Threads.cpp revision 1258c1ab592a899fabb1e31eb5db2ef413b6f38a
1/* 2** 3** Copyright 2012, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21#define ATRACE_TAG ATRACE_TAG_AUDIO 22 23#include "Configuration.h" 24#include <math.h> 25#include <fcntl.h> 26#include <sys/stat.h> 27#include <cutils/properties.h> 28#include <media/AudioParameter.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31 32#include <private/media/AudioTrackShared.h> 33#include <hardware/audio.h> 34#include <audio_effects/effect_ns.h> 35#include <audio_effects/effect_aec.h> 36#include <audio_utils/primitives.h> 37#include <audio_utils/format.h> 38#include <audio_utils/minifloat.h> 39 40// NBAIO implementations 41#include <media/nbaio/AudioStreamOutSink.h> 42#include <media/nbaio/MonoPipe.h> 43#include <media/nbaio/MonoPipeReader.h> 44#include <media/nbaio/Pipe.h> 45#include <media/nbaio/PipeReader.h> 46#include <media/nbaio/SourceAudioBufferProvider.h> 47 48#include <powermanager/PowerManager.h> 49 50#include <common_time/cc_helper.h> 51#include <common_time/local_clock.h> 52 53#include "AudioFlinger.h" 54#include "AudioMixer.h" 55#include "FastMixer.h" 56#include "ServiceUtilities.h" 57#include "SchedulingPolicyService.h" 58 59#ifdef ADD_BATTERY_DATA 60#include <media/IMediaPlayerService.h> 61#include <media/IMediaDeathNotifier.h> 62#endif 63 64#ifdef DEBUG_CPU_USAGE 65#include <cpustats/CentralTendencyStatistics.h> 66#include <cpustats/ThreadCpuUsage.h> 67#endif 68 69// ---------------------------------------------------------------------------- 70 71// Note: the following macro is used for extremely verbose logging message. In 72// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 73// 0; but one side effect of this is to turn all LOGV's as well. Some messages 74// are so verbose that we want to suppress them even when we have ALOG_ASSERT 75// turned on. Do not uncomment the #def below unless you really know what you 76// are doing and want to see all of the extremely verbose messages. 77//#define VERY_VERY_VERBOSE_LOGGING 78#ifdef VERY_VERY_VERBOSE_LOGGING 79#define ALOGVV ALOGV 80#else 81#define ALOGVV(a...) do { } while(0) 82#endif 83 84namespace android { 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95// don't warn about blocked writes or record buffer overflows more often than this 96static const nsecs_t kWarningThrottleNs = seconds(5); 97 98// RecordThread loop sleep time upon application overrun or audio HAL read error 99static const int kRecordThreadSleepUs = 5000; 100 101// maximum time to wait in sendConfigEvent_l() for a status to be received 102static const nsecs_t kConfigEventTimeoutNs = seconds(2); 103 104// minimum sleep time for the mixer thread loop when tracks are active but in underrun 105static const uint32_t kMinThreadSleepTimeUs = 5000; 106// maximum divider applied to the active sleep time in the mixer thread loop 107static const uint32_t kMaxThreadSleepTimeShift = 2; 108 109// minimum normal sink buffer size, expressed in milliseconds rather than frames 110static const uint32_t kMinNormalSinkBufferSizeMs = 20; 111// maximum normal sink buffer size 112static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 113 114// Offloaded output thread standby delay: allows track transition without going to standby 115static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 116 117// Whether to use fast mixer 118static const enum { 119 FastMixer_Never, // never initialize or use: for debugging only 120 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 121 // normal mixer multiplier is 1 122 FastMixer_Static, // initialize if needed, then use all the time if initialized, 123 // multiplier is calculated based on min & max normal mixer buffer size 124 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 125 // multiplier is calculated based on min & max normal mixer buffer size 126 // FIXME for FastMixer_Dynamic: 127 // Supporting this option will require fixing HALs that can't handle large writes. 128 // For example, one HAL implementation returns an error from a large write, 129 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 130 // We could either fix the HAL implementations, or provide a wrapper that breaks 131 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 132} kUseFastMixer = FastMixer_Static; 133 134// Priorities for requestPriority 135static const int kPriorityAudioApp = 2; 136static const int kPriorityFastMixer = 3; 137 138// IAudioFlinger::createTrack() reports back to client the total size of shared memory area 139// for the track. The client then sub-divides this into smaller buffers for its use. 140// Currently the client uses N-buffering by default, but doesn't tell us about the value of N. 141// So for now we just assume that client is double-buffered for fast tracks. 142// FIXME It would be better for client to tell AudioFlinger the value of N, 143// so AudioFlinger could allocate the right amount of memory. 144// See the client's minBufCount and mNotificationFramesAct calculations for details. 145 146// This is the default value, if not specified by property. 147static const int kFastTrackMultiplier = 2; 148 149// The minimum and maximum allowed values 150static const int kFastTrackMultiplierMin = 1; 151static const int kFastTrackMultiplierMax = 2; 152 153// The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 154static int sFastTrackMultiplier = kFastTrackMultiplier; 155 156// See Thread::readOnlyHeap(). 157// Initially this heap is used to allocate client buffers for "fast" AudioRecord. 158// Eventually it will be the single buffer that FastCapture writes into via HAL read(), 159// and that all "fast" AudioRecord clients read from. In either case, the size can be small. 160static const size_t kRecordThreadReadOnlyHeapSize = 0x1000; 161 162// ---------------------------------------------------------------------------- 163 164static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 165 166static void sFastTrackMultiplierInit() 167{ 168 char value[PROPERTY_VALUE_MAX]; 169 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 170 char *endptr; 171 unsigned long ul = strtoul(value, &endptr, 0); 172 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 173 sFastTrackMultiplier = (int) ul; 174 } 175 } 176} 177 178// ---------------------------------------------------------------------------- 179 180#ifdef ADD_BATTERY_DATA 181// To collect the amplifier usage 182static void addBatteryData(uint32_t params) { 183 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 184 if (service == NULL) { 185 // it already logged 186 return; 187 } 188 189 service->addBatteryData(params); 190} 191#endif 192 193 194// ---------------------------------------------------------------------------- 195// CPU Stats 196// ---------------------------------------------------------------------------- 197 198class CpuStats { 199public: 200 CpuStats(); 201 void sample(const String8 &title); 202#ifdef DEBUG_CPU_USAGE 203private: 204 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 205 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 206 207 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 208 209 int mCpuNum; // thread's current CPU number 210 int mCpukHz; // frequency of thread's current CPU in kHz 211#endif 212}; 213 214CpuStats::CpuStats() 215#ifdef DEBUG_CPU_USAGE 216 : mCpuNum(-1), mCpukHz(-1) 217#endif 218{ 219} 220 221void CpuStats::sample(const String8 &title 222#ifndef DEBUG_CPU_USAGE 223 __unused 224#endif 225 ) { 226#ifdef DEBUG_CPU_USAGE 227 // get current thread's delta CPU time in wall clock ns 228 double wcNs; 229 bool valid = mCpuUsage.sampleAndEnable(wcNs); 230 231 // record sample for wall clock statistics 232 if (valid) { 233 mWcStats.sample(wcNs); 234 } 235 236 // get the current CPU number 237 int cpuNum = sched_getcpu(); 238 239 // get the current CPU frequency in kHz 240 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 241 242 // check if either CPU number or frequency changed 243 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 244 mCpuNum = cpuNum; 245 mCpukHz = cpukHz; 246 // ignore sample for purposes of cycles 247 valid = false; 248 } 249 250 // if no change in CPU number or frequency, then record sample for cycle statistics 251 if (valid && mCpukHz > 0) { 252 double cycles = wcNs * cpukHz * 0.000001; 253 mHzStats.sample(cycles); 254 } 255 256 unsigned n = mWcStats.n(); 257 // mCpuUsage.elapsed() is expensive, so don't call it every loop 258 if ((n & 127) == 1) { 259 long long elapsed = mCpuUsage.elapsed(); 260 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 261 double perLoop = elapsed / (double) n; 262 double perLoop100 = perLoop * 0.01; 263 double perLoop1k = perLoop * 0.001; 264 double mean = mWcStats.mean(); 265 double stddev = mWcStats.stddev(); 266 double minimum = mWcStats.minimum(); 267 double maximum = mWcStats.maximum(); 268 double meanCycles = mHzStats.mean(); 269 double stddevCycles = mHzStats.stddev(); 270 double minCycles = mHzStats.minimum(); 271 double maxCycles = mHzStats.maximum(); 272 mCpuUsage.resetElapsed(); 273 mWcStats.reset(); 274 mHzStats.reset(); 275 ALOGD("CPU usage for %s over past %.1f secs\n" 276 " (%u mixer loops at %.1f mean ms per loop):\n" 277 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 278 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 279 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 280 title.string(), 281 elapsed * .000000001, n, perLoop * .000001, 282 mean * .001, 283 stddev * .001, 284 minimum * .001, 285 maximum * .001, 286 mean / perLoop100, 287 stddev / perLoop100, 288 minimum / perLoop100, 289 maximum / perLoop100, 290 meanCycles / perLoop1k, 291 stddevCycles / perLoop1k, 292 minCycles / perLoop1k, 293 maxCycles / perLoop1k); 294 295 } 296 } 297#endif 298}; 299 300// ---------------------------------------------------------------------------- 301// ThreadBase 302// ---------------------------------------------------------------------------- 303 304AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 305 audio_devices_t outDevice, audio_devices_t inDevice, type_t type) 306 : Thread(false /*canCallJava*/), 307 mType(type), 308 mAudioFlinger(audioFlinger), 309 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 310 // are set by PlaybackThread::readOutputParameters_l() or 311 // RecordThread::readInputParameters_l() 312 //FIXME: mStandby should be true here. Is this some kind of hack? 313 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 314 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 315 // mName will be set by concrete (non-virtual) subclass 316 mDeathRecipient(new PMDeathRecipient(this)) 317{ 318} 319 320AudioFlinger::ThreadBase::~ThreadBase() 321{ 322 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 323 mConfigEvents.clear(); 324 325 // do not lock the mutex in destructor 326 releaseWakeLock_l(); 327 if (mPowerManager != 0) { 328 sp<IBinder> binder = mPowerManager->asBinder(); 329 binder->unlinkToDeath(mDeathRecipient); 330 } 331} 332 333status_t AudioFlinger::ThreadBase::readyToRun() 334{ 335 status_t status = initCheck(); 336 if (status == NO_ERROR) { 337 ALOGI("AudioFlinger's thread %p ready to run", this); 338 } else { 339 ALOGE("No working audio driver found."); 340 } 341 return status; 342} 343 344void AudioFlinger::ThreadBase::exit() 345{ 346 ALOGV("ThreadBase::exit"); 347 // do any cleanup required for exit to succeed 348 preExit(); 349 { 350 // This lock prevents the following race in thread (uniprocessor for illustration): 351 // if (!exitPending()) { 352 // // context switch from here to exit() 353 // // exit() calls requestExit(), what exitPending() observes 354 // // exit() calls signal(), which is dropped since no waiters 355 // // context switch back from exit() to here 356 // mWaitWorkCV.wait(...); 357 // // now thread is hung 358 // } 359 AutoMutex lock(mLock); 360 requestExit(); 361 mWaitWorkCV.broadcast(); 362 } 363 // When Thread::requestExitAndWait is made virtual and this method is renamed to 364 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 365 requestExitAndWait(); 366} 367 368status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 369{ 370 status_t status; 371 372 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 373 Mutex::Autolock _l(mLock); 374 375 return sendSetParameterConfigEvent_l(keyValuePairs); 376} 377 378// sendConfigEvent_l() must be called with ThreadBase::mLock held 379// Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 380status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 381{ 382 status_t status = NO_ERROR; 383 384 mConfigEvents.add(event); 385 ALOGV("sendConfigEvent_l() num events %d event %d", mConfigEvents.size(), event->mType); 386 mWaitWorkCV.signal(); 387 mLock.unlock(); 388 { 389 Mutex::Autolock _l(event->mLock); 390 while (event->mWaitStatus) { 391 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 392 event->mStatus = TIMED_OUT; 393 event->mWaitStatus = false; 394 } 395 } 396 status = event->mStatus; 397 } 398 mLock.lock(); 399 return status; 400} 401 402void AudioFlinger::ThreadBase::sendIoConfigEvent(int event, int param) 403{ 404 Mutex::Autolock _l(mLock); 405 sendIoConfigEvent_l(event, param); 406} 407 408// sendIoConfigEvent_l() must be called with ThreadBase::mLock held 409void AudioFlinger::ThreadBase::sendIoConfigEvent_l(int event, int param) 410{ 411 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, param); 412 sendConfigEvent_l(configEvent); 413} 414 415// sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 416void AudioFlinger::ThreadBase::sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio) 417{ 418 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio); 419 sendConfigEvent_l(configEvent); 420} 421 422// sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 423status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 424{ 425 sp<ConfigEvent> configEvent = (ConfigEvent *)new SetParameterConfigEvent(keyValuePair); 426 return sendConfigEvent_l(configEvent); 427} 428 429status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 430 const struct audio_patch *patch, 431 audio_patch_handle_t *handle) 432{ 433 Mutex::Autolock _l(mLock); 434 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 435 status_t status = sendConfigEvent_l(configEvent); 436 if (status == NO_ERROR) { 437 CreateAudioPatchConfigEventData *data = 438 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 439 *handle = data->mHandle; 440 } 441 return status; 442} 443 444status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 445 const audio_patch_handle_t handle) 446{ 447 Mutex::Autolock _l(mLock); 448 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 449 return sendConfigEvent_l(configEvent); 450} 451 452 453// post condition: mConfigEvents.isEmpty() 454void AudioFlinger::ThreadBase::processConfigEvents_l() 455{ 456 bool configChanged = false; 457 458 while (!mConfigEvents.isEmpty()) { 459 ALOGV("processConfigEvents_l() remaining events %d", mConfigEvents.size()); 460 sp<ConfigEvent> event = mConfigEvents[0]; 461 mConfigEvents.removeAt(0); 462 switch (event->mType) { 463 case CFG_EVENT_PRIO: { 464 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 465 // FIXME Need to understand why this has to be done asynchronously 466 int err = requestPriority(data->mPid, data->mTid, data->mPrio, 467 true /*asynchronous*/); 468 if (err != 0) { 469 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 470 data->mPrio, data->mPid, data->mTid, err); 471 } 472 } break; 473 case CFG_EVENT_IO: { 474 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 475 audioConfigChanged(data->mEvent, data->mParam); 476 } break; 477 case CFG_EVENT_SET_PARAMETER: { 478 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 479 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 480 configChanged = true; 481 } 482 } break; 483 case CFG_EVENT_CREATE_AUDIO_PATCH: { 484 CreateAudioPatchConfigEventData *data = 485 (CreateAudioPatchConfigEventData *)event->mData.get(); 486 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 487 } break; 488 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 489 ReleaseAudioPatchConfigEventData *data = 490 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 491 event->mStatus = releaseAudioPatch_l(data->mHandle); 492 } break; 493 default: 494 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 495 break; 496 } 497 { 498 Mutex::Autolock _l(event->mLock); 499 if (event->mWaitStatus) { 500 event->mWaitStatus = false; 501 event->mCond.signal(); 502 } 503 } 504 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 505 } 506 507 if (configChanged) { 508 cacheParameters_l(); 509 } 510} 511 512String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 513 String8 s; 514 if (output) { 515 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 516 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 517 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 518 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 519 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 520 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 521 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 522 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 523 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 524 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 525 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 526 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 527 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 528 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 529 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 530 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 531 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 532 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 533 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 534 } else { 535 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 536 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 537 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 538 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 539 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 540 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 541 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 542 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 543 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 544 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 545 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 546 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 547 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 548 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 549 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 550 } 551 int len = s.length(); 552 if (s.length() > 2) { 553 char *str = s.lockBuffer(len); 554 s.unlockBuffer(len - 2); 555 } 556 return s; 557} 558 559void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 560{ 561 const size_t SIZE = 256; 562 char buffer[SIZE]; 563 String8 result; 564 565 bool locked = AudioFlinger::dumpTryLock(mLock); 566 if (!locked) { 567 dprintf(fd, "thread %p maybe dead locked\n", this); 568 } 569 570 dprintf(fd, " I/O handle: %d\n", mId); 571 dprintf(fd, " TID: %d\n", getTid()); 572 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 573 dprintf(fd, " Sample rate: %u\n", mSampleRate); 574 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 575 dprintf(fd, " HAL buffer size: %u bytes\n", mBufferSize); 576 dprintf(fd, " Channel Count: %u\n", mChannelCount); 577 dprintf(fd, " Channel Mask: 0x%08x (%s)\n", mChannelMask, 578 channelMaskToString(mChannelMask, mType != RECORD).string()); 579 dprintf(fd, " Format: 0x%x (%s)\n", mFormat, formatToString(mFormat)); 580 dprintf(fd, " Frame size: %zu\n", mFrameSize); 581 dprintf(fd, " Pending config events:"); 582 size_t numConfig = mConfigEvents.size(); 583 if (numConfig) { 584 for (size_t i = 0; i < numConfig; i++) { 585 mConfigEvents[i]->dump(buffer, SIZE); 586 dprintf(fd, "\n %s", buffer); 587 } 588 dprintf(fd, "\n"); 589 } else { 590 dprintf(fd, " none\n"); 591 } 592 593 if (locked) { 594 mLock.unlock(); 595 } 596} 597 598void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 599{ 600 const size_t SIZE = 256; 601 char buffer[SIZE]; 602 String8 result; 603 604 size_t numEffectChains = mEffectChains.size(); 605 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 606 write(fd, buffer, strlen(buffer)); 607 608 for (size_t i = 0; i < numEffectChains; ++i) { 609 sp<EffectChain> chain = mEffectChains[i]; 610 if (chain != 0) { 611 chain->dump(fd, args); 612 } 613 } 614} 615 616void AudioFlinger::ThreadBase::acquireWakeLock(int uid) 617{ 618 Mutex::Autolock _l(mLock); 619 acquireWakeLock_l(uid); 620} 621 622String16 AudioFlinger::ThreadBase::getWakeLockTag() 623{ 624 switch (mType) { 625 case MIXER: 626 return String16("AudioMix"); 627 case DIRECT: 628 return String16("AudioDirectOut"); 629 case DUPLICATING: 630 return String16("AudioDup"); 631 case RECORD: 632 return String16("AudioIn"); 633 case OFFLOAD: 634 return String16("AudioOffload"); 635 default: 636 ALOG_ASSERT(false); 637 return String16("AudioUnknown"); 638 } 639} 640 641void AudioFlinger::ThreadBase::acquireWakeLock_l(int uid) 642{ 643 getPowerManager_l(); 644 if (mPowerManager != 0) { 645 sp<IBinder> binder = new BBinder(); 646 status_t status; 647 if (uid >= 0) { 648 status = mPowerManager->acquireWakeLockWithUid(POWERMANAGER_PARTIAL_WAKE_LOCK, 649 binder, 650 getWakeLockTag(), 651 String16("media"), 652 uid); 653 } else { 654 status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 655 binder, 656 getWakeLockTag(), 657 String16("media")); 658 } 659 if (status == NO_ERROR) { 660 mWakeLockToken = binder; 661 } 662 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 663 } 664} 665 666void AudioFlinger::ThreadBase::releaseWakeLock() 667{ 668 Mutex::Autolock _l(mLock); 669 releaseWakeLock_l(); 670} 671 672void AudioFlinger::ThreadBase::releaseWakeLock_l() 673{ 674 if (mWakeLockToken != 0) { 675 ALOGV("releaseWakeLock_l() %s", mName); 676 if (mPowerManager != 0) { 677 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 678 } 679 mWakeLockToken.clear(); 680 } 681} 682 683void AudioFlinger::ThreadBase::updateWakeLockUids(const SortedVector<int> &uids) { 684 Mutex::Autolock _l(mLock); 685 updateWakeLockUids_l(uids); 686} 687 688void AudioFlinger::ThreadBase::getPowerManager_l() { 689 690 if (mPowerManager == 0) { 691 // use checkService() to avoid blocking if power service is not up yet 692 sp<IBinder> binder = 693 defaultServiceManager()->checkService(String16("power")); 694 if (binder == 0) { 695 ALOGW("Thread %s cannot connect to the power manager service", mName); 696 } else { 697 mPowerManager = interface_cast<IPowerManager>(binder); 698 binder->linkToDeath(mDeathRecipient); 699 } 700 } 701} 702 703void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<int> &uids) { 704 705 getPowerManager_l(); 706 if (mWakeLockToken == NULL) { 707 ALOGE("no wake lock to update!"); 708 return; 709 } 710 if (mPowerManager != 0) { 711 sp<IBinder> binder = new BBinder(); 712 status_t status; 713 status = mPowerManager->updateWakeLockUids(mWakeLockToken, uids.size(), uids.array()); 714 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 715 } 716} 717 718void AudioFlinger::ThreadBase::clearPowerManager() 719{ 720 Mutex::Autolock _l(mLock); 721 releaseWakeLock_l(); 722 mPowerManager.clear(); 723} 724 725void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 726{ 727 sp<ThreadBase> thread = mThread.promote(); 728 if (thread != 0) { 729 thread->clearPowerManager(); 730 } 731 ALOGW("power manager service died !!!"); 732} 733 734void AudioFlinger::ThreadBase::setEffectSuspended( 735 const effect_uuid_t *type, bool suspend, int sessionId) 736{ 737 Mutex::Autolock _l(mLock); 738 setEffectSuspended_l(type, suspend, sessionId); 739} 740 741void AudioFlinger::ThreadBase::setEffectSuspended_l( 742 const effect_uuid_t *type, bool suspend, int sessionId) 743{ 744 sp<EffectChain> chain = getEffectChain_l(sessionId); 745 if (chain != 0) { 746 if (type != NULL) { 747 chain->setEffectSuspended_l(type, suspend); 748 } else { 749 chain->setEffectSuspendedAll_l(suspend); 750 } 751 } 752 753 updateSuspendedSessions_l(type, suspend, sessionId); 754} 755 756void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 757{ 758 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 759 if (index < 0) { 760 return; 761 } 762 763 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 764 mSuspendedSessions.valueAt(index); 765 766 for (size_t i = 0; i < sessionEffects.size(); i++) { 767 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 768 for (int j = 0; j < desc->mRefCount; j++) { 769 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 770 chain->setEffectSuspendedAll_l(true); 771 } else { 772 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 773 desc->mType.timeLow); 774 chain->setEffectSuspended_l(&desc->mType, true); 775 } 776 } 777 } 778} 779 780void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 781 bool suspend, 782 int sessionId) 783{ 784 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 785 786 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 787 788 if (suspend) { 789 if (index >= 0) { 790 sessionEffects = mSuspendedSessions.valueAt(index); 791 } else { 792 mSuspendedSessions.add(sessionId, sessionEffects); 793 } 794 } else { 795 if (index < 0) { 796 return; 797 } 798 sessionEffects = mSuspendedSessions.valueAt(index); 799 } 800 801 802 int key = EffectChain::kKeyForSuspendAll; 803 if (type != NULL) { 804 key = type->timeLow; 805 } 806 index = sessionEffects.indexOfKey(key); 807 808 sp<SuspendedSessionDesc> desc; 809 if (suspend) { 810 if (index >= 0) { 811 desc = sessionEffects.valueAt(index); 812 } else { 813 desc = new SuspendedSessionDesc(); 814 if (type != NULL) { 815 desc->mType = *type; 816 } 817 sessionEffects.add(key, desc); 818 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 819 } 820 desc->mRefCount++; 821 } else { 822 if (index < 0) { 823 return; 824 } 825 desc = sessionEffects.valueAt(index); 826 if (--desc->mRefCount == 0) { 827 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 828 sessionEffects.removeItemsAt(index); 829 if (sessionEffects.isEmpty()) { 830 ALOGV("updateSuspendedSessions_l() restore removing session %d", 831 sessionId); 832 mSuspendedSessions.removeItem(sessionId); 833 } 834 } 835 } 836 if (!sessionEffects.isEmpty()) { 837 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 838 } 839} 840 841void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 842 bool enabled, 843 int sessionId) 844{ 845 Mutex::Autolock _l(mLock); 846 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 847} 848 849void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 850 bool enabled, 851 int sessionId) 852{ 853 if (mType != RECORD) { 854 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 855 // another session. This gives the priority to well behaved effect control panels 856 // and applications not using global effects. 