Threads.cpp revision c62476f0c0c1cf9283a38852bde0a4c9434df712
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38#include <audio_utils/minifloat.h> 39 40// NBAIO implementations 41#include <media/nbaio/AudioStreamOutSink.h> 42#include <media/nbaio/MonoPipe.h> 43#include <media/nbaio/MonoPipeReader.h> 44#include <media/nbaio/Pipe.h> 45#include <media/nbaio/PipeReader.h> 46#include <media/nbaio/SourceAudioBufferProvider.h> 47 48#include <powermanager/PowerManager.h> 49 50#include <common_time/cc_helper.h> 51#include <common_time/local_clock.h> 52 53#include "AudioFlinger.h" 54#include "AudioMixer.h" 55#include "FastMixer.h" 56#include "ServiceUtilities.h" 57#include "SchedulingPolicyService.h" 58 59#ifdef ADD_BATTERY_DATA 60#include <media/IMediaPlayerService.h> 61#include <media/IMediaDeathNotifier.h> 62#endif 63 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait in sendConfigEvent_l() for a status to be received 102static const nsecs_t kConfigEventTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal sink buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalSinkBufferSizeMs = 20; 111// maximum normal sink buffer size 112static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 113 114// Offloaded output thread standby delay: allows track transition without going to standby 115static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 116 117// Whether to use fast mixer 118static const enum { 119 FastMixer_Never, // never initialize or use: for debugging only 120 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 121 // normal mixer multiplier is 1 122 FastMixer_Static, // initialize if needed, then use all the time if initialized, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 125 // multiplier is calculated based on min & max normal mixer buffer size 126 // FIXME for FastMixer_Dynamic: 127 // Supporting this option will require fixing HALs that can't handle large writes. 128 // For example, one HAL implementation returns an error from a large write, 129 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 130 // We could either fix the HAL implementations, or provide a wrapper that breaks 131 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 132} kUseFastMixer = FastMixer_Static; 133 134// Priorities for requestPriority 135static const int kPriorityAudioApp = 2; 136static const int kPriorityFastMixer = 3; 137 138// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 139// for the track. The client then sub-divides this into smaller buffers for its use. 140// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 141// So for now we just assume that client is double-buffered for fast tracks. 142// FIXME It would be better for client to tell AudioFlinger the value of N, 143// so AudioFlinger could allocate the right amount of memory. 144// See the client's minBufCount and mNotificationFramesAct calculations for details. 145static const int kFastTrackMultiplier = 2; 146 147// See Thread::readOnlyHeap(). 148// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 149// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 150// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 151static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 152 153// ---------------------------------------------------------------------------- 154 155#ifdef ADD_BATTERY_DATA 156// To collect the amplifier usage 157static void addBatteryData(uint32_t params) { 158 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 159 if (service == NULL) { 160 // it already logged 161 return; 162 } 163 164 service->addBatteryData(params); 165} 166#endif 167 168 169// ---------------------------------------------------------------------------- 170// CPU Stats 171// ---------------------------------------------------------------------------- 172 173class CpuStats { 174public: 175 CpuStats(); 176 void sample(const String8 &title); 177#ifdef DEBUG_CPU_USAGE 178private: 179 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 180 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 181 182 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 183 184 int mCpuNum; // thread's current CPU number 185 int mCpukHz; // frequency of thread's current CPU in kHz 186#endif 187}; 188 189CpuStats::CpuStats() 190#ifdef DEBUG_CPU_USAGE 191 : mCpuNum(-1), mCpukHz(-1) 192#endif 193{ 194} 195 196void CpuStats::sample(const String8 &title 197#ifndef DEBUG_CPU_USAGE 198 __unused 199#endif 200 ) { 201#ifdef DEBUG_CPU_USAGE 202 // get current thread's delta CPU time in wall clock ns 203 double wcNs; 204 bool valid = mCpuUsage.sampleAndEnable(wcNs); 205 206 // record sample for wall clock statistics 207 if (valid) { 208 mWcStats.sample(wcNs); 209 } 210 211 // get the current CPU number 212 int cpuNum = sched_getcpu(); 213 214 // get the current CPU frequency in kHz 215 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 216 217 // check if either CPU number or frequency changed 218 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 219 mCpuNum = cpuNum; 220 mCpukHz = cpukHz; 221 // ignore sample for purposes of cycles 222 valid = false; 223 } 224 225 // if no change in CPU number or frequency, then record sample for cycle statistics 226 if (valid && mCpukHz > 0) { 227 double cycles = wcNs * cpukHz * 0.000001; 228 mHzStats.sample(cycles); 229 } 230 231 unsigned n = mWcStats.n(); 232 // mCpuUsage.elapsed() is expensive, so don't call it every loop 233 if ((n & 127) == 1) { 234 long long elapsed = mCpuUsage.elapsed(); 235 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 236 double perLoop = elapsed / (double) n; 237 double perLoop100 = perLoop * 0.01; 238 double perLoop1k = perLoop * 0.001; 239 double mean = mWcStats.mean(); 240 double stddev = mWcStats.stddev(); 241 double minimum = mWcStats.minimum(); 242 double maximum = mWcStats.maximum(); 243 double meanCycles = mHzStats.mean(); 244 double stddevCycles = mHzStats.stddev(); 245 double minCycles = mHzStats.minimum(); 246 double maxCycles = mHzStats.maximum(); 247 mCpuUsage.resetElapsed(); 248 mWcStats.reset(); 249 mHzStats.reset(); 250 ALOGD("CPU usage for %s over past %.1f secs\n" 251 " (%u mixer loops at %.1f mean ms per loop):\n" 252 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 253 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 254 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 255 title.string(), 256 elapsed * .000000001, n, perLoop * .000001, 257 mean * .001, 258 stddev * .001, 259 minimum * .001, 260 maximum * .001, 261 mean / perLoop100, 262 stddev / perLoop100, 263 minimum / perLoop100, 264 maximum / perLoop100, 265 meanCycles / perLoop1k, 266 stddevCycles / perLoop1k, 267 minCycles / perLoop1k, 268 maxCycles / perLoop1k); 269 270 } 271 } 272#endif 273}; 274 275// ---------------------------------------------------------------------------- 276// ThreadBase 277// ---------------------------------------------------------------------------- 278 279AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 280 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 281 : Thread(false /*canCallJava*/), 282 mType(type), 283 mAudioFlinger(audioFlinger), 284 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 285 // are set by PlaybackThread::readOutputParameters_l() or 286 // RecordThread::readInputParameters_l() 287 //FIXME: mStandby should be true here. Is this some kind of hack? 288 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 289 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 290 // mName will be set by concrete (non-virtual) subclass 291 mDeathRecipient(new PMDeathRecipient(this)) 292{ 293} 294 295AudioFlinger::ThreadBase::~ThreadBase() 296{ 297 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 298 mConfigEvents.clear(); 299 300 // do not lock the mutex in destructor 301 releaseWakeLock_l(); 302 if (mPowerManager != 0) { 303 sp<IBinder> binder = mPowerManager->asBinder(); 304 binder->unlinkToDeath(mDeathRecipient); 305 } 306} 307 308status_t AudioFlinger::ThreadBase::readyToRun() 309{ 310 status_t status = initCheck(); 311 if (status == NO_ERROR) { 312 ALOGI("AudioFlinger's thread %p ready to run", this); 313 } else { 314 ALOGE("No working audio driver found."); 315 } 316 return status; 317} 318 319void AudioFlinger::ThreadBase::exit() 320{ 321 ALOGV("ThreadBase::exit"); 322 // do any cleanup required for exit to succeed 323 preExit(); 324 { 325 // This lock prevents the following race in thread (uniprocessor for illustration): 326 // if (!exitPending()) { 327 // // context switch from here to exit() 328 // // exit() calls requestExit(), what exitPending() observes 329 // // exit() calls signal(), which is dropped since no waiters 330 // // context switch back from exit() to here 331 // mWaitWorkCV.wait(...); 332 // // now thread is hung 333 // } 334 AutoMutex lock(mLock); 335 requestExit(); 336 mWaitWorkCV.broadcast(); 337 } 338 // When Thread::requestExitAndWait is made virtual and this method is renamed to 339 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 340 requestExitAndWait(); 341} 342 343status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 344{ 345 status_t status; 346 347 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 348 Mutex::Autolock _l(mLock); 349 350 return sendSetParameterConfigEvent_l(keyValuePairs); 351} 352 353// sendConfigEvent_l() must be called with ThreadBase::mLock held 354// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 355status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 356{ 357 status_t status = NO_ERROR; 358 359 mConfigEvents.add(event); 360 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 361 mWaitWorkCV.signal(); 362 mLock.unlock(); 363 { 364 Mutex::Autolock _l(event->mLock); 365 while (event->mWaitStatus) { 366 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 367 event->mStatus = TIMED_OUT; 368 event->mWaitStatus = false; 369 } 370 } 371 status = event->mStatus; 372 } 373 mLock.lock(); 374 return status; 375} 376 377void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 378{ 379 Mutex::Autolock _l(mLock); 380 sendIoConfigEvent_l(event, param); 381} 382 383// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 384void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 385{ 386 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 387 sendConfigEvent_l(configEvent); 388} 389 390// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 391void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 392{ 393 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 394 sendConfigEvent_l(configEvent); 395} 396 397// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 398status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 399{ 400 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 401 return sendConfigEvent_l(configEvent); 402} 403 404// post condition: mConfigEvents.isEmpty() 405void AudioFlinger::ThreadBase::processConfigEvents_l() 406{ 407 bool configChanged = false; 408 409 while (!mConfigEvents.isEmpty()) { 410 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 411 sp<ConfigEvent> event = mConfigEvents[0]; 412 mConfigEvents.removeAt(0); 413 switch (event->mType) { 414 case CFG_EVENT_PRIO: { 415 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 416 // FIXME Need to understand why this has to be done asynchronously 417 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 418 true /*asynchronous*/); 419 if (err != 0) { 420 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 421 data->mPrio, data->mPid, data->mTid, err); 422 } 423 } break; 424 case CFG_EVENT_IO: { 425 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 426 audioConfigChanged(data->mEvent, data->mParam); 427 } break; 428 case CFG_EVENT_SET_PARAMETER: { 429 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 430 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 431 configChanged = true; 432 } 433 } break; 434 default: 435 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 436 break; 437 } 438 { 439 Mutex::Autolock _l(event->mLock); 440 if (event->mWaitStatus) { 441 event->mWaitStatus = false; 442 event->mCond.signal(); 443 } 444 } 445 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 446 } 447 448 if (configChanged) { 449 cacheParameters_l(); 450 } 451} 452 453String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 454 String8 s; 455 if (output) { 456 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 457 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 458 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 459 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 460 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 461 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 462 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 463 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 464 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 465 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 466 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 467 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 468 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 469 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 470 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 471 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 472 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 473 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 474 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 475 } else { 476 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 477 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 478 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 479 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 480 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 481 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 482 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 483 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 484 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 485 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 486 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 487 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 488 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 489 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 490 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 491 } 492 int len = s.length(); 493 if (s.length() > 2) { 494 char *str = s.lockBuffer(len); 495 s.unlockBuffer(len - 2); 496 } 497 return s; 498} 499 500void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 501{ 502 const size_t SIZE = 256; 503 char buffer[SIZE]; 504 String8 result; 505 506 bool locked = AudioFlinger::dumpTryLock(mLock); 507 if (!locked) { 508 dprintf(fd, "thread %p maybe dead locked\n", this); 509 } 510 511 dprintf(fd, " I/O handle: %d\n", mId); 512 dprintf(fd, " TID: %d\n", getTid()); 513 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 514 dprintf(fd, " Sample rate: %u\n", mSampleRate); 515 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 516 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 517 dprintf(fd, " Channel Count: %u\n", mChannelCount); 518 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 519 channelMaskToString(mChannelMask, mType != RECORD).string()); 520 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 521 dprintf(fd, " Frame size: %zu\n", mFrameSize); 522 dprintf(fd, " Pending config events:"); 523 size_t numConfig = mConfigEvents.size(); 524 if (numConfig) { 525 for (size_t i = 0; i < numConfig; i++) { 526 mConfigEvents[i]->dump(buffer, SIZE); 527 dprintf(fd, "\n %s", buffer); 528 } 529 dprintf(fd, "\n"); 530 } else { 531 dprintf(fd, " none\n"); 532 } 533 534 if (locked) { 535 mLock.unlock(); 536 } 537} 538 539void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 540{ 541 const size_t SIZE = 256; 542 char buffer[SIZE]; 543 String8 result; 544 545 size_t numEffectChains = mEffectChains.size(); 546 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 547 write(fd, buffer, strlen(buffer)); 548 549 for (size_t i = 0; i < numEffectChains; ++i) { 550 sp<EffectChain> chain = mEffectChains[i]; 551 if (chain != 0) { 552 chain->dump(fd, args); 553 } 554 } 555} 556 557void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 558{ 559 Mutex::Autolock _l(mLock); 560 acquireWakeLock_l(uid); 561} 562 563String16 AudioFlinger::ThreadBase::getWakeLockTag() 564{ 565 switch (mType) { 566 case MIXER: 567 return String16("AudioMix"); 568 case DIRECT: 569 return String16("AudioDirectOut"); 570 case DUPLICATING: 571 return String16("AudioDup"); 572 case RECORD: 573 return String16("AudioIn"); 574 case OFFLOAD: 575 return String16("AudioOffload"); 576 default: 577 ALOG_ASSERT(false); 578 return String16("AudioUnknown"); 579 } 580} 581 582void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 583{ 584 getPowerManager_l(); 585 if (mPowerManager != 0) { 586 sp<IBinder> binder = new BBinder(); 587 status_t status; 588 if (uid >= 0) { 589 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 590 binder, 591 getWakeLockTag(), 592 String16("media"), 593 uid); 594 } else { 595 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 596 binder, 597 getWakeLockTag(), 598 String16("media")); 599 } 600 if (status == NO_ERROR) { 601 mWakeLockToken = binder; 602 } 603 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 604 } 605} 606 607void AudioFlinger::ThreadBase::releaseWakeLock() 608{ 609 Mutex::Autolock _l(mLock); 610 releaseWakeLock_l(); 611} 612 613void AudioFlinger::ThreadBase::releaseWakeLock_l() 614{ 615 if (mWakeLockToken != 0) { 616 ALOGV("releaseWakeLock_l() %s", mName); 617 if (mPowerManager != 0) { 618 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 619 } 620 mWakeLockToken.clear(); 621 } 622} 623 624void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 625 Mutex::Autolock _l(mLock); 626 updateWakeLockUids_l(uids); 627} 628 629void AudioFlinger::ThreadBase::getPowerManager_l() { 630 631 if (mPowerManager == 0) { 632 // use checkService() to avoid blocking if power service is not up yet 633 sp<IBinder> binder = 634 defaultServiceManager()->checkService(String16("power")); 635 if (binder == 0) { 636 ALOGW("Thread %s cannot connect to the power manager service", mName); 637 } else { 638 mPowerManager = interface_cast<IPowerManager>(binder); 639 binder->linkToDeath(mDeathRecipient); 640 } 641 } 642} 643 644void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 645 646 getPowerManager_l(); 647 if (mWakeLockToken == NULL) { 648 ALOGE("no wake lock to update!"); 649 return; 650 } 651 if (mPowerManager != 0) { 652 sp<IBinder> binder = new BBinder(); 653 status_t status; 654 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 655 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 656 } 657} 658 659void AudioFlinger::ThreadBase::clearPowerManager() 660{ 661 Mutex::Autolock _l(mLock); 662 releaseWakeLock_l(); 663 mPowerManager.clear(); 664} 665 666void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 667{ 668 sp<ThreadBase> thread = mThread.promote(); 669 if (thread != 0) { 670 thread->clearPowerManager(); 671 } 672 ALOGW("power manager service died !!!"); 673} 674 675void AudioFlinger::ThreadBase::setEffectSuspended( 676 const effect_uuid_t *type, bool suspend, int sessionId) 677{ 678 Mutex::Autolock _l(mLock); 679 setEffectSuspended_l(type, suspend, sessionId); 680} 681 682void AudioFlinger::ThreadBase::setEffectSuspended_l( 683 const effect_uuid_t *type, bool suspend, int sessionId) 684{ 685 sp<EffectChain> chain = getEffectChain_l(sessionId); 686 if (chain != 0) { 687 if (type != NULL) { 688 chain->setEffectSuspended_l(type, suspend); 689 } else { 690 chain->setEffectSuspendedAll_l(suspend); 691 } 692 } 693 694 updateSuspendedSessions_l(type, suspend, sessionId); 695} 696 697void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 698{ 699 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 700 if (index < 0) { 701 return; 702 } 703 704 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 705 mSuspendedSessions.valueAt(index); 706 707 for (size_t i = 0; i < sessionEffects.size(); i++) { 708 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 709 for (int j = 0; j < desc->mRefCount; j++) { 710 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 711 chain->setEffectSuspendedAll_l(true); 712 } else { 713 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 714 desc->mType.timeLow); 715 chain->setEffectSuspended_l(&desc->mType, true); 716 } 717 } 718 } 719} 720 721void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 722 bool suspend, 723 int sessionId) 724{ 725 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 726 727 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 728 729 if (suspend) { 730 if (index >= 0) { 731 sessionEffects = mSuspendedSessions.valueAt(index); 732 } else { 733 mSuspendedSessions.add(sessionId, sessionEffects); 734 } 735 } else { 736 if (index < 0) { 737 return; 738 } 739 sessionEffects = mSuspendedSessions.valueAt(index); 740 } 741 742 743 int key = EffectChain::kKeyForSuspendAll; 744 if (type != NULL) { 745 key = type->timeLow; 746 } 747 index = sessionEffects.indexOfKey(key); 748 749 sp<SuspendedSessionDesc> desc; 750 if (suspend) { 751 if (index >= 0) { 752 desc = sessionEffects.valueAt(index); 753 } else { 754 desc = new SuspendedSessionDesc(); 755 if (type != NULL) { 756 desc->mType = *type; 757 } 758 sessionEffects.add(key, desc); 759 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 760 } 761 desc->mRefCount++; 762 } else { 763 if (index < 0) { 764 return; 765 } 766 desc = sessionEffects.valueAt(index); 767 if (--desc->mRefCount == 0) { 768 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 769 sessionEffects.removeItemsAt(index); 770 if (sessionEffects.isEmpty()) { 771 ALOGV("updateSuspendedSessions_l() restore removing session %d", 772 sessionId); 773 mSuspendedSessions.removeItem(sessionId); 774 } 775 } 776 } 777 if (!sessionEffects.isEmpty()) { 778 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 779 } 780} 781 782void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 783 bool enabled, 784 int sessionId) 785{ 786 Mutex::Autolock _l(mLock); 787 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 788} 789 790void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 791 bool enabled, 792 int sessionId) 793{ 794 if (mType != RECORD) { 795 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 796 // another session. This gives the priority to well behaved effect control panels 797 // and applications not using global effects. 