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 <linux/futex.h> 27 #include <sys/stat.h> 28 #include <sys/syscall.h> 29 #include <cutils/properties.h> 30 #include <media/AudioParameter.h> 31 #include <media/AudioResamplerPublic.h> 32 #include <media/RecordBufferConverter.h> 33 #include <media/TypeConverter.h> 34 #include <utils/Log.h> 35 #include <utils/Trace.h> 36 37 #include <private/media/AudioTrackShared.h> 38 #include <private/android_filesystem_config.h> 39 #include <audio_utils/mono_blend.h> 40 #include <audio_utils/primitives.h> 41 #include <audio_utils/format.h> 42 #include <audio_utils/minifloat.h> 43 #include <system/audio_effects/effect_ns.h> 44 #include <system/audio_effects/effect_aec.h> 45 #include <system/audio.h> 46 47 // NBAIO implementations 48 #include <media/nbaio/AudioStreamInSource.h> 49 #include <media/nbaio/AudioStreamOutSink.h> 50 #include <media/nbaio/MonoPipe.h> 51 #include <media/nbaio/MonoPipeReader.h> 52 #include <media/nbaio/Pipe.h> 53 #include <media/nbaio/PipeReader.h> 54 #include <media/nbaio/SourceAudioBufferProvider.h> 55 #include <mediautils/BatteryNotifier.h> 56 57 #include <powermanager/PowerManager.h> 58 59 #include <media/audiohal/EffectsFactoryHalInterface.h> 60 #include <media/audiohal/StreamHalInterface.h> 61 62 #include "AudioFlinger.h" 63 #include "FastMixer.h" 64 #include "FastCapture.h" 65 #include "ServiceUtilities.h" 66 #include "mediautils/SchedulingPolicyService.h" 67 68 #ifdef ADD_BATTERY_DATA 69 #include <media/IMediaPlayerService.h> 70 #include <media/IMediaDeathNotifier.h> 71 #endif 72 73 #ifdef DEBUG_CPU_USAGE 74 #include <cpustats/CentralTendencyStatistics.h> 75 #include <cpustats/ThreadCpuUsage.h> 76 #endif 77 78 #include "AutoPark.h" 79 80 #include <pthread.h> 81 #include "TypedLogger.h" 82 83 // ---------------------------------------------------------------------------- 84 85 // Note: the following macro is used for extremely verbose logging message. In 86 // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 87 // 0; but one side effect of this is to turn all LOGV's as well. Some messages 88 // are so verbose that we want to suppress them even when we have ALOG_ASSERT 89 // turned on. Do not uncomment the #def below unless you really know what you 90 // are doing and want to see all of the extremely verbose messages. 91 //#define VERY_VERY_VERBOSE_LOGGING 92 #ifdef VERY_VERY_VERBOSE_LOGGING 93 #define ALOGVV ALOGV 94 #else 95 #define ALOGVV(a...) do { } while(0) 96 #endif 97 98 // TODO: Move these macro/inlines to a header file. 99 #define max(a, b) ((a) > (b) ? (a) : (b)) 100 template <typename T> 101 static inline T min(const T& a, const T& b) 102 { 103 return a < b ? a : b; 104 } 105 106 namespace android { 107 108 // retry counts for buffer fill timeout 109 // 50 * ~20msecs = 1 second 110 static const int8_t kMaxTrackRetries = 50; 111 static const int8_t kMaxTrackStartupRetries = 50; 112 // allow less retry attempts on direct output thread. 113 // direct outputs can be a scarce resource in audio hardware and should 114 // be released as quickly as possible. 115 static const int8_t kMaxTrackRetriesDirect = 2; 116 117 118 119 // don't warn about blocked writes or record buffer overflows more often than this 120 static const nsecs_t kWarningThrottleNs = seconds(5); 121 122 // RecordThread loop sleep time upon application overrun or audio HAL read error 123 static const int kRecordThreadSleepUs = 5000; 124 125 // maximum time to wait in sendConfigEvent_l() for a status to be received 126 static const nsecs_t kConfigEventTimeoutNs = seconds(2); 127 128 // minimum sleep time for the mixer thread loop when tracks are active but in underrun 129 static const uint32_t kMinThreadSleepTimeUs = 5000; 130 // maximum divider applied to the active sleep time in the mixer thread loop 131 static const uint32_t kMaxThreadSleepTimeShift = 2; 132 133 // minimum normal sink buffer size, expressed in milliseconds rather than frames 134 // FIXME This should be based on experimentally observed scheduling jitter 135 static const uint32_t kMinNormalSinkBufferSizeMs = 20; 136 // maximum normal sink buffer size 137 static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 138 139 // minimum capture buffer size in milliseconds to _not_ need a fast capture thread 140 // FIXME This should be based on experimentally observed scheduling jitter 141 static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 142 143 // Offloaded output thread standby delay: allows track transition without going to standby 144 static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 145 146 // Direct output thread minimum sleep time in idle or active(underrun) state 147 static const nsecs_t kDirectMinSleepTimeUs = 10000; 148 149 // The universal constant for ubiquitous 20ms value. The value of 20ms seems to provide a good 150 // balance between power consumption and latency, and allows threads to be scheduled reliably 151 // by the CFS scheduler. 152 // FIXME Express other hardcoded references to 20ms with references to this constant and move 153 // it appropriately. 154 #define FMS_20 20 155 156 // Whether to use fast mixer 157 static const enum { 158 FastMixer_Never, // never initialize or use: for debugging only 159 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 160 // normal mixer multiplier is 1 161 FastMixer_Static, // initialize if needed, then use all the time if initialized, 162 // multiplier is calculated based on min & max normal mixer buffer size 163 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 164 // multiplier is calculated based on min & max normal mixer buffer size 165 // FIXME for FastMixer_Dynamic: 166 // Supporting this option will require fixing HALs that can't handle large writes. 167 // For example, one HAL implementation returns an error from a large write, 168 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 169 // We could either fix the HAL implementations, or provide a wrapper that breaks 170 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 171 } kUseFastMixer = FastMixer_Static; 172 173 // Whether to use fast capture 174 static const enum { 175 FastCapture_Never, // never initialize or use: for debugging only 176 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 177 FastCapture_Static, // initialize if needed, then use all the time if initialized 178 } kUseFastCapture = FastCapture_Static; 179 180 // Priorities for requestPriority 181 static const int kPriorityAudioApp = 2; 182 static const int kPriorityFastMixer = 3; 183 static const int kPriorityFastCapture = 3; 184 185 // IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the 186 // track buffer in shared memory. Zero on input means to use a default value. For fast tracks, 187 // AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'. 188 189 // This is the default value, if not specified by property. 190 static const int kFastTrackMultiplier = 2; 191 192 // The minimum and maximum allowed values 193 static const int kFastTrackMultiplierMin = 1; 194 static const int kFastTrackMultiplierMax = 2; 195 196 // The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 197 static int sFastTrackMultiplier = kFastTrackMultiplier; 198 199 // See Thread::readOnlyHeap(). 200 // Initially this heap is used to allocate client buffers for "fast" AudioRecord. 201 // Eventually it will be the single buffer that FastCapture writes into via HAL read(), 202 // and that all "fast" AudioRecord clients read from. In either case, the size can be small. 203 static const size_t kRecordThreadReadOnlyHeapSize = 0x4000; 204 205 // ---------------------------------------------------------------------------- 206 207 static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 208 209 static void sFastTrackMultiplierInit() 210 { 211 char value[PROPERTY_VALUE_MAX]; 212 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 213 char *endptr; 214 unsigned long ul = strtoul(value, &endptr, 0); 215 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 216 sFastTrackMultiplier = (int) ul; 217 } 218 } 219 } 220 221 // ---------------------------------------------------------------------------- 222 223 #ifdef ADD_BATTERY_DATA 224 // To collect the amplifier usage 225 static void addBatteryData(uint32_t params) { 226 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 227 if (service == NULL) { 228 // it already logged 229 return; 230 } 231 232 service->addBatteryData(params); 233 } 234 #endif 235 236 // Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 237 struct { 238 // call when you acquire a partial wakelock 239 void acquire(const sp<IBinder> &wakeLockToken) { 240 pthread_mutex_lock(&mLock); 241 if (wakeLockToken.get() == nullptr) { 242 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 243 } else { 244 if (mCount == 0) { 245 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 246 } 247 ++mCount; 248 } 249 pthread_mutex_unlock(&mLock); 250 } 251 252 // call when you release a partial wakelock. 253 void release(const sp<IBinder> &wakeLockToken) { 254 if (wakeLockToken.get() == nullptr) { 255 return; 256 } 257 pthread_mutex_lock(&mLock); 258 if (--mCount < 0) { 259 ALOGE("negative wakelock count"); 260 mCount = 0; 261 } 262 pthread_mutex_unlock(&mLock); 263 } 264 265 // retrieves the boottime timebase offset from monotonic. 266 int64_t getBoottimeOffset() { 267 pthread_mutex_lock(&mLock); 268 int64_t boottimeOffset = mBoottimeOffset; 269 pthread_mutex_unlock(&mLock); 270 return boottimeOffset; 271 } 272 273 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 274 // and the selected timebase. 275 // Currently only TIMEBASE_BOOTTIME is allowed. 276 // 277 // This only needs to be called upon acquiring the first partial wakelock 278 // after all other partial wakelocks are released. 279 // 280 // We do an empirical measurement of the offset rather than parsing 281 // /proc/timer_list since the latter is not a formal kernel ABI. 282 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 283 int clockbase; 284 switch (timebase) { 285 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 286 clockbase = SYSTEM_TIME_BOOTTIME; 287 break; 288 default: 289 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 290 break; 291 } 292 // try three times to get the clock offset, choose the one 293 // with the minimum gap in measurements. 294 const int tries = 3; 295 nsecs_t bestGap, measured; 296 for (int i = 0; i < tries; ++i) { 297 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 298 const nsecs_t tbase = systemTime(clockbase); 299 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 300 const nsecs_t gap = tmono2 - tmono; 301 if (i == 0 || gap < bestGap) { 302 bestGap = gap; 303 measured = tbase - ((tmono + tmono2) >> 1); 304 } 305 } 306 307 // to avoid micro-adjusting, we don't change the timebase 308 // unless it is significantly different. 309 // 310 // Assumption: It probably takes more than toleranceNs to 311 // suspend and resume the device. 312 static int64_t toleranceNs = 10000; // 10 us 313 if (llabs(*offset - measured) > toleranceNs) { 314 ALOGV("Adjusting timebase offset old: %lld new: %lld", 315 (long long)*offset, (long long)measured); 316 *offset = measured; 317 } 318 } 319 320 pthread_mutex_t mLock; 321 int32_t mCount; 322 int64_t mBoottimeOffset; 323 } gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 324 325 // ---------------------------------------------------------------------------- 326 // CPU Stats 327 // ---------------------------------------------------------------------------- 328 329 class CpuStats { 330 public: 331 CpuStats(); 332 void sample(const String8 &title); 333 #ifdef DEBUG_CPU_USAGE 334 private: 335 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 336 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 337 338 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 339 340 int mCpuNum; // thread's current CPU number 341 int mCpukHz; // frequency of thread's current CPU in kHz 342 #endif 343 }; 344 345 CpuStats::CpuStats() 346 #ifdef DEBUG_CPU_USAGE 347 : mCpuNum(-1), mCpukHz(-1) 348 #endif 349 { 350 } 351 352 void CpuStats::sample(const String8 &title 353 #ifndef DEBUG_CPU_USAGE 354 __unused 355 #endif 356 ) { 357 #ifdef DEBUG_CPU_USAGE 358 // get current thread's delta CPU time in wall clock ns 359 double wcNs; 360 bool valid = mCpuUsage.sampleAndEnable(wcNs); 361 362 // record sample for wall clock statistics 363 if (valid) { 364 mWcStats.sample(wcNs); 365 } 366 367 // get the current CPU number 368 int cpuNum = sched_getcpu(); 369 370 // get the current CPU frequency in kHz 371 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 372 373 // check if either CPU number or frequency changed 374 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 375 mCpuNum = cpuNum; 376 mCpukHz = cpukHz; 377 // ignore sample for purposes of cycles 378 valid = false; 379 } 380 381 // if no change in CPU number or frequency, then record sample for cycle statistics 382 if (valid && mCpukHz > 0) { 383 double cycles = wcNs * cpukHz * 0.000001; 384 mHzStats.sample(cycles); 385 } 386 387 unsigned n = mWcStats.n(); 388 // mCpuUsage.elapsed() is expensive, so don't call it every loop 389 if ((n & 127) == 1) { 390 long long elapsed = mCpuUsage.elapsed(); 391 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 392 double perLoop = elapsed / (double) n; 393 double perLoop100 = perLoop * 0.01; 394 double perLoop1k = perLoop * 0.001; 395 double mean = mWcStats.mean(); 396 double stddev = mWcStats.stddev(); 397 double minimum = mWcStats.minimum(); 398 double maximum = mWcStats.maximum(); 399 double meanCycles = mHzStats.mean(); 400 double stddevCycles = mHzStats.stddev(); 401 double minCycles = mHzStats.minimum(); 402 double maxCycles = mHzStats.maximum(); 403 mCpuUsage.resetElapsed(); 404 mWcStats.reset(); 405 mHzStats.reset(); 406 ALOGD("CPU usage for %s over past %.1f secs\n" 407 " (%u mixer loops at %.1f mean ms per loop):\n" 408 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 409 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 410 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 411 title.string(), 412 elapsed * .000000001, n, perLoop * .000001, 413 mean * .001, 414 stddev * .001, 415 minimum * .001, 416 maximum * .001, 417 mean / perLoop100, 418 stddev / perLoop100, 419 minimum / perLoop100, 420 maximum / perLoop100, 421 meanCycles / perLoop1k, 422 stddevCycles / perLoop1k, 423 minCycles / perLoop1k, 424 maxCycles / perLoop1k); 425 426 } 427 } 428 #endif 429 }; 430 431 // ---------------------------------------------------------------------------- 432 // ThreadBase 433 // ---------------------------------------------------------------------------- 434 435 // static 436 const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 437 { 438 switch (type) { 439 case MIXER: 440 return "MIXER"; 441 case DIRECT: 442 return "DIRECT"; 443 case DUPLICATING: 444 return "DUPLICATING"; 445 case RECORD: 446 return "RECORD"; 447 case OFFLOAD: 448 return "OFFLOAD"; 449 case MMAP: 450 return "MMAP"; 451 default: 452 return "unknown"; 453 } 454 } 455 456 std::string devicesToString(audio_devices_t devices) 457 { 458 std::string result; 459 if (devices & AUDIO_DEVICE_BIT_IN) { 460 InputDeviceConverter::maskToString(devices, result); 461 } else { 462 OutputDeviceConverter::maskToString(devices, result); 463 } 464 return result; 465 } 466 467 std::string inputFlagsToString(audio_input_flags_t flags) 468 { 469 std::string result; 470 InputFlagConverter::maskToString(flags, result); 471 return result; 472 } 473 474 std::string outputFlagsToString(audio_output_flags_t flags) 475 { 476 std::string result; 477 OutputFlagConverter::maskToString(flags, result); 478 return result; 479 } 480 481 const char *sourceToString(audio_source_t source) 482 { 483 switch (source) { 484 case AUDIO_SOURCE_DEFAULT: return "default"; 485 case AUDIO_SOURCE_MIC: return "mic"; 486 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 487 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 488 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 489 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 490 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 491 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 492 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 493 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 494 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 495 case AUDIO_SOURCE_HOTWORD: return "hotword"; 496 default: return "unknown"; 497 } 498 } 499 500 AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 501 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 502 : Thread(false /*canCallJava*/), 503 mType(type), 504 mAudioFlinger(audioFlinger), 505 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 506 // are set by PlaybackThread::readOutputParameters_l() or 507 // RecordThread::readInputParameters_l() 508 //FIXME: mStandby should be true here. Is this some kind of hack? 509 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 510 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 511 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 512 // mName will be set by concrete (non-virtual) subclass 513 mDeathRecipient(new PMDeathRecipient(this)), 514 mSystemReady(systemReady), 515 mSignalPending(false) 516 { 517 memset(&mPatch, 0, sizeof(struct audio_patch)); 518 } 519 520 AudioFlinger::ThreadBase::~ThreadBase() 521 { 522 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 523 mConfigEvents.clear(); 524 525 // do not lock the mutex in destructor 526 releaseWakeLock_l(); 527 if (mPowerManager != 0) { 528 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 529 binder->unlinkToDeath(mDeathRecipient); 530 } 531 } 532 533 status_t AudioFlinger::ThreadBase::readyToRun() 534 { 535 status_t status = initCheck(); 536 if (status == NO_ERROR) { 537 ALOGI("AudioFlinger's thread %p tid=%d ready to run", this, getTid()); 538 } else { 539 ALOGE("No working audio driver found."); 540 } 541 return status; 542 } 543 544 void AudioFlinger::ThreadBase::exit() 545 { 546 ALOGV("ThreadBase::exit"); 547 // do any cleanup required for exit to succeed 548 preExit(); 549 { 550 // This lock prevents the following race in thread (uniprocessor for illustration): 551 // if (!exitPending()) { 552 // // context switch from here to exit() 553 // // exit() calls requestExit(), what exitPending() observes 554 // // exit() calls signal(), which is dropped since no waiters 555 // // context switch back from exit() to here 556 // mWaitWorkCV.wait(...); 557 // // now thread is hung 558 // } 559 AutoMutex lock(mLock); 560 requestExit(); 561 mWaitWorkCV.broadcast(); 562 } 563 // When Thread::requestExitAndWait is made virtual and this method is renamed to 564 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 565 requestExitAndWait(); 566 } 567 568 status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 569 { 570 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 571 Mutex::Autolock _l(mLock); 572 573 return sendSetParameterConfigEvent_l(keyValuePairs); 574 } 575 576 // sendConfigEvent_l() must be called with ThreadBase::mLock held 577 // Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 578 status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 579 { 580 status_t status = NO_ERROR; 581 582 if (event->mRequiresSystemReady && !mSystemReady) { 583 event->mWaitStatus = false; 584 mPendingConfigEvents.add(event); 585 return status; 586 } 587 mConfigEvents.add(event); 588 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 589 mWaitWorkCV.signal(); 590 mLock.unlock(); 591 { 592 Mutex::Autolock _l(event->mLock); 593 while (event->mWaitStatus) { 594 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 595 event->mStatus = TIMED_OUT; 596 event->mWaitStatus = false; 597 } 598 } 599 status = event->mStatus; 600 } 601 mLock.lock(); 602 return status; 603 } 604 605 void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 606 { 607 Mutex::Autolock _l(mLock); 608 sendIoConfigEvent_l(event, pid); 609 } 610 611 // sendIoConfigEvent_l() must be called with ThreadBase::mLock held 612 void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 613 { 614 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 615 sendConfigEvent_l(configEvent); 616 } 617 618 void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) 619 { 620 Mutex::Autolock _l(mLock); 621 sendPrioConfigEvent_l(pid, tid, prio, forApp); 622 } 623 624 // sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 625 void AudioFlinger::ThreadBase::sendPrioConfigEvent_l( 626 pid_t pid, pid_t tid, int32_t prio, bool forApp) 627 { 628 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio, forApp); 629 sendConfigEvent_l(configEvent); 630 } 631 632 // sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 633 status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 634 { 635 sp<ConfigEvent> configEvent; 636 AudioParameter param(keyValuePair); 637 int value; 638 if (param.getInt(String8(AudioParameter::keyMonoOutput), value) == NO_ERROR) { 639 setMasterMono_l(value != 0); 640 if (param.size() == 1) { 641 return NO_ERROR; // should be a solo parameter - we don't pass down 642 } 643 param.remove(String8(AudioParameter::keyMonoOutput)); 644 configEvent = new SetParameterConfigEvent(param.toString()); 645 } else { 646 configEvent = new SetParameterConfigEvent(keyValuePair); 647 } 648 return sendConfigEvent_l(configEvent); 649 } 650 651 status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 652 const struct audio_patch *patch, 653 audio_patch_handle_t *handle) 654 { 655 Mutex::Autolock _l(mLock); 656 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 657 status_t status = sendConfigEvent_l(configEvent); 658 if (status == NO_ERROR) { 659 CreateAudioPatchConfigEventData *data = 660 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 661 *handle = data->mHandle; 662 } 663 return status; 664 } 665 666 status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 667 const audio_patch_handle_t handle) 668 { 669 Mutex::Autolock _l(mLock); 670 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 671 return sendConfigEvent_l(configEvent); 672 } 673 674 675 // post condition: mConfigEvents.isEmpty() 676 void AudioFlinger::ThreadBase::processConfigEvents_l() 677 { 678 bool configChanged = false; 679 680 while (!mConfigEvents.isEmpty()) { 681 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 682 sp<ConfigEvent> event = mConfigEvents[0]; 683 mConfigEvents.removeAt(0); 684 switch (event->mType) { 685 case CFG_EVENT_PRIO: { 686 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 687 // FIXME Need to understand why this has to be done asynchronously 688 int err = requestPriority(data->mPid, data->mTid, data->mPrio, data->mForApp, 689 true /*asynchronous*/); 690 if (err != 0) { 691 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 692 data->mPrio, data->mPid, data->mTid, err); 693 } 694 } break; 695 case CFG_EVENT_IO: { 696 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 697 ioConfigChanged(data->mEvent, data->mPid); 698 } break; 699 case CFG_EVENT_SET_PARAMETER: { 700 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 701 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 702 configChanged = true; 703 mLocalLog.log("CFG_EVENT_SET_PARAMETER: (%s) configuration changed", 704 data->mKeyValuePairs.string()); 705 } 706 } break; 707 case CFG_EVENT_CREATE_AUDIO_PATCH: { 708 const audio_devices_t oldDevice = getDevice(); 709 CreateAudioPatchConfigEventData *data = 710 (CreateAudioPatchConfigEventData *)event->mData.get(); 711 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 712 const audio_devices_t newDevice = getDevice(); 713 mLocalLog.log("CFG_EVENT_CREATE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)", 714 (unsigned)oldDevice, devicesToString(oldDevice).c_str(), 715 (unsigned)newDevice, devicesToString(newDevice).c_str()); 716 } break; 717 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 718 const audio_devices_t oldDevice = getDevice(); 719 ReleaseAudioPatchConfigEventData *data = 720 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 721 event->mStatus = releaseAudioPatch_l(data->mHandle); 722 const audio_devices_t newDevice = getDevice(); 723 mLocalLog.log("CFG_EVENT_RELEASE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)", 724 (unsigned)oldDevice, devicesToString(oldDevice).c_str(), 725 (unsigned)newDevice, devicesToString(newDevice).c_str()); 726 } break; 727 default: 728 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 729 break; 730 } 731 { 732 Mutex::Autolock _l(event->mLock); 733 if (event->mWaitStatus) { 734 event->mWaitStatus = false; 735 event->mCond.signal(); 736 } 737 } 738 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 739 } 740 741 if (configChanged) { 742 cacheParameters_l(); 743 } 744 } 745 746 String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 747 String8 s; 748 const audio_channel_representation_t representation = 749 audio_channel_mask_get_representation(mask); 750 751 switch (representation) { 752 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 753 if (output) { 754 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 755 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 756 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 757 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 758 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 759 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 760 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 761 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 762 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 763 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 764 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 765 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 766 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 767 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 768 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 769 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 770 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 771 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 772 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 773 } else { 774 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 775 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 776 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 777 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 778 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 779 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 780 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 781 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 782 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 783 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 784 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 785 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 786 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 787 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 788 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 789 } 790 const int len = s.length(); 791 if (len > 2) { 792 (void) s.lockBuffer(len); // needed? 793 s.unlockBuffer(len - 2); // remove trailing ", " 794 } 795 return s; 796 } 797 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 798 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 799 return s; 800 default: 801 s.appendFormat("unknown mask, representation:%d bits:%#x", 802 representation, audio_channel_mask_get_bits(mask)); 803 return s; 804 } 805 } 806 807 void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 808 { 809 const size_t SIZE = 256; 810 char buffer[SIZE]; 811 String8 result; 812 813 dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input", 814 this, mThreadName, getTid(), type(), threadTypeToString(type())); 815 816 bool locked = AudioFlinger::dumpTryLock(mLock); 817 if (!locked) { 818 dprintf(fd, " Thread may be deadlocked\n"); 819 } 820 821 dprintf(fd, " I/O handle: %d\n", mId); 822 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 823 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 824 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 825 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat).c_str()); 826 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 827 dprintf(fd, " Channel count: %u\n", mChannelCount); 828 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 829 channelMaskToString(mChannelMask, mType != RECORD).string()); 830 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat).c_str()); 831 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 832 dprintf(fd, " Pending config events:"); 833 size_t numConfig = mConfigEvents.size(); 834 if (numConfig) { 835 for (size_t i = 0; i < numConfig; i++) { 836 mConfigEvents[i]->dump(buffer, SIZE); 837 dprintf(fd, "\n %s", buffer); 838 } 839 dprintf(fd, "\n"); 840 } else { 841 dprintf(fd, " none\n"); 842 } 843 // Note: output device may be used by capture threads for effects such as AEC. 844 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).c_str()); 845 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).c_str()); 846 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 847 848 if (locked) { 849 mLock.unlock(); 850 } 851 } 852 853 void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 854 { 855 const size_t SIZE = 256; 856 char buffer[SIZE]; 857 String8 result; 858 859 size_t numEffectChains = mEffectChains.size(); 860 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 861 write(fd, buffer, strlen(buffer)); 862 863 for (size_t i = 0; i < numEffectChains; ++i) { 864 sp<EffectChain> chain = mEffectChains[i]; 865 if (chain != 0) { 866 chain->dump(fd, args); 867 } 868 } 869 } 870 871 void AudioFlinger::ThreadBase::acquireWakeLock() 872 { 873 Mutex::Autolock _l(mLock); 874 acquireWakeLock_l(); 875 } 876 877 String16 AudioFlinger::ThreadBase::getWakeLockTag() 878 { 879 switch (mType) { 880 case MIXER: 881 return String16("AudioMix"); 882 case DIRECT: 883 return String16("AudioDirectOut"); 884 case DUPLICATING: 885 return String16("AudioDup"); 886 case RECORD: 887 return String16("AudioIn"); 888 case OFFLOAD: 889 return String16("AudioOffload"); 890 case MMAP: 891 return String16("Mmap"); 892 default: 893 ALOG_ASSERT(false); 894 return String16("AudioUnknown"); 895 } 896 } 897 898 void AudioFlinger::ThreadBase::acquireWakeLock_l() 899 { 900 getPowerManager_l(); 901 if (mPowerManager != 0) { 902 sp<IBinder> binder = new BBinder(); 903 // Uses AID_AUDIOSERVER for wakelock. updateWakeLockUids_l() updates with client uids. 904 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 905 binder, 906 getWakeLockTag(), 907 String16("audioserver"), 908 true /* FIXME force oneway contrary to .aidl */); 909 if (status == NO_ERROR) { 910 mWakeLockToken = binder; 911 } 912 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 913 } 914 915 gBoottime.acquire(mWakeLockToken); 916 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 917 gBoottime.getBoottimeOffset(); 918 } 919 920 void AudioFlinger::ThreadBase::releaseWakeLock() 921 { 922 Mutex::Autolock _l(mLock); 923 releaseWakeLock_l(); 924 } 925 926 void AudioFlinger::ThreadBase::releaseWakeLock_l() 927 { 928 gBoottime.release(mWakeLockToken); 929 if (mWakeLockToken != 0) { 930 ALOGV("releaseWakeLock_l() %s", mThreadName); 931 if (mPowerManager != 0) { 932 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 933 true /* FIXME force oneway contrary to .aidl */); 934 } 935 mWakeLockToken.clear(); 936 } 937 } 938 939 void AudioFlinger::ThreadBase::getPowerManager_l() { 940 if (mSystemReady && mPowerManager == 0) { 941 // use checkService() to avoid blocking if power service is not up yet 942 sp<IBinder> binder = 943 defaultServiceManager()->checkService(String16("power")); 944 if (binder == 0) { 945 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 946 } else { 947 mPowerManager = interface_cast<IPowerManager>(binder); 948 binder->linkToDeath(mDeathRecipient); 949 } 950 } 951 } 952 953 void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t> &uids) { 954 getPowerManager_l(); 955 956 #if !LOG_NDEBUG 957 std::stringstream s; 958 for (uid_t uid : uids) { 959 s << uid << " "; 960 } 961 ALOGD("updateWakeLockUids_l %s uids:%s", mThreadName, s.str().c_str()); 962 #endif 963 964 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 965 if (mSystemReady) { 966 ALOGE("no wake lock to update, but system ready!"); 967 } else { 968 ALOGW("no wake lock to update, system not ready yet"); 969 } 970 return; 971 } 972 if (mPowerManager != 0) { 973 std::vector<int> uidsAsInt(uids.begin(), uids.end()); // powermanager expects uids as ints 974 status_t status = mPowerManager->updateWakeLockUids( 975 mWakeLockToken, uidsAsInt.size(), uidsAsInt.data(), 976 true /* FIXME force oneway contrary to .aidl */); 977 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 978 } 979 } 980 981 void AudioFlinger::ThreadBase::clearPowerManager() 982 { 983 Mutex::Autolock _l(mLock); 984 releaseWakeLock_l(); 985 mPowerManager.clear(); 986 } 987 988 void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 989 { 990 sp<ThreadBase> thread = mThread.promote(); 991 if (thread != 0) { 992 thread->clearPowerManager(); 993 } 994 ALOGW("power manager service died !!!"); 995 } 996 997 void AudioFlinger::ThreadBase::setEffectSuspended_l( 998 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 999 { 1000 sp<EffectChain> chain = getEffectChain_l(sessionId); 1001 if (chain != 0) { 1002 if (type != NULL) { 1003 chain->setEffectSuspended_l(type, suspend); 1004 } else { 1005 chain->setEffectSuspendedAll_l(suspend); 1006 } 1007 } 1008 1009 updateSuspendedSessions_l(type, suspend, sessionId); 1010 } 1011 1012 void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1013 { 1014 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1015 if (index < 0) { 1016 return; 1017 } 1018 1019 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1020 mSuspendedSessions.valueAt(index); 1021 1022 for (size_t i = 0; i < sessionEffects.size(); i++) { 1023 const sp<SuspendedSessionDesc>& desc = sessionEffects.valueAt(i); 1024 for (int j = 0; j < desc->mRefCount; j++) { 1025 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1026 chain->setEffectSuspendedAll_l(true); 1027 } else { 1028 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1029 desc->mType.timeLow); 1030 chain->setEffectSuspended_l(&desc->mType, true); 1031 } 1032 } 1033 } 1034 } 1035 1036 void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1037 bool suspend, 1038 audio_session_t sessionId) 1039 { 1040 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1041 1042 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1043 1044 if (suspend) { 1045 if (index >= 0) { 1046 sessionEffects = mSuspendedSessions.valueAt(index); 1047 } else { 1048 mSuspendedSessions.add(sessionId, sessionEffects); 1049 } 1050 } else { 1051 if (index < 0) { 1052 return; 1053 } 1054 sessionEffects = mSuspendedSessions.valueAt(index); 1055 } 1056 1057 1058 int key = EffectChain::kKeyForSuspendAll; 1059 if (type != NULL) { 1060 key = type->timeLow; 1061 } 1062 index = sessionEffects.indexOfKey(key); 1063 1064 sp<SuspendedSessionDesc> desc; 1065 if (suspend) { 1066 if (index >= 0) { 1067 desc = sessionEffects.valueAt(index); 1068 } else { 1069 desc = new SuspendedSessionDesc(); 1070 if (type != NULL) { 1071 desc->mType = *type; 1072 } 1073 sessionEffects.add(key, desc); 1074 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1075 } 1076 desc->mRefCount++; 1077 } else { 1078 if (index < 0) { 1079 return; 1080 } 1081 desc = sessionEffects.valueAt(index); 1082 if (--desc->mRefCount == 0) { 1083 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1084 sessionEffects.removeItemsAt(index); 1085 if (sessionEffects.isEmpty()) { 1086 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1087 sessionId); 1088 mSuspendedSessions.removeItem(sessionId); 1089 } 1090 } 1091 } 1092 if (!sessionEffects.isEmpty()) { 1093 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1094 } 1095 } 1096 1097 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1098 bool enabled, 1099 audio_session_t sessionId) 1100 { 1101 Mutex::Autolock _l(mLock); 1102 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1103 } 1104 1105 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1106 bool enabled, 1107 audio_session_t sessionId) 1108 { 1109 if (mType != RECORD) { 1110 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1111 // another session. This gives the priority to well behaved effect control panels 1112 // and applications not using global effects. 