Threads.cpp revision 0349009fd19f89f8414c428f6b71b369f7546085
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. 145 146// This is the default value, if not specified by property. 147static const int kFastTrackMultiplier = 2; 148 149// The minimum and maximum allowed values 150static const int kFastTrackMultiplierMin = 1; 151static const int kFastTrackMultiplierMax = 2; 152 153// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 154static int sFastTrackMultiplier = kFastTrackMultiplier; 155 156// See Thread::readOnlyHeap(). 157// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 158// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 159// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 160static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 161 162// ---------------------------------------------------------------------------- 163 164static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 165 166static void sFastTrackMultiplierInit() 167{ 168 char value[PROPERTY_VALUE_MAX]; 169 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 170 char *endptr; 171 unsigned long ul = strtoul(value, &endptr, 0); 172 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 173 sFastTrackMultiplier = (int) ul; 174 } 175 } 176} 177 178// ---------------------------------------------------------------------------- 179 180#ifdef ADD_BATTERY_DATA 181// To collect the amplifier usage 182static void addBatteryData(uint32_t params) { 183 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 184 if (service == NULL) { 185 // it already logged 186 return; 187 } 188 189 service->addBatteryData(params); 190} 191#endif 192 193 194// ---------------------------------------------------------------------------- 195// CPU Stats 196// ---------------------------------------------------------------------------- 197 198class CpuStats { 199public: 200 CpuStats(); 201 void sample(const String8 &title); 202#ifdef DEBUG_CPU_USAGE 203private: 204 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 205 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 206 207 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 208 209 int mCpuNum; // thread's current CPU number 210 int mCpukHz; // frequency of thread's current CPU in kHz 211#endif 212}; 213 214CpuStats::CpuStats() 215#ifdef DEBUG_CPU_USAGE 216 : mCpuNum(-1), mCpukHz(-1) 217#endif 218{ 219} 220 221void CpuStats::sample(const String8 &title 222#ifndef DEBUG_CPU_USAGE 223 __unused 224#endif 225 ) { 226#ifdef DEBUG_CPU_USAGE 227 // get current thread's delta CPU time in wall clock ns 228 double wcNs; 229 bool valid = mCpuUsage.sampleAndEnable(wcNs); 230 231 // record sample for wall clock statistics 232 if (valid) { 233 mWcStats.sample(wcNs); 234 } 235 236 // get the current CPU number 237 int cpuNum = sched_getcpu(); 238 239 // get the current CPU frequency in kHz 240 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 241 242 // check if either CPU number or frequency changed 243 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 244 mCpuNum = cpuNum; 245 mCpukHz = cpukHz; 246 // ignore sample for purposes of cycles 247 valid = false; 248 } 249 250 // if no change in CPU number or frequency, then record sample for cycle statistics 251 if (valid && mCpukHz > 0) { 252 double cycles = wcNs * cpukHz * 0.000001; 253 mHzStats.sample(cycles); 254 } 255 256 unsigned n = mWcStats.n(); 257 // mCpuUsage.elapsed() is expensive, so don't call it every loop 258 if ((n & 127) == 1) { 259 long long elapsed = mCpuUsage.elapsed(); 260 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 261 double perLoop = elapsed / (double) n; 262 double perLoop100 = perLoop * 0.01; 263 double perLoop1k = perLoop * 0.001; 264 double mean = mWcStats.mean(); 265 double stddev = mWcStats.stddev(); 266 double minimum = mWcStats.minimum(); 267 double maximum = mWcStats.maximum(); 268 double meanCycles = mHzStats.mean(); 269 double stddevCycles = mHzStats.stddev(); 270 double minCycles = mHzStats.minimum(); 271 double maxCycles = mHzStats.maximum(); 272 mCpuUsage.resetElapsed(); 273 mWcStats.reset(); 274 mHzStats.reset(); 275 ALOGD("CPU usage for %s over past %.1f secs\n" 276 " (%u mixer loops at %.1f mean ms per loop):\n" 277 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 278 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 279 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 280 title.string(), 281 elapsed * .000000001, n, perLoop * .000001, 282 mean * .001, 283 stddev * .001, 284 minimum * .001, 285 maximum * .001, 286 mean / perLoop100, 287 stddev / perLoop100, 288 minimum / perLoop100, 289 maximum / perLoop100, 290 meanCycles / perLoop1k, 291 stddevCycles / perLoop1k, 292 minCycles / perLoop1k, 293 maxCycles / perLoop1k); 294 295 } 296 } 297#endif 298}; 299 300// ---------------------------------------------------------------------------- 301// ThreadBase 302// ---------------------------------------------------------------------------- 303 304AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 305 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 306 : Thread(false /*canCallJava*/), 307 mType(type), 308 mAudioFlinger(audioFlinger), 309 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 310 // are set by PlaybackThread::readOutputParameters_l() or 311 // RecordThread::readInputParameters_l() 312 //FIXME: mStandby should be true here. Is this some kind of hack? 313 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 314 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 315 // mName will be set by concrete (non-virtual) subclass 316 mDeathRecipient(new PMDeathRecipient(this)) 317{ 318} 319 320AudioFlinger::ThreadBase::~ThreadBase() 321{ 322 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 323 mConfigEvents.clear(); 324 325 // do not lock the mutex in destructor 326 releaseWakeLock_l(); 327 if (mPowerManager != 0) { 328 sp<IBinder> binder = mPowerManager->asBinder(); 329 binder->unlinkToDeath(mDeathRecipient); 330 } 331} 332 333status_t AudioFlinger::ThreadBase::readyToRun() 334{ 335 status_t status = initCheck(); 336 if (status == NO_ERROR) { 337 ALOGI("AudioFlinger's thread %p ready to run", this); 338 } else { 339 ALOGE("No working audio driver found."); 340 } 341 return status; 342} 343 344void AudioFlinger::ThreadBase::exit() 345{ 346 ALOGV("ThreadBase::exit"); 347 // do any cleanup required for exit to succeed 348 preExit(); 349 { 350 // This lock prevents the following race in thread (uniprocessor for illustration): 351 // if (!exitPending()) { 352 // // context switch from here to exit() 353 // // exit() calls requestExit(), what exitPending() observes 354 // // exit() calls signal(), which is dropped since no waiters 355 // // context switch back from exit() to here 356 // mWaitWorkCV.wait(...); 357 // // now thread is hung 358 // } 359 AutoMutex lock(mLock); 360 requestExit(); 361 mWaitWorkCV.broadcast(); 362 } 363 // When Thread::requestExitAndWait is made virtual and this method is renamed to 364 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 365 requestExitAndWait(); 366} 367 368status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 369{ 370 status_t status; 371 372 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 373 Mutex::Autolock _l(mLock); 374 375 return sendSetParameterConfigEvent_l(keyValuePairs); 376} 377 378// sendConfigEvent_l() must be called with ThreadBase::mLock held 379// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 380status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 381{ 382 status_t status = NO_ERROR; 383 384 mConfigEvents.add(event); 385 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 386 mWaitWorkCV.signal(); 387 mLock.unlock(); 388 { 389 Mutex::Autolock _l(event->mLock); 390 while (event->mWaitStatus) { 391 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 392 event->mStatus = TIMED_OUT; 393 event->mWaitStatus = false; 394 } 395 } 396 status = event->mStatus; 397 } 398 mLock.lock(); 399 return status; 400} 401 402void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 403{ 404 Mutex::Autolock _l(mLock); 405 sendIoConfigEvent_l(event, param); 406} 407 408// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 409void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 410{ 411 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 412 sendConfigEvent_l(configEvent); 413} 414 415// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 416void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 417{ 418 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 419 sendConfigEvent_l(configEvent); 420} 421 422// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 423status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 424{ 425 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 426 return sendConfigEvent_l(configEvent); 427} 428 429// post condition: mConfigEvents.isEmpty() 430void AudioFlinger::ThreadBase::processConfigEvents_l() 431{ 432 bool configChanged = false; 433 434 while (!mConfigEvents.isEmpty()) { 435 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 436 sp<ConfigEvent> event = mConfigEvents[0]; 437 mConfigEvents.removeAt(0); 438 switch (event->mType) { 439 case CFG_EVENT_PRIO: { 440 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 441 // FIXME Need to understand why this has to be done asynchronously 442 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 443 true /*asynchronous*/); 444 if (err != 0) { 445 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 446 data->mPrio, data->mPid, data->mTid, err); 447 } 448 } break; 449 case CFG_EVENT_IO: { 450 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 451 audioConfigChanged(data->mEvent, data->mParam); 452 } break; 453 case CFG_EVENT_SET_PARAMETER: { 454 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 455 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 456 configChanged = true; 457 } 458 } break; 459 default: 460 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 461 break; 462 } 463 { 464 Mutex::Autolock _l(event->mLock); 465 if (event->mWaitStatus) { 466 event->mWaitStatus = false; 467 event->mCond.signal(); 468 } 469 } 470 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 471 } 472 473 if (configChanged) { 474 cacheParameters_l(); 475 } 476} 477 478String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 479 String8 s; 480 if (output) { 481 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 482 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 483 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 484 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 485 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 486 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 487 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 488 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 489 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 490 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 491 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 492 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 493 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 494 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 495 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 496 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 497 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 498 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 499 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 500 } else { 501 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 502 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 503 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 504 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 505 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 506 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 507 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 508 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 509 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 510 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 511 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 512 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 513 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 514 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 515 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 516 } 517 int len = s.length(); 518 if (s.length() > 2) { 519 char *str = s.lockBuffer(len); 520 s.unlockBuffer(len - 2); 521 } 522 return s; 523} 524 525void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 526{ 527 const size_t SIZE = 256; 528 char buffer[SIZE]; 529 String8 result; 530 531 bool locked = AudioFlinger::dumpTryLock(mLock); 532 if (!locked) { 533 dprintf(fd, "thread %p maybe dead locked\n", this); 534 } 535 536 dprintf(fd, " I/O handle: %d\n", mId); 537 dprintf(fd, " TID: %d\n", getTid()); 538 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 539 dprintf(fd, " Sample rate: %u\n", mSampleRate); 540 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 541 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 542 dprintf(fd, " Channel Count: %u\n", mChannelCount); 543 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 544 channelMaskToString(mChannelMask, mType != RECORD).string()); 545 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 546 dprintf(fd, " Frame size: %zu\n", mFrameSize); 547 dprintf(fd, " Pending config events:"); 548 size_t numConfig = mConfigEvents.size(); 549 if (numConfig) { 550 for (size_t i = 0; i < numConfig; i++) { 551 mConfigEvents[i]->dump(buffer, SIZE); 552 dprintf(fd, "\n %s", buffer); 553 } 554 dprintf(fd, "\n"); 555 } else { 556 dprintf(fd, " none\n"); 557 } 558 559 if (locked) { 560 mLock.unlock(); 561 } 562} 563 564void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 565{ 566 const size_t SIZE = 256; 567 char buffer[SIZE]; 568 String8 result; 569 570 size_t numEffectChains = mEffectChains.size(); 571 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 572 write(fd, buffer, strlen(buffer)); 573 574 for (size_t i = 0; i < numEffectChains; ++i) { 575 sp<EffectChain> chain = mEffectChains[i]; 576 if (chain != 0) { 577 chain->dump(fd, args); 578 } 579 } 580} 581 582void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 583{ 584 Mutex::Autolock _l(mLock); 585 acquireWakeLock_l(uid); 586} 587 588String16 AudioFlinger::ThreadBase::getWakeLockTag() 589{ 590 switch (mType) { 591 case MIXER: 592 return String16("AudioMix"); 593 case DIRECT: 594 return String16("AudioDirectOut"); 595 case DUPLICATING: 596 return String16("AudioDup"); 597 case RECORD: 598 return String16("AudioIn"); 599 case OFFLOAD: 600 return String16("AudioOffload"); 601 default: 602 ALOG_ASSERT(false); 603 return String16("AudioUnknown"); 604 } 605} 606 607void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 608{ 609 getPowerManager_l(); 610 if (mPowerManager != 0) { 611 sp<IBinder> binder = new BBinder(); 612 status_t status; 613 if (uid >= 0) { 614 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 615 binder, 616 getWakeLockTag(), 617 String16("media"), 618 uid); 619 } else { 620 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 621 binder, 622 getWakeLockTag(), 623 String16("media")); 624 } 625 if (status == NO_ERROR) { 626 mWakeLockToken = binder; 627 } 628 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 629 } 630} 631 632void AudioFlinger::ThreadBase::releaseWakeLock() 633{ 634 Mutex::Autolock _l(mLock); 635 releaseWakeLock_l(); 636} 637 638void AudioFlinger::ThreadBase::releaseWakeLock_l() 639{ 640 if (mWakeLockToken != 0) { 641 ALOGV("releaseWakeLock_l() %s", mName); 642 if (mPowerManager != 0) { 643 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 644 } 645 mWakeLockToken.clear(); 646 } 647} 648 649void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 650 Mutex::Autolock _l(mLock); 651 updateWakeLockUids_l(uids); 652} 653 654void AudioFlinger::ThreadBase::getPowerManager_l() { 655 656 if (mPowerManager == 0) { 657 // use checkService() to avoid blocking if power service is not up yet 658 sp<IBinder> binder = 659 defaultServiceManager()->checkService(String16("power")); 660 if (binder == 0) { 661 ALOGW("Thread %s cannot connect to the power manager service", mName); 662 } else { 663 mPowerManager = interface_cast<IPowerManager>(binder); 664 binder->linkToDeath(mDeathRecipient); 665 } 666 } 667} 668 669void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 670 671 getPowerManager_l(); 672 if (mWakeLockToken == NULL) { 673 ALOGE("no wake lock to update!"); 674 return; 675 } 676 if (mPowerManager != 0) { 677 sp<IBinder> binder = new BBinder(); 678 status_t status; 679 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 680 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 681 } 682} 683 684void AudioFlinger::ThreadBase::clearPowerManager() 685{ 686 Mutex::Autolock _l(mLock); 687 releaseWakeLock_l(); 688 mPowerManager.clear(); 689} 690 691void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 692{ 693 sp<ThreadBase> thread = mThread.promote(); 694 if (thread != 0) { 695 thread->clearPowerManager(); 696 } 697 ALOGW("power manager service died !!!"); 698} 699 700void AudioFlinger::ThreadBase::setEffectSuspended( 701 const effect_uuid_t *type, bool suspend, int sessionId) 702{ 703 Mutex::Autolock _l(mLock); 704 setEffectSuspended_l(type, suspend, sessionId); 705} 706 707void AudioFlinger::ThreadBase::setEffectSuspended_l( 708 const effect_uuid_t *type, bool suspend, int sessionId) 709{ 710 sp<EffectChain> chain = getEffectChain_l(sessionId); 711 if (chain != 0) { 712 if (type != NULL) { 713 chain->setEffectSuspended_l(type, suspend); 714 } else { 715 chain->setEffectSuspendedAll_l(suspend); 716 } 717 } 718 719 updateSuspendedSessions_l(type, suspend, sessionId); 720} 721 722void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 723{ 724 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 725 if (index < 0) { 726 return; 727 } 728 729 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 730 mSuspendedSessions.valueAt(index); 731 732 for (size_t i = 0; i < sessionEffects.size(); i++) { 733 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 734 for (int j = 0; j < desc->mRefCount; j++) { 735 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 736 chain->setEffectSuspendedAll_l(true); 737 } else { 738 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 739 desc->mType.timeLow); 740 chain->setEffectSuspended_l(&desc->mType, true); 741 } 742 } 743 } 744} 745 746void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 747 bool suspend, 748 int sessionId) 749{ 750 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 751 752 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 753 754 if (suspend) { 755 if (index >= 0) { 756 sessionEffects = mSuspendedSessions.valueAt(index); 757 } else { 758 mSuspendedSessions.add(sessionId, sessionEffects); 759 } 760 } else { 761 if (index < 0) { 762 return; 763 } 764 sessionEffects = mSuspendedSessions.valueAt(index); 765 } 766 767 768 int key = EffectChain::kKeyForSuspendAll; 769 if (type != NULL) { 770 key = type->timeLow; 771 } 772 index = sessionEffects.indexOfKey(key); 773 774 sp<SuspendedSessionDesc> desc; 775 if (suspend) { 776 if (index >= 0) { 777 desc = sessionEffects.valueAt(index); 778 } else { 779 desc = new SuspendedSessionDesc(); 780 if (type != NULL) { 781 desc->mType = *type; 782 } 783 sessionEffects.add(key, desc); 784 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 785 } 786 desc->mRefCount++; 787 } else { 788 if (index < 0) { 789 return; 790 } 791 desc = sessionEffects.valueAt(index); 792 if (--desc->mRefCount == 0) { 793 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 794 sessionEffects.removeItemsAt(index); 795 if (sessionEffects.isEmpty()) { 796 ALOGV("updateSuspendedSessions_l() restore removing session %d", 797 sessionId); 798 mSuspendedSessions.removeItem(sessionId); 799 } 800 } 801 } 802 if (!sessionEffects.isEmpty()) { 803 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 804 } 805} 806 807void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 808 bool enabled, 809 int sessionId) 810{ 811 Mutex::Autolock _l(mLock); 812 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 813} 814 815void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 816 bool enabled, 817 int sessionId) 818{ 819 if (mType != RECORD) { 820 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 821 // another session. This gives the priority to well behaved effect control panels 822 // and applications not using global effects. 