798 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 799 // global effects 800 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 801 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 802 } 803 } 804 805 sp<EffectChain> chain = getEffectChain_l(sessionId); 806 if (chain != 0) { 807 chain->checkSuspendOnEffectEnabled(effect, enabled); 808 } 809} 810 811// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 812sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 813 const sp<AudioFlinger::Client>& client, 814 const sp<IEffectClient>& effectClient, 815 int32_t priority, 816 int sessionId, 817 effect_descriptor_t *desc, 818 int *enabled, 819 status_t *status) 820{ 821 sp<EffectModule> effect; 822 sp<EffectHandle> handle; 823 status_t lStatus; 824 sp<EffectChain> chain; 825 bool chainCreated = false; 826 bool effectCreated = false; 827 bool effectRegistered = false; 828 829 lStatus = initCheck(); 830 if (lStatus != NO_ERROR) { 831 ALOGW("createEffect_l() Audio driver not initialized."); 832 goto Exit; 833 } 834 835 // Reject any effect on Direct output threads for now, since the format of 836 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 837 if (mType == DIRECT) { 838 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 839 desc->name, mName); 840 lStatus = BAD_VALUE; 841 goto Exit; 842 } 843 844 // Allow global effects only on offloaded and mixer threads 845 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 846 switch (mType) { 847 case MIXER: 848 case OFFLOAD: 849 break; 850 case DIRECT: 851 case DUPLICATING: 852 case RECORD: 853 default: 854 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 855 lStatus = BAD_VALUE; 856 goto Exit; 857 } 858 } 859 860 // Only Pre processor effects are allowed on input threads and only on input threads 861 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 862 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 863 desc->name, desc->flags, mType); 864 lStatus = BAD_VALUE; 865 goto Exit; 866 } 867 868 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 869 870 { // scope for mLock 871 Mutex::Autolock _l(mLock); 872 873 // check for existing effect chain with the requested audio session 874 chain = getEffectChain_l(sessionId); 875 if (chain == 0) { 876 // create a new chain for this session 877 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 878 chain = new EffectChain(this, sessionId); 879 addEffectChain_l(chain); 880 chain->setStrategy(getStrategyForSession_l(sessionId)); 881 chainCreated = true; 882 } else { 883 effect = chain->getEffectFromDesc_l(desc); 884 } 885 886 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 887 888 if (effect == 0) { 889 int id = mAudioFlinger->nextUniqueId(); 890 // Check CPU and memory usage 891 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 892 if (lStatus != NO_ERROR) { 893 goto Exit; 894 } 895 effectRegistered = true; 896 // create a new effect module if none present in the chain 897 effect = new EffectModule(this, chain, desc, id, sessionId); 898 lStatus = effect->status(); 899 if (lStatus != NO_ERROR) { 900 goto Exit; 901 } 902 effect->setOffloaded(mType == OFFLOAD, mId); 903 904 lStatus = chain->addEffect_l(effect); 905 if (lStatus != NO_ERROR) { 906 goto Exit; 907 } 908 effectCreated = true; 909 910 effect->setDevice(mOutDevice); 911 effect->setDevice(mInDevice); 912 effect->setMode(mAudioFlinger->getMode()); 913 effect->setAudioSource(mAudioSource); 914 } 915 // create effect handle and connect it to effect module 916 handle = new EffectHandle(effect, client, effectClient, priority); 917 lStatus = handle->initCheck(); 918 if (lStatus == OK) { 919 lStatus = effect->addHandle(handle.get()); 920 } 921 if (enabled != NULL) { 922 *enabled = (int)effect->isEnabled(); 923 } 924 } 925 926Exit: 927 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 928 Mutex::Autolock _l(mLock); 929 if (effectCreated) { 930 chain->removeEffect_l(effect); 931 } 932 if (effectRegistered) { 933 AudioSystem::unregisterEffect(effect->id()); 934 } 935 if (chainCreated) { 936 removeEffectChain_l(chain); 937 } 938 handle.clear(); 939 } 940 941 *status = lStatus; 942 return handle; 943} 944 945sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 946{ 947 Mutex::Autolock _l(mLock); 948 return getEffect_l(sessionId, effectId); 949} 950 951sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 952{ 953 sp<EffectChain> chain = getEffectChain_l(sessionId); 954 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 955} 956 957// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 958// PlaybackThread::mLock held 959status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 960{ 961 // check for existing effect chain with the requested audio session 962 int sessionId = effect->sessionId(); 963 sp<EffectChain> chain = getEffectChain_l(sessionId); 964 bool chainCreated = false; 965 966 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 967 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 968 this, effect->desc().name, effect->desc().flags); 969 970 if (chain == 0) { 971 // create a new chain for this session 972 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 973 chain = new EffectChain(this, sessionId); 974 addEffectChain_l(chain); 975 chain->setStrategy(getStrategyForSession_l(sessionId)); 976 chainCreated = true; 977 } 978 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 979 980 if (chain->getEffectFromId_l(effect->id()) != 0) { 981 ALOGW("addEffect_l() %p effect %s already present in chain %p", 982 this, effect->desc().name, chain.get()); 983 return BAD_VALUE; 984 } 985 986 effect->setOffloaded(mType == OFFLOAD, mId); 987 988 status_t status = chain->addEffect_l(effect); 989 if (status != NO_ERROR) { 990 if (chainCreated) { 991 removeEffectChain_l(chain); 992 } 993 return status; 994 } 995 996 effect->setDevice(mOutDevice); 997 effect->setDevice(mInDevice); 998 effect->setMode(mAudioFlinger->getMode()); 999 effect->setAudioSource(mAudioSource); 1000 return NO_ERROR; 1001} 1002 1003void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1004 1005 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1006 effect_descriptor_t desc = effect->desc(); 1007 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1008 detachAuxEffect_l(effect->id()); 1009 } 1010 1011 sp<EffectChain> chain = effect->chain().promote(); 1012 if (chain != 0) { 1013 // remove effect chain if removing last effect 1014 if (chain->removeEffect_l(effect) == 0) { 1015 removeEffectChain_l(chain); 1016 } 1017 } else { 1018 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1019 } 1020} 1021 1022void AudioFlinger::ThreadBase::lockEffectChains_l( 1023 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1024{ 1025 effectChains = mEffectChains; 1026 for (size_t i = 0; i < mEffectChains.size(); i++) { 1027 mEffectChains[i]->lock(); 1028 } 1029} 1030 1031void AudioFlinger::ThreadBase::unlockEffectChains( 1032 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1033{ 1034 for (size_t i = 0; i < effectChains.size(); i++) { 1035 effectChains[i]->unlock(); 1036 } 1037} 1038 1039sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1040{ 1041 Mutex::Autolock _l(mLock); 1042 return getEffectChain_l(sessionId); 1043} 1044 1045sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1046{ 1047 size_t size = mEffectChains.size(); 1048 for (size_t i = 0; i < size; i++) { 1049 if (mEffectChains[i]->sessionId() == sessionId) { 1050 return mEffectChains[i]; 1051 } 1052 } 1053 return 0; 1054} 1055 1056void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1057{ 1058 Mutex::Autolock _l(mLock); 1059 size_t size = mEffectChains.size(); 1060 for (size_t i = 0; i < size; i++) { 1061 mEffectChains[i]->setMode_l(mode); 1062 } 1063} 1064 1065void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1066 EffectHandle *handle, 1067 bool unpinIfLast) { 1068 1069 Mutex::Autolock _l(mLock); 1070 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1071 // delete the effect module if removing last handle on it 1072 if (effect->removeHandle(handle) == 0) { 1073 if (!effect->isPinned() || unpinIfLast) { 1074 removeEffect_l(effect); 1075 AudioSystem::unregisterEffect(effect->id()); 1076 } 1077 } 1078} 1079 1080// ---------------------------------------------------------------------------- 1081// Playback 1082// ---------------------------------------------------------------------------- 1083 1084AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1085 AudioStreamOut* output, 1086 audio_io_handle_t id, 1087 audio_devices_t device, 1088 type_t type) 1089 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1090 mNormalFrameCount(0), mSinkBuffer(NULL), 1091 mMixerBufferEnabled(false), 1092 mMixerBuffer(NULL), 1093 mMixerBufferSize(0), 1094 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1095 mMixerBufferValid(false), 1096 mEffectBufferEnabled(false), 1097 mEffectBuffer(NULL), 1098 mEffectBufferSize(0), 1099 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1100 mEffectBufferValid(false), 1101 mSuspended(0), mBytesWritten(0), 1102 mActiveTracksGeneration(0), 1103 // mStreamTypes[] initialized in constructor body 1104 mOutput(output), 1105 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1106 mMixerStatus(MIXER_IDLE), 1107 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1108 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1109 mBytesRemaining(0), 1110 mCurrentWriteLength(0), 1111 mUseAsyncWrite(false), 1112 mWriteAckSequence(0), 1113 mDrainSequence(0), 1114 mSignalPending(false), 1115 mScreenState(AudioFlinger::mScreenState), 1116 // index 0 is reserved for normal mixer's submix 1117 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1118 // mLatchD, mLatchQ, 1119 mLatchDValid(false), mLatchQValid(false) 1120{ 1121 snprintf(mName, kNameLength, "AudioOut_%X", id); 1122 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1123 1124 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1125 // it would be safer to explicitly pass initial masterVolume/masterMute as 1126 // parameter. 1127 // 1128 // If the HAL we are using has support for master volume or master mute, 1129 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1130 // and the mute set to false). 1131 mMasterVolume = audioFlinger->masterVolume_l(); 1132 mMasterMute = audioFlinger->masterMute_l(); 1133 if (mOutput && mOutput->audioHwDev) { 1134 if (mOutput->audioHwDev->canSetMasterVolume()) { 1135 mMasterVolume = 1.0; 1136 } 1137 1138 if (mOutput->audioHwDev->canSetMasterMute()) { 1139 mMasterMute = false; 1140 } 1141 } 1142 1143 readOutputParameters_l(); 1144 1145 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1146 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1147 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1148 stream = (audio_stream_type_t) (stream + 1)) { 1149 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1150 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1151 } 1152 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1153 // because mAudioFlinger doesn't have one to copy from 1154} 1155 1156AudioFlinger::PlaybackThread::~PlaybackThread() 1157{ 1158 mAudioFlinger->unregisterWriter(mNBLogWriter); 1159 free(mSinkBuffer); 1160 free(mMixerBuffer); 1161 free(mEffectBuffer); 1162} 1163 1164void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1165{ 1166 dumpInternals(fd, args); 1167 dumpTracks(fd, args); 1168 dumpEffectChains(fd, args); 1169} 1170 1171void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1172{ 1173 const size_t SIZE = 256; 1174 char buffer[SIZE]; 1175 String8 result; 1176 1177 result.appendFormat(" Stream volumes in dB: "); 1178 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1179 const stream_type_t *st = &mStreamTypes[i]; 1180 if (i > 0) { 1181 result.appendFormat(", "); 1182 } 1183 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1184 if (st->mute) { 1185 result.append("M"); 1186 } 1187 } 1188 result.append("\n"); 1189 write(fd, result.string(), result.length()); 1190 result.clear(); 1191 1192 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1193 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1194 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1195 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1196 1197 size_t numtracks = mTracks.size(); 1198 size_t numactive = mActiveTracks.size(); 1199 dprintf(fd, " %d Tracks", numtracks); 1200 size_t numactiveseen = 0; 1201 if (numtracks) { 1202 dprintf(fd, " of which %d are active\n", numactive); 1203 Track::appendDumpHeader(result); 1204 for (size_t i = 0; i < numtracks; ++i) { 1205 sp<Track> track = mTracks[i]; 1206 if (track != 0) { 1207 bool active = mActiveTracks.indexOf(track) >= 0; 1208 if (active) { 1209 numactiveseen++; 1210 } 1211 track->dump(buffer, SIZE, active); 1212 result.append(buffer); 1213 } 1214 } 1215 } else { 1216 result.append("\n"); 1217 } 1218 if (numactiveseen != numactive) { 1219 // some tracks in the active list were not in the tracks list 1220 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1221 " not in the track list\n"); 1222 result.append(buffer); 1223 Track::appendDumpHeader(result); 1224 for (size_t i = 0; i < numactive; ++i) { 1225 sp<Track> track = mActiveTracks[i].promote(); 1226 if (track != 0 && mTracks.indexOf(track) < 0) { 1227 track->dump(buffer, SIZE, true); 1228 result.append(buffer); 1229 } 1230 } 1231 } 1232 1233 write(fd, result.string(), result.size()); 1234} 1235 1236void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1237{ 1238 dprintf(fd, "\nOutput thread %p:\n", this); 1239 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1240 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1241 dprintf(fd, " Total writes: %d\n", mNumWrites); 1242 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1243 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1244 dprintf(fd, " Suspend count: %d\n", mSuspended); 1245 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1246 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1247 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1248 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1249 1250 dumpBase(fd, args); 1251} 1252 1253// Thread virtuals 1254 1255void AudioFlinger::PlaybackThread::onFirstRef() 1256{ 1257 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1258} 1259 1260// ThreadBase virtuals 1261void AudioFlinger::PlaybackThread::preExit() 1262{ 1263 ALOGV(" preExit()"); 1264 // FIXME this is using hard-coded strings but in the future, this functionality will be 1265 // converted to use audio HAL extensions required to support tunneling 1266 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1267} 1268 1269// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1270sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1271 const sp<AudioFlinger::Client>& client, 1272 audio_stream_type_t streamType, 1273 uint32_t sampleRate, 1274 audio_format_t format, 1275 audio_channel_mask_t channelMask, 1276 size_t *pFrameCount, 1277 const sp<IMemory>& sharedBuffer, 1278 int sessionId, 1279 IAudioFlinger::track_flags_t *flags, 1280 pid_t tid, 1281 int uid, 1282 status_t *status) 1283{ 1284 size_t frameCount = *pFrameCount; 1285 sp<Track> track; 1286 status_t lStatus; 1287 1288 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1289 1290 // client expresses a preference for FAST, but we get the final say 1291 if (*flags & IAudioFlinger::TRACK_FAST) { 1292 if ( 1293 // not timed 1294 (!isTimed) && 1295 // either of these use cases: 1296 ( 1297 // use case 1: shared buffer with any frame count 1298 ( 1299 (sharedBuffer != 0) 1300 ) || 1301 // use case 2: callback handler and frame count is default or at least as large as HAL 1302 ( 1303 (tid != -1) && 1304 ((frameCount == 0) || 1305 (frameCount >= mFrameCount)) 1306 ) 1307 ) && 1308 // PCM data 1309 audio_is_linear_pcm(format) && 1310 // mono or stereo 1311 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1312 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1313 // hardware sample rate 1314 (sampleRate == mSampleRate) && 1315 // normal mixer has an associated fast mixer 1316 hasFastMixer() && 1317 // there are sufficient fast track slots available 1318 (mFastTrackAvailMask != 0) 1319 // FIXME test that MixerThread for this fast track has a capable output HAL 1320 // FIXME add a permission test also? 1321 ) { 1322 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1323 if (frameCount == 0) { 1324 frameCount = mFrameCount * kFastTrackMultiplier; 1325 } 1326 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1327 frameCount, mFrameCount); 1328 } else { 1329 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1330 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1331 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1332 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1333 audio_is_linear_pcm(format), 1334 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1335 *flags &= ~IAudioFlinger::TRACK_FAST; 1336 // For compatibility with AudioTrack calculation, buffer depth is forced 1337 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1338 // This is probably too conservative, but legacy application code may depend on it. 1339 // If you change this calculation, also review the start threshold which is related. 1340 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1341 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1342 if (minBufCount < 2) { 1343 minBufCount = 2; 1344 } 1345 size_t minFrameCount = mNormalFrameCount * minBufCount; 1346 if (frameCount < minFrameCount) { 1347 frameCount = minFrameCount; 1348 } 1349 } 1350 } 1351 *pFrameCount = frameCount; 1352 1353 switch (mType) { 1354 1355 case DIRECT: 1356 if (audio_is_linear_pcm(format)) { 1357 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1358 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1359 "for output %p with format %#x", 1360 sampleRate, format, channelMask, mOutput, mFormat); 1361 lStatus = BAD_VALUE; 1362 goto Exit; 1363 } 1364 } 1365 break; 1366 1367 case OFFLOAD: 1368 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1369 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1370 "for output %p with format %#x", 1371 sampleRate, format, channelMask, mOutput, mFormat); 1372 lStatus = BAD_VALUE; 1373 goto Exit; 1374 } 1375 break; 1376 1377 default: 1378 if (!audio_is_linear_pcm(format)) { 1379 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1380 "for output %p with format %#x", 1381 format, mOutput, mFormat); 1382 lStatus = BAD_VALUE; 1383 goto Exit; 1384 } 1385 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1386 if (sampleRate > mSampleRate*2) { 1387 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1388 lStatus = BAD_VALUE; 1389 goto Exit; 1390 } 1391 break; 1392 1393 } 1394 1395 lStatus = initCheck(); 1396 if (lStatus != NO_ERROR) { 1397 ALOGE("createTrack_l() audio driver not initialized"); 1398 goto Exit; 1399 } 1400 1401 { // scope for mLock 1402 Mutex::Autolock _l(mLock); 1403 1404 // all tracks in same audio session must share the same routing strategy otherwise 1405 // conflicts will happen when tracks are moved from one output to another by audio policy 1406 // manager 1407 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1408 for (size_t i = 0; i < mTracks.size(); ++i) { 1409 sp<Track> t = mTracks[i]; 1410 if (t != 0 && !t->isOutputTrack()) { 1411 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1412 if (sessionId == t->sessionId() && strategy != actual) { 1413 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1414 strategy, actual); 1415 lStatus = BAD_VALUE; 1416 goto Exit; 1417 } 1418 } 1419 } 1420 1421 if (!isTimed) { 1422 track = new Track(this, client, streamType, sampleRate, format, 1423 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1424 } else { 1425 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1426 channelMask, frameCount, sharedBuffer, sessionId, uid); 1427 } 1428 1429 // new Track always returns non-NULL, 1430 // but TimedTrack::create() is a factory that could fail by returning NULL 1431 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1432 if (lStatus != NO_ERROR) { 1433 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1434 // track must be cleared from the caller as the caller has the AF lock 1435 goto Exit; 1436 } 1437 mTracks.add(track); 1438 1439 sp<EffectChain> chain = getEffectChain_l(sessionId); 1440 if (chain != 0) { 1441 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1442 track->setMainBuffer(chain->inBuffer()); 1443 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1444 chain->incTrackCnt(); 1445 } 1446 1447 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1448 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1449 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1450 // so ask activity manager to do this on our behalf 1451 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1452 } 1453 } 1454 1455 lStatus = NO_ERROR; 1456 1457Exit: 1458 *status = lStatus; 1459 return track; 1460} 1461 1462uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1463{ 1464 return latency; 1465} 1466 1467uint32_t AudioFlinger::PlaybackThread::latency() const 1468{ 1469 Mutex::Autolock _l(mLock); 1470 return latency_l(); 1471} 1472uint32_t AudioFlinger::PlaybackThread::latency_l() const 1473{ 1474 if (initCheck() == NO_ERROR) { 1475 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1476 } else { 1477 return 0; 1478 } 1479} 1480 1481void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1482{ 1483 Mutex::Autolock _l(mLock); 1484 // Don't apply master volume in SW if our HAL can do it for us. 1485 if (mOutput && mOutput->audioHwDev && 1486 mOutput->audioHwDev->canSetMasterVolume()) { 1487 mMasterVolume = 1.0; 1488 } else { 1489 mMasterVolume = value; 1490 } 1491} 1492 1493void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1494{ 1495 Mutex::Autolock _l(mLock); 1496 // Don't apply master mute in SW if our HAL can do it for us. 1497 if (mOutput && mOutput->audioHwDev && 1498 mOutput->audioHwDev->canSetMasterMute()) { 1499 mMasterMute = false; 1500 } else { 1501 mMasterMute = muted; 1502 } 1503} 1504 1505void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1506{ 1507 Mutex::Autolock _l(mLock); 1508 mStreamTypes[stream].volume = value; 1509 broadcast_l(); 1510} 1511 1512void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1513{ 1514 Mutex::Autolock _l(mLock); 1515 mStreamTypes[stream].mute = muted; 1516 broadcast_l(); 1517} 1518 1519float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1520{ 1521 Mutex::Autolock _l(mLock); 1522 return mStreamTypes[stream].volume; 1523} 1524 1525// addTrack_l() must be called with ThreadBase::mLock held 1526status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1527{ 1528 status_t status = ALREADY_EXISTS; 1529 1530 // set retry count for buffer fill 1531 track->mRetryCount = kMaxTrackStartupRetries; 1532 if (mActiveTracks.indexOf(track) < 0) { 1533 // the track is newly added, make sure it fills up all its 1534 // buffers before playing. This is to ensure the client will 1535 // effectively get the latency it requested. 