857 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 858 // global effects 859 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 860 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 861 } 862 } 863 864 sp<EffectChain> chain = getEffectChain_l(sessionId); 865 if (chain != 0) { 866 chain->checkSuspendOnEffectEnabled(effect, enabled); 867 } 868} 869 870// ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 871sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 872 const sp<AudioFlinger::Client>& client, 873 const sp<IEffectClient>& effectClient, 874 int32_t priority, 875 int sessionId, 876 effect_descriptor_t *desc, 877 int *enabled, 878 status_t *status) 879{ 880 sp<EffectModule> effect; 881 sp<EffectHandle> handle; 882 status_t lStatus; 883 sp<EffectChain> chain; 884 bool chainCreated = false; 885 bool effectCreated = false; 886 bool effectRegistered = false; 887 888 lStatus = initCheck(); 889 if (lStatus != NO_ERROR) { 890 ALOGW("createEffect_l() Audio driver not initialized."); 891 goto Exit; 892 } 893 894 // Reject any effect on Direct output threads for now, since the format of 895 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 896 if (mType == DIRECT) { 897 ALOGW("createEffect_l() Cannot add effect %s on Direct output type thread %s", 898 desc->name, mName); 899 lStatus = BAD_VALUE; 900 goto Exit; 901 } 902 903 // Allow global effects only on offloaded and mixer threads 904 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 905 switch (mType) { 906 case MIXER: 907 case OFFLOAD: 908 break; 909 case DIRECT: 910 case DUPLICATING: 911 case RECORD: 912 default: 913 ALOGW("createEffect_l() Cannot add global effect %s on thread %s", desc->name, mName); 914 lStatus = BAD_VALUE; 915 goto Exit; 916 } 917 } 918 919 // Only Pre processor effects are allowed on input threads and only on input threads 920 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 921 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 922 desc->name, desc->flags, mType); 923 lStatus = BAD_VALUE; 924 goto Exit; 925 } 926 927 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 928 929 { // scope for mLock 930 Mutex::Autolock _l(mLock); 931 932 // check for existing effect chain with the requested audio session 933 chain = getEffectChain_l(sessionId); 934 if (chain == 0) { 935 // create a new chain for this session 936 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 937 chain = new EffectChain(this, sessionId); 938 addEffectChain_l(chain); 939 chain->setStrategy(getStrategyForSession_l(sessionId)); 940 chainCreated = true; 941 } else { 942 effect = chain->getEffectFromDesc_l(desc); 943 } 944 945 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 946 947 if (effect == 0) { 948 int id = mAudioFlinger->nextUniqueId(); 949 // Check CPU and memory usage 950 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 951 if (lStatus != NO_ERROR) { 952 goto Exit; 953 } 954 effectRegistered = true; 955 // create a new effect module if none present in the chain 956 effect = new EffectModule(this, chain, desc, id, sessionId); 957 lStatus = effect->status(); 958 if (lStatus != NO_ERROR) { 959 goto Exit; 960 } 961 effect->setOffloaded(mType == OFFLOAD, mId); 962 963 lStatus = chain->addEffect_l(effect); 964 if (lStatus != NO_ERROR) { 965 goto Exit; 966 } 967 effectCreated = true; 968 969 effect->setDevice(mOutDevice); 970 effect->setDevice(mInDevice); 971 effect->setMode(mAudioFlinger->getMode()); 972 effect->setAudioSource(mAudioSource); 973 } 974 // create effect handle and connect it to effect module 975 handle = new EffectHandle(effect, client, effectClient, priority); 976 lStatus = handle->initCheck(); 977 if (lStatus == OK) { 978 lStatus = effect->addHandle(handle.get()); 979 } 980 if (enabled != NULL) { 981 *enabled = (int)effect->isEnabled(); 982 } 983 } 984 985Exit: 986 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 987 Mutex::Autolock _l(mLock); 988 if (effectCreated) { 989 chain->removeEffect_l(effect); 990 } 991 if (effectRegistered) { 992 AudioSystem::unregisterEffect(effect->id()); 993 } 994 if (chainCreated) { 995 removeEffectChain_l(chain); 996 } 997 handle.clear(); 998 } 999 1000 *status = lStatus; 1001 return handle; 1002} 1003 1004sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 1005{ 1006 Mutex::Autolock _l(mLock); 1007 return getEffect_l(sessionId, effectId); 1008} 1009 1010sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 1011{ 1012 sp<EffectChain> chain = getEffectChain_l(sessionId); 1013 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1014} 1015 1016// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1017// PlaybackThread::mLock held 1018status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1019{ 1020 // check for existing effect chain with the requested audio session 1021 int sessionId = effect->sessionId(); 1022 sp<EffectChain> chain = getEffectChain_l(sessionId); 1023 bool chainCreated = false; 1024 1025 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1026 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1027 this, effect->desc().name, effect->desc().flags); 1028 1029 if (chain == 0) { 1030 // create a new chain for this session 1031 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1032 chain = new EffectChain(this, sessionId); 1033 addEffectChain_l(chain); 1034 chain->setStrategy(getStrategyForSession_l(sessionId)); 1035 chainCreated = true; 1036 } 1037 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1038 1039 if (chain->getEffectFromId_l(effect->id()) != 0) { 1040 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1041 this, effect->desc().name, chain.get()); 1042 return BAD_VALUE; 1043 } 1044 1045 effect->setOffloaded(mType == OFFLOAD, mId); 1046 1047 status_t status = chain->addEffect_l(effect); 1048 if (status != NO_ERROR) { 1049 if (chainCreated) { 1050 removeEffectChain_l(chain); 1051 } 1052 return status; 1053 } 1054 1055 effect->setDevice(mOutDevice); 1056 effect->setDevice(mInDevice); 1057 effect->setMode(mAudioFlinger->getMode()); 1058 effect->setAudioSource(mAudioSource); 1059 return NO_ERROR; 1060} 1061 1062void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 1063 1064 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 1065 effect_descriptor_t desc = effect->desc(); 1066 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1067 detachAuxEffect_l(effect->id()); 1068 } 1069 1070 sp<EffectChain> chain = effect->chain().promote(); 1071 if (chain != 0) { 1072 // remove effect chain if removing last effect 1073 if (chain->removeEffect_l(effect) == 0) { 1074 removeEffectChain_l(chain); 1075 } 1076 } else { 1077 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1078 } 1079} 1080 1081void AudioFlinger::ThreadBase::lockEffectChains_l( 1082 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1083{ 1084 effectChains = mEffectChains; 1085 for (size_t i = 0; i < mEffectChains.size(); i++) { 1086 mEffectChains[i]->lock(); 1087 } 1088} 1089 1090void AudioFlinger::ThreadBase::unlockEffectChains( 1091 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1092{ 1093 for (size_t i = 0; i < effectChains.size(); i++) { 1094 effectChains[i]->unlock(); 1095 } 1096} 1097 1098sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 1099{ 1100 Mutex::Autolock _l(mLock); 1101 return getEffectChain_l(sessionId); 1102} 1103 1104sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) const 1105{ 1106 size_t size = mEffectChains.size(); 1107 for (size_t i = 0; i < size; i++) { 1108 if (mEffectChains[i]->sessionId() == sessionId) { 1109 return mEffectChains[i]; 1110 } 1111 } 1112 return 0; 1113} 1114 1115void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1116{ 1117 Mutex::Autolock _l(mLock); 1118 size_t size = mEffectChains.size(); 1119 for (size_t i = 0; i < size; i++) { 1120 mEffectChains[i]->setMode_l(mode); 1121 } 1122} 1123 1124void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 1125 EffectHandle *handle, 1126 bool unpinIfLast) { 1127 1128 Mutex::Autolock _l(mLock); 1129 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 1130 // delete the effect module if removing last handle on it 1131 if (effect->removeHandle(handle) == 0) { 1132 if (!effect->isPinned() || unpinIfLast) { 1133 removeEffect_l(effect); 1134 AudioSystem::unregisterEffect(effect->id()); 1135 } 1136 } 1137} 1138 1139// ---------------------------------------------------------------------------- 1140// Playback 1141// ---------------------------------------------------------------------------- 1142 1143AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1144 AudioStreamOut* output, 1145 audio_io_handle_t id, 1146 audio_devices_t device, 1147 type_t type) 1148 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type), 1149 mNormalFrameCount(0), mSinkBuffer(NULL), 1150 mMixerBufferEnabled(false), 1151 mMixerBuffer(NULL), 1152 mMixerBufferSize(0), 1153 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1154 mMixerBufferValid(false), 1155 mEffectBufferEnabled(false), 1156 mEffectBuffer(NULL), 1157 mEffectBufferSize(0), 1158 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1159 mEffectBufferValid(false), 1160 mSuspended(0), mBytesWritten(0), 1161 mActiveTracksGeneration(0), 1162 // mStreamTypes[] initialized in constructor body 1163 mOutput(output), 1164 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1165 mMixerStatus(MIXER_IDLE), 1166 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1167 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1168 mBytesRemaining(0), 1169 mCurrentWriteLength(0), 1170 mUseAsyncWrite(false), 1171 mWriteAckSequence(0), 1172 mDrainSequence(0), 1173 mSignalPending(false), 1174 mScreenState(AudioFlinger::mScreenState), 1175 // index 0 is reserved for normal mixer's submix 1176 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1), 1177 // mLatchD, mLatchQ, 1178 mLatchDValid(false), mLatchQValid(false) 1179{ 1180 snprintf(mName, kNameLength, "AudioOut_%X", id); 1181 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 1182 1183 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1184 // it would be safer to explicitly pass initial masterVolume/masterMute as 1185 // parameter. 1186 // 1187 // If the HAL we are using has support for master volume or master mute, 1188 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1189 // and the mute set to false). 1190 mMasterVolume = audioFlinger->masterVolume_l(); 1191 mMasterMute = audioFlinger->masterMute_l(); 1192 if (mOutput && mOutput->audioHwDev) { 1193 if (mOutput->audioHwDev->canSetMasterVolume()) { 1194 mMasterVolume = 1.0; 1195 } 1196 1197 if (mOutput->audioHwDev->canSetMasterMute()) { 1198 mMasterMute = false; 1199 } 1200 } 1201 1202 readOutputParameters_l(); 1203 1204 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1205 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1206 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1207 stream = (audio_stream_type_t) (stream + 1)) { 1208 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1209 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1210 } 1211 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1212 // because mAudioFlinger doesn't have one to copy from 1213} 1214 1215AudioFlinger::PlaybackThread::~PlaybackThread() 1216{ 1217 mAudioFlinger->unregisterWriter(mNBLogWriter); 1218 free(mSinkBuffer); 1219 free(mMixerBuffer); 1220 free(mEffectBuffer); 1221} 1222 1223void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1224{ 1225 dumpInternals(fd, args); 1226 dumpTracks(fd, args); 1227 dumpEffectChains(fd, args); 1228} 1229 1230void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1231{ 1232 const size_t SIZE = 256; 1233 char buffer[SIZE]; 1234 String8 result; 1235 1236 result.appendFormat(" Stream volumes in dB: "); 1237 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1238 const stream_type_t *st = &mStreamTypes[i]; 1239 if (i > 0) { 1240 result.appendFormat(", "); 1241 } 1242 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1243 if (st->mute) { 1244 result.append("M"); 1245 } 1246 } 1247 result.append("\n"); 1248 write(fd, result.string(), result.length()); 1249 result.clear(); 1250 1251 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1252 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1253 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1254 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1255 1256 size_t numtracks = mTracks.size(); 1257 size_t numactive = mActiveTracks.size(); 1258 dprintf(fd, " %d Tracks", numtracks); 1259 size_t numactiveseen = 0; 1260 if (numtracks) { 1261 dprintf(fd, " of which %d are active\n", numactive); 1262 Track::appendDumpHeader(result); 1263 for (size_t i = 0; i < numtracks; ++i) { 1264 sp<Track> track = mTracks[i]; 1265 if (track != 0) { 1266 bool active = mActiveTracks.indexOf(track) >= 0; 1267 if (active) { 1268 numactiveseen++; 1269 } 1270 track->dump(buffer, SIZE, active); 1271 result.append(buffer); 1272 } 1273 } 1274 } else { 1275 result.append("\n"); 1276 } 1277 if (numactiveseen != numactive) { 1278 // some tracks in the active list were not in the tracks list 1279 snprintf(buffer, SIZE, " The following tracks are in the active list but" 1280 " not in the track list\n"); 1281 result.append(buffer); 1282 Track::appendDumpHeader(result); 1283 for (size_t i = 0; i < numactive; ++i) { 1284 sp<Track> track = mActiveTracks[i].promote(); 1285 if (track != 0 && mTracks.indexOf(track) < 0) { 1286 track->dump(buffer, SIZE, true); 1287 result.append(buffer); 1288 } 1289 } 1290 } 1291 1292 write(fd, result.string(), result.size()); 1293} 1294 1295void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1296{ 1297 dprintf(fd, "\nOutput thread %p:\n", this); 1298 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1299 dprintf(fd, " Last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1300 dprintf(fd, " Total writes: %d\n", mNumWrites); 1301 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1302 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1303 dprintf(fd, " Suspend count: %d\n", mSuspended); 1304 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1305 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1306 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1307 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1308 1309 dumpBase(fd, args); 1310} 1311 1312// Thread virtuals 1313 1314void AudioFlinger::PlaybackThread::onFirstRef() 1315{ 1316 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1317} 1318 1319// ThreadBase virtuals 1320void AudioFlinger::PlaybackThread::preExit() 1321{ 1322 ALOGV(" preExit()"); 1323 // FIXME this is using hard-coded strings but in the future, this functionality will be 1324 // converted to use audio HAL extensions required to support tunneling 1325 mOutput->stream->common.set_parameters(&mOutput->stream->common, "exiting=1"); 1326} 1327 1328// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1329sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1330 const sp<AudioFlinger::Client>& client, 1331 audio_stream_type_t streamType, 1332 uint32_t sampleRate, 1333 audio_format_t format, 1334 audio_channel_mask_t channelMask, 1335 size_t *pFrameCount, 1336 const sp<IMemory>& sharedBuffer, 1337 int sessionId, 1338 IAudioFlinger::track_flags_t *flags, 1339 pid_t tid, 1340 int uid, 1341 status_t *status) 1342{ 1343 size_t frameCount = *pFrameCount; 1344 sp<Track> track; 1345 status_t lStatus; 1346 1347 bool isTimed = (*flags & IAudioFlinger::TRACK_TIMED) != 0; 1348 1349 // client expresses a preference for FAST, but we get the final say 1350 if (*flags & IAudioFlinger::TRACK_FAST) { 1351 if ( 1352 // not timed 1353 (!isTimed) && 1354 // either of these use cases: 1355 ( 1356 // use case 1: shared buffer with any frame count 1357 ( 1358 (sharedBuffer != 0) 1359 ) || 1360 // use case 2: callback handler and frame count is default or at least as large as HAL 1361 ( 1362 (tid != -1) && 1363 ((frameCount == 0) || 1364 (frameCount >= mFrameCount)) 1365 ) 1366 ) && 1367 // PCM data 1368 audio_is_linear_pcm(format) && 1369 // mono or stereo 1370 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1371 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1372 // hardware sample rate 1373 (sampleRate == mSampleRate) && 1374 // normal mixer has an associated fast mixer 1375 hasFastMixer() && 1376 // there are sufficient fast track slots available 1377 (mFastTrackAvailMask != 0) 1378 // FIXME test that MixerThread for this fast track has a capable output HAL 1379 // FIXME add a permission test also? 1380 ) { 1381 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1382 if (frameCount == 0) { 1383 // read the fast track multiplier property the first time it is needed 1384 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1385 if (ok != 0) { 1386 ALOGE("%s pthread_once failed: %d", __func__, ok); 1387 } 1388 frameCount = mFrameCount * sFastTrackMultiplier; 1389 } 1390 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1391 frameCount, mFrameCount); 1392 } else { 1393 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1394 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 1395 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1396 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1397 audio_is_linear_pcm(format), 1398 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1399 *flags &= ~IAudioFlinger::TRACK_FAST; 1400 // For compatibility with AudioTrack calculation, buffer depth is forced 1401 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1402 // This is probably too conservative, but legacy application code may depend on it. 1403 // If you change this calculation, also review the start threshold which is related. 1404 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1405 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1406 if (minBufCount < 2) { 1407 minBufCount = 2; 1408 } 1409 size_t minFrameCount = mNormalFrameCount * minBufCount; 1410 if (frameCount < minFrameCount) { 1411 frameCount = minFrameCount; 1412 } 1413 } 1414 } 1415 *pFrameCount = frameCount; 1416 1417 switch (mType) { 1418 1419 case DIRECT: 1420 if (audio_is_linear_pcm(format)) { 1421 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1422 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1423 "for output %p with format %#x", 1424 sampleRate, format, channelMask, mOutput, mFormat); 1425 lStatus = BAD_VALUE; 1426 goto Exit; 1427 } 1428 } 1429 break; 1430 1431 case OFFLOAD: 1432 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1433 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1434 "for output %p with format %#x", 1435 sampleRate, format, channelMask, mOutput, mFormat); 1436 lStatus = BAD_VALUE; 1437 goto Exit; 1438 } 1439 break; 1440 1441 default: 1442 if (!audio_is_linear_pcm(format)) { 1443 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1444 "for output %p with format %#x", 1445 format, mOutput, mFormat); 1446 lStatus = BAD_VALUE; 1447 goto Exit; 1448 } 1449 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1450 if (sampleRate > mSampleRate*2) { 1451 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1452 lStatus = BAD_VALUE; 1453 goto Exit; 1454 } 1455 break; 1456 1457 } 1458 1459 lStatus = initCheck(); 1460 if (lStatus != NO_ERROR) { 1461 ALOGE("createTrack_l() audio driver not initialized"); 1462 goto Exit; 1463 } 1464 1465 { // scope for mLock 1466 Mutex::Autolock _l(mLock); 1467 1468 // all tracks in same audio session must share the same routing strategy otherwise 1469 // conflicts will happen when tracks are moved from one output to another by audio policy 1470 // manager 1471 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1472 for (size_t i = 0; i < mTracks.size(); ++i) { 1473 sp<Track> t = mTracks[i]; 1474 if (t != 0 && !t->isOutputTrack()) { 1475 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1476 if (sessionId == t->sessionId() && strategy != actual) { 1477 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1478 strategy, actual); 1479 lStatus = BAD_VALUE; 1480 goto Exit; 1481 } 1482 } 1483 } 1484 1485 if (!isTimed) { 1486 track = new Track(this, client, streamType, sampleRate, format, 1487 channelMask, frameCount, sharedBuffer, sessionId, uid, *flags); 1488 } else { 1489 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1490 channelMask, frameCount, sharedBuffer, sessionId, uid); 1491 } 1492 1493 // new Track always returns non-NULL, 1494 // but TimedTrack::create() is a factory that could fail by returning NULL 1495 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 1496 if (lStatus != NO_ERROR) { 1497 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 1498 // track must be cleared from the caller as the caller has the AF lock 1499 goto Exit; 1500 } 1501 mTracks.add(track); 1502 1503 sp<EffectChain> chain = getEffectChain_l(sessionId); 1504 if (chain != 0) { 1505 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1506 track->setMainBuffer(chain->inBuffer()); 1507 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1508 chain->incTrackCnt(); 1509 } 1510 1511 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1512 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1513 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1514 // so ask activity manager to do this on our behalf 1515 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 1516 } 1517 } 1518 1519 lStatus = NO_ERROR; 1520 1521Exit: 1522 *status = lStatus; 1523 return track; 1524} 1525 1526uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 1527{ 1528 return latency; 1529} 1530 1531uint32_t AudioFlinger::PlaybackThread::latency() const 1532{ 1533 Mutex::Autolock _l(mLock); 1534 return latency_l(); 1535} 1536uint32_t AudioFlinger::PlaybackThread::latency_l() const 1537{ 1538 if (initCheck() == NO_ERROR) { 1539 return correctLatency_l(mOutput->stream->get_latency(mOutput->stream)); 1540 } else { 1541 return 0; 1542 } 1543} 1544 1545void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1546{ 1547 Mutex::Autolock _l(mLock); 1548 // Don't apply master volume in SW if our HAL can do it for us. 1549 if (mOutput && mOutput->audioHwDev && 1550 mOutput->audioHwDev->canSetMasterVolume()) { 1551 mMasterVolume = 1.0; 1552 } else { 1553 mMasterVolume = value; 1554 } 1555} 1556 1557void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1558{ 1559 Mutex::Autolock _l(mLock); 1560 // Don't apply master mute in SW if our HAL can do it for us. 1561 if (mOutput && mOutput->audioHwDev && 1562 mOutput->audioHwDev->canSetMasterMute()) { 1563 mMasterMute = false; 1564 } else { 1565 mMasterMute = muted; 1566 } 1567} 1568 1569void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1570{ 1571 Mutex::Autolock _l(mLock); 1572 mStreamTypes[stream].volume = value; 1573 broadcast_l(); 1574} 1575 1576void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1577{ 1578 Mutex::Autolock _l(mLock); 1579 mStreamTypes[stream].mute = muted; 1580 broadcast_l(); 1581} 1582 1583float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1584{ 1585 Mutex::Autolock _l(mLock); 1586 return mStreamTypes[stream].volume; 1587} 1588 1589// addTrack_l() must be called with ThreadBase::mLock held 1590status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1591{ 1592 status_t status = ALREADY_EXISTS; 1593 1594 // set retry count for buffer fill 1595 track->mRetryCount = kMaxTrackStartupRetries; 1596 if (mActiveTracks.indexOf(track) < 0) { 1597 // the track is newly added, make sure it fills up all its 1598 // buffers before playing. This is to ensure the client will 1599 // effectively get the latency it requested. 