823 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 824 // global effects 825 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 826 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 827 } 828 } 829 830 sp<EffectChain> chain = getEffectChain_l(sessionId); 831 if (chain != 0) { 832 chain->checkSuspendOnEffectEnabled(effect, enabled); 833 } 834} 835 836// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 837sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 838 const sp<AudioFlinger::Client>& client, 839 const sp<IEffectClient>& effectClient, 840 int32_t priority, 841 int sessionId, 842 effect_descriptor_t *desc, 843 int *enabled, 844 status_t *status) 845{ 846 sp<EffectModule> effect; 847 sp<EffectHandle> handle; 848 status_t lStatus; 849 sp<EffectChain> chain; 850 bool chainCreated = false; 851 bool effectCreated = false; 852 bool effectRegistered = false; 853 854 lStatus = initCheck(); 855 if (lStatus != NO_ERROR) { 856 ALOGW("createEffect_l() Audio driver not initialized."); 857 goto Exit; 858 } 859 860 // Reject any effect on Direct output threads for now, since the format of 861 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 862 if (mType == DIRECT) { 863 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 864 desc->name, mName); 865 lStatus = BAD_VALUE; 866 goto Exit; 867 } 868 869 // Allow global effects only on offloaded and mixer threads 870 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 871 switch (mType) { 872 case MIXER: 873 case OFFLOAD: 874 break; 875 case DIRECT: 876 case DUPLICATING: 877 case RECORD: 878 default: 879 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 880 lStatus = BAD_VALUE; 881 goto Exit; 882 } 883 } 884 885 // Only Pre processor effects are allowed on input threads and only on input threads 886 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 887 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 888 desc->name, desc->flags, mType); 889 lStatus = BAD_VALUE; 890 goto Exit; 891 } 892 893 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 894 895 { // scope for mLock 896 Mutex::Autolock _l(mLock); 897 898 // check for existing effect chain with the requested audio session 899 chain = getEffectChain_l(sessionId); 900 if (chain == 0) { 901 // create a new chain for this session 902 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 903 chain = new EffectChain(this, sessionId); 904 addEffectChain_l(chain); 905 chain->setStrategy(getStrategyForSession_l(sessionId)); 906 chainCreated = true; 907 } else { 908 effect = chain->getEffectFromDesc_l(desc); 909 } 910 911 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 912 913 if (effect == 0) { 914 int id = mAudioFlinger->nextUniqueId(); 915 // Check CPU and memory usage 916 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 917 if (lStatus != NO_ERROR) { 918 goto Exit; 919 } 920 effectRegistered = true; 921 // create a new effect module if none present in the chain 922 effect = new EffectModule(this, chain, desc, id, sessionId); 923 lStatus = effect->status(); 924 if (lStatus != NO_ERROR) { 925 goto Exit; 926 } 927 effect->setOffloaded(mType == OFFLOAD, mId); 928 929 lStatus = chain->addEffect_l(effect); 930 if (lStatus != NO_ERROR) { 931 goto Exit; 932 } 933 effectCreated = true; 934 935 effect->setDevice(mOutDevice); 936 effect->setDevice(mInDevice); 937 effect->setMode(mAudioFlinger->getMode()); 938 effect->setAudioSource(mAudioSource); 939 } 940 // create effect handle and connect it to effect module 941 handle = new EffectHandle(effect, client, effectClient, priority); 942 lStatus = handle->initCheck(); 943 if (lStatus == OK) { 944 lStatus = effect->addHandle(handle.get()); 945 } 946 if (enabled != NULL) { 947 *enabled = (int)effect->isEnabled(); 948 } 949 } 950 951Exit: 952 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 953 Mutex::Autolock _l(mLock); 954 if (effectCreated) { 955 chain->removeEffect_l(effect); 956 } 957 if (effectRegistered) { 958 AudioSystem::unregisterEffect(effect->id()); 959 } 960 if (chainCreated) { 961 removeEffectChain_l(chain); 962 } 963 handle.clear(); 964 } 965 966 *status = lStatus; 967 return handle; 968} 969 970sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 971{ 972 Mutex::Autolock _l(mLock); 973 return getEffect_l(sessionId, effectId); 974} 975 976sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 977{ 978 sp<EffectChain> chain = getEffectChain_l(sessionId); 979 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 980} 981 982// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 983// PlaybackThread::mLock held 984status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 985{ 986 // check for existing effect chain with the requested audio session 987 int sessionId = effect->sessionId(); 988 sp<EffectChain> chain = getEffectChain_l(sessionId); 989 bool chainCreated = false; 990 991 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 992 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 993 this, effect->desc().name, effect->desc().flags); 994 995 if (chain == 0) { 996 // create a new chain for this session 997 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 998 chain = new EffectChain(this, sessionId); 999 addEffectChain_l(chain); 1000 chain->setStrategy(getStrategyForSession_l(sessionId)); 1001 chainCreated = true; 1002 } 1003 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1004 1005 if (chain->getEffectFromId_l(effect->id()) != 0) { 1006 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1007 this, effect->desc().name, chain.get()); 1008 return BAD_VALUE; 1009 } 1010 1011 effect->setOffloaded(mType == OFFLOAD, mId); 1012 1013 status_t status = chain->addEffect_l(effect); 1014 if (status != NO_ERROR) { 1015 if (chainCreated) { 1016 removeEffectChain_l(chain); 1017 } 1018 return status; 1019 } 1020 1021 effect->setDevice(mOutDevice); 1022 effect->setDevice(mInDevice); 1023 effect->setMode(mAudioFlinger->getMode()); 1024 effect->setAudioSource(mAudioSource); 1025 return NO_ERROR; 1026} 1027 1028void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1029 1030 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1031 effect_descriptor_t desc = effect->desc(); 1032 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1033 detachAuxEffect_l(effect->id()); 1034 } 1035 1036 sp<EffectChain> chain = effect->chain().promote(); 1037 if (chain != 0) { 1038 // remove effect chain if removing last effect 1039 if (chain->removeEffect_l(effect) == 0) { 1040 removeEffectChain_l(chain); 1041 } 1042 } else { 1043 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1044 } 1045} 1046 1047void AudioFlinger::ThreadBase::lockEffectChains_l( 1048 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1049{ 1050 effectChains = mEffectChains; 1051 for (size_t i = 0; i < mEffectChains.size(); i++) { 1052 mEffectChains[i]->lock(); 1053 } 1054} 1055 1056void AudioFlinger::ThreadBase::unlockEffectChains( 1057 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1058{ 1059 for (size_t i = 0; i < effectChains.size(); i++) { 1060 effectChains[i]->unlock(); 1061 } 1062} 1063 1064sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1065{ 1066 Mutex::Autolock _l(mLock); 1067 return getEffectChain_l(sessionId); 1068} 1069 1070sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1071{ 1072 size_t size = mEffectChains.size(); 1073 for (size_t i = 0; i < size; i++) { 1074 if (mEffectChains[i]->sessionId() == sessionId) { 1075 return mEffectChains[i]; 1076 } 1077 } 1078 return 0; 1079} 1080 1081void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1082{ 1083 Mutex::Autolock _l(mLock); 1084 size_t size = mEffectChains.size(); 1085 for (size_t i = 0; i < size; i++) { 1086 mEffectChains[i]->setMode_l(mode); 1087 } 1088} 1089 1090void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1091 EffectHandle *handle, 1092 bool unpinIfLast) { 1093 1094 Mutex::Autolock _l(mLock); 1095 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1096 // delete the effect module if removing last handle on it 1097 if (effect->removeHandle(handle) == 0) { 1098 if (!effect->isPinned() || unpinIfLast) { 1099 removeEffect_l(effect); 1100 AudioSystem::unregisterEffect(effect->id()); 1101 } 1102 } 1103} 1104 1105// ---------------------------------------------------------------------------- 1106// Playback 1107// ---------------------------------------------------------------------------- 1108 1109AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1110 AudioStreamOut* output, 1111 audio_io_handle_t id, 1112 audio_devices_t device, 1113 type_t type) 1114 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1115 mNormalFrameCount(0), mSinkBuffer(NULL), 1116 mMixerBufferEnabled(false), 1117 mMixerBuffer(NULL), 1118 mMixerBufferSize(0), 1119 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1120 mMixerBufferValid(false), 1121 mEffectBufferEnabled(false), 1122 mEffectBuffer(NULL), 1123 mEffectBufferSize(0), 1124 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1125 mEffectBufferValid(false), 1126 mSuspended(0), mBytesWritten(0), 1127 mActiveTracksGeneration(0), 1128 // mStreamTypes[] initialized in constructor body 1129 mOutput(output), 1130 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1131 mMixerStatus(MIXER_IDLE), 1132 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1133 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1134 mBytesRemaining(0), 1135 mCurrentWriteLength(0), 1136 mUseAsyncWrite(false), 1137 mWriteAckSequence(0), 1138 mDrainSequence(0), 1139 mSignalPending(false), 1140 mScreenState(AudioFlinger::mScreenState), 1141 // index 0 is reserved for normal mixer's submix 1142 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1143 // mLatchD, mLatchQ, 1144 mLatchDValid(false), mLatchQValid(false) 1145{ 1146 snprintf(mName, kNameLength, "AudioOut_%X", id); 1147 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1148 1149 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1150 // it would be safer to explicitly pass initial masterVolume/masterMute as 1151 // parameter. 1152 // 1153 // If the HAL we are using has support for master volume or master mute, 1154 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1155 // and the mute set to false). 1156 mMasterVolume = audioFlinger->masterVolume_l(); 1157 mMasterMute = audioFlinger->masterMute_l(); 1158 if (mOutput && mOutput->audioHwDev) { 1159 if (mOutput->audioHwDev->canSetMasterVolume()) { 1160 mMasterVolume = 1.0; 1161 } 1162 1163 if (mOutput->audioHwDev->canSetMasterMute()) { 1164 mMasterMute = false; 1165 } 1166 } 1167 1168 readOutputParameters_l(); 1169 1170 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1171 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1172 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1173 stream = (audio_stream_type_t) (stream + 1)) { 1174 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1175 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1176 } 1177 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1178 // because mAudioFlinger doesn't have one to copy from 1179} 1180 1181AudioFlinger::PlaybackThread::~PlaybackThread() 1182{ 1183 mAudioFlinger->unregisterWriter(mNBLogWriter); 1184 free(mSinkBuffer); 1185 free(mMixerBuffer); 1186 free(mEffectBuffer); 1187} 1188 1189void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1190{ 1191 dumpInternals(fd, args); 1192 dumpTracks(fd, args); 1193 dumpEffectChains(fd, args); 1194} 1195 1196void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1197{ 1198 const size_t SIZE = 256; 1199 char buffer[SIZE]; 1200 String8 result; 1201 1202 result.appendFormat(" Stream volumes in dB: "); 1203 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1204 const stream_type_t *st = &mStreamTypes[i]; 1205 if (i > 0) { 1206 result.appendFormat(", "); 1207 } 1208 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1209 if (st->mute) { 1210 result.append("M"); 1211 } 1212 } 1213 result.append("\n"); 1214 write(fd, result.string(), result.length()); 1215 result.clear(); 1216 1217 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1218 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1219 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1220 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1221 1222 size_t numtracks = mTracks.size(); 1223 size_t numactive = mActiveTracks.size(); 1224 dprintf(fd, " %d Tracks", numtracks); 1225 size_t numactiveseen = 0; 1226 if (numtracks) { 1227 dprintf(fd, " of which %d are active\n", numactive); 1228 Track::appendDumpHeader(result); 1229 for (size_t i = 0; i < numtracks; ++i) { 1230 sp<Track> track = mTracks[i]; 1231 if (track != 0) { 1232 bool active = mActiveTracks.indexOf(track) >= 0; 1233 if (active) { 1234 numactiveseen++; 1235 } 1236 track->dump(buffer, SIZE, active); 1237 result.append(buffer); 1238 } 1239 } 1240 } else { 1241 result.append("\n"); 1242 } 1243 if (numactiveseen != numactive) { 1244 // some tracks in the active list were not in the tracks list 1245 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1246 " not in the track list\n"); 1247 result.append(buffer); 1248 Track::appendDumpHeader(result); 1249 for (size_t i = 0; i < numactive; ++i) { 1250 sp<Track> track = mActiveTracks[i].promote(); 1251 if (track != 0 && mTracks.indexOf(track) < 0) { 1252 track->dump(buffer, SIZE, true); 1253 result.append(buffer); 1254 } 1255 } 1256 } 1257 1258 write(fd, result.string(), result.size()); 1259} 1260 1261void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1262{ 1263 dprintf(fd, "\nOutput thread %p:\n", this); 1264 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1265 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1266 dprintf(fd, " Total writes: %d\n", mNumWrites); 1267 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1268 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1269 dprintf(fd, " Suspend count: %d\n", mSuspended); 1270 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1271 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1272 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1273 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1274 1275 dumpBase(fd, args); 1276} 1277 1278// Thread virtuals 1279 1280void AudioFlinger::PlaybackThread::onFirstRef() 1281{ 1282 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1283} 1284 1285// ThreadBase virtuals 1286void AudioFlinger::PlaybackThread::preExit() 1287{ 1288 ALOGV(" preExit()"); 1289 // FIXME this is using hard-coded strings but in the future, this functionality will be 1290 // converted to use audio HAL extensions required to support tunneling 1291 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1292} 1293 1294// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1295sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1296 const sp<AudioFlinger::Client>& client, 1297 audio_stream_type_t streamType, 1298 uint32_t sampleRate, 1299 audio_format_t format, 1300 audio_channel_mask_t channelMask, 1301 size_t *pFrameCount, 1302 const sp<IMemory>& sharedBuffer, 1303 int sessionId, 1304 IAudioFlinger::track_flags_t *flags, 1305 pid_t tid, 1306 int uid, 1307 status_t *status) 1308{ 1309 size_t frameCount = *pFrameCount; 1310 sp<Track> track; 1311 status_t lStatus; 1312 1313 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1314 1315 // client expresses a preference for FAST, but we get the final say 1316 if (*flags & IAudioFlinger::TRACK_FAST) { 1317 if ( 1318 // not timed 1319 (!isTimed) && 1320 // either of these use cases: 1321 ( 1322 // use case 1: shared buffer with any frame count 1323 ( 1324 (sharedBuffer != 0) 1325 ) || 1326 // use case 2: callback handler and frame count is default or at least as large as HAL 1327 ( 1328 (tid != -1) && 1329 ((frameCount == 0) || 1330 (frameCount >= mFrameCount)) 1331 ) 1332 ) && 1333 // PCM data 1334 audio_is_linear_pcm(format) && 1335 // mono or stereo 1336 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1337 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1338 // hardware sample rate 1339 (sampleRate == mSampleRate) && 1340 // normal mixer has an associated fast mixer 1341 hasFastMixer() && 1342 // there are sufficient fast track slots available 1343 (mFastTrackAvailMask != 0) 1344 // FIXME test that MixerThread for this fast track has a capable output HAL 1345 // FIXME add a permission test also? 1346 ) { 1347 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1348 if (frameCount == 0) { 1349 // read the fast track multiplier property the first time it is needed 1350 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1351 if (ok != 0) { 1352 ALOGE("%s pthread_once failed: %d", __func__, ok); 1353 } 1354 frameCount = mFrameCount * sFastTrackMultiplier; 1355 } 1356 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1357 frameCount, mFrameCount); 1358 } else { 1359 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1360 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1361 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1362 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1363 audio_is_linear_pcm(format), 1364 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1365 *flags &= ~IAudioFlinger::TRACK_FAST; 1366 // For compatibility with AudioTrack calculation, buffer depth is forced 1367 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1368 // This is probably too conservative, but legacy application code may depend on it. 1369 // If you change this calculation, also review the start threshold which is related. 1370 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1371 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1372 if (minBufCount < 2) { 1373 minBufCount = 2; 1374 } 1375 size_t minFrameCount = mNormalFrameCount * minBufCount; 1376 if (frameCount < minFrameCount) { 1377 frameCount = minFrameCount; 1378 } 1379 } 1380 } 1381 *pFrameCount = frameCount; 1382 1383 switch (mType) { 1384 1385 case DIRECT: 1386 if (audio_is_linear_pcm(format)) { 1387 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1388 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1389 "for output %p with format %#x", 1390 sampleRate, format, channelMask, mOutput, mFormat); 1391 lStatus = BAD_VALUE; 1392 goto Exit; 1393 } 1394 } 1395 break; 1396 1397 case OFFLOAD: 1398 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1399 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1400 "for output %p with format %#x", 1401 sampleRate, format, channelMask, mOutput, mFormat); 1402 lStatus = BAD_VALUE; 1403 goto Exit; 1404 } 1405 break; 1406 1407 default: 1408 if (!audio_is_linear_pcm(format)) { 1409 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1410 "for output %p with format %#x", 1411 format, mOutput, mFormat); 1412 lStatus = BAD_VALUE; 1413 goto Exit; 1414 } 1415 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1416 if (sampleRate > mSampleRate*2) { 1417 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1418 lStatus = BAD_VALUE; 1419 goto Exit; 1420 } 1421 break; 1422 1423 } 1424 1425 lStatus = initCheck(); 1426 if (lStatus != NO_ERROR) { 1427 ALOGE("createTrack_l() audio driver not initialized"); 1428 goto Exit; 1429 } 1430 1431 { // scope for mLock 1432 Mutex::Autolock _l(mLock); 1433 1434 // all tracks in same audio session must share the same routing strategy otherwise 1435 // conflicts will happen when tracks are moved from one output to another by audio policy 1436 // manager 1437 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1438 for (size_t i = 0; i < mTracks.size(); ++i) { 1439 sp<Track> t = mTracks[i]; 1440 if (t != 0 && !t->isOutputTrack()) { 1441 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1442 if (sessionId == t->sessionId() && strategy != actual) { 1443 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1444 strategy, actual); 1445 lStatus = BAD_VALUE; 1446 goto Exit; 1447 } 1448 } 1449 } 1450 1451 if (!isTimed) { 1452 track = new Track(this, client, streamType, sampleRate, format, 1453 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1454 } else { 1455 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1456 channelMask, frameCount, sharedBuffer, sessionId, uid); 1457 } 1458 1459 // new Track always returns non-NULL, 1460 // but TimedTrack::create() is a factory that could fail by returning NULL 1461 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1462 if (lStatus != NO_ERROR) { 1463 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1464 // track must be cleared from the caller as the caller has the AF lock 1465 goto Exit; 1466 } 1467 mTracks.add(track); 1468 1469 sp<EffectChain> chain = getEffectChain_l(sessionId); 1470 if (chain != 0) { 1471 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1472 track->setMainBuffer(chain->inBuffer()); 1473 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1474 chain->incTrackCnt(); 1475 } 1476 1477 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1478 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1479 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1480 // so ask activity manager to do this on our behalf 1481 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1482 } 1483 } 1484 1485 lStatus = NO_ERROR; 1486 1487Exit: 1488 *status = lStatus; 1489 return track; 1490} 1491 1492uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1493{ 1494 return latency; 1495} 1496 1497uint32_t AudioFlinger::PlaybackThread::latency() const 1498{ 1499 Mutex::Autolock _l(mLock); 1500 return latency_l(); 1501} 1502uint32_t AudioFlinger::PlaybackThread::latency_l() const 1503{ 1504 if (initCheck() == NO_ERROR) { 1505 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1506 } else { 1507 return 0; 1508 } 1509} 1510 1511void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1512{ 1513 Mutex::Autolock _l(mLock); 1514 // Don't apply master volume in SW if our HAL can do it for us. 1515 if (mOutput && mOutput->audioHwDev && 1516 mOutput->audioHwDev->canSetMasterVolume()) { 1517 mMasterVolume = 1.0; 1518 } else { 1519 mMasterVolume = value; 1520 } 1521} 1522 1523void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1524{ 1525 Mutex::Autolock _l(mLock); 1526 // Don't apply master mute in SW if our HAL can do it for us. 1527 if (mOutput && mOutput->audioHwDev && 1528 mOutput->audioHwDev->canSetMasterMute()) { 1529 mMasterMute = false; 1530 } else { 1531 mMasterMute = muted; 1532 } 1533} 1534 1535void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1536{ 1537 Mutex::Autolock _l(mLock); 1538 mStreamTypes[stream].volume = value; 1539 broadcast_l(); 1540} 1541 1542void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1543{ 1544 Mutex::Autolock _l(mLock); 1545 mStreamTypes[stream].mute = muted; 1546 broadcast_l(); 1547} 1548 1549float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1550{ 1551 Mutex::Autolock _l(mLock); 1552 return mStreamTypes[stream].volume; 1553} 1554 1555// addTrack_l() must be called with ThreadBase::mLock held 1556status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1557{ 1558 status_t status = ALREADY_EXISTS; 1559 1560 // set retry count for buffer fill 1561 track->mRetryCount = kMaxTrackStartupRetries; 1562 if (mActiveTracks.indexOf(track) < 0) { 1563 // the track is newly added, make sure it fills up all its 1564 // buffers before playing. This is to ensure the client will 1565 // effectively get the latency it requested. 