1536 if (!track->isOutputTrack()) { 1537 TrackBase::track_state state = track->mState; 1538 mLock.unlock(); 1539 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1540 mLock.lock(); 1541 // abort track was stopped/paused while we released the lock 1542 if (state != track->mState) { 1543 if (status == NO_ERROR) { 1544 mLock.unlock(); 1545 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1546 mLock.lock(); 1547 } 1548 return INVALID_OPERATION; 1549 } 1550 // abort if start is rejected by audio policy manager 1551 if (status != NO_ERROR) { 1552 return PERMISSION_DENIED; 1553 } 1554#ifdef ADD_BATTERY_DATA 1555 // to track the speaker usage 1556 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1557#endif 1558 } 1559 1560 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1561 track->mResetDone = false; 1562 track->mPresentationCompleteFrames = 0; 1563 mActiveTracks.add(track); 1564 mWakeLockUids.add(track->uid()); 1565 mActiveTracksGeneration++; 1566 mLatestActiveTrack = track; 1567 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1568 if (chain != 0) { 1569 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1570 track->sessionId()); 1571 chain->incActiveTrackCnt(); 1572 } 1573 1574 status = NO_ERROR; 1575 } 1576 1577 onAddNewTrack_l(); 1578 return status; 1579} 1580 1581bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1582{ 1583 track->terminate(); 1584 // active tracks are removed by threadLoop() 1585 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1586 track->mState = TrackBase::STOPPED; 1587 if (!trackActive) { 1588 removeTrack_l(track); 1589 } else if (track->isFastTrack() || track->isOffloaded()) { 1590 track->mState = TrackBase::STOPPING_1; 1591 } 1592 1593 return trackActive; 1594} 1595 1596void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1597{ 1598 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1599 mTracks.remove(track); 1600 deleteTrackName_l(track->name()); 1601 // redundant as track is about to be destroyed, for dumpsys only 1602 track->mName = -1; 1603 if (track->isFastTrack()) { 1604 int index = track->mFastIndex; 1605 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1606 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1607 mFastTrackAvailMask |= 1 << index; 1608 // redundant as track is about to be destroyed, for dumpsys only 1609 track->mFastIndex = -1; 1610 } 1611 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1612 if (chain != 0) { 1613 chain->decTrackCnt(); 1614 } 1615} 1616 1617void AudioFlinger::PlaybackThread::broadcast_l() 1618{ 1619 // Thread could be blocked waiting for async 1620 // so signal it to handle state changes immediately 1621 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1622 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1623 mSignalPending = true; 1624 mWaitWorkCV.broadcast(); 1625} 1626 1627String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1628{ 1629 Mutex::Autolock _l(mLock); 1630 if (initCheck() != NO_ERROR) { 1631 return String8(); 1632 } 1633 1634 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1635 const String8 out_s8(s); 1636 free(s); 1637 return out_s8; 1638} 1639 1640void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1641 AudioSystem::OutputDescriptor desc; 1642 void *param2 = NULL; 1643 1644 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1645 param); 1646 1647 switch (event) { 1648 case AudioSystem::OUTPUT_OPENED: 1649 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1650 desc.channelMask = mChannelMask; 1651 desc.samplingRate = mSampleRate; 1652 desc.format = mFormat; 1653 desc.frameCount = mNormalFrameCount; // FIXME see 1654 // AudioFlinger::frameCount(audio_io_handle_t) 1655 desc.latency = latency_l(); 1656 param2 = &desc; 1657 break; 1658 1659 case AudioSystem::STREAM_CONFIG_CHANGED: 1660 param2 = ¶m; 1661 case AudioSystem::OUTPUT_CLOSED: 1662 default: 1663 break; 1664 } 1665 mAudioFlinger->audioConfigChanged(event, mId, param2); 1666} 1667 1668void AudioFlinger::PlaybackThread::writeCallback() 1669{ 1670 ALOG_ASSERT(mCallbackThread != 0); 1671 mCallbackThread->resetWriteBlocked(); 1672} 1673 1674void AudioFlinger::PlaybackThread::drainCallback() 1675{ 1676 ALOG_ASSERT(mCallbackThread != 0); 1677 mCallbackThread->resetDraining(); 1678} 1679 1680void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1681{ 1682 Mutex::Autolock _l(mLock); 1683 // reject out of sequence requests 1684 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1685 mWriteAckSequence &= ~1; 1686 mWaitWorkCV.signal(); 1687 } 1688} 1689 1690void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1691{ 1692 Mutex::Autolock _l(mLock); 1693 // reject out of sequence requests 1694 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1695 mDrainSequence &= ~1; 1696 mWaitWorkCV.signal(); 1697 } 1698} 1699 1700// static 1701int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1702 void *param __unused, 1703 void *cookie) 1704{ 1705 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1706 ALOGV("asyncCallback() event %d", event); 1707 switch (event) { 1708 case STREAM_CBK_EVENT_WRITE_READY: 1709 me->writeCallback(); 1710 break; 1711 case STREAM_CBK_EVENT_DRAIN_READY: 1712 me->drainCallback(); 1713 break; 1714 default: 1715 ALOGW("asyncCallback() unknown event %d", event); 1716 break; 1717 } 1718 return 0; 1719} 1720 1721void AudioFlinger::PlaybackThread::readOutputParameters_l() 1722{ 1723 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1724 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1725 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1726 if (!audio_is_output_channel(mChannelMask)) { 1727 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1728 } 1729 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1730 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1731 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1732 } 1733 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1734 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1735 if (!audio_is_valid_format(mFormat)) { 1736 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1737 } 1738 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1739 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1740 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1741 } 1742 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1743 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1744 mFrameCount = mBufferSize / mFrameSize; 1745 if (mFrameCount & 15) { 1746 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1747 mFrameCount); 1748 } 1749 1750 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1751 (mOutput->stream->set_callback != NULL)) { 1752 if (mOutput->stream->set_callback(mOutput->stream, 1753 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1754 mUseAsyncWrite = true; 1755 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1756 } 1757 } 1758 1759 // Calculate size of normal sink buffer relative to the HAL output buffer size 1760 double multiplier = 1.0; 1761 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1762 kUseFastMixer == FastMixer_Dynamic)) { 1763 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1764 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1765 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1766 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1767 maxNormalFrameCount = maxNormalFrameCount & ~15; 1768 if (maxNormalFrameCount < minNormalFrameCount) { 1769 maxNormalFrameCount = minNormalFrameCount; 1770 } 1771 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1772 if (multiplier <= 1.0) { 1773 multiplier = 1.0; 1774 } else if (multiplier <= 2.0) { 1775 if (2 * mFrameCount <= maxNormalFrameCount) { 1776 multiplier = 2.0; 1777 } else { 1778 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1779 } 1780 } else { 1781 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1782 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1783 // track, but we sometimes have to do this to satisfy the maximum frame count 1784 // constraint) 1785 // FIXME this rounding up should not be done if no HAL SRC 1786 uint32_t truncMult = (uint32_t) multiplier; 1787 if ((truncMult & 1)) { 1788 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1789 ++truncMult; 1790 } 1791 } 1792 multiplier = (double) truncMult; 1793 } 1794 } 1795 mNormalFrameCount = multiplier * mFrameCount; 1796 // round up to nearest 16 frames to satisfy AudioMixer 1797 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1798 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1799 mNormalFrameCount); 1800 1801 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1802 // Originally this was int16_t[] array, need to remove legacy implications. 1803 free(mSinkBuffer); 1804 mSinkBuffer = NULL; 1805 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1806 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1807 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1808 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1809 1810 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1811 // drives the output. 1812 free(mMixerBuffer); 1813 mMixerBuffer = NULL; 1814 if (mMixerBufferEnabled) { 1815 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1816 mMixerBufferSize = mNormalFrameCount * mChannelCount 1817 * audio_bytes_per_sample(mMixerBufferFormat); 1818 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1819 } 1820 free(mEffectBuffer); 1821 mEffectBuffer = NULL; 1822 if (mEffectBufferEnabled) { 1823 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1824 mEffectBufferSize = mNormalFrameCount * mChannelCount 1825 * audio_bytes_per_sample(mEffectBufferFormat); 1826 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1827 } 1828 1829 // force reconfiguration of effect chains and engines to take new buffer size and audio 1830 // parameters into account 1831 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1832 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1833 // matter. 1834 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1835 Vector< sp<EffectChain> > effectChains = mEffectChains; 1836 for (size_t i = 0; i < effectChains.size(); i ++) { 1837 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1838 } 1839} 1840 1841 1842status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1843{ 1844 if (halFrames == NULL || dspFrames == NULL) { 1845 return BAD_VALUE; 1846 } 1847 Mutex::Autolock _l(mLock); 1848 if (initCheck() != NO_ERROR) { 1849 return INVALID_OPERATION; 1850 } 1851 size_t framesWritten = mBytesWritten / mFrameSize; 1852 *halFrames = framesWritten; 1853 1854 if (isSuspended()) { 1855 // return an estimation of rendered frames when the output is suspended 1856 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1857 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1858 return NO_ERROR; 1859 } else { 1860 status_t status; 1861 uint32_t frames; 1862 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1863 *dspFrames = (size_t)frames; 1864 return status; 1865 } 1866} 1867 1868uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1869{ 1870 Mutex::Autolock _l(mLock); 1871 uint32_t result = 0; 1872 if (getEffectChain_l(sessionId) != 0) { 1873 result = EFFECT_SESSION; 1874 } 1875 1876 for (size_t i = 0; i < mTracks.size(); ++i) { 1877 sp<Track> track = mTracks[i]; 1878 if (sessionId == track->sessionId() && !track->isInvalid()) { 1879 result |= TRACK_SESSION; 1880 break; 1881 } 1882 } 1883 1884 return result; 1885} 1886 1887uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1888{ 1889 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1890 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1891 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1892 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1893 } 1894 for (size_t i = 0; i < mTracks.size(); i++) { 1895 sp<Track> track = mTracks[i]; 1896 if (sessionId == track->sessionId() && !track->isInvalid()) { 1897 return AudioSystem::getStrategyForStream(track->streamType()); 1898 } 1899 } 1900 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1901} 1902 1903 1904AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1905{ 1906 Mutex::Autolock _l(mLock); 1907 return mOutput; 1908} 1909 1910AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1911{ 1912 Mutex::Autolock _l(mLock); 1913 AudioStreamOut *output = mOutput; 1914 mOutput = NULL; 1915 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1916 // must push a NULL and wait for ack 1917 mOutputSink.clear(); 1918 mPipeSink.clear(); 1919 mNormalSink.clear(); 1920 return output; 1921} 1922 1923// this method must always be called either with ThreadBase mLock held or inside the thread loop 1924audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1925{ 1926 if (mOutput == NULL) { 1927 return NULL; 1928 } 1929 return &mOutput->stream->common; 1930} 1931 1932uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1933{ 1934 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1935} 1936 1937status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1938{ 1939 if (!isValidSyncEvent(event)) { 1940 return BAD_VALUE; 1941 } 1942 1943 Mutex::Autolock _l(mLock); 1944 1945 for (size_t i = 0; i < mTracks.size(); ++i) { 1946 sp<Track> track = mTracks[i]; 1947 if (event->triggerSession() == track->sessionId()) { 1948 (void) track->setSyncEvent(event); 1949 return NO_ERROR; 1950 } 1951 } 1952 1953 return NAME_NOT_FOUND; 1954} 1955 1956bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1957{ 1958 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1959} 1960 1961void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1962 const Vector< sp<Track> >& tracksToRemove) 1963{ 1964 size_t count = tracksToRemove.size(); 1965 if (count > 0) { 1966 for (size_t i = 0 ; i < count ; i++) { 1967 const sp<Track>& track = tracksToRemove.itemAt(i); 1968 if (!track->isOutputTrack()) { 1969 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1970#ifdef ADD_BATTERY_DATA 1971 // to track the speaker usage 1972 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1973#endif 1974 if (track->isTerminated()) { 1975 AudioSystem::releaseOutput(mId); 1976 } 1977 } 1978 } 1979 } 1980} 1981 1982void AudioFlinger::PlaybackThread::checkSilentMode_l() 1983{ 1984 if (!mMasterMute) { 1985 char value[PROPERTY_VALUE_MAX]; 1986 if (property_get("ro.audio.silent", value, "0") > 0) { 1987 char *endptr; 1988 unsigned long ul = strtoul(value, &endptr, 0); 1989 if (*endptr == '\0' && ul != 0) { 1990 ALOGD("Silence is golden"); 1991 // The setprop command will not allow a property to be changed after 1992 // the first time it is set, so we don't have to worry about un-muting. 1993 setMasterMute_l(true); 1994 } 1995 } 1996 } 1997} 1998 1999// shared by MIXER and DIRECT, overridden by DUPLICATING 2000ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2001{ 2002 // FIXME rewrite to reduce number of system calls 2003 mLastWriteTime = systemTime(); 2004 mInWrite = true; 2005 ssize_t bytesWritten; 2006 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2007 2008 // If an NBAIO sink is present, use it to write the normal mixer's submix 2009 if (mNormalSink != 0) { 2010 const size_t count = mBytesRemaining / mFrameSize; 2011 2012 ATRACE_BEGIN("write"); 2013 // update the setpoint when AudioFlinger::mScreenState changes 2014 uint32_t screenState = AudioFlinger::mScreenState; 2015 if (screenState != mScreenState) { 2016 mScreenState = screenState; 2017 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2018 if (pipe != NULL) { 2019 pipe->setAvgFrames((mScreenState & 1) ? 2020 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2021 } 2022 } 2023 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2024 ATRACE_END(); 2025 if (framesWritten > 0) { 2026 bytesWritten = framesWritten * mFrameSize; 2027 } else { 2028 bytesWritten = framesWritten; 2029 } 2030 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2031 if (status == NO_ERROR) { 2032 size_t totalFramesWritten = mNormalSink->framesWritten(); 2033 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2034 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2035 mLatchDValid = true; 2036 } 2037 } 2038 // otherwise use the HAL / AudioStreamOut directly 2039 } else { 2040 // Direct output and offload threads 2041 2042 if (mUseAsyncWrite) { 2043 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2044 mWriteAckSequence += 2; 2045 mWriteAckSequence |= 1; 2046 ALOG_ASSERT(mCallbackThread != 0); 2047 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2048 } 2049 // FIXME We should have an implementation of timestamps for direct output threads. 2050 // They are used e.g for multichannel PCM playback over HDMI. 2051 bytesWritten = mOutput->stream->write(mOutput->stream, 2052 (char *)mSinkBuffer + offset, mBytesRemaining); 2053 if (mUseAsyncWrite && 2054 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2055 // do not wait for async callback in case of error of full write 2056 mWriteAckSequence &= ~1; 2057 ALOG_ASSERT(mCallbackThread != 0); 2058 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2059 } 2060 } 2061 2062 mNumWrites++; 2063 mInWrite = false; 2064 mStandby = false; 2065 return bytesWritten; 2066} 2067 2068void AudioFlinger::PlaybackThread::threadLoop_drain() 2069{ 2070 if (mOutput->stream->drain) { 2071 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2072 if (mUseAsyncWrite) { 2073 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2074 mDrainSequence |= 1; 2075 ALOG_ASSERT(mCallbackThread != 0); 2076 mCallbackThread->setDraining(mDrainSequence); 2077 } 2078 mOutput->stream->drain(mOutput->stream, 2079 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2080 : AUDIO_DRAIN_ALL); 2081 } 2082} 2083 2084void AudioFlinger::PlaybackThread::threadLoop_exit() 2085{ 2086 // Default implementation has nothing to do 2087} 2088 2089/* 2090The derived values that are cached: 2091 - mSinkBufferSize from frame count * frame size 2092 - activeSleepTime from activeSleepTimeUs() 2093 - idleSleepTime from idleSleepTimeUs() 2094 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2095 - maxPeriod from frame count and sample rate (MIXER only) 2096 2097The parameters that affect these derived values are: 2098 - frame count 2099 - frame size 2100 - sample rate 2101 - device type: A2DP or not 2102 - device latency 2103 - format: PCM or not 2104 - active sleep time 2105 - idle sleep time 2106*/ 2107 2108void AudioFlinger::PlaybackThread::cacheParameters_l() 2109{ 2110 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2111 activeSleepTime = activeSleepTimeUs(); 2112 idleSleepTime = idleSleepTimeUs(); 2113} 2114 2115void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2116{ 2117 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2118 this, streamType, mTracks.size()); 2119 Mutex::Autolock _l(mLock); 2120 2121 size_t size = mTracks.size(); 2122 for (size_t i = 0; i < size; i++) { 2123 sp<Track> t = mTracks[i]; 2124 if (t->streamType() == streamType) { 2125 t->invalidate(); 2126 } 2127 } 2128} 2129 2130status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2131{ 2132 int session = chain->sessionId(); 2133 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2134 ? mEffectBuffer : mSinkBuffer); 2135 bool ownsBuffer = false; 2136 2137 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2138 if (session > 0) { 2139 // Only one effect chain can be present in direct output thread and it uses 2140 // the sink buffer as input 2141 if (mType != DIRECT) { 2142 size_t numSamples = mNormalFrameCount * mChannelCount; 2143 buffer = new int16_t[numSamples]; 2144 memset(buffer, 0, numSamples * sizeof(int16_t)); 2145 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2146 ownsBuffer = true; 2147 } 2148 2149 // Attach all tracks with same session ID to this chain. 2150 for (size_t i = 0; i < mTracks.size(); ++i) { 2151 sp<Track> track = mTracks[i]; 2152 if (session == track->sessionId()) { 2153 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2154 buffer); 2155 track->setMainBuffer(buffer); 2156 chain->incTrackCnt(); 2157 } 2158 } 2159 2160 // indicate all active tracks in the chain 2161 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2162 sp<Track> track = mActiveTracks[i].promote(); 2163 if (track == 0) { 2164 continue; 2165 } 2166 if (session == track->sessionId()) { 2167 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2168 chain->incActiveTrackCnt(); 2169 } 2170 } 2171 } 2172 2173 chain->setInBuffer(buffer, ownsBuffer); 2174 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2175 ? mEffectBuffer : mSinkBuffer)); 2176 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2177 // chains list in order to be processed last as it contains output stage effects 2178 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2179 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2180 // after track specific effects and before output stage 2181 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2182 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2183 // Effect chain for other sessions are inserted at beginning of effect 2184 // chains list to be processed before output mix effects. Relative order between other 2185 // sessions is not important 2186 size_t size = mEffectChains.size(); 2187 size_t i = 0; 2188 for (i = 0; i < size; i++) { 2189 if (mEffectChains[i]->sessionId() < session) { 2190 break; 2191 } 2192 } 2193 mEffectChains.insertAt(chain, i); 2194 checkSuspendOnAddEffectChain_l(chain); 2195 2196 return NO_ERROR; 2197} 2198 2199size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2200{ 2201 int session = chain->sessionId(); 2202 2203 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2204 2205 for (size_t i = 0; i < mEffectChains.size(); i++) { 2206 if (chain == mEffectChains[i]) { 2207 mEffectChains.removeAt(i); 2208 // detach all active tracks from the chain 2209 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2210 sp<Track> track = mActiveTracks[i].promote(); 2211 if (track == 0) { 2212 continue; 2213 } 2214 if (session == track->sessionId()) { 2215 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2216 chain.get(), session); 2217 chain->decActiveTrackCnt(); 2218 } 2219 } 2220 2221 // detach all tracks with same session ID from this chain 2222 for (size_t i = 0; i < mTracks.size(); ++i) { 2223 sp<Track> track = mTracks[i]; 2224 if (session == track->sessionId()) { 2225 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2226 chain->decTrackCnt(); 2227 } 2228 } 2229 break; 2230 } 2231 } 2232 return mEffectChains.size(); 2233} 2234 2235status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2236 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2237{ 2238 Mutex::Autolock _l(mLock); 2239 return attachAuxEffect_l(track, EffectId); 2240} 2241 2242status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2243 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2244{ 2245 status_t status = NO_ERROR; 2246 2247 if (EffectId == 0) { 2248 track->setAuxBuffer(0, NULL); 2249 } else { 2250 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2251 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2252 if (effect != 0) { 2253 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2254 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2255 } else { 2256 status = INVALID_OPERATION; 2257 } 2258 } else { 2259 status = BAD_VALUE; 2260 } 2261 } 2262 return status; 2263} 2264 2265void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2266{ 2267 for (size_t i = 0; i < mTracks.size(); ++i) { 2268 sp<Track> track = mTracks[i]; 2269 if (track->auxEffectId() == effectId) { 2270 attachAuxEffect_l(track, 0); 2271 } 2272 } 2273} 2274 2275bool AudioFlinger::PlaybackThread::threadLoop() 2276{ 2277 Vector< sp<Track> > tracksToRemove; 2278 2279 standbyTime = systemTime(); 2280 2281 // MIXER 2282 nsecs_t lastWarning = 0; 2283 2284 // DUPLICATING 2285 // FIXME could this be made local to while loop? 2286 writeFrames = 0; 2287 2288 int lastGeneration = 0; 2289 2290 cacheParameters_l(); 2291 sleepTime = idleSleepTime; 2292 2293 if (mType == MIXER) { 2294 sleepTimeShift = 0; 2295 } 2296 2297 CpuStats cpuStats; 2298 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2299 2300 acquireWakeLock(); 2301 2302 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2303 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2304 // and then that string will be logged at the next convenient opportunity. 