812 const sp<AudioFlinger::Client>& client, 813 const sp<IEffectClient>& effectClient, 814 int32_t priority, 815 int sessionId, 816 effect_descriptor_t *desc, 817 int *enabled, 818 status_t *status) 819{ 820 sp<EffectModule> effect; 821 sp<EffectHandle> handle; 822 status_t lStatus; 823 sp<EffectChain> chain; 824 bool chainCreated = false; 825 bool effectCreated = false; 826 bool effectRegistered = false; 827 828 lStatus = initCheck(); 829 if (lStatus != NO_ERROR) { 830 ALOGW("createEffect_l() Audio driver not initialized."); 831 goto Exit; 832 } 833 834 // Reject any effect on Direct output threads for now, since the format of 835 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 836 if (mType == DIRECT) { 837 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 838 desc->name, mName); 839 lStatus = BAD_VALUE; 840 goto Exit; 841 } 842 843 // Allow global effects only on offloaded and mixer threads 844 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 845 switch (mType) { 846 case MIXER: 847 case OFFLOAD: 848 break; 849 case DIRECT: 850 case DUPLICATING: 851 case RECORD: 852 default: 853 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 854 lStatus = BAD_VALUE; 855 goto Exit; 856 } 857 } 858 859 // Only Pre processor effects are allowed on input threads and only on input threads 860 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 861 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 862 desc->name, desc->flags, mType); 863 lStatus = BAD_VALUE; 864 goto Exit; 865 } 866 867 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 868 869 { // scope for mLock 870 Mutex::Autolock _l(mLock); 871 872 // check for existing effect chain with the requested audio session 873 chain = getEffectChain_l(sessionId); 874 if (chain == 0) { 875 // create a new chain for this session 876 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 877 chain = new EffectChain(this, sessionId); 878 addEffectChain_l(chain); 879 chain->setStrategy(getStrategyForSession_l(sessionId)); 880 chainCreated = true; 881 } else { 882 effect = chain->getEffectFromDesc_l(desc); 883 } 884 885 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 886 887 if (effect == 0) { 888 int id = mAudioFlinger->nextUniqueId(); 889 // Check CPU and memory usage 890 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 891 if (lStatus != NO_ERROR) { 892 goto Exit; 893 } 894 effectRegistered = true; 895 // create a new effect module if none present in the chain 896 effect = new EffectModule(this, chain, desc, id, sessionId); 897 lStatus = effect->status(); 898 if (lStatus != NO_ERROR) { 899 goto Exit; 900 } 901 effect->setOffloaded(mType == OFFLOAD, mId); 902 903 lStatus = chain->addEffect_l(effect); 904 if (lStatus != NO_ERROR) { 905 goto Exit; 906 } 907 effectCreated = true; 908 909 effect->setDevice(mOutDevice); 910 effect->setDevice(mInDevice); 911 effect->setMode(mAudioFlinger->getMode()); 912 effect->setAudioSource(mAudioSource); 913 } 914 // create effect handle and connect it to effect module 915 handle = new EffectHandle(effect, client, effectClient, priority); 916 lStatus = handle->initCheck(); 917 if (lStatus == OK) { 918 lStatus = effect->addHandle(handle.get()); 919 } 920 if (enabled != NULL) { 921 *enabled = (int)effect->isEnabled(); 922 } 923 } 924 925Exit: 926 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 927 Mutex::Autolock _l(mLock); 928 if (effectCreated) { 929 chain->removeEffect_l(effect); 930 } 931 if (effectRegistered) { 932 AudioSystem::unregisterEffect(effect->id()); 933 } 934 if (chainCreated) { 935 removeEffectChain_l(chain); 936 } 937 handle.clear(); 938 } 939 940 *status = lStatus; 941 return handle; 942} 943 944sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 945{ 946 Mutex::Autolock _l(mLock); 947 return getEffect_l(sessionId, effectId); 948} 949 950sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 951{ 952 sp<EffectChain> chain = getEffectChain_l(sessionId); 953 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 954} 955 956// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 957// PlaybackThread::mLock held 958status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 959{ 960 // check for existing effect chain with the requested audio session 961 int sessionId = effect->sessionId(); 962 sp<EffectChain> chain = getEffectChain_l(sessionId); 963 bool chainCreated = false; 964 965 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 966 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 967 this, effect->desc().name, effect->desc().flags); 968 969 if (chain == 0) { 970 // create a new chain for this session 971 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 972 chain = new EffectChain(this, sessionId); 973 addEffectChain_l(chain); 974 chain->setStrategy(getStrategyForSession_l(sessionId)); 975 chainCreated = true; 976 } 977 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 978 979 if (chain->getEffectFromId_l(effect->id()) != 0) { 980 ALOGW("addEffect_l() %p effect %s already present in chain %p", 981 this, effect->desc().name, chain.get()); 982 return BAD_VALUE; 983 } 984 985 effect->setOffloaded(mType == OFFLOAD, mId); 986 987 status_t status = chain->addEffect_l(effect); 988 if (status != NO_ERROR) { 989 if (chainCreated) { 990 removeEffectChain_l(chain); 991 } 992 return status; 993 } 994 995 effect->setDevice(mOutDevice); 996 effect->setDevice(mInDevice); 997 effect->setMode(mAudioFlinger->getMode()); 998 effect->setAudioSource(mAudioSource); 999 return NO_ERROR; 1000} 1001 1002void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1003 1004 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1005 effect_descriptor_t desc = effect->desc(); 1006 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1007 detachAuxEffect_l(effect->id()); 1008 } 1009 1010 sp<EffectChain> chain = effect->chain().promote(); 1011 if (chain != 0) { 1012 // remove effect chain if removing last effect 1013 if (chain->removeEffect_l(effect) == 0) { 1014 removeEffectChain_l(chain); 1015 } 1016 } else { 1017 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1018 } 1019} 1020 1021void AudioFlinger::ThreadBase::lockEffectChains_l( 1022 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1023{ 1024 effectChains = mEffectChains; 1025 for (size_t i = 0; i < mEffectChains.size(); i++) { 1026 mEffectChains[i]->lock(); 1027 } 1028} 1029 1030void AudioFlinger::ThreadBase::unlockEffectChains( 1031 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1032{ 1033 for (size_t i = 0; i < effectChains.size(); i++) { 1034 effectChains[i]->unlock(); 1035 } 1036} 1037 1038sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1039{ 1040 Mutex::Autolock _l(mLock); 1041 return getEffectChain_l(sessionId); 1042} 1043 1044sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1045{ 1046 size_t size = mEffectChains.size(); 1047 for (size_t i = 0; i < size; i++) { 1048 if (mEffectChains[i]->sessionId() == sessionId) { 1049 return mEffectChains[i]; 1050 } 1051 } 1052 return 0; 1053} 1054 1055void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1056{ 1057 Mutex::Autolock _l(mLock); 1058 size_t size = mEffectChains.size(); 1059 for (size_t i = 0; i < size; i++) { 1060 mEffectChains[i]->setMode_l(mode); 1061 } 1062} 1063 1064void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1065 EffectHandle *handle, 1066 bool unpinIfLast) { 1067 1068 Mutex::Autolock _l(mLock); 1069 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1070 // delete the effect module if removing last handle on it 1071 if (effect->removeHandle(handle) == 0) { 1072 if (!effect->isPinned() || unpinIfLast) { 1073 removeEffect_l(effect); 1074 AudioSystem::unregisterEffect(effect->id()); 1075 } 1076 } 1077} 1078 1079// ---------------------------------------------------------------------------- 1080// Playback 1081// ---------------------------------------------------------------------------- 1082 1083AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1084 AudioStreamOut* output, 1085 audio_io_handle_t id, 1086 audio_devices_t device, 1087 type_t type) 1088 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1089 mNormalFrameCount(0), mSinkBuffer(NULL), 1090 mMixerBufferEnabled(false), 1091 mMixerBuffer(NULL), 1092 mMixerBufferSize(0), 1093 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1094 mMixerBufferValid(false), 1095 mEffectBufferEnabled(false), 1096 mEffectBuffer(NULL), 1097 mEffectBufferSize(0), 1098 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1099 mEffectBufferValid(false), 1100 mSuspended(0), mBytesWritten(0), 1101 mActiveTracksGeneration(0), 1102 // mStreamTypes[] initialized in constructor body 1103 mOutput(output), 1104 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1105 mMixerStatus(MIXER_IDLE), 1106 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1107 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1108 mBytesRemaining(0), 1109 mCurrentWriteLength(0), 1110 mUseAsyncWrite(false), 1111 mWriteAckSequence(0), 1112 mDrainSequence(0), 1113 mSignalPending(false), 1114 mScreenState(AudioFlinger::mScreenState), 1115 // index 0 is reserved for normal mixer's submix 1116 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1117 // mLatchD, mLatchQ, 1118 mLatchDValid(false), mLatchQValid(false) 1119{ 1120 snprintf(mName, kNameLength, "AudioOut_%X", id); 1121 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1122 1123 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1124 // it would be safer to explicitly pass initial masterVolume/masterMute as 1125 // parameter. 1126 // 1127 // If the HAL we are using has support for master volume or master mute, 1128 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1129 // and the mute set to false). 1130 mMasterVolume = audioFlinger->masterVolume_l(); 1131 mMasterMute = audioFlinger->masterMute_l(); 1132 if (mOutput && mOutput->audioHwDev) { 1133 if (mOutput->audioHwDev->canSetMasterVolume()) { 1134 mMasterVolume = 1.0; 1135 } 1136 1137 if (mOutput->audioHwDev->canSetMasterMute()) { 1138 mMasterMute = false; 1139 } 1140 } 1141 1142 readOutputParameters_l(); 1143 1144 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1145 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1146 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1147 stream = (audio_stream_type_t) (stream + 1)) { 1148 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1149 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1150 } 1151 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1152 // because mAudioFlinger doesn't have one to copy from 1153} 1154 1155AudioFlinger::PlaybackThread::~PlaybackThread() 1156{ 1157 mAudioFlinger->unregisterWriter(mNBLogWriter); 1158 free(mSinkBuffer); 1159 free(mMixerBuffer); 1160 free(mEffectBuffer); 1161} 1162 1163void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1164{ 1165 dumpInternals(fd, args); 1166 dumpTracks(fd, args); 1167 dumpEffectChains(fd, args); 1168} 1169 1170void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1171{ 1172 const size_t SIZE = 256; 1173 char buffer[SIZE]; 1174 String8 result; 1175 1176 result.appendFormat(" Stream volumes in dB: "); 1177 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1178 const stream_type_t *st = &mStreamTypes[i]; 1179 if (i > 0) { 1180 result.appendFormat(", "); 1181 } 1182 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1183 if (st->mute) { 1184 result.append("M"); 1185 } 1186 } 1187 result.append("\n"); 1188 write(fd, result.string(), result.length()); 1189 result.clear(); 1190 1191 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1192 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1193 fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1194 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1195 1196 size_t numtracks = mTracks.size(); 1197 size_t numactive = mActiveTracks.size(); 1198 fdprintf(fd, " %d Tracks", numtracks); 1199 size_t numactiveseen = 0; 1200 if (numtracks) { 1201 fdprintf(fd, " of which %d are active\n", numactive); 1202 Track::appendDumpHeader(result); 1203 for (size_t i = 0; i < numtracks; ++i) { 1204 sp<Track> track = mTracks[i]; 1205 if (track != 0) { 1206 bool active = mActiveTracks.indexOf(track) >= 0; 1207 if (active) { 1208 numactiveseen++; 1209 } 1210 track->dump(buffer, SIZE, active); 1211 result.append(buffer); 1212 } 1213 } 1214 } else { 1215 result.append("\n"); 1216 } 1217 if (numactiveseen != numactive) { 1218 // some tracks in the active list were not in the tracks list 1219 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1220 " not in the track list\n"); 1221 result.append(buffer); 1222 Track::appendDumpHeader(result); 1223 for (size_t i = 0; i < numactive; ++i) { 1224 sp<Track> track = mActiveTracks[i].promote(); 1225 if (track != 0 && mTracks.indexOf(track) < 0) { 1226 track->dump(buffer, SIZE, true); 1227 result.append(buffer); 1228 } 1229 } 1230 } 1231 1232 write(fd, result.string(), result.size()); 1233 1234} 1235 1236void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1237{ 1238 fdprintf(fd, "\nOutput thread %p:\n", this); 1239 fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1240 fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1241 fdprintf(fd, " Total writes: %d\n", mNumWrites); 1242 fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1243 fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1244 fdprintf(fd, " Suspend count: %d\n", mSuspended); 1245 fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1246 fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1247 fdprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1248 fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1249 1250 dumpBase(fd, args); 1251} 1252 1253// Thread virtuals 1254 1255void AudioFlinger::PlaybackThread::onFirstRef() 1256{ 1257 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1258} 1259 1260// ThreadBase virtuals 1261void AudioFlinger::PlaybackThread::preExit() 1262{ 1263 ALOGV(" preExit()"); 1264 // FIXME this is using hard-coded strings but in the future, this functionality will be 1265 // converted to use audio HAL extensions required to support tunneling 1266 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1267} 1268 1269// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1270sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1271 const sp<AudioFlinger::Client>& client, 1272 audio_stream_type_t streamType, 1273 uint32_t sampleRate, 1274 audio_format_t format, 1275 audio_channel_mask_t channelMask, 1276 size_t *pFrameCount, 1277 const sp<IMemory>& sharedBuffer, 1278 int sessionId, 1279 IAudioFlinger::track_flags_t *flags, 1280 pid_t tid, 1281 int uid, 1282 status_t *status) 1283{ 1284 size_t frameCount = *pFrameCount; 1285 sp<Track> track; 1286 status_t lStatus; 1287 1288 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1289 1290 // client expresses a preference for FAST, but we get the final say 1291 if (*flags & IAudioFlinger::TRACK_FAST) { 1292 if ( 1293 // not timed 1294 (!isTimed) && 1295 // either of these use cases: 1296 ( 1297 // use case 1: shared buffer with any frame count 1298 ( 1299 (sharedBuffer != 0) 1300 ) || 1301 // use case 2: callback handler and frame count is default or at least as large as HAL 1302 ( 1303 (tid != -1) && 1304 ((frameCount == 0) || 1305 (frameCount >= mFrameCount)) 1306 ) 1307 ) && 1308 // PCM data 1309 audio_is_linear_pcm(format) && 1310 // mono or stereo 1311 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1312 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1313 // hardware sample rate 1314 (sampleRate == mSampleRate) && 1315 // normal mixer has an associated fast mixer 1316 hasFastMixer() && 1317 // there are sufficient fast track slots available 1318 (mFastTrackAvailMask != 0) 1319 // FIXME test that MixerThread for this fast track has a capable output HAL 1320 // FIXME add a permission test also? 1321 ) { 1322 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1323 if (frameCount == 0) { 1324 frameCount = mFrameCount * kFastTrackMultiplier; 1325 } 1326 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1327 frameCount, mFrameCount); 1328 } else { 1329 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1330 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1331 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1332 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1333 audio_is_linear_pcm(format), 1334 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1335 *flags &= ~IAudioFlinger::TRACK_FAST; 1336 // For compatibility with AudioTrack calculation, buffer depth is forced 1337 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1338 // This is probably too conservative, but legacy application code may depend on it. 1339 // If you change this calculation, also review the start threshold which is related. 1340 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1341 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1342 if (minBufCount < 2) { 1343 minBufCount = 2; 1344 } 1345 size_t minFrameCount = mNormalFrameCount * minBufCount; 1346 if (frameCount < minFrameCount) { 1347 frameCount = minFrameCount; 1348 } 1349 } 1350 } 1351 *pFrameCount = frameCount; 1352 1353 switch (mType) { 1354 1355 case DIRECT: 1356 if (audio_is_linear_pcm(format)) { 1357 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1358 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1359 "for output %p with format %#x", 1360 sampleRate, format, channelMask, mOutput, mFormat); 1361 lStatus = BAD_VALUE; 1362 goto Exit; 1363 } 1364 } 1365 break; 1366 1367 case OFFLOAD: 1368 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1369 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1370 "for output %p with format %#x", 1371 sampleRate, format, channelMask, mOutput, mFormat); 1372 lStatus = BAD_VALUE; 1373 goto Exit; 1374 } 1375 break; 1376 1377 default: 1378 if (!audio_is_linear_pcm(format)) { 1379 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1380 "for output %p with format %#x", 1381 format, mOutput, mFormat); 1382 lStatus = BAD_VALUE; 1383 goto Exit; 1384 } 1385 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1386 if (sampleRate > mSampleRate*2) { 1387 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1388 lStatus = BAD_VALUE; 1389 goto Exit; 1390 } 1391 break; 1392 1393 } 1394 1395 lStatus = initCheck(); 1396 if (lStatus != NO_ERROR) { 1397 ALOGE("createTrack_l() audio driver not initialized"); 1398 goto Exit; 1399 } 1400 1401 { // scope for mLock 1402 Mutex::Autolock _l(mLock); 1403 1404 // all tracks in same audio session must share the same routing strategy otherwise 1405 // conflicts will happen when tracks are moved from one output to another by audio policy 1406 // manager 1407 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1408 for (size_t i = 0; i < mTracks.size(); ++i) { 1409 sp<Track> t = mTracks[i]; 1410 if (t != 0 && !t->isOutputTrack()) { 1411 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1412 if (sessionId == t->sessionId() && strategy != actual) { 1413 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1414 strategy, actual); 1415 lStatus = BAD_VALUE; 1416 goto Exit; 1417 } 1418 } 1419 } 1420 1421 if (!isTimed) { 1422 track = new Track(this, client, streamType, sampleRate, format, 1423 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1424 } else { 1425 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1426 channelMask, frameCount, sharedBuffer, sessionId, uid); 1427 } 1428 1429 // new Track always returns non-NULL, 1430 // but TimedTrack::create() is a factory that could fail by returning NULL 1431 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1432 if (lStatus != NO_ERROR) { 1433 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1434 // track must be cleared from the caller as the caller has the AF lock 1435 goto Exit; 1436 } 1437 mTracks.add(track); 1438 1439 sp<EffectChain> chain = getEffectChain_l(sessionId); 1440 if (chain != 0) { 1441 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1442 track->setMainBuffer(chain->inBuffer()); 1443 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1444 chain->incTrackCnt(); 1445 } 1446 1447 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1448 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1449 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1450 // so ask activity manager to do this on our behalf 1451 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1452 } 1453 } 1454 1455 lStatus = NO_ERROR; 1456 1457Exit: 1458 *status = lStatus; 1459 return track; 1460} 1461 1462uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1463{ 1464 return latency; 1465} 1466 1467uint32_t AudioFlinger::PlaybackThread::latency() const 1468{ 1469 Mutex::Autolock _l(mLock); 1470 return latency_l(); 1471} 1472uint32_t AudioFlinger::PlaybackThread::latency_l() const 1473{ 1474 if (initCheck() == NO_ERROR) { 1475 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1476 } else { 1477 return 0; 1478 } 1479} 1480 1481void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1482{ 1483 Mutex::Autolock _l(mLock); 1484 // Don't apply master volume in SW if our HAL can do it for us. 1485 if (mOutput && mOutput->audioHwDev && 1486 mOutput->audioHwDev->canSetMasterVolume()) { 1487 mMasterVolume = 1.0; 1488 } else { 1489 mMasterVolume = value; 1490 } 1491} 1492 1493void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1494{ 1495 Mutex::Autolock _l(mLock); 1496 // Don't apply master mute in SW if our HAL can do it for us. 1497 if (mOutput && mOutput->audioHwDev && 1498 mOutput->audioHwDev->canSetMasterMute()) { 1499 mMasterMute = false; 1500 } else { 1501 mMasterMute = muted; 1502 } 1503} 1504 1505void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1506{ 1507 Mutex::Autolock _l(mLock); 1508 mStreamTypes[stream].volume = value; 1509 broadcast_l(); 1510} 1511 1512void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1513{ 1514 Mutex::Autolock _l(mLock); 1515 mStreamTypes[stream].mute = muted; 1516 broadcast_l(); 1517} 1518 1519float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1520{ 1521 Mutex::Autolock _l(mLock); 1522 return mStreamTypes[stream].volume; 1523} 1524 1525// addTrack_l() must be called with ThreadBase::mLock held 1526status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1527{ 1528 status_t status = ALREADY_EXISTS; 1529 1530 // set retry count for buffer fill 1531 track->mRetryCount = kMaxTrackStartupRetries; 1532 if (mActiveTracks.indexOf(track) < 0) { 1533 // the track is newly added, make sure it fills up all its 1534 // buffers before playing. This is to ensure the client will 1535 // effectively get the latency it requested. 