1600 if (!track->isOutputTrack()) { 1601 TrackBase::track_state state = track->mState; 1602 mLock.unlock(); 1603 status = AudioSystem::startOutput(mId, track->streamType(), track->sessionId()); 1604 mLock.lock(); 1605 // abort track was stopped/paused while we released the lock 1606 if (state != track->mState) { 1607 if (status == NO_ERROR) { 1608 mLock.unlock(); 1609 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 1610 mLock.lock(); 1611 } 1612 return INVALID_OPERATION; 1613 } 1614 // abort if start is rejected by audio policy manager 1615 if (status != NO_ERROR) { 1616 return PERMISSION_DENIED; 1617 } 1618#ifdef ADD_BATTERY_DATA 1619 // to track the speaker usage 1620 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 1621#endif 1622 } 1623 1624 track->mFillingUpStatus = track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 1625 track->mResetDone = false; 1626 track->mPresentationCompleteFrames = 0; 1627 mActiveTracks.add(track); 1628 mWakeLockUids.add(track->uid()); 1629 mActiveTracksGeneration++; 1630 mLatestActiveTrack = track; 1631 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1632 if (chain != 0) { 1633 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 1634 track->sessionId()); 1635 chain->incActiveTrackCnt(); 1636 } 1637 1638 status = NO_ERROR; 1639 } 1640 1641 onAddNewTrack_l(); 1642 return status; 1643} 1644 1645bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1646{ 1647 track->terminate(); 1648 // active tracks are removed by threadLoop() 1649 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 1650 track->mState = TrackBase::STOPPED; 1651 if (!trackActive) { 1652 removeTrack_l(track); 1653 } else if (track->isFastTrack() || track->isOffloaded()) { 1654 track->mState = TrackBase::STOPPING_1; 1655 } 1656 1657 return trackActive; 1658} 1659 1660void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1661{ 1662 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1663 mTracks.remove(track); 1664 deleteTrackName_l(track->name()); 1665 // redundant as track is about to be destroyed, for dumpsys only 1666 track->mName = -1; 1667 if (track->isFastTrack()) { 1668 int index = track->mFastIndex; 1669 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1670 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1671 mFastTrackAvailMask |= 1 << index; 1672 // redundant as track is about to be destroyed, for dumpsys only 1673 track->mFastIndex = -1; 1674 } 1675 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1676 if (chain != 0) { 1677 chain->decTrackCnt(); 1678 } 1679} 1680 1681void AudioFlinger::PlaybackThread::broadcast_l() 1682{ 1683 // Thread could be blocked waiting for async 1684 // so signal it to handle state changes immediately 1685 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1686 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1687 mSignalPending = true; 1688 mWaitWorkCV.broadcast(); 1689} 1690 1691String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1692{ 1693 Mutex::Autolock _l(mLock); 1694 if (initCheck() != NO_ERROR) { 1695 return String8(); 1696 } 1697 1698 char *s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1699 const String8 out_s8(s); 1700 free(s); 1701 return out_s8; 1702} 1703 1704void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) { 1705 AudioSystem::OutputDescriptor desc; 1706 void *param2 = NULL; 1707 1708 ALOGV("PlaybackThread::audioConfigChanged, thread %p, event %d, param %d", this, event, 1709 param); 1710 1711 switch (event) { 1712 case AudioSystem::OUTPUT_OPENED: 1713 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1714 desc.channelMask = mChannelMask; 1715 desc.samplingRate = mSampleRate; 1716 desc.format = mFormat; 1717 desc.frameCount = mNormalFrameCount; // FIXME see 1718 // AudioFlinger::frameCount(audio_io_handle_t) 1719 desc.latency = latency_l(); 1720 param2 = &desc; 1721 break; 1722 1723 case AudioSystem::STREAM_CONFIG_CHANGED: 1724 param2 = ¶m; 1725 case AudioSystem::OUTPUT_CLOSED: 1726 default: 1727 break; 1728 } 1729 mAudioFlinger->audioConfigChanged(event, mId, param2); 1730} 1731 1732void AudioFlinger::PlaybackThread::writeCallback() 1733{ 1734 ALOG_ASSERT(mCallbackThread != 0); 1735 mCallbackThread->resetWriteBlocked(); 1736} 1737 1738void AudioFlinger::PlaybackThread::drainCallback() 1739{ 1740 ALOG_ASSERT(mCallbackThread != 0); 1741 mCallbackThread->resetDraining(); 1742} 1743 1744void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 1745{ 1746 Mutex::Autolock _l(mLock); 1747 // reject out of sequence requests 1748 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 1749 mWriteAckSequence &= ~1; 1750 mWaitWorkCV.signal(); 1751 } 1752} 1753 1754void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 1755{ 1756 Mutex::Autolock _l(mLock); 1757 // reject out of sequence requests 1758 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 1759 mDrainSequence &= ~1; 1760 mWaitWorkCV.signal(); 1761 } 1762} 1763 1764// static 1765int AudioFlinger::PlaybackThread::asyncCallback(stream_callback_event_t event, 1766 void *param __unused, 1767 void *cookie) 1768{ 1769 AudioFlinger::PlaybackThread *me = (AudioFlinger::PlaybackThread *)cookie; 1770 ALOGV("asyncCallback() event %d", event); 1771 switch (event) { 1772 case STREAM_CBK_EVENT_WRITE_READY: 1773 me->writeCallback(); 1774 break; 1775 case STREAM_CBK_EVENT_DRAIN_READY: 1776 me->drainCallback(); 1777 break; 1778 default: 1779 ALOGW("asyncCallback() unknown event %d", event); 1780 break; 1781 } 1782 return 0; 1783} 1784 1785void AudioFlinger::PlaybackThread::readOutputParameters_l() 1786{ 1787 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 1788 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1789 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1790 if (!audio_is_output_channel(mChannelMask)) { 1791 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 1792 } 1793 if ((mType == MIXER || mType == DUPLICATING) && mChannelMask != AUDIO_CHANNEL_OUT_STEREO) { 1794 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output; " 1795 "must be AUDIO_CHANNEL_OUT_STEREO", mChannelMask); 1796 } 1797 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 1798 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1799 if (!audio_is_valid_format(mFormat)) { 1800 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 1801 } 1802 if ((mType == MIXER || mType == DUPLICATING) && mFormat != AUDIO_FORMAT_PCM_16_BIT) { 1803 LOG_ALWAYS_FATAL("HAL format %#x not supported for mixed output; " 1804 "must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 1805 } 1806 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1807 mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); 1808 mFrameCount = mBufferSize / mFrameSize; 1809 if (mFrameCount & 15) { 1810 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1811 mFrameCount); 1812 } 1813 1814 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) && 1815 (mOutput->stream->set_callback != NULL)) { 1816 if (mOutput->stream->set_callback(mOutput->stream, 1817 AudioFlinger::PlaybackThread::asyncCallback, this) == 0) { 1818 mUseAsyncWrite = true; 1819 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 1820 } 1821 } 1822 1823 // Calculate size of normal sink buffer relative to the HAL output buffer size 1824 double multiplier = 1.0; 1825 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 1826 kUseFastMixer == FastMixer_Dynamic)) { 1827 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 1828 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 1829 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1830 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1831 maxNormalFrameCount = maxNormalFrameCount & ~15; 1832 if (maxNormalFrameCount < minNormalFrameCount) { 1833 maxNormalFrameCount = minNormalFrameCount; 1834 } 1835 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1836 if (multiplier <= 1.0) { 1837 multiplier = 1.0; 1838 } else if (multiplier <= 2.0) { 1839 if (2 * mFrameCount <= maxNormalFrameCount) { 1840 multiplier = 2.0; 1841 } else { 1842 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1843 } 1844 } else { 1845 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL 1846 // SRC (it would be unusual for the normal sink buffer size to not be a multiple of fast 1847 // track, but we sometimes have to do this to satisfy the maximum frame count 1848 // constraint) 1849 // FIXME this rounding up should not be done if no HAL SRC 1850 uint32_t truncMult = (uint32_t) multiplier; 1851 if ((truncMult & 1)) { 1852 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1853 ++truncMult; 1854 } 1855 } 1856 multiplier = (double) truncMult; 1857 } 1858 } 1859 mNormalFrameCount = multiplier * mFrameCount; 1860 // round up to nearest 16 frames to satisfy AudioMixer 1861 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 1862 ALOGI("HAL output buffer size %u frames, normal sink buffer size %u frames", mFrameCount, 1863 mNormalFrameCount); 1864 1865 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 1866 // Originally this was int16_t[] array, need to remove legacy implications. 1867 free(mSinkBuffer); 1868 mSinkBuffer = NULL; 1869 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 1870 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 1871 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 1872 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 1873 1874 // We resize the mMixerBuffer according to the requirements of the sink buffer which 1875 // drives the output. 1876 free(mMixerBuffer); 1877 mMixerBuffer = NULL; 1878 if (mMixerBufferEnabled) { 1879 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 1880 mMixerBufferSize = mNormalFrameCount * mChannelCount 1881 * audio_bytes_per_sample(mMixerBufferFormat); 1882 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 1883 } 1884 free(mEffectBuffer); 1885 mEffectBuffer = NULL; 1886 if (mEffectBufferEnabled) { 1887 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 1888 mEffectBufferSize = mNormalFrameCount * mChannelCount 1889 * audio_bytes_per_sample(mEffectBufferFormat); 1890 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 1891 } 1892 1893 // force reconfiguration of effect chains and engines to take new buffer size and audio 1894 // parameters into account 1895 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 1896 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1897 // matter. 1898 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1899 Vector< sp<EffectChain> > effectChains = mEffectChains; 1900 for (size_t i = 0; i < effectChains.size(); i ++) { 1901 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1902 } 1903} 1904 1905 1906status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1907{ 1908 if (halFrames == NULL || dspFrames == NULL) { 1909 return BAD_VALUE; 1910 } 1911 Mutex::Autolock _l(mLock); 1912 if (initCheck() != NO_ERROR) { 1913 return INVALID_OPERATION; 1914 } 1915 size_t framesWritten = mBytesWritten / mFrameSize; 1916 *halFrames = framesWritten; 1917 1918 if (isSuspended()) { 1919 // return an estimation of rendered frames when the output is suspended 1920 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 1921 *dspFrames = framesWritten >= latencyFrames ? framesWritten - latencyFrames : 0; 1922 return NO_ERROR; 1923 } else { 1924 status_t status; 1925 uint32_t frames; 1926 status = mOutput->stream->get_render_position(mOutput->stream, &frames); 1927 *dspFrames = (size_t)frames; 1928 return status; 1929 } 1930} 1931 1932uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) const 1933{ 1934 Mutex::Autolock _l(mLock); 1935 uint32_t result = 0; 1936 if (getEffectChain_l(sessionId) != 0) { 1937 result = EFFECT_SESSION; 1938 } 1939 1940 for (size_t i = 0; i < mTracks.size(); ++i) { 1941 sp<Track> track = mTracks[i]; 1942 if (sessionId == track->sessionId() && !track->isInvalid()) { 1943 result |= TRACK_SESSION; 1944 break; 1945 } 1946 } 1947 1948 return result; 1949} 1950 1951uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1952{ 1953 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1954 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1955 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1956 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1957 } 1958 for (size_t i = 0; i < mTracks.size(); i++) { 1959 sp<Track> track = mTracks[i]; 1960 if (sessionId == track->sessionId() && !track->isInvalid()) { 1961 return AudioSystem::getStrategyForStream(track->streamType()); 1962 } 1963 } 1964 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1965} 1966 1967 1968AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1969{ 1970 Mutex::Autolock _l(mLock); 1971 return mOutput; 1972} 1973 1974AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1975{ 1976 Mutex::Autolock _l(mLock); 1977 AudioStreamOut *output = mOutput; 1978 mOutput = NULL; 1979 // FIXME FastMixer might also have a raw ptr to mOutputSink; 1980 // must push a NULL and wait for ack 1981 mOutputSink.clear(); 1982 mPipeSink.clear(); 1983 mNormalSink.clear(); 1984 return output; 1985} 1986 1987// this method must always be called either with ThreadBase mLock held or inside the thread loop 1988audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1989{ 1990 if (mOutput == NULL) { 1991 return NULL; 1992 } 1993 return &mOutput->stream->common; 1994} 1995 1996uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1997{ 1998 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 1999} 2000 2001status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2002{ 2003 if (!isValidSyncEvent(event)) { 2004 return BAD_VALUE; 2005 } 2006 2007 Mutex::Autolock _l(mLock); 2008 2009 for (size_t i = 0; i < mTracks.size(); ++i) { 2010 sp<Track> track = mTracks[i]; 2011 if (event->triggerSession() == track->sessionId()) { 2012 (void) track->setSyncEvent(event); 2013 return NO_ERROR; 2014 } 2015 } 2016 2017 return NAME_NOT_FOUND; 2018} 2019 2020bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2021{ 2022 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2023} 2024 2025void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2026 const Vector< sp<Track> >& tracksToRemove) 2027{ 2028 size_t count = tracksToRemove.size(); 2029 if (count > 0) { 2030 for (size_t i = 0 ; i < count ; i++) { 2031 const sp<Track>& track = tracksToRemove.itemAt(i); 2032 if (!track->isOutputTrack()) { 2033 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2034#ifdef ADD_BATTERY_DATA 2035 // to track the speaker usage 2036 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2037#endif 2038 if (track->isTerminated()) { 2039 AudioSystem::releaseOutput(mId); 2040 } 2041 } 2042 } 2043 } 2044} 2045 2046void AudioFlinger::PlaybackThread::checkSilentMode_l() 2047{ 2048 if (!mMasterMute) { 2049 char value[PROPERTY_VALUE_MAX]; 2050 if (property_get("ro.audio.silent", value, "0") > 0) { 2051 char *endptr; 2052 unsigned long ul = strtoul(value, &endptr, 0); 2053 if (*endptr == '\0' && ul != 0) { 2054 ALOGD("Silence is golden"); 2055 // The setprop command will not allow a property to be changed after 2056 // the first time it is set, so we don't have to worry about un-muting. 2057 setMasterMute_l(true); 2058 } 2059 } 2060 } 2061} 2062 2063// shared by MIXER and DIRECT, overridden by DUPLICATING 2064ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2065{ 2066 // FIXME rewrite to reduce number of system calls 2067 mLastWriteTime = systemTime(); 2068 mInWrite = true; 2069 ssize_t bytesWritten; 2070 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2071 2072 // If an NBAIO sink is present, use it to write the normal mixer's submix 2073 if (mNormalSink != 0) { 2074 const size_t count = mBytesRemaining / mFrameSize; 2075 2076 ATRACE_BEGIN("write"); 2077 // update the setpoint when AudioFlinger::mScreenState changes 2078 uint32_t screenState = AudioFlinger::mScreenState; 2079 if (screenState != mScreenState) { 2080 mScreenState = screenState; 2081 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2082 if (pipe != NULL) { 2083 pipe->setAvgFrames((mScreenState & 1) ? 2084 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2085 } 2086 } 2087 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2088 ATRACE_END(); 2089 if (framesWritten > 0) { 2090 bytesWritten = framesWritten * mFrameSize; 2091 } else { 2092 bytesWritten = framesWritten; 2093 } 2094 status_t status = mNormalSink->getTimestamp(mLatchD.mTimestamp); 2095 if (status == NO_ERROR) { 2096 size_t totalFramesWritten = mNormalSink->framesWritten(); 2097 if (totalFramesWritten >= mLatchD.mTimestamp.mPosition) { 2098 mLatchD.mUnpresentedFrames = totalFramesWritten - mLatchD.mTimestamp.mPosition; 2099 mLatchDValid = true; 2100 } 2101 } 2102 // otherwise use the HAL / AudioStreamOut directly 2103 } else { 2104 // Direct output and offload threads 2105 2106 if (mUseAsyncWrite) { 2107 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2108 mWriteAckSequence += 2; 2109 mWriteAckSequence |= 1; 2110 ALOG_ASSERT(mCallbackThread != 0); 2111 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2112 } 2113 // FIXME We should have an implementation of timestamps for direct output threads. 2114 // They are used e.g for multichannel PCM playback over HDMI. 2115 bytesWritten = mOutput->stream->write(mOutput->stream, 2116 (char *)mSinkBuffer + offset, mBytesRemaining); 2117 if (mUseAsyncWrite && 2118 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2119 // do not wait for async callback in case of error of full write 2120 mWriteAckSequence &= ~1; 2121 ALOG_ASSERT(mCallbackThread != 0); 2122 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2123 } 2124 } 2125 2126 mNumWrites++; 2127 mInWrite = false; 2128 mStandby = false; 2129 return bytesWritten; 2130} 2131 2132void AudioFlinger::PlaybackThread::threadLoop_drain() 2133{ 2134 if (mOutput->stream->drain) { 2135 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2136 if (mUseAsyncWrite) { 2137 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2138 mDrainSequence |= 1; 2139 ALOG_ASSERT(mCallbackThread != 0); 2140 mCallbackThread->setDraining(mDrainSequence); 2141 } 2142 mOutput->stream->drain(mOutput->stream, 2143 (mMixerStatus == MIXER_DRAIN_TRACK) ? AUDIO_DRAIN_EARLY_NOTIFY 2144 : AUDIO_DRAIN_ALL); 2145 } 2146} 2147 2148void AudioFlinger::PlaybackThread::threadLoop_exit() 2149{ 2150 // Default implementation has nothing to do 2151} 2152 2153/* 2154The derived values that are cached: 2155 - mSinkBufferSize from frame count * frame size 2156 - activeSleepTime from activeSleepTimeUs() 2157 - idleSleepTime from idleSleepTimeUs() 2158 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2159 - maxPeriod from frame count and sample rate (MIXER only) 2160 2161The parameters that affect these derived values are: 2162 - frame count 2163 - frame size 2164 - sample rate 2165 - device type: A2DP or not 2166 - device latency 2167 - format: PCM or not 2168 - active sleep time 2169 - idle sleep time 2170*/ 2171 2172void AudioFlinger::PlaybackThread::cacheParameters_l() 2173{ 2174 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2175 activeSleepTime = activeSleepTimeUs(); 2176 idleSleepTime = idleSleepTimeUs(); 2177} 2178 2179void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2180{ 2181 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2182 this, streamType, mTracks.size()); 2183 Mutex::Autolock _l(mLock); 2184 2185 size_t size = mTracks.size(); 2186 for (size_t i = 0; i < size; i++) { 2187 sp<Track> t = mTracks[i]; 2188 if (t->streamType() == streamType) { 2189 t->invalidate(); 2190 } 2191 } 2192} 2193 2194status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2195{ 2196 int session = chain->sessionId(); 2197 int16_t* buffer = reinterpret_cast<int16_t*>(mEffectBufferEnabled 2198 ? mEffectBuffer : mSinkBuffer); 2199 bool ownsBuffer = false; 2200 2201 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2202 if (session > 0) { 2203 // Only one effect chain can be present in direct output thread and it uses 2204 // the sink buffer as input 2205 if (mType != DIRECT) { 2206 size_t numSamples = mNormalFrameCount * mChannelCount; 2207 buffer = new int16_t[numSamples]; 2208 memset(buffer, 0, numSamples * sizeof(int16_t)); 2209 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 2210 ownsBuffer = true; 2211 } 2212 2213 // Attach all tracks with same session ID to this chain. 