2305 const char *logString = NULL; 2306 2307 checkSilentMode_l(); 2308 2309 while (!exitPending()) 2310 { 2311 cpuStats.sample(myName); 2312 2313 Vector< sp<EffectChain> > effectChains; 2314 2315 { // scope for mLock 2316 2317 Mutex::Autolock _l(mLock); 2318 2319 processConfigEvents_l(); 2320 2321 if (logString != NULL) { 2322 mNBLogWriter->logTimestamp(); 2323 mNBLogWriter->log(logString); 2324 logString = NULL; 2325 } 2326 2327 if (mLatchDValid) { 2328 mLatchQ = mLatchD; 2329 mLatchDValid = false; 2330 mLatchQValid = true; 2331 } 2332 2333 saveOutputTracks(); 2334 if (mSignalPending) { 2335 // A signal was raised while we were unlocked 2336 mSignalPending = false; 2337 } else if (waitingAsyncCallback_l()) { 2338 if (exitPending()) { 2339 break; 2340 } 2341 releaseWakeLock_l(); 2342 mWakeLockUids.clear(); 2343 mActiveTracksGeneration++; 2344 ALOGV("wait async completion"); 2345 mWaitWorkCV.wait(mLock); 2346 ALOGV("async completion/wake"); 2347 acquireWakeLock_l(); 2348 standbyTime = systemTime() + standbyDelay; 2349 sleepTime = 0; 2350 2351 continue; 2352 } 2353 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2354 isSuspended()) { 2355 // put audio hardware into standby after short delay 2356 if (shouldStandby_l()) { 2357 2358 threadLoop_standby(); 2359 2360 mStandby = true; 2361 } 2362 2363 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2364 // we're about to wait, flush the binder command buffer 2365 IPCThreadState::self()->flushCommands(); 2366 2367 clearOutputTracks(); 2368 2369 if (exitPending()) { 2370 break; 2371 } 2372 2373 releaseWakeLock_l(); 2374 mWakeLockUids.clear(); 2375 mActiveTracksGeneration++; 2376 // wait until we have something to do... 2377 ALOGV("%s going to sleep", myName.string()); 2378 mWaitWorkCV.wait(mLock); 2379 ALOGV("%s waking up", myName.string()); 2380 acquireWakeLock_l(); 2381 2382 mMixerStatus = MIXER_IDLE; 2383 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2384 mBytesWritten = 0; 2385 mBytesRemaining = 0; 2386 checkSilentMode_l(); 2387 2388 standbyTime = systemTime() + standbyDelay; 2389 sleepTime = idleSleepTime; 2390 if (mType == MIXER) { 2391 sleepTimeShift = 0; 2392 } 2393 2394 continue; 2395 } 2396 } 2397 // mMixerStatusIgnoringFastTracks is also updated internally 2398 mMixerStatus = prepareTracks_l(&tracksToRemove); 2399 2400 // compare with previously applied list 2401 if (lastGeneration != mActiveTracksGeneration) { 2402 // update wakelock 2403 updateWakeLockUids_l(mWakeLockUids); 2404 lastGeneration = mActiveTracksGeneration; 2405 } 2406 2407 // prevent any changes in effect chain list and in each effect chain 2408 // during mixing and effect process as the audio buffers could be deleted 2409 // or modified if an effect is created or deleted 2410 lockEffectChains_l(effectChains); 2411 } // mLock scope ends 2412 2413 if (mBytesRemaining == 0) { 2414 mCurrentWriteLength = 0; 2415 if (mMixerStatus == MIXER_TRACKS_READY) { 2416 // threadLoop_mix() sets mCurrentWriteLength 2417 threadLoop_mix(); 2418 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2419 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2420 // threadLoop_sleepTime sets sleepTime to 0 if data 2421 // must be written to HAL 2422 threadLoop_sleepTime(); 2423 if (sleepTime == 0) { 2424 mCurrentWriteLength = mSinkBufferSize; 2425 } 2426 } 2427 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2428 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2429 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2430 // or mSinkBuffer (if there are no effects). 2431 // 2432 // This is done pre-effects computation; if effects change to 2433 // support higher precision, this needs to move. 2434 // 2435 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2436 // TODO use sleepTime == 0 as an additional condition. 2437 if (mMixerBufferValid) { 2438 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2439 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2440 2441 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2442 mNormalFrameCount * mChannelCount); 2443 } 2444 2445 mBytesRemaining = mCurrentWriteLength; 2446 if (isSuspended()) { 2447 sleepTime = suspendSleepTimeUs(); 2448 // simulate write to HAL when suspended 2449 mBytesWritten += mSinkBufferSize; 2450 mBytesRemaining = 0; 2451 } 2452 2453 // only process effects if we're going to write 2454 if (sleepTime == 0 && mType != OFFLOAD) { 2455 for (size_t i = 0; i < effectChains.size(); i ++) { 2456 effectChains[i]->process_l(); 2457 } 2458 } 2459 } 2460 // Process effect chains for offloaded thread even if no audio 2461 // was read from audio track: process only updates effect state 2462 // and thus does have to be synchronized with audio writes but may have 2463 // to be called while waiting for async write callback 2464 if (mType == OFFLOAD) { 2465 for (size_t i = 0; i < effectChains.size(); i ++) { 2466 effectChains[i]->process_l(); 2467 } 2468 } 2469 2470 // Only if the Effects buffer is enabled and there is data in the 2471 // Effects buffer (buffer valid), we need to 2472 // copy into the sink buffer. 2473 // TODO use sleepTime == 0 as an additional condition. 2474 if (mEffectBufferValid) { 2475 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2476 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2477 mNormalFrameCount * mChannelCount); 2478 } 2479 2480 // enable changes in effect chain 2481 unlockEffectChains(effectChains); 2482 2483 if (!waitingAsyncCallback()) { 2484 // sleepTime == 0 means we must write to audio hardware 2485 if (sleepTime == 0) { 2486 if (mBytesRemaining) { 2487 ssize_t ret = threadLoop_write(); 2488 if (ret < 0) { 2489 mBytesRemaining = 0; 2490 } else { 2491 mBytesWritten += ret; 2492 mBytesRemaining -= ret; 2493 } 2494 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2495 (mMixerStatus == MIXER_DRAIN_ALL)) { 2496 threadLoop_drain(); 2497 } 2498 if (mType == MIXER) { 2499 // write blocked detection 2500 nsecs_t now = systemTime(); 2501 nsecs_t delta = now - mLastWriteTime; 2502 if (!mStandby && delta > maxPeriod) { 2503 mNumDelayedWrites++; 2504 if ((now - lastWarning) > kWarningThrottleNs) { 2505 ATRACE_NAME("underrun"); 2506 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2507 ns2ms(delta), mNumDelayedWrites, this); 2508 lastWarning = now; 2509 } 2510 } 2511 } 2512 2513 } else { 2514 usleep(sleepTime); 2515 } 2516 } 2517 2518 // Finally let go of removed track(s), without the lock held 2519 // since we can't guarantee the destructors won't acquire that 2520 // same lock. This will also mutate and push a new fast mixer state. 2521 threadLoop_removeTracks(tracksToRemove); 2522 tracksToRemove.clear(); 2523 2524 // FIXME I don't understand the need for this here; 2525 // it was in the original code but maybe the 2526 // assignment in saveOutputTracks() makes this unnecessary? 2527 clearOutputTracks(); 2528 2529 // Effect chains will be actually deleted here if they were removed from 2530 // mEffectChains list during mixing or effects processing 2531 effectChains.clear(); 2532 2533 // FIXME Note that the above .clear() is no longer necessary since effectChains 2534 // is now local to this block, but will keep it for now (at least until merge done). 2535 } 2536 2537 threadLoop_exit(); 2538 2539 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2540 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2541 // put output stream into standby mode 2542 if (!mStandby) { 2543 mOutput->stream->common.standby(&mOutput->stream->common); 2544 } 2545 } 2546 2547 releaseWakeLock(); 2548 mWakeLockUids.clear(); 2549 mActiveTracksGeneration++; 2550 2551 ALOGV("Thread %p type %d exiting", this, mType); 2552 return false; 2553} 2554 2555// removeTracks_l() must be called with ThreadBase::mLock held 2556void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2557{ 2558 size_t count = tracksToRemove.size(); 2559 if (count > 0) { 2560 for (size_t i=0 ; i<count ; i++) { 2561 const sp<Track>& track = tracksToRemove.itemAt(i); 2562 mActiveTracks.remove(track); 2563 mWakeLockUids.remove(track->uid()); 2564 mActiveTracksGeneration++; 2565 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2566 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2567 if (chain != 0) { 2568 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2569 track->sessionId()); 2570 chain->decActiveTrackCnt(); 2571 } 2572 if (track->isTerminated()) { 2573 removeTrack_l(track); 2574 } 2575 } 2576 } 2577 2578} 2579 2580status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2581{ 2582 if (mNormalSink != 0) { 2583 return mNormalSink->getTimestamp(timestamp); 2584 } 2585 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2586 uint64_t position64; 2587 int ret = mOutput->stream->get_presentation_position( 2588 mOutput->stream, &position64, ×tamp.mTime); 2589 if (ret == 0) { 2590 timestamp.mPosition = (uint32_t)position64; 2591 return NO_ERROR; 2592 } 2593 } 2594 return INVALID_OPERATION; 2595} 2596// ---------------------------------------------------------------------------- 2597 2598AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2599 audio_io_handle_t id, audio_devices_t device, type_t type) 2600 : PlaybackThread(audioFlinger, output, id, device, type), 2601 // mAudioMixer below 2602 // mFastMixer below 2603 mFastMixerFutex(0) 2604 // mOutputSink below 2605 // mPipeSink below 2606 // mNormalSink below 2607{ 2608 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2609 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2610 "mFrameCount=%d, mNormalFrameCount=%d", 2611 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2612 mNormalFrameCount); 2613 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2614 2615 // FIXME - Current mixer implementation only supports stereo output 2616 if (mChannelCount != FCC_2) { 2617 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2618 } 2619 2620 // create an NBAIO sink for the HAL output stream, and negotiate 2621 mOutputSink = new AudioStreamOutSink(output->stream); 2622 size_t numCounterOffers = 0; 2623 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2624 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2625 ALOG_ASSERT(index == 0); 2626 2627 // initialize fast mixer depending on configuration 2628 bool initFastMixer; 2629 switch (kUseFastMixer) { 2630 case FastMixer_Never: 2631 initFastMixer = false; 2632 break; 2633 case FastMixer_Always: 2634 initFastMixer = true; 2635 break; 2636 case FastMixer_Static: 2637 case FastMixer_Dynamic: 2638 initFastMixer = mFrameCount < mNormalFrameCount; 2639 break; 2640 } 2641 if (initFastMixer) { 2642 2643 // create a MonoPipe to connect our submix to FastMixer 2644 NBAIO_Format format = mOutputSink->format(); 2645 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2646 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2647 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2648 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2649 const NBAIO_Format offers[1] = {format}; 2650 size_t numCounterOffers = 0; 2651 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2652 ALOG_ASSERT(index == 0); 2653 monoPipe->setAvgFrames((mScreenState & 1) ? 2654 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2655 mPipeSink = monoPipe; 2656 2657#ifdef TEE_SINK 2658 if (mTeeSinkOutputEnabled) { 2659 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2660 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2661 numCounterOffers = 0; 2662 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2663 ALOG_ASSERT(index == 0); 2664 mTeeSink = teeSink; 2665 PipeReader *teeSource = new PipeReader(*teeSink); 2666 numCounterOffers = 0; 2667 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2668 ALOG_ASSERT(index == 0); 2669 mTeeSource = teeSource; 2670 } 2671#endif 2672 2673 // create fast mixer and configure it initially with just one fast track for our submix 2674 mFastMixer = new FastMixer(); 2675 FastMixerStateQueue *sq = mFastMixer->sq(); 2676#ifdef STATE_QUEUE_DUMP 2677 sq->setObserverDump(&mStateQueueObserverDump); 2678 sq->setMutatorDump(&mStateQueueMutatorDump); 2679#endif 2680 FastMixerState *state = sq->begin(); 2681 FastTrack *fastTrack = &state->mFastTracks[0]; 2682 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2683 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2684 fastTrack->mVolumeProvider = NULL; 2685 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2686 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2687 fastTrack->mGeneration++; 2688 state->mFastTracksGen++; 2689 state->mTrackMask = 1; 2690 // fast mixer will use the HAL output sink 2691 state->mOutputSink = mOutputSink.get(); 2692 state->mOutputSinkGen++; 2693 state->mFrameCount = mFrameCount; 2694 state->mCommand = FastMixerState::COLD_IDLE; 2695 // already done in constructor initialization list 2696 //mFastMixerFutex = 0; 2697 state->mColdFutexAddr = &mFastMixerFutex; 2698 state->mColdGen++; 2699 state->mDumpState = &mFastMixerDumpState; 2700#ifdef TEE_SINK 2701 state->mTeeSink = mTeeSink.get(); 2702#endif 2703 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2704 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2705 sq->end(); 2706 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2707 2708 // start the fast mixer 2709 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2710 pid_t tid = mFastMixer->getTid(); 2711 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2712 if (err != 0) { 2713 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2714 kPriorityFastMixer, getpid_cached, tid, err); 2715 } 2716 2717#ifdef AUDIO_WATCHDOG 2718 // create and start the watchdog 2719 mAudioWatchdog = new AudioWatchdog(); 2720 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2721 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2722 tid = mAudioWatchdog->getTid(); 2723 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2724 if (err != 0) { 2725 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2726 kPriorityFastMixer, getpid_cached, tid, err); 2727 } 2728#endif 2729 2730 } else { 2731 mFastMixer = NULL; 2732 } 2733 2734 switch (kUseFastMixer) { 2735 case FastMixer_Never: 2736 case FastMixer_Dynamic: 2737 mNormalSink = mOutputSink; 2738 break; 2739 case FastMixer_Always: 2740 mNormalSink = mPipeSink; 2741 break; 2742 case FastMixer_Static: 2743 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2744 break; 2745 } 2746} 2747 2748AudioFlinger::MixerThread::~MixerThread() 2749{ 2750 if (mFastMixer != NULL) { 2751 FastMixerStateQueue *sq = mFastMixer->sq(); 2752 FastMixerState *state = sq->begin(); 2753 if (state->mCommand == FastMixerState::COLD_IDLE) { 2754 int32_t old = android_atomic_inc(&mFastMixerFutex); 2755 if (old == -1) { 2756 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2757 } 2758 } 2759 state->mCommand = FastMixerState::EXIT; 2760 sq->end(); 2761 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2762 mFastMixer->join(); 2763 // Though the fast mixer thread has exited, it's state queue is still valid. 2764 // We'll use that extract the final state which contains one remaining fast track 2765 // corresponding to our sub-mix. 2766 state = sq->begin(); 2767 ALOG_ASSERT(state->mTrackMask == 1); 2768 FastTrack *fastTrack = &state->mFastTracks[0]; 2769 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2770 delete fastTrack->mBufferProvider; 2771 sq->end(false /*didModify*/); 2772 delete mFastMixer; 2773#ifdef AUDIO_WATCHDOG 2774 if (mAudioWatchdog != 0) { 2775 mAudioWatchdog->requestExit(); 2776 mAudioWatchdog->requestExitAndWait(); 2777 mAudioWatchdog.clear(); 2778 } 2779#endif 2780 } 2781 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2782 delete mAudioMixer; 2783} 2784 2785 2786uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2787{ 2788 if (mFastMixer != NULL) { 2789 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2790 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2791 } 2792 return latency; 2793} 2794 2795 2796void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2797{ 2798 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2799} 2800 2801ssize_t AudioFlinger::MixerThread::threadLoop_write() 2802{ 2803 // FIXME we should only do one push per cycle; confirm this is true 2804 // Start the fast mixer if it's not already running 2805 if (mFastMixer != NULL) { 2806 FastMixerStateQueue *sq = mFastMixer->sq(); 2807 FastMixerState *state = sq->begin(); 2808 if (state->mCommand != FastMixerState::MIX_WRITE && 2809 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2810 if (state->mCommand == FastMixerState::COLD_IDLE) { 2811 int32_t old = android_atomic_inc(&mFastMixerFutex); 2812 if (old == -1) { 2813 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2814 } 2815#ifdef AUDIO_WATCHDOG 2816 if (mAudioWatchdog != 0) { 2817 mAudioWatchdog->resume(); 2818 } 2819#endif 2820 } 2821 state->mCommand = FastMixerState::MIX_WRITE; 2822 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2823 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2824 sq->end(); 2825 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2826 if (kUseFastMixer == FastMixer_Dynamic) { 2827 mNormalSink = mPipeSink; 2828 } 2829 } else { 2830 sq->end(false /*didModify*/); 2831 } 2832 } 2833 return PlaybackThread::threadLoop_write(); 2834} 2835 2836void AudioFlinger::MixerThread::threadLoop_standby() 2837{ 2838 // Idle the fast mixer if it's currently running 2839 if (mFastMixer != NULL) { 2840 FastMixerStateQueue *sq = mFastMixer->sq(); 2841 FastMixerState *state = sq->begin(); 2842 if (!(state->mCommand & FastMixerState::IDLE)) { 2843 state->mCommand = FastMixerState::COLD_IDLE; 2844 state->mColdFutexAddr = &mFastMixerFutex; 2845 state->mColdGen++; 2846 mFastMixerFutex = 0; 2847 sq->end(); 2848 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2849 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2850 if (kUseFastMixer == FastMixer_Dynamic) { 2851 mNormalSink = mOutputSink; 2852 } 2853#ifdef AUDIO_WATCHDOG 2854 if (mAudioWatchdog != 0) { 2855 mAudioWatchdog->pause(); 2856 } 2857#endif 2858 } else { 2859 sq->end(false /*didModify*/); 2860 } 2861 } 2862 PlaybackThread::threadLoop_standby(); 2863} 2864 2865bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2866{ 2867 return false; 2868} 2869 2870bool AudioFlinger::PlaybackThread::shouldStandby_l() 2871{ 2872 return !mStandby; 2873} 2874 2875bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2876{ 2877 Mutex::Autolock _l(mLock); 2878 return waitingAsyncCallback_l(); 2879} 2880 2881// shared by MIXER and DIRECT, overridden by DUPLICATING 2882void AudioFlinger::PlaybackThread::threadLoop_standby() 2883{ 2884 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2885 mOutput->stream->common.standby(&mOutput->stream->common); 2886 if (mUseAsyncWrite != 0) { 2887 // discard any pending drain or write ack by incrementing sequence 2888 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2889 mDrainSequence = (mDrainSequence + 2) & ~1; 2890 ALOG_ASSERT(mCallbackThread != 0); 2891 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2892 mCallbackThread->setDraining(mDrainSequence); 2893 } 2894} 2895 2896void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2897{ 2898 ALOGV("signal playback thread"); 2899 broadcast_l(); 2900} 2901 2902void AudioFlinger::MixerThread::threadLoop_mix() 2903{ 2904 // obtain the presentation timestamp of the next output buffer 2905 int64_t pts; 2906 status_t status = INVALID_OPERATION; 2907 2908 if (mNormalSink != 0) { 2909 status = mNormalSink->getNextWriteTimestamp(&pts); 2910 } else { 2911 status = mOutputSink->getNextWriteTimestamp(&pts); 2912 } 2913 2914 if (status != NO_ERROR) { 2915 pts = AudioBufferProvider::kInvalidPTS; 2916 } 2917 2918 // mix buffers... 2919 mAudioMixer->process(pts); 2920 mCurrentWriteLength = mSinkBufferSize; 2921 // increase sleep time progressively when application underrun condition clears. 2922 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2923 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2924 // such that we would underrun the audio HAL. 2925 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2926 sleepTimeShift--; 2927 } 2928 sleepTime = 0; 2929 standbyTime = systemTime() + standbyDelay; 2930 //TODO: delay standby when effects have a tail 2931} 2932 2933void AudioFlinger::MixerThread::threadLoop_sleepTime() 2934{ 2935 // If no tracks are ready, sleep once for the duration of an output 2936 // buffer size, then write 0s to the output 2937 if (sleepTime == 0) { 2938 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2939 sleepTime = activeSleepTime >> sleepTimeShift; 2940 if (sleepTime < kMinThreadSleepTimeUs) { 2941 sleepTime = kMinThreadSleepTimeUs; 2942 } 2943 // reduce sleep time in case of consecutive application underruns to avoid 2944 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2945 // duration we would end up writing less data than needed by the audio HAL if 2946 // the condition persists. 2947 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2948 sleepTimeShift++; 2949 } 2950 } else { 2951 sleepTime = idleSleepTime; 2952 } 2953 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2954 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2955 // before effects processing or output. 