1566 if (!track->isOutputTrack()) { 1567 TrackBase::track_state state = track->mState; 1568 mLock.unlock(); 1569 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1570 mLock.lock(); 1571 // abort track was stopped/paused while we released the lock 1572 if (state != track->mState) { 1573 if (status == NO_ERROR) { 1574 mLock.unlock(); 1575 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1576 mLock.lock(); 1577 } 1578 return INVALID_OPERATION; 1579 } 1580 // abort if start is rejected by audio policy manager 1581 if (status != NO_ERROR) { 1582 return PERMISSION_DENIED; 1583 } 1584#ifdef ADD_BATTERY_DATA 1585 // to track the speaker usage 1586 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1587#endif 1588 } 1589 1590 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1591 track->mResetDone = false; 1592 track->mPresentationCompleteFrames = 0; 1593 mActiveTracks.add(track); 1594 mWakeLockUids.add(track->uid()); 1595 mActiveTracksGeneration++; 1596 mLatestActiveTrack = track; 1597 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1598 if (chain != 0) { 1599 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1600 track->sessionId()); 1601 chain->incActiveTrackCnt(); 1602 } 1603 1604 status = NO_ERROR; 1605 } 1606 1607 onAddNewTrack_l(); 1608 return status; 1609} 1610 1611bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1612{ 1613 track->terminate(); 1614 // active tracks are removed by threadLoop() 1615 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1616 track->mState = TrackBase::STOPPED; 1617 if (!trackActive) { 1618 removeTrack_l(track); 1619 } else if (track->isFastTrack() || track->isOffloaded()) { 1620 track->mState = TrackBase::STOPPING_1; 1621 } 1622 1623 return trackActive; 1624} 1625 1626void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1627{ 1628 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1629 mTracks.remove(track); 1630 deleteTrackName_l(track->name()); 1631 // redundant as track is about to be destroyed, for dumpsys only 1632 track->mName = -1; 1633 if (track->isFastTrack()) { 1634 int index = track->mFastIndex; 1635 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1636 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1637 mFastTrackAvailMask |= 1 << index; 1638 // redundant as track is about to be destroyed, for dumpsys only 1639 track->mFastIndex = -1; 1640 } 1641 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1642 if (chain != 0) { 1643 chain->decTrackCnt(); 1644 } 1645} 1646 1647void AudioFlinger::PlaybackThread::broadcast_l() 1648{ 1649 // Thread could be blocked waiting for async 1650 // so signal it to handle state changes immediately 1651 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1652 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1653 mSignalPending = true; 1654 mWaitWorkCV.broadcast(); 1655} 1656 1657String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1658{ 1659 Mutex::Autolock _l(mLock); 1660 if (initCheck() != NO_ERROR) { 1661 return String8(); 1662 } 1663 1664 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1665 const String8 out_s8(s); 1666 free(s); 1667 return out_s8; 1668} 1669 1670void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1671 AudioSystem::OutputDescriptor desc; 1672 void *param2 = NULL; 1673 1674 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1675 param); 1676 1677 switch (event) { 1678 case AudioSystem::OUTPUT_OPENED: 1679 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1680 desc.channelMask = mChannelMask; 1681 desc.samplingRate = mSampleRate; 1682 desc.format = mFormat; 1683 desc.frameCount = mNormalFrameCount; // FIXME see 1684 // AudioFlinger::frameCount(audio_io_handle_t) 1685 desc.latency = latency_l(); 1686 param2 = &desc; 1687 break; 1688 1689 case AudioSystem::STREAM_CONFIG_CHANGED: 1690 param2 = ¶m; 1691 case AudioSystem::OUTPUT_CLOSED: 1692 default: 1693 break; 1694 } 1695 mAudioFlinger->audioConfigChanged(event, mId, param2); 1696} 1697 1698void AudioFlinger::PlaybackThread::writeCallback() 1699{ 1700 ALOG_ASSERT(mCallbackThread != 0); 1701 mCallbackThread->resetWriteBlocked(); 1702} 1703 1704void AudioFlinger::PlaybackThread::drainCallback() 1705{ 1706 ALOG_ASSERT(mCallbackThread != 0); 1707 mCallbackThread->resetDraining(); 1708} 1709 1710void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1711{ 1712 Mutex::Autolock _l(mLock); 1713 // reject out of sequence requests 1714 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1715 mWriteAckSequence &= ~1; 1716 mWaitWorkCV.signal(); 1717 } 1718} 1719 1720void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1721{ 1722 Mutex::Autolock _l(mLock); 1723 // reject out of sequence requests 1724 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1725 mDrainSequence &= ~1; 1726 mWaitWorkCV.signal(); 1727 } 1728} 1729 1730// static 1731int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1732 void *param __unused, 1733 void *cookie) 1734{ 1735 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1736 ALOGV("asyncCallback() event %d", event); 1737 switch (event) { 1738 case STREAM_CBK_EVENT_WRITE_READY: 1739 me->writeCallback(); 1740 break; 1741 case STREAM_CBK_EVENT_DRAIN_READY: 1742 me->drainCallback(); 1743 break; 1744 default: 1745 ALOGW("asyncCallback() unknown event %d", event); 1746 break; 1747 } 1748 return 0; 1749} 1750 1751void AudioFlinger::PlaybackThread::readOutputParameters_l() 1752{ 1753 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1754 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1755 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1756 if (!audio_is_output_channel(mChannelMask)) { 1757 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1758 } 1759 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1760 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1761 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1762 } 1763 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1764 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1765 if (!audio_is_valid_format(mFormat)) { 1766 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1767 } 1768 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1769 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1770 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1771 } 1772 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1773 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1774 mFrameCount = mBufferSize / mFrameSize; 1775 if (mFrameCount & 15) { 1776 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1777 mFrameCount); 1778 } 1779 1780 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1781 (mOutput->stream->set_callback != NULL)) { 1782 if (mOutput->stream->set_callback(mOutput->stream, 1783 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1784 mUseAsyncWrite = true; 1785 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1786 } 1787 } 1788 1789 // Calculate size of normal sink buffer relative to the HAL output buffer size 1790 double multiplier = 1.0; 1791 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1792 kUseFastMixer == FastMixer_Dynamic)) { 1793 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1794 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1795 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1796 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1797 maxNormalFrameCount = maxNormalFrameCount & ~15; 1798 if (maxNormalFrameCount < minNormalFrameCount) { 1799 maxNormalFrameCount = minNormalFrameCount; 1800 } 1801 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1802 if (multiplier <= 1.0) { 1803 multiplier = 1.0; 1804 } else if (multiplier <= 2.0) { 1805 if (2 * mFrameCount <= maxNormalFrameCount) { 1806 multiplier = 2.0; 1807 } else { 1808 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1809 } 1810 } else { 1811 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1812 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1813 // track, but we sometimes have to do this to satisfy the maximum frame count 1814 // constraint) 1815 // FIXME this rounding up should not be done if no HAL SRC 1816 uint32_t truncMult = (uint32_t) multiplier; 1817 if ((truncMult & 1)) { 1818 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1819 ++truncMult; 1820 } 1821 } 1822 multiplier = (double) truncMult; 1823 } 1824 } 1825 mNormalFrameCount = multiplier * mFrameCount; 1826 // round up to nearest 16 frames to satisfy AudioMixer 1827 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1828 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1829 mNormalFrameCount); 1830 1831 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1832 // Originally this was int16_t[] array, need to remove legacy implications. 1833 free(mSinkBuffer); 1834 mSinkBuffer = NULL; 1835 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1836 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1837 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1838 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1839 1840 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1841 // drives the output. 1842 free(mMixerBuffer); 1843 mMixerBuffer = NULL; 1844 if (mMixerBufferEnabled) { 1845 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1846 mMixerBufferSize = mNormalFrameCount * mChannelCount 1847 * audio_bytes_per_sample(mMixerBufferFormat); 1848 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1849 } 1850 free(mEffectBuffer); 1851 mEffectBuffer = NULL; 1852 if (mEffectBufferEnabled) { 1853 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1854 mEffectBufferSize = mNormalFrameCount * mChannelCount 1855 * audio_bytes_per_sample(mEffectBufferFormat); 1856 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1857 } 1858 1859 // force reconfiguration of effect chains and engines to take new buffer size and audio 1860 // parameters into account 1861 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1862 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1863 // matter. 1864 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1865 Vector< sp<EffectChain> > effectChains = mEffectChains; 1866 for (size_t i = 0; i < effectChains.size(); i ++) { 1867 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1868 } 1869} 1870 1871 1872status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1873{ 1874 if (halFrames == NULL || dspFrames == NULL) { 1875 return BAD_VALUE; 1876 } 1877 Mutex::Autolock _l(mLock); 1878 if (initCheck() != NO_ERROR) { 1879 return INVALID_OPERATION; 1880 } 1881 size_t framesWritten = mBytesWritten / mFrameSize; 1882 *halFrames = framesWritten; 1883 1884 if (isSuspended()) { 1885 // return an estimation of rendered frames when the output is suspended 1886 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1887 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1888 return NO_ERROR; 1889 } else { 1890 status_t status; 1891 uint32_t frames; 1892 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1893 *dspFrames = (size_t)frames; 1894 return status; 1895 } 1896} 1897 1898uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1899{ 1900 Mutex::Autolock _l(mLock); 1901 uint32_t result = 0; 1902 if (getEffectChain_l(sessionId) != 0) { 1903 result = EFFECT_SESSION; 1904 } 1905 1906 for (size_t i = 0; i < mTracks.size(); ++i) { 1907 sp<Track> track = mTracks[i]; 1908 if (sessionId == track->sessionId() && !track->isInvalid()) { 1909 result |= TRACK_SESSION; 1910 break; 1911 } 1912 } 1913 1914 return result; 1915} 1916 1917uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1918{ 1919 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1920 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1921 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1922 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1923 } 1924 for (size_t i = 0; i < mTracks.size(); i++) { 1925 sp<Track> track = mTracks[i]; 1926 if (sessionId == track->sessionId() && !track->isInvalid()) { 1927 return AudioSystem::getStrategyForStream(track->streamType()); 1928 } 1929 } 1930 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1931} 1932 1933 1934AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1935{ 1936 Mutex::Autolock _l(mLock); 1937 return mOutput; 1938} 1939 1940AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1941{ 1942 Mutex::Autolock _l(mLock); 1943 AudioStreamOut *output = mOutput; 1944 mOutput = NULL; 1945 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1946 // must push a NULL and wait for ack 1947 mOutputSink.clear(); 1948 mPipeSink.clear(); 1949 mNormalSink.clear(); 1950 return output; 1951} 1952 1953// this method must always be called either with ThreadBase mLock held or inside the thread loop 1954audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1955{ 1956 if (mOutput == NULL) { 1957 return NULL; 1958 } 1959 return &mOutput->stream->common; 1960} 1961 1962uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1963{ 1964 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1965} 1966 1967status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1968{ 1969 if (!isValidSyncEvent(event)) { 1970 return BAD_VALUE; 1971 } 1972 1973 Mutex::Autolock _l(mLock); 1974 1975 for (size_t i = 0; i < mTracks.size(); ++i) { 1976 sp<Track> track = mTracks[i]; 1977 if (event->triggerSession() == track->sessionId()) { 1978 (void) track->setSyncEvent(event); 1979 return NO_ERROR; 1980 } 1981 } 1982 1983 return NAME_NOT_FOUND; 1984} 1985 1986bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1987{ 1988 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1989} 1990 1991void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1992 const Vector< sp<Track> >& tracksToRemove) 1993{ 1994 size_t count = tracksToRemove.size(); 1995 if (count > 0) { 1996 for (size_t i = 0 ; i < count ; i++) { 1997 const sp<Track>& track = tracksToRemove.itemAt(i); 1998 if (!track->isOutputTrack()) { 1999 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2000#ifdef ADD_BATTERY_DATA 2001 // to track the speaker usage 2002 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2003#endif 2004 if (track->isTerminated()) { 2005 AudioSystem::releaseOutput(mId); 2006 } 2007 } 2008 } 2009 } 2010} 2011 2012void AudioFlinger::PlaybackThread::checkSilentMode_l() 2013{ 2014 if (!mMasterMute) { 2015 char value[PROPERTY_VALUE_MAX]; 2016 if (property_get("ro.audio.silent", value, "0") > 0) { 2017 char *endptr; 2018 unsigned long ul = strtoul(value, &endptr, 0); 2019 if (*endptr == '\0' && ul != 0) { 2020 ALOGD("Silence is golden"); 2021 // The setprop command will not allow a property to be changed after 2022 // the first time it is set, so we don't have to worry about un-muting. 2023 setMasterMute_l(true); 2024 } 2025 } 2026 } 2027} 2028 2029// shared by MIXER and DIRECT, overridden by DUPLICATING 2030ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2031{ 2032 // FIXME rewrite to reduce number of system calls 2033 mLastWriteTime = systemTime(); 2034 mInWrite = true; 2035 ssize_t bytesWritten; 2036 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2037 2038 // If an NBAIO sink is present, use it to write the normal mixer's submix 2039 if (mNormalSink != 0) { 2040 const size_t count = mBytesRemaining / mFrameSize; 2041 2042 ATRACE_BEGIN("write"); 2043 // update the setpoint when AudioFlinger::mScreenState changes 2044 uint32_t screenState = AudioFlinger::mScreenState; 2045 if (screenState != mScreenState) { 2046 mScreenState = screenState; 2047 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2048 if (pipe != NULL) { 2049 pipe->setAvgFrames((mScreenState & 1) ? 2050 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2051 } 2052 } 2053 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2054 ATRACE_END(); 2055 if (framesWritten > 0) { 2056 bytesWritten = framesWritten * mFrameSize; 2057 } else { 2058 bytesWritten = framesWritten; 2059 } 2060 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2061 if (status == NO_ERROR) { 2062 size_t totalFramesWritten = mNormalSink->framesWritten(); 2063 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2064 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2065 mLatchDValid = true; 2066 } 2067 } 2068 // otherwise use the HAL / AudioStreamOut directly 2069 } else { 2070 // Direct output and offload threads 2071 2072 if (mUseAsyncWrite) { 2073 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2074 mWriteAckSequence += 2; 2075 mWriteAckSequence |= 1; 2076 ALOG_ASSERT(mCallbackThread != 0); 2077 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2078 } 2079 // FIXME We should have an implementation of timestamps for direct output threads. 2080 // They are used e.g for multichannel PCM playback over HDMI. 2081 bytesWritten = mOutput->stream->write(mOutput->stream, 2082 (char *)mSinkBuffer + offset, mBytesRemaining); 2083 if (mUseAsyncWrite && 2084 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2085 // do not wait for async callback in case of error of full write 2086 mWriteAckSequence &= ~1; 2087 ALOG_ASSERT(mCallbackThread != 0); 2088 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2089 } 2090 } 2091 2092 mNumWrites++; 2093 mInWrite = false; 2094 mStandby = false; 2095 return bytesWritten; 2096} 2097 2098void AudioFlinger::PlaybackThread::threadLoop_drain() 2099{ 2100 if (mOutput->stream->drain) { 2101 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2102 if (mUseAsyncWrite) { 2103 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2104 mDrainSequence |= 1; 2105 ALOG_ASSERT(mCallbackThread != 0); 2106 mCallbackThread->setDraining(mDrainSequence); 2107 } 2108 mOutput->stream->drain(mOutput->stream, 2109 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2110 : AUDIO_DRAIN_ALL); 2111 } 2112} 2113 2114void AudioFlinger::PlaybackThread::threadLoop_exit() 2115{ 2116 // Default implementation has nothing to do 2117} 2118 2119/* 2120The derived values that are cached: 2121 - mSinkBufferSize from frame count * frame size 2122 - activeSleepTime from activeSleepTimeUs() 2123 - idleSleepTime from idleSleepTimeUs() 2124 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2125 - maxPeriod from frame count and sample rate (MIXER only) 2126 2127The parameters that affect these derived values are: 2128 - frame count 2129 - frame size 2130 - sample rate 2131 - device type: A2DP or not 2132 - device latency 2133 - format: PCM or not 2134 - active sleep time 2135 - idle sleep time 2136*/ 2137 2138void AudioFlinger::PlaybackThread::cacheParameters_l() 2139{ 2140 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2141 activeSleepTime = activeSleepTimeUs(); 2142 idleSleepTime = idleSleepTimeUs(); 2143} 2144 2145void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2146{ 2147 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2148 this, streamType, mTracks.size()); 2149 Mutex::Autolock _l(mLock); 2150 2151 size_t size = mTracks.size(); 2152 for (size_t i = 0; i < size; i++) { 2153 sp<Track> t = mTracks[i]; 2154 if (t->streamType() == streamType) { 2155 t->invalidate(); 2156 } 2157 } 2158} 2159 2160status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2161{ 2162 int session = chain->sessionId(); 2163 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2164 ? mEffectBuffer : mSinkBuffer); 2165 bool ownsBuffer = false; 2166 2167 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2168 if (session > 0) { 2169 // Only one effect chain can be present in direct output thread and it uses 2170 // the sink buffer as input 2171 if (mType != DIRECT) { 2172 size_t numSamples = mNormalFrameCount * mChannelCount; 2173 buffer = new int16_t[numSamples]; 2174 memset(buffer, 0, numSamples * sizeof(int16_t)); 2175 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2176 ownsBuffer = true; 2177 } 2178 2179 // Attach all tracks with same session ID to this chain. 2180 for (size_t i = 0; i < mTracks.size(); ++i) { 2181 sp<Track> track = mTracks[i]; 2182 if (session == track->sessionId()) { 2183 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2184 buffer); 2185 track->setMainBuffer(buffer); 2186 chain->incTrackCnt(); 2187 } 2188 } 2189 2190 // indicate all active tracks in the chain 2191 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2192 sp<Track> track = mActiveTracks[i].promote(); 2193 if (track == 0) { 2194 continue; 2195 } 2196 if (session == track->sessionId()) { 2197 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2198 chain->incActiveTrackCnt(); 2199 } 2200 } 2201 } 2202 2203 chain->setInBuffer(buffer, ownsBuffer); 2204 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2205 ? mEffectBuffer : mSinkBuffer)); 2206 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2207 // chains list in order to be processed last as it contains output stage effects 2208 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2209 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2210 // after track specific effects and before output stage 2211 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2212 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2213 // Effect chain for other sessions are inserted at beginning of effect 2214 // chains list to be processed before output mix effects. Relative order between other 2215 // sessions is not important 2216 size_t size = mEffectChains.size(); 2217 size_t i = 0; 2218 for (i = 0; i < size; i++) { 2219 if (mEffectChains[i]->sessionId() < session) { 2220 break; 2221 } 2222 } 2223 mEffectChains.insertAt(chain, i); 2224 checkSuspendOnAddEffectChain_l(chain); 2225 2226 return NO_ERROR; 2227} 2228 2229size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2230{ 2231 int session = chain->sessionId(); 2232 2233 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2234 2235 for (size_t i = 0; i < mEffectChains.size(); i++) { 2236 if (chain == mEffectChains[i]) { 2237 mEffectChains.removeAt(i); 2238 // detach all active tracks from the chain 2239 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2240 sp<Track> track = mActiveTracks[i].promote(); 2241 if (track == 0) { 2242 continue; 2243 } 2244 if (session == track->sessionId()) { 2245 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2246 chain.get(), session); 2247 chain->decActiveTrackCnt(); 2248 } 2249 } 2250 2251 // detach all tracks with same session ID from this chain 2252 for (size_t i = 0; i < mTracks.size(); ++i) { 2253 sp<Track> track = mTracks[i]; 2254 if (session == track->sessionId()) { 2255 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2256 chain->decTrackCnt(); 2257 } 2258 } 2259 break; 2260 } 2261 } 2262 return mEffectChains.size(); 2263} 2264 2265status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2266 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2267{ 2268 Mutex::Autolock _l(mLock); 2269 return attachAuxEffect_l(track, EffectId); 2270} 2271 2272status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2273 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2274{ 2275 status_t status = NO_ERROR; 2276 2277 if (EffectId == 0) { 2278 track->setAuxBuffer(0, NULL); 2279 } else { 2280 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2281 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2282 if (effect != 0) { 2283 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2284 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2285 } else { 2286 status = INVALID_OPERATION; 2287 } 2288 } else { 2289 status = BAD_VALUE; 2290 } 2291 } 2292 return status; 2293} 2294 2295void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2296{ 2297 for (size_t i = 0; i < mTracks.size(); ++i) { 2298 sp<Track> track = mTracks[i]; 2299 if (track->auxEffectId() == effectId) { 2300 attachAuxEffect_l(track, 0); 2301 } 2302 } 2303} 2304 2305bool AudioFlinger::PlaybackThread::threadLoop() 2306{ 2307 Vector< sp<Track> > tracksToRemove; 2308 2309 standbyTime = systemTime(); 2310 2311 // MIXER 2312 nsecs_t lastWarning = 0; 2313 2314 // DUPLICATING 2315 // FIXME could this be made local to while loop? 2316 writeFrames = 0; 2317 2318 int lastGeneration = 0; 2319 2320 cacheParameters_l(); 2321 sleepTime = idleSleepTime; 2322 2323 if (mType == MIXER) { 2324 sleepTimeShift = 0; 2325 } 2326 2327 CpuStats cpuStats; 2328 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2329 2330 acquireWakeLock(); 2331 2332 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2333 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2334 // and then that string will be logged at the next convenient opportunity. 