Threads.cpp revision 69aed5f0f4a3be3996d1e78a0473e1a72c1547da
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 38// NBAIO implementations 39#include <media/nbaio/AudioStreamOutSink.h> 40#include <media/nbaio/MonoPipe.h> 41#include <media/nbaio/MonoPipeReader.h> 42#include <media/nbaio/Pipe.h> 43#include <media/nbaio/PipeReader.h> 44#include <media/nbaio/SourceAudioBufferProvider.h> 45 46#include <powermanager/PowerManager.h> 47 48#include <common_time/cc_helper.h> 49#include <common_time/local_clock.h> 50 51#include "AudioFlinger.h" 52#include "AudioMixer.h" 53#include "FastMixer.h" 54#include "ServiceUtilities.h" 55#include "SchedulingPolicyService.h" 56 57#ifdef ADD_BATTERY_DATA 58#include <media/IMediaPlayerService.h> 59#include <media/IMediaDeathNotifier.h> 60#endif 61 62#ifdef DEBUG_CPU_USAGE 63#include <cpustats/CentralTendencyStatistics.h> 64#include <cpustats/ThreadCpuUsage.h> 65#endif 66 67// ---------------------------------------------------------------------------- 68 69// Note: the following macro is used for extremely verbose logging message. In 70// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 71// 0; but one side effect of this is to turn all LOGV's as well. Some messages 72// are so verbose that we want to suppress them even when we have ALOG_ASSERT 73// turned on. Do not uncomment the #def below unless you really know what you 74// are doing and want to see all of the extremely verbose messages. 75//#define VERY_VERY_VERBOSE_LOGGING 76#ifdef VERY_VERY_VERBOSE_LOGGING 77#define ALOGVV ALOGV 78#else 79#define ALOGVV(a...) do { } while(0) 80#endif 81 82namespace android { 83 84// retry counts for buffer fill timeout 85// 50 * ~20msecs = 1 second 86static const int8_t kMaxTrackRetries = 50; 87static const int8_t kMaxTrackStartupRetries = 50; 88// allow less retry attempts on direct output thread. 89// direct outputs can be a scarce resource in audio hardware and should 90// be released as quickly as possible. 91static const int8_t kMaxTrackRetriesDirect = 2; 92 93// don't warn about blocked writes or record buffer overflows more often than this 94static const nsecs_t kWarningThrottleNs = seconds(5); 95 96// RecordThread loop sleep time upon application overrun or audio HAL read error 97static const int kRecordThreadSleepUs = 5000; 98 99// maximum time to wait for setParameters to complete 100static const nsecs_t kSetParametersTimeoutNs = seconds(2); 101 102// minimum sleep time for the mixer thread loop when tracks are active but in underrun 103static const uint32_t kMinThreadSleepTimeUs = 5000; 104// maximum divider applied to the active sleep time in the mixer thread loop 105static const uint32_t kMaxThreadSleepTimeShift = 2; 106 107// minimum normal sink buffer size, expressed in milliseconds rather than frames 108static const uint32_t kMinNormalSinkBufferSizeMs = 20; 109// maximum normal sink buffer size 110static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 111 112// Offloaded output thread standby delay: allows track transition without going to standby 113static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 114 115// Whether to use fast mixer 116static const enum { 117 FastMixer_Never, // never initialize or use: for debugging only 118 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 119 // normal mixer multiplier is 1 120 FastMixer_Static, // initialize if needed, then use all the time if initialized, 121 // multiplier is calculated based on min & max normal mixer buffer size 122 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 // FIXME for FastMixer_Dynamic: 125 // Supporting this option will require fixing HALs that can't handle large writes. 126 // For example, one HAL implementation returns an error from a large write, 127 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 128 // We could either fix the HAL implementations, or provide a wrapper that breaks 129 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 130} kUseFastMixer = FastMixer_Static; 131 132// Priorities for requestPriority 133static const int kPriorityAudioApp = 2; 134static const int kPriorityFastMixer = 3; 135 136// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 137// for the track. The client then sub-divides this into smaller buffers for its use. 138// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 139// So for now we just assume that client is double-buffered for fast tracks. 140// FIXME It would be better for client to tell AudioFlinger the value of N, 141// so AudioFlinger could allocate the right amount of memory. 142// See the client's minBufCount and mNotificationFramesAct calculations for details. 143static const int kFastTrackMultiplier = 2; 144 145// ---------------------------------------------------------------------------- 146 147#ifdef ADD_BATTERY_DATA 148// To collect the amplifier usage 149static void addBatteryData(uint32_t params) { 150 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 151 if (service == NULL) { 152 // it already logged 153 return; 154 } 155 156 service->addBatteryData(params); 157} 158#endif 159 160 161// ---------------------------------------------------------------------------- 162// CPU Stats 163// ---------------------------------------------------------------------------- 164 165class CpuStats { 166public: 167 CpuStats(); 168 void sample(const String8 &title); 169#ifdef DEBUG_CPU_USAGE 170private: 171 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 172 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 173 174 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 175 176 int mCpuNum; // thread's current CPU number 177 int mCpukHz; // frequency of thread's current CPU in kHz 178#endif 179}; 180 181CpuStats::CpuStats() 182#ifdef DEBUG_CPU_USAGE 183 : mCpuNum(-1), mCpukHz(-1) 184#endif 185{ 186} 187 188void CpuStats::sample(const String8 &title 189#ifndef DEBUG_CPU_USAGE 190 __unused 191#endif 192 ) { 193#ifdef DEBUG_CPU_USAGE 194 // get current thread's delta CPU time in wall clock ns 195 double wcNs; 196 bool valid = mCpuUsage.sampleAndEnable(wcNs); 197 198 // record sample for wall clock statistics 199 if (valid) { 200 mWcStats.sample(wcNs); 201 } 202 203 // get the current CPU number 204 int cpuNum = sched_getcpu(); 205 206 // get the current CPU frequency in kHz 207 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 208 209 // check if either CPU number or frequency changed 210 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 211 mCpuNum = cpuNum; 212 mCpukHz = cpukHz; 213 // ignore sample for purposes of cycles 214 valid = false; 215 } 216 217 // if no change in CPU number or frequency, then record sample for cycle statistics 218 if (valid && mCpukHz > 0) { 219 double cycles = wcNs * cpukHz * 0.000001; 220 mHzStats.sample(cycles); 221 } 222 223 unsigned n = mWcStats.n(); 224 // mCpuUsage.elapsed() is expensive, so don't call it every loop 225 if ((n & 127) == 1) { 226 long long elapsed = mCpuUsage.elapsed(); 227 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 228 double perLoop = elapsed / (double) n; 229 double perLoop100 = perLoop * 0.01; 230 double perLoop1k = perLoop * 0.001; 231 double mean = mWcStats.mean(); 232 double stddev = mWcStats.stddev(); 233 double minimum = mWcStats.minimum(); 234 double maximum = mWcStats.maximum(); 235 double meanCycles = mHzStats.mean(); 236 double stddevCycles = mHzStats.stddev(); 237 double minCycles = mHzStats.minimum(); 238 double maxCycles = mHzStats.maximum(); 239 mCpuUsage.resetElapsed(); 240 mWcStats.reset(); 241 mHzStats.reset(); 242 ALOGD("CPU usage for %s over past %.1f secs\n" 243 " (%u mixer loops at %.1f mean ms per loop):\n" 244 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 245 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 246 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 247 title.string(), 248 elapsed * .000000001, n, perLoop * .000001, 249 mean * .001, 250 stddev * .001, 251 minimum * .001, 252 maximum * .001, 253 mean / perLoop100, 254 stddev / perLoop100, 255 minimum / perLoop100, 256 maximum / perLoop100, 257 meanCycles / perLoop1k, 258 stddevCycles / perLoop1k, 259 minCycles / perLoop1k, 260 maxCycles / perLoop1k); 261 262 } 263 } 264#endif 265}; 266 267// ---------------------------------------------------------------------------- 268// ThreadBase 269// ---------------------------------------------------------------------------- 270 271AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 272 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 273 : Thread(false /*canCallJava*/), 274 mType(type), 275 mAudioFlinger(audioFlinger), 276 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 277 // are set by PlaybackThread::readOutputParameters_l() or 278 // RecordThread::readInputParameters_l() 279 mParamStatus(NO_ERROR), 280 //FIXME: mStandby should be true here. Is this some kind of hack? 281 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 282 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 283 // mName will be set by concrete (non-virtual) subclass 284 mDeathRecipient(new PMDeathRecipient(this)) 285{ 286} 287 288AudioFlinger::ThreadBase::~ThreadBase() 289{ 290 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 291 for (size_t i = 0; i < mConfigEvents.size(); i++) { 292 delete mConfigEvents[i]; 293 } 294 mConfigEvents.clear(); 295 296 mParamCond.broadcast(); 297 // do not lock the mutex in destructor 298 releaseWakeLock_l(); 299 if (mPowerManager != 0) { 300 sp<IBinder> binder = mPowerManager->asBinder(); 301 binder->unlinkToDeath(mDeathRecipient); 302 } 303} 304 305status_t AudioFlinger::ThreadBase::readyToRun() 306{ 307 status_t status = initCheck(); 308 if (status == NO_ERROR) { 309 ALOGI("AudioFlinger's thread %p ready to run", this); 310 } else { 311 ALOGE("No working audio driver found."); 312 } 313 return status; 314} 315 316void AudioFlinger::ThreadBase::exit() 317{ 318 ALOGV("ThreadBase::exit"); 319 // do any cleanup required for exit to succeed 320 preExit(); 321 { 322 // This lock prevents the following race in thread (uniprocessor for illustration): 323 // if (!exitPending()) { 324 // // context switch from here to exit() 325 // // exit() calls requestExit(), what exitPending() observes 326 // // exit() calls signal(), which is dropped since no waiters 327 // // context switch back from exit() to here 328 // mWaitWorkCV.wait(...); 329 // // now thread is hung 330 // } 331 AutoMutex lock(mLock); 332 requestExit(); 333 mWaitWorkCV.broadcast(); 334 } 335 // When Thread::requestExitAndWait is made virtual and this method is renamed to 336 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 337 requestExitAndWait(); 338} 339 340status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 341{ 342 status_t status; 343 344 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 345 Mutex::Autolock _l(mLock); 346 347 mNewParameters.add(keyValuePairs); 348 mWaitWorkCV.signal(); 349 // wait condition with timeout in case the thread loop has exited 350 // before the request could be processed 351 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 352 status = mParamStatus; 353 mWaitWorkCV.signal(); 354 } else { 355 status = TIMED_OUT; 356 } 357 return status; 358} 359 360void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 361{ 362 Mutex::Autolock _l(mLock); 363 sendIoConfigEvent_l(event, param); 364} 365 366// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 367void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 368{ 369 IoConfigEvent *ioEvent = new IoConfigEvent(event, param); 370 mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent)); 371 ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, 372 param); 373 mWaitWorkCV.signal(); 374} 375 376// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 377void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 378{ 379 PrioConfigEvent *prioEvent = new PrioConfigEvent(pid, tid, prio); 380 mConfigEvents.add(static_cast<ConfigEvent *>(prioEvent)); 381 ALOGV("sendPrioConfigEvent_l() num events %d pid %d, tid %d prio %d", 382 mConfigEvents.size(), pid, tid, prio); 383 mWaitWorkCV.signal(); 384} 385 386void AudioFlinger::ThreadBase::processConfigEvents() 387{ 388 Mutex::Autolock _l(mLock); 389 processConfigEvents_l(); 390} 391 392// post condition: mConfigEvents.isEmpty() 393void AudioFlinger::ThreadBase::processConfigEvents_l() 394{ 395 while (!mConfigEvents.isEmpty()) { 396 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 397 ConfigEvent *event = mConfigEvents[0]; 398 mConfigEvents.removeAt(0); 399 // release mLock before locking AudioFlinger mLock: lock order is always 400 // AudioFlinger then ThreadBase to avoid cross deadlock 401 mLock.unlock(); 402 switch (event->type()) { 403 case CFG_EVENT_PRIO: { 404 PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event); 405 // FIXME Need to understand why this has be done asynchronously 406 int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio(), 407 true /*asynchronous*/); 408 if (err != 0) { 409 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 410 prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err); 411 } 412 } break; 413 case CFG_EVENT_IO: { 414 IoConfigEvent *ioEvent = static_cast<IoConfigEvent *>(event); 415 { 416 Mutex::Autolock _l(mAudioFlinger->mLock); 417 audioConfigChanged_l(ioEvent->event(), ioEvent->param()); 418 } 419 } break; 420 default: 421 ALOGE("processConfigEvents() unknown event type %d", event->type()); 422 break; 423 } 424 delete event; 425 mLock.lock(); 426 } 427} 428 429String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 430 String8 s; 431 if (output) { 432 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 433 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 434 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 435 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 436 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 437 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 438 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 439 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 440 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 441 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 442 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 443 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 444 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 445 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 446 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 447 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 448 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 449 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 450 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 451 } else { 452 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 453 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 454 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 455 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 456 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 457 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 458 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 459 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 460 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 461 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 462 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 463 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 464 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 465 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 466 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 467 } 468 int len = s.length(); 469 if (s.length() > 2) { 470 char *str = s.lockBuffer(len); 471 s.unlockBuffer(len - 2); 472 } 473 return s; 474} 475 476void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 477{ 478 const size_t SIZE = 256; 479 char buffer[SIZE]; 480 String8 result; 481 482 bool locked = AudioFlinger::dumpTryLock(mLock); 483 if (!locked) { 484 fdprintf(fd, "thread %p maybe dead locked\n", this); 485 } 486 487 fdprintf(fd, " I/O handle: %d\n", mId); 488 fdprintf(fd, " TID: %d\n", getTid()); 489 fdprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 490 fdprintf(fd, " Sample rate: %u\n", mSampleRate); 491 fdprintf(fd, " HAL frame count: %zu\n", mFrameCount); 492 fdprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 493 fdprintf(fd, " Channel Count: %u\n", mChannelCount); 494 fdprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 495 channelMaskToString(mChannelMask, mType != RECORD).string()); 496 fdprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 497 fdprintf(fd, " Frame size: %zu\n", mFrameSize); 498 fdprintf(fd, " Pending setParameters commands:"); 499 size_t numParams = mNewParameters.size(); 500 if (numParams) { 501 fdprintf(fd, "\n Index Command"); 502 for (size_t i = 0; i < numParams; ++i) { 503 fdprintf(fd, "\n %02zu ", i); 504 fdprintf(fd, mNewParameters[i]); 505 } 506 fdprintf(fd, "\n"); 507 } else { 508 fdprintf(fd, " none\n"); 509 } 510 fdprintf(fd, " Pending config events:"); 511 size_t numConfig = mConfigEvents.size(); 512 if (numConfig) { 513 for (size_t i = 0; i < numConfig; i++) { 514 mConfigEvents[i]->dump(buffer, SIZE); 515 fdprintf(fd, "\n %s", buffer); 516 } 517 fdprintf(fd, "\n"); 518 } else { 519 fdprintf(fd, " none\n"); 520 } 521 522 if (locked) { 523 mLock.unlock(); 524 } 525} 526 527void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 528{ 529 const size_t SIZE = 256; 530 char buffer[SIZE]; 531 String8 result; 532 533 size_t numEffectChains = mEffectChains.size(); 534 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 535 write(fd, buffer, strlen(buffer)); 536 537 for (size_t i = 0; i < numEffectChains; ++i) { 538 sp<EffectChain> chain = mEffectChains[i]; 539 if (chain != 0) { 540 chain->dump(fd, args); 541 } 542 } 543} 544 545void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 546{ 547 Mutex::Autolock _l(mLock); 548 acquireWakeLock_l(uid); 549} 550 551String16 AudioFlinger::ThreadBase::getWakeLockTag() 552{ 553 switch (mType) { 554 case MIXER: 555 return String16("AudioMix"); 556 case DIRECT: 557 return String16("AudioDirectOut"); 558 case DUPLICATING: 559 return String16("AudioDup"); 560 case RECORD: 561 return String16("AudioIn"); 562 case OFFLOAD: 563 return String16("AudioOffload"); 564 default: 565 ALOG_ASSERT(false); 566 return String16("AudioUnknown"); 567 } 568} 569 570void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 571{ 572 getPowerManager_l(); 573 if (mPowerManager != 0) { 574 sp<IBinder> binder = new BBinder(); 575 status_t status; 576 if (uid >= 0) { 577 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 578 binder, 579 getWakeLockTag(), 580 String16("media"), 581 uid); 582 } else { 583 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 584 binder, 585 getWakeLockTag(), 586 String16("media")); 587 } 588 if (status == NO_ERROR) { 589 mWakeLockToken = binder; 590 } 591 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 592 } 593} 594 595void AudioFlinger::ThreadBase::releaseWakeLock() 596{ 597 Mutex::Autolock _l(mLock); 598 releaseWakeLock_l(); 599} 600 601void AudioFlinger::ThreadBase::releaseWakeLock_l() 602{ 603 if (mWakeLockToken != 0) { 604 ALOGV("releaseWakeLock_l() %s", mName); 605 if (mPowerManager != 0) { 606 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 607 } 608 mWakeLockToken.clear(); 609 } 610} 611 612void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 613 Mutex::Autolock _l(mLock); 614 updateWakeLockUids_l(uids); 615} 616 617void AudioFlinger::ThreadBase::getPowerManager_l() { 618 619 if (mPowerManager == 0) { 620 // use checkService() to avoid blocking if power service is not up yet 621 sp<IBinder> binder = 622 defaultServiceManager()->checkService(String16("power")); 623 if (binder == 0) { 624 ALOGW("Thread %s cannot connect to the power manager service", mName); 625 } else { 626 mPowerManager = interface_cast<IPowerManager>(binder); 627 binder->linkToDeath(mDeathRecipient); 628 } 629 } 630} 631 632void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 633 634 getPowerManager_l(); 635 if (mWakeLockToken == NULL) { 636 ALOGE("no wake lock to update!"); 637 return; 638 } 639 if (mPowerManager != 0) { 640 sp<IBinder> binder = new BBinder(); 641 status_t status; 642 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 643 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 644 } 645} 646 647void AudioFlinger::ThreadBase::clearPowerManager() 648{ 649 Mutex::Autolock _l(mLock); 650 releaseWakeLock_l(); 651 mPowerManager.clear(); 652} 653 654void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 655{ 656 sp<ThreadBase> thread = mThread.promote(); 657 if (thread != 0) { 658 thread->clearPowerManager(); 659 } 660 ALOGW("power manager service died !!!"); 661} 662 663void AudioFlinger::ThreadBase::setEffectSuspended( 664 const effect_uuid_t *type, bool suspend, int sessionId) 665{ 666 Mutex::Autolock _l(mLock); 667 setEffectSuspended_l(type, suspend, sessionId); 668} 669 670void AudioFlinger::ThreadBase::setEffectSuspended_l( 671 const effect_uuid_t *type, bool suspend, int sessionId) 672{ 673 sp<EffectChain> chain = getEffectChain_l(sessionId); 674 if (chain != 0) { 675 if (type != NULL) { 676 chain->setEffectSuspended_l(type, suspend); 677 } else { 678 chain->setEffectSuspendedAll_l(suspend); 679 } 680 } 681 682 updateSuspendedSessions_l(type, suspend, sessionId); 683} 684 685void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 686{ 687 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 688 if (index < 0) { 689 return; 690 } 691 692 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 693 mSuspendedSessions.valueAt(index); 694 695 for (size_t i = 0; i < sessionEffects.size(); i++) { 696 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 697 for (int j = 0; j < desc->mRefCount; j++) { 698 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 699 chain->setEffectSuspendedAll_l(true); 700 } else { 701 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 702 desc->mType.timeLow); 703 chain->setEffectSuspended_l(&desc->mType, true); 704 } 705 } 706 } 707} 708 709void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 710 bool suspend, 711 int sessionId) 712{ 713 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 714 715 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 716 717 if (suspend) { 718 if (index >= 0) { 719 sessionEffects = mSuspendedSessions.valueAt(index); 720 } else { 721 mSuspendedSessions.add(sessionId, sessionEffects); 722 } 723 } else { 724 if (index < 0) { 725 return; 726 } 727 sessionEffects = mSuspendedSessions.valueAt(index); 728 } 729 730 731 int key = EffectChain::kKeyForSuspendAll; 732 if (type != NULL) { 733 key = type->timeLow; 734 } 735 index = sessionEffects.indexOfKey(key); 736 737 sp<SuspendedSessionDesc> desc; 738 if (suspend) { 739 if (index >= 0) { 740 desc = sessionEffects.valueAt(index); 741 } else { 742 desc = new SuspendedSessionDesc(); 743 if (type != NULL) { 744 desc->mType = *type; 745 } 746 sessionEffects.add(key, desc); 747 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 748 } 749 desc->mRefCount++; 750 } else { 751 if (index < 0) { 752 return; 753 } 754 desc = sessionEffects.valueAt(index); 755 if (--desc->mRefCount == 0) { 756 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 757 sessionEffects.removeItemsAt(index); 758 if (sessionEffects.isEmpty()) { 759 ALOGV("updateSuspendedSessions_l() restore removing session %d", 760 sessionId); 761 mSuspendedSessions.removeItem(sessionId); 762 } 763 } 764 } 765 if (!sessionEffects.isEmpty()) { 766 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 767 } 768} 769 770void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 771 bool enabled, 772 int sessionId) 773{ 774 Mutex::Autolock _l(mLock); 775 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 776} 777 778void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 779 bool enabled, 780 int sessionId) 781{ 782 if (mType != RECORD) { 783 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 784 // another session. This gives the priority to well behaved effect control panels 785 // and applications not using global effects. 