1536 if (!track->isOutputTrack()) { 1537 TrackBase::track_state state = track->mState; 1538 mLock.unlock(); 1539 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1540 mLock.lock(); 1541 // abort track was stopped/paused while we released the lock 1542 if (state != track->mState) { 1543 if (status == NO_ERROR) { 1544 mLock.unlock(); 1545 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1546 mLock.lock(); 1547 } 1548 return INVALID_OPERATION; 1549 } 1550 // abort if start is rejected by audio policy manager 1551 if (status != NO_ERROR) { 1552 return PERMISSION_DENIED; 1553 } 1554#ifdef ADD_BATTERY_DATA 1555 // to track the speaker usage 1556 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1557#endif 1558 } 1559 1560 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1561 track->mResetDone = false; 1562 track->mPresentationCompleteFrames = 0; 1563 mActiveTracks.add(track); 1564 mWakeLockUids.add(track->uid()); 1565 mActiveTracksGeneration++; 1566 mLatestActiveTrack = track; 1567 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1568 if (chain != 0) { 1569 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1570 track->sessionId()); 1571 chain->incActiveTrackCnt(); 1572 } 1573 1574 status = NO_ERROR; 1575 } 1576 1577 onAddNewTrack_l(); 1578 return status; 1579} 1580 1581bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1582{ 1583 track->terminate(); 1584 // active tracks are removed by threadLoop() 1585 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1586 track->mState = TrackBase::STOPPED; 1587 if (!trackActive) { 1588 removeTrack_l(track); 1589 } else if (track->isFastTrack() || track->isOffloaded()) { 1590 track->mState = TrackBase::STOPPING_1; 1591 } 1592 1593 return trackActive; 1594} 1595 1596void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1597{ 1598 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1599 mTracks.remove(track); 1600 deleteTrackName_l(track->name()); 1601 // redundant as track is about to be destroyed, for dumpsys only 1602 track->mName = -1; 1603 if (track->isFastTrack()) { 1604 int index = track->mFastIndex; 1605 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1606 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1607 mFastTrackAvailMask |= 1 << index; 1608 // redundant as track is about to be destroyed, for dumpsys only 1609 track->mFastIndex = -1; 1610 } 1611 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1612 if (chain != 0) { 1613 chain->decTrackCnt(); 1614 } 1615} 1616 1617void AudioFlinger::PlaybackThread::broadcast_l() 1618{ 1619 // Thread could be blocked waiting for async 1620 // so signal it to handle state changes immediately 1621 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1622 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1623 mSignalPending = true; 1624 mWaitWorkCV.broadcast(); 1625} 1626 1627String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1628{ 1629 Mutex::Autolock _l(mLock); 1630 if (initCheck() != NO_ERROR) { 1631 return String8(); 1632 } 1633 1634 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1635 const String8 out_s8(s); 1636 free(s); 1637 return out_s8; 1638} 1639 1640void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1641 AudioSystem::OutputDescriptor desc; 1642 void *param2 = NULL; 1643 1644 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1645 param); 1646 1647 switch (event) { 1648 case AudioSystem::OUTPUT_OPENED: 1649 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1650 desc.channelMask = mChannelMask; 1651 desc.samplingRate = mSampleRate; 1652 desc.format = mFormat; 1653 desc.frameCount = mNormalFrameCount; // FIXME see 1654 // AudioFlinger::frameCount(audio_io_handle_t) 1655 desc.latency = latency_l(); 1656 param2 = &desc; 1657 break; 1658 1659 case AudioSystem::STREAM_CONFIG_CHANGED: 1660 param2 = ¶m; 1661 case AudioSystem::OUTPUT_CLOSED: 1662 default: 1663 break; 1664 } 1665 mAudioFlinger->audioConfigChanged(event, mId, param2); 1666} 1667 1668void AudioFlinger::PlaybackThread::writeCallback() 1669{ 1670 ALOG_ASSERT(mCallbackThread != 0); 1671 mCallbackThread->resetWriteBlocked(); 1672} 1673 1674void AudioFlinger::PlaybackThread::drainCallback() 1675{ 1676 ALOG_ASSERT(mCallbackThread != 0); 1677 mCallbackThread->resetDraining(); 1678} 1679 1680void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1681{ 1682 Mutex::Autolock _l(mLock); 1683 // reject out of sequence requests 1684 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1685 mWriteAckSequence &= ~1; 1686 mWaitWorkCV.signal(); 1687 } 1688} 1689 1690void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1691{ 1692 Mutex::Autolock _l(mLock); 1693 // reject out of sequence requests 1694 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1695 mDrainSequence &= ~1; 1696 mWaitWorkCV.signal(); 1697 } 1698} 1699 1700// static 1701int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1702 void *param __unused, 1703 void *cookie) 1704{ 1705 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1706 ALOGV("asyncCallback() event %d", event); 1707 switch (event) { 1708 case STREAM_CBK_EVENT_WRITE_READY: 1709 me->writeCallback(); 1710 break; 1711 case STREAM_CBK_EVENT_DRAIN_READY: 1712 me->drainCallback(); 1713 break; 1714 default: 1715 ALOGW("asyncCallback() unknown event %d", event); 1716 break; 1717 } 1718 return 0; 1719} 1720 1721void AudioFlinger::PlaybackThread::readOutputParameters_l() 1722{ 1723 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1724 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1725 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1726 if (!audio_is_output_channel(mChannelMask)) { 1727 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1728 } 1729 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1730 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1731 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1732 } 1733 mChannelCount = popcount(mChannelMask); 1734 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1735 if (!audio_is_valid_format(mFormat)) { 1736 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1737 } 1738 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1739 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1740 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1741 } 1742 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1743 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1744 mFrameCount = mBufferSize / mFrameSize; 1745 if (mFrameCount & 15) { 1746 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1747 mFrameCount); 1748 } 1749 1750 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1751 (mOutput->stream->set_callback != NULL)) { 1752 if (mOutput->stream->set_callback(mOutput->stream, 1753 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1754 mUseAsyncWrite = true; 1755 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1756 } 1757 } 1758 1759 // Calculate size of normal sink buffer relative to the HAL output buffer size 1760 double multiplier = 1.0; 1761 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1762 kUseFastMixer == FastMixer_Dynamic)) { 1763 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1764 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1765 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1766 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1767 maxNormalFrameCount = maxNormalFrameCount & ~15; 1768 if (maxNormalFrameCount < minNormalFrameCount) { 1769 maxNormalFrameCount = minNormalFrameCount; 1770 } 1771 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1772 if (multiplier <= 1.0) { 1773 multiplier = 1.0; 1774 } else if (multiplier <= 2.0) { 1775 if (2 * mFrameCount <= maxNormalFrameCount) { 1776 multiplier = 2.0; 1777 } else { 1778 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1779 } 1780 } else { 1781 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1782 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1783 // track, but we sometimes have to do this to satisfy the maximum frame count 1784 // constraint) 1785 // FIXME this rounding up should not be done if no HAL SRC 1786 uint32_t truncMult = (uint32_t) multiplier; 1787 if ((truncMult & 1)) { 1788 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1789 ++truncMult; 1790 } 1791 } 1792 multiplier = (double) truncMult; 1793 } 1794 } 1795 mNormalFrameCount = multiplier * mFrameCount; 1796 // round up to nearest 16 frames to satisfy AudioMixer 1797 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1798 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1799 mNormalFrameCount); 1800 1801 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1802 // Originally this was int16_t[] array, need to remove legacy implications. 1803 free(mSinkBuffer); 1804 mSinkBuffer = NULL; 1805 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1806 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1807 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1808 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1809 1810 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1811 // drives the output. 1812 free(mMixerBuffer); 1813 mMixerBuffer = NULL; 1814 if (mMixerBufferEnabled) { 1815 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1816 mMixerBufferSize = mNormalFrameCount * mChannelCount 1817 * audio_bytes_per_sample(mMixerBufferFormat); 1818 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1819 } 1820 free(mEffectBuffer); 1821 mEffectBuffer = NULL; 1822 if (mEffectBufferEnabled) { 1823 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1824 mEffectBufferSize = mNormalFrameCount * mChannelCount 1825 * audio_bytes_per_sample(mEffectBufferFormat); 1826 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1827 } 1828 1829 // force reconfiguration of effect chains and engines to take new buffer size and audio 1830 // parameters into account 1831 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1832 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1833 // matter. 1834 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1835 Vector< sp<EffectChain> > effectChains = mEffectChains; 1836 for (size_t i = 0; i < effectChains.size(); i ++) { 1837 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1838 } 1839} 1840 1841 1842status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1843{ 1844 if (halFrames == NULL || dspFrames == NULL) { 1845 return BAD_VALUE; 1846 } 1847 Mutex::Autolock _l(mLock); 1848 if (initCheck() != NO_ERROR) { 1849 return INVALID_OPERATION; 1850 } 1851 size_t framesWritten = mBytesWritten / mFrameSize; 1852 *halFrames = framesWritten; 1853 1854 if (isSuspended()) { 1855 // return an estimation of rendered frames when the output is suspended 1856 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1857 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1858 return NO_ERROR; 1859 } else { 1860 status_t status; 1861 uint32_t frames; 1862 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1863 *dspFrames = (size_t)frames; 1864 return status; 1865 } 1866} 1867 1868uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1869{ 1870 Mutex::Autolock _l(mLock); 1871 uint32_t result = 0; 1872 if (getEffectChain_l(sessionId) != 0) { 1873 result = EFFECT_SESSION; 1874 } 1875 1876 for (size_t i = 0; i < mTracks.size(); ++i) { 1877 sp<Track> track = mTracks[i]; 1878 if (sessionId == track->sessionId() && !track->isInvalid()) { 1879 result |= TRACK_SESSION; 1880 break; 1881 } 1882 } 1883 1884 return result; 1885} 1886 1887uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1888{ 1889 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1890 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1891 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1892 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1893 } 1894 for (size_t i = 0; i < mTracks.size(); i++) { 1895 sp<Track> track = mTracks[i]; 1896 if (sessionId == track->sessionId() && !track->isInvalid()) { 1897 return AudioSystem::getStrategyForStream(track->streamType()); 1898 } 1899 } 1900 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1901} 1902 1903 1904AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1905{ 1906 Mutex::Autolock _l(mLock); 1907 return mOutput; 1908} 1909 1910AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1911{ 1912 Mutex::Autolock _l(mLock); 1913 AudioStreamOut *output = mOutput; 1914 mOutput = NULL; 1915 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1916 // must push a NULL and wait for ack 1917 mOutputSink.clear(); 1918 mPipeSink.clear(); 1919 mNormalSink.clear(); 1920 return output; 1921} 1922 1923// this method must always be called either with ThreadBase mLock held or inside the thread loop 1924audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1925{ 1926 if (mOutput == NULL) { 1927 return NULL; 1928 } 1929 return &mOutput->stream->common; 1930} 1931 1932uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1933{ 1934 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1935} 1936 1937status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1938{ 1939 if (!isValidSyncEvent(event)) { 1940 return BAD_VALUE; 1941 } 1942 1943 Mutex::Autolock _l(mLock); 1944 1945 for (size_t i = 0; i < mTracks.size(); ++i) { 1946 sp<Track> track = mTracks[i]; 1947 if (event->triggerSession() == track->sessionId()) { 1948 (void) track->setSyncEvent(event); 1949 return NO_ERROR; 1950 } 1951 } 1952 1953 return NAME_NOT_FOUND; 1954} 1955 1956bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1957{ 1958 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1959} 1960 1961void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1962 const Vector< sp<Track> >& tracksToRemove) 1963{ 1964 size_t count = tracksToRemove.size(); 1965 if (count > 0) { 1966 for (size_t i = 0 ; i < count ; i++) { 1967 const sp<Track>& track = tracksToRemove.itemAt(i); 1968 if (!track->isOutputTrack()) { 1969 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1970#ifdef ADD_BATTERY_DATA 1971 // to track the speaker usage 1972 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1973#endif 1974 if (track->isTerminated()) { 1975 AudioSystem::releaseOutput(mId); 1976 } 1977 } 1978 } 1979 } 1980} 1981 1982void AudioFlinger::PlaybackThread::checkSilentMode_l() 1983{ 1984 if (!mMasterMute) { 1985 char value[PROPERTY_VALUE_MAX]; 1986 if (property_get("ro.audio.silent", value, "0") > 0) { 1987 char *endptr; 1988 unsigned long ul = strtoul(value, &endptr, 0); 1989 if (*endptr == '\0' && ul != 0) { 1990 ALOGD("Silence is golden"); 1991 // The setprop command will not allow a property to be changed after 1992 // the first time it is set, so we don't have to worry about un-muting. 1993 setMasterMute_l(true); 1994 } 1995 } 1996 } 1997} 1998 1999// shared by MIXER and DIRECT, overridden by DUPLICATING 2000ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2001{ 2002 // FIXME rewrite to reduce number of system calls 2003 mLastWriteTime = systemTime(); 2004 mInWrite = true; 2005 ssize_t bytesWritten; 2006 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2007 2008 // If an NBAIO sink is present, use it to write the normal mixer's submix 2009 if (mNormalSink != 0) { 2010 const size_t count = mBytesRemaining / mFrameSize; 2011 2012 ATRACE_BEGIN("write"); 2013 // update the setpoint when AudioFlinger::mScreenState changes 2014 uint32_t screenState = AudioFlinger::mScreenState; 2015 if (screenState != mScreenState) { 2016 mScreenState = screenState; 2017 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2018 if (pipe != NULL) { 2019 pipe->setAvgFrames((mScreenState & 1) ? 2020 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2021 } 2022 } 2023 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2024 ATRACE_END(); 2025 if (framesWritten > 0) { 2026 bytesWritten = framesWritten * mFrameSize; 2027 } else { 2028 bytesWritten = framesWritten; 2029 } 2030 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2031 if (status == NO_ERROR) { 2032 size_t totalFramesWritten = mNormalSink->framesWritten(); 2033 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2034 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2035 mLatchDValid = true; 2036 } 2037 } 2038 // otherwise use the HAL / AudioStreamOut directly 2039 } else { 2040 // Direct output and offload threads 2041 2042 if (mUseAsyncWrite) { 2043 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2044 mWriteAckSequence += 2; 2045 mWriteAckSequence |= 1; 2046 ALOG_ASSERT(mCallbackThread != 0); 2047 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2048 } 2049 // FIXME We should have an implementation of timestamps for direct output threads. 