2214 for (size_t i = 0; i < mTracks.size(); ++i) { 2215 sp<Track> track = mTracks[i]; 2216 if (session == track->sessionId()) { 2217 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2218 buffer); 2219 track->setMainBuffer(buffer); 2220 chain->incTrackCnt(); 2221 } 2222 } 2223 2224 // indicate all active tracks in the chain 2225 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2226 sp<Track> track = mActiveTracks[i].promote(); 2227 if (track == 0) { 2228 continue; 2229 } 2230 if (session == track->sessionId()) { 2231 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2232 chain->incActiveTrackCnt(); 2233 } 2234 } 2235 } 2236 2237 chain->setInBuffer(buffer, ownsBuffer); 2238 chain->setOutBuffer(reinterpret_cast<int16_t*>(mEffectBufferEnabled 2239 ? mEffectBuffer : mSinkBuffer)); 2240 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2241 // chains list in order to be processed last as it contains output stage effects 2242 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2243 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2244 // after track specific effects and before output stage 2245 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2246 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 2247 // Effect chain for other sessions are inserted at beginning of effect 2248 // chains list to be processed before output mix effects. Relative order between other 2249 // sessions is not important 2250 size_t size = mEffectChains.size(); 2251 size_t i = 0; 2252 for (i = 0; i < size; i++) { 2253 if (mEffectChains[i]->sessionId() < session) { 2254 break; 2255 } 2256 } 2257 mEffectChains.insertAt(chain, i); 2258 checkSuspendOnAddEffectChain_l(chain); 2259 2260 return NO_ERROR; 2261} 2262 2263size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2264{ 2265 int session = chain->sessionId(); 2266 2267 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2268 2269 for (size_t i = 0; i < mEffectChains.size(); i++) { 2270 if (chain == mEffectChains[i]) { 2271 mEffectChains.removeAt(i); 2272 // detach all active tracks from the chain 2273 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 2274 sp<Track> track = mActiveTracks[i].promote(); 2275 if (track == 0) { 2276 continue; 2277 } 2278 if (session == track->sessionId()) { 2279 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2280 chain.get(), session); 2281 chain->decActiveTrackCnt(); 2282 } 2283 } 2284 2285 // detach all tracks with same session ID from this chain 2286 for (size_t i = 0; i < mTracks.size(); ++i) { 2287 sp<Track> track = mTracks[i]; 2288 if (session == track->sessionId()) { 2289 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2290 chain->decTrackCnt(); 2291 } 2292 } 2293 break; 2294 } 2295 } 2296 return mEffectChains.size(); 2297} 2298 2299status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2300 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2301{ 2302 Mutex::Autolock _l(mLock); 2303 return attachAuxEffect_l(track, EffectId); 2304} 2305 2306status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2307 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 2308{ 2309 status_t status = NO_ERROR; 2310 2311 if (EffectId == 0) { 2312 track->setAuxBuffer(0, NULL); 2313 } else { 2314 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2315 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2316 if (effect != 0) { 2317 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2318 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2319 } else { 2320 status = INVALID_OPERATION; 2321 } 2322 } else { 2323 status = BAD_VALUE; 2324 } 2325 } 2326 return status; 2327} 2328 2329void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2330{ 2331 for (size_t i = 0; i < mTracks.size(); ++i) { 2332 sp<Track> track = mTracks[i]; 2333 if (track->auxEffectId() == effectId) { 2334 attachAuxEffect_l(track, 0); 2335 } 2336 } 2337} 2338 2339bool AudioFlinger::PlaybackThread::threadLoop() 2340{ 2341 Vector< sp<Track> > tracksToRemove; 2342 2343 standbyTime = systemTime(); 2344 2345 // MIXER 2346 nsecs_t lastWarning = 0; 2347 2348 // DUPLICATING 2349 // FIXME could this be made local to while loop? 2350 writeFrames = 0; 2351 2352 int lastGeneration = 0; 2353 2354 cacheParameters_l(); 2355 sleepTime = idleSleepTime; 2356 2357 if (mType == MIXER) { 2358 sleepTimeShift = 0; 2359 } 2360 2361 CpuStats cpuStats; 2362 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2363 2364 acquireWakeLock(); 2365 2366 // mNBLogWriter->log can only be called while thread mutex mLock is held. 2367 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2368 // and then that string will be logged at the next convenient opportunity. 2369 const char *logString = NULL; 2370 2371 checkSilentMode_l(); 2372 2373 while (!exitPending()) 2374 { 2375 cpuStats.sample(myName); 2376 2377 Vector< sp<EffectChain> > effectChains; 2378 2379 { // scope for mLock 2380 2381 Mutex::Autolock _l(mLock); 2382 2383 processConfigEvents_l(); 2384 2385 if (logString != NULL) { 2386 mNBLogWriter->logTimestamp(); 2387 mNBLogWriter->log(logString); 2388 logString = NULL; 2389 } 2390 2391 if (mLatchDValid) { 2392 mLatchQ = mLatchD; 2393 mLatchDValid = false; 2394 mLatchQValid = true; 2395 } 2396 2397 saveOutputTracks(); 2398 if (mSignalPending) { 2399 // A signal was raised while we were unlocked 2400 mSignalPending = false; 2401 } else if (waitingAsyncCallback_l()) { 2402 if (exitPending()) { 2403 break; 2404 } 2405 releaseWakeLock_l(); 2406 mWakeLockUids.clear(); 2407 mActiveTracksGeneration++; 2408 ALOGV("wait async completion"); 2409 mWaitWorkCV.wait(mLock); 2410 ALOGV("async completion/wake"); 2411 acquireWakeLock_l(); 2412 standbyTime = systemTime() + standbyDelay; 2413 sleepTime = 0; 2414 2415 continue; 2416 } 2417 if ((!mActiveTracks.size() && systemTime() > standbyTime) || 2418 isSuspended()) { 2419 // put audio hardware into standby after short delay 2420 if (shouldStandby_l()) { 2421 2422 threadLoop_standby(); 2423 2424 mStandby = true; 2425 } 2426 2427 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2428 // we're about to wait, flush the binder command buffer 2429 IPCThreadState::self()->flushCommands(); 2430 2431 clearOutputTracks(); 2432 2433 if (exitPending()) { 2434 break; 2435 } 2436 2437 releaseWakeLock_l(); 2438 mWakeLockUids.clear(); 2439 mActiveTracksGeneration++; 2440 // wait until we have something to do... 2441 ALOGV("%s going to sleep", myName.string()); 2442 mWaitWorkCV.wait(mLock); 2443 ALOGV("%s waking up", myName.string()); 2444 acquireWakeLock_l(); 2445 2446 mMixerStatus = MIXER_IDLE; 2447 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2448 mBytesWritten = 0; 2449 mBytesRemaining = 0; 2450 checkSilentMode_l(); 2451 2452 standbyTime = systemTime() + standbyDelay; 2453 sleepTime = idleSleepTime; 2454 if (mType == MIXER) { 2455 sleepTimeShift = 0; 2456 } 2457 2458 continue; 2459 } 2460 } 2461 // mMixerStatusIgnoringFastTracks is also updated internally 2462 mMixerStatus = prepareTracks_l(&tracksToRemove); 2463 2464 // compare with previously applied list 2465 if (lastGeneration != mActiveTracksGeneration) { 2466 // update wakelock 2467 updateWakeLockUids_l(mWakeLockUids); 2468 lastGeneration = mActiveTracksGeneration; 2469 } 2470 2471 // prevent any changes in effect chain list and in each effect chain 2472 // during mixing and effect process as the audio buffers could be deleted 2473 // or modified if an effect is created or deleted 2474 lockEffectChains_l(effectChains); 2475 } // mLock scope ends 2476 2477 if (mBytesRemaining == 0) { 2478 mCurrentWriteLength = 0; 2479 if (mMixerStatus == MIXER_TRACKS_READY) { 2480 // threadLoop_mix() sets mCurrentWriteLength 2481 threadLoop_mix(); 2482 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 2483 && (mMixerStatus != MIXER_DRAIN_ALL)) { 2484 // threadLoop_sleepTime sets sleepTime to 0 if data 2485 // must be written to HAL 2486 threadLoop_sleepTime(); 2487 if (sleepTime == 0) { 2488 mCurrentWriteLength = mSinkBufferSize; 2489 } 2490 } 2491 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 2492 // mMixerBuffer with data if mMixerBufferValid is true and sleepTime == 0. 2493 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 2494 // or mSinkBuffer (if there are no effects). 2495 // 2496 // This is done pre-effects computation; if effects change to 2497 // support higher precision, this needs to move. 2498 // 2499 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 2500 // TODO use sleepTime == 0 as an additional condition. 2501 if (mMixerBufferValid) { 2502 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 2503 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 2504 2505 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 2506 mNormalFrameCount * mChannelCount); 2507 } 2508 2509 mBytesRemaining = mCurrentWriteLength; 2510 if (isSuspended()) { 2511 sleepTime = suspendSleepTimeUs(); 2512 // simulate write to HAL when suspended 2513 mBytesWritten += mSinkBufferSize; 2514 mBytesRemaining = 0; 2515 } 2516 2517 // only process effects if we're going to write 2518 if (sleepTime == 0 && mType != OFFLOAD) { 2519 for (size_t i = 0; i < effectChains.size(); i ++) { 2520 effectChains[i]->process_l(); 2521 } 2522 } 2523 } 2524 // Process effect chains for offloaded thread even if no audio 2525 // was read from audio track: process only updates effect state 2526 // and thus does have to be synchronized with audio writes but may have 2527 // to be called while waiting for async write callback 2528 if (mType == OFFLOAD) { 2529 for (size_t i = 0; i < effectChains.size(); i ++) { 2530 effectChains[i]->process_l(); 2531 } 2532 } 2533 2534 // Only if the Effects buffer is enabled and there is data in the 2535 // Effects buffer (buffer valid), we need to 2536 // copy into the sink buffer. 2537 // TODO use sleepTime == 0 as an additional condition. 2538 if (mEffectBufferValid) { 2539 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 2540 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 2541 mNormalFrameCount * mChannelCount); 2542 } 2543 2544 // enable changes in effect chain 2545 unlockEffectChains(effectChains); 2546 2547 if (!waitingAsyncCallback()) { 2548 // sleepTime == 0 means we must write to audio hardware 2549 if (sleepTime == 0) { 2550 if (mBytesRemaining) { 2551 ssize_t ret = threadLoop_write(); 2552 if (ret < 0) { 2553 mBytesRemaining = 0; 2554 } else { 2555 mBytesWritten += ret; 2556 mBytesRemaining -= ret; 2557 } 2558 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 2559 (mMixerStatus == MIXER_DRAIN_ALL)) { 2560 threadLoop_drain(); 2561 } 2562 if (mType == MIXER) { 2563 // write blocked detection 2564 nsecs_t now = systemTime(); 2565 nsecs_t delta = now - mLastWriteTime; 2566 if (!mStandby && delta > maxPeriod) { 2567 mNumDelayedWrites++; 2568 if ((now - lastWarning) > kWarningThrottleNs) { 2569 ATRACE_NAME("underrun"); 2570 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2571 ns2ms(delta), mNumDelayedWrites, this); 2572 lastWarning = now; 2573 } 2574 } 2575 } 2576 2577 } else { 2578 usleep(sleepTime); 2579 } 2580 } 2581 2582 // Finally let go of removed track(s), without the lock held 2583 // since we can't guarantee the destructors won't acquire that 2584 // same lock. This will also mutate and push a new fast mixer state. 2585 threadLoop_removeTracks(tracksToRemove); 2586 tracksToRemove.clear(); 2587 2588 // FIXME I don't understand the need for this here; 2589 // it was in the original code but maybe the 2590 // assignment in saveOutputTracks() makes this unnecessary? 2591 clearOutputTracks(); 2592 2593 // Effect chains will be actually deleted here if they were removed from 2594 // mEffectChains list during mixing or effects processing 2595 effectChains.clear(); 2596 2597 // FIXME Note that the above .clear() is no longer necessary since effectChains 2598 // is now local to this block, but will keep it for now (at least until merge done). 2599 } 2600 2601 threadLoop_exit(); 2602 2603 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2604 if (mType == MIXER || mType == DIRECT || mType == OFFLOAD) { 2605 // put output stream into standby mode 2606 if (!mStandby) { 2607 mOutput->stream->common.standby(&mOutput->stream->common); 2608 } 2609 } 2610 2611 releaseWakeLock(); 2612 mWakeLockUids.clear(); 2613 mActiveTracksGeneration++; 2614 2615 ALOGV("Thread %p type %d exiting", this, mType); 2616 return false; 2617} 2618 2619// removeTracks_l() must be called with ThreadBase::mLock held 2620void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 2621{ 2622 size_t count = tracksToRemove.size(); 2623 if (count > 0) { 2624 for (size_t i=0 ; i<count ; i++) { 2625 const sp<Track>& track = tracksToRemove.itemAt(i); 2626 mActiveTracks.remove(track); 2627 mWakeLockUids.remove(track->uid()); 2628 mActiveTracksGeneration++; 2629 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 2630 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2631 if (chain != 0) { 2632 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 2633 track->sessionId()); 2634 chain->decActiveTrackCnt(); 2635 } 2636 if (track->isTerminated()) { 2637 removeTrack_l(track); 2638 } 2639 } 2640 } 2641 2642} 2643 2644status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 2645{ 2646 if (mNormalSink != 0) { 2647 return mNormalSink->getTimestamp(timestamp); 2648 } 2649 if (mType == OFFLOAD && mOutput->stream->get_presentation_position) { 2650 uint64_t position64; 2651 int ret = mOutput->stream->get_presentation_position( 2652 mOutput->stream, &position64, ×tamp.mTime); 2653 if (ret == 0) { 2654 timestamp.mPosition = (uint32_t)position64; 2655 return NO_ERROR; 2656 } 2657 } 2658 return INVALID_OPERATION; 2659} 2660 2661status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 2662 audio_patch_handle_t *handle) 2663{ 2664 status_t status = NO_ERROR; 2665 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2666 // store new device and send to effects 2667 audio_devices_t type = AUDIO_DEVICE_NONE; 2668 for (unsigned int i = 0; i < patch->num_sinks; i++) { 2669 type |= patch->sinks[i].ext.device.type; 2670 } 2671 mOutDevice = type; 2672 for (size_t i = 0; i < mEffectChains.size(); i++) { 2673 mEffectChains[i]->setDevice_l(mOutDevice); 2674 } 2675 2676 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2677 status = hwDevice->create_audio_patch(hwDevice, 2678 patch->num_sources, 2679 patch->sources, 2680 patch->num_sinks, 2681 patch->sinks, 2682 handle); 2683 } else { 2684 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 2685 } 2686 return status; 2687} 2688 2689status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 2690{ 2691 status_t status = NO_ERROR; 2692 if (mOutput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 2693 audio_hw_device_t *hwDevice = mOutput->audioHwDev->hwDevice(); 2694 status = hwDevice->release_audio_patch(hwDevice, handle); 2695 } else { 2696 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 2697 } 2698 return status; 2699} 2700 2701// ---------------------------------------------------------------------------- 2702 2703AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2704 audio_io_handle_t id, audio_devices_t device, type_t type) 2705 : PlaybackThread(audioFlinger, output, id, device, type), 2706 // mAudioMixer below 2707 // mFastMixer below 2708 mFastMixerFutex(0) 2709 // mOutputSink below 2710 // mPipeSink below 2711 // mNormalSink below 2712{ 2713 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2714 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%u, " 2715 "mFrameCount=%d, mNormalFrameCount=%d", 2716 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2717 mNormalFrameCount); 2718 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2719 2720 // FIXME - Current mixer implementation only supports stereo output 2721 if (mChannelCount != FCC_2) { 2722 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2723 } 2724 2725 // create an NBAIO sink for the HAL output stream, and negotiate 2726 mOutputSink = new AudioStreamOutSink(output->stream); 2727 size_t numCounterOffers = 0; 2728 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 2729 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2730 ALOG_ASSERT(index == 0); 2731 2732 // initialize fast mixer depending on configuration 2733 bool initFastMixer; 2734 switch (kUseFastMixer) { 2735 case FastMixer_Never: 2736 initFastMixer = false; 2737 break; 2738 case FastMixer_Always: 2739 initFastMixer = true; 2740 break; 2741 case FastMixer_Static: 2742 case FastMixer_Dynamic: 2743 initFastMixer = mFrameCount < mNormalFrameCount; 2744 break; 2745 } 2746 if (initFastMixer) { 2747 audio_format_t fastMixerFormat; 2748 if (mMixerBufferEnabled && mEffectBufferEnabled) { 2749 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 2750 } else { 2751 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 2752 } 2753 if (mFormat != fastMixerFormat) { 2754 // change our Sink format to accept our intermediate precision 2755 mFormat = fastMixerFormat; 2756 free(mSinkBuffer); 2757 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2758 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2759 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2760 } 2761 2762 // create a MonoPipe to connect our submix to FastMixer 2763 NBAIO_Format format = mOutputSink->format(); 2764 // adjust format to match that of the Fast Mixer 2765 format.mFormat = fastMixerFormat; 2766 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 2767 2768 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2769 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2770 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2771 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2772 const NBAIO_Format offers[1] = {format}; 2773 size_t numCounterOffers = 0; 2774 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2775 ALOG_ASSERT(index == 0); 2776 monoPipe->setAvgFrames((mScreenState & 1) ? 2777 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2778 mPipeSink = monoPipe; 2779 2780#ifdef TEE_SINK 2781 if (mTeeSinkOutputEnabled) { 2782 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2783 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, format); 2784 numCounterOffers = 0; 2785 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2786 ALOG_ASSERT(index == 0); 2787 mTeeSink = teeSink; 2788 PipeReader *teeSource = new PipeReader(*teeSink); 2789 numCounterOffers = 0; 2790 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2791 ALOG_ASSERT(index == 0); 2792 mTeeSource = teeSource; 2793 } 2794#endif 2795 2796 // create fast mixer and configure it initially with just one fast track for our submix 2797 mFastMixer = new FastMixer(); 2798 FastMixerStateQueue *sq = mFastMixer->sq(); 2799#ifdef STATE_QUEUE_DUMP 2800 sq->setObserverDump(&mStateQueueObserverDump); 2801 sq->setMutatorDump(&mStateQueueMutatorDump); 2802#endif 2803 FastMixerState *state = sq->begin(); 2804 FastTrack *fastTrack = &state->mFastTracks[0]; 2805 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2806 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2807 fastTrack->mVolumeProvider = NULL; 2808 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 2809 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 2810 fastTrack->mGeneration++; 2811 state->mFastTracksGen++; 2812 state->mTrackMask = 1; 2813 // fast mixer will use the HAL output sink 2814 state->mOutputSink = mOutputSink.get(); 2815 state->mOutputSinkGen++; 2816 state->mFrameCount = mFrameCount; 2817 state->mCommand = FastMixerState::COLD_IDLE; 2818 // already done in constructor initialization list 2819 //mFastMixerFutex = 0; 2820 state->mColdFutexAddr = &mFastMixerFutex; 2821 state->mColdGen++; 2822 state->mDumpState = &mFastMixerDumpState; 2823#ifdef TEE_SINK 2824 state->mTeeSink = mTeeSink.get(); 2825#endif 2826 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 2827 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 2828 sq->end(); 2829 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2830 2831 // start the fast mixer 2832 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2833 pid_t tid = mFastMixer->getTid(); 2834 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2835 if (err != 0) { 2836 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2837 kPriorityFastMixer, getpid_cached, tid, err); 2838 } 2839 2840#ifdef AUDIO_WATCHDOG 2841 // create and start the watchdog 2842 mAudioWatchdog = new AudioWatchdog(); 2843 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2844 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2845 tid = mAudioWatchdog->getTid(); 2846 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2847 if (err != 0) { 2848 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2849 kPriorityFastMixer, getpid_cached, tid, err); 2850 } 2851#endif 2852 2853 } else { 2854 mFastMixer = NULL; 2855 } 2856 2857 switch (kUseFastMixer) { 2858 case FastMixer_Never: 2859 case FastMixer_Dynamic: 2860 mNormalSink = mOutputSink; 2861 break; 2862 case FastMixer_Always: 2863 mNormalSink = mPipeSink; 2864 break; 2865 case FastMixer_Static: 2866 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2867 break; 2868 } 2869} 2870 2871AudioFlinger::MixerThread::~MixerThread() 2872{ 2873 if (mFastMixer != NULL) { 2874 FastMixerStateQueue *sq = mFastMixer->sq(); 2875 FastMixerState *state = sq->begin(); 2876 if (state->mCommand == FastMixerState::COLD_IDLE) { 2877 int32_t old = android_atomic_inc(&mFastMixerFutex); 2878 if (old == -1) { 2879 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2880 } 2881 } 2882 state->mCommand = FastMixerState::EXIT; 2883 sq->end(); 2884 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2885 mFastMixer->join(); 2886 // Though the fast mixer thread has exited, it's state queue is still valid. 2887 // We'll use that extract the final state which contains one remaining fast track 2888 // corresponding to our sub-mix. 2889 state = sq->begin(); 2890 ALOG_ASSERT(state->mTrackMask == 1); 2891 FastTrack *fastTrack = &state->mFastTracks[0]; 2892 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2893 delete fastTrack->mBufferProvider; 2894 sq->end(false /*didModify*/); 2895 delete mFastMixer; 2896#ifdef AUDIO_WATCHDOG 2897 if (mAudioWatchdog != 0) { 2898 mAudioWatchdog->requestExit(); 2899 mAudioWatchdog->requestExitAndWait(); 2900 mAudioWatchdog.clear(); 2901 } 2902#endif 2903 } 2904 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 2905 delete mAudioMixer; 2906} 2907 2908 2909uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 2910{ 2911 if (mFastMixer != NULL) { 2912 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2913 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 2914 } 2915 return latency; 2916} 2917 2918 2919void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2920{ 2921 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2922} 2923 2924ssize_t AudioFlinger::MixerThread::threadLoop_write() 2925{ 2926 // FIXME we should only do one push per cycle; confirm this is true 2927 // Start the fast mixer if it's not already running 2928 if (mFastMixer != NULL) { 2929 FastMixerStateQueue *sq = mFastMixer->sq(); 2930 FastMixerState *state = sq->begin(); 2931 if (state->mCommand != FastMixerState::MIX_WRITE && 2932 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2933 if (state->mCommand == FastMixerState::COLD_IDLE) { 2934 int32_t old = android_atomic_inc(&mFastMixerFutex); 2935 if (old == -1) { 2936 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2937 } 2938#ifdef AUDIO_WATCHDOG 2939 if (mAudioWatchdog != 0) { 2940 mAudioWatchdog->resume(); 2941 } 2942#endif 2943 } 2944 state->mCommand = FastMixerState::MIX_WRITE; 2945 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 2946 FastMixerDumpState::kSamplingNforLowRamDevice : FastMixerDumpState::kSamplingN); 2947 sq->end(); 2948 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2949 if (kUseFastMixer == FastMixer_Dynamic) { 2950 mNormalSink = mPipeSink; 2951 } 2952 } else { 2953 sq->end(false /*didModify*/); 2954 } 2955 } 2956 return PlaybackThread::threadLoop_write(); 2957} 2958 2959void AudioFlinger::MixerThread::threadLoop_standby() 2960{ 2961 // Idle the fast mixer if it's currently running 2962 if (mFastMixer != NULL) { 2963 FastMixerStateQueue *sq = mFastMixer->sq(); 2964 FastMixerState *state = sq->begin(); 2965 if (!(state->mCommand & FastMixerState::IDLE)) { 2966 state->mCommand = FastMixerState::COLD_IDLE; 2967 state->mColdFutexAddr = &mFastMixerFutex; 2968 state->mColdGen++; 2969 mFastMixerFutex = 0; 2970 sq->end(); 2971 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2972 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2973 if (kUseFastMixer == FastMixer_Dynamic) { 2974 mNormalSink = mOutputSink; 2975 } 2976#ifdef AUDIO_WATCHDOG 2977 if (mAudioWatchdog != 0) { 2978 mAudioWatchdog->pause(); 2979 } 2980#endif 2981 } else { 2982 sq->end(false /*didModify*/); 2983 } 2984 } 2985 PlaybackThread::threadLoop_standby(); 2986} 2987 2988bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 2989{ 2990 return false; 2991} 2992 2993bool AudioFlinger::PlaybackThread::shouldStandby_l() 2994{ 2995 return !mStandby; 2996} 2997 2998bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 2999{ 3000 Mutex::Autolock _l(mLock); 3001 return waitingAsyncCallback_l(); 3002} 3003 3004// shared by MIXER and DIRECT, overridden by DUPLICATING 3005void AudioFlinger::PlaybackThread::threadLoop_standby() 3006{ 3007 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3008 mOutput->stream->common.standby(&mOutput->stream->common); 3009 if (mUseAsyncWrite != 0) { 3010 // discard any pending drain or write ack by incrementing sequence 3011 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3012 mDrainSequence = (mDrainSequence + 2) & ~1; 3013 ALOG_ASSERT(mCallbackThread != 0); 3014 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3015 mCallbackThread->setDraining(mDrainSequence); 3016 } 3017} 3018 3019void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3020{ 3021 ALOGV("signal playback thread"); 3022 broadcast_l(); 3023} 3024 3025void AudioFlinger::MixerThread::threadLoop_mix() 3026{ 3027 // obtain the presentation timestamp of the next output buffer 3028 int64_t pts; 3029 status_t status = INVALID_OPERATION; 3030 3031 if (mNormalSink != 0) { 3032 status = mNormalSink->getNextWriteTimestamp(&pts); 3033 } else { 3034 status = mOutputSink->getNextWriteTimestamp(&pts); 3035 } 3036 3037 if (status != NO_ERROR) { 3038 pts = AudioBufferProvider::kInvalidPTS; 3039 } 3040 3041 // mix buffers... 