786 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 787 // global effects 788 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 789 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 790 } 791 } 792 793 sp<EffectChain> chain = getEffectChain_l(sessionId); 794 if (chain != 0) { 795 chain->checkSuspendOnEffectEnabled(effect, enabled); 796 } 797} 798 799// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 800sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 801 const sp<AudioFlinger::Client>& client, 802 const sp<IEffectClient>& effectClient, 803 int32_t priority, 804 int sessionId, 805 effect_descriptor_t *desc, 806 int *enabled, 807 status_t *status) 808{ 809 sp<EffectModule> effect; 810 sp<EffectHandle> handle; 811 status_t lStatus; 812 sp<EffectChain> chain; 813 bool chainCreated = false; 814 bool effectCreated = false; 815 bool effectRegistered = false; 816 817 lStatus = initCheck(); 818 if (lStatus != NO_ERROR) { 819 ALOGW("createEffect_l() Audio driver not initialized."); 820 goto Exit; 821 } 822 823 // Allow global effects only on offloaded and mixer threads 824 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 825 switch (mType) { 826 case MIXER: 827 case OFFLOAD: 828 break; 829 case DIRECT: 830 case DUPLICATING: 831 case RECORD: 832 default: 833 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 834 lStatus = BAD_VALUE; 835 goto Exit; 836 } 837 } 838 839 // Only Pre processor effects are allowed on input threads and only on input threads 840 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 841 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 842 desc->name, desc->flags, mType); 843 lStatus = BAD_VALUE; 844 goto Exit; 845 } 846 847 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 848 849 { // scope for mLock 850 Mutex::Autolock _l(mLock); 851 852 // check for existing effect chain with the requested audio session 853 chain = getEffectChain_l(sessionId); 854 if (chain == 0) { 855 // create a new chain for this session 856 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 857 chain = new EffectChain(this, sessionId); 858 addEffectChain_l(chain); 859 chain->setStrategy(getStrategyForSession_l(sessionId)); 860 chainCreated = true; 861 } else { 862 effect = chain->getEffectFromDesc_l(desc); 863 } 864 865 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 866 867 if (effect == 0) { 868 int id = mAudioFlinger->nextUniqueId(); 869 // Check CPU and memory usage 870 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 871 if (lStatus != NO_ERROR) { 872 goto Exit; 873 } 874 effectRegistered = true; 875 // create a new effect module if none present in the chain 876 effect = new EffectModule(this, chain, desc, id, sessionId); 877 lStatus = effect->status(); 878 if (lStatus != NO_ERROR) { 879 goto Exit; 880 } 881 effect->setOffloaded(mType == OFFLOAD, mId); 882 883 lStatus = chain->addEffect_l(effect); 884 if (lStatus != NO_ERROR) { 885 goto Exit; 886 } 887 effectCreated = true; 888 889 effect->setDevice(mOutDevice); 890 effect->setDevice(mInDevice); 891 effect->setMode(mAudioFlinger->getMode()); 892 effect->setAudioSource(mAudioSource); 893 } 894 // create effect handle and connect it to effect module 895 handle = new EffectHandle(effect, client, effectClient, priority); 896 lStatus = handle->initCheck(); 897 if (lStatus == OK) { 898 lStatus = effect->addHandle(handle.get()); 899 } 900 if (enabled != NULL) { 901 *enabled = (int)effect->isEnabled(); 902 } 903 } 904 905Exit: 906 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 907 Mutex::Autolock _l(mLock); 908 if (effectCreated) { 909 chain->removeEffect_l(effect); 910 } 911 if (effectRegistered) { 912 AudioSystem::unregisterEffect(effect->id()); 913 } 914 if (chainCreated) { 915 removeEffectChain_l(chain); 916 } 917 handle.clear(); 918 } 919 920 *status = lStatus; 921 return handle; 922} 923 924sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 925{ 926 Mutex::Autolock _l(mLock); 927 return getEffect_l(sessionId, effectId); 928} 929 930sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 931{ 932 sp<EffectChain> chain = getEffectChain_l(sessionId); 933 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 934} 935 936// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 937// PlaybackThread::mLock held 938status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 939{ 940 // check for existing effect chain with the requested audio session 941 int sessionId = effect->sessionId(); 942 sp<EffectChain> chain = getEffectChain_l(sessionId); 943 bool chainCreated = false; 944 945 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 946 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 947 this, effect->desc().name, effect->desc().flags); 948 949 if (chain == 0) { 950 // create a new chain for this session 951 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 952 chain = new EffectChain(this, sessionId); 953 addEffectChain_l(chain); 954 chain->setStrategy(getStrategyForSession_l(sessionId)); 955 chainCreated = true; 956 } 957 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 958 959 if (chain->getEffectFromId_l(effect->id()) != 0) { 960 ALOGW("addEffect_l() %p effect %s already present in chain %p", 961 this, effect->desc().name, chain.get()); 962 return BAD_VALUE; 963 } 964 965 effect->setOffloaded(mType == OFFLOAD, mId); 966 967 status_t status = chain->addEffect_l(effect); 968 if (status != NO_ERROR) { 969 if (chainCreated) { 970 removeEffectChain_l(chain); 971 } 972 return status; 973 } 974 975 effect->setDevice(mOutDevice); 976 effect->setDevice(mInDevice); 977 effect->setMode(mAudioFlinger->getMode()); 978 effect->setAudioSource(mAudioSource); 979 return NO_ERROR; 980} 981 982void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 983 984 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 985 effect_descriptor_t desc = effect->desc(); 986 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 987 detachAuxEffect_l(effect->id()); 988 } 989 990 sp<EffectChain> chain = effect->chain().promote(); 991 if (chain != 0) { 992 // remove effect chain if removing last effect 993 if (chain->removeEffect_l(effect) == 0) { 994 removeEffectChain_l(chain); 995 } 996 } else { 997 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 998 } 999} 1000 1001void AudioFlinger::ThreadBase::lockEffectChains_l( 1002 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1003{ 1004 effectChains = mEffectChains; 1005 for (size_t i = 0; i < mEffectChains.size(); i++) { 1006 mEffectChains[i]->lock(); 1007 } 1008} 1009 1010void AudioFlinger::ThreadBase::unlockEffectChains( 1011 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1012{ 1013 for (size_t i = 0; i < effectChains.size(); i++) { 1014 effectChains[i]->unlock(); 1015 } 1016} 1017 1018sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1019{ 1020 Mutex::Autolock _l(mLock); 1021 return getEffectChain_l(sessionId); 1022} 1023 1024sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1025{ 1026 size_t size = mEffectChains.size(); 1027 for (size_t i = 0; i < size; i++) { 1028 if (mEffectChains[i]->sessionId() == sessionId) { 1029 return mEffectChains[i]; 1030 } 1031 } 1032 return 0; 1033} 1034 1035void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1036{ 1037 Mutex::Autolock _l(mLock); 1038 size_t size = mEffectChains.size(); 1039 for (size_t i = 0; i < size; i++) { 1040 mEffectChains[i]->setMode_l(mode); 1041 } 1042} 1043 1044void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1045 EffectHandle *handle, 1046 bool unpinIfLast) { 1047 1048 Mutex::Autolock _l(mLock); 1049 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1050 // delete the effect module if removing last handle on it 1051 if (effect->removeHandle(handle) == 0) { 1052 if (!effect->isPinned() || unpinIfLast) { 1053 removeEffect_l(effect); 1054 AudioSystem::unregisterEffect(effect->id()); 1055 } 1056 } 1057} 1058 1059// ---------------------------------------------------------------------------- 1060// Playback 1061// ---------------------------------------------------------------------------- 1062 1063AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1064 AudioStreamOut* output, 1065 audio_io_handle_t id, 1066 audio_devices_t device, 1067 type_t type) 1068 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1069 mNormalFrameCount(0), mSinkBuffer(NULL), 1070 mMixerBufferEnabled(false), 1071 mMixerBuffer(NULL), 1072 mMixerBufferSize(0), 1073 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1074 mMixerBufferValid(false), 1075 mSuspended(0), mBytesWritten(0), 1076 mActiveTracksGeneration(0), 1077 // mStreamTypes[] initialized in constructor body 1078 mOutput(output), 1079 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1080 mMixerStatus(MIXER_IDLE), 1081 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1082 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1083 mBytesRemaining(0), 1084 mCurrentWriteLength(0), 1085 mUseAsyncWrite(false), 1086 mWriteAckSequence(0), 1087 mDrainSequence(0), 1088 mSignalPending(false), 1089 mScreenState(AudioFlinger::mScreenState), 1090 // index 0 is reserved for normal mixer's submix 1091 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1092 // mLatchD, mLatchQ, 1093 mLatchDValid(false), mLatchQValid(false) 1094{ 1095 snprintf(mName, kNameLength, "AudioOut_%X", id); 1096 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1097 1098 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1099 // it would be safer to explicitly pass initial masterVolume/masterMute as 1100 // parameter. 1101 // 1102 // If the HAL we are using has support for master volume or master mute, 1103 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1104 // and the mute set to false). 1105 mMasterVolume = audioFlinger->masterVolume_l(); 1106 mMasterMute = audioFlinger->masterMute_l(); 1107 if (mOutput && mOutput->audioHwDev) { 1108 if (mOutput->audioHwDev->canSetMasterVolume()) { 1109 mMasterVolume = 1.0; 1110 } 1111 1112 if (mOutput->audioHwDev->canSetMasterMute()) { 1113 mMasterMute = false; 1114 } 1115 } 1116 1117 readOutputParameters_l(); 1118 1119 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1120 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1121 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1122 stream = (audio_stream_type_t) (stream + 1)) { 1123 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1124 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1125 } 1126 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1127 // because mAudioFlinger doesn't have one to copy from 1128} 1129 1130AudioFlinger::PlaybackThread::~PlaybackThread() 1131{ 1132 mAudioFlinger->unregisterWriter(mNBLogWriter); 1133 delete[] mSinkBuffer; 1134 free(mMixerBuffer); 1135} 1136 1137void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1138{ 1139 dumpInternals(fd, args); 1140 dumpTracks(fd, args); 1141 dumpEffectChains(fd, args); 1142} 1143 1144void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1145{ 1146 const size_t SIZE = 256; 1147 char buffer[SIZE]; 1148 String8 result; 1149 1150 result.appendFormat(" Stream volumes in dB: "); 1151 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1152 const stream_type_t *st = &mStreamTypes[i]; 1153 if (i > 0) { 1154 result.appendFormat(", "); 1155 } 1156 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1157 if (st->mute) { 1158 result.append("M"); 1159 } 1160 } 1161 result.append("\n"); 1162 write(fd, result.string(), result.length()); 1163 result.clear(); 1164 1165 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1166 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1167 fdprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1168 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1169 1170 size_t numtracks = mTracks.size(); 1171 size_t numactive = mActiveTracks.size(); 1172 fdprintf(fd, " %d Tracks", numtracks); 1173 size_t numactiveseen = 0; 1174 if (numtracks) { 1175 fdprintf(fd, " of which %d are active\n", numactive); 1176 Track::appendDumpHeader(result); 1177 for (size_t i = 0; i < numtracks; ++i) { 1178 sp<Track> track = mTracks[i]; 1179 if (track != 0) { 1180 bool active = mActiveTracks.indexOf(track) >= 0; 1181 if (active) { 1182 numactiveseen++; 1183 } 1184 track->dump(buffer, SIZE, active); 1185 result.append(buffer); 1186 } 1187 } 1188 } else { 1189 result.append("\n"); 1190 } 1191 if (numactiveseen != numactive) { 1192 // some tracks in the active list were not in the tracks list 1193 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1194 " not in the track list\n"); 1195 result.append(buffer); 1196 Track::appendDumpHeader(result); 1197 for (size_t i = 0; i < numactive; ++i) { 1198 sp<Track> track = mActiveTracks[i].promote(); 1199 if (track != 0 && mTracks.indexOf(track) < 0) { 1200 track->dump(buffer, SIZE, true); 1201 result.append(buffer); 1202 } 1203 } 1204 } 1205 1206 write(fd, result.string(), result.size()); 1207 1208} 1209 1210void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1211{ 1212 fdprintf(fd, "\nOutput thread %p:\n", this); 1213 fdprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1214 fdprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1215 fdprintf(fd, " Total writes: %d\n", mNumWrites); 1216 fdprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1217 fdprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1218 fdprintf(fd, " Suspend count: %d\n", mSuspended); 1219 fdprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1220 fdprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1221 fdprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1222 1223 dumpBase(fd, args); 1224} 1225 1226// Thread virtuals 1227 1228void AudioFlinger::PlaybackThread::onFirstRef() 1229{ 1230 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1231} 1232 1233// ThreadBase virtuals 1234void AudioFlinger::PlaybackThread::preExit() 1235{ 1236 ALOGV(" preExit()"); 1237 // FIXME this is using hard-coded strings but in the future, this functionality will be 1238 // converted to use audio HAL extensions required to support tunneling 1239 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1240} 1241 1242// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1243sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1244 const sp<AudioFlinger::Client>& client, 1245 audio_stream_type_t streamType, 1246 uint32_t sampleRate, 1247 audio_format_t format, 1248 audio_channel_mask_t channelMask, 1249 size_t *pFrameCount, 1250 const sp<IMemory>& sharedBuffer, 1251 int sessionId, 1252 IAudioFlinger::track_flags_t *flags, 1253 pid_t tid, 1254 int uid, 1255 status_t *status) 1256{ 1257 size_t frameCount = *pFrameCount; 1258 sp<Track> track; 1259 status_t lStatus; 1260 1261 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1262 1263 // client expresses a preference for FAST, but we get the final say 1264 if (*flags & IAudioFlinger::TRACK_FAST) { 1265 if ( 1266 // not timed 1267 (!isTimed) && 1268 // either of these use cases: 1269 ( 1270 // use case 1: shared buffer with any frame count 1271 ( 1272 (sharedBuffer != 0) 1273 ) || 1274 // use case 2: callback handler and frame count is default or at least as large as HAL 1275 ( 1276 (tid != -1) && 1277 ((frameCount == 0) || 1278 (frameCount >= mFrameCount)) 1279 ) 1280 ) && 1281 // PCM data 1282 audio_is_linear_pcm(format) && 1283 // mono or stereo 1284 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1285 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1286 // hardware sample rate 1287 (sampleRate == mSampleRate) && 1288 // normal mixer has an associated fast mixer 1289 hasFastMixer() && 1290 // there are sufficient fast track slots available 1291 (mFastTrackAvailMask != 0) 1292 // FIXME test that MixerThread for this fast track has a capable output HAL 1293 // FIXME add a permission test also? 1294 ) { 1295 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1296 if (frameCount == 0) { 1297 frameCount = mFrameCount * kFastTrackMultiplier; 1298 } 1299 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1300 frameCount, mFrameCount); 1301 } else { 1302 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1303 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1304 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1305 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1306 audio_is_linear_pcm(format), 1307 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1308 *flags &= ~IAudioFlinger::TRACK_FAST; 1309 // For compatibility with AudioTrack calculation, buffer depth is forced 1310 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1311 // This is probably too conservative, but legacy application code may depend on it. 1312 // If you change this calculation, also review the start threshold which is related. 1313 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1314 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1315 if (minBufCount < 2) { 1316 minBufCount = 2; 1317 } 1318 size_t minFrameCount = mNormalFrameCount * minBufCount; 1319 if (frameCount < minFrameCount) { 1320 frameCount = minFrameCount; 1321 } 1322 } 1323 } 1324 *pFrameCount = frameCount; 1325 1326 if (mType == DIRECT) { 1327 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1328 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1329 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1330 "for output %p with format %#x", 1331 sampleRate, format, channelMask, mOutput, mFormat); 1332 lStatus = BAD_VALUE; 1333 goto Exit; 1334 } 1335 } 1336 } else if (mType == OFFLOAD) { 1337 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1338 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1339 "for output %p with format %#x", 1340 sampleRate, format, channelMask, mOutput, mFormat); 1341 lStatus = BAD_VALUE; 1342 goto Exit; 1343 } 1344 } else { 1345 if ((format & AUDIO_FORMAT_MAIN_MASK) != AUDIO_FORMAT_PCM) { 1346 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1347 "for output %p with format %#x", 1348 format, mOutput, mFormat); 1349 lStatus = BAD_VALUE; 1350 goto Exit; 1351 } 1352 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1353 if (sampleRate > mSampleRate*2) { 1354 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1355 lStatus = BAD_VALUE; 1356 goto Exit; 1357 } 1358 } 1359 1360 lStatus = initCheck(); 1361 if (lStatus != NO_ERROR) { 1362 ALOGE("Audio driver not initialized."); 1363 goto Exit; 1364 } 1365 1366 { // scope for mLock 1367 Mutex::Autolock _l(mLock); 1368 1369 // all tracks in same audio session must share the same routing strategy otherwise 1370 // conflicts will happen when tracks are moved from one output to another by audio policy 1371 // manager 1372 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1373 for (size_t i = 0; i < mTracks.size(); ++i) { 1374 sp<Track> t = mTracks[i]; 1375 if (t != 0 && !t->isOutputTrack()) { 1376 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1377 if (sessionId == t->sessionId() && strategy != actual) { 1378 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1379 strategy, actual); 1380 lStatus = BAD_VALUE; 1381 goto Exit; 1382 } 1383 } 1384 } 1385 1386 if (!isTimed) { 1387 track = new Track(this, client, streamType, sampleRate, format, 1388 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1389 } else { 1390 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1391 channelMask, frameCount, sharedBuffer, sessionId, uid); 1392 } 1393 1394 // new Track always returns non-NULL, 1395 // but TimedTrack::create() is a factory that could fail by returning NULL 1396 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1397 if (lStatus != NO_ERROR) { 1398 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1399 // track must be cleared from the caller as the caller has the AF lock 1400 goto Exit; 1401 } 1402 1403 mTracks.add(track); 1404 1405 sp<EffectChain> chain = getEffectChain_l(sessionId); 1406 if (chain != 0) { 1407 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1408 track->setMainBuffer(chain->inBuffer()); 1409 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1410 chain->incTrackCnt(); 1411 } 1412 1413 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1414 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1415 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1416 // so ask activity manager to do this on our behalf 1417 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1418 } 1419 } 1420 1421 lStatus = NO_ERROR; 1422 1423Exit: 1424 *status = lStatus; 1425 return track; 1426} 1427 1428uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1429{ 1430 return latency; 1431} 1432 1433uint32_t AudioFlinger::PlaybackThread::latency() const 1434{ 1435 Mutex::Autolock _l(mLock); 1436 return latency_l(); 1437} 1438uint32_t AudioFlinger::PlaybackThread::latency_l() const 1439{ 1440 if (initCheck() == NO_ERROR) { 1441 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1442 } else { 1443 return 0; 1444 } 1445} 1446 1447void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1448{ 1449 Mutex::Autolock _l(mLock); 1450 // Don't apply master volume in SW if our HAL can do it for us. 1451 if (mOutput && mOutput->audioHwDev && 1452 mOutput->audioHwDev->canSetMasterVolume()) { 1453 mMasterVolume = 1.0; 1454 } else { 1455 mMasterVolume = value; 1456 } 1457} 1458 1459void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1460{ 1461 Mutex::Autolock _l(mLock); 1462 // Don't apply master mute in SW if our HAL can do it for us. 1463 if (mOutput && mOutput->audioHwDev && 1464 mOutput->audioHwDev->canSetMasterMute()) { 1465 mMasterMute = false; 1466 } else { 1467 mMasterMute = muted; 1468 } 1469} 1470 1471void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1472{ 1473 Mutex::Autolock _l(mLock); 1474 mStreamTypes[stream].volume = value; 1475 broadcast_l(); 1476} 1477 1478void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1479{ 1480 Mutex::Autolock _l(mLock); 1481 mStreamTypes[stream].mute = muted; 1482 broadcast_l(); 1483} 1484 1485float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1486{ 1487 Mutex::Autolock _l(mLock); 1488 return mStreamTypes[stream].volume; 1489} 1490 1491// addTrack_l() must be called with ThreadBase::mLock held 1492status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1493{ 1494 status_t status = ALREADY_EXISTS; 1495 1496 // set retry count for buffer fill 1497 track->mRetryCount = kMaxTrackStartupRetries; 1498 if (mActiveTracks.indexOf(track) < 0) { 1499 // the track is newly added, make sure it fills up all its 1500 // buffers before playing. This is to ensure the client will 1501 // effectively get the latency it requested. 