1113 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1114 // global effects 1115 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1116 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1117 } 1118 } 1119 1120 sp<EffectChain> chain = getEffectChain_l(sessionId); 1121 if (chain != 0) { 1122 chain->checkSuspendOnEffectEnabled(effect, enabled); 1123 } 1124 } 1125 1126 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1127 status_t AudioFlinger::RecordThread::checkEffectCompatibility_l( 1128 const effect_descriptor_t *desc, audio_session_t sessionId) 1129 { 1130 // No global effect sessions on record threads 1131 if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1132 ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s", 1133 desc->name, mThreadName); 1134 return BAD_VALUE; 1135 } 1136 // only pre processing effects on record thread 1137 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) { 1138 ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s", 1139 desc->name, mThreadName); 1140 return BAD_VALUE; 1141 } 1142 1143 // always allow effects without processing load or latency 1144 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1145 return NO_ERROR; 1146 } 1147 1148 audio_input_flags_t flags = mInput->flags; 1149 if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) { 1150 if (flags & AUDIO_INPUT_FLAG_RAW) { 1151 ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode", 1152 desc->name, mThreadName); 1153 return BAD_VALUE; 1154 } 1155 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1156 ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode", 1157 desc->name, mThreadName); 1158 return BAD_VALUE; 1159 } 1160 } 1161 return NO_ERROR; 1162 } 1163 1164 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1165 status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l( 1166 const effect_descriptor_t *desc, audio_session_t sessionId) 1167 { 1168 // no preprocessing on playback threads 1169 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) { 1170 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback" 1171 " thread %s", desc->name, mThreadName); 1172 return BAD_VALUE; 1173 } 1174 1175 // always allow effects without processing load or latency 1176 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1177 return NO_ERROR; 1178 } 1179 1180 switch (mType) { 1181 case MIXER: { 1182 #ifndef MULTICHANNEL_EFFECT_CHAIN 1183 // Reject any effect on mixer multichannel sinks. 1184 // TODO: fix both format and multichannel issues with effects. 1185 if (mChannelCount != FCC_2) { 1186 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER" 1187 " thread %s", desc->name, mChannelCount, mThreadName); 1188 return BAD_VALUE; 1189 } 1190 #endif 1191 audio_output_flags_t flags = mOutput->flags; 1192 if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) { 1193 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1194 // global effects are applied only to non fast tracks if they are SW 1195 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1196 break; 1197 } 1198 } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1199 // only post processing on output stage session 1200 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) { 1201 ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed" 1202 " on output stage session", desc->name); 1203 return BAD_VALUE; 1204 } 1205 } else { 1206 // no restriction on effects applied on non fast tracks 1207 if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) { 1208 break; 1209 } 1210 } 1211 1212 if (flags & AUDIO_OUTPUT_FLAG_RAW) { 1213 ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode", 1214 desc->name); 1215 return BAD_VALUE; 1216 } 1217 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1218 ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread" 1219 " in fast mode", desc->name); 1220 return BAD_VALUE; 1221 } 1222 } 1223 } break; 1224 case OFFLOAD: 1225 // nothing actionable on offload threads, if the effect: 1226 // - is offloadable: the effect can be created 1227 // - is NOT offloadable: the effect should still be created, but EffectHandle::enable() 1228 // will take care of invalidating the tracks of the thread 1229 break; 1230 case DIRECT: 1231 // Reject any effect on Direct output threads for now, since the format of 1232 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1233 ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s", 1234 desc->name, mThreadName); 1235 return BAD_VALUE; 1236 case DUPLICATING: 1237 #ifndef MULTICHANNEL_EFFECT_CHAIN 1238 // Reject any effect on mixer multichannel sinks. 1239 // TODO: fix both format and multichannel issues with effects. 1240 if (mChannelCount != FCC_2) { 1241 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)" 1242 " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName); 1243 return BAD_VALUE; 1244 } 1245 #endif 1246 if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) { 1247 ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING" 1248 " thread %s", desc->name, mThreadName); 1249 return BAD_VALUE; 1250 } 1251 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 1252 ALOGW("checkEffectCompatibility_l(): post processing effect %s on" 1253 " DUPLICATING thread %s", desc->name, mThreadName); 1254 return BAD_VALUE; 1255 } 1256 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) { 1257 ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on" 1258 " DUPLICATING thread %s", desc->name, mThreadName); 1259 return BAD_VALUE; 1260 } 1261 break; 1262 default: 1263 LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType); 1264 } 1265 1266 return NO_ERROR; 1267 } 1268 1269 // ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1270 sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1271 const sp<AudioFlinger::Client>& client, 1272 const sp<IEffectClient>& effectClient, 1273 int32_t priority, 1274 audio_session_t sessionId, 1275 effect_descriptor_t *desc, 1276 int *enabled, 1277 status_t *status, 1278 bool pinned) 1279 { 1280 sp<EffectModule> effect; 1281 sp<EffectHandle> handle; 1282 status_t lStatus; 1283 sp<EffectChain> chain; 1284 bool chainCreated = false; 1285 bool effectCreated = false; 1286 bool effectRegistered = false; 1287 audio_unique_id_t effectId = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; 1288 1289 lStatus = initCheck(); 1290 if (lStatus != NO_ERROR) { 1291 ALOGW("createEffect_l() Audio driver not initialized."); 1292 goto Exit; 1293 } 1294 1295 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1296 1297 { // scope for mLock 1298 Mutex::Autolock _l(mLock); 1299 1300 lStatus = checkEffectCompatibility_l(desc, sessionId); 1301 if (lStatus != NO_ERROR) { 1302 goto Exit; 1303 } 1304 1305 // check for existing effect chain with the requested audio session 1306 chain = getEffectChain_l(sessionId); 1307 if (chain == 0) { 1308 // create a new chain for this session 1309 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1310 chain = new EffectChain(this, sessionId); 1311 addEffectChain_l(chain); 1312 chain->setStrategy(getStrategyForSession_l(sessionId)); 1313 chainCreated = true; 1314 } else { 1315 effect = chain->getEffectFromDesc_l(desc); 1316 } 1317 1318 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1319 1320 if (effect == 0) { 1321 effectId = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1322 // Check CPU and memory usage 1323 lStatus = AudioSystem::registerEffect( 1324 desc, mId, chain->strategy(), sessionId, effectId); 1325 if (lStatus != NO_ERROR) { 1326 goto Exit; 1327 } 1328 effectRegistered = true; 1329 // create a new effect module if none present in the chain 1330 lStatus = chain->createEffect_l(effect, this, desc, effectId, sessionId, pinned); 1331 if (lStatus != NO_ERROR) { 1332 goto Exit; 1333 } 1334 effectCreated = true; 1335 1336 effect->setDevice(mOutDevice); 1337 effect->setDevice(mInDevice); 1338 effect->setMode(mAudioFlinger->getMode()); 1339 effect->setAudioSource(mAudioSource); 1340 } 1341 // create effect handle and connect it to effect module 1342 handle = new EffectHandle(effect, client, effectClient, priority); 1343 lStatus = handle->initCheck(); 1344 if (lStatus == OK) { 1345 lStatus = effect->addHandle(handle.get()); 1346 } 1347 if (enabled != NULL) { 1348 *enabled = (int)effect->isEnabled(); 1349 } 1350 } 1351 1352 Exit: 1353 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1354 Mutex::Autolock _l(mLock); 1355 if (effectCreated) { 1356 chain->removeEffect_l(effect); 1357 } 1358 if (effectRegistered) { 1359 AudioSystem::unregisterEffect(effectId); 1360 } 1361 if (chainCreated) { 1362 removeEffectChain_l(chain); 1363 } 1364 // handle must be cleared by caller to avoid deadlock. 1365 } 1366 1367 *status = lStatus; 1368 return handle; 1369 } 1370 1371 void AudioFlinger::ThreadBase::disconnectEffectHandle(EffectHandle *handle, 1372 bool unpinIfLast) 1373 { 1374 bool remove = false; 1375 sp<EffectModule> effect; 1376 { 1377 Mutex::Autolock _l(mLock); 1378 1379 effect = handle->effect().promote(); 1380 if (effect == 0) { 1381 return; 1382 } 1383 // restore suspended effects if the disconnected handle was enabled and the last one. 1384 remove = (effect->removeHandle(handle) == 0) && (!effect->isPinned() || unpinIfLast); 1385 if (remove) { 1386 removeEffect_l(effect, true); 1387 } 1388 } 1389 if (remove) { 1390 mAudioFlinger->updateOrphanEffectChains(effect); 1391 AudioSystem::unregisterEffect(effect->id()); 1392 if (handle->enabled()) { 1393 checkSuspendOnEffectEnabled(effect, false, effect->sessionId()); 1394 } 1395 } 1396 } 1397 1398 sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1399 int effectId) 1400 { 1401 Mutex::Autolock _l(mLock); 1402 return getEffect_l(sessionId, effectId); 1403 } 1404 1405 sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1406 int effectId) 1407 { 1408 sp<EffectChain> chain = getEffectChain_l(sessionId); 1409 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1410 } 1411 1412 // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1413 // PlaybackThread::mLock held 1414 status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1415 { 1416 // check for existing effect chain with the requested audio session 1417 audio_session_t sessionId = effect->sessionId(); 1418 sp<EffectChain> chain = getEffectChain_l(sessionId); 1419 bool chainCreated = false; 1420 1421 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1422 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %#x", 1423 this, effect->desc().name, effect->desc().flags); 1424 1425 if (chain == 0) { 1426 // create a new chain for this session 1427 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1428 chain = new EffectChain(this, sessionId); 1429 addEffectChain_l(chain); 1430 chain->setStrategy(getStrategyForSession_l(sessionId)); 1431 chainCreated = true; 1432 } 1433 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1434 1435 if (chain->getEffectFromId_l(effect->id()) != 0) { 1436 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1437 this, effect->desc().name, chain.get()); 1438 return BAD_VALUE; 1439 } 1440 1441 effect->setOffloaded(mType == OFFLOAD, mId); 1442 1443 status_t status = chain->addEffect_l(effect); 1444 if (status != NO_ERROR) { 1445 if (chainCreated) { 1446 removeEffectChain_l(chain); 1447 } 1448 return status; 1449 } 1450 1451 effect->setDevice(mOutDevice); 1452 effect->setDevice(mInDevice); 1453 effect->setMode(mAudioFlinger->getMode()); 1454 effect->setAudioSource(mAudioSource); 1455 1456 return NO_ERROR; 1457 } 1458 1459 void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect, bool release) { 1460 1461 ALOGV("%s %p effect %p", __FUNCTION__, this, effect.get()); 1462 effect_descriptor_t desc = effect->desc(); 1463 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1464 detachAuxEffect_l(effect->id()); 1465 } 1466 1467 sp<EffectChain> chain = effect->chain().promote(); 1468 if (chain != 0) { 1469 // remove effect chain if removing last effect 1470 if (chain->removeEffect_l(effect, release) == 0) { 1471 removeEffectChain_l(chain); 1472 } 1473 } else { 1474 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1475 } 1476 } 1477 1478 void AudioFlinger::ThreadBase::lockEffectChains_l( 1479 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1480 { 1481 effectChains = mEffectChains; 1482 for (size_t i = 0; i < mEffectChains.size(); i++) { 1483 mEffectChains[i]->lock(); 1484 } 1485 } 1486 1487 void AudioFlinger::ThreadBase::unlockEffectChains( 1488 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1489 { 1490 for (size_t i = 0; i < effectChains.size(); i++) { 1491 effectChains[i]->unlock(); 1492 } 1493 } 1494 1495 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1496 { 1497 Mutex::Autolock _l(mLock); 1498 return getEffectChain_l(sessionId); 1499 } 1500 1501 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1502 const 1503 { 1504 size_t size = mEffectChains.size(); 1505 for (size_t i = 0; i < size; i++) { 1506 if (mEffectChains[i]->sessionId() == sessionId) { 1507 return mEffectChains[i]; 1508 } 1509 } 1510 return 0; 1511 } 1512 1513 void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1514 { 1515 Mutex::Autolock _l(mLock); 1516 size_t size = mEffectChains.size(); 1517 for (size_t i = 0; i < size; i++) { 1518 mEffectChains[i]->setMode_l(mode); 1519 } 1520 } 1521 1522 void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1523 { 1524 config->type = AUDIO_PORT_TYPE_MIX; 1525 config->ext.mix.handle = mId; 1526 config->sample_rate = mSampleRate; 1527 config->format = mFormat; 1528 config->channel_mask = mChannelMask; 1529 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1530 AUDIO_PORT_CONFIG_FORMAT; 1531 } 1532 1533 void AudioFlinger::ThreadBase::systemReady() 1534 { 1535 Mutex::Autolock _l(mLock); 1536 if (mSystemReady) { 1537 return; 1538 } 1539 mSystemReady = true; 1540 1541 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1542 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1543 } 1544 mPendingConfigEvents.clear(); 1545 } 1546 1547 template <typename T> 1548 ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::add(const sp<T> &track) { 1549 ssize_t index = mActiveTracks.indexOf(track); 1550 if (index >= 0) { 1551 ALOGW("ActiveTracks<T>::add track %p already there", track.get()); 1552 return index; 1553 } 1554 logTrack("add", track); 1555 mActiveTracksGeneration++; 1556 mLatestActiveTrack = track; 1557 ++mBatteryCounter[track->uid()].second; 1558 mHasChanged = true; 1559 return mActiveTracks.add(track); 1560 } 1561 1562 template <typename T> 1563 ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::remove(const sp<T> &track) { 1564 ssize_t index = mActiveTracks.remove(track); 1565 if (index < 0) { 1566 ALOGW("ActiveTracks<T>::remove nonexistent track %p", track.get()); 1567 return index; 1568 } 1569 logTrack("remove", track); 1570 mActiveTracksGeneration++; 1571 --mBatteryCounter[track->uid()].second; 1572 // mLatestActiveTrack is not cleared even if is the same as track. 1573 mHasChanged = true; 1574 return index; 1575 } 1576 1577 template <typename T> 1578 void AudioFlinger::ThreadBase::ActiveTracks<T>::clear() { 1579 for (const sp<T> &track : mActiveTracks) { 1580 BatteryNotifier::getInstance().noteStopAudio(track->uid()); 1581 logTrack("clear", track); 1582 } 1583 mLastActiveTracksGeneration = mActiveTracksGeneration; 1584 if (!mActiveTracks.empty()) { mHasChanged = true; } 1585 mActiveTracks.clear(); 1586 mLatestActiveTrack.clear(); 1587 mBatteryCounter.clear(); 1588 } 1589 1590 template <typename T> 1591 void AudioFlinger::ThreadBase::ActiveTracks<T>::updatePowerState( 1592 sp<ThreadBase> thread, bool force) { 1593 // Updates ActiveTracks client uids to the thread wakelock. 1594 if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) { 1595 thread->updateWakeLockUids_l(getWakeLockUids()); 1596 mLastActiveTracksGeneration = mActiveTracksGeneration; 1597 } 1598 1599 // Updates BatteryNotifier uids 1600 for (auto it = mBatteryCounter.begin(); it != mBatteryCounter.end();) { 1601 const uid_t uid = it->first; 1602 ssize_t &previous = it->second.first; 1603 ssize_t ¤t = it->second.second; 1604 if (current > 0) { 1605 if (previous == 0) { 1606 BatteryNotifier::getInstance().noteStartAudio(uid); 1607 } 1608 previous = current; 1609 ++it; 1610 } else if (current == 0) { 1611 if (previous > 0) { 1612 BatteryNotifier::getInstance().noteStopAudio(uid); 1613 } 1614 it = mBatteryCounter.erase(it); // std::map<> is stable on iterator erase. 1615 } else /* (current < 0) */ { 1616 LOG_ALWAYS_FATAL("negative battery count %zd", current); 1617 } 1618 } 1619 } 1620 1621 template <typename T> 1622 bool AudioFlinger::ThreadBase::ActiveTracks<T>::readAndClearHasChanged() { 1623 const bool hasChanged = mHasChanged; 1624 mHasChanged = false; 1625 return hasChanged; 1626 } 1627 1628 template <typename T> 1629 void AudioFlinger::ThreadBase::ActiveTracks<T>::logTrack( 1630 const char *funcName, const sp<T> &track) const { 1631 if (mLocalLog != nullptr) { 1632 String8 result; 1633 track->appendDump(result, false /* active */); 1634 mLocalLog->log("AT::%-10s(%p) %s", funcName, track.get(), result.string()); 1635 } 1636 } 1637 1638 void AudioFlinger::ThreadBase::broadcast_l() 1639 { 1640 // Thread could be blocked waiting for async 1641 // so signal it to handle state changes immediately 1642 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1643 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1644 mSignalPending = true; 1645 mWaitWorkCV.broadcast(); 1646 } 1647 1648 // ---------------------------------------------------------------------------- 1649 // Playback 1650 // ---------------------------------------------------------------------------- 1651 1652 AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1653 AudioStreamOut* output, 1654 audio_io_handle_t id, 1655 audio_devices_t device, 1656 type_t type, 1657 bool systemReady) 1658 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1659 mNormalFrameCount(0), mSinkBuffer(NULL), 1660 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1661 mMixerBuffer(NULL), 1662 mMixerBufferSize(0), 1663 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1664 mMixerBufferValid(false), 1665 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1666 mEffectBuffer(NULL), 1667 mEffectBufferSize(0), 1668 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1669 mEffectBufferValid(false), 1670 mSuspended(0), mBytesWritten(0), 1671 mFramesWritten(0), 1672 mSuspendedFrames(0), 1673 mActiveTracks(&this->mLocalLog), 1674 // mStreamTypes[] initialized in constructor body 1675 mTracks(type == MIXER), 1676 mOutput(output), 1677 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1678 mMixerStatus(MIXER_IDLE), 1679 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1680 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1681 mBytesRemaining(0), 1682 mCurrentWriteLength(0), 1683 mUseAsyncWrite(false), 1684 mWriteAckSequence(0), 1685 mDrainSequence(0), 1686 mScreenState(AudioFlinger::mScreenState), 1687 // index 0 is reserved for normal mixer's submix 1688 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1689 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1690 mLeftVolFloat(-1.0), mRightVolFloat(-1.0) 1691 { 1692 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1693 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1694 1695 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1696 // it would be safer to explicitly pass initial masterVolume/masterMute as 1697 // parameter. 1698 // 1699 // If the HAL we are using has support for master volume or master mute, 1700 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1701 // and the mute set to false). 1702 mMasterVolume = audioFlinger->masterVolume_l(); 1703 mMasterMute = audioFlinger->masterMute_l(); 1704 if (mOutput && mOutput->audioHwDev) { 1705 if (mOutput->audioHwDev->canSetMasterVolume()) { 1706 mMasterVolume = 1.0; 1707 } 1708 1709 if (mOutput->audioHwDev->canSetMasterMute()) { 1710 mMasterMute = false; 1711 } 1712 } 1713 1714 readOutputParameters_l(); 1715 1716 // ++ operator does not compile 1717 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_FOR_POLICY_CNT; 1718 stream = (audio_stream_type_t) (stream + 1)) { 1719 mStreamTypes[stream].volume = 0.0f; 1720 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1721 } 1722 // Audio patch volume is always max 1723 mStreamTypes[AUDIO_STREAM_PATCH].volume = 1.0f; 1724 mStreamTypes[AUDIO_STREAM_PATCH].mute = false; 1725 } 1726 1727 AudioFlinger::PlaybackThread::~PlaybackThread() 1728 { 1729 mAudioFlinger->unregisterWriter(mNBLogWriter); 1730 free(mSinkBuffer); 1731 free(mMixerBuffer); 1732 free(mEffectBuffer); 1733 } 1734 1735 void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1736 { 1737 dumpInternals(fd, args); 1738 dumpTracks(fd, args); 1739 dumpEffectChains(fd, args); 1740 dprintf(fd, " Local log:\n"); 1741 mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */); 1742 } 1743 1744 void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1745 { 1746 String8 result; 1747 1748 result.appendFormat(" Stream volumes in dB: "); 1749 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1750 const stream_type_t *st = &mStreamTypes[i]; 1751 if (i > 0) { 1752 result.appendFormat(", "); 1753 } 1754 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1755 if (st->mute) { 1756 result.append("M"); 1757 } 1758 } 1759 result.append("\n"); 1760 write(fd, result.string(), result.length()); 1761 result.clear(); 1762 1763 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1764 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1765 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1766 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1767 1768 size_t numtracks = mTracks.size(); 1769 size_t numactive = mActiveTracks.size(); 1770 dprintf(fd, " %zu Tracks", numtracks); 1771 size_t numactiveseen = 0; 1772 const char *prefix = " "; 1773 if (numtracks) { 1774 dprintf(fd, " of which %zu are active\n", numactive); 1775 result.append(prefix); 1776 Track::appendDumpHeader(result); 1777 for (size_t i = 0; i < numtracks; ++i) { 1778 sp<Track> track = mTracks[i]; 1779 if (track != 0) { 1780 bool active = mActiveTracks.indexOf(track) >= 0; 1781 if (active) { 1782 numactiveseen++; 1783 } 1784 result.append(prefix); 1785 track->appendDump(result, active); 1786 } 1787 } 1788 } else { 1789 result.append("\n"); 1790 } 1791 if (numactiveseen != numactive) { 1792 // some tracks in the active list were not in the tracks list 1793 result.append(" The following tracks are in the active list but" 1794 " not in the track list\n"); 1795 result.append(prefix); 1796 Track::appendDumpHeader(result); 1797 for (size_t i = 0; i < numactive; ++i) { 1798 sp<Track> track = mActiveTracks[i]; 1799 if (mTracks.indexOf(track) < 0) { 1800 result.append(prefix); 1801 track->appendDump(result, true /* active */); 1802 } 1803 } 1804 } 1805 1806 write(fd, result.string(), result.size()); 1807 } 1808 1809 void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1810 { 1811 dumpBase(fd, args); 1812 1813 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1814 dprintf(fd, " Last write occurred (msecs): %llu\n", 1815 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1816 dprintf(fd, " Total writes: %d\n", mNumWrites); 1817 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1818 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1819 dprintf(fd, " Suspend count: %d\n", mSuspended); 1820 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1821 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1822 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1823 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1824 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1825 AudioStreamOut *output = mOutput; 1826 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1827 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", 1828 output, flags, outputFlagsToString(flags).c_str()); 1829 dprintf(fd, " Frames written: %lld\n", (long long)mFramesWritten); 1830 dprintf(fd, " Suspended frames: %lld\n", (long long)mSuspendedFrames); 1831 if (mPipeSink.get() != nullptr) { 1832 dprintf(fd, " PipeSink frames written: %lld\n", (long long)mPipeSink->framesWritten()); 1833 } 1834 if (output != nullptr) { 1835 dprintf(fd, " Hal stream dump:\n"); 1836 (void)output->stream->dump(fd); 1837 } 1838 } 1839 1840 // Thread virtuals 1841 1842 void AudioFlinger::PlaybackThread::onFirstRef() 1843 { 1844 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1845 } 1846 1847 // ThreadBase virtuals 1848 void AudioFlinger::PlaybackThread::preExit() 1849 { 1850 ALOGV(" preExit()"); 1851 // FIXME this is using hard-coded strings but in the future, this functionality will be 1852 // converted to use audio HAL extensions required to support tunneling 1853 status_t result = mOutput->stream->setParameters(String8("exiting=1")); 1854 ALOGE_IF(result != OK, "Error when setting parameters on exit: %d", result); 1855 } 1856 1857 // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1858 sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1859 const sp<AudioFlinger::Client>& client, 1860 audio_stream_type_t streamType, 1861 const audio_attributes_t& attr, 1862 uint32_t *pSampleRate, 1863 audio_format_t format, 1864 audio_channel_mask_t channelMask, 1865 size_t *pFrameCount, 1866 size_t *pNotificationFrameCount, 1867 uint32_t notificationsPerBuffer, 1868 float speed, 1869 const sp<IMemory>& sharedBuffer, 1870 audio_session_t sessionId, 1871 audio_output_flags_t *flags, 1872 pid_t tid, 1873 uid_t uid, 1874 status_t *status, 1875 audio_port_handle_t portId) 1876 { 1877 size_t frameCount = *pFrameCount; 1878 size_t notificationFrameCount = *pNotificationFrameCount; 1879 sp<Track> track; 1880 status_t lStatus; 1881 audio_output_flags_t outputFlags = mOutput->flags; 1882 audio_output_flags_t requestedFlags = *flags; 1883 1884 if (*pSampleRate == 0) { 1885 *pSampleRate = mSampleRate; 1886 } 1887 uint32_t sampleRate = *pSampleRate; 1888 1889 // special case for FAST flag considered OK if fast mixer is present 1890 if (hasFastMixer()) { 1891 outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST); 1892 } 1893 1894 // Check if requested flags are compatible with output stream flags 1895 if ((*flags & outputFlags) != *flags) { 1896 ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)", 1897 *flags, outputFlags); 1898 *flags = (audio_output_flags_t)(*flags & outputFlags); 1899 } 1900 1901 // client expresses a preference for FAST, but we get the final say 1902 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 1903 if ( 1904 // PCM data 1905 audio_is_linear_pcm(format) && 1906 // TODO: extract as a data library function that checks that a computationally 1907 // expensive downmixer is not required: isFastOutputChannelConversion() 1908 (channelMask == mChannelMask || 1909 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1910 (channelMask == AUDIO_CHANNEL_OUT_MONO 1911 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1912 // hardware sample rate 1913 (sampleRate == mSampleRate) && 1914 // normal mixer has an associated fast mixer 1915 hasFastMixer() && 1916 // there are sufficient fast track slots available 1917 (mFastTrackAvailMask != 0) 1918 // FIXME test that MixerThread for this fast track has a capable output HAL 1919 // FIXME add a permission test also? 1920 ) { 1921 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1922 if (sharedBuffer == 0) { 1923 // read the fast track multiplier property the first time it is needed 1924 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1925 if (ok != 0) { 1926 ALOGE("%s pthread_once failed: %d", __func__, ok); 1927 } 1928 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1929 } 1930 1931 // check compatibility with audio effects. 1932 { // scope for mLock 1933 Mutex::Autolock _l(mLock); 1934 for (audio_session_t session : { 1935 AUDIO_SESSION_OUTPUT_STAGE, 1936 AUDIO_SESSION_OUTPUT_MIX, 1937 sessionId, 1938 }) { 1939 sp<EffectChain> chain = getEffectChain_l(session); 1940 if (chain.get() != nullptr) { 1941 audio_output_flags_t old = *flags; 1942 chain->checkOutputFlagCompatibility(flags); 1943 if (old != *flags) { 1944 ALOGV("AUDIO_OUTPUT_FLAGS denied by effect, session=%d old=%#x new=%#x", 1945 (int)session, (int)old, (int)*flags); 1946 } 1947 } 1948 } 1949 } 1950 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0, 1951 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1952 frameCount, mFrameCount); 1953 } else { 1954 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1955 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1956 "sampleRate=%u mSampleRate=%u " 1957 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1958 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1959 audio_is_linear_pcm(format), 1960 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1961 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1962 } 1963 } 1964 1965 if (!audio_has_proportional_frames(format)) { 1966 if (sharedBuffer != 0) { 1967 // Same comment as below about ignoring frameCount parameter for set() 1968 frameCount = sharedBuffer->size(); 1969 } else if (frameCount == 0) { 1970 frameCount = mNormalFrameCount; 1971 } 1972 if (notificationFrameCount != frameCount) { 1973 notificationFrameCount = frameCount; 1974 } 1975 } else if (sharedBuffer != 0) { 1976 // FIXME: Ensure client side memory buffers need 1977 // not have additional alignment beyond sample 1978 // (e.g. 16 bit stereo accessed as 32 bit frame). 1979 size_t alignment = audio_bytes_per_sample(format); 1980 if (alignment & 1) { 1981 // for AUDIO_FORMAT_PCM_24_BIT_PACKED (not exposed through Java). 1982 alignment = 1; 1983 } 1984 uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 1985 size_t frameSize = channelCount * audio_bytes_per_sample(format); 1986 if (channelCount > 1) { 1987 // More than 2 channels does not require stronger alignment than stereo 1988 alignment <<= 1; 1989 } 1990 if (((uintptr_t)sharedBuffer->pointer() & (alignment - 1)) != 0) { 1991 ALOGE("Invalid buffer alignment: address %p, channel count %u", 1992 sharedBuffer->pointer(), channelCount); 1993 lStatus = BAD_VALUE; 1994 goto Exit; 1995 } 1996 1997 // When initializing a shared buffer AudioTrack via constructors, 1998 // there's no frameCount parameter. 1999 // But when initializing a shared buffer AudioTrack via set(), 2000 // there _is_ a frameCount parameter. We silently ignore it. 2001 frameCount = sharedBuffer->size() / frameSize; 2002 } else { 2003 size_t minFrameCount = 0; 2004 // For fast tracks we try to respect the application's request for notifications per buffer. 2005 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 2006 if (notificationsPerBuffer > 0) { 2007 // Avoid possible arithmetic overflow during multiplication. 2008 if (notificationsPerBuffer > SIZE_MAX / mFrameCount) { 2009 ALOGE("Requested notificationPerBuffer=%u ignored for HAL frameCount=%zu", 2010 notificationsPerBuffer, mFrameCount); 2011 } else { 2012 minFrameCount = mFrameCount * notificationsPerBuffer; 2013 } 2014 } 2015 } else { 2016 // For normal PCM streaming tracks, update minimum frame count. 2017 // Buffer depth is forced to be at least 2 x the normal mixer frame count and 2018 // cover audio hardware latency. 2019 // This is probably too conservative, but legacy application code may depend on it. 2020 // If you change this calculation, also review the start threshold which is related. 