2050 // They are used e.g for multichannel PCM playback over HDMI. 2051 bytesWritten = mOutput->stream->write(mOutput->stream, 2052 (char *)mSinkBuffer + offset, mBytesRemaining); 2053 if (mUseAsyncWrite && 2054 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2055 // do not wait for async callback in case of error of full write 2056 mWriteAckSequence &= ~1; 2057 ALOG_ASSERT(mCallbackThread != 0); 2058 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2059 } 2060 } 2061 2062 mNumWrites++; 2063 mInWrite = false; 2064 mStandby = false; 2065 return bytesWritten; 2066} 2067 2068void AudioFlinger::PlaybackThread::threadLoop_drain() 2069{ 2070 if (mOutput->stream->drain) { 2071 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2072 if (mUseAsyncWrite) { 2073 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2074 mDrainSequence |= 1; 2075 ALOG_ASSERT(mCallbackThread != 0); 2076 mCallbackThread->setDraining(mDrainSequence); 2077 } 2078 mOutput->stream->drain(mOutput->stream, 2079 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2080 : AUDIO_DRAIN_ALL); 2081 } 2082} 2083 2084void AudioFlinger::PlaybackThread::threadLoop_exit() 2085{ 2086 // Default implementation has nothing to do 2087} 2088 2089/* 2090The derived values that are cached: 2091 - mSinkBufferSize from frame count * frame size 2092 - activeSleepTime from activeSleepTimeUs() 2093 - idleSleepTime from idleSleepTimeUs() 2094 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2095 - maxPeriod from frame count and sample rate (MIXER only) 2096 2097The parameters that affect these derived values are: 2098 - frame count 2099 - frame size 2100 - sample rate 2101 - device type: A2DP or not 2102 - device latency 2103 - format: PCM or not 2104 - active sleep time 2105 - idle sleep time 2106*/ 2107 2108void AudioFlinger::PlaybackThread::cacheParameters_l() 2109{ 2110 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2111 activeSleepTime = activeSleepTimeUs(); 2112 idleSleepTime = idleSleepTimeUs(); 2113} 2114 2115void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2116{ 2117 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2118 this, streamType, mTracks.size()); 2119 Mutex::Autolock _l(mLock); 2120 2121 size_t size = mTracks.size(); 2122 for (size_t i = 0; i < size; i++) { 2123 sp<Track> t = mTracks[i]; 2124 if (t->streamType() == streamType) { 2125 t->invalidate(); 2126 } 2127 } 2128} 2129 2130status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2131{ 2132 int session = chain->sessionId(); 2133 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2134 ? mEffectBuffer : mSinkBuffer); 2135 bool ownsBuffer = false; 2136 2137 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2138 if (session > 0) { 2139 // Only one effect chain can be present in direct output thread and it uses 2140 // the sink buffer as input 2141 if (mType != DIRECT) { 2142 size_t numSamples = mNormalFrameCount * mChannelCount; 2143 buffer = new int16_t[numSamples]; 2144 memset(buffer, 0, numSamples * sizeof(int16_t)); 2145 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2146 ownsBuffer = true; 2147 } 2148 2149 // Attach all tracks with same session ID to this chain. 2150 for (size_t i = 0; i < mTracks.size(); ++i) { 2151 sp<Track> track = mTracks[i]; 2152 if (session == track->sessionId()) { 2153 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2154 buffer); 2155 track->setMainBuffer(buffer); 2156 chain->incTrackCnt(); 2157 } 2158 } 2159 2160 // indicate all active tracks in the chain 2161 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2162 sp<Track> track = mActiveTracks[i].promote(); 2163 if (track == 0) { 2164 continue; 2165 } 2166 if (session == track->sessionId()) { 2167 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2168 chain->incActiveTrackCnt(); 2169 } 2170 } 2171 } 2172 2173 chain->setInBuffer(buffer, ownsBuffer); 2174 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2175 ? mEffectBuffer : mSinkBuffer)); 2176 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2177 // chains list in order to be processed last as it contains output stage effects 2178 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2179 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2180 // after track specific effects and before output stage 2181 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2182 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2183 // Effect chain for other sessions are inserted at beginning of effect 2184 // chains list to be processed before output mix effects. Relative order between other 2185 // sessions is not important 2186 size_t size = mEffectChains.size(); 2187 size_t i = 0; 2188 for (i = 0; i < size; i++) { 2189 if (mEffectChains[i]->sessionId() < session) { 2190 break; 2191 } 2192 } 2193 mEffectChains.insertAt(chain, i); 2194 checkSuspendOnAddEffectChain_l(chain); 2195 2196 return NO_ERROR; 2197} 2198 2199size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2200{ 2201 int session = chain->sessionId(); 2202 2203 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2204 2205 for (size_t i = 0; i < mEffectChains.size(); i++) { 2206 if (chain == mEffectChains[i]) { 2207 mEffectChains.removeAt(i); 2208 // detach all active tracks from the chain 2209 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2210 sp<Track> track = mActiveTracks[i].promote(); 2211 if (track == 0) { 2212 continue; 2213 } 2214 if (session == track->sessionId()) { 2215 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2216 chain.get(), session); 2217 chain->decActiveTrackCnt(); 2218 } 2219 } 2220 2221 // detach all tracks with same session ID from this chain 2222 for (size_t i = 0; i < mTracks.size(); ++i) { 2223 sp<Track> track = mTracks[i]; 2224 if (session == track->sessionId()) { 2225 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2226 chain->decTrackCnt(); 2227 } 2228 } 2229 break; 2230 } 2231 } 2232 return mEffectChains.size(); 2233} 2234 2235status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2236 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2237{ 2238 Mutex::Autolock _l(mLock); 2239 return attachAuxEffect_l(track, EffectId); 2240} 2241 2242status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2243 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2244{ 2245 status_t status = NO_ERROR; 2246 2247 if (EffectId == 0) { 2248 track->setAuxBuffer(0, NULL); 2249 } else { 2250 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2251 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2252 if (effect != 0) { 2253 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2254 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2255 } else { 2256 status = INVALID_OPERATION; 2257 } 2258 } else { 2259 status = BAD_VALUE; 2260 } 2261 } 2262 return status; 2263} 2264 2265void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2266{ 2267 for (size_t i = 0; i < mTracks.size(); ++i) { 2268 sp<Track> track = mTracks[i]; 2269 if (track->auxEffectId() == effectId) { 2270 attachAuxEffect_l(track, 0); 2271 } 2272 } 2273} 2274 2275bool AudioFlinger::PlaybackThread::threadLoop() 2276{ 2277 Vector< sp<Track> > tracksToRemove; 2278 2279 standbyTime = systemTime(); 2280 2281 // MIXER 2282 nsecs_t lastWarning = 0; 2283 2284 // DUPLICATING 2285 // FIXME could this be made local to while loop? 2286 writeFrames = 0; 2287 2288 int lastGeneration = 0; 2289 2290 cacheParameters_l(); 2291 sleepTime = idleSleepTime; 2292 2293 if (mType == MIXER) { 2294 sleepTimeShift = 0; 2295 } 2296 2297 CpuStats cpuStats; 2298 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2299 2300 acquireWakeLock(); 2301 2302 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2303 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2304 // and then that string will be logged at the next convenient opportunity. 2305 const char *logString = NULL; 2306 2307 checkSilentMode_l(); 2308 2309 while (!exitPending()) 2310 { 2311 cpuStats.sample(myName); 2312 2313 Vector< sp<EffectChain> > effectChains; 2314 2315 { // scope for mLock 2316 2317 Mutex::Autolock _l(mLock); 2318 2319 processConfigEvents_l(); 2320 2321 if (logString != NULL) { 2322 mNBLogWriter->logTimestamp(); 2323 mNBLogWriter->log(logString); 2324 logString = NULL; 2325 } 2326 2327 if (mLatchDValid) { 2328 mLatchQ = mLatchD; 2329 mLatchDValid = false; 2330 mLatchQValid = true; 2331 } 2332 2333 saveOutputTracks(); 2334 if (mSignalPending) { 2335 // A signal was raised while we were unlocked 2336 mSignalPending = false; 2337 } else if (waitingAsyncCallback_l()) { 2338 if (exitPending()) { 2339 break; 2340 } 2341 releaseWakeLock_l(); 2342 mWakeLockUids.clear(); 2343 mActiveTracksGeneration++; 2344 ALOGV("wait async completion"); 2345 mWaitWorkCV.wait(mLock); 2346 ALOGV("async completion/wake"); 2347 acquireWakeLock_l(); 2348 standbyTime = systemTime() + standbyDelay; 2349 sleepTime = 0; 2350 2351 continue; 2352 } 2353 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2354 isSuspended()) { 2355 // put audio hardware into standby after short delay 2356 if (shouldStandby_l()) { 2357 2358 threadLoop_standby(); 2359 2360 mStandby = true; 2361 } 2362 2363 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2364 // we're about to wait, flush the binder command buffer 2365 IPCThreadState::self()->flushCommands(); 2366 2367 clearOutputTracks(); 2368 2369 if (exitPending()) { 2370 break; 2371 } 2372 2373 releaseWakeLock_l(); 2374 mWakeLockUids.clear(); 2375 mActiveTracksGeneration++; 2376 // wait until we have something to do... 2377 ALOGV("%s going to sleep", myName.string()); 2378 mWaitWorkCV.wait(mLock); 2379 ALOGV("%s waking up", myName.string()); 2380 acquireWakeLock_l(); 2381 2382 mMixerStatus = MIXER_IDLE; 2383 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2384 mBytesWritten = 0; 2385 mBytesRemaining = 0; 2386 checkSilentMode_l(); 2387 2388 standbyTime = systemTime() + standbyDelay; 2389 sleepTime = idleSleepTime; 2390 if (mType == MIXER) { 2391 sleepTimeShift = 0; 2392 } 2393 2394 continue; 2395 } 2396 } 2397 // mMixerStatusIgnoringFastTracks is also updated internally 2398 mMixerStatus = prepareTracks_l(&tracksToRemove); 2399 2400 // compare with previously applied list 2401 if (lastGeneration != mActiveTracksGeneration) { 2402 // update wakelock 2403 updateWakeLockUids_l(mWakeLockUids); 2404 lastGeneration = mActiveTracksGeneration; 2405 } 2406 2407 // prevent any changes in effect chain list and in each effect chain 2408 // during mixing and effect process as the audio buffers could be deleted 2409 // or modified if an effect is created or deleted 2410 lockEffectChains_l(effectChains); 2411 } // mLock scope ends 2412 2413 if (mBytesRemaining == 0) { 2414 mCurrentWriteLength = 0; 2415 if (mMixerStatus == MIXER_TRACKS_READY) { 2416 // threadLoop_mix() sets mCurrentWriteLength 2417 threadLoop_mix(); 2418 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2419 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2420 // threadLoop_sleepTime sets sleepTime to 0 if data 2421 // must be written to HAL 2422 threadLoop_sleepTime(); 2423 if (sleepTime == 0) { 2424 mCurrentWriteLength = mSinkBufferSize; 2425 } 2426 } 2427 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2428 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2429 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2430 // or mSinkBuffer (if there are no effects). 2431 // 2432 // This is done pre-effects computation; if effects change to 2433 // support higher precision, this needs to move. 2434 // 2435 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2436 // TODO use sleepTime == 0 as an additional condition. 2437 if (mMixerBufferValid) { 2438 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2439 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2440 2441 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2442 mNormalFrameCount * mChannelCount); 2443 } 2444 2445 mBytesRemaining = mCurrentWriteLength; 2446 if (isSuspended()) { 2447 sleepTime = suspendSleepTimeUs(); 2448 // simulate write to HAL when suspended 2449 mBytesWritten += mSinkBufferSize; 2450 mBytesRemaining = 0; 2451 } 2452 2453 // only process effects if we're going to write 2454 if (sleepTime == 0 && mType != OFFLOAD) { 2455 for (size_t i = 0; i < effectChains.size(); i ++) { 2456 effectChains[i]->process_l(); 2457 } 2458 } 2459 } 2460 // Process effect chains for offloaded thread even if no audio 2461 // was read from audio track: process only updates effect state 2462 // and thus does have to be synchronized with audio writes but may have 2463 // to be called while waiting for async write callback 2464 if (mType == OFFLOAD) { 2465 for (size_t i = 0; i < effectChains.size(); i ++) { 2466 effectChains[i]->process_l(); 2467 } 2468 } 2469 2470 // Only if the Effects buffer is enabled and there is data in the 2471 // Effects buffer (buffer valid), we need to 2472 // copy into the sink buffer. 2473 // TODO use sleepTime == 0 as an additional condition. 2474 if (mEffectBufferValid) { 2475 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2476 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2477 mNormalFrameCount * mChannelCount); 2478 } 2479 2480 // enable changes in effect chain 2481 unlockEffectChains(effectChains); 2482 2483 if (!waitingAsyncCallback()) { 2484 // sleepTime == 0 means we must write to audio hardware 2485 if (sleepTime == 0) { 2486 if (mBytesRemaining) { 2487 ssize_t ret = threadLoop_write(); 2488 if (ret < 0) { 2489 mBytesRemaining = 0; 2490 } else { 2491 mBytesWritten += ret; 2492 mBytesRemaining -= ret; 2493 } 2494 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2495 (mMixerStatus == MIXER_DRAIN_ALL)) { 2496 threadLoop_drain(); 2497 } 2498 if (mType == MIXER) { 2499 // write blocked detection 2500 nsecs_t now = systemTime(); 2501 nsecs_t delta = now - mLastWriteTime; 2502 if (!mStandby && delta > maxPeriod) { 2503 mNumDelayedWrites++; 2504 if ((now - lastWarning) > kWarningThrottleNs) { 2505 ATRACE_NAME("underrun"); 2506 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2507 ns2ms(delta), mNumDelayedWrites, this); 2508 lastWarning = now; 2509 } 2510 } 2511 } 2512 2513 } else { 2514 usleep(sleepTime); 2515 } 2516 } 2517 2518 // Finally let go of removed track(s), without the lock held 2519 // since we can't guarantee the destructors won't acquire that 2520 // same lock. This will also mutate and push a new fast mixer state. 2521 threadLoop_removeTracks(tracksToRemove); 2522 tracksToRemove.clear(); 2523 2524 // FIXME I don't understand the need for this here; 2525 // it was in the original code but maybe the 2526 // assignment in saveOutputTracks() makes this unnecessary? 2527 clearOutputTracks(); 2528 2529 // Effect chains will be actually deleted here if they were removed from 2530 // mEffectChains list during mixing or effects processing 2531 effectChains.clear(); 2532 2533 // FIXME Note that the above .clear() is no longer necessary since effectChains 2534 // is now local to this block, but will keep it for now (at least until merge done). 2535 } 2536 2537 threadLoop_exit(); 2538 2539 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2540 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2541 // put output stream into standby mode 2542 if (!mStandby) { 2543 mOutput->stream->common.standby(&mOutput->stream->common); 2544 } 2545 } 2546 2547 releaseWakeLock(); 2548 mWakeLockUids.clear(); 2549 mActiveTracksGeneration++; 2550 2551 ALOGV("Thread %p type %d exiting", this, mType); 2552 return false; 2553} 2554 2555// removeTracks_l() must be called with ThreadBase::mLock held 2556void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2557{ 2558 size_t count = tracksToRemove.size(); 2559 if (count > 0) { 2560 for (size_t i=0 ; i<count ; i++) { 2561 const sp<Track>& track = tracksToRemove.itemAt(i); 2562 mActiveTracks.remove(track); 2563 mWakeLockUids.remove(track->uid()); 2564 mActiveTracksGeneration++; 2565 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2566 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2567 if (chain != 0) { 2568 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2569 track->sessionId()); 2570 chain->decActiveTrackCnt(); 2571 } 2572 if (track->isTerminated()) { 2573 removeTrack_l(track); 2574 } 2575 } 2576 } 2577 2578} 2579 2580status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2581{ 2582 if (mNormalSink != 0) { 2583 return mNormalSink->getTimestamp(timestamp); 2584 } 2585 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2586 uint64_t position64; 2587 int ret = mOutput->stream->get_presentation_position( 2588 mOutput->stream, &position64, ×tamp.mTime); 2589 if (ret == 0) { 2590 timestamp.mPosition = (uint32_t)position64; 2591 return NO_ERROR; 2592 } 2593 } 2594 return INVALID_OPERATION; 2595} 2596// ---------------------------------------------------------------------------- 2597 2598AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2599 audio_io_handle_t id, audio_devices_t device, type_t type) 2600 : PlaybackThread(audioFlinger, output, id, device, type), 2601 // mAudioMixer below 2602 // mFastMixer below 2603 mFastMixerFutex(0) 2604 // mOutputSink below 2605 // mPipeSink below 2606 // mNormalSink below 2607{ 2608 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2609 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2610 "mFrameCount=%d, mNormalFrameCount=%d", 2611 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2612 mNormalFrameCount); 2613 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2614 2615 // FIXME - Current mixer implementation only supports stereo output 2616 if (mChannelCount != FCC_2) { 2617 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2618 } 2619 2620 // create an NBAIO sink for the HAL output stream, and negotiate 2621 mOutputSink = new AudioStreamOutSink(output->stream); 2622 size_t numCounterOffers = 0; 2623 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2624 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2625 ALOG_ASSERT(index == 0); 2626 2627 // initialize fast mixer depending on configuration 2628 bool initFastMixer; 2629 switch (kUseFastMixer) { 2630 case FastMixer_Never: 2631 initFastMixer = false; 2632 break; 2633 case FastMixer_Always: 2634 initFastMixer = true; 2635 break; 2636 case FastMixer_Static: 2637 case FastMixer_Dynamic: 2638 initFastMixer = mFrameCount < mNormalFrameCount; 2639 break; 2640 } 2641 if (initFastMixer) { 2642 2643 // create a MonoPipe to connect our submix to FastMixer 2644 NBAIO_Format format = mOutputSink->format(); 2645 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2646 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2647 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2648 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2649 const NBAIO_Format offers[1] = {format}; 2650 size_t numCounterOffers = 0; 2651 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2652 ALOG_ASSERT(index == 0); 2653 monoPipe->setAvgFrames((mScreenState & 1) ? 