1502 if (!track->isOutputTrack()) { 1503 TrackBase::track_state state = track->mState; 1504 mLock.unlock(); 1505 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1506 mLock.lock(); 1507 // abort track was stopped/paused while we released the lock 1508 if (state != track->mState) { 1509 if (status == NO_ERROR) { 1510 mLock.unlock(); 1511 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1512 mLock.lock(); 1513 } 1514 return INVALID_OPERATION; 1515 } 1516 // abort if start is rejected by audio policy manager 1517 if (status != NO_ERROR) { 1518 return PERMISSION_DENIED; 1519 } 1520#ifdef ADD_BATTERY_DATA 1521 // to track the speaker usage 1522 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1523#endif 1524 } 1525 1526 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1527 track->mResetDone = false; 1528 track->mPresentationCompleteFrames = 0; 1529 mActiveTracks.add(track); 1530 mWakeLockUids.add(track->uid()); 1531 mActiveTracksGeneration++; 1532 mLatestActiveTrack = track; 1533 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1534 if (chain != 0) { 1535 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1536 track->sessionId()); 1537 chain->incActiveTrackCnt(); 1538 } 1539 1540 status = NO_ERROR; 1541 } 1542 1543 onAddNewTrack_l(); 1544 return status; 1545} 1546 1547bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1548{ 1549 track->terminate(); 1550 // active tracks are removed by threadLoop() 1551 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1552 track->mState = TrackBase::STOPPED; 1553 if (!trackActive) { 1554 removeTrack_l(track); 1555 } else if (track->isFastTrack() || track->isOffloaded()) { 1556 track->mState = TrackBase::STOPPING_1; 1557 } 1558 1559 return trackActive; 1560} 1561 1562void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1563{ 1564 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1565 mTracks.remove(track); 1566 deleteTrackName_l(track->name()); 1567 // redundant as track is about to be destroyed, for dumpsys only 1568 track->mName = -1; 1569 if (track->isFastTrack()) { 1570 int index = track->mFastIndex; 1571 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1572 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1573 mFastTrackAvailMask |= 1 << index; 1574 // redundant as track is about to be destroyed, for dumpsys only 1575 track->mFastIndex = -1; 1576 } 1577 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1578 if (chain != 0) { 1579 chain->decTrackCnt(); 1580 } 1581} 1582 1583void AudioFlinger::PlaybackThread::broadcast_l() 1584{ 1585 // Thread could be blocked waiting for async 1586 // so signal it to handle state changes immediately 1587 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1588 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1589 mSignalPending = true; 1590 mWaitWorkCV.broadcast(); 1591} 1592 1593String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1594{ 1595 Mutex::Autolock _l(mLock); 1596 if (initCheck() != NO_ERROR) { 1597 return String8(); 1598 } 1599 1600 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1601 const String8 out_s8(s); 1602 free(s); 1603 return out_s8; 1604} 1605 1606// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1607void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1608 AudioSystem::OutputDescriptor desc; 1609 void *param2 = NULL; 1610 1611 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, 1612 param); 1613 1614 switch (event) { 1615 case AudioSystem::OUTPUT_OPENED: 1616 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1617 desc.channelMask = mChannelMask; 1618 desc.samplingRate = mSampleRate; 1619 desc.format = mFormat; 1620 desc.frameCount = mNormalFrameCount; // FIXME see 1621 // AudioFlinger::frameCount(audio_io_handle_t) 1622 desc.latency = latency(); 1623 param2 = &desc; 1624 break; 1625 1626 case AudioSystem::STREAM_CONFIG_CHANGED: 1627 param2 = ¶m; 1628 case AudioSystem::OUTPUT_CLOSED: 1629 default: 1630 break; 1631 } 1632 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1633} 1634 1635void AudioFlinger::PlaybackThread::writeCallback() 1636{ 1637 ALOG_ASSERT(mCallbackThread != 0); 1638 mCallbackThread->resetWriteBlocked(); 1639} 1640 1641void AudioFlinger::PlaybackThread::drainCallback() 1642{ 1643 ALOG_ASSERT(mCallbackThread != 0); 1644 mCallbackThread->resetDraining(); 1645} 1646 1647void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1648{ 1649 Mutex::Autolock _l(mLock); 1650 // reject out of sequence requests 1651 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1652 mWriteAckSequence &= ~1; 1653 mWaitWorkCV.signal(); 1654 } 1655} 1656 1657void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1658{ 1659 Mutex::Autolock _l(mLock); 1660 // reject out of sequence requests 1661 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1662 mDrainSequence &= ~1; 1663 mWaitWorkCV.signal(); 1664 } 1665} 1666 1667// static 1668int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1669 void *param __unused, 1670 void *cookie) 1671{ 1672 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1673 ALOGV("asyncCallback() event %d", event); 1674 switch (event) { 1675 case STREAM_CBK_EVENT_WRITE_READY: 1676 me->writeCallback(); 1677 break; 1678 case STREAM_CBK_EVENT_DRAIN_READY: 1679 me->drainCallback(); 1680 break; 1681 default: 1682 ALOGW("asyncCallback() unknown event %d", event); 1683 break; 1684 } 1685 return 0; 1686} 1687 1688void AudioFlinger::PlaybackThread::readOutputParameters_l() 1689{ 1690 // unfortunately we have no way of recovering from errors here, hence the LOG_FATAL 1691 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1692 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1693 if (!audio_is_output_channel(mChannelMask)) { 1694 LOG_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1695 } 1696 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1697 LOG_FATAL("HAL channel mask %#x not supported for mixed output; " 1698 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1699 } 1700 mChannelCount = popcount(mChannelMask); 1701 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1702 if (!audio_is_valid_format(mFormat)) { 1703 LOG_FATAL("HAL format %#x not valid for output", mFormat); 1704 } 1705 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1706 LOG_FATAL("HAL format %#x not supported for mixed output; must be AUDIO_FORMAT_PCM_16_BIT", 1707 mFormat); 1708 } 1709 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1710 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1711 mFrameCount = mBufferSize / mFrameSize; 1712 if (mFrameCount & 15) { 1713 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1714 mFrameCount); 1715 } 1716 1717 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1718 (mOutput->stream->set_callback != NULL)) { 1719 if (mOutput->stream->set_callback(mOutput->stream, 1720 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1721 mUseAsyncWrite = true; 1722 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1723 } 1724 } 1725 1726 // Calculate size of normal sink buffer relative to the HAL output buffer size 1727 double multiplier = 1.0; 1728 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1729 kUseFastMixer == FastMixer_Dynamic)) { 1730 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1731 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1732 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1733 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1734 maxNormalFrameCount = maxNormalFrameCount & ~15; 1735 if (maxNormalFrameCount < minNormalFrameCount) { 1736 maxNormalFrameCount = minNormalFrameCount; 1737 } 1738 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1739 if (multiplier <= 1.0) { 1740 multiplier = 1.0; 1741 } else if (multiplier <= 2.0) { 1742 if (2 * mFrameCount <= maxNormalFrameCount) { 1743 multiplier = 2.0; 1744 } else { 1745 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1746 } 1747 } else { 1748 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1749 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1750 // track, but we sometimes have to do this to satisfy the maximum frame count 1751 // constraint) 1752 // FIXME this rounding up should not be done if no HAL SRC 1753 uint32_t truncMult = (uint32_t) multiplier; 1754 if ((truncMult & 1)) { 1755 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1756 ++truncMult; 1757 } 1758 } 1759 multiplier = (double) truncMult; 1760 } 1761 } 1762 mNormalFrameCount = multiplier * mFrameCount; 1763 // round up to nearest 16 frames to satisfy AudioMixer 1764 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1765 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1766 mNormalFrameCount); 1767 1768 delete[] mSinkBuffer; 1769 size_t normalBufferSize = mNormalFrameCount * mFrameSize; 1770 // For historical reasons mSinkBuffer is int16_t[], but mFrameSize can be odd (such as 1) 1771 mSinkBuffer = new int16_t[(normalBufferSize + 1) >> 1]; 1772 memset(mSinkBuffer, 0, normalBufferSize); 1773 1774 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1775 // drives the output. 1776 free(mMixerBuffer); 1777 mMixerBuffer = NULL; 1778 if (mMixerBufferEnabled) { 1779 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1780 mMixerBufferSize = mNormalFrameCount * mChannelCount 1781 * audio_bytes_per_sample(mMixerBufferFormat); 1782 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1783 } 1784 1785 // force reconfiguration of effect chains and engines to take new buffer size and audio 1786 // parameters into account 1787 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1788 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1789 // matter. 1790 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1791 Vector< sp<EffectChain> > effectChains = mEffectChains; 1792 for (size_t i = 0; i < effectChains.size(); i ++) { 1793 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1794 } 1795} 1796 1797 1798status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1799{ 1800 if (halFrames == NULL || dspFrames == NULL) { 1801 return BAD_VALUE; 1802 } 1803 Mutex::Autolock _l(mLock); 1804 if (initCheck() != NO_ERROR) { 1805 return INVALID_OPERATION; 1806 } 1807 size_t framesWritten = mBytesWritten / mFrameSize; 1808 *halFrames = framesWritten; 1809 1810 if (isSuspended()) { 1811 // return an estimation of rendered frames when the output is suspended 1812 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1813 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1814 return NO_ERROR; 1815 } else { 1816 status_t status; 1817 uint32_t frames; 1818 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1819 *dspFrames = (size_t)frames; 1820 return status; 1821 } 1822} 1823 1824uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1825{ 1826 Mutex::Autolock _l(mLock); 1827 uint32_t result = 0; 1828 if (getEffectChain_l(sessionId) != 0) { 1829 result = EFFECT_SESSION; 1830 } 1831 1832 for (size_t i = 0; i < mTracks.size(); ++i) { 1833 sp<Track> track = mTracks[i]; 1834 if (sessionId == track->sessionId() && !track->isInvalid()) { 1835 result |= TRACK_SESSION; 1836 break; 1837 } 1838 } 1839 1840 return result; 1841} 1842 1843uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1844{ 1845 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1846 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1847 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1848 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1849 } 1850 for (size_t i = 0; i < mTracks.size(); i++) { 1851 sp<Track> track = mTracks[i]; 1852 if (sessionId == track->sessionId() && !track->isInvalid()) { 1853 return AudioSystem::getStrategyForStream(track->streamType()); 1854 } 1855 } 1856 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1857} 1858 1859 1860AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1861{ 1862 Mutex::Autolock _l(mLock); 1863 return mOutput; 1864} 1865 1866AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1867{ 1868 Mutex::Autolock _l(mLock); 1869 AudioStreamOut *output = mOutput; 1870 mOutput = NULL; 1871 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1872 // must push a NULL and wait for ack 1873 mOutputSink.clear(); 1874 mPipeSink.clear(); 1875 mNormalSink.clear(); 1876 return output; 1877} 1878 1879// this method must always be called either with ThreadBase mLock held or inside the thread loop 1880audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1881{ 1882 if (mOutput == NULL) { 1883 return NULL; 1884 } 1885 return &mOutput->stream->common; 1886} 1887 1888uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1889{ 1890 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1891} 1892 1893status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1894{ 1895 if (!isValidSyncEvent(event)) { 1896 return BAD_VALUE; 1897 } 1898 1899 Mutex::Autolock _l(mLock); 1900 1901 for (size_t i = 0; i < mTracks.size(); ++i) { 1902 sp<Track> track = mTracks[i]; 1903 if (event->triggerSession() == track->sessionId()) { 1904 (void) track->setSyncEvent(event); 1905 return NO_ERROR; 1906 } 1907 } 1908 1909 return NAME_NOT_FOUND; 1910} 1911 1912bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1913{ 1914 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1915} 1916 1917void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1918 const Vector< sp<Track> >& tracksToRemove) 1919{ 1920 size_t count = tracksToRemove.size(); 1921 if (count > 0) { 1922 for (size_t i = 0 ; i < count ; i++) { 1923 const sp<Track>& track = tracksToRemove.itemAt(i); 1924 if (!track->isOutputTrack()) { 1925 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1926#ifdef ADD_BATTERY_DATA 1927 // to track the speaker usage 1928 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 1929#endif 1930 if (track->isTerminated()) { 1931 AudioSystem::releaseOutput(mId); 1932 } 1933 } 1934 } 1935 } 1936} 1937 1938void AudioFlinger::PlaybackThread::checkSilentMode_l() 1939{ 1940 if (!mMasterMute) { 1941 char value[PROPERTY_VALUE_MAX]; 1942 if (property_get("ro.audio.silent", value, "0") > 0) { 1943 char *endptr; 1944 unsigned long ul = strtoul(value, &endptr, 0); 1945 if (*endptr == '\0' && ul != 0) { 1946 ALOGD("Silence is golden"); 1947 // The setprop command will not allow a property to be changed after 1948 // the first time it is set, so we don't have to worry about un-muting. 1949 setMasterMute_l(true); 1950 } 1951 } 1952 } 1953} 1954 1955// shared by MIXER and DIRECT, overridden by DUPLICATING 1956ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 1957{ 1958 // FIXME rewrite to reduce number of system calls 1959 mLastWriteTime = systemTime(); 1960 mInWrite = true; 1961 ssize_t bytesWritten; 1962 1963 // If an NBAIO sink is present, use it to write the normal mixer's submix 1964 if (mNormalSink != 0) { 1965#define mBitShift 2 // FIXME 1966 size_t count = mBytesRemaining >> mBitShift; 1967 size_t offset = (mCurrentWriteLength - mBytesRemaining) >> 1; 1968 ATRACE_BEGIN("write"); 1969 // update the setpoint when AudioFlinger::mScreenState changes 1970 uint32_t screenState = AudioFlinger::mScreenState; 1971 if (screenState != mScreenState) { 1972 mScreenState = screenState; 1973 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1974 if (pipe != NULL) { 1975 pipe->setAvgFrames((mScreenState & 1) ? 1976 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 1977 } 1978 } 1979 ssize_t framesWritten = mNormalSink->write(mSinkBuffer + offset, count); 1980 ATRACE_END(); 1981 if (framesWritten > 0) { 1982 bytesWritten = framesWritten << mBitShift; 1983 } else { 1984 bytesWritten = framesWritten; 1985 } 1986 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 1987 if (status == NO_ERROR) { 1988 size_t totalFramesWritten = mNormalSink->framesWritten(); 1989 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 1990 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 1991 mLatchDValid = true; 1992 } 1993 } 1994 // otherwise use the HAL / AudioStreamOut directly 1995 } else { 1996 // Direct output and offload threads 1997 size_t offset = (mCurrentWriteLength - mBytesRemaining); 1998 if (mUseAsyncWrite) { 1999 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2000 mWriteAckSequence += 2; 2001 mWriteAckSequence |= 1; 2002 ALOG_ASSERT(mCallbackThread != 0); 2003 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2004 } 2005 // FIXME We should have an implementation of timestamps for direct output threads. 2006 // They are used e.g for multichannel PCM playback over HDMI. 2007 bytesWritten = mOutput->stream->write(mOutput->stream, 2008 (char *)mSinkBuffer + offset, mBytesRemaining); 2009 if (mUseAsyncWrite && 2010 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2011 // do not wait for async callback in case of error of full write 2012 mWriteAckSequence &= ~1; 2013 ALOG_ASSERT(mCallbackThread != 0); 2014 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2015 } 2016 } 2017 2018 mNumWrites++; 2019 mInWrite = false; 2020 mStandby = false; 2021 return bytesWritten; 2022} 2023 2024void AudioFlinger::PlaybackThread::threadLoop_drain() 2025{ 2026 if (mOutput->stream->drain) { 2027 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2028 if (mUseAsyncWrite) { 2029 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2030 mDrainSequence |= 1; 2031 ALOG_ASSERT(mCallbackThread != 0); 2032 mCallbackThread->setDraining(mDrainSequence); 2033 } 2034 mOutput->stream->drain(mOutput->stream, 2035 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2036 : AUDIO_DRAIN_ALL); 2037 } 2038} 2039 2040void AudioFlinger::PlaybackThread::threadLoop_exit() 2041{ 2042 // Default implementation has nothing to do 2043} 2044 2045/* 2046The derived values that are cached: 2047 - mSinkBufferSize from frame count * frame size 2048 - activeSleepTime from activeSleepTimeUs() 2049 - idleSleepTime from idleSleepTimeUs() 2050 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2051 - maxPeriod from frame count and sample rate (MIXER only) 2052 2053The parameters that affect these derived values are: 2054 - frame count 2055 - frame size 2056 - sample rate 2057 - device type: A2DP or not 2058 - device latency 2059 - format: PCM or not 2060 - active sleep time 2061 - idle sleep time 2062*/ 2063 2064void AudioFlinger::PlaybackThread::cacheParameters_l() 2065{ 2066 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2067 activeSleepTime = activeSleepTimeUs(); 2068 idleSleepTime = idleSleepTimeUs(); 2069} 2070 2071void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2072{ 2073 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2074 this, streamType, mTracks.size()); 2075 Mutex::Autolock _l(mLock); 2076 2077 size_t size = mTracks.size(); 2078 for (size_t i = 0; i < size; i++) { 2079 sp<Track> t = mTracks[i]; 2080 if (t->streamType() == streamType) { 2081 t->invalidate(); 2082 } 2083 } 2084} 2085 2086status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2087{ 2088 int session = chain->sessionId(); 2089 int16_t *buffer = mSinkBuffer; 2090 bool ownsBuffer = false; 2091 2092 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2093 if (session > 0) { 2094 // Only one effect chain can be present in direct output thread and it uses 2095 // the sink buffer as input 2096 if (mType != DIRECT) { 2097 size_t numSamples = mNormalFrameCount * mChannelCount; 2098 buffer = new int16_t[numSamples]; 2099 memset(buffer, 0, numSamples * sizeof(int16_t)); 2100 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2101 ownsBuffer = true; 2102 } 2103 2104 // Attach all tracks with same session ID to this chain. 