2021 uint32_t latencyMs = latency_l(); 2022 if (latencyMs == 0) { 2023 ALOGE("Error when retrieving output stream latency"); 2024 lStatus = UNKNOWN_ERROR; 2025 goto Exit; 2026 } 2027 2028 minFrameCount = AudioSystem::calculateMinFrameCount(latencyMs, mNormalFrameCount, 2029 mSampleRate, sampleRate, speed /*, 0 mNotificationsPerBufferReq*/); 2030 2031 } 2032 if (frameCount < minFrameCount) { 2033 frameCount = minFrameCount; 2034 } 2035 } 2036 2037 // Make sure that application is notified with sufficient margin before underrun. 2038 // The client can divide the AudioTrack buffer into sub-buffers, 2039 // and expresses its desire to server as the notification frame count. 2040 if (sharedBuffer == 0 && audio_is_linear_pcm(format)) { 2041 size_t maxNotificationFrames; 2042 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 2043 // notify every HAL buffer, regardless of the size of the track buffer 2044 maxNotificationFrames = mFrameCount; 2045 } else { 2046 // For normal tracks, use at least double-buffering if no sample rate conversion, 2047 // or at least triple-buffering if there is sample rate conversion 2048 const int nBuffering = sampleRate == mSampleRate ? 2 : 3; 2049 maxNotificationFrames = frameCount / nBuffering; 2050 // If client requested a fast track but this was denied, then use the smaller maximum. 2051 if (requestedFlags & AUDIO_OUTPUT_FLAG_FAST) { 2052 size_t maxNotificationFramesFastDenied = FMS_20 * sampleRate / 1000; 2053 if (maxNotificationFrames > maxNotificationFramesFastDenied) { 2054 maxNotificationFrames = maxNotificationFramesFastDenied; 2055 } 2056 } 2057 } 2058 if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) { 2059 if (notificationFrameCount == 0) { 2060 ALOGD("Client defaulted notificationFrames to %zu for frameCount %zu", 2061 maxNotificationFrames, frameCount); 2062 } else { 2063 ALOGW("Client adjusted notificationFrames from %zu to %zu for frameCount %zu", 2064 notificationFrameCount, maxNotificationFrames, frameCount); 2065 } 2066 notificationFrameCount = maxNotificationFrames; 2067 } 2068 } 2069 2070 *pFrameCount = frameCount; 2071 *pNotificationFrameCount = notificationFrameCount; 2072 2073 switch (mType) { 2074 2075 case DIRECT: 2076 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 2077 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 2078 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 2079 "for output %p with format %#x", 2080 sampleRate, format, channelMask, mOutput, mFormat); 2081 lStatus = BAD_VALUE; 2082 goto Exit; 2083 } 2084 } 2085 break; 2086 2087 case OFFLOAD: 2088 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 2089 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 2090 "for output %p with format %#x", 2091 sampleRate, format, channelMask, mOutput, mFormat); 2092 lStatus = BAD_VALUE; 2093 goto Exit; 2094 } 2095 break; 2096 2097 default: 2098 if (!audio_is_linear_pcm(format)) { 2099 ALOGE("createTrack_l() Bad parameter: format %#x \"" 2100 "for output %p with format %#x", 2101 format, mOutput, mFormat); 2102 lStatus = BAD_VALUE; 2103 goto Exit; 2104 } 2105 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 2106 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 2107 lStatus = BAD_VALUE; 2108 goto Exit; 2109 } 2110 break; 2111 2112 } 2113 2114 lStatus = initCheck(); 2115 if (lStatus != NO_ERROR) { 2116 ALOGE("createTrack_l() audio driver not initialized"); 2117 goto Exit; 2118 } 2119 2120 { // scope for mLock 2121 Mutex::Autolock _l(mLock); 2122 2123 // all tracks in same audio session must share the same routing strategy otherwise 2124 // conflicts will happen when tracks are moved from one output to another by audio policy 2125 // manager 2126 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 2127 for (size_t i = 0; i < mTracks.size(); ++i) { 2128 sp<Track> t = mTracks[i]; 2129 if (t != 0 && t->isExternalTrack()) { 2130 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 2131 if (sessionId == t->sessionId() && strategy != actual) { 2132 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 2133 strategy, actual); 2134 lStatus = BAD_VALUE; 2135 goto Exit; 2136 } 2137 } 2138 } 2139 2140 track = new Track(this, client, streamType, attr, sampleRate, format, 2141 channelMask, frameCount, 2142 nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer, 2143 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT, portId); 2144 2145 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 2146 if (lStatus != NO_ERROR) { 2147 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 2148 // track must be cleared from the caller as the caller has the AF lock 2149 goto Exit; 2150 } 2151 mTracks.add(track); 2152 2153 sp<EffectChain> chain = getEffectChain_l(sessionId); 2154 if (chain != 0) { 2155 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 2156 track->setMainBuffer(chain->inBuffer()); 2157 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 2158 chain->incTrackCnt(); 2159 } 2160 2161 if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) { 2162 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2163 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 2164 // so ask activity manager to do this on our behalf 2165 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/); 2166 } 2167 } 2168 2169 lStatus = NO_ERROR; 2170 2171 Exit: 2172 *status = lStatus; 2173 return track; 2174 } 2175 2176 template<typename T> 2177 ssize_t AudioFlinger::PlaybackThread::Tracks<T>::add(const sp<T> &track) 2178 { 2179 const ssize_t index = mTracks.add(track); 2180 if (index >= 0) { 2181 // set name for track when adding. 2182 int name; 2183 if (mUnusedTrackNames.empty()) { 2184 name = mTracks.size() - 1; // new name {0 ... size-1}. 2185 } else { 2186 // reuse smallest name for deleted track. 2187 auto it = mUnusedTrackNames.begin(); 2188 name = *it; 2189 (void)mUnusedTrackNames.erase(it); 2190 } 2191 track->setName(name); 2192 } else { 2193 LOG_ALWAYS_FATAL("cannot add track"); 2194 } 2195 return index; 2196 } 2197 2198 template<typename T> 2199 ssize_t AudioFlinger::PlaybackThread::Tracks<T>::remove(const sp<T> &track) 2200 { 2201 const int name = track->name(); 2202 const ssize_t index = mTracks.remove(track); 2203 if (index >= 0) { 2204 // invalidate name when removing from mTracks. 2205 LOG_ALWAYS_FATAL_IF(name < 0, "invalid name %d for track on mTracks", name); 2206 2207 if (mSaveDeletedTrackNames) { 2208 // We can't directly access mAudioMixer since the caller may be outside of threadLoop. 2209 // Instead, we add to mDeletedTrackNames which is solely used for mAudioMixer update, 2210 // to be handled when MixerThread::prepareTracks_l() next changes mAudioMixer. 2211 mDeletedTrackNames.emplace(name); 2212 } 2213 2214 mUnusedTrackNames.emplace(name); 2215 track->setName(T::TRACK_NAME_PENDING); 2216 } else { 2217 LOG_ALWAYS_FATAL_IF(name >= 0, 2218 "valid name %d for track not in mTracks (returned %zd)", name, index); 2219 } 2220 return index; 2221 } 2222 2223 uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 2224 { 2225 return latency; 2226 } 2227 2228 uint32_t AudioFlinger::PlaybackThread::latency() const 2229 { 2230 Mutex::Autolock _l(mLock); 2231 return latency_l(); 2232 } 2233 uint32_t AudioFlinger::PlaybackThread::latency_l() const 2234 { 2235 uint32_t latency; 2236 if (initCheck() == NO_ERROR && mOutput->stream->getLatency(&latency) == OK) { 2237 return correctLatency_l(latency); 2238 } 2239 return 0; 2240 } 2241 2242 void AudioFlinger::PlaybackThread::setMasterVolume(float value) 2243 { 2244 Mutex::Autolock _l(mLock); 2245 // Don't apply master volume in SW if our HAL can do it for us. 2246 if (mOutput && mOutput->audioHwDev && 2247 mOutput->audioHwDev->canSetMasterVolume()) { 2248 mMasterVolume = 1.0; 2249 } else { 2250 mMasterVolume = value; 2251 } 2252 } 2253 2254 void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 2255 { 2256 if (isDuplicating()) { 2257 return; 2258 } 2259 Mutex::Autolock _l(mLock); 2260 // Don't apply master mute in SW if our HAL can do it for us. 2261 if (mOutput && mOutput->audioHwDev && 2262 mOutput->audioHwDev->canSetMasterMute()) { 2263 mMasterMute = false; 2264 } else { 2265 mMasterMute = muted; 2266 } 2267 } 2268 2269 void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2270 { 2271 Mutex::Autolock _l(mLock); 2272 mStreamTypes[stream].volume = value; 2273 broadcast_l(); 2274 } 2275 2276 void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2277 { 2278 Mutex::Autolock _l(mLock); 2279 mStreamTypes[stream].mute = muted; 2280 broadcast_l(); 2281 } 2282 2283 float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2284 { 2285 Mutex::Autolock _l(mLock); 2286 return mStreamTypes[stream].volume; 2287 } 2288 2289 // addTrack_l() must be called with ThreadBase::mLock held 2290 status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2291 { 2292 status_t status = ALREADY_EXISTS; 2293 2294 if (mActiveTracks.indexOf(track) < 0) { 2295 // the track is newly added, make sure it fills up all its 2296 // buffers before playing. This is to ensure the client will 2297 // effectively get the latency it requested. 2298 if (track->isExternalTrack()) { 2299 TrackBase::track_state state = track->mState; 2300 mLock.unlock(); 2301 status = AudioSystem::startOutput(mId, track->streamType(), 2302 track->sessionId()); 2303 mLock.lock(); 2304 // abort track was stopped/paused while we released the lock 2305 if (state != track->mState) { 2306 if (status == NO_ERROR) { 2307 mLock.unlock(); 2308 AudioSystem::stopOutput(mId, track->streamType(), 2309 track->sessionId()); 2310 mLock.lock(); 2311 } 2312 return INVALID_OPERATION; 2313 } 2314 // abort if start is rejected by audio policy manager 2315 if (status != NO_ERROR) { 2316 return PERMISSION_DENIED; 2317 } 2318 #ifdef ADD_BATTERY_DATA 2319 // to track the speaker usage 2320 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2321 #endif 2322 } 2323 2324 // set retry count for buffer fill 2325 if (track->isOffloaded()) { 2326 if (track->isStopping_1()) { 2327 track->mRetryCount = kMaxTrackStopRetriesOffload; 2328 } else { 2329 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2330 } 2331 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED; 2332 } else { 2333 track->mRetryCount = kMaxTrackStartupRetries; 2334 track->mFillingUpStatus = 2335 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2336 } 2337 2338 track->mResetDone = false; 2339 track->mPresentationCompleteFrames = 0; 2340 mActiveTracks.add(track); 2341 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2342 if (chain != 0) { 2343 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2344 track->sessionId()); 2345 chain->incActiveTrackCnt(); 2346 } 2347 2348 status = NO_ERROR; 2349 } 2350 2351 onAddNewTrack_l(); 2352 return status; 2353 } 2354 2355 bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2356 { 2357 track->terminate(); 2358 // active tracks are removed by threadLoop() 2359 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2360 track->mState = TrackBase::STOPPED; 2361 if (!trackActive) { 2362 removeTrack_l(track); 2363 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2364 track->mState = TrackBase::STOPPING_1; 2365 } 2366 2367 return trackActive; 2368 } 2369 2370 void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2371 { 2372 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2373 2374 String8 result; 2375 track->appendDump(result, false /* active */); 2376 mLocalLog.log("removeTrack_l (%p) %s", track.get(), result.string()); 2377 2378 mTracks.remove(track); 2379 if (track->isFastTrack()) { 2380 int index = track->mFastIndex; 2381 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2382 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2383 mFastTrackAvailMask |= 1 << index; 2384 // redundant as track is about to be destroyed, for dumpsys only 2385 track->mFastIndex = -1; 2386 } 2387 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2388 if (chain != 0) { 2389 chain->decTrackCnt(); 2390 } 2391 } 2392 2393 String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2394 { 2395 Mutex::Autolock _l(mLock); 2396 String8 out_s8; 2397 if (initCheck() == NO_ERROR && mOutput->stream->getParameters(keys, &out_s8) == OK) { 2398 return out_s8; 2399 } 2400 return String8(); 2401 } 2402 2403 void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2404 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2405 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2406 2407 desc->mIoHandle = mId; 2408 2409 switch (event) { 2410 case AUDIO_OUTPUT_OPENED: 2411 case AUDIO_OUTPUT_REGISTERED: 2412 case AUDIO_OUTPUT_CONFIG_CHANGED: 2413 desc->mPatch = mPatch; 2414 desc->mChannelMask = mChannelMask; 2415 desc->mSamplingRate = mSampleRate; 2416 desc->mFormat = mFormat; 2417 desc->mFrameCount = mNormalFrameCount; // FIXME see 2418 // AudioFlinger::frameCount(audio_io_handle_t) 2419 desc->mFrameCountHAL = mFrameCount; 2420 desc->mLatency = latency_l(); 2421 break; 2422 2423 case AUDIO_OUTPUT_CLOSED: 2424 default: 2425 break; 2426 } 2427 mAudioFlinger->ioConfigChanged(event, desc, pid); 2428 } 2429 2430 void AudioFlinger::PlaybackThread::onWriteReady() 2431 { 2432 mCallbackThread->resetWriteBlocked(); 2433 } 2434 2435 void AudioFlinger::PlaybackThread::onDrainReady() 2436 { 2437 mCallbackThread->resetDraining(); 2438 } 2439 2440 void AudioFlinger::PlaybackThread::onError() 2441 { 2442 mCallbackThread->setAsyncError(); 2443 } 2444 2445 void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2446 { 2447 Mutex::Autolock _l(mLock); 2448 // reject out of sequence requests 2449 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2450 mWriteAckSequence &= ~1; 2451 mWaitWorkCV.signal(); 2452 } 2453 } 2454 2455 void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2456 { 2457 Mutex::Autolock _l(mLock); 2458 // reject out of sequence requests 2459 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2460 mDrainSequence &= ~1; 2461 mWaitWorkCV.signal(); 2462 } 2463 } 2464 2465 void AudioFlinger::PlaybackThread::readOutputParameters_l() 2466 { 2467 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2468 mSampleRate = mOutput->getSampleRate(); 2469 mChannelMask = mOutput->getChannelMask(); 2470 if (!audio_is_output_channel(mChannelMask)) { 2471 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2472 } 2473 if ((mType == MIXER || mType == DUPLICATING) 2474 && !isValidPcmSinkChannelMask(mChannelMask)) { 2475 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2476 mChannelMask); 2477 } 2478 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2479 2480 // Get actual HAL format. 2481 status_t result = mOutput->stream->getFormat(&mHALFormat); 2482 LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result); 2483 // Get format from the shim, which will be different than the HAL format 2484 // if playing compressed audio over HDMI passthrough. 2485 mFormat = mOutput->getFormat(); 2486 if (!audio_is_valid_format(mFormat)) { 2487 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2488 } 2489 if ((mType == MIXER || mType == DUPLICATING) 2490 && !isValidPcmSinkFormat(mFormat)) { 2491 LOG_FATAL("HAL format %#x not supported for mixed output", 2492 mFormat); 2493 } 2494 mFrameSize = mOutput->getFrameSize(); 2495 result = mOutput->stream->getBufferSize(&mBufferSize); 2496 LOG_ALWAYS_FATAL_IF(result != OK, 2497 "Error when retrieving output stream buffer size: %d", result); 2498 mFrameCount = mBufferSize / mFrameSize; 2499 if (mFrameCount & 15) { 2500 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2501 mFrameCount); 2502 } 2503 2504 if (mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) { 2505 if (mOutput->stream->setCallback(this) == OK) { 2506 mUseAsyncWrite = true; 2507 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2508 } 2509 } 2510 2511 mHwSupportsPause = false; 2512 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2513 bool supportsPause = false, supportsResume = false; 2514 if (mOutput->stream->supportsPauseAndResume(&supportsPause, &supportsResume) == OK) { 2515 if (supportsPause && supportsResume) { 2516 mHwSupportsPause = true; 2517 } else if (supportsPause) { 2518 ALOGW("direct output implements pause but not resume"); 2519 } else if (supportsResume) { 2520 ALOGW("direct output implements resume but not pause"); 2521 } 2522 } 2523 } 2524 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2525 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2526 } 2527 2528 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2529 // For best precision, we use float instead of the associated output 2530 // device format (typically PCM 16 bit). 2531 2532 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2533 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2534 mBufferSize = mFrameSize * mFrameCount; 2535 2536 // TODO: We currently use the associated output device channel mask and sample rate. 2537 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2538 // (if a valid mask) to avoid premature downmix. 2539 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2540 // instead of the output device sample rate to avoid loss of high frequency information. 2541 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2542 } 2543 2544 // Calculate size of normal sink buffer relative to the HAL output buffer size 2545 double multiplier = 1.0; 2546 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2547 kUseFastMixer == FastMixer_Dynamic)) { 2548 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2549 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2550 2551 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2552 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2553 maxNormalFrameCount = maxNormalFrameCount & ~15; 2554 if (maxNormalFrameCount < minNormalFrameCount) { 2555 maxNormalFrameCount = minNormalFrameCount; 2556 } 2557 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2558 if (multiplier <= 1.0) { 2559 multiplier = 1.0; 2560 } else if (multiplier <= 2.0) { 2561 if (2 * mFrameCount <= maxNormalFrameCount) { 2562 multiplier = 2.0; 2563 } else { 2564 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2565 } 2566 } else { 2567 multiplier = floor(multiplier); 2568 } 2569 } 2570 mNormalFrameCount = multiplier * mFrameCount; 2571 // round up to nearest 16 frames to satisfy AudioMixer 2572 if (mType == MIXER || mType == DUPLICATING) { 2573 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2574 } 2575 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2576 mNormalFrameCount); 2577 2578 // Check if we want to throttle the processing to no more than 2x normal rate 2579 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2580 mThreadThrottleTimeMs = 0; 2581 mThreadThrottleEndMs = 0; 2582 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2583 2584 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2585 // Originally this was int16_t[] array, need to remove legacy implications. 2586 free(mSinkBuffer); 2587 mSinkBuffer = NULL; 2588 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2589 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2590 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2591 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2592 2593 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2594 // drives the output. 2595 free(mMixerBuffer); 2596 mMixerBuffer = NULL; 2597 if (mMixerBufferEnabled) { 2598 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2599 mMixerBufferSize = mNormalFrameCount * mChannelCount 2600 * audio_bytes_per_sample(mMixerBufferFormat); 2601 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2602 } 2603 free(mEffectBuffer); 2604 mEffectBuffer = NULL; 2605 if (mEffectBufferEnabled) { 2606 mEffectBufferFormat = EFFECT_BUFFER_FORMAT; 2607 mEffectBufferSize = mNormalFrameCount * mChannelCount 2608 * audio_bytes_per_sample(mEffectBufferFormat); 2609 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2610 } 2611 2612 // force reconfiguration of effect chains and engines to take new buffer size and audio 2613 // parameters into account 2614 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2615 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2616 // matter. 2617 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2618 Vector< sp<EffectChain> > effectChains = mEffectChains; 2619 for (size_t i = 0; i < effectChains.size(); i ++) { 2620 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2621 } 2622 } 2623 2624 void AudioFlinger::PlaybackThread::updateMetadata_l() 2625 { 2626 if (mOutput == nullptr || mOutput->stream == nullptr ) { 2627 return; // That should not happen 2628 } 2629 bool hasChanged = mActiveTracks.readAndClearHasChanged(); 2630 for (const sp<Track> &track : mActiveTracks) { 2631 // Do not short-circuit as all hasChanged states must be reset 2632 // as all the metadata are going to be sent 2633 hasChanged |= track->readAndClearHasChanged(); 2634 } 2635 if (!hasChanged) { 2636 return; // nothing to do 2637 } 2638 StreamOutHalInterface::SourceMetadata metadata; 2639 auto backInserter = std::back_inserter(metadata.tracks); 2640 for (const sp<Track> &track : mActiveTracks) { 2641 // No track is invalid as this is called after prepareTrack_l in the same critical section 2642 track->copyMetadataTo(backInserter); 2643 } 2644 sendMetadataToBackend_l(metadata); 2645 } 2646 2647 void AudioFlinger::PlaybackThread::sendMetadataToBackend_l( 2648 const StreamOutHalInterface::SourceMetadata& metadata) 2649 { 2650 mOutput->stream->updateSourceMetadata(metadata); 2651 }; 2652 2653 status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2654 { 2655 if (halFrames == NULL || dspFrames == NULL) { 2656 return BAD_VALUE; 2657 } 2658 Mutex::Autolock _l(mLock); 2659 if (initCheck() != NO_ERROR) { 2660 return INVALID_OPERATION; 2661 } 2662 int64_t framesWritten = mBytesWritten / mFrameSize; 2663 *halFrames = framesWritten; 2664 2665 if (isSuspended()) { 2666 // return an estimation of rendered frames when the output is suspended 2667 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2668 *dspFrames = (uint32_t) 2669 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2670 return NO_ERROR; 2671 } else { 2672 status_t status; 2673 uint32_t frames; 2674 status = mOutput->getRenderPosition(&frames); 2675 *dspFrames = (size_t)frames; 2676 return status; 2677 } 2678 } 2679 2680 // hasAudioSession_l() must be called with ThreadBase::mLock held 2681 uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const 2682 { 2683 uint32_t result = 0; 2684 if (getEffectChain_l(sessionId) != 0) { 2685 result = EFFECT_SESSION; 2686 } 2687 2688 for (size_t i = 0; i < mTracks.size(); ++i) { 2689 sp<Track> track = mTracks[i]; 2690 if (sessionId == track->sessionId() && !track->isInvalid()) { 2691 result |= TRACK_SESSION; 2692 if (track->isFastTrack()) { 2693 result |= FAST_SESSION; 2694 } 2695 break; 2696 } 2697 } 2698 2699 return result; 2700 } 2701 2702 uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2703 { 2704 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2705 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2706 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2707 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2708 } 2709 for (size_t i = 0; i < mTracks.size(); i++) { 2710 sp<Track> track = mTracks[i]; 2711 if (sessionId == track->sessionId() && !track->isInvalid()) { 2712 return AudioSystem::getStrategyForStream(track->streamType()); 2713 } 2714 } 2715 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2716 } 2717 2718 2719 AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2720 { 2721 Mutex::Autolock _l(mLock); 2722 return mOutput; 2723 } 2724 2725 AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2726 { 2727 Mutex::Autolock _l(mLock); 2728 AudioStreamOut *output = mOutput; 2729 mOutput = NULL; 2730 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2731 // must push a NULL and wait for ack 2732 mOutputSink.clear(); 2733 mPipeSink.clear(); 2734 mNormalSink.clear(); 2735 return output; 2736 } 2737 2738 // this method must always be called either with ThreadBase mLock held or inside the thread loop 2739 sp<StreamHalInterface> AudioFlinger::PlaybackThread::stream() const 2740 { 2741 if (mOutput == NULL) { 2742 return NULL; 2743 } 2744 return mOutput->stream; 2745 } 2746 2747 uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2748 { 2749 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2750 } 2751 2752 status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2753 { 2754 if (!isValidSyncEvent(event)) { 2755 return BAD_VALUE; 2756 } 2757 2758 Mutex::Autolock _l(mLock); 2759 2760 for (size_t i = 0; i < mTracks.size(); ++i) { 2761 sp<Track> track = mTracks[i]; 2762 if (event->triggerSession() == track->sessionId()) { 2763 (void) track->setSyncEvent(event); 2764 return NO_ERROR; 2765 } 2766 } 2767 2768 return NAME_NOT_FOUND; 2769 } 2770 2771 bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2772 { 2773 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2774 } 2775 2776 void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2777 const Vector< sp<Track> >& tracksToRemove) 2778 { 2779 size_t count = tracksToRemove.size(); 2780 if (count > 0) { 2781 for (size_t i = 0 ; i < count ; i++) { 2782 const sp<Track>& track = tracksToRemove.itemAt(i); 2783 if (track->isExternalTrack()) { 2784 AudioSystem::stopOutput(mId, track->streamType(), 2785 track->sessionId()); 2786 #ifdef ADD_BATTERY_DATA 2787 // to track the speaker usage 2788 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2789 #endif 2790 if (track->isTerminated()) { 2791 AudioSystem::releaseOutput(mId, track->streamType(), 2792 track->sessionId()); 2793 } 2794 } 2795 } 2796 } 2797 } 2798 2799 void AudioFlinger::PlaybackThread::checkSilentMode_l() 2800 { 2801 if (!mMasterMute) { 2802 char value[PROPERTY_VALUE_MAX]; 2803 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2804 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2805 return; 2806 } 2807 if (property_get("ro.audio.silent", value, "0") > 0) { 2808 char *endptr; 2809 unsigned long ul = strtoul(value, &endptr, 0); 2810 if (*endptr == '\0' && ul != 0) { 2811 ALOGD("Silence is golden"); 2812 // The setprop command will not allow a property to be changed after 2813 // the first time it is set, so we don't have to worry about un-muting. 2814 setMasterMute_l(true); 2815 } 2816 } 2817 } 2818 } 2819 2820 // shared by MIXER and DIRECT, overridden by DUPLICATING 2821 ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2822 { 2823 LOG_HIST_TS(); 2824 mInWrite = true; 2825 ssize_t bytesWritten; 2826 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2827 2828 // If an NBAIO sink is present, use it to write the normal mixer's submix 2829 if (mNormalSink != 0) { 2830 2831 const size_t count = mBytesRemaining / mFrameSize; 2832 2833 ATRACE_BEGIN("write"); 2834 // update the setpoint when AudioFlinger::mScreenState changes 2835 uint32_t screenState = AudioFlinger::mScreenState; 2836 if (screenState != mScreenState) { 2837 mScreenState = screenState; 2838 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2839 if (pipe != NULL) { 2840 pipe->setAvgFrames((mScreenState & 1) ? 2841 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2842 } 2843 } 2844 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2845 ATRACE_END(); 2846 if (framesWritten > 0) { 2847 bytesWritten = framesWritten * mFrameSize; 2848 } else { 2849 bytesWritten = framesWritten; 2850 } 2851 // otherwise use the HAL / AudioStreamOut directly 2852 } else { 2853 // Direct output and offload threads 2854 2855 if (mUseAsyncWrite) { 2856 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2857 mWriteAckSequence += 2; 2858 mWriteAckSequence |= 1; 2859 ALOG_ASSERT(mCallbackThread != 0); 2860 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2861 } 2862 // FIXME We should have an implementation of timestamps for direct output threads. 2863 // They are used e.g for multichannel PCM playback over HDMI. 2864 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2865 2866 if (mUseAsyncWrite && 2867 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2868 // do not wait for async callback in case of error of full write 2869 mWriteAckSequence &= ~1; 2870 ALOG_ASSERT(mCallbackThread != 0); 2871 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2872 } 2873 } 2874 2875 mNumWrites++; 2876 mInWrite = false; 2877 mStandby = false; 2878 return bytesWritten; 2879 } 2880 2881 void AudioFlinger::PlaybackThread::threadLoop_drain() 2882 { 2883 bool supportsDrain = false; 2884 if (mOutput->stream->supportsDrain(&supportsDrain) == OK && supportsDrain) { 2885 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2886 if (mUseAsyncWrite) { 2887 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2888 mDrainSequence |= 1; 2889 ALOG_ASSERT(mCallbackThread != 0); 2890 mCallbackThread->setDraining(mDrainSequence); 2891 } 2892 status_t result = mOutput->stream->drain(mMixerStatus == MIXER_DRAIN_TRACK); 2893 ALOGE_IF(result != OK, "Error when draining stream: %d", result); 2894 } 2895 } 2896 2897 void AudioFlinger::PlaybackThread::threadLoop_exit() 2898 { 2899 { 2900 Mutex::Autolock _l(mLock); 2901 for (size_t i = 0; i < mTracks.size(); i++) { 2902 sp<Track> track = mTracks[i]; 2903 track->invalidate(); 2904 } 2905 // Clear ActiveTracks to update BatteryNotifier in case active tracks remain. 2906 // After we exit there are no more track changes sent to BatteryNotifier 2907 // because that requires an active threadLoop. 2908 // TODO: should we decActiveTrackCnt() of the cleared track effect chain? 2909 mActiveTracks.clear(); 2910 } 2911 } 2912 2913 /* 2914 The derived values that are cached: 2915 - mSinkBufferSize from frame count * frame size 2916 - mActiveSleepTimeUs from activeSleepTimeUs() 2917 - mIdleSleepTimeUs from idleSleepTimeUs() 2918 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2919 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2920 - maxPeriod from frame count and sample rate (MIXER only) 2921 2922 The parameters that affect these derived values are: 2923 - frame count 2924 - frame size 2925 - sample rate 2926 - device type: A2DP or not 2927 - device latency 2928 - format: PCM or not 2929 - active sleep time 2930 - idle sleep time 2931 */ 2932 2933 void AudioFlinger::PlaybackThread::cacheParameters_l() 2934 { 2935 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2936 mActiveSleepTimeUs = activeSleepTimeUs(); 2937 mIdleSleepTimeUs = idleSleepTimeUs(); 2938 2939 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2940 // truncating audio when going to standby. 2941 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2942 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2943 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2944 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2945 } 2946 } 2947 } 2948 2949 bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2950 { 2951 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2952 this, streamType, mTracks.size()); 2953 bool trackMatch = false; 2954 size_t size = mTracks.size(); 2955 for (size_t i = 0; i < size; i++) { 2956 sp<Track> t = mTracks[i]; 2957 if (t->streamType() == streamType && t->isExternalTrack()) { 2958 t->invalidate(); 2959 trackMatch = true; 2960 } 2961 } 2962 return trackMatch; 2963 } 2964 2965 void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2966 { 2967 Mutex::Autolock _l(mLock); 2968 invalidateTracks_l(streamType); 2969 } 2970 2971 status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2972 { 2973 audio_session_t session = chain->sessionId(); 2974 sp<EffectBufferHalInterface> halInBuffer, halOutBuffer; 2975 status_t result = mAudioFlinger->mEffectsFactoryHal->mirrorBuffer( 2976 mEffectBufferEnabled ? mEffectBuffer : mSinkBuffer, 2977 mEffectBufferEnabled ? mEffectBufferSize : mSinkBufferSize, 2978 &halInBuffer); 2979 if (result != OK) return result; 2980 halOutBuffer = halInBuffer; 2981 effect_buffer_t *buffer = reinterpret_cast<effect_buffer_t*>(halInBuffer->externalData()); 2982 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2983 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2984 // Only one effect chain can be present in direct output thread and it uses 2985 // the sink buffer as input 2986 if (mType != DIRECT) { 2987 size_t numSamples = mNormalFrameCount * mChannelCount; 2988 status_t result = mAudioFlinger->mEffectsFactoryHal->allocateBuffer( 2989 numSamples * sizeof(effect_buffer_t), 2990 &halInBuffer); 2991 if (result != OK) return result; 2992 #ifdef FLOAT_EFFECT_CHAIN 2993 buffer = halInBuffer->audioBuffer()->f32; 2994 #else 2995 buffer = halInBuffer->audioBuffer()->s16; 2996 #endif 2997 ALOGV("addEffectChain_l() creating new input buffer %p session %d", 2998 buffer, session); 2999 } 3000 3001 // Attach all tracks with same session ID to this chain. 3002 for (size_t i = 0; i < mTracks.size(); ++i) { 3003 sp<Track> track = mTracks[i]; 3004 if (session == track->sessionId()) { 3005 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 3006 buffer); 3007 track->setMainBuffer(buffer); 3008 chain->incTrackCnt(); 3009 } 3010 } 3011 3012 // indicate all active tracks in the chain 3013 for (const sp<Track> &track : mActiveTracks) { 3014 if (session == track->sessionId()) { 3015 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 3016 chain->incActiveTrackCnt(); 3017 } 3018 } 3019 } 3020 chain->setThread(this); 3021 chain->setInBuffer(halInBuffer); 3022 chain->setOutBuffer(halOutBuffer); 3023 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 3024 // chains list in order to be processed last as it contains output stage effects. 3025 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 3026 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 3027 // after track specific effects and before output stage. 3028 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 3029 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 3030 // Effect chain for other sessions are inserted at beginning of effect 3031 // chains list to be processed before output mix effects. Relative order between other 3032 // sessions is not important. 