2654 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2655 mPipeSink = monoPipe; 2656 2657#ifdef TEE_SINK 2658 if (mTeeSinkOutputEnabled) { 2659 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2660 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2661 numCounterOffers = 0; 2662 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2663 ALOG_ASSERT(index == 0); 2664 mTeeSink = teeSink; 2665 PipeReader *teeSource = new PipeReader(*teeSink); 2666 numCounterOffers = 0; 2667 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2668 ALOG_ASSERT(index == 0); 2669 mTeeSource = teeSource; 2670 } 2671#endif 2672 2673 // create fast mixer and configure it initially with just one fast track for our submix 2674 mFastMixer = new FastMixer(); 2675 FastMixerStateQueue *sq = mFastMixer->sq(); 2676#ifdef STATE_QUEUE_DUMP 2677 sq->setObserverDump(&mStateQueueObserverDump); 2678 sq->setMutatorDump(&mStateQueueMutatorDump); 2679#endif 2680 FastMixerState *state = sq->begin(); 2681 FastTrack *fastTrack = &state->mFastTracks[0]; 2682 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2683 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2684 fastTrack->mVolumeProvider = NULL; 2685 fastTrack->mGeneration++; 2686 state->mFastTracksGen++; 2687 state->mTrackMask = 1; 2688 // fast mixer will use the HAL output sink 2689 state->mOutputSink = mOutputSink.get(); 2690 state->mOutputSinkGen++; 2691 state->mFrameCount = mFrameCount; 2692 state->mCommand = FastMixerState::COLD_IDLE; 2693 // already done in constructor initialization list 2694 //mFastMixerFutex = 0; 2695 state->mColdFutexAddr = &mFastMixerFutex; 2696 state->mColdGen++; 2697 state->mDumpState = &mFastMixerDumpState; 2698#ifdef TEE_SINK 2699 state->mTeeSink = mTeeSink.get(); 2700#endif 2701 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2702 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2703 sq->end(); 2704 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2705 2706 // start the fast mixer 2707 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2708 pid_t tid = mFastMixer->getTid(); 2709 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2710 if (err != 0) { 2711 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2712 kPriorityFastMixer, getpid_cached, tid, err); 2713 } 2714 2715#ifdef AUDIO_WATCHDOG 2716 // create and start the watchdog 2717 mAudioWatchdog = new AudioWatchdog(); 2718 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2719 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2720 tid = mAudioWatchdog->getTid(); 2721 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2722 if (err != 0) { 2723 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2724 kPriorityFastMixer, getpid_cached, tid, err); 2725 } 2726#endif 2727 2728 } else { 2729 mFastMixer = NULL; 2730 } 2731 2732 switch (kUseFastMixer) { 2733 case FastMixer_Never: 2734 case FastMixer_Dynamic: 2735 mNormalSink = mOutputSink; 2736 break; 2737 case FastMixer_Always: 2738 mNormalSink = mPipeSink; 2739 break; 2740 case FastMixer_Static: 2741 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2742 break; 2743 } 2744} 2745 2746AudioFlinger::MixerThread::~MixerThread() 2747{ 2748 if (mFastMixer != NULL) { 2749 FastMixerStateQueue *sq = mFastMixer->sq(); 2750 FastMixerState *state = sq->begin(); 2751 if (state->mCommand == FastMixerState::COLD_IDLE) { 2752 int32_t old = android_atomic_inc(&mFastMixerFutex); 2753 if (old == -1) { 2754 (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2755 } 2756 } 2757 state->mCommand = FastMixerState::EXIT; 2758 sq->end(); 2759 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2760 mFastMixer->join(); 2761 // Though the fast mixer thread has exited, it's state queue is still valid. 2762 // We'll use that extract the final state which contains one remaining fast track 2763 // corresponding to our sub-mix. 2764 state = sq->begin(); 2765 ALOG_ASSERT(state->mTrackMask == 1); 2766 FastTrack *fastTrack = &state->mFastTracks[0]; 2767 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2768 delete fastTrack->mBufferProvider; 2769 sq->end(false /*didModify*/); 2770 delete mFastMixer; 2771#ifdef AUDIO_WATCHDOG 2772 if (mAudioWatchdog != 0) { 2773 mAudioWatchdog->requestExit(); 2774 mAudioWatchdog->requestExitAndWait(); 2775 mAudioWatchdog.clear(); 2776 } 2777#endif 2778 } 2779 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2780 delete mAudioMixer; 2781} 2782 2783 2784uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2785{ 2786 if (mFastMixer != NULL) { 2787 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2788 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2789 } 2790 return latency; 2791} 2792 2793 2794void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2795{ 2796 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2797} 2798 2799ssize_t AudioFlinger::MixerThread::threadLoop_write() 2800{ 2801 // FIXME we should only do one push per cycle; confirm this is true 2802 // Start the fast mixer if it's not already running 2803 if (mFastMixer != NULL) { 2804 FastMixerStateQueue *sq = mFastMixer->sq(); 2805 FastMixerState *state = sq->begin(); 2806 if (state->mCommand != FastMixerState::MIX_WRITE && 2807 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2808 if (state->mCommand == FastMixerState::COLD_IDLE) { 2809 int32_t old = android_atomic_inc(&mFastMixerFutex); 2810 if (old == -1) { 2811 (void) __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2812 } 2813#ifdef AUDIO_WATCHDOG 2814 if (mAudioWatchdog != 0) { 2815 mAudioWatchdog->resume(); 2816 } 2817#endif 2818 } 2819 state->mCommand = FastMixerState::MIX_WRITE; 2820 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2821 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2822 sq->end(); 2823 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2824 if (kUseFastMixer == FastMixer_Dynamic) { 2825 mNormalSink = mPipeSink; 2826 } 2827 } else { 2828 sq->end(false /*didModify*/); 2829 } 2830 } 2831 return PlaybackThread::threadLoop_write(); 2832} 2833 2834void AudioFlinger::MixerThread::threadLoop_standby() 2835{ 2836 // Idle the fast mixer if it's currently running 2837 if (mFastMixer != NULL) { 2838 FastMixerStateQueue *sq = mFastMixer->sq(); 2839 FastMixerState *state = sq->begin(); 2840 if (!(state->mCommand & FastMixerState::IDLE)) { 2841 state->mCommand = FastMixerState::COLD_IDLE; 2842 state->mColdFutexAddr = &mFastMixerFutex; 2843 state->mColdGen++; 2844 mFastMixerFutex = 0; 2845 sq->end(); 2846 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2847 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2848 if (kUseFastMixer == FastMixer_Dynamic) { 2849 mNormalSink = mOutputSink; 2850 } 2851#ifdef AUDIO_WATCHDOG 2852 if (mAudioWatchdog != 0) { 2853 mAudioWatchdog->pause(); 2854 } 2855#endif 2856 } else { 2857 sq->end(false /*didModify*/); 2858 } 2859 } 2860 PlaybackThread::threadLoop_standby(); 2861} 2862 2863bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2864{ 2865 return false; 2866} 2867 2868bool AudioFlinger::PlaybackThread::shouldStandby_l() 2869{ 2870 return !mStandby; 2871} 2872 2873bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2874{ 2875 Mutex::Autolock _l(mLock); 2876 return waitingAsyncCallback_l(); 2877} 2878 2879// shared by MIXER and DIRECT, overridden by DUPLICATING 2880void AudioFlinger::PlaybackThread::threadLoop_standby() 2881{ 2882 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2883 mOutput->stream->common.standby(&mOutput->stream->common); 2884 if (mUseAsyncWrite != 0) { 2885 // discard any pending drain or write ack by incrementing sequence 2886 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2887 mDrainSequence = (mDrainSequence + 2) & ~1; 2888 ALOG_ASSERT(mCallbackThread != 0); 2889 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2890 mCallbackThread->setDraining(mDrainSequence); 2891 } 2892} 2893 2894void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2895{ 2896 ALOGV("signal playback thread"); 2897 broadcast_l(); 2898} 2899 2900void AudioFlinger::MixerThread::threadLoop_mix() 2901{ 2902 // obtain the presentation timestamp of the next output buffer 2903 int64_t pts; 2904 status_t status = INVALID_OPERATION; 2905 2906 if (mNormalSink != 0) { 2907 status = mNormalSink->getNextWriteTimestamp(&pts); 2908 } else { 2909 status = mOutputSink->getNextWriteTimestamp(&pts); 2910 } 2911 2912 if (status != NO_ERROR) { 2913 pts = AudioBufferProvider::kInvalidPTS; 2914 } 2915 2916 // mix buffers... 2917 mAudioMixer->process(pts); 2918 mCurrentWriteLength = mSinkBufferSize; 2919 // increase sleep time progressively when application underrun condition clears. 2920 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2921 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2922 // such that we would underrun the audio HAL. 2923 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2924 sleepTimeShift--; 2925 } 2926 sleepTime = 0; 2927 standbyTime = systemTime() + standbyDelay; 2928 //TODO: delay standby when effects have a tail 2929} 2930 2931void AudioFlinger::MixerThread::threadLoop_sleepTime() 2932{ 2933 // If no tracks are ready, sleep once for the duration of an output 2934 // buffer size, then write 0s to the output 2935 if (sleepTime == 0) { 2936 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2937 sleepTime = activeSleepTime >> sleepTimeShift; 2938 if (sleepTime < kMinThreadSleepTimeUs) { 2939 sleepTime = kMinThreadSleepTimeUs; 2940 } 2941 // reduce sleep time in case of consecutive application underruns to avoid 2942 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2943 // duration we would end up writing less data than needed by the audio HAL if 2944 // the condition persists. 2945 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2946 sleepTimeShift++; 2947 } 2948 } else { 2949 sleepTime = idleSleepTime; 2950 } 2951 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2952 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2953 // before effects processing or output. 2954 if (mMixerBufferValid) { 2955 memset(mMixerBuffer, 0, mMixerBufferSize); 2956 } else { 2957 memset(mSinkBuffer, 0, mSinkBufferSize); 2958 } 2959 sleepTime = 0; 2960 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2961 "anticipated start"); 2962 } 2963 // TODO add standby time extension fct of effect tail 2964} 2965 2966// prepareTracks_l() must be called with ThreadBase::mLock held 2967AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2968 Vector< sp<Track> > *tracksToRemove) 2969{ 2970 2971 mixer_state mixerStatus = MIXER_IDLE; 2972 // find out which tracks need to be processed 2973 size_t count = mActiveTracks.size(); 2974 size_t mixedTracks = 0; 2975 size_t tracksWithEffect = 0; 2976 // counts only _active_ fast tracks 2977 size_t fastTracks = 0; 2978 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2979 2980 float masterVolume = mMasterVolume; 2981 bool masterMute = mMasterMute; 2982 2983 if (masterMute) { 2984 masterVolume = 0; 2985 } 2986 // Delegate master volume control to effect in output mix effect chain if needed 2987 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2988 if (chain != 0) { 2989 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2990 chain->setVolume_l(&v, &v); 2991 masterVolume = (float)((v + (1 << 23)) >> 24); 2992 chain.clear(); 2993 } 2994 2995 // prepare a new state to push 2996 FastMixerStateQueue *sq = NULL; 2997 FastMixerState *state = NULL; 2998 bool didModify = false; 2999 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3000 if (mFastMixer != NULL) { 3001 sq = mFastMixer->sq(); 3002 state = sq->begin(); 3003 } 3004 3005 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3006 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3007 3008 for (size_t i=0 ; i<count ; i++) { 3009 const sp<Track> t = mActiveTracks[i].promote(); 3010 if (t == 0) { 3011 continue; 3012 } 3013 3014 // this const just means the local variable doesn't change 3015 Track* const track = t.get(); 3016 3017 // process fast tracks 3018 if (track->isFastTrack()) { 3019 3020 // It's theoretically possible (though unlikely) for a fast track to be created 3021 // and then removed within the same normal mix cycle. This is not a problem, as 3022 // the track never becomes active so it's fast mixer slot is never touched. 3023 // The converse, of removing an (active) track and then creating a new track 3024 // at the identical fast mixer slot within the same normal mix cycle, 3025 // is impossible because the slot isn't marked available until the end of each cycle. 3026 int j = track->mFastIndex; 3027 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3028 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3029 FastTrack *fastTrack = &state->mFastTracks[j]; 3030 3031 // Determine whether the track is currently in underrun condition, 3032 // and whether it had a recent underrun. 3033 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3034 FastTrackUnderruns underruns = ftDump->mUnderruns; 3035 uint32_t recentFull = (underruns.mBitFields.mFull - 3036 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3037 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3038 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3039 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3040 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3041 uint32_t recentUnderruns = recentPartial + recentEmpty; 3042 track->mObservedUnderruns = underruns; 3043 // don't count underruns that occur while stopping or pausing 3044 // or stopped which can occur when flush() is called while active 3045 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3046 recentUnderruns > 0) { 3047 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3048 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3049 } 3050 3051 // This is similar to the state machine for normal tracks, 3052 // with a few modifications for fast tracks. 3053 bool isActive = true; 3054 switch (track->mState) { 3055 case TrackBase::STOPPING_1: 3056 // track stays active in STOPPING_1 state until first underrun 3057 if (recentUnderruns > 0 || track->isTerminated()) { 3058 track->mState = TrackBase::STOPPING_2; 3059 } 3060 break; 3061 case TrackBase::PAUSING: 3062 // ramp down is not yet implemented 3063 track->setPaused(); 3064 break; 3065 case TrackBase::RESUMING: 3066 // ramp up is not yet implemented 3067 track->mState = TrackBase::ACTIVE; 3068 break; 3069 case TrackBase::ACTIVE: 3070 if (recentFull > 0 || recentPartial > 0) { 3071 // track has provided at least some frames recently: reset retry count 3072 track->mRetryCount = kMaxTrackRetries; 3073 } 3074 if (recentUnderruns == 0) { 3075 // no recent underruns: stay active 3076 break; 3077 } 3078 // there has recently been an underrun of some kind 3079 if (track->sharedBuffer() == 0) { 3080 // were any of the recent underruns "empty" (no frames available)? 3081 if (recentEmpty == 0) { 3082 // no, then ignore the partial underruns as they are allowed indefinitely 3083 break; 3084 } 3085 // there has recently been an "empty" underrun: decrement the retry counter 3086 if (--(track->mRetryCount) > 0) { 3087 break; 3088 } 3089 // indicate to client process that the track was disabled because of underrun; 3090 // it will then automatically call start() when data is available 3091 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3092 // remove from active list, but state remains ACTIVE [confusing but true] 3093 isActive = false; 3094 break; 3095 } 3096 // fall through 3097 case TrackBase::STOPPING_2: 3098 case TrackBase::PAUSED: 3099 case TrackBase::STOPPED: 3100 case TrackBase::FLUSHED: // flush() while active 3101 // Check for presentation complete if track is inactive 3102 // We have consumed all the buffers of this track. 3103 // This would be incomplete if we auto-paused on underrun 3104 { 3105 size_t audioHALFrames = 3106 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3107 size_t framesWritten = mBytesWritten / mFrameSize; 3108 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3109 // track stays in active list until presentation is complete 3110 break; 3111 } 3112 } 3113 if (track->isStopping_2()) { 3114 track->mState = TrackBase::STOPPED; 3115 } 3116 if (track->isStopped()) { 3117 // Can't reset directly, as fast mixer is still polling this track 3118 // track->reset(); 3119 // So instead mark this track as needing to be reset after push with ack 3120 resetMask |= 1 << i; 3121 } 3122 isActive = false; 3123 break; 3124 case TrackBase::IDLE: 3125 default: 3126 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3127 } 3128 3129 if (isActive) { 3130 // was it previously inactive? 3131 if (!(state->mTrackMask & (1 << j))) { 3132 ExtendedAudioBufferProvider *eabp = track; 3133 VolumeProvider *vp = track; 3134 fastTrack->mBufferProvider = eabp; 3135 fastTrack->mVolumeProvider = vp; 3136 fastTrack->mChannelMask = track->mChannelMask; 3137 fastTrack->mGeneration++; 3138 state->mTrackMask |= 1 << j; 3139 didModify = true; 3140 // no acknowledgement required for newly active tracks 3141 } 3142 // cache the combined master volume and stream type volume for fast mixer; this 3143 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3144 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3145 ++fastTracks; 3146 } else { 3147 // was it previously active? 3148 if (state->mTrackMask & (1 << j)) { 3149 fastTrack->mBufferProvider = NULL; 3150 fastTrack->mGeneration++; 3151 state->mTrackMask &= ~(1 << j); 3152 didModify = true; 3153 // If any fast tracks were removed, we must wait for acknowledgement 3154 // because we're about to decrement the last sp<> on those tracks. 3155 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3156 } else { 3157 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3158 } 3159 tracksToRemove->add(track); 3160 // Avoids a misleading display in dumpsys 3161 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3162 } 3163 continue; 3164 } 3165 3166 { // local variable scope to avoid goto warning 3167 3168 audio_track_cblk_t* cblk = track->cblk(); 3169 3170 // The first time a track is added we wait 3171 // for all its buffers to be filled before processing it 3172 int name = track->name(); 3173 // make sure that we have enough frames to mix one full buffer. 3174 // enforce this condition only once to enable draining the buffer in case the client 3175 // app does not call stop() and relies on underrun to stop: 3176 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3177 // during last round 3178 size_t desiredFrames; 3179 uint32_t sr = track->sampleRate(); 3180 if (sr == mSampleRate) { 3181 desiredFrames = mNormalFrameCount; 3182 } else { 3183 // +1 for rounding and +1 for additional sample needed for interpolation 3184 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3185 // add frames already consumed but not yet released by the resampler 3186 // because mAudioTrackServerProxy->framesReady() will include these frames 3187 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3188#if 0 3189 // the minimum track buffer size is normally twice the number of frames necessary 3190 // to fill one buffer and the resampler should not leave more than one buffer worth 3191 // of unreleased frames after each pass, but just in case... 3192 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3193#endif 3194 } 3195 uint32_t minFrames = 1; 3196 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3197 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3198 minFrames = desiredFrames; 3199 } 3200 3201 size_t framesReady = track->framesReady(); 3202 if ((framesReady >= minFrames) && track->isReady() && 3203 !track->isPaused() && !track->isTerminated()) 3204 { 3205 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3206 3207 mixedTracks++; 3208 3209 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3210 // there is an effect chain connected to the track 3211 chain.clear(); 3212 if (track->mainBuffer() != mSinkBuffer && 3213 track->mainBuffer() != mMixerBuffer) { 3214 if (mEffectBufferEnabled) { 3215 mEffectBufferValid = true; // Later can set directly. 