3042 mAudioMixer->process(pts); 3043 mCurrentWriteLength = mSinkBufferSize; 3044 // increase sleep time progressively when application underrun condition clears. 3045 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3046 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3047 // such that we would underrun the audio HAL. 3048 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 3049 sleepTimeShift--; 3050 } 3051 sleepTime = 0; 3052 standbyTime = systemTime() + standbyDelay; 3053 //TODO: delay standby when effects have a tail 3054} 3055 3056void AudioFlinger::MixerThread::threadLoop_sleepTime() 3057{ 3058 // If no tracks are ready, sleep once for the duration of an output 3059 // buffer size, then write 0s to the output 3060 if (sleepTime == 0) { 3061 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3062 sleepTime = activeSleepTime >> sleepTimeShift; 3063 if (sleepTime < kMinThreadSleepTimeUs) { 3064 sleepTime = kMinThreadSleepTimeUs; 3065 } 3066 // reduce sleep time in case of consecutive application underruns to avoid 3067 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3068 // duration we would end up writing less data than needed by the audio HAL if 3069 // the condition persists. 3070 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3071 sleepTimeShift++; 3072 } 3073 } else { 3074 sleepTime = idleSleepTime; 3075 } 3076 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3077 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3078 // before effects processing or output. 3079 if (mMixerBufferValid) { 3080 memset(mMixerBuffer, 0, mMixerBufferSize); 3081 } else { 3082 memset(mSinkBuffer, 0, mSinkBufferSize); 3083 } 3084 sleepTime = 0; 3085 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3086 "anticipated start"); 3087 } 3088 // TODO add standby time extension fct of effect tail 3089} 3090 3091// prepareTracks_l() must be called with ThreadBase::mLock held 3092AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 3093 Vector< sp<Track> > *tracksToRemove) 3094{ 3095 3096 mixer_state mixerStatus = MIXER_IDLE; 3097 // find out which tracks need to be processed 3098 size_t count = mActiveTracks.size(); 3099 size_t mixedTracks = 0; 3100 size_t tracksWithEffect = 0; 3101 // counts only _active_ fast tracks 3102 size_t fastTracks = 0; 3103 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 3104 3105 float masterVolume = mMasterVolume; 3106 bool masterMute = mMasterMute; 3107 3108 if (masterMute) { 3109 masterVolume = 0; 3110 } 3111 // Delegate master volume control to effect in output mix effect chain if needed 3112 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 3113 if (chain != 0) { 3114 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 3115 chain->setVolume_l(&v, &v); 3116 masterVolume = (float)((v + (1 << 23)) >> 24); 3117 chain.clear(); 3118 } 3119 3120 // prepare a new state to push 3121 FastMixerStateQueue *sq = NULL; 3122 FastMixerState *state = NULL; 3123 bool didModify = false; 3124 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 3125 if (mFastMixer != NULL) { 3126 sq = mFastMixer->sq(); 3127 state = sq->begin(); 3128 } 3129 3130 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 3131 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 3132 3133 for (size_t i=0 ; i<count ; i++) { 3134 const sp<Track> t = mActiveTracks[i].promote(); 3135 if (t == 0) { 3136 continue; 3137 } 3138 3139 // this const just means the local variable doesn't change 3140 Track* const track = t.get(); 3141 3142 // process fast tracks 3143 if (track->isFastTrack()) { 3144 3145 // It's theoretically possible (though unlikely) for a fast track to be created 3146 // and then removed within the same normal mix cycle. This is not a problem, as 3147 // the track never becomes active so it's fast mixer slot is never touched. 3148 // The converse, of removing an (active) track and then creating a new track 3149 // at the identical fast mixer slot within the same normal mix cycle, 3150 // is impossible because the slot isn't marked available until the end of each cycle. 3151 int j = track->mFastIndex; 3152 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 3153 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 3154 FastTrack *fastTrack = &state->mFastTracks[j]; 3155 3156 // Determine whether the track is currently in underrun condition, 3157 // and whether it had a recent underrun. 3158 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 3159 FastTrackUnderruns underruns = ftDump->mUnderruns; 3160 uint32_t recentFull = (underruns.mBitFields.mFull - 3161 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 3162 uint32_t recentPartial = (underruns.mBitFields.mPartial - 3163 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 3164 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 3165 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 3166 uint32_t recentUnderruns = recentPartial + recentEmpty; 3167 track->mObservedUnderruns = underruns; 3168 // don't count underruns that occur while stopping or pausing 3169 // or stopped which can occur when flush() is called while active 3170 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 3171 recentUnderruns > 0) { 3172 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 3173 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 3174 } 3175 3176 // This is similar to the state machine for normal tracks, 3177 // with a few modifications for fast tracks. 3178 bool isActive = true; 3179 switch (track->mState) { 3180 case TrackBase::STOPPING_1: 3181 // track stays active in STOPPING_1 state until first underrun 3182 if (recentUnderruns > 0 || track->isTerminated()) { 3183 track->mState = TrackBase::STOPPING_2; 3184 } 3185 break; 3186 case TrackBase::PAUSING: 3187 // ramp down is not yet implemented 3188 track->setPaused(); 3189 break; 3190 case TrackBase::RESUMING: 3191 // ramp up is not yet implemented 3192 track->mState = TrackBase::ACTIVE; 3193 break; 3194 case TrackBase::ACTIVE: 3195 if (recentFull > 0 || recentPartial > 0) { 3196 // track has provided at least some frames recently: reset retry count 3197 track->mRetryCount = kMaxTrackRetries; 3198 } 3199 if (recentUnderruns == 0) { 3200 // no recent underruns: stay active 3201 break; 3202 } 3203 // there has recently been an underrun of some kind 3204 if (track->sharedBuffer() == 0) { 3205 // were any of the recent underruns "empty" (no frames available)? 3206 if (recentEmpty == 0) { 3207 // no, then ignore the partial underruns as they are allowed indefinitely 3208 break; 3209 } 3210 // there has recently been an "empty" underrun: decrement the retry counter 3211 if (--(track->mRetryCount) > 0) { 3212 break; 3213 } 3214 // indicate to client process that the track was disabled because of underrun; 3215 // it will then automatically call start() when data is available 3216 android_atomic_or(CBLK_DISABLED, &track->mCblk->mFlags); 3217 // remove from active list, but state remains ACTIVE [confusing but true] 3218 isActive = false; 3219 break; 3220 } 3221 // fall through 3222 case TrackBase::STOPPING_2: 3223 case TrackBase::PAUSED: 3224 case TrackBase::STOPPED: 3225 case TrackBase::FLUSHED: // flush() while active 3226 // Check for presentation complete if track is inactive 3227 // We have consumed all the buffers of this track. 3228 // This would be incomplete if we auto-paused on underrun 3229 { 3230 size_t audioHALFrames = 3231 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3232 size_t framesWritten = mBytesWritten / mFrameSize; 3233 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 3234 // track stays in active list until presentation is complete 3235 break; 3236 } 3237 } 3238 if (track->isStopping_2()) { 3239 track->mState = TrackBase::STOPPED; 3240 } 3241 if (track->isStopped()) { 3242 // Can't reset directly, as fast mixer is still polling this track 3243 // track->reset(); 3244 // So instead mark this track as needing to be reset after push with ack 3245 resetMask |= 1 << i; 3246 } 3247 isActive = false; 3248 break; 3249 case TrackBase::IDLE: 3250 default: 3251 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 3252 } 3253 3254 if (isActive) { 3255 // was it previously inactive? 3256 if (!(state->mTrackMask & (1 << j))) { 3257 ExtendedAudioBufferProvider *eabp = track; 3258 VolumeProvider *vp = track; 3259 fastTrack->mBufferProvider = eabp; 3260 fastTrack->mVolumeProvider = vp; 3261 fastTrack->mChannelMask = track->mChannelMask; 3262 fastTrack->mFormat = track->mFormat; 3263 fastTrack->mGeneration++; 3264 state->mTrackMask |= 1 << j; 3265 didModify = true; 3266 // no acknowledgement required for newly active tracks 3267 } 3268 // cache the combined master volume and stream type volume for fast mixer; this 3269 // lacks any synchronization or barrier so VolumeProvider may read a stale value 3270 track->mCachedVolume = masterVolume * mStreamTypes[track->streamType()].volume; 3271 ++fastTracks; 3272 } else { 3273 // was it previously active? 3274 if (state->mTrackMask & (1 << j)) { 3275 fastTrack->mBufferProvider = NULL; 3276 fastTrack->mGeneration++; 3277 state->mTrackMask &= ~(1 << j); 3278 didModify = true; 3279 // If any fast tracks were removed, we must wait for acknowledgement 3280 // because we're about to decrement the last sp<> on those tracks. 3281 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3282 } else { 3283 LOG_ALWAYS_FATAL("fast track %d should have been active", j); 3284 } 3285 tracksToRemove->add(track); 3286 // Avoids a misleading display in dumpsys 3287 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3288 } 3289 continue; 3290 } 3291 3292 { // local variable scope to avoid goto warning 3293 3294 audio_track_cblk_t* cblk = track->cblk(); 3295 3296 // The first time a track is added we wait 3297 // for all its buffers to be filled before processing it 3298 int name = track->name(); 3299 // make sure that we have enough frames to mix one full buffer. 3300 // enforce this condition only once to enable draining the buffer in case the client 3301 // app does not call stop() and relies on underrun to stop: 3302 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3303 // during last round 3304 size_t desiredFrames; 3305 uint32_t sr = track->sampleRate(); 3306 if (sr == mSampleRate) { 3307 desiredFrames = mNormalFrameCount; 3308 } else { 3309 // +1 for rounding and +1 for additional sample needed for interpolation 3310 desiredFrames = (mNormalFrameCount * sr) / mSampleRate + 1 + 1; 3311 // add frames already consumed but not yet released by the resampler 3312 // because mAudioTrackServerProxy->framesReady() will include these frames 3313 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3314#if 0 3315 // the minimum track buffer size is normally twice the number of frames necessary 3316 // to fill one buffer and the resampler should not leave more than one buffer worth 3317 // of unreleased frames after each pass, but just in case... 3318 ALOG_ASSERT(desiredFrames <= cblk->frameCount_); 3319#endif 3320 } 3321 uint32_t minFrames = 1; 3322 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3323 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3324 minFrames = desiredFrames; 3325 } 3326 3327 size_t framesReady = track->framesReady(); 3328 if ((framesReady >= minFrames) && track->isReady() && 3329 !track->isPaused() && !track->isTerminated()) 3330 { 3331 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 3332 3333 mixedTracks++; 3334 3335 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 3336 // there is an effect chain connected to the track 3337 chain.clear(); 3338 if (track->mainBuffer() != mSinkBuffer && 3339 track->mainBuffer() != mMixerBuffer) { 3340 if (mEffectBufferEnabled) { 3341 mEffectBufferValid = true; // Later can set directly. 3342 } 3343 chain = getEffectChain_l(track->sessionId()); 3344 // Delegate volume control to effect in track effect chain if needed 3345 if (chain != 0) { 3346 tracksWithEffect++; 3347 } else { 3348 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 3349 "session %d", 3350 name, track->sessionId()); 3351 } 3352 } 3353 3354 3355 int param = AudioMixer::VOLUME; 3356 if (track->mFillingUpStatus == Track::FS_FILLED) { 3357 // no ramp for the first volume setting 3358 track->mFillingUpStatus = Track::FS_ACTIVE; 3359 if (track->mState == TrackBase::RESUMING) { 3360 track->mState = TrackBase::ACTIVE; 3361 param = AudioMixer::RAMP_VOLUME; 3362 } 3363 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3364 // FIXME should not make a decision based on mServer 3365 } else if (cblk->mServer != 0) { 3366 // If the track is stopped before the first frame was mixed, 3367 // do not apply ramp 3368 param = AudioMixer::RAMP_VOLUME; 3369 } 3370 3371 // compute volume for this track 3372 uint32_t vl, vr, va; 3373 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 3374 vl = vr = va = 0; 3375 if (track->isPausing()) { 3376 track->setPaused(); 3377 } 3378 } else { 3379 3380 // read original volumes with volume control 3381 float typeVolume = mStreamTypes[track->streamType()].volume; 3382 float v = masterVolume * typeVolume; 3383 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3384 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3385 float vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 3386 float vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 3387 // track volumes come from shared memory, so can't be trusted and must be clamped 3388 if (vlf > GAIN_FLOAT_UNITY) { 3389 ALOGV("Track left volume out of range: %.3g", vlf); 3390 vlf = GAIN_FLOAT_UNITY; 3391 } 3392 if (vrf > GAIN_FLOAT_UNITY) { 3393 ALOGV("Track right volume out of range: %.3g", vrf); 3394 vrf = GAIN_FLOAT_UNITY; 3395 } 3396 // now apply the master volume and stream type volume 3397 // FIXME we're losing the wonderful dynamic range in the minifloat representation 3398 float v8_24 = v * (MAX_GAIN_INT * MAX_GAIN_INT); 3399 vl = (uint32_t) (v8_24 * vlf); 3400 vr = (uint32_t) (v8_24 * vrf); 3401 // assuming master volume and stream type volume each go up to 1.0, 3402 // vl and vr are now in 8.24 format 3403 3404 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 3405 // send level comes from shared memory and so may be corrupt 3406 if (sendLevel > MAX_GAIN_INT) { 3407 ALOGV("Track send level out of range: %04X", sendLevel); 3408 sendLevel = MAX_GAIN_INT; 3409 } 3410 va = (uint32_t)(v * sendLevel); 3411 } 3412 3413 // Delegate volume control to effect in track effect chain if needed 3414 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3415 // Do not ramp volume if volume is controlled by effect 3416 param = AudioMixer::VOLUME; 3417 track->mHasVolumeController = true; 3418 } else { 3419 // force no volume ramp when volume controller was just disabled or removed 3420 // from effect chain to avoid volume spike 3421 if (track->mHasVolumeController) { 3422 param = AudioMixer::VOLUME; 3423 } 3424 track->mHasVolumeController = false; 3425 } 3426 3427 // FIXME Use float 3428 // Convert volumes from 8.24 to 4.12 format 3429 // This additional clamping is needed in case chain->setVolume_l() overshot 3430 vl = (vl + (1 << 11)) >> 12; 3431 if (vl > MAX_GAIN_INT) { 3432 vl = MAX_GAIN_INT; 3433 } 3434 vr = (vr + (1 << 11)) >> 12; 3435 if (vr > MAX_GAIN_INT) { 3436 vr = MAX_GAIN_INT; 3437 } 3438 3439 if (va > MAX_GAIN_INT) { 3440 va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3441 } 3442 3443 // XXX: these things DON'T need to be done each time 3444 mAudioMixer->setBufferProvider(name, track); 3445 mAudioMixer->enable(name); 3446 3447 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)(uintptr_t)vl); 3448 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)(uintptr_t)vr); 3449 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)(uintptr_t)va); 3450 mAudioMixer->setParameter( 3451 name, 3452 AudioMixer::TRACK, 3453 AudioMixer::FORMAT, (void *)track->format()); 3454 mAudioMixer->setParameter( 3455 name, 3456 AudioMixer::TRACK, 3457 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 3458 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 3459 uint32_t maxSampleRate = mSampleRate * 2; 3460 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 3461 if (reqSampleRate == 0) { 3462 reqSampleRate = mSampleRate; 3463 } else if (reqSampleRate > maxSampleRate) { 3464 reqSampleRate = maxSampleRate; 3465 } 3466 mAudioMixer->setParameter( 3467 name, 3468 AudioMixer::RESAMPLE, 3469 AudioMixer::SAMPLE_RATE, 3470 (void *)(uintptr_t)reqSampleRate); 3471 /* 3472 * Select the appropriate output buffer for the track. 3473 * 3474 * Tracks with effects go into their own effects chain buffer 3475 * and from there into either mEffectBuffer or mSinkBuffer. 3476 * 3477 * Other tracks can use mMixerBuffer for higher precision 3478 * channel accumulation. If this buffer is enabled 3479 * (mMixerBufferEnabled true), then selected tracks will accumulate 3480 * into it. 3481 * 3482 */ 3483 if (mMixerBufferEnabled 3484 && (track->mainBuffer() == mSinkBuffer 3485 || track->mainBuffer() == mMixerBuffer)) { 3486 mAudioMixer->setParameter( 3487 name, 3488 AudioMixer::TRACK, 3489 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 3490 mAudioMixer->setParameter( 3491 name, 3492 AudioMixer::TRACK, 3493 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 3494 // TODO: override track->mainBuffer()? 3495 mMixerBufferValid = true; 3496 } else { 3497 mAudioMixer->setParameter( 3498 name, 3499 AudioMixer::TRACK, 3500 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 3501 mAudioMixer->setParameter( 3502 name, 3503 AudioMixer::TRACK, 3504 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3505 } 3506 mAudioMixer->setParameter( 3507 name, 3508 AudioMixer::TRACK, 3509 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3510 3511 // reset retry count 3512 track->mRetryCount = kMaxTrackRetries; 3513 3514 // If one track is ready, set the mixer ready if: 3515 // - the mixer was not ready during previous round OR 3516 // - no other track is not ready 3517 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3518 mixerStatus != MIXER_TRACKS_ENABLED) { 3519 mixerStatus = MIXER_TRACKS_READY; 3520 } 3521 } else { 3522 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 3523 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 3524 } 3525 // clear effect chain input buffer if an active track underruns to avoid sending 3526 // previous audio buffer again to effects 3527 chain = getEffectChain_l(track->sessionId()); 3528 if (chain != 0) { 3529 chain->clearInputBuffer(); 3530 } 3531 3532 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 3533 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3534 track->isStopped() || track->isPaused()) { 3535 // We have consumed all the buffers of this track. 3536 // Remove it from the list of active tracks. 3537 // TODO: use actual buffer filling status instead of latency when available from 3538 // audio HAL 3539 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3540 size_t framesWritten = mBytesWritten / mFrameSize; 3541 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 3542 if (track->isStopped()) { 3543 track->reset(); 3544 } 3545 tracksToRemove->add(track); 3546 } 3547 } else { 3548 // No buffers for this track. Give it a few chances to 3549 // fill a buffer, then remove it from active list. 3550 if (--(track->mRetryCount) <= 0) { 3551 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3552 tracksToRemove->add(track); 3553 // indicate to client process that the track was disabled because of underrun; 3554 // it will then automatically call start() when data is available 3555 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 3556 // If one track is not ready, mark the mixer also not ready if: 3557 // - the mixer was ready during previous round OR 3558 // - no other track is ready 3559 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3560 mixerStatus != MIXER_TRACKS_READY) { 3561 mixerStatus = MIXER_TRACKS_ENABLED; 3562 } 3563 } 3564 mAudioMixer->disable(name); 3565 } 3566 3567 } // local variable scope to avoid goto warning 3568track_is_ready: ; 3569 3570 } 3571 3572 // Push the new FastMixer state if necessary 3573 bool pauseAudioWatchdog = false; 3574 if (didModify) { 3575 state->mFastTracksGen++; 3576 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3577 if (kUseFastMixer == FastMixer_Dynamic && 3578 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3579 state->mCommand = FastMixerState::COLD_IDLE; 3580 state->mColdFutexAddr = &mFastMixerFutex; 3581 state->mColdGen++; 3582 mFastMixerFutex = 0; 3583 if (kUseFastMixer == FastMixer_Dynamic) { 3584 mNormalSink = mOutputSink; 3585 } 3586 // If we go into cold idle, need to wait for acknowledgement 3587 // so that fast mixer stops doing I/O. 3588 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3589 pauseAudioWatchdog = true; 3590 } 3591 } 3592 if (sq != NULL) { 3593 sq->end(didModify); 3594 sq->push(block); 3595 } 3596#ifdef AUDIO_WATCHDOG 3597 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3598 mAudioWatchdog->pause(); 3599 } 3600#endif 3601 3602 // Now perform the deferred reset on fast tracks that have stopped 3603 while (resetMask != 0) { 3604 size_t i = __builtin_ctz(resetMask); 3605 ALOG_ASSERT(i < count); 3606 resetMask &= ~(1 << i); 3607 sp<Track> t = mActiveTracks[i].promote(); 3608 if (t == 0) { 3609 continue; 3610 } 3611 Track* track = t.get(); 3612 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3613 track->reset(); 3614 } 3615 3616 // remove all the tracks that need to be... 3617 removeTracks_l(*tracksToRemove); 3618 3619 // sink or mix buffer must be cleared if all tracks are connected to an 3620 // effect chain as in this case the mixer will not write to the sink or mix buffer 3621 // and track effects will accumulate into it 3622 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3623 (mixedTracks == 0 && fastTracks > 0))) { 3624 // FIXME as a performance optimization, should remember previous zero status 3625 if (mMixerBufferValid) { 3626 memset(mMixerBuffer, 0, mMixerBufferSize); 3627 // TODO: In testing, mSinkBuffer below need not be cleared because 3628 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 3629 // after mixing. 3630 // 3631 // To enforce this guarantee: 3632 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 3633 // (mixedTracks == 0 && fastTracks > 0)) 3634 // must imply MIXER_TRACKS_READY. 3635 // Later, we may clear buffers regardless, and skip much of this logic. 3636 } 3637 // TODO - either mEffectBuffer or mSinkBuffer needs to be cleared. 3638 if (mEffectBufferValid) { 3639 memset(mEffectBuffer, 0, mEffectBufferSize); 3640 } 3641 // FIXME as a performance optimization, should remember previous zero status 3642 memset(mSinkBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3643 } 3644 3645 // if any fast tracks, then status is ready 3646 mMixerStatusIgnoringFastTracks = mixerStatus; 3647 if (fastTracks > 0) { 3648 mixerStatus = MIXER_TRACKS_READY; 3649 } 3650 return mixerStatus; 3651} 3652 3653// getTrackName_l() must be called with ThreadBase::mLock held 3654int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 3655 audio_format_t format, int sessionId) 3656{ 3657 return mAudioMixer->getTrackName(channelMask, format, sessionId); 3658} 3659 3660// deleteTrackName_l() must be called with ThreadBase::mLock held 3661void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3662{ 3663 ALOGV("remove track (%d) and delete from mixer", name); 3664 mAudioMixer->deleteTrackName(name); 3665} 3666 3667// checkForNewParameter_l() must be called with ThreadBase::mLock held 3668bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 3669 status_t& status) 3670{ 3671 bool reconfig = false; 3672 3673 status = NO_ERROR; 3674 3675 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3676 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3677 if (mFastMixer != NULL) { 3678 FastMixerStateQueue *sq = mFastMixer->sq(); 3679 FastMixerState *state = sq->begin(); 3680 if (!