Threads.cpp revision 1c333e252cbca3337c1bedbc57a005f3b7d23fdb
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38#include <audio_utils/minifloat.h> 39 40// NBAIO implementations 41#include <media/nbaio/AudioStreamOutSink.h> 42#include <media/nbaio/MonoPipe.h> 43#include <media/nbaio/MonoPipeReader.h> 44#include <media/nbaio/Pipe.h> 45#include <media/nbaio/PipeReader.h> 46#include <media/nbaio/SourceAudioBufferProvider.h> 47 48#include <powermanager/PowerManager.h> 49 50#include <common_time/cc_helper.h> 51#include <common_time/local_clock.h> 52 53#include "AudioFlinger.h" 54#include "AudioMixer.h" 55#include "FastMixer.h" 56#include "ServiceUtilities.h" 57#include "SchedulingPolicyService.h" 58 59#ifdef ADD_BATTERY_DATA 60#include <media/IMediaPlayerService.h> 61#include <media/IMediaDeathNotifier.h> 62#endif 63 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait in sendConfigEvent_l() for a status to be received 102static const nsecs_t kConfigEventTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal sink buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalSinkBufferSizeMs = 20; 111// maximum normal sink buffer size 112static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 113 114// Offloaded output thread standby delay: allows track transition without going to standby 115static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 116 117// Whether to use fast mixer 118static const enum { 119 FastMixer_Never, // never initialize or use: for debugging only 120 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 121 // normal mixer multiplier is 1 122 FastMixer_Static, // initialize if needed, then use all the time if initialized, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 125 // multiplier is calculated based on min & max normal mixer buffer size 126 // FIXME for FastMixer_Dynamic: 127 // Supporting this option will require fixing HALs that can't handle large writes. 128 // For example, one HAL implementation returns an error from a large write, 129 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 130 // We could either fix the HAL implementations, or provide a wrapper that breaks 131 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 132} kUseFastMixer = FastMixer_Static; 133 134// Priorities for requestPriority 135static const int kPriorityAudioApp = 2; 136static const int kPriorityFastMixer = 3; 137 138// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 139// for the track. The client then sub-divides this into smaller buffers for its use. 140// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 141// So for now we just assume that client is double-buffered for fast tracks. 142// FIXME It would be better for client to tell AudioFlinger the value of N, 143// so AudioFlinger could allocate the right amount of memory. 144// See the client's minBufCount and mNotificationFramesAct calculations for details. 145static const int kFastTrackMultiplier = 2; 146 147// See Thread::readOnlyHeap(). 148// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 149// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 150// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 151static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 152 153// ---------------------------------------------------------------------------- 154 155#ifdef ADD_BATTERY_DATA 156// To collect the amplifier usage 157static void addBatteryData(uint32_t params) { 158 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 159 if (service == NULL) { 160 // it already logged 161 return; 162 } 163 164 service->addBatteryData(params); 165} 166#endif 167 168 169// ---------------------------------------------------------------------------- 170// CPU Stats 171// ---------------------------------------------------------------------------- 172 173class CpuStats { 174public: 175 CpuStats(); 176 void sample(const String8 &title); 177#ifdef DEBUG_CPU_USAGE 178private: 179 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 180 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 181 182 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 183 184 int mCpuNum; // thread's current CPU number 185 int mCpukHz; // frequency of thread's current CPU in kHz 186#endif 187}; 188 189CpuStats::CpuStats() 190#ifdef DEBUG_CPU_USAGE 191 : mCpuNum(-1), mCpukHz(-1) 192#endif 193{ 194} 195 196void CpuStats::sample(const String8 &title 197#ifndef DEBUG_CPU_USAGE 198 __unused 199#endif 200 ) { 201#ifdef DEBUG_CPU_USAGE 202 // get current thread's delta CPU time in wall clock ns 203 double wcNs; 204 bool valid = mCpuUsage.sampleAndEnable(wcNs); 205 206 // record sample for wall clock statistics 207 if (valid) { 208 mWcStats.sample(wcNs); 209 } 210 211 // get the current CPU number 212 int cpuNum = sched_getcpu(); 213 214 // get the current CPU frequency in kHz 215 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 216 217 // check if either CPU number or frequency changed 218 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 219 mCpuNum = cpuNum; 220 mCpukHz = cpukHz; 221 // ignore sample for purposes of cycles 222 valid = false; 223 } 224 225 // if no change in CPU number or frequency, then record sample for cycle statistics 226 if (valid && mCpukHz > 0) { 227 double cycles = wcNs * cpukHz * 0.000001; 228 mHzStats.sample(cycles); 229 } 230 231 unsigned n = mWcStats.n(); 232 // mCpuUsage.elapsed() is expensive, so don't call it every loop 233 if ((n & 127) == 1) { 234 long long elapsed = mCpuUsage.elapsed(); 235 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 236 double perLoop = elapsed / (double) n; 237 double perLoop100 = perLoop * 0.01; 238 double perLoop1k = perLoop * 0.001; 239 double mean = mWcStats.mean(); 240 double stddev = mWcStats.stddev(); 241 double minimum = mWcStats.minimum(); 242 double maximum = mWcStats.maximum(); 243 double meanCycles = mHzStats.mean(); 244 double stddevCycles = mHzStats.stddev(); 245 double minCycles = mHzStats.minimum(); 246 double maxCycles = mHzStats.maximum(); 247 mCpuUsage.resetElapsed(); 248 mWcStats.reset(); 249 mHzStats.reset(); 250 ALOGD("CPU usage for %s over past %.1f secs\n" 251 " (%u mixer loops at %.1f mean ms per loop):\n" 252 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 253 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 254 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 255 title.string(), 256 elapsed * .000000001, n, perLoop * .000001, 257 mean * .001, 258 stddev * .001, 259 minimum * .001, 260 maximum * .001, 261 mean / perLoop100, 262 stddev / perLoop100, 263 minimum / perLoop100, 264 maximum / perLoop100, 265 meanCycles / perLoop1k, 266 stddevCycles / perLoop1k, 267 minCycles / perLoop1k, 268 maxCycles / perLoop1k); 269 270 } 271 } 272#endif 273}; 274 275// ---------------------------------------------------------------------------- 276// ThreadBase 277// ---------------------------------------------------------------------------- 278 279AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 280 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 281 : Thread(false /*canCallJava*/), 282 mType(type), 283 mAudioFlinger(audioFlinger), 284 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 285 // are set by PlaybackThread::readOutputParameters_l() or 286 // RecordThread::readInputParameters_l() 287 //FIXME: mStandby should be true here. Is this some kind of hack? 288 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 289 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 290 // mName will be set by concrete (non-virtual) subclass 291 mDeathRecipient(new PMDeathRecipient(this)) 292{ 293} 294 295AudioFlinger::ThreadBase::~ThreadBase() 296{ 297 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 298 mConfigEvents.clear(); 299 300 // do not lock the mutex in destructor 301 releaseWakeLock_l(); 302 if (mPowerManager != 0) { 303 sp<IBinder> binder = mPowerManager->asBinder(); 304 binder->unlinkToDeath(mDeathRecipient); 305 } 306} 307 308status_t AudioFlinger::ThreadBase::readyToRun() 309{ 310 status_t status = initCheck(); 311 if (status == NO_ERROR) { 312 ALOGI("AudioFlinger's thread %p ready to run", this); 313 } else { 314 ALOGE("No working audio driver found."); 315 } 316 return status; 317} 318 319void AudioFlinger::ThreadBase::exit() 320{ 321 ALOGV("ThreadBase::exit"); 322 // do any cleanup required for exit to succeed 323 preExit(); 324 { 325 // This lock prevents the following race in thread (uniprocessor for illustration): 326 // if (!exitPending()) { 327 // // context switch from here to exit() 328 // // exit() calls requestExit(), what exitPending() observes 329 // // exit() calls signal(), which is dropped since no waiters 330 // // context switch back from exit() to here 331 // mWaitWorkCV.wait(...); 332 // // now thread is hung 333 // } 334 AutoMutex lock(mLock); 335 requestExit(); 336 mWaitWorkCV.broadcast(); 337 } 338 // When Thread::requestExitAndWait is made virtual and this method is renamed to 339 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 340 requestExitAndWait(); 341} 342 343status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 344{ 345 status_t status; 346 347 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 348 Mutex::Autolock _l(mLock); 349 350 return sendSetParameterConfigEvent_l(keyValuePairs); 351} 352 353// sendConfigEvent_l() must be called with ThreadBase::mLock held 354// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 355status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 356{ 357 status_t status = NO_ERROR; 358 359 mConfigEvents.add(event); 360 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 361 mWaitWorkCV.signal(); 362 mLock.unlock(); 363 { 364 Mutex::Autolock _l(event->mLock); 365 while (event->mWaitStatus) { 366 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 367 event->mStatus = TIMED_OUT; 368 event->mWaitStatus = false; 369 } 370 } 371 status = event->mStatus; 372 } 373 mLock.lock(); 374 return status; 375} 376 377void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 378{ 379 Mutex::Autolock _l(mLock); 380 sendIoConfigEvent_l(event, param); 381} 382 383// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 384void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 385{ 386 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 387 sendConfigEvent_l(configEvent); 388} 389 390// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 391void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 392{ 393 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 394 sendConfigEvent_l(configEvent); 395} 396 397// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 398status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 399{ 400 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 401 return sendConfigEvent_l(configEvent); 402} 403 404status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 405 const struct audio_patch *patch, 406 audio_patch_handle_t *handle) 407{ 408 Mutex::Autolock _l(mLock); 409 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 410 status_t status = sendConfigEvent_l(configEvent); 411 if (status == NO_ERROR) { 412 CreateAudioPatchConfigEventData *data = 413 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 414 *handle = data->mHandle; 415 } 416 return status; 417} 418 419status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 420 const audio_patch_handle_t handle) 421{ 422 Mutex::Autolock _l(mLock); 423 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 424 return sendConfigEvent_l(configEvent); 425} 426 427 428// post condition: mConfigEvents.isEmpty() 429void AudioFlinger::ThreadBase::processConfigEvents_l() 430{ 431 bool configChanged = false; 432 433 while (!mConfigEvents.isEmpty()) { 434 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 435 sp<ConfigEvent> event = mConfigEvents[0]; 436 mConfigEvents.removeAt(0); 437 switch (event->mType) { 438 case CFG_EVENT_PRIO: { 439 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 440 // FIXME Need to understand why this has to be done asynchronously 441 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 442 true /*asynchronous*/); 443 if (err != 0) { 444 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 445 data->mPrio, data->mPid, data->mTid, err); 446 } 447 } break; 448 case CFG_EVENT_IO: { 449 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 450 audioConfigChanged(data->mEvent, data->mParam); 451 } break; 452 case CFG_EVENT_SET_PARAMETER: { 453 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 454 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 455 configChanged = true; 456 } 457 } break; 458 case CFG_EVENT_CREATE_AUDIO_PATCH: { 459 CreateAudioPatchConfigEventData *data = 460 (CreateAudioPatchConfigEventData *)event->mData.get(); 461 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 462 } break; 463 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 464 ReleaseAudioPatchConfigEventData *data = 465 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 466 event->mStatus = releaseAudioPatch_l(data->mHandle); 467 } break; 468 default: 469 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 470 break; 471 } 472 { 473 Mutex::Autolock _l(event->mLock); 474 if (event->mWaitStatus) { 475 event->mWaitStatus = false; 476 event->mCond.signal(); 477 } 478 } 479 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 480 } 481 482 if (configChanged) { 483 cacheParameters_l(); 484 } 485} 486 487String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 488 String8 s; 489 if (output) { 490 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 491 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 492 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 493 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 494 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 495 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 496 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 497 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 498 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 499 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 500 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 501 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 502 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 503 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 504 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 505 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 506 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 507 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 508 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 509 } else { 510 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 511 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 512 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 513 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 514 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 515 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 516 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 517 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 518 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 519 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 520 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 521 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 522 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 523 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 524 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 525 } 526 int len = s.length(); 527 if (s.length() > 2) { 528 char *str = s.lockBuffer(len); 529 s.unlockBuffer(len - 2); 530 } 531 return s; 532} 533 534void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 535{ 536 const size_t SIZE = 256; 537 char buffer[SIZE]; 538 String8 result; 539 540 bool locked = AudioFlinger::dumpTryLock(mLock); 541 if (!locked) { 542 dprintf(fd, "thread %p maybe dead locked\n", this); 543 } 544 545 dprintf(fd, " I/O handle: %d\n", mId); 546 dprintf(fd, " TID: %d\n", getTid()); 547 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 548 dprintf(fd, " Sample rate: %u\n", mSampleRate); 549 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 550 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 551 dprintf(fd, " Channel Count: %u\n", mChannelCount); 552 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 553 channelMaskToString(mChannelMask, mType != RECORD).string()); 554 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 555 dprintf(fd, " Frame size: %zu\n", mFrameSize); 556 dprintf(fd, " Pending config events:"); 557 size_t numConfig = mConfigEvents.size(); 558 if (numConfig) { 559 for (size_t i = 0; i < numConfig; i++) { 560 mConfigEvents[i]->dump(buffer, SIZE); 561 dprintf(fd, "\n %s", buffer); 562 } 563 dprintf(fd, "\n"); 564 } else { 565 dprintf(fd, " none\n"); 566 } 567 568 if (locked) { 569 mLock.unlock(); 570 } 571} 572 573void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 574{ 575 const size_t SIZE = 256; 576 char buffer[SIZE]; 577 String8 result; 578 579 size_t numEffectChains = mEffectChains.size(); 580 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 581 write(fd, buffer, strlen(buffer)); 582 583 for (size_t i = 0; i < numEffectChains; ++i) { 584 sp<EffectChain> chain = mEffectChains[i]; 585 if (chain != 0) { 586 chain->dump(fd, args); 587 } 588 } 589} 590 591void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 592{ 593 Mutex::Autolock _l(mLock); 594 acquireWakeLock_l(uid); 595} 596 597String16 AudioFlinger::ThreadBase::getWakeLockTag() 598{ 599 switch (mType) { 600 case MIXER: 601 return String16("AudioMix"); 602 case DIRECT: 603 return String16("AudioDirectOut"); 604 case DUPLICATING: 605 return String16("AudioDup"); 606 case RECORD: 607 return String16("AudioIn"); 608 case OFFLOAD: 609 return String16("AudioOffload"); 610 default: 611 ALOG_ASSERT(false); 612 return String16("AudioUnknown"); 613 } 614} 615 616void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 617{ 618 getPowerManager_l(); 619 if (mPowerManager != 0) { 620 sp<IBinder> binder = new BBinder(); 621 status_t status; 622 if (uid >= 0) { 623 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 624 binder, 625 getWakeLockTag(), 626 String16("media"), 627 uid); 628 } else { 629 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 630 binder, 631 getWakeLockTag(), 632 String16("media")); 633 } 634 if (status == NO_ERROR) { 635 mWakeLockToken = binder; 636 } 637 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 638 } 639} 640 641void AudioFlinger::ThreadBase::releaseWakeLock() 642{ 643 Mutex::Autolock _l(mLock); 644 releaseWakeLock_l(); 645} 646 647void AudioFlinger::ThreadBase::releaseWakeLock_l() 648{ 649 if (mWakeLockToken != 0) { 650 ALOGV("releaseWakeLock_l() %s", mName); 651 if (mPowerManager != 0) { 652 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 653 } 654 mWakeLockToken.clear(); 655 } 656} 657 658void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 659 Mutex::Autolock _l(mLock); 660 updateWakeLockUids_l(uids); 661} 662 663void AudioFlinger::ThreadBase::getPowerManager_l() { 664 665 if (mPowerManager == 0) { 666 // use checkService() to avoid blocking if power service is not up yet 667 sp<IBinder> binder = 668 defaultServiceManager()->checkService(String16("power")); 669 if (binder == 0) { 670 ALOGW("Thread %s cannot connect to the power manager service", mName); 671 } else { 672 mPowerManager = interface_cast<IPowerManager>(binder); 673 binder->linkToDeath(mDeathRecipient); 674 } 675 } 676} 677 678void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 679 680 getPowerManager_l(); 681 if (mWakeLockToken == NULL) { 682 ALOGE("no wake lock to update!"); 683 return; 684 } 685 if (mPowerManager != 0) { 686 sp<IBinder> binder = new BBinder(); 687 status_t status; 688 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 689 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 690 } 691} 692 693void AudioFlinger::ThreadBase::clearPowerManager() 694{ 695 Mutex::Autolock _l(mLock); 696 releaseWakeLock_l(); 697 mPowerManager.clear(); 698} 699 700void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 701{ 702 sp<ThreadBase> thread = mThread.promote(); 703 if (thread != 0) { 704 thread->clearPowerManager(); 705 } 706 ALOGW("power manager service died !!!"); 707} 708 709void AudioFlinger::ThreadBase::setEffectSuspended( 710 const effect_uuid_t *type, bool suspend, int sessionId) 711{ 712 Mutex::Autolock _l(mLock); 713 setEffectSuspended_l(type, suspend, sessionId); 714} 715 716void AudioFlinger::ThreadBase::setEffectSuspended_l( 717 const effect_uuid_t *type, bool suspend, int sessionId) 718{ 719 sp<EffectChain> chain = getEffectChain_l(sessionId); 720 if (chain != 0) { 721 if (type != NULL) { 722 chain->setEffectSuspended_l(type, suspend); 723 } else { 724 chain->setEffectSuspendedAll_l(suspend); 725 } 726 } 727 728 updateSuspendedSessions_l(type, suspend, sessionId); 729} 730 731void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 732{ 733 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 734 if (index < 0) { 735 return; 736 } 737 738 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 739 mSuspendedSessions.valueAt(index); 740 741 for (size_t i = 0; i < sessionEffects.size(); i++) { 742 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 743 for (int j = 0; j < desc->mRefCount; j++) { 744 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 745 chain->setEffectSuspendedAll_l(true); 746 } else { 747 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 748 desc->mType.timeLow); 749 chain->setEffectSuspended_l(&desc->mType, true); 750 } 751 } 752 } 753} 754 755void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 756 bool suspend, 757 int sessionId) 758{ 759 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 760 761 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 762 763 if (suspend) { 764 if (index >= 0) { 765 sessionEffects = mSuspendedSessions.valueAt(index); 766 } else { 767 mSuspendedSessions.add(sessionId, sessionEffects); 768 } 769 } else { 770 if (index < 0) { 771 return; 772 } 773 sessionEffects = mSuspendedSessions.valueAt(index); 774 } 775 776 777 int key = EffectChain::kKeyForSuspendAll; 778 if (type != NULL) { 779 key = type->timeLow; 780 } 781 index = sessionEffects.indexOfKey(key); 782 783 sp<SuspendedSessionDesc> desc; 784 if (suspend) { 785 if (index >= 0) { 786 desc = sessionEffects.valueAt(index); 787 } else { 788 desc = new SuspendedSessionDesc(); 789 if (type != NULL) { 790 desc->mType = *type; 791 } 792 sessionEffects.add(key, desc); 793 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 794 } 795 desc->mRefCount++; 796 } else { 797 if (index < 0) { 798 return; 799 } 800 desc = sessionEffects.valueAt(index); 801 if (--desc->mRefCount == 0) { 802 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 803 sessionEffects.removeItemsAt(index); 804 if (sessionEffects.isEmpty()) { 805 ALOGV("updateSuspendedSessions_l() restore removing session %d", 806 sessionId); 807 mSuspendedSessions.removeItem(sessionId); 808 } 809 } 810 } 811 if (!sessionEffects.isEmpty()) { 812 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 813 } 814} 815 816void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 817 bool enabled, 818 int sessionId) 819{ 820 Mutex::Autolock _l(mLock); 821 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 822} 823 824void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 825 bool enabled, 826 int sessionId) 827{ 828 if (mType != RECORD) { 829 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 830 // another session. This gives the priority to well behaved effect control panels 831 // and applications not using global effects. 