2105 for (size_t i = 0; i < mTracks.size(); ++i) { 2106 sp<Track> track = mTracks[i]; 2107 if (session == track->sessionId()) { 2108 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2109 buffer); 2110 track->setMainBuffer(buffer); 2111 chain->incTrackCnt(); 2112 } 2113 } 2114 2115 // indicate all active tracks in the chain 2116 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2117 sp<Track> track = mActiveTracks[i].promote(); 2118 if (track == 0) { 2119 continue; 2120 } 2121 if (session == track->sessionId()) { 2122 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2123 chain->incActiveTrackCnt(); 2124 } 2125 } 2126 } 2127 2128 chain->setInBuffer(buffer, ownsBuffer); 2129 chain->setOutBuffer(mSinkBuffer); 2130 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2131 // chains list in order to be processed last as it contains output stage effects 2132 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2133 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2134 // after track specific effects and before output stage 2135 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2136 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2137 // Effect chain for other sessions are inserted at beginning of effect 2138 // chains list to be processed before output mix effects. Relative order between other 2139 // sessions is not important 2140 size_t size = mEffectChains.size(); 2141 size_t i = 0; 2142 for (i = 0; i < size; i++) { 2143 if (mEffectChains[i]->sessionId() < session) { 2144 break; 2145 } 2146 } 2147 mEffectChains.insertAt(chain, i); 2148 checkSuspendOnAddEffectChain_l(chain); 2149 2150 return NO_ERROR; 2151} 2152 2153size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2154{ 2155 int session = chain->sessionId(); 2156 2157 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2158 2159 for (size_t i = 0; i < mEffectChains.size(); i++) { 2160 if (chain == mEffectChains[i]) { 2161 mEffectChains.removeAt(i); 2162 // detach all active tracks from the chain 2163 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2164 sp<Track> track = mActiveTracks[i].promote(); 2165 if (track == 0) { 2166 continue; 2167 } 2168 if (session == track->sessionId()) { 2169 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2170 chain.get(), session); 2171 chain->decActiveTrackCnt(); 2172 } 2173 } 2174 2175 // detach all tracks with same session ID from this chain 2176 for (size_t i = 0; i < mTracks.size(); ++i) { 2177 sp<Track> track = mTracks[i]; 2178 if (session == track->sessionId()) { 2179 track->setMainBuffer(mSinkBuffer); 2180 chain->decTrackCnt(); 2181 } 2182 } 2183 break; 2184 } 2185 } 2186 return mEffectChains.size(); 2187} 2188 2189status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2190 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2191{ 2192 Mutex::Autolock _l(mLock); 2193 return attachAuxEffect_l(track, EffectId); 2194} 2195 2196status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2197 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2198{ 2199 status_t status = NO_ERROR; 2200 2201 if (EffectId == 0) { 2202 track->setAuxBuffer(0, NULL); 2203 } else { 2204 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2205 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2206 if (effect != 0) { 2207 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2208 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2209 } else { 2210 status = INVALID_OPERATION; 2211 } 2212 } else { 2213 status = BAD_VALUE; 2214 } 2215 } 2216 return status; 2217} 2218 2219void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2220{ 2221 for (size_t i = 0; i < mTracks.size(); ++i) { 2222 sp<Track> track = mTracks[i]; 2223 if (track->auxEffectId() == effectId) { 2224 attachAuxEffect_l(track, 0); 2225 } 2226 } 2227} 2228 2229bool AudioFlinger::PlaybackThread::threadLoop() 2230{ 2231 Vector< sp<Track> > tracksToRemove; 2232 2233 standbyTime = systemTime(); 2234 2235 // MIXER 2236 nsecs_t lastWarning = 0; 2237 2238 // DUPLICATING 2239 // FIXME could this be made local to while loop? 2240 writeFrames = 0; 2241 2242 int lastGeneration = 0; 2243 2244 cacheParameters_l(); 2245 sleepTime = idleSleepTime; 2246 2247 if (mType == MIXER) { 2248 sleepTimeShift = 0; 2249 } 2250 2251 CpuStats cpuStats; 2252 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2253 2254 acquireWakeLock(); 2255 2256 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2257 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2258 // and then that string will be logged at the next convenient opportunity. 2259 const char *logString = NULL; 2260 2261 checkSilentMode_l(); 2262 2263 while (!exitPending()) 2264 { 2265 cpuStats.sample(myName); 2266 2267 Vector< sp<EffectChain> > effectChains; 2268 2269 processConfigEvents(); 2270 2271 { // scope for mLock 2272 2273 Mutex::Autolock _l(mLock); 2274 2275 if (logString != NULL) { 2276 mNBLogWriter->logTimestamp(); 2277 mNBLogWriter->log(logString); 2278 logString = NULL; 2279 } 2280 2281 if (mLatchDValid) { 2282 mLatchQ = mLatchD; 2283 mLatchDValid = false; 2284 mLatchQValid = true; 2285 } 2286 2287 if (checkForNewParameters_l()) { 2288 cacheParameters_l(); 2289 } 2290 2291 saveOutputTracks(); 2292 if (mSignalPending) { 2293 // A signal was raised while we were unlocked 2294 mSignalPending = false; 2295 } else if (waitingAsyncCallback_l()) { 2296 if (exitPending()) { 2297 break; 2298 } 2299 releaseWakeLock_l(); 2300 mWakeLockUids.clear(); 2301 mActiveTracksGeneration++; 2302 ALOGV("wait async completion"); 2303 mWaitWorkCV.wait(mLock); 2304 ALOGV("async completion/wake"); 2305 acquireWakeLock_l(); 2306 standbyTime = systemTime() + standbyDelay; 2307 sleepTime = 0; 2308 2309 continue; 2310 } 2311 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2312 isSuspended()) { 2313 // put audio hardware into standby after short delay 2314 if (shouldStandby_l()) { 2315 2316 threadLoop_standby(); 2317 2318 mStandby = true; 2319 } 2320 2321 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2322 // we're about to wait, flush the binder command buffer 2323 IPCThreadState::self()->flushCommands(); 2324 2325 clearOutputTracks(); 2326 2327 if (exitPending()) { 2328 break; 2329 } 2330 2331 releaseWakeLock_l(); 2332 mWakeLockUids.clear(); 2333 mActiveTracksGeneration++; 2334 // wait until we have something to do... 2335 ALOGV("%s going to sleep", myName.string()); 2336 mWaitWorkCV.wait(mLock); 2337 ALOGV("%s waking up", myName.string()); 2338 acquireWakeLock_l(); 2339 2340 mMixerStatus = MIXER_IDLE; 2341 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2342 mBytesWritten = 0; 2343 mBytesRemaining = 0; 2344 checkSilentMode_l(); 2345 2346 standbyTime = systemTime() + standbyDelay; 2347 sleepTime = idleSleepTime; 2348 if (mType == MIXER) { 2349 sleepTimeShift = 0; 2350 } 2351 2352 continue; 2353 } 2354 } 2355 // mMixerStatusIgnoringFastTracks is also updated internally 2356 mMixerStatus = prepareTracks_l(&tracksToRemove); 2357 2358 // compare with previously applied list 2359 if (lastGeneration != mActiveTracksGeneration) { 2360 // update wakelock 2361 updateWakeLockUids_l(mWakeLockUids); 2362 lastGeneration = mActiveTracksGeneration; 2363 } 2364 2365 // prevent any changes in effect chain list and in each effect chain 2366 // during mixing and effect process as the audio buffers could be deleted 2367 // or modified if an effect is created or deleted 2368 lockEffectChains_l(effectChains); 2369 } // mLock scope ends 2370 2371 if (mBytesRemaining == 0) { 2372 mCurrentWriteLength = 0; 2373 if (mMixerStatus == MIXER_TRACKS_READY) { 2374 // threadLoop_mix() sets mCurrentWriteLength 2375 threadLoop_mix(); 2376 2377 // Merge mMixerBuffer data into mSinkBuffer 2378 // This is done pre-effects computation; if effects change to 2379 // support higher precision, this needs to move. 2380 // 2381 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2382 if (mMixerBufferValid) { 2383 if (mMixerBufferFormat == AUDIO_FORMAT_PCM_FLOAT) { 2384 memcpy_to_i16_from_float(mSinkBuffer, 2385 reinterpret_cast<float*>(mMixerBuffer), 2386 mNormalFrameCount * mChannelCount); 2387 } else { // mMixerBufferFormat == AUDIO_FORMAT_PCM_16_BIT 2388 memcpy(mSinkBuffer, 2389 mMixerBuffer, 2390 mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2391 } 2392 } 2393 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2394 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2395 // threadLoop_sleepTime sets sleepTime to 0 if data 2396 // must be written to HAL 2397 threadLoop_sleepTime(); 2398 if (sleepTime == 0) { 2399 mCurrentWriteLength = mSinkBufferSize; 2400 } 2401 } 2402 mBytesRemaining = mCurrentWriteLength; 2403 if (isSuspended()) { 2404 sleepTime = suspendSleepTimeUs(); 2405 // simulate write to HAL when suspended 2406 mBytesWritten += mSinkBufferSize; 2407 mBytesRemaining = 0; 2408 } 2409 2410 // only process effects if we're going to write 2411 if (sleepTime == 0 && mType != OFFLOAD) { 2412 for (size_t i = 0; i < effectChains.size(); i ++) { 2413 effectChains[i]->process_l(); 2414 } 2415 } 2416 } 2417 // Process effect chains for offloaded thread even if no audio 2418 // was read from audio track: process only updates effect state 2419 // and thus does have to be synchronized with audio writes but may have 2420 // to be called while waiting for async write callback 2421 if (mType == OFFLOAD) { 2422 for (size_t i = 0; i < effectChains.size(); i ++) { 2423 effectChains[i]->process_l(); 2424 } 2425 } 2426 2427 // enable changes in effect chain 2428 unlockEffectChains(effectChains); 2429 2430 if (!waitingAsyncCallback()) { 2431 // sleepTime == 0 means we must write to audio hardware 2432 if (sleepTime == 0) { 2433 if (mBytesRemaining) { 2434 ssize_t ret = threadLoop_write(); 2435 if (ret < 0) { 2436 mBytesRemaining = 0; 2437 } else { 2438 mBytesWritten += ret; 2439 mBytesRemaining -= ret; 2440 } 2441 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2442 (mMixerStatus == MIXER_DRAIN_ALL)) { 2443 threadLoop_drain(); 2444 } 2445 if (mType == MIXER) { 2446 // write blocked detection 2447 nsecs_t now = systemTime(); 2448 nsecs_t delta = now - mLastWriteTime; 2449 if (!mStandby && delta > maxPeriod) { 2450 mNumDelayedWrites++; 2451 if ((now - lastWarning) > kWarningThrottleNs) { 2452 ATRACE_NAME("underrun"); 2453 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2454 ns2ms(delta), mNumDelayedWrites, this); 2455 lastWarning = now; 2456 } 2457 } 2458 } 2459 2460 } else { 2461 usleep(sleepTime); 2462 } 2463 } 2464 2465 // Finally let go of removed track(s), without the lock held 2466 // since we can't guarantee the destructors won't acquire that 2467 // same lock. This will also mutate and push a new fast mixer state. 2468 threadLoop_removeTracks(tracksToRemove); 2469 tracksToRemove.clear(); 2470 2471 // FIXME I don't understand the need for this here; 2472 // it was in the original code but maybe the 2473 // assignment in saveOutputTracks() makes this unnecessary? 2474 clearOutputTracks(); 2475 2476 // Effect chains will be actually deleted here if they were removed from 2477 // mEffectChains list during mixing or effects processing 2478 effectChains.clear(); 2479 2480 // FIXME Note that the above .clear() is no longer necessary since effectChains 2481 // is now local to this block, but will keep it for now (at least until merge done). 2482 } 2483 2484 threadLoop_exit(); 2485 2486 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2487 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2488 // put output stream into standby mode 2489 if (!mStandby) { 2490 mOutput->stream->common.standby(&mOutput->stream->common); 2491 } 2492 } 2493 2494 releaseWakeLock(); 2495 mWakeLockUids.clear(); 2496 mActiveTracksGeneration++; 2497 2498 ALOGV("Thread %p type %d exiting", this, mType); 2499 return false; 2500} 2501 2502// removeTracks_l() must be called with ThreadBase::mLock held 2503void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2504{ 2505 size_t count = tracksToRemove.size(); 2506 if (count > 0) { 2507 for (size_t i=0 ; i<count ; i++) { 2508 const sp<Track>& track = tracksToRemove.itemAt(i); 2509 mActiveTracks.remove(track); 2510 mWakeLockUids.remove(track->uid()); 2511 mActiveTracksGeneration++; 2512 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2513 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2514 if (chain != 0) { 2515 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2516 track->sessionId()); 2517 chain->decActiveTrackCnt(); 2518 } 2519 if (track->isTerminated()) { 2520 removeTrack_l(track); 2521 } 2522 } 2523 } 2524 2525} 2526 2527status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2528{ 2529 if (mNormalSink != 0) { 2530 return mNormalSink->getTimestamp(timestamp); 2531 } 2532 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2533 uint64_t position64; 2534 int ret = mOutput->stream->get_presentation_position( 2535 mOutput->stream, &position64, ×tamp.mTime); 2536 if (ret == 0) { 2537 timestamp.mPosition = (uint32_t)position64; 2538 return NO_ERROR; 2539 } 2540 } 2541 return INVALID_OPERATION; 2542} 2543// ---------------------------------------------------------------------------- 2544 2545AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2546 audio_io_handle_t id, audio_devices_t device, type_t type) 2547 : PlaybackThread(audioFlinger, output, id, device, type), 2548 // mAudioMixer below 2549 // mFastMixer below 2550 mFastMixerFutex(0) 2551 // mOutputSink below 2552 // mPipeSink below 2553 // mNormalSink below 2554{ 2555 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2556 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2557 "mFrameCount=%d, mNormalFrameCount=%d", 2558 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2559 mNormalFrameCount); 2560 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2561 2562 // FIXME - Current mixer implementation only supports stereo output 2563 if (mChannelCount != FCC_2) { 2564 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2565 } 2566 2567 // create an NBAIO sink for the HAL output stream, and negotiate 2568 mOutputSink = new AudioStreamOutSink(output->stream); 2569 size_t numCounterOffers = 0; 2570 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2571 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2572 ALOG_ASSERT(index == 0); 2573 2574 // initialize fast mixer depending on configuration 2575 bool initFastMixer; 2576 switch (kUseFastMixer) { 2577 case FastMixer_Never: 2578 initFastMixer = false; 2579 break; 2580 case FastMixer_Always: 2581 initFastMixer = true; 2582 break; 2583 case FastMixer_Static: 2584 case FastMixer_Dynamic: 2585 initFastMixer = mFrameCount < mNormalFrameCount; 2586 break; 2587 } 2588 if (initFastMixer) { 2589 2590 // create a MonoPipe to connect our submix to FastMixer 2591 NBAIO_Format format = mOutputSink->format(); 2592 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2593 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2594 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2595 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2596 const NBAIO_Format offers[1] = {format}; 2597 size_t numCounterOffers = 0; 2598 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2599 ALOG_ASSERT(index == 0); 2600 monoPipe->setAvgFrames((mScreenState & 1) ? 2601 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2602 mPipeSink = monoPipe; 2603 2604#ifdef TEE_SINK 2605 if (mTeeSinkOutputEnabled) { 2606 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2607 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2608 numCounterOffers = 0; 2609 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2610 ALOG_ASSERT(index == 0); 2611 mTeeSink = teeSink; 2612 PipeReader *teeSource = new PipeReader(*teeSink); 2613 numCounterOffers = 0; 2614 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2615 ALOG_ASSERT(index == 0); 2616 mTeeSource = teeSource; 2617 } 2618#endif 2619 2620 // create fast mixer and configure it initially with just one fast track for our submix 2621 mFastMixer = new FastMixer(); 2622 FastMixerStateQueue *sq = mFastMixer->sq(); 2623#ifdef STATE_QUEUE_DUMP 2624 sq->setObserverDump(&mStateQueueObserverDump); 2625 sq->setMutatorDump(&mStateQueueMutatorDump); 2626#endif 2627 FastMixerState *state = sq->begin(); 2628 FastTrack *fastTrack = &state->mFastTracks[0]; 2629 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2630 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2631 fastTrack->mVolumeProvider = NULL; 2632 fastTrack->mGeneration++; 2633 state->mFastTracksGen++; 2634 state->mTrackMask = 1; 2635 // fast mixer will use the HAL output sink 2636 state->mOutputSink = mOutputSink.get(); 2637 state->mOutputSinkGen++; 2638 state->mFrameCount = mFrameCount; 2639 state->mCommand = FastMixerState::COLD_IDLE; 2640 // already done in constructor initialization list 2641 //mFastMixerFutex = 0; 2642 state->mColdFutexAddr = &mFastMixerFutex; 2643 state->mColdGen++; 2644 state->mDumpState = &mFastMixerDumpState; 2645#ifdef TEE_SINK 2646 state->mTeeSink = mTeeSink.get(); 2647#endif 2648 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2649 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2650 sq->end(); 2651 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2652 2653 // start the fast mixer 2654 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2655 pid_t tid = mFastMixer->getTid(); 2656 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2657 if (err != 0) { 2658 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2659 kPriorityFastMixer, getpid_cached, tid, err); 2660 } 2661 2662#ifdef AUDIO_WATCHDOG 2663 // create and start the watchdog 2664 mAudioWatchdog = new AudioWatchdog(); 2665 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2666 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2667 tid = mAudioWatchdog->getTid(); 2668 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2669 if (err != 0) { 2670 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2671 kPriorityFastMixer, getpid_cached, tid, err); 2672 } 2673#endif 2674 2675 } else { 2676 mFastMixer = NULL; 2677 } 2678 2679 switch (kUseFastMixer) { 2680 case FastMixer_Never: 2681 case FastMixer_Dynamic: 2682 mNormalSink = mOutputSink; 2683 break; 2684 case FastMixer_Always: 2685 mNormalSink = mPipeSink; 2686 break; 2687 case FastMixer_Static: 2688 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2689 break; 2690 } 2691} 2692 2693AudioFlinger::MixerThread::~MixerThread() 2694{ 2695 if (mFastMixer != NULL) { 2696 FastMixerStateQueue *sq = mFastMixer->sq(); 2697 FastMixerState *state = sq->begin(); 2698 if (state->mCommand == FastMixerState::COLD_IDLE) { 2699 int32_t old = android_atomic_inc(&mFastMixerFutex); 2700 if (old == -1) { 2701 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2702 } 2703 } 2704 state->mCommand = FastMixerState::EXIT; 2705 sq->end(); 2706 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2707 mFastMixer->join(); 2708 // Though the fast mixer thread has exited, it's state queue is still valid. 2709 // We'll use that extract the final state which contains one remaining fast track 2710 // corresponding to our sub-mix. 2711 state = sq->begin(); 2712 ALOG_ASSERT(state->mTrackMask == 1); 2713 FastTrack *fastTrack = &state->mFastTracks[0]; 2714 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2715 delete fastTrack->mBufferProvider; 2716 sq->end(false /*didModify*/); 2717 delete mFastMixer; 2718#ifdef AUDIO_WATCHDOG 2719 if (mAudioWatchdog != 0) { 2720 mAudioWatchdog->requestExit(); 2721 mAudioWatchdog->requestExitAndWait(); 2722 mAudioWatchdog.clear(); 2723 } 2724#endif 2725 } 2726 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2727 delete mAudioMixer; 2728} 2729 2730 2731uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2732{ 2733 if (mFastMixer != NULL) { 2734 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2735 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2736 } 2737 return latency; 2738} 2739 2740 2741void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2742{ 2743 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2744} 2745 2746ssize_t AudioFlinger::MixerThread::threadLoop_write() 2747{ 2748 // FIXME we should only do one push per cycle; confirm this is true 2749 // Start the fast mixer if it's not already running 2750 if (mFastMixer != NULL) { 2751 FastMixerStateQueue *sq = mFastMixer->sq(); 2752 FastMixerState *state = sq->begin(); 2753 if (state->mCommand != FastMixerState::MIX_WRITE && 2754 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2755 if (state->mCommand == FastMixerState::COLD_IDLE) { 2756 int32_t old = android_atomic_inc(&mFastMixerFutex); 2757 if (old == -1) { 2758 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2759 } 2760#ifdef AUDIO_WATCHDOG 2761 if (mAudioWatchdog != 0) { 2762 mAudioWatchdog->resume(); 2763 } 2764#endif 2765 } 2766 state->mCommand = FastMixerState::MIX_WRITE; 2767 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2768 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2769 sq->end(); 2770 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2771 if (kUseFastMixer == FastMixer_Dynamic) { 2772 mNormalSink = mPipeSink; 2773 } 2774 } else { 2775 sq->end(false /*didModify*/); 2776 } 2777 } 2778 return PlaybackThread::threadLoop_write(); 2779} 2780 2781void AudioFlinger::MixerThread::threadLoop_standby() 2782{ 2783 // Idle the fast mixer if it's currently running 2784 if (mFastMixer != NULL) { 2785 FastMixerStateQueue *sq = mFastMixer->sq(); 2786 FastMixerState *state = sq->begin(); 2787 if (!(state->mCommand & FastMixerState::IDLE)) { 2788 state->mCommand = FastMixerState::COLD_IDLE; 2789 state->mColdFutexAddr = &mFastMixerFutex; 2790 state->mColdGen++; 2791 mFastMixerFutex = 0; 2792 sq->end(); 2793 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2794 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2795 if (kUseFastMixer == FastMixer_Dynamic) { 2796 mNormalSink = mOutputSink; 2797 } 2798#ifdef AUDIO_WATCHDOG 2799 if (mAudioWatchdog != 0) { 2800 mAudioWatchdog->pause(); 2801 } 2802#endif 2803 } else { 2804 sq->end(false /*didModify*/); 2805 } 2806 } 2807 PlaybackThread::threadLoop_standby(); 2808} 2809 2810bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2811{ 2812 return false; 2813} 2814 2815bool AudioFlinger::PlaybackThread::shouldStandby_l() 2816{ 2817 return !mStandby; 2818} 2819 2820bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2821{ 2822 Mutex::Autolock _l(mLock); 2823 return waitingAsyncCallback_l(); 2824} 2825 2826// shared by MIXER and DIRECT, overridden by DUPLICATING 2827void AudioFlinger::PlaybackThread::threadLoop_standby() 2828{ 2829 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2830 mOutput->stream->common.standby(&mOutput->stream->common); 2831 if (mUseAsyncWrite != 0) { 2832 // discard any pending drain or write ack by incrementing sequence 2833 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2834 mDrainSequence = (mDrainSequence + 2) & ~1; 2835 ALOG_ASSERT(mCallbackThread != 0); 2836 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2837 mCallbackThread->setDraining(mDrainSequence); 2838 } 2839} 2840 2841void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2842{ 2843 ALOGV("signal playback thread"); 2844 broadcast_l(); 2845} 2846 2847void AudioFlinger::MixerThread::threadLoop_mix() 2848{ 2849 // obtain the presentation timestamp of the next output buffer 2850 int64_t pts; 2851 status_t status = INVALID_OPERATION; 2852 2853 if (mNormalSink != 0) { 2854 status = mNormalSink->getNextWriteTimestamp(&pts); 2855 } else { 2856 status = mOutputSink->getNextWriteTimestamp(&pts); 2857 } 2858 2859 if (status != NO_ERROR) { 2860 pts = AudioBufferProvider::kInvalidPTS; 2861 } 2862 2863 // mix buffers... 2864 mAudioMixer->process(pts); 2865 mCurrentWriteLength = mSinkBufferSize; 2866 // increase sleep time progressively when application underrun condition clears. 2867 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2868 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2869 // such that we would underrun the audio HAL. 2870 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2871 sleepTimeShift--; 2872 } 2873 sleepTime = 0; 2874 standbyTime = systemTime() + standbyDelay; 2875 //TODO: delay standby when effects have a tail 2876} 2877 2878void AudioFlinger::MixerThread::threadLoop_sleepTime() 2879{ 2880 // If no tracks are ready, sleep once for the duration of an output 2881 // buffer size, then write 0s to the output 2882 if (sleepTime == 0) { 2883 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2884 sleepTime = activeSleepTime >> sleepTimeShift; 2885 if (sleepTime < kMinThreadSleepTimeUs) { 2886 sleepTime = kMinThreadSleepTimeUs; 2887 } 2888 // reduce sleep time in case of consecutive application underruns to avoid 2889 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2890 // duration we would end up writing less data than needed by the audio HAL if 2891 // the condition persists. 