(state->mCommand & FastMixerState::IDLE)) { 3681 previousCommand = state->mCommand; 3682 state->mCommand = FastMixerState::HOT_IDLE; 3683 sq->end(); 3684 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3685 } else { 3686 sq->end(false /*didModify*/); 3687 } 3688 } 3689 3690 AudioParameter param = AudioParameter(keyValuePair); 3691 int value; 3692 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3693 reconfig = true; 3694 } 3695 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3696 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3697 status = BAD_VALUE; 3698 } else { 3699 // no need to save value, since it's constant 3700 reconfig = true; 3701 } 3702 } 3703 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3704 if ((audio_channel_mask_t) value != AUDIO_CHANNEL_OUT_STEREO) { 3705 status = BAD_VALUE; 3706 } else { 3707 // no need to save value, since it's constant 3708 reconfig = true; 3709 } 3710 } 3711 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3712 // do not accept frame count changes if tracks are open as the track buffer 3713 // size depends on frame count and correct behavior would not be guaranteed 3714 // if frame count is changed after track creation 3715 if (!mTracks.isEmpty()) { 3716 status = INVALID_OPERATION; 3717 } else { 3718 reconfig = true; 3719 } 3720 } 3721 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3722#ifdef ADD_BATTERY_DATA 3723 // when changing the audio output device, call addBatteryData to notify 3724 // the change 3725 if (mOutDevice != value) { 3726 uint32_t params = 0; 3727 // check whether speaker is on 3728 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3729 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3730 } 3731 3732 audio_devices_t deviceWithoutSpeaker 3733 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3734 // check if any other device (except speaker) is on 3735 if (value & deviceWithoutSpeaker ) { 3736 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3737 } 3738 3739 if (params != 0) { 3740 addBatteryData(params); 3741 } 3742 } 3743#endif 3744 3745 // forward device change to effects that have requested to be 3746 // aware of attached audio device. 3747 if (value != AUDIO_DEVICE_NONE) { 3748 mOutDevice = value; 3749 for (size_t i = 0; i < mEffectChains.size(); i++) { 3750 mEffectChains[i]->setDevice_l(mOutDevice); 3751 } 3752 } 3753 } 3754 3755 if (status == NO_ERROR) { 3756 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3757 keyValuePair.string()); 3758 if (!mStandby && status == INVALID_OPERATION) { 3759 mOutput->stream->common.standby(&mOutput->stream->common); 3760 mStandby = true; 3761 mBytesWritten = 0; 3762 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3763 keyValuePair.string()); 3764 } 3765 if (status == NO_ERROR && reconfig) { 3766 readOutputParameters_l(); 3767 delete mAudioMixer; 3768 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3769 for (size_t i = 0; i < mTracks.size() ; i++) { 3770 int name = getTrackName_l(mTracks[i]->mChannelMask, 3771 mTracks[i]->mFormat, mTracks[i]->mSessionId); 3772 if (name < 0) { 3773 break; 3774 } 3775 mTracks[i]->mName = name; 3776 } 3777 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3778 } 3779 } 3780 3781 if (!(previousCommand & FastMixerState::IDLE)) { 3782 ALOG_ASSERT(mFastMixer != NULL); 3783 FastMixerStateQueue *sq = mFastMixer->sq(); 3784 FastMixerState *state = sq->begin(); 3785 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3786 state->mCommand = previousCommand; 3787 sq->end(); 3788 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3789 } 3790 3791 return reconfig; 3792} 3793 3794 3795void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3796{ 3797 const size_t SIZE = 256; 3798 char buffer[SIZE]; 3799 String8 result; 3800 3801 PlaybackThread::dumpInternals(fd, args); 3802 3803 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 3804 3805 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3806 const FastMixerDumpState copy(mFastMixerDumpState); 3807 copy.dump(fd); 3808 3809#ifdef STATE_QUEUE_DUMP 3810 // Similar for state queue 3811 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3812 observerCopy.dump(fd); 3813 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3814 mutatorCopy.dump(fd); 3815#endif 3816 3817#ifdef TEE_SINK 3818 // Write the tee output to a .wav file 3819 dumpTee(fd, mTeeSource, mId); 3820#endif 3821 3822#ifdef AUDIO_WATCHDOG 3823 if (mAudioWatchdog != 0) { 3824 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3825 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3826 wdCopy.dump(fd); 3827 } 3828#endif 3829} 3830 3831uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3832{ 3833 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3834} 3835 3836uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3837{ 3838 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3839} 3840 3841void AudioFlinger::MixerThread::cacheParameters_l() 3842{ 3843 PlaybackThread::cacheParameters_l(); 3844 3845 // FIXME: Relaxed timing because of a certain device that can't meet latency 3846 // Should be reduced to 2x after the vendor fixes the driver issue 3847 // increase threshold again due to low power audio mode. The way this warning 3848 // threshold is calculated and its usefulness should be reconsidered anyway. 3849 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3850} 3851 3852// ---------------------------------------------------------------------------- 3853 3854AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3855 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3856 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3857 // mLeftVolFloat, mRightVolFloat 3858{ 3859} 3860 3861AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3862 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 3863 ThreadBase::type_t type) 3864 : PlaybackThread(audioFlinger, output, id, device, type) 3865 // mLeftVolFloat, mRightVolFloat 3866{ 3867} 3868 3869AudioFlinger::DirectOutputThread::~DirectOutputThread() 3870{ 3871} 3872 3873void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 3874{ 3875 audio_track_cblk_t* cblk = track->cblk(); 3876 float left, right; 3877 3878 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 3879 left = right = 0; 3880 } else { 3881 float typeVolume = mStreamTypes[track->streamType()].volume; 3882 float v = mMasterVolume * typeVolume; 3883 AudioTrackServerProxy *proxy = track->mAudioTrackServerProxy; 3884 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 3885 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 3886 if (left > GAIN_FLOAT_UNITY) { 3887 left = GAIN_FLOAT_UNITY; 3888 } 3889 left *= v; 3890 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 3891 if (right > GAIN_FLOAT_UNITY) { 3892 right = GAIN_FLOAT_UNITY; 3893 } 3894 right *= v; 3895 } 3896 3897 if (lastTrack) { 3898 if (left != mLeftVolFloat || right != mRightVolFloat) { 3899 mLeftVolFloat = left; 3900 mRightVolFloat = right; 3901 3902 // Convert volumes from float to 8.24 3903 uint32_t vl = (uint32_t)(left * (1 << 24)); 3904 uint32_t vr = (uint32_t)(right * (1 << 24)); 3905 3906 // Delegate volume control to effect in track effect chain if needed 3907 // only one effect chain can be present on DirectOutputThread, so if 3908 // there is one, the track is connected to it 3909 if (!mEffectChains.isEmpty()) { 3910 mEffectChains[0]->setVolume_l(&vl, &vr); 3911 left = (float)vl / (1 << 24); 3912 right = (float)vr / (1 << 24); 3913 } 3914 if (mOutput->stream->set_volume) { 3915 mOutput->stream->set_volume(mOutput->stream, left, right); 3916 } 3917 } 3918 } 3919} 3920 3921 3922AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3923 Vector< sp<Track> > *tracksToRemove 3924) 3925{ 3926 size_t count = mActiveTracks.size(); 3927 mixer_state mixerStatus = MIXER_IDLE; 3928 3929 // find out which tracks need to be processed 3930 for (size_t i = 0; i < count; i++) { 3931 sp<Track> t = mActiveTracks[i].promote(); 3932 // The track died recently 3933 if (t == 0) { 3934 continue; 3935 } 3936 3937 Track* const track = t.get(); 3938 audio_track_cblk_t* cblk = track->cblk(); 3939 // Only consider last track started for volume and mixer state control. 3940 // In theory an older track could underrun and restart after the new one starts 3941 // but as we only care about the transition phase between two tracks on a 3942 // direct output, it is not a problem to ignore the underrun case. 3943 sp<Track> l = mLatestActiveTrack.promote(); 3944 bool last = l.get() == track; 3945 3946 // The first time a track is added we wait 3947 // for all its buffers to be filled before processing it 3948 uint32_t minFrames; 3949 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3950 minFrames = mNormalFrameCount; 3951 } else { 3952 minFrames = 1; 3953 } 3954 3955 if ((track->framesReady() >= minFrames) && track->isReady() && 3956 !track->isPaused() && !track->isTerminated()) 3957 { 3958 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 3959 3960 if (track->mFillingUpStatus == Track::FS_FILLED) { 3961 track->mFillingUpStatus = Track::FS_ACTIVE; 3962 // make sure processVolume_l() will apply new volume even if 0 3963 mLeftVolFloat = mRightVolFloat = -1.0; 3964 if (track->mState == TrackBase::RESUMING) { 3965 track->mState = TrackBase::ACTIVE; 3966 } 3967 } 3968 3969 // compute volume for this track 3970 processVolume_l(track, last); 3971 if (last) { 3972 // reset retry count 3973 track->mRetryCount = kMaxTrackRetriesDirect; 3974 mActiveTrack = t; 3975 mixerStatus = MIXER_TRACKS_READY; 3976 } 3977 } else { 3978 // clear effect chain input buffer if the last active track started underruns 3979 // to avoid sending previous audio buffer again to effects 3980 if (!mEffectChains.isEmpty() && last) { 3981 mEffectChains[0]->clearInputBuffer(); 3982 } 3983 3984 ALOGVV("track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 3985 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3986 track->isStopped() || track->isPaused()) { 3987 // We have consumed all the buffers of this track. 3988 // Remove it from the list of active tracks. 3989 // TODO: implement behavior for compressed audio 3990 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 3991 size_t framesWritten = mBytesWritten / mFrameSize; 3992 if (mStandby || !last || 3993 track->presentationComplete(framesWritten, audioHALFrames)) { 3994 if (track->isStopped()) { 3995 track->reset(); 3996 } 3997 tracksToRemove->add(track); 3998 } 3999 } else { 4000 // No buffers for this track. Give it a few chances to 4001 // fill a buffer, then remove it from active list. 4002 // Only consider last track started for mixer state control 4003 if (--(track->mRetryCount) <= 0) { 4004 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 4005 tracksToRemove->add(track); 4006 // indicate to client process that the track was disabled because of underrun; 4007 // it will then automatically call start() when data is available 4008 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4009 } else if (last) { 4010 mixerStatus = MIXER_TRACKS_ENABLED; 4011 } 4012 } 4013 } 4014 } 4015 4016 // remove all the tracks that need to be... 4017 removeTracks_l(*tracksToRemove); 4018 4019 return mixerStatus; 4020} 4021 4022void AudioFlinger::DirectOutputThread::threadLoop_mix() 4023{ 4024 size_t frameCount = mFrameCount; 4025 int8_t *curBuf = (int8_t *)mSinkBuffer; 4026 // output audio to hardware 4027 while (frameCount) { 4028 AudioBufferProvider::Buffer buffer; 4029 buffer.frameCount = frameCount; 4030 mActiveTrack->getNextBuffer(&buffer); 4031 if (buffer.raw == NULL) { 4032 memset(curBuf, 0, frameCount * mFrameSize); 4033 break; 4034 } 4035 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 4036 frameCount -= buffer.frameCount; 4037 curBuf += buffer.frameCount * mFrameSize; 4038 mActiveTrack->releaseBuffer(&buffer); 4039 } 4040 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 4041 sleepTime = 0; 4042 standbyTime = systemTime() + standbyDelay; 4043 mActiveTrack.clear(); 4044} 4045 4046void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 4047{ 4048 if (sleepTime == 0) { 4049 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4050 sleepTime = activeSleepTime; 4051 } else { 4052 sleepTime = idleSleepTime; 4053 } 4054 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 4055 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 4056 sleepTime = 0; 4057 } 4058} 4059 4060// getTrackName_l() must be called with ThreadBase::mLock held 4061int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 4062 audio_format_t format __unused, int sessionId __unused) 4063{ 4064 return 0; 4065} 4066 4067// deleteTrackName_l() must be called with ThreadBase::mLock held 4068void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 4069{ 4070} 4071 4072// checkForNewParameter_l() must be called with ThreadBase::mLock held 4073bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 4074 status_t& status) 4075{ 4076 bool reconfig = false; 4077 4078 status = NO_ERROR; 4079 4080 AudioParameter param = AudioParameter(keyValuePair); 4081 int value; 4082 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4083 // forward device change to effects that have requested to be 4084 // aware of attached audio device. 4085 if (value != AUDIO_DEVICE_NONE) { 4086 mOutDevice = value; 4087 for (size_t i = 0; i < mEffectChains.size(); i++) { 4088 mEffectChains[i]->setDevice_l(mOutDevice); 4089 } 4090 } 4091 } 4092 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4093 // do not accept frame count changes if tracks are open as the track buffer 4094 // size depends on frame count and correct behavior would not be garantied 4095 // if frame count is changed after track creation 4096 if (!mTracks.isEmpty()) { 4097 status = INVALID_OPERATION; 4098 } else { 4099 reconfig = true; 4100 } 4101 } 4102 if (status == NO_ERROR) { 4103 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4104 keyValuePair.string()); 4105 if (!mStandby && status == INVALID_OPERATION) { 4106 mOutput->stream->common.standby(&mOutput->stream->common); 4107 mStandby = true; 4108 mBytesWritten = 0; 4109 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 4110 keyValuePair.string()); 4111 } 4112 if (status == NO_ERROR && reconfig) { 4113 readOutputParameters_l(); 4114 sendIoConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 4115 } 4116 } 4117 4118 return reconfig; 4119} 4120 4121uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 4122{ 4123 uint32_t time; 4124 if (audio_is_linear_pcm(mFormat)) { 4125 time = PlaybackThread::activeSleepTimeUs(); 4126 } else { 4127 time = 10000; 4128 } 4129 return time; 4130} 4131 4132uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 4133{ 4134 uint32_t time; 4135 if (audio_is_linear_pcm(mFormat)) { 4136 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 4137 } else { 4138 time = 10000; 4139 } 4140 return time; 4141} 4142 4143uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 4144{ 4145 uint32_t time; 4146 if (audio_is_linear_pcm(mFormat)) { 4147 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 4148 } else { 4149 time = 10000; 4150 } 4151 return time; 4152} 4153 4154void AudioFlinger::DirectOutputThread::cacheParameters_l() 4155{ 4156 PlaybackThread::cacheParameters_l(); 4157 4158 // use shorter standby delay as on normal output to release 4159 // hardware resources as soon as possible 4160 if (audio_is_linear_pcm(mFormat)) { 4161 standbyDelay = microseconds(activeSleepTime*2); 4162 } else { 4163 standbyDelay = kOffloadStandbyDelayNs; 4164 } 4165} 4166 4167// ---------------------------------------------------------------------------- 4168 4169AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 4170 const wp<AudioFlinger::PlaybackThread>& playbackThread) 4171 : Thread(false /*canCallJava*/), 4172 mPlaybackThread(playbackThread), 4173 mWriteAckSequence(0), 4174 mDrainSequence(0) 4175{ 4176} 4177 4178AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 4179{ 4180} 4181 4182void AudioFlinger::AsyncCallbackThread::onFirstRef() 4183{ 4184 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 4185} 4186 4187bool AudioFlinger::AsyncCallbackThread::threadLoop() 4188{ 4189 while (!exitPending()) { 4190 uint32_t writeAckSequence; 4191 uint32_t drainSequence; 4192 4193 { 4194 Mutex::Autolock _l(mLock); 4195 while (!((mWriteAckSequence & 1) || 4196 (mDrainSequence & 1) || 4197 exitPending())) { 4198 mWaitWorkCV.wait(mLock); 4199 } 4200 4201 if (exitPending()) { 4202 break; 4203 } 4204 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 4205 mWriteAckSequence, mDrainSequence); 4206 writeAckSequence = mWriteAckSequence; 4207 mWriteAckSequence &= ~1; 4208 drainSequence = mDrainSequence; 4209 mDrainSequence &= ~1; 4210 } 4211 { 4212 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 4213 if (playbackThread != 0) { 4214 if (writeAckSequence & 1) { 4215 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 4216 } 4217 if (drainSequence & 1) { 4218 playbackThread->resetDraining(drainSequence >> 1); 4219 } 4220 } 4221 } 4222 } 4223 return false; 4224} 4225 4226void AudioFlinger::AsyncCallbackThread::exit() 4227{ 4228 ALOGV("AsyncCallbackThread::exit"); 4229 Mutex::Autolock _l(mLock); 4230 requestExit(); 4231 mWaitWorkCV.broadcast(); 4232} 4233 4234void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 4235{ 4236 Mutex::Autolock _l(mLock); 4237 // bit 0 is cleared 4238 mWriteAckSequence = sequence << 1; 4239} 4240 4241void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 4242{ 4243 Mutex::Autolock _l(mLock); 4244 // ignore unexpected callbacks 4245 if (mWriteAckSequence & 2) { 4246 mWriteAckSequence |= 1; 4247 mWaitWorkCV.signal(); 4248 } 4249} 4250 4251void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 4252{ 4253 Mutex::Autolock _l(mLock); 4254 // bit 0 is cleared 4255 mDrainSequence = sequence << 1; 4256} 4257 4258void AudioFlinger::AsyncCallbackThread::resetDraining() 4259{ 4260 Mutex::Autolock _l(mLock); 4261 // ignore unexpected callbacks 4262 if (mDrainSequence & 2) { 4263 mDrainSequence |= 1; 4264 mWaitWorkCV.signal(); 4265 } 4266} 4267 4268 4269// ---------------------------------------------------------------------------- 4270AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 4271 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 4272 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD), 4273 mHwPaused(false), 4274 mFlushPending(false), 4275 mPausedBytesRemaining(0) 4276{ 4277 //FIXME: mStandby should be set to true by ThreadBase constructor 4278 mStandby = true; 4279} 4280 4281void AudioFlinger::OffloadThread::threadLoop_exit() 4282{ 4283 if (mFlushPending || mHwPaused) { 4284 // If a flush is pending or track was paused, just discard buffered data 4285 flushHw_l(); 4286 } else { 4287 mMixerStatus = MIXER_DRAIN_ALL; 4288 threadLoop_drain(); 4289 } 4290 if (mUseAsyncWrite) { 4291 ALOG_ASSERT(mCallbackThread != 0); 4292 mCallbackThread->exit(); 4293 } 4294 PlaybackThread::threadLoop_exit(); 4295} 4296 4297AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 4298 Vector< sp<Track> > *tracksToRemove 4299) 4300{ 4301 size_t count = mActiveTracks.size(); 4302 4303 mixer_state mixerStatus = MIXER_IDLE; 4304 bool doHwPause = false; 4305 bool doHwResume = false; 4306 4307 ALOGV("OffloadThread::prepareTracks_l active tracks %d", count); 4308 4309 // find out which tracks need to be processed 4310 for (size_t i = 0; i < count; i++) { 4311 sp<Track> t = mActiveTracks[i].promote(); 4312 // The track died recently 4313 if (t == 0) { 4314 continue; 4315 } 4316 Track* const track = t.get(); 4317 audio_track_cblk_t* cblk = track->cblk(); 4318 // Only consider last track started for volume and mixer state control. 4319 // In theory an older track could underrun and restart after the new one starts 4320 // but as we only care about the transition phase between two tracks on a 4321 // direct output, it is not a problem to ignore the underrun case. 4322 sp<Track> l = mLatestActiveTrack.promote(); 4323 bool last = l.get() == track; 4324 4325 if (track->isInvalid()) { 4326 ALOGW("An invalidated track shouldn't be in active list"); 4327 tracksToRemove->add(track); 4328 continue; 4329 } 4330 4331 if (track->mState == TrackBase::IDLE) { 4332 ALOGW("An idle track shouldn't be in active list"); 4333 continue; 4334 } 4335 4336 if (track->isPausing()) { 4337 track->setPaused(); 4338 if (last) { 4339 if (!mHwPaused) { 4340 doHwPause = true; 4341 mHwPaused = true; 4342 } 4343 // If we were part way through writing the mixbuffer to 4344 // the HAL we must save this until we resume 4345 // BUG - this will be wrong if a different track is made active, 4346 // in that case we want to discard the pending data in the 4347 // mixbuffer and tell the client to present it again when the 4348 // track is resumed 4349 mPausedWriteLength = mCurrentWriteLength; 4350 mPausedBytesRemaining = mBytesRemaining; 4351 mBytesRemaining = 0; // stop writing 4352 } 4353 tracksToRemove->add(track); 4354 } else if (track->isFlushPending()) { 4355 track->flushAck(); 4356 if (last) { 4357 mFlushPending = true; 4358 } 4359 } else if (track->isResumePending()){ 4360 track->resumeAck(); 4361 if (last) { 4362 if (mPausedBytesRemaining) { 4363 // Need to continue write that was interrupted 4364 mCurrentWriteLength = mPausedWriteLength; 4365 mBytesRemaining = mPausedBytesRemaining; 4366 mPausedBytesRemaining = 0; 4367 } 4368 if (mHwPaused) { 4369 doHwResume = true; 4370 mHwPaused = false; 4371 // threadLoop_mix() will handle the case that we need to 4372 // resume an interrupted write 4373 } 4374 // enable write to audio HAL 4375 sleepTime = 0; 4376 4377 // Do not handle new data in this iteration even if track->framesReady() 4378 mixerStatus = MIXER_TRACKS_ENABLED; 4379 } 4380 } else if (track->framesReady() && track->isReady() && 4381 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 4382 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 4383 if (track->mFillingUpStatus == Track::FS_FILLED) { 4384 track->mFillingUpStatus = Track::FS_ACTIVE; 4385 // make sure processVolume_l() will apply new volume even if 0 4386 mLeftVolFloat = mRightVolFloat = -1.0; 4387 } 4388 4389 if (last) { 4390 sp<Track> previousTrack = mPreviousTrack.promote(); 4391 if (previousTrack != 0) { 4392 if (track != previousTrack.get()) { 4393 // Flush any data still being written from last track 4394 mBytesRemaining = 0; 4395 if (mPausedBytesRemaining) { 4396 // Last track was paused so we also need to flush saved 4397 // mixbuffer state and invalidate track so that it will 4398 // re-submit that unwritten data when it is next resumed 4399 mPausedBytesRemaining = 0; 4400 // Invalidate is a bit drastic - would be more efficient 4401 // to have a flag to tell client that some of the 4402 // previously written data was lost 4403 previousTrack->invalidate(); 4404 } 4405 // flush data already sent to the DSP if changing audio session as audio 4406 // comes from a different source. Also invalidate previous track to force a 4407 // seek when resuming. 4408 if (previousTrack->sessionId() != track->sessionId()) { 4409 previousTrack->invalidate(); 4410 } 4411 } 4412 } 4413 mPreviousTrack = track; 4414 // reset retry count 4415 track->mRetryCount = kMaxTrackRetriesOffload; 4416 mActiveTrack = t; 4417 mixerStatus = MIXER_TRACKS_READY; 4418 } 4419 } else { 4420 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 4421 if (track->isStopping_1()) { 4422 // Hardware buffer can hold a large amount of audio so we must 4423 // wait for all current track's data to drain before we say 4424 // that the track is stopped. 4425 if (mBytesRemaining == 0) { 4426 // Only start draining when all data in mixbuffer 4427 // has been written 4428 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 4429 track->mState = TrackBase::STOPPING_2; // so presentation completes after drain 4430 // do not drain if no data was ever sent to HAL (mStandby == true) 4431 if (last && !mStandby) { 4432 // do not modify drain sequence if we are already draining. This happens 4433 // when resuming from pause after drain. 