832 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 833 // global effects 834 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 835 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 836 } 837 } 838 839 sp<EffectChain> chain = getEffectChain_l(sessionId); 840 if (chain != 0) { 841 chain->checkSuspendOnEffectEnabled(effect, enabled); 842 } 843} 844 845// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 846sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 847 const sp<AudioFlinger::Client>& client, 848 const sp<IEffectClient>& effectClient, 849 int32_t priority, 850 int sessionId, 851 effect_descriptor_t *desc, 852 int *enabled, 853 status_t *status) 854{ 855 sp<EffectModule> effect; 856 sp<EffectHandle> handle; 857 status_t lStatus; 858 sp<EffectChain> chain; 859 bool chainCreated = false; 860 bool effectCreated = false; 861 bool effectRegistered = false; 862 863 lStatus = initCheck(); 864 if (lStatus != NO_ERROR) { 865 ALOGW("createEffect_l() Audio driver not initialized."); 866 goto Exit; 867 } 868 869 // Reject any effect on Direct output threads for now, since the format of 870 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 871 if (mType == DIRECT) { 872 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 873 desc->name, mName); 874 lStatus = BAD_VALUE; 875 goto Exit; 876 } 877 878 // Allow global effects only on offloaded and mixer threads 879 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 880 switch (mType) { 881 case MIXER: 882 case OFFLOAD: 883 break; 884 case DIRECT: 885 case DUPLICATING: 886 case RECORD: 887 default: 888 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 889 lStatus = BAD_VALUE; 890 goto Exit; 891 } 892 } 893 894 // Only Pre processor effects are allowed on input threads and only on input threads 895 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 896 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 897 desc->name, desc->flags, mType); 898 lStatus = BAD_VALUE; 899 goto Exit; 900 } 901 902 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 903 904 { // scope for mLock 905 Mutex::Autolock _l(mLock); 906 907 // check for existing effect chain with the requested audio session 908 chain = getEffectChain_l(sessionId); 909 if (chain == 0) { 910 // create a new chain for this session 911 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 912 chain = new EffectChain(this, sessionId); 913 addEffectChain_l(chain); 914 chain->setStrategy(getStrategyForSession_l(sessionId)); 915 chainCreated = true; 916 } else { 917 effect = chain->getEffectFromDesc_l(desc); 918 } 919 920 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 921 922 if (effect == 0) { 923 int id = mAudioFlinger->nextUniqueId(); 924 // Check CPU and memory usage 925 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 926 if (lStatus != NO_ERROR) { 927 goto Exit; 928 } 929 effectRegistered = true; 930 // create a new effect module if none present in the chain 931 effect = new EffectModule(this, chain, desc, id, sessionId); 932 lStatus = effect->status(); 933 if (lStatus != NO_ERROR) { 934 goto Exit; 935 } 936 effect->setOffloaded(mType == OFFLOAD, mId); 937 938 lStatus = chain->addEffect_l(effect); 939 if (lStatus != NO_ERROR) { 940 goto Exit; 941 } 942 effectCreated = true; 943 944 effect->setDevice(mOutDevice); 945 effect->setDevice(mInDevice); 946 effect->setMode(mAudioFlinger->getMode()); 947 effect->setAudioSource(mAudioSource); 948 } 949 // create effect handle and connect it to effect module 950 handle = new EffectHandle(effect, client, effectClient, priority); 951 lStatus = handle->initCheck(); 952 if (lStatus == OK) { 953 lStatus = effect->addHandle(handle.get()); 954 } 955 if (enabled != NULL) { 956 *enabled = (int)effect->isEnabled(); 957 } 958 } 959 960Exit: 961 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 962 Mutex::Autolock _l(mLock); 963 if (effectCreated) { 964 chain->removeEffect_l(effect); 965 } 966 if (effectRegistered) { 967 AudioSystem::unregisterEffect(effect->id()); 968 } 969 if (chainCreated) { 970 removeEffectChain_l(chain); 971 } 972 handle.clear(); 973 } 974 975 *status = lStatus; 976 return handle; 977} 978 979sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 980{ 981 Mutex::Autolock _l(mLock); 982 return getEffect_l(sessionId, effectId); 983} 984 985sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 986{ 987 sp<EffectChain> chain = getEffectChain_l(sessionId); 988 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 989} 990 991// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 992// PlaybackThread::mLock held 993status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 994{ 995 // check for existing effect chain with the requested audio session 996 int sessionId = effect->sessionId(); 997 sp<EffectChain> chain = getEffectChain_l(sessionId); 998 bool chainCreated = false; 999 1000 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1001 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1002 this, effect->desc().name, effect->desc().flags); 1003 1004 if (chain == 0) { 1005 // create a new chain for this session 1006 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1007 chain = new EffectChain(this, sessionId); 1008 addEffectChain_l(chain); 1009 chain->setStrategy(getStrategyForSession_l(sessionId)); 1010 chainCreated = true; 1011 } 1012 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1013 1014 if (chain->getEffectFromId_l(effect->id()) != 0) { 1015 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1016 this, effect->desc().name, chain.get()); 1017 return BAD_VALUE; 1018 } 1019 1020 effect->setOffloaded(mType == OFFLOAD, mId); 1021 1022 status_t status = chain->addEffect_l(effect); 1023 if (status != NO_ERROR) { 1024 if (chainCreated) { 1025 removeEffectChain_l(chain); 1026 } 1027 return status; 1028 } 1029 1030 effect->setDevice(mOutDevice); 1031 effect->setDevice(mInDevice); 1032 effect->setMode(mAudioFlinger->getMode()); 1033 effect->setAudioSource(mAudioSource); 1034 return NO_ERROR; 1035} 1036 1037void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1038 1039 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1040 effect_descriptor_t desc = effect->desc(); 1041 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1042 detachAuxEffect_l(effect->id()); 1043 } 1044 1045 sp<EffectChain> chain = effect->chain().promote(); 1046 if (chain != 0) { 1047 // remove effect chain if removing last effect 1048 if (chain->removeEffect_l(effect) == 0) { 1049 removeEffectChain_l(chain); 1050 } 1051 } else { 1052 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1053 } 1054} 1055 1056void AudioFlinger::ThreadBase::lockEffectChains_l( 1057 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1058{ 1059 effectChains = mEffectChains; 1060 for (size_t i = 0; i < mEffectChains.size(); i++) { 1061 mEffectChains[i]->lock(); 1062 } 1063} 1064 1065void AudioFlinger::ThreadBase::unlockEffectChains( 1066 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1067{ 1068 for (size_t i = 0; i < effectChains.size(); i++) { 1069 effectChains[i]->unlock(); 1070 } 1071} 1072 1073sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1074{ 1075 Mutex::Autolock _l(mLock); 1076 return getEffectChain_l(sessionId); 1077} 1078 1079sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1080{ 1081 size_t size = mEffectChains.size(); 1082 for (size_t i = 0; i < size; i++) { 1083 if (mEffectChains[i]->sessionId() == sessionId) { 1084 return mEffectChains[i]; 1085 } 1086 } 1087 return 0; 1088} 1089 1090void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1091{ 1092 Mutex::Autolock _l(mLock); 1093 size_t size = mEffectChains.size(); 1094 for (size_t i = 0; i < size; i++) { 1095 mEffectChains[i]->setMode_l(mode); 1096 } 1097} 1098 1099void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1100 EffectHandle *handle, 1101 bool unpinIfLast) { 1102 1103 Mutex::Autolock _l(mLock); 1104 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1105 // delete the effect module if removing last handle on it 1106 if (effect->removeHandle(handle) == 0) { 1107 if (!effect->isPinned() || unpinIfLast) { 1108 removeEffect_l(effect); 1109 AudioSystem::unregisterEffect(effect->id()); 1110 } 1111 } 1112} 1113 1114// ---------------------------------------------------------------------------- 1115// Playback 1116// ---------------------------------------------------------------------------- 1117 1118AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1119 AudioStreamOut* output, 1120 audio_io_handle_t id, 1121 audio_devices_t device, 1122 type_t type) 1123 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1124 mNormalFrameCount(0), mSinkBuffer(NULL), 1125 mMixerBufferEnabled(false), 1126 mMixerBuffer(NULL), 1127 mMixerBufferSize(0), 1128 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1129 mMixerBufferValid(false), 1130 mEffectBufferEnabled(false), 1131 mEffectBuffer(NULL), 1132 mEffectBufferSize(0), 1133 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1134 mEffectBufferValid(false), 1135 mSuspended(0), mBytesWritten(0), 1136 mActiveTracksGeneration(0), 1137 // mStreamTypes[] initialized in constructor body 1138 mOutput(output), 1139 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1140 mMixerStatus(MIXER_IDLE), 1141 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1142 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1143 mBytesRemaining(0), 1144 mCurrentWriteLength(0), 1145 mUseAsyncWrite(false), 1146 mWriteAckSequence(0), 1147 mDrainSequence(0), 1148 mSignalPending(false), 1149 mScreenState(AudioFlinger::mScreenState), 1150 // index 0 is reserved for normal mixer's submix 1151 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1152 // mLatchD, mLatchQ, 1153 mLatchDValid(false), mLatchQValid(false) 1154{ 1155 snprintf(mName, kNameLength, "AudioOut_%X", id); 1156 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1157 1158 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1159 // it would be safer to explicitly pass initial masterVolume/masterMute as 1160 // parameter. 1161 // 1162 // If the HAL we are using has support for master volume or master mute, 1163 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1164 // and the mute set to false). 1165 mMasterVolume = audioFlinger->masterVolume_l(); 1166 mMasterMute = audioFlinger->masterMute_l(); 1167 if (mOutput && mOutput->audioHwDev) { 1168 if (mOutput->audioHwDev->canSetMasterVolume()) { 1169 mMasterVolume = 1.0; 1170 } 1171 1172 if (mOutput->audioHwDev->canSetMasterMute()) { 1173 mMasterMute = false; 1174 } 1175 } 1176 1177 readOutputParameters_l(); 1178 1179 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1180 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1181 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1182 stream = (audio_stream_type_t) (stream + 1)) { 1183 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1184 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1185 } 1186 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1187 // because mAudioFlinger doesn't have one to copy from 1188} 1189 1190AudioFlinger::PlaybackThread::~PlaybackThread() 1191{ 1192 mAudioFlinger->unregisterWriter(mNBLogWriter); 1193 free(mSinkBuffer); 1194 free(mMixerBuffer); 1195 free(mEffectBuffer); 1196} 1197 1198void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1199{ 1200 dumpInternals(fd, args); 1201 dumpTracks(fd, args); 1202 dumpEffectChains(fd, args); 1203} 1204 1205void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1206{ 1207 const size_t SIZE = 256; 1208 char buffer[SIZE]; 1209 String8 result; 1210 1211 result.appendFormat(" Stream volumes in dB: "); 1212 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1213 const stream_type_t *st = &mStreamTypes[i]; 1214 if (i > 0) { 1215 result.appendFormat(", "); 1216 } 1217 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1218 if (st->mute) { 1219 result.append("M"); 1220 } 1221 } 1222 result.append("\n"); 1223 write(fd, result.string(), result.length()); 1224 result.clear(); 1225 1226 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1227 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1228 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1229 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1230 1231 size_t numtracks = mTracks.size(); 1232 size_t numactive = mActiveTracks.size(); 1233 dprintf(fd, " %d Tracks", numtracks); 1234 size_t numactiveseen = 0; 1235 if (numtracks) { 1236 dprintf(fd, " of which %d are active\n", numactive); 1237 Track::appendDumpHeader(result); 1238 for (size_t i = 0; i < numtracks; ++i) { 1239 sp<Track> track = mTracks[i]; 1240 if (track != 0) { 1241 bool active = mActiveTracks.indexOf(track) >= 0; 1242 if (active) { 1243 numactiveseen++; 1244 } 1245 track->dump(buffer, SIZE, active); 1246 result.append(buffer); 1247 } 1248 } 1249 } else { 1250 result.append("\n"); 1251 } 1252 if (numactiveseen != numactive) { 1253 // some tracks in the active list were not in the tracks list 1254 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1255 " not in the track list\n"); 1256 result.append(buffer); 1257 Track::appendDumpHeader(result); 1258 for (size_t i = 0; i < numactive; ++i) { 1259 sp<Track> track = mActiveTracks[i].promote(); 1260 if (track != 0 && mTracks.indexOf(track) < 0) { 1261 track->dump(buffer, SIZE, true); 1262 result.append(buffer); 1263 } 1264 } 1265 } 1266 1267 write(fd, result.string(), result.size()); 1268} 1269 1270void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1271{ 1272 dprintf(fd, "\nOutput thread %p:\n", this); 1273 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1274 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1275 dprintf(fd, " Total writes: %d\n", mNumWrites); 1276 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1277 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1278 dprintf(fd, " Suspend count: %d\n", mSuspended); 1279 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1280 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1281 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1282 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1283 1284 dumpBase(fd, args); 1285} 1286 1287// Thread virtuals 1288 1289void AudioFlinger::PlaybackThread::onFirstRef() 1290{ 1291 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1292} 1293 1294// ThreadBase virtuals 1295void AudioFlinger::PlaybackThread::preExit() 1296{ 1297 ALOGV(" preExit()"); 1298 // FIXME this is using hard-coded strings but in the future, this functionality will be 1299 // converted to use audio HAL extensions required to support tunneling 1300 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1301} 1302 1303// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1304sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1305 const sp<AudioFlinger::Client>& client, 1306 audio_stream_type_t streamType, 1307 uint32_t sampleRate, 1308 audio_format_t format, 1309 audio_channel_mask_t channelMask, 1310 size_t *pFrameCount, 1311 const sp<IMemory>& sharedBuffer, 1312 int sessionId, 1313 IAudioFlinger::track_flags_t *flags, 1314 pid_t tid, 1315 int uid, 1316 status_t *status) 1317{ 1318 size_t frameCount = *pFrameCount; 1319 sp<Track> track; 1320 status_t lStatus; 1321 1322 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1323 1324 // client expresses a preference for FAST, but we get the final say 1325 if (*flags & IAudioFlinger::TRACK_FAST) { 1326 if ( 1327 // not timed 1328 (!isTimed) && 1329 // either of these use cases: 1330 ( 1331 // use case 1: shared buffer with any frame count 1332 ( 1333 (sharedBuffer != 0) 1334 ) || 1335 // use case 2: callback handler and frame count is default or at least as large as HAL 1336 ( 1337 (tid != -1) && 1338 ((frameCount == 0) || 1339 (frameCount >= mFrameCount)) 1340 ) 1341 ) && 1342 // PCM data 1343 audio_is_linear_pcm(format) && 1344 // mono or stereo 1345 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1346 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1347 // hardware sample rate 1348 (sampleRate == mSampleRate) && 1349 // normal mixer has an associated fast mixer 1350 hasFastMixer() && 1351 // there are sufficient fast track slots available 1352 (mFastTrackAvailMask != 0) 1353 // FIXME test that MixerThread for this fast track has a capable output HAL 1354 // FIXME add a permission test also? 1355 ) { 1356 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1357 if (frameCount == 0) { 1358 frameCount = mFrameCount * kFastTrackMultiplier; 1359 } 1360 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1361 frameCount, mFrameCount); 1362 } else { 1363 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1364 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1365 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1366 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1367 audio_is_linear_pcm(format), 1368 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1369 *flags &= ~IAudioFlinger::TRACK_FAST; 1370 // For compatibility with AudioTrack calculation, buffer depth is forced 1371 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1372 // This is probably too conservative, but legacy application code may depend on it. 1373 // If you change this calculation, also review the start threshold which is related. 1374 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1375 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1376 if (minBufCount < 2) { 1377 minBufCount = 2; 1378 } 1379 size_t minFrameCount = mNormalFrameCount * minBufCount; 1380 if (frameCount < minFrameCount) { 1381 frameCount = minFrameCount; 1382 } 1383 } 1384 } 1385 *pFrameCount = frameCount; 1386 1387 switch (mType) { 1388 1389 case DIRECT: 1390 if (audio_is_linear_pcm(format)) { 1391 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1392 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1393 "for output %p with format %#x", 1394 sampleRate, format, channelMask, mOutput, mFormat); 1395 lStatus = BAD_VALUE; 1396 goto Exit; 1397 } 1398 } 1399 break; 1400 1401 case OFFLOAD: 1402 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1403 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1404 "for output %p with format %#x", 1405 sampleRate, format, channelMask, mOutput, mFormat); 1406 lStatus = BAD_VALUE; 1407 goto Exit; 1408 } 1409 break; 1410 1411 default: 1412 if (!audio_is_linear_pcm(format)) { 1413 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1414 "for output %p with format %#x", 1415 format, mOutput, mFormat); 1416 lStatus = BAD_VALUE; 1417 goto Exit; 1418 } 1419 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1420 if (sampleRate > mSampleRate*2) { 1421 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1422 lStatus = BAD_VALUE; 1423 goto Exit; 1424 } 1425 break; 1426 1427 } 1428 1429 lStatus = initCheck(); 1430 if (lStatus != NO_ERROR) { 1431 ALOGE("createTrack_l() audio driver not initialized"); 1432 goto Exit; 1433 } 1434 1435 { // scope for mLock 1436 Mutex::Autolock _l(mLock); 1437 1438 // all tracks in same audio session must share the same routing strategy otherwise 1439 // conflicts will happen when tracks are moved from one output to another by audio policy 1440 // manager 1441 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1442 for (size_t i = 0; i < mTracks.size(); ++i) { 1443 sp<Track> t = mTracks[i]; 1444 if (t != 0 && !t->isOutputTrack()) { 1445 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1446 if (sessionId == t->sessionId() && strategy != actual) { 1447 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1448 strategy, actual); 1449 lStatus = BAD_VALUE; 1450 goto Exit; 1451 } 1452 } 1453 } 1454 1455 if (!isTimed) { 1456 track = new Track(this, client, streamType, sampleRate, format, 1457 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1458 } else { 1459 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1460 channelMask, frameCount, sharedBuffer, sessionId, uid); 1461 } 1462 1463 // new Track always returns non-NULL, 1464 // but TimedTrack::create() is a factory that could fail by returning NULL 1465 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1466 if (lStatus != NO_ERROR) { 1467 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1468 // track must be cleared from the caller as the caller has the AF lock 1469 goto Exit; 1470 } 1471 mTracks.add(track); 1472 1473 sp<EffectChain> chain = getEffectChain_l(sessionId); 1474 if (chain != 0) { 1475 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1476 track->setMainBuffer(chain->inBuffer()); 1477 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1478 chain->incTrackCnt(); 1479 } 1480 1481 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1482 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1483 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1484 // so ask activity manager to do this on our behalf 1485 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1486 } 1487 } 1488 1489 lStatus = NO_ERROR; 1490 1491Exit: 1492 *status = lStatus; 1493 return track; 1494} 1495 1496uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1497{ 1498 return latency; 1499} 1500 1501uint32_t AudioFlinger::PlaybackThread::latency() const 1502{ 1503 Mutex::Autolock _l(mLock); 1504 return latency_l(); 1505} 1506uint32_t AudioFlinger::PlaybackThread::latency_l() const 1507{ 1508 if (initCheck() == NO_ERROR) { 1509 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1510 } else { 1511 return 0; 1512 } 1513} 1514 1515void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1516{ 1517 Mutex::Autolock _l(mLock); 1518 // Don't apply master volume in SW if our HAL can do it for us. 1519 if (mOutput && mOutput->audioHwDev && 1520 mOutput->audioHwDev->canSetMasterVolume()) { 1521 mMasterVolume = 1.0; 1522 } else { 1523 mMasterVolume = value; 1524 } 1525} 1526 1527void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1528{ 1529 Mutex::Autolock _l(mLock); 1530 // Don't apply master mute in SW if our HAL can do it for us. 1531 if (mOutput && mOutput->audioHwDev && 1532 mOutput->audioHwDev->canSetMasterMute()) { 1533 mMasterMute = false; 1534 } else { 1535 mMasterMute = muted; 1536 } 1537} 1538 1539void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1540{ 1541 Mutex::Autolock _l(mLock); 1542 mStreamTypes[stream].volume = value; 1543 broadcast_l(); 1544} 1545 1546void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1547{ 1548 Mutex::Autolock _l(mLock); 1549 mStreamTypes[stream].mute = muted; 1550 broadcast_l(); 1551} 1552 1553float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1554{ 1555 Mutex::Autolock _l(mLock); 1556 return mStreamTypes[stream].volume; 1557} 1558 1559// addTrack_l() must be called with ThreadBase::mLock held 1560status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1561{ 1562 status_t status = ALREADY_EXISTS; 1563 1564 // set retry count for buffer fill 1565 track->mRetryCount = kMaxTrackStartupRetries; 1566 if (mActiveTracks.indexOf(track) < 0) { 1567 // the track is newly added, make sure it fills up all its 1568 // buffers before playing. This is to ensure the client will 1569 // effectively get the latency it requested. 