2892 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2893 sleepTimeShift++; 2894 } 2895 } else { 2896 sleepTime = idleSleepTime; 2897 } 2898 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2899 memset(mSinkBuffer, 0, mSinkBufferSize); 2900 sleepTime = 0; 2901 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 2902 "anticipated start"); 2903 } 2904 // TODO add standby time extension fct of effect tail 2905} 2906 2907// prepareTracks_l() must be called with ThreadBase::mLock held 2908AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2909 Vector< sp<Track> > *tracksToRemove) 2910{ 2911 2912 mixer_state mixerStatus = MIXER_IDLE; 2913 // find out which tracks need to be processed 2914 size_t count = mActiveTracks.size(); 2915 size_t mixedTracks = 0; 2916 size_t tracksWithEffect = 0; 2917 // counts only _active_ fast tracks 2918 size_t fastTracks = 0; 2919 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2920 2921 float masterVolume = mMasterVolume; 2922 bool masterMute = mMasterMute; 2923 2924 if (masterMute) { 2925 masterVolume = 0; 2926 } 2927 // Delegate master volume control to effect in output mix effect chain if needed 2928 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2929 if (chain != 0) { 2930 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2931 chain->setVolume_l(&v, &v); 2932 masterVolume = (float)((v + (1 << 23)) >> 24); 2933 chain.clear(); 2934 } 2935 2936 // prepare a new state to push 2937 FastMixerStateQueue *sq = NULL; 2938 FastMixerState *state = NULL; 2939 bool didModify = false; 2940 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2941 if (mFastMixer != NULL) { 2942 sq = mFastMixer->sq(); 2943 state = sq->begin(); 2944 } 2945 2946 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 2947 2948 for (size_t i=0 ; i<count ; i++) { 2949 const sp<Track> t = mActiveTracks[i].promote(); 2950 if (t == 0) { 2951 continue; 2952 } 2953 2954 // this const just means the local variable doesn't change 2955 Track* const track = t.get(); 2956 2957 // process fast tracks 2958 if (track->isFastTrack()) { 2959 2960 // It's theoretically possible (though unlikely) for a fast track to be created 2961 // and then removed within the same normal mix cycle. This is not a problem, as 2962 // the track never becomes active so it's fast mixer slot is never touched. 2963 // The converse, of removing an (active) track and then creating a new track 2964 // at the identical fast mixer slot within the same normal mix cycle, 2965 // is impossible because the slot isn't marked available until the end of each cycle. 2966 int j = track->mFastIndex; 2967 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2968 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2969 FastTrack *fastTrack = &state->mFastTracks[j]; 2970 2971 // Determine whether the track is currently in underrun condition, 2972 // and whether it had a recent underrun. 2973 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2974 FastTrackUnderruns underruns = ftDump->mUnderruns; 2975 uint32_t recentFull = (underruns.mBitFields.mFull - 2976 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2977 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2978 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2979 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2980 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2981 uint32_t recentUnderruns = recentPartial + recentEmpty; 2982 track->mObservedUnderruns = underruns; 2983 // don't count underruns that occur while stopping or pausing 2984 // or stopped which can occur when flush() is called while active 2985 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 2986 recentUnderruns > 0) { 2987 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 2988 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 2989 } 2990 2991 // This is similar to the state machine for normal tracks, 2992 // with a few modifications for fast tracks. 2993 bool isActive = true; 2994 switch (track->mState) { 2995 case TrackBase::STOPPING_1: 2996 // track stays active in STOPPING_1 state until first underrun 2997 if (recentUnderruns > 0 || track->isTerminated()) { 2998 track->mState = TrackBase::STOPPING_2; 2999 } 3000 break; 3001 case TrackBase::PAUSING: 3002 // ramp down is not yet implemented 3003 track->setPaused(); 3004 break; 3005 case TrackBase::RESUMING: 3006 // ramp up is not yet implemented 3007 track->mState = TrackBase::ACTIVE; 3008 break; 3009 case TrackBase::ACTIVE: 3010 if (recentFull > 0 || recentPartial > 0) { 3011 // track has provided at least some frames recently: reset retry count 3012 track->mRetryCount = kMaxTrackRetries; 3013 } 3014 if (recentUnderruns == 0) { 3015 // no recent underruns: stay active 3016 break; 3017 } 3018 // there has recently been an underrun of some kind 3019 if (track->sharedBuffer() == 0) { 3020 // were any of the recent underruns "empty" (no frames available)? 3021 if (recentEmpty == 0) { 3022 // no, then ignore the partial underruns as they are allowed indefinitely 3023 break; 3024 } 3025 // there has recently been an "empty" underrun: decrement the retry counter 3026 if (--(track->mRetryCount) > 0) { 3027 break; 3028 } 3029 // indicate to client process that the track was disabled because of underrun; 3030 // it will then automatically call start() when data is available 3031 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3032 // remove from active list, but state remains ACTIVE [confusing but true] 3033 isActive = false; 3034 break; 3035 } 3036 // fall through 3037 case TrackBase::STOPPING_2: 3038 case TrackBase::PAUSED: 3039 case TrackBase::STOPPED: 3040 case TrackBase::FLUSHED: // flush() while active 3041 // Check for presentation complete if track is inactive 3042 // We have consumed all the buffers of this track. 3043 // This would be incomplete if we auto-paused on underrun 3044 { 3045 size_t audioHALFrames = 3046 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3047 size_t framesWritten = mBytesWritten / mFrameSize; 3048 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3049 // track stays in active list until presentation is complete 3050 break; 3051 } 3052 } 3053 if (track->isStopping_2()) { 3054 track->mState = TrackBase::STOPPED; 3055 } 3056 if (track->isStopped()) { 3057 // Can't reset directly, as fast mixer is still polling this track 3058 // track->reset(); 3059 // So instead mark this track as needing to be reset after push with ack 3060 resetMask |= 1 << i; 3061 } 3062 isActive = false; 3063 break; 3064 case TrackBase::IDLE: 3065 default: 3066 LOG_FATAL("unexpected track state %d", track->mState); 3067 } 3068 3069 if (isActive) { 3070 // was it previously inactive? 3071 if (!(state->mTrackMask & (1 << j))) { 3072 ExtendedAudioBufferProvider *eabp = track; 3073 VolumeProvider *vp = track; 3074 fastTrack->mBufferProvider = eabp; 3075 fastTrack->mVolumeProvider = vp; 3076 fastTrack->mChannelMask = track->mChannelMask; 3077 fastTrack->mGeneration++; 3078 state->mTrackMask |= 1 << j; 3079 didModify = true; 3080 // no acknowledgement required for newly active tracks 3081 } 3082 // cache the combined master volume and stream type volume for fast mixer; this 3083 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3084 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3085 ++fastTracks; 3086 } else { 3087 // was it previously active? 3088 if (state->mTrackMask & (1 << j)) { 3089 fastTrack->mBufferProvider = NULL; 3090 fastTrack->mGeneration++; 3091 state->mTrackMask &= ~(1 << j); 3092 didModify = true; 3093 // If any fast tracks were removed, we must wait for acknowledgement 3094 // because we're about to decrement the last sp<> on those tracks. 3095 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3096 } else { 3097 LOG_FATAL("fast track %d should have been active", j); 3098 } 3099 tracksToRemove->add(track); 3100 // Avoids a misleading display in dumpsys 3101 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3102 } 3103 continue; 3104 } 3105 3106 { // local variable scope to avoid goto warning 3107 3108 audio_track_cblk_t* cblk = track->cblk(); 3109 3110 // The first time a track is added we wait 3111 // for all its buffers to be filled before processing it 3112 int name = track->name(); 3113 // make sure that we have enough frames to mix one full buffer. 3114 // enforce this condition only once to enable draining the buffer in case the client 3115 // app does not call stop() and relies on underrun to stop: 3116 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3117 // during last round 3118 size_t desiredFrames; 3119 uint32_t sr = track->sampleRate(); 3120 if (sr == mSampleRate) { 3121 desiredFrames = mNormalFrameCount; 3122 } else { 3123 // +1 for rounding and +1 for additional sample needed for interpolation 3124 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3125 // add frames already consumed but not yet released by the resampler 3126 // because mAudioTrackServerProxy->framesReady() will include these frames 3127 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3128#if 0 3129 // the minimum track buffer size is normally twice the number of frames necessary 3130 // to fill one buffer and the resampler should not leave more than one buffer worth 3131 // of unreleased frames after each pass, but just in case... 3132 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3133#endif 3134 } 3135 uint32_t minFrames = 1; 3136 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3137 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3138 minFrames = desiredFrames; 3139 } 3140 3141 size_t framesReady = track->framesReady(); 3142 if ((framesReady >= minFrames) && track->isReady() && 3143 !track->isPaused() && !track->isTerminated()) 3144 { 3145 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3146 3147 mixedTracks++; 3148 3149 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3150 // there is an effect chain connected to the track 3151 chain.clear(); 3152 if (track->mainBuffer() != mSinkBuffer && 3153 track->mainBuffer() != mMixerBuffer) { 3154 chain = getEffectChain_l(track->sessionId()); 3155 // Delegate volume control to effect in track effect chain if needed 3156 if (chain != 0) { 3157 tracksWithEffect++; 3158 } else { 3159 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3160 "session %d", 3161 name, track->sessionId()); 3162 } 3163 } 3164 3165 3166 int param = AudioMixer::VOLUME; 3167 if (track->mFillingUpStatus == Track::FS_FILLED) { 3168 // no ramp for the first volume setting 3169 track->mFillingUpStatus = Track::FS_ACTIVE; 3170 if (track->mState == TrackBase::RESUMING) { 3171 track->mState = TrackBase::ACTIVE; 3172 param = AudioMixer::RAMP_VOLUME; 3173 } 3174 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3175 // FIXME should not make a decision based on mServer 3176 } else if (cblk->mServer != 0) { 3177 // If the track is stopped before the first frame was mixed, 3178 // do not apply ramp 3179 param = AudioMixer::RAMP_VOLUME; 3180 } 3181 3182 // compute volume for this track 3183 uint32_t vl, vr, va; 3184 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3185 vl = vr = va = 0; 3186 if (track->isPausing()) { 3187 track->setPaused(); 3188 } 3189 } else { 3190 3191 // read original volumes with volume control 3192 float typeVolume = mStreamTypes[track->streamType()].volume; 3193 float v = masterVolume * typeVolume; 3194 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3195 uint32_t vlr = proxy->getVolumeLR(); 3196 vl = vlr & 0xFFFF; 3197 vr = vlr >> 16; 3198 // track volumes come from shared memory, so can't be trusted and must be clamped 3199 if (vl > MAX_GAIN_INT) { 3200 ALOGV("Track left volume out of range: %04X", vl); 3201 vl = MAX_GAIN_INT; 3202 } 3203 if (vr > MAX_GAIN_INT) { 3204 ALOGV("Track right volume out of range: %04X", vr); 3205 vr = MAX_GAIN_INT; 3206 } 3207 // now apply the master volume and stream type volume 3208 vl = (uint32_t)(v * vl) << 12; 3209 vr = (uint32_t)(v * vr) << 12; 3210 // assuming master volume and stream type volume each go up to 1.0, 3211 // vl and vr are now in 8.24 format 3212 3213 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3214 // send level comes from shared memory and so may be corrupt 3215 if (sendLevel > MAX_GAIN_INT) { 3216 ALOGV("Track send level out of range: %04X", sendLevel); 3217 sendLevel = MAX_GAIN_INT; 3218 } 3219 va = (uint32_t)(v * sendLevel); 3220 } 3221 3222 // Delegate volume control to effect in track effect chain if needed 3223 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3224 // Do not ramp volume if volume is controlled by effect 3225 param = AudioMixer::VOLUME; 3226 track->mHasVolumeController = true; 3227 } else { 3228 // force no volume ramp when volume controller was just disabled or removed 3229 // from effect chain to avoid volume spike 3230 if (track->mHasVolumeController) { 3231 param = AudioMixer::VOLUME; 3232 } 3233 track->mHasVolumeController = false; 3234 } 3235 3236 // Convert volumes from 8.24 to 4.12 format 3237 // This additional clamping is needed in case chain->setVolume_l() overshot 3238 vl = (vl + (1 << 11)) >> 12; 3239 if (vl > MAX_GAIN_INT) { 3240 vl = MAX_GAIN_INT; 3241 } 3242 vr = (vr + (1 << 11)) >> 12; 3243 if (vr > MAX_GAIN_INT) { 3244 vr = MAX_GAIN_INT; 3245 } 3246 3247 if (va > MAX_GAIN_INT) { 3248 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3249 } 3250 3251 // XXX: these things DON'T need to be done each time 3252 mAudioMixer->setBufferProvider(name, track); 3253 mAudioMixer->enable(name); 3254 3255 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3256 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3257 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3258 mAudioMixer->setParameter( 3259 name, 3260 AudioMixer::TRACK, 3261 AudioMixer::FORMAT, (void *)track->format()); 3262 mAudioMixer->setParameter( 3263 name, 3264 AudioMixer::TRACK, 3265 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3266 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3267 uint32_t maxSampleRate = mSampleRate * 2; 3268 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3269 if (reqSampleRate == 0) { 3270 reqSampleRate = mSampleRate; 3271 } else if (reqSampleRate > maxSampleRate) { 3272 reqSampleRate = maxSampleRate; 3273 } 3274 mAudioMixer->setParameter( 3275 name, 3276 AudioMixer::RESAMPLE, 3277 AudioMixer::SAMPLE_RATE, 3278 (void *)(uintptr_t)reqSampleRate); 3279 /* 3280 * Select the appropriate output buffer for the track. 3281 * 3282 * For tracks with effects, only mSinkBuffer can be used (at this time). 3283 * 3284 * Other tracks can use mMixerBuffer for higher precision 3285 * channel accumulation. If this buffer is enabled 3286 * (mMixerBufferEnabled true), then selected tracks will accumulate 3287 * into it. 3288 * 3289 */ 3290 if (mMixerBufferEnabled 3291 && (track->mainBuffer() == mSinkBuffer 3292 || track->mainBuffer() == mMixerBuffer)) { 3293 mAudioMixer->setParameter( 3294 name, 3295 AudioMixer::TRACK, 3296 AudioMixer::SINK_FORMAT, (void *)mMixerBufferFormat); 3297 mAudioMixer->setParameter( 3298 name, 3299 AudioMixer::TRACK, 3300 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3301 // TODO: override track->mainBuffer()? 3302 mMixerBufferValid = true; 3303 } else { 3304 mAudioMixer->setParameter( 3305 name, 3306 AudioMixer::TRACK, 3307 AudioMixer::SINK_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3308 mAudioMixer->setParameter( 3309 name, 3310 AudioMixer::TRACK, 3311 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3312 } 3313 mAudioMixer->setParameter( 3314 name, 3315 AudioMixer::TRACK, 3316 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3317 3318 // reset retry count 3319 track->mRetryCount = kMaxTrackRetries; 3320 3321 // If one track is ready, set the mixer ready if: 3322 // - the mixer was not ready during previous round OR 3323 // - no other track is not ready 3324 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3325 mixerStatus != MIXER_TRACKS_ENABLED) { 3326 mixerStatus = MIXER_TRACKS_READY; 3327 } 3328 } else { 3329 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3330 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3331 } 3332 // clear effect chain input buffer if an active track underruns to avoid sending 3333 // previous audio buffer again to effects 3334 chain = getEffectChain_l(track->sessionId()); 3335 if (chain != 0) { 3336 chain->clearInputBuffer(); 3337 } 3338 3339 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3340 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3341 track->isStopped() || track->isPaused()) { 3342 // We have consumed all the buffers of this track. 3343 // Remove it from the list of active tracks. 3344 // TODO: use actual buffer filling status instead of latency when available from 3345 // audio HAL 3346 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3347 size_t framesWritten = mBytesWritten / mFrameSize; 3348 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3349 if (track->isStopped()) { 3350 track->reset(); 3351 } 3352 tracksToRemove->add(track); 3353 } 3354 } else { 3355 // No buffers for this track. Give it a few chances to 3356 // fill a buffer, then remove it from active list. 3357 if (--(track->mRetryCount) <= 0) { 3358 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3359 tracksToRemove->add(track); 3360 // indicate to client process that the track was disabled because of underrun; 3361 // it will then automatically call start() when data is available 3362 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3363 // If one track is not ready, mark the mixer also not ready if: 3364 // - the mixer was ready during previous round OR 3365 // - no other track is ready 3366 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3367 mixerStatus != MIXER_TRACKS_READY) { 3368 mixerStatus = MIXER_TRACKS_ENABLED; 3369 } 3370 } 3371 mAudioMixer->disable(name); 3372 } 3373 3374 } // local variable scope to avoid goto warning 3375track_is_ready: ; 3376 3377 } 3378 3379 // Push the new FastMixer state if necessary 3380 bool pauseAudioWatchdog = false; 3381 if (didModify) { 3382 state->mFastTracksGen++; 3383 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3384 if (kUseFastMixer == FastMixer_Dynamic && 3385 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3386 state->mCommand = FastMixerState::COLD_IDLE; 3387 state->mColdFutexAddr = &mFastMixerFutex; 3388 state->mColdGen++; 3389 mFastMixerFutex = 0; 3390 if (kUseFastMixer == FastMixer_Dynamic) { 3391 mNormalSink = mOutputSink; 3392 } 3393 // If we go into cold idle, need to wait for acknowledgement 3394 // so that fast mixer stops doing I/O. 3395 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3396 pauseAudioWatchdog = true; 3397 } 3398 } 3399 if (sq != NULL) { 3400 sq->end(didModify); 3401 sq->push(block); 3402 } 3403#ifdef AUDIO_WATCHDOG 3404 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3405 mAudioWatchdog->pause(); 3406 } 3407#endif 3408 3409 // Now perform the deferred reset on fast tracks that have stopped 3410 while (resetMask != 0) { 3411 size_t i = __builtin_ctz(resetMask); 3412 ALOG_ASSERT(i < count); 3413 resetMask &= ~(1 << i); 3414 sp<Track> t = mActiveTracks[i].promote(); 3415 if (t == 0) { 3416 continue; 3417 } 3418 Track* track = t.get(); 3419 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3420 track->reset(); 3421 } 3422 3423 // remove all the tracks that need to be... 3424 removeTracks_l(*tracksToRemove); 3425 3426 // sink or mix buffer must be cleared if all tracks are connected to an 3427 // effect chain as in this case the mixer will not write to the sink or mix buffer 3428 // and track effects will accumulate into it 3429 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3430 (mixedTracks == 0 && fastTracks > 0))) { 3431 // FIXME as a performance optimization, should remember previous zero status 3432 if (mMixerBufferValid) { 3433 memset(mMixerBuffer, 0, mMixerBufferSize); 3434 // TODO: In testing, mSinkBuffer below need not be cleared because 3435 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3436 // after mixing. 3437 // 3438 // To enforce this guarantee: 3439 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3440 // (mixedTracks == 0 && fastTracks > 0)) 3441 // must imply MIXER_TRACKS_READY. 3442 // Later, we may clear buffers regardless, and skip much of this logic. 3443 } 3444 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3445 } 3446 3447 // if any fast tracks, then status is ready 3448 mMixerStatusIgnoringFastTracks = mixerStatus; 3449 if (fastTracks > 0) { 3450 mixerStatus = MIXER_TRACKS_READY; 3451 } 3452 return mixerStatus; 3453} 3454 3455// getTrackName_l() must be called with ThreadBase::mLock held 3456int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, int sessionId) 3457{ 3458 return mAudioMixer->getTrackName(channelMask, sessionId); 3459} 3460 3461// deleteTrackName_l() must be called with ThreadBase::mLock held 3462void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3463{ 3464 ALOGV("remove track (%d) and delete from mixer", name); 3465 mAudioMixer->deleteTrackName(name); 3466} 3467 3468// checkForNewParameters_l() must be called with ThreadBase::mLock held 3469bool AudioFlinger::MixerThread::checkForNewParameters_l() 3470{ 3471 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3472 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3473 bool reconfig = false; 3474 3475 while (!mNewParameters.isEmpty()) { 3476 3477 if (mFastMixer != NULL) { 3478 FastMixerStateQueue *sq = mFastMixer->sq(); 3479 FastMixerState *state = sq->begin(); 3480 if (!