4434 if ((mDrainSequence & 1) == 0) { 4435 sleepTime = 0; 4436 standbyTime = systemTime() + standbyDelay; 4437 mixerStatus = MIXER_DRAIN_TRACK; 4438 mDrainSequence += 2; 4439 } 4440 if (mHwPaused) { 4441 // It is possible to move from PAUSED to STOPPING_1 without 4442 // a resume so we must ensure hardware is running 4443 doHwResume = true; 4444 mHwPaused = false; 4445 } 4446 } 4447 } 4448 } else if (track->isStopping_2()) { 4449 // Drain has completed or we are in standby, signal presentation complete 4450 if (!(mDrainSequence & 1) || !last || mStandby) { 4451 track->mState = TrackBase::STOPPED; 4452 size_t audioHALFrames = 4453 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 4454 size_t framesWritten = 4455 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 4456 track->presentationComplete(framesWritten, audioHALFrames); 4457 track->reset(); 4458 tracksToRemove->add(track); 4459 } 4460 } else { 4461 // No buffers for this track. Give it a few chances to 4462 // fill a buffer, then remove it from active list. 4463 if (--(track->mRetryCount) <= 0) { 4464 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 4465 track->name()); 4466 tracksToRemove->add(track); 4467 // indicate to client process that the track was disabled because of underrun; 4468 // it will then automatically call start() when data is available 4469 android_atomic_or(CBLK_DISABLED, &cblk->mFlags); 4470 } else if (last){ 4471 mixerStatus = MIXER_TRACKS_ENABLED; 4472 } 4473 } 4474 } 4475 // compute volume for this track 4476 processVolume_l(track, last); 4477 } 4478 4479 // make sure the pause/flush/resume sequence is executed in the right order. 4480 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 4481 // before flush and then resume HW. This can happen in case of pause/flush/resume 4482 // if resume is received before pause is executed. 4483 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 4484 mOutput->stream->pause(mOutput->stream); 4485 } 4486 if (mFlushPending) { 4487 flushHw_l(); 4488 mFlushPending = false; 4489 } 4490 if (!mStandby && doHwResume) { 4491 mOutput->stream->resume(mOutput->stream); 4492 } 4493 4494 // remove all the tracks that need to be... 4495 removeTracks_l(*tracksToRemove); 4496 4497 return mixerStatus; 4498} 4499 4500// must be called with thread mutex locked 4501bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 4502{ 4503 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 4504 mWriteAckSequence, mDrainSequence); 4505 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 4506 return true; 4507 } 4508 return false; 4509} 4510 4511// must be called with thread mutex locked 4512bool AudioFlinger::OffloadThread::shouldStandby_l() 4513{ 4514 bool trackPaused = false; 4515 4516 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 4517 // after a timeout and we will enter standby then. 4518 if (mTracks.size() > 0) { 4519 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 4520 } 4521 4522 return !mStandby && !trackPaused; 4523} 4524 4525 4526bool AudioFlinger::OffloadThread::waitingAsyncCallback() 4527{ 4528 Mutex::Autolock _l(mLock); 4529 return waitingAsyncCallback_l(); 4530} 4531 4532void AudioFlinger::OffloadThread::flushHw_l() 4533{ 4534 mOutput->stream->flush(mOutput->stream); 4535 // Flush anything still waiting in the mixbuffer 4536 mCurrentWriteLength = 0; 4537 mBytesRemaining = 0; 4538 mPausedWriteLength = 0; 4539 mPausedBytesRemaining = 0; 4540 mHwPaused = false; 4541 4542 if (mUseAsyncWrite) { 4543 // discard any pending drain or write ack by incrementing sequence 4544 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4545 mDrainSequence = (mDrainSequence + 2) & ~1; 4546 ALOG_ASSERT(mCallbackThread != 0); 4547 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4548 mCallbackThread->setDraining(mDrainSequence); 4549 } 4550} 4551 4552void AudioFlinger::OffloadThread::onAddNewTrack_l() 4553{ 4554 sp<Track> previousTrack = mPreviousTrack.promote(); 4555 sp<Track> latestTrack = mLatestActiveTrack.promote(); 4556 4557 if (previousTrack != 0 && latestTrack != 0 && 4558 (previousTrack->sessionId() != latestTrack->sessionId())) { 4559 mFlushPending = true; 4560 } 4561 PlaybackThread::onAddNewTrack_l(); 4562} 4563 4564// ---------------------------------------------------------------------------- 4565 4566AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 4567 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 4568 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 4569 DUPLICATING), 4570 mWaitTimeMs(UINT_MAX) 4571{ 4572 addOutputTrack(mainThread); 4573} 4574 4575AudioFlinger::DuplicatingThread::~DuplicatingThread() 4576{ 4577 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4578 mOutputTracks[i]->destroy(); 4579 } 4580} 4581 4582void AudioFlinger::DuplicatingThread::threadLoop_mix() 4583{ 4584 // mix buffers... 4585 if (outputsReady(outputTracks)) { 4586 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 4587 } else { 4588 memset(mSinkBuffer, 0, mSinkBufferSize); 4589 } 4590 sleepTime = 0; 4591 writeFrames = mNormalFrameCount; 4592 mCurrentWriteLength = mSinkBufferSize; 4593 standbyTime = systemTime() + standbyDelay; 4594} 4595 4596void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 4597{ 4598 if (sleepTime == 0) { 4599 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4600 sleepTime = activeSleepTime; 4601 } else { 4602 sleepTime = idleSleepTime; 4603 } 4604 } else if (mBytesWritten != 0) { 4605 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4606 writeFrames = mNormalFrameCount; 4607 memset(mSinkBuffer, 0, mSinkBufferSize); 4608 } else { 4609 // flush remaining overflow buffers in output tracks 4610 writeFrames = 0; 4611 } 4612 sleepTime = 0; 4613 } 4614} 4615 4616ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 4617{ 4618 for (size_t i = 0; i < outputTracks.size(); i++) { 4619 // We convert the duplicating thread format to AUDIO_FORMAT_PCM_16_BIT 4620 // for delivery downstream as needed. This in-place conversion is safe as 4621 // AUDIO_FORMAT_PCM_16_BIT is smaller than any other supported format 4622 // (AUDIO_FORMAT_PCM_8_BIT is not allowed here). 4623 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4624 memcpy_by_audio_format(mSinkBuffer, AUDIO_FORMAT_PCM_16_BIT, 4625 mSinkBuffer, mFormat, writeFrames * mChannelCount); 4626 } 4627 outputTracks[i]->write(reinterpret_cast<int16_t*>(mSinkBuffer), writeFrames); 4628 } 4629 mStandby = false; 4630 return (ssize_t)mSinkBufferSize; 4631} 4632 4633void AudioFlinger::DuplicatingThread::threadLoop_standby() 4634{ 4635 // DuplicatingThread implements standby by stopping all tracks 4636 for (size_t i = 0; i < outputTracks.size(); i++) { 4637 outputTracks[i]->stop(); 4638 } 4639} 4640 4641void AudioFlinger::DuplicatingThread::saveOutputTracks() 4642{ 4643 outputTracks = mOutputTracks; 4644} 4645 4646void AudioFlinger::DuplicatingThread::clearOutputTracks() 4647{ 4648 outputTracks.clear(); 4649} 4650 4651void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 4652{ 4653 Mutex::Autolock _l(mLock); 4654 // FIXME explain this formula 4655 size_t frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 4656 // OutputTrack is forced to AUDIO_FORMAT_PCM_16_BIT regardless of mFormat 4657 // due to current usage case and restrictions on the AudioBufferProvider. 4658 // Actual buffer conversion is done in threadLoop_write(). 4659 // 4660 // TODO: This may change in the future, depending on multichannel 4661 // (and non int16_t*) support on AF::PlaybackThread::OutputTrack 4662 OutputTrack *outputTrack = new OutputTrack(thread, 4663 this, 4664 mSampleRate, 4665 AUDIO_FORMAT_PCM_16_BIT, 4666 mChannelMask, 4667 frameCount, 4668 IPCThreadState::self()->getCallingUid()); 4669 if (outputTrack->cblk() != NULL) { 4670 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 4671 mOutputTracks.add(outputTrack); 4672 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 4673 updateWaitTime_l(); 4674 } 4675} 4676 4677void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 4678{ 4679 Mutex::Autolock _l(mLock); 4680 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4681 if (mOutputTracks[i]->thread() == thread) { 4682 mOutputTracks[i]->destroy(); 4683 mOutputTracks.removeAt(i); 4684 updateWaitTime_l(); 4685 return; 4686 } 4687 } 4688 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4689} 4690 4691// caller must hold mLock 4692void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4693{ 4694 mWaitTimeMs = UINT_MAX; 4695 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4696 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4697 if (strong != 0) { 4698 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4699 if (waitTimeMs < mWaitTimeMs) { 4700 mWaitTimeMs = waitTimeMs; 4701 } 4702 } 4703 } 4704} 4705 4706 4707bool AudioFlinger::DuplicatingThread::outputsReady( 4708 const SortedVector< sp<OutputTrack> > &outputTracks) 4709{ 4710 for (size_t i = 0; i < outputTracks.size(); i++) { 4711 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4712 if (thread == 0) { 4713 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 4714 outputTracks[i].get()); 4715 return false; 4716 } 4717 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4718 // see note at standby() declaration 4719 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4720 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 4721 thread.get()); 4722 return false; 4723 } 4724 } 4725 return true; 4726} 4727 4728uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4729{ 4730 return (mWaitTimeMs * 1000) / 2; 4731} 4732 4733void AudioFlinger::DuplicatingThread::cacheParameters_l() 4734{ 4735 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4736 updateWaitTime_l(); 4737 4738 MixerThread::cacheParameters_l(); 4739} 4740 4741// ---------------------------------------------------------------------------- 4742// Record 4743// ---------------------------------------------------------------------------- 4744 4745AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4746 AudioStreamIn *input, 4747 audio_io_handle_t id, 4748 audio_devices_t outDevice, 4749 audio_devices_t inDevice 4750#ifdef TEE_SINK 4751 , const sp<NBAIO_Sink>& teeSink 4752#endif 4753 ) : 4754 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), 4755 mInput(input), mActiveTracksGen(0), mRsmpInBuffer(NULL), 4756 // mRsmpInFrames and mRsmpInFramesP2 are set by readInputParameters_l() 4757 mRsmpInRear(0) 4758#ifdef TEE_SINK 4759 , mTeeSink(teeSink) 4760#endif 4761 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 4762 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 4763{ 4764 snprintf(mName, kNameLength, "AudioIn_%X", id); 4765 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mName); 4766 4767 readInputParameters_l(); 4768} 4769 4770 4771AudioFlinger::RecordThread::~RecordThread() 4772{ 4773 mAudioFlinger->unregisterWriter(mNBLogWriter); 4774 delete[] mRsmpInBuffer; 4775} 4776 4777void AudioFlinger::RecordThread::onFirstRef() 4778{ 4779 run(mName, PRIORITY_URGENT_AUDIO); 4780} 4781 4782bool AudioFlinger::RecordThread::threadLoop() 4783{ 4784 nsecs_t lastWarning = 0; 4785 4786 inputStandBy(); 4787 4788reacquire_wakelock: 4789 sp<RecordTrack> activeTrack; 4790 int activeTracksGen; 4791 { 4792 Mutex::Autolock _l(mLock); 4793 size_t size = mActiveTracks.size(); 4794 activeTracksGen = mActiveTracksGen; 4795 if (size > 0) { 4796 // FIXME an arbitrary choice 4797 activeTrack = mActiveTracks[0]; 4798 acquireWakeLock_l(activeTrack->uid()); 4799 if (size > 1) { 4800 SortedVector<int> tmp; 4801 for (size_t i = 0; i < size; i++) { 4802 tmp.add(mActiveTracks[i]->uid()); 4803 } 4804 updateWakeLockUids_l(tmp); 4805 } 4806 } else { 4807 acquireWakeLock_l(-1); 4808 } 4809 } 4810 4811 // used to request a deferred sleep, to be executed later while mutex is unlocked 4812 uint32_t sleepUs = 0; 4813 4814 // loop while there is work to do 4815 for (;;) { 4816 Vector< sp<EffectChain> > effectChains; 4817 4818 // sleep with mutex unlocked 4819 if (sleepUs > 0) { 4820 usleep(sleepUs); 4821 sleepUs = 0; 4822 } 4823 4824 // activeTracks accumulates a copy of a subset of mActiveTracks 4825 Vector< sp<RecordTrack> > activeTracks; 4826 4827 4828 { // scope for mLock 4829 Mutex::Autolock _l(mLock); 4830 4831 processConfigEvents_l(); 4832 4833 // check exitPending here because checkForNewParameters_l() and 4834 // checkForNewParameters_l() can temporarily release mLock 4835 if (exitPending()) { 4836 break; 4837 } 4838 4839 // if no active track(s), then standby and release wakelock 4840 size_t size = mActiveTracks.size(); 4841 if (size == 0) { 4842 standbyIfNotAlreadyInStandby(); 4843 // exitPending() can't become true here 4844 releaseWakeLock_l(); 4845 ALOGV("RecordThread: loop stopping"); 4846 // go to sleep 4847 mWaitWorkCV.wait(mLock); 4848 ALOGV("RecordThread: loop starting"); 4849 goto reacquire_wakelock; 4850 } 4851 4852 if (mActiveTracksGen != activeTracksGen) { 4853 activeTracksGen = mActiveTracksGen; 4854 SortedVector<int> tmp; 4855 for (size_t i = 0; i < size; i++) { 4856 tmp.add(mActiveTracks[i]->uid()); 4857 } 4858 updateWakeLockUids_l(tmp); 4859 } 4860 4861 bool doBroadcast = false; 4862 for (size_t i = 0; i < size; ) { 4863 4864 activeTrack = mActiveTracks[i]; 4865 if (activeTrack->isTerminated()) { 4866 removeTrack_l(activeTrack); 4867 mActiveTracks.remove(activeTrack); 4868 mActiveTracksGen++; 4869 size--; 4870 continue; 4871 } 4872 4873 TrackBase::track_state activeTrackState = activeTrack->mState; 4874 switch (activeTrackState) { 4875 4876 case TrackBase::PAUSING: 4877 mActiveTracks.remove(activeTrack); 4878 mActiveTracksGen++; 4879 doBroadcast = true; 4880 size--; 4881 continue; 4882 4883 case TrackBase::STARTING_1: 4884 sleepUs = 10000; 4885 i++; 4886 continue; 4887 4888 case TrackBase::STARTING_2: 4889 doBroadcast = true; 4890 mStandby = false; 4891 activeTrack->mState = TrackBase::ACTIVE; 4892 break; 4893 4894 case TrackBase::ACTIVE: 4895 break; 4896 4897 case TrackBase::IDLE: 4898 i++; 4899 continue; 4900 4901 default: 4902 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 4903 } 4904 4905 activeTracks.add(activeTrack); 4906 i++; 4907 4908 } 4909 if (doBroadcast) { 4910 mStartStopCond.broadcast(); 4911 } 4912 4913 // sleep if there are no active tracks to process 4914 if (activeTracks.size() == 0) { 4915 if (sleepUs == 0) { 4916 sleepUs = kRecordThreadSleepUs; 4917 } 4918 continue; 4919 } 4920 sleepUs = 0; 4921 4922 lockEffectChains_l(effectChains); 4923 } 4924 4925 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 4926 4927 size_t size = effectChains.size(); 4928 for (size_t i = 0; i < size; i++) { 4929 // thread mutex is not locked, but effect chain is locked 4930 effectChains[i]->process_l(); 4931 } 4932 4933 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 4934 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 4935 // slow, then this RecordThread will overrun by not calling HAL read often enough. 4936 // If destination is non-contiguous, first read past the nominal end of buffer, then 4937 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 4938 4939 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 4940 ssize_t bytesRead = mInput->stream->read(mInput->stream, 4941 &mRsmpInBuffer[rear * mChannelCount], mBufferSize); 4942 if (bytesRead <= 0) { 4943 ALOGE("read failed: bytesRead=%d < %u", bytesRead, mBufferSize); 4944 // Force input into standby so that it tries to recover at next read attempt 4945 inputStandBy(); 4946 sleepUs = kRecordThreadSleepUs; 4947 continue; 4948 } 4949 ALOG_ASSERT((size_t) bytesRead <= mBufferSize); 4950 size_t framesRead = bytesRead / mFrameSize; 4951 ALOG_ASSERT(framesRead > 0); 4952 if (mTeeSink != 0) { 4953 (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead); 4954 } 4955 // If destination is non-contiguous, we now correct for reading past end of buffer. 4956 size_t part1 = mRsmpInFramesP2 - rear; 4957 if (framesRead > part1) { 4958 memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount], 4959 (framesRead - part1) * mFrameSize); 4960 } 4961 rear = mRsmpInRear += framesRead; 4962 4963 size = activeTracks.size(); 4964 // loop over each active track 4965 for (size_t i = 0; i < size; i++) { 4966 activeTrack = activeTracks[i]; 4967 4968 enum { 4969 OVERRUN_UNKNOWN, 4970 OVERRUN_TRUE, 4971 OVERRUN_FALSE 4972 } overrun = OVERRUN_UNKNOWN; 4973 4974 // loop over getNextBuffer to handle circular sink 4975 for (;;) { 4976 4977 activeTrack->mSink.frameCount = ~0; 4978 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 4979 size_t framesOut = activeTrack->mSink.frameCount; 4980 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 4981 4982 int32_t front = activeTrack->mRsmpInFront; 4983 ssize_t filled = rear - front; 4984 size_t framesIn; 4985 4986 if (filled < 0) { 4987 // should not happen, but treat like a massive overrun and re-sync 4988 framesIn = 0; 4989 activeTrack->mRsmpInFront = rear; 4990 overrun = OVERRUN_TRUE; 4991 } else if ((size_t) filled <= mRsmpInFrames) { 4992 framesIn = (size_t) filled; 4993 } else { 4994 // client is not keeping up with server, but give it latest data 4995 framesIn = mRsmpInFrames; 4996 activeTrack->mRsmpInFront = front = rear - framesIn; 4997 overrun = OVERRUN_TRUE; 4998 } 4999 5000 if (framesOut == 0 || framesIn == 0) { 5001 break; 5002 } 5003 5004 if (activeTrack->mResampler == NULL) { 5005 // no resampling 5006 if (framesIn > framesOut) { 5007 framesIn = framesOut; 5008 } else { 5009 framesOut = framesIn; 5010 } 5011 int8_t *dst = activeTrack->mSink.i8; 5012 while (framesIn > 0) { 5013 front &= mRsmpInFramesP2 - 1; 5014 size_t part1 = mRsmpInFramesP2 - front; 5015 if (part1 > framesIn) { 5016 part1 = framesIn; 5017 } 5018 int8_t *src = (int8_t *)mRsmpInBuffer + (front * mFrameSize); 5019 if (mChannelCount == activeTrack->mChannelCount) { 5020 memcpy(dst, src, part1 * mFrameSize); 5021 } else if (mChannelCount == 1) { 5022 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, (int16_t *)src, 5023 part1); 5024 } else { 5025 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, (int16_t *)src, 5026 part1); 5027 } 5028 dst += part1 * activeTrack->mFrameSize; 5029 front += part1; 5030 framesIn -= part1; 5031 } 5032 activeTrack->mRsmpInFront += framesOut; 5033 5034 } else { 5035 // resampling 5036 // FIXME framesInNeeded should really be part of resampler API, and should 5037 // depend on the SRC ratio 5038 // to keep mRsmpInBuffer full so resampler always has sufficient input 5039 size_t framesInNeeded; 5040 // FIXME only re-calculate when it changes, and optimize for common ratios 5041 double inOverOut = (double) mSampleRate / activeTrack->mSampleRate; 5042 double outOverIn = (double) activeTrack->mSampleRate / mSampleRate; 5043 framesInNeeded = ceil(framesOut * inOverOut) + 1; 5044 ALOGV("need %u frames in to produce %u out given in/out ratio of %.4g", 5045 framesInNeeded, framesOut, inOverOut); 5046 // Although we theoretically have framesIn in circular buffer, some of those are 5047 // unreleased frames, and thus must be discounted for purpose of budgeting. 5048 size_t unreleased = activeTrack->mRsmpInUnrel; 5049 framesIn = framesIn > unreleased ? framesIn - unreleased : 0; 5050 if (framesIn < framesInNeeded) { 5051 ALOGV("not enough to resample: have %u frames in but need %u in to " 5052 "produce %u out given in/out ratio of %.4g", 5053 framesIn, framesInNeeded, framesOut, inOverOut); 5054 size_t newFramesOut = framesIn > 0 ? floor((framesIn - 1) * outOverIn) : 0; 5055 LOG_ALWAYS_FATAL_IF(newFramesOut >= framesOut); 5056 if (newFramesOut == 0) { 5057 break; 5058 } 5059 framesInNeeded = ceil(newFramesOut * inOverOut) + 1; 5060 ALOGV("now need %u frames in to produce %u out given out/in ratio of %.4g", 5061 framesInNeeded, newFramesOut, outOverIn); 5062 LOG_ALWAYS_FATAL_IF(framesIn < framesInNeeded); 5063 ALOGV("success 2: have %u frames in and need %u in to produce %u out " 5064 "given in/out ratio of %.4g", 5065 framesIn, framesInNeeded, newFramesOut, inOverOut); 5066 framesOut = newFramesOut; 5067 } else { 5068 ALOGV("success 1: have %u in and need %u in to produce %u out " 5069 "given in/out ratio of %.4g", 5070 framesIn, framesInNeeded, framesOut, inOverOut); 5071 } 5072 5073 // reallocate mRsmpOutBuffer as needed; we will grow but never shrink 5074 if (activeTrack->mRsmpOutFrameCount < framesOut) { 5075 // FIXME why does each track need it's own mRsmpOutBuffer? can't they share? 5076 delete[] activeTrack->mRsmpOutBuffer; 5077 // resampler always outputs stereo 5078 activeTrack->mRsmpOutBuffer = new int32_t[framesOut * FCC_2]; 5079 activeTrack->mRsmpOutFrameCount = framesOut; 5080 } 5081 5082 // resampler accumulates, but we only have one source track 5083 memset(activeTrack->mRsmpOutBuffer, 0, framesOut * FCC_2 * sizeof(int32_t)); 5084 activeTrack->mResampler->resample(activeTrack->mRsmpOutBuffer, framesOut, 5085 // FIXME how about having activeTrack implement this interface itself? 5086 activeTrack->mResamplerBufferProvider 5087 /*this*/ /* AudioBufferProvider* */); 5088 // ditherAndClamp() works as long as all buffers returned by 5089 // activeTrack->getNextBuffer() are 32 bit aligned which should be always true. 5090 if (activeTrack->mChannelCount == 1) { 5091 // temporarily type pun mRsmpOutBuffer from Q4.27 to int16_t 5092 ditherAndClamp(activeTrack->mRsmpOutBuffer, activeTrack->mRsmpOutBuffer, 5093 framesOut); 5094 // the resampler always outputs stereo samples: 5095 // do post stereo to mono conversion 5096 downmix_to_mono_i16_from_stereo_i16(activeTrack->mSink.i16, 5097 (int16_t *)activeTrack->mRsmpOutBuffer, framesOut); 5098 } else { 5099 ditherAndClamp((int32_t *)activeTrack->mSink.raw, 5100 activeTrack->mRsmpOutBuffer, framesOut); 5101 } 5102 // now done with mRsmpOutBuffer 5103 5104 } 5105 5106 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 5107 overrun = OVERRUN_FALSE; 5108 } 5109 5110 if (activeTrack->mFramesToDrop == 0) { 5111 if (framesOut > 0) { 5112 activeTrack->mSink.frameCount = framesOut; 5113 activeTrack->releaseBuffer(&activeTrack->mSink); 5114 } 5115 } else { 5116 // FIXME could do a partial drop of framesOut 5117 if (activeTrack->mFramesToDrop > 0) { 5118 activeTrack->mFramesToDrop -= framesOut; 5119 if (activeTrack->mFramesToDrop <= 0) { 5120 activeTrack->clearSyncStartEvent(); 5121 } 5122 } else { 5123 activeTrack->mFramesToDrop += framesOut; 5124 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 5125 activeTrack->mSyncStartEvent->isCancelled()) { 5126 ALOGW("Synced record %s, session %d, trigger session %d", 5127 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 5128 activeTrack->sessionId(), 5129 (activeTrack->mSyncStartEvent != 0) ? 5130 activeTrack->mSyncStartEvent->triggerSession() : 0); 5131 activeTrack->clearSyncStartEvent(); 5132 } 5133 } 5134 } 5135 5136 if (framesOut == 0) { 5137 break; 5138 } 5139 } 5140 5141 switch (overrun) { 5142 case OVERRUN_TRUE: 5143 // client isn't retrieving buffers fast enough 5144 if (!activeTrack->setOverflow()) { 5145 nsecs_t now = systemTime(); 5146 // FIXME should lastWarning per track? 