1570 if (!track->isOutputTrack()) { 1571 TrackBase::track_state state = track->mState; 1572 mLock.unlock(); 1573 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1574 mLock.lock(); 1575 // abort track was stopped/paused while we released the lock 1576 if (state != track->mState) { 1577 if (status == NO_ERROR) { 1578 mLock.unlock(); 1579 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1580 mLock.lock(); 1581 } 1582 return INVALID_OPERATION; 1583 } 1584 // abort if start is rejected by audio policy manager 1585 if (status != NO_ERROR) { 1586 return PERMISSION_DENIED; 1587 } 1588#ifdef ADD_BATTERY_DATA 1589 // to track the speaker usage 1590 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1591#endif 1592 } 1593 1594 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1595 track->mResetDone = false; 1596 track->mPresentationCompleteFrames = 0; 1597 mActiveTracks.add(track); 1598 mWakeLockUids.add(track->uid()); 1599 mActiveTracksGeneration++; 1600 mLatestActiveTrack = track; 1601 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1602 if (chain != 0) { 1603 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1604 track->sessionId()); 1605 chain->incActiveTrackCnt(); 1606 } 1607 1608 status = NO_ERROR; 1609 } 1610 1611 onAddNewTrack_l(); 1612 return status; 1613} 1614 1615bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1616{ 1617 track->terminate(); 1618 // active tracks are removed by threadLoop() 1619 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1620 track->mState = TrackBase::STOPPED; 1621 if (!trackActive) { 1622 removeTrack_l(track); 1623 } else if (track->isFastTrack() || track->isOffloaded()) { 1624 track->mState = TrackBase::STOPPING_1; 1625 } 1626 1627 return trackActive; 1628} 1629 1630void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1631{ 1632 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1633 mTracks.remove(track); 1634 deleteTrackName_l(track->name()); 1635 // redundant as track is about to be destroyed, for dumpsys only 1636 track->mName = -1; 1637 if (track->isFastTrack()) { 1638 int index = track->mFastIndex; 1639 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1640 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1641 mFastTrackAvailMask |= 1 << index; 1642 // redundant as track is about to be destroyed, for dumpsys only 1643 track->mFastIndex = -1; 1644 } 1645 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1646 if (chain != 0) { 1647 chain->decTrackCnt(); 1648 } 1649} 1650 1651void AudioFlinger::PlaybackThread::broadcast_l() 1652{ 1653 // Thread could be blocked waiting for async 1654 // so signal it to handle state changes immediately 1655 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1656 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1657 mSignalPending = true; 1658 mWaitWorkCV.broadcast(); 1659} 1660 1661String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1662{ 1663 Mutex::Autolock _l(mLock); 1664 if (initCheck() != NO_ERROR) { 1665 return String8(); 1666 } 1667 1668 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1669 const String8 out_s8(s); 1670 free(s); 1671 return out_s8; 1672} 1673 1674void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1675 AudioSystem::OutputDescriptor desc; 1676 void *param2 = NULL; 1677 1678 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1679 param); 1680 1681 switch (event) { 1682 case AudioSystem::OUTPUT_OPENED: 1683 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1684 desc.channelMask = mChannelMask; 1685 desc.samplingRate = mSampleRate; 1686 desc.format = mFormat; 1687 desc.frameCount = mNormalFrameCount; // FIXME see 1688 // AudioFlinger::frameCount(audio_io_handle_t) 1689 desc.latency = latency_l(); 1690 param2 = &desc; 1691 break; 1692 1693 case AudioSystem::STREAM_CONFIG_CHANGED: 1694 param2 = ¶m; 1695 case AudioSystem::OUTPUT_CLOSED: 1696 default: 1697 break; 1698 } 1699 mAudioFlinger->audioConfigChanged(event, mId, param2); 1700} 1701 1702void AudioFlinger::PlaybackThread::writeCallback() 1703{ 1704 ALOG_ASSERT(mCallbackThread != 0); 1705 mCallbackThread->resetWriteBlocked(); 1706} 1707 1708void AudioFlinger::PlaybackThread::drainCallback() 1709{ 1710 ALOG_ASSERT(mCallbackThread != 0); 1711 mCallbackThread->resetDraining(); 1712} 1713 1714void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1715{ 1716 Mutex::Autolock _l(mLock); 1717 // reject out of sequence requests 1718 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1719 mWriteAckSequence &= ~1; 1720 mWaitWorkCV.signal(); 1721 } 1722} 1723 1724void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1725{ 1726 Mutex::Autolock _l(mLock); 1727 // reject out of sequence requests 1728 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1729 mDrainSequence &= ~1; 1730 mWaitWorkCV.signal(); 1731 } 1732} 1733 1734// static 1735int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1736 void *param __unused, 1737 void *cookie) 1738{ 1739 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1740 ALOGV("asyncCallback() event %d", event); 1741 switch (event) { 1742 case STREAM_CBK_EVENT_WRITE_READY: 1743 me->writeCallback(); 1744 break; 1745 case STREAM_CBK_EVENT_DRAIN_READY: 1746 me->drainCallback(); 1747 break; 1748 default: 1749 ALOGW("asyncCallback() unknown event %d", event); 1750 break; 1751 } 1752 return 0; 1753} 1754 1755void AudioFlinger::PlaybackThread::readOutputParameters_l() 1756{ 1757 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1758 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1759 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1760 if (!audio_is_output_channel(mChannelMask)) { 1761 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1762 } 1763 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1764 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1765 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1766 } 1767 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1768 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1769 if (!audio_is_valid_format(mFormat)) { 1770 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1771 } 1772 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1773 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1774 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1775 } 1776 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1777 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1778 mFrameCount = mBufferSize / mFrameSize; 1779 if (mFrameCount & 15) { 1780 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1781 mFrameCount); 1782 } 1783 1784 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1785 (mOutput->stream->set_callback != NULL)) { 1786 if (mOutput->stream->set_callback(mOutput->stream, 1787 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1788 mUseAsyncWrite = true; 1789 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1790 } 1791 } 1792 1793 // Calculate size of normal sink buffer relative to the HAL output buffer size 1794 double multiplier = 1.0; 1795 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1796 kUseFastMixer == FastMixer_Dynamic)) { 1797 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1798 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1799 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1800 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1801 maxNormalFrameCount = maxNormalFrameCount & ~15; 1802 if (maxNormalFrameCount < minNormalFrameCount) { 1803 maxNormalFrameCount = minNormalFrameCount; 1804 } 1805 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1806 if (multiplier <= 1.0) { 1807 multiplier = 1.0; 1808 } else if (multiplier <= 2.0) { 1809 if (2 * mFrameCount <= maxNormalFrameCount) { 1810 multiplier = 2.0; 1811 } else { 1812 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1813 } 1814 } else { 1815 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1816 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1817 // track, but we sometimes have to do this to satisfy the maximum frame count 1818 // constraint) 1819 // FIXME this rounding up should not be done if no HAL SRC 1820 uint32_t truncMult = (uint32_t) multiplier; 1821 if ((truncMult & 1)) { 1822 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1823 ++truncMult; 1824 } 1825 } 1826 multiplier = (double) truncMult; 1827 } 1828 } 1829 mNormalFrameCount = multiplier * mFrameCount; 1830 // round up to nearest 16 frames to satisfy AudioMixer 1831 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1832 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1833 mNormalFrameCount); 1834 1835 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1836 // Originally this was int16_t[] array, need to remove legacy implications. 1837 free(mSinkBuffer); 1838 mSinkBuffer = NULL; 1839 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1840 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1841 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1842 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1843 1844 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1845 // drives the output. 1846 free(mMixerBuffer); 1847 mMixerBuffer = NULL; 1848 if (mMixerBufferEnabled) { 1849 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1850 mMixerBufferSize = mNormalFrameCount * mChannelCount 1851 * audio_bytes_per_sample(mMixerBufferFormat); 1852 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1853 } 1854 free(mEffectBuffer); 1855 mEffectBuffer = NULL; 1856 if (mEffectBufferEnabled) { 1857 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1858 mEffectBufferSize = mNormalFrameCount * mChannelCount 1859 * audio_bytes_per_sample(mEffectBufferFormat); 1860 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1861 } 1862 1863 // force reconfiguration of effect chains and engines to take new buffer size and audio 1864 // parameters into account 1865 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1866 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1867 // matter. 1868 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1869 Vector< sp<EffectChain> > effectChains = mEffectChains; 1870 for (size_t i = 0; i < effectChains.size(); i ++) { 1871 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1872 } 1873} 1874 1875 1876status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1877{ 1878 if (halFrames == NULL || dspFrames == NULL) { 1879 return BAD_VALUE; 1880 } 1881 Mutex::Autolock _l(mLock); 1882 if (initCheck() != NO_ERROR) { 1883 return INVALID_OPERATION; 1884 } 1885 size_t framesWritten = mBytesWritten / mFrameSize; 1886 *halFrames = framesWritten; 1887 1888 if (isSuspended()) { 1889 // return an estimation of rendered frames when the output is suspended 1890 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1891 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1892 return NO_ERROR; 1893 } else { 1894 status_t status; 1895 uint32_t frames; 1896 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1897 *dspFrames = (size_t)frames; 1898 return status; 1899 } 1900} 1901 1902uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1903{ 1904 Mutex::Autolock _l(mLock); 1905 uint32_t result = 0; 1906 if (getEffectChain_l(sessionId) != 0) { 1907 result = EFFECT_SESSION; 1908 } 1909 1910 for (size_t i = 0; i < mTracks.size(); ++i) { 1911 sp<Track> track = mTracks[i]; 1912 if (sessionId == track->sessionId() && !track->isInvalid()) { 1913 result |= TRACK_SESSION; 1914 break; 1915 } 1916 } 1917 1918 return result; 1919} 1920 1921uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1922{ 1923 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1924 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1925 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1926 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1927 } 1928 for (size_t i = 0; i < mTracks.size(); i++) { 1929 sp<Track> track = mTracks[i]; 1930 if (sessionId == track->sessionId() && !track->isInvalid()) { 1931 return AudioSystem::getStrategyForStream(track->streamType()); 1932 } 1933 } 1934 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1935} 1936 1937 1938AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1939{ 1940 Mutex::Autolock _l(mLock); 1941 return mOutput; 1942} 1943 1944AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1945{ 1946 Mutex::Autolock _l(mLock); 1947 AudioStreamOut *output = mOutput; 1948 mOutput = NULL; 1949 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1950 // must push a NULL and wait for ack 1951 mOutputSink.clear(); 1952 mPipeSink.clear(); 1953 mNormalSink.clear(); 1954 return output; 1955} 1956 1957// this method must always be called either with ThreadBase mLock held or inside the thread loop 1958audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1959{ 1960 if (mOutput == NULL) { 1961 return NULL; 1962 } 1963 return &mOutput->stream->common; 1964} 1965 1966uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1967{ 1968 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1969} 1970 1971status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1972{ 1973 if (!isValidSyncEvent(event)) { 1974 return BAD_VALUE; 1975 } 1976 1977 Mutex::Autolock _l(mLock); 1978 1979 for (size_t i = 0; i < mTracks.size(); ++i) { 1980 sp<Track> track = mTracks[i]; 1981 if (event->triggerSession() == track->sessionId()) { 1982 (void) track->setSyncEvent(event); 1983 return NO_ERROR; 1984 } 1985 } 1986 1987 return NAME_NOT_FOUND; 1988} 1989 1990bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 1991{ 1992 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 1993} 1994 1995void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 1996 const Vector< sp<Track> >& tracksToRemove) 1997{ 1998 size_t count = tracksToRemove.size(); 1999 if (count > 0) { 2000 for (size_t i = 0 ; i < count ; i++) { 2001 const sp<Track>& track = tracksToRemove.itemAt(i); 2002 if (!track->isOutputTrack()) { 2003 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2004#ifdef ADD_BATTERY_DATA 2005 // to track the speaker usage 2006 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2007#endif 2008 if (track->isTerminated()) { 2009 AudioSystem::releaseOutput(mId); 2010 } 2011 } 2012 } 2013 } 2014} 2015 2016void AudioFlinger::PlaybackThread::checkSilentMode_l() 2017{ 2018 if (!mMasterMute) { 2019 char value[PROPERTY_VALUE_MAX]; 2020 if (property_get("ro.audio.silent", value, "0") > 0) { 2021 char *endptr; 2022 unsigned long ul = strtoul(value, &endptr, 0); 2023 if (*endptr == '\0' && ul != 0) { 2024 ALOGD("Silence is golden"); 2025 // The setprop command will not allow a property to be changed after 2026 // the first time it is set, so we don't have to worry about un-muting. 2027 setMasterMute_l(true); 2028 } 2029 } 2030 } 2031} 2032 2033// shared by MIXER and DIRECT, overridden by DUPLICATING 2034ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2035{ 2036 // FIXME rewrite to reduce number of system calls 2037 mLastWriteTime = systemTime(); 2038 mInWrite = true; 2039 ssize_t bytesWritten; 2040 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2041 2042 // If an NBAIO sink is present, use it to write the normal mixer's submix 2043 if (mNormalSink != 0) { 2044 const size_t count = mBytesRemaining / mFrameSize; 2045 2046 ATRACE_BEGIN("write"); 2047 // update the setpoint when AudioFlinger::mScreenState changes 2048 uint32_t screenState = AudioFlinger::mScreenState; 2049 if (screenState != mScreenState) { 2050 mScreenState = screenState; 2051 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2052 if (pipe != NULL) { 2053 pipe->setAvgFrames((mScreenState & 1) ? 2054 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2055 } 2056 } 2057 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2058 ATRACE_END(); 2059 if (framesWritten > 0) { 2060 bytesWritten = framesWritten * mFrameSize; 2061 } else { 2062 bytesWritten = framesWritten; 2063 } 2064 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2065 if (status == NO_ERROR) { 2066 size_t totalFramesWritten = mNormalSink->framesWritten(); 2067 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2068 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2069 mLatchDValid = true; 2070 } 2071 } 2072 // otherwise use the HAL / AudioStreamOut directly 2073 } else { 2074 // Direct output and offload threads 2075 2076 if (mUseAsyncWrite) { 2077 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2078 mWriteAckSequence += 2; 2079 mWriteAckSequence |= 1; 2080 ALOG_ASSERT(mCallbackThread != 0); 2081 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2082 } 2083 // FIXME We should have an implementation of timestamps for direct output threads. 2084 // They are used e.g for multichannel PCM playback over HDMI. 2085 bytesWritten = mOutput->stream->write(mOutput->stream, 2086 (char *)mSinkBuffer + offset, mBytesRemaining); 2087 if (mUseAsyncWrite && 2088 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2089 // do not wait for async callback in case of error of full write 2090 mWriteAckSequence &= ~1; 2091 ALOG_ASSERT(mCallbackThread != 0); 2092 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2093 } 2094 } 2095 2096 mNumWrites++; 2097 mInWrite = false; 2098 mStandby = false; 2099 return bytesWritten; 2100} 2101 2102void AudioFlinger::PlaybackThread::threadLoop_drain() 2103{ 2104 if (mOutput->stream->drain) { 2105 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2106 if (mUseAsyncWrite) { 2107 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2108 mDrainSequence |= 1; 2109 ALOG_ASSERT(mCallbackThread != 0); 2110 mCallbackThread->setDraining(mDrainSequence); 2111 } 2112 mOutput->stream->drain(mOutput->stream, 2113 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2114 : AUDIO_DRAIN_ALL); 2115 } 2116} 2117 2118void AudioFlinger::PlaybackThread::threadLoop_exit() 2119{ 2120 // Default implementation has nothing to do 2121} 2122 2123/* 2124The derived values that are cached: 2125 - mSinkBufferSize from frame count * frame size 2126 - activeSleepTime from activeSleepTimeUs() 2127 - idleSleepTime from idleSleepTimeUs() 2128 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2129 - maxPeriod from frame count and sample rate (MIXER only) 2130 2131The parameters that affect these derived values are: 2132 - frame count 2133 - frame size 2134 - sample rate 2135 - device type: A2DP or not 2136 - device latency 2137 - format: PCM or not 2138 - active sleep time 2139 - idle sleep time 2140*/ 2141 2142void AudioFlinger::PlaybackThread::cacheParameters_l() 2143{ 2144 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2145 activeSleepTime = activeSleepTimeUs(); 2146 idleSleepTime = idleSleepTimeUs(); 2147} 2148 2149void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2150{ 2151 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2152 this, streamType, mTracks.size()); 2153 Mutex::Autolock _l(mLock); 2154 2155 size_t size = mTracks.size(); 2156 for (size_t i = 0; i < size; i++) { 2157 sp<Track> t = mTracks[i]; 2158 if (t->streamType() == streamType) { 2159 t->invalidate(); 2160 } 2161 } 2162} 2163 2164status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2165{ 2166 int session = chain->sessionId(); 2167 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2168 ? mEffectBuffer : mSinkBuffer); 2169 bool ownsBuffer = false; 2170 2171 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2172 if (session > 0) { 2173 // Only one effect chain can be present in direct output thread and it uses 2174 // the sink buffer as input 2175 if (mType != DIRECT) { 2176 size_t numSamples = mNormalFrameCount * mChannelCount; 2177 buffer = new int16_t[numSamples]; 2178 memset(buffer, 0, numSamples * sizeof(int16_t)); 2179 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2180 ownsBuffer = true; 2181 } 2182 2183 // Attach all tracks with same session ID to this chain. 2184 for (size_t i = 0; i < mTracks.size(); ++i) { 2185 sp<Track> track = mTracks[i]; 2186 if (session == track->sessionId()) { 2187 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2188 buffer); 2189 track->setMainBuffer(buffer); 2190 chain->incTrackCnt(); 2191 } 2192 } 2193 2194 // indicate all active tracks in the chain 2195 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2196 sp<Track> track = mActiveTracks[i].promote(); 2197 if (track == 0) { 2198 continue; 2199 } 2200 if (session == track->sessionId()) { 2201 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2202 chain->incActiveTrackCnt(); 2203 } 2204 } 2205 } 2206 2207 chain->setInBuffer(buffer, ownsBuffer); 2208 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2209 ? mEffectBuffer : mSinkBuffer)); 2210 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2211 // chains list in order to be processed last as it contains output stage effects 2212 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2213 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2214 // after track specific effects and before output stage 2215 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2216 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2217 // Effect chain for other sessions are inserted at beginning of effect 2218 // chains list to be processed before output mix effects. Relative order between other 2219 // sessions is not important 2220 size_t size = mEffectChains.size(); 2221 size_t i = 0; 2222 for (i = 0; i < size; i++) { 2223 if (mEffectChains[i]->sessionId() < session) { 2224 break; 2225 } 2226 } 2227 mEffectChains.insertAt(chain, i); 2228 checkSuspendOnAddEffectChain_l(chain); 2229 2230 return NO_ERROR; 2231} 2232 2233size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2234{ 2235 int session = chain->sessionId(); 2236 2237 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2238 2239 for (size_t i = 0; i < mEffectChains.size(); i++) { 2240 if (chain == mEffectChains[i]) { 2241 mEffectChains.removeAt(i); 2242 // detach all active tracks from the chain 2243 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2244 sp<Track> track = mActiveTracks[i].promote(); 2245 if (track == 0) { 2246 continue; 2247 } 2248 if (session == track->sessionId()) { 2249 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2250 chain.get(), session); 2251 chain->decActiveTrackCnt(); 2252 } 2253 } 2254 2255 // detach all tracks with same session ID from this chain 2256 for (size_t i = 0; i < mTracks.size(); ++i) { 2257 sp<Track> track = mTracks[i]; 2258 if (session == track->sessionId()) { 2259 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2260 chain->decTrackCnt(); 2261 } 2262 } 2263 break; 2264 } 2265 } 2266 return mEffectChains.size(); 2267} 2268 2269status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2270 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2271{ 2272 Mutex::Autolock _l(mLock); 2273 return attachAuxEffect_l(track, EffectId); 2274} 2275 2276status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2277 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2278{ 2279 status_t status = NO_ERROR; 2280 2281 if (EffectId == 0) { 2282 track->setAuxBuffer(0, NULL); 2283 } else { 2284 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2285 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2286 if (effect != 0) { 2287 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2288 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2289 } else { 2290 status = INVALID_OPERATION; 2291 } 2292 } else { 2293 status = BAD_VALUE; 2294 } 2295 } 2296 return status; 2297} 2298 2299void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2300{ 2301 for (size_t i = 0; i < mTracks.size(); ++i) { 2302 sp<Track> track = mTracks[i]; 2303 if (track->auxEffectId() == effectId) { 2304 attachAuxEffect_l(track, 0); 2305 } 2306 } 2307} 2308 2309bool AudioFlinger::PlaybackThread::threadLoop() 2310{ 2311 Vector< sp<Track> > tracksToRemove; 2312 2313 standbyTime = systemTime(); 2314 2315 // MIXER 2316 nsecs_t lastWarning = 0; 2317 2318 // DUPLICATING 2319 // FIXME could this be made local to while loop? 2320 writeFrames = 0; 2321 2322 int lastGeneration = 0; 2323 2324 cacheParameters_l(); 2325 sleepTime = idleSleepTime; 2326 2327 if (mType == MIXER) { 2328 sleepTimeShift = 0; 2329 } 2330 2331 CpuStats cpuStats; 2332 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2333 2334 acquireWakeLock(); 2335 2336 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2337 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2338 // and then that string will be logged at the next convenient opportunity. 