3216 } 3217 chain = getEffectChain_l(track->sessionId()); 3218 // Delegate volume control to effect in track effect chain if needed 3219 if (chain != 0) { 3220 tracksWithEffect++; 3221 } else { 3222 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3223 "session %d", 3224 name, track->sessionId()); 3225 } 3226 } 3227 3228 3229 int param = AudioMixer::VOLUME; 3230 if (track->mFillingUpStatus == Track::FS_FILLED) { 3231 // no ramp for the first volume setting 3232 track->mFillingUpStatus = Track::FS_ACTIVE; 3233 if (track->mState == TrackBase::RESUMING) { 3234 track->mState = TrackBase::ACTIVE; 3235 param = AudioMixer::RAMP_VOLUME; 3236 } 3237 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3238 // FIXME should not make a decision based on mServer 3239 } else if (cblk->mServer != 0) { 3240 // If the track is stopped before the first frame was mixed, 3241 // do not apply ramp 3242 param = AudioMixer::RAMP_VOLUME; 3243 } 3244 3245 // compute volume for this track 3246 uint32_t vl, vr, va; 3247 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3248 vl = vr = va = 0; 3249 if (track->isPausing()) { 3250 track->setPaused(); 3251 } 3252 } else { 3253 3254 // read original volumes with volume control 3255 float typeVolume = mStreamTypes[track->streamType()].volume; 3256 float v = masterVolume * typeVolume; 3257 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3258 uint32_t vlr = proxy->getVolumeLR(); 3259 vl = vlr & 0xFFFF; 3260 vr = vlr >> 16; 3261 // track volumes come from shared memory, so can't be trusted and must be clamped 3262 if (vl > MAX_GAIN_INT) { 3263 ALOGV("Track left volume out of range: %04X", vl); 3264 vl = MAX_GAIN_INT; 3265 } 3266 if (vr > MAX_GAIN_INT) { 3267 ALOGV("Track right volume out of range: %04X", vr); 3268 vr = MAX_GAIN_INT; 3269 } 3270 // now apply the master volume and stream type volume 3271 vl = (uint32_t)(v * vl) << 12; 3272 vr = (uint32_t)(v * vr) << 12; 3273 // assuming master volume and stream type volume each go up to 1.0, 3274 // vl and vr are now in 8.24 format 3275 3276 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3277 // send level comes from shared memory and so may be corrupt 3278 if (sendLevel > MAX_GAIN_INT) { 3279 ALOGV("Track send level out of range: %04X", sendLevel); 3280 sendLevel = MAX_GAIN_INT; 3281 } 3282 va = (uint32_t)(v * sendLevel); 3283 } 3284 3285 // Delegate volume control to effect in track effect chain if needed 3286 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3287 // Do not ramp volume if volume is controlled by effect 3288 param = AudioMixer::VOLUME; 3289 track->mHasVolumeController = true; 3290 } else { 3291 // force no volume ramp when volume controller was just disabled or removed 3292 // from effect chain to avoid volume spike 3293 if (track->mHasVolumeController) { 3294 param = AudioMixer::VOLUME; 3295 } 3296 track->mHasVolumeController = false; 3297 } 3298 3299 // Convert volumes from 8.24 to 4.12 format 3300 // This additional clamping is needed in case chain->setVolume_l() overshot 3301 vl = (vl + (1 << 11)) >> 12; 3302 if (vl > MAX_GAIN_INT) { 3303 vl = MAX_GAIN_INT; 3304 } 3305 vr = (vr + (1 << 11)) >> 12; 3306 if (vr > MAX_GAIN_INT) { 3307 vr = MAX_GAIN_INT; 3308 } 3309 3310 if (va > MAX_GAIN_INT) { 3311 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3312 } 3313 3314 // XXX: these things DON'T need to be done each time 3315 mAudioMixer->setBufferProvider(name, track); 3316 mAudioMixer->enable(name); 3317 3318 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3319 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3320 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3321 mAudioMixer->setParameter( 3322 name, 3323 AudioMixer::TRACK, 3324 AudioMixer::FORMAT, (void *)track->format()); 3325 mAudioMixer->setParameter( 3326 name, 3327 AudioMixer::TRACK, 3328 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3329 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3330 uint32_t maxSampleRate = mSampleRate * 2; 3331 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3332 if (reqSampleRate == 0) { 3333 reqSampleRate = mSampleRate; 3334 } else if (reqSampleRate > maxSampleRate) { 3335 reqSampleRate = maxSampleRate; 3336 } 3337 mAudioMixer->setParameter( 3338 name, 3339 AudioMixer::RESAMPLE, 3340 AudioMixer::SAMPLE_RATE, 3341 (void *)(uintptr_t)reqSampleRate); 3342 /* 3343 * Select the appropriate output buffer for the track. 3344 * 3345 * Tracks with effects go into their own effects chain buffer 3346 * and from there into either mEffectBuffer or mSinkBuffer. 3347 * 3348 * Other tracks can use mMixerBuffer for higher precision 3349 * channel accumulation. If this buffer is enabled 3350 * (mMixerBufferEnabled true), then selected tracks will accumulate 3351 * into it. 3352 * 3353 */ 3354 if (mMixerBufferEnabled 3355 && (track->mainBuffer() == mSinkBuffer 3356 || track->mainBuffer() == mMixerBuffer)) { 3357 mAudioMixer->setParameter( 3358 name, 3359 AudioMixer::TRACK, 3360 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3361 mAudioMixer->setParameter( 3362 name, 3363 AudioMixer::TRACK, 3364 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3365 // TODO: override track->mainBuffer()? 3366 mMixerBufferValid = true; 3367 } else { 3368 mAudioMixer->setParameter( 3369 name, 3370 AudioMixer::TRACK, 3371 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3372 mAudioMixer->setParameter( 3373 name, 3374 AudioMixer::TRACK, 3375 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3376 } 3377 mAudioMixer->setParameter( 3378 name, 3379 AudioMixer::TRACK, 3380 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3381 3382 // reset retry count 3383 track->mRetryCount = kMaxTrackRetries; 3384 3385 // If one track is ready, set the mixer ready if: 3386 // - the mixer was not ready during previous round OR 3387 // - no other track is not ready 3388 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3389 mixerStatus != MIXER_TRACKS_ENABLED) { 3390 mixerStatus = MIXER_TRACKS_READY; 3391 } 3392 } else { 3393 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3394 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3395 } 3396 // clear effect chain input buffer if an active track underruns to avoid sending 3397 // previous audio buffer again to effects 3398 chain = getEffectChain_l(track->sessionId()); 3399 if (chain != 0) { 3400 chain->clearInputBuffer(); 3401 } 3402 3403 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3404 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3405 track->isStopped() || track->isPaused()) { 3406 // We have consumed all the buffers of this track. 3407 // Remove it from the list of active tracks. 3408 // TODO: use actual buffer filling status instead of latency when available from 3409 // audio HAL 3410 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3411 size_t framesWritten = mBytesWritten / mFrameSize; 3412 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3413 if (track->isStopped()) { 3414 track->reset(); 3415 } 3416 tracksToRemove->add(track); 3417 } 3418 } else { 3419 // No buffers for this track. Give it a few chances to 3420 // fill a buffer, then remove it from active list. 3421 if (--(track->mRetryCount) <= 0) { 3422 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3423 tracksToRemove->add(track); 3424 // indicate to client process that the track was disabled because of underrun; 3425 // it will then automatically call start() when data is available 3426 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3427 // If one track is not ready, mark the mixer also not ready if: 3428 // - the mixer was ready during previous round OR 3429 // - no other track is ready 3430 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3431 mixerStatus != MIXER_TRACKS_READY) { 3432 mixerStatus = MIXER_TRACKS_ENABLED; 3433 } 3434 } 3435 mAudioMixer->disable(name); 3436 } 3437 3438 } // local variable scope to avoid goto warning 3439track_is_ready: ; 3440 3441 } 3442 3443 // Push the new FastMixer state if necessary 3444 bool pauseAudioWatchdog = false; 3445 if (didModify) { 3446 state->mFastTracksGen++; 3447 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3448 if (kUseFastMixer == FastMixer_Dynamic && 3449 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3450 state->mCommand = FastMixerState::COLD_IDLE; 3451 state->mColdFutexAddr = &mFastMixerFutex; 3452 state->mColdGen++; 3453 mFastMixerFutex = 0; 3454 if (kUseFastMixer == FastMixer_Dynamic) { 3455 mNormalSink = mOutputSink; 3456 } 3457 // If we go into cold idle, need to wait for acknowledgement 3458 // so that fast mixer stops doing I/O. 3459 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3460 pauseAudioWatchdog = true; 3461 } 3462 } 3463 if (sq != NULL) { 3464 sq->end(didModify); 3465 sq->push(block); 3466 } 3467#ifdef AUDIO_WATCHDOG 3468 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3469 mAudioWatchdog->pause(); 3470 } 3471#endif 3472 3473 // Now perform the deferred reset on fast tracks that have stopped 3474 while (resetMask != 0) { 3475 size_t i = __builtin_ctz(resetMask); 3476 ALOG_ASSERT(i < count); 3477 resetMask &= ~(1 << i); 3478 sp<Track> t = mActiveTracks[i].promote(); 3479 if (t == 0) { 3480 continue; 3481 } 3482 Track* track = t.get(); 3483 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3484 track->reset(); 3485 } 3486 3487 // remove all the tracks that need to be... 3488 removeTracks_l(*tracksToRemove); 3489 3490 // sink or mix buffer must be cleared if all tracks are connected to an 3491 // effect chain as in this case the mixer will not write to the sink or mix buffer 3492 // and track effects will accumulate into it 3493 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3494 (mixedTracks == 0 && fastTracks > 0))) { 3495 // FIXME as a performance optimization, should remember previous zero status 3496 if (mMixerBufferValid) { 3497 memset(mMixerBuffer, 0, mMixerBufferSize); 3498 // TODO: In testing, mSinkBuffer below need not be cleared because 3499 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3500 // after mixing. 3501 // 3502 // To enforce this guarantee: 3503 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3504 // (mixedTracks == 0 && fastTracks > 0)) 3505 // must imply MIXER_TRACKS_READY. 3506 // Later, we may clear buffers regardless, and skip much of this logic. 3507 } 3508 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3509 if (mEffectBufferValid) { 3510 memset(mEffectBuffer, 0, mEffectBufferSize); 3511 } 3512 // FIXME as a performance optimization, should remember previous zero status 3513 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3514 } 3515 3516 // if any fast tracks, then status is ready 3517 mMixerStatusIgnoringFastTracks = mixerStatus; 3518 if (fastTracks > 0) { 3519 mixerStatus = MIXER_TRACKS_READY; 3520 } 3521 return mixerStatus; 3522} 3523 3524// getTrackName_l() must be called with ThreadBase::mLock held 3525int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3526{ 3527 return mAudioMixer->getTrackName(channelMask, sessionId); 3528} 3529 3530// deleteTrackName_l() must be called with ThreadBase::mLock held 3531void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3532{ 3533 ALOGV("remove track (%d) and delete from mixer", name); 3534 mAudioMixer->deleteTrackName(name); 3535} 3536 3537// checkForNewParameter_l() must be called with ThreadBase::mLock held 3538bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3539 status_t& status) 3540{ 3541 bool reconfig = false; 3542 3543 status = NO_ERROR; 3544 3545 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3546 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3547 if (mFastMixer != NULL) { 3548 FastMixerStateQueue *sq = mFastMixer->sq(); 3549 FastMixerState *state = sq->begin(); 3550 if (!(state->mCommand & FastMixerState::IDLE)) { 3551 previousCommand = state->mCommand; 3552 state->mCommand = FastMixerState::HOT_IDLE; 3553 sq->end(); 3554 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3555 } else { 3556 sq->end(false /*didModify*/); 3557 } 3558 } 3559 3560 AudioParameter param = AudioParameter(keyValuePair); 3561 int value; 3562 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3563 reconfig = true; 3564 } 3565 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3566 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3567 status = BAD_VALUE; 3568 } else { 3569 // no need to save value, since it's constant 3570 reconfig = true; 3571 } 3572 } 3573 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3574 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3575 status = BAD_VALUE; 3576 } else { 3577 // no need to save value, since it's constant 3578 reconfig = true; 3579 } 3580 } 3581 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3582 // do not accept frame count changes if tracks are open as the track buffer 3583 // size depends on frame count and correct behavior would not be guaranteed 3584 // if frame count is changed after track creation 3585 if (!mTracks.isEmpty()) { 3586 status = INVALID_OPERATION; 3587 } else { 3588 reconfig = true; 3589 } 3590 } 3591 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3592#ifdef ADD_BATTERY_DATA 3593 // when changing the audio output device, call addBatteryData to notify 3594 // the change 3595 if (mOutDevice != value) { 3596 uint32_t params = 0; 3597 // check whether speaker is on 3598 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3599 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3600 } 3601 3602 audio_devices_t deviceWithoutSpeaker 3603 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3604 // check if any other device (except speaker) is on 3605 if (value & deviceWithoutSpeaker ) { 3606 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3607 } 3608 3609 if (params != 0) { 3610 addBatteryData(params); 3611 } 3612 } 3613#endif 3614 3615 // forward device change to effects that have requested to be 3616 // aware of attached audio device. 3617 if (value != AUDIO_DEVICE_NONE) { 3618 mOutDevice = value; 3619 for (size_t i = 0; i < mEffectChains.size(); i++) { 3620 mEffectChains[i]->setDevice_l(mOutDevice); 3621 } 3622 } 3623 } 3624 3625 if (status == NO_ERROR) { 3626 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3627 keyValuePair.string()); 3628 if (!mStandby && status == INVALID_OPERATION) { 3629 mOutput->stream->common.standby(&mOutput->stream->common); 3630 mStandby = true; 3631 mBytesWritten = 0; 3632 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3633 keyValuePair.string()); 3634 } 3635 if (status == NO_ERROR && reconfig) { 3636 readOutputParameters_l(); 3637 delete mAudioMixer; 3638 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3639 for (size_t i = 0; i < mTracks.size() ; i++) { 3640 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3641 if (name < 0) { 3642 break; 3643 } 3644 mTracks[i]->mName = name; 3645 } 3646 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3647 } 3648 } 3649 3650 if (!(previousCommand & FastMixerState::IDLE)) { 3651 ALOG_ASSERT(mFastMixer != NULL); 3652 FastMixerStateQueue *sq = mFastMixer->sq(); 3653 FastMixerState *state = sq->begin(); 3654 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3655 state->mCommand = previousCommand; 3656 sq->end(); 3657 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3658 } 3659 3660 return reconfig; 3661} 3662 3663 3664void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3665{ 3666 const size_t SIZE = 256; 3667 char buffer[SIZE]; 3668 String8 result; 3669 3670 PlaybackThread::dumpInternals(fd, args); 3671 3672 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3673 3674 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3675 const FastMixerDumpState copy(mFastMixerDumpState); 3676 copy.dump(fd); 3677 3678#ifdef STATE_QUEUE_DUMP 3679 // Similar for state queue 3680 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3681 observerCopy.dump(fd); 3682 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3683 mutatorCopy.dump(fd); 3684#endif 3685 3686#ifdef TEE_SINK 3687 // Write the tee output to a .wav file 3688 dumpTee(fd, mTeeSource, mId); 3689#endif 3690 3691#ifdef AUDIO_WATCHDOG 3692 if (mAudioWatchdog != 0) { 3693 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3694 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3695 wdCopy.dump(fd); 3696 } 3697#endif 3698} 3699 3700uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3701{ 3702 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3703} 3704 3705uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3706{ 3707 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3708} 3709 3710void AudioFlinger::MixerThread::cacheParameters_l() 3711{ 3712 PlaybackThread::cacheParameters_l(); 3713 3714 // FIXME: Relaxed timing because of a certain device that can't meet latency 3715 // Should be reduced to 2x after the vendor fixes the driver issue 3716 // increase threshold again due to low power audio mode. The way this warning 3717 // threshold is calculated and its usefulness should be reconsidered anyway. 3718 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3719} 3720 3721// ---------------------------------------------------------------------------- 3722 3723AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3724 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3725 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3726 // mLeftVolFloat, mRightVolFloat 3727{ 3728} 3729 3730AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3731 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3732 ThreadBase::type_t type) 3733 : PlaybackThread(audioFlinger, output, id, device, type) 3734 // mLeftVolFloat, mRightVolFloat 3735{ 3736} 3737 3738AudioFlinger::DirectOutputThread::~DirectOutputThread() 3739{ 3740} 3741 3742void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3743{ 3744 audio_track_cblk_t* cblk = track->cblk(); 3745 float left, right; 3746 3747 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3748 left = right = 0; 3749 } else { 3750 float typeVolume = mStreamTypes[track->streamType()].volume; 3751 float v = mMasterVolume * typeVolume; 3752 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3753 uint32_t vlr = proxy->getVolumeLR(); 3754 float v_clamped = v * (vlr & 0xFFFF); 3755 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3756 left = v_clamped/MAX_GAIN; 3757 v_clamped = v * (vlr >> 16); 3758 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3759 right = v_clamped/MAX_GAIN; 3760 } 3761 3762 if (lastTrack) { 3763 if (left != mLeftVolFloat || right != mRightVolFloat) { 3764 mLeftVolFloat = left; 3765 mRightVolFloat = right; 3766 3767 // Convert volumes from float to 8.24 3768 uint32_t vl = (uint32_t)(left * (1 << 24)); 3769 uint32_t vr = (uint32_t)(right * (1 << 24)); 3770 3771 // Delegate volume control to effect in track effect chain if needed 3772 // only one effect chain can be present on DirectOutputThread, so if 3773 // there is one, the track is connected to it 3774 if (!mEffectChains.isEmpty()) { 3775 mEffectChains[0]->setVolume_l(&vl, &vr); 3776 left = (float)vl / (1 << 24); 3777 right = (float)vr / (1 << 24); 3778 } 3779 if (mOutput->stream->set_volume) { 3780 mOutput->stream->set_volume(mOutput->stream, left, right); 3781 } 3782 } 3783 } 3784} 3785 3786 3787AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3788 Vector< sp<Track> > *tracksToRemove 3789) 3790{ 3791 size_t count = mActiveTracks.size(); 3792 mixer_state mixerStatus = MIXER_IDLE; 3793 3794 // find out which tracks need to be processed 3795 for (size_t i = 0; i < count; i++) { 3796 sp<Track> t = mActiveTracks[i].promote(); 3797 // The track died recently 3798 if (t == 0) { 3799 continue; 3800 } 3801 3802 Track* const track = t.get(); 3803 audio_track_cblk_t* cblk = track->cblk(); 3804 // Only consider last track started for volume and mixer state control. 3805 // In theory an older track could underrun and restart after the new one starts 3806 // but as we only care about the transition phase between two tracks on a 3807 // direct output, it is not a problem to ignore the underrun case. 3808 sp<Track> l = mLatestActiveTrack.promote(); 3809 bool last = l.get() == track; 3810 3811 // The first time a track is added we wait 3812 // for all its buffers to be filled before processing it 3813 uint32_t minFrames; 3814 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3815 minFrames = mNormalFrameCount; 3816 } else { 3817 minFrames = 1; 3818 } 3819 3820 if ((track->framesReady() >= minFrames) && track->isReady() && 3821 !track->isPaused() && !track->isTerminated()) 3822 { 3823 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3824 3825 if (track->mFillingUpStatus == Track::FS_FILLED) { 3826 track->mFillingUpStatus = Track::FS_ACTIVE; 3827 // make sure processVolume_l() will apply new volume even if 0 3828 mLeftVolFloat = mRightVolFloat = -1.0; 3829 if (track->mState == TrackBase::RESUMING) { 3830 track->mState = TrackBase::ACTIVE; 3831 } 3832 } 3833 3834 // compute volume for this track 3835 processVolume_l(track, last); 3836 if (last) { 3837 // reset retry count 3838 track->mRetryCount = kMaxTrackRetriesDirect; 3839 mActiveTrack = t; 3840 mixerStatus = MIXER_TRACKS_READY; 3841 } 3842 } else { 3843 // clear effect chain input buffer if the last active track started underruns 3844 // to avoid sending previous audio buffer again to effects 3845 if (!mEffectChains.isEmpty() && last) { 3846 mEffectChains[0]->clearInputBuffer(); 3847 } 3848 3849 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3850 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3851 track->isStopped() || track->isPaused()) { 3852 // We have consumed all the buffers of this track. 