2335 const char *logString = NULL; 2336 2337 checkSilentMode_l(); 2338 2339 while (!exitPending()) 2340 { 2341 cpuStats.sample(myName); 2342 2343 Vector< sp<EffectChain> > effectChains; 2344 2345 { // scope for mLock 2346 2347 Mutex::Autolock _l(mLock); 2348 2349 processConfigEvents_l(); 2350 2351 if (logString != NULL) { 2352 mNBLogWriter->logTimestamp(); 2353 mNBLogWriter->log(logString); 2354 logString = NULL; 2355 } 2356 2357 if (mLatchDValid) { 2358 mLatchQ = mLatchD; 2359 mLatchDValid = false; 2360 mLatchQValid = true; 2361 } 2362 2363 saveOutputTracks(); 2364 if (mSignalPending) { 2365 // A signal was raised while we were unlocked 2366 mSignalPending = false; 2367 } else if (waitingAsyncCallback_l()) { 2368 if (exitPending()) { 2369 break; 2370 } 2371 releaseWakeLock_l(); 2372 mWakeLockUids.clear(); 2373 mActiveTracksGeneration++; 2374 ALOGV("wait async completion"); 2375 mWaitWorkCV.wait(mLock); 2376 ALOGV("async completion/wake"); 2377 acquireWakeLock_l(); 2378 standbyTime = systemTime() + standbyDelay; 2379 sleepTime = 0; 2380 2381 continue; 2382 } 2383 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2384 isSuspended()) { 2385 // put audio hardware into standby after short delay 2386 if (shouldStandby_l()) { 2387 2388 threadLoop_standby(); 2389 2390 mStandby = true; 2391 } 2392 2393 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2394 // we're about to wait, flush the binder command buffer 2395 IPCThreadState::self()->flushCommands(); 2396 2397 clearOutputTracks(); 2398 2399 if (exitPending()) { 2400 break; 2401 } 2402 2403 releaseWakeLock_l(); 2404 mWakeLockUids.clear(); 2405 mActiveTracksGeneration++; 2406 // wait until we have something to do... 2407 ALOGV("%s going to sleep", myName.string()); 2408 mWaitWorkCV.wait(mLock); 2409 ALOGV("%s waking up", myName.string()); 2410 acquireWakeLock_l(); 2411 2412 mMixerStatus = MIXER_IDLE; 2413 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2414 mBytesWritten = 0; 2415 mBytesRemaining = 0; 2416 checkSilentMode_l(); 2417 2418 standbyTime = systemTime() + standbyDelay; 2419 sleepTime = idleSleepTime; 2420 if (mType == MIXER) { 2421 sleepTimeShift = 0; 2422 } 2423 2424 continue; 2425 } 2426 } 2427 // mMixerStatusIgnoringFastTracks is also updated internally 2428 mMixerStatus = prepareTracks_l(&tracksToRemove); 2429 2430 // compare with previously applied list 2431 if (lastGeneration != mActiveTracksGeneration) { 2432 // update wakelock 2433 updateWakeLockUids_l(mWakeLockUids); 2434 lastGeneration = mActiveTracksGeneration; 2435 } 2436 2437 // prevent any changes in effect chain list and in each effect chain 2438 // during mixing and effect process as the audio buffers could be deleted 2439 // or modified if an effect is created or deleted 2440 lockEffectChains_l(effectChains); 2441 } // mLock scope ends 2442 2443 if (mBytesRemaining == 0) { 2444 mCurrentWriteLength = 0; 2445 if (mMixerStatus == MIXER_TRACKS_READY) { 2446 // threadLoop_mix() sets mCurrentWriteLength 2447 threadLoop_mix(); 2448 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2449 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2450 // threadLoop_sleepTime sets sleepTime to 0 if data 2451 // must be written to HAL 2452 threadLoop_sleepTime(); 2453 if (sleepTime == 0) { 2454 mCurrentWriteLength = mSinkBufferSize; 2455 } 2456 } 2457 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2458 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2459 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2460 // or mSinkBuffer (if there are no effects). 2461 // 2462 // This is done pre-effects computation; if effects change to 2463 // support higher precision, this needs to move. 2464 // 2465 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2466 // TODO use sleepTime == 0 as an additional condition. 2467 if (mMixerBufferValid) { 2468 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2469 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2470 2471 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2472 mNormalFrameCount * mChannelCount); 2473 } 2474 2475 mBytesRemaining = mCurrentWriteLength; 2476 if (isSuspended()) { 2477 sleepTime = suspendSleepTimeUs(); 2478 // simulate write to HAL when suspended 2479 mBytesWritten += mSinkBufferSize; 2480 mBytesRemaining = 0; 2481 } 2482 2483 // only process effects if we're going to write 2484 if (sleepTime == 0 && mType != OFFLOAD) { 2485 for (size_t i = 0; i < effectChains.size(); i ++) { 2486 effectChains[i]->process_l(); 2487 } 2488 } 2489 } 2490 // Process effect chains for offloaded thread even if no audio 2491 // was read from audio track: process only updates effect state 2492 // and thus does have to be synchronized with audio writes but may have 2493 // to be called while waiting for async write callback 2494 if (mType == OFFLOAD) { 2495 for (size_t i = 0; i < effectChains.size(); i ++) { 2496 effectChains[i]->process_l(); 2497 } 2498 } 2499 2500 // Only if the Effects buffer is enabled and there is data in the 2501 // Effects buffer (buffer valid), we need to 2502 // copy into the sink buffer. 2503 // TODO use sleepTime == 0 as an additional condition. 2504 if (mEffectBufferValid) { 2505 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2506 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2507 mNormalFrameCount * mChannelCount); 2508 } 2509 2510 // enable changes in effect chain 2511 unlockEffectChains(effectChains); 2512 2513 if (!waitingAsyncCallback()) { 2514 // sleepTime == 0 means we must write to audio hardware 2515 if (sleepTime == 0) { 2516 if (mBytesRemaining) { 2517 ssize_t ret = threadLoop_write(); 2518 if (ret < 0) { 2519 mBytesRemaining = 0; 2520 } else { 2521 mBytesWritten += ret; 2522 mBytesRemaining -= ret; 2523 } 2524 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2525 (mMixerStatus == MIXER_DRAIN_ALL)) { 2526 threadLoop_drain(); 2527 } 2528 if (mType == MIXER) { 2529 // write blocked detection 2530 nsecs_t now = systemTime(); 2531 nsecs_t delta = now - mLastWriteTime; 2532 if (!mStandby && delta > maxPeriod) { 2533 mNumDelayedWrites++; 2534 if ((now - lastWarning) > kWarningThrottleNs) { 2535 ATRACE_NAME("underrun"); 2536 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2537 ns2ms(delta), mNumDelayedWrites, this); 2538 lastWarning = now; 2539 } 2540 } 2541 } 2542 2543 } else { 2544 usleep(sleepTime); 2545 } 2546 } 2547 2548 // Finally let go of removed track(s), without the lock held 2549 // since we can't guarantee the destructors won't acquire that 2550 // same lock. This will also mutate and push a new fast mixer state. 2551 threadLoop_removeTracks(tracksToRemove); 2552 tracksToRemove.clear(); 2553 2554 // FIXME I don't understand the need for this here; 2555 // it was in the original code but maybe the 2556 // assignment in saveOutputTracks() makes this unnecessary? 2557 clearOutputTracks(); 2558 2559 // Effect chains will be actually deleted here if they were removed from 2560 // mEffectChains list during mixing or effects processing 2561 effectChains.clear(); 2562 2563 // FIXME Note that the above .clear() is no longer necessary since effectChains 2564 // is now local to this block, but will keep it for now (at least until merge done). 2565 } 2566 2567 threadLoop_exit(); 2568 2569 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2570 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2571 // put output stream into standby mode 2572 if (!mStandby) { 2573 mOutput->stream->common.standby(&mOutput->stream->common); 2574 } 2575 } 2576 2577 releaseWakeLock(); 2578 mWakeLockUids.clear(); 2579 mActiveTracksGeneration++; 2580 2581 ALOGV("Thread %p type %d exiting", this, mType); 2582 return false; 2583} 2584 2585// removeTracks_l() must be called with ThreadBase::mLock held 2586void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2587{ 2588 size_t count = tracksToRemove.size(); 2589 if (count > 0) { 2590 for (size_t i=0 ; i<count ; i++) { 2591 const sp<Track>& track = tracksToRemove.itemAt(i); 2592 mActiveTracks.remove(track); 2593 mWakeLockUids.remove(track->uid()); 2594 mActiveTracksGeneration++; 2595 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2596 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2597 if (chain != 0) { 2598 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2599 track->sessionId()); 2600 chain->decActiveTrackCnt(); 2601 } 2602 if (track->isTerminated()) { 2603 removeTrack_l(track); 2604 } 2605 } 2606 } 2607 2608} 2609 2610status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2611{ 2612 if (mNormalSink != 0) { 2613 return mNormalSink->getTimestamp(timestamp); 2614 } 2615 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2616 uint64_t position64; 2617 int ret = mOutput->stream->get_presentation_position( 2618 mOutput->stream, &position64, ×tamp.mTime); 2619 if (ret == 0) { 2620 timestamp.mPosition = (uint32_t)position64; 2621 return NO_ERROR; 2622 } 2623 } 2624 return INVALID_OPERATION; 2625} 2626// ---------------------------------------------------------------------------- 2627 2628AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2629 audio_io_handle_t id, audio_devices_t device, type_t type) 2630 : PlaybackThread(audioFlinger, output, id, device, type), 2631 // mAudioMixer below 2632 // mFastMixer below 2633 mFastMixerFutex(0) 2634 // mOutputSink below 2635 // mPipeSink below 2636 // mNormalSink below 2637{ 2638 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2639 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2640 "mFrameCount=%d, mNormalFrameCount=%d", 2641 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2642 mNormalFrameCount); 2643 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2644 2645 // FIXME - Current mixer implementation only supports stereo output 2646 if (mChannelCount != FCC_2) { 2647 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2648 } 2649 2650 // create an NBAIO sink for the HAL output stream, and negotiate 2651 mOutputSink = new AudioStreamOutSink(output->stream); 2652 size_t numCounterOffers = 0; 2653 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2654 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2655 ALOG_ASSERT(index == 0); 2656 2657 // initialize fast mixer depending on configuration 2658 bool initFastMixer; 2659 switch (kUseFastMixer) { 2660 case FastMixer_Never: 2661 initFastMixer = false; 2662 break; 2663 case FastMixer_Always: 2664 initFastMixer = true; 2665 break; 2666 case FastMixer_Static: 2667 case FastMixer_Dynamic: 2668 initFastMixer = mFrameCount < mNormalFrameCount; 2669 break; 2670 } 2671 if (initFastMixer) { 2672 2673 // create a MonoPipe to connect our submix to FastMixer 2674 NBAIO_Format format = mOutputSink->format(); 2675 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2676 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2677 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2678 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2679 const NBAIO_Format offers[1] = {format}; 2680 size_t numCounterOffers = 0; 2681 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2682 ALOG_ASSERT(index == 0); 2683 monoPipe->setAvgFrames((mScreenState & 1) ? 2684 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2685 mPipeSink = monoPipe; 2686 2687#ifdef TEE_SINK 2688 if (mTeeSinkOutputEnabled) { 2689 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2690 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2691 numCounterOffers = 0; 2692 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2693 ALOG_ASSERT(index == 0); 2694 mTeeSink = teeSink; 2695 PipeReader *teeSource = new PipeReader(*teeSink); 2696 numCounterOffers = 0; 2697 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2698 ALOG_ASSERT(index == 0); 2699 mTeeSource = teeSource; 2700 } 2701#endif 2702 2703 // create fast mixer and configure it initially with just one fast track for our submix 2704 mFastMixer = new FastMixer(); 2705 FastMixerStateQueue *sq = mFastMixer->sq(); 2706#ifdef STATE_QUEUE_DUMP 2707 sq->setObserverDump(&mStateQueueObserverDump); 2708 sq->setMutatorDump(&mStateQueueMutatorDump); 2709#endif 2710 FastMixerState *state = sq->begin(); 2711 FastTrack *fastTrack = &state->mFastTracks[0]; 2712 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2713 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2714 fastTrack->mVolumeProvider = NULL; 2715 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2716 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2717 fastTrack->mGeneration++; 2718 state->mFastTracksGen++; 2719 state->mTrackMask = 1; 2720 // fast mixer will use the HAL output sink 2721 state->mOutputSink = mOutputSink.get(); 2722 state->mOutputSinkGen++; 2723 state->mFrameCount = mFrameCount; 2724 state->mCommand = FastMixerState::COLD_IDLE; 2725 // already done in constructor initialization list 2726 //mFastMixerFutex = 0; 2727 state->mColdFutexAddr = &mFastMixerFutex; 2728 state->mColdGen++; 2729 state->mDumpState = &mFastMixerDumpState; 2730#ifdef TEE_SINK 2731 state->mTeeSink = mTeeSink.get(); 2732#endif 2733 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2734 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2735 sq->end(); 2736 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2737 2738 // start the fast mixer 2739 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2740 pid_t tid = mFastMixer->getTid(); 2741 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2742 if (err != 0) { 2743 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2744 kPriorityFastMixer, getpid_cached, tid, err); 2745 } 2746 2747#ifdef AUDIO_WATCHDOG 2748 // create and start the watchdog 2749 mAudioWatchdog = new AudioWatchdog(); 2750 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2751 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2752 tid = mAudioWatchdog->getTid(); 2753 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2754 if (err != 0) { 2755 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2756 kPriorityFastMixer, getpid_cached, tid, err); 2757 } 2758#endif 2759 2760 } else { 2761 mFastMixer = NULL; 2762 } 2763 2764 switch (kUseFastMixer) { 2765 case FastMixer_Never: 2766 case FastMixer_Dynamic: 2767 mNormalSink = mOutputSink; 2768 break; 2769 case FastMixer_Always: 2770 mNormalSink = mPipeSink; 2771 break; 2772 case FastMixer_Static: 2773 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2774 break; 2775 } 2776} 2777 2778AudioFlinger::MixerThread::~MixerThread() 2779{ 2780 if (mFastMixer != NULL) { 2781 FastMixerStateQueue *sq = mFastMixer->sq(); 2782 FastMixerState *state = sq->begin(); 2783 if (state->mCommand == FastMixerState::COLD_IDLE) { 2784 int32_t old = android_atomic_inc(&mFastMixerFutex); 2785 if (old == -1) { 2786 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2787 } 2788 } 2789 state->mCommand = FastMixerState::EXIT; 2790 sq->end(); 2791 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2792 mFastMixer->join(); 2793 // Though the fast mixer thread has exited, it's state queue is still valid. 2794 // We'll use that extract the final state which contains one remaining fast track 2795 // corresponding to our sub-mix. 2796 state = sq->begin(); 2797 ALOG_ASSERT(state->mTrackMask == 1); 2798 FastTrack *fastTrack = &state->mFastTracks[0]; 2799 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2800 delete fastTrack->mBufferProvider; 2801 sq->end(false /*didModify*/); 2802 delete mFastMixer; 2803#ifdef AUDIO_WATCHDOG 2804 if (mAudioWatchdog != 0) { 2805 mAudioWatchdog->requestExit(); 2806 mAudioWatchdog->requestExitAndWait(); 2807 mAudioWatchdog.clear(); 2808 } 2809#endif 2810 } 2811 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2812 delete mAudioMixer; 2813} 2814 2815 2816uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2817{ 2818 if (mFastMixer != NULL) { 2819 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2820 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2821 } 2822 return latency; 2823} 2824 2825 2826void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2827{ 2828 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2829} 2830 2831ssize_t AudioFlinger::MixerThread::threadLoop_write() 2832{ 2833 // FIXME we should only do one push per cycle; confirm this is true 2834 // Start the fast mixer if it's not already running 2835 if (mFastMixer != NULL) { 2836 FastMixerStateQueue *sq = mFastMixer->sq(); 2837 FastMixerState *state = sq->begin(); 2838 if (state->mCommand != FastMixerState::MIX_WRITE && 2839 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2840 if (state->mCommand == FastMixerState::COLD_IDLE) { 2841 int32_t old = android_atomic_inc(&mFastMixerFutex); 2842 if (old == -1) { 2843 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2844 } 2845#ifdef AUDIO_WATCHDOG 2846 if (mAudioWatchdog != 0) { 2847 mAudioWatchdog->resume(); 2848 } 2849#endif 2850 } 2851 state->mCommand = FastMixerState::MIX_WRITE; 2852 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2853 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2854 sq->end(); 2855 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2856 if (kUseFastMixer == FastMixer_Dynamic) { 2857 mNormalSink = mPipeSink; 2858 } 2859 } else { 2860 sq->end(false /*didModify*/); 2861 } 2862 } 2863 return PlaybackThread::threadLoop_write(); 2864} 2865 2866void AudioFlinger::MixerThread::threadLoop_standby() 2867{ 2868 // Idle the fast mixer if it's currently running 2869 if (mFastMixer != NULL) { 2870 FastMixerStateQueue *sq = mFastMixer->sq(); 2871 FastMixerState *state = sq->begin(); 2872 if (!(state->mCommand & FastMixerState::IDLE)) { 2873 state->mCommand = FastMixerState::COLD_IDLE; 2874 state->mColdFutexAddr = &mFastMixerFutex; 2875 state->mColdGen++; 2876 mFastMixerFutex = 0; 2877 sq->end(); 2878 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2879 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2880 if (kUseFastMixer == FastMixer_Dynamic) { 2881 mNormalSink = mOutputSink; 2882 } 2883#ifdef AUDIO_WATCHDOG 2884 if (mAudioWatchdog != 0) { 2885 mAudioWatchdog->pause(); 2886 } 2887#endif 2888 } else { 2889 sq->end(false /*didModify*/); 2890 } 2891 } 2892 PlaybackThread::threadLoop_standby(); 2893} 2894 2895bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2896{ 2897 return false; 2898} 2899 2900bool AudioFlinger::PlaybackThread::shouldStandby_l() 2901{ 2902 return !mStandby; 2903} 2904 2905bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2906{ 2907 Mutex::Autolock _l(mLock); 2908 return waitingAsyncCallback_l(); 2909} 2910 2911// shared by MIXER and DIRECT, overridden by DUPLICATING 2912void AudioFlinger::PlaybackThread::threadLoop_standby() 2913{ 2914 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2915 mOutput->stream->common.standby(&mOutput->stream->common); 2916 if (mUseAsyncWrite != 0) { 2917 // discard any pending drain or write ack by incrementing sequence 2918 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2919 mDrainSequence = (mDrainSequence + 2) & ~1; 2920 ALOG_ASSERT(mCallbackThread != 0); 2921 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2922 mCallbackThread->setDraining(mDrainSequence); 2923 } 2924} 2925 2926void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2927{ 2928 ALOGV("signal playback thread"); 2929 broadcast_l(); 2930} 2931 2932void AudioFlinger::MixerThread::threadLoop_mix() 2933{ 2934 // obtain the presentation timestamp of the next output buffer 2935 int64_t pts; 2936 status_t status = INVALID_OPERATION; 2937 2938 if (mNormalSink != 0) { 2939 status = mNormalSink->getNextWriteTimestamp(&pts); 2940 } else { 2941 status = mOutputSink->getNextWriteTimestamp(&pts); 2942 } 2943 2944 if (status != NO_ERROR) { 2945 pts = AudioBufferProvider::kInvalidPTS; 2946 } 2947 2948 // mix buffers... 2949 mAudioMixer->process(pts); 2950 mCurrentWriteLength = mSinkBufferSize; 2951 // increase sleep time progressively when application underrun condition clears. 2952 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2953 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2954 // such that we would underrun the audio HAL. 2955 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2956 sleepTimeShift--; 2957 } 2958 sleepTime = 0; 2959 standbyTime = systemTime() + standbyDelay; 2960 //TODO: delay standby when effects have a tail 2961} 2962 2963void AudioFlinger::MixerThread::threadLoop_sleepTime() 2964{ 2965 // If no tracks are ready, sleep once for the duration of an output 2966 // buffer size, then write 0s to the output 2967 if (sleepTime == 0) { 2968 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2969 sleepTime = activeSleepTime >> sleepTimeShift; 2970 if (sleepTime < kMinThreadSleepTimeUs) { 2971 sleepTime = kMinThreadSleepTimeUs; 2972 } 2973 // reduce sleep time in case of consecutive application underruns to avoid 2974 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2975 // duration we would end up writing less data than needed by the audio HAL if 2976 // the condition persists. 2977 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2978 sleepTimeShift++; 2979 } 2980 } else { 2981 sleepTime = idleSleepTime; 2982 } 2983 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2984 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 2985 // before effects processing or output. 