2339 const char *logString = NULL; 2340 2341 checkSilentMode_l(); 2342 2343 while (!exitPending()) 2344 { 2345 cpuStats.sample(myName); 2346 2347 Vector< sp<EffectChain> > effectChains; 2348 2349 { // scope for mLock 2350 2351 Mutex::Autolock _l(mLock); 2352 2353 processConfigEvents_l(); 2354 2355 if (logString != NULL) { 2356 mNBLogWriter->logTimestamp(); 2357 mNBLogWriter->log(logString); 2358 logString = NULL; 2359 } 2360 2361 if (mLatchDValid) { 2362 mLatchQ = mLatchD; 2363 mLatchDValid = false; 2364 mLatchQValid = true; 2365 } 2366 2367 saveOutputTracks(); 2368 if (mSignalPending) { 2369 // A signal was raised while we were unlocked 2370 mSignalPending = false; 2371 } else if (waitingAsyncCallback_l()) { 2372 if (exitPending()) { 2373 break; 2374 } 2375 releaseWakeLock_l(); 2376 mWakeLockUids.clear(); 2377 mActiveTracksGeneration++; 2378 ALOGV("wait async completion"); 2379 mWaitWorkCV.wait(mLock); 2380 ALOGV("async completion/wake"); 2381 acquireWakeLock_l(); 2382 standbyTime = systemTime() + standbyDelay; 2383 sleepTime = 0; 2384 2385 continue; 2386 } 2387 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2388 isSuspended()) { 2389 // put audio hardware into standby after short delay 2390 if (shouldStandby_l()) { 2391 2392 threadLoop_standby(); 2393 2394 mStandby = true; 2395 } 2396 2397 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2398 // we're about to wait, flush the binder command buffer 2399 IPCThreadState::self()->flushCommands(); 2400 2401 clearOutputTracks(); 2402 2403 if (exitPending()) { 2404 break; 2405 } 2406 2407 releaseWakeLock_l(); 2408 mWakeLockUids.clear(); 2409 mActiveTracksGeneration++; 2410 // wait until we have something to do... 2411 ALOGV("%s going to sleep", myName.string()); 2412 mWaitWorkCV.wait(mLock); 2413 ALOGV("%s waking up", myName.string()); 2414 acquireWakeLock_l(); 2415 2416 mMixerStatus = MIXER_IDLE; 2417 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2418 mBytesWritten = 0; 2419 mBytesRemaining = 0; 2420 checkSilentMode_l(); 2421 2422 standbyTime = systemTime() + standbyDelay; 2423 sleepTime = idleSleepTime; 2424 if (mType == MIXER) { 2425 sleepTimeShift = 0; 2426 } 2427 2428 continue; 2429 } 2430 } 2431 // mMixerStatusIgnoringFastTracks is also updated internally 2432 mMixerStatus = prepareTracks_l(&tracksToRemove); 2433 2434 // compare with previously applied list 2435 if (lastGeneration != mActiveTracksGeneration) { 2436 // update wakelock 2437 updateWakeLockUids_l(mWakeLockUids); 2438 lastGeneration = mActiveTracksGeneration; 2439 } 2440 2441 // prevent any changes in effect chain list and in each effect chain 2442 // during mixing and effect process as the audio buffers could be deleted 2443 // or modified if an effect is created or deleted 2444 lockEffectChains_l(effectChains); 2445 } // mLock scope ends 2446 2447 if (mBytesRemaining == 0) { 2448 mCurrentWriteLength = 0; 2449 if (mMixerStatus == MIXER_TRACKS_READY) { 2450 // threadLoop_mix() sets mCurrentWriteLength 2451 threadLoop_mix(); 2452 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2453 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2454 // threadLoop_sleepTime sets sleepTime to 0 if data 2455 // must be written to HAL 2456 threadLoop_sleepTime(); 2457 if (sleepTime == 0) { 2458 mCurrentWriteLength = mSinkBufferSize; 2459 } 2460 } 2461 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2462 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2463 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2464 // or mSinkBuffer (if there are no effects). 2465 // 2466 // This is done pre-effects computation; if effects change to 2467 // support higher precision, this needs to move. 2468 // 2469 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2470 // TODO use sleepTime == 0 as an additional condition. 2471 if (mMixerBufferValid) { 2472 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2473 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2474 2475 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2476 mNormalFrameCount * mChannelCount); 2477 } 2478 2479 mBytesRemaining = mCurrentWriteLength; 2480 if (isSuspended()) { 2481 sleepTime = suspendSleepTimeUs(); 2482 // simulate write to HAL when suspended 2483 mBytesWritten += mSinkBufferSize; 2484 mBytesRemaining = 0; 2485 } 2486 2487 // only process effects if we're going to write 2488 if (sleepTime == 0 && mType != OFFLOAD) { 2489 for (size_t i = 0; i < effectChains.size(); i ++) { 2490 effectChains[i]->process_l(); 2491 } 2492 } 2493 } 2494 // Process effect chains for offloaded thread even if no audio 2495 // was read from audio track: process only updates effect state 2496 // and thus does have to be synchronized with audio writes but may have 2497 // to be called while waiting for async write callback 2498 if (mType == OFFLOAD) { 2499 for (size_t i = 0; i < effectChains.size(); i ++) { 2500 effectChains[i]->process_l(); 2501 } 2502 } 2503 2504 // Only if the Effects buffer is enabled and there is data in the 2505 // Effects buffer (buffer valid), we need to 2506 // copy into the sink buffer. 2507 // TODO use sleepTime == 0 as an additional condition. 2508 if (mEffectBufferValid) { 2509 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2510 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2511 mNormalFrameCount * mChannelCount); 2512 } 2513 2514 // enable changes in effect chain 2515 unlockEffectChains(effectChains); 2516 2517 if (!waitingAsyncCallback()) { 2518 // sleepTime == 0 means we must write to audio hardware 2519 if (sleepTime == 0) { 2520 if (mBytesRemaining) { 2521 ssize_t ret = threadLoop_write(); 2522 if (ret < 0) { 2523 mBytesRemaining = 0; 2524 } else { 2525 mBytesWritten += ret; 2526 mBytesRemaining -= ret; 2527 } 2528 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2529 (mMixerStatus == MIXER_DRAIN_ALL)) { 2530 threadLoop_drain(); 2531 } 2532 if (mType == MIXER) { 2533 // write blocked detection 2534 nsecs_t now = systemTime(); 2535 nsecs_t delta = now - mLastWriteTime; 2536 if (!mStandby && delta > maxPeriod) { 2537 mNumDelayedWrites++; 2538 if ((now - lastWarning) > kWarningThrottleNs) { 2539 ATRACE_NAME("underrun"); 2540 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2541 ns2ms(delta), mNumDelayedWrites, this); 2542 lastWarning = now; 2543 } 2544 } 2545 } 2546 2547 } else { 2548 usleep(sleepTime); 2549 } 2550 } 2551 2552 // Finally let go of removed track(s), without the lock held 2553 // since we can't guarantee the destructors won't acquire that 2554 // same lock. This will also mutate and push a new fast mixer state. 2555 threadLoop_removeTracks(tracksToRemove); 2556 tracksToRemove.clear(); 2557 2558 // FIXME I don't understand the need for this here; 2559 // it was in the original code but maybe the 2560 // assignment in saveOutputTracks() makes this unnecessary? 2561 clearOutputTracks(); 2562 2563 // Effect chains will be actually deleted here if they were removed from 2564 // mEffectChains list during mixing or effects processing 2565 effectChains.clear(); 2566 2567 // FIXME Note that the above .clear() is no longer necessary since effectChains 2568 // is now local to this block, but will keep it for now (at least until merge done). 2569 } 2570 2571 threadLoop_exit(); 2572 2573 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2574 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2575 // put output stream into standby mode 2576 if (!mStandby) { 2577 mOutput->stream->common.standby(&mOutput->stream->common); 2578 } 2579 } 2580 2581 releaseWakeLock(); 2582 mWakeLockUids.clear(); 2583 mActiveTracksGeneration++; 2584 2585 ALOGV("Thread %p type %d exiting", this, mType); 2586 return false; 2587} 2588 2589// removeTracks_l() must be called with ThreadBase::mLock held 2590void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2591{ 2592 size_t count = tracksToRemove.size(); 2593 if (count > 0) { 2594 for (size_t i=0 ; i<count ; i++) { 2595 const sp<Track>& track = tracksToRemove.itemAt(i); 2596 mActiveTracks.remove(track); 2597 mWakeLockUids.remove(track->uid()); 2598 mActiveTracksGeneration++; 2599 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2600 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2601 if (chain != 0) { 2602 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2603 track->sessionId()); 2604 chain->decActiveTrackCnt(); 2605 } 2606 if (track->isTerminated()) { 2607 removeTrack_l(track); 2608 } 2609 } 2610 } 2611 2612} 2613 2614status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2615{ 2616 if (mNormalSink != 0) { 2617 return mNormalSink->getTimestamp(timestamp); 2618 } 2619 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2620 uint64_t position64; 2621 int ret = mOutput->stream->get_presentation_position( 2622 mOutput->stream, &position64, ×tamp.mTime); 2623 if (ret == 0) { 2624 timestamp.mPosition = (uint32_t)position64; 2625 return NO_ERROR; 2626 } 2627 } 2628 return INVALID_OPERATION; 2629} 2630 2631status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2632 audio_patch_handle_t *handle) 2633{ 2634 status_t status = NO_ERROR; 2635 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2636 // store new device and send to effects 2637 audio_devices_t type = AUDIO_DEVICE_NONE; 2638 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2639 type |= patch->sinks[i].ext.device.type; 2640 } 2641 mOutDevice = type; 2642 for (size_t i = 0; i < mEffectChains.size(); i++) { 2643 mEffectChains[i]->setDevice_l(mOutDevice); 2644 } 2645 2646 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2647 status = hwDevice->create_audio_patch(hwDevice, 2648 patch->num_sources, 2649 patch->sources, 2650 patch->num_sinks, 2651 patch->sinks, 2652 handle); 2653 } else { 2654 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2655 } 2656 return status; 2657} 2658 2659status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2660{ 2661 status_t status = NO_ERROR; 2662 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2663 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2664 status = hwDevice->release_audio_patch(hwDevice, handle); 2665 } else { 2666 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2667 } 2668 return status; 2669} 2670 2671// ---------------------------------------------------------------------------- 2672 2673AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2674 audio_io_handle_t id, audio_devices_t device, type_t type) 2675 : PlaybackThread(audioFlinger, output, id, device, type), 2676 // mAudioMixer below 2677 // mFastMixer below 2678 mFastMixerFutex(0) 2679 // mOutputSink below 2680 // mPipeSink below 2681 // mNormalSink below 2682{ 2683 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2684 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2685 "mFrameCount=%d, mNormalFrameCount=%d", 2686 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2687 mNormalFrameCount); 2688 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2689 2690 // FIXME - Current mixer implementation only supports stereo output 2691 if (mChannelCount != FCC_2) { 2692 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2693 } 2694 2695 // create an NBAIO sink for the HAL output stream, and negotiate 2696 mOutputSink = new AudioStreamOutSink(output->stream); 2697 size_t numCounterOffers = 0; 2698 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2699 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2700 ALOG_ASSERT(index == 0); 2701 2702 // initialize fast mixer depending on configuration 2703 bool initFastMixer; 2704 switch (kUseFastMixer) { 2705 case FastMixer_Never: 2706 initFastMixer = false; 2707 break; 2708 case FastMixer_Always: 2709 initFastMixer = true; 2710 break; 2711 case FastMixer_Static: 2712 case FastMixer_Dynamic: 2713 initFastMixer = mFrameCount < mNormalFrameCount; 2714 break; 2715 } 2716 if (initFastMixer) { 2717 2718 // create a MonoPipe to connect our submix to FastMixer 2719 NBAIO_Format format = mOutputSink->format(); 2720 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2721 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2722 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2723 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2724 const NBAIO_Format offers[1] = {format}; 2725 size_t numCounterOffers = 0; 2726 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2727 ALOG_ASSERT(index == 0); 2728 monoPipe->setAvgFrames((mScreenState & 1) ? 2729 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2730 mPipeSink = monoPipe; 2731 2732#ifdef TEE_SINK 2733 if (mTeeSinkOutputEnabled) { 2734 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2735 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2736 numCounterOffers = 0; 2737 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2738 ALOG_ASSERT(index == 0); 2739 mTeeSink = teeSink; 2740 PipeReader *teeSource = new PipeReader(*teeSink); 2741 numCounterOffers = 0; 2742 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2743 ALOG_ASSERT(index == 0); 2744 mTeeSource = teeSource; 2745 } 2746#endif 2747 2748 // create fast mixer and configure it initially with just one fast track for our submix 2749 mFastMixer = new FastMixer(); 2750 FastMixerStateQueue *sq = mFastMixer->sq(); 2751#ifdef STATE_QUEUE_DUMP 2752 sq->setObserverDump(&mStateQueueObserverDump); 2753 sq->setMutatorDump(&mStateQueueMutatorDump); 2754#endif 2755 FastMixerState *state = sq->begin(); 2756 FastTrack *fastTrack = &state->mFastTracks[0]; 2757 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2758 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2759 fastTrack->mVolumeProvider = NULL; 2760 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2761 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2762 fastTrack->mGeneration++; 2763 state->mFastTracksGen++; 2764 state->mTrackMask = 1; 2765 // fast mixer will use the HAL output sink 2766 state->mOutputSink = mOutputSink.get(); 2767 state->mOutputSinkGen++; 2768 state->mFrameCount = mFrameCount; 2769 state->mCommand = FastMixerState::COLD_IDLE; 2770 // already done in constructor initialization list 2771 //mFastMixerFutex = 0; 2772 state->mColdFutexAddr = &mFastMixerFutex; 2773 state->mColdGen++; 2774 state->mDumpState = &mFastMixerDumpState; 2775#ifdef TEE_SINK 2776 state->mTeeSink = mTeeSink.get(); 2777#endif 2778 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2779 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2780 sq->end(); 2781 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2782 2783 // start the fast mixer 2784 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2785 pid_t tid = mFastMixer->getTid(); 2786 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2787 if (err != 0) { 2788 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2789 kPriorityFastMixer, getpid_cached, tid, err); 2790 } 2791 2792#ifdef AUDIO_WATCHDOG 2793 // create and start the watchdog 2794 mAudioWatchdog = new AudioWatchdog(); 2795 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2796 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2797 tid = mAudioWatchdog->getTid(); 2798 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2799 if (err != 0) { 2800 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2801 kPriorityFastMixer, getpid_cached, tid, err); 2802 } 2803#endif 2804 2805 } else { 2806 mFastMixer = NULL; 2807 } 2808 2809 switch (kUseFastMixer) { 2810 case FastMixer_Never: 2811 case FastMixer_Dynamic: 2812 mNormalSink = mOutputSink; 2813 break; 2814 case FastMixer_Always: 2815 mNormalSink = mPipeSink; 2816 break; 2817 case FastMixer_Static: 2818 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2819 break; 2820 } 2821} 2822 2823AudioFlinger::MixerThread::~MixerThread() 2824{ 2825 if (mFastMixer != NULL) { 2826 FastMixerStateQueue *sq = mFastMixer->sq(); 2827 FastMixerState *state = sq->begin(); 2828 if (state->mCommand == FastMixerState::COLD_IDLE) { 2829 int32_t old = android_atomic_inc(&mFastMixerFutex); 2830 if (old == -1) { 2831 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2832 } 2833 } 2834 state->mCommand = FastMixerState::EXIT; 2835 sq->end(); 2836 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2837 mFastMixer->join(); 2838 // Though the fast mixer thread has exited, it's state queue is still valid. 2839 // We'll use that extract the final state which contains one remaining fast track 2840 // corresponding to our sub-mix. 2841 state = sq->begin(); 2842 ALOG_ASSERT(state->mTrackMask == 1); 2843 FastTrack *fastTrack = &state->mFastTracks[0]; 2844 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2845 delete fastTrack->mBufferProvider; 2846 sq->end(false /*didModify*/); 2847 delete mFastMixer; 2848#ifdef AUDIO_WATCHDOG 2849 if (mAudioWatchdog != 0) { 2850 mAudioWatchdog->requestExit(); 2851 mAudioWatchdog->requestExitAndWait(); 2852 mAudioWatchdog.clear(); 2853 } 2854#endif 2855 } 2856 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2857 delete mAudioMixer; 2858} 2859 2860 2861uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2862{ 2863 if (mFastMixer != NULL) { 2864 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2865 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2866 } 2867 return latency; 2868} 2869 2870 2871void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2872{ 2873 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2874} 2875 2876ssize_t AudioFlinger::MixerThread::threadLoop_write() 2877{ 2878 // FIXME we should only do one push per cycle; confirm this is true 2879 // Start the fast mixer if it's not already running 2880 if (mFastMixer != NULL) { 2881 FastMixerStateQueue *sq = mFastMixer->sq(); 2882 FastMixerState *state = sq->begin(); 2883 if (state->mCommand != FastMixerState::MIX_WRITE && 2884 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2885 if (state->mCommand == FastMixerState::COLD_IDLE) { 2886 int32_t old = android_atomic_inc(&mFastMixerFutex); 2887 if (old == -1) { 2888 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2889 } 2890#ifdef AUDIO_WATCHDOG 2891 if (mAudioWatchdog != 0) { 2892 mAudioWatchdog->resume(); 2893 } 2894#endif 2895 } 2896 state->mCommand = FastMixerState::MIX_WRITE; 2897 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2898 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2899 sq->end(); 2900 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2901 if (kUseFastMixer == FastMixer_Dynamic) { 2902 mNormalSink = mPipeSink; 2903 } 2904 } else { 2905 sq->end(false /*didModify*/); 2906 } 2907 } 2908 return PlaybackThread::threadLoop_write(); 2909} 2910 2911void AudioFlinger::MixerThread::threadLoop_standby() 2912{ 2913 // Idle the fast mixer if it's currently running 2914 if (mFastMixer != NULL) { 2915 FastMixerStateQueue *sq = mFastMixer->sq(); 2916 FastMixerState *state = sq->begin(); 2917 if (!(state->mCommand & FastMixerState::IDLE)) { 2918 state->mCommand = FastMixerState::COLD_IDLE; 2919 state->mColdFutexAddr = &mFastMixerFutex; 2920 state->mColdGen++; 2921 mFastMixerFutex = 0; 2922 sq->end(); 2923 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2924 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2925 if (kUseFastMixer == FastMixer_Dynamic) { 2926 mNormalSink = mOutputSink; 2927 } 2928#ifdef AUDIO_WATCHDOG 2929 if (mAudioWatchdog != 0) { 2930 mAudioWatchdog->pause(); 2931 } 2932#endif 2933 } else { 2934 sq->end(false /*didModify*/); 2935 } 2936 } 2937 PlaybackThread::threadLoop_standby(); 2938} 2939 2940bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2941{ 2942 return false; 2943} 2944 2945bool AudioFlinger::PlaybackThread::shouldStandby_l() 2946{ 2947 return !mStandby; 2948} 2949 2950bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2951{ 2952 Mutex::Autolock _l(mLock); 2953 return waitingAsyncCallback_l(); 2954} 2955 2956// shared by MIXER and DIRECT, overridden by DUPLICATING 2957void AudioFlinger::PlaybackThread::threadLoop_standby() 2958{ 2959 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2960 mOutput->stream->common.standby(&mOutput->stream->common); 2961 if (mUseAsyncWrite != 0) { 2962 // discard any pending drain or write ack by incrementing sequence 2963 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 2964 mDrainSequence = (mDrainSequence + 2) & ~1; 2965 ALOG_ASSERT(mCallbackThread != 0); 2966 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2967 mCallbackThread->setDraining(mDrainSequence); 2968 } 2969} 2970 2971void AudioFlinger::PlaybackThread::onAddNewTrack_l() 2972{ 2973 ALOGV("signal playback thread"); 2974 broadcast_l(); 2975} 2976 2977void AudioFlinger::MixerThread::threadLoop_mix() 2978{ 2979 // obtain the presentation timestamp of the next output buffer 2980 int64_t pts; 2981 status_t status = INVALID_OPERATION; 2982 2983 if (mNormalSink != 0) { 2984 status = mNormalSink->getNextWriteTimestamp(&pts); 2985 } else { 2986 status = mOutputSink->getNextWriteTimestamp(&pts); 2987 } 2988 2989 if (status != NO_ERROR) { 2990 pts = AudioBufferProvider::kInvalidPTS; 2991 } 2992 2993 // mix buffers... 2994 mAudioMixer->process(pts); 2995 mCurrentWriteLength = mSinkBufferSize; 2996 // increase sleep time progressively when application underrun condition clears. 2997 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2998 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2999 // such that we would underrun the audio HAL. 3000 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3001 sleepTimeShift--; 3002 } 3003 sleepTime = 0; 3004 standbyTime = systemTime() + standbyDelay; 3005 //TODO: delay standby when effects have a tail 3006} 3007 3008void AudioFlinger::MixerThread::threadLoop_sleepTime() 3009{ 3010 // If no tracks are ready, sleep once for the duration of an output 3011 // buffer size, then write 0s to the output 3012 if (sleepTime == 0) { 3013 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3014 sleepTime = activeSleepTime >> sleepTimeShift; 3015 if (sleepTime < kMinThreadSleepTimeUs) { 3016 sleepTime = kMinThreadSleepTimeUs; 3017 } 3018 // reduce sleep time in case of consecutive application underruns to avoid 3019 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3020 // duration we would end up writing less data than needed by the audio HAL if 3021 // the condition persists. 3022 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3023 sleepTimeShift++; 3024 } 3025 } else { 3026 sleepTime = idleSleepTime; 3027 } 3028 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3029 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3030 // before effects processing or output. 