2956 if (mMixerBufferValid) { 2957 memset(mMixerBuffer, 0, mMixerBufferSize); 2958 } else { 2959 memset(mSinkBuffer, 0, mSinkBufferSize); 2960 } 2961 sleepTime = 0; 2962 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2963 "anticipated start"); 2964 } 2965 // TODO add standby time extension fct of effect tail 2966} 2967 2968// prepareTracks_l() must be called with ThreadBase::mLock held 2969AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2970 Vector< sp<Track> > *tracksToRemove) 2971{ 2972 2973 mixer_state mixerStatus = MIXER_IDLE; 2974 // find out which tracks need to be processed 2975 size_t count = mActiveTracks.size(); 2976 size_t mixedTracks = 0; 2977 size_t tracksWithEffect = 0; 2978 // counts only _active_ fast tracks 2979 size_t fastTracks = 0; 2980 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2981 2982 float masterVolume = mMasterVolume; 2983 bool masterMute = mMasterMute; 2984 2985 if (masterMute) { 2986 masterVolume = 0; 2987 } 2988 // Delegate master volume control to effect in output mix effect chain if needed 2989 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2990 if (chain != 0) { 2991 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2992 chain->setVolume_l(&v, &v); 2993 masterVolume = (float)((v + (1 << 23)) >> 24); 2994 chain.clear(); 2995 } 2996 2997 // prepare a new state to push 2998 FastMixerStateQueue *sq = NULL; 2999 FastMixerState *state = NULL; 3000 bool didModify = false; 3001 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3002 if (mFastMixer != NULL) { 3003 sq = mFastMixer->sq(); 3004 state = sq->begin(); 3005 } 3006 3007 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3008 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3009 3010 for (size_t i=0 ; i<count ; i++) { 3011 const sp<Track> t = mActiveTracks[i].promote(); 3012 if (t == 0) { 3013 continue; 3014 } 3015 3016 // this const just means the local variable doesn't change 3017 Track* const track = t.get(); 3018 3019 // process fast tracks 3020 if (track->isFastTrack()) { 3021 3022 // It's theoretically possible (though unlikely) for a fast track to be created 3023 // and then removed within the same normal mix cycle. This is not a problem, as 3024 // the track never becomes active so it's fast mixer slot is never touched. 3025 // The converse, of removing an (active) track and then creating a new track 3026 // at the identical fast mixer slot within the same normal mix cycle, 3027 // is impossible because the slot isn't marked available until the end of each cycle. 3028 int j = track->mFastIndex; 3029 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3030 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3031 FastTrack *fastTrack = &state->mFastTracks[j]; 3032 3033 // Determine whether the track is currently in underrun condition, 3034 // and whether it had a recent underrun. 3035 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3036 FastTrackUnderruns underruns = ftDump->mUnderruns; 3037 uint32_t recentFull = (underruns.mBitFields.mFull - 3038 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3039 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3040 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3041 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3042 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3043 uint32_t recentUnderruns = recentPartial + recentEmpty; 3044 track->mObservedUnderruns = underruns; 3045 // don't count underruns that occur while stopping or pausing 3046 // or stopped which can occur when flush() is called while active 3047 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3048 recentUnderruns > 0) { 3049 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3050 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3051 } 3052 3053 // This is similar to the state machine for normal tracks, 3054 // with a few modifications for fast tracks. 3055 bool isActive = true; 3056 switch (track->mState) { 3057 case TrackBase::STOPPING_1: 3058 // track stays active in STOPPING_1 state until first underrun 3059 if (recentUnderruns > 0 || track->isTerminated()) { 3060 track->mState = TrackBase::STOPPING_2; 3061 } 3062 break; 3063 case TrackBase::PAUSING: 3064 // ramp down is not yet implemented 3065 track->setPaused(); 3066 break; 3067 case TrackBase::RESUMING: 3068 // ramp up is not yet implemented 3069 track->mState = TrackBase::ACTIVE; 3070 break; 3071 case TrackBase::ACTIVE: 3072 if (recentFull > 0 || recentPartial > 0) { 3073 // track has provided at least some frames recently: reset retry count 3074 track->mRetryCount = kMaxTrackRetries; 3075 } 3076 if (recentUnderruns == 0) { 3077 // no recent underruns: stay active 3078 break; 3079 } 3080 // there has recently been an underrun of some kind 3081 if (track->sharedBuffer() == 0) { 3082 // were any of the recent underruns "empty" (no frames available)? 3083 if (recentEmpty == 0) { 3084 // no, then ignore the partial underruns as they are allowed indefinitely 3085 break; 3086 } 3087 // there has recently been an "empty" underrun: decrement the retry counter 3088 if (--(track->mRetryCount) > 0) { 3089 break; 3090 } 3091 // indicate to client process that the track was disabled because of underrun; 3092 // it will then automatically call start() when data is available 3093 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3094 // remove from active list, but state remains ACTIVE [confusing but true] 3095 isActive = false; 3096 break; 3097 } 3098 // fall through 3099 case TrackBase::STOPPING_2: 3100 case TrackBase::PAUSED: 3101 case TrackBase::STOPPED: 3102 case TrackBase::FLUSHED: // flush() while active 3103 // Check for presentation complete if track is inactive 3104 // We have consumed all the buffers of this track. 3105 // This would be incomplete if we auto-paused on underrun 3106 { 3107 size_t audioHALFrames = 3108 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3109 size_t framesWritten = mBytesWritten / mFrameSize; 3110 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3111 // track stays in active list until presentation is complete 3112 break; 3113 } 3114 } 3115 if (track->isStopping_2()) { 3116 track->mState = TrackBase::STOPPED; 3117 } 3118 if (track->isStopped()) { 3119 // Can't reset directly, as fast mixer is still polling this track 3120 // track->reset(); 3121 // So instead mark this track as needing to be reset after push with ack 3122 resetMask |= 1 << i; 3123 } 3124 isActive = false; 3125 break; 3126 case TrackBase::IDLE: 3127 default: 3128 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3129 } 3130 3131 if (isActive) { 3132 // was it previously inactive? 3133 if (!(state->mTrackMask & (1 << j))) { 3134 ExtendedAudioBufferProvider *eabp = track; 3135 VolumeProvider *vp = track; 3136 fastTrack->mBufferProvider = eabp; 3137 fastTrack->mVolumeProvider = vp; 3138 fastTrack->mChannelMask = track->mChannelMask; 3139 fastTrack->mFormat = track->mFormat; 3140 fastTrack->mGeneration++; 3141 state->mTrackMask |= 1 << j; 3142 didModify = true; 3143 // no acknowledgement required for newly active tracks 3144 } 3145 // cache the combined master volume and stream type volume for fast mixer; this 3146 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3147 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3148 ++fastTracks; 3149 } else { 3150 // was it previously active? 3151 if (state->mTrackMask & (1 << j)) { 3152 fastTrack->mBufferProvider = NULL; 3153 fastTrack->mGeneration++; 3154 state->mTrackMask &= ~(1 << j); 3155 didModify = true; 3156 // If any fast tracks were removed, we must wait for acknowledgement 3157 // because we're about to decrement the last sp<> on those tracks. 3158 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3159 } else { 3160 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3161 } 3162 tracksToRemove->add(track); 3163 // Avoids a misleading display in dumpsys 3164 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3165 } 3166 continue; 3167 } 3168 3169 { // local variable scope to avoid goto warning 3170 3171 audio_track_cblk_t* cblk = track->cblk(); 3172 3173 // The first time a track is added we wait 3174 // for all its buffers to be filled before processing it 3175 int name = track->name(); 3176 // make sure that we have enough frames to mix one full buffer. 3177 // enforce this condition only once to enable draining the buffer in case the client 3178 // app does not call stop() and relies on underrun to stop: 3179 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3180 // during last round 3181 size_t desiredFrames; 3182 uint32_t sr = track->sampleRate(); 3183 if (sr == mSampleRate) { 3184 desiredFrames = mNormalFrameCount; 3185 } else { 3186 // +1 for rounding and +1 for additional sample needed for interpolation 3187 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3188 // add frames already consumed but not yet released by the resampler 3189 // because mAudioTrackServerProxy->framesReady() will include these frames 3190 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3191#if 0 3192 // the minimum track buffer size is normally twice the number of frames necessary 3193 // to fill one buffer and the resampler should not leave more than one buffer worth 3194 // of unreleased frames after each pass, but just in case... 3195 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3196#endif 3197 } 3198 uint32_t minFrames = 1; 3199 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3200 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3201 minFrames = desiredFrames; 3202 } 3203 3204 size_t framesReady = track->framesReady(); 3205 if ((framesReady >= minFrames) && track->isReady() && 3206 !track->isPaused() && !track->isTerminated()) 3207 { 3208 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3209 3210 mixedTracks++; 3211 3212 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3213 // there is an effect chain connected to the track 3214 chain.clear(); 3215 if (track->mainBuffer() != mSinkBuffer && 3216 track->mainBuffer() != mMixerBuffer) { 3217 if (mEffectBufferEnabled) { 3218 mEffectBufferValid = true; // Later can set directly. 3219 } 3220 chain = getEffectChain_l(track->sessionId()); 3221 // Delegate volume control to effect in track effect chain if needed 3222 if (chain != 0) { 3223 tracksWithEffect++; 3224 } else { 3225 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3226 "session %d", 3227 name, track->sessionId()); 3228 } 3229 } 3230 3231 3232 int param = AudioMixer::VOLUME; 3233 if (track->mFillingUpStatus == Track::FS_FILLED) { 3234 // no ramp for the first volume setting 3235 track->mFillingUpStatus = Track::FS_ACTIVE; 3236 if (track->mState == TrackBase::RESUMING) { 3237 track->mState = TrackBase::ACTIVE; 3238 param = AudioMixer::RAMP_VOLUME; 3239 } 3240 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3241 // FIXME should not make a decision based on mServer 3242 } else if (cblk->mServer != 0) { 3243 // If the track is stopped before the first frame was mixed, 3244 // do not apply ramp 3245 param = AudioMixer::RAMP_VOLUME; 3246 } 3247 3248 // compute volume for this track 3249 uint32_t vl, vr, va; 3250 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3251 vl = vr = va = 0; 3252 if (track->isPausing()) { 3253 track->setPaused(); 3254 } 3255 } else { 3256 3257 // read original volumes with volume control 3258 float typeVolume = mStreamTypes[track->streamType()].volume; 3259 float v = masterVolume * typeVolume; 3260 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3261 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3262 float vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3263 float vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3264 // track volumes come from shared memory, so can't be trusted and must be clamped 3265 if (vlf > GAIN_FLOAT_UNITY) { 3266 ALOGV("Track left volume out of range: %.3g", vlf); 3267 vlf = GAIN_FLOAT_UNITY; 3268 } 3269 if (vrf > GAIN_FLOAT_UNITY) { 3270 ALOGV("Track right volume out of range: %.3g", vrf); 3271 vrf = GAIN_FLOAT_UNITY; 3272 } 3273 // now apply the master volume and stream type volume 3274 // FIXME we're losing the wonderful dynamic range in the minifloat representation 3275 float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT); 3276 vl = (uint32_t) (v8_24 * vlf); 3277 vr = (uint32_t) (v8_24 * vrf); 3278 // assuming master volume and stream type volume each go up to 1.0, 3279 // vl and vr are now in 8.24 format 3280 3281 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3282 // send level comes from shared memory and so may be corrupt 3283 if (sendLevel > MAX_GAIN_INT) { 3284 ALOGV("Track send level out of range: %04X", sendLevel); 3285 sendLevel = MAX_GAIN_INT; 3286 } 3287 va = (uint32_t)(v * sendLevel); 3288 } 3289 3290 // Delegate volume control to effect in track effect chain if needed 3291 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3292 // Do not ramp volume if volume is controlled by effect 3293 param = AudioMixer::VOLUME; 3294 track->mHasVolumeController = true; 3295 } else { 3296 // force no volume ramp when volume controller was just disabled or removed 3297 // from effect chain to avoid volume spike 3298 if (track->mHasVolumeController) { 3299 param = AudioMixer::VOLUME; 3300 } 3301 track->mHasVolumeController = false; 3302 } 3303 3304 // FIXME Use float 3305 // Convert volumes from 8.24 to 4.12 format 3306 // This additional clamping is needed in case chain->setVolume_l() overshot 3307 vl = (vl + (1 << 11)) >> 12; 3308 if (vl > MAX_GAIN_INT) { 3309 vl = MAX_GAIN_INT; 3310 } 3311 vr = (vr + (1 << 11)) >> 12; 3312 if (vr > MAX_GAIN_INT) { 3313 vr = MAX_GAIN_INT; 3314 } 3315 3316 if (va > MAX_GAIN_INT) { 3317 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3318 } 3319 3320 // XXX: these things DON'T need to be done each time 3321 mAudioMixer->setBufferProvider(name, track); 3322 mAudioMixer->enable(name); 3323 3324 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3325 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3326 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3327 mAudioMixer->setParameter( 3328 name, 3329 AudioMixer::TRACK, 3330 AudioMixer::FORMAT, (void *)track->format()); 3331 mAudioMixer->setParameter( 3332 name, 3333 AudioMixer::TRACK, 3334 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3335 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3336 uint32_t maxSampleRate = mSampleRate * 2; 3337 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3338 if (reqSampleRate == 0) { 3339 reqSampleRate = mSampleRate; 3340 } else if (reqSampleRate > maxSampleRate) { 3341 reqSampleRate = maxSampleRate; 3342 } 3343 mAudioMixer->setParameter( 3344 name, 3345 AudioMixer::RESAMPLE, 3346 AudioMixer::SAMPLE_RATE, 3347 (void *)(uintptr_t)reqSampleRate); 3348 /* 3349 * Select the appropriate output buffer for the track. 3350 * 3351 * Tracks with effects go into their own effects chain buffer 3352 * and from there into either mEffectBuffer or mSinkBuffer. 3353 * 3354 * Other tracks can use mMixerBuffer for higher precision 3355 * channel accumulation. If this buffer is enabled 3356 * (mMixerBufferEnabled true), then selected tracks will accumulate 3357 * into it. 3358 * 3359 */ 3360 if (mMixerBufferEnabled 3361 && (track->mainBuffer() == mSinkBuffer 3362 || track->mainBuffer() == mMixerBuffer)) { 3363 mAudioMixer->setParameter( 3364 name, 3365 AudioMixer::TRACK, 3366 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3367 mAudioMixer->setParameter( 3368 name, 3369 AudioMixer::TRACK, 3370 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3371 // TODO: override track->mainBuffer()? 3372 mMixerBufferValid = true; 3373 } else { 3374 mAudioMixer->setParameter( 3375 name, 3376 AudioMixer::TRACK, 3377 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3378 mAudioMixer->setParameter( 3379 name, 3380 AudioMixer::TRACK, 3381 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3382 } 3383 mAudioMixer->setParameter( 3384 name, 3385 AudioMixer::TRACK, 3386 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3387 3388 // reset retry count 3389 track->mRetryCount = kMaxTrackRetries; 3390 3391 // If one track is ready, set the mixer ready if: 3392 // - the mixer was not ready during previous round OR 3393 // - no other track is not ready 3394 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3395 mixerStatus != MIXER_TRACKS_ENABLED) { 3396 mixerStatus = MIXER_TRACKS_READY; 3397 } 3398 } else { 3399 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3400 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3401 } 3402 // clear effect chain input buffer if an active track underruns to avoid sending 3403 // previous audio buffer again to effects 3404 chain = getEffectChain_l(track->sessionId()); 3405 if (chain != 0) { 3406 chain->clearInputBuffer(); 3407 } 3408 3409 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3410 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3411 track->isStopped() || track->isPaused()) { 3412 // We have consumed all the buffers of this track. 3413 // Remove it from the list of active tracks. 3414 // TODO: use actual buffer filling status instead of latency when available from 3415 // audio HAL 3416 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3417 size_t framesWritten = mBytesWritten / mFrameSize; 3418 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3419 if (track->isStopped()) { 3420 track->reset(); 3421 } 3422 tracksToRemove->add(track); 3423 } 3424 } else { 3425 // No buffers for this track. Give it a few chances to 3426 // fill a buffer, then remove it from active list. 3427 if (--(track->mRetryCount) <= 0) { 3428 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3429 tracksToRemove->add(track); 3430 // indicate to client process that the track was disabled because of underrun; 3431 // it will then automatically call start() when data is available 3432 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3433 // If one track is not ready, mark the mixer also not ready if: 3434 // - the mixer was ready during previous round OR 3435 // - no other track is ready 3436 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3437 mixerStatus != MIXER_TRACKS_READY) { 3438 mixerStatus = MIXER_TRACKS_ENABLED; 3439 } 3440 } 3441 mAudioMixer->disable(name); 3442 } 3443 3444 } // local variable scope to avoid goto warning 3445track_is_ready: ; 3446 3447 } 3448 3449 // Push the new FastMixer state if necessary 3450 bool pauseAudioWatchdog = false; 3451 if (didModify) { 3452 state->mFastTracksGen++; 3453 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3454 if (kUseFastMixer == FastMixer_Dynamic && 3455 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3456 state->mCommand = FastMixerState::COLD_IDLE; 3457 state->mColdFutexAddr = &mFastMixerFutex; 3458 state->mColdGen++; 3459 mFastMixerFutex = 0; 3460 if (kUseFastMixer == FastMixer_Dynamic) { 3461 mNormalSink = mOutputSink; 3462 } 3463 // If we go into cold idle, need to wait for acknowledgement 3464 // so that fast mixer stops doing I/O. 3465 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3466 pauseAudioWatchdog = true; 3467 } 3468 } 3469 if (sq != NULL) { 3470 sq->end(didModify); 3471 sq->push(block); 3472 } 3473#ifdef AUDIO_WATCHDOG 3474 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3475 mAudioWatchdog->pause(); 3476 } 3477#endif 3478 3479 // Now perform the deferred reset on fast tracks that have stopped 3480 while (resetMask != 0) { 3481 size_t i = __builtin_ctz(resetMask); 3482 ALOG_ASSERT(i < count); 3483 resetMask &= ~(1 << i); 3484 sp<Track> t = mActiveTracks[i].promote(); 3485 if (t == 0) { 3486 continue; 3487 } 3488 Track* track = t.get(); 3489 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3490 track->reset(); 3491 } 3492 3493 // remove all the tracks that need to be... 3494 removeTracks_l(*tracksToRemove); 3495 3496 // sink or mix buffer must be cleared if all tracks are connected to an 3497 // effect chain as in this case the mixer will not write to the sink or mix buffer 3498 // and track effects will accumulate into it 3499 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3500 (mixedTracks == 0 && fastTracks > 0))) { 3501 // FIXME as a performance optimization, should remember previous zero status 3502 if (mMixerBufferValid) { 3503 memset(mMixerBuffer, 0, mMixerBufferSize); 3504 // TODO: In testing, mSinkBuffer below need not be cleared because 3505 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3506 // after mixing. 3507 // 3508 // To enforce this guarantee: 3509 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3510 // (mixedTracks == 0 && fastTracks > 0)) 3511 // must imply MIXER_TRACKS_READY. 3512 // Later, we may clear buffers regardless, and skip much of this logic. 3513 } 3514 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3515 if (mEffectBufferValid) { 3516 memset(mEffectBuffer, 0, mEffectBufferSize); 3517 } 3518 // FIXME as a performance optimization, should remember previous zero status 3519 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3520 } 3521 3522 // if any fast tracks, then status is ready 3523 mMixerStatusIgnoringFastTracks = mixerStatus; 3524 if (fastTracks > 0) { 3525 mixerStatus = MIXER_TRACKS_READY; 3526 } 3527 return mixerStatus; 3528} 3529 3530// getTrackName_l() must be called with ThreadBase::mLock held 3531int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3532 audio_format_t format, int sessionId) 3533{ 3534 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3535} 3536 3537// deleteTrackName_l() must be called with ThreadBase::mLock held 3538void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3539{ 3540 ALOGV("remove track (%d) and delete from mixer", name); 3541 mAudioMixer->deleteTrackName(name); 3542} 3543 3544// checkForNewParameter_l() must be called with ThreadBase::mLock held 3545bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3546 status_t& status) 3547{ 3548 bool reconfig = false; 3549 3550 status = NO_ERROR; 3551 3552 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3553 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3554 if (mFastMixer != NULL) { 3555 FastMixerStateQueue *sq = mFastMixer->sq(); 3556 FastMixerState *state = sq->begin(); 3557 if (!