3033 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 3034 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 3035 "audio_session_t constants misdefined"); 3036 size_t size = mEffectChains.size(); 3037 size_t i = 0; 3038 for (i = 0; i < size; i++) { 3039 if (mEffectChains[i]->sessionId() < session) { 3040 break; 3041 } 3042 } 3043 mEffectChains.insertAt(chain, i); 3044 checkSuspendOnAddEffectChain_l(chain); 3045 3046 return NO_ERROR; 3047 } 3048 3049 size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 3050 { 3051 audio_session_t session = chain->sessionId(); 3052 3053 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 3054 3055 for (size_t i = 0; i < mEffectChains.size(); i++) { 3056 if (chain == mEffectChains[i]) { 3057 mEffectChains.removeAt(i); 3058 // detach all active tracks from the chain 3059 for (const sp<Track> &track : mActiveTracks) { 3060 if (session == track->sessionId()) { 3061 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 3062 chain.get(), session); 3063 chain->decActiveTrackCnt(); 3064 } 3065 } 3066 3067 // detach all tracks with same session ID from this chain 3068 for (size_t i = 0; i < mTracks.size(); ++i) { 3069 sp<Track> track = mTracks[i]; 3070 if (session == track->sessionId()) { 3071 track->setMainBuffer(reinterpret_cast<effect_buffer_t*>(mSinkBuffer)); 3072 chain->decTrackCnt(); 3073 } 3074 } 3075 break; 3076 } 3077 } 3078 return mEffectChains.size(); 3079 } 3080 3081 status_t AudioFlinger::PlaybackThread::attachAuxEffect( 3082 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId) 3083 { 3084 Mutex::Autolock _l(mLock); 3085 return attachAuxEffect_l(track, EffectId); 3086 } 3087 3088 status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 3089 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId) 3090 { 3091 status_t status = NO_ERROR; 3092 3093 if (EffectId == 0) { 3094 track->setAuxBuffer(0, NULL); 3095 } else { 3096 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 3097 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 3098 if (effect != 0) { 3099 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 3100 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 3101 } else { 3102 status = INVALID_OPERATION; 3103 } 3104 } else { 3105 status = BAD_VALUE; 3106 } 3107 } 3108 return status; 3109 } 3110 3111 void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 3112 { 3113 for (size_t i = 0; i < mTracks.size(); ++i) { 3114 sp<Track> track = mTracks[i]; 3115 if (track->auxEffectId() == effectId) { 3116 attachAuxEffect_l(track, 0); 3117 } 3118 } 3119 } 3120 3121 bool AudioFlinger::PlaybackThread::threadLoop() 3122 { 3123 tlNBLogWriter = mNBLogWriter.get(); 3124 3125 Vector< sp<Track> > tracksToRemove; 3126 3127 mStandbyTimeNs = systemTime(); 3128 nsecs_t lastWriteFinished = -1; // time last server write completed 3129 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written 3130 3131 // MIXER 3132 nsecs_t lastWarning = 0; 3133 3134 // DUPLICATING 3135 // FIXME could this be made local to while loop? 3136 writeFrames = 0; 3137 3138 cacheParameters_l(); 3139 mSleepTimeUs = mIdleSleepTimeUs; 3140 3141 if (mType == MIXER) { 3142 sleepTimeShift = 0; 3143 } 3144 3145 CpuStats cpuStats; 3146 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 3147 3148 acquireWakeLock(); 3149 3150 // mNBLogWriter logging APIs can only be called by a single thread, typically the 3151 // thread associated with this PlaybackThread. 3152 // If you want to share the mNBLogWriter with other threads (for example, binder threads) 3153 // then all such threads must agree to hold a common mutex before logging. 3154 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 3155 // and then that string will be logged at the next convenient opportunity. 3156 // See reference to logString below. 3157 const char *logString = NULL; 3158 3159 // Estimated time for next buffer to be written to hal. This is used only on 3160 // suspended mode (for now) to help schedule the wait time until next iteration. 3161 nsecs_t timeLoopNextNs = 0; 3162 3163 checkSilentMode_l(); 3164 3165 while (!exitPending()) 3166 { 3167 // Log merge requests are performed during AudioFlinger binder transactions, but 3168 // that does not cover audio playback. It's requested here for that reason. 3169 mAudioFlinger->requestLogMerge(); 3170 3171 cpuStats.sample(myName); 3172 3173 Vector< sp<EffectChain> > effectChains; 3174 3175 { // scope for mLock 3176 3177 Mutex::Autolock _l(mLock); 3178 3179 processConfigEvents_l(); 3180 3181 // See comment at declaration of logString for why this is done under mLock 3182 if (logString != NULL) { 3183 mNBLogWriter->logTimestamp(); 3184 mNBLogWriter->log(logString); 3185 logString = NULL; 3186 } 3187 3188 // Gather the framesReleased counters for all active tracks, 3189 // and associate with the sink frames written out. We need 3190 // this to convert the sink timestamp to the track timestamp. 3191 bool kernelLocationUpdate = false; 3192 if (mNormalSink != 0) { 3193 // Note: The DuplicatingThread may not have a mNormalSink. 3194 // We always fetch the timestamp here because often the downstream 3195 // sink will block while writing. 3196 ExtendedTimestamp timestamp; // use private copy to fetch 3197 (void) mNormalSink->getTimestamp(timestamp); 3198 3199 // We keep track of the last valid kernel position in case we are in underrun 3200 // and the normal mixer period is the same as the fast mixer period, or there 3201 // is some error from the HAL. 3202 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3203 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3204 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 3205 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3206 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3207 3208 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3209 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 3210 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3211 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 3212 } 3213 3214 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3215 kernelLocationUpdate = true; 3216 } else { 3217 ALOGVV("getTimestamp error - no valid kernel position"); 3218 } 3219 3220 // copy over kernel info 3221 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 3222 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] 3223 + mSuspendedFrames; // add frames discarded when suspended 3224 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 3225 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3226 } 3227 // mFramesWritten for non-offloaded tracks are contiguous 3228 // even after standby() is called. This is useful for the track frame 3229 // to sink frame mapping. 3230 bool serverLocationUpdate = false; 3231 if (mFramesWritten != lastFramesWritten) { 3232 serverLocationUpdate = true; 3233 lastFramesWritten = mFramesWritten; 3234 } 3235 // Only update timestamps if there is a meaningful change. 3236 // Either the kernel timestamp must be valid or we have written something. 3237 if (kernelLocationUpdate || serverLocationUpdate) { 3238 if (serverLocationUpdate) { 3239 // use the time before we called the HAL write - it is a bit more accurate 3240 // to when the server last read data than the current time here. 3241 // 3242 // If we haven't written anything, mLastWriteTime will be -1 3243 // and we use systemTime(). 3244 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 3245 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1 3246 ? systemTime() : mLastWriteTime; 3247 } 3248 3249 for (const sp<Track> &t : mActiveTracks) { 3250 if (!t->isFastTrack()) { 3251 t->updateTrackFrameInfo( 3252 t->mAudioTrackServerProxy->framesReleased(), 3253 mFramesWritten, 3254 mTimestamp); 3255 } 3256 } 3257 } 3258 #if 0 3259 // logFormat example 3260 if (z % 100 == 0) { 3261 timespec ts; 3262 clock_gettime(CLOCK_MONOTONIC, &ts); 3263 LOGT("This is an integer %d, this is a float %f, this is my " 3264 "pid %p %% %s %t", 42, 3.14, "and this is a timestamp", ts); 3265 LOGT("A deceptive null-terminated string %\0"); 3266 } 3267 ++z; 3268 #endif 3269 saveOutputTracks(); 3270 if (mSignalPending) { 3271 // A signal was raised while we were unlocked 3272 mSignalPending = false; 3273 } else if (waitingAsyncCallback_l()) { 3274 if (exitPending()) { 3275 break; 3276 } 3277 bool released = false; 3278 if (!keepWakeLock()) { 3279 releaseWakeLock_l(); 3280 released = true; 3281 } 3282 3283 const int64_t waitNs = computeWaitTimeNs_l(); 3284 ALOGV("wait async completion (wait time: %lld)", (long long)waitNs); 3285 status_t status = mWaitWorkCV.waitRelative(mLock, waitNs); 3286 if (status == TIMED_OUT) { 3287 mSignalPending = true; // if timeout recheck everything 3288 } 3289 ALOGV("async completion/wake"); 3290 if (released) { 3291 acquireWakeLock_l(); 3292 } 3293 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3294 mSleepTimeUs = 0; 3295 3296 continue; 3297 } 3298 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 3299 isSuspended()) { 3300 // put audio hardware into standby after short delay 3301 if (shouldStandby_l()) { 3302 3303 threadLoop_standby(); 3304 3305 // This is where we go into standby 3306 if (!mStandby) { 3307 LOG_AUDIO_STATE(); 3308 } 3309 mStandby = true; 3310 } 3311 3312 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 3313 // we're about to wait, flush the binder command buffer 3314 IPCThreadState::self()->flushCommands(); 3315 3316 clearOutputTracks(); 3317 3318 if (exitPending()) { 3319 break; 3320 } 3321 3322 releaseWakeLock_l(); 3323 // wait until we have something to do... 3324 ALOGV("%s going to sleep", myName.string()); 3325 mWaitWorkCV.wait(mLock); 3326 ALOGV("%s waking up", myName.string()); 3327 acquireWakeLock_l(); 3328 3329 mMixerStatus = MIXER_IDLE; 3330 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 3331 mBytesWritten = 0; 3332 mBytesRemaining = 0; 3333 checkSilentMode_l(); 3334 3335 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3336 mSleepTimeUs = mIdleSleepTimeUs; 3337 if (mType == MIXER) { 3338 sleepTimeShift = 0; 3339 } 3340 3341 continue; 3342 } 3343 } 3344 // mMixerStatusIgnoringFastTracks is also updated internally 3345 mMixerStatus = prepareTracks_l(&tracksToRemove); 3346 3347 mActiveTracks.updatePowerState(this); 3348 3349 updateMetadata_l(); 3350 3351 // prevent any changes in effect chain list and in each effect chain 3352 // during mixing and effect process as the audio buffers could be deleted 3353 // or modified if an effect is created or deleted 3354 lockEffectChains_l(effectChains); 3355 } // mLock scope ends 3356 3357 if (mBytesRemaining == 0) { 3358 mCurrentWriteLength = 0; 3359 if (mMixerStatus == MIXER_TRACKS_READY) { 3360 // threadLoop_mix() sets mCurrentWriteLength 3361 threadLoop_mix(); 3362 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3363 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3364 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3365 // must be written to HAL 3366 threadLoop_sleepTime(); 3367 if (mSleepTimeUs == 0) { 3368 mCurrentWriteLength = mSinkBufferSize; 3369 } 3370 } 3371 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3372 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3373 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3374 // or mSinkBuffer (if there are no effects). 3375 // 3376 // This is done pre-effects computation; if effects change to 3377 // support higher precision, this needs to move. 3378 // 3379 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3380 // TODO use mSleepTimeUs == 0 as an additional condition. 3381 if (mMixerBufferValid) { 3382 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3383 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3384 3385 // mono blend occurs for mixer threads only (not direct or offloaded) 3386 // and is handled here if we're going directly to the sink. 3387 if (requireMonoBlend() && !mEffectBufferValid) { 3388 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3389 true /*limit*/); 3390 } 3391 3392 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3393 mNormalFrameCount * mChannelCount); 3394 } 3395 3396 mBytesRemaining = mCurrentWriteLength; 3397 if (isSuspended()) { 3398 // Simulate write to HAL when suspended (e.g. BT SCO phone call). 3399 mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer. 3400 const size_t framesRemaining = mBytesRemaining / mFrameSize; 3401 mBytesWritten += mBytesRemaining; 3402 mFramesWritten += framesRemaining; 3403 mSuspendedFrames += framesRemaining; // to adjust kernel HAL position 3404 mBytesRemaining = 0; 3405 } 3406 3407 // only process effects if we're going to write 3408 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3409 for (size_t i = 0; i < effectChains.size(); i ++) { 3410 effectChains[i]->process_l(); 3411 } 3412 } 3413 } 3414 // Process effect chains for offloaded thread even if no audio 3415 // was read from audio track: process only updates effect state 3416 // and thus does have to be synchronized with audio writes but may have 3417 // to be called while waiting for async write callback 3418 if (mType == OFFLOAD) { 3419 for (size_t i = 0; i < effectChains.size(); i ++) { 3420 effectChains[i]->process_l(); 3421 } 3422 } 3423 3424 // Only if the Effects buffer is enabled and there is data in the 3425 // Effects buffer (buffer valid), we need to 3426 // copy into the sink buffer. 3427 // TODO use mSleepTimeUs == 0 as an additional condition. 3428 if (mEffectBufferValid) { 3429 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3430 3431 if (requireMonoBlend()) { 3432 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3433 true /*limit*/); 3434 } 3435 3436 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3437 mNormalFrameCount * mChannelCount); 3438 } 3439 3440 // enable changes in effect chain 3441 unlockEffectChains(effectChains); 3442 3443 if (!waitingAsyncCallback()) { 3444 // mSleepTimeUs == 0 means we must write to audio hardware 3445 if (mSleepTimeUs == 0) { 3446 ssize_t ret = 0; 3447 // We save lastWriteFinished here, as previousLastWriteFinished, 3448 // for throttling. On thread start, previousLastWriteFinished will be 3449 // set to -1, which properly results in no throttling after the first write. 3450 nsecs_t previousLastWriteFinished = lastWriteFinished; 3451 nsecs_t delta = 0; 3452 if (mBytesRemaining) { 3453 // FIXME rewrite to reduce number of system calls 3454 mLastWriteTime = systemTime(); // also used for dumpsys 3455 ret = threadLoop_write(); 3456 lastWriteFinished = systemTime(); 3457 delta = lastWriteFinished - mLastWriteTime; 3458 if (ret < 0) { 3459 mBytesRemaining = 0; 3460 } else { 3461 mBytesWritten += ret; 3462 mBytesRemaining -= ret; 3463 mFramesWritten += ret / mFrameSize; 3464 } 3465 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3466 (mMixerStatus == MIXER_DRAIN_ALL)) { 3467 threadLoop_drain(); 3468 } 3469 if (mType == MIXER && !mStandby) { 3470 // write blocked detection 3471 if (delta > maxPeriod) { 3472 mNumDelayedWrites++; 3473 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) { 3474 ATRACE_NAME("underrun"); 3475 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3476 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3477 lastWarning = lastWriteFinished; 3478 } 3479 } 3480 3481 if (mThreadThrottle 3482 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3483 && ret > 0) { // we wrote something 3484 // Limit MixerThread data processing to no more than twice the 3485 // expected processing rate. 3486 // 3487 // This helps prevent underruns with NuPlayer and other applications 3488 // which may set up buffers that are close to the minimum size, or use 3489 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3490 // 3491 // The throttle smooths out sudden large data drains from the device, 3492 // e.g. when it comes out of standby, which often causes problems with 3493 // (1) mixer threads without a fast mixer (which has its own warm-up) 3494 // (2) minimum buffer sized tracks (even if the track is full, 3495 // the app won't fill fast enough to handle the sudden draw). 3496 // 3497 // Total time spent in last processing cycle equals time spent in 3498 // 1. threadLoop_write, as well as time spent in 3499 // 2. threadLoop_mix (significant for heavy mixing, especially 3500 // on low tier processors) 3501 3502 // it's OK if deltaMs (and deltaNs) is an overestimate. 3503 nsecs_t deltaNs; 3504 // deltaNs = lastWriteFinished - previousLastWriteFinished; 3505 __builtin_sub_overflow( 3506 lastWriteFinished,previousLastWriteFinished, &deltaNs); 3507 const int32_t deltaMs = deltaNs / 1000000; 3508 3509 const int32_t throttleMs = (int32_t)mHalfBufferMs - deltaMs; 3510 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3511 usleep(throttleMs * 1000); 3512 // notify of throttle start on verbose log 3513 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3514 "mixer(%p) throttle begin:" 3515 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3516 this, ret, deltaMs, throttleMs); 3517 mThreadThrottleTimeMs += throttleMs; 3518 // Throttle must be attributed to the previous mixer loop's write time 3519 // to allow back-to-back throttling. 3520 lastWriteFinished += throttleMs * 1000000; 3521 } else { 3522 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3523 if (diff > 0) { 3524 // notify of throttle end on debug log 3525 // but prevent spamming for bluetooth 3526 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()) && 3527 !audio_is_hearing_aid_out_device(outDevice()), 3528 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3529 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3530 } 3531 } 3532 } 3533 } 3534 3535 } else { 3536 ATRACE_BEGIN("sleep"); 3537 Mutex::Autolock _l(mLock); 3538 // suspended requires accurate metering of sleep time. 3539 if (isSuspended()) { 3540 // advance by expected sleepTime 3541 timeLoopNextNs += microseconds((nsecs_t)mSleepTimeUs); 3542 const nsecs_t nowNs = systemTime(); 3543 3544 // compute expected next time vs current time. 3545 // (negative deltas are treated as delays). 3546 nsecs_t deltaNs = timeLoopNextNs - nowNs; 3547 if (deltaNs < -kMaxNextBufferDelayNs) { 3548 // Delays longer than the max allowed trigger a reset. 3549 ALOGV("DelayNs: %lld, resetting timeLoopNextNs", (long long) deltaNs); 3550 deltaNs = microseconds((nsecs_t)mSleepTimeUs); 3551 timeLoopNextNs = nowNs + deltaNs; 3552 } else if (deltaNs < 0) { 3553 // Delays within the max delay allowed: zero the delta/sleepTime 3554 // to help the system catch up in the next iteration(s) 3555 ALOGV("DelayNs: %lld, catching-up", (long long) deltaNs); 3556 deltaNs = 0; 3557 } 3558 // update sleep time (which is >= 0) 3559 mSleepTimeUs = deltaNs / 1000; 3560 } 3561 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) { 3562 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs)); 3563 } 3564 ATRACE_END(); 3565 } 3566 } 3567 3568 // Finally let go of removed track(s), without the lock held 3569 // since we can't guarantee the destructors won't acquire that 3570 // same lock. This will also mutate and push a new fast mixer state. 3571 threadLoop_removeTracks(tracksToRemove); 3572 tracksToRemove.clear(); 3573 3574 // FIXME I don't understand the need for this here; 3575 // it was in the original code but maybe the 3576 // assignment in saveOutputTracks() makes this unnecessary? 3577 clearOutputTracks(); 3578 3579 // Effect chains will be actually deleted here if they were removed from 3580 // mEffectChains list during mixing or effects processing 3581 effectChains.clear(); 3582 3583 // FIXME Note that the above .clear() is no longer necessary since effectChains 3584 // is now local to this block, but will keep it for now (at least until merge done). 3585 } 3586 3587 threadLoop_exit(); 3588 3589 if (!mStandby) { 3590 threadLoop_standby(); 3591 mStandby = true; 3592 } 3593 3594 releaseWakeLock(); 3595 3596 ALOGV("Thread %p type %d exiting", this, mType); 3597 return false; 3598 } 3599 3600 // removeTracks_l() must be called with ThreadBase::mLock held 3601 void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3602 { 3603 size_t count = tracksToRemove.size(); 3604 if (count > 0) { 3605 for (size_t i=0 ; i<count ; i++) { 3606 const sp<Track>& track = tracksToRemove.itemAt(i); 3607 mActiveTracks.remove(track); 3608 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3609 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3610 if (chain != 0) { 3611 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3612 track->sessionId()); 3613 chain->decActiveTrackCnt(); 3614 } 3615 if (track->isTerminated()) { 3616 removeTrack_l(track); 3617 } 3618 } 3619 } 3620 3621 } 3622 3623 status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3624 { 3625 if (mNormalSink != 0) { 3626 ExtendedTimestamp ets; 3627 status_t status = mNormalSink->getTimestamp(ets); 3628 if (status == NO_ERROR) { 3629 status = ets.getBestTimestamp(×tamp); 3630 } 3631 return status; 3632 } 3633 if ((mType == OFFLOAD || mType == DIRECT) && mOutput != NULL) { 3634 uint64_t position64; 3635 if (mOutput->getPresentationPosition(&position64, ×tamp.mTime) == OK) { 3636 timestamp.mPosition = (uint32_t)position64; 3637 return NO_ERROR; 3638 } 3639 } 3640 return INVALID_OPERATION; 3641 } 3642 3643 status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3644 audio_patch_handle_t *handle) 3645 { 3646 status_t status; 3647 if (property_get_bool("af.patch_park", false /* default_value */)) { 3648 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3649 // or if HAL does not properly lock against access. 3650 AutoPark<FastMixer> park(mFastMixer); 3651 status = PlaybackThread::createAudioPatch_l(patch, handle); 3652 } else { 3653 status = PlaybackThread::createAudioPatch_l(patch, handle); 3654 } 3655 return status; 3656 } 3657 3658 status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3659 audio_patch_handle_t *handle) 3660 { 3661 status_t status = NO_ERROR; 3662 3663 // store new device and send to effects 3664 audio_devices_t type = AUDIO_DEVICE_NONE; 3665 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3666 type |= patch->sinks[i].ext.device.type; 3667 } 3668 3669 #ifdef ADD_BATTERY_DATA 3670 // when changing the audio output device, call addBatteryData to notify 3671 // the change 3672 if (mOutDevice != type) { 3673 uint32_t params = 0; 3674 // check whether speaker is on 3675 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3676 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3677 } 3678 3679 audio_devices_t deviceWithoutSpeaker 3680 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3681 // check if any other device (except speaker) is on 3682 if (type & deviceWithoutSpeaker) { 3683 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3684 } 3685 3686 if (params != 0) { 3687 addBatteryData(params); 3688 } 3689 } 3690 #endif 3691 3692 for (size_t i = 0; i < mEffectChains.size(); i++) { 3693 mEffectChains[i]->setDevice_l(type); 3694 } 3695 3696 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3697 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3698 bool configChanged = mPrevOutDevice != type; 3699 mOutDevice = type; 3700 mPatch = *patch; 3701 3702 if (mOutput->audioHwDev->supportsAudioPatches()) { 3703 sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice(); 3704 status = hwDevice->createAudioPatch(patch->num_sources, 3705 patch->sources, 3706 patch->num_sinks, 3707 patch->sinks, 3708 handle); 3709 } else { 3710 char *address; 3711 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3712 //FIXME: we only support address on first sink with HAL version < 3.0 3713 address = audio_device_address_to_parameter( 3714 patch->sinks[0].ext.device.type, 3715 patch->sinks[0].ext.device.address); 3716 } else { 3717 address = (char *)calloc(1, 1); 3718 } 3719 AudioParameter param = AudioParameter(String8(address)); 3720 free(address); 3721 param.addInt(String8(AudioParameter::keyRouting), (int)type); 3722 status = mOutput->stream->setParameters(param.toString()); 3723 *handle = AUDIO_PATCH_HANDLE_NONE; 3724 } 3725 if (configChanged) { 3726 mPrevOutDevice = type; 3727 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3728 } 3729 return status; 3730 } 3731 3732 status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3733 { 3734 status_t status; 3735 if (property_get_bool("af.patch_park", false /* default_value */)) { 3736 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3737 // or if HAL does not properly lock against access. 3738 AutoPark<FastMixer> park(mFastMixer); 3739 status = PlaybackThread::releaseAudioPatch_l(handle); 3740 } else { 3741 status = PlaybackThread::releaseAudioPatch_l(handle); 3742 } 3743 return status; 3744 } 3745 3746 status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3747 { 3748 status_t status = NO_ERROR; 3749 3750 mOutDevice = AUDIO_DEVICE_NONE; 3751 3752 if (mOutput->audioHwDev->supportsAudioPatches()) { 3753 sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice(); 3754 status = hwDevice->releaseAudioPatch(handle); 3755 } else { 3756 AudioParameter param; 3757 param.addInt(String8(AudioParameter::keyRouting), 0); 3758 status = mOutput->stream->setParameters(param.toString()); 3759 } 3760 return status; 3761 } 3762 3763 void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3764 { 3765 Mutex::Autolock _l(mLock); 3766 mTracks.add(track); 3767 } 3768 3769 void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3770 { 3771 Mutex::Autolock _l(mLock); 3772 destroyTrack_l(track); 3773 } 3774 3775 void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3776 { 3777 ThreadBase::getAudioPortConfig(config); 3778 config->role = AUDIO_PORT_ROLE_SOURCE; 3779 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3780 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3781 } 3782 3783 // ---------------------------------------------------------------------------- 3784 3785 AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3786 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3787 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3788 // mAudioMixer below 3789 // mFastMixer below 3790 mFastMixerFutex(0), 3791 mMasterMono(false) 3792 // mOutputSink below 3793 // mPipeSink below 3794 // mNormalSink below 3795 { 3796 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3797 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, " 3798 "mFrameCount=%zu, mNormalFrameCount=%zu", 3799 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3800 mNormalFrameCount); 3801 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3802 3803 if (type == DUPLICATING) { 3804 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3805 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3806 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3807 return; 3808 } 3809 // create an NBAIO sink for the HAL output stream, and negotiate 3810 mOutputSink = new AudioStreamOutSink(output->stream); 3811 size_t numCounterOffers = 0; 3812 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3813 #if !LOG_NDEBUG 3814 ssize_t index = 3815 #else 3816 (void) 3817 #endif 3818 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3819 ALOG_ASSERT(index == 0); 3820 3821 // initialize fast mixer depending on configuration 3822 bool initFastMixer; 3823 switch (kUseFastMixer) { 3824 case FastMixer_Never: 3825 initFastMixer = false; 3826 break; 3827 case FastMixer_Always: 3828 initFastMixer = true; 3829 break; 3830 case FastMixer_Static: 3831 case FastMixer_Dynamic: 3832 // FastMixer was designed to operate with a HAL that pulls at a regular rate, 3833 // where the period is less than an experimentally determined threshold that can be 3834 // scheduled reliably with CFS. However, the BT A2DP HAL is 3835 // bursty (does not pull at a regular rate) and so cannot operate with FastMixer. 3836 initFastMixer = mFrameCount < mNormalFrameCount 3837 && (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) == 0; 3838 break; 3839 } 3840 ALOGW_IF(initFastMixer == false && mFrameCount < mNormalFrameCount, 3841 "FastMixer is preferred for this sink as frameCount %zu is less than threshold %zu", 3842 mFrameCount, mNormalFrameCount); 3843 if (initFastMixer) { 3844 audio_format_t fastMixerFormat; 3845 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3846 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3847 } else { 3848 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3849 } 3850 if (mFormat != fastMixerFormat) { 3851 // change our Sink format to accept our intermediate precision 3852 mFormat = fastMixerFormat; 3853 free(mSinkBuffer); 3854 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3855 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3856 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3857 } 3858 3859 // create a MonoPipe to connect our submix to FastMixer 3860 NBAIO_Format format = mOutputSink->format(); 3861 #ifdef TEE_SINK 3862 NBAIO_Format origformat = format; 3863 #endif 3864 // adjust format to match that of the Fast Mixer 3865 ALOGV("format changed from %#x to %#x", format.mFormat, fastMixerFormat); 3866 format.mFormat = fastMixerFormat; 3867 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3868 3869 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3870 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3871 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3872 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3873 const NBAIO_Format offers[1] = {format}; 3874 size_t numCounterOffers = 0; 3875 #if !LOG_NDEBUG || defined(TEE_SINK) 3876 ssize_t index = 3877 #else 3878 (void) 3879 #endif 3880 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3881 ALOG_ASSERT(index == 0); 3882 monoPipe->setAvgFrames((mScreenState & 1) ? 3883 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3884 mPipeSink = monoPipe; 3885 3886 #ifdef TEE_SINK 3887 if (mTeeSinkOutputEnabled) { 3888 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3889 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3890 const NBAIO_Format offers2[1] = {origformat}; 3891 numCounterOffers = 0; 3892 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3893 ALOG_ASSERT(index == 0); 3894 mTeeSink = teeSink; 3895 PipeReader *teeSource = new PipeReader(*teeSink); 3896 numCounterOffers = 0; 3897 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3898 ALOG_ASSERT(index == 0); 3899 mTeeSource = teeSource; 3900 } 3901 #endif 3902 3903 // create fast mixer and configure it initially with just one fast track for our submix 3904 mFastMixer = new FastMixer(); 3905 FastMixerStateQueue *sq = mFastMixer->sq(); 3906 #ifdef STATE_QUEUE_DUMP 3907 sq->setObserverDump(&mStateQueueObserverDump); 3908 sq->setMutatorDump(&mStateQueueMutatorDump); 3909 #endif 3910 FastMixerState *state = sq->begin(); 3911 FastTrack *fastTrack = &state->mFastTracks[0]; 3912 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3913 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3914 fastTrack->mVolumeProvider = NULL; 3915 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3916 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3917 fastTrack->mGeneration++; 3918 state->mFastTracksGen++; 3919 state->mTrackMask = 1; 3920 // fast mixer will use the HAL output sink 3921 state->mOutputSink = mOutputSink.get(); 3922 state->mOutputSinkGen++; 3923 state->mFrameCount = mFrameCount; 3924 state->mCommand = FastMixerState::COLD_IDLE; 3925 // already done in constructor initialization list 3926 //mFastMixerFutex = 0; 3927 state->mColdFutexAddr = &mFastMixerFutex; 3928 state->mColdGen++; 3929 state->mDumpState = &mFastMixerDumpState; 3930 #ifdef TEE_SINK 3931 state->mTeeSink = mTeeSink.get(); 3932 #endif 3933 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3934 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3935 sq->end(); 3936 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3937 3938 // start the fast mixer 3939 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3940 pid_t tid = mFastMixer->getTid(); 3941 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer, false /*forApp*/); 3942 stream()->setHalThreadPriority(kPriorityFastMixer); 3943 3944 #ifdef AUDIO_WATCHDOG 3945 // create and start the watchdog 3946 mAudioWatchdog = new AudioWatchdog(); 3947 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3948 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3949 tid = mAudioWatchdog->getTid(); 3950 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer, false /*forApp*/); 3951 #endif 3952 3953 } 3954 3955 switch (kUseFastMixer) { 3956 case FastMixer_Never: 3957 case FastMixer_Dynamic: 3958 mNormalSink = mOutputSink; 3959 break; 3960 case FastMixer_Always: 3961 mNormalSink = mPipeSink; 3962 break; 3963 case FastMixer_Static: 3964 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3965 break; 3966 } 3967 } 3968 3969 AudioFlinger::MixerThread::~MixerThread() 3970 { 3971 if (mFastMixer != 0) { 3972 FastMixerStateQueue *sq = mFastMixer->sq(); 3973 FastMixerState *state = sq->begin(); 3974 if (state->mCommand == FastMixerState::COLD_IDLE) { 3975 int32_t old = android_atomic_inc(&mFastMixerFutex); 3976 if (old == -1) { 3977 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3978 } 3979 } 3980 state->mCommand = FastMixerState::EXIT; 3981 sq->end(); 3982 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3983 mFastMixer->join(); 3984 // Though the fast mixer thread has exited, it's state queue is still valid. 