5147 if ((now - lastWarning) > kWarningThrottleNs) { 5148 ALOGW("RecordThread: buffer overflow"); 5149 lastWarning = now; 5150 } 5151 } 5152 break; 5153 case OVERRUN_FALSE: 5154 activeTrack->clearOverflow(); 5155 break; 5156 case OVERRUN_UNKNOWN: 5157 break; 5158 } 5159 5160 } 5161 5162 // enable changes in effect chain 5163 unlockEffectChains(effectChains); 5164 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 5165 } 5166 5167 standbyIfNotAlreadyInStandby(); 5168 5169 { 5170 Mutex::Autolock _l(mLock); 5171 for (size_t i = 0; i < mTracks.size(); i++) { 5172 sp<RecordTrack> track = mTracks[i]; 5173 track->invalidate(); 5174 } 5175 mActiveTracks.clear(); 5176 mActiveTracksGen++; 5177 mStartStopCond.broadcast(); 5178 } 5179 5180 releaseWakeLock(); 5181 5182 ALOGV("RecordThread %p exiting", this); 5183 return false; 5184} 5185 5186void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 5187{ 5188 if (!mStandby) { 5189 inputStandBy(); 5190 mStandby = true; 5191 } 5192} 5193 5194void AudioFlinger::RecordThread::inputStandBy() 5195{ 5196 mInput->stream->common.standby(&mInput->stream->common); 5197} 5198 5199// RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 5200sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5201 const sp<AudioFlinger::Client>& client, 5202 uint32_t sampleRate, 5203 audio_format_t format, 5204 audio_channel_mask_t channelMask, 5205 size_t *pFrameCount, 5206 int sessionId, 5207 int uid, 5208 IAudioFlinger::track_flags_t *flags, 5209 pid_t tid, 5210 status_t *status) 5211{ 5212 size_t frameCount = *pFrameCount; 5213 sp<RecordTrack> track; 5214 status_t lStatus; 5215 5216 // client expresses a preference for FAST, but we get the final say 5217 if (*flags & IAudioFlinger::TRACK_FAST) { 5218 if ( 5219 // use case: callback handler and frame count is default or at least as large as HAL 5220 ( 5221 (tid != -1) && 5222 ((frameCount == 0) || 5223 // FIXME not necessarily true, should be native frame count for native SR! 5224 (frameCount >= mFrameCount)) 5225 ) && 5226 // PCM data 5227 audio_is_linear_pcm(format) && 5228 // mono or stereo 5229 ( (channelMask == AUDIO_CHANNEL_IN_MONO) || 5230 (channelMask == AUDIO_CHANNEL_IN_STEREO) ) && 5231 // hardware sample rate 5232 // FIXME actually the native hardware sample rate 5233 (sampleRate == mSampleRate) && 5234 // record thread has an associated fast capture 5235 hasFastCapture() 5236 // fast capture does not require slots 5237 ) { 5238 // if frameCount not specified, then it defaults to fast capture (HAL) frame count 5239 if (frameCount == 0) { 5240 // FIXME wrong mFrameCount 5241 frameCount = mFrameCount * kFastTrackMultiplier; 5242 } 5243 ALOGV("AUDIO_INPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 5244 frameCount, mFrameCount); 5245 } else { 5246 ALOGV("AUDIO_INPUT_FLAG_FAST denied: frameCount=%d " 5247 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%u mSampleRate=%u " 5248 "hasFastCapture=%d tid=%d", 5249 frameCount, mFrameCount, format, 5250 audio_is_linear_pcm(format), 5251 channelMask, sampleRate, mSampleRate, hasFastCapture(), tid); 5252 *flags &= ~IAudioFlinger::TRACK_FAST; 5253 // FIXME It's not clear that we need to enforce this any more, since we have a pipe. 5254 // For compatibility with AudioRecord calculation, buffer depth is forced 5255 // to be at least 2 x the record thread frame count and cover audio hardware latency. 5256 // This is probably too conservative, but legacy application code may depend on it. 5257 // If you change this calculation, also review the start threshold which is related. 5258 // FIXME It's not clear how input latency actually matters. Perhaps this should be 0. 5259 uint32_t latencyMs = 50; // FIXME mInput->stream->get_latency(mInput->stream); 5260 size_t mNormalFrameCount = 2048; // FIXME 5261 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 5262 if (minBufCount < 2) { 5263 minBufCount = 2; 5264 } 5265 size_t minFrameCount = mNormalFrameCount * minBufCount; 5266 if (frameCount < minFrameCount) { 5267 frameCount = minFrameCount; 5268 } 5269 } 5270 } 5271 *pFrameCount = frameCount; 5272 5273 lStatus = initCheck(); 5274 if (lStatus != NO_ERROR) { 5275 ALOGE("createRecordTrack_l() audio driver not initialized"); 5276 goto Exit; 5277 } 5278 5279 { // scope for mLock 5280 Mutex::Autolock _l(mLock); 5281 5282 track = new RecordTrack(this, client, sampleRate, 5283 format, channelMask, frameCount, sessionId, uid, 5284 *flags); 5285 5286 lStatus = track->initCheck(); 5287 if (lStatus != NO_ERROR) { 5288 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 5289 // track must be cleared from the caller as the caller has the AF lock 5290 goto Exit; 5291 } 5292 mTracks.add(track); 5293 5294 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5295 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5296 mAudioFlinger->btNrecIsOff(); 5297 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5298 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5299 5300 if ((*flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 5301 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 5302 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 5303 // so ask activity manager to do this on our behalf 5304 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp); 5305 } 5306 } 5307 5308 lStatus = NO_ERROR; 5309 5310Exit: 5311 *status = lStatus; 5312 return track; 5313} 5314 5315status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5316 AudioSystem::sync_event_t event, 5317 int triggerSession) 5318{ 5319 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 5320 sp<ThreadBase> strongMe = this; 5321 status_t status = NO_ERROR; 5322 5323 if (event == AudioSystem::SYNC_EVENT_NONE) { 5324 recordTrack->clearSyncStartEvent(); 5325 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5326 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5327 triggerSession, 5328 recordTrack->sessionId(), 5329 syncStartEventCallback, 5330 recordTrack); 5331 // Sync event can be cancelled by the trigger session if the track is not in a 5332 // compatible state in which case we start record immediately 5333 if (recordTrack->mSyncStartEvent->isCancelled()) { 5334 recordTrack->clearSyncStartEvent(); 5335 } else { 5336 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 5337 recordTrack->mFramesToDrop = - 5338 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 5339 } 5340 } 5341 5342 { 5343 // This section is a rendezvous between binder thread executing start() and RecordThread 5344 AutoMutex lock(mLock); 5345 if (mActiveTracks.indexOf(recordTrack) >= 0) { 5346 if (recordTrack->mState == TrackBase::PAUSING) { 5347 ALOGV("active record track PAUSING -> ACTIVE"); 5348 recordTrack->mState = TrackBase::ACTIVE; 5349 } else { 5350 ALOGV("active record track state %d", recordTrack->mState); 5351 } 5352 return status; 5353 } 5354 5355 // TODO consider other ways of handling this, such as changing the state to :STARTING and 5356 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 5357 // or using a separate command thread 5358 recordTrack->mState = TrackBase::STARTING_1; 5359 mActiveTracks.add(recordTrack); 5360 mActiveTracksGen++; 5361 mLock.unlock(); 5362 status_t status = AudioSystem::startInput(mId); 5363 mLock.lock(); 5364 // FIXME should verify that recordTrack is still in mActiveTracks 5365 if (status != NO_ERROR) { 5366 mActiveTracks.remove(recordTrack); 5367 mActiveTracksGen++; 5368 recordTrack->clearSyncStartEvent(); 5369 return status; 5370 } 5371 // Catch up with current buffer indices if thread is already running. 5372 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 5373 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 5374 // see previously buffered data before it called start(), but with greater risk of overrun. 5375 5376 recordTrack->mRsmpInFront = mRsmpInRear; 5377 recordTrack->mRsmpInUnrel = 0; 5378 // FIXME why reset? 5379 if (recordTrack->mResampler != NULL) { 5380 recordTrack->mResampler->reset(); 5381 } 5382 recordTrack->mState = TrackBase::STARTING_2; 5383 // signal thread to start 5384 mWaitWorkCV.broadcast(); 5385 if (mActiveTracks.indexOf(recordTrack) < 0) { 5386 ALOGV("Record failed to start"); 5387 status = BAD_VALUE; 5388 goto startError; 5389 } 5390 return status; 5391 } 5392 5393startError: 5394 AudioSystem::stopInput(mId); 5395 recordTrack->clearSyncStartEvent(); 5396 // FIXME I wonder why we do not reset the state here? 5397 return status; 5398} 5399 5400void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5401{ 5402 sp<SyncEvent> strongEvent = event.promote(); 5403 5404 if (strongEvent != 0) { 5405 sp<RefBase> ptr = strongEvent->cookie().promote(); 5406 if (ptr != 0) { 5407 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 5408 recordTrack->handleSyncStartEvent(strongEvent); 5409 } 5410 } 5411} 5412 5413bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5414 ALOGV("RecordThread::stop"); 5415 AutoMutex _l(mLock); 5416 if (mActiveTracks.indexOf(recordTrack) != 0 || recordTrack->mState == TrackBase::PAUSING) { 5417 return false; 5418 } 5419 // note that threadLoop may still be processing the track at this point [without lock] 5420 recordTrack->mState = TrackBase::PAUSING; 5421 // do not wait for mStartStopCond if exiting 5422 if (exitPending()) { 5423 return true; 5424 } 5425 // FIXME incorrect usage of wait: no explicit predicate or loop 5426 mStartStopCond.wait(mLock); 5427 // if we have been restarted, recordTrack is in mActiveTracks here 5428 if (exitPending() || mActiveTracks.indexOf(recordTrack) != 0) { 5429 ALOGV("Record stopped OK"); 5430 return true; 5431 } 5432 return false; 5433} 5434 5435bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 5436{ 5437 return false; 5438} 5439 5440status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 5441{ 5442#if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 5443 if (!isValidSyncEvent(event)) { 5444 return BAD_VALUE; 5445 } 5446 5447 int eventSession = event->triggerSession(); 5448 status_t ret = NAME_NOT_FOUND; 5449 5450 Mutex::Autolock _l(mLock); 5451 5452 for (size_t i = 0; i < mTracks.size(); i++) { 5453 sp<RecordTrack> track = mTracks[i]; 5454 if (eventSession == track->sessionId()) { 5455 (void) track->setSyncEvent(event); 5456 ret = NO_ERROR; 5457 } 5458 } 5459 return ret; 5460#else 5461 return BAD_VALUE; 5462#endif 5463} 5464 5465// destroyTrack_l() must be called with ThreadBase::mLock held 5466void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 5467{ 5468 track->terminate(); 5469 track->mState = TrackBase::STOPPED; 5470 // active tracks are removed by threadLoop() 5471 if (mActiveTracks.indexOf(track) < 0) { 5472 removeTrack_l(track); 5473 } 5474} 5475 5476void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 5477{ 5478 mTracks.remove(track); 5479 // need anything related to effects here? 5480} 5481 5482void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5483{ 5484 dumpInternals(fd, args); 5485 dumpTracks(fd, args); 5486 dumpEffectChains(fd, args); 5487} 5488 5489void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 5490{ 5491 dprintf(fd, "\nInput thread %p:\n", this); 5492 5493 if (mActiveTracks.size() > 0) { 5494 dprintf(fd, " Buffer size: %zu bytes\n", mBufferSize); 5495 } else { 5496 dprintf(fd, " No active record clients\n"); 5497 } 5498 5499 dumpBase(fd, args); 5500} 5501 5502void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 5503{ 5504 const size_t SIZE = 256; 5505 char buffer[SIZE]; 5506 String8 result; 5507 5508 size_t numtracks = mTracks.size(); 5509 size_t numactive = mActiveTracks.size(); 5510 size_t numactiveseen = 0; 5511 dprintf(fd, " %d Tracks", numtracks); 5512 if (numtracks) { 5513 dprintf(fd, " of which %d are active\n", numactive); 5514 RecordTrack::appendDumpHeader(result); 5515 for (size_t i = 0; i < numtracks ; ++i) { 5516 sp<RecordTrack> track = mTracks[i]; 5517 if (track != 0) { 5518 bool active = mActiveTracks.indexOf(track) >= 0; 5519 if (active) { 5520 numactiveseen++; 5521 } 5522 track->dump(buffer, SIZE, active); 5523 result.append(buffer); 5524 } 5525 } 5526 } else { 5527 dprintf(fd, "\n"); 5528 } 5529 5530 if (numactiveseen != numactive) { 5531 snprintf(buffer, SIZE, " The following tracks are in the active list but" 5532 " not in the track list\n"); 5533 result.append(buffer); 5534 RecordTrack::appendDumpHeader(result); 5535 for (size_t i = 0; i < numactive; ++i) { 5536 sp<RecordTrack> track = mActiveTracks[i]; 5537 if (mTracks.indexOf(track) < 0) { 5538 track->dump(buffer, SIZE, true); 5539 result.append(buffer); 5540 } 5541 } 5542 5543 } 5544 write(fd, result.string(), result.size()); 5545} 5546 5547// AudioBufferProvider interface 5548status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 5549 AudioBufferProvider::Buffer* buffer, int64_t pts __unused) 5550{ 5551 RecordTrack *activeTrack = mRecordTrack; 5552 sp<ThreadBase> threadBase = activeTrack->mThread.promote(); 5553 if (threadBase == 0) { 5554 buffer->frameCount = 0; 5555 buffer->raw = NULL; 5556 return NOT_ENOUGH_DATA; 5557 } 5558 RecordThread *recordThread = (RecordThread *) threadBase.get(); 5559 int32_t rear = recordThread->mRsmpInRear; 5560 int32_t front = activeTrack->mRsmpInFront; 5561 ssize_t filled = rear - front; 5562 // FIXME should not be P2 (don't want to increase latency) 5563 // FIXME if client not keeping up, discard 5564 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 5565 // 'filled' may be non-contiguous, so return only the first contiguous chunk 5566 front &= recordThread->mRsmpInFramesP2 - 1; 5567 size_t part1 = recordThread->mRsmpInFramesP2 - front; 5568 if (part1 > (size_t) filled) { 5569 part1 = filled; 5570 } 5571 size_t ask = buffer->frameCount; 5572 ALOG_ASSERT(ask > 0); 5573 if (part1 > ask) { 5574 part1 = ask; 5575 } 5576 if (part1 == 0) { 5577 // Higher-level should keep mRsmpInBuffer full, and not call resampler if empty 5578 LOG_ALWAYS_FATAL("RecordThread::getNextBuffer() starved"); 5579 buffer->raw = NULL; 5580 buffer->frameCount = 0; 5581 activeTrack->mRsmpInUnrel = 0; 5582 return NOT_ENOUGH_DATA; 5583 } 5584 5585 buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount; 5586 buffer->frameCount = part1; 5587 activeTrack->mRsmpInUnrel = part1; 5588 return NO_ERROR; 5589} 5590 5591// AudioBufferProvider interface 5592void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 5593 AudioBufferProvider::Buffer* buffer) 5594{ 5595 RecordTrack *activeTrack = mRecordTrack; 5596 size_t stepCount = buffer->frameCount; 5597 if (stepCount == 0) { 5598 return; 5599 } 5600 ALOG_ASSERT(stepCount <= activeTrack->mRsmpInUnrel); 5601 activeTrack->mRsmpInUnrel -= stepCount; 5602 activeTrack->mRsmpInFront += stepCount; 5603 buffer->raw = NULL; 5604 buffer->frameCount = 0; 5605} 5606 5607bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 5608 status_t& status) 5609{ 5610 bool reconfig = false; 5611 5612 status = NO_ERROR; 5613 5614 audio_format_t reqFormat = mFormat; 5615 uint32_t samplingRate = mSampleRate; 5616 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 5617 5618 AudioParameter param = AudioParameter(keyValuePair); 5619 int value; 5620 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 5621 // channel count change can be requested. Do we mandate the first client defines the 5622 // HAL sampling rate and channel count or do we allow changes on the fly? 5623 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5624 samplingRate = value; 5625 reconfig = true; 5626 } 5627 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5628 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 5629 status = BAD_VALUE; 5630 } else { 5631 reqFormat = (audio_format_t) value; 5632 reconfig = true; 5633 } 5634 } 5635 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5636 audio_channel_mask_t mask = (audio_channel_mask_t) value; 5637 if (mask != AUDIO_CHANNEL_IN_MONO && mask != AUDIO_CHANNEL_IN_STEREO) { 5638 status = BAD_VALUE; 5639 } else { 5640 channelMask = mask; 5641 reconfig = true; 5642 } 5643 } 5644 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5645 // do not accept frame count changes if tracks are open as the track buffer 5646 // size depends on frame count and correct behavior would not be guaranteed 5647 // if frame count is changed after track creation 5648 if (mActiveTracks.size() > 0) { 5649 status = INVALID_OPERATION; 5650 } else { 5651 reconfig = true; 5652 } 5653 } 5654 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5655 // forward device change to effects that have requested to be 5656 // aware of attached audio device. 5657 for (size_t i = 0; i < mEffectChains.size(); i++) { 5658 mEffectChains[i]->setDevice_l(value); 5659 } 5660 5661 // store input device and output device but do not forward output device to audio HAL. 5662 // Note that status is ignored by the caller for output device 5663 // (see AudioFlinger::setParameters() 5664 if (audio_is_output_devices(value)) { 5665 mOutDevice = value; 5666 status = BAD_VALUE; 5667 } else { 5668 mInDevice = value; 5669 // disable AEC and NS if the device is a BT SCO headset supporting those 5670 // pre processings 5671 if (mTracks.size() > 0) { 5672 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5673 mAudioFlinger->btNrecIsOff(); 5674 for (size_t i = 0; i < mTracks.size(); i++) { 5675 sp<RecordTrack> track = mTracks[i]; 5676 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5677 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5678 } 5679 } 5680 } 5681 } 5682 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 5683 mAudioSource != (audio_source_t)value) { 5684 // forward device change to effects that have requested to be 5685 // aware of attached audio device. 5686 for (size_t i = 0; i < mEffectChains.size(); i++) { 5687 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 5688 } 5689 mAudioSource = (audio_source_t)value; 5690 } 5691 5692 if (status == NO_ERROR) { 5693 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5694 keyValuePair.string()); 5695 if (status == INVALID_OPERATION) { 5696 inputStandBy(); 5697 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5698 keyValuePair.string()); 5699 } 5700 if (reconfig) { 5701 if (status == BAD_VALUE && 5702 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5703 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5704 (mInput->stream->common.get_sample_rate(&mInput->stream->common) 5705 <= (2 * samplingRate)) && 5706 audio_channel_count_from_in_mask( 5707 mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5708 (channelMask == AUDIO_CHANNEL_IN_MONO || 5709 channelMask == AUDIO_CHANNEL_IN_STEREO)) { 5710 status = NO_ERROR; 5711 } 5712 if (status == NO_ERROR) { 5713 readInputParameters_l(); 5714 sendIoConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5715 } 5716 } 5717 } 5718 5719 return reconfig; 5720} 5721 5722String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5723{ 5724 Mutex::Autolock _l(mLock); 5725 if (initCheck() != NO_ERROR) { 5726 return String8(); 5727 } 5728 5729 char *s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5730 const String8 out_s8(s); 5731 free(s); 5732 return out_s8; 5733} 5734 5735void AudioFlinger::RecordThread::audioConfigChanged(int event, int param __unused) { 5736 AudioSystem::OutputDescriptor desc; 5737 const void *param2 = NULL; 5738 5739 switch (event) { 5740 case AudioSystem::INPUT_OPENED: 5741 case AudioSystem::INPUT_CONFIG_CHANGED: 5742 desc.channelMask = mChannelMask; 5743 desc.samplingRate = mSampleRate; 5744 desc.format = mFormat; 5745 desc.frameCount = mFrameCount; 5746 desc.latency = 0; 5747 param2 = &desc; 5748 break; 5749 5750 case AudioSystem::INPUT_CLOSED: 5751 default: 5752 break; 5753 } 5754 mAudioFlinger->audioConfigChanged(event, mId, param2); 5755} 5756 5757void AudioFlinger::RecordThread::readInputParameters_l() 5758{ 5759 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5760 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5761 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 5762 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5763 if (mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5764 ALOGE("HAL format %#x not supported; must be AUDIO_FORMAT_PCM_16_BIT", mFormat); 5765 } 5766 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5767 mBufferSize = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5768 mFrameCount = mBufferSize / mFrameSize; 5769 // This is the formula for calculating the temporary buffer size. 5770 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 5771 // 1 full output buffer, regardless of the alignment of the available input. 5772 // The value is somewhat arbitrary, and could probably be even larger. 5773 // A larger value should allow more old data to be read after a track calls start(), 5774 // without increasing latency. 5775 mRsmpInFrames = mFrameCount * 7; 5776 mRsmpInFramesP2 = roundup(mRsmpInFrames); 5777 delete[] mRsmpInBuffer; 5778 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 5779 mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount]; 5780 5781 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 5782 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 5783} 5784 5785uint32_t AudioFlinger::RecordThread::getInputFramesLost() 5786{ 5787 Mutex::Autolock _l(mLock); 5788 if (initCheck() != NO_ERROR) { 5789 return 0; 5790 } 5791 5792 return mInput->stream->get_input_frames_lost(mInput->stream); 5793} 5794 5795uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) const 5796{ 5797 Mutex::Autolock _l(mLock); 5798 uint32_t result = 0; 5799 if (getEffectChain_l(sessionId) != 0) { 5800 result = EFFECT_SESSION; 5801 } 5802 5803 for (size_t i = 0; i < mTracks.size(); ++i) { 5804 if (sessionId == mTracks[i]->sessionId()) { 5805 result |= TRACK_SESSION; 5806 break; 5807 } 5808 } 5809 5810 return result; 5811} 5812 5813KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() const 5814{ 5815 KeyedVector<int, bool> ids; 5816 Mutex::Autolock _l(mLock); 5817 for (size_t j = 0; j < mTracks.size(); ++j) { 5818 sp<RecordThread::RecordTrack> track = mTracks[j]; 5819 int sessionId = track->sessionId(); 5820 if (ids.indexOfKey(sessionId) < 0) { 5821 ids.add(sessionId, true); 5822 } 5823 } 5824 return ids; 5825} 5826 5827AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5828{ 5829 Mutex::Autolock _l(mLock); 5830 AudioStreamIn *input = mInput; 5831 mInput = NULL; 5832 return input; 5833} 5834 5835// this method must always be called either with ThreadBase mLock held or inside the thread loop 5836audio_stream_t* AudioFlinger::RecordThread::stream() const 5837{ 5838 if (mInput == NULL) { 5839 return NULL; 5840 } 5841 return &mInput->stream->common; 5842} 5843 5844status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5845{ 5846 // only one chain per input thread 5847 if (mEffectChains.size() != 0) { 5848 return INVALID_OPERATION; 5849 } 5850 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5851 5852 chain->setInBuffer(NULL); 5853 chain->setOutBuffer(NULL); 5854 5855 checkSuspendOnAddEffectChain_l(chain); 5856 5857 mEffectChains.add(chain); 5858 5859 return NO_ERROR; 5860} 5861 5862size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5863{ 5864 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5865 ALOGW_IF(mEffectChains.size() != 1, 5866 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5867 chain.get(), mEffectChains.size(), this); 5868 if (mEffectChains.size() == 1) { 5869 mEffectChains.removeAt(0); 5870 } 5871 return 0; 5872} 5873 5874status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 5875 audio_patch_handle_t *handle) 5876{ 5877 status_t status = NO_ERROR; 5878 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 5879 // store new device and send to effects 5880 mInDevice = patch->sources[0].ext.device.type; 5881 for (size_t i = 0; i < mEffectChains.size(); i++) { 5882 mEffectChains[i]->setDevice_l(mInDevice); 5883 } 5884 5885 // disable AEC and NS if the device is a BT SCO headset supporting those 5886 // pre processings 5887 if (mTracks.size() > 0) { 5888 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 5889 mAudioFlinger->btNrecIsOff(); 5890 for (size_t i = 0; i < mTracks.size(); i++) { 5891 sp<RecordTrack> track = mTracks[i]; 5892 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 5893 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 5894 } 5895 } 5896 5897 // store new source and send to effects 5898 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 5899 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 5900 for (size_t i = 0; i < mEffectChains.size(); i++) { 5901 mEffectChains[i]->setAudioSource_l(mAudioSource); 5902 } 5903 } 5904 5905 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 5906 status = hwDevice->create_audio_patch(hwDevice, 5907 patch->num_sources, 5908 patch->sources, 5909 patch->num_sinks, 5910 patch->sinks, 5911 handle); 5912 } else { 5913 ALOG_ASSERT(false, "createAudioPatch_l() called on a pre 3.0 HAL"); 5914 } 5915 return status; 5916} 5917 5918status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 5919{ 5920 status_t status = NO_ERROR; 5921 if (mInput->audioHwDev->version() >= AUDIO_DEVICE_API_VERSION_3_0) { 5922 audio_hw_device_t *hwDevice = mInput->audioHwDev->hwDevice(); 5923 status = hwDevice->release_audio_patch(hwDevice, handle); 5924 } else { 5925 ALOG_ASSERT(false, "releaseAudioPatch_l() called on a pre 3.0 HAL"); 5926 } 5927 return status; 5928} 5929 5930 5931}; // namespace android 5932