(state->mCommand & FastMixerState::IDLE)) { 3558 previousCommand = state->mCommand; 3559 state->mCommand = FastMixerState::HOT_IDLE; 3560 sq->end(); 3561 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3562 } else { 3563 sq->end(false /*didModify*/); 3564 } 3565 } 3566 3567 AudioParameter param = AudioParameter(keyValuePair); 3568 int value; 3569 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3570 reconfig = true; 3571 } 3572 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3573 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3574 status = BAD_VALUE; 3575 } else { 3576 // no need to save value, since it's constant 3577 reconfig = true; 3578 } 3579 } 3580 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3581 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3582 status = BAD_VALUE; 3583 } else { 3584 // no need to save value, since it's constant 3585 reconfig = true; 3586 } 3587 } 3588 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3589 // do not accept frame count changes if tracks are open as the track buffer 3590 // size depends on frame count and correct behavior would not be guaranteed 3591 // if frame count is changed after track creation 3592 if (!mTracks.isEmpty()) { 3593 status = INVALID_OPERATION; 3594 } else { 3595 reconfig = true; 3596 } 3597 } 3598 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3599#ifdef ADD_BATTERY_DATA 3600 // when changing the audio output device, call addBatteryData to notify 3601 // the change 3602 if (mOutDevice != value) { 3603 uint32_t params = 0; 3604 // check whether speaker is on 3605 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3606 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3607 } 3608 3609 audio_devices_t deviceWithoutSpeaker 3610 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3611 // check if any other device (except speaker) is on 3612 if (value & deviceWithoutSpeaker ) { 3613 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3614 } 3615 3616 if (params != 0) { 3617 addBatteryData(params); 3618 } 3619 } 3620#endif 3621 3622 // forward device change to effects that have requested to be 3623 // aware of attached audio device. 3624 if (value != AUDIO_DEVICE_NONE) { 3625 mOutDevice = value; 3626 for (size_t i = 0; i < mEffectChains.size(); i++) { 3627 mEffectChains[i]->setDevice_l(mOutDevice); 3628 } 3629 } 3630 } 3631 3632 if (status == NO_ERROR) { 3633 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3634 keyValuePair.string()); 3635 if (!mStandby && status == INVALID_OPERATION) { 3636 mOutput->stream->common.standby(&mOutput->stream->common); 3637 mStandby = true; 3638 mBytesWritten = 0; 3639 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3640 keyValuePair.string()); 3641 } 3642 if (status == NO_ERROR && reconfig) { 3643 readOutputParameters_l(); 3644 delete mAudioMixer; 3645 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3646 for (size_t i = 0; i < mTracks.size() ; i++) { 3647 int name = getTrackName_l(mTracks[i]->mChannelMask, 3648 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3649 if (name < 0) { 3650 break; 3651 } 3652 mTracks[i]->mName = name; 3653 } 3654 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3655 } 3656 } 3657 3658 if (!(previousCommand & FastMixerState::IDLE)) { 3659 ALOG_ASSERT(mFastMixer != NULL); 3660 FastMixerStateQueue *sq = mFastMixer->sq(); 3661 FastMixerState *state = sq->begin(); 3662 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3663 state->mCommand = previousCommand; 3664 sq->end(); 3665 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3666 } 3667 3668 return reconfig; 3669} 3670 3671 3672void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3673{ 3674 const size_t SIZE = 256; 3675 char buffer[SIZE]; 3676 String8 result; 3677 3678 PlaybackThread::dumpInternals(fd, args); 3679 3680 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3681 3682 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3683 const FastMixerDumpState copy(mFastMixerDumpState); 3684 copy.dump(fd); 3685 3686#ifdef STATE_QUEUE_DUMP 3687 // Similar for state queue 3688 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3689 observerCopy.dump(fd); 3690 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3691 mutatorCopy.dump(fd); 3692#endif 3693 3694#ifdef TEE_SINK 3695 // Write the tee output to a .wav file 3696 dumpTee(fd, mTeeSource, mId); 3697#endif 3698 3699#ifdef AUDIO_WATCHDOG 3700 if (mAudioWatchdog != 0) { 3701 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3702 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3703 wdCopy.dump(fd); 3704 } 3705#endif 3706} 3707 3708uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3709{ 3710 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3711} 3712 3713uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3714{ 3715 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3716} 3717 3718void AudioFlinger::MixerThread::cacheParameters_l() 3719{ 3720 PlaybackThread::cacheParameters_l(); 3721 3722 // FIXME: Relaxed timing because of a certain device that can't meet latency 3723 // Should be reduced to 2x after the vendor fixes the driver issue 3724 // increase threshold again due to low power audio mode. The way this warning 3725 // threshold is calculated and its usefulness should be reconsidered anyway. 3726 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3727} 3728 3729// ---------------------------------------------------------------------------- 3730 3731AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3732 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3733 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3734 // mLeftVolFloat, mRightVolFloat 3735{ 3736} 3737 3738AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3739 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3740 ThreadBase::type_t type) 3741 : PlaybackThread(audioFlinger, output, id, device, type) 3742 // mLeftVolFloat, mRightVolFloat 3743{ 3744} 3745 3746AudioFlinger::DirectOutputThread::~DirectOutputThread() 3747{ 3748} 3749 3750void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3751{ 3752 audio_track_cblk_t* cblk = track->cblk(); 3753 float left, right; 3754 3755 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3756 left = right = 0; 3757 } else { 3758 float typeVolume = mStreamTypes[track->streamType()].volume; 3759 float v = mMasterVolume * typeVolume; 3760 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3761 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3762 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3763 if (left > GAIN_FLOAT_UNITY) { 3764 left = GAIN_FLOAT_UNITY; 3765 } 3766 left *= v; 3767 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3768 if (right > GAIN_FLOAT_UNITY) { 3769 right = GAIN_FLOAT_UNITY; 3770 } 3771 right *= v; 3772 } 3773 3774 if (lastTrack) { 3775 if (left != mLeftVolFloat || right != mRightVolFloat) { 3776 mLeftVolFloat = left; 3777 mRightVolFloat = right; 3778 3779 // Convert volumes from float to 8.24 3780 uint32_t vl = (uint32_t)(left * (1 << 24)); 3781 uint32_t vr = (uint32_t)(right * (1 << 24)); 3782 3783 // Delegate volume control to effect in track effect chain if needed 3784 // only one effect chain can be present on DirectOutputThread, so if 3785 // there is one, the track is connected to it 3786 if (!mEffectChains.isEmpty()) { 3787 mEffectChains[0]->setVolume_l(&vl, &vr); 3788 left = (float)vl / (1 << 24); 3789 right = (float)vr / (1 << 24); 3790 } 3791 if (mOutput->stream->set_volume) { 3792 mOutput->stream->set_volume(mOutput->stream, left, right); 3793 } 3794 } 3795 } 3796} 3797 3798 3799AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3800 Vector< sp<Track> > *tracksToRemove 3801) 3802{ 3803 size_t count = mActiveTracks.size(); 3804 mixer_state mixerStatus = MIXER_IDLE; 3805 3806 // find out which tracks need to be processed 3807 for (size_t i = 0; i < count; i++) { 3808 sp<Track> t = mActiveTracks[i].promote(); 3809 // The track died recently 3810 if (t == 0) { 3811 continue; 3812 } 3813 3814 Track* const track = t.get(); 3815 audio_track_cblk_t* cblk = track->cblk(); 3816 // Only consider last track started for volume and mixer state control. 3817 // In theory an older track could underrun and restart after the new one starts 3818 // but as we only care about the transition phase between two tracks on a 3819 // direct output, it is not a problem to ignore the underrun case. 3820 sp<Track> l = mLatestActiveTrack.promote(); 3821 bool last = l.get() == track; 3822 3823 // The first time a track is added we wait 3824 // for all its buffers to be filled before processing it 3825 uint32_t minFrames; 3826 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3827 minFrames = mNormalFrameCount; 3828 } else { 3829 minFrames = 1; 3830 } 3831 3832 if ((track->framesReady() >= minFrames) && track->isReady() && 3833 !track->isPaused() && !track->isTerminated()) 3834 { 3835 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3836 3837 if (track->mFillingUpStatus == Track::FS_FILLED) { 3838 track->mFillingUpStatus = Track::FS_ACTIVE; 3839 // make sure processVolume_l() will apply new volume even if 0 3840 mLeftVolFloat = mRightVolFloat = -1.0; 3841 if (track->mState == TrackBase::RESUMING) { 3842 track->mState = TrackBase::ACTIVE; 3843 } 3844 } 3845 3846 // compute volume for this track 3847 processVolume_l(track, last); 3848 if (last) { 3849 // reset retry count 3850 track->mRetryCount = kMaxTrackRetriesDirect; 3851 mActiveTrack = t; 3852 mixerStatus = MIXER_TRACKS_READY; 3853 } 3854 } else { 3855 // clear effect chain input buffer if the last active track started underruns 3856 // to avoid sending previous audio buffer again to effects 3857 if (!mEffectChains.isEmpty() && last) { 3858 mEffectChains[0]->clearInputBuffer(); 3859 } 3860 3861 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3862 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3863 track->isStopped() || track->isPaused()) { 3864 // We have consumed all the buffers of this track. 3865 // Remove it from the list of active tracks. 3866 // TODO: implement behavior for compressed audio 3867 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3868 size_t framesWritten = mBytesWritten / mFrameSize; 3869 if (mStandby || !last || 3870 track->presentationComplete(framesWritten, audioHALFrames)) { 3871 if (track->isStopped()) { 3872 track->reset(); 3873 } 3874 tracksToRemove->add(track); 3875 } 3876 } else { 3877 // No buffers for this track. Give it a few chances to 3878 // fill a buffer, then remove it from active list. 3879 // Only consider last track started for mixer state control 3880 if (--(track->mRetryCount) <= 0) { 3881 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3882 tracksToRemove->add(track); 3883 // indicate to client process that the track was disabled because of underrun; 3884 // it will then automatically call start() when data is available 3885 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3886 } else if (last) { 3887 mixerStatus = MIXER_TRACKS_ENABLED; 3888 } 3889 } 3890 } 3891 } 3892 3893 // remove all the tracks that need to be... 3894 removeTracks_l(*tracksToRemove); 3895 3896 return mixerStatus; 3897} 3898 3899void AudioFlinger::DirectOutputThread::threadLoop_mix() 3900{ 3901 size_t frameCount = mFrameCount; 3902 int8_t *curBuf = (int8_t *)mSinkBuffer; 3903 // output audio to hardware 3904 while (frameCount) { 3905 AudioBufferProvider::Buffer buffer; 3906 buffer.frameCount = frameCount; 3907 mActiveTrack->getNextBuffer(&buffer); 3908 if (buffer.raw == NULL) { 3909 memset(curBuf, 0, frameCount * mFrameSize); 3910 break; 3911 } 3912 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3913 frameCount -= buffer.frameCount; 3914 curBuf += buffer.frameCount * mFrameSize; 3915 mActiveTrack->releaseBuffer(&buffer); 3916 } 3917 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3918 sleepTime = 0; 3919 standbyTime = systemTime() + standbyDelay; 3920 mActiveTrack.clear(); 3921} 3922 3923void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3924{ 3925 if (sleepTime == 0) { 3926 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3927 sleepTime = activeSleepTime; 3928 } else { 3929 sleepTime = idleSleepTime; 3930 } 3931 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3932 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3933 sleepTime = 0; 3934 } 3935} 3936 3937// getTrackName_l() must be called with ThreadBase::mLock held 3938int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3939 audio_format_t format __unused, int sessionId __unused) 3940{ 3941 return 0; 3942} 3943 3944// deleteTrackName_l() must be called with ThreadBase::mLock held 3945void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3946{ 3947} 3948 3949// checkForNewParameter_l() must be called with ThreadBase::mLock held 3950bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 3951 status_t& status) 3952{ 3953 bool reconfig = false; 3954 3955 status = NO_ERROR; 3956 3957 AudioParameter param = AudioParameter(keyValuePair); 3958 int value; 3959 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3960 // forward device change to effects that have requested to be 3961 // aware of attached audio device. 3962 if (value != AUDIO_DEVICE_NONE) { 3963 mOutDevice = value; 3964 for (size_t i = 0; i < mEffectChains.size(); i++) { 3965 mEffectChains[i]->setDevice_l(mOutDevice); 3966 } 3967 } 3968 } 3969 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3970 // do not accept frame count changes if tracks are open as the track buffer 3971 // size depends on frame count and correct behavior would not be garantied 3972 // if frame count is changed after track creation 3973 if (!mTracks.isEmpty()) { 3974 status = INVALID_OPERATION; 3975 } else { 3976 reconfig = true; 3977 } 3978 } 3979 if (status == NO_ERROR) { 3980 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3981 keyValuePair.string()); 3982 if (!mStandby && status == INVALID_OPERATION) { 3983 mOutput->stream->common.standby(&mOutput->stream->common); 3984 mStandby = true; 3985 mBytesWritten = 0; 3986 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3987 keyValuePair.string()); 3988 } 3989 if (status == NO_ERROR && reconfig) { 3990 readOutputParameters_l(); 3991 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3992 } 3993 } 3994 3995 return reconfig; 3996} 3997 3998uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3999{ 4000 uint32_t time; 4001 if (audio_is_linear_pcm(mFormat)) { 4002 time = PlaybackThread::activeSleepTimeUs(); 4003 } else { 4004 time = 10000; 4005 } 4006 return time; 4007} 4008 4009uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4010{ 4011 uint32_t time; 4012 if (audio_is_linear_pcm(mFormat)) { 4013 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4014 } else { 4015 time = 10000; 4016 } 4017 return time; 4018} 4019 4020uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4021{ 4022 uint32_t time; 4023 if (audio_is_linear_pcm(mFormat)) { 4024 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4025 } else { 4026 time = 10000; 4027 } 4028 return time; 4029} 4030 4031void AudioFlinger::DirectOutputThread::cacheParameters_l() 4032{ 4033 PlaybackThread::cacheParameters_l(); 4034 4035 // use shorter standby delay as on normal output to release 4036 // hardware resources as soon as possible 4037 if (audio_is_linear_pcm(mFormat)) { 4038 standbyDelay = microseconds(activeSleepTime*2); 4039 } else { 4040 standbyDelay = kOffloadStandbyDelayNs; 4041 } 4042} 4043 4044// ---------------------------------------------------------------------------- 4045 4046AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4047 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4048 : Thread(false /*canCallJava*/), 4049 mPlaybackThread(playbackThread), 4050 mWriteAckSequence(0), 4051 mDrainSequence(0) 4052{ 4053} 4054 4055AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4056{ 4057} 4058 4059void AudioFlinger::AsyncCallbackThread::onFirstRef() 4060{ 4061 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4062} 4063 4064bool AudioFlinger::AsyncCallbackThread::threadLoop() 4065{ 4066 while (!exitPending()) { 4067 uint32_t writeAckSequence; 4068 uint32_t drainSequence; 4069 4070 { 4071 Mutex::Autolock _l(mLock); 4072 while (!((mWriteAckSequence & 1) || 4073 (mDrainSequence & 1) || 4074 exitPending())) { 4075 mWaitWorkCV.wait(mLock); 4076 } 4077 4078 if (exitPending()) { 4079 break; 4080 } 4081 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4082 mWriteAckSequence, mDrainSequence); 4083 writeAckSequence = mWriteAckSequence; 4084 mWriteAckSequence &= ~1; 4085 drainSequence = mDrainSequence; 4086 mDrainSequence &= ~1; 4087 } 4088 { 4089 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4090 if (playbackThread != 0) { 4091 if (writeAckSequence & 1) { 4092 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4093 } 4094 if (drainSequence & 1) { 4095 playbackThread->resetDraining(drainSequence >> 1); 4096 } 4097 } 4098 } 4099 } 4100 return false; 4101} 4102 4103void AudioFlinger::AsyncCallbackThread::exit() 4104{ 4105 ALOGV("AsyncCallbackThread::exit"); 4106 Mutex::Autolock _l(mLock); 4107 requestExit(); 4108 mWaitWorkCV.broadcast(); 4109} 4110 4111void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4112{ 4113 Mutex::Autolock _l(mLock); 4114 // bit 0 is cleared 4115 mWriteAckSequence = sequence << 1; 4116} 4117 4118void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4119{ 4120 Mutex::Autolock _l(mLock); 4121 // ignore unexpected callbacks 4122 if (mWriteAckSequence & 2) { 4123 mWriteAckSequence |= 1; 4124 mWaitWorkCV.signal(); 4125 } 4126} 4127 4128void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4129{ 4130 Mutex::Autolock _l(mLock); 4131 // bit 0 is cleared 4132 mDrainSequence = sequence << 1; 4133} 4134 4135void AudioFlinger::AsyncCallbackThread::resetDraining() 4136{ 4137 Mutex::Autolock _l(mLock); 4138 // ignore unexpected callbacks 4139 if (mDrainSequence & 2) { 4140 mDrainSequence |= 1; 4141 mWaitWorkCV.signal(); 4142 } 4143} 4144 4145 4146// ---------------------------------------------------------------------------- 4147AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4148 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4149 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4150 mHwPaused(false), 4151 mFlushPending(false), 4152 mPausedBytesRemaining(0) 4153{ 4154 //FIXME: mStandby should be set to true by ThreadBase constructor 4155 mStandby = true; 4156} 4157 4158void AudioFlinger::OffloadThread::threadLoop_exit() 4159{ 4160 if (mFlushPending || mHwPaused) { 4161 // If a flush is pending or track was paused, just discard buffered data 4162 flushHw_l(); 4163 } else { 4164 mMixerStatus = MIXER_DRAIN_ALL; 4165 threadLoop_drain(); 4166 } 4167 if (mUseAsyncWrite) { 4168 ALOG_ASSERT(mCallbackThread != 0); 4169 mCallbackThread->exit(); 4170 } 4171 PlaybackThread::threadLoop_exit(); 4172} 4173 4174AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4175 Vector< sp<Track> > *tracksToRemove 4176) 4177{ 4178 size_t count = mActiveTracks.size(); 4179 4180 mixer_state mixerStatus = MIXER_IDLE; 4181 bool doHwPause = false; 4182 bool doHwResume = false; 4183 4184 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4185 4186 // find out which tracks need to be processed 4187 for (size_t i = 0; i < count; i++) { 4188 sp<Track> t = mActiveTracks[i].promote(); 4189 // The track died recently 4190 if (t == 0) { 4191 continue; 4192 } 4193 Track* const track = t.get(); 4194 audio_track_cblk_t* cblk = track->cblk(); 4195 // Only consider last track started for volume and mixer state control. 4196 // In theory an older track could underrun and restart after the new one starts 4197 // but as we only care about the transition phase between two tracks on a 4198 // direct output, it is not a problem to ignore the underrun case. 4199 sp<Track> l = mLatestActiveTrack.promote(); 4200 bool last = l.get() == track; 4201 4202 if (track->isInvalid()) { 4203 ALOGW("An invalidated track shouldn't be in active list"); 4204 tracksToRemove->add(track); 4205 continue; 4206 } 4207 4208 if (track->mState == TrackBase::IDLE) { 4209 ALOGW("An idle track shouldn't be in active list"); 4210 continue; 4211 } 4212 4213 if (track->isPausing()) { 4214 track->setPaused(); 4215 if (last) { 4216 if (!mHwPaused) { 4217 doHwPause = true; 4218 mHwPaused = true; 4219 } 4220 // If we were part way through writing the mixbuffer to 4221 // the HAL we must save this until we resume 4222 // BUG - this will be wrong if a different track is made active, 4223 // in that case we want to discard the pending data in the 4224 // mixbuffer and tell the client to present it again when the 4225 // track is resumed 4226 mPausedWriteLength = mCurrentWriteLength; 4227 mPausedBytesRemaining = mBytesRemaining; 4228 mBytesRemaining = 0; // stop writing 4229 } 4230 tracksToRemove->add(track); 4231 } else if (track->isFlushPending()) { 4232 track->flushAck(); 4233 if (last) { 4234 mFlushPending = true; 4235 } 4236 } else if (track->isResumePending()){ 4237 track->resumeAck(); 4238 if (last) { 4239 if (mPausedBytesRemaining) { 4240 // Need to continue write that was interrupted 4241 mCurrentWriteLength = mPausedWriteLength; 4242 mBytesRemaining = mPausedBytesRemaining; 4243 mPausedBytesRemaining = 0; 4244 } 4245 if (mHwPaused) { 4246 doHwResume = true; 4247 mHwPaused = false; 4248 // threadLoop_mix() will handle the case that we need to 4249 // resume an interrupted write 4250 } 4251 // enable write to audio HAL 4252 sleepTime = 0; 4253 4254 // Do not handle new data in this iteration even if track->framesReady() 4255 mixerStatus = MIXER_TRACKS_ENABLED; 4256 } 4257 } else if (track->framesReady() && track->isReady() && 4258 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4259 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4260 if (track->mFillingUpStatus == Track::FS_FILLED) { 4261 track->mFillingUpStatus = Track::FS_ACTIVE; 4262 // make sure processVolume_l() will apply new volume even if 0 4263 mLeftVolFloat = mRightVolFloat = -1.0; 4264 } 4265 4266 if (last) { 4267 sp<Track> previousTrack = mPreviousTrack.promote(); 4268 if (previousTrack != 0) { 4269 if (track != previousTrack.get()) { 4270 // Flush any data still being written from last track 4271 mBytesRemaining = 0; 4272 if (mPausedBytesRemaining) { 4273 // Last track was paused so we also need to flush saved 4274 // mixbuffer state and invalidate track so that it will 4275 // re-submit that unwritten data when it is next resumed 4276 mPausedBytesRemaining = 0; 4277 // Invalidate is a bit drastic - would be more efficient 4278 // to have a flag to tell client that some of the 4279 // previously written data was lost 4280 previousTrack->invalidate(); 4281 } 4282 // flush data already sent to the DSP if changing audio session as audio 4283 // comes from a different source. Also invalidate previous track to force a 4284 // seek when resuming. 4285 if (previousTrack->sessionId() != track->sessionId()) { 4286 previousTrack->invalidate(); 4287 } 4288 } 4289 } 4290 mPreviousTrack = track; 4291 // reset retry count 4292 track->mRetryCount = kMaxTrackRetriesOffload; 4293 mActiveTrack = t; 4294 mixerStatus = MIXER_TRACKS_READY; 4295 } 4296 } else { 4297 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4298 if (track->isStopping_1()) { 4299 // Hardware buffer can hold a large amount of audio so we must 4300 // wait for all current track's data to drain before we say 4301 // that the track is stopped. 