3985 // We'll use that extract the final state which contains one remaining fast track 3986 // corresponding to our sub-mix. 3987 state = sq->begin(); 3988 ALOG_ASSERT(state->mTrackMask == 1); 3989 FastTrack *fastTrack = &state->mFastTracks[0]; 3990 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3991 delete fastTrack->mBufferProvider; 3992 sq->end(false /*didModify*/); 3993 mFastMixer.clear(); 3994 #ifdef AUDIO_WATCHDOG 3995 if (mAudioWatchdog != 0) { 3996 mAudioWatchdog->requestExit(); 3997 mAudioWatchdog->requestExitAndWait(); 3998 mAudioWatchdog.clear(); 3999 } 4000 #endif 4001 } 4002 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 4003 delete mAudioMixer; 4004 } 4005 4006 4007 uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 4008 { 4009 if (mFastMixer != 0) { 4010 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 4011 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 4012 } 4013 return latency; 4014 } 4015 4016 4017 void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 4018 { 4019 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 4020 } 4021 4022 ssize_t AudioFlinger::MixerThread::threadLoop_write() 4023 { 4024 // FIXME we should only do one push per cycle; confirm this is true 4025 // Start the fast mixer if it's not already running 4026 if (mFastMixer != 0) { 4027 FastMixerStateQueue *sq = mFastMixer->sq(); 4028 FastMixerState *state = sq->begin(); 4029 if (state->mCommand != FastMixerState::MIX_WRITE && 4030 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 4031 if (state->mCommand == FastMixerState::COLD_IDLE) { 4032 4033 // FIXME workaround for first HAL write being CPU bound on some devices 4034 ATRACE_BEGIN("write"); 4035 mOutput->write((char *)mSinkBuffer, 0); 4036 ATRACE_END(); 4037 4038 int32_t old = android_atomic_inc(&mFastMixerFutex); 4039 if (old == -1) { 4040 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 4041 } 4042 #ifdef AUDIO_WATCHDOG 4043 if (mAudioWatchdog != 0) { 4044 mAudioWatchdog->resume(); 4045 } 4046 #endif 4047 } 4048 state->mCommand = FastMixerState::MIX_WRITE; 4049 #ifdef FAST_THREAD_STATISTICS 4050 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 4051 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 4052 #endif 4053 sq->end(); 4054 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 4055 if (kUseFastMixer == FastMixer_Dynamic) { 4056 mNormalSink = mPipeSink; 4057 } 4058 } else { 4059 sq->end(false /*didModify*/); 4060 } 4061 } 4062 return PlaybackThread::threadLoop_write(); 4063 } 4064 4065 void AudioFlinger::MixerThread::threadLoop_standby() 4066 { 4067 // Idle the fast mixer if it's currently running 4068 if (mFastMixer != 0) { 4069 FastMixerStateQueue *sq = mFastMixer->sq(); 4070 FastMixerState *state = sq->begin(); 4071 if (!(state->mCommand & FastMixerState::IDLE)) { 4072 // Report any frames trapped in the Monopipe 4073 MonoPipe *monoPipe = (MonoPipe *)mPipeSink.get(); 4074 const long long pipeFrames = monoPipe->maxFrames() - monoPipe->availableToWrite(); 4075 mLocalLog.log("threadLoop_standby: framesWritten:%lld suspendedFrames:%lld " 4076 "monoPipeWritten:%lld monoPipeLeft:%lld", 4077 (long long)mFramesWritten, (long long)mSuspendedFrames, 4078 (long long)mPipeSink->framesWritten(), pipeFrames); 4079 mLocalLog.log("threadLoop_standby: %s", mTimestamp.toString().c_str()); 4080 4081 state->mCommand = FastMixerState::COLD_IDLE; 4082 state->mColdFutexAddr = &mFastMixerFutex; 4083 state->mColdGen++; 4084 mFastMixerFutex = 0; 4085 sq->end(); 4086 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 4087 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 4088 if (kUseFastMixer == FastMixer_Dynamic) { 4089 mNormalSink = mOutputSink; 4090 } 4091 #ifdef AUDIO_WATCHDOG 4092 if (mAudioWatchdog != 0) { 4093 mAudioWatchdog->pause(); 4094 } 4095 #endif 4096 } else { 4097 sq->end(false /*didModify*/); 4098 } 4099 } 4100 PlaybackThread::threadLoop_standby(); 4101 } 4102 4103 bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 4104 { 4105 return false; 4106 } 4107 4108 bool AudioFlinger::PlaybackThread::shouldStandby_l() 4109 { 4110 return !mStandby; 4111 } 4112 4113 bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 4114 { 4115 Mutex::Autolock _l(mLock); 4116 return waitingAsyncCallback_l(); 4117 } 4118 4119 // shared by MIXER and DIRECT, overridden by DUPLICATING 4120 void AudioFlinger::PlaybackThread::threadLoop_standby() 4121 { 4122 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 4123 mOutput->standby(); 4124 if (mUseAsyncWrite != 0) { 4125 // discard any pending drain or write ack by incrementing sequence 4126 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 4127 mDrainSequence = (mDrainSequence + 2) & ~1; 4128 ALOG_ASSERT(mCallbackThread != 0); 4129 mCallbackThread->setWriteBlocked(mWriteAckSequence); 4130 mCallbackThread->setDraining(mDrainSequence); 4131 } 4132 mHwPaused = false; 4133 } 4134 4135 void AudioFlinger::PlaybackThread::onAddNewTrack_l() 4136 { 4137 ALOGV("signal playback thread"); 4138 broadcast_l(); 4139 } 4140 4141 void AudioFlinger::PlaybackThread::onAsyncError() 4142 { 4143 for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { 4144 invalidateTracks((audio_stream_type_t)i); 4145 } 4146 } 4147 4148 void AudioFlinger::MixerThread::threadLoop_mix() 4149 { 4150 // mix buffers... 4151 mAudioMixer->process(); 4152 mCurrentWriteLength = mSinkBufferSize; 4153 // increase sleep time progressively when application underrun condition clears. 4154 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 4155 // that a steady state of alternating ready/not ready conditions keeps the sleep time 4156 // such that we would underrun the audio HAL. 4157 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 4158 sleepTimeShift--; 4159 } 4160 mSleepTimeUs = 0; 4161 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 4162 //TODO: delay standby when effects have a tail 4163 4164 } 4165 4166 void AudioFlinger::MixerThread::threadLoop_sleepTime() 4167 { 4168 // If no tracks are ready, sleep once for the duration of an output 4169 // buffer size, then write 0s to the output 4170 if (mSleepTimeUs == 0) { 4171 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 4172 if (mPipeSink.get() != nullptr && mPipeSink == mNormalSink) { 4173 // Using the Monopipe availableToWrite, we estimate the 4174 // sleep time to retry for more data (before we underrun). 4175 MonoPipe *monoPipe = static_cast<MonoPipe *>(mPipeSink.get()); 4176 const ssize_t availableToWrite = mPipeSink->availableToWrite(); 4177 const size_t pipeFrames = monoPipe->maxFrames(); 4178 const size_t framesLeft = pipeFrames - max(availableToWrite, 0); 4179 // HAL_framecount <= framesDelay ~ framesLeft / 2 <= Normal_Mixer_framecount 4180 const size_t framesDelay = std::min( 4181 mNormalFrameCount, max(framesLeft / 2, mFrameCount)); 4182 ALOGV("pipeFrames:%zu framesLeft:%zu framesDelay:%zu", 4183 pipeFrames, framesLeft, framesDelay); 4184 mSleepTimeUs = framesDelay * MICROS_PER_SECOND / mSampleRate; 4185 } else { 4186 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 4187 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 4188 mSleepTimeUs = kMinThreadSleepTimeUs; 4189 } 4190 // reduce sleep time in case of consecutive application underruns to avoid 4191 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 4192 // duration we would end up writing less data than needed by the audio HAL if 4193 // the condition persists. 4194 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 4195 sleepTimeShift++; 4196 } 4197 } 4198 } else { 4199 mSleepTimeUs = mIdleSleepTimeUs; 4200 } 4201 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 4202 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 4203 // before effects processing or output. 4204 if (mMixerBufferValid) { 4205 memset(mMixerBuffer, 0, mMixerBufferSize); 4206 } else { 4207 memset(mSinkBuffer, 0, mSinkBufferSize); 4208 } 4209 mSleepTimeUs = 0; 4210 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 4211 "anticipated start"); 4212 } 4213 // TODO add standby time extension fct of effect tail 4214 } 4215 4216 // prepareTracks_l() must be called with ThreadBase::mLock held 4217 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 4218 Vector< sp<Track> > *tracksToRemove) 4219 { 4220 // clean up deleted track names in AudioMixer before allocating new tracks 4221 (void)mTracks.processDeletedTrackNames([this](int name) { 4222 // for each name, destroy it in the AudioMixer 4223 if (mAudioMixer->exists(name)) { 4224 mAudioMixer->destroy(name); 4225 } 4226 }); 4227 mTracks.clearDeletedTrackNames(); 4228 4229 mixer_state mixerStatus = MIXER_IDLE; 4230 // find out which tracks need to be processed 4231 size_t count = mActiveTracks.size(); 4232 size_t mixedTracks = 0; 4233 size_t tracksWithEffect = 0; 4234 // counts only _active_ fast tracks 4235 size_t fastTracks = 0; 4236 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 4237 4238 float masterVolume = mMasterVolume; 4239 bool masterMute = mMasterMute; 4240 4241 if (masterMute) { 4242 masterVolume = 0; 4243 } 4244 // Delegate master volume control to effect in output mix effect chain if needed 4245 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4246 if (chain != 0) { 4247 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 4248 chain->setVolume_l(&v, &v); 4249 masterVolume = (float)((v + (1 << 23)) >> 24); 4250 chain.clear(); 4251 } 4252 4253 // prepare a new state to push 4254 FastMixerStateQueue *sq = NULL; 4255 FastMixerState *state = NULL; 4256 bool didModify = false; 4257 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 4258 bool coldIdle = false; 4259 if (mFastMixer != 0) { 4260 sq = mFastMixer->sq(); 4261 state = sq->begin(); 4262 coldIdle = state->mCommand == FastMixerState::COLD_IDLE; 4263 } 4264 4265 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 4266 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 4267 4268 for (size_t i=0 ; i<count ; i++) { 4269 const sp<Track> t = mActiveTracks[i]; 4270 4271 // this const just means the local variable doesn't change 4272 Track* const track = t.get(); 4273 4274 // process fast tracks 4275 if (track->isFastTrack()) { 4276 4277 // It's theoretically possible (though unlikely) for a fast track to be created 4278 // and then removed within the same normal mix cycle. This is not a problem, as 4279 // the track never becomes active so it's fast mixer slot is never touched. 4280 // The converse, of removing an (active) track and then creating a new track 4281 // at the identical fast mixer slot within the same normal mix cycle, 4282 // is impossible because the slot isn't marked available until the end of each cycle. 4283 int j = track->mFastIndex; 4284 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 4285 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 4286 FastTrack *fastTrack = &state->mFastTracks[j]; 4287 4288 // Determine whether the track is currently in underrun condition, 4289 // and whether it had a recent underrun. 4290 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 4291 FastTrackUnderruns underruns = ftDump->mUnderruns; 4292 uint32_t recentFull = (underruns.mBitFields.mFull - 4293 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 4294 uint32_t recentPartial = (underruns.mBitFields.mPartial - 4295 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 4296 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 4297 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 4298 uint32_t recentUnderruns = recentPartial + recentEmpty; 4299 track->mObservedUnderruns = underruns; 4300 // don't count underruns that occur while stopping or pausing 4301 // or stopped which can occur when flush() is called while active 4302 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 4303 recentUnderruns > 0) { 4304 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 4305 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 4306 } else { 4307 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4308 } 4309 4310 // This is similar to the state machine for normal tracks, 4311 // with a few modifications for fast tracks. 4312 bool isActive = true; 4313 switch (track->mState) { 4314 case TrackBase::STOPPING_1: 4315 // track stays active in STOPPING_1 state until first underrun 4316 if (recentUnderruns > 0 || track->isTerminated()) { 4317 track->mState = TrackBase::STOPPING_2; 4318 } 4319 break; 4320 case TrackBase::PAUSING: 4321 // ramp down is not yet implemented 4322 track->setPaused(); 4323 break; 4324 case TrackBase::RESUMING: 4325 // ramp up is not yet implemented 4326 track->mState = TrackBase::ACTIVE; 4327 break; 4328 case TrackBase::ACTIVE: 4329 if (recentFull > 0 || recentPartial > 0) { 4330 // track has provided at least some frames recently: reset retry count 4331 track->mRetryCount = kMaxTrackRetries; 4332 } 4333 if (recentUnderruns == 0) { 4334 // no recent underruns: stay active 4335 break; 4336 } 4337 // there has recently been an underrun of some kind 4338 if (track->sharedBuffer() == 0) { 4339 // were any of the recent underruns "empty" (no frames available)? 4340 if (recentEmpty == 0) { 4341 // no, then ignore the partial underruns as they are allowed indefinitely 4342 break; 4343 } 4344 // there has recently been an "empty" underrun: decrement the retry counter 4345 if (--(track->mRetryCount) > 0) { 4346 break; 4347 } 4348 // indicate to client process that the track was disabled because of underrun; 4349 // it will then automatically call start() when data is available 4350 track->disable(); 4351 // remove from active list, but state remains ACTIVE [confusing but true] 4352 isActive = false; 4353 break; 4354 } 4355 // fall through 4356 case TrackBase::STOPPING_2: 4357 case TrackBase::PAUSED: 4358 case TrackBase::STOPPED: 4359 case TrackBase::FLUSHED: // flush() while active 4360 // Check for presentation complete if track is inactive 4361 // We have consumed all the buffers of this track. 4362 // This would be incomplete if we auto-paused on underrun 4363 { 4364 uint32_t latency = 0; 4365 status_t result = mOutput->stream->getLatency(&latency); 4366 ALOGE_IF(result != OK, 4367 "Error when retrieving output stream latency: %d", result); 4368 size_t audioHALFrames = (latency * mSampleRate) / 1000; 4369 int64_t framesWritten = mBytesWritten / mFrameSize; 4370 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 4371 // track stays in active list until presentation is complete 4372 break; 4373 } 4374 } 4375 if (track->isStopping_2()) { 4376 track->mState = TrackBase::STOPPED; 4377 } 4378 if (track->isStopped()) { 4379 // Can't reset directly, as fast mixer is still polling this track 4380 // track->reset(); 4381 // So instead mark this track as needing to be reset after push with ack 4382 resetMask |= 1 << i; 4383 } 4384 isActive = false; 4385 break; 4386 case TrackBase::IDLE: 4387 default: 4388 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4389 } 4390 4391 if (isActive) { 4392 // was it previously inactive? 4393 if (!(state->mTrackMask & (1 << j))) { 4394 ExtendedAudioBufferProvider *eabp = track; 4395 VolumeProvider *vp = track; 4396 fastTrack->mBufferProvider = eabp; 4397 fastTrack->mVolumeProvider = vp; 4398 fastTrack->mChannelMask = track->mChannelMask; 4399 fastTrack->mFormat = track->mFormat; 4400 fastTrack->mGeneration++; 4401 state->mTrackMask |= 1 << j; 4402 didModify = true; 4403 // no acknowledgement required for newly active tracks 4404 } 4405 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy; 4406 // cache the combined master volume and stream type volume for fast mixer; this 4407 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4408 const float vh = track->getVolumeHandler()->getVolume( 4409 proxy->framesReleased()).first; 4410 float volume = masterVolume 4411 * mStreamTypes[track->streamType()].volume 4412 * vh; 4413 track->mCachedVolume = volume; 4414 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4415 float vlf = volume * float_from_gain(gain_minifloat_unpack_left(vlr)); 4416 float vrf = volume * float_from_gain(gain_minifloat_unpack_right(vlr)); 4417 track->setFinalVolume((vlf + vrf) / 2.f); 4418 ++fastTracks; 4419 } else { 4420 // was it previously active? 4421 if (state->mTrackMask & (1 << j)) { 4422 fastTrack->mBufferProvider = NULL; 4423 fastTrack->mGeneration++; 4424 state->mTrackMask &= ~(1 << j); 4425 didModify = true; 4426 // If any fast tracks were removed, we must wait for acknowledgement 4427 // because we're about to decrement the last sp<> on those tracks. 4428 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4429 } else { 4430 // ALOGW rather than LOG_ALWAYS_FATAL because it seems there are cases where an 4431 // AudioTrack may start (which may not be with a start() but with a write() 4432 // after underrun) and immediately paused or released. In that case the 4433 // FastTrack state hasn't had time to update. 4434 // TODO Remove the ALOGW when this theory is confirmed. 4435 ALOGW("fast track %d should have been active; " 4436 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4437 j, track->mState, state->mTrackMask, recentUnderruns, 4438 track->sharedBuffer() != 0); 4439 // Since the FastMixer state already has the track inactive, do nothing here. 4440 } 4441 tracksToRemove->add(track); 4442 // Avoids a misleading display in dumpsys 4443 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4444 } 4445 continue; 4446 } 4447 4448 { // local variable scope to avoid goto warning 4449 4450 audio_track_cblk_t* cblk = track->cblk(); 4451 4452 // The first time a track is added we wait 4453 // for all its buffers to be filled before processing it 4454 int name = track->name(); 4455 4456 // if an active track doesn't exist in the AudioMixer, create it. 4457 if (!mAudioMixer->exists(name)) { 4458 status_t status = mAudioMixer->create( 4459 name, 4460 track->mChannelMask, 4461 track->mFormat, 4462 track->mSessionId); 4463 if (status != OK) { 4464 ALOGW("%s: cannot create track name" 4465 " %d, mask %#x, format %#x, sessionId %d in AudioMixer", 4466 __func__, name, track->mChannelMask, track->mFormat, track->mSessionId); 4467 tracksToRemove->add(track); 4468 track->invalidate(); // consider it dead. 4469 continue; 4470 } 4471 } 4472 4473 // make sure that we have enough frames to mix one full buffer. 4474 // enforce this condition only once to enable draining the buffer in case the client 4475 // app does not call stop() and relies on underrun to stop: 4476 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4477 // during last round 4478 size_t desiredFrames; 4479 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4480 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4481 4482 desiredFrames = sourceFramesNeededWithTimestretch( 4483 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4484 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4485 // add frames already consumed but not yet released by the resampler 4486 // because mAudioTrackServerProxy->framesReady() will include these frames 4487 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4488 4489 uint32_t minFrames = 1; 4490 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4491 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4492 minFrames = desiredFrames; 4493 } 4494 4495 size_t framesReady = track->framesReady(); 4496 if (ATRACE_ENABLED()) { 4497 // I wish we had formatted trace names 4498 std::string traceName("nRdy"); 4499 traceName += std::to_string(track->name()); 4500 ATRACE_INT(traceName.c_str(), framesReady); 4501 } 4502 if ((framesReady >= minFrames) && track->isReady() && 4503 !track->isPaused() && !track->isTerminated()) 4504 { 4505 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4506 4507 mixedTracks++; 4508 4509 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4510 // there is an effect chain connected to the track 4511 chain.clear(); 4512 if (track->mainBuffer() != mSinkBuffer && 4513 track->mainBuffer() != mMixerBuffer) { 4514 if (mEffectBufferEnabled) { 4515 mEffectBufferValid = true; // Later can set directly. 4516 } 4517 chain = getEffectChain_l(track->sessionId()); 4518 // Delegate volume control to effect in track effect chain if needed 4519 if (chain != 0) { 4520 tracksWithEffect++; 4521 } else { 4522 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4523 "session %d", 4524 name, track->sessionId()); 4525 } 4526 } 4527 4528 4529 int param = AudioMixer::VOLUME; 4530 if (track->mFillingUpStatus == Track::FS_FILLED) { 4531 // no ramp for the first volume setting 4532 track->mFillingUpStatus = Track::FS_ACTIVE; 4533 if (track->mState == TrackBase::RESUMING) { 4534 track->mState = TrackBase::ACTIVE; 4535 param = AudioMixer::RAMP_VOLUME; 4536 } 4537 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4538 mLeftVolFloat = -1.0; 4539 // FIXME should not make a decision based on mServer 4540 } else if (cblk->mServer != 0) { 4541 // If the track is stopped before the first frame was mixed, 4542 // do not apply ramp 4543 param = AudioMixer::RAMP_VOLUME; 4544 } 4545 4546 // compute volume for this track 4547 uint32_t vl, vr; // in U8.24 integer format 4548 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4549 // read original volumes with volume control 4550 float typeVolume = mStreamTypes[track->streamType()].volume; 4551 float v = masterVolume * typeVolume; 4552 4553 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4554 vl = vr = 0; 4555 vlf = vrf = vaf = 0.; 4556 if (track->isPausing()) { 4557 track->setPaused(); 4558 } 4559 } else { 4560 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy; 4561 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4562 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4563 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4564 // track volumes come from shared memory, so can't be trusted and must be clamped 4565 if (vlf > GAIN_FLOAT_UNITY) { 4566 ALOGV("Track left volume out of range: %.3g", vlf); 4567 vlf = GAIN_FLOAT_UNITY; 4568 } 4569 if (vrf > GAIN_FLOAT_UNITY) { 4570 ALOGV("Track right volume out of range: %.3g", vrf); 4571 vrf = GAIN_FLOAT_UNITY; 4572 } 4573 const float vh = track->getVolumeHandler()->getVolume( 4574 track->mAudioTrackServerProxy->framesReleased()).first; 4575 // now apply the master volume and stream type volume and shaper volume 4576 vlf *= v * vh; 4577 vrf *= v * vh; 4578 // assuming master volume and stream type volume each go up to 1.0, 4579 // then derive vl and vr as U8.24 versions for the effect chain 4580 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4581 vl = (uint32_t) (scaleto8_24 * vlf); 4582 vr = (uint32_t) (scaleto8_24 * vrf); 4583 // vl and vr are now in U8.24 format 4584 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4585 // send level comes from shared memory and so may be corrupt 4586 if (sendLevel > MAX_GAIN_INT) { 4587 ALOGV("Track send level out of range: %04X", sendLevel); 4588 sendLevel = MAX_GAIN_INT; 4589 } 4590 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4591 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4592 } 4593 4594 track->setFinalVolume((vrf + vlf) / 2.f); 4595 4596 // Delegate volume control to effect in track effect chain if needed 4597 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4598 // Do not ramp volume if volume is controlled by effect 4599 param = AudioMixer::VOLUME; 4600 // Update remaining floating point volume levels 4601 vlf = (float)vl / (1 << 24); 4602 vrf = (float)vr / (1 << 24); 4603 track->mHasVolumeController = true; 4604 } else { 4605 // force no volume ramp when volume controller was just disabled or removed 4606 // from effect chain to avoid volume spike 4607 if (track->mHasVolumeController) { 4608 param = AudioMixer::VOLUME; 4609 } 4610 track->mHasVolumeController = false; 4611 } 4612 4613 // For dedicated VoIP outputs, let the HAL apply the stream volume. Track volume is 4614 // still applied by the mixer. 4615 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) { 4616 v = mStreamTypes[track->streamType()].mute ? 0.0f : v; 4617 if (v != mLeftVolFloat) { 4618 status_t result = mOutput->stream->setVolume(v, v); 4619 ALOGE_IF(result != OK, "Error when setting output stream volume: %d", result); 4620 if (result == OK) { 4621 mLeftVolFloat = v; 4622 } 4623 } 4624 // if stream volume was successfully sent to the HAL, mLeftVolFloat == v here and we 4625 // remove stream volume contribution from software volume. 4626 if (v != 0.0f && mLeftVolFloat == v) { 4627 vlf = min(1.0f, vlf / v); 4628 vrf = min(1.0f, vrf / v); 4629 vaf = min(1.0f, vaf / v); 4630 } 4631 } 4632 // XXX: these things DON'T need to be done each time 4633 mAudioMixer->setBufferProvider(name, track); 4634 mAudioMixer->enable(name); 4635 4636 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4637 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4638 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4639 mAudioMixer->setParameter( 4640 name, 4641 AudioMixer::TRACK, 4642 AudioMixer::FORMAT, (void *)track->format()); 4643 mAudioMixer->setParameter( 4644 name, 4645 AudioMixer::TRACK, 4646 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4647 mAudioMixer->setParameter( 4648 name, 4649 AudioMixer::TRACK, 4650 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4651 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4652 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4653 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4654 if (reqSampleRate == 0) { 4655 reqSampleRate = mSampleRate; 4656 } else if (reqSampleRate > maxSampleRate) { 4657 reqSampleRate = maxSampleRate; 4658 } 4659 mAudioMixer->setParameter( 4660 name, 4661 AudioMixer::RESAMPLE, 4662 AudioMixer::SAMPLE_RATE, 4663 (void *)(uintptr_t)reqSampleRate); 4664 4665 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4666 mAudioMixer->setParameter( 4667 name, 4668 AudioMixer::TIMESTRETCH, 4669 AudioMixer::PLAYBACK_RATE, 4670 &playbackRate); 4671 4672 /* 4673 * Select the appropriate output buffer for the track. 4674 * 4675 * Tracks with effects go into their own effects chain buffer 4676 * and from there into either mEffectBuffer or mSinkBuffer. 4677 * 4678 * Other tracks can use mMixerBuffer for higher precision 4679 * channel accumulation. If this buffer is enabled 4680 * (mMixerBufferEnabled true), then selected tracks will accumulate 4681 * into it. 4682 * 4683 */ 4684 if (mMixerBufferEnabled 4685 && (track->mainBuffer() == mSinkBuffer 4686 || track->mainBuffer() == mMixerBuffer)) { 4687 mAudioMixer->setParameter( 4688 name, 4689 AudioMixer::TRACK, 4690 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4691 mAudioMixer->setParameter( 4692 name, 4693 AudioMixer::TRACK, 4694 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4695 // TODO: override track->mainBuffer()? 4696 mMixerBufferValid = true; 4697 } else { 4698 mAudioMixer->setParameter( 4699 name, 4700 AudioMixer::TRACK, 4701 AudioMixer::MIXER_FORMAT, (void *)EFFECT_BUFFER_FORMAT); 4702 mAudioMixer->setParameter( 4703 name, 4704 AudioMixer::TRACK, 4705 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4706 } 4707 mAudioMixer->setParameter( 4708 name, 4709 AudioMixer::TRACK, 4710 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4711 4712 // reset retry count 4713 track->mRetryCount = kMaxTrackRetries; 4714 4715 // If one track is ready, set the mixer ready if: 4716 // - the mixer was not ready during previous round OR 4717 // - no other track is not ready 4718 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4719 mixerStatus != MIXER_TRACKS_ENABLED) { 4720 mixerStatus = MIXER_TRACKS_READY; 4721 } 4722 } else { 4723 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4724 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4725 track, framesReady, desiredFrames); 4726 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4727 } else { 4728 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4729 } 4730 4731 // clear effect chain input buffer if an active track underruns to avoid sending 4732 // previous audio buffer again to effects 4733 chain = getEffectChain_l(track->sessionId()); 4734 if (chain != 0) { 4735 chain->clearInputBuffer(); 4736 } 4737 4738 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4739 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4740 track->isStopped() || track->isPaused()) { 4741 // We have consumed all the buffers of this track. 4742 // Remove it from the list of active tracks. 4743 // TODO: use actual buffer filling status instead of latency when available from 4744 // audio HAL 4745 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4746 int64_t framesWritten = mBytesWritten / mFrameSize; 4747 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4748 if (track->isStopped()) { 4749 track->reset(); 4750 } 4751 tracksToRemove->add(track); 4752 } 4753 } else { 4754 // No buffers for this track. Give it a few chances to 4755 // fill a buffer, then remove it from active list. 4756 if (--(track->mRetryCount) <= 0) { 4757 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4758 tracksToRemove->add(track); 4759 // indicate to client process that the track was disabled because of underrun; 4760 // it will then automatically call start() when data is available 4761 track->disable(); 4762 // If one track is not ready, mark the mixer also not ready if: 4763 // - the mixer was ready during previous round OR 4764 // - no other track is ready 4765 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4766 mixerStatus != MIXER_TRACKS_READY) { 4767 mixerStatus = MIXER_TRACKS_ENABLED; 4768 } 4769 } 4770 mAudioMixer->disable(name); 4771 } 4772 4773 } // local variable scope to avoid goto warning 4774 4775 } 4776 4777 // Push the new FastMixer state if necessary 4778 bool pauseAudioWatchdog = false; 4779 if (didModify) { 4780 state->mFastTracksGen++; 4781 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4782 if (kUseFastMixer == FastMixer_Dynamic && 4783 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4784 state->mCommand = FastMixerState::COLD_IDLE; 4785 state->mColdFutexAddr = &mFastMixerFutex; 4786 state->mColdGen++; 4787 mFastMixerFutex = 0; 4788 if (kUseFastMixer == FastMixer_Dynamic) { 4789 mNormalSink = mOutputSink; 4790 } 4791 // If we go into cold idle, need to wait for acknowledgement 4792 // so that fast mixer stops doing I/O. 4793 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4794 pauseAudioWatchdog = true; 4795 } 4796 } 4797 if (sq != NULL) { 4798 sq->end(didModify); 4799 // No need to block if the FastMixer is in COLD_IDLE as the FastThread 4800 // is not active. (We BLOCK_UNTIL_ACKED when entering COLD_IDLE 4801 // when bringing the output sink into standby.) 4802 // 4803 // We will get the latest FastMixer state when we come out of COLD_IDLE. 4804 // 4805 // This occurs with BT suspend when we idle the FastMixer with 4806 // active tracks, which may be added or removed. 4807 sq->push(coldIdle ? FastMixerStateQueue::BLOCK_NEVER : block); 4808 } 4809 #ifdef AUDIO_WATCHDOG 4810 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4811 mAudioWatchdog->pause(); 4812 } 4813 #endif 4814 4815 // Now perform the deferred reset on fast tracks that have stopped 4816 while (resetMask != 0) { 4817 size_t i = __builtin_ctz(resetMask); 4818 ALOG_ASSERT(i < count); 4819 resetMask &= ~(1 << i); 4820 sp<Track> track = mActiveTracks[i]; 4821 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4822 track->reset(); 4823 } 4824 4825 // Track destruction may occur outside of threadLoop once it is removed from active tracks. 4826 // Ensure the AudioMixer doesn't have a raw "buffer provider" pointer to the track if 4827 // it ceases to be active, to allow safe removal from the AudioMixer at the start 4828 // of prepareTracks_l(); this releases any outstanding buffer back to the track. 4829 // See also the implementation of destroyTrack_l(). 4830 for (const auto &track : *tracksToRemove) { 4831 const int name = track->name(); 4832 if (mAudioMixer->exists(name)) { // Normal tracks here, fast tracks in FastMixer. 4833 mAudioMixer->setBufferProvider(name, nullptr /* bufferProvider */); 4834 } 4835 } 4836 4837 // remove all the tracks that need to be... 4838 removeTracks_l(*tracksToRemove); 4839 4840 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4841 mEffectBufferValid = true; 4842 } 4843 4844 if (mEffectBufferValid) { 4845 // as long as there are effects we should clear the effects buffer, to avoid 4846 // passing a non-clean buffer to the effect chain 4847 memset(mEffectBuffer, 0, mEffectBufferSize); 4848 } 4849 // sink or mix buffer must be cleared if all tracks are connected to an 4850 // effect chain as in this case the mixer will not write to the sink or mix buffer 4851 // and track effects will accumulate into it 4852 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4853 (mixedTracks == 0 && fastTracks > 0))) { 4854 // FIXME as a performance optimization, should remember previous zero status 4855 if (mMixerBufferValid) { 4856 memset(mMixerBuffer, 0, mMixerBufferSize); 4857 // TODO: In testing, mSinkBuffer below need not be cleared because 4858 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4859 // after mixing. 4860 // 4861 // To enforce this guarantee: 4862 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4863 // (mixedTracks == 0 && fastTracks > 0)) 4864 // must imply MIXER_TRACKS_READY. 4865 // Later, we may clear buffers regardless, and skip much of this logic. 