2986 if (mMixerBufferValid) { 2987 memset(mMixerBuffer, 0, mMixerBufferSize); 2988 } else { 2989 memset(mSinkBuffer, 0, mSinkBufferSize); 2990 } 2991 sleepTime = 0; 2992 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2993 "anticipated start"); 2994 } 2995 // TODO add standby time extension fct of effect tail 2996} 2997 2998// prepareTracks_l() must be called with ThreadBase::mLock held 2999AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3000 Vector< sp<Track> > *tracksToRemove) 3001{ 3002 3003 mixer_state mixerStatus = MIXER_IDLE; 3004 // find out which tracks need to be processed 3005 size_t count = mActiveTracks.size(); 3006 size_t mixedTracks = 0; 3007 size_t tracksWithEffect = 0; 3008 // counts only _active_ fast tracks 3009 size_t fastTracks = 0; 3010 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3011 3012 float masterVolume = mMasterVolume; 3013 bool masterMute = mMasterMute; 3014 3015 if (masterMute) { 3016 masterVolume = 0; 3017 } 3018 // Delegate master volume control to effect in output mix effect chain if needed 3019 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3020 if (chain != 0) { 3021 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3022 chain->setVolume_l(&v, &v); 3023 masterVolume = (float)((v + (1 << 23)) >> 24); 3024 chain.clear(); 3025 } 3026 3027 // prepare a new state to push 3028 FastMixerStateQueue *sq = NULL; 3029 FastMixerState *state = NULL; 3030 bool didModify = false; 3031 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3032 if (mFastMixer != NULL) { 3033 sq = mFastMixer->sq(); 3034 state = sq->begin(); 3035 } 3036 3037 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3038 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3039 3040 for (size_t i=0 ; i<count ; i++) { 3041 const sp<Track> t = mActiveTracks[i].promote(); 3042 if (t == 0) { 3043 continue; 3044 } 3045 3046 // this const just means the local variable doesn't change 3047 Track* const track = t.get(); 3048 3049 // process fast tracks 3050 if (track->isFastTrack()) { 3051 3052 // It's theoretically possible (though unlikely) for a fast track to be created 3053 // and then removed within the same normal mix cycle. This is not a problem, as 3054 // the track never becomes active so it's fast mixer slot is never touched. 3055 // The converse, of removing an (active) track and then creating a new track 3056 // at the identical fast mixer slot within the same normal mix cycle, 3057 // is impossible because the slot isn't marked available until the end of each cycle. 3058 int j = track->mFastIndex; 3059 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3060 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3061 FastTrack *fastTrack = &state->mFastTracks[j]; 3062 3063 // Determine whether the track is currently in underrun condition, 3064 // and whether it had a recent underrun. 3065 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3066 FastTrackUnderruns underruns = ftDump->mUnderruns; 3067 uint32_t recentFull = (underruns.mBitFields.mFull - 3068 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3069 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3070 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3071 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3072 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3073 uint32_t recentUnderruns = recentPartial + recentEmpty; 3074 track->mObservedUnderruns = underruns; 3075 // don't count underruns that occur while stopping or pausing 3076 // or stopped which can occur when flush() is called while active 3077 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3078 recentUnderruns > 0) { 3079 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3080 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3081 } 3082 3083 // This is similar to the state machine for normal tracks, 3084 // with a few modifications for fast tracks. 3085 bool isActive = true; 3086 switch (track->mState) { 3087 case TrackBase::STOPPING_1: 3088 // track stays active in STOPPING_1 state until first underrun 3089 if (recentUnderruns > 0 || track->isTerminated()) { 3090 track->mState = TrackBase::STOPPING_2; 3091 } 3092 break; 3093 case TrackBase::PAUSING: 3094 // ramp down is not yet implemented 3095 track->setPaused(); 3096 break; 3097 case TrackBase::RESUMING: 3098 // ramp up is not yet implemented 3099 track->mState = TrackBase::ACTIVE; 3100 break; 3101 case TrackBase::ACTIVE: 3102 if (recentFull > 0 || recentPartial > 0) { 3103 // track has provided at least some frames recently: reset retry count 3104 track->mRetryCount = kMaxTrackRetries; 3105 } 3106 if (recentUnderruns == 0) { 3107 // no recent underruns: stay active 3108 break; 3109 } 3110 // there has recently been an underrun of some kind 3111 if (track->sharedBuffer() == 0) { 3112 // were any of the recent underruns "empty" (no frames available)? 3113 if (recentEmpty == 0) { 3114 // no, then ignore the partial underruns as they are allowed indefinitely 3115 break; 3116 } 3117 // there has recently been an "empty" underrun: decrement the retry counter 3118 if (--(track->mRetryCount) > 0) { 3119 break; 3120 } 3121 // indicate to client process that the track was disabled because of underrun; 3122 // it will then automatically call start() when data is available 3123 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3124 // remove from active list, but state remains ACTIVE [confusing but true] 3125 isActive = false; 3126 break; 3127 } 3128 // fall through 3129 case TrackBase::STOPPING_2: 3130 case TrackBase::PAUSED: 3131 case TrackBase::STOPPED: 3132 case TrackBase::FLUSHED: // flush() while active 3133 // Check for presentation complete if track is inactive 3134 // We have consumed all the buffers of this track. 3135 // This would be incomplete if we auto-paused on underrun 3136 { 3137 size_t audioHALFrames = 3138 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3139 size_t framesWritten = mBytesWritten / mFrameSize; 3140 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3141 // track stays in active list until presentation is complete 3142 break; 3143 } 3144 } 3145 if (track->isStopping_2()) { 3146 track->mState = TrackBase::STOPPED; 3147 } 3148 if (track->isStopped()) { 3149 // Can't reset directly, as fast mixer is still polling this track 3150 // track->reset(); 3151 // So instead mark this track as needing to be reset after push with ack 3152 resetMask |= 1 << i; 3153 } 3154 isActive = false; 3155 break; 3156 case TrackBase::IDLE: 3157 default: 3158 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3159 } 3160 3161 if (isActive) { 3162 // was it previously inactive? 3163 if (!(state->mTrackMask & (1 << j))) { 3164 ExtendedAudioBufferProvider *eabp = track; 3165 VolumeProvider *vp = track; 3166 fastTrack->mBufferProvider = eabp; 3167 fastTrack->mVolumeProvider = vp; 3168 fastTrack->mChannelMask = track->mChannelMask; 3169 fastTrack->mFormat = track->mFormat; 3170 fastTrack->mGeneration++; 3171 state->mTrackMask |= 1 << j; 3172 didModify = true; 3173 // no acknowledgement required for newly active tracks 3174 } 3175 // cache the combined master volume and stream type volume for fast mixer; this 3176 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3177 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3178 ++fastTracks; 3179 } else { 3180 // was it previously active? 3181 if (state->mTrackMask & (1 << j)) { 3182 fastTrack->mBufferProvider = NULL; 3183 fastTrack->mGeneration++; 3184 state->mTrackMask &= ~(1 << j); 3185 didModify = true; 3186 // If any fast tracks were removed, we must wait for acknowledgement 3187 // because we're about to decrement the last sp<> on those tracks. 3188 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3189 } else { 3190 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3191 } 3192 tracksToRemove->add(track); 3193 // Avoids a misleading display in dumpsys 3194 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3195 } 3196 continue; 3197 } 3198 3199 { // local variable scope to avoid goto warning 3200 3201 audio_track_cblk_t* cblk = track->cblk(); 3202 3203 // The first time a track is added we wait 3204 // for all its buffers to be filled before processing it 3205 int name = track->name(); 3206 // make sure that we have enough frames to mix one full buffer. 3207 // enforce this condition only once to enable draining the buffer in case the client 3208 // app does not call stop() and relies on underrun to stop: 3209 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3210 // during last round 3211 size_t desiredFrames; 3212 uint32_t sr = track->sampleRate(); 3213 if (sr == mSampleRate) { 3214 desiredFrames = mNormalFrameCount; 3215 } else { 3216 // +1 for rounding and +1 for additional sample needed for interpolation 3217 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3218 // add frames already consumed but not yet released by the resampler 3219 // because mAudioTrackServerProxy->framesReady() will include these frames 3220 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3221#if 0 3222 // the minimum track buffer size is normally twice the number of frames necessary 3223 // to fill one buffer and the resampler should not leave more than one buffer worth 3224 // of unreleased frames after each pass, but just in case... 3225 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3226#endif 3227 } 3228 uint32_t minFrames = 1; 3229 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3230 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3231 minFrames = desiredFrames; 3232 } 3233 3234 size_t framesReady = track->framesReady(); 3235 if ((framesReady >= minFrames) && track->isReady() && 3236 !track->isPaused() && !track->isTerminated()) 3237 { 3238 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3239 3240 mixedTracks++; 3241 3242 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3243 // there is an effect chain connected to the track 3244 chain.clear(); 3245 if (track->mainBuffer() != mSinkBuffer && 3246 track->mainBuffer() != mMixerBuffer) { 3247 if (mEffectBufferEnabled) { 3248 mEffectBufferValid = true; // Later can set directly. 3249 } 3250 chain = getEffectChain_l(track->sessionId()); 3251 // Delegate volume control to effect in track effect chain if needed 3252 if (chain != 0) { 3253 tracksWithEffect++; 3254 } else { 3255 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3256 "session %d", 3257 name, track->sessionId()); 3258 } 3259 } 3260 3261 3262 int param = AudioMixer::VOLUME; 3263 if (track->mFillingUpStatus == Track::FS_FILLED) { 3264 // no ramp for the first volume setting 3265 track->mFillingUpStatus = Track::FS_ACTIVE; 3266 if (track->mState == TrackBase::RESUMING) { 3267 track->mState = TrackBase::ACTIVE; 3268 param = AudioMixer::RAMP_VOLUME; 3269 } 3270 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3271 // FIXME should not make a decision based on mServer 3272 } else if (cblk->mServer != 0) { 3273 // If the track is stopped before the first frame was mixed, 3274 // do not apply ramp 3275 param = AudioMixer::RAMP_VOLUME; 3276 } 3277 3278 // compute volume for this track 3279 uint32_t vl, vr, va; 3280 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3281 vl = vr = va = 0; 3282 if (track->isPausing()) { 3283 track->setPaused(); 3284 } 3285 } else { 3286 3287 // read original volumes with volume control 3288 float typeVolume = mStreamTypes[track->streamType()].volume; 3289 float v = masterVolume * typeVolume; 3290 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3291 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3292 float vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3293 float vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3294 // track volumes come from shared memory, so can't be trusted and must be clamped 3295 if (vlf > GAIN_FLOAT_UNITY) { 3296 ALOGV("Track left volume out of range: %.3g", vlf); 3297 vlf = GAIN_FLOAT_UNITY; 3298 } 3299 if (vrf > GAIN_FLOAT_UNITY) { 3300 ALOGV("Track right volume out of range: %.3g", vrf); 3301 vrf = GAIN_FLOAT_UNITY; 3302 } 3303 // now apply the master volume and stream type volume 3304 // FIXME we're losing the wonderful dynamic range in the minifloat representation 3305 float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT); 3306 vl = (uint32_t) (v8_24 * vlf); 3307 vr = (uint32_t) (v8_24 * vrf); 3308 // assuming master volume and stream type volume each go up to 1.0, 3309 // vl and vr are now in 8.24 format 3310 3311 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3312 // send level comes from shared memory and so may be corrupt 3313 if (sendLevel > MAX_GAIN_INT) { 3314 ALOGV("Track send level out of range: %04X", sendLevel); 3315 sendLevel = MAX_GAIN_INT; 3316 } 3317 va = (uint32_t)(v * sendLevel); 3318 } 3319 3320 // Delegate volume control to effect in track effect chain if needed 3321 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3322 // Do not ramp volume if volume is controlled by effect 3323 param = AudioMixer::VOLUME; 3324 track->mHasVolumeController = true; 3325 } else { 3326 // force no volume ramp when volume controller was just disabled or removed 3327 // from effect chain to avoid volume spike 3328 if (track->mHasVolumeController) { 3329 param = AudioMixer::VOLUME; 3330 } 3331 track->mHasVolumeController = false; 3332 } 3333 3334 // FIXME Use float 3335 // Convert volumes from 8.24 to 4.12 format 3336 // This additional clamping is needed in case chain->setVolume_l() overshot 3337 vl = (vl + (1 << 11)) >> 12; 3338 if (vl > MAX_GAIN_INT) { 3339 vl = MAX_GAIN_INT; 3340 } 3341 vr = (vr + (1 << 11)) >> 12; 3342 if (vr > MAX_GAIN_INT) { 3343 vr = MAX_GAIN_INT; 3344 } 3345 3346 if (va > MAX_GAIN_INT) { 3347 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3348 } 3349 3350 // XXX: these things DON'T need to be done each time 3351 mAudioMixer->setBufferProvider(name, track); 3352 mAudioMixer->enable(name); 3353 3354 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3355 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3356 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3357 mAudioMixer->setParameter( 3358 name, 3359 AudioMixer::TRACK, 3360 AudioMixer::FORMAT, (void *)track->format()); 3361 mAudioMixer->setParameter( 3362 name, 3363 AudioMixer::TRACK, 3364 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3365 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3366 uint32_t maxSampleRate = mSampleRate * 2; 3367 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3368 if (reqSampleRate == 0) { 3369 reqSampleRate = mSampleRate; 3370 } else if (reqSampleRate > maxSampleRate) { 3371 reqSampleRate = maxSampleRate; 3372 } 3373 mAudioMixer->setParameter( 3374 name, 3375 AudioMixer::RESAMPLE, 3376 AudioMixer::SAMPLE_RATE, 3377 (void *)(uintptr_t)reqSampleRate); 3378 /* 3379 * Select the appropriate output buffer for the track. 3380 * 3381 * Tracks with effects go into their own effects chain buffer 3382 * and from there into either mEffectBuffer or mSinkBuffer. 3383 * 3384 * Other tracks can use mMixerBuffer for higher precision 3385 * channel accumulation. If this buffer is enabled 3386 * (mMixerBufferEnabled true), then selected tracks will accumulate 3387 * into it. 3388 * 3389 */ 3390 if (mMixerBufferEnabled 3391 && (track->mainBuffer() == mSinkBuffer 3392 || track->mainBuffer() == mMixerBuffer)) { 3393 mAudioMixer->setParameter( 3394 name, 3395 AudioMixer::TRACK, 3396 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3397 mAudioMixer->setParameter( 3398 name, 3399 AudioMixer::TRACK, 3400 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3401 // TODO: override track->mainBuffer()? 3402 mMixerBufferValid = true; 3403 } else { 3404 mAudioMixer->setParameter( 3405 name, 3406 AudioMixer::TRACK, 3407 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3408 mAudioMixer->setParameter( 3409 name, 3410 AudioMixer::TRACK, 3411 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3412 } 3413 mAudioMixer->setParameter( 3414 name, 3415 AudioMixer::TRACK, 3416 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3417 3418 // reset retry count 3419 track->mRetryCount = kMaxTrackRetries; 3420 3421 // If one track is ready, set the mixer ready if: 3422 // - the mixer was not ready during previous round OR 3423 // - no other track is not ready 3424 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3425 mixerStatus != MIXER_TRACKS_ENABLED) { 3426 mixerStatus = MIXER_TRACKS_READY; 3427 } 3428 } else { 3429 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3430 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3431 } 3432 // clear effect chain input buffer if an active track underruns to avoid sending 3433 // previous audio buffer again to effects 3434 chain = getEffectChain_l(track->sessionId()); 3435 if (chain != 0) { 3436 chain->clearInputBuffer(); 3437 } 3438 3439 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3440 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3441 track->isStopped() || track->isPaused()) { 3442 // We have consumed all the buffers of this track. 3443 // Remove it from the list of active tracks. 3444 // TODO: use actual buffer filling status instead of latency when available from 3445 // audio HAL 3446 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3447 size_t framesWritten = mBytesWritten / mFrameSize; 3448 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3449 if (track->isStopped()) { 3450 track->reset(); 3451 } 3452 tracksToRemove->add(track); 3453 } 3454 } else { 3455 // No buffers for this track. Give it a few chances to 3456 // fill a buffer, then remove it from active list. 3457 if (--(track->mRetryCount) <= 0) { 3458 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3459 tracksToRemove->add(track); 3460 // indicate to client process that the track was disabled because of underrun; 3461 // it will then automatically call start() when data is available 3462 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3463 // If one track is not ready, mark the mixer also not ready if: 3464 // - the mixer was ready during previous round OR 3465 // - no other track is ready 3466 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3467 mixerStatus != MIXER_TRACKS_READY) { 3468 mixerStatus = MIXER_TRACKS_ENABLED; 3469 } 3470 } 3471 mAudioMixer->disable(name); 3472 } 3473 3474 } // local variable scope to avoid goto warning 3475track_is_ready: ; 3476 3477 } 3478 3479 // Push the new FastMixer state if necessary 3480 bool pauseAudioWatchdog = false; 3481 if (didModify) { 3482 state->mFastTracksGen++; 3483 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3484 if (kUseFastMixer == FastMixer_Dynamic && 3485 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3486 state->mCommand = FastMixerState::COLD_IDLE; 3487 state->mColdFutexAddr = &mFastMixerFutex; 3488 state->mColdGen++; 3489 mFastMixerFutex = 0; 3490 if (kUseFastMixer == FastMixer_Dynamic) { 3491 mNormalSink = mOutputSink; 3492 } 3493 // If we go into cold idle, need to wait for acknowledgement 3494 // so that fast mixer stops doing I/O. 3495 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3496 pauseAudioWatchdog = true; 3497 } 3498 } 3499 if (sq != NULL) { 3500 sq->end(didModify); 3501 sq->push(block); 3502 } 3503#ifdef AUDIO_WATCHDOG 3504 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3505 mAudioWatchdog->pause(); 3506 } 3507#endif 3508 3509 // Now perform the deferred reset on fast tracks that have stopped 3510 while (resetMask != 0) { 3511 size_t i = __builtin_ctz(resetMask); 3512 ALOG_ASSERT(i < count); 3513 resetMask &= ~(1 << i); 3514 sp<Track> t = mActiveTracks[i].promote(); 3515 if (t == 0) { 3516 continue; 3517 } 3518 Track* track = t.get(); 3519 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3520 track->reset(); 3521 } 3522 3523 // remove all the tracks that need to be... 3524 removeTracks_l(*tracksToRemove); 3525 3526 // sink or mix buffer must be cleared if all tracks are connected to an 3527 // effect chain as in this case the mixer will not write to the sink or mix buffer 3528 // and track effects will accumulate into it 3529 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3530 (mixedTracks == 0 && fastTracks > 0))) { 3531 // FIXME as a performance optimization, should remember previous zero status 3532 if (mMixerBufferValid) { 3533 memset(mMixerBuffer, 0, mMixerBufferSize); 3534 // TODO: In testing, mSinkBuffer below need not be cleared because 3535 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3536 // after mixing. 3537 // 3538 // To enforce this guarantee: 3539 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3540 // (mixedTracks == 0 && fastTracks > 0)) 3541 // must imply MIXER_TRACKS_READY. 3542 // Later, we may clear buffers regardless, and skip much of this logic. 3543 } 3544 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3545 if (mEffectBufferValid) { 3546 memset(mEffectBuffer, 0, mEffectBufferSize); 3547 } 3548 // FIXME as a performance optimization, should remember previous zero status 3549 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3550 } 3551 3552 // if any fast tracks, then status is ready 3553 mMixerStatusIgnoringFastTracks = mixerStatus; 3554 if (fastTracks > 0) { 3555 mixerStatus = MIXER_TRACKS_READY; 3556 } 3557 return mixerStatus; 3558} 3559 3560// getTrackName_l() must be called with ThreadBase::mLock held 3561int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3562 audio_format_t format, int sessionId) 3563{ 3564 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3565} 3566 3567// deleteTrackName_l() must be called with ThreadBase::mLock held 3568void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3569{ 3570 ALOGV("remove track (%d) and delete from mixer", name); 3571 mAudioMixer->deleteTrackName(name); 3572} 3573 3574// checkForNewParameter_l() must be called with ThreadBase::mLock held 3575bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3576 status_t& status) 3577{ 3578 bool reconfig = false; 3579 3580 status = NO_ERROR; 3581 3582 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3583 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3584 if (mFastMixer != NULL) { 3585 FastMixerStateQueue *sq = mFastMixer->sq(); 3586 FastMixerState *state = sq->begin(); 3587 if (!(state->mCommand & FastMixerState::IDLE)) { 3588 previousCommand = state->mCommand; 3589 state->mCommand = FastMixerState::HOT_IDLE; 3590 sq->end(); 3591 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3592 } else { 3593 sq->end(false /*didModify*/); 3594 } 3595 } 3596 3597 AudioParameter param = AudioParameter(keyValuePair); 3598 int value; 3599 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3600 reconfig = true; 3601 } 3602 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3603 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3604 status = BAD_VALUE; 3605 } else { 3606 // no need to save value, since it's constant 3607 reconfig = true; 3608 } 3609 } 3610 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3611 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3612 status = BAD_VALUE; 3613 } else { 3614 // no need to save value, since it's constant 3615 reconfig = true; 3616 } 3617 } 3618 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3619 // do not accept frame count changes if tracks are open as the track buffer 3620 // size depends on frame count and correct behavior would not be guaranteed 3621 // if frame count is changed after track creation 3622 if (!mTracks.isEmpty()) { 3623 status = INVALID_OPERATION; 3624 } else { 3625 reconfig = true; 3626 } 3627 } 3628 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3629#ifdef ADD_BATTERY_DATA 3630 // when changing the audio output device, call addBatteryData to notify 3631 // the change 3632 if (mOutDevice != value) { 3633 uint32_t params = 0; 3634 // check whether speaker is on 3635 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3636 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3637 } 3638 3639 audio_devices_t deviceWithoutSpeaker 3640 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3641 // check if any other device (except speaker) is on 3642 if (value & deviceWithoutSpeaker ) { 3643 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3644 } 3645 3646 if (params != 0) { 3647 addBatteryData(params); 3648 } 3649 } 3650#endif 3651 3652 // forward device change to effects that have requested to be 3653 // aware of attached audio device. 