3031 if (mMixerBufferValid) { 3032 memset(mMixerBuffer, 0, mMixerBufferSize); 3033 } else { 3034 memset(mSinkBuffer, 0, mSinkBufferSize); 3035 } 3036 sleepTime = 0; 3037 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3038 "anticipated start"); 3039 } 3040 // TODO add standby time extension fct of effect tail 3041} 3042 3043// prepareTracks_l() must be called with ThreadBase::mLock held 3044AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3045 Vector< sp<Track> > *tracksToRemove) 3046{ 3047 3048 mixer_state mixerStatus = MIXER_IDLE; 3049 // find out which tracks need to be processed 3050 size_t count = mActiveTracks.size(); 3051 size_t mixedTracks = 0; 3052 size_t tracksWithEffect = 0; 3053 // counts only _active_ fast tracks 3054 size_t fastTracks = 0; 3055 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3056 3057 float masterVolume = mMasterVolume; 3058 bool masterMute = mMasterMute; 3059 3060 if (masterMute) { 3061 masterVolume = 0; 3062 } 3063 // Delegate master volume control to effect in output mix effect chain if needed 3064 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3065 if (chain != 0) { 3066 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3067 chain->setVolume_l(&v, &v); 3068 masterVolume = (float)((v + (1 << 23)) >> 24); 3069 chain.clear(); 3070 } 3071 3072 // prepare a new state to push 3073 FastMixerStateQueue *sq = NULL; 3074 FastMixerState *state = NULL; 3075 bool didModify = false; 3076 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3077 if (mFastMixer != NULL) { 3078 sq = mFastMixer->sq(); 3079 state = sq->begin(); 3080 } 3081 3082 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3083 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3084 3085 for (size_t i=0 ; i<count ; i++) { 3086 const sp<Track> t = mActiveTracks[i].promote(); 3087 if (t == 0) { 3088 continue; 3089 } 3090 3091 // this const just means the local variable doesn't change 3092 Track* const track = t.get(); 3093 3094 // process fast tracks 3095 if (track->isFastTrack()) { 3096 3097 // It's theoretically possible (though unlikely) for a fast track to be created 3098 // and then removed within the same normal mix cycle. This is not a problem, as 3099 // the track never becomes active so it's fast mixer slot is never touched. 3100 // The converse, of removing an (active) track and then creating a new track 3101 // at the identical fast mixer slot within the same normal mix cycle, 3102 // is impossible because the slot isn't marked available until the end of each cycle. 3103 int j = track->mFastIndex; 3104 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3105 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3106 FastTrack *fastTrack = &state->mFastTracks[j]; 3107 3108 // Determine whether the track is currently in underrun condition, 3109 // and whether it had a recent underrun. 3110 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3111 FastTrackUnderruns underruns = ftDump->mUnderruns; 3112 uint32_t recentFull = (underruns.mBitFields.mFull - 3113 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3114 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3115 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3116 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3117 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3118 uint32_t recentUnderruns = recentPartial + recentEmpty; 3119 track->mObservedUnderruns = underruns; 3120 // don't count underruns that occur while stopping or pausing 3121 // or stopped which can occur when flush() is called while active 3122 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3123 recentUnderruns > 0) { 3124 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3125 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3126 } 3127 3128 // This is similar to the state machine for normal tracks, 3129 // with a few modifications for fast tracks. 3130 bool isActive = true; 3131 switch (track->mState) { 3132 case TrackBase::STOPPING_1: 3133 // track stays active in STOPPING_1 state until first underrun 3134 if (recentUnderruns > 0 || track->isTerminated()) { 3135 track->mState = TrackBase::STOPPING_2; 3136 } 3137 break; 3138 case TrackBase::PAUSING: 3139 // ramp down is not yet implemented 3140 track->setPaused(); 3141 break; 3142 case TrackBase::RESUMING: 3143 // ramp up is not yet implemented 3144 track->mState = TrackBase::ACTIVE; 3145 break; 3146 case TrackBase::ACTIVE: 3147 if (recentFull > 0 || recentPartial > 0) { 3148 // track has provided at least some frames recently: reset retry count 3149 track->mRetryCount = kMaxTrackRetries; 3150 } 3151 if (recentUnderruns == 0) { 3152 // no recent underruns: stay active 3153 break; 3154 } 3155 // there has recently been an underrun of some kind 3156 if (track->sharedBuffer() == 0) { 3157 // were any of the recent underruns "empty" (no frames available)? 3158 if (recentEmpty == 0) { 3159 // no, then ignore the partial underruns as they are allowed indefinitely 3160 break; 3161 } 3162 // there has recently been an "empty" underrun: decrement the retry counter 3163 if (--(track->mRetryCount) > 0) { 3164 break; 3165 } 3166 // indicate to client process that the track was disabled because of underrun; 3167 // it will then automatically call start() when data is available 3168 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3169 // remove from active list, but state remains ACTIVE [confusing but true] 3170 isActive = false; 3171 break; 3172 } 3173 // fall through 3174 case TrackBase::STOPPING_2: 3175 case TrackBase::PAUSED: 3176 case TrackBase::STOPPED: 3177 case TrackBase::FLUSHED: // flush() while active 3178 // Check for presentation complete if track is inactive 3179 // We have consumed all the buffers of this track. 3180 // This would be incomplete if we auto-paused on underrun 3181 { 3182 size_t audioHALFrames = 3183 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3184 size_t framesWritten = mBytesWritten / mFrameSize; 3185 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3186 // track stays in active list until presentation is complete 3187 break; 3188 } 3189 } 3190 if (track->isStopping_2()) { 3191 track->mState = TrackBase::STOPPED; 3192 } 3193 if (track->isStopped()) { 3194 // Can't reset directly, as fast mixer is still polling this track 3195 // track->reset(); 3196 // So instead mark this track as needing to be reset after push with ack 3197 resetMask |= 1 << i; 3198 } 3199 isActive = false; 3200 break; 3201 case TrackBase::IDLE: 3202 default: 3203 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3204 } 3205 3206 if (isActive) { 3207 // was it previously inactive? 3208 if (!(state->mTrackMask & (1 << j))) { 3209 ExtendedAudioBufferProvider *eabp = track; 3210 VolumeProvider *vp = track; 3211 fastTrack->mBufferProvider = eabp; 3212 fastTrack->mVolumeProvider = vp; 3213 fastTrack->mChannelMask = track->mChannelMask; 3214 fastTrack->mFormat = track->mFormat; 3215 fastTrack->mGeneration++; 3216 state->mTrackMask |= 1 << j; 3217 didModify = true; 3218 // no acknowledgement required for newly active tracks 3219 } 3220 // cache the combined master volume and stream type volume for fast mixer; this 3221 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3222 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3223 ++fastTracks; 3224 } else { 3225 // was it previously active? 3226 if (state->mTrackMask & (1 << j)) { 3227 fastTrack->mBufferProvider = NULL; 3228 fastTrack->mGeneration++; 3229 state->mTrackMask &= ~(1 << j); 3230 didModify = true; 3231 // If any fast tracks were removed, we must wait for acknowledgement 3232 // because we're about to decrement the last sp<> on those tracks. 3233 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3234 } else { 3235 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3236 } 3237 tracksToRemove->add(track); 3238 // Avoids a misleading display in dumpsys 3239 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3240 } 3241 continue; 3242 } 3243 3244 { // local variable scope to avoid goto warning 3245 3246 audio_track_cblk_t* cblk = track->cblk(); 3247 3248 // The first time a track is added we wait 3249 // for all its buffers to be filled before processing it 3250 int name = track->name(); 3251 // make sure that we have enough frames to mix one full buffer. 3252 // enforce this condition only once to enable draining the buffer in case the client 3253 // app does not call stop() and relies on underrun to stop: 3254 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3255 // during last round 3256 size_t desiredFrames; 3257 uint32_t sr = track->sampleRate(); 3258 if (sr == mSampleRate) { 3259 desiredFrames = mNormalFrameCount; 3260 } else { 3261 // +1 for rounding and +1 for additional sample needed for interpolation 3262 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3263 // add frames already consumed but not yet released by the resampler 3264 // because mAudioTrackServerProxy->framesReady() will include these frames 3265 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3266#if 0 3267 // the minimum track buffer size is normally twice the number of frames necessary 3268 // to fill one buffer and the resampler should not leave more than one buffer worth 3269 // of unreleased frames after each pass, but just in case... 3270 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3271#endif 3272 } 3273 uint32_t minFrames = 1; 3274 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3275 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3276 minFrames = desiredFrames; 3277 } 3278 3279 size_t framesReady = track->framesReady(); 3280 if ((framesReady >= minFrames) && track->isReady() && 3281 !track->isPaused() && !track->isTerminated()) 3282 { 3283 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3284 3285 mixedTracks++; 3286 3287 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3288 // there is an effect chain connected to the track 3289 chain.clear(); 3290 if (track->mainBuffer() != mSinkBuffer && 3291 track->mainBuffer() != mMixerBuffer) { 3292 if (mEffectBufferEnabled) { 3293 mEffectBufferValid = true; // Later can set directly. 3294 } 3295 chain = getEffectChain_l(track->sessionId()); 3296 // Delegate volume control to effect in track effect chain if needed 3297 if (chain != 0) { 3298 tracksWithEffect++; 3299 } else { 3300 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3301 "session %d", 3302 name, track->sessionId()); 3303 } 3304 } 3305 3306 3307 int param = AudioMixer::VOLUME; 3308 if (track->mFillingUpStatus == Track::FS_FILLED) { 3309 // no ramp for the first volume setting 3310 track->mFillingUpStatus = Track::FS_ACTIVE; 3311 if (track->mState == TrackBase::RESUMING) { 3312 track->mState = TrackBase::ACTIVE; 3313 param = AudioMixer::RAMP_VOLUME; 3314 } 3315 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3316 // FIXME should not make a decision based on mServer 3317 } else if (cblk->mServer != 0) { 3318 // If the track is stopped before the first frame was mixed, 3319 // do not apply ramp 3320 param = AudioMixer::RAMP_VOLUME; 3321 } 3322 3323 // compute volume for this track 3324 uint32_t vl, vr, va; 3325 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3326 vl = vr = va = 0; 3327 if (track->isPausing()) { 3328 track->setPaused(); 3329 } 3330 } else { 3331 3332 // read original volumes with volume control 3333 float typeVolume = mStreamTypes[track->streamType()].volume; 3334 float v = masterVolume * typeVolume; 3335 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3336 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3337 float vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3338 float vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3339 // track volumes come from shared memory, so can't be trusted and must be clamped 3340 if (vlf > GAIN_FLOAT_UNITY) { 3341 ALOGV("Track left volume out of range: %.3g", vlf); 3342 vlf = GAIN_FLOAT_UNITY; 3343 } 3344 if (vrf > GAIN_FLOAT_UNITY) { 3345 ALOGV("Track right volume out of range: %.3g", vrf); 3346 vrf = GAIN_FLOAT_UNITY; 3347 } 3348 // now apply the master volume and stream type volume 3349 // FIXME we're losing the wonderful dynamic range in the minifloat representation 3350 float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT); 3351 vl = (uint32_t) (v8_24 * vlf); 3352 vr = (uint32_t) (v8_24 * vrf); 3353 // assuming master volume and stream type volume each go up to 1.0, 3354 // vl and vr are now in 8.24 format 3355 3356 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3357 // send level comes from shared memory and so may be corrupt 3358 if (sendLevel > MAX_GAIN_INT) { 3359 ALOGV("Track send level out of range: %04X", sendLevel); 3360 sendLevel = MAX_GAIN_INT; 3361 } 3362 va = (uint32_t)(v * sendLevel); 3363 } 3364 3365 // Delegate volume control to effect in track effect chain if needed 3366 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3367 // Do not ramp volume if volume is controlled by effect 3368 param = AudioMixer::VOLUME; 3369 track->mHasVolumeController = true; 3370 } else { 3371 // force no volume ramp when volume controller was just disabled or removed 3372 // from effect chain to avoid volume spike 3373 if (track->mHasVolumeController) { 3374 param = AudioMixer::VOLUME; 3375 } 3376 track->mHasVolumeController = false; 3377 } 3378 3379 // FIXME Use float 3380 // Convert volumes from 8.24 to 4.12 format 3381 // This additional clamping is needed in case chain->setVolume_l() overshot 3382 vl = (vl + (1 << 11)) >> 12; 3383 if (vl > MAX_GAIN_INT) { 3384 vl = MAX_GAIN_INT; 3385 } 3386 vr = (vr + (1 << 11)) >> 12; 3387 if (vr > MAX_GAIN_INT) { 3388 vr = MAX_GAIN_INT; 3389 } 3390 3391 if (va > MAX_GAIN_INT) { 3392 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3393 } 3394 3395 // XXX: these things DON'T need to be done each time 3396 mAudioMixer->setBufferProvider(name, track); 3397 mAudioMixer->enable(name); 3398 3399 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3400 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3401 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3402 mAudioMixer->setParameter( 3403 name, 3404 AudioMixer::TRACK, 3405 AudioMixer::FORMAT, (void *)track->format()); 3406 mAudioMixer->setParameter( 3407 name, 3408 AudioMixer::TRACK, 3409 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3410 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3411 uint32_t maxSampleRate = mSampleRate * 2; 3412 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3413 if (reqSampleRate == 0) { 3414 reqSampleRate = mSampleRate; 3415 } else if (reqSampleRate > maxSampleRate) { 3416 reqSampleRate = maxSampleRate; 3417 } 3418 mAudioMixer->setParameter( 3419 name, 3420 AudioMixer::RESAMPLE, 3421 AudioMixer::SAMPLE_RATE, 3422 (void *)(uintptr_t)reqSampleRate); 3423 /* 3424 * Select the appropriate output buffer for the track. 3425 * 3426 * Tracks with effects go into their own effects chain buffer 3427 * and from there into either mEffectBuffer or mSinkBuffer. 3428 * 3429 * Other tracks can use mMixerBuffer for higher precision 3430 * channel accumulation. If this buffer is enabled 3431 * (mMixerBufferEnabled true), then selected tracks will accumulate 3432 * into it. 3433 * 3434 */ 3435 if (mMixerBufferEnabled 3436 && (track->mainBuffer() == mSinkBuffer 3437 || track->mainBuffer() == mMixerBuffer)) { 3438 mAudioMixer->setParameter( 3439 name, 3440 AudioMixer::TRACK, 3441 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3442 mAudioMixer->setParameter( 3443 name, 3444 AudioMixer::TRACK, 3445 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3446 // TODO: override track->mainBuffer()? 3447 mMixerBufferValid = true; 3448 } else { 3449 mAudioMixer->setParameter( 3450 name, 3451 AudioMixer::TRACK, 3452 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3453 mAudioMixer->setParameter( 3454 name, 3455 AudioMixer::TRACK, 3456 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3457 } 3458 mAudioMixer->setParameter( 3459 name, 3460 AudioMixer::TRACK, 3461 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3462 3463 // reset retry count 3464 track->mRetryCount = kMaxTrackRetries; 3465 3466 // If one track is ready, set the mixer ready if: 3467 // - the mixer was not ready during previous round OR 3468 // - no other track is not ready 3469 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3470 mixerStatus != MIXER_TRACKS_ENABLED) { 3471 mixerStatus = MIXER_TRACKS_READY; 3472 } 3473 } else { 3474 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3475 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3476 } 3477 // clear effect chain input buffer if an active track underruns to avoid sending 3478 // previous audio buffer again to effects 3479 chain = getEffectChain_l(track->sessionId()); 3480 if (chain != 0) { 3481 chain->clearInputBuffer(); 3482 } 3483 3484 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3485 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3486 track->isStopped() || track->isPaused()) { 3487 // We have consumed all the buffers of this track. 3488 // Remove it from the list of active tracks. 3489 // TODO: use actual buffer filling status instead of latency when available from 3490 // audio HAL 3491 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3492 size_t framesWritten = mBytesWritten / mFrameSize; 3493 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3494 if (track->isStopped()) { 3495 track->reset(); 3496 } 3497 tracksToRemove->add(track); 3498 } 3499 } else { 3500 // No buffers for this track. Give it a few chances to 3501 // fill a buffer, then remove it from active list. 3502 if (--(track->mRetryCount) <= 0) { 3503 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3504 tracksToRemove->add(track); 3505 // indicate to client process that the track was disabled because of underrun; 3506 // it will then automatically call start() when data is available 3507 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3508 // If one track is not ready, mark the mixer also not ready if: 3509 // - the mixer was ready during previous round OR 3510 // - no other track is ready 3511 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3512 mixerStatus != MIXER_TRACKS_READY) { 3513 mixerStatus = MIXER_TRACKS_ENABLED; 3514 } 3515 } 3516 mAudioMixer->disable(name); 3517 } 3518 3519 } // local variable scope to avoid goto warning 3520track_is_ready: ; 3521 3522 } 3523 3524 // Push the new FastMixer state if necessary 3525 bool pauseAudioWatchdog = false; 3526 if (didModify) { 3527 state->mFastTracksGen++; 3528 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3529 if (kUseFastMixer == FastMixer_Dynamic && 3530 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3531 state->mCommand = FastMixerState::COLD_IDLE; 3532 state->mColdFutexAddr = &mFastMixerFutex; 3533 state->mColdGen++; 3534 mFastMixerFutex = 0; 3535 if (kUseFastMixer == FastMixer_Dynamic) { 3536 mNormalSink = mOutputSink; 3537 } 3538 // If we go into cold idle, need to wait for acknowledgement 3539 // so that fast mixer stops doing I/O. 3540 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3541 pauseAudioWatchdog = true; 3542 } 3543 } 3544 if (sq != NULL) { 3545 sq->end(didModify); 3546 sq->push(block); 3547 } 3548#ifdef AUDIO_WATCHDOG 3549 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3550 mAudioWatchdog->pause(); 3551 } 3552#endif 3553 3554 // Now perform the deferred reset on fast tracks that have stopped 3555 while (resetMask != 0) { 3556 size_t i = __builtin_ctz(resetMask); 3557 ALOG_ASSERT(i < count); 3558 resetMask &= ~(1 << i); 3559 sp<Track> t = mActiveTracks[i].promote(); 3560 if (t == 0) { 3561 continue; 3562 } 3563 Track* track = t.get(); 3564 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3565 track->reset(); 3566 } 3567 3568 // remove all the tracks that need to be... 3569 removeTracks_l(*tracksToRemove); 3570 3571 // sink or mix buffer must be cleared if all tracks are connected to an 3572 // effect chain as in this case the mixer will not write to the sink or mix buffer 3573 // and track effects will accumulate into it 3574 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3575 (mixedTracks == 0 && fastTracks > 0))) { 3576 // FIXME as a performance optimization, should remember previous zero status 3577 if (mMixerBufferValid) { 3578 memset(mMixerBuffer, 0, mMixerBufferSize); 3579 // TODO: In testing, mSinkBuffer below need not be cleared because 3580 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3581 // after mixing. 3582 // 3583 // To enforce this guarantee: 3584 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3585 // (mixedTracks == 0 && fastTracks > 0)) 3586 // must imply MIXER_TRACKS_READY. 3587 // Later, we may clear buffers regardless, and skip much of this logic. 3588 } 3589 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3590 if (mEffectBufferValid) { 3591 memset(mEffectBuffer, 0, mEffectBufferSize); 3592 } 3593 // FIXME as a performance optimization, should remember previous zero status 3594 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3595 } 3596 3597 // if any fast tracks, then status is ready 3598 mMixerStatusIgnoringFastTracks = mixerStatus; 3599 if (fastTracks > 0) { 3600 mixerStatus = MIXER_TRACKS_READY; 3601 } 3602 return mixerStatus; 3603} 3604 3605// getTrackName_l() must be called with ThreadBase::mLock held 3606int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3607 audio_format_t format, int sessionId) 3608{ 3609 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3610} 3611 3612// deleteTrackName_l() must be called with ThreadBase::mLock held 3613void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3614{ 3615 ALOGV("remove track (%d) and delete from mixer", name); 3616 mAudioMixer->deleteTrackName(name); 3617} 3618 3619// checkForNewParameter_l() must be called with ThreadBase::mLock held 3620bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3621 status_t& status) 3622{ 3623 bool reconfig = false; 3624 3625 status = NO_ERROR; 3626 3627 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3628 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3629 if (mFastMixer != NULL) { 3630 FastMixerStateQueue *sq = mFastMixer->sq(); 3631 FastMixerState *state = sq->begin(); 3632 if (!