4302 if (mBytesRemaining == 0) { 4303 // Only start draining when all data in mixbuffer 4304 // has been written 4305 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4306 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4307 // do not drain if no data was ever sent to HAL (mStandby == true) 4308 if (last && !mStandby) { 4309 // do not modify drain sequence if we are already draining. This happens 4310 // when resuming from pause after drain. 4311 if ((mDrainSequence & 1) == 0) { 4312 sleepTime = 0; 4313 standbyTime = systemTime() + standbyDelay; 4314 mixerStatus = MIXER_DRAIN_TRACK; 4315 mDrainSequence += 2; 4316 } 4317 if (mHwPaused) { 4318 // It is possible to move from PAUSED to STOPPING_1 without 4319 // a resume so we must ensure hardware is running 4320 doHwResume = true; 4321 mHwPaused = false; 4322 } 4323 } 4324 } 4325 } else if (track->isStopping_2()) { 4326 // Drain has completed or we are in standby, signal presentation complete 4327 if (!(mDrainSequence & 1) || !last || mStandby) { 4328 track->mState = TrackBase::STOPPED; 4329 size_t audioHALFrames = 4330 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4331 size_t framesWritten = 4332 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4333 track->presentationComplete(framesWritten, audioHALFrames); 4334 track->reset(); 4335 tracksToRemove->add(track); 4336 } 4337 } else { 4338 // No buffers for this track. Give it a few chances to 4339 // fill a buffer, then remove it from active list. 4340 if (--(track->mRetryCount) <= 0) { 4341 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4342 track->name()); 4343 tracksToRemove->add(track); 4344 // indicate to client process that the track was disabled because of underrun; 4345 // it will then automatically call start() when data is available 4346 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4347 } else if (last){ 4348 mixerStatus = MIXER_TRACKS_ENABLED; 4349 } 4350 } 4351 } 4352 // compute volume for this track 4353 processVolume_l(track, last); 4354 } 4355 4356 // make sure the pause/flush/resume sequence is executed in the right order. 4357 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4358 // before flush and then resume HW. This can happen in case of pause/flush/resume 4359 // if resume is received before pause is executed. 4360 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4361 mOutput->stream->pause(mOutput->stream); 4362 } 4363 if (mFlushPending) { 4364 flushHw_l(); 4365 mFlushPending = false; 4366 } 4367 if (!mStandby && doHwResume) { 4368 mOutput->stream->resume(mOutput->stream); 4369 } 4370 4371 // remove all the tracks that need to be... 4372 removeTracks_l(*tracksToRemove); 4373 4374 return mixerStatus; 4375} 4376 4377// must be called with thread mutex locked 4378bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4379{ 4380 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4381 mWriteAckSequence, mDrainSequence); 4382 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4383 return true; 4384 } 4385 return false; 4386} 4387 4388// must be called with thread mutex locked 4389bool AudioFlinger::OffloadThread::shouldStandby_l() 4390{ 4391 bool trackPaused = false; 4392 4393 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4394 // after a timeout and we will enter standby then. 4395 if (mTracks.size() > 0) { 4396 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4397 } 4398 4399 return !mStandby && !trackPaused; 4400} 4401 4402 4403bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4404{ 4405 Mutex::Autolock _l(mLock); 4406 return waitingAsyncCallback_l(); 4407} 4408 4409void AudioFlinger::OffloadThread::flushHw_l() 4410{ 4411 mOutput->stream->flush(mOutput->stream); 4412 // Flush anything still waiting in the mixbuffer 4413 mCurrentWriteLength = 0; 4414 mBytesRemaining = 0; 4415 mPausedWriteLength = 0; 4416 mPausedBytesRemaining = 0; 4417 mHwPaused = false; 4418 4419 if (mUseAsyncWrite) { 4420 // discard any pending drain or write ack by incrementing sequence 4421 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4422 mDrainSequence = (mDrainSequence + 2) & ~1; 4423 ALOG_ASSERT(mCallbackThread != 0); 4424 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4425 mCallbackThread->setDraining(mDrainSequence); 4426 } 4427} 4428 4429void AudioFlinger::OffloadThread::onAddNewTrack_l() 4430{ 4431 sp<Track> previousTrack = mPreviousTrack.promote(); 4432 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4433 4434 if (previousTrack != 0 && latestTrack != 0 && 4435 (previousTrack->sessionId() != latestTrack->sessionId())) { 4436 mFlushPending = true; 4437 } 4438 PlaybackThread::onAddNewTrack_l(); 4439} 4440 4441// ---------------------------------------------------------------------------- 4442 4443AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4444 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4445 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4446 DUPLICATING), 4447 mWaitTimeMs(UINT_MAX) 4448{ 4449 addOutputTrack(mainThread); 4450} 4451 4452AudioFlinger::DuplicatingThread::~DuplicatingThread() 4453{ 4454 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4455 mOutputTracks[i]->destroy(); 4456 } 4457} 4458 4459void AudioFlinger::DuplicatingThread::threadLoop_mix() 4460{ 4461 // mix buffers... 4462 if (outputsReady(outputTracks)) { 4463 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4464 } else { 4465 memset(mSinkBuffer, 0, mSinkBufferSize); 4466 } 4467 sleepTime = 0; 4468 writeFrames = mNormalFrameCount; 4469 mCurrentWriteLength = mSinkBufferSize; 4470 standbyTime = systemTime() + standbyDelay; 4471} 4472 4473void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4474{ 4475 if (sleepTime == 0) { 4476 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4477 sleepTime = activeSleepTime; 4478 } else { 4479 sleepTime = idleSleepTime; 4480 } 4481 } else if (mBytesWritten != 0) { 4482 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4483 writeFrames = mNormalFrameCount; 4484 memset(mSinkBuffer, 0, mSinkBufferSize); 4485 } else { 4486 // flush remaining overflow buffers in output tracks 4487 writeFrames = 0; 4488 } 4489 sleepTime = 0; 4490 } 4491} 4492 4493ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4494{ 4495 for (size_t i = 0; i < outputTracks.size(); i++) { 4496 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4497 // for delivery downstream as needed. This in-place conversion is safe as 4498 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4499 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4500 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4501 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4502 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4503 } 4504 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4505 } 4506 mStandby = false; 4507 return (ssize_t)mSinkBufferSize; 4508} 4509 4510void AudioFlinger::DuplicatingThread::threadLoop_standby() 4511{ 4512 // DuplicatingThread implements standby by stopping all tracks 4513 for (size_t i = 0; i < outputTracks.size(); i++) { 4514 outputTracks[i]->stop(); 4515 } 4516} 4517 4518void AudioFlinger::DuplicatingThread::saveOutputTracks() 4519{ 4520 outputTracks = mOutputTracks; 4521} 4522 4523void AudioFlinger::DuplicatingThread::clearOutputTracks() 4524{ 4525 outputTracks.clear(); 4526} 4527 4528void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4529{ 4530 Mutex::Autolock _l(mLock); 4531 // FIXME explain this formula 4532 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4533 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4534 // due to current usage case and restrictions on the AudioBufferProvider. 4535 // Actual buffer conversion is done in threadLoop_write(). 4536 // 4537 // TODO: This may change in the future, depending on multichannel 4538 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4539 OutputTrack *outputTrack = new OutputTrack(thread, 4540 this, 4541 mSampleRate, 4542 AUDIO_FORMAT_PCM_16_BIT, 4543 mChannelMask, 4544 frameCount, 4545 IPCThreadState::self()->getCallingUid()); 4546 if (outputTrack->cblk() != NULL) { 4547 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4548 mOutputTracks.add(outputTrack); 4549 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4550 updateWaitTime_l(); 4551 } 4552} 4553 4554void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4555{ 4556 Mutex::Autolock _l(mLock); 4557 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4558 if (mOutputTracks[i]->thread() == thread) { 4559 mOutputTracks[i]->destroy(); 4560 mOutputTracks.removeAt(i); 4561 updateWaitTime_l(); 4562 return; 4563 } 4564 } 4565 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4566} 4567 4568// caller must hold mLock 4569void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4570{ 4571 mWaitTimeMs = UINT_MAX; 4572 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4573 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4574 if (strong != 0) { 4575 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4576 if (waitTimeMs < mWaitTimeMs) { 4577 mWaitTimeMs = waitTimeMs; 4578 } 4579 } 4580 } 4581} 4582 4583 4584bool AudioFlinger::DuplicatingThread::outputsReady( 4585 const SortedVector< sp<OutputTrack> > &outputTracks) 4586{ 4587 for (size_t i = 0; i < outputTracks.size(); i++) { 4588 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4589 if (thread == 0) { 4590 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4591 outputTracks[i].get()); 4592 return false; 4593 } 4594 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4595 // see note at standby() declaration 4596 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4597 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4598 thread.get()); 4599 return false; 4600 } 4601 } 4602 return true; 4603} 4604 4605uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4606{ 4607 return (mWaitTimeMs * 1000) / 2; 4608} 4609 4610void AudioFlinger::DuplicatingThread::cacheParameters_l() 4611{ 4612 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4613 updateWaitTime_l(); 4614 4615 MixerThread::cacheParameters_l(); 4616} 4617 4618// ---------------------------------------------------------------------------- 4619// Record 4620// ---------------------------------------------------------------------------- 4621 4622AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4623 AudioStreamIn *input, 4624 audio_io_handle_t id, 4625 audio_devices_t outDevice, 4626 audio_devices_t inDevice 4627#ifdef TEE_SINK 4628 , const sp<NBAIO_Sink>& teeSink 4629#endif 4630 ) : 4631 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4632 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4633 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4634 mRsmpInRear(0) 4635#ifdef TEE_SINK 4636 , mTeeSink(teeSink) 4637#endif 4638 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4639 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4640{ 4641 snprintf(mName, kNameLength, "AudioIn_%X", id); 4642 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4643 4644 readInputParameters_l(); 4645} 4646 4647 4648AudioFlinger::RecordThread::~RecordThread() 4649{ 4650 mAudioFlinger->unregisterWriter(mNBLogWriter); 4651 delete[] mRsmpInBuffer; 4652} 4653 4654void AudioFlinger::RecordThread::onFirstRef() 4655{ 4656 run(mName, PRIORITY_URGENT_AUDIO); 4657} 4658 4659bool AudioFlinger::RecordThread::threadLoop() 4660{ 4661 nsecs_t lastWarning = 0; 4662 4663 inputStandBy(); 4664 4665reacquire_wakelock: 4666 sp<RecordTrack> activeTrack; 4667 int activeTracksGen; 4668 { 4669 Mutex::Autolock _l(mLock); 4670 size_t size = mActiveTracks.size(); 4671 activeTracksGen = mActiveTracksGen; 4672 if (size > 0) { 4673 // FIXME an arbitrary choice 4674 activeTrack = mActiveTracks[0]; 4675 acquireWakeLock_l(activeTrack->uid()); 4676 if (size > 1) { 4677 SortedVector<int> tmp; 4678 for (size_t i = 0; i < size; i++) { 4679 tmp.add(mActiveTracks[i]->uid()); 4680 } 4681 updateWakeLockUids_l(tmp); 4682 } 4683 } else { 4684 acquireWakeLock_l(-1); 4685 } 4686 } 4687 4688 // used to request a deferred sleep, to be executed later while mutex is unlocked 4689 uint32_t sleepUs = 0; 4690 4691 // loop while there is work to do 4692 for (;;) { 4693 Vector< sp<EffectChain> > effectChains; 4694 4695 // sleep with mutex unlocked 4696 if (sleepUs > 0) { 4697 usleep(sleepUs); 4698 sleepUs = 0; 4699 } 4700 4701 // activeTracks accumulates a copy of a subset of mActiveTracks 4702 Vector< sp<RecordTrack> > activeTracks; 4703 4704 4705 { // scope for mLock 4706 Mutex::Autolock _l(mLock); 4707 4708 processConfigEvents_l(); 4709 4710 // check exitPending here because checkForNewParameters_l() and 4711 // checkForNewParameters_l() can temporarily release mLock 4712 if (exitPending()) { 4713 break; 4714 } 4715 4716 // if no active track(s), then standby and release wakelock 4717 size_t size = mActiveTracks.size(); 4718 if (size == 0) { 4719 standbyIfNotAlreadyInStandby(); 4720 // exitPending() can't become true here 4721 releaseWakeLock_l(); 4722 ALOGV("RecordThread: loop stopping"); 4723 // go to sleep 4724 mWaitWorkCV.wait(mLock); 4725 ALOGV("RecordThread: loop starting"); 4726 goto reacquire_wakelock; 4727 } 4728 4729 if (mActiveTracksGen != activeTracksGen) { 4730 activeTracksGen = mActiveTracksGen; 4731 SortedVector<int> tmp; 4732 for (size_t i = 0; i < size; i++) { 4733 tmp.add(mActiveTracks[i]->uid()); 4734 } 4735 updateWakeLockUids_l(tmp); 4736 } 4737 4738 bool doBroadcast = false; 4739 for (size_t i = 0; i < size; ) { 4740 4741 activeTrack = mActiveTracks[i]; 4742 if (activeTrack->isTerminated()) { 4743 removeTrack_l(activeTrack); 4744 mActiveTracks.remove(activeTrack); 4745 mActiveTracksGen++; 4746 size--; 4747 continue; 4748 } 4749 4750 TrackBase::track_state activeTrackState = activeTrack->mState; 4751 switch (activeTrackState) { 4752 4753 case TrackBase::PAUSING: 4754 mActiveTracks.remove(activeTrack); 4755 mActiveTracksGen++; 4756 doBroadcast = true; 4757 size--; 4758 continue; 4759 4760 case TrackBase::STARTING_1: 4761 sleepUs = 10000; 4762 i++; 4763 continue; 4764 4765 case TrackBase::STARTING_2: 4766 doBroadcast = true; 4767 mStandby = false; 4768 activeTrack->mState = TrackBase::ACTIVE; 4769 break; 4770 4771 case TrackBase::ACTIVE: 4772 break; 4773 4774 case TrackBase::IDLE: 4775 i++; 4776 continue; 4777 4778 default: 4779 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4780 } 4781 4782 activeTracks.add(activeTrack); 4783 i++; 4784 4785 } 4786 if (doBroadcast) { 4787 mStartStopCond.broadcast(); 4788 } 4789 4790 // sleep if there are no active tracks to process 4791 if (activeTracks.size() == 0) { 4792 if (sleepUs == 0) { 4793 sleepUs = kRecordThreadSleepUs; 4794 } 4795 continue; 4796 } 4797 sleepUs = 0; 4798 4799 lockEffectChains_l(effectChains); 4800 } 4801 4802 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4803 4804 size_t size = effectChains.size(); 4805 for (size_t i = 0; i < size; i++) { 4806 // thread mutex is not locked, but effect chain is locked 4807 effectChains[i]->process_l(); 4808 } 4809 4810 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4811 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4812 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4813 // If destination is non-contiguous, first read past the nominal end of buffer, then 4814 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4815 4816 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4817 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4818 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4819 if (bytesRead <= 0) { 4820 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4821 // Force input into standby so that it tries to recover at next read attempt 4822 inputStandBy(); 4823 sleepUs = kRecordThreadSleepUs; 4824 continue; 4825 } 4826 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4827 size_t framesRead = bytesRead / mFrameSize; 4828 ALOG_ASSERT(framesRead > 0); 4829 if (mTeeSink != 0) { 4830 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4831 } 4832 // If destination is non-contiguous, we now correct for reading past end of buffer. 4833 size_t part1 = mRsmpInFramesP2 - rear; 4834 if (framesRead > part1) { 4835 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4836 (framesRead - part1) * mFrameSize); 4837 } 4838 rear = mRsmpInRear += framesRead; 4839 4840 size = activeTracks.size(); 4841 // loop over each active track 4842 for (size_t i = 0; i < size; i++) { 4843 activeTrack = activeTracks[i]; 4844 4845 enum { 4846 OVERRUN_UNKNOWN, 4847 OVERRUN_TRUE, 4848 OVERRUN_FALSE 4849 } overrun = OVERRUN_UNKNOWN; 4850 4851 // loop over getNextBuffer to handle circular sink 4852 for (;;) { 4853 4854 activeTrack->mSink.frameCount = ~0; 4855 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4856 size_t framesOut = activeTrack->mSink.frameCount; 4857 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4858 4859 int32_t front = activeTrack->mRsmpInFront; 4860 ssize_t filled = rear - front; 4861 size_t framesIn; 4862 4863 if (filled < 0) { 4864 // should not happen, but treat like a massive overrun and re-sync 4865 framesIn = 0; 4866 activeTrack->mRsmpInFront = rear; 4867 overrun = OVERRUN_TRUE; 4868 } else if ((size_t) filled <= mRsmpInFrames) { 4869 framesIn = (size_t) filled; 4870 } else { 4871 // client is not keeping up with server, but give it latest data 4872 framesIn = mRsmpInFrames; 4873 activeTrack->mRsmpInFront = front = rear - framesIn; 4874 overrun = OVERRUN_TRUE; 4875 } 4876 4877 if (framesOut == 0 || framesIn == 0) { 4878 break; 4879 } 4880 4881 if (activeTrack->mResampler == NULL) { 4882 // no resampling 4883 if (framesIn > framesOut) { 4884 framesIn = framesOut; 4885 } else { 4886 framesOut = framesIn; 4887 } 4888 int8_t *dst = activeTrack->mSink.i8; 4889 while (framesIn > 0) { 4890 front &= mRsmpInFramesP2 - 1; 4891 size_t part1 = mRsmpInFramesP2 - front; 4892 if (part1 > framesIn) { 4893 part1 = framesIn; 4894 } 4895 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4896 if (mChannelCount == activeTrack->mChannelCount) { 4897 memcpy(dst, src, part1 * mFrameSize); 4898 } else if (mChannelCount == 1) { 4899 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4900 part1); 4901 } else { 4902 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4903 part1); 4904 } 4905 dst += part1 * activeTrack->mFrameSize; 4906 front += part1; 4907 framesIn -= part1; 4908 } 4909 activeTrack->mRsmpInFront += framesOut; 4910 4911 } else { 4912 // resampling 4913 // FIXME framesInNeeded should really be part of resampler API, and should 4914 // depend on the SRC ratio 4915 // to keep mRsmpInBuffer full so resampler always has sufficient input 4916 size_t framesInNeeded; 4917 // FIXME only re-calculate when it changes, and optimize for common ratios 4918 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4919 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4920 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4921 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4922 framesInNeeded, framesOut, inOverOut); 4923 // Although we theoretically have framesIn in circular buffer, some of those are 4924 // unreleased frames, and thus must be discounted for purpose of budgeting. 4925 size_t unreleased = activeTrack->mRsmpInUnrel; 4926 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4927 if (framesIn < framesInNeeded) { 4928 ALOGV("not enough to resample: have %u frames in but need %u in to " 4929 "produce %u out given in/out ratio of %.4g", 4930 framesIn, framesInNeeded, framesOut, inOverOut); 4931 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4932 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4933 if (newFramesOut == 0) { 4934 break; 4935 } 4936 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4937 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4938 framesInNeeded, newFramesOut, outOverIn); 4939 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4940 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4941 "given in/out ratio of %.4g", 4942 framesIn, framesInNeeded, newFramesOut, inOverOut); 4943 framesOut = newFramesOut; 4944 } else { 4945 ALOGV("success 1: have %u in and need %u in to produce %u out " 4946 "given in/out ratio of %.4g", 4947 framesIn, framesInNeeded, framesOut, inOverOut); 4948 } 4949 4950 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4951 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4952 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4953 delete[] activeTrack->mRsmpOutBuffer; 4954 // resampler always outputs stereo 4955 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4956 activeTrack->mRsmpOutFrameCount = framesOut; 4957 } 4958 4959 // resampler accumulates, but we only have one source track 4960 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4961 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4962 // FIXME how about having activeTrack implement this interface itself? 4963 activeTrack->mResamplerBufferProvider 4964 /*this*/ /* AudioBufferProvider* */); 4965 // ditherAndClamp() works as long as all buffers returned by 4966 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4967 if (activeTrack->mChannelCount == 1) { 4968 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 4969 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4970 framesOut); 4971 // the resampler always outputs stereo samples: 4972 // do post stereo to mono conversion 4973 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4974 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4975 } else { 4976 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4977 activeTrack->mRsmpOutBuffer, framesOut); 4978 } 4979 // now done with mRsmpOutBuffer 4980 4981 } 4982 4983 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4984 overrun = OVERRUN_FALSE; 4985 } 4986 4987 if (activeTrack->mFramesToDrop == 0) { 4988 if (framesOut > 0) { 4989 activeTrack->mSink.frameCount = framesOut; 4990 activeTrack->releaseBuffer(&activeTrack->mSink); 4991 } 4992 } else { 4993 // FIXME could do a partial drop of framesOut 4994 if (activeTrack->mFramesToDrop > 0) { 4995 activeTrack->mFramesToDrop -= framesOut; 4996 if (activeTrack->mFramesToDrop <= 0) { 4997 activeTrack->clearSyncStartEvent(); 4998 } 4999 } else { 5000 activeTrack->mFramesToDrop += framesOut; 5001 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5002 activeTrack->mSyncStartEvent->isCancelled()) { 5003 ALOGW("Synced record %s, session %d, trigger session %d", 5004 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5005 activeTrack->sessionId(), 5006 (activeTrack->mSyncStartEvent != 0) ? 5007 activeTrack->mSyncStartEvent->triggerSession() : 0); 5008 activeTrack->clearSyncStartEvent(); 5009 } 5010 } 5011 } 5012 5013 if (framesOut == 0) { 5014 break; 5015 } 5016 } 5017 5018 switch (overrun) { 5019 case OVERRUN_TRUE: 5020 // client isn't retrieving buffers fast enough 5021 if (!activeTrack->setOverflow()) { 5022 nsecs_t now = systemTime(); 5023 // FIXME should lastWarning per track? 