3654 if (value != AUDIO_DEVICE_NONE) { 3655 mOutDevice = value; 3656 for (size_t i = 0; i < mEffectChains.size(); i++) { 3657 mEffectChains[i]->setDevice_l(mOutDevice); 3658 } 3659 } 3660 } 3661 3662 if (status == NO_ERROR) { 3663 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3664 keyValuePair.string()); 3665 if (!mStandby && status == INVALID_OPERATION) { 3666 mOutput->stream->common.standby(&mOutput->stream->common); 3667 mStandby = true; 3668 mBytesWritten = 0; 3669 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3670 keyValuePair.string()); 3671 } 3672 if (status == NO_ERROR && reconfig) { 3673 readOutputParameters_l(); 3674 delete mAudioMixer; 3675 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3676 for (size_t i = 0; i < mTracks.size() ; i++) { 3677 int name = getTrackName_l(mTracks[i]->mChannelMask, 3678 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3679 if (name < 0) { 3680 break; 3681 } 3682 mTracks[i]->mName = name; 3683 } 3684 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3685 } 3686 } 3687 3688 if (!(previousCommand & FastMixerState::IDLE)) { 3689 ALOG_ASSERT(mFastMixer != NULL); 3690 FastMixerStateQueue *sq = mFastMixer->sq(); 3691 FastMixerState *state = sq->begin(); 3692 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3693 state->mCommand = previousCommand; 3694 sq->end(); 3695 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3696 } 3697 3698 return reconfig; 3699} 3700 3701 3702void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3703{ 3704 const size_t SIZE = 256; 3705 char buffer[SIZE]; 3706 String8 result; 3707 3708 PlaybackThread::dumpInternals(fd, args); 3709 3710 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3711 3712 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3713 const FastMixerDumpState copy(mFastMixerDumpState); 3714 copy.dump(fd); 3715 3716#ifdef STATE_QUEUE_DUMP 3717 // Similar for state queue 3718 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3719 observerCopy.dump(fd); 3720 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3721 mutatorCopy.dump(fd); 3722#endif 3723 3724#ifdef TEE_SINK 3725 // Write the tee output to a .wav file 3726 dumpTee(fd, mTeeSource, mId); 3727#endif 3728 3729#ifdef AUDIO_WATCHDOG 3730 if (mAudioWatchdog != 0) { 3731 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3732 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3733 wdCopy.dump(fd); 3734 } 3735#endif 3736} 3737 3738uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3739{ 3740 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3741} 3742 3743uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3744{ 3745 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3746} 3747 3748void AudioFlinger::MixerThread::cacheParameters_l() 3749{ 3750 PlaybackThread::cacheParameters_l(); 3751 3752 // FIXME: Relaxed timing because of a certain device that can't meet latency 3753 // Should be reduced to 2x after the vendor fixes the driver issue 3754 // increase threshold again due to low power audio mode. The way this warning 3755 // threshold is calculated and its usefulness should be reconsidered anyway. 3756 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3757} 3758 3759// ---------------------------------------------------------------------------- 3760 3761AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3762 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3763 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3764 // mLeftVolFloat, mRightVolFloat 3765{ 3766} 3767 3768AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3769 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3770 ThreadBase::type_t type) 3771 : PlaybackThread(audioFlinger, output, id, device, type) 3772 // mLeftVolFloat, mRightVolFloat 3773{ 3774} 3775 3776AudioFlinger::DirectOutputThread::~DirectOutputThread() 3777{ 3778} 3779 3780void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3781{ 3782 audio_track_cblk_t* cblk = track->cblk(); 3783 float left, right; 3784 3785 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3786 left = right = 0; 3787 } else { 3788 float typeVolume = mStreamTypes[track->streamType()].volume; 3789 float v = mMasterVolume * typeVolume; 3790 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3791 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3792 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3793 if (left > GAIN_FLOAT_UNITY) { 3794 left = GAIN_FLOAT_UNITY; 3795 } 3796 left *= v; 3797 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3798 if (right > GAIN_FLOAT_UNITY) { 3799 right = GAIN_FLOAT_UNITY; 3800 } 3801 right *= v; 3802 } 3803 3804 if (lastTrack) { 3805 if (left != mLeftVolFloat || right != mRightVolFloat) { 3806 mLeftVolFloat = left; 3807 mRightVolFloat = right; 3808 3809 // Convert volumes from float to 8.24 3810 uint32_t vl = (uint32_t)(left * (1 << 24)); 3811 uint32_t vr = (uint32_t)(right * (1 << 24)); 3812 3813 // Delegate volume control to effect in track effect chain if needed 3814 // only one effect chain can be present on DirectOutputThread, so if 3815 // there is one, the track is connected to it 3816 if (!mEffectChains.isEmpty()) { 3817 mEffectChains[0]->setVolume_l(&vl, &vr); 3818 left = (float)vl / (1 << 24); 3819 right = (float)vr / (1 << 24); 3820 } 3821 if (mOutput->stream->set_volume) { 3822 mOutput->stream->set_volume(mOutput->stream, left, right); 3823 } 3824 } 3825 } 3826} 3827 3828 3829AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3830 Vector< sp<Track> > *tracksToRemove 3831) 3832{ 3833 size_t count = mActiveTracks.size(); 3834 mixer_state mixerStatus = MIXER_IDLE; 3835 3836 // find out which tracks need to be processed 3837 for (size_t i = 0; i < count; i++) { 3838 sp<Track> t = mActiveTracks[i].promote(); 3839 // The track died recently 3840 if (t == 0) { 3841 continue; 3842 } 3843 3844 Track* const track = t.get(); 3845 audio_track_cblk_t* cblk = track->cblk(); 3846 // Only consider last track started for volume and mixer state control. 3847 // In theory an older track could underrun and restart after the new one starts 3848 // but as we only care about the transition phase between two tracks on a 3849 // direct output, it is not a problem to ignore the underrun case. 3850 sp<Track> l = mLatestActiveTrack.promote(); 3851 bool last = l.get() == track; 3852 3853 // The first time a track is added we wait 3854 // for all its buffers to be filled before processing it 3855 uint32_t minFrames; 3856 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3857 minFrames = mNormalFrameCount; 3858 } else { 3859 minFrames = 1; 3860 } 3861 3862 if ((track->framesReady() >= minFrames) && track->isReady() && 3863 !track->isPaused() && !track->isTerminated()) 3864 { 3865 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3866 3867 if (track->mFillingUpStatus == Track::FS_FILLED) { 3868 track->mFillingUpStatus = Track::FS_ACTIVE; 3869 // make sure processVolume_l() will apply new volume even if 0 3870 mLeftVolFloat = mRightVolFloat = -1.0; 3871 if (track->mState == TrackBase::RESUMING) { 3872 track->mState = TrackBase::ACTIVE; 3873 } 3874 } 3875 3876 // compute volume for this track 3877 processVolume_l(track, last); 3878 if (last) { 3879 // reset retry count 3880 track->mRetryCount = kMaxTrackRetriesDirect; 3881 mActiveTrack = t; 3882 mixerStatus = MIXER_TRACKS_READY; 3883 } 3884 } else { 3885 // clear effect chain input buffer if the last active track started underruns 3886 // to avoid sending previous audio buffer again to effects 3887 if (!mEffectChains.isEmpty() && last) { 3888 mEffectChains[0]->clearInputBuffer(); 3889 } 3890 3891 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3892 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3893 track->isStopped() || track->isPaused()) { 3894 // We have consumed all the buffers of this track. 3895 // Remove it from the list of active tracks. 3896 // TODO: implement behavior for compressed audio 3897 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3898 size_t framesWritten = mBytesWritten / mFrameSize; 3899 if (mStandby || !last || 3900 track->presentationComplete(framesWritten, audioHALFrames)) { 3901 if (track->isStopped()) { 3902 track->reset(); 3903 } 3904 tracksToRemove->add(track); 3905 } 3906 } else { 3907 // No buffers for this track. Give it a few chances to 3908 // fill a buffer, then remove it from active list. 3909 // Only consider last track started for mixer state control 3910 if (--(track->mRetryCount) <= 0) { 3911 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3912 tracksToRemove->add(track); 3913 // indicate to client process that the track was disabled because of underrun; 3914 // it will then automatically call start() when data is available 3915 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3916 } else if (last) { 3917 mixerStatus = MIXER_TRACKS_ENABLED; 3918 } 3919 } 3920 } 3921 } 3922 3923 // remove all the tracks that need to be... 3924 removeTracks_l(*tracksToRemove); 3925 3926 return mixerStatus; 3927} 3928 3929void AudioFlinger::DirectOutputThread::threadLoop_mix() 3930{ 3931 size_t frameCount = mFrameCount; 3932 int8_t *curBuf = (int8_t *)mSinkBuffer; 3933 // output audio to hardware 3934 while (frameCount) { 3935 AudioBufferProvider::Buffer buffer; 3936 buffer.frameCount = frameCount; 3937 mActiveTrack->getNextBuffer(&buffer); 3938 if (buffer.raw == NULL) { 3939 memset(curBuf, 0, frameCount * mFrameSize); 3940 break; 3941 } 3942 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3943 frameCount -= buffer.frameCount; 3944 curBuf += buffer.frameCount * mFrameSize; 3945 mActiveTrack->releaseBuffer(&buffer); 3946 } 3947 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3948 sleepTime = 0; 3949 standbyTime = systemTime() + standbyDelay; 3950 mActiveTrack.clear(); 3951} 3952 3953void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3954{ 3955 if (sleepTime == 0) { 3956 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3957 sleepTime = activeSleepTime; 3958 } else { 3959 sleepTime = idleSleepTime; 3960 } 3961 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3962 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3963 sleepTime = 0; 3964 } 3965} 3966 3967// getTrackName_l() must be called with ThreadBase::mLock held 3968int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3969 audio_format_t format __unused, int sessionId __unused) 3970{ 3971 return 0; 3972} 3973 3974// deleteTrackName_l() must be called with ThreadBase::mLock held 3975void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3976{ 3977} 3978 3979// checkForNewParameter_l() must be called with ThreadBase::mLock held 3980bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 3981 status_t& status) 3982{ 3983 bool reconfig = false; 3984 3985 status = NO_ERROR; 3986 3987 AudioParameter param = AudioParameter(keyValuePair); 3988 int value; 3989 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3990 // forward device change to effects that have requested to be 3991 // aware of attached audio device. 3992 if (value != AUDIO_DEVICE_NONE) { 3993 mOutDevice = value; 3994 for (size_t i = 0; i < mEffectChains.size(); i++) { 3995 mEffectChains[i]->setDevice_l(mOutDevice); 3996 } 3997 } 3998 } 3999 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4000 // do not accept frame count changes if tracks are open as the track buffer 4001 // size depends on frame count and correct behavior would not be garantied 4002 // if frame count is changed after track creation 4003 if (!mTracks.isEmpty()) { 4004 status = INVALID_OPERATION; 4005 } else { 4006 reconfig = true; 4007 } 4008 } 4009 if (status == NO_ERROR) { 4010 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4011 keyValuePair.string()); 4012 if (!mStandby && status == INVALID_OPERATION) { 4013 mOutput->stream->common.standby(&mOutput->stream->common); 4014 mStandby = true; 4015 mBytesWritten = 0; 4016 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4017 keyValuePair.string()); 4018 } 4019 if (status == NO_ERROR && reconfig) { 4020 readOutputParameters_l(); 4021 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4022 } 4023 } 4024 4025 return reconfig; 4026} 4027 4028uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4029{ 4030 uint32_t time; 4031 if (audio_is_linear_pcm(mFormat)) { 4032 time = PlaybackThread::activeSleepTimeUs(); 4033 } else { 4034 time = 10000; 4035 } 4036 return time; 4037} 4038 4039uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4040{ 4041 uint32_t time; 4042 if (audio_is_linear_pcm(mFormat)) { 4043 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4044 } else { 4045 time = 10000; 4046 } 4047 return time; 4048} 4049 4050uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4051{ 4052 uint32_t time; 4053 if (audio_is_linear_pcm(mFormat)) { 4054 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4055 } else { 4056 time = 10000; 4057 } 4058 return time; 4059} 4060 4061void AudioFlinger::DirectOutputThread::cacheParameters_l() 4062{ 4063 PlaybackThread::cacheParameters_l(); 4064 4065 // use shorter standby delay as on normal output to release 4066 // hardware resources as soon as possible 4067 if (audio_is_linear_pcm(mFormat)) { 4068 standbyDelay = microseconds(activeSleepTime*2); 4069 } else { 4070 standbyDelay = kOffloadStandbyDelayNs; 4071 } 4072} 4073 4074// ---------------------------------------------------------------------------- 4075 4076AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4077 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4078 : Thread(false /*canCallJava*/), 4079 mPlaybackThread(playbackThread), 4080 mWriteAckSequence(0), 4081 mDrainSequence(0) 4082{ 4083} 4084 4085AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4086{ 4087} 4088 4089void AudioFlinger::AsyncCallbackThread::onFirstRef() 4090{ 4091 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4092} 4093 4094bool AudioFlinger::AsyncCallbackThread::threadLoop() 4095{ 4096 while (!exitPending()) { 4097 uint32_t writeAckSequence; 4098 uint32_t drainSequence; 4099 4100 { 4101 Mutex::Autolock _l(mLock); 4102 while (!((mWriteAckSequence & 1) || 4103 (mDrainSequence & 1) || 4104 exitPending())) { 4105 mWaitWorkCV.wait(mLock); 4106 } 4107 4108 if (exitPending()) { 4109 break; 4110 } 4111 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4112 mWriteAckSequence, mDrainSequence); 4113 writeAckSequence = mWriteAckSequence; 4114 mWriteAckSequence &= ~1; 4115 drainSequence = mDrainSequence; 4116 mDrainSequence &= ~1; 4117 } 4118 { 4119 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4120 if (playbackThread != 0) { 4121 if (writeAckSequence & 1) { 4122 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4123 } 4124 if (drainSequence & 1) { 4125 playbackThread->resetDraining(drainSequence >> 1); 4126 } 4127 } 4128 } 4129 } 4130 return false; 4131} 4132 4133void AudioFlinger::AsyncCallbackThread::exit() 4134{ 4135 ALOGV("AsyncCallbackThread::exit"); 4136 Mutex::Autolock _l(mLock); 4137 requestExit(); 4138 mWaitWorkCV.broadcast(); 4139} 4140 4141void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4142{ 4143 Mutex::Autolock _l(mLock); 4144 // bit 0 is cleared 4145 mWriteAckSequence = sequence << 1; 4146} 4147 4148void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4149{ 4150 Mutex::Autolock _l(mLock); 4151 // ignore unexpected callbacks 4152 if (mWriteAckSequence & 2) { 4153 mWriteAckSequence |= 1; 4154 mWaitWorkCV.signal(); 4155 } 4156} 4157 4158void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4159{ 4160 Mutex::Autolock _l(mLock); 4161 // bit 0 is cleared 4162 mDrainSequence = sequence << 1; 4163} 4164 4165void AudioFlinger::AsyncCallbackThread::resetDraining() 4166{ 4167 Mutex::Autolock _l(mLock); 4168 // ignore unexpected callbacks 4169 if (mDrainSequence & 2) { 4170 mDrainSequence |= 1; 4171 mWaitWorkCV.signal(); 4172 } 4173} 4174 4175 4176// ---------------------------------------------------------------------------- 4177AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4178 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4179 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4180 mHwPaused(false), 4181 mFlushPending(false), 4182 mPausedBytesRemaining(0) 4183{ 4184 //FIXME: mStandby should be set to true by ThreadBase constructor 4185 mStandby = true; 4186} 4187 4188void AudioFlinger::OffloadThread::threadLoop_exit() 4189{ 4190 if (mFlushPending || mHwPaused) { 4191 // If a flush is pending or track was paused, just discard buffered data 4192 flushHw_l(); 4193 } else { 4194 mMixerStatus = MIXER_DRAIN_ALL; 4195 threadLoop_drain(); 4196 } 4197 if (mUseAsyncWrite) { 4198 ALOG_ASSERT(mCallbackThread != 0); 4199 mCallbackThread->exit(); 4200 } 4201 PlaybackThread::threadLoop_exit(); 4202} 4203 4204AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4205 Vector< sp<Track> > *tracksToRemove 4206) 4207{ 4208 size_t count = mActiveTracks.size(); 4209 4210 mixer_state mixerStatus = MIXER_IDLE; 4211 bool doHwPause = false; 4212 bool doHwResume = false; 4213 4214 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4215 4216 // find out which tracks need to be processed 4217 for (size_t i = 0; i < count; i++) { 4218 sp<Track> t = mActiveTracks[i].promote(); 4219 // The track died recently 4220 if (t == 0) { 4221 continue; 4222 } 4223 Track* const track = t.get(); 4224 audio_track_cblk_t* cblk = track->cblk(); 4225 // Only consider last track started for volume and mixer state control. 4226 // In theory an older track could underrun and restart after the new one starts 4227 // but as we only care about the transition phase between two tracks on a 4228 // direct output, it is not a problem to ignore the underrun case. 4229 sp<Track> l = mLatestActiveTrack.promote(); 4230 bool last = l.get() == track; 4231 4232 if (track->isInvalid()) { 4233 ALOGW("An invalidated track shouldn't be in active list"); 4234 tracksToRemove->add(track); 4235 continue; 4236 } 4237 4238 if (track->mState == TrackBase::IDLE) { 4239 ALOGW("An idle track shouldn't be in active list"); 4240 continue; 4241 } 4242 4243 if (track->isPausing()) { 4244 track->setPaused(); 4245 if (last) { 4246 if (!mHwPaused) { 4247 doHwPause = true; 4248 mHwPaused = true; 4249 } 4250 // If we were part way through writing the mixbuffer to 4251 // the HAL we must save this until we resume 4252 // BUG - this will be wrong if a different track is made active, 4253 // in that case we want to discard the pending data in the 4254 // mixbuffer and tell the client to present it again when the 4255 // track is resumed 4256 mPausedWriteLength = mCurrentWriteLength; 4257 mPausedBytesRemaining = mBytesRemaining; 4258 mBytesRemaining = 0; // stop writing 4259 } 4260 tracksToRemove->add(track); 4261 } else if (track->isFlushPending()) { 4262 track->flushAck(); 4263 if (last) { 4264 mFlushPending = true; 4265 } 4266 } else if (track->isResumePending()){ 4267 track->resumeAck(); 4268 if (last) { 4269 if (mPausedBytesRemaining) { 4270 // Need to continue write that was interrupted 4271 mCurrentWriteLength = mPausedWriteLength; 4272 mBytesRemaining = mPausedBytesRemaining; 4273 mPausedBytesRemaining = 0; 4274 } 4275 if (mHwPaused) { 4276 doHwResume = true; 4277 mHwPaused = false; 4278 // threadLoop_mix() will handle the case that we need to 4279 // resume an interrupted write 4280 } 4281 // enable write to audio HAL 4282 sleepTime = 0; 4283 4284 // Do not handle new data in this iteration even if track->framesReady() 4285 mixerStatus = MIXER_TRACKS_ENABLED; 4286 } 4287 } else if (track->framesReady() && track->isReady() && 4288 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4289 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4290 if (track->mFillingUpStatus == Track::FS_FILLED) { 4291 track->mFillingUpStatus = Track::FS_ACTIVE; 4292 // make sure processVolume_l() will apply new volume even if 0 4293 mLeftVolFloat = mRightVolFloat = -1.0; 4294 } 4295 4296 if (last) { 4297 sp<Track> previousTrack = mPreviousTrack.promote(); 4298 if (previousTrack != 0) { 4299 if (track != previousTrack.get()) { 4300 // Flush any data still being written from last track 4301 mBytesRemaining = 0; 4302 if (mPausedBytesRemaining) { 4303 // Last track was paused so we also need to flush saved 4304 // mixbuffer state and invalidate track so that it will 4305 // re-submit that unwritten data when it is next resumed 4306 mPausedBytesRemaining = 0; 4307 // Invalidate is a bit drastic - would be more efficient 4308 // to have a flag to tell client that some of the 4309 // previously written data was lost 4310 previousTrack->invalidate(); 4311 } 4312 // flush data already sent to the DSP if changing audio session as audio 4313 // comes from a different source. Also invalidate previous track to force a 4314 // seek when resuming. 4315 if (previousTrack->sessionId() != track->sessionId()) { 4316 previousTrack->invalidate(); 4317 } 4318 } 4319 } 4320 mPreviousTrack = track; 4321 // reset retry count 4322 track->mRetryCount = kMaxTrackRetriesOffload; 4323 mActiveTrack = t; 4324 mixerStatus = MIXER_TRACKS_READY; 4325 } 4326 } else { 4327 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4328 if (track->isStopping_1()) { 4329 // Hardware buffer can hold a large amount of audio so we must 4330 // wait for all current track's data to drain before we say 4331 // that the track is stopped. 4332 if (mBytesRemaining == 0) { 4333 // Only start draining when all data in mixbuffer 4334 // has been written 4335 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4336 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4337 // do not drain if no data was ever sent to HAL (mStandby == true) 4338 if (last && !mStandby) { 4339 // do not modify drain sequence if we are already draining. This happens 4340 // when resuming from pause after drain. 