(state->mCommand & FastMixerState::IDLE)) { 3633 previousCommand = state->mCommand; 3634 state->mCommand = FastMixerState::HOT_IDLE; 3635 sq->end(); 3636 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3637 } else { 3638 sq->end(false /*didModify*/); 3639 } 3640 } 3641 3642 AudioParameter param = AudioParameter(keyValuePair); 3643 int value; 3644 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3645 reconfig = true; 3646 } 3647 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3648 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3649 status = BAD_VALUE; 3650 } else { 3651 // no need to save value, since it's constant 3652 reconfig = true; 3653 } 3654 } 3655 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3656 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3657 status = BAD_VALUE; 3658 } else { 3659 // no need to save value, since it's constant 3660 reconfig = true; 3661 } 3662 } 3663 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3664 // do not accept frame count changes if tracks are open as the track buffer 3665 // size depends on frame count and correct behavior would not be guaranteed 3666 // if frame count is changed after track creation 3667 if (!mTracks.isEmpty()) { 3668 status = INVALID_OPERATION; 3669 } else { 3670 reconfig = true; 3671 } 3672 } 3673 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3674#ifdef ADD_BATTERY_DATA 3675 // when changing the audio output device, call addBatteryData to notify 3676 // the change 3677 if (mOutDevice != value) { 3678 uint32_t params = 0; 3679 // check whether speaker is on 3680 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3681 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3682 } 3683 3684 audio_devices_t deviceWithoutSpeaker 3685 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3686 // check if any other device (except speaker) is on 3687 if (value & deviceWithoutSpeaker ) { 3688 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3689 } 3690 3691 if (params != 0) { 3692 addBatteryData(params); 3693 } 3694 } 3695#endif 3696 3697 // forward device change to effects that have requested to be 3698 // aware of attached audio device. 3699 if (value != AUDIO_DEVICE_NONE) { 3700 mOutDevice = value; 3701 for (size_t i = 0; i < mEffectChains.size(); i++) { 3702 mEffectChains[i]->setDevice_l(mOutDevice); 3703 } 3704 } 3705 } 3706 3707 if (status == NO_ERROR) { 3708 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3709 keyValuePair.string()); 3710 if (!mStandby && status == INVALID_OPERATION) { 3711 mOutput->stream->common.standby(&mOutput->stream->common); 3712 mStandby = true; 3713 mBytesWritten = 0; 3714 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3715 keyValuePair.string()); 3716 } 3717 if (status == NO_ERROR && reconfig) { 3718 readOutputParameters_l(); 3719 delete mAudioMixer; 3720 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3721 for (size_t i = 0; i < mTracks.size() ; i++) { 3722 int name = getTrackName_l(mTracks[i]->mChannelMask, 3723 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3724 if (name < 0) { 3725 break; 3726 } 3727 mTracks[i]->mName = name; 3728 } 3729 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3730 } 3731 } 3732 3733 if (!(previousCommand & FastMixerState::IDLE)) { 3734 ALOG_ASSERT(mFastMixer != NULL); 3735 FastMixerStateQueue *sq = mFastMixer->sq(); 3736 FastMixerState *state = sq->begin(); 3737 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3738 state->mCommand = previousCommand; 3739 sq->end(); 3740 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3741 } 3742 3743 return reconfig; 3744} 3745 3746 3747void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3748{ 3749 const size_t SIZE = 256; 3750 char buffer[SIZE]; 3751 String8 result; 3752 3753 PlaybackThread::dumpInternals(fd, args); 3754 3755 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3756 3757 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3758 const FastMixerDumpState copy(mFastMixerDumpState); 3759 copy.dump(fd); 3760 3761#ifdef STATE_QUEUE_DUMP 3762 // Similar for state queue 3763 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3764 observerCopy.dump(fd); 3765 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3766 mutatorCopy.dump(fd); 3767#endif 3768 3769#ifdef TEE_SINK 3770 // Write the tee output to a .wav file 3771 dumpTee(fd, mTeeSource, mId); 3772#endif 3773 3774#ifdef AUDIO_WATCHDOG 3775 if (mAudioWatchdog != 0) { 3776 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3777 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3778 wdCopy.dump(fd); 3779 } 3780#endif 3781} 3782 3783uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3784{ 3785 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3786} 3787 3788uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3789{ 3790 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3791} 3792 3793void AudioFlinger::MixerThread::cacheParameters_l() 3794{ 3795 PlaybackThread::cacheParameters_l(); 3796 3797 // FIXME: Relaxed timing because of a certain device that can't meet latency 3798 // Should be reduced to 2x after the vendor fixes the driver issue 3799 // increase threshold again due to low power audio mode. The way this warning 3800 // threshold is calculated and its usefulness should be reconsidered anyway. 3801 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3802} 3803 3804// ---------------------------------------------------------------------------- 3805 3806AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3807 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3808 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3809 // mLeftVolFloat, mRightVolFloat 3810{ 3811} 3812 3813AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3814 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3815 ThreadBase::type_t type) 3816 : PlaybackThread(audioFlinger, output, id, device, type) 3817 // mLeftVolFloat, mRightVolFloat 3818{ 3819} 3820 3821AudioFlinger::DirectOutputThread::~DirectOutputThread() 3822{ 3823} 3824 3825void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3826{ 3827 audio_track_cblk_t* cblk = track->cblk(); 3828 float left, right; 3829 3830 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3831 left = right = 0; 3832 } else { 3833 float typeVolume = mStreamTypes[track->streamType()].volume; 3834 float v = mMasterVolume * typeVolume; 3835 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3836 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3837 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3838 if (left > GAIN_FLOAT_UNITY) { 3839 left = GAIN_FLOAT_UNITY; 3840 } 3841 left *= v; 3842 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3843 if (right > GAIN_FLOAT_UNITY) { 3844 right = GAIN_FLOAT_UNITY; 3845 } 3846 right *= v; 3847 } 3848 3849 if (lastTrack) { 3850 if (left != mLeftVolFloat || right != mRightVolFloat) { 3851 mLeftVolFloat = left; 3852 mRightVolFloat = right; 3853 3854 // Convert volumes from float to 8.24 3855 uint32_t vl = (uint32_t)(left * (1 << 24)); 3856 uint32_t vr = (uint32_t)(right * (1 << 24)); 3857 3858 // Delegate volume control to effect in track effect chain if needed 3859 // only one effect chain can be present on DirectOutputThread, so if 3860 // there is one, the track is connected to it 3861 if (!mEffectChains.isEmpty()) { 3862 mEffectChains[0]->setVolume_l(&vl, &vr); 3863 left = (float)vl / (1 << 24); 3864 right = (float)vr / (1 << 24); 3865 } 3866 if (mOutput->stream->set_volume) { 3867 mOutput->stream->set_volume(mOutput->stream, left, right); 3868 } 3869 } 3870 } 3871} 3872 3873 3874AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3875 Vector< sp<Track> > *tracksToRemove 3876) 3877{ 3878 size_t count = mActiveTracks.size(); 3879 mixer_state mixerStatus = MIXER_IDLE; 3880 3881 // find out which tracks need to be processed 3882 for (size_t i = 0; i < count; i++) { 3883 sp<Track> t = mActiveTracks[i].promote(); 3884 // The track died recently 3885 if (t == 0) { 3886 continue; 3887 } 3888 3889 Track* const track = t.get(); 3890 audio_track_cblk_t* cblk = track->cblk(); 3891 // Only consider last track started for volume and mixer state control. 3892 // In theory an older track could underrun and restart after the new one starts 3893 // but as we only care about the transition phase between two tracks on a 3894 // direct output, it is not a problem to ignore the underrun case. 3895 sp<Track> l = mLatestActiveTrack.promote(); 3896 bool last = l.get() == track; 3897 3898 // The first time a track is added we wait 3899 // for all its buffers to be filled before processing it 3900 uint32_t minFrames; 3901 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3902 minFrames = mNormalFrameCount; 3903 } else { 3904 minFrames = 1; 3905 } 3906 3907 if ((track->framesReady() >= minFrames) && track->isReady() && 3908 !track->isPaused() && !track->isTerminated()) 3909 { 3910 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3911 3912 if (track->mFillingUpStatus == Track::FS_FILLED) { 3913 track->mFillingUpStatus = Track::FS_ACTIVE; 3914 // make sure processVolume_l() will apply new volume even if 0 3915 mLeftVolFloat = mRightVolFloat = -1.0; 3916 if (track->mState == TrackBase::RESUMING) { 3917 track->mState = TrackBase::ACTIVE; 3918 } 3919 } 3920 3921 // compute volume for this track 3922 processVolume_l(track, last); 3923 if (last) { 3924 // reset retry count 3925 track->mRetryCount = kMaxTrackRetriesDirect; 3926 mActiveTrack = t; 3927 mixerStatus = MIXER_TRACKS_READY; 3928 } 3929 } else { 3930 // clear effect chain input buffer if the last active track started underruns 3931 // to avoid sending previous audio buffer again to effects 3932 if (!mEffectChains.isEmpty() && last) { 3933 mEffectChains[0]->clearInputBuffer(); 3934 } 3935 3936 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3937 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3938 track->isStopped() || track->isPaused()) { 3939 // We have consumed all the buffers of this track. 3940 // Remove it from the list of active tracks. 3941 // TODO: implement behavior for compressed audio 3942 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3943 size_t framesWritten = mBytesWritten / mFrameSize; 3944 if (mStandby || !last || 3945 track->presentationComplete(framesWritten, audioHALFrames)) { 3946 if (track->isStopped()) { 3947 track->reset(); 3948 } 3949 tracksToRemove->add(track); 3950 } 3951 } else { 3952 // No buffers for this track. Give it a few chances to 3953 // fill a buffer, then remove it from active list. 3954 // Only consider last track started for mixer state control 3955 if (--(track->mRetryCount) <= 0) { 3956 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3957 tracksToRemove->add(track); 3958 // indicate to client process that the track was disabled because of underrun; 3959 // it will then automatically call start() when data is available 3960 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3961 } else if (last) { 3962 mixerStatus = MIXER_TRACKS_ENABLED; 3963 } 3964 } 3965 } 3966 } 3967 3968 // remove all the tracks that need to be... 3969 removeTracks_l(*tracksToRemove); 3970 3971 return mixerStatus; 3972} 3973 3974void AudioFlinger::DirectOutputThread::threadLoop_mix() 3975{ 3976 size_t frameCount = mFrameCount; 3977 int8_t *curBuf = (int8_t *)mSinkBuffer; 3978 // output audio to hardware 3979 while (frameCount) { 3980 AudioBufferProvider::Buffer buffer; 3981 buffer.frameCount = frameCount; 3982 mActiveTrack->getNextBuffer(&buffer); 3983 if (buffer.raw == NULL) { 3984 memset(curBuf, 0, frameCount * mFrameSize); 3985 break; 3986 } 3987 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3988 frameCount -= buffer.frameCount; 3989 curBuf += buffer.frameCount * mFrameSize; 3990 mActiveTrack->releaseBuffer(&buffer); 3991 } 3992 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 3993 sleepTime = 0; 3994 standbyTime = systemTime() + standbyDelay; 3995 mActiveTrack.clear(); 3996} 3997 3998void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3999{ 4000 if (sleepTime == 0) { 4001 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4002 sleepTime = activeSleepTime; 4003 } else { 4004 sleepTime = idleSleepTime; 4005 } 4006 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4007 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4008 sleepTime = 0; 4009 } 4010} 4011 4012// getTrackName_l() must be called with ThreadBase::mLock held 4013int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4014 audio_format_t format __unused, int sessionId __unused) 4015{ 4016 return 0; 4017} 4018 4019// deleteTrackName_l() must be called with ThreadBase::mLock held 4020void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4021{ 4022} 4023 4024// checkForNewParameter_l() must be called with ThreadBase::mLock held 4025bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4026 status_t& status) 4027{ 4028 bool reconfig = false; 4029 4030 status = NO_ERROR; 4031 4032 AudioParameter param = AudioParameter(keyValuePair); 4033 int value; 4034 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4035 // forward device change to effects that have requested to be 4036 // aware of attached audio device. 4037 if (value != AUDIO_DEVICE_NONE) { 4038 mOutDevice = value; 4039 for (size_t i = 0; i < mEffectChains.size(); i++) { 4040 mEffectChains[i]->setDevice_l(mOutDevice); 4041 } 4042 } 4043 } 4044 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4045 // do not accept frame count changes if tracks are open as the track buffer 4046 // size depends on frame count and correct behavior would not be garantied 4047 // if frame count is changed after track creation 4048 if (!mTracks.isEmpty()) { 4049 status = INVALID_OPERATION; 4050 } else { 4051 reconfig = true; 4052 } 4053 } 4054 if (status == NO_ERROR) { 4055 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4056 keyValuePair.string()); 4057 if (!mStandby && status == INVALID_OPERATION) { 4058 mOutput->stream->common.standby(&mOutput->stream->common); 4059 mStandby = true; 4060 mBytesWritten = 0; 4061 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4062 keyValuePair.string()); 4063 } 4064 if (status == NO_ERROR && reconfig) { 4065 readOutputParameters_l(); 4066 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4067 } 4068 } 4069 4070 return reconfig; 4071} 4072 4073uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4074{ 4075 uint32_t time; 4076 if (audio_is_linear_pcm(mFormat)) { 4077 time = PlaybackThread::activeSleepTimeUs(); 4078 } else { 4079 time = 10000; 4080 } 4081 return time; 4082} 4083 4084uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4085{ 4086 uint32_t time; 4087 if (audio_is_linear_pcm(mFormat)) { 4088 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4089 } else { 4090 time = 10000; 4091 } 4092 return time; 4093} 4094 4095uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4096{ 4097 uint32_t time; 4098 if (audio_is_linear_pcm(mFormat)) { 4099 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4100 } else { 4101 time = 10000; 4102 } 4103 return time; 4104} 4105 4106void AudioFlinger::DirectOutputThread::cacheParameters_l() 4107{ 4108 PlaybackThread::cacheParameters_l(); 4109 4110 // use shorter standby delay as on normal output to release 4111 // hardware resources as soon as possible 4112 if (audio_is_linear_pcm(mFormat)) { 4113 standbyDelay = microseconds(activeSleepTime*2); 4114 } else { 4115 standbyDelay = kOffloadStandbyDelayNs; 4116 } 4117} 4118 4119// ---------------------------------------------------------------------------- 4120 4121AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4122 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4123 : Thread(false /*canCallJava*/), 4124 mPlaybackThread(playbackThread), 4125 mWriteAckSequence(0), 4126 mDrainSequence(0) 4127{ 4128} 4129 4130AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4131{ 4132} 4133 4134void AudioFlinger::AsyncCallbackThread::onFirstRef() 4135{ 4136 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4137} 4138 4139bool AudioFlinger::AsyncCallbackThread::threadLoop() 4140{ 4141 while (!exitPending()) { 4142 uint32_t writeAckSequence; 4143 uint32_t drainSequence; 4144 4145 { 4146 Mutex::Autolock _l(mLock); 4147 while (!((mWriteAckSequence & 1) || 4148 (mDrainSequence & 1) || 4149 exitPending())) { 4150 mWaitWorkCV.wait(mLock); 4151 } 4152 4153 if (exitPending()) { 4154 break; 4155 } 4156 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4157 mWriteAckSequence, mDrainSequence); 4158 writeAckSequence = mWriteAckSequence; 4159 mWriteAckSequence &= ~1; 4160 drainSequence = mDrainSequence; 4161 mDrainSequence &= ~1; 4162 } 4163 { 4164 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4165 if (playbackThread != 0) { 4166 if (writeAckSequence & 1) { 4167 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4168 } 4169 if (drainSequence & 1) { 4170 playbackThread->resetDraining(drainSequence >> 1); 4171 } 4172 } 4173 } 4174 } 4175 return false; 4176} 4177 4178void AudioFlinger::AsyncCallbackThread::exit() 4179{ 4180 ALOGV("AsyncCallbackThread::exit"); 4181 Mutex::Autolock _l(mLock); 4182 requestExit(); 4183 mWaitWorkCV.broadcast(); 4184} 4185 4186void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4187{ 4188 Mutex::Autolock _l(mLock); 4189 // bit 0 is cleared 4190 mWriteAckSequence = sequence << 1; 4191} 4192 4193void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4194{ 4195 Mutex::Autolock _l(mLock); 4196 // ignore unexpected callbacks 4197 if (mWriteAckSequence & 2) { 4198 mWriteAckSequence |= 1; 4199 mWaitWorkCV.signal(); 4200 } 4201} 4202 4203void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4204{ 4205 Mutex::Autolock _l(mLock); 4206 // bit 0 is cleared 4207 mDrainSequence = sequence << 1; 4208} 4209 4210void AudioFlinger::AsyncCallbackThread::resetDraining() 4211{ 4212 Mutex::Autolock _l(mLock); 4213 // ignore unexpected callbacks 4214 if (mDrainSequence & 2) { 4215 mDrainSequence |= 1; 4216 mWaitWorkCV.signal(); 4217 } 4218} 4219 4220 4221// ---------------------------------------------------------------------------- 4222AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4223 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4224 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4225 mHwPaused(false), 4226 mFlushPending(false), 4227 mPausedBytesRemaining(0) 4228{ 4229 //FIXME: mStandby should be set to true by ThreadBase constructor 4230 mStandby = true; 4231} 4232 4233void AudioFlinger::OffloadThread::threadLoop_exit() 4234{ 4235 if (mFlushPending || mHwPaused) { 4236 // If a flush is pending or track was paused, just discard buffered data 4237 flushHw_l(); 4238 } else { 4239 mMixerStatus = MIXER_DRAIN_ALL; 4240 threadLoop_drain(); 4241 } 4242 if (mUseAsyncWrite) { 4243 ALOG_ASSERT(mCallbackThread != 0); 4244 mCallbackThread->exit(); 4245 } 4246 PlaybackThread::threadLoop_exit(); 4247} 4248 4249AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4250 Vector< sp<Track> > *tracksToRemove 4251) 4252{ 4253 size_t count = mActiveTracks.size(); 4254 4255 mixer_state mixerStatus = MIXER_IDLE; 4256 bool doHwPause = false; 4257 bool doHwResume = false; 4258 4259 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4260 4261 // find out which tracks need to be processed 4262 for (size_t i = 0; i < count; i++) { 4263 sp<Track> t = mActiveTracks[i].promote(); 4264 // The track died recently 4265 if (t == 0) { 4266 continue; 4267 } 4268 Track* const track = t.get(); 4269 audio_track_cblk_t* cblk = track->cblk(); 4270 // Only consider last track started for volume and mixer state control. 4271 // In theory an older track could underrun and restart after the new one starts 4272 // but as we only care about the transition phase between two tracks on a 4273 // direct output, it is not a problem to ignore the underrun case. 4274 sp<Track> l = mLatestActiveTrack.promote(); 4275 bool last = l.get() == track; 4276 4277 if (track->isInvalid()) { 4278 ALOGW("An invalidated track shouldn't be in active list"); 4279 tracksToRemove->add(track); 4280 continue; 4281 } 4282 4283 if (track->mState == TrackBase::IDLE) { 4284 ALOGW("An idle track shouldn't be in active list"); 4285 continue; 4286 } 4287 4288 if (track->isPausing()) { 4289 track->setPaused(); 4290 if (last) { 4291 if (!mHwPaused) { 4292 doHwPause = true; 4293 mHwPaused = true; 4294 } 4295 // If we were part way through writing the mixbuffer to 4296 // the HAL we must save this until we resume 4297 // BUG - this will be wrong if a different track is made active, 4298 // in that case we want to discard the pending data in the 4299 // mixbuffer and tell the client to present it again when the 4300 // track is resumed 4301 mPausedWriteLength = mCurrentWriteLength; 4302 mPausedBytesRemaining = mBytesRemaining; 4303 mBytesRemaining = 0; // stop writing 4304 } 4305 tracksToRemove->add(track); 4306 } else if (track->isFlushPending()) { 4307 track->flushAck(); 4308 if (last) { 4309 mFlushPending = true; 4310 } 4311 } else if (track->isResumePending()){ 4312 track->resumeAck(); 4313 if (last) { 4314 if (mPausedBytesRemaining) { 4315 // Need to continue write that was interrupted 4316 mCurrentWriteLength = mPausedWriteLength; 4317 mBytesRemaining = mPausedBytesRemaining; 4318 mPausedBytesRemaining = 0; 4319 } 4320 if (mHwPaused) { 4321 doHwResume = true; 4322 mHwPaused = false; 4323 // threadLoop_mix() will handle the case that we need to 4324 // resume an interrupted write 4325 } 4326 // enable write to audio HAL 4327 sleepTime = 0; 4328 4329 // Do not handle new data in this iteration even if track->framesReady() 4330 mixerStatus = MIXER_TRACKS_ENABLED; 4331 } 4332 } else if (track->framesReady() && track->isReady() && 4333 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4334 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4335 if (track->mFillingUpStatus == Track::FS_FILLED) { 4336 track->mFillingUpStatus = Track::FS_ACTIVE; 4337 // make sure processVolume_l() will apply new volume even if 0 4338 mLeftVolFloat = mRightVolFloat = -1.0; 4339 } 4340 4341 if (last) { 4342 sp<Track> previousTrack = mPreviousTrack.promote(); 4343 if (previousTrack != 0) { 4344 if (track != previousTrack.get()) { 4345 // Flush any data still being written from last track 4346 mBytesRemaining = 0; 4347 if (mPausedBytesRemaining) { 4348 // Last track was paused so we also need to flush saved 4349 // mixbuffer state and invalidate track so that it will 4350 // re-submit that unwritten data when it is next resumed 4351 mPausedBytesRemaining = 0; 4352 // Invalidate is a bit drastic - would be more efficient 4353 // to have a flag to tell client that some of the 4354 // previously written data was lost 4355 previousTrack->invalidate(); 4356 } 4357 // flush data already sent to the DSP if changing audio session as audio 4358 // comes from a different source. Also invalidate previous track to force a 4359 // seek when resuming. 4360 if (previousTrack->sessionId() != track->sessionId()) { 4361 previousTrack->invalidate(); 4362 } 4363 } 4364 } 4365 mPreviousTrack = track; 4366 // reset retry count 4367 track->mRetryCount = kMaxTrackRetriesOffload; 4368 mActiveTrack = t; 4369 mixerStatus = MIXER_TRACKS_READY; 4370 } 4371 } else { 4372 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4373 if (track->isStopping_1()) { 4374 // Hardware buffer can hold a large amount of audio so we must 4375 // wait for all current track's data to drain before we say 4376 // that the track is stopped. 4377 if (mBytesRemaining == 0) { 4378 // Only start draining when all data in mixbuffer 4379 // has been written 4380 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4381 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4382 // do not drain if no data was ever sent to HAL (mStandby == true) 4383 if (last && !mStandby) { 4384 // do not modify drain sequence if we are already draining. This happens 4385 // when resuming from pause after drain. 4386 if ((mDrainSequence & 1) == 0) { 4387 sleepTime = 0; 4388 standbyTime = systemTime() + standbyDelay; 4389 mixerStatus = MIXER_DRAIN_TRACK; 4390 mDrainSequence += 2; 4391 } 4392 if (mHwPaused) { 4393 // It is possible to move from PAUSED to STOPPING_1 without 4394 // a resume so we must ensure hardware is running 4395 doHwResume = true; 4396 mHwPaused = false; 4397 } 4398 } 4399 } 4400 } else if (track->isStopping_2()) { 4401 // Drain has completed or we are in standby, signal presentation complete 4402 if (!