5024 if ((now - lastWarning) > kWarningThrottleNs) { 5025 ALOGW("RecordThread: buffer overflow"); 5026 lastWarning = now; 5027 } 5028 } 5029 break; 5030 case OVERRUN_FALSE: 5031 activeTrack->clearOverflow(); 5032 break; 5033 case OVERRUN_UNKNOWN: 5034 break; 5035 } 5036 5037 } 5038 5039 // enable changes in effect chain 5040 unlockEffectChains(effectChains); 5041 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5042 } 5043 5044 standbyIfNotAlreadyInStandby(); 5045 5046 { 5047 Mutex::Autolock _l(mLock); 5048 for (size_t i = 0; i < mTracks.size(); i++) { 5049 sp<RecordTrack> track = mTracks[i]; 5050 track->invalidate(); 5051 } 5052 mActiveTracks.clear(); 5053 mActiveTracksGen++; 5054 mStartStopCond.broadcast(); 5055 } 5056 5057 releaseWakeLock(); 5058 5059 ALOGV("RecordThread %p exiting", this); 5060 return false; 5061} 5062 5063void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5064{ 5065 if (!mStandby) { 5066 inputStandBy(); 5067 mStandby = true; 5068 } 5069} 5070 5071void AudioFlinger::RecordThread::inputStandBy() 5072{ 5073 mInput->stream->common.standby(&mInput->stream->common); 5074} 5075 5076// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5077sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5078 const sp<AudioFlinger::Client>& client, 5079 uint32_t sampleRate, 5080 audio_format_t format, 5081 audio_channel_mask_t channelMask, 5082 size_t *pFrameCount, 5083 int sessionId, 5084 int uid, 5085 IAudioFlinger::track_flags_t *flags, 5086 pid_t tid, 5087 status_t *status) 5088{ 5089 size_t frameCount = *pFrameCount; 5090 sp<RecordTrack> track; 5091 status_t lStatus; 5092 5093 // client expresses a preference for FAST, but we get the final say 5094 if (*flags & IAudioFlinger::TRACK_FAST) { 5095 if ( 5096 // use case: callback handler and frame count is default or at least as large as HAL 5097 ( 5098 (tid != -1) && 5099 ((frameCount == 0) || 5100 // FIXME not necessarily true, should be native frame count for native SR! 5101 (frameCount >= mFrameCount)) 5102 ) && 5103 // PCM data 5104 audio_is_linear_pcm(format) && 5105 // mono or stereo 5106 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5107 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5108 // hardware sample rate 5109 // FIXME actually the native hardware sample rate 5110 (sampleRate == mSampleRate) && 5111 // record thread has an associated fast capture 5112 hasFastCapture() 5113 // fast capture does not require slots 5114 ) { 5115 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5116 if (frameCount == 0) { 5117 // FIXME wrong mFrameCount 5118 frameCount = mFrameCount * kFastTrackMultiplier; 5119 } 5120 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5121 frameCount, mFrameCount); 5122 } else { 5123 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5124 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5125 "hasFastCapture=%d tid=%d", 5126 frameCount, mFrameCount, format, 5127 audio_is_linear_pcm(format), 5128 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5129 *flags &= ~IAudioFlinger::TRACK_FAST; 5130 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5131 // For compatibility with AudioRecord calculation, buffer depth is forced 5132 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5133 // This is probably too conservative, but legacy application code may depend on it. 5134 // If you change this calculation, also review the start threshold which is related. 5135 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5136 size_t mNormalFrameCount = 2048; // FIXME 5137 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5138 if (minBufCount < 2) { 5139 minBufCount = 2; 5140 } 5141 size_t minFrameCount = mNormalFrameCount * minBufCount; 5142 if (frameCount < minFrameCount) { 5143 frameCount = minFrameCount; 5144 } 5145 } 5146 } 5147 *pFrameCount = frameCount; 5148 5149 lStatus = initCheck(); 5150 if (lStatus != NO_ERROR) { 5151 ALOGE("createRecordTrack_l() audio driver not initialized"); 5152 goto Exit; 5153 } 5154 5155 { // scope for mLock 5156 Mutex::Autolock _l(mLock); 5157 5158 track = new RecordTrack(this, client, sampleRate, 5159 format, channelMask, frameCount, sessionId, uid, 5160 *flags); 5161 5162 lStatus = track->initCheck(); 5163 if (lStatus != NO_ERROR) { 5164 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5165 // track must be cleared from the caller as the caller has the AF lock 5166 goto Exit; 5167 } 5168 mTracks.add(track); 5169 5170 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5171 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5172 mAudioFlinger->btNrecIsOff(); 5173 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5174 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5175 5176 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5177 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5178 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5179 // so ask activity manager to do this on our behalf 5180 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5181 } 5182 } 5183 5184 lStatus = NO_ERROR; 5185 5186Exit: 5187 *status = lStatus; 5188 return track; 5189} 5190 5191status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5192 AudioSystem::sync_event_t event, 5193 int triggerSession) 5194{ 5195 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5196 sp<ThreadBase> strongMe = this; 5197 status_t status = NO_ERROR; 5198 5199 if (event == AudioSystem::SYNC_EVENT_NONE) { 5200 recordTrack->clearSyncStartEvent(); 5201 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5202 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5203 triggerSession, 5204 recordTrack->sessionId(), 5205 syncStartEventCallback, 5206 recordTrack); 5207 // Sync event can be cancelled by the trigger session if the track is not in a 5208 // compatible state in which case we start record immediately 5209 if (recordTrack->mSyncStartEvent->isCancelled()) { 5210 recordTrack->clearSyncStartEvent(); 5211 } else { 5212 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5213 recordTrack->mFramesToDrop = - 5214 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5215 } 5216 } 5217 5218 { 5219 // This section is a rendezvous between binder thread executing start() and RecordThread 5220 AutoMutex lock(mLock); 5221 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5222 if (recordTrack->mState == TrackBase::PAUSING) { 5223 ALOGV("active record track PAUSING -> ACTIVE"); 5224 recordTrack->mState = TrackBase::ACTIVE; 5225 } else { 5226 ALOGV("active record track state %d", recordTrack->mState); 5227 } 5228 return status; 5229 } 5230 5231 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5232 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5233 // or using a separate command thread 5234 recordTrack->mState = TrackBase::STARTING_1; 5235 mActiveTracks.add(recordTrack); 5236 mActiveTracksGen++; 5237 mLock.unlock(); 5238 status_t status = AudioSystem::startInput(mId); 5239 mLock.lock(); 5240 // FIXME should verify that recordTrack is still in mActiveTracks 5241 if (status != NO_ERROR) { 5242 mActiveTracks.remove(recordTrack); 5243 mActiveTracksGen++; 5244 recordTrack->clearSyncStartEvent(); 5245 return status; 5246 } 5247 // Catch up with current buffer indices if thread is already running. 5248 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5249 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5250 // see previously buffered data before it called start(), but with greater risk of overrun. 5251 5252 recordTrack->mRsmpInFront = mRsmpInRear; 5253 recordTrack->mRsmpInUnrel = 0; 5254 // FIXME why reset? 5255 if (recordTrack->mResampler != NULL) { 5256 recordTrack->mResampler->reset(); 5257 } 5258 recordTrack->mState = TrackBase::STARTING_2; 5259 // signal thread to start 5260 mWaitWorkCV.broadcast(); 5261 if (mActiveTracks.indexOf(recordTrack) < 0) { 5262 ALOGV("Record failed to start"); 5263 status = BAD_VALUE; 5264 goto startError; 5265 } 5266 return status; 5267 } 5268 5269startError: 5270 AudioSystem::stopInput(mId); 5271 recordTrack->clearSyncStartEvent(); 5272 // FIXME I wonder why we do not reset the state here? 5273 return status; 5274} 5275 5276void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5277{ 5278 sp<SyncEvent> strongEvent = event.promote(); 5279 5280 if (strongEvent != 0) { 5281 sp<RefBase> ptr = strongEvent->cookie().promote(); 5282 if (ptr != 0) { 5283 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5284 recordTrack->handleSyncStartEvent(strongEvent); 5285 } 5286 } 5287} 5288 5289bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5290 ALOGV("RecordThread::stop"); 5291 AutoMutex _l(mLock); 5292 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5293 return false; 5294 } 5295 // note that threadLoop may still be processing the track at this point [without lock] 5296 recordTrack->mState = TrackBase::PAUSING; 5297 // do not wait for mStartStopCond if exiting 5298 if (exitPending()) { 5299 return true; 5300 } 5301 // FIXME incorrect usage of wait: no explicit predicate or loop 5302 mStartStopCond.wait(mLock); 5303 // if we have been restarted, recordTrack is in mActiveTracks here 5304 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5305 ALOGV("Record stopped OK"); 5306 return true; 5307 } 5308 return false; 5309} 5310 5311bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5312{ 5313 return false; 5314} 5315 5316status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5317{ 5318#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5319 if (!isValidSyncEvent(event)) { 5320 return BAD_VALUE; 5321 } 5322 5323 int eventSession = event->triggerSession(); 5324 status_t ret = NAME_NOT_FOUND; 5325 5326 Mutex::Autolock _l(mLock); 5327 5328 for (size_t i = 0; i < mTracks.size(); i++) { 5329 sp<RecordTrack> track = mTracks[i]; 5330 if (eventSession == track->sessionId()) { 5331 (void) track->setSyncEvent(event); 5332 ret = NO_ERROR; 5333 } 5334 } 5335 return ret; 5336#else 5337 return BAD_VALUE; 5338#endif 5339} 5340 5341// destroyTrack_l() must be called with ThreadBase::mLock held 5342void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5343{ 5344 track->terminate(); 5345 track->mState = TrackBase::STOPPED; 5346 // active tracks are removed by threadLoop() 5347 if (mActiveTracks.indexOf(track) < 0) { 5348 removeTrack_l(track); 5349 } 5350} 5351 5352void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5353{ 5354 mTracks.remove(track); 5355 // need anything related to effects here? 5356} 5357 5358void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5359{ 5360 dumpInternals(fd, args); 5361 dumpTracks(fd, args); 5362 dumpEffectChains(fd, args); 5363} 5364 5365void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5366{ 5367 dprintf(fd, "\nInput thread %p:\n", this); 5368 5369 if (mActiveTracks.size() > 0) { 5370 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5371 } else { 5372 dprintf(fd, " No active record clients\n"); 5373 } 5374 5375 dumpBase(fd, args); 5376} 5377 5378void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5379{ 5380 const size_t SIZE = 256; 5381 char buffer[SIZE]; 5382 String8 result; 5383 5384 size_t numtracks = mTracks.size(); 5385 size_t numactive = mActiveTracks.size(); 5386 size_t numactiveseen = 0; 5387 dprintf(fd, " %d Tracks", numtracks); 5388 if (numtracks) { 5389 dprintf(fd, " of which %d are active\n", numactive); 5390 RecordTrack::appendDumpHeader(result); 5391 for (size_t i = 0; i < numtracks ; ++i) { 5392 sp<RecordTrack> track = mTracks[i]; 5393 if (track != 0) { 5394 bool active = mActiveTracks.indexOf(track) >= 0; 5395 if (active) { 5396 numactiveseen++; 5397 } 5398 track->dump(buffer, SIZE, active); 5399 result.append(buffer); 5400 } 5401 } 5402 } else { 5403 dprintf(fd, "\n"); 5404 } 5405 5406 if (numactiveseen != numactive) { 5407 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5408 " not in the track list\n"); 5409 result.append(buffer); 5410 RecordTrack::appendDumpHeader(result); 5411 for (size_t i = 0; i < numactive; ++i) { 5412 sp<RecordTrack> track = mActiveTracks[i]; 5413 if (mTracks.indexOf(track) < 0) { 5414 track->dump(buffer, SIZE, true); 5415 result.append(buffer); 5416 } 5417 } 5418 5419 } 5420 write(fd, result.string(), result.size()); 5421} 5422 5423// AudioBufferProvider interface 5424status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5425 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5426{ 5427 RecordTrack *activeTrack = mRecordTrack; 5428 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5429 if (threadBase == 0) { 5430 buffer->frameCount = 0; 5431 buffer->raw = NULL; 5432 return NOT_ENOUGH_DATA; 5433 } 5434 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5435 int32_t rear = recordThread->mRsmpInRear; 5436 int32_t front = activeTrack->mRsmpInFront; 5437 ssize_t filled = rear - front; 5438 // FIXME should not be P2 (don't want to increase latency) 5439 // FIXME if client not keeping up, discard 5440 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5441 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5442 front &= recordThread->mRsmpInFramesP2 - 1; 5443 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5444 if (part1 > (size_t) filled) { 5445 part1 = filled; 5446 } 5447 size_t ask = buffer->frameCount; 5448 ALOG_ASSERT(ask > 0); 5449 if (part1 > ask) { 5450 part1 = ask; 5451 } 5452 if (part1 == 0) { 5453 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5454 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5455 buffer->raw = NULL; 5456 buffer->frameCount = 0; 5457 activeTrack->mRsmpInUnrel = 0; 5458 return NOT_ENOUGH_DATA; 5459 } 5460 5461 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5462 buffer->frameCount = part1; 5463 activeTrack->mRsmpInUnrel = part1; 5464 return NO_ERROR; 5465} 5466 5467// AudioBufferProvider interface 5468void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5469 AudioBufferProvider::Buffer* buffer) 5470{ 5471 RecordTrack *activeTrack = mRecordTrack; 5472 size_t stepCount = buffer->frameCount; 5473 if (stepCount == 0) { 5474 return; 5475 } 5476 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5477 activeTrack->mRsmpInUnrel -= stepCount; 5478 activeTrack->mRsmpInFront += stepCount; 5479 buffer->raw = NULL; 5480 buffer->frameCount = 0; 5481} 5482 5483bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5484 status_t& status) 5485{ 5486 bool reconfig = false; 5487 5488 status = NO_ERROR; 5489 5490 audio_format_t reqFormat = mFormat; 5491 uint32_t samplingRate = mSampleRate; 5492 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5493 5494 AudioParameter param = AudioParameter(keyValuePair); 5495 int value; 5496 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5497 // channel count change can be requested. Do we mandate the first client defines the 5498 // HAL sampling rate and channel count or do we allow changes on the fly? 5499 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5500 samplingRate = value; 5501 reconfig = true; 5502 } 5503 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5504 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5505 status = BAD_VALUE; 5506 } else { 5507 reqFormat = (audio_format_t) value; 5508 reconfig = true; 5509 } 5510 } 5511 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5512 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5513 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5514 status = BAD_VALUE; 5515 } else { 5516 channelMask = mask; 5517 reconfig = true; 5518 } 5519 } 5520 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5521 // do not accept frame count changes if tracks are open as the track buffer 5522 // size depends on frame count and correct behavior would not be guaranteed 5523 // if frame count is changed after track creation 5524 if (mActiveTracks.size() > 0) { 5525 status = INVALID_OPERATION; 5526 } else { 5527 reconfig = true; 5528 } 5529 } 5530 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5531 // forward device change to effects that have requested to be 5532 // aware of attached audio device. 5533 for (size_t i = 0; i < mEffectChains.size(); i++) { 5534 mEffectChains[i]->setDevice_l(value); 5535 } 5536 5537 // store input device and output device but do not forward output device to audio HAL. 5538 // Note that status is ignored by the caller for output device 5539 // (see AudioFlinger::setParameters() 5540 if (audio_is_output_devices(value)) { 5541 mOutDevice = value; 5542 status = BAD_VALUE; 5543 } else { 5544 mInDevice = value; 5545 // disable AEC and NS if the device is a BT SCO headset supporting those 5546 // pre processings 5547 if (mTracks.size() > 0) { 5548 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5549 mAudioFlinger->btNrecIsOff(); 5550 for (size_t i = 0; i < mTracks.size(); i++) { 5551 sp<RecordTrack> track = mTracks[i]; 5552 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5553 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5554 } 5555 } 5556 } 5557 } 5558 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5559 mAudioSource != (audio_source_t)value) { 5560 // forward device change to effects that have requested to be 5561 // aware of attached audio device. 5562 for (size_t i = 0; i < mEffectChains.size(); i++) { 5563 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5564 } 5565 mAudioSource = (audio_source_t)value; 5566 } 5567 5568 if (status == NO_ERROR) { 5569 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5570 keyValuePair.string()); 5571 if (status == INVALID_OPERATION) { 5572 inputStandBy(); 5573 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5574 keyValuePair.string()); 5575 } 5576 if (reconfig) { 5577 if (status == BAD_VALUE && 5578 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5579 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5580 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5581 <= (2 * samplingRate)) && 5582 audio_channel_count_from_in_mask( 5583 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5584 (channelMask == AUDIO_CHANNEL_IN_MONO || 5585 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5586 status = NO_ERROR; 5587 } 5588 if (status == NO_ERROR) { 5589 readInputParameters_l(); 5590 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5591 } 5592 } 5593 } 5594 5595 return reconfig; 5596} 5597 5598String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5599{ 5600 Mutex::Autolock _l(mLock); 5601 if (initCheck() != NO_ERROR) { 5602 return String8(); 5603 } 5604 5605 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5606 const String8 out_s8(s); 5607 free(s); 5608 return out_s8; 5609} 5610 5611void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5612 AudioSystem::OutputDescriptor desc; 5613 const void *param2 = NULL; 5614 5615 switch (event) { 5616 case AudioSystem::INPUT_OPENED: 5617 case AudioSystem::INPUT_CONFIG_CHANGED: 5618 desc.channelMask = mChannelMask; 5619 desc.samplingRate = mSampleRate; 5620 desc.format = mFormat; 5621 desc.frameCount = mFrameCount; 5622 desc.latency = 0; 5623 param2 = &desc; 5624 break; 5625 5626 case AudioSystem::INPUT_CLOSED: 5627 default: 5628 break; 5629 } 5630 mAudioFlinger->audioConfigChanged(event, mId, param2); 5631} 5632 5633void AudioFlinger::RecordThread::readInputParameters_l() 5634{ 5635 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5636 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5637 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5638 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5639 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5640 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5641 } 5642 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5643 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5644 mFrameCount = mBufferSize / mFrameSize; 5645 // This is the formula for calculating the temporary buffer size. 5646 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5647 // 1 full output buffer, regardless of the alignment of the available input. 5648 // The value is somewhat arbitrary, and could probably be even larger. 5649 // A larger value should allow more old data to be read after a track calls start(), 5650 // without increasing latency. 5651 mRsmpInFrames = mFrameCount * 7; 5652 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5653 delete[] mRsmpInBuffer; 5654 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5655 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5656 5657 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5658 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5659} 5660 5661uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5662{ 5663 Mutex::Autolock _l(mLock); 5664 if (initCheck() != NO_ERROR) { 5665 return 0; 5666 } 5667 5668 return mInput->stream->get_input_frames_lost(mInput->stream); 5669} 5670 5671uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5672{ 5673 Mutex::Autolock _l(mLock); 5674 uint32_t result = 0; 5675 if (getEffectChain_l(sessionId) != 0) { 5676 result = EFFECT_SESSION; 5677 } 5678 5679 for (size_t i = 0; i < mTracks.size(); ++i) { 5680 if (sessionId == mTracks[i]->sessionId()) { 5681 result |= TRACK_SESSION; 5682 break; 5683 } 5684 } 5685 5686 return result; 5687} 5688 5689KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5690{ 5691 KeyedVector<int, bool> ids; 5692 Mutex::Autolock _l(mLock); 5693 for (size_t j = 0; j < mTracks.size(); ++j) { 5694 sp<RecordThread::RecordTrack> track = mTracks[j]; 5695 int sessionId = track->sessionId(); 5696 if (ids.indexOfKey(sessionId) < 0) { 5697 ids.add(sessionId, true); 5698 } 5699 } 5700 return ids; 5701} 5702 5703AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5704{ 5705 Mutex::Autolock _l(mLock); 5706 AudioStreamIn *input = mInput; 5707 mInput = NULL; 5708 return input; 5709} 5710 5711// this method must always be called either with ThreadBase mLock held or inside the thread loop 5712audio_stream_t* AudioFlinger::RecordThread::stream() const 5713{ 5714 if (mInput == NULL) { 5715 return NULL; 5716 } 5717 return &mInput->stream->common; 5718} 5719 5720status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5721{ 5722 // only one chain per input thread 5723 if (mEffectChains.size() != 0) { 5724 return INVALID_OPERATION; 5725 } 5726 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5727 5728 chain->setInBuffer(NULL); 5729 chain->setOutBuffer(NULL); 5730 5731 checkSuspendOnAddEffectChain_l(chain); 5732 5733 mEffectChains.add(chain); 5734 5735 return NO_ERROR; 5736} 5737 5738size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5739{ 5740 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5741 ALOGW_IF(mEffectChains.size() != 1, 5742 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5743 chain.get(), mEffectChains.size(), this); 5744 if (mEffectChains.size() == 1) { 5745 mEffectChains.removeAt(0); 5746 } 5747 return 0; 5748} 5749 5750}; // namespace android 5751