3853 // Remove it from the list of active tracks. 3854 // TODO: implement behavior for compressed audio 3855 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3856 size_t framesWritten = mBytesWritten / mFrameSize; 3857 if (mStandby || !last || 3858 track->presentationComplete(framesWritten, audioHALFrames)) { 3859 if (track->isStopped()) { 3860 track->reset(); 3861 } 3862 tracksToRemove->add(track); 3863 } 3864 } else { 3865 // No buffers for this track. Give it a few chances to 3866 // fill a buffer, then remove it from active list. 3867 // Only consider last track started for mixer state control 3868 if (--(track->mRetryCount) <= 0) { 3869 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3870 tracksToRemove->add(track); 3871 // indicate to client process that the track was disabled because of underrun; 3872 // it will then automatically call start() when data is available 3873 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3874 } else if (last) { 3875 mixerStatus = MIXER_TRACKS_ENABLED; 3876 } 3877 } 3878 } 3879 } 3880 3881 // remove all the tracks that need to be... 3882 removeTracks_l(*tracksToRemove); 3883 3884 return mixerStatus; 3885} 3886 3887void AudioFlinger::DirectOutputThread::threadLoop_mix() 3888{ 3889 size_t frameCount = mFrameCount; 3890 int8_t *curBuf = (int8_t *)mSinkBuffer; 3891 // output audio to hardware 3892 while (frameCount) { 3893 AudioBufferProvider::Buffer buffer; 3894 buffer.frameCount = frameCount; 3895 mActiveTrack->getNextBuffer(&buffer); 3896 if (buffer.raw == NULL) { 3897 memset(curBuf, 0, frameCount * mFrameSize); 3898 break; 3899 } 3900 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3901 frameCount -= buffer.frameCount; 3902 curBuf += buffer.frameCount * mFrameSize; 3903 mActiveTrack->releaseBuffer(&buffer); 3904 } 3905 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3906 sleepTime = 0; 3907 standbyTime = systemTime() + standbyDelay; 3908 mActiveTrack.clear(); 3909} 3910 3911void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3912{ 3913 if (sleepTime == 0) { 3914 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3915 sleepTime = activeSleepTime; 3916 } else { 3917 sleepTime = idleSleepTime; 3918 } 3919 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3920 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3921 sleepTime = 0; 3922 } 3923} 3924 3925// getTrackName_l() must be called with ThreadBase::mLock held 3926int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3927 int sessionId __unused) 3928{ 3929 return 0; 3930} 3931 3932// deleteTrackName_l() must be called with ThreadBase::mLock held 3933void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3934{ 3935} 3936 3937// checkForNewParameter_l() must be called with ThreadBase::mLock held 3938bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 3939 status_t& status) 3940{ 3941 bool reconfig = false; 3942 3943 status = NO_ERROR; 3944 3945 AudioParameter param = AudioParameter(keyValuePair); 3946 int value; 3947 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3948 // forward device change to effects that have requested to be 3949 // aware of attached audio device. 3950 if (value != AUDIO_DEVICE_NONE) { 3951 mOutDevice = value; 3952 for (size_t i = 0; i < mEffectChains.size(); i++) { 3953 mEffectChains[i]->setDevice_l(mOutDevice); 3954 } 3955 } 3956 } 3957 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3958 // do not accept frame count changes if tracks are open as the track buffer 3959 // size depends on frame count and correct behavior would not be garantied 3960 // if frame count is changed after track creation 3961 if (!mTracks.isEmpty()) { 3962 status = INVALID_OPERATION; 3963 } else { 3964 reconfig = true; 3965 } 3966 } 3967 if (status == NO_ERROR) { 3968 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3969 keyValuePair.string()); 3970 if (!mStandby && status == INVALID_OPERATION) { 3971 mOutput->stream->common.standby(&mOutput->stream->common); 3972 mStandby = true; 3973 mBytesWritten = 0; 3974 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3975 keyValuePair.string()); 3976 } 3977 if (status == NO_ERROR && reconfig) { 3978 readOutputParameters_l(); 3979 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3980 } 3981 } 3982 3983 return reconfig; 3984} 3985 3986uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3987{ 3988 uint32_t time; 3989 if (audio_is_linear_pcm(mFormat)) { 3990 time = PlaybackThread::activeSleepTimeUs(); 3991 } else { 3992 time = 10000; 3993 } 3994 return time; 3995} 3996 3997uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3998{ 3999 uint32_t time; 4000 if (audio_is_linear_pcm(mFormat)) { 4001 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4002 } else { 4003 time = 10000; 4004 } 4005 return time; 4006} 4007 4008uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4009{ 4010 uint32_t time; 4011 if (audio_is_linear_pcm(mFormat)) { 4012 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4013 } else { 4014 time = 10000; 4015 } 4016 return time; 4017} 4018 4019void AudioFlinger::DirectOutputThread::cacheParameters_l() 4020{ 4021 PlaybackThread::cacheParameters_l(); 4022 4023 // use shorter standby delay as on normal output to release 4024 // hardware resources as soon as possible 4025 if (audio_is_linear_pcm(mFormat)) { 4026 standbyDelay = microseconds(activeSleepTime*2); 4027 } else { 4028 standbyDelay = kOffloadStandbyDelayNs; 4029 } 4030} 4031 4032// ---------------------------------------------------------------------------- 4033 4034AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4035 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4036 : Thread(false /*canCallJava*/), 4037 mPlaybackThread(playbackThread), 4038 mWriteAckSequence(0), 4039 mDrainSequence(0) 4040{ 4041} 4042 4043AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4044{ 4045} 4046 4047void AudioFlinger::AsyncCallbackThread::onFirstRef() 4048{ 4049 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4050} 4051 4052bool AudioFlinger::AsyncCallbackThread::threadLoop() 4053{ 4054 while (!exitPending()) { 4055 uint32_t writeAckSequence; 4056 uint32_t drainSequence; 4057 4058 { 4059 Mutex::Autolock _l(mLock); 4060 while (!((mWriteAckSequence & 1) || 4061 (mDrainSequence & 1) || 4062 exitPending())) { 4063 mWaitWorkCV.wait(mLock); 4064 } 4065 4066 if (exitPending()) { 4067 break; 4068 } 4069 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4070 mWriteAckSequence, mDrainSequence); 4071 writeAckSequence = mWriteAckSequence; 4072 mWriteAckSequence &= ~1; 4073 drainSequence = mDrainSequence; 4074 mDrainSequence &= ~1; 4075 } 4076 { 4077 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4078 if (playbackThread != 0) { 4079 if (writeAckSequence & 1) { 4080 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4081 } 4082 if (drainSequence & 1) { 4083 playbackThread->resetDraining(drainSequence >> 1); 4084 } 4085 } 4086 } 4087 } 4088 return false; 4089} 4090 4091void AudioFlinger::AsyncCallbackThread::exit() 4092{ 4093 ALOGV("AsyncCallbackThread::exit"); 4094 Mutex::Autolock _l(mLock); 4095 requestExit(); 4096 mWaitWorkCV.broadcast(); 4097} 4098 4099void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4100{ 4101 Mutex::Autolock _l(mLock); 4102 // bit 0 is cleared 4103 mWriteAckSequence = sequence << 1; 4104} 4105 4106void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4107{ 4108 Mutex::Autolock _l(mLock); 4109 // ignore unexpected callbacks 4110 if (mWriteAckSequence & 2) { 4111 mWriteAckSequence |= 1; 4112 mWaitWorkCV.signal(); 4113 } 4114} 4115 4116void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4117{ 4118 Mutex::Autolock _l(mLock); 4119 // bit 0 is cleared 4120 mDrainSequence = sequence << 1; 4121} 4122 4123void AudioFlinger::AsyncCallbackThread::resetDraining() 4124{ 4125 Mutex::Autolock _l(mLock); 4126 // ignore unexpected callbacks 4127 if (mDrainSequence & 2) { 4128 mDrainSequence |= 1; 4129 mWaitWorkCV.signal(); 4130 } 4131} 4132 4133 4134// ---------------------------------------------------------------------------- 4135AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4136 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4137 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4138 mHwPaused(false), 4139 mFlushPending(false), 4140 mPausedBytesRemaining(0) 4141{ 4142 //FIXME: mStandby should be set to true by ThreadBase constructor 4143 mStandby = true; 4144} 4145 4146void AudioFlinger::OffloadThread::threadLoop_exit() 4147{ 4148 if (mFlushPending || mHwPaused) { 4149 // If a flush is pending or track was paused, just discard buffered data 4150 flushHw_l(); 4151 } else { 4152 mMixerStatus = MIXER_DRAIN_ALL; 4153 threadLoop_drain(); 4154 } 4155 mCallbackThread->exit(); 4156 PlaybackThread::threadLoop_exit(); 4157} 4158 4159AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4160 Vector< sp<Track> > *tracksToRemove 4161) 4162{ 4163 size_t count = mActiveTracks.size(); 4164 4165 mixer_state mixerStatus = MIXER_IDLE; 4166 bool doHwPause = false; 4167 bool doHwResume = false; 4168 4169 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4170 4171 // find out which tracks need to be processed 4172 for (size_t i = 0; i < count; i++) { 4173 sp<Track> t = mActiveTracks[i].promote(); 4174 // The track died recently 4175 if (t == 0) { 4176 continue; 4177 } 4178 Track* const track = t.get(); 4179 audio_track_cblk_t* cblk = track->cblk(); 4180 // Only consider last track started for volume and mixer state control. 4181 // In theory an older track could underrun and restart after the new one starts 4182 // but as we only care about the transition phase between two tracks on a 4183 // direct output, it is not a problem to ignore the underrun case. 4184 sp<Track> l = mLatestActiveTrack.promote(); 4185 bool last = l.get() == track; 4186 4187 if (track->isInvalid()) { 4188 ALOGW("An invalidated track shouldn't be in active list"); 4189 tracksToRemove->add(track); 4190 continue; 4191 } 4192 4193 if (track->mState == TrackBase::IDLE) { 4194 ALOGW("An idle track shouldn't be in active list"); 4195 continue; 4196 } 4197 4198 if (track->isPausing()) { 4199 track->setPaused(); 4200 if (last) { 4201 if (!mHwPaused) { 4202 doHwPause = true; 4203 mHwPaused = true; 4204 } 4205 // If we were part way through writing the mixbuffer to 4206 // the HAL we must save this until we resume 4207 // BUG - this will be wrong if a different track is made active, 4208 // in that case we want to discard the pending data in the 4209 // mixbuffer and tell the client to present it again when the 4210 // track is resumed 4211 mPausedWriteLength = mCurrentWriteLength; 4212 mPausedBytesRemaining = mBytesRemaining; 4213 mBytesRemaining = 0; // stop writing 4214 } 4215 tracksToRemove->add(track); 4216 } else if (track->isFlushPending()) { 4217 track->flushAck(); 4218 if (last) { 4219 mFlushPending = true; 4220 } 4221 } else if (track->isResumePending()){ 4222 track->resumeAck(); 4223 if (last) { 4224 if (mPausedBytesRemaining) { 4225 // Need to continue write that was interrupted 4226 mCurrentWriteLength = mPausedWriteLength; 4227 mBytesRemaining = mPausedBytesRemaining; 4228 mPausedBytesRemaining = 0; 4229 } 4230 if (mHwPaused) { 4231 doHwResume = true; 4232 mHwPaused = false; 4233 // threadLoop_mix() will handle the case that we need to 4234 // resume an interrupted write 4235 } 4236 // enable write to audio HAL 4237 sleepTime = 0; 4238 4239 // Do not handle new data in this iteration even if track->framesReady() 4240 mixerStatus = MIXER_TRACKS_ENABLED; 4241 } 4242 } else if (track->framesReady() && track->isReady() && 4243 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4244 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4245 if (track->mFillingUpStatus == Track::FS_FILLED) { 4246 track->mFillingUpStatus = Track::FS_ACTIVE; 4247 // make sure processVolume_l() will apply new volume even if 0 4248 mLeftVolFloat = mRightVolFloat = -1.0; 4249 } 4250 4251 if (last) { 4252 sp<Track> previousTrack = mPreviousTrack.promote(); 4253 if (previousTrack != 0) { 4254 if (track != previousTrack.get()) { 4255 // Flush any data still being written from last track 4256 mBytesRemaining = 0; 4257 if (mPausedBytesRemaining) { 4258 // Last track was paused so we also need to flush saved 4259 // mixbuffer state and invalidate track so that it will 4260 // re-submit that unwritten data when it is next resumed 4261 mPausedBytesRemaining = 0; 4262 // Invalidate is a bit drastic - would be more efficient 4263 // to have a flag to tell client that some of the 4264 // previously written data was lost 4265 previousTrack->invalidate(); 4266 } 4267 // flush data already sent to the DSP if changing audio session as audio 4268 // comes from a different source. Also invalidate previous track to force a 4269 // seek when resuming. 4270 if (previousTrack->sessionId() != track->sessionId()) { 4271 previousTrack->invalidate(); 4272 } 4273 } 4274 } 4275 mPreviousTrack = track; 4276 // reset retry count 4277 track->mRetryCount = kMaxTrackRetriesOffload; 4278 mActiveTrack = t; 4279 mixerStatus = MIXER_TRACKS_READY; 4280 } 4281 } else { 4282 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4283 if (track->isStopping_1()) { 4284 // Hardware buffer can hold a large amount of audio so we must 4285 // wait for all current track's data to drain before we say 4286 // that the track is stopped. 4287 if (mBytesRemaining == 0) { 4288 // Only start draining when all data in mixbuffer 4289 // has been written 4290 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4291 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4292 // do not drain if no data was ever sent to HAL (mStandby == true) 4293 if (last && !mStandby) { 4294 // do not modify drain sequence if we are already draining. This happens 4295 // when resuming from pause after drain. 4296 if ((mDrainSequence & 1) == 0) { 4297 sleepTime = 0; 4298 standbyTime = systemTime() + standbyDelay; 4299 mixerStatus = MIXER_DRAIN_TRACK; 4300 mDrainSequence += 2; 4301 } 4302 if (mHwPaused) { 4303 // It is possible to move from PAUSED to STOPPING_1 without 4304 // a resume so we must ensure hardware is running 4305 doHwResume = true; 4306 mHwPaused = false; 4307 } 4308 } 4309 } 4310 } else if (track->isStopping_2()) { 4311 // Drain has completed or we are in standby, signal presentation complete 4312 if (!(mDrainSequence & 1) || !last || mStandby) { 4313 track->mState = TrackBase::STOPPED; 4314 size_t audioHALFrames = 4315 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4316 size_t framesWritten = 4317 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4318 track->presentationComplete(framesWritten, audioHALFrames); 4319 track->reset(); 4320 tracksToRemove->add(track); 4321 } 4322 } else { 4323 // No buffers for this track. Give it a few chances to 4324 // fill a buffer, then remove it from active list. 4325 if (--(track->mRetryCount) <= 0) { 4326 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4327 track->name()); 4328 tracksToRemove->add(track); 4329 // indicate to client process that the track was disabled because of underrun; 4330 // it will then automatically call start() when data is available 4331 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4332 } else if (last){ 4333 mixerStatus = MIXER_TRACKS_ENABLED; 4334 } 4335 } 4336 } 4337 // compute volume for this track 4338 processVolume_l(track, last); 4339 } 4340 4341 // make sure the pause/flush/resume sequence is executed in the right order. 4342 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4343 // before flush and then resume HW. This can happen in case of pause/flush/resume 4344 // if resume is received before pause is executed. 4345 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4346 mOutput->stream->pause(mOutput->stream); 4347 } 4348 if (mFlushPending) { 4349 flushHw_l(); 4350 mFlushPending = false; 4351 } 4352 if (!mStandby && doHwResume) { 4353 mOutput->stream->resume(mOutput->stream); 4354 } 4355 4356 // remove all the tracks that need to be... 4357 removeTracks_l(*tracksToRemove); 4358 4359 return mixerStatus; 4360} 4361 4362// must be called with thread mutex locked 4363bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4364{ 4365 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4366 mWriteAckSequence, mDrainSequence); 4367 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4368 return true; 4369 } 4370 return false; 4371} 4372 4373// must be called with thread mutex locked 4374bool AudioFlinger::OffloadThread::shouldStandby_l() 4375{ 4376 bool trackPaused = false; 4377 4378 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4379 // after a timeout and we will enter standby then. 4380 if (mTracks.size() > 0) { 4381 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4382 } 4383 4384 return !mStandby && !trackPaused; 4385} 4386 4387 4388bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4389{ 4390 Mutex::Autolock _l(mLock); 4391 return waitingAsyncCallback_l(); 4392} 4393 4394void AudioFlinger::OffloadThread::flushHw_l() 4395{ 4396 mOutput->stream->flush(mOutput->stream); 4397 // Flush anything still waiting in the mixbuffer 4398 mCurrentWriteLength = 0; 4399 mBytesRemaining = 0; 4400 mPausedWriteLength = 0; 4401 mPausedBytesRemaining = 0; 4402 mHwPaused = false; 4403 4404 if (mUseAsyncWrite) { 4405 // discard any pending drain or write ack by incrementing sequence 4406 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4407 mDrainSequence = (mDrainSequence + 2) & ~1; 4408 ALOG_ASSERT(mCallbackThread != 0); 4409 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4410 mCallbackThread->setDraining(mDrainSequence); 4411 } 4412} 4413 4414void AudioFlinger::OffloadThread::onAddNewTrack_l() 4415{ 4416 sp<Track> previousTrack = mPreviousTrack.promote(); 4417 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4418 4419 if (previousTrack != 0 && latestTrack != 0 && 4420 (previousTrack->sessionId() != latestTrack->sessionId())) { 4421 mFlushPending = true; 4422 } 4423 PlaybackThread::onAddNewTrack_l(); 4424} 4425 4426// ---------------------------------------------------------------------------- 4427 4428AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4429 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4430 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4431 DUPLICATING), 4432 mWaitTimeMs(UINT_MAX) 4433{ 4434 addOutputTrack(mainThread); 4435} 4436 4437AudioFlinger::DuplicatingThread::~DuplicatingThread() 4438{ 4439 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4440 mOutputTracks[i]->destroy(); 4441 } 4442} 4443 4444void AudioFlinger::DuplicatingThread::threadLoop_mix() 4445{ 4446 // mix buffers... 