(mDrainSequence & 1) || !last || mStandby) { 4403 track->mState = TrackBase::STOPPED; 4404 size_t audioHALFrames = 4405 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4406 size_t framesWritten = 4407 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4408 track->presentationComplete(framesWritten, audioHALFrames); 4409 track->reset(); 4410 tracksToRemove->add(track); 4411 } 4412 } else { 4413 // No buffers for this track. Give it a few chances to 4414 // fill a buffer, then remove it from active list. 4415 if (--(track->mRetryCount) <= 0) { 4416 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4417 track->name()); 4418 tracksToRemove->add(track); 4419 // indicate to client process that the track was disabled because of underrun; 4420 // it will then automatically call start() when data is available 4421 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4422 } else if (last){ 4423 mixerStatus = MIXER_TRACKS_ENABLED; 4424 } 4425 } 4426 } 4427 // compute volume for this track 4428 processVolume_l(track, last); 4429 } 4430 4431 // make sure the pause/flush/resume sequence is executed in the right order. 4432 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4433 // before flush and then resume HW. This can happen in case of pause/flush/resume 4434 // if resume is received before pause is executed. 4435 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4436 mOutput->stream->pause(mOutput->stream); 4437 } 4438 if (mFlushPending) { 4439 flushHw_l(); 4440 mFlushPending = false; 4441 } 4442 if (!mStandby && doHwResume) { 4443 mOutput->stream->resume(mOutput->stream); 4444 } 4445 4446 // remove all the tracks that need to be... 4447 removeTracks_l(*tracksToRemove); 4448 4449 return mixerStatus; 4450} 4451 4452// must be called with thread mutex locked 4453bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4454{ 4455 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4456 mWriteAckSequence, mDrainSequence); 4457 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4458 return true; 4459 } 4460 return false; 4461} 4462 4463// must be called with thread mutex locked 4464bool AudioFlinger::OffloadThread::shouldStandby_l() 4465{ 4466 bool trackPaused = false; 4467 4468 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4469 // after a timeout and we will enter standby then. 4470 if (mTracks.size() > 0) { 4471 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4472 } 4473 4474 return !mStandby && !trackPaused; 4475} 4476 4477 4478bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4479{ 4480 Mutex::Autolock _l(mLock); 4481 return waitingAsyncCallback_l(); 4482} 4483 4484void AudioFlinger::OffloadThread::flushHw_l() 4485{ 4486 mOutput->stream->flush(mOutput->stream); 4487 // Flush anything still waiting in the mixbuffer 4488 mCurrentWriteLength = 0; 4489 mBytesRemaining = 0; 4490 mPausedWriteLength = 0; 4491 mPausedBytesRemaining = 0; 4492 mHwPaused = false; 4493 4494 if (mUseAsyncWrite) { 4495 // discard any pending drain or write ack by incrementing sequence 4496 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4497 mDrainSequence = (mDrainSequence + 2) & ~1; 4498 ALOG_ASSERT(mCallbackThread != 0); 4499 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4500 mCallbackThread->setDraining(mDrainSequence); 4501 } 4502} 4503 4504void AudioFlinger::OffloadThread::onAddNewTrack_l() 4505{ 4506 sp<Track> previousTrack = mPreviousTrack.promote(); 4507 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4508 4509 if (previousTrack != 0 && latestTrack != 0 && 4510 (previousTrack->sessionId() != latestTrack->sessionId())) { 4511 mFlushPending = true; 4512 } 4513 PlaybackThread::onAddNewTrack_l(); 4514} 4515 4516// ---------------------------------------------------------------------------- 4517 4518AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4519 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4520 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4521 DUPLICATING), 4522 mWaitTimeMs(UINT_MAX) 4523{ 4524 addOutputTrack(mainThread); 4525} 4526 4527AudioFlinger::DuplicatingThread::~DuplicatingThread() 4528{ 4529 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4530 mOutputTracks[i]->destroy(); 4531 } 4532} 4533 4534void AudioFlinger::DuplicatingThread::threadLoop_mix() 4535{ 4536 // mix buffers... 4537 if (outputsReady(outputTracks)) { 4538 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4539 } else { 4540 memset(mSinkBuffer, 0, mSinkBufferSize); 4541 } 4542 sleepTime = 0; 4543 writeFrames = mNormalFrameCount; 4544 mCurrentWriteLength = mSinkBufferSize; 4545 standbyTime = systemTime() + standbyDelay; 4546} 4547 4548void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4549{ 4550 if (sleepTime == 0) { 4551 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4552 sleepTime = activeSleepTime; 4553 } else { 4554 sleepTime = idleSleepTime; 4555 } 4556 } else if (mBytesWritten != 0) { 4557 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4558 writeFrames = mNormalFrameCount; 4559 memset(mSinkBuffer, 0, mSinkBufferSize); 4560 } else { 4561 // flush remaining overflow buffers in output tracks 4562 writeFrames = 0; 4563 } 4564 sleepTime = 0; 4565 } 4566} 4567 4568ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4569{ 4570 for (size_t i = 0; i < outputTracks.size(); i++) { 4571 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4572 // for delivery downstream as needed. This in-place conversion is safe as 4573 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4574 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4575 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4576 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4577 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4578 } 4579 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4580 } 4581 mStandby = false; 4582 return (ssize_t)mSinkBufferSize; 4583} 4584 4585void AudioFlinger::DuplicatingThread::threadLoop_standby() 4586{ 4587 // DuplicatingThread implements standby by stopping all tracks 4588 for (size_t i = 0; i < outputTracks.size(); i++) { 4589 outputTracks[i]->stop(); 4590 } 4591} 4592 4593void AudioFlinger::DuplicatingThread::saveOutputTracks() 4594{ 4595 outputTracks = mOutputTracks; 4596} 4597 4598void AudioFlinger::DuplicatingThread::clearOutputTracks() 4599{ 4600 outputTracks.clear(); 4601} 4602 4603void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4604{ 4605 Mutex::Autolock _l(mLock); 4606 // FIXME explain this formula 4607 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4608 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4609 // due to current usage case and restrictions on the AudioBufferProvider. 4610 // Actual buffer conversion is done in threadLoop_write(). 4611 // 4612 // TODO: This may change in the future, depending on multichannel 4613 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4614 OutputTrack *outputTrack = new OutputTrack(thread, 4615 this, 4616 mSampleRate, 4617 AUDIO_FORMAT_PCM_16_BIT, 4618 mChannelMask, 4619 frameCount, 4620 IPCThreadState::self()->getCallingUid()); 4621 if (outputTrack->cblk() != NULL) { 4622 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4623 mOutputTracks.add(outputTrack); 4624 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4625 updateWaitTime_l(); 4626 } 4627} 4628 4629void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4630{ 4631 Mutex::Autolock _l(mLock); 4632 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4633 if (mOutputTracks[i]->thread() == thread) { 4634 mOutputTracks[i]->destroy(); 4635 mOutputTracks.removeAt(i); 4636 updateWaitTime_l(); 4637 return; 4638 } 4639 } 4640 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4641} 4642 4643// caller must hold mLock 4644void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4645{ 4646 mWaitTimeMs = UINT_MAX; 4647 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4648 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4649 if (strong != 0) { 4650 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4651 if (waitTimeMs < mWaitTimeMs) { 4652 mWaitTimeMs = waitTimeMs; 4653 } 4654 } 4655 } 4656} 4657 4658 4659bool AudioFlinger::DuplicatingThread::outputsReady( 4660 const SortedVector< sp<OutputTrack> > &outputTracks) 4661{ 4662 for (size_t i = 0; i < outputTracks.size(); i++) { 4663 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4664 if (thread == 0) { 4665 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4666 outputTracks[i].get()); 4667 return false; 4668 } 4669 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4670 // see note at standby() declaration 4671 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4672 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4673 thread.get()); 4674 return false; 4675 } 4676 } 4677 return true; 4678} 4679 4680uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4681{ 4682 return (mWaitTimeMs * 1000) / 2; 4683} 4684 4685void AudioFlinger::DuplicatingThread::cacheParameters_l() 4686{ 4687 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4688 updateWaitTime_l(); 4689 4690 MixerThread::cacheParameters_l(); 4691} 4692 4693// ---------------------------------------------------------------------------- 4694// Record 4695// ---------------------------------------------------------------------------- 4696 4697AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4698 AudioStreamIn *input, 4699 audio_io_handle_t id, 4700 audio_devices_t outDevice, 4701 audio_devices_t inDevice 4702#ifdef TEE_SINK 4703 , const sp<NBAIO_Sink>& teeSink 4704#endif 4705 ) : 4706 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4707 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4708 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4709 mRsmpInRear(0) 4710#ifdef TEE_SINK 4711 , mTeeSink(teeSink) 4712#endif 4713 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4714 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4715{ 4716 snprintf(mName, kNameLength, "AudioIn_%X", id); 4717 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4718 4719 readInputParameters_l(); 4720} 4721 4722 4723AudioFlinger::RecordThread::~RecordThread() 4724{ 4725 mAudioFlinger->unregisterWriter(mNBLogWriter); 4726 delete[] mRsmpInBuffer; 4727} 4728 4729void AudioFlinger::RecordThread::onFirstRef() 4730{ 4731 run(mName, PRIORITY_URGENT_AUDIO); 4732} 4733 4734bool AudioFlinger::RecordThread::threadLoop() 4735{ 4736 nsecs_t lastWarning = 0; 4737 4738 inputStandBy(); 4739 4740reacquire_wakelock: 4741 sp<RecordTrack> activeTrack; 4742 int activeTracksGen; 4743 { 4744 Mutex::Autolock _l(mLock); 4745 size_t size = mActiveTracks.size(); 4746 activeTracksGen = mActiveTracksGen; 4747 if (size > 0) { 4748 // FIXME an arbitrary choice 4749 activeTrack = mActiveTracks[0]; 4750 acquireWakeLock_l(activeTrack->uid()); 4751 if (size > 1) { 4752 SortedVector<int> tmp; 4753 for (size_t i = 0; i < size; i++) { 4754 tmp.add(mActiveTracks[i]->uid()); 4755 } 4756 updateWakeLockUids_l(tmp); 4757 } 4758 } else { 4759 acquireWakeLock_l(-1); 4760 } 4761 } 4762 4763 // used to request a deferred sleep, to be executed later while mutex is unlocked 4764 uint32_t sleepUs = 0; 4765 4766 // loop while there is work to do 4767 for (;;) { 4768 Vector< sp<EffectChain> > effectChains; 4769 4770 // sleep with mutex unlocked 4771 if (sleepUs > 0) { 4772 usleep(sleepUs); 4773 sleepUs = 0; 4774 } 4775 4776 // activeTracks accumulates a copy of a subset of mActiveTracks 4777 Vector< sp<RecordTrack> > activeTracks; 4778 4779 4780 { // scope for mLock 4781 Mutex::Autolock _l(mLock); 4782 4783 processConfigEvents_l(); 4784 4785 // check exitPending here because checkForNewParameters_l() and 4786 // checkForNewParameters_l() can temporarily release mLock 4787 if (exitPending()) { 4788 break; 4789 } 4790 4791 // if no active track(s), then standby and release wakelock 4792 size_t size = mActiveTracks.size(); 4793 if (size == 0) { 4794 standbyIfNotAlreadyInStandby(); 4795 // exitPending() can't become true here 4796 releaseWakeLock_l(); 4797 ALOGV("RecordThread: loop stopping"); 4798 // go to sleep 4799 mWaitWorkCV.wait(mLock); 4800 ALOGV("RecordThread: loop starting"); 4801 goto reacquire_wakelock; 4802 } 4803 4804 if (mActiveTracksGen != activeTracksGen) { 4805 activeTracksGen = mActiveTracksGen; 4806 SortedVector<int> tmp; 4807 for (size_t i = 0; i < size; i++) { 4808 tmp.add(mActiveTracks[i]->uid()); 4809 } 4810 updateWakeLockUids_l(tmp); 4811 } 4812 4813 bool doBroadcast = false; 4814 for (size_t i = 0; i < size; ) { 4815 4816 activeTrack = mActiveTracks[i]; 4817 if (activeTrack->isTerminated()) { 4818 removeTrack_l(activeTrack); 4819 mActiveTracks.remove(activeTrack); 4820 mActiveTracksGen++; 4821 size--; 4822 continue; 4823 } 4824 4825 TrackBase::track_state activeTrackState = activeTrack->mState; 4826 switch (activeTrackState) { 4827 4828 case TrackBase::PAUSING: 4829 mActiveTracks.remove(activeTrack); 4830 mActiveTracksGen++; 4831 doBroadcast = true; 4832 size--; 4833 continue; 4834 4835 case TrackBase::STARTING_1: 4836 sleepUs = 10000; 4837 i++; 4838 continue; 4839 4840 case TrackBase::STARTING_2: 4841 doBroadcast = true; 4842 mStandby = false; 4843 activeTrack->mState = TrackBase::ACTIVE; 4844 break; 4845 4846 case TrackBase::ACTIVE: 4847 break; 4848 4849 case TrackBase::IDLE: 4850 i++; 4851 continue; 4852 4853 default: 4854 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4855 } 4856 4857 activeTracks.add(activeTrack); 4858 i++; 4859 4860 } 4861 if (doBroadcast) { 4862 mStartStopCond.broadcast(); 4863 } 4864 4865 // sleep if there are no active tracks to process 4866 if (activeTracks.size() == 0) { 4867 if (sleepUs == 0) { 4868 sleepUs = kRecordThreadSleepUs; 4869 } 4870 continue; 4871 } 4872 sleepUs = 0; 4873 4874 lockEffectChains_l(effectChains); 4875 } 4876 4877 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4878 4879 size_t size = effectChains.size(); 4880 for (size_t i = 0; i < size; i++) { 4881 // thread mutex is not locked, but effect chain is locked 4882 effectChains[i]->process_l(); 4883 } 4884 4885 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4886 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4887 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4888 // If destination is non-contiguous, first read past the nominal end of buffer, then 4889 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4890 4891 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4892 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4893 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4894 if (bytesRead <= 0) { 4895 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4896 // Force input into standby so that it tries to recover at next read attempt 4897 inputStandBy(); 4898 sleepUs = kRecordThreadSleepUs; 4899 continue; 4900 } 4901 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4902 size_t framesRead = bytesRead / mFrameSize; 4903 ALOG_ASSERT(framesRead > 0); 4904 if (mTeeSink != 0) { 4905 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4906 } 4907 // If destination is non-contiguous, we now correct for reading past end of buffer. 4908 size_t part1 = mRsmpInFramesP2 - rear; 4909 if (framesRead > part1) { 4910 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4911 (framesRead - part1) * mFrameSize); 4912 } 4913 rear = mRsmpInRear += framesRead; 4914 4915 size = activeTracks.size(); 4916 // loop over each active track 4917 for (size_t i = 0; i < size; i++) { 4918 activeTrack = activeTracks[i]; 4919 4920 enum { 4921 OVERRUN_UNKNOWN, 4922 OVERRUN_TRUE, 4923 OVERRUN_FALSE 4924 } overrun = OVERRUN_UNKNOWN; 4925 4926 // loop over getNextBuffer to handle circular sink 4927 for (;;) { 4928 4929 activeTrack->mSink.frameCount = ~0; 4930 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4931 size_t framesOut = activeTrack->mSink.frameCount; 4932 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4933 4934 int32_t front = activeTrack->mRsmpInFront; 4935 ssize_t filled = rear - front; 4936 size_t framesIn; 4937 4938 if (filled < 0) { 4939 // should not happen, but treat like a massive overrun and re-sync 4940 framesIn = 0; 4941 activeTrack->mRsmpInFront = rear; 4942 overrun = OVERRUN_TRUE; 4943 } else if ((size_t) filled <= mRsmpInFrames) { 4944 framesIn = (size_t) filled; 4945 } else { 4946 // client is not keeping up with server, but give it latest data 4947 framesIn = mRsmpInFrames; 4948 activeTrack->mRsmpInFront = front = rear - framesIn; 4949 overrun = OVERRUN_TRUE; 4950 } 4951 4952 if (framesOut == 0 || framesIn == 0) { 4953 break; 4954 } 4955 4956 if (activeTrack->mResampler == NULL) { 4957 // no resampling 4958 if (framesIn > framesOut) { 4959 framesIn = framesOut; 4960 } else { 4961 framesOut = framesIn; 4962 } 4963 int8_t *dst = activeTrack->mSink.i8; 4964 while (framesIn > 0) { 4965 front &= mRsmpInFramesP2 - 1; 4966 size_t part1 = mRsmpInFramesP2 - front; 4967 if (part1 > framesIn) { 4968 part1 = framesIn; 4969 } 4970 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 4971 if (mChannelCount == activeTrack->mChannelCount) { 4972 memcpy(dst, src, part1 * mFrameSize); 4973 } else if (mChannelCount == 1) { 4974 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 4975 part1); 4976 } else { 4977 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 4978 part1); 4979 } 4980 dst += part1 * activeTrack->mFrameSize; 4981 front += part1; 4982 framesIn -= part1; 4983 } 4984 activeTrack->mRsmpInFront += framesOut; 4985 4986 } else { 4987 // resampling 4988 // FIXME framesInNeeded should really be part of resampler API, and should 4989 // depend on the SRC ratio 4990 // to keep mRsmpInBuffer full so resampler always has sufficient input 4991 size_t framesInNeeded; 4992 // FIXME only re-calculate when it changes, and optimize for common ratios 4993 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 4994 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 4995 framesInNeeded = ceil(framesOut * inOverOut) + 1; 4996 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 4997 framesInNeeded, framesOut, inOverOut); 4998 // Although we theoretically have framesIn in circular buffer, some of those are 4999 // unreleased frames, and thus must be discounted for purpose of budgeting. 5000 size_t unreleased = activeTrack->mRsmpInUnrel; 5001 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5002 if (framesIn < framesInNeeded) { 5003 ALOGV("not enough to resample: have %u frames in but need %u in to " 5004 "produce %u out given in/out ratio of %.4g", 5005 framesIn, framesInNeeded, framesOut, inOverOut); 5006 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5007 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5008 if (newFramesOut == 0) { 5009 break; 5010 } 5011 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5012 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5013 framesInNeeded, newFramesOut, outOverIn); 5014 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5015 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5016 "given in/out ratio of %.4g", 5017 framesIn, framesInNeeded, newFramesOut, inOverOut); 5018 framesOut = newFramesOut; 5019 } else { 5020 ALOGV("success 1: have %u in and need %u in to produce %u out " 5021 "given in/out ratio of %.4g", 5022 framesIn, framesInNeeded, framesOut, inOverOut); 5023 } 5024 5025 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5026 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5027 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5028 delete[] activeTrack->mRsmpOutBuffer; 5029 // resampler always outputs stereo 5030 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5031 activeTrack->mRsmpOutFrameCount = framesOut; 5032 } 5033 5034 // resampler accumulates, but we only have one source track 5035 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5036 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5037 // FIXME how about having activeTrack implement this interface itself? 5038 activeTrack->mResamplerBufferProvider 5039 /*this*/ /* AudioBufferProvider* */); 5040 // ditherAndClamp() works as long as all buffers returned by 5041 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5042 if (activeTrack->mChannelCount == 1) { 5043 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5044 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5045 framesOut); 5046 // the resampler always outputs stereo samples: 5047 // do post stereo to mono conversion 5048 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5049 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5050 } else { 5051 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5052 activeTrack->mRsmpOutBuffer, framesOut); 5053 } 5054 // now done with mRsmpOutBuffer 5055 5056 } 5057 5058 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5059 overrun = OVERRUN_FALSE; 5060 } 5061 5062 if (activeTrack->mFramesToDrop == 0) { 5063 if (framesOut > 0) { 5064 activeTrack->mSink.frameCount = framesOut; 5065 activeTrack->releaseBuffer(&activeTrack->mSink); 5066 } 5067 } else { 5068 // FIXME could do a partial drop of framesOut 5069 if (activeTrack->mFramesToDrop > 0) { 5070 activeTrack->mFramesToDrop -= framesOut; 5071 if (activeTrack->mFramesToDrop <= 0) { 5072 activeTrack->clearSyncStartEvent(); 5073 } 5074 } else { 5075 activeTrack->mFramesToDrop += framesOut; 5076 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5077 activeTrack->mSyncStartEvent->isCancelled()) { 5078 ALOGW("Synced record %s, session %d, trigger session %d", 5079 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5080 activeTrack->sessionId(), 5081 (activeTrack->mSyncStartEvent != 0) ? 5082 activeTrack->mSyncStartEvent->triggerSession() : 0); 5083 activeTrack->clearSyncStartEvent(); 5084 } 5085 } 5086 } 5087 5088 if (framesOut == 0) { 5089 break; 5090 } 5091 } 5092 5093 switch (overrun) { 5094 case OVERRUN_TRUE: 5095 // client isn't retrieving buffers fast enough 5096 if (!activeTrack->setOverflow()) { 5097 nsecs_t now = systemTime(); 5098 // FIXME should lastWarning per track? 