(state->mCommand & FastMixerState::IDLE)) { 3481 previousCommand = state->mCommand; 3482 state->mCommand = FastMixerState::HOT_IDLE; 3483 sq->end(); 3484 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3485 } else { 3486 sq->end(false /*didModify*/); 3487 } 3488 } 3489 3490 status_t status = NO_ERROR; 3491 String8 keyValuePair = mNewParameters[0]; 3492 AudioParameter param = AudioParameter(keyValuePair); 3493 int value; 3494 3495 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3496 reconfig = true; 3497 } 3498 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3499 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3500 status = BAD_VALUE; 3501 } else { 3502 // no need to save value, since it's constant 3503 reconfig = true; 3504 } 3505 } 3506 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3507 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3508 status = BAD_VALUE; 3509 } else { 3510 // no need to save value, since it's constant 3511 reconfig = true; 3512 } 3513 } 3514 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3515 // do not accept frame count changes if tracks are open as the track buffer 3516 // size depends on frame count and correct behavior would not be guaranteed 3517 // if frame count is changed after track creation 3518 if (!mTracks.isEmpty()) { 3519 status = INVALID_OPERATION; 3520 } else { 3521 reconfig = true; 3522 } 3523 } 3524 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3525#ifdef ADD_BATTERY_DATA 3526 // when changing the audio output device, call addBatteryData to notify 3527 // the change 3528 if (mOutDevice != value) { 3529 uint32_t params = 0; 3530 // check whether speaker is on 3531 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3532 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3533 } 3534 3535 audio_devices_t deviceWithoutSpeaker 3536 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3537 // check if any other device (except speaker) is on 3538 if (value & deviceWithoutSpeaker ) { 3539 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3540 } 3541 3542 if (params != 0) { 3543 addBatteryData(params); 3544 } 3545 } 3546#endif 3547 3548 // forward device change to effects that have requested to be 3549 // aware of attached audio device. 3550 if (value != AUDIO_DEVICE_NONE) { 3551 mOutDevice = value; 3552 for (size_t i = 0; i < mEffectChains.size(); i++) { 3553 mEffectChains[i]->setDevice_l(mOutDevice); 3554 } 3555 } 3556 } 3557 3558 if (status == NO_ERROR) { 3559 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3560 keyValuePair.string()); 3561 if (!mStandby && status == INVALID_OPERATION) { 3562 mOutput->stream->common.standby(&mOutput->stream->common); 3563 mStandby = true; 3564 mBytesWritten = 0; 3565 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3566 keyValuePair.string()); 3567 } 3568 if (status == NO_ERROR && reconfig) { 3569 readOutputParameters_l(); 3570 delete mAudioMixer; 3571 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3572 for (size_t i = 0; i < mTracks.size() ; i++) { 3573 int name = getTrackName_l(mTracks[i]->mChannelMask, mTracks[i]->mSessionId); 3574 if (name < 0) { 3575 break; 3576 } 3577 mTracks[i]->mName = name; 3578 } 3579 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3580 } 3581 } 3582 3583 mNewParameters.removeAt(0); 3584 3585 mParamStatus = status; 3586 mParamCond.signal(); 3587 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3588 // already timed out waiting for the status and will never signal the condition. 3589 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3590 } 3591 3592 if (!(previousCommand & FastMixerState::IDLE)) { 3593 ALOG_ASSERT(mFastMixer != NULL); 3594 FastMixerStateQueue *sq = mFastMixer->sq(); 3595 FastMixerState *state = sq->begin(); 3596 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3597 state->mCommand = previousCommand; 3598 sq->end(); 3599 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3600 } 3601 3602 return reconfig; 3603} 3604 3605 3606void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3607{ 3608 const size_t SIZE = 256; 3609 char buffer[SIZE]; 3610 String8 result; 3611 3612 PlaybackThread::dumpInternals(fd, args); 3613 3614 fdprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3615 3616 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3617 const FastMixerDumpState copy(mFastMixerDumpState); 3618 copy.dump(fd); 3619 3620#ifdef STATE_QUEUE_DUMP 3621 // Similar for state queue 3622 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3623 observerCopy.dump(fd); 3624 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3625 mutatorCopy.dump(fd); 3626#endif 3627 3628#ifdef TEE_SINK 3629 // Write the tee output to a .wav file 3630 dumpTee(fd, mTeeSource, mId); 3631#endif 3632 3633#ifdef AUDIO_WATCHDOG 3634 if (mAudioWatchdog != 0) { 3635 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3636 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3637 wdCopy.dump(fd); 3638 } 3639#endif 3640} 3641 3642uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3643{ 3644 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3645} 3646 3647uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3648{ 3649 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3650} 3651 3652void AudioFlinger::MixerThread::cacheParameters_l() 3653{ 3654 PlaybackThread::cacheParameters_l(); 3655 3656 // FIXME: Relaxed timing because of a certain device that can't meet latency 3657 // Should be reduced to 2x after the vendor fixes the driver issue 3658 // increase threshold again due to low power audio mode. The way this warning 3659 // threshold is calculated and its usefulness should be reconsidered anyway. 3660 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3661} 3662 3663// ---------------------------------------------------------------------------- 3664 3665AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3666 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3667 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3668 // mLeftVolFloat, mRightVolFloat 3669{ 3670} 3671 3672AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3673 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3674 ThreadBase::type_t type) 3675 : PlaybackThread(audioFlinger, output, id, device, type) 3676 // mLeftVolFloat, mRightVolFloat 3677{ 3678} 3679 3680AudioFlinger::DirectOutputThread::~DirectOutputThread() 3681{ 3682} 3683 3684void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3685{ 3686 audio_track_cblk_t* cblk = track->cblk(); 3687 float left, right; 3688 3689 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3690 left = right = 0; 3691 } else { 3692 float typeVolume = mStreamTypes[track->streamType()].volume; 3693 float v = mMasterVolume * typeVolume; 3694 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3695 uint32_t vlr = proxy->getVolumeLR(); 3696 float v_clamped = v * (vlr & 0xFFFF); 3697 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3698 left = v_clamped/MAX_GAIN; 3699 v_clamped = v * (vlr >> 16); 3700 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3701 right = v_clamped/MAX_GAIN; 3702 } 3703 3704 if (lastTrack) { 3705 if (left != mLeftVolFloat || right != mRightVolFloat) { 3706 mLeftVolFloat = left; 3707 mRightVolFloat = right; 3708 3709 // Convert volumes from float to 8.24 3710 uint32_t vl = (uint32_t)(left * (1 << 24)); 3711 uint32_t vr = (uint32_t)(right * (1 << 24)); 3712 3713 // Delegate volume control to effect in track effect chain if needed 3714 // only one effect chain can be present on DirectOutputThread, so if 3715 // there is one, the track is connected to it 3716 if (!mEffectChains.isEmpty()) { 3717 mEffectChains[0]->setVolume_l(&vl, &vr); 3718 left = (float)vl / (1 << 24); 3719 right = (float)vr / (1 << 24); 3720 } 3721 if (mOutput->stream->set_volume) { 3722 mOutput->stream->set_volume(mOutput->stream, left, right); 3723 } 3724 } 3725 } 3726} 3727 3728 3729AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3730 Vector< sp<Track> > *tracksToRemove 3731) 3732{ 3733 size_t count = mActiveTracks.size(); 3734 mixer_state mixerStatus = MIXER_IDLE; 3735 3736 // find out which tracks need to be processed 3737 for (size_t i = 0; i < count; i++) { 3738 sp<Track> t = mActiveTracks[i].promote(); 3739 // The track died recently 3740 if (t == 0) { 3741 continue; 3742 } 3743 3744 Track* const track = t.get(); 3745 audio_track_cblk_t* cblk = track->cblk(); 3746 // Only consider last track started for volume and mixer state control. 3747 // In theory an older track could underrun and restart after the new one starts 3748 // but as we only care about the transition phase between two tracks on a 3749 // direct output, it is not a problem to ignore the underrun case. 3750 sp<Track> l = mLatestActiveTrack.promote(); 3751 bool last = l.get() == track; 3752 3753 // The first time a track is added we wait 3754 // for all its buffers to be filled before processing it 3755 uint32_t minFrames; 3756 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3757 minFrames = mNormalFrameCount; 3758 } else { 3759 minFrames = 1; 3760 } 3761 3762 if ((track->framesReady() >= minFrames) && track->isReady() && 3763 !track->isPaused() && !track->isTerminated()) 3764 { 3765 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3766 3767 if (track->mFillingUpStatus == Track::FS_FILLED) { 3768 track->mFillingUpStatus = Track::FS_ACTIVE; 3769 // make sure processVolume_l() will apply new volume even if 0 3770 mLeftVolFloat = mRightVolFloat = -1.0; 3771 if (track->mState == TrackBase::RESUMING) { 3772 track->mState = TrackBase::ACTIVE; 3773 } 3774 } 3775 3776 // compute volume for this track 3777 processVolume_l(track, last); 3778 if (last) { 3779 // reset retry count 3780 track->mRetryCount = kMaxTrackRetriesDirect; 3781 mActiveTrack = t; 3782 mixerStatus = MIXER_TRACKS_READY; 3783 } 3784 } else { 3785 // clear effect chain input buffer if the last active track started underruns 3786 // to avoid sending previous audio buffer again to effects 3787 if (!mEffectChains.isEmpty() && last) { 3788 mEffectChains[0]->clearInputBuffer(); 3789 } 3790 3791 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3792 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3793 track->isStopped() || track->isPaused()) { 3794 // We have consumed all the buffers of this track. 3795 // Remove it from the list of active tracks. 3796 // TODO: implement behavior for compressed audio 3797 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3798 size_t framesWritten = mBytesWritten / mFrameSize; 3799 if (mStandby || !last || 3800 track->presentationComplete(framesWritten, audioHALFrames)) { 3801 if (track->isStopped()) { 3802 track->reset(); 3803 } 3804 tracksToRemove->add(track); 3805 } 3806 } else { 3807 // No buffers for this track. Give it a few chances to 3808 // fill a buffer, then remove it from active list. 3809 // Only consider last track started for mixer state control 3810 if (--(track->mRetryCount) <= 0) { 3811 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3812 tracksToRemove->add(track); 3813 // indicate to client process that the track was disabled because of underrun; 3814 // it will then automatically call start() when data is available 3815 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3816 } else if (last) { 3817 mixerStatus = MIXER_TRACKS_ENABLED; 3818 } 3819 } 3820 } 3821 } 3822 3823 // remove all the tracks that need to be... 3824 removeTracks_l(*tracksToRemove); 3825 3826 return mixerStatus; 3827} 3828 3829void AudioFlinger::DirectOutputThread::threadLoop_mix() 3830{ 3831 size_t frameCount = mFrameCount; 3832 int8_t *curBuf = (int8_t *)mSinkBuffer; 3833 // output audio to hardware 3834 while (frameCount) { 3835 AudioBufferProvider::Buffer buffer; 3836 buffer.frameCount = frameCount; 3837 mActiveTrack->getNextBuffer(&buffer); 3838 if (buffer.raw == NULL) { 3839 memset(curBuf, 0, frameCount * mFrameSize); 3840 break; 3841 } 3842 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3843 frameCount -= buffer.frameCount; 3844 curBuf += buffer.frameCount * mFrameSize; 3845 mActiveTrack->releaseBuffer(&buffer); 3846 } 3847 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3848 sleepTime = 0; 3849 standbyTime = systemTime() + standbyDelay; 3850 mActiveTrack.clear(); 3851} 3852 3853void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3854{ 3855 if (sleepTime == 0) { 3856 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3857 sleepTime = activeSleepTime; 3858 } else { 3859 sleepTime = idleSleepTime; 3860 } 3861 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3862 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 3863 sleepTime = 0; 3864 } 3865} 3866 3867// getTrackName_l() must be called with ThreadBase::mLock held 3868int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 3869 int sessionId __unused) 3870{ 3871 return 0; 3872} 3873 3874// deleteTrackName_l() must be called with ThreadBase::mLock held 3875void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 3876{ 3877} 3878 3879// checkForNewParameters_l() must be called with ThreadBase::mLock held 3880bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3881{ 3882 bool reconfig = false; 3883 3884 while (!mNewParameters.isEmpty()) { 3885 status_t status = NO_ERROR; 3886 String8 keyValuePair = mNewParameters[0]; 3887 AudioParameter param = AudioParameter(keyValuePair); 3888 int value; 3889 3890 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3891 // do not accept frame count changes if tracks are open as the track buffer 3892 // size depends on frame count and correct behavior would not be garantied 3893 // if frame count is changed after track creation 3894 if (!mTracks.isEmpty()) { 3895 status = INVALID_OPERATION; 3896 } else { 3897 reconfig = true; 3898 } 3899 } 3900 if (status == NO_ERROR) { 3901 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3902 keyValuePair.string()); 3903 if (!mStandby && status == INVALID_OPERATION) { 3904 mOutput->stream->common.standby(&mOutput->stream->common); 3905 mStandby = true; 3906 mBytesWritten = 0; 3907 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3908 keyValuePair.string()); 3909 } 3910 if (status == NO_ERROR && reconfig) { 3911 readOutputParameters_l(); 3912 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3913 } 3914 } 3915 3916 mNewParameters.removeAt(0); 3917 3918 mParamStatus = status; 3919 mParamCond.signal(); 3920 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3921 // already timed out waiting for the status and will never signal the condition. 3922 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3923 } 3924 return reconfig; 3925} 3926 3927uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3928{ 3929 uint32_t time; 3930 if (audio_is_linear_pcm(mFormat)) { 3931 time = PlaybackThread::activeSleepTimeUs(); 3932 } else { 3933 time = 10000; 3934 } 3935 return time; 3936} 3937 3938uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3939{ 3940 uint32_t time; 3941 if (audio_is_linear_pcm(mFormat)) { 3942 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3943 } else { 3944 time = 10000; 3945 } 3946 return time; 3947} 3948 3949uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3950{ 3951 uint32_t time; 3952 if (audio_is_linear_pcm(mFormat)) { 3953 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3954 } else { 3955 time = 10000; 3956 } 3957 return time; 3958} 3959 3960void AudioFlinger::DirectOutputThread::cacheParameters_l() 3961{ 3962 PlaybackThread::cacheParameters_l(); 3963 3964 // use shorter standby delay as on normal output to release 3965 // hardware resources as soon as possible 3966 if (audio_is_linear_pcm(mFormat)) { 3967 standbyDelay = microseconds(activeSleepTime*2); 3968 } else { 3969 standbyDelay = kOffloadStandbyDelayNs; 3970 } 3971} 3972 3973// ---------------------------------------------------------------------------- 3974 3975AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 3976 const wp<AudioFlinger::PlaybackThread>& playbackThread) 3977 : Thread(false /*canCallJava*/), 3978 mPlaybackThread(playbackThread), 3979 mWriteAckSequence(0), 3980 mDrainSequence(0) 3981{ 3982} 3983 3984AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 3985{ 3986} 3987 3988void AudioFlinger::AsyncCallbackThread::onFirstRef() 3989{ 3990 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 3991} 3992 3993bool AudioFlinger::AsyncCallbackThread::threadLoop() 3994{ 3995 while (!exitPending()) { 3996 uint32_t writeAckSequence; 3997 uint32_t drainSequence; 3998 3999 { 4000 Mutex::Autolock _l(mLock); 4001 while (!((mWriteAckSequence & 1) || 4002 (mDrainSequence & 1) || 4003 exitPending())) { 4004 mWaitWorkCV.wait(mLock); 4005 } 4006 4007 if (exitPending()) { 4008 break; 4009 } 4010 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4011 mWriteAckSequence, mDrainSequence); 4012 writeAckSequence = mWriteAckSequence; 4013 mWriteAckSequence &= ~1; 4014 drainSequence = mDrainSequence; 4015 mDrainSequence &= ~1; 4016 } 4017 { 4018 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4019 if (playbackThread != 0) { 4020 if (writeAckSequence & 1) { 4021 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4022 } 4023 if (drainSequence & 1) { 4024 playbackThread->resetDraining(drainSequence >> 1); 4025 } 4026 } 4027 } 4028 } 4029 return false; 4030} 4031 4032void AudioFlinger::AsyncCallbackThread::exit() 4033{ 4034 ALOGV("AsyncCallbackThread::exit"); 4035 Mutex::Autolock _l(mLock); 4036 requestExit(); 4037 mWaitWorkCV.broadcast(); 4038} 4039 4040void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4041{ 4042 Mutex::Autolock _l(mLock); 4043 // bit 0 is cleared 4044 mWriteAckSequence = sequence << 1; 4045} 4046 4047void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4048{ 4049 Mutex::Autolock _l(mLock); 4050 // ignore unexpected callbacks 4051 if (mWriteAckSequence & 2) { 4052 mWriteAckSequence |= 1; 4053 mWaitWorkCV.signal(); 4054 } 4055} 4056 4057void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4058{ 4059 Mutex::Autolock _l(mLock); 4060 // bit 0 is cleared 4061 mDrainSequence = sequence << 1; 4062} 4063 4064void AudioFlinger::AsyncCallbackThread::resetDraining() 4065{ 4066 Mutex::Autolock _l(mLock); 4067 // ignore unexpected callbacks 4068 if (mDrainSequence & 2) { 4069 mDrainSequence |= 1; 4070 mWaitWorkCV.signal(); 4071 } 4072} 4073 4074 4075// ---------------------------------------------------------------------------- 4076AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4077 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4078 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4079 mHwPaused(false), 4080 mFlushPending(false), 4081 mPausedBytesRemaining(0) 4082{ 4083 //FIXME: mStandby should be set to true by ThreadBase constructor 4084 mStandby = true; 4085} 4086 4087void AudioFlinger::OffloadThread::threadLoop_exit() 4088{ 4089 if (mFlushPending || mHwPaused) { 4090 // If a flush is pending or track was paused, just discard buffered data 4091 flushHw_l(); 4092 } else { 4093 mMixerStatus = MIXER_DRAIN_ALL; 4094 threadLoop_drain(); 4095 } 4096 mCallbackThread->exit(); 4097 PlaybackThread::threadLoop_exit(); 4098} 4099 4100AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4101 Vector< sp<Track> > *tracksToRemove 4102) 4103{ 4104 size_t count = mActiveTracks.size(); 4105 4106 mixer_state mixerStatus = MIXER_IDLE; 4107 bool doHwPause = false; 4108 bool doHwResume = false; 4109 4110 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4111 4112 // find out which tracks need to be processed 4113 for (size_t i = 0; i < count; i++) { 4114 sp<Track> t = mActiveTracks[i].promote(); 4115 // The track died recently 4116 if (t == 0) { 4117 continue; 4118 } 4119 Track* const track = t.get(); 4120 audio_track_cblk_t* cblk = track->cblk(); 4121 // Only consider last track started for volume and mixer state control. 4122 // In theory an older track could underrun and restart after the new one starts 4123 // but as we only care about the transition phase between two tracks on a 4124 // direct output, it is not a problem to ignore the underrun case. 4125 sp<Track> l = mLatestActiveTrack.promote(); 4126 bool last = l.get() == track; 4127 4128 if (track->isInvalid()) { 4129 ALOGW("An invalidated track shouldn't be in active list"); 4130 tracksToRemove->add(track); 4131 continue; 4132 } 4133 4134 if (track->mState == TrackBase::IDLE) { 4135 ALOGW("An idle track shouldn't be in active list"); 4136 continue; 4137 } 4138 4139 if (track->isPausing()) { 4140 track->setPaused(); 4141 if (last) { 4142 if (!mHwPaused) { 4143 doHwPause = true; 4144 mHwPaused = true; 4145 } 4146 // If we were part way through writing the mixbuffer to 4147 // the HAL we must save this until we resume 4148 // BUG - this will be wrong if a different track is made active, 4149 // in that case we want to discard the pending data in the 4150 // mixbuffer and tell the client to present it again when the 4151 // track is resumed 4152 mPausedWriteLength = mCurrentWriteLength; 4153 mPausedBytesRemaining = mBytesRemaining; 4154 mBytesRemaining = 0; // stop writing 4155 } 4156 tracksToRemove->add(track); 4157 } else if (track->isFlushPending()) { 4158 track->flushAck(); 4159 if (last) { 4160 mFlushPending = true; 4161 } 4162 } else if (track->framesReady() && track->isReady() && 4163 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4164 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4165 if (track->mFillingUpStatus == Track::FS_FILLED) { 4166 track->mFillingUpStatus = Track::FS_ACTIVE; 4167 // make sure processVolume_l() will apply new volume even if 0 4168 mLeftVolFloat = mRightVolFloat = -1.0; 4169 if (track->mState == TrackBase::RESUMING) { 4170 track->mState = TrackBase::ACTIVE; 4171 if (last) { 4172 if (mPausedBytesRemaining) { 4173 // Need to continue write that was interrupted 4174 mCurrentWriteLength = mPausedWriteLength; 4175 mBytesRemaining = mPausedBytesRemaining; 4176 mPausedBytesRemaining = 0; 4177 } 4178 if (mHwPaused) { 4179 doHwResume = true; 4180 mHwPaused = false; 4181 // threadLoop_mix() will handle the case that we need to 4182 // resume an interrupted write 4183 } 4184 // enable write to audio HAL 4185 sleepTime = 0; 4186 } 4187 } 4188 } 4189 4190 if (last) { 4191 sp<Track> previousTrack = mPreviousTrack.promote(); 4192 if (previousTrack != 0) { 4193 if (track != previousTrack.get()) { 4194 // Flush any data still being written from last track 4195 mBytesRemaining = 0; 4196 if (mPausedBytesRemaining) { 4197 // Last track was paused so we also need to flush saved 4198 // mixbuffer state and invalidate track so that it will 4199 // re-submit that unwritten data when it is next resumed 4200 mPausedBytesRemaining = 0; 4201 // Invalidate is a bit drastic - would be more efficient 4202 // to have a flag to tell client that some of the 4203 // previously written data was lost 4204 previousTrack->invalidate(); 4205 } 4206 // flush data already sent to the DSP if changing audio session as audio 4207 // comes from a different source. Also invalidate previous track to force a 4208 // seek when resuming. 4209 if (previousTrack->sessionId() != track->sessionId()) { 4210 previousTrack->invalidate(); 4211 } 4212 } 4213 } 4214 mPreviousTrack = track; 4215 // reset retry count 4216 track->mRetryCount = kMaxTrackRetriesOffload; 4217 mActiveTrack = t; 4218 mixerStatus = MIXER_TRACKS_READY; 4219 } 4220 } else { 4221 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4222 if (track->isStopping_1()) { 4223 // Hardware buffer can hold a large amount of audio so we must 4224 // wait for all current track's data to drain before we say 4225 // that the track is stopped. 