4866 } 4867 // FIXME as a performance optimization, should remember previous zero status 4868 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4869 } 4870 4871 // if any fast tracks, then status is ready 4872 mMixerStatusIgnoringFastTracks = mixerStatus; 4873 if (fastTracks > 0) { 4874 mixerStatus = MIXER_TRACKS_READY; 4875 } 4876 return mixerStatus; 4877 } 4878 4879 // trackCountForUid_l() must be called with ThreadBase::mLock held 4880 uint32_t AudioFlinger::PlaybackThread::trackCountForUid_l(uid_t uid) const 4881 { 4882 uint32_t trackCount = 0; 4883 for (size_t i = 0; i < mTracks.size() ; i++) { 4884 if (mTracks[i]->uid() == uid) { 4885 trackCount++; 4886 } 4887 } 4888 return trackCount; 4889 } 4890 4891 // isTrackAllowed_l() must be called with ThreadBase::mLock held 4892 bool AudioFlinger::MixerThread::isTrackAllowed_l( 4893 audio_channel_mask_t channelMask, audio_format_t format, 4894 audio_session_t sessionId, uid_t uid) const 4895 { 4896 if (!PlaybackThread::isTrackAllowed_l(channelMask, format, sessionId, uid)) { 4897 return false; 4898 } 4899 // Check validity as we don't call AudioMixer::create() here. 4900 if (!AudioMixer::isValidFormat(format)) { 4901 ALOGW("%s: invalid format: %#x", __func__, format); 4902 return false; 4903 } 4904 if (!AudioMixer::isValidChannelMask(channelMask)) { 4905 ALOGW("%s: invalid channelMask: %#x", __func__, channelMask); 4906 return false; 4907 } 4908 return true; 4909 } 4910 4911 // checkForNewParameter_l() must be called with ThreadBase::mLock held 4912 bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4913 status_t& status) 4914 { 4915 bool reconfig = false; 4916 bool a2dpDeviceChanged = false; 4917 4918 status = NO_ERROR; 4919 4920 AutoPark<FastMixer> park(mFastMixer); 4921 4922 AudioParameter param = AudioParameter(keyValuePair); 4923 int value; 4924 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4925 reconfig = true; 4926 } 4927 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4928 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4929 status = BAD_VALUE; 4930 } else { 4931 // no need to save value, since it's constant 4932 reconfig = true; 4933 } 4934 } 4935 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4936 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4937 status = BAD_VALUE; 4938 } else { 4939 // no need to save value, since it's constant 4940 reconfig = true; 4941 } 4942 } 4943 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4944 // do not accept frame count changes if tracks are open as the track buffer 4945 // size depends on frame count and correct behavior would not be guaranteed 4946 // if frame count is changed after track creation 4947 if (!mTracks.isEmpty()) { 4948 status = INVALID_OPERATION; 4949 } else { 4950 reconfig = true; 4951 } 4952 } 4953 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4954 #ifdef ADD_BATTERY_DATA 4955 // when changing the audio output device, call addBatteryData to notify 4956 // the change 4957 if (mOutDevice != value) { 4958 uint32_t params = 0; 4959 // check whether speaker is on 4960 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4961 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4962 } 4963 4964 audio_devices_t deviceWithoutSpeaker 4965 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4966 // check if any other device (except speaker) is on 4967 if (value & deviceWithoutSpeaker) { 4968 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4969 } 4970 4971 if (params != 0) { 4972 addBatteryData(params); 4973 } 4974 } 4975 #endif 4976 4977 // forward device change to effects that have requested to be 4978 // aware of attached audio device. 4979 if (value != AUDIO_DEVICE_NONE) { 4980 a2dpDeviceChanged = 4981 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4982 mOutDevice = value; 4983 for (size_t i = 0; i < mEffectChains.size(); i++) { 4984 mEffectChains[i]->setDevice_l(mOutDevice); 4985 } 4986 } 4987 } 4988 4989 if (status == NO_ERROR) { 4990 status = mOutput->stream->setParameters(keyValuePair); 4991 if (!mStandby && status == INVALID_OPERATION) { 4992 mOutput->standby(); 4993 mStandby = true; 4994 mBytesWritten = 0; 4995 status = mOutput->stream->setParameters(keyValuePair); 4996 } 4997 if (status == NO_ERROR && reconfig) { 4998 readOutputParameters_l(); 4999 delete mAudioMixer; 5000 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 5001 for (const auto &track : mTracks) { 5002 const int name = track->name(); 5003 status_t status = mAudioMixer->create( 5004 name, 5005 track->mChannelMask, 5006 track->mFormat, 5007 track->mSessionId); 5008 ALOGW_IF(status != NO_ERROR, 5009 "%s: cannot create track name" 5010 " %d, mask %#x, format %#x, sessionId %d in AudioMixer", 5011 __func__, 5012 name, track->mChannelMask, track->mFormat, track->mSessionId); 5013 } 5014 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5015 } 5016 } 5017 5018 return reconfig || a2dpDeviceChanged; 5019 } 5020 5021 5022 void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 5023 { 5024 PlaybackThread::dumpInternals(fd, args); 5025 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 5026 dprintf(fd, " AudioMixer tracks: %s\n", mAudioMixer->trackNames().c_str()); 5027 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 5028 5029 if (hasFastMixer()) { 5030 dprintf(fd, " FastMixer thread %p tid=%d", mFastMixer.get(), mFastMixer->getTid()); 5031 5032 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 5033 // while we are dumping it. It may be inconsistent, but it won't mutate! 5034 // This is a large object so we place it on the heap. 5035 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 5036 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 5037 copy->dump(fd); 5038 delete copy; 5039 5040 #ifdef STATE_QUEUE_DUMP 5041 // Similar for state queue 5042 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 5043 observerCopy.dump(fd); 5044 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 5045 mutatorCopy.dump(fd); 5046 #endif 5047 5048 #ifdef AUDIO_WATCHDOG 5049 if (mAudioWatchdog != 0) { 5050 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 5051 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 5052 wdCopy.dump(fd); 5053 } 5054 #endif 5055 5056 } else { 5057 dprintf(fd, " No FastMixer\n"); 5058 } 5059 5060 #ifdef TEE_SINK 5061 // Write the tee output to a .wav file 5062 dumpTee(fd, mTeeSource, mId, 'M'); 5063 #endif 5064 5065 } 5066 5067 uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 5068 { 5069 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 5070 } 5071 5072 uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 5073 { 5074 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 5075 } 5076 5077 void AudioFlinger::MixerThread::cacheParameters_l() 5078 { 5079 PlaybackThread::cacheParameters_l(); 5080 5081 // FIXME: Relaxed timing because of a certain device that can't meet latency 5082 // Should be reduced to 2x after the vendor fixes the driver issue 5083 // increase threshold again due to low power audio mode. The way this warning 5084 // threshold is calculated and its usefulness should be reconsidered anyway. 5085 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 5086 } 5087 5088 // ---------------------------------------------------------------------------- 5089 5090 AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 5091 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 5092 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 5093 { 5094 } 5095 5096 AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 5097 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 5098 ThreadBase::type_t type, bool systemReady) 5099 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 5100 , mVolumeShaperActive(false) 5101 { 5102 } 5103 5104 AudioFlinger::DirectOutputThread::~DirectOutputThread() 5105 { 5106 } 5107 5108 void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 5109 { 5110 float left, right; 5111 5112 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 5113 left = right = 0; 5114 } else { 5115 float typeVolume = mStreamTypes[track->streamType()].volume; 5116 float v = mMasterVolume * typeVolume; 5117 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy; 5118 5119 // Get volumeshaper scaling 5120 std::pair<float /* volume */, bool /* active */> 5121 vh = track->getVolumeHandler()->getVolume( 5122 track->mAudioTrackServerProxy->framesReleased()); 5123 v *= vh.first; 5124 mVolumeShaperActive = vh.second; 5125 5126 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 5127 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 5128 if (left > GAIN_FLOAT_UNITY) { 5129 left = GAIN_FLOAT_UNITY; 5130 } 5131 left *= v; 5132 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 5133 if (right > GAIN_FLOAT_UNITY) { 5134 right = GAIN_FLOAT_UNITY; 5135 } 5136 right *= v; 5137 } 5138 5139 if (lastTrack) { 5140 track->setFinalVolume((left + right) / 2.f); 5141 if (left != mLeftVolFloat || right != mRightVolFloat) { 5142 mLeftVolFloat = left; 5143 mRightVolFloat = right; 5144 5145 // Convert volumes from float to 8.24 5146 uint32_t vl = (uint32_t)(left * (1 << 24)); 5147 uint32_t vr = (uint32_t)(right * (1 << 24)); 5148 5149 // Delegate volume control to effect in track effect chain if needed 5150 // only one effect chain can be present on DirectOutputThread, so if 5151 // there is one, the track is connected to it 5152 if (!mEffectChains.isEmpty()) { 5153 mEffectChains[0]->setVolume_l(&vl, &vr); 5154 left = (float)vl / (1 << 24); 5155 right = (float)vr / (1 << 24); 5156 } 5157 status_t result = mOutput->stream->setVolume(left, right); 5158 ALOGE_IF(result != OK, "Error when setting output stream volume: %d", result); 5159 } 5160 } 5161 } 5162 5163 void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 5164 { 5165 sp<Track> previousTrack = mPreviousTrack.promote(); 5166 sp<Track> latestTrack = mActiveTracks.getLatest(); 5167 5168 if (previousTrack != 0 && latestTrack != 0) { 5169 if (mType == DIRECT) { 5170 if (previousTrack.get() != latestTrack.get()) { 5171 mFlushPending = true; 5172 } 5173 } else /* mType == OFFLOAD */ { 5174 if (previousTrack->sessionId() != latestTrack->sessionId()) { 5175 mFlushPending = true; 5176 } 5177 } 5178 } 5179 PlaybackThread::onAddNewTrack_l(); 5180 } 5181 5182 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 5183 Vector< sp<Track> > *tracksToRemove 5184 ) 5185 { 5186 size_t count = mActiveTracks.size(); 5187 mixer_state mixerStatus = MIXER_IDLE; 5188 bool doHwPause = false; 5189 bool doHwResume = false; 5190 5191 // find out which tracks need to be processed 5192 for (const sp<Track> &t : mActiveTracks) { 5193 if (t->isInvalid()) { 5194 ALOGW("An invalidated track shouldn't be in active list"); 5195 tracksToRemove->add(t); 5196 continue; 5197 } 5198 5199 Track* const track = t.get(); 5200 #ifdef VERY_VERY_VERBOSE_LOGGING 5201 audio_track_cblk_t* cblk = track->cblk(); 5202 #endif 5203 // Only consider last track started for volume and mixer state control. 5204 // In theory an older track could underrun and restart after the new one starts 5205 // but as we only care about the transition phase between two tracks on a 5206 // direct output, it is not a problem to ignore the underrun case. 5207 sp<Track> l = mActiveTracks.getLatest(); 5208 bool last = l.get() == track; 5209 5210 if (track->isPausing()) { 5211 track->setPaused(); 5212 if (mHwSupportsPause && last && !mHwPaused) { 5213 doHwPause = true; 5214 mHwPaused = true; 5215 } 5216 tracksToRemove->add(track); 5217 } else if (track->isFlushPending()) { 5218 track->flushAck(); 5219 if (last) { 5220 mFlushPending = true; 5221 } 5222 } else if (track->isResumePending()) { 5223 track->resumeAck(); 5224 if (last) { 5225 mLeftVolFloat = mRightVolFloat = -1.0; 5226 if (mHwPaused) { 5227 doHwResume = true; 5228 mHwPaused = false; 5229 } 5230 } 5231 } 5232 5233 // The first time a track is added we wait 5234 // for all its buffers to be filled before processing it. 5235 // Allow draining the buffer in case the client 5236 // app does not call stop() and relies on underrun to stop: 5237 // hence the test on (track->mRetryCount > 1). 5238 // If retryCount<=1 then track is about to underrun and be removed. 5239 // Do not use a high threshold for compressed audio. 5240 uint32_t minFrames; 5241 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 5242 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 5243 minFrames = mNormalFrameCount; 5244 } else { 5245 minFrames = 1; 5246 } 5247 5248 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 5249 !track->isStopping_2() && !track->isStopped()) 5250 { 5251 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 5252 5253 if (track->mFillingUpStatus == Track::FS_FILLED) { 5254 track->mFillingUpStatus = Track::FS_ACTIVE; 5255 if (last) { 5256 // make sure processVolume_l() will apply new volume even if 0 5257 mLeftVolFloat = mRightVolFloat = -1.0; 5258 } 5259 if (!mHwSupportsPause) { 5260 track->resumeAck(); 5261 } 5262 } 5263 5264 // compute volume for this track 5265 processVolume_l(track, last); 5266 if (last) { 5267 sp<Track> previousTrack = mPreviousTrack.promote(); 5268 if (previousTrack != 0) { 5269 if (track != previousTrack.get()) { 5270 // Flush any data still being written from last track 5271 mBytesRemaining = 0; 5272 // Invalidate previous track to force a seek when resuming. 5273 previousTrack->invalidate(); 5274 } 5275 } 5276 mPreviousTrack = track; 5277 5278 // reset retry count 5279 track->mRetryCount = kMaxTrackRetriesDirect; 5280 mActiveTrack = t; 5281 mixerStatus = MIXER_TRACKS_READY; 5282 if (mHwPaused) { 5283 doHwResume = true; 5284 mHwPaused = false; 5285 } 5286 } 5287 } else { 5288 // clear effect chain input buffer if the last active track started underruns 5289 // to avoid sending previous audio buffer again to effects 5290 if (!mEffectChains.isEmpty() && last) { 5291 mEffectChains[0]->clearInputBuffer(); 5292 } 5293 if (track->isStopping_1()) { 5294 track->mState = TrackBase::STOPPING_2; 5295 if (last && mHwPaused) { 5296 doHwResume = true; 5297 mHwPaused = false; 5298 } 5299 } 5300 if ((track->sharedBuffer() != 0) || track->isStopped() || 5301 track->isStopping_2() || track->isPaused()) { 5302 // We have consumed all the buffers of this track. 5303 // Remove it from the list of active tracks. 5304 size_t audioHALFrames; 5305 if (audio_has_proportional_frames(mFormat)) { 5306 audioHALFrames = (latency_l() * mSampleRate) / 1000; 5307 } else { 5308 audioHALFrames = 0; 5309 } 5310 5311 int64_t framesWritten = mBytesWritten / mFrameSize; 5312 if (mStandby || !last || 5313 track->presentationComplete(framesWritten, audioHALFrames)) { 5314 if (track->isStopping_2()) { 5315 track->mState = TrackBase::STOPPED; 5316 } 5317 if (track->isStopped()) { 5318 track->reset(); 5319 } 5320 tracksToRemove->add(track); 5321 } 5322 } else { 5323 // No buffers for this track. Give it a few chances to 5324 // fill a buffer, then remove it from active list. 5325 // Only consider last track started for mixer state control 5326 if (--(track->mRetryCount) <= 0) { 5327 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 5328 tracksToRemove->add(track); 5329 // indicate to client process that the track was disabled because of underrun; 5330 // it will then automatically call start() when data is available 5331 track->disable(); 5332 } else if (last) { 5333 ALOGW("pause because of UNDERRUN, framesReady = %zu," 5334 "minFrames = %u, mFormat = %#x", 5335 track->framesReady(), minFrames, mFormat); 5336 mixerStatus = MIXER_TRACKS_ENABLED; 5337 if (mHwSupportsPause && !mHwPaused && !mStandby) { 5338 doHwPause = true; 5339 mHwPaused = true; 5340 } 5341 } 5342 } 5343 } 5344 } 5345 5346 // if an active track did not command a flush, check for pending flush on stopped tracks 5347 if (!mFlushPending) { 5348 for (size_t i = 0; i < mTracks.size(); i++) { 5349 if (mTracks[i]->isFlushPending()) { 5350 mTracks[i]->flushAck(); 5351 mFlushPending = true; 5352 } 5353 } 5354 } 5355 5356 // make sure the pause/flush/resume sequence is executed in the right order. 5357 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5358 // before flush and then resume HW. This can happen in case of pause/flush/resume 5359 // if resume is received before pause is executed. 5360 if (mHwSupportsPause && !mStandby && 5361 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5362 status_t result = mOutput->stream->pause(); 5363 ALOGE_IF(result != OK, "Error when pausing output stream: %d", result); 5364 } 5365 if (mFlushPending) { 5366 flushHw_l(); 5367 } 5368 if (mHwSupportsPause && !mStandby && doHwResume) { 5369 status_t result = mOutput->stream->resume(); 5370 ALOGE_IF(result != OK, "Error when resuming output stream: %d", result); 5371 } 5372 // remove all the tracks that need to be... 5373 removeTracks_l(*tracksToRemove); 5374 5375 return mixerStatus; 5376 } 5377 5378 void AudioFlinger::DirectOutputThread::threadLoop_mix() 5379 { 5380 size_t frameCount = mFrameCount; 5381 int8_t *curBuf = (int8_t *)mSinkBuffer; 5382 // output audio to hardware 5383 while (frameCount) { 5384 AudioBufferProvider::Buffer buffer; 5385 buffer.frameCount = frameCount; 5386 status_t status = mActiveTrack->getNextBuffer(&buffer); 5387 if (status != NO_ERROR || buffer.raw == NULL) { 5388 // no need to pad with 0 for compressed audio 5389 if (audio_has_proportional_frames(mFormat)) { 5390 memset(curBuf, 0, frameCount * mFrameSize); 5391 } 5392 break; 5393 } 5394 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 5395 frameCount -= buffer.frameCount; 5396 curBuf += buffer.frameCount * mFrameSize; 5397 mActiveTrack->releaseBuffer(&buffer); 5398 } 5399 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 5400 mSleepTimeUs = 0; 5401 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5402 mActiveTrack.clear(); 5403 } 5404 5405 void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 5406 { 5407 // do not write to HAL when paused 5408 if (mHwPaused || (usesHwAvSync() && mStandby)) { 5409 mSleepTimeUs = mIdleSleepTimeUs; 5410 return; 5411 } 5412 if (mSleepTimeUs == 0) { 5413 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5414 mSleepTimeUs = mActiveSleepTimeUs; 5415 } else { 5416 mSleepTimeUs = mIdleSleepTimeUs; 5417 } 5418 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 5419 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 5420 mSleepTimeUs = 0; 5421 } 5422 } 5423 5424 void AudioFlinger::DirectOutputThread::threadLoop_exit() 5425 { 5426 { 5427 Mutex::Autolock _l(mLock); 5428 for (size_t i = 0; i < mTracks.size(); i++) { 5429 if (mTracks[i]->isFlushPending()) { 5430 mTracks[i]->flushAck(); 5431 mFlushPending = true; 5432 } 5433 } 5434 if (mFlushPending) { 5435 flushHw_l(); 5436 } 5437 } 5438 PlaybackThread::threadLoop_exit(); 5439 } 5440 5441 // must be called with thread mutex locked 5442 bool AudioFlinger::DirectOutputThread::shouldStandby_l() 5443 { 5444 bool trackPaused = false; 5445 bool trackStopped = false; 5446 5447 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 5448 return !mStandby; 5449 } 5450 5451 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 5452 // after a timeout and we will enter standby then. 5453 if (mTracks.size() > 0) { 5454 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5455 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5456 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5457 } 5458 5459 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5460 } 5461 5462 // checkForNewParameter_l() must be called with ThreadBase::mLock held 5463 bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5464 status_t& status) 5465 { 5466 bool reconfig = false; 5467 bool a2dpDeviceChanged = false; 5468 5469 status = NO_ERROR; 5470 5471 AudioParameter param = AudioParameter(keyValuePair); 5472 int value; 5473 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5474 // forward device change to effects that have requested to be 5475 // aware of attached audio device. 5476 if (value != AUDIO_DEVICE_NONE) { 5477 a2dpDeviceChanged = 5478 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5479 mOutDevice = value; 5480 for (size_t i = 0; i < mEffectChains.size(); i++) { 5481 mEffectChains[i]->setDevice_l(mOutDevice); 5482 } 5483 } 5484 } 5485 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5486 // do not accept frame count changes if tracks are open as the track buffer 5487 // size depends on frame count and correct behavior would not be garantied 5488 // if frame count is changed after track creation 5489 if (!mTracks.isEmpty()) { 5490 status = INVALID_OPERATION; 5491 } else { 5492 reconfig = true; 5493 } 5494 } 5495 if (status == NO_ERROR) { 5496 status = mOutput->stream->setParameters(keyValuePair); 5497 if (!mStandby && status == INVALID_OPERATION) { 5498 mOutput->standby(); 5499 mStandby = true; 5500 mBytesWritten = 0; 5501 status = mOutput->stream->setParameters(keyValuePair); 5502 } 5503 if (status == NO_ERROR && reconfig) { 5504 readOutputParameters_l(); 5505 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5506 } 5507 } 5508 5509 return reconfig || a2dpDeviceChanged; 5510 } 5511 5512 uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5513 { 5514 uint32_t time; 5515 if (audio_has_proportional_frames(mFormat)) { 5516 time = PlaybackThread::activeSleepTimeUs(); 5517 } else { 5518 time = kDirectMinSleepTimeUs; 5519 } 5520 return time; 5521 } 5522 5523 uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5524 { 5525 uint32_t time; 5526 if (audio_has_proportional_frames(mFormat)) { 5527 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5528 } else { 5529 time = kDirectMinSleepTimeUs; 5530 } 5531 return time; 5532 } 5533 5534 uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5535 { 5536 uint32_t time; 5537 if (audio_has_proportional_frames(mFormat)) { 5538 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5539 } else { 5540 time = kDirectMinSleepTimeUs; 5541 } 5542 return time; 5543 } 5544 5545 void AudioFlinger::DirectOutputThread::cacheParameters_l() 5546 { 5547 PlaybackThread::cacheParameters_l(); 5548 5549 // use shorter standby delay as on normal output to release 5550 // hardware resources as soon as possible 5551 // no delay on outputs with HW A/V sync 5552 if (usesHwAvSync()) { 5553 mStandbyDelayNs = 0; 5554 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5555 mStandbyDelayNs = kOffloadStandbyDelayNs; 5556 } else { 5557 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5558 } 5559 } 5560 5561 void AudioFlinger::DirectOutputThread::flushHw_l() 5562 { 5563 mOutput->flush(); 5564 mHwPaused = false; 5565 mFlushPending = false; 5566 } 5567 5568 int64_t AudioFlinger::DirectOutputThread::computeWaitTimeNs_l() const { 5569 // If a VolumeShaper is active, we must wake up periodically to update volume. 5570 const int64_t NS_PER_MS = 1000000; 5571 return mVolumeShaperActive ? 5572 kMinNormalSinkBufferSizeMs * NS_PER_MS : PlaybackThread::computeWaitTimeNs_l(); 5573 } 5574 5575 // ---------------------------------------------------------------------------- 5576 5577 AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5578 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5579 : Thread(false /*canCallJava*/), 5580 mPlaybackThread(playbackThread), 5581 mWriteAckSequence(0), 5582 mDrainSequence(0), 5583 mAsyncError(false) 5584 { 5585 } 5586 5587 AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5588 { 5589 } 5590 5591 void AudioFlinger::AsyncCallbackThread::onFirstRef() 5592 { 5593 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5594 } 5595 5596 bool AudioFlinger::AsyncCallbackThread::threadLoop() 5597 { 5598 while (!exitPending()) { 5599 uint32_t writeAckSequence; 5600 uint32_t drainSequence; 5601 bool asyncError; 5602 5603 { 5604 Mutex::Autolock _l(mLock); 5605 while (!((mWriteAckSequence & 1) || 5606 (mDrainSequence & 1) || 5607 mAsyncError || 5608 exitPending())) { 5609 mWaitWorkCV.wait(mLock); 5610 } 5611 5612 if (exitPending()) { 5613 break; 5614 } 5615 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5616 mWriteAckSequence, mDrainSequence); 5617 writeAckSequence = mWriteAckSequence; 5618 mWriteAckSequence &= ~1; 5619 drainSequence = mDrainSequence; 5620 mDrainSequence &= ~1; 5621 asyncError = mAsyncError; 5622 mAsyncError = false; 5623 } 5624 { 5625 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5626 if (playbackThread != 0) { 5627 if (writeAckSequence & 1) { 5628 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5629 } 5630 if (drainSequence & 1) { 5631 playbackThread->resetDraining(drainSequence >> 1); 5632 } 5633 if (asyncError) { 5634 playbackThread->onAsyncError(); 5635 } 5636 } 5637 } 5638 } 5639 return false; 5640 } 5641 5642 void AudioFlinger::AsyncCallbackThread::exit() 5643 { 5644 ALOGV("AsyncCallbackThread::exit"); 5645 Mutex::Autolock _l(mLock); 5646 requestExit(); 5647 mWaitWorkCV.broadcast(); 5648 } 5649 5650 void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5651 { 5652 Mutex::Autolock _l(mLock); 5653 // bit 0 is cleared 5654 mWriteAckSequence = sequence << 1; 5655 } 5656 5657 void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5658 { 5659 Mutex::Autolock _l(mLock); 5660 // ignore unexpected callbacks 5661 if (mWriteAckSequence & 2) { 5662 mWriteAckSequence |= 1; 5663 mWaitWorkCV.signal(); 5664 } 5665 } 5666 5667 void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5668 { 5669 Mutex::Autolock _l(mLock); 5670 // bit 0 is cleared 5671 mDrainSequence = sequence << 1; 5672 } 5673 5674 void AudioFlinger::AsyncCallbackThread::resetDraining() 5675 { 5676 Mutex::Autolock _l(mLock); 5677 // ignore unexpected callbacks 5678 if (mDrainSequence & 2) { 5679 mDrainSequence |= 1; 5680 mWaitWorkCV.signal(); 5681 } 5682 } 5683 5684 void AudioFlinger::AsyncCallbackThread::setAsyncError() 5685 { 5686 Mutex::Autolock _l(mLock); 5687 mAsyncError = true; 5688 mWaitWorkCV.signal(); 5689 } 5690 5691 5692 // ---------------------------------------------------------------------------- 5693 AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5694 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5695 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5696 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true), 5697 mOffloadUnderrunPosition(~0LL) 5698 { 5699 //FIXME: mStandby should be set to true by ThreadBase constructo 5700 mStandby = true; 5701 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5702 } 5703 5704 void AudioFlinger::OffloadThread::threadLoop_exit() 5705 { 5706 if (mFlushPending || mHwPaused) { 5707 // If a flush is pending or track was paused, just discard buffered data 5708 flushHw_l(); 5709 } else { 5710 mMixerStatus = MIXER_DRAIN_ALL; 5711 threadLoop_drain(); 5712 } 5713 if (mUseAsyncWrite) { 5714 ALOG_ASSERT(mCallbackThread != 0); 5715 mCallbackThread->exit(); 5716 } 5717 PlaybackThread::threadLoop_exit(); 5718 } 5719 5720 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5721 Vector< sp<Track> > *tracksToRemove 5722 ) 5723 { 5724 size_t count = mActiveTracks.size(); 5725 5726 mixer_state mixerStatus = MIXER_IDLE; 5727 bool doHwPause = false; 5728 bool doHwResume = false; 5729 5730 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5731 5732 // find out which tracks need to be processed 5733 for (const sp<Track> &t : mActiveTracks) { 5734 Track* const track = t.get(); 5735 #ifdef VERY_VERY_VERBOSE_LOGGING 5736 audio_track_cblk_t* cblk = track->cblk(); 5737 #endif 5738 // Only consider last track started for volume and mixer state control. 5739 // In theory an older track could underrun and restart after the new one starts 5740 // but as we only care about the transition phase between two tracks on a 5741 // direct output, it is not a problem to ignore the underrun case. 5742 sp<Track> l = mActiveTracks.getLatest(); 5743 bool last = l.get() == track; 5744 5745 if (track->isInvalid()) { 5746 ALOGW("An invalidated track shouldn't be in active list"); 5747 tracksToRemove->add(track); 5748 continue; 5749 } 5750 5751 if (track->mState == TrackBase::IDLE) { 5752 ALOGW("An idle track shouldn't be in active list"); 5753 continue; 5754 } 5755 5756 if (track->isPausing()) { 5757 track->setPaused(); 5758 if (last) { 5759 if (mHwSupportsPause && !mHwPaused) { 5760 doHwPause = true; 5761 mHwPaused = true; 5762 } 5763 // If we were part way through writing the mixbuffer to 5764 // the HAL we must save this until we resume 5765 // BUG - this will be wrong if a different track is made active, 5766 // in that case we want to discard the pending data in the 5767 // mixbuffer and tell the client to present it again when the 5768 // track is resumed 5769 mPausedWriteLength = mCurrentWriteLength; 5770 mPausedBytesRemaining = mBytesRemaining; 5771 mBytesRemaining = 0; // stop writing 5772 } 5773 tracksToRemove->add(track); 5774 } else if (track->isFlushPending()) { 5775 if (track->isStopping_1()) { 5776 track->mRetryCount = kMaxTrackStopRetriesOffload; 5777 } else { 5778 track->mRetryCount = kMaxTrackRetriesOffload; 5779 } 5780 track->flushAck(); 5781 if (last) { 5782 mFlushPending = true; 5783 } 5784 } else if (track->isResumePending()){ 5785 track->resumeAck(); 5786 if (last) { 5787 if (mPausedBytesRemaining) { 5788 // Need to continue write that was interrupted 5789 mCurrentWriteLength = mPausedWriteLength; 5790 mBytesRemaining = mPausedBytesRemaining; 5791 mPausedBytesRemaining = 0; 5792 } 5793 if (mHwPaused) { 5794 doHwResume = true; 5795 mHwPaused = false; 5796 // threadLoop_mix() will handle the case that we need to 5797 // resume an interrupted write 5798 } 5799 // enable write to audio HAL 5800 mSleepTimeUs = 0; 5801 5802 mLeftVolFloat = mRightVolFloat = -1.0; 5803 5804 // Do not handle new data in this iteration even if track->framesReady() 5805 mixerStatus = MIXER_TRACKS_ENABLED; 5806 } 5807 } else if (track->framesReady() && track->isReady() && 5808 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5809 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5810 if (track->mFillingUpStatus == Track::FS_FILLED) { 5811 track->mFillingUpStatus = Track::FS_ACTIVE; 5812 if (last) { 5813 // make sure processVolume_l() will apply new volume even if 0 5814 mLeftVolFloat = mRightVolFloat = -1.0; 5815 } 5816 } 5817 5818 if (last) { 5819 sp<Track> previousTrack = mPreviousTrack.promote(); 5820 if (previousTrack != 0) { 5821 if (track != previousTrack.get()) { 5822 // Flush any data still being written from last track 5823 mBytesRemaining = 0; 5824 if (mPausedBytesRemaining) { 5825 // Last track was paused so we also need to flush saved 5826 // mixbuffer state and invalidate track so that it will 5827 // re-submit that unwritten data when it is next resumed 5828 mPausedBytesRemaining = 0; 5829 // Invalidate is a bit drastic - would be more efficient 5830 // to have a flag to tell client that some of the 5831 // previously written data was lost 5832 previousTrack->invalidate(); 5833 } 5834 // flush data already sent to the DSP if changing audio session as audio 5835 // comes from a different source. Also invalidate previous track to force a 5836 // seek when resuming. 5837 if (previousTrack->sessionId() != track->sessionId()) { 5838 previousTrack->invalidate(); 5839 } 5840 } 5841 } 5842 mPreviousTrack = track; 5843 // reset retry count 5844 if (track->isStopping_1()) { 5845 track->mRetryCount = kMaxTrackStopRetriesOffload; 5846 } else { 5847 track->mRetryCount = kMaxTrackRetriesOffload; 5848 } 5849 mActiveTrack = t; 5850 mixerStatus = MIXER_TRACKS_READY; 5851 } 5852 } else { 5853 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5854 if (track->isStopping_1()) { 5855 if (--(track->mRetryCount) <= 0) { 5856 // Hardware buffer can hold a large amount of audio so we must 5857 // wait for all current track's data to drain before we say 5858 // that the track is stopped. 5859 if (mBytesRemaining == 0) { 5860 // Only start draining when all data in mixbuffer 5861 // has been written 5862 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5863 track->mState = TrackBase::STOPPING_2; // so presentation completes after 5864 // drain do not drain if no data was ever sent to HAL (mStandby == true) 5865 if (last && !mStandby) { 5866 // do not modify drain sequence if we are already draining. This happens 5867 // when resuming from pause after drain. 5868 if ((mDrainSequence & 1) == 0) { 5869 mSleepTimeUs = 0; 5870 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5871 mixerStatus = MIXER_DRAIN_TRACK; 5872 mDrainSequence += 2; 5873 } 5874 if (mHwPaused) { 5875 // It is possible to move from PAUSED to STOPPING_1 without 5876 // a resume so we must ensure hardware is running 5877 doHwResume = true; 5878 mHwPaused = false; 5879 } 5880 } 5881 } 5882 } else if (last) { 5883 ALOGV("stopping1 underrun retries left %d", track->mRetryCount); 5884 mixerStatus = MIXER_TRACKS_ENABLED; 5885 } 5886 } else if (track->isStopping_2()) { 5887 // Drain has completed or we are in standby, signal presentation complete 5888 if (!(mDrainSequence & 1) || !last || mStandby) { 5889 track->mState = TrackBase::STOPPED; 5890 uint32_t latency = 0; 5891 status_t result = mOutput->stream->getLatency(&latency); 5892 ALOGE_IF(result != OK, 5893 "Error when retrieving output stream latency: %d", result); 5894 size_t audioHALFrames = (latency * mSampleRate) / 1000; 5895 int64_t framesWritten = 5896 mBytesWritten / mOutput->getFrameSize(); 5897 track->presentationComplete(framesWritten, audioHALFrames); 5898 track->reset(); 5899 tracksToRemove->add(track); 5900 } 5901 } else { 5902 // No buffers for this track. Give it a few chances to 5903 // fill a buffer, then remove it from active list. 5904 if (--(track->mRetryCount) <= 0) { 5905 bool running = false; 5906 uint64_t position = 0; 5907 struct timespec unused; 5908 // The running check restarts the retry counter at least once. 5909 status_t ret = mOutput->stream->getPresentationPosition(&position, &unused); 5910 if (ret == NO_ERROR && position != mOffloadUnderrunPosition) { 5911 running = true; 5912 mOffloadUnderrunPosition = position; 5913 } 5914 if (ret == NO_ERROR) { 5915 ALOGVV("underrun counter, running(%d): %lld vs %lld", running, 5916 (long long)position, (long long)mOffloadUnderrunPosition); 5917 } 5918 if (running) { // still running, give us more time. 5919 track->mRetryCount = kMaxTrackRetriesOffload; 5920 } else { 5921 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5922 track->name()); 5923 tracksToRemove->add(track); 5924 // tell client process that the track was disabled because of underrun; 5925 // it will then automatically call start() when data is available 5926 track->disable(); 5927 } 5928 } else if (last){ 5929 mixerStatus = MIXER_TRACKS_ENABLED; 5930 } 5931 } 5932 } 5933 // compute volume for this track 5934 processVolume_l(track, last); 5935 } 5936 5937 // make sure the pause/flush/resume sequence is executed in the right order. 5938 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5939 // before flush and then resume HW. This can happen in case of pause/flush/resume 5940 // if resume is received before pause is executed. 5941 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5942 status_t result = mOutput->stream->pause(); 5943 ALOGE_IF(result != OK, "Error when pausing output stream: %d", result); 5944 } 5945 if (mFlushPending) { 5946 flushHw_l(); 5947 } 5948 if (!mStandby && doHwResume) { 5949 status_t result = mOutput->stream->resume(); 5950 ALOGE_IF(result != OK, "Error when resuming output stream: %d", result); 5951 } 5952 5953 // remove all the tracks that need to be... 5954 removeTracks_l(*tracksToRemove); 5955 5956 return mixerStatus; 5957 } 5958 5959 // must be called with thread mutex locked 5960 bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5961 { 5962 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5963 mWriteAckSequence, mDrainSequence); 5964 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5965 return true; 5966 } 5967 return false; 5968 } 5969 5970 bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5971 { 5972 Mutex::Autolock _l(mLock); 5973 return waitingAsyncCallback_l(); 5974 } 5975 5976 void AudioFlinger::OffloadThread::flushHw_l() 5977 { 5978 DirectOutputThread::flushHw_l(); 5979 // Flush anything still waiting in the mixbuffer 5980 mCurrentWriteLength = 0; 5981 mBytesRemaining = 0; 5982 mPausedWriteLength = 0; 5983 mPausedBytesRemaining = 0; 5984 // reset bytes written count to reflect that DSP buffers are empty after flush. 