5099 if ((now - lastWarning) > kWarningThrottleNs) { 5100 ALOGW("RecordThread: buffer overflow"); 5101 lastWarning = now; 5102 } 5103 } 5104 break; 5105 case OVERRUN_FALSE: 5106 activeTrack->clearOverflow(); 5107 break; 5108 case OVERRUN_UNKNOWN: 5109 break; 5110 } 5111 5112 } 5113 5114 // enable changes in effect chain 5115 unlockEffectChains(effectChains); 5116 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5117 } 5118 5119 standbyIfNotAlreadyInStandby(); 5120 5121 { 5122 Mutex::Autolock _l(mLock); 5123 for (size_t i = 0; i < mTracks.size(); i++) { 5124 sp<RecordTrack> track = mTracks[i]; 5125 track->invalidate(); 5126 } 5127 mActiveTracks.clear(); 5128 mActiveTracksGen++; 5129 mStartStopCond.broadcast(); 5130 } 5131 5132 releaseWakeLock(); 5133 5134 ALOGV("RecordThread %p exiting", this); 5135 return false; 5136} 5137 5138void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5139{ 5140 if (!mStandby) { 5141 inputStandBy(); 5142 mStandby = true; 5143 } 5144} 5145 5146void AudioFlinger::RecordThread::inputStandBy() 5147{ 5148 mInput->stream->common.standby(&mInput->stream->common); 5149} 5150 5151// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5152sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5153 const sp<AudioFlinger::Client>& client, 5154 uint32_t sampleRate, 5155 audio_format_t format, 5156 audio_channel_mask_t channelMask, 5157 size_t *pFrameCount, 5158 int sessionId, 5159 int uid, 5160 IAudioFlinger::track_flags_t *flags, 5161 pid_t tid, 5162 status_t *status) 5163{ 5164 size_t frameCount = *pFrameCount; 5165 sp<RecordTrack> track; 5166 status_t lStatus; 5167 5168 // client expresses a preference for FAST, but we get the final say 5169 if (*flags & IAudioFlinger::TRACK_FAST) { 5170 if ( 5171 // use case: callback handler and frame count is default or at least as large as HAL 5172 ( 5173 (tid != -1) && 5174 ((frameCount == 0) || 5175 // FIXME not necessarily true, should be native frame count for native SR! 5176 (frameCount >= mFrameCount)) 5177 ) && 5178 // PCM data 5179 audio_is_linear_pcm(format) && 5180 // mono or stereo 5181 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5182 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5183 // hardware sample rate 5184 // FIXME actually the native hardware sample rate 5185 (sampleRate == mSampleRate) && 5186 // record thread has an associated fast capture 5187 hasFastCapture() 5188 // fast capture does not require slots 5189 ) { 5190 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5191 if (frameCount == 0) { 5192 // FIXME wrong mFrameCount 5193 frameCount = mFrameCount * kFastTrackMultiplier; 5194 } 5195 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5196 frameCount, mFrameCount); 5197 } else { 5198 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5199 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5200 "hasFastCapture=%d tid=%d", 5201 frameCount, mFrameCount, format, 5202 audio_is_linear_pcm(format), 5203 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5204 *flags &= ~IAudioFlinger::TRACK_FAST; 5205 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5206 // For compatibility with AudioRecord calculation, buffer depth is forced 5207 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5208 // This is probably too conservative, but legacy application code may depend on it. 5209 // If you change this calculation, also review the start threshold which is related. 5210 // FIXME It's not clear how input latency actually matters. Perhaps this should be 0. 5211 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5212 size_t mNormalFrameCount = 2048; // FIXME 5213 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5214 if (minBufCount < 2) { 5215 minBufCount = 2; 5216 } 5217 size_t minFrameCount = mNormalFrameCount * minBufCount; 5218 if (frameCount < minFrameCount) { 5219 frameCount = minFrameCount; 5220 } 5221 } 5222 } 5223 *pFrameCount = frameCount; 5224 5225 lStatus = initCheck(); 5226 if (lStatus != NO_ERROR) { 5227 ALOGE("createRecordTrack_l() audio driver not initialized"); 5228 goto Exit; 5229 } 5230 5231 { // scope for mLock 5232 Mutex::Autolock _l(mLock); 5233 5234 track = new RecordTrack(this, client, sampleRate, 5235 format, channelMask, frameCount, sessionId, uid, 5236 *flags); 5237 5238 lStatus = track->initCheck(); 5239 if (lStatus != NO_ERROR) { 5240 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5241 // track must be cleared from the caller as the caller has the AF lock 5242 goto Exit; 5243 } 5244 mTracks.add(track); 5245 5246 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5247 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5248 mAudioFlinger->btNrecIsOff(); 5249 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5250 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5251 5252 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5253 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5254 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5255 // so ask activity manager to do this on our behalf 5256 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5257 } 5258 } 5259 5260 lStatus = NO_ERROR; 5261 5262Exit: 5263 *status = lStatus; 5264 return track; 5265} 5266 5267status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5268 AudioSystem::sync_event_t event, 5269 int triggerSession) 5270{ 5271 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5272 sp<ThreadBase> strongMe = this; 5273 status_t status = NO_ERROR; 5274 5275 if (event == AudioSystem::SYNC_EVENT_NONE) { 5276 recordTrack->clearSyncStartEvent(); 5277 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5278 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5279 triggerSession, 5280 recordTrack->sessionId(), 5281 syncStartEventCallback, 5282 recordTrack); 5283 // Sync event can be cancelled by the trigger session if the track is not in a 5284 // compatible state in which case we start record immediately 5285 if (recordTrack->mSyncStartEvent->isCancelled()) { 5286 recordTrack->clearSyncStartEvent(); 5287 } else { 5288 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5289 recordTrack->mFramesToDrop = - 5290 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5291 } 5292 } 5293 5294 { 5295 // This section is a rendezvous between binder thread executing start() and RecordThread 5296 AutoMutex lock(mLock); 5297 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5298 if (recordTrack->mState == TrackBase::PAUSING) { 5299 ALOGV("active record track PAUSING -> ACTIVE"); 5300 recordTrack->mState = TrackBase::ACTIVE; 5301 } else { 5302 ALOGV("active record track state %d", recordTrack->mState); 5303 } 5304 return status; 5305 } 5306 5307 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5308 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5309 // or using a separate command thread 5310 recordTrack->mState = TrackBase::STARTING_1; 5311 mActiveTracks.add(recordTrack); 5312 mActiveTracksGen++; 5313 mLock.unlock(); 5314 status_t status = AudioSystem::startInput(mId); 5315 mLock.lock(); 5316 // FIXME should verify that recordTrack is still in mActiveTracks 5317 if (status != NO_ERROR) { 5318 mActiveTracks.remove(recordTrack); 5319 mActiveTracksGen++; 5320 recordTrack->clearSyncStartEvent(); 5321 return status; 5322 } 5323 // Catch up with current buffer indices if thread is already running. 5324 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5325 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5326 // see previously buffered data before it called start(), but with greater risk of overrun. 5327 5328 recordTrack->mRsmpInFront = mRsmpInRear; 5329 recordTrack->mRsmpInUnrel = 0; 5330 // FIXME why reset? 5331 if (recordTrack->mResampler != NULL) { 5332 recordTrack->mResampler->reset(); 5333 } 5334 recordTrack->mState = TrackBase::STARTING_2; 5335 // signal thread to start 5336 mWaitWorkCV.broadcast(); 5337 if (mActiveTracks.indexOf(recordTrack) < 0) { 5338 ALOGV("Record failed to start"); 5339 status = BAD_VALUE; 5340 goto startError; 5341 } 5342 return status; 5343 } 5344 5345startError: 5346 AudioSystem::stopInput(mId); 5347 recordTrack->clearSyncStartEvent(); 5348 // FIXME I wonder why we do not reset the state here? 5349 return status; 5350} 5351 5352void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5353{ 5354 sp<SyncEvent> strongEvent = event.promote(); 5355 5356 if (strongEvent != 0) { 5357 sp<RefBase> ptr = strongEvent->cookie().promote(); 5358 if (ptr != 0) { 5359 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5360 recordTrack->handleSyncStartEvent(strongEvent); 5361 } 5362 } 5363} 5364 5365bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5366 ALOGV("RecordThread::stop"); 5367 AutoMutex _l(mLock); 5368 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5369 return false; 5370 } 5371 // note that threadLoop may still be processing the track at this point [without lock] 5372 recordTrack->mState = TrackBase::PAUSING; 5373 // do not wait for mStartStopCond if exiting 5374 if (exitPending()) { 5375 return true; 5376 } 5377 // FIXME incorrect usage of wait: no explicit predicate or loop 5378 mStartStopCond.wait(mLock); 5379 // if we have been restarted, recordTrack is in mActiveTracks here 5380 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5381 ALOGV("Record stopped OK"); 5382 return true; 5383 } 5384 return false; 5385} 5386 5387bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5388{ 5389 return false; 5390} 5391 5392status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5393{ 5394#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5395 if (!isValidSyncEvent(event)) { 5396 return BAD_VALUE; 5397 } 5398 5399 int eventSession = event->triggerSession(); 5400 status_t ret = NAME_NOT_FOUND; 5401 5402 Mutex::Autolock _l(mLock); 5403 5404 for (size_t i = 0; i < mTracks.size(); i++) { 5405 sp<RecordTrack> track = mTracks[i]; 5406 if (eventSession == track->sessionId()) { 5407 (void) track->setSyncEvent(event); 5408 ret = NO_ERROR; 5409 } 5410 } 5411 return ret; 5412#else 5413 return BAD_VALUE; 5414#endif 5415} 5416 5417// destroyTrack_l() must be called with ThreadBase::mLock held 5418void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5419{ 5420 track->terminate(); 5421 track->mState = TrackBase::STOPPED; 5422 // active tracks are removed by threadLoop() 5423 if (mActiveTracks.indexOf(track) < 0) { 5424 removeTrack_l(track); 5425 } 5426} 5427 5428void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5429{ 5430 mTracks.remove(track); 5431 // need anything related to effects here? 5432} 5433 5434void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5435{ 5436 dumpInternals(fd, args); 5437 dumpTracks(fd, args); 5438 dumpEffectChains(fd, args); 5439} 5440 5441void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5442{ 5443 dprintf(fd, "\nInput thread %p:\n", this); 5444 5445 if (mActiveTracks.size() > 0) { 5446 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5447 } else { 5448 dprintf(fd, " No active record clients\n"); 5449 } 5450 5451 dumpBase(fd, args); 5452} 5453 5454void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5455{ 5456 const size_t SIZE = 256; 5457 char buffer[SIZE]; 5458 String8 result; 5459 5460 size_t numtracks = mTracks.size(); 5461 size_t numactive = mActiveTracks.size(); 5462 size_t numactiveseen = 0; 5463 dprintf(fd, " %d Tracks", numtracks); 5464 if (numtracks) { 5465 dprintf(fd, " of which %d are active\n", numactive); 5466 RecordTrack::appendDumpHeader(result); 5467 for (size_t i = 0; i < numtracks ; ++i) { 5468 sp<RecordTrack> track = mTracks[i]; 5469 if (track != 0) { 5470 bool active = mActiveTracks.indexOf(track) >= 0; 5471 if (active) { 5472 numactiveseen++; 5473 } 5474 track->dump(buffer, SIZE, active); 5475 result.append(buffer); 5476 } 5477 } 5478 } else { 5479 dprintf(fd, "\n"); 5480 } 5481 5482 if (numactiveseen != numactive) { 5483 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5484 " not in the track list\n"); 5485 result.append(buffer); 5486 RecordTrack::appendDumpHeader(result); 5487 for (size_t i = 0; i < numactive; ++i) { 5488 sp<RecordTrack> track = mActiveTracks[i]; 5489 if (mTracks.indexOf(track) < 0) { 5490 track->dump(buffer, SIZE, true); 5491 result.append(buffer); 5492 } 5493 } 5494 5495 } 5496 write(fd, result.string(), result.size()); 5497} 5498 5499// AudioBufferProvider interface 5500status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5501 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5502{ 5503 RecordTrack *activeTrack = mRecordTrack; 5504 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5505 if (threadBase == 0) { 5506 buffer->frameCount = 0; 5507 buffer->raw = NULL; 5508 return NOT_ENOUGH_DATA; 5509 } 5510 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5511 int32_t rear = recordThread->mRsmpInRear; 5512 int32_t front = activeTrack->mRsmpInFront; 5513 ssize_t filled = rear - front; 5514 // FIXME should not be P2 (don't want to increase latency) 5515 // FIXME if client not keeping up, discard 5516 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5517 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5518 front &= recordThread->mRsmpInFramesP2 - 1; 5519 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5520 if (part1 > (size_t) filled) { 5521 part1 = filled; 5522 } 5523 size_t ask = buffer->frameCount; 5524 ALOG_ASSERT(ask > 0); 5525 if (part1 > ask) { 5526 part1 = ask; 5527 } 5528 if (part1 == 0) { 5529 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5530 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5531 buffer->raw = NULL; 5532 buffer->frameCount = 0; 5533 activeTrack->mRsmpInUnrel = 0; 5534 return NOT_ENOUGH_DATA; 5535 } 5536 5537 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5538 buffer->frameCount = part1; 5539 activeTrack->mRsmpInUnrel = part1; 5540 return NO_ERROR; 5541} 5542 5543// AudioBufferProvider interface 5544void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5545 AudioBufferProvider::Buffer* buffer) 5546{ 5547 RecordTrack *activeTrack = mRecordTrack; 5548 size_t stepCount = buffer->frameCount; 5549 if (stepCount == 0) { 5550 return; 5551 } 5552 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5553 activeTrack->mRsmpInUnrel -= stepCount; 5554 activeTrack->mRsmpInFront += stepCount; 5555 buffer->raw = NULL; 5556 buffer->frameCount = 0; 5557} 5558 5559bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5560 status_t& status) 5561{ 5562 bool reconfig = false; 5563 5564 status = NO_ERROR; 5565 5566 audio_format_t reqFormat = mFormat; 5567 uint32_t samplingRate = mSampleRate; 5568 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5569 5570 AudioParameter param = AudioParameter(keyValuePair); 5571 int value; 5572 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5573 // channel count change can be requested. Do we mandate the first client defines the 5574 // HAL sampling rate and channel count or do we allow changes on the fly? 5575 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5576 samplingRate = value; 5577 reconfig = true; 5578 } 5579 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5580 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5581 status = BAD_VALUE; 5582 } else { 5583 reqFormat = (audio_format_t) value; 5584 reconfig = true; 5585 } 5586 } 5587 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5588 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5589 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5590 status = BAD_VALUE; 5591 } else { 5592 channelMask = mask; 5593 reconfig = true; 5594 } 5595 } 5596 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5597 // do not accept frame count changes if tracks are open as the track buffer 5598 // size depends on frame count and correct behavior would not be guaranteed 5599 // if frame count is changed after track creation 5600 if (mActiveTracks.size() > 0) { 5601 status = INVALID_OPERATION; 5602 } else { 5603 reconfig = true; 5604 } 5605 } 5606 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5607 // forward device change to effects that have requested to be 5608 // aware of attached audio device. 5609 for (size_t i = 0; i < mEffectChains.size(); i++) { 5610 mEffectChains[i]->setDevice_l(value); 5611 } 5612 5613 // store input device and output device but do not forward output device to audio HAL. 5614 // Note that status is ignored by the caller for output device 5615 // (see AudioFlinger::setParameters() 5616 if (audio_is_output_devices(value)) { 5617 mOutDevice = value; 5618 status = BAD_VALUE; 5619 } else { 5620 mInDevice = value; 5621 // disable AEC and NS if the device is a BT SCO headset supporting those 5622 // pre processings 5623 if (mTracks.size() > 0) { 5624 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5625 mAudioFlinger->btNrecIsOff(); 5626 for (size_t i = 0; i < mTracks.size(); i++) { 5627 sp<RecordTrack> track = mTracks[i]; 5628 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5629 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5630 } 5631 } 5632 } 5633 } 5634 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5635 mAudioSource != (audio_source_t)value) { 5636 // forward device change to effects that have requested to be 5637 // aware of attached audio device. 5638 for (size_t i = 0; i < mEffectChains.size(); i++) { 5639 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5640 } 5641 mAudioSource = (audio_source_t)value; 5642 } 5643 5644 if (status == NO_ERROR) { 5645 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5646 keyValuePair.string()); 5647 if (status == INVALID_OPERATION) { 5648 inputStandBy(); 5649 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5650 keyValuePair.string()); 5651 } 5652 if (reconfig) { 5653 if (status == BAD_VALUE && 5654 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5655 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5656 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5657 <= (2 * samplingRate)) && 5658 audio_channel_count_from_in_mask( 5659 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5660 (channelMask == AUDIO_CHANNEL_IN_MONO || 5661 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5662 status = NO_ERROR; 5663 } 5664 if (status == NO_ERROR) { 5665 readInputParameters_l(); 5666 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5667 } 5668 } 5669 } 5670 5671 return reconfig; 5672} 5673 5674String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5675{ 5676 Mutex::Autolock _l(mLock); 5677 if (initCheck() != NO_ERROR) { 5678 return String8(); 5679 } 5680 5681 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5682 const String8 out_s8(s); 5683 free(s); 5684 return out_s8; 5685} 5686 5687void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5688 AudioSystem::OutputDescriptor desc; 5689 const void *param2 = NULL; 5690 5691 switch (event) { 5692 case AudioSystem::INPUT_OPENED: 5693 case AudioSystem::INPUT_CONFIG_CHANGED: 5694 desc.channelMask = mChannelMask; 5695 desc.samplingRate = mSampleRate; 5696 desc.format = mFormat; 5697 desc.frameCount = mFrameCount; 5698 desc.latency = 0; 5699 param2 = &desc; 5700 break; 5701 5702 case AudioSystem::INPUT_CLOSED: 5703 default: 5704 break; 5705 } 5706 mAudioFlinger->audioConfigChanged(event, mId, param2); 5707} 5708 5709void AudioFlinger::RecordThread::readInputParameters_l() 5710{ 5711 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5712 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5713 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5714 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5715 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5716 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5717 } 5718 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5719 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5720 mFrameCount = mBufferSize / mFrameSize; 5721 // This is the formula for calculating the temporary buffer size. 5722 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5723 // 1 full output buffer, regardless of the alignment of the available input. 5724 // The value is somewhat arbitrary, and could probably be even larger. 5725 // A larger value should allow more old data to be read after a track calls start(), 5726 // without increasing latency. 5727 mRsmpInFrames = mFrameCount * 7; 5728 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5729 delete[] mRsmpInBuffer; 5730 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5731 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5732 5733 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5734 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5735} 5736 5737uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5738{ 5739 Mutex::Autolock _l(mLock); 5740 if (initCheck() != NO_ERROR) { 5741 return 0; 5742 } 5743 5744 return mInput->stream->get_input_frames_lost(mInput->stream); 5745} 5746 5747uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5748{ 5749 Mutex::Autolock _l(mLock); 5750 uint32_t result = 0; 5751 if (getEffectChain_l(sessionId) != 0) { 5752 result = EFFECT_SESSION; 5753 } 5754 5755 for (size_t i = 0; i < mTracks.size(); ++i) { 5756 if (sessionId == mTracks[i]->sessionId()) { 5757 result |= TRACK_SESSION; 5758 break; 5759 } 5760 } 5761 5762 return result; 5763} 5764 5765KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5766{ 5767 KeyedVector<int, bool> ids; 5768 Mutex::Autolock _l(mLock); 5769 for (size_t j = 0; j < mTracks.size(); ++j) { 5770 sp<RecordThread::RecordTrack> track = mTracks[j]; 5771 int sessionId = track->sessionId(); 5772 if (ids.indexOfKey(sessionId) < 0) { 5773 ids.add(sessionId, true); 5774 } 5775 } 5776 return ids; 5777} 5778 5779AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5780{ 5781 Mutex::Autolock _l(mLock); 5782 AudioStreamIn *input = mInput; 5783 mInput = NULL; 5784 return input; 5785} 5786 5787// this method must always be called either with ThreadBase mLock held or inside the thread loop 5788audio_stream_t* AudioFlinger::RecordThread::stream() const 5789{ 5790 if (mInput == NULL) { 5791 return NULL; 5792 } 5793 return &mInput->stream->common; 5794} 5795 5796status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5797{ 5798 // only one chain per input thread 5799 if (mEffectChains.size() != 0) { 5800 return INVALID_OPERATION; 5801 } 5802 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5803 5804 chain->setInBuffer(NULL); 5805 chain->setOutBuffer(NULL); 5806 5807 checkSuspendOnAddEffectChain_l(chain); 5808 5809 mEffectChains.add(chain); 5810 5811 return NO_ERROR; 5812} 5813 5814size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5815{ 5816 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5817 ALOGW_IF(mEffectChains.size() != 1, 5818 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5819 chain.get(), mEffectChains.size(), this); 5820 if (mEffectChains.size() == 1) { 5821 mEffectChains.removeAt(0); 5822 } 5823 return 0; 5824} 5825 5826status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 5827 audio_patch_handle_t *handle) 5828{ 5829 status_t status = NO_ERROR; 5830 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 5831 // store new device and send to effects 5832 mInDevice = patch->sources[0].ext.device.type; 5833 for (size_t i = 0; i < mEffectChains.size(); i++) { 5834 mEffectChains[i]->setDevice_l(mInDevice); 5835 } 5836 5837 // disable AEC and NS if the device is a BT SCO headset supporting those 5838 // pre processings 5839 if (mTracks.size() > 0) { 5840 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5841 mAudioFlinger->btNrecIsOff(); 5842 for (size_t i = 0; i < mTracks.size(); i++) { 5843 sp<RecordTrack> track = mTracks[i]; 5844 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5845 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5846 } 5847 } 5848 5849 // store new source and send to effects 5850 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 5851 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 5852 for (size_t i = 0; i < mEffectChains.size(); i++) { 5853 mEffectChains[i]->setAudioSource_l(mAudioSource); 5854 } 5855 } 5856 5857 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 5858 status = hwDevice->create_audio_patch(hwDevice, 5859 patch->num_sources, 5860 patch->sources, 5861 patch->num_sinks, 5862 patch->sinks, 5863 handle); 5864 } else { 5865 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 5866 } 5867 return status; 5868} 5869 5870status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 5871{ 5872 status_t status = NO_ERROR; 5873 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 5874 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 5875 status = hwDevice->release_audio_patch(hwDevice, handle); 5876 } else { 5877 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 5878 } 5879 return status; 5880} 5881 5882 5883}; // namespace android 5884