4226 if (mBytesRemaining == 0) { 4227 // Only start draining when all data in mixbuffer 4228 // has been written 4229 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4230 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4231 // do not drain if no data was ever sent to HAL (mStandby == true) 4232 if (last && !mStandby) { 4233 // do not modify drain sequence if we are already draining. This happens 4234 // when resuming from pause after drain. 4235 if ((mDrainSequence & 1) == 0) { 4236 sleepTime = 0; 4237 standbyTime = systemTime() + standbyDelay; 4238 mixerStatus = MIXER_DRAIN_TRACK; 4239 mDrainSequence += 2; 4240 } 4241 if (mHwPaused) { 4242 // It is possible to move from PAUSED to STOPPING_1 without 4243 // a resume so we must ensure hardware is running 4244 doHwResume = true; 4245 mHwPaused = false; 4246 } 4247 } 4248 } 4249 } else if (track->isStopping_2()) { 4250 // Drain has completed or we are in standby, signal presentation complete 4251 if (!(mDrainSequence & 1) || !last || mStandby) { 4252 track->mState = TrackBase::STOPPED; 4253 size_t audioHALFrames = 4254 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4255 size_t framesWritten = 4256 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4257 track->presentationComplete(framesWritten, audioHALFrames); 4258 track->reset(); 4259 tracksToRemove->add(track); 4260 } 4261 } else { 4262 // No buffers for this track. Give it a few chances to 4263 // fill a buffer, then remove it from active list. 4264 if (--(track->mRetryCount) <= 0) { 4265 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4266 track->name()); 4267 tracksToRemove->add(track); 4268 // indicate to client process that the track was disabled because of underrun; 4269 // it will then automatically call start() when data is available 4270 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4271 } else if (last){ 4272 mixerStatus = MIXER_TRACKS_ENABLED; 4273 } 4274 } 4275 } 4276 // compute volume for this track 4277 processVolume_l(track, last); 4278 } 4279 4280 // make sure the pause/flush/resume sequence is executed in the right order. 4281 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4282 // before flush and then resume HW. This can happen in case of pause/flush/resume 4283 // if resume is received before pause is executed. 4284 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4285 mOutput->stream->pause(mOutput->stream); 4286 } 4287 if (mFlushPending) { 4288 flushHw_l(); 4289 mFlushPending = false; 4290 } 4291 if (!mStandby && doHwResume) { 4292 mOutput->stream->resume(mOutput->stream); 4293 } 4294 4295 // remove all the tracks that need to be... 4296 removeTracks_l(*tracksToRemove); 4297 4298 return mixerStatus; 4299} 4300 4301// must be called with thread mutex locked 4302bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4303{ 4304 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4305 mWriteAckSequence, mDrainSequence); 4306 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4307 return true; 4308 } 4309 return false; 4310} 4311 4312// must be called with thread mutex locked 4313bool AudioFlinger::OffloadThread::shouldStandby_l() 4314{ 4315 bool trackPaused = false; 4316 4317 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4318 // after a timeout and we will enter standby then. 4319 if (mTracks.size() > 0) { 4320 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4321 } 4322 4323 return !mStandby && !trackPaused; 4324} 4325 4326 4327bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4328{ 4329 Mutex::Autolock _l(mLock); 4330 return waitingAsyncCallback_l(); 4331} 4332 4333void AudioFlinger::OffloadThread::flushHw_l() 4334{ 4335 mOutput->stream->flush(mOutput->stream); 4336 // Flush anything still waiting in the mixbuffer 4337 mCurrentWriteLength = 0; 4338 mBytesRemaining = 0; 4339 mPausedWriteLength = 0; 4340 mPausedBytesRemaining = 0; 4341 mHwPaused = false; 4342 4343 if (mUseAsyncWrite) { 4344 // discard any pending drain or write ack by incrementing sequence 4345 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4346 mDrainSequence = (mDrainSequence + 2) & ~1; 4347 ALOG_ASSERT(mCallbackThread != 0); 4348 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4349 mCallbackThread->setDraining(mDrainSequence); 4350 } 4351} 4352 4353void AudioFlinger::OffloadThread::onAddNewTrack_l() 4354{ 4355 sp<Track> previousTrack = mPreviousTrack.promote(); 4356 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4357 4358 if (previousTrack != 0 && latestTrack != 0 && 4359 (previousTrack->sessionId() != latestTrack->sessionId())) { 4360 mFlushPending = true; 4361 } 4362 PlaybackThread::onAddNewTrack_l(); 4363} 4364 4365// ---------------------------------------------------------------------------- 4366 4367AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4368 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4369 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4370 DUPLICATING), 4371 mWaitTimeMs(UINT_MAX) 4372{ 4373 addOutputTrack(mainThread); 4374} 4375 4376AudioFlinger::DuplicatingThread::~DuplicatingThread() 4377{ 4378 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4379 mOutputTracks[i]->destroy(); 4380 } 4381} 4382 4383void AudioFlinger::DuplicatingThread::threadLoop_mix() 4384{ 4385 // mix buffers... 4386 if (outputsReady(outputTracks)) { 4387 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4388 } else { 4389 memset(mSinkBuffer, 0, mSinkBufferSize); 4390 } 4391 sleepTime = 0; 4392 writeFrames = mNormalFrameCount; 4393 mCurrentWriteLength = mSinkBufferSize; 4394 standbyTime = systemTime() + standbyDelay; 4395} 4396 4397void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4398{ 4399 if (sleepTime == 0) { 4400 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4401 sleepTime = activeSleepTime; 4402 } else { 4403 sleepTime = idleSleepTime; 4404 } 4405 } else if (mBytesWritten != 0) { 4406 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4407 writeFrames = mNormalFrameCount; 4408 memset(mSinkBuffer, 0, mSinkBufferSize); 4409 } else { 4410 // flush remaining overflow buffers in output tracks 4411 writeFrames = 0; 4412 } 4413 sleepTime = 0; 4414 } 4415} 4416 4417ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4418{ 4419 for (size_t i = 0; i < outputTracks.size(); i++) { 4420 outputTracks[i]->write(mSinkBuffer, writeFrames); 4421 } 4422 mStandby = false; 4423 return (ssize_t)mSinkBufferSize; 4424} 4425 4426void AudioFlinger::DuplicatingThread::threadLoop_standby() 4427{ 4428 // DuplicatingThread implements standby by stopping all tracks 4429 for (size_t i = 0; i < outputTracks.size(); i++) { 4430 outputTracks[i]->stop(); 4431 } 4432} 4433 4434void AudioFlinger::DuplicatingThread::saveOutputTracks() 4435{ 4436 outputTracks = mOutputTracks; 4437} 4438 4439void AudioFlinger::DuplicatingThread::clearOutputTracks() 4440{ 4441 outputTracks.clear(); 4442} 4443 4444void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4445{ 4446 Mutex::Autolock _l(mLock); 4447 // FIXME explain this formula 4448 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4449 OutputTrack *outputTrack = new OutputTrack(thread, 4450 this, 4451 mSampleRate, 4452 mFormat, 4453 mChannelMask, 4454 frameCount, 4455 IPCThreadState::self()->getCallingUid()); 4456 if (outputTrack->cblk() != NULL) { 4457 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4458 mOutputTracks.add(outputTrack); 4459 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4460 updateWaitTime_l(); 4461 } 4462} 4463 4464void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4465{ 4466 Mutex::Autolock _l(mLock); 4467 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4468 if (mOutputTracks[i]->thread() == thread) { 4469 mOutputTracks[i]->destroy(); 4470 mOutputTracks.removeAt(i); 4471 updateWaitTime_l(); 4472 return; 4473 } 4474 } 4475 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4476} 4477 4478// caller must hold mLock 4479void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4480{ 4481 mWaitTimeMs = UINT_MAX; 4482 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4483 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4484 if (strong != 0) { 4485 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4486 if (waitTimeMs < mWaitTimeMs) { 4487 mWaitTimeMs = waitTimeMs; 4488 } 4489 } 4490 } 4491} 4492 4493 4494bool AudioFlinger::DuplicatingThread::outputsReady( 4495 const SortedVector< sp<OutputTrack> > &outputTracks) 4496{ 4497 for (size_t i = 0; i < outputTracks.size(); i++) { 4498 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4499 if (thread == 0) { 4500 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4501 outputTracks[i].get()); 4502 return false; 4503 } 4504 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4505 // see note at standby() declaration 4506 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4507 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4508 thread.get()); 4509 return false; 4510 } 4511 } 4512 return true; 4513} 4514 4515uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4516{ 4517 return (mWaitTimeMs * 1000) / 2; 4518} 4519 4520void AudioFlinger::DuplicatingThread::cacheParameters_l() 4521{ 4522 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4523 updateWaitTime_l(); 4524 4525 MixerThread::cacheParameters_l(); 4526} 4527 4528// ---------------------------------------------------------------------------- 4529// Record 4530// ---------------------------------------------------------------------------- 4531 4532AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4533 AudioStreamIn *input, 4534 audio_io_handle_t id, 4535 audio_devices_t outDevice, 4536 audio_devices_t inDevice 4537#ifdef TEE_SINK 4538 , const sp<NBAIO_Sink>& teeSink 4539#endif 4540 ) : 4541 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4542 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4543 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4544 mRsmpInRear(0) 4545#ifdef TEE_SINK 4546 , mTeeSink(teeSink) 4547#endif 4548{ 4549 snprintf(mName, kNameLength, "AudioIn_%X", id); 4550 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4551 4552 readInputParameters_l(); 4553} 4554 4555 4556AudioFlinger::RecordThread::~RecordThread() 4557{ 4558 mAudioFlinger->unregisterWriter(mNBLogWriter); 4559 delete[] mRsmpInBuffer; 4560} 4561 4562void AudioFlinger::RecordThread::onFirstRef() 4563{ 4564 run(mName, PRIORITY_URGENT_AUDIO); 4565} 4566 4567bool AudioFlinger::RecordThread::threadLoop() 4568{ 4569 nsecs_t lastWarning = 0; 4570 4571 inputStandBy(); 4572 4573reacquire_wakelock: 4574 sp<RecordTrack> activeTrack; 4575 int activeTracksGen; 4576 { 4577 Mutex::Autolock _l(mLock); 4578 size_t size = mActiveTracks.size(); 4579 activeTracksGen = mActiveTracksGen; 4580 if (size > 0) { 4581 // FIXME an arbitrary choice 4582 activeTrack = mActiveTracks[0]; 4583 acquireWakeLock_l(activeTrack->uid()); 4584 if (size > 1) { 4585 SortedVector<int> tmp; 4586 for (size_t i = 0; i < size; i++) { 4587 tmp.add(mActiveTracks[i]->uid()); 4588 } 4589 updateWakeLockUids_l(tmp); 4590 } 4591 } else { 4592 acquireWakeLock_l(-1); 4593 } 4594 } 4595 4596 // used to request a deferred sleep, to be executed later while mutex is unlocked 4597 uint32_t sleepUs = 0; 4598 4599 // loop while there is work to do 4600 for (;;) { 4601 Vector< sp<EffectChain> > effectChains; 4602 4603 // sleep with mutex unlocked 4604 if (sleepUs > 0) { 4605 usleep(sleepUs); 4606 sleepUs = 0; 4607 } 4608 4609 // activeTracks accumulates a copy of a subset of mActiveTracks 4610 Vector< sp<RecordTrack> > activeTracks; 4611 4612 { // scope for mLock 4613 Mutex::Autolock _l(mLock); 4614 4615 processConfigEvents_l(); 4616 // return value 'reconfig' is currently unused 4617 bool reconfig = checkForNewParameters_l(); 4618 4619 // check exitPending here because checkForNewParameters_l() and 4620 // checkForNewParameters_l() can temporarily release mLock 4621 if (exitPending()) { 4622 break; 4623 } 4624 4625 // if no active track(s), then standby and release wakelock 4626 size_t size = mActiveTracks.size(); 4627 if (size == 0) { 4628 standbyIfNotAlreadyInStandby(); 4629 // exitPending() can't become true here 4630 releaseWakeLock_l(); 4631 ALOGV("RecordThread: loop stopping"); 4632 // go to sleep 4633 mWaitWorkCV.wait(mLock); 4634 ALOGV("RecordThread: loop starting"); 4635 goto reacquire_wakelock; 4636 } 4637 4638 if (mActiveTracksGen != activeTracksGen) { 4639 activeTracksGen = mActiveTracksGen; 4640 SortedVector<int> tmp; 4641 for (size_t i = 0; i < size; i++) { 4642 tmp.add(mActiveTracks[i]->uid()); 4643 } 4644 updateWakeLockUids_l(tmp); 4645 } 4646 4647 bool doBroadcast = false; 4648 for (size_t i = 0; i < size; ) { 4649 4650 activeTrack = mActiveTracks[i]; 4651 if (activeTrack->isTerminated()) { 4652 removeTrack_l(activeTrack); 4653 mActiveTracks.remove(activeTrack); 4654 mActiveTracksGen++; 4655 size--; 4656 continue; 4657 } 4658 4659 TrackBase::track_state activeTrackState = activeTrack->mState; 4660 switch (activeTrackState) { 4661 4662 case TrackBase::PAUSING: 4663 mActiveTracks.remove(activeTrack); 4664 mActiveTracksGen++; 4665 doBroadcast = true; 4666 size--; 4667 continue; 4668 4669 case TrackBase::STARTING_1: 4670 sleepUs = 10000; 4671 i++; 4672 continue; 4673 4674 case TrackBase::STARTING_2: 4675 doBroadcast = true; 4676 mStandby = false; 4677 activeTrack->mState = TrackBase::ACTIVE; 4678 break; 4679 4680 case TrackBase::ACTIVE: 4681 break; 4682 4683 case TrackBase::IDLE: 4684 i++; 4685 continue; 4686 4687 default: 4688 LOG_FATAL("Unexpected activeTrackState %d", activeTrackState); 4689 } 4690 4691 activeTracks.add(activeTrack); 4692 i++; 4693 4694 } 4695 if (doBroadcast) { 4696 mStartStopCond.broadcast(); 4697 } 4698 4699 // sleep if there are no active tracks to process 4700 if (activeTracks.size() == 0) { 4701 if (sleepUs == 0) { 4702 sleepUs = kRecordThreadSleepUs; 4703 } 4704 continue; 4705 } 4706 sleepUs = 0; 4707 4708 lockEffectChains_l(effectChains); 4709 } 4710 4711 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4712 4713 size_t size = effectChains.size(); 4714 for (size_t i = 0; i < size; i++) { 4715 // thread mutex is not locked, but effect chain is locked 4716 effectChains[i]->process_l(); 4717 } 4718 4719 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4720 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4721 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4722 // If destination is non-contiguous, first read past the nominal end of buffer, then 4723 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4724 4725 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4726 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4727 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4728 if (bytesRead <= 0) { 4729 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4730 // Force input into standby so that it tries to recover at next read attempt 4731 inputStandBy(); 4732 sleepUs = kRecordThreadSleepUs; 4733 continue; 4734 } 4735 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4736 size_t framesRead = bytesRead / mFrameSize; 4737 ALOG_ASSERT(framesRead > 0); 4738 if (mTeeSink != 0) { 4739 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4740 } 4741 // If destination is non-contiguous, we now correct for reading past end of buffer. 4742 size_t part1 = mRsmpInFramesP2 - rear; 4743 if (framesRead > part1) { 4744 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4745 (framesRead - part1) * mFrameSize); 4746 } 4747 rear = mRsmpInRear += framesRead; 4748 4749 size = activeTracks.size(); 4750 // loop over each active track 4751 for (size_t i = 0; i < size; i++) { 4752 activeTrack = activeTracks[i]; 4753 4754 enum { 4755 OVERRUN_UNKNOWN, 4756 OVERRUN_TRUE, 4757 OVERRUN_FALSE 4758 } overrun = OVERRUN_UNKNOWN; 4759 4760 // loop over getNextBuffer to handle circular sink 4761 for (;;) { 4762 4763 activeTrack->mSink.frameCount = ~0; 4764 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4765 size_t framesOut = activeTrack->mSink.frameCount; 4766 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4767 4768 int32_t front = activeTrack->mRsmpInFront; 4769 ssize_t filled = rear - front; 4770 size_t framesIn; 4771 4772 if (filled < 0) { 4773 // should not happen, but treat like a massive overrun and re-sync 4774 framesIn = 0; 4775 activeTrack->mRsmpInFront = rear; 4776 overrun = OVERRUN_TRUE; 4777 } else if ((size_t) filled <= mRsmpInFrames) { 4778 framesIn = (size_t) filled; 4779 } else { 4780 // client is not keeping up with server, but give it latest data 4781 framesIn = mRsmpInFrames; 4782 activeTrack->mRsmpInFront = front = rear - framesIn; 4783 overrun = OVERRUN_TRUE; 4784 } 4785 4786 if (framesOut == 0 || framesIn == 0) { 4787 break; 4788 } 4789 4790 if (activeTrack->mResampler == NULL) { 4791 // no resampling 4792 if (framesIn > framesOut) { 4793 framesIn = framesOut; 4794 } else { 4795 framesOut = framesIn; 4796 } 4797 int8_t *dst = activeTrack->mSink.i8; 4798 while (framesIn > 0) { 4799 front &= mRsmpInFramesP2 - 1; 4800 size_t part1 = mRsmpInFramesP2 - front; 4801 if (part1 > framesIn) { 4802 part1 = framesIn; 4803 } 4804 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4805 if (mChannelCount == activeTrack->mChannelCount) { 4806 memcpy(dst, src, part1 * mFrameSize); 4807 } else if (mChannelCount == 1) { 4808 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4809 part1); 4810 } else { 4811 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4812 part1); 4813 } 4814 dst += part1 * activeTrack->mFrameSize; 4815 front += part1; 4816 framesIn -= part1; 4817 } 4818 activeTrack->mRsmpInFront += framesOut; 4819 4820 } else { 4821 // resampling 4822 // FIXME framesInNeeded should really be part of resampler API, and should 4823 // depend on the SRC ratio 4824 // to keep mRsmpInBuffer full so resampler always has sufficient input 4825 size_t framesInNeeded; 4826 // FIXME only re-calculate when it changes, and optimize for common ratios 4827 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4828 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4829 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4830 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4831 framesInNeeded, framesOut, inOverOut); 4832 // Although we theoretically have framesIn in circular buffer, some of those are 4833 // unreleased frames, and thus must be discounted for purpose of budgeting. 4834 size_t unreleased = activeTrack->mRsmpInUnrel; 4835 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 4836 if (framesIn < framesInNeeded) { 4837 ALOGV("not enough to resample: have %u frames in but need %u in to " 4838 "produce %u out given in/out ratio of %.4g", 4839 framesIn, framesInNeeded, framesOut, inOverOut); 4840 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 4841 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 4842 if (newFramesOut == 0) { 4843 break; 4844 } 4845 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 4846 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 4847 framesInNeeded, newFramesOut, outOverIn); 4848 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 4849 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 4850 "given in/out ratio of %.4g", 4851 framesIn, framesInNeeded, newFramesOut, inOverOut); 4852 framesOut = newFramesOut; 4853 } else { 4854 ALOGV("success 1: have %u in and need %u in to produce %u out " 4855 "given in/out ratio of %.4g", 4856 framesIn, framesInNeeded, framesOut, inOverOut); 4857 } 4858 4859 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 4860 if (activeTrack->mRsmpOutFrameCount < framesOut) { 4861 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 4862 delete[] activeTrack->mRsmpOutBuffer; 4863 // resampler always outputs stereo 4864 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 4865 activeTrack->mRsmpOutFrameCount = framesOut; 4866 } 4867 4868 // resampler accumulates, but we only have one source track 4869 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 4870 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 4871 // FIXME how about having activeTrack implement this interface itself? 4872 activeTrack->mResamplerBufferProvider 4873 /*this*/ /* AudioBufferProvider* */); 4874 // ditherAndClamp() works as long as all buffers returned by 4875 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 4876 if (activeTrack->mChannelCount == 1) { 4877 // temporarily type pun mRsmpOutBuffer from Q19.12 to int16_t 4878 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 4879 framesOut); 4880 // the resampler always outputs stereo samples: 4881 // do post stereo to mono conversion 4882 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 4883 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 4884 } else { 4885 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 4886 activeTrack->mRsmpOutBuffer, framesOut); 4887 } 4888 // now done with mRsmpOutBuffer 4889 4890 } 4891 4892 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 4893 overrun = OVERRUN_FALSE; 4894 } 4895 4896 if (activeTrack->mFramesToDrop == 0) { 4897 if (framesOut > 0) { 4898 activeTrack->mSink.frameCount = framesOut; 4899 activeTrack->releaseBuffer(&activeTrack->mSink); 4900 } 4901 } else { 4902 // FIXME could do a partial drop of framesOut 4903 if (activeTrack->mFramesToDrop > 0) { 4904 activeTrack->mFramesToDrop -= framesOut; 4905 if (activeTrack->mFramesToDrop <= 0) { 4906 activeTrack->clearSyncStartEvent(); 4907 } 4908 } else { 4909 activeTrack->mFramesToDrop += framesOut; 4910 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 4911 activeTrack->mSyncStartEvent->isCancelled()) { 4912 ALOGW("Synced record %s, session %d, trigger session %d", 4913 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 4914 activeTrack->sessionId(), 4915 (activeTrack->mSyncStartEvent != 0) ? 4916 activeTrack->mSyncStartEvent->triggerSession() : 0); 4917 activeTrack->clearSyncStartEvent(); 4918 } 4919 } 4920 } 4921 4922 if (framesOut == 0) { 4923 break; 4924 } 4925 } 4926 4927 switch (overrun) { 4928 case OVERRUN_TRUE: 4929 // client isn't retrieving buffers fast enough 4930 if (!activeTrack->setOverflow()) { 4931 nsecs_t now = systemTime(); 4932 // FIXME should lastWarning per track? 