5985 mBytesWritten = 0; 5986 mOffloadUnderrunPosition = ~0LL; 5987 5988 if (mUseAsyncWrite) { 5989 // discard any pending drain or write ack by incrementing sequence 5990 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5991 mDrainSequence = (mDrainSequence + 2) & ~1; 5992 ALOG_ASSERT(mCallbackThread != 0); 5993 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5994 mCallbackThread->setDraining(mDrainSequence); 5995 } 5996 } 5997 5998 void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5999 { 6000 Mutex::Autolock _l(mLock); 6001 if (PlaybackThread::invalidateTracks_l(streamType)) { 6002 mFlushPending = true; 6003 } 6004 } 6005 6006 // ---------------------------------------------------------------------------- 6007 6008 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 6009 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 6010 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 6011 systemReady, DUPLICATING), 6012 mWaitTimeMs(UINT_MAX) 6013 { 6014 addOutputTrack(mainThread); 6015 } 6016 6017 AudioFlinger::DuplicatingThread::~DuplicatingThread() 6018 { 6019 for (size_t i = 0; i < mOutputTracks.size(); i++) { 6020 mOutputTracks[i]->destroy(); 6021 } 6022 } 6023 6024 void AudioFlinger::DuplicatingThread::threadLoop_mix() 6025 { 6026 // mix buffers... 6027 if (outputsReady(outputTracks)) { 6028 mAudioMixer->process(); 6029 } else { 6030 if (mMixerBufferValid) { 6031 memset(mMixerBuffer, 0, mMixerBufferSize); 6032 } else { 6033 memset(mSinkBuffer, 0, mSinkBufferSize); 6034 } 6035 } 6036 mSleepTimeUs = 0; 6037 writeFrames = mNormalFrameCount; 6038 mCurrentWriteLength = mSinkBufferSize; 6039 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 6040 } 6041 6042 void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 6043 { 6044 if (mSleepTimeUs == 0) { 6045 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 6046 mSleepTimeUs = mActiveSleepTimeUs; 6047 } else { 6048 mSleepTimeUs = mIdleSleepTimeUs; 6049 } 6050 } else if (mBytesWritten != 0) { 6051 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 6052 writeFrames = mNormalFrameCount; 6053 memset(mSinkBuffer, 0, mSinkBufferSize); 6054 } else { 6055 // flush remaining overflow buffers in output tracks 6056 writeFrames = 0; 6057 } 6058 mSleepTimeUs = 0; 6059 } 6060 } 6061 6062 ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 6063 { 6064 for (size_t i = 0; i < outputTracks.size(); i++) { 6065 outputTracks[i]->write(mSinkBuffer, writeFrames); 6066 } 6067 mStandby = false; 6068 return (ssize_t)mSinkBufferSize; 6069 } 6070 6071 void AudioFlinger::DuplicatingThread::threadLoop_standby() 6072 { 6073 // DuplicatingThread implements standby by stopping all tracks 6074 for (size_t i = 0; i < outputTracks.size(); i++) { 6075 outputTracks[i]->stop(); 6076 } 6077 } 6078 6079 void AudioFlinger::DuplicatingThread::dumpInternals(int fd, const Vector<String16>& args __unused) 6080 { 6081 MixerThread::dumpInternals(fd, args); 6082 6083 std::stringstream ss; 6084 const size_t numTracks = mOutputTracks.size(); 6085 ss << " " << numTracks << " OutputTracks"; 6086 if (numTracks > 0) { 6087 ss << ":"; 6088 for (const auto &track : mOutputTracks) { 6089 const sp<ThreadBase> thread = track->thread().promote(); 6090 ss << " (" << track->name() << " : "; 6091 if (thread.get() != nullptr) { 6092 ss << thread.get() << ", " << thread->id(); 6093 } else { 6094 ss << "null"; 6095 } 6096 ss << ")"; 6097 } 6098 } 6099 ss << "\n"; 6100 std::string result = ss.str(); 6101 write(fd, result.c_str(), result.size()); 6102 } 6103 6104 void AudioFlinger::DuplicatingThread::saveOutputTracks() 6105 { 6106 outputTracks = mOutputTracks; 6107 } 6108 6109 void AudioFlinger::DuplicatingThread::clearOutputTracks() 6110 { 6111 outputTracks.clear(); 6112 } 6113 6114 void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 6115 { 6116 Mutex::Autolock _l(mLock); 6117 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 6118 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 6119 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 6120 const size_t frameCount = 6121 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 6122 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 6123 // from different OutputTracks and their associated MixerThreads (e.g. one may 6124 // nearly empty and the other may be dropping data). 6125 6126 sp<OutputTrack> outputTrack = new OutputTrack(thread, 6127 this, 6128 mSampleRate, 6129 mFormat, 6130 mChannelMask, 6131 frameCount, 6132 IPCThreadState::self()->getCallingUid()); 6133 status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY; 6134 if (status != NO_ERROR) { 6135 ALOGE("addOutputTrack() initCheck failed %d", status); 6136 return; 6137 } 6138 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 6139 mOutputTracks.add(outputTrack); 6140 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 6141 updateWaitTime_l(); 6142 } 6143 6144 void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 6145 { 6146 Mutex::Autolock _l(mLock); 6147 for (size_t i = 0; i < mOutputTracks.size(); i++) { 6148 if (mOutputTracks[i]->thread() == thread) { 6149 mOutputTracks[i]->destroy(); 6150 mOutputTracks.removeAt(i); 6151 updateWaitTime_l(); 6152 if (thread->getOutput() == mOutput) { 6153 mOutput = NULL; 6154 } 6155 return; 6156 } 6157 } 6158 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 6159 } 6160 6161 // caller must hold mLock 6162 void AudioFlinger::DuplicatingThread::updateWaitTime_l() 6163 { 6164 mWaitTimeMs = UINT_MAX; 6165 for (size_t i = 0; i < mOutputTracks.size(); i++) { 6166 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 6167 if (strong != 0) { 6168 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 6169 if (waitTimeMs < mWaitTimeMs) { 6170 mWaitTimeMs = waitTimeMs; 6171 } 6172 } 6173 } 6174 } 6175 6176 6177 bool AudioFlinger::DuplicatingThread::outputsReady( 6178 const SortedVector< sp<OutputTrack> > &outputTracks) 6179 { 6180 for (size_t i = 0; i < outputTracks.size(); i++) { 6181 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 6182 if (thread == 0) { 6183 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 6184 outputTracks[i].get()); 6185 return false; 6186 } 6187 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 6188 // see note at standby() declaration 6189 if (playbackThread->standby() && !playbackThread->isSuspended()) { 6190 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 6191 thread.get()); 6192 return false; 6193 } 6194 } 6195 return true; 6196 } 6197 6198 void AudioFlinger::DuplicatingThread::sendMetadataToBackend_l( 6199 const StreamOutHalInterface::SourceMetadata& metadata) 6200 { 6201 for (auto& outputTrack : outputTracks) { // not mOutputTracks 6202 outputTrack->setMetadatas(metadata.tracks); 6203 } 6204 } 6205 6206 uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 6207 { 6208 return (mWaitTimeMs * 1000) / 2; 6209 } 6210 6211 void AudioFlinger::DuplicatingThread::cacheParameters_l() 6212 { 6213 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 6214 updateWaitTime_l(); 6215 6216 MixerThread::cacheParameters_l(); 6217 } 6218 6219 6220 // ---------------------------------------------------------------------------- 6221 // Record 6222 // ---------------------------------------------------------------------------- 6223 6224 AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 6225 AudioStreamIn *input, 6226 audio_io_handle_t id, 6227 audio_devices_t outDevice, 6228 audio_devices_t inDevice, 6229 bool systemReady 6230 #ifdef TEE_SINK 6231 , const sp<NBAIO_Sink>& teeSink 6232 #endif 6233 ) : 6234 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 6235 mInput(input), 6236 mActiveTracks(&this->mLocalLog), 6237 mRsmpInBuffer(NULL), 6238 // mRsmpInFrames, mRsmpInFramesP2, and mRsmpInFramesOA are set by readInputParameters_l() 6239 mRsmpInRear(0) 6240 #ifdef TEE_SINK 6241 , mTeeSink(teeSink) 6242 #endif 6243 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 6244 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 6245 // mFastCapture below 6246 , mFastCaptureFutex(0) 6247 // mInputSource 6248 // mPipeSink 6249 // mPipeSource 6250 , mPipeFramesP2(0) 6251 // mPipeMemory 6252 // mFastCaptureNBLogWriter 6253 , mFastTrackAvail(false) 6254 , mBtNrecSuspended(false) 6255 { 6256 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 6257 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 6258 6259 readInputParameters_l(); 6260 6261 // create an NBAIO source for the HAL input stream, and negotiate 6262 mInputSource = new AudioStreamInSource(input->stream); 6263 size_t numCounterOffers = 0; 6264 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 6265 #if !LOG_NDEBUG 6266 ssize_t index = 6267 #else 6268 (void) 6269 #endif 6270 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 6271 ALOG_ASSERT(index == 0); 6272 6273 // initialize fast capture depending on configuration 6274 bool initFastCapture; 6275 switch (kUseFastCapture) { 6276 case FastCapture_Never: 6277 initFastCapture = false; 6278 ALOGV("%p kUseFastCapture = Never, initFastCapture = false", this); 6279 break; 6280 case FastCapture_Always: 6281 initFastCapture = true; 6282 ALOGV("%p kUseFastCapture = Always, initFastCapture = true", this); 6283 break; 6284 case FastCapture_Static: 6285 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 6286 ALOGV("%p kUseFastCapture = Static, (%lld * 1000) / %u vs %u, initFastCapture = %d", 6287 this, (long long)mFrameCount, mSampleRate, kMinNormalCaptureBufferSizeMs, 6288 initFastCapture); 6289 break; 6290 // case FastCapture_Dynamic: 6291 } 6292 6293 if (initFastCapture) { 6294 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 6295 NBAIO_Format format = mInputSource->format(); 6296 // quadruple-buffering of 20 ms each; this ensures we can sleep for 20ms in RecordThread 6297 size_t pipeFramesP2 = roundup(4 * FMS_20 * mSampleRate / 1000); 6298 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 6299 void *pipeBuffer = nullptr; 6300 const sp<MemoryDealer> roHeap(readOnlyHeap()); 6301 sp<IMemory> pipeMemory; 6302 if ((roHeap == 0) || 6303 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 6304 (pipeBuffer = pipeMemory->pointer()) == nullptr) { 6305 ALOGE("not enough memory for pipe buffer size=%zu; " 6306 "roHeap=%p, pipeMemory=%p, pipeBuffer=%p; roHeapSize: %lld", 6307 pipeSize, roHeap.get(), pipeMemory.get(), pipeBuffer, 6308 (long long)kRecordThreadReadOnlyHeapSize); 6309 goto failed; 6310 } 6311 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 6312 memset(pipeBuffer, 0, pipeSize); 6313 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 6314 const NBAIO_Format offers[1] = {format}; 6315 size_t numCounterOffers = 0; 6316 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 6317 ALOG_ASSERT(index == 0); 6318 mPipeSink = pipe; 6319 PipeReader *pipeReader = new PipeReader(*pipe); 6320 numCounterOffers = 0; 6321 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 6322 ALOG_ASSERT(index == 0); 6323 mPipeSource = pipeReader; 6324 mPipeFramesP2 = pipeFramesP2; 6325 mPipeMemory = pipeMemory; 6326 6327 // create fast capture 6328 mFastCapture = new FastCapture(); 6329 FastCaptureStateQueue *sq = mFastCapture->sq(); 6330 #ifdef STATE_QUEUE_DUMP 6331 // FIXME 6332 #endif 6333 FastCaptureState *state = sq->begin(); 6334 state->mCblk = NULL; 6335 state->mInputSource = mInputSource.get(); 6336 state->mInputSourceGen++; 6337 state->mPipeSink = pipe; 6338 state->mPipeSinkGen++; 6339 state->mFrameCount = mFrameCount; 6340 state->mCommand = FastCaptureState::COLD_IDLE; 6341 // already done in constructor initialization list 6342 //mFastCaptureFutex = 0; 6343 state->mColdFutexAddr = &mFastCaptureFutex; 6344 state->mColdGen++; 6345 state->mDumpState = &mFastCaptureDumpState; 6346 #ifdef TEE_SINK 6347 // FIXME 6348 #endif 6349 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 6350 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 6351 sq->end(); 6352 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6353 6354 // start the fast capture 6355 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 6356 pid_t tid = mFastCapture->getTid(); 6357 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture, false /*forApp*/); 6358 stream()->setHalThreadPriority(kPriorityFastCapture); 6359 #ifdef AUDIO_WATCHDOG 6360 // FIXME 6361 #endif 6362 6363 mFastTrackAvail = true; 6364 } 6365 failed: ; 6366 6367 // FIXME mNormalSource 6368 } 6369 6370 AudioFlinger::RecordThread::~RecordThread() 6371 { 6372 if (mFastCapture != 0) { 6373 FastCaptureStateQueue *sq = mFastCapture->sq(); 6374 FastCaptureState *state = sq->begin(); 6375 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6376 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6377 if (old == -1) { 6378 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6379 } 6380 } 6381 state->mCommand = FastCaptureState::EXIT; 6382 sq->end(); 6383 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6384 mFastCapture->join(); 6385 mFastCapture.clear(); 6386 } 6387 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 6388 mAudioFlinger->unregisterWriter(mNBLogWriter); 6389 free(mRsmpInBuffer); 6390 } 6391 6392 void AudioFlinger::RecordThread::onFirstRef() 6393 { 6394 run(mThreadName, PRIORITY_URGENT_AUDIO); 6395 } 6396 6397 void AudioFlinger::RecordThread::preExit() 6398 { 6399 ALOGV(" preExit()"); 6400 Mutex::Autolock _l(mLock); 6401 for (size_t i = 0; i < mTracks.size(); i++) { 6402 sp<RecordTrack> track = mTracks[i]; 6403 track->invalidate(); 6404 } 6405 mActiveTracks.clear(); 6406 mStartStopCond.broadcast(); 6407 } 6408 6409 bool AudioFlinger::RecordThread::threadLoop() 6410 { 6411 nsecs_t lastWarning = 0; 6412 6413 inputStandBy(); 6414 6415 reacquire_wakelock: 6416 sp<RecordTrack> activeTrack; 6417 { 6418 Mutex::Autolock _l(mLock); 6419 acquireWakeLock_l(); 6420 } 6421 6422 // used to request a deferred sleep, to be executed later while mutex is unlocked 6423 uint32_t sleepUs = 0; 6424 6425 // loop while there is work to do 6426 for (;;) { 6427 Vector< sp<EffectChain> > effectChains; 6428 6429 // activeTracks accumulates a copy of a subset of mActiveTracks 6430 Vector< sp<RecordTrack> > activeTracks; 6431 6432 // reference to the (first and only) active fast track 6433 sp<RecordTrack> fastTrack; 6434 6435 // reference to a fast track which is about to be removed 6436 sp<RecordTrack> fastTrackToRemove; 6437 6438 { // scope for mLock 6439 Mutex::Autolock _l(mLock); 6440 6441 processConfigEvents_l(); 6442 6443 // check exitPending here because checkForNewParameters_l() and 6444 // checkForNewParameters_l() can temporarily release mLock 6445 if (exitPending()) { 6446 break; 6447 } 6448 6449 // sleep with mutex unlocked 6450 if (sleepUs > 0) { 6451 ATRACE_BEGIN("sleepC"); 6452 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)sleepUs)); 6453 ATRACE_END(); 6454 sleepUs = 0; 6455 continue; 6456 } 6457 6458 // if no active track(s), then standby and release wakelock 6459 size_t size = mActiveTracks.size(); 6460 if (size == 0) { 6461 standbyIfNotAlreadyInStandby(); 6462 // exitPending() can't become true here 6463 releaseWakeLock_l(); 6464 ALOGV("RecordThread: loop stopping"); 6465 // go to sleep 6466 mWaitWorkCV.wait(mLock); 6467 ALOGV("RecordThread: loop starting"); 6468 goto reacquire_wakelock; 6469 } 6470 6471 bool doBroadcast = false; 6472 bool allStopped = true; 6473 for (size_t i = 0; i < size; ) { 6474 6475 activeTrack = mActiveTracks[i]; 6476 if (activeTrack->isTerminated()) { 6477 if (activeTrack->isFastTrack()) { 6478 ALOG_ASSERT(fastTrackToRemove == 0); 6479 fastTrackToRemove = activeTrack; 6480 } 6481 removeTrack_l(activeTrack); 6482 mActiveTracks.remove(activeTrack); 6483 size--; 6484 continue; 6485 } 6486 6487 TrackBase::track_state activeTrackState = activeTrack->mState; 6488 switch (activeTrackState) { 6489 6490 case TrackBase::PAUSING: 6491 mActiveTracks.remove(activeTrack); 6492 doBroadcast = true; 6493 size--; 6494 continue; 6495 6496 case TrackBase::STARTING_1: 6497 sleepUs = 10000; 6498 i++; 6499 allStopped = false; 6500 continue; 6501 6502 case TrackBase::STARTING_2: 6503 doBroadcast = true; 6504 mStandby = false; 6505 activeTrack->mState = TrackBase::ACTIVE; 6506 allStopped = false; 6507 break; 6508 6509 case TrackBase::ACTIVE: 6510 allStopped = false; 6511 break; 6512 6513 case TrackBase::IDLE: 6514 i++; 6515 continue; 6516 6517 default: 6518 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 6519 } 6520 6521 activeTracks.add(activeTrack); 6522 i++; 6523 6524 if (activeTrack->isFastTrack()) { 6525 ALOG_ASSERT(!mFastTrackAvail); 6526 ALOG_ASSERT(fastTrack == 0); 6527 fastTrack = activeTrack; 6528 } 6529 } 6530 6531 mActiveTracks.updatePowerState(this); 6532 6533 updateMetadata_l(); 6534 6535 if (allStopped) { 6536 standbyIfNotAlreadyInStandby(); 6537 } 6538 if (doBroadcast) { 6539 mStartStopCond.broadcast(); 6540 } 6541 6542 // sleep if there are no active tracks to process 6543 if (activeTracks.size() == 0) { 6544 if (sleepUs == 0) { 6545 sleepUs = kRecordThreadSleepUs; 6546 } 6547 continue; 6548 } 6549 sleepUs = 0; 6550 6551 lockEffectChains_l(effectChains); 6552 } 6553 6554 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6555 6556 size_t size = effectChains.size(); 6557 for (size_t i = 0; i < size; i++) { 6558 // thread mutex is not locked, but effect chain is locked 6559 effectChains[i]->process_l(); 6560 } 6561 6562 // Push a new fast capture state if fast capture is not already running, or cblk change 6563 if (mFastCapture != 0) { 6564 FastCaptureStateQueue *sq = mFastCapture->sq(); 6565 FastCaptureState *state = sq->begin(); 6566 bool didModify = false; 6567 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6568 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6569 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6570 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6571 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6572 if (old == -1) { 6573 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6574 } 6575 } 6576 state->mCommand = FastCaptureState::READ_WRITE; 6577 #if 0 // FIXME 6578 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6579 FastThreadDumpState::kSamplingNforLowRamDevice : 6580 FastThreadDumpState::kSamplingN); 6581 #endif 6582 didModify = true; 6583 } 6584 audio_track_cblk_t *cblkOld = state->mCblk; 6585 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6586 if (cblkNew != cblkOld) { 6587 state->mCblk = cblkNew; 6588 // block until acked if removing a fast track 6589 if (cblkOld != NULL) { 6590 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6591 } 6592 didModify = true; 6593 } 6594 sq->end(didModify); 6595 if (didModify) { 6596 sq->push(block); 6597 #if 0 6598 if (kUseFastCapture == FastCapture_Dynamic) { 6599 mNormalSource = mPipeSource; 6600 } 6601 #endif 6602 } 6603 } 6604 6605 // now run the fast track destructor with thread mutex unlocked 6606 fastTrackToRemove.clear(); 6607 6608 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6609 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6610 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6611 // If destination is non-contiguous, first read past the nominal end of buffer, then 6612 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6613 6614 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6615 ssize_t framesRead; 6616 6617 // If an NBAIO source is present, use it to read the normal capture's data 6618 if (mPipeSource != 0) { 6619 size_t framesToRead = mBufferSize / mFrameSize; 6620 framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2); 6621 6622 // The audio fifo read() returns OVERRUN on overflow, and advances the read pointer 6623 // to the full buffer point (clearing the overflow condition). Upon OVERRUN error, 6624 // we immediately retry the read() to get data and prevent another overflow. 6625 for (int retries = 0; retries <= 2; ++retries) { 6626 ALOGW_IF(retries > 0, "overrun on read from pipe, retry #%d", retries); 6627 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6628 framesToRead); 6629 if (framesRead != OVERRUN) break; 6630 } 6631 6632 const ssize_t availableToRead = mPipeSource->availableToRead(); 6633 if (availableToRead >= 0) { 6634 // PipeSource is the master clock. It is up to the AudioRecord client to keep up. 6635 LOG_ALWAYS_FATAL_IF((size_t)availableToRead > mPipeFramesP2, 6636 "more frames to read than fifo size, %zd > %zu", 6637 availableToRead, mPipeFramesP2); 6638 const size_t pipeFramesFree = mPipeFramesP2 - availableToRead; 6639 const size_t sleepFrames = min(pipeFramesFree, mRsmpInFramesP2) / 2; 6640 ALOGVV("mPipeFramesP2:%zu mRsmpInFramesP2:%zu sleepFrames:%zu availableToRead:%zd", 6641 mPipeFramesP2, mRsmpInFramesP2, sleepFrames, availableToRead); 6642 sleepUs = (sleepFrames * 1000000LL) / mSampleRate; 6643 } 6644 if (framesRead < 0) { 6645 status_t status = (status_t) framesRead; 6646 switch (status) { 6647 case OVERRUN: 6648 ALOGW("overrun on read from pipe"); 6649 framesRead = 0; 6650 break; 6651 case NEGOTIATE: 6652 ALOGE("re-negotiation is needed"); 6653 framesRead = -1; // Will cause an attempt to recover. 6654 break; 6655 default: 6656 ALOGE("unknown error %d on read from pipe", status); 6657 break; 6658 } 6659 } 6660 // otherwise use the HAL / AudioStreamIn directly 6661 } else { 6662 ATRACE_BEGIN("read"); 6663 size_t bytesRead; 6664 status_t result = mInput->stream->read( 6665 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize, &bytesRead); 6666 ATRACE_END(); 6667 if (result < 0) { 6668 framesRead = result; 6669 } else { 6670 framesRead = bytesRead / mFrameSize; 6671 } 6672 } 6673 6674 // Update server timestamp with server stats 6675 // systemTime() is optional if the hardware supports timestamps. 6676 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6677 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6678 6679 // Update server timestamp with kernel stats 6680 if (mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) { 6681 int64_t position, time; 6682 int ret = mInput->stream->getCapturePosition(&position, &time); 6683 if (ret == NO_ERROR) { 6684 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6685 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6686 // Note: In general record buffers should tend to be empty in 6687 // a properly running pipeline. 6688 // 6689 // Also, it is not advantageous to call get_presentation_position during the read 6690 // as the read obtains a lock, preventing the timestamp call from executing. 6691 } 6692 } 6693 // Use this to track timestamp information 6694 // ALOGD("%s", mTimestamp.toString().c_str()); 6695 6696 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6697 ALOGE("read failed: framesRead=%zd", framesRead); 6698 // Force input into standby so that it tries to recover at next read attempt 6699 inputStandBy(); 6700 sleepUs = kRecordThreadSleepUs; 6701 } 6702 if (framesRead <= 0) { 6703 goto unlock; 6704 } 6705 ALOG_ASSERT(framesRead > 0); 6706 6707 if (mTeeSink != 0) { 6708 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6709 } 6710 // If destination is non-contiguous, we now correct for reading past end of buffer. 6711 { 6712 size_t part1 = mRsmpInFramesP2 - rear; 6713 if ((size_t) framesRead > part1) { 6714 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6715 (framesRead - part1) * mFrameSize); 6716 } 6717 } 6718 rear = mRsmpInRear += framesRead; 6719 6720 size = activeTracks.size(); 6721 6722 // loop over each active track 6723 for (size_t i = 0; i < size; i++) { 6724 activeTrack = activeTracks[i]; 6725 6726 // skip fast tracks, as those are handled directly by FastCapture 6727 if (activeTrack->isFastTrack()) { 6728 continue; 6729 } 6730 6731 // TODO: This code probably should be moved to RecordTrack. 6732 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6733 6734 enum { 6735 OVERRUN_UNKNOWN, 6736 OVERRUN_TRUE, 6737 OVERRUN_FALSE 6738 } overrun = OVERRUN_UNKNOWN; 6739 6740 // loop over getNextBuffer to handle circular sink 6741 for (;;) { 6742 6743 activeTrack->mSink.frameCount = ~0; 6744 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6745 size_t framesOut = activeTrack->mSink.frameCount; 6746 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6747 6748 // check available frames and handle overrun conditions 6749 // if the record track isn't draining fast enough. 6750 bool hasOverrun; 6751 size_t framesIn; 6752 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6753 if (hasOverrun) { 6754 overrun = OVERRUN_TRUE; 6755 } 6756 if (framesOut == 0 || framesIn == 0) { 6757 break; 6758 } 6759 6760 // Don't allow framesOut to be larger than what is possible with resampling 6761 // from framesIn. 6762 // This isn't strictly necessary but helps limit buffer resizing in 6763 // RecordBufferConverter. TODO: remove when no longer needed. 6764 framesOut = min(framesOut, 6765 destinationFramesPossible( 6766 framesIn, mSampleRate, activeTrack->mSampleRate)); 6767 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6768 framesOut = activeTrack->mRecordBufferConverter->convert( 6769 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6770 6771 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6772 overrun = OVERRUN_FALSE; 6773 } 6774 6775 if (activeTrack->mFramesToDrop == 0) { 6776 if (framesOut > 0) { 6777 activeTrack->mSink.frameCount = framesOut; 6778 // Sanitize before releasing if the track has no access to the source data 6779 // An idle UID receives silence from non virtual devices until active 6780 if (activeTrack->isSilenced()) { 6781 memset(activeTrack->mSink.raw, 0, framesOut * mFrameSize); 6782 } 6783 activeTrack->releaseBuffer(&activeTrack->mSink); 6784 } 6785 } else { 6786 // FIXME could do a partial drop of framesOut 6787 if (activeTrack->mFramesToDrop > 0) { 6788 activeTrack->mFramesToDrop -= framesOut; 6789 if (activeTrack->mFramesToDrop <= 0) { 6790 activeTrack->clearSyncStartEvent(); 6791 } 6792 } else { 6793 activeTrack->mFramesToDrop += framesOut; 6794 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6795 activeTrack->mSyncStartEvent->isCancelled()) { 6796 ALOGW("Synced record %s, session %d, trigger session %d", 6797 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6798 activeTrack->sessionId(), 6799 (activeTrack->mSyncStartEvent != 0) ? 6800 activeTrack->mSyncStartEvent->triggerSession() : 6801 AUDIO_SESSION_NONE); 6802 activeTrack->clearSyncStartEvent(); 6803 } 6804 } 6805 } 6806 6807 if (framesOut == 0) { 6808 break; 6809 } 6810 } 6811 6812 switch (overrun) { 6813 case OVERRUN_TRUE: 6814 // client isn't retrieving buffers fast enough 6815 if (!activeTrack->setOverflow()) { 6816 nsecs_t now = systemTime(); 6817 // FIXME should lastWarning per track? 6818 if ((now - lastWarning) > kWarningThrottleNs) { 6819 ALOGW("RecordThread: buffer overflow"); 6820 lastWarning = now; 6821 } 6822 } 6823 break; 6824 case OVERRUN_FALSE: 6825 activeTrack->clearOverflow(); 6826 break; 6827 case OVERRUN_UNKNOWN: 6828 break; 6829 } 6830 6831 // update frame information and push timestamp out 6832 activeTrack->updateTrackFrameInfo( 6833 activeTrack->mServerProxy->framesReleased(), 6834 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6835 mSampleRate, mTimestamp); 6836 } 6837 6838 unlock: 6839 // enable changes in effect chain 6840 unlockEffectChains(effectChains); 6841 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6842 } 6843 6844 standbyIfNotAlreadyInStandby(); 6845 6846 { 6847 Mutex::Autolock _l(mLock); 6848 for (size_t i = 0; i < mTracks.size(); i++) { 6849 sp<RecordTrack> track = mTracks[i]; 6850 track->invalidate(); 6851 } 6852 mActiveTracks.clear(); 6853 mStartStopCond.broadcast(); 6854 } 6855 6856 releaseWakeLock(); 6857 6858 ALOGV("RecordThread %p exiting", this); 6859 return false; 6860 } 6861 6862 void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6863 { 6864 if (!mStandby) { 6865 inputStandBy(); 6866 mStandby = true; 6867 } 6868 } 6869 6870 void AudioFlinger::RecordThread::inputStandBy() 6871 { 6872 // Idle the fast capture if it's currently running 6873 if (mFastCapture != 0) { 6874 FastCaptureStateQueue *sq = mFastCapture->sq(); 6875 FastCaptureState *state = sq->begin(); 6876 if (!(state->mCommand & FastCaptureState::IDLE)) { 6877 state->mCommand = FastCaptureState::COLD_IDLE; 6878 state->mColdFutexAddr = &mFastCaptureFutex; 6879 state->mColdGen++; 6880 mFastCaptureFutex = 0; 6881 sq->end(); 6882 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6883 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6884 #if 0 6885 if (kUseFastCapture == FastCapture_Dynamic) { 6886 // FIXME 6887 } 6888 #endif 6889 #ifdef AUDIO_WATCHDOG 6890 // FIXME 6891 #endif 6892 } else { 6893 sq->end(false /*didModify*/); 6894 } 6895 } 6896 status_t result = mInput->stream->standby(); 6897 ALOGE_IF(result != OK, "Error when putting input stream into standby: %d", result); 6898 6899 // If going into standby, flush the pipe source. 6900 if (mPipeSource.get() != nullptr) { 6901 const ssize_t flushed = mPipeSource->flush(); 6902 if (flushed > 0) { 6903 ALOGV("Input standby flushed PipeSource %zd frames", flushed); 6904 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += flushed; 6905 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6906 } 6907 } 6908 } 6909 6910 // RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6911 sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6912 const sp<AudioFlinger::Client>& client, 6913 const audio_attributes_t& attr, 6914 uint32_t *pSampleRate, 6915 audio_format_t format, 6916 audio_channel_mask_t channelMask, 6917 size_t *pFrameCount, 6918 audio_session_t sessionId, 6919 size_t *pNotificationFrameCount, 6920 uid_t uid, 6921 audio_input_flags_t *flags, 6922 pid_t tid, 6923 status_t *status, 6924 audio_port_handle_t portId) 6925 { 6926 size_t frameCount = *pFrameCount; 6927 size_t notificationFrameCount = *pNotificationFrameCount; 6928 sp<RecordTrack> track; 6929 status_t lStatus; 6930 audio_input_flags_t inputFlags = mInput->flags; 6931 audio_input_flags_t requestedFlags = *flags; 6932 uint32_t sampleRate; 6933 6934 lStatus = initCheck(); 6935 if (lStatus != NO_ERROR) { 6936 ALOGE("createRecordTrack_l() audio driver not initialized"); 6937 goto Exit; 6938 } 6939 6940 if (*pSampleRate == 0) { 6941 *pSampleRate = mSampleRate; 6942 } 6943 sampleRate = *pSampleRate; 6944 6945 // special case for FAST flag considered OK if fast capture is present 6946 if (hasFastCapture()) { 6947 inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST); 6948 } 6949 6950 // Check if requested flags are compatible with input stream flags 6951 if ((*flags & inputFlags) != *flags) { 6952 ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and" 6953 " input flags (%08x)", 6954 *flags, inputFlags); 6955 *flags = (audio_input_flags_t)(*flags & inputFlags); 6956 } 6957 6958 // client expresses a preference for FAST, but we get the final say 6959 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6960 if ( 6961 // we formerly checked for a callback handler (non-0 tid), 6962 // but that is no longer required for TRANSFER_OBTAIN mode 6963 // 6964 // frame count is not specified, or is exactly the pipe depth 6965 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6966 // PCM data 6967 audio_is_linear_pcm(format) && 6968 // hardware format 6969 (format == mFormat) && 6970 // hardware channel mask 6971 (channelMask == mChannelMask) && 6972 // hardware sample rate 6973 (sampleRate == mSampleRate) && 6974 // record thread has an associated fast capture 6975 hasFastCapture() && 6976 // there are sufficient fast track slots available 6977 mFastTrackAvail 6978 ) { 6979 // check compatibility with audio effects. 