4447 if (outputsReady(outputTracks)) { 4448 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4449 } else { 4450 memset(mSinkBuffer, 0, mSinkBufferSize); 4451 } 4452 sleepTime = 0; 4453 writeFrames = mNormalFrameCount; 4454 mCurrentWriteLength = mSinkBufferSize; 4455 standbyTime = systemTime() + standbyDelay; 4456} 4457 4458void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4459{ 4460 if (sleepTime == 0) { 4461 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4462 sleepTime = activeSleepTime; 4463 } else { 4464 sleepTime = idleSleepTime; 4465 } 4466 } else if (mBytesWritten != 0) { 4467 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4468 writeFrames = mNormalFrameCount; 4469 memset(mSinkBuffer, 0, mSinkBufferSize); 4470 } else { 4471 // flush remaining overflow buffers in output tracks 4472 writeFrames = 0; 4473 } 4474 sleepTime = 0; 4475 } 4476} 4477 4478ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4479{ 4480 for (size_t i = 0; i < outputTracks.size(); i++) { 4481 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4482 // for delivery downstream as needed. This in-place conversion is safe as 4483 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4484 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4485 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4486 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4487 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4488 } 4489 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4490 } 4491 mStandby = false; 4492 return (ssize_t)mSinkBufferSize; 4493} 4494 4495void AudioFlinger::DuplicatingThread::threadLoop_standby() 4496{ 4497 // DuplicatingThread implements standby by stopping all tracks 4498 for (size_t i = 0; i < outputTracks.size(); i++) { 4499 outputTracks[i]->stop(); 4500 } 4501} 4502 4503void AudioFlinger::DuplicatingThread::saveOutputTracks() 4504{ 4505 outputTracks = mOutputTracks; 4506} 4507 4508void AudioFlinger::DuplicatingThread::clearOutputTracks() 4509{ 4510 outputTracks.clear(); 4511} 4512 4513void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4514{ 4515 Mutex::Autolock _l(mLock); 4516 // FIXME explain this formula 4517 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4518 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4519 // due to current usage case and restrictions on the AudioBufferProvider. 4520 // Actual buffer conversion is done in threadLoop_write(). 4521 // 4522 // TODO: This may change in the future, depending on multichannel 4523 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4524 OutputTrack *outputTrack = new OutputTrack(thread, 4525 this, 4526 mSampleRate, 4527 AUDIO_FORMAT_PCM_16_BIT, 4528 mChannelMask, 4529 frameCount, 4530 IPCThreadState::self()->getCallingUid()); 4531 if (outputTrack->cblk() != NULL) { 4532 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4533 mOutputTracks.add(outputTrack); 4534 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4535 updateWaitTime_l(); 4536 } 4537} 4538 4539void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4540{ 4541 Mutex::Autolock _l(mLock); 4542 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4543 if (mOutputTracks[i]->thread() == thread) { 4544 mOutputTracks[i]->destroy(); 4545 mOutputTracks.removeAt(i); 4546 updateWaitTime_l(); 4547 return; 4548 } 4549 } 4550 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4551} 4552 4553// caller must hold mLock 4554void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4555{ 4556 mWaitTimeMs = UINT_MAX; 4557 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4558 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4559 if (strong != 0) { 4560 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4561 if (waitTimeMs < mWaitTimeMs) { 4562 mWaitTimeMs = waitTimeMs; 4563 } 4564 } 4565 } 4566} 4567 4568 4569bool AudioFlinger::DuplicatingThread::outputsReady( 4570 const SortedVector< sp<OutputTrack> > &outputTracks) 4571{ 4572 for (size_t i = 0; i < outputTracks.size(); i++) { 4573 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4574 if (thread == 0) { 4575 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4576 outputTracks[i].get()); 4577 return false; 4578 } 4579 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4580 // see note at standby() declaration 4581 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4582 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4583 thread.get()); 4584 return false; 4585 } 4586 } 4587 return true; 4588} 4589 4590uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4591{ 4592 return (mWaitTimeMs * 1000) / 2; 4593} 4594 4595void AudioFlinger::DuplicatingThread::cacheParameters_l() 4596{ 4597 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4598 updateWaitTime_l(); 4599 4600 MixerThread::cacheParameters_l(); 4601} 4602 4603// ---------------------------------------------------------------------------- 4604// Record 4605// ---------------------------------------------------------------------------- 4606 4607AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4608 AudioStreamIn *input, 4609 audio_io_handle_t id, 4610 audio_devices_t outDevice, 4611 audio_devices_t inDevice 4612#ifdef TEE_SINK 4613 , const sp<NBAIO_Sink>& teeSink 4614#endif 4615 ) : 4616 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4617 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4618 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4619 mRsmpInRear(0) 4620#ifdef TEE_SINK 4621 , mTeeSink(teeSink) 4622#endif 4623 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4624 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4625{ 4626 snprintf(mName, kNameLength, "AudioIn_%X", id); 4627 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4628 4629 readInputParameters_l(); 4630} 4631 4632 4633AudioFlinger::RecordThread::~RecordThread() 4634{ 4635 mAudioFlinger->unregisterWriter(mNBLogWriter); 4636 delete[] mRsmpInBuffer; 4637} 4638 4639void AudioFlinger::RecordThread::onFirstRef() 4640{ 4641 run(mName, PRIORITY_URGENT_AUDIO); 4642} 4643 4644bool AudioFlinger::RecordThread::threadLoop() 4645{ 4646 nsecs_t lastWarning = 0; 4647 4648 inputStandBy(); 4649 4650reacquire_wakelock: 4651 sp<RecordTrack> activeTrack; 4652 int activeTracksGen; 4653 { 4654 Mutex::Autolock _l(mLock); 4655 size_t size = mActiveTracks.size(); 4656 activeTracksGen = mActiveTracksGen; 4657 if (size > 0) { 4658 // FIXME an arbitrary choice 4659 activeTrack = mActiveTracks[0]; 4660 acquireWakeLock_l(activeTrack->uid()); 4661 if (size > 1) { 4662 SortedVector<int> tmp; 4663 for (size_t i = 0; i < size; i++) { 4664 tmp.add(mActiveTracks[i]->uid()); 4665 } 4666 updateWakeLockUids_l(tmp); 4667 } 4668 } else { 4669 acquireWakeLock_l(-1); 4670 } 4671 } 4672 4673 // used to request a deferred sleep, to be executed later while mutex is unlocked 4674 uint32_t sleepUs = 0; 4675 4676 // loop while there is work to do 4677 for (;;) { 4678 Vector< sp<EffectChain> > effectChains; 4679 4680 // sleep with mutex unlocked 4681 if (sleepUs > 0) { 4682 usleep(sleepUs); 4683 sleepUs = 0; 4684 } 4685 4686 // activeTracks accumulates a copy of a subset of mActiveTracks 4687 Vector< sp<RecordTrack> > activeTracks; 4688 4689 4690 { // scope for mLock 4691 Mutex::Autolock _l(mLock); 4692 4693 processConfigEvents_l(); 4694 4695 // check exitPending here because checkForNewParameters_l() and 4696 // checkForNewParameters_l() can temporarily release mLock 4697 if (exitPending()) { 4698 break; 4699 } 4700 4701 // if no active track(s), then standby and release wakelock 4702 size_t size = mActiveTracks.size(); 4703 if (size == 0) { 4704 standbyIfNotAlreadyInStandby(); 4705 // exitPending() can't become true here 4706 releaseWakeLock_l(); 4707 ALOGV("RecordThread: loop stopping"); 4708 // go to sleep 4709 mWaitWorkCV.wait(mLock); 4710 ALOGV("RecordThread: loop starting"); 4711 goto reacquire_wakelock; 4712 } 4713 4714 if (mActiveTracksGen != activeTracksGen) { 4715 activeTracksGen = mActiveTracksGen; 4716 SortedVector<int> tmp; 4717 for (size_t i = 0; i < size; i++) { 4718 tmp.add(mActiveTracks[i]->uid()); 4719 } 4720 updateWakeLockUids_l(tmp); 4721 } 4722 4723 bool doBroadcast = false; 4724 for (size_t i = 0; i < size; ) { 4725 4726 activeTrack = mActiveTracks[i]; 4727 if (activeTrack->isTerminated()) { 4728 removeTrack_l(activeTrack); 4729 mActiveTracks.remove(activeTrack); 4730 mActiveTracksGen++; 4731 size--; 4732 continue; 4733 } 4734 4735 TrackBase::track_state activeTrackState = activeTrack->mState; 4736 switch (activeTrackState) { 4737 4738 case TrackBase::PAUSING: 4739 mActiveTracks.remove(activeTrack); 4740 mActiveTracksGen++; 4741 doBroadcast = true; 4742 size--; 4743 continue; 4744 4745 case TrackBase::STARTING_1: 4746 sleepUs = 10000; 4747 i++; 4748 continue; 4749 4750 case TrackBase::STARTING_2: 4751 doBroadcast = true; 4752 mStandby = false; 4753 activeTrack->mState = TrackBase::ACTIVE; 4754 break; 4755 4756 case TrackBase::ACTIVE: 4757 break; 4758 4759 case TrackBase::IDLE: 4760 i++; 4761 continue; 4762 4763 default: 4764 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4765 } 4766 4767 activeTracks.add(activeTrack); 4768 i++; 4769 4770 } 4771 if (doBroadcast) { 4772 mStartStopCond.broadcast(); 4773 } 4774 4775 // sleep if there are no active tracks to process 4776 if (activeTracks.size() == 0) { 4777 if (sleepUs == 0) { 4778 sleepUs = kRecordThreadSleepUs; 4779 } 4780 continue; 4781 } 4782 sleepUs = 0; 4783 4784 lockEffectChains_l(effectChains); 4785 } 4786 4787 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4788 4789 size_t size = effectChains.size(); 4790 for (size_t i = 0; i < size; i++) { 4791 // thread mutex is not locked, but effect chain is locked 4792 effectChains[i]->process_l(); 4793 } 4794 4795 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4796 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4797 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4798 // If destination is non-contiguous, first read past the nominal end of buffer, then 4799 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4800 4801 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4802 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4803 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4804 if (bytesRead <= 0) { 4805 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4806 // Force input into standby so that it tries to recover at next read attempt 4807 inputStandBy(); 4808 sleepUs = kRecordThreadSleepUs; 4809 continue; 4810 } 4811 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4812 size_t framesRead = bytesRead / mFrameSize; 4813 ALOG_ASSERT(framesRead > 0); 4814 if (mTeeSink != 0) { 4815 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4816 } 4817 // If destination is non-contiguous, we now correct for reading past end of buffer. 4818 size_t part1 = mRsmpInFramesP2 - rear; 4819 if (framesRead > part1) { 4820 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4821 (framesRead - part1) * mFrameSize); 4822 } 4823 rear = mRsmpInRear += framesRead; 4824 4825 size = activeTracks.size(); 4826 // loop over each active track 4827 for (size_t i = 0; i < size; i++) { 4828 activeTrack = activeTracks[i]; 4829 4830 enum { 4831 OVERRUN_UNKNOWN, 4832 OVERRUN_TRUE, 4833 OVERRUN_FALSE 4834 } overrun = OVERRUN_UNKNOWN; 4835 4836 // loop over getNextBuffer to handle circular sink 4837 for (;;) { 4838 4839 activeTrack->mSink.frameCount = ~0; 4840 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4841 size_t framesOut = activeTrack->mSink.frameCount; 4842 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4843 4844 int32_t front = activeTrack->mRsmpInFront; 4845 ssize_t filled = rear - front; 4846 size_t framesIn; 4847 4848 if (filled < 0) { 4849 // should not happen, but treat like a massive overrun and re-sync 4850 framesIn = 0; 4851 activeTrack->mRsmpInFront = rear; 4852 overrun = OVERRUN_TRUE; 4853 } else if ((size_t) filled <= mRsmpInFrames) { 4854 framesIn = (size_t) filled; 4855 } else { 4856 // client is not keeping up with server, but give it latest data 4857 framesIn = mRsmpInFrames; 4858 activeTrack->mRsmpInFront = front = rear - framesIn; 4859 overrun = OVERRUN_TRUE; 4860 } 4861 4862 if (framesOut == 0 || framesIn == 0) { 4863 break; 4864 } 4865 4866 if (activeTrack->mResampler == NULL) { 4867 // no resampling 4868 if (framesIn > framesOut) { 4869 framesIn = framesOut; 4870 } else { 4871 framesOut = framesIn; 4872 } 4873 int8_t *dst = activeTrack->mSink.i8; 4874 while (framesIn > 0) { 4875 front &= mRsmpInFramesP2 - 1; 4876 size_t part1 = mRsmpInFramesP2 - front; 4877 if (part1 > framesIn) { 4878 part1 = framesIn; 4879 } 4880 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4881 if (mChannelCount == activeTrack->mChannelCount) { 4882 memcpy(dst, src, part1 * mFrameSize); 4883 } else if (mChannelCount == 1) { 4884 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4885 part1); 4886 } else { 4887 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4888 part1); 4889 } 4890 dst += part1 * activeTrack->mFrameSize; 4891 front += part1; 4892 framesIn -= part1; 4893 } 4894 activeTrack->mRsmpInFront += framesOut; 4895 4896 } else { 4897 // resampling 4898 // FIXME framesInNeeded should really be part of resampler API, and should 4899 // depend on the SRC ratio 4900 // to keep mRsmpInBuffer full so resampler always has sufficient input 4901 size_t framesInNeeded; 4902 // FIXME only re-calculate when it changes, and optimize for common ratios 4903 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4904 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4905 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4906 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4907 framesInNeeded, framesOut, inOverOut); 4908 // Although we theoretically have framesIn in circular buffer, some of those are 4909 // unreleased frames, and thus must be discounted for purpose of budgeting. 4910 size_t unreleased = activeTrack->mRsmpInUnrel; 4911 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4912 if (framesIn < framesInNeeded) { 4913 ALOGV("not enough to resample: have %u frames in but need %u in to " 4914 "produce %u out given in/out ratio of %.4g", 4915 framesIn, framesInNeeded, framesOut, inOverOut); 4916 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4917 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4918 if (newFramesOut == 0) { 4919 break; 4920 } 4921 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4922 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4923 framesInNeeded, newFramesOut, outOverIn); 4924 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4925 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4926 "given in/out ratio of %.4g", 4927 framesIn, framesInNeeded, newFramesOut, inOverOut); 4928 framesOut = newFramesOut; 4929 } else { 4930 ALOGV("success 1: have %u in and need %u in to produce %u out " 4931 "given in/out ratio of %.4g", 4932 framesIn, framesInNeeded, framesOut, inOverOut); 4933 } 4934 4935 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4936 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4937 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4938 delete[] activeTrack->mRsmpOutBuffer; 4939 // resampler always outputs stereo 4940 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4941 activeTrack->mRsmpOutFrameCount = framesOut; 4942 } 4943 4944 // resampler accumulates, but we only have one source track 4945 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4946 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4947 // FIXME how about having activeTrack implement this interface itself? 4948 activeTrack->mResamplerBufferProvider 4949 /*this*/ /* AudioBufferProvider* */); 4950 // ditherAndClamp() works as long as all buffers returned by 4951 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4952 if (activeTrack->mChannelCount == 1) { 4953 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 4954 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4955 framesOut); 4956 // the resampler always outputs stereo samples: 4957 // do post stereo to mono conversion 4958 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4959 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4960 } else { 4961 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4962 activeTrack->mRsmpOutBuffer, framesOut); 4963 } 4964 // now done with mRsmpOutBuffer 4965 4966 } 4967 4968 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4969 overrun = OVERRUN_FALSE; 4970 } 4971 4972 if (activeTrack->mFramesToDrop == 0) { 4973 if (framesOut > 0) { 4974 activeTrack->mSink.frameCount = framesOut; 4975 activeTrack->releaseBuffer(&activeTrack->mSink); 4976 } 4977 } else { 4978 // FIXME could do a partial drop of framesOut 4979 if (activeTrack->mFramesToDrop > 0) { 4980 activeTrack->mFramesToDrop -= framesOut; 4981 if (activeTrack->mFramesToDrop <= 0) { 4982 activeTrack->clearSyncStartEvent(); 4983 } 4984 } else { 4985 activeTrack->mFramesToDrop += framesOut; 4986 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4987 activeTrack->mSyncStartEvent->isCancelled()) { 4988 ALOGW("Synced record %s, session %d, trigger session %d", 4989 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 4990 activeTrack->sessionId(), 4991 (activeTrack->mSyncStartEvent != 0) ? 4992 activeTrack->mSyncStartEvent->triggerSession() : 0); 4993 activeTrack->clearSyncStartEvent(); 4994 } 4995 } 4996 } 4997 4998 if (framesOut == 0) { 4999 break; 5000 } 5001 } 5002 5003 switch (overrun) { 5004 case OVERRUN_TRUE: 5005 // client isn't retrieving buffers fast enough 5006 if (!activeTrack->setOverflow()) { 5007 nsecs_t now = systemTime(); 5008 // FIXME should lastWarning per track? 5009 if ((now - lastWarning) > kWarningThrottleNs) { 5010 ALOGW("RecordThread: buffer overflow"); 5011 lastWarning = now; 5012 } 5013 } 5014 break; 5015 case OVERRUN_FALSE: 5016 activeTrack->clearOverflow(); 5017 break; 5018 case OVERRUN_UNKNOWN: 5019 break; 5020 } 5021 5022 } 5023 5024 // enable changes in effect chain 5025 unlockEffectChains(effectChains); 5026 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5027 } 5028 5029 standbyIfNotAlreadyInStandby(); 5030 5031 { 5032 Mutex::Autolock _l(mLock); 5033 for (size_t i = 0; i < mTracks.size(); i++) { 5034 sp<RecordTrack> track = mTracks[i]; 5035 track->invalidate(); 5036 } 5037 mActiveTracks.clear(); 5038 mActiveTracksGen++; 5039 mStartStopCond.broadcast(); 5040 } 5041 5042 releaseWakeLock(); 5043 5044 ALOGV("RecordThread %p exiting", this); 5045 return false; 5046} 5047 5048void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5049{ 5050 if (!mStandby) { 5051 inputStandBy(); 5052 mStandby = true; 5053 } 5054} 5055 5056void AudioFlinger::RecordThread::inputStandBy() 5057{ 5058 mInput->stream->common.standby(&mInput->stream->common); 5059} 5060 5061// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5062sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5063 const sp<AudioFlinger::Client>& client, 5064 uint32_t sampleRate, 5065 audio_format_t format, 5066 audio_channel_mask_t channelMask, 5067 size_t *pFrameCount, 5068 int sessionId, 5069 int uid, 5070 IAudioFlinger::track_flags_t *flags, 5071 pid_t tid, 5072 status_t *status) 5073{ 5074 size_t frameCount = *pFrameCount; 5075 sp<RecordTrack> track; 5076 status_t lStatus; 5077 5078 // client expresses a preference for FAST, but we get the final say 5079 if (*flags & IAudioFlinger::TRACK_FAST) { 5080 if ( 5081 // use case: callback handler and frame count is default or at least as large as HAL 5082 ( 5083 (tid != -1) && 5084 ((frameCount == 0) || 5085 // FIXME not necessarily true, should be native frame count for native SR! 5086 (frameCount >= mFrameCount)) 5087 ) && 5088 // PCM data 5089 audio_is_linear_pcm(format) && 5090 // mono or stereo 5091 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5092 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5093 // hardware sample rate 5094 // FIXME actually the native hardware sample rate 5095 (sampleRate == mSampleRate) && 5096 // record thread has an associated fast capture 5097 hasFastCapture() 5098 // fast capture does not require slots 5099 ) { 5100 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5101 if (frameCount == 0) { 5102 // FIXME wrong mFrameCount 5103 frameCount = mFrameCount * kFastTrackMultiplier; 5104 } 5105 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5106 frameCount, mFrameCount); 5107 } else { 5108 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5109 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5110 "hasFastCapture=%d tid=%d", 5111 frameCount, mFrameCount, format, 5112 audio_is_linear_pcm(format), 5113 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5114 *flags &= ~IAudioFlinger::TRACK_FAST; 5115 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5116 // For compatibility with AudioRecord calculation, buffer depth is forced 5117 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5118 // This is probably too conservative, but legacy application code may depend on it. 5119 // If you change this calculation, also review the start threshold which is related. 