4341 if ((mDrainSequence & 1) == 0) { 4342 sleepTime = 0; 4343 standbyTime = systemTime() + standbyDelay; 4344 mixerStatus = MIXER_DRAIN_TRACK; 4345 mDrainSequence += 2; 4346 } 4347 if (mHwPaused) { 4348 // It is possible to move from PAUSED to STOPPING_1 without 4349 // a resume so we must ensure hardware is running 4350 doHwResume = true; 4351 mHwPaused = false; 4352 } 4353 } 4354 } 4355 } else if (track->isStopping_2()) { 4356 // Drain has completed or we are in standby, signal presentation complete 4357 if (!(mDrainSequence & 1) || !last || mStandby) { 4358 track->mState = TrackBase::STOPPED; 4359 size_t audioHALFrames = 4360 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4361 size_t framesWritten = 4362 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4363 track->presentationComplete(framesWritten, audioHALFrames); 4364 track->reset(); 4365 tracksToRemove->add(track); 4366 } 4367 } else { 4368 // No buffers for this track. Give it a few chances to 4369 // fill a buffer, then remove it from active list. 4370 if (--(track->mRetryCount) <= 0) { 4371 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4372 track->name()); 4373 tracksToRemove->add(track); 4374 // indicate to client process that the track was disabled because of underrun; 4375 // it will then automatically call start() when data is available 4376 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4377 } else if (last){ 4378 mixerStatus = MIXER_TRACKS_ENABLED; 4379 } 4380 } 4381 } 4382 // compute volume for this track 4383 processVolume_l(track, last); 4384 } 4385 4386 // make sure the pause/flush/resume sequence is executed in the right order. 4387 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4388 // before flush and then resume HW. This can happen in case of pause/flush/resume 4389 // if resume is received before pause is executed. 4390 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4391 mOutput->stream->pause(mOutput->stream); 4392 } 4393 if (mFlushPending) { 4394 flushHw_l(); 4395 mFlushPending = false; 4396 } 4397 if (!mStandby && doHwResume) { 4398 mOutput->stream->resume(mOutput->stream); 4399 } 4400 4401 // remove all the tracks that need to be... 4402 removeTracks_l(*tracksToRemove); 4403 4404 return mixerStatus; 4405} 4406 4407// must be called with thread mutex locked 4408bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4409{ 4410 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4411 mWriteAckSequence, mDrainSequence); 4412 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4413 return true; 4414 } 4415 return false; 4416} 4417 4418// must be called with thread mutex locked 4419bool AudioFlinger::OffloadThread::shouldStandby_l() 4420{ 4421 bool trackPaused = false; 4422 4423 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4424 // after a timeout and we will enter standby then. 4425 if (mTracks.size() > 0) { 4426 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4427 } 4428 4429 return !mStandby && !trackPaused; 4430} 4431 4432 4433bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4434{ 4435 Mutex::Autolock _l(mLock); 4436 return waitingAsyncCallback_l(); 4437} 4438 4439void AudioFlinger::OffloadThread::flushHw_l() 4440{ 4441 mOutput->stream->flush(mOutput->stream); 4442 // Flush anything still waiting in the mixbuffer 4443 mCurrentWriteLength = 0; 4444 mBytesRemaining = 0; 4445 mPausedWriteLength = 0; 4446 mPausedBytesRemaining = 0; 4447 mHwPaused = false; 4448 4449 if (mUseAsyncWrite) { 4450 // discard any pending drain or write ack by incrementing sequence 4451 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4452 mDrainSequence = (mDrainSequence + 2) & ~1; 4453 ALOG_ASSERT(mCallbackThread != 0); 4454 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4455 mCallbackThread->setDraining(mDrainSequence); 4456 } 4457} 4458 4459void AudioFlinger::OffloadThread::onAddNewTrack_l() 4460{ 4461 sp<Track> previousTrack = mPreviousTrack.promote(); 4462 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4463 4464 if (previousTrack != 0 && latestTrack != 0 && 4465 (previousTrack->sessionId() != latestTrack->sessionId())) { 4466 mFlushPending = true; 4467 } 4468 PlaybackThread::onAddNewTrack_l(); 4469} 4470 4471// ---------------------------------------------------------------------------- 4472 4473AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4474 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4475 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4476 DUPLICATING), 4477 mWaitTimeMs(UINT_MAX) 4478{ 4479 addOutputTrack(mainThread); 4480} 4481 4482AudioFlinger::DuplicatingThread::~DuplicatingThread() 4483{ 4484 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4485 mOutputTracks[i]->destroy(); 4486 } 4487} 4488 4489void AudioFlinger::DuplicatingThread::threadLoop_mix() 4490{ 4491 // mix buffers... 4492 if (outputsReady(outputTracks)) { 4493 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4494 } else { 4495 memset(mSinkBuffer, 0, mSinkBufferSize); 4496 } 4497 sleepTime = 0; 4498 writeFrames = mNormalFrameCount; 4499 mCurrentWriteLength = mSinkBufferSize; 4500 standbyTime = systemTime() + standbyDelay; 4501} 4502 4503void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4504{ 4505 if (sleepTime == 0) { 4506 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4507 sleepTime = activeSleepTime; 4508 } else { 4509 sleepTime = idleSleepTime; 4510 } 4511 } else if (mBytesWritten != 0) { 4512 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4513 writeFrames = mNormalFrameCount; 4514 memset(mSinkBuffer, 0, mSinkBufferSize); 4515 } else { 4516 // flush remaining overflow buffers in output tracks 4517 writeFrames = 0; 4518 } 4519 sleepTime = 0; 4520 } 4521} 4522 4523ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4524{ 4525 for (size_t i = 0; i < outputTracks.size(); i++) { 4526 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4527 // for delivery downstream as needed. This in-place conversion is safe as 4528 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4529 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4530 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4531 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4532 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4533 } 4534 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4535 } 4536 mStandby = false; 4537 return (ssize_t)mSinkBufferSize; 4538} 4539 4540void AudioFlinger::DuplicatingThread::threadLoop_standby() 4541{ 4542 // DuplicatingThread implements standby by stopping all tracks 4543 for (size_t i = 0; i < outputTracks.size(); i++) { 4544 outputTracks[i]->stop(); 4545 } 4546} 4547 4548void AudioFlinger::DuplicatingThread::saveOutputTracks() 4549{ 4550 outputTracks = mOutputTracks; 4551} 4552 4553void AudioFlinger::DuplicatingThread::clearOutputTracks() 4554{ 4555 outputTracks.clear(); 4556} 4557 4558void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4559{ 4560 Mutex::Autolock _l(mLock); 4561 // FIXME explain this formula 4562 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4563 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4564 // due to current usage case and restrictions on the AudioBufferProvider. 4565 // Actual buffer conversion is done in threadLoop_write(). 4566 // 4567 // TODO: This may change in the future, depending on multichannel 4568 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4569 OutputTrack *outputTrack = new OutputTrack(thread, 4570 this, 4571 mSampleRate, 4572 AUDIO_FORMAT_PCM_16_BIT, 4573 mChannelMask, 4574 frameCount, 4575 IPCThreadState::self()->getCallingUid()); 4576 if (outputTrack->cblk() != NULL) { 4577 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4578 mOutputTracks.add(outputTrack); 4579 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4580 updateWaitTime_l(); 4581 } 4582} 4583 4584void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4585{ 4586 Mutex::Autolock _l(mLock); 4587 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4588 if (mOutputTracks[i]->thread() == thread) { 4589 mOutputTracks[i]->destroy(); 4590 mOutputTracks.removeAt(i); 4591 updateWaitTime_l(); 4592 return; 4593 } 4594 } 4595 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4596} 4597 4598// caller must hold mLock 4599void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4600{ 4601 mWaitTimeMs = UINT_MAX; 4602 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4603 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4604 if (strong != 0) { 4605 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4606 if (waitTimeMs < mWaitTimeMs) { 4607 mWaitTimeMs = waitTimeMs; 4608 } 4609 } 4610 } 4611} 4612 4613 4614bool AudioFlinger::DuplicatingThread::outputsReady( 4615 const SortedVector< sp<OutputTrack> > &outputTracks) 4616{ 4617 for (size_t i = 0; i < outputTracks.size(); i++) { 4618 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4619 if (thread == 0) { 4620 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4621 outputTracks[i].get()); 4622 return false; 4623 } 4624 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4625 // see note at standby() declaration 4626 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4627 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4628 thread.get()); 4629 return false; 4630 } 4631 } 4632 return true; 4633} 4634 4635uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4636{ 4637 return (mWaitTimeMs * 1000) / 2; 4638} 4639 4640void AudioFlinger::DuplicatingThread::cacheParameters_l() 4641{ 4642 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4643 updateWaitTime_l(); 4644 4645 MixerThread::cacheParameters_l(); 4646} 4647 4648// ---------------------------------------------------------------------------- 4649// Record 4650// ---------------------------------------------------------------------------- 4651 4652AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4653 AudioStreamIn *input, 4654 audio_io_handle_t id, 4655 audio_devices_t outDevice, 4656 audio_devices_t inDevice 4657#ifdef TEE_SINK 4658 , const sp<NBAIO_Sink>& teeSink 4659#endif 4660 ) : 4661 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4662 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4663 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4664 mRsmpInRear(0) 4665#ifdef TEE_SINK 4666 , mTeeSink(teeSink) 4667#endif 4668 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4669 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4670{ 4671 snprintf(mName, kNameLength, "AudioIn_%X", id); 4672 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4673 4674 readInputParameters_l(); 4675} 4676 4677 4678AudioFlinger::RecordThread::~RecordThread() 4679{ 4680 mAudioFlinger->unregisterWriter(mNBLogWriter); 4681 delete[] mRsmpInBuffer; 4682} 4683 4684void AudioFlinger::RecordThread::onFirstRef() 4685{ 4686 run(mName, PRIORITY_URGENT_AUDIO); 4687} 4688 4689bool AudioFlinger::RecordThread::threadLoop() 4690{ 4691 nsecs_t lastWarning = 0; 4692 4693 inputStandBy(); 4694 4695reacquire_wakelock: 4696 sp<RecordTrack> activeTrack; 4697 int activeTracksGen; 4698 { 4699 Mutex::Autolock _l(mLock); 4700 size_t size = mActiveTracks.size(); 4701 activeTracksGen = mActiveTracksGen; 4702 if (size > 0) { 4703 // FIXME an arbitrary choice 4704 activeTrack = mActiveTracks[0]; 4705 acquireWakeLock_l(activeTrack->uid()); 4706 if (size > 1) { 4707 SortedVector<int> tmp; 4708 for (size_t i = 0; i < size; i++) { 4709 tmp.add(mActiveTracks[i]->uid()); 4710 } 4711 updateWakeLockUids_l(tmp); 4712 } 4713 } else { 4714 acquireWakeLock_l(-1); 4715 } 4716 } 4717 4718 // used to request a deferred sleep, to be executed later while mutex is unlocked 4719 uint32_t sleepUs = 0; 4720 4721 // loop while there is work to do 4722 for (;;) { 4723 Vector< sp<EffectChain> > effectChains; 4724 4725 // sleep with mutex unlocked 4726 if (sleepUs > 0) { 4727 usleep(sleepUs); 4728 sleepUs = 0; 4729 } 4730 4731 // activeTracks accumulates a copy of a subset of mActiveTracks 4732 Vector< sp<RecordTrack> > activeTracks; 4733 4734 4735 { // scope for mLock 4736 Mutex::Autolock _l(mLock); 4737 4738 processConfigEvents_l(); 4739 4740 // check exitPending here because checkForNewParameters_l() and 4741 // checkForNewParameters_l() can temporarily release mLock 4742 if (exitPending()) { 4743 break; 4744 } 4745 4746 // if no active track(s), then standby and release wakelock 4747 size_t size = mActiveTracks.size(); 4748 if (size == 0) { 4749 standbyIfNotAlreadyInStandby(); 4750 // exitPending() can't become true here 4751 releaseWakeLock_l(); 4752 ALOGV("RecordThread: loop stopping"); 4753 // go to sleep 4754 mWaitWorkCV.wait(mLock); 4755 ALOGV("RecordThread: loop starting"); 4756 goto reacquire_wakelock; 4757 } 4758 4759 if (mActiveTracksGen != activeTracksGen) { 4760 activeTracksGen = mActiveTracksGen; 4761 SortedVector<int> tmp; 4762 for (size_t i = 0; i < size; i++) { 4763 tmp.add(mActiveTracks[i]->uid()); 4764 } 4765 updateWakeLockUids_l(tmp); 4766 } 4767 4768 bool doBroadcast = false; 4769 for (size_t i = 0; i < size; ) { 4770 4771 activeTrack = mActiveTracks[i]; 4772 if (activeTrack->isTerminated()) { 4773 removeTrack_l(activeTrack); 4774 mActiveTracks.remove(activeTrack); 4775 mActiveTracksGen++; 4776 size--; 4777 continue; 4778 } 4779 4780 TrackBase::track_state activeTrackState = activeTrack->mState; 4781 switch (activeTrackState) { 4782 4783 case TrackBase::PAUSING: 4784 mActiveTracks.remove(activeTrack); 4785 mActiveTracksGen++; 4786 doBroadcast = true; 4787 size--; 4788 continue; 4789 4790 case TrackBase::STARTING_1: 4791 sleepUs = 10000; 4792 i++; 4793 continue; 4794 4795 case TrackBase::STARTING_2: 4796 doBroadcast = true; 4797 mStandby = false; 4798 activeTrack->mState = TrackBase::ACTIVE; 4799 break; 4800 4801 case TrackBase::ACTIVE: 4802 break; 4803 4804 case TrackBase::IDLE: 4805 i++; 4806 continue; 4807 4808 default: 4809 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4810 } 4811 4812 activeTracks.add(activeTrack); 4813 i++; 4814 4815 } 4816 if (doBroadcast) { 4817 mStartStopCond.broadcast(); 4818 } 4819 4820 // sleep if there are no active tracks to process 4821 if (activeTracks.size() == 0) { 4822 if (sleepUs == 0) { 4823 sleepUs = kRecordThreadSleepUs; 4824 } 4825 continue; 4826 } 4827 sleepUs = 0; 4828 4829 lockEffectChains_l(effectChains); 4830 } 4831 4832 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4833 4834 size_t size = effectChains.size(); 4835 for (size_t i = 0; i < size; i++) { 4836 // thread mutex is not locked, but effect chain is locked 4837 effectChains[i]->process_l(); 4838 } 4839 4840 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4841 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4842 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4843 // If destination is non-contiguous, first read past the nominal end of buffer, then 4844 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4845 4846 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4847 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4848 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4849 if (bytesRead <= 0) { 4850 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4851 // Force input into standby so that it tries to recover at next read attempt 4852 inputStandBy(); 4853 sleepUs = kRecordThreadSleepUs; 4854 continue; 4855 } 4856 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4857 size_t framesRead = bytesRead / mFrameSize; 4858 ALOG_ASSERT(framesRead > 0); 4859 if (mTeeSink != 0) { 4860 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4861 } 4862 // If destination is non-contiguous, we now correct for reading past end of buffer. 4863 size_t part1 = mRsmpInFramesP2 - rear; 4864 if (framesRead > part1) { 4865 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4866 (framesRead - part1) * mFrameSize); 4867 } 4868 rear = mRsmpInRear += framesRead; 4869 4870 size = activeTracks.size(); 4871 // loop over each active track 4872 for (size_t i = 0; i < size; i++) { 4873 activeTrack = activeTracks[i]; 4874 4875 enum { 4876 OVERRUN_UNKNOWN, 4877 OVERRUN_TRUE, 4878 OVERRUN_FALSE 4879 } overrun = OVERRUN_UNKNOWN; 4880 4881 // loop over getNextBuffer to handle circular sink 4882 for (;;) { 4883 4884 activeTrack->mSink.frameCount = ~0; 4885 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4886 size_t framesOut = activeTrack->mSink.frameCount; 4887 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4888 4889 int32_t front = activeTrack->mRsmpInFront; 4890 ssize_t filled = rear - front; 4891 size_t framesIn; 4892 4893 if (filled < 0) { 4894 // should not happen, but treat like a massive overrun and re-sync 4895 framesIn = 0; 4896 activeTrack->mRsmpInFront = rear; 4897 overrun = OVERRUN_TRUE; 4898 } else if ((size_t) filled <= mRsmpInFrames) { 4899 framesIn = (size_t) filled; 4900 } else { 4901 // client is not keeping up with server, but give it latest data 4902 framesIn = mRsmpInFrames; 4903 activeTrack->mRsmpInFront = front = rear - framesIn; 4904 overrun = OVERRUN_TRUE; 4905 } 4906 4907 if (framesOut == 0 || framesIn == 0) { 4908 break; 4909 } 4910 4911 if (activeTrack->mResampler == NULL) { 4912 // no resampling 4913 if (framesIn > framesOut) { 4914 framesIn = framesOut; 4915 } else { 4916 framesOut = framesIn; 4917 } 4918 int8_t *dst = activeTrack->mSink.i8; 4919 while (framesIn > 0) { 4920 front &= mRsmpInFramesP2 - 1; 4921 size_t part1 = mRsmpInFramesP2 - front; 4922 if (part1 > framesIn) { 4923 part1 = framesIn; 4924 } 4925 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4926 if (mChannelCount == activeTrack->mChannelCount) { 4927 memcpy(dst, src, part1 * mFrameSize); 4928 } else if (mChannelCount == 1) { 4929 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4930 part1); 4931 } else { 4932 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4933 part1); 4934 } 4935 dst += part1 * activeTrack->mFrameSize; 4936 front += part1; 4937 framesIn -= part1; 4938 } 4939 activeTrack->mRsmpInFront += framesOut; 4940 4941 } else { 4942 // resampling 4943 // FIXME framesInNeeded should really be part of resampler API, and should 4944 // depend on the SRC ratio 4945 // to keep mRsmpInBuffer full so resampler always has sufficient input 4946 size_t framesInNeeded; 4947 // FIXME only re-calculate when it changes, and optimize for common ratios 4948 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4949 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4950 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4951 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4952 framesInNeeded, framesOut, inOverOut); 4953 // Although we theoretically have framesIn in circular buffer, some of those are 4954 // unreleased frames, and thus must be discounted for purpose of budgeting. 4955 size_t unreleased = activeTrack->mRsmpInUnrel; 4956 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4957 if (framesIn < framesInNeeded) { 4958 ALOGV("not enough to resample: have %u frames in but need %u in to " 4959 "produce %u out given in/out ratio of %.4g", 4960 framesIn, framesInNeeded, framesOut, inOverOut); 4961 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4962 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4963 if (newFramesOut == 0) { 4964 break; 4965 } 4966 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4967 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4968 framesInNeeded, newFramesOut, outOverIn); 4969 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4970 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4971 "given in/out ratio of %.4g", 4972 framesIn, framesInNeeded, newFramesOut, inOverOut); 4973 framesOut = newFramesOut; 4974 } else { 4975 ALOGV("success 1: have %u in and need %u in to produce %u out " 4976 "given in/out ratio of %.4g", 4977 framesIn, framesInNeeded, framesOut, inOverOut); 4978 } 4979 4980 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4981 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4982 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4983 delete[] activeTrack->mRsmpOutBuffer; 4984 // resampler always outputs stereo 4985 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4986 activeTrack->mRsmpOutFrameCount = framesOut; 4987 } 4988 4989 // resampler accumulates, but we only have one source track 4990 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4991 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4992 // FIXME how about having activeTrack implement this interface itself? 4993 activeTrack->mResamplerBufferProvider 4994 /*this*/ /* AudioBufferProvider* */); 4995 // ditherAndClamp() works as long as all buffers returned by 4996 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4997 if (activeTrack->mChannelCount == 1) { 4998 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 4999 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5000 framesOut); 5001 // the resampler always outputs stereo samples: 5002 // do post stereo to mono conversion 5003 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5004 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5005 } else { 5006 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5007 activeTrack->mRsmpOutBuffer, framesOut); 5008 } 5009 // now done with mRsmpOutBuffer 5010 5011 } 5012 5013 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5014 overrun = OVERRUN_FALSE; 5015 } 5016 5017 if (activeTrack->mFramesToDrop == 0) { 5018 if (framesOut > 0) { 5019 activeTrack->mSink.frameCount = framesOut; 5020 activeTrack->releaseBuffer(&activeTrack->mSink); 5021 } 5022 } else { 5023 // FIXME could do a partial drop of framesOut 5024 if (activeTrack->mFramesToDrop > 0) { 5025 activeTrack->mFramesToDrop -= framesOut; 5026 if (activeTrack->mFramesToDrop <= 0) { 5027 activeTrack->clearSyncStartEvent(); 5028 } 5029 } else { 5030 activeTrack->mFramesToDrop += framesOut; 5031 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5032 activeTrack->mSyncStartEvent->isCancelled()) { 5033 ALOGW("Synced record %s, session %d, trigger session %d", 5034 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5035 activeTrack->sessionId(), 5036 (activeTrack->mSyncStartEvent != 0) ? 5037 activeTrack->mSyncStartEvent->triggerSession() : 0); 5038 activeTrack->clearSyncStartEvent(); 5039 } 5040 } 5041 } 5042 5043 if (framesOut == 0) { 5044 break; 5045 } 5046 } 5047 5048 switch (overrun) { 5049 case OVERRUN_TRUE: 5050 // client isn't retrieving buffers fast enough 5051 if (!activeTrack->setOverflow()) { 5052 nsecs_t now = systemTime(); 5053 // FIXME should lastWarning per track? 