6980 Mutex::Autolock _l(mLock); 6981 // Do not accept FAST flag if the session has software effects 6982 sp<EffectChain> chain = getEffectChain_l(sessionId); 6983 if (chain != 0) { 6984 audio_input_flags_t old = *flags; 6985 chain->checkInputFlagCompatibility(flags); 6986 if (old != *flags) { 6987 ALOGV("%p AUDIO_INPUT_FLAGS denied by effect old=%#x new=%#x", 6988 this, (int)old, (int)*flags); 6989 } 6990 } 6991 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0, 6992 "%p AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6993 this, frameCount, mFrameCount); 6994 } else { 6995 ALOGV("%p AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6996 "format=%#x isLinear=%d mFormat=%#x channelMask=%#x sampleRate=%u mSampleRate=%u " 6997 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6998 this, frameCount, mFrameCount, mPipeFramesP2, 6999 format, audio_is_linear_pcm(format), mFormat, channelMask, sampleRate, mSampleRate, 7000 hasFastCapture(), tid, mFastTrackAvail); 7001 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 7002 } 7003 } 7004 7005 // If FAST or RAW flags were corrected, ask caller to request new input from audio policy 7006 if ((*flags & AUDIO_INPUT_FLAG_FAST) != 7007 (requestedFlags & AUDIO_INPUT_FLAG_FAST)) { 7008 *flags = (audio_input_flags_t) (*flags & ~(AUDIO_INPUT_FLAG_FAST | AUDIO_INPUT_FLAG_RAW)); 7009 lStatus = BAD_TYPE; 7010 goto Exit; 7011 } 7012 7013 // compute track buffer size in frames, and suggest the notification frame count 7014 if (*flags & AUDIO_INPUT_FLAG_FAST) { 7015 // fast track: frame count is exactly the pipe depth 7016 frameCount = mPipeFramesP2; 7017 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 7018 notificationFrameCount = mFrameCount; 7019 } else { 7020 // not fast track: max notification period is resampled equivalent of one HAL buffer time 7021 // or 20 ms if there is a fast capture 7022 // TODO This could be a roundupRatio inline, and const 7023 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 7024 * sampleRate + mSampleRate - 1) / mSampleRate; 7025 // minimum number of notification periods is at least kMinNotifications, 7026 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 7027 static const size_t kMinNotifications = 3; 7028 static const uint32_t kMinMs = 30; 7029 // TODO This could be a roundupRatio inline 7030 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 7031 // TODO This could be a roundupRatio inline 7032 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 7033 maxNotificationFrames; 7034 const size_t minFrameCount = maxNotificationFrames * 7035 max(kMinNotifications, minNotificationsByMs); 7036 frameCount = max(frameCount, minFrameCount); 7037 if (notificationFrameCount == 0 || notificationFrameCount > maxNotificationFrames) { 7038 notificationFrameCount = maxNotificationFrames; 7039 } 7040 } 7041 *pFrameCount = frameCount; 7042 *pNotificationFrameCount = notificationFrameCount; 7043 7044 { // scope for mLock 7045 Mutex::Autolock _l(mLock); 7046 7047 track = new RecordTrack(this, client, attr, sampleRate, 7048 format, channelMask, frameCount, 7049 nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, uid, 7050 *flags, TrackBase::TYPE_DEFAULT, portId); 7051 7052 lStatus = track->initCheck(); 7053 if (lStatus != NO_ERROR) { 7054 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 7055 // track must be cleared from the caller as the caller has the AF lock 7056 goto Exit; 7057 } 7058 mTracks.add(track); 7059 7060 if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) { 7061 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 7062 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 7063 // so ask activity manager to do this on our behalf 7064 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/); 7065 } 7066 } 7067 7068 lStatus = NO_ERROR; 7069 7070 Exit: 7071 *status = lStatus; 7072 return track; 7073 } 7074 7075 status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 7076 AudioSystem::sync_event_t event, 7077 audio_session_t triggerSession) 7078 { 7079 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 7080 sp<ThreadBase> strongMe = this; 7081 status_t status = NO_ERROR; 7082 7083 if (event == AudioSystem::SYNC_EVENT_NONE) { 7084 recordTrack->clearSyncStartEvent(); 7085 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 7086 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 7087 triggerSession, 7088 recordTrack->sessionId(), 7089 syncStartEventCallback, 7090 recordTrack); 7091 // Sync event can be cancelled by the trigger session if the track is not in a 7092 // compatible state in which case we start record immediately 7093 if (recordTrack->mSyncStartEvent->isCancelled()) { 7094 recordTrack->clearSyncStartEvent(); 7095 } else { 7096 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 7097 recordTrack->mFramesToDrop = -(ssize_t) 7098 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 7099 } 7100 } 7101 7102 { 7103 // This section is a rendezvous between binder thread executing start() and RecordThread 7104 AutoMutex lock(mLock); 7105 if (mActiveTracks.indexOf(recordTrack) >= 0) { 7106 if (recordTrack->mState == TrackBase::PAUSING) { 7107 ALOGV("active record track PAUSING -> ACTIVE"); 7108 recordTrack->mState = TrackBase::ACTIVE; 7109 } else { 7110 ALOGV("active record track state %d", recordTrack->mState); 7111 } 7112 return status; 7113 } 7114 7115 // TODO consider other ways of handling this, such as changing the state to :STARTING and 7116 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 7117 // or using a separate command thread 7118 recordTrack->mState = TrackBase::STARTING_1; 7119 mActiveTracks.add(recordTrack); 7120 status_t status = NO_ERROR; 7121 if (recordTrack->isExternalTrack()) { 7122 mLock.unlock(); 7123 bool silenced; 7124 status = AudioSystem::startInput(recordTrack->portId(), &silenced); 7125 mLock.lock(); 7126 // FIXME should verify that recordTrack is still in mActiveTracks 7127 if (status != NO_ERROR) { 7128 mActiveTracks.remove(recordTrack); 7129 recordTrack->clearSyncStartEvent(); 7130 ALOGV("RecordThread::start error %d", status); 7131 return status; 7132 } 7133 recordTrack->setSilenced(silenced); 7134 } 7135 // Catch up with current buffer indices if thread is already running. 7136 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 7137 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 7138 // see previously buffered data before it called start(), but with greater risk of overrun. 7139 7140 recordTrack->mResamplerBufferProvider->reset(); 7141 // clear any converter state as new data will be discontinuous 7142 recordTrack->mRecordBufferConverter->reset(); 7143 recordTrack->mState = TrackBase::STARTING_2; 7144 // signal thread to start 7145 mWaitWorkCV.broadcast(); 7146 if (mActiveTracks.indexOf(recordTrack) < 0) { 7147 ALOGV("Record failed to start"); 7148 status = BAD_VALUE; 7149 goto startError; 7150 } 7151 return status; 7152 } 7153 7154 startError: 7155 if (recordTrack->isExternalTrack()) { 7156 AudioSystem::stopInput(recordTrack->portId()); 7157 } 7158 recordTrack->clearSyncStartEvent(); 7159 // FIXME I wonder why we do not reset the state here? 7160 return status; 7161 } 7162 7163 void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 7164 { 7165 sp<SyncEvent> strongEvent = event.promote(); 7166 7167 if (strongEvent != 0) { 7168 sp<RefBase> ptr = strongEvent->cookie().promote(); 7169 if (ptr != 0) { 7170 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 7171 recordTrack->handleSyncStartEvent(strongEvent); 7172 } 7173 } 7174 } 7175 7176 bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 7177 ALOGV("RecordThread::stop"); 7178 AutoMutex _l(mLock); 7179 if (mActiveTracks.indexOf(recordTrack) < 0 || recordTrack->mState == TrackBase::PAUSING) { 7180 return false; 7181 } 7182 // note that threadLoop may still be processing the track at this point [without lock] 7183 recordTrack->mState = TrackBase::PAUSING; 7184 // signal thread to stop 7185 mWaitWorkCV.broadcast(); 7186 // do not wait for mStartStopCond if exiting 7187 if (exitPending()) { 7188 return true; 7189 } 7190 // FIXME incorrect usage of wait: no explicit predicate or loop 7191 mStartStopCond.wait(mLock); 7192 // if we have been restarted, recordTrack is in mActiveTracks here 7193 if (exitPending() || mActiveTracks.indexOf(recordTrack) < 0) { 7194 ALOGV("Record stopped OK"); 7195 return true; 7196 } 7197 return false; 7198 } 7199 7200 bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 7201 { 7202 return false; 7203 } 7204 7205 status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 7206 { 7207 #if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 7208 if (!isValidSyncEvent(event)) { 7209 return BAD_VALUE; 7210 } 7211 7212 audio_session_t eventSession = event->triggerSession(); 7213 status_t ret = NAME_NOT_FOUND; 7214 7215 Mutex::Autolock _l(mLock); 7216 7217 for (size_t i = 0; i < mTracks.size(); i++) { 7218 sp<RecordTrack> track = mTracks[i]; 7219 if (eventSession == track->sessionId()) { 7220 (void) track->setSyncEvent(event); 7221 ret = NO_ERROR; 7222 } 7223 } 7224 return ret; 7225 #else 7226 return BAD_VALUE; 7227 #endif 7228 } 7229 7230 status_t AudioFlinger::RecordThread::getActiveMicrophones( 7231 std::vector<media::MicrophoneInfo>* activeMicrophones) 7232 { 7233 ALOGV("RecordThread::getActiveMicrophones"); 7234 AutoMutex _l(mLock); 7235 status_t status = mInput->stream->getActiveMicrophones(activeMicrophones); 7236 return status; 7237 } 7238 7239 void AudioFlinger::RecordThread::updateMetadata_l() 7240 { 7241 if (mInput == nullptr || mInput->stream == nullptr || 7242 !mActiveTracks.readAndClearHasChanged()) { 7243 return; 7244 } 7245 StreamInHalInterface::SinkMetadata metadata; 7246 for (const sp<RecordTrack> &track : mActiveTracks) { 7247 // No track is invalid as this is called after prepareTrack_l in the same critical section 7248 metadata.tracks.push_back({ 7249 .source = track->attributes().source, 7250 .gain = 1, // capture tracks do not have volumes 7251 }); 7252 } 7253 mInput->stream->updateSinkMetadata(metadata); 7254 } 7255 7256 // destroyTrack_l() must be called with ThreadBase::mLock held 7257 void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 7258 { 7259 track->terminate(); 7260 track->mState = TrackBase::STOPPED; 7261 // active tracks are removed by threadLoop() 7262 if (mActiveTracks.indexOf(track) < 0) { 7263 removeTrack_l(track); 7264 } 7265 } 7266 7267 void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 7268 { 7269 String8 result; 7270 track->appendDump(result, false /* active */); 7271 mLocalLog.log("removeTrack_l (%p) %s", track.get(), result.string()); 7272 7273 mTracks.remove(track); 7274 // need anything related to effects here? 7275 if (track->isFastTrack()) { 7276 ALOG_ASSERT(!mFastTrackAvail); 7277 mFastTrackAvail = true; 7278 } 7279 } 7280 7281 void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 7282 { 7283 dumpInternals(fd, args); 7284 dumpTracks(fd, args); 7285 dumpEffectChains(fd, args); 7286 dprintf(fd, " Local log:\n"); 7287 mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */); 7288 } 7289 7290 void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 7291 { 7292 dumpBase(fd, args); 7293 7294 AudioStreamIn *input = mInput; 7295 audio_input_flags_t flags = input != NULL ? input->flags : AUDIO_INPUT_FLAG_NONE; 7296 dprintf(fd, " AudioStreamIn: %p flags %#x (%s)\n", 7297 input, flags, inputFlagsToString(flags).c_str()); 7298 if (mActiveTracks.size() == 0) { 7299 dprintf(fd, " No active record clients\n"); 7300 } 7301 7302 if (input != nullptr) { 7303 dprintf(fd, " Hal stream dump:\n"); 7304 (void)input->stream->dump(fd); 7305 } 7306 7307 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 7308 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 7309 7310 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 7311 // while we are dumping it. It may be inconsistent, but it won't mutate! 7312 // This is a large object so we place it on the heap. 7313 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 7314 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 7315 copy->dump(fd); 7316 delete copy; 7317 } 7318 7319 void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 7320 { 7321 String8 result; 7322 size_t numtracks = mTracks.size(); 7323 size_t numactive = mActiveTracks.size(); 7324 size_t numactiveseen = 0; 7325 dprintf(fd, " %zu Tracks", numtracks); 7326 const char *prefix = " "; 7327 if (numtracks) { 7328 dprintf(fd, " of which %zu are active\n", numactive); 7329 result.append(prefix); 7330 RecordTrack::appendDumpHeader(result); 7331 for (size_t i = 0; i < numtracks ; ++i) { 7332 sp<RecordTrack> track = mTracks[i]; 7333 if (track != 0) { 7334 bool active = mActiveTracks.indexOf(track) >= 0; 7335 if (active) { 7336 numactiveseen++; 7337 } 7338 result.append(prefix); 7339 track->appendDump(result, active); 7340 } 7341 } 7342 } else { 7343 dprintf(fd, "\n"); 7344 } 7345 7346 if (numactiveseen != numactive) { 7347 result.append(" The following tracks are in the active list but" 7348 " not in the track list\n"); 7349 result.append(prefix); 7350 RecordTrack::appendDumpHeader(result); 7351 for (size_t i = 0; i < numactive; ++i) { 7352 sp<RecordTrack> track = mActiveTracks[i]; 7353 if (mTracks.indexOf(track) < 0) { 7354 result.append(prefix); 7355 track->appendDump(result, true /* active */); 7356 } 7357 } 7358 7359 } 7360 write(fd, result.string(), result.size()); 7361 } 7362 7363 void AudioFlinger::RecordThread::setRecordSilenced(uid_t uid, bool silenced) 7364 { 7365 Mutex::Autolock _l(mLock); 7366 for (size_t i = 0; i < mTracks.size() ; i++) { 7367 sp<RecordTrack> track = mTracks[i]; 7368 if (track != 0 && track->uid() == uid) { 7369 track->setSilenced(silenced); 7370 } 7371 } 7372 } 7373 7374 void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 7375 { 7376 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 7377 RecordThread *recordThread = (RecordThread *) threadBase.get(); 7378 mRsmpInFront = recordThread->mRsmpInRear; 7379 mRsmpInUnrel = 0; 7380 } 7381 7382 void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 7383 size_t *framesAvailable, bool *hasOverrun) 7384 { 7385 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 7386 RecordThread *recordThread = (RecordThread *) threadBase.get(); 7387 const int32_t rear = recordThread->mRsmpInRear; 7388 const int32_t front = mRsmpInFront; 7389 const ssize_t filled = rear - front; 7390 7391 size_t framesIn; 7392 bool overrun = false; 7393 if (filled < 0) { 7394 // should not happen, but treat like a massive overrun and re-sync 7395 framesIn = 0; 7396 mRsmpInFront = rear; 7397 overrun = true; 7398 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 7399 framesIn = (size_t) filled; 7400 } else { 7401 // client is not keeping up with server, but give it latest data 7402 framesIn = recordThread->mRsmpInFrames; 7403 mRsmpInFront = /* front = */ rear - framesIn; 7404 overrun = true; 7405 } 7406 if (framesAvailable != NULL) { 7407 *framesAvailable = framesIn; 7408 } 7409 if (hasOverrun != NULL) { 7410 *hasOverrun = overrun; 7411 } 7412 } 7413 7414 // AudioBufferProvider interface 7415 status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 7416 AudioBufferProvider::Buffer* buffer) 7417 { 7418 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 7419 if (threadBase == 0) { 7420 buffer->frameCount = 0; 7421 buffer->raw = NULL; 7422 return NOT_ENOUGH_DATA; 7423 } 7424 RecordThread *recordThread = (RecordThread *) threadBase.get(); 7425 int32_t rear = recordThread->mRsmpInRear; 7426 int32_t front = mRsmpInFront; 7427 ssize_t filled = rear - front; 7428 // FIXME should not be P2 (don't want to increase latency) 7429 // FIXME if client not keeping up, discard 7430 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 7431 // 'filled' may be non-contiguous, so return only the first contiguous chunk 7432 front &= recordThread->mRsmpInFramesP2 - 1; 7433 size_t part1 = recordThread->mRsmpInFramesP2 - front; 7434 if (part1 > (size_t) filled) { 7435 part1 = filled; 7436 } 7437 size_t ask = buffer->frameCount; 7438 ALOG_ASSERT(ask > 0); 7439 if (part1 > ask) { 7440 part1 = ask; 7441 } 7442 if (part1 == 0) { 7443 // out of data is fine since the resampler will return a short-count. 7444 buffer->raw = NULL; 7445 buffer->frameCount = 0; 7446 mRsmpInUnrel = 0; 7447 return NOT_ENOUGH_DATA; 7448 } 7449 7450 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 7451 buffer->frameCount = part1; 7452 mRsmpInUnrel = part1; 7453 return NO_ERROR; 7454 } 7455 7456 // AudioBufferProvider interface 7457 void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 7458 AudioBufferProvider::Buffer* buffer) 7459 { 7460 size_t stepCount = buffer->frameCount; 7461 if (stepCount == 0) { 7462 return; 7463 } 7464 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 7465 mRsmpInUnrel -= stepCount; 7466 mRsmpInFront += stepCount; 7467 buffer->raw = NULL; 7468 buffer->frameCount = 0; 7469 } 7470 7471 void AudioFlinger::RecordThread::checkBtNrec() 7472 { 7473 Mutex::Autolock _l(mLock); 7474 checkBtNrec_l(); 7475 } 7476 7477 void AudioFlinger::RecordThread::checkBtNrec_l() 7478 { 7479 // disable AEC and NS if the device is a BT SCO headset supporting those 7480 // pre processings 7481 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7482 mAudioFlinger->btNrecIsOff(); 7483 if (mBtNrecSuspended.exchange(suspend) != suspend) { 7484 for (size_t i = 0; i < mEffectChains.size(); i++) { 7485 setEffectSuspended_l(FX_IID_AEC, suspend, mEffectChains[i]->sessionId()); 7486 setEffectSuspended_l(FX_IID_NS, suspend, mEffectChains[i]->sessionId()); 7487 } 7488 } 7489 } 7490 7491 7492 bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7493 status_t& status) 7494 { 7495 bool reconfig = false; 7496 7497 status = NO_ERROR; 7498 7499 audio_format_t reqFormat = mFormat; 7500 uint32_t samplingRate = mSampleRate; 7501 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7502 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7503 7504 AudioParameter param = AudioParameter(keyValuePair); 7505 int value; 7506 7507 // scope for AutoPark extends to end of method 7508 AutoPark<FastCapture> park(mFastCapture); 7509 7510 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7511 // channel count change can be requested. Do we mandate the first client defines the 7512 // HAL sampling rate and channel count or do we allow changes on the fly? 7513 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7514 samplingRate = value; 7515 reconfig = true; 7516 } 7517 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7518 if (!audio_is_linear_pcm((audio_format_t) value)) { 7519 status = BAD_VALUE; 7520 } else { 7521 reqFormat = (audio_format_t) value; 7522 reconfig = true; 7523 } 7524 } 7525 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7526 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7527 if (!audio_is_input_channel(mask) || 7528 audio_channel_count_from_in_mask(mask) > FCC_8) { 7529 status = BAD_VALUE; 7530 } else { 7531 channelMask = mask; 7532 reconfig = true; 7533 } 7534 } 7535 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7536 // do not accept frame count changes if tracks are open as the track buffer 7537 // size depends on frame count and correct behavior would not be guaranteed 7538 // if frame count is changed after track creation 7539 if (mActiveTracks.size() > 0) { 7540 status = INVALID_OPERATION; 7541 } else { 7542 reconfig = true; 7543 } 7544 } 7545 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7546 // forward device change to effects that have requested to be 7547 // aware of attached audio device. 7548 for (size_t i = 0; i < mEffectChains.size(); i++) { 7549 mEffectChains[i]->setDevice_l(value); 7550 } 7551 7552 // store input device and output device but do not forward output device to audio HAL. 7553 // Note that status is ignored by the caller for output device 7554 // (see AudioFlinger::setParameters() 7555 if (audio_is_output_devices(value)) { 7556 mOutDevice = value; 7557 status = BAD_VALUE; 7558 } else { 7559 mInDevice = value; 7560 if (value != AUDIO_DEVICE_NONE) { 7561 mPrevInDevice = value; 7562 } 7563 checkBtNrec_l(); 7564 } 7565 } 7566 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7567 mAudioSource != (audio_source_t)value) { 7568 // forward device change to effects that have requested to be 7569 // aware of attached audio device. 7570 for (size_t i = 0; i < mEffectChains.size(); i++) { 7571 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7572 } 7573 mAudioSource = (audio_source_t)value; 7574 } 7575 7576 if (status == NO_ERROR) { 7577 status = mInput->stream->setParameters(keyValuePair); 7578 if (status == INVALID_OPERATION) { 7579 inputStandBy(); 7580 status = mInput->stream->setParameters(keyValuePair); 7581 } 7582 if (reconfig) { 7583 if (status == BAD_VALUE) { 7584 uint32_t sRate; 7585 audio_channel_mask_t channelMask; 7586 audio_format_t format; 7587 if (mInput->stream->getAudioProperties(&sRate, &channelMask, &format) == OK && 7588 audio_is_linear_pcm(format) && audio_is_linear_pcm(reqFormat) && 7589 sRate <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate) && 7590 audio_channel_count_from_in_mask(channelMask) <= FCC_8) { 7591 status = NO_ERROR; 7592 } 7593 } 7594 if (status == NO_ERROR) { 7595 readInputParameters_l(); 7596 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7597 } 7598 } 7599 } 7600 7601 return reconfig; 7602 } 7603 7604 String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7605 { 7606 Mutex::Autolock _l(mLock); 7607 if (initCheck() == NO_ERROR) { 7608 String8 out_s8; 7609 if (mInput->stream->getParameters(keys, &out_s8) == OK) { 7610 return out_s8; 7611 } 7612 } 7613 return String8(); 7614 } 7615 7616 void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7617 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7618 7619 desc->mIoHandle = mId; 7620 7621 switch (event) { 7622 case AUDIO_INPUT_OPENED: 7623 case AUDIO_INPUT_REGISTERED: 7624 case AUDIO_INPUT_CONFIG_CHANGED: 7625 desc->mPatch = mPatch; 7626 desc->mChannelMask = mChannelMask; 7627 desc->mSamplingRate = mSampleRate; 7628 desc->mFormat = mFormat; 7629 desc->mFrameCount = mFrameCount; 7630 desc->mFrameCountHAL = mFrameCount; 7631 desc->mLatency = 0; 7632 break; 7633 7634 case AUDIO_INPUT_CLOSED: 7635 default: 7636 break; 7637 } 7638 mAudioFlinger->ioConfigChanged(event, desc, pid); 7639 } 7640 7641 void AudioFlinger::RecordThread::readInputParameters_l() 7642 { 7643 status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat); 7644 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result); 7645 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7646 LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_8, "HAL channel count %d > %d", mChannelCount, FCC_8); 7647 mFormat = mHALFormat; 7648 LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat); 7649 result = mInput->stream->getFrameSize(&mFrameSize); 7650 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result); 7651 result = mInput->stream->getBufferSize(&mBufferSize); 7652 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result); 7653 mFrameCount = mBufferSize / mFrameSize; 7654 ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%lld, " 7655 "mBufferSize=%lld, mFrameCount=%lld", 7656 this, mChannelCount, mFormat, (long long)mFrameSize, (long long)mBufferSize, 7657 (long long)mFrameCount); 7658 // This is the formula for calculating the temporary buffer size. 7659 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7660 // 1 full output buffer, regardless of the alignment of the available input. 7661 // The value is somewhat arbitrary, and could probably be even larger. 7662 // A larger value should allow more old data to be read after a track calls start(), 7663 // without increasing latency. 7664 // 7665 // Note this is independent of the maximum downsampling ratio permitted for capture. 7666 mRsmpInFrames = mFrameCount * 7; 7667 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7668 free(mRsmpInBuffer); 7669 mRsmpInBuffer = NULL; 7670 7671 // TODO optimize audio capture buffer sizes ... 7672 // Here we calculate the size of the sliding buffer used as a source 7673 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7674 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7675 // be better to have it derived from the pipe depth in the long term. 7676 // The current value is higher than necessary. However it should not add to latency. 7677 7678 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7679 mRsmpInFramesOA = mRsmpInFramesP2 + mFrameCount - 1; 7680 (void)posix_memalign(&mRsmpInBuffer, 32, mRsmpInFramesOA * mFrameSize); 7681 // if posix_memalign fails, will segv here. 7682 memset(mRsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize); 7683 7684 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7685 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7686 } 7687 7688 uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7689 { 7690 Mutex::Autolock _l(mLock); 7691 uint32_t result; 7692 if (initCheck() == NO_ERROR && mInput->stream->getInputFramesLost(&result) == OK) { 7693 return result; 7694 } 7695 return 0; 7696 } 7697 7698 // hasAudioSession_l() must be called with ThreadBase::mLock held 7699 uint32_t AudioFlinger::RecordThread::hasAudioSession_l(audio_session_t sessionId) const 7700 { 7701 uint32_t result = 0; 7702 if (getEffectChain_l(sessionId) != 0) { 7703 result = EFFECT_SESSION; 7704 } 7705 7706 for (size_t i = 0; i < mTracks.size(); ++i) { 7707 if (sessionId == mTracks[i]->sessionId()) { 7708 result |= TRACK_SESSION; 7709 if (mTracks[i]->isFastTrack()) { 7710 result |= FAST_SESSION; 7711 } 7712 break; 7713 } 7714 } 7715 7716 return result; 7717 } 7718 7719 KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7720 { 7721 KeyedVector<audio_session_t, bool> ids; 7722 Mutex::Autolock _l(mLock); 7723 for (size_t j = 0; j < mTracks.size(); ++j) { 7724 sp<RecordThread::RecordTrack> track = mTracks[j]; 7725 audio_session_t sessionId = track->sessionId(); 7726 if (ids.indexOfKey(sessionId) < 0) { 7727 ids.add(sessionId, true); 7728 } 7729 } 7730 return ids; 7731 } 7732 7733 AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7734 { 7735 Mutex::Autolock _l(mLock); 7736 AudioStreamIn *input = mInput; 7737 mInput = NULL; 7738 return input; 7739 } 7740 7741 // this method must always be called either with ThreadBase mLock held or inside the thread loop 7742 sp<StreamHalInterface> AudioFlinger::RecordThread::stream() const 7743 { 7744 if (mInput == NULL) { 7745 return NULL; 7746 } 7747 return mInput->stream; 7748 } 7749 7750 status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7751 { 7752 // only one chain per input thread 7753 if (mEffectChains.size() != 0) { 7754 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7755 return INVALID_OPERATION; 7756 } 7757 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7758 chain->setThread(this); 7759 chain->setInBuffer(NULL); 7760 chain->setOutBuffer(NULL); 7761 7762 checkSuspendOnAddEffectChain_l(chain); 7763 7764 // make sure enabled pre processing effects state is communicated to the HAL as we 7765 // just moved them to a new input stream. 7766 chain->syncHalEffectsState(); 7767 7768 mEffectChains.add(chain); 7769 7770 return NO_ERROR; 7771 } 7772 7773 size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7774 { 7775 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7776 ALOGW_IF(mEffectChains.size() != 1, 7777 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7778 chain.get(), mEffectChains.size(), this); 7779 if (mEffectChains.size() == 1) { 7780 mEffectChains.removeAt(0); 7781 } 7782 return 0; 7783 } 7784 7785 status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7786 audio_patch_handle_t *handle) 7787 { 7788 status_t status = NO_ERROR; 7789 7790 // store new device and send to effects 7791 mInDevice = patch->sources[0].ext.device.type; 7792 mPatch = *patch; 7793 for (size_t i = 0; i < mEffectChains.size(); i++) { 7794 mEffectChains[i]->setDevice_l(mInDevice); 7795 } 7796 7797 checkBtNrec_l(); 7798 7799 // store new source and send to effects 7800 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7801 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7802 for (size_t i = 0; i < mEffectChains.size(); i++) { 7803 mEffectChains[i]->setAudioSource_l(mAudioSource); 7804 } 7805 } 7806 7807 if (mInput->audioHwDev->supportsAudioPatches()) { 7808 sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice(); 7809 status = hwDevice->createAudioPatch(patch->num_sources, 7810 patch->sources, 7811 patch->num_sinks, 7812 patch->sinks, 7813 handle); 7814 } else { 7815 char *address; 7816 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7817 address = audio_device_address_to_parameter( 7818 patch->sources[0].ext.device.type, 7819 patch->sources[0].ext.device.address); 7820 } else { 7821 address = (char *)calloc(1, 1); 7822 } 7823 AudioParameter param = AudioParameter(String8(address)); 7824 free(address); 7825 param.addInt(String8(AudioParameter::keyRouting), 7826 (int)patch->sources[0].ext.device.type); 7827 param.addInt(String8(AudioParameter::keyInputSource), 7828 (int)patch->sinks[0].ext.mix.usecase.source); 7829 status = mInput->stream->setParameters(param.toString()); 7830 *handle = AUDIO_PATCH_HANDLE_NONE; 7831 } 7832 7833 if (mInDevice != mPrevInDevice) { 7834 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7835 mPrevInDevice = mInDevice; 7836 } 7837 7838 return status; 7839 } 7840 7841 status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7842 { 7843 status_t status = NO_ERROR; 7844 7845 mInDevice = AUDIO_DEVICE_NONE; 7846 7847 if (mInput->audioHwDev->supportsAudioPatches()) { 7848 sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice(); 7849 status = hwDevice->releaseAudioPatch(handle); 7850 } else { 7851 AudioParameter param; 7852 param.addInt(String8(AudioParameter::keyRouting), 0); 7853 status = mInput->stream->setParameters(param.toString()); 7854 } 7855 return status; 7856 } 7857 7858 void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7859 { 7860 Mutex::Autolock _l(mLock); 7861 mTracks.add(record); 7862 } 7863 7864 void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7865 { 7866 Mutex::Autolock _l(mLock); 7867 destroyTrack_l(record); 7868 } 7869 7870 void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7871 { 7872 ThreadBase::getAudioPortConfig(config); 7873 config->role = AUDIO_PORT_ROLE_SINK; 7874 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7875 config->ext.mix.usecase.source = mAudioSource; 7876 } 7877 7878 // ---------------------------------------------------------------------------- 7879 // Mmap 7880 // ---------------------------------------------------------------------------- 7881 7882 AudioFlinger::MmapThreadHandle::MmapThreadHandle(const sp<MmapThread>& thread) 7883 : mThread(thread) 7884 { 7885 assert(thread != 0); // thread must start non-null and stay non-null 7886 } 7887 7888 AudioFlinger::MmapThreadHandle::~MmapThreadHandle() 7889 { 7890 mThread->disconnect(); 7891 } 7892 7893 status_t AudioFlinger::MmapThreadHandle::createMmapBuffer(int32_t minSizeFrames, 7894 struct audio_mmap_buffer_info *info) 7895 { 7896 return mThread->createMmapBuffer(minSizeFrames, info); 7897 } 7898 7899 status_t AudioFlinger::MmapThreadHandle::getMmapPosition(struct audio_mmap_position *position) 7900 { 7901 return mThread->getMmapPosition(position); 7902 } 7903 7904 status_t AudioFlinger::MmapThreadHandle::start(const AudioClient& client, 7905 audio_port_handle_t *handle) 7906 7907 { 7908 return mThread->start(client, handle); 7909 } 7910 7911 status_t AudioFlinger::MmapThreadHandle::stop(audio_port_handle_t handle) 7912 { 7913 return mThread->stop(handle); 7914 } 7915 7916 status_t AudioFlinger::MmapThreadHandle::standby() 7917 { 7918 return mThread->standby(); 7919 } 7920 7921 7922 AudioFlinger::MmapThread::MmapThread( 7923 const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 7924 AudioHwDevice *hwDev, sp<StreamHalInterface> stream, 7925 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady) 7926 : ThreadBase(audioFlinger, id, outDevice, inDevice, MMAP, systemReady), 7927 mSessionId(AUDIO_SESSION_NONE), 7928 mDeviceId(AUDIO_PORT_HANDLE_NONE), mPortId(AUDIO_PORT_HANDLE_NONE), 7929 mHalStream(stream), mHalDevice(hwDev->hwDevice()), mAudioHwDev(hwDev), 7930 mActiveTracks(&this->mLocalLog), 7931 mHalVolFloat(-1.0f), // Initialize to illegal value so it always gets set properly later. 7932 mNoCallbackWarningCount(0) 7933 { 7934 mStandby = true; 7935 readHalParameters_l(); 7936 } 7937 7938 AudioFlinger::MmapThread::~MmapThread() 7939 { 7940 releaseWakeLock_l(); 7941 } 7942 7943 void AudioFlinger::MmapThread::onFirstRef() 7944 { 7945 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 7946 } 7947 7948 void AudioFlinger::MmapThread::disconnect() 7949 { 7950 ActiveTracks<MmapTrack> activeTracks; 7951 { 7952 Mutex::Autolock _l(mLock); 7953 for (const sp<MmapTrack> &t : mActiveTracks) { 7954 activeTracks.add(t); 7955 } 7956 } 7957 for (const sp<MmapTrack> &t : activeTracks) { 7958 stop(t->portId()); 7959 } 7960 // This will decrement references and may cause the destruction of this thread. 7961 if (isOutput()) { 7962 AudioSystem::releaseOutput(mId, streamType(), mSessionId); 7963 } else { 7964 AudioSystem::releaseInput(mPortId); 7965 } 7966 } 7967 7968 7969 void AudioFlinger::MmapThread::configure(const audio_attributes_t *attr, 7970 audio_stream_type_t streamType __unused, 7971 audio_session_t sessionId, 7972 const sp<MmapStreamCallback>& callback, 7973 audio_port_handle_t deviceId, 7974 audio_port_handle_t portId) 7975 { 7976 mAttr = *attr; 7977 mSessionId = sessionId; 7978 mCallback = callback; 7979 mDeviceId = deviceId; 7980 mPortId = portId; 7981 } 7982 7983 status_t AudioFlinger::MmapThread::createMmapBuffer(int32_t minSizeFrames, 7984 struct audio_mmap_buffer_info *info) 7985 { 7986 if (mHalStream == 0) { 7987 return NO_INIT; 7988 } 7989 mStandby = true; 7990 acquireWakeLock(); 7991 return mHalStream->createMmapBuffer(minSizeFrames, info); 7992 } 7993 7994 status_t AudioFlinger::MmapThread::getMmapPosition(struct audio_mmap_position *position) 7995 { 7996 if (mHalStream == 0) { 7997 return NO_INIT; 7998 } 7999 return mHalStream->getMmapPosition(position); 8000 } 8001 8002 status_t AudioFlinger::MmapThread::exitStandby() 8003 { 8004 status_t ret = mHalStream->start(); 8005 if (ret != NO_ERROR) { 8006 ALOGE("%s: error mHalStream->start() = %d for first track", __FUNCTION__, ret); 8007 return ret; 8008 } 8009 mStandby = false; 8010 return NO_ERROR; 8011 } 8012 8013 status_t AudioFlinger::MmapThread::start(const AudioClient& client, 8014 audio_port_handle_t *handle) 8015 { 8016 ALOGV("%s clientUid %d mStandby %d mPortId %d *handle %d", __FUNCTION__, 8017 client.clientUid, mStandby, mPortId, *handle); 8018 if (mHalStream == 0) { 8019 return NO_INIT; 8020 } 8021 8022 status_t ret; 8023 8024 if (*handle == mPortId) { 8025 // for the first track, reuse portId and session allocated when the stream was opened 8026 return exitStandby(); 8027 } 8028 8029 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE; 8030 8031 audio_io_handle_t io = mId; 8032 if (isOutput()) { 8033 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 8034 config.sample_rate = mSampleRate; 8035 config.channel_mask = mChannelMask; 8036 config.format = mFormat; 8037 audio_stream_type_t stream = streamType(); 8038 audio_output_flags_t flags = 8039 (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT); 8040 audio_port_handle_t deviceId = mDeviceId; 8041 ret = AudioSystem::getOutputForAttr(&mAttr, &io, 8042 mSessionId, 8043 &stream, 8044 client.clientPid, 8045 client.clientUid, 8046 &config, 8047 flags, 8048 &deviceId, 8049 &portId); 8050 } else { 8051 audio_config_base_t config; 8052 config.sample_rate = mSampleRate; 8053 config.channel_mask = mChannelMask; 8054 config.format = mFormat; 8055 audio_port_handle_t deviceId = mDeviceId; 8056 ret = AudioSystem::getInputForAttr(