4933 if ((now - lastWarning) > kWarningThrottleNs) { 4934 ALOGW("RecordThread: buffer overflow"); 4935 lastWarning = now; 4936 } 4937 } 4938 break; 4939 case OVERRUN_FALSE: 4940 activeTrack->clearOverflow(); 4941 break; 4942 case OVERRUN_UNKNOWN: 4943 break; 4944 } 4945 4946 } 4947 4948 // enable changes in effect chain 4949 unlockEffectChains(effectChains); 4950 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 4951 } 4952 4953 standbyIfNotAlreadyInStandby(); 4954 4955 { 4956 Mutex::Autolock _l(mLock); 4957 for (size_t i = 0; i < mTracks.size(); i++) { 4958 sp<RecordTrack> track = mTracks[i]; 4959 track->invalidate(); 4960 } 4961 mActiveTracks.clear(); 4962 mActiveTracksGen++; 4963 mStartStopCond.broadcast(); 4964 } 4965 4966 releaseWakeLock(); 4967 4968 ALOGV("RecordThread %p exiting", this); 4969 return false; 4970} 4971 4972void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 4973{ 4974 if (!mStandby) { 4975 inputStandBy(); 4976 mStandby = true; 4977 } 4978} 4979 4980void AudioFlinger::RecordThread::inputStandBy() 4981{ 4982 mInput->stream->common.standby(&mInput->stream->common); 4983} 4984 4985sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4986 const sp<AudioFlinger::Client>& client, 4987 uint32_t sampleRate, 4988 audio_format_t format, 4989 audio_channel_mask_t channelMask, 4990 size_t *pFrameCount, 4991 int sessionId, 4992 int uid, 4993 IAudioFlinger::track_flags_t *flags, 4994 pid_t tid, 4995 status_t *status) 4996{ 4997 size_t frameCount = *pFrameCount; 4998 sp<RecordTrack> track; 4999 status_t lStatus; 5000 5001 lStatus = initCheck(); 5002 if (lStatus != NO_ERROR) { 5003 ALOGE("createRecordTrack_l() audio driver not initialized"); 5004 goto Exit; 5005 } 5006 5007 // client expresses a preference for FAST, but we get the final say 5008 if (*flags & IAudioFlinger::TRACK_FAST) { 5009 if ( 5010 // use case: callback handler and frame count is default or at least as large as HAL 5011 ( 5012 (tid != -1) && 5013 ((frameCount == 0) || 5014 (frameCount >= mFrameCount)) 5015 ) && 5016 // FIXME when record supports non-PCM data, also check for audio_is_linear_pcm(format) 5017 // mono or stereo 5018 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 5019 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 5020 // hardware sample rate 5021 (sampleRate == mSampleRate) && 5022 // record thread has an associated fast recorder 5023 hasFastRecorder() 5024 // FIXME test that RecordThread for this fast track has a capable output HAL 5025 // FIXME add a permission test also? 5026 ) { 5027 // if frameCount not specified, then it defaults to fast recorder (HAL) frame count 5028 if (frameCount == 0) { 5029 frameCount = mFrameCount * kFastTrackMultiplier; 5030 } 5031 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5032 frameCount, mFrameCount); 5033 } else { 5034 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5035 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5036 "hasFastRecorder=%d tid=%d", 5037 frameCount, mFrameCount, format, 5038 audio_is_linear_pcm(format), 5039 channelMask, sampleRate, mSampleRate, hasFastRecorder(), tid); 5040 *flags &= ~IAudioFlinger::TRACK_FAST; 5041 // For compatibility with AudioRecord calculation, buffer depth is forced 5042 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5043 // This is probably too conservative, but legacy application code may depend on it. 5044 // If you change this calculation, also review the start threshold which is related. 5045 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5046 size_t mNormalFrameCount = 2048; // FIXME 5047 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5048 if (minBufCount < 2) { 5049 minBufCount = 2; 5050 } 5051 size_t minFrameCount = mNormalFrameCount * minBufCount; 5052 if (frameCount < minFrameCount) { 5053 frameCount = minFrameCount; 5054 } 5055 } 5056 } 5057 *pFrameCount = frameCount; 5058 5059 // FIXME use flags and tid similar to createTrack_l() 5060 5061 { // scope for mLock 5062 Mutex::Autolock _l(mLock); 5063 5064 track = new RecordTrack(this, client, sampleRate, 5065 format, channelMask, frameCount, sessionId, uid); 5066 5067 lStatus = track->initCheck(); 5068 if (lStatus != NO_ERROR) { 5069 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5070 // track must be cleared from the caller as the caller has the AF lock 5071 goto Exit; 5072 } 5073 mTracks.add(track); 5074 5075 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5076 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5077 mAudioFlinger->btNrecIsOff(); 5078 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5079 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5080 5081 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5082 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5083 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5084 // so ask activity manager to do this on our behalf 5085 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5086 } 5087 } 5088 lStatus = NO_ERROR; 5089 5090Exit: 5091 *status = lStatus; 5092 return track; 5093} 5094 5095status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5096 AudioSystem::sync_event_t event, 5097 int triggerSession) 5098{ 5099 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5100 sp<ThreadBase> strongMe = this; 5101 status_t status = NO_ERROR; 5102 5103 if (event == AudioSystem::SYNC_EVENT_NONE) { 5104 recordTrack->clearSyncStartEvent(); 5105 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5106 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5107 triggerSession, 5108 recordTrack->sessionId(), 5109 syncStartEventCallback, 5110 recordTrack); 5111 // Sync event can be cancelled by the trigger session if the track is not in a 5112 // compatible state in which case we start record immediately 5113 if (recordTrack->mSyncStartEvent->isCancelled()) { 5114 recordTrack->clearSyncStartEvent(); 5115 } else { 5116 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5117 recordTrack->mFramesToDrop = - 5118 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5119 } 5120 } 5121 5122 { 5123 // This section is a rendezvous between binder thread executing start() and RecordThread 5124 AutoMutex lock(mLock); 5125 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5126 if (recordTrack->mState == TrackBase::PAUSING) { 5127 ALOGV("active record track PAUSING -> ACTIVE"); 5128 recordTrack->mState = TrackBase::ACTIVE; 5129 } else { 5130 ALOGV("active record track state %d", recordTrack->mState); 5131 } 5132 return status; 5133 } 5134 5135 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5136 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5137 // or using a separate command thread 5138 recordTrack->mState = TrackBase::STARTING_1; 5139 mActiveTracks.add(recordTrack); 5140 mActiveTracksGen++; 5141 mLock.unlock(); 5142 status_t status = AudioSystem::startInput(mId); 5143 mLock.lock(); 5144 // FIXME should verify that recordTrack is still in mActiveTracks 5145 if (status != NO_ERROR) { 5146 mActiveTracks.remove(recordTrack); 5147 mActiveTracksGen++; 5148 recordTrack->clearSyncStartEvent(); 5149 return status; 5150 } 5151 // Catch up with current buffer indices if thread is already running. 5152 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5153 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5154 // see previously buffered data before it called start(), but with greater risk of overrun. 5155 5156 recordTrack->mRsmpInFront = mRsmpInRear; 5157 recordTrack->mRsmpInUnrel = 0; 5158 // FIXME why reset? 5159 if (recordTrack->mResampler != NULL) { 5160 recordTrack->mResampler->reset(); 5161 } 5162 recordTrack->mState = TrackBase::STARTING_2; 5163 // signal thread to start 5164 mWaitWorkCV.broadcast(); 5165 if (mActiveTracks.indexOf(recordTrack) < 0) { 5166 ALOGV("Record failed to start"); 5167 status = BAD_VALUE; 5168 goto startError; 5169 } 5170 return status; 5171 } 5172 5173startError: 5174 AudioSystem::stopInput(mId); 5175 recordTrack->clearSyncStartEvent(); 5176 // FIXME I wonder why we do not reset the state here? 5177 return status; 5178} 5179 5180void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5181{ 5182 sp<SyncEvent> strongEvent = event.promote(); 5183 5184 if (strongEvent != 0) { 5185 sp<RefBase> ptr = strongEvent->cookie().promote(); 5186 if (ptr != 0) { 5187 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5188 recordTrack->handleSyncStartEvent(strongEvent); 5189 } 5190 } 5191} 5192 5193bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5194 ALOGV("RecordThread::stop"); 5195 AutoMutex _l(mLock); 5196 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5197 return false; 5198 } 5199 // note that threadLoop may still be processing the track at this point [without lock] 5200 recordTrack->mState = TrackBase::PAUSING; 5201 // do not wait for mStartStopCond if exiting 5202 if (exitPending()) { 5203 return true; 5204 } 5205 // FIXME incorrect usage of wait: no explicit predicate or loop 5206 mStartStopCond.wait(mLock); 5207 // if we have been restarted, recordTrack is in mActiveTracks here 5208 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5209 ALOGV("Record stopped OK"); 5210 return true; 5211 } 5212 return false; 5213} 5214 5215bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5216{ 5217 return false; 5218} 5219 5220status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5221{ 5222#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5223 if (!isValidSyncEvent(event)) { 5224 return BAD_VALUE; 5225 } 5226 5227 int eventSession = event->triggerSession(); 5228 status_t ret = NAME_NOT_FOUND; 5229 5230 Mutex::Autolock _l(mLock); 5231 5232 for (size_t i = 0; i < mTracks.size(); i++) { 5233 sp<RecordTrack> track = mTracks[i]; 5234 if (eventSession == track->sessionId()) { 5235 (void) track->setSyncEvent(event); 5236 ret = NO_ERROR; 5237 } 5238 } 5239 return ret; 5240#else 5241 return BAD_VALUE; 5242#endif 5243} 5244 5245// destroyTrack_l() must be called with ThreadBase::mLock held 5246void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5247{ 5248 track->terminate(); 5249 track->mState = TrackBase::STOPPED; 5250 // active tracks are removed by threadLoop() 5251 if (mActiveTracks.indexOf(track) < 0) { 5252 removeTrack_l(track); 5253 } 5254} 5255 5256void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5257{ 5258 mTracks.remove(track); 5259 // need anything related to effects here? 5260} 5261 5262void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5263{ 5264 dumpInternals(fd, args); 5265 dumpTracks(fd, args); 5266 dumpEffectChains(fd, args); 5267} 5268 5269void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5270{ 5271 fdprintf(fd, "\nInput thread %p:\n", this); 5272 5273 if (mActiveTracks.size() > 0) { 5274 fdprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5275 } else { 5276 fdprintf(fd, " No active record clients\n"); 5277 } 5278 5279 dumpBase(fd, args); 5280} 5281 5282void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5283{ 5284 const size_t SIZE = 256; 5285 char buffer[SIZE]; 5286 String8 result; 5287 5288 size_t numtracks = mTracks.size(); 5289 size_t numactive = mActiveTracks.size(); 5290 size_t numactiveseen = 0; 5291 fdprintf(fd, " %d Tracks", numtracks); 5292 if (numtracks) { 5293 fdprintf(fd, " of which %d are active\n", numactive); 5294 RecordTrack::appendDumpHeader(result); 5295 for (size_t i = 0; i < numtracks ; ++i) { 5296 sp<RecordTrack> track = mTracks[i]; 5297 if (track != 0) { 5298 bool active = mActiveTracks.indexOf(track) >= 0; 5299 if (active) { 5300 numactiveseen++; 5301 } 5302 track->dump(buffer, SIZE, active); 5303 result.append(buffer); 5304 } 5305 } 5306 } else { 5307 fdprintf(fd, "\n"); 5308 } 5309 5310 if (numactiveseen != numactive) { 5311 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5312 " not in the track list\n"); 5313 result.append(buffer); 5314 RecordTrack::appendDumpHeader(result); 5315 for (size_t i = 0; i < numactive; ++i) { 5316 sp<RecordTrack> track = mActiveTracks[i]; 5317 if (mTracks.indexOf(track) < 0) { 5318 track->dump(buffer, SIZE, true); 5319 result.append(buffer); 5320 } 5321 } 5322 5323 } 5324 write(fd, result.string(), result.size()); 5325} 5326 5327// AudioBufferProvider interface 5328status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5329 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5330{ 5331 RecordTrack *activeTrack = mRecordTrack; 5332 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5333 if (threadBase == 0) { 5334 buffer->frameCount = 0; 5335 buffer->raw = NULL; 5336 return NOT_ENOUGH_DATA; 5337 } 5338 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5339 int32_t rear = recordThread->mRsmpInRear; 5340 int32_t front = activeTrack->mRsmpInFront; 5341 ssize_t filled = rear - front; 5342 // FIXME should not be P2 (don't want to increase latency) 5343 // FIXME if client not keeping up, discard 5344 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5345 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5346 front &= recordThread->mRsmpInFramesP2 - 1; 5347 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5348 if (part1 > (size_t) filled) { 5349 part1 = filled; 5350 } 5351 size_t ask = buffer->frameCount; 5352 ALOG_ASSERT(ask > 0); 5353 if (part1 > ask) { 5354 part1 = ask; 5355 } 5356 if (part1 == 0) { 5357 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5358 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5359 buffer->raw = NULL; 5360 buffer->frameCount = 0; 5361 activeTrack->mRsmpInUnrel = 0; 5362 return NOT_ENOUGH_DATA; 5363 } 5364 5365 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5366 buffer->frameCount = part1; 5367 activeTrack->mRsmpInUnrel = part1; 5368 return NO_ERROR; 5369} 5370 5371// AudioBufferProvider interface 5372void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5373 AudioBufferProvider::Buffer* buffer) 5374{ 5375 RecordTrack *activeTrack = mRecordTrack; 5376 size_t stepCount = buffer->frameCount; 5377 if (stepCount == 0) { 5378 return; 5379 } 5380 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5381 activeTrack->mRsmpInUnrel -= stepCount; 5382 activeTrack->mRsmpInFront += stepCount; 5383 buffer->raw = NULL; 5384 buffer->frameCount = 0; 5385} 5386 5387bool AudioFlinger::RecordThread::checkForNewParameters_l() 5388{ 5389 bool reconfig = false; 5390 5391 while (!mNewParameters.isEmpty()) { 5392 status_t status = NO_ERROR; 5393 String8 keyValuePair = mNewParameters[0]; 5394 AudioParameter param = AudioParameter(keyValuePair); 5395 int value; 5396 audio_format_t reqFormat = mFormat; 5397 uint32_t samplingRate = mSampleRate; 5398 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5399 5400 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5401 // channel count change can be requested. Do we mandate the first client defines the 5402 // HAL sampling rate and channel count or do we allow changes on the fly? 5403 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5404 samplingRate = value; 5405 reconfig = true; 5406 } 5407 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5408 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5409 status = BAD_VALUE; 5410 } else { 5411 reqFormat = (audio_format_t) value; 5412 reconfig = true; 5413 } 5414 } 5415 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5416 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5417 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5418 status = BAD_VALUE; 5419 } else { 5420 channelMask = mask; 5421 reconfig = true; 5422 } 5423 } 5424 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5425 // do not accept frame count changes if tracks are open as the track buffer 5426 // size depends on frame count and correct behavior would not be guaranteed 5427 // if frame count is changed after track creation 5428 if (mActiveTracks.size() > 0) { 5429 status = INVALID_OPERATION; 5430 } else { 5431 reconfig = true; 5432 } 5433 } 5434 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5435 // forward device change to effects that have requested to be 5436 // aware of attached audio device. 5437 for (size_t i = 0; i < mEffectChains.size(); i++) { 5438 mEffectChains[i]->setDevice_l(value); 5439 } 5440 5441 // store input device and output device but do not forward output device to audio HAL. 5442 // Note that status is ignored by the caller for output device 5443 // (see AudioFlinger::setParameters() 5444 if (audio_is_output_devices(value)) { 5445 mOutDevice = value; 5446 status = BAD_VALUE; 5447 } else { 5448 mInDevice = value; 5449 // disable AEC and NS if the device is a BT SCO headset supporting those 5450 // pre processings 5451 if (mTracks.size() > 0) { 5452 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5453 mAudioFlinger->btNrecIsOff(); 5454 for (size_t i = 0; i < mTracks.size(); i++) { 5455 sp<RecordTrack> track = mTracks[i]; 5456 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5457 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5458 } 5459 } 5460 } 5461 } 5462 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5463 mAudioSource != (audio_source_t)value) { 5464 // forward device change to effects that have requested to be 5465 // aware of attached audio device. 5466 for (size_t i = 0; i < mEffectChains.size(); i++) { 5467 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5468 } 5469 mAudioSource = (audio_source_t)value; 5470 } 5471 5472 if (status == NO_ERROR) { 5473 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5474 keyValuePair.string()); 5475 if (status == INVALID_OPERATION) { 5476 inputStandBy(); 5477 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5478 keyValuePair.string()); 5479 } 5480 if (reconfig) { 5481 if (status == BAD_VALUE && 5482 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5483 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5484 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5485 <= (2 * samplingRate)) && 5486 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) 5487 <= FCC_2 && 5488 (channelMask == AUDIO_CHANNEL_IN_MONO || 5489 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5490 status = NO_ERROR; 5491 } 5492 if (status == NO_ERROR) { 5493 readInputParameters_l(); 5494 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5495 } 5496 } 5497 } 5498 5499 mNewParameters.removeAt(0); 5500 5501 mParamStatus = status; 5502 mParamCond.signal(); 5503 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5504 // already timed out waiting for the status and will never signal the condition. 5505 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5506 } 5507 return reconfig; 5508} 5509 5510String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5511{ 5512 Mutex::Autolock _l(mLock); 5513 if (initCheck() != NO_ERROR) { 5514 return String8(); 5515 } 5516 5517 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5518 const String8 out_s8(s); 5519 free(s); 5520 return out_s8; 5521} 5522 5523void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param __unused) { 5524 AudioSystem::OutputDescriptor desc; 5525 const void *param2 = NULL; 5526 5527 switch (event) { 5528 case AudioSystem::INPUT_OPENED: 5529 case AudioSystem::INPUT_CONFIG_CHANGED: 5530 desc.channelMask = mChannelMask; 5531 desc.samplingRate = mSampleRate; 5532 desc.format = mFormat; 5533 desc.frameCount = mFrameCount; 5534 desc.latency = 0; 5535 param2 = &desc; 5536 break; 5537 5538 case AudioSystem::INPUT_CLOSED: 5539 default: 5540 break; 5541 } 5542 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5543} 5544 5545void AudioFlinger::RecordThread::readInputParameters_l() 5546{ 5547 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5548 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5549 mChannelCount = popcount(mChannelMask); 5550 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5551 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5552 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5553 } 5554 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5555 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5556 mFrameCount = mBufferSize / mFrameSize; 5557 // This is the formula for calculating the temporary buffer size. 5558 // With 3 HAL buffers, we can guarantee ability to down-sample the input by ratio of 2:1 to 5559 // 1 full output buffer, regardless of the alignment of the available input. 5560 // The "3" is somewhat arbitrary, and could probably be larger. 5561 // A larger value should allow more old data to be read after a track calls start(), 5562 // without increasing latency. 5563 mRsmpInFrames = mFrameCount * 3; 5564 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5565 delete[] mRsmpInBuffer; 5566 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5567 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5568 5569 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5570 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5571} 5572 5573uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5574{ 5575 Mutex::Autolock _l(mLock); 5576 if (initCheck() != NO_ERROR) { 5577 return 0; 5578 } 5579 5580 return mInput->stream->get_input_frames_lost(mInput->stream); 5581} 5582 5583uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5584{ 5585 Mutex::Autolock _l(mLock); 5586 uint32_t result = 0; 5587 if (getEffectChain_l(sessionId) != 0) { 5588 result = EFFECT_SESSION; 5589 } 5590 5591 for (size_t i = 0; i < mTracks.size(); ++i) { 5592 if (sessionId == mTracks[i]->sessionId()) { 5593 result |= TRACK_SESSION; 5594 break; 5595 } 5596 } 5597 5598 return result; 5599} 5600 5601KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5602{ 5603 KeyedVector<int, bool> ids; 5604 Mutex::Autolock _l(mLock); 5605 for (size_t j = 0; j < mTracks.size(); ++j) { 5606 sp<RecordThread::RecordTrack> track = mTracks[j]; 5607 int sessionId = track->sessionId(); 5608 if (ids.indexOfKey(sessionId) < 0) { 5609 ids.add(sessionId, true); 5610 } 5611 } 5612 return ids; 5613} 5614 5615AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5616{ 5617 Mutex::Autolock _l(mLock); 5618 AudioStreamIn *input = mInput; 5619 mInput = NULL; 5620 return input; 5621} 5622 5623// this method must always be called either with ThreadBase mLock held or inside the thread loop 5624audio_stream_t* AudioFlinger::RecordThread::stream() const 5625{ 5626 if (mInput == NULL) { 5627 return NULL; 5628 } 5629 return &mInput->stream->common; 5630} 5631 5632status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5633{ 5634 // only one chain per input thread 5635 if (mEffectChains.size() != 0) { 5636 return INVALID_OPERATION; 5637 } 5638 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5639 5640 chain->setInBuffer(NULL); 5641 chain->setOutBuffer(NULL); 5642 5643 checkSuspendOnAddEffectChain_l(chain); 5644 5645 mEffectChains.add(chain); 5646 5647 return NO_ERROR; 5648} 5649 5650size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5651{ 5652 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5653 ALOGW_IF(mEffectChains.size() != 1, 5654 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5655 chain.get(), mEffectChains.size(), this); 5656 if (mEffectChains.size() == 1) { 5657 mEffectChains.removeAt(0); 5658 } 5659 return 0; 5660} 5661 5662}; // namespace android 5663