5120 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5121 size_t mNormalFrameCount = 2048; // FIXME 5122 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5123 if (minBufCount < 2) { 5124 minBufCount = 2; 5125 } 5126 size_t minFrameCount = mNormalFrameCount * minBufCount; 5127 if (frameCount < minFrameCount) { 5128 frameCount = minFrameCount; 5129 } 5130 } 5131 } 5132 *pFrameCount = frameCount; 5133 5134 lStatus = initCheck(); 5135 if (lStatus != NO_ERROR) { 5136 ALOGE("createRecordTrack_l() audio driver not initialized"); 5137 goto Exit; 5138 } 5139 5140 { // scope for mLock 5141 Mutex::Autolock _l(mLock); 5142 5143 track = new RecordTrack(this, client, sampleRate, 5144 format, channelMask, frameCount, sessionId, uid, 5145 (*flags & IAudioFlinger::TRACK_FAST) != 0); 5146 5147 lStatus = track->initCheck(); 5148 if (lStatus != NO_ERROR) { 5149 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5150 // track must be cleared from the caller as the caller has the AF lock 5151 goto Exit; 5152 } 5153 mTracks.add(track); 5154 5155 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5156 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5157 mAudioFlinger->btNrecIsOff(); 5158 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5159 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5160 5161 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5162 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5163 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5164 // so ask activity manager to do this on our behalf 5165 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5166 } 5167 } 5168 5169 lStatus = NO_ERROR; 5170 5171Exit: 5172 *status = lStatus; 5173 return track; 5174} 5175 5176status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5177 AudioSystem::sync_event_t event, 5178 int triggerSession) 5179{ 5180 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5181 sp<ThreadBase> strongMe = this; 5182 status_t status = NO_ERROR; 5183 5184 if (event == AudioSystem::SYNC_EVENT_NONE) { 5185 recordTrack->clearSyncStartEvent(); 5186 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5187 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5188 triggerSession, 5189 recordTrack->sessionId(), 5190 syncStartEventCallback, 5191 recordTrack); 5192 // Sync event can be cancelled by the trigger session if the track is not in a 5193 // compatible state in which case we start record immediately 5194 if (recordTrack->mSyncStartEvent->isCancelled()) { 5195 recordTrack->clearSyncStartEvent(); 5196 } else { 5197 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5198 recordTrack->mFramesToDrop = - 5199 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5200 } 5201 } 5202 5203 { 5204 // This section is a rendezvous between binder thread executing start() and RecordThread 5205 AutoMutex lock(mLock); 5206 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5207 if (recordTrack->mState == TrackBase::PAUSING) { 5208 ALOGV("active record track PAUSING -> ACTIVE"); 5209 recordTrack->mState = TrackBase::ACTIVE; 5210 } else { 5211 ALOGV("active record track state %d", recordTrack->mState); 5212 } 5213 return status; 5214 } 5215 5216 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5217 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5218 // or using a separate command thread 5219 recordTrack->mState = TrackBase::STARTING_1; 5220 mActiveTracks.add(recordTrack); 5221 mActiveTracksGen++; 5222 mLock.unlock(); 5223 status_t status = AudioSystem::startInput(mId); 5224 mLock.lock(); 5225 // FIXME should verify that recordTrack is still in mActiveTracks 5226 if (status != NO_ERROR) { 5227 mActiveTracks.remove(recordTrack); 5228 mActiveTracksGen++; 5229 recordTrack->clearSyncStartEvent(); 5230 return status; 5231 } 5232 // Catch up with current buffer indices if thread is already running. 5233 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5234 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5235 // see previously buffered data before it called start(), but with greater risk of overrun. 5236 5237 recordTrack->mRsmpInFront = mRsmpInRear; 5238 recordTrack->mRsmpInUnrel = 0; 5239 // FIXME why reset? 5240 if (recordTrack->mResampler != NULL) { 5241 recordTrack->mResampler->reset(); 5242 } 5243 recordTrack->mState = TrackBase::STARTING_2; 5244 // signal thread to start 5245 mWaitWorkCV.broadcast(); 5246 if (mActiveTracks.indexOf(recordTrack) < 0) { 5247 ALOGV("Record failed to start"); 5248 status = BAD_VALUE; 5249 goto startError; 5250 } 5251 return status; 5252 } 5253 5254startError: 5255 AudioSystem::stopInput(mId); 5256 recordTrack->clearSyncStartEvent(); 5257 // FIXME I wonder why we do not reset the state here? 5258 return status; 5259} 5260 5261void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5262{ 5263 sp<SyncEvent> strongEvent = event.promote(); 5264 5265 if (strongEvent != 0) { 5266 sp<RefBase> ptr = strongEvent->cookie().promote(); 5267 if (ptr != 0) { 5268 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5269 recordTrack->handleSyncStartEvent(strongEvent); 5270 } 5271 } 5272} 5273 5274bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5275 ALOGV("RecordThread::stop"); 5276 AutoMutex _l(mLock); 5277 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5278 return false; 5279 } 5280 // note that threadLoop may still be processing the track at this point [without lock] 5281 recordTrack->mState = TrackBase::PAUSING; 5282 // do not wait for mStartStopCond if exiting 5283 if (exitPending()) { 5284 return true; 5285 } 5286 // FIXME incorrect usage of wait: no explicit predicate or loop 5287 mStartStopCond.wait(mLock); 5288 // if we have been restarted, recordTrack is in mActiveTracks here 5289 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5290 ALOGV("Record stopped OK"); 5291 return true; 5292 } 5293 return false; 5294} 5295 5296bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5297{ 5298 return false; 5299} 5300 5301status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5302{ 5303#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5304 if (!isValidSyncEvent(event)) { 5305 return BAD_VALUE; 5306 } 5307 5308 int eventSession = event->triggerSession(); 5309 status_t ret = NAME_NOT_FOUND; 5310 5311 Mutex::Autolock _l(mLock); 5312 5313 for (size_t i = 0; i < mTracks.size(); i++) { 5314 sp<RecordTrack> track = mTracks[i]; 5315 if (eventSession == track->sessionId()) { 5316 (void) track->setSyncEvent(event); 5317 ret = NO_ERROR; 5318 } 5319 } 5320 return ret; 5321#else 5322 return BAD_VALUE; 5323#endif 5324} 5325 5326// destroyTrack_l() must be called with ThreadBase::mLock held 5327void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5328{ 5329 track->terminate(); 5330 track->mState = TrackBase::STOPPED; 5331 // active tracks are removed by threadLoop() 5332 if (mActiveTracks.indexOf(track) < 0) { 5333 removeTrack_l(track); 5334 } 5335} 5336 5337void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5338{ 5339 mTracks.remove(track); 5340 // need anything related to effects here? 5341} 5342 5343void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5344{ 5345 dumpInternals(fd, args); 5346 dumpTracks(fd, args); 5347 dumpEffectChains(fd, args); 5348} 5349 5350void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5351{ 5352 fdprintf(fd, "\nInput thread %p:\n", this); 5353 5354 if (mActiveTracks.size() > 0) { 5355 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5356 } else { 5357 fdprintf(fd, " No active record clients\n"); 5358 } 5359 5360 dumpBase(fd, args); 5361} 5362 5363void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5364{ 5365 const size_t SIZE = 256; 5366 char buffer[SIZE]; 5367 String8 result; 5368 5369 size_t numtracks = mTracks.size(); 5370 size_t numactive = mActiveTracks.size(); 5371 size_t numactiveseen = 0; 5372 fdprintf(fd, " %d Tracks", numtracks); 5373 if (numtracks) { 5374 fdprintf(fd, " of which %d are active\n", numactive); 5375 RecordTrack::appendDumpHeader(result); 5376 for (size_t i = 0; i < numtracks ; ++i) { 5377 sp<RecordTrack> track = mTracks[i]; 5378 if (track != 0) { 5379 bool active = mActiveTracks.indexOf(track) >= 0; 5380 if (active) { 5381 numactiveseen++; 5382 } 5383 track->dump(buffer, SIZE, active); 5384 result.append(buffer); 5385 } 5386 } 5387 } else { 5388 fdprintf(fd, "\n"); 5389 } 5390 5391 if (numactiveseen != numactive) { 5392 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5393 " not in the track list\n"); 5394 result.append(buffer); 5395 RecordTrack::appendDumpHeader(result); 5396 for (size_t i = 0; i < numactive; ++i) { 5397 sp<RecordTrack> track = mActiveTracks[i]; 5398 if (mTracks.indexOf(track) < 0) { 5399 track->dump(buffer, SIZE, true); 5400 result.append(buffer); 5401 } 5402 } 5403 5404 } 5405 write(fd, result.string(), result.size()); 5406} 5407 5408// AudioBufferProvider interface 5409status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5410 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5411{ 5412 RecordTrack *activeTrack = mRecordTrack; 5413 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5414 if (threadBase == 0) { 5415 buffer->frameCount = 0; 5416 buffer->raw = NULL; 5417 return NOT_ENOUGH_DATA; 5418 } 5419 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5420 int32_t rear = recordThread->mRsmpInRear; 5421 int32_t front = activeTrack->mRsmpInFront; 5422 ssize_t filled = rear - front; 5423 // FIXME should not be P2 (don't want to increase latency) 5424 // FIXME if client not keeping up, discard 5425 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5426 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5427 front &= recordThread->mRsmpInFramesP2 - 1; 5428 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5429 if (part1 > (size_t) filled) { 5430 part1 = filled; 5431 } 5432 size_t ask = buffer->frameCount; 5433 ALOG_ASSERT(ask > 0); 5434 if (part1 > ask) { 5435 part1 = ask; 5436 } 5437 if (part1 == 0) { 5438 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5439 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5440 buffer->raw = NULL; 5441 buffer->frameCount = 0; 5442 activeTrack->mRsmpInUnrel = 0; 5443 return NOT_ENOUGH_DATA; 5444 } 5445 5446 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5447 buffer->frameCount = part1; 5448 activeTrack->mRsmpInUnrel = part1; 5449 return NO_ERROR; 5450} 5451 5452// AudioBufferProvider interface 5453void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5454 AudioBufferProvider::Buffer* buffer) 5455{ 5456 RecordTrack *activeTrack = mRecordTrack; 5457 size_t stepCount = buffer->frameCount; 5458 if (stepCount == 0) { 5459 return; 5460 } 5461 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5462 activeTrack->mRsmpInUnrel -= stepCount; 5463 activeTrack->mRsmpInFront += stepCount; 5464 buffer->raw = NULL; 5465 buffer->frameCount = 0; 5466} 5467 5468bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5469 status_t& status) 5470{ 5471 bool reconfig = false; 5472 5473 status = NO_ERROR; 5474 5475 audio_format_t reqFormat = mFormat; 5476 uint32_t samplingRate = mSampleRate; 5477 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5478 5479 AudioParameter param = AudioParameter(keyValuePair); 5480 int value; 5481 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5482 // channel count change can be requested. Do we mandate the first client defines the 5483 // HAL sampling rate and channel count or do we allow changes on the fly? 5484 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5485 samplingRate = value; 5486 reconfig = true; 5487 } 5488 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5489 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5490 status = BAD_VALUE; 5491 } else { 5492 reqFormat = (audio_format_t) value; 5493 reconfig = true; 5494 } 5495 } 5496 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5497 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5498 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5499 status = BAD_VALUE; 5500 } else { 5501 channelMask = mask; 5502 reconfig = true; 5503 } 5504 } 5505 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5506 // do not accept frame count changes if tracks are open as the track buffer 5507 // size depends on frame count and correct behavior would not be guaranteed 5508 // if frame count is changed after track creation 5509 if (mActiveTracks.size() > 0) { 5510 status = INVALID_OPERATION; 5511 } else { 5512 reconfig = true; 5513 } 5514 } 5515 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5516 // forward device change to effects that have requested to be 5517 // aware of attached audio device. 5518 for (size_t i = 0; i < mEffectChains.size(); i++) { 5519 mEffectChains[i]->setDevice_l(value); 5520 } 5521 5522 // store input device and output device but do not forward output device to audio HAL. 5523 // Note that status is ignored by the caller for output device 5524 // (see AudioFlinger::setParameters() 5525 if (audio_is_output_devices(value)) { 5526 mOutDevice = value; 5527 status = BAD_VALUE; 5528 } else { 5529 mInDevice = value; 5530 // disable AEC and NS if the device is a BT SCO headset supporting those 5531 // pre processings 5532 if (mTracks.size() > 0) { 5533 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5534 mAudioFlinger->btNrecIsOff(); 5535 for (size_t i = 0; i < mTracks.size(); i++) { 5536 sp<RecordTrack> track = mTracks[i]; 5537 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5538 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5539 } 5540 } 5541 } 5542 } 5543 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5544 mAudioSource != (audio_source_t)value) { 5545 // forward device change to effects that have requested to be 5546 // aware of attached audio device. 5547 for (size_t i = 0; i < mEffectChains.size(); i++) { 5548 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5549 } 5550 mAudioSource = (audio_source_t)value; 5551 } 5552 5553 if (status == NO_ERROR) { 5554 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5555 keyValuePair.string()); 5556 if (status == INVALID_OPERATION) { 5557 inputStandBy(); 5558 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5559 keyValuePair.string()); 5560 } 5561 if (reconfig) { 5562 if (status == BAD_VALUE && 5563 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5564 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5565 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5566 <= (2 * samplingRate)) && 5567 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5568 <= FCC_2 && 5569 (channelMask == AUDIO_CHANNEL_IN_MONO || 5570 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5571 status = NO_ERROR; 5572 } 5573 if (status == NO_ERROR) { 5574 readInputParameters_l(); 5575 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5576 } 5577 } 5578 } 5579 5580 return reconfig; 5581} 5582 5583String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5584{ 5585 Mutex::Autolock _l(mLock); 5586 if (initCheck() != NO_ERROR) { 5587 return String8(); 5588 } 5589 5590 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5591 const String8 out_s8(s); 5592 free(s); 5593 return out_s8; 5594} 5595 5596void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5597 AudioSystem::OutputDescriptor desc; 5598 const void *param2 = NULL; 5599 5600 switch (event) { 5601 case AudioSystem::INPUT_OPENED: 5602 case AudioSystem::INPUT_CONFIG_CHANGED: 5603 desc.channelMask = mChannelMask; 5604 desc.samplingRate = mSampleRate; 5605 desc.format = mFormat; 5606 desc.frameCount = mFrameCount; 5607 desc.latency = 0; 5608 param2 = &desc; 5609 break; 5610 5611 case AudioSystem::INPUT_CLOSED: 5612 default: 5613 break; 5614 } 5615 mAudioFlinger->audioConfigChanged(event, mId, param2); 5616} 5617 5618void AudioFlinger::RecordThread::readInputParameters_l() 5619{ 5620 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5621 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5622 mChannelCount = popcount(mChannelMask); 5623 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5624 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5625 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5626 } 5627 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5628 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5629 mFrameCount = mBufferSize / mFrameSize; 5630 // This is the formula for calculating the temporary buffer size. 5631 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5632 // 1 full output buffer, regardless of the alignment of the available input. 5633 // The value is somewhat arbitrary, and could probably be even larger. 5634 // A larger value should allow more old data to be read after a track calls start(), 5635 // without increasing latency. 5636 mRsmpInFrames = mFrameCount * 7; 5637 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5638 delete[] mRsmpInBuffer; 5639 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5640 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5641 5642 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5643 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5644} 5645 5646uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5647{ 5648 Mutex::Autolock _l(mLock); 5649 if (initCheck() != NO_ERROR) { 5650 return 0; 5651 } 5652 5653 return mInput->stream->get_input_frames_lost(mInput->stream); 5654} 5655 5656uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5657{ 5658 Mutex::Autolock _l(mLock); 5659 uint32_t result = 0; 5660 if (getEffectChain_l(sessionId) != 0) { 5661 result = EFFECT_SESSION; 5662 } 5663 5664 for (size_t i = 0; i < mTracks.size(); ++i) { 5665 if (sessionId == mTracks[i]->sessionId()) { 5666 result |= TRACK_SESSION; 5667 break; 5668 } 5669 } 5670 5671 return result; 5672} 5673 5674KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5675{ 5676 KeyedVector<int, bool> ids; 5677 Mutex::Autolock _l(mLock); 5678 for (size_t j = 0; j < mTracks.size(); ++j) { 5679 sp<RecordThread::RecordTrack> track = mTracks[j]; 5680 int sessionId = track->sessionId(); 5681 if (ids.indexOfKey(sessionId) < 0) { 5682 ids.add(sessionId, true); 5683 } 5684 } 5685 return ids; 5686} 5687 5688AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5689{ 5690 Mutex::Autolock _l(mLock); 5691 AudioStreamIn *input = mInput; 5692 mInput = NULL; 5693 return input; 5694} 5695 5696// this method must always be called either with ThreadBase mLock held or inside the thread loop 5697audio_stream_t* AudioFlinger::RecordThread::stream() const 5698{ 5699 if (mInput == NULL) { 5700 return NULL; 5701 } 5702 return &mInput->stream->common; 5703} 5704 5705status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5706{ 5707 // only one chain per input thread 5708 if (mEffectChains.size() != 0) { 5709 return INVALID_OPERATION; 5710 } 5711 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5712 5713 chain->setInBuffer(NULL); 5714 chain->setOutBuffer(NULL); 5715 5716 checkSuspendOnAddEffectChain_l(chain); 5717 5718 mEffectChains.add(chain); 5719 5720 return NO_ERROR; 5721} 5722 5723size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5724{ 5725 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5726 ALOGW_IF(mEffectChains.size() != 1, 5727 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5728 chain.get(), mEffectChains.size(), this); 5729 if (mEffectChains.size() == 1) { 5730 mEffectChains.removeAt(0); 5731 } 5732 return 0; 5733} 5734 5735}; // namespace android 5736