5054 if ((now - lastWarning) > kWarningThrottleNs) { 5055 ALOGW("RecordThread: buffer overflow"); 5056 lastWarning = now; 5057 } 5058 } 5059 break; 5060 case OVERRUN_FALSE: 5061 activeTrack->clearOverflow(); 5062 break; 5063 case OVERRUN_UNKNOWN: 5064 break; 5065 } 5066 5067 } 5068 5069 // enable changes in effect chain 5070 unlockEffectChains(effectChains); 5071 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5072 } 5073 5074 standbyIfNotAlreadyInStandby(); 5075 5076 { 5077 Mutex::Autolock _l(mLock); 5078 for (size_t i = 0; i < mTracks.size(); i++) { 5079 sp<RecordTrack> track = mTracks[i]; 5080 track->invalidate(); 5081 } 5082 mActiveTracks.clear(); 5083 mActiveTracksGen++; 5084 mStartStopCond.broadcast(); 5085 } 5086 5087 releaseWakeLock(); 5088 5089 ALOGV("RecordThread %p exiting", this); 5090 return false; 5091} 5092 5093void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5094{ 5095 if (!mStandby) { 5096 inputStandBy(); 5097 mStandby = true; 5098 } 5099} 5100 5101void AudioFlinger::RecordThread::inputStandBy() 5102{ 5103 mInput->stream->common.standby(&mInput->stream->common); 5104} 5105 5106// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5107sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5108 const sp<AudioFlinger::Client>& client, 5109 uint32_t sampleRate, 5110 audio_format_t format, 5111 audio_channel_mask_t channelMask, 5112 size_t *pFrameCount, 5113 int sessionId, 5114 int uid, 5115 IAudioFlinger::track_flags_t *flags, 5116 pid_t tid, 5117 status_t *status) 5118{ 5119 size_t frameCount = *pFrameCount; 5120 sp<RecordTrack> track; 5121 status_t lStatus; 5122 5123 // client expresses a preference for FAST, but we get the final say 5124 if (*flags & IAudioFlinger::TRACK_FAST) { 5125 if ( 5126 // use case: callback handler and frame count is default or at least as large as HAL 5127 ( 5128 (tid != -1) && 5129 ((frameCount == 0) || 5130 // FIXME not necessarily true, should be native frame count for native SR! 5131 (frameCount >= mFrameCount)) 5132 ) && 5133 // PCM data 5134 audio_is_linear_pcm(format) && 5135 // mono or stereo 5136 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5137 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5138 // hardware sample rate 5139 // FIXME actually the native hardware sample rate 5140 (sampleRate == mSampleRate) && 5141 // record thread has an associated fast capture 5142 hasFastCapture() 5143 // fast capture does not require slots 5144 ) { 5145 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5146 if (frameCount == 0) { 5147 // FIXME wrong mFrameCount 5148 frameCount = mFrameCount * kFastTrackMultiplier; 5149 } 5150 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5151 frameCount, mFrameCount); 5152 } else { 5153 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5154 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5155 "hasFastCapture=%d tid=%d", 5156 frameCount, mFrameCount, format, 5157 audio_is_linear_pcm(format), 5158 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5159 *flags &= ~IAudioFlinger::TRACK_FAST; 5160 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5161 // For compatibility with AudioRecord calculation, buffer depth is forced 5162 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5163 // This is probably too conservative, but legacy application code may depend on it. 5164 // If you change this calculation, also review the start threshold which is related. 5165 // FIXME It's not clear how input latency actually matters. Perhaps this should be 0. 5166 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5167 size_t mNormalFrameCount = 2048; // FIXME 5168 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5169 if (minBufCount < 2) { 5170 minBufCount = 2; 5171 } 5172 size_t minFrameCount = mNormalFrameCount * minBufCount; 5173 if (frameCount < minFrameCount) { 5174 frameCount = minFrameCount; 5175 } 5176 } 5177 } 5178 *pFrameCount = frameCount; 5179 5180 lStatus = initCheck(); 5181 if (lStatus != NO_ERROR) { 5182 ALOGE("createRecordTrack_l() audio driver not initialized"); 5183 goto Exit; 5184 } 5185 5186 { // scope for mLock 5187 Mutex::Autolock _l(mLock); 5188 5189 track = new RecordTrack(this, client, sampleRate, 5190 format, channelMask, frameCount, sessionId, uid, 5191 *flags); 5192 5193 lStatus = track->initCheck(); 5194 if (lStatus != NO_ERROR) { 5195 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5196 // track must be cleared from the caller as the caller has the AF lock 5197 goto Exit; 5198 } 5199 mTracks.add(track); 5200 5201 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5202 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5203 mAudioFlinger->btNrecIsOff(); 5204 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5205 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5206 5207 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5208 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5209 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5210 // so ask activity manager to do this on our behalf 5211 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5212 } 5213 } 5214 5215 lStatus = NO_ERROR; 5216 5217Exit: 5218 *status = lStatus; 5219 return track; 5220} 5221 5222status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5223 AudioSystem::sync_event_t event, 5224 int triggerSession) 5225{ 5226 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5227 sp<ThreadBase> strongMe = this; 5228 status_t status = NO_ERROR; 5229 5230 if (event == AudioSystem::SYNC_EVENT_NONE) { 5231 recordTrack->clearSyncStartEvent(); 5232 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5233 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5234 triggerSession, 5235 recordTrack->sessionId(), 5236 syncStartEventCallback, 5237 recordTrack); 5238 // Sync event can be cancelled by the trigger session if the track is not in a 5239 // compatible state in which case we start record immediately 5240 if (recordTrack->mSyncStartEvent->isCancelled()) { 5241 recordTrack->clearSyncStartEvent(); 5242 } else { 5243 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5244 recordTrack->mFramesToDrop = - 5245 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5246 } 5247 } 5248 5249 { 5250 // This section is a rendezvous between binder thread executing start() and RecordThread 5251 AutoMutex lock(mLock); 5252 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5253 if (recordTrack->mState == TrackBase::PAUSING) { 5254 ALOGV("active record track PAUSING -> ACTIVE"); 5255 recordTrack->mState = TrackBase::ACTIVE; 5256 } else { 5257 ALOGV("active record track state %d", recordTrack->mState); 5258 } 5259 return status; 5260 } 5261 5262 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5263 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5264 // or using a separate command thread 5265 recordTrack->mState = TrackBase::STARTING_1; 5266 mActiveTracks.add(recordTrack); 5267 mActiveTracksGen++; 5268 mLock.unlock(); 5269 status_t status = AudioSystem::startInput(mId); 5270 mLock.lock(); 5271 // FIXME should verify that recordTrack is still in mActiveTracks 5272 if (status != NO_ERROR) { 5273 mActiveTracks.remove(recordTrack); 5274 mActiveTracksGen++; 5275 recordTrack->clearSyncStartEvent(); 5276 return status; 5277 } 5278 // Catch up with current buffer indices if thread is already running. 5279 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5280 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5281 // see previously buffered data before it called start(), but with greater risk of overrun. 5282 5283 recordTrack->mRsmpInFront = mRsmpInRear; 5284 recordTrack->mRsmpInUnrel = 0; 5285 // FIXME why reset? 5286 if (recordTrack->mResampler != NULL) { 5287 recordTrack->mResampler->reset(); 5288 } 5289 recordTrack->mState = TrackBase::STARTING_2; 5290 // signal thread to start 5291 mWaitWorkCV.broadcast(); 5292 if (mActiveTracks.indexOf(recordTrack) < 0) { 5293 ALOGV("Record failed to start"); 5294 status = BAD_VALUE; 5295 goto startError; 5296 } 5297 return status; 5298 } 5299 5300startError: 5301 AudioSystem::stopInput(mId); 5302 recordTrack->clearSyncStartEvent(); 5303 // FIXME I wonder why we do not reset the state here? 5304 return status; 5305} 5306 5307void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5308{ 5309 sp<SyncEvent> strongEvent = event.promote(); 5310 5311 if (strongEvent != 0) { 5312 sp<RefBase> ptr = strongEvent->cookie().promote(); 5313 if (ptr != 0) { 5314 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5315 recordTrack->handleSyncStartEvent(strongEvent); 5316 } 5317 } 5318} 5319 5320bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5321 ALOGV("RecordThread::stop"); 5322 AutoMutex _l(mLock); 5323 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5324 return false; 5325 } 5326 // note that threadLoop may still be processing the track at this point [without lock] 5327 recordTrack->mState = TrackBase::PAUSING; 5328 // do not wait for mStartStopCond if exiting 5329 if (exitPending()) { 5330 return true; 5331 } 5332 // FIXME incorrect usage of wait: no explicit predicate or loop 5333 mStartStopCond.wait(mLock); 5334 // if we have been restarted, recordTrack is in mActiveTracks here 5335 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5336 ALOGV("Record stopped OK"); 5337 return true; 5338 } 5339 return false; 5340} 5341 5342bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5343{ 5344 return false; 5345} 5346 5347status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5348{ 5349#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5350 if (!isValidSyncEvent(event)) { 5351 return BAD_VALUE; 5352 } 5353 5354 int eventSession = event->triggerSession(); 5355 status_t ret = NAME_NOT_FOUND; 5356 5357 Mutex::Autolock _l(mLock); 5358 5359 for (size_t i = 0; i < mTracks.size(); i++) { 5360 sp<RecordTrack> track = mTracks[i]; 5361 if (eventSession == track->sessionId()) { 5362 (void) track->setSyncEvent(event); 5363 ret = NO_ERROR; 5364 } 5365 } 5366 return ret; 5367#else 5368 return BAD_VALUE; 5369#endif 5370} 5371 5372// destroyTrack_l() must be called with ThreadBase::mLock held 5373void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5374{ 5375 track->terminate(); 5376 track->mState = TrackBase::STOPPED; 5377 // active tracks are removed by threadLoop() 5378 if (mActiveTracks.indexOf(track) < 0) { 5379 removeTrack_l(track); 5380 } 5381} 5382 5383void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5384{ 5385 mTracks.remove(track); 5386 // need anything related to effects here? 5387} 5388 5389void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5390{ 5391 dumpInternals(fd, args); 5392 dumpTracks(fd, args); 5393 dumpEffectChains(fd, args); 5394} 5395 5396void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5397{ 5398 dprintf(fd, "\nInput thread %p:\n", this); 5399 5400 if (mActiveTracks.size() > 0) { 5401 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5402 } else { 5403 dprintf(fd, " No active record clients\n"); 5404 } 5405 5406 dumpBase(fd, args); 5407} 5408 5409void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5410{ 5411 const size_t SIZE = 256; 5412 char buffer[SIZE]; 5413 String8 result; 5414 5415 size_t numtracks = mTracks.size(); 5416 size_t numactive = mActiveTracks.size(); 5417 size_t numactiveseen = 0; 5418 dprintf(fd, " %d Tracks", numtracks); 5419 if (numtracks) { 5420 dprintf(fd, " of which %d are active\n", numactive); 5421 RecordTrack::appendDumpHeader(result); 5422 for (size_t i = 0; i < numtracks ; ++i) { 5423 sp<RecordTrack> track = mTracks[i]; 5424 if (track != 0) { 5425 bool active = mActiveTracks.indexOf(track) >= 0; 5426 if (active) { 5427 numactiveseen++; 5428 } 5429 track->dump(buffer, SIZE, active); 5430 result.append(buffer); 5431 } 5432 } 5433 } else { 5434 dprintf(fd, "\n"); 5435 } 5436 5437 if (numactiveseen != numactive) { 5438 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5439 " not in the track list\n"); 5440 result.append(buffer); 5441 RecordTrack::appendDumpHeader(result); 5442 for (size_t i = 0; i < numactive; ++i) { 5443 sp<RecordTrack> track = mActiveTracks[i]; 5444 if (mTracks.indexOf(track) < 0) { 5445 track->dump(buffer, SIZE, true); 5446 result.append(buffer); 5447 } 5448 } 5449 5450 } 5451 write(fd, result.string(), result.size()); 5452} 5453 5454// AudioBufferProvider interface 5455status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5456 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5457{ 5458 RecordTrack *activeTrack = mRecordTrack; 5459 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5460 if (threadBase == 0) { 5461 buffer->frameCount = 0; 5462 buffer->raw = NULL; 5463 return NOT_ENOUGH_DATA; 5464 } 5465 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5466 int32_t rear = recordThread->mRsmpInRear; 5467 int32_t front = activeTrack->mRsmpInFront; 5468 ssize_t filled = rear - front; 5469 // FIXME should not be P2 (don't want to increase latency) 5470 // FIXME if client not keeping up, discard 5471 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5472 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5473 front &= recordThread->mRsmpInFramesP2 - 1; 5474 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5475 if (part1 > (size_t) filled) { 5476 part1 = filled; 5477 } 5478 size_t ask = buffer->frameCount; 5479 ALOG_ASSERT(ask > 0); 5480 if (part1 > ask) { 5481 part1 = ask; 5482 } 5483 if (part1 == 0) { 5484 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5485 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5486 buffer->raw = NULL; 5487 buffer->frameCount = 0; 5488 activeTrack->mRsmpInUnrel = 0; 5489 return NOT_ENOUGH_DATA; 5490 } 5491 5492 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5493 buffer->frameCount = part1; 5494 activeTrack->mRsmpInUnrel = part1; 5495 return NO_ERROR; 5496} 5497 5498// AudioBufferProvider interface 5499void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5500 AudioBufferProvider::Buffer* buffer) 5501{ 5502 RecordTrack *activeTrack = mRecordTrack; 5503 size_t stepCount = buffer->frameCount; 5504 if (stepCount == 0) { 5505 return; 5506 } 5507 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5508 activeTrack->mRsmpInUnrel -= stepCount; 5509 activeTrack->mRsmpInFront += stepCount; 5510 buffer->raw = NULL; 5511 buffer->frameCount = 0; 5512} 5513 5514bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5515 status_t& status) 5516{ 5517 bool reconfig = false; 5518 5519 status = NO_ERROR; 5520 5521 audio_format_t reqFormat = mFormat; 5522 uint32_t samplingRate = mSampleRate; 5523 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5524 5525 AudioParameter param = AudioParameter(keyValuePair); 5526 int value; 5527 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5528 // channel count change can be requested. Do we mandate the first client defines the 5529 // HAL sampling rate and channel count or do we allow changes on the fly? 5530 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5531 samplingRate = value; 5532 reconfig = true; 5533 } 5534 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5535 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5536 status = BAD_VALUE; 5537 } else { 5538 reqFormat = (audio_format_t) value; 5539 reconfig = true; 5540 } 5541 } 5542 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5543 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5544 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5545 status = BAD_VALUE; 5546 } else { 5547 channelMask = mask; 5548 reconfig = true; 5549 } 5550 } 5551 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5552 // do not accept frame count changes if tracks are open as the track buffer 5553 // size depends on frame count and correct behavior would not be guaranteed 5554 // if frame count is changed after track creation 5555 if (mActiveTracks.size() > 0) { 5556 status = INVALID_OPERATION; 5557 } else { 5558 reconfig = true; 5559 } 5560 } 5561 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5562 // forward device change to effects that have requested to be 5563 // aware of attached audio device. 5564 for (size_t i = 0; i < mEffectChains.size(); i++) { 5565 mEffectChains[i]->setDevice_l(value); 5566 } 5567 5568 // store input device and output device but do not forward output device to audio HAL. 5569 // Note that status is ignored by the caller for output device 5570 // (see AudioFlinger::setParameters() 5571 if (audio_is_output_devices(value)) { 5572 mOutDevice = value; 5573 status = BAD_VALUE; 5574 } else { 5575 mInDevice = value; 5576 // disable AEC and NS if the device is a BT SCO headset supporting those 5577 // pre processings 5578 if (mTracks.size() > 0) { 5579 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5580 mAudioFlinger->btNrecIsOff(); 5581 for (size_t i = 0; i < mTracks.size(); i++) { 5582 sp<RecordTrack> track = mTracks[i]; 5583 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5584 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5585 } 5586 } 5587 } 5588 } 5589 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5590 mAudioSource != (audio_source_t)value) { 5591 // forward device change to effects that have requested to be 5592 // aware of attached audio device. 5593 for (size_t i = 0; i < mEffectChains.size(); i++) { 5594 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5595 } 5596 mAudioSource = (audio_source_t)value; 5597 } 5598 5599 if (status == NO_ERROR) { 5600 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5601 keyValuePair.string()); 5602 if (status == INVALID_OPERATION) { 5603 inputStandBy(); 5604 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5605 keyValuePair.string()); 5606 } 5607 if (reconfig) { 5608 if (status == BAD_VALUE && 5609 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5610 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5611 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5612 <= (2 * samplingRate)) && 5613 audio_channel_count_from_in_mask( 5614 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5615 (channelMask == AUDIO_CHANNEL_IN_MONO || 5616 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5617 status = NO_ERROR; 5618 } 5619 if (status == NO_ERROR) { 5620 readInputParameters_l(); 5621 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5622 } 5623 } 5624 } 5625 5626 return reconfig; 5627} 5628 5629String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5630{ 5631 Mutex::Autolock _l(mLock); 5632 if (initCheck() != NO_ERROR) { 5633 return String8(); 5634 } 5635 5636 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5637 const String8 out_s8(s); 5638 free(s); 5639 return out_s8; 5640} 5641 5642void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5643 AudioSystem::OutputDescriptor desc; 5644 const void *param2 = NULL; 5645 5646 switch (event) { 5647 case AudioSystem::INPUT_OPENED: 5648 case AudioSystem::INPUT_CONFIG_CHANGED: 5649 desc.channelMask = mChannelMask; 5650 desc.samplingRate = mSampleRate; 5651 desc.format = mFormat; 5652 desc.frameCount = mFrameCount; 5653 desc.latency = 0; 5654 param2 = &desc; 5655 break; 5656 5657 case AudioSystem::INPUT_CLOSED: 5658 default: 5659 break; 5660 } 5661 mAudioFlinger->audioConfigChanged(event, mId, param2); 5662} 5663 5664void AudioFlinger::RecordThread::readInputParameters_l() 5665{ 5666 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5667 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5668 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5669 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5670 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5671 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5672 } 5673 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5674 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5675 mFrameCount = mBufferSize / mFrameSize; 5676 // This is the formula for calculating the temporary buffer size. 5677 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5678 // 1 full output buffer, regardless of the alignment of the available input. 5679 // The value is somewhat arbitrary, and could probably be even larger. 5680 // A larger value should allow more old data to be read after a track calls start(), 5681 // without increasing latency. 5682 mRsmpInFrames = mFrameCount * 7; 5683 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5684 delete[] mRsmpInBuffer; 5685 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5686 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5687 5688 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5689 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5690} 5691 5692uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5693{ 5694 Mutex::Autolock _l(mLock); 5695 if (initCheck() != NO_ERROR) { 5696 return 0; 5697 } 5698 5699 return mInput->stream->get_input_frames_lost(mInput->stream); 5700} 5701 5702uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5703{ 5704 Mutex::Autolock _l(mLock); 5705 uint32_t result = 0; 5706 if (getEffectChain_l(sessionId) != 0) { 5707 result = EFFECT_SESSION; 5708 } 5709 5710 for (size_t i = 0; i < mTracks.size(); ++i) { 5711 if (sessionId == mTracks[i]->sessionId()) { 5712 result |= TRACK_SESSION; 5713 break; 5714 } 5715 } 5716 5717 return result; 5718} 5719 5720KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5721{ 5722 KeyedVector<int, bool> ids; 5723 Mutex::Autolock _l(mLock); 5724 for (size_t j = 0; j < mTracks.size(); ++j) { 5725 sp<RecordThread::RecordTrack> track = mTracks[j]; 5726 int sessionId = track->sessionId(); 5727 if (ids.indexOfKey(sessionId) < 0) { 5728 ids.add(sessionId, true); 5729 } 5730 } 5731 return ids; 5732} 5733 5734AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5735{ 5736 Mutex::Autolock _l(mLock); 5737 AudioStreamIn *input = mInput; 5738 mInput = NULL; 5739 return input; 5740} 5741 5742// this method must always be called either with ThreadBase mLock held or inside the thread loop 5743audio_stream_t* AudioFlinger::RecordThread::stream() const 5744{ 5745 if (mInput == NULL) { 5746 return NULL; 5747 } 5748 return &mInput->stream->common; 5749} 5750 5751status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5752{ 5753 // only one chain per input thread 5754 if (mEffectChains.size() != 0) { 5755 return INVALID_OPERATION; 5756 } 5757 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5758 5759 chain->setInBuffer(NULL); 5760 chain->setOutBuffer(NULL); 5761 5762 checkSuspendOnAddEffectChain_l(chain); 5763 5764 mEffectChains.add(chain); 5765 5766 return NO_ERROR; 5767} 5768 5769size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5770{ 5771 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5772 ALOGW_IF(mEffectChains.size() != 1, 5773 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5774 chain.get(), mEffectChains.size(), this); 5775 if (mEffectChains.size() == 1) { 5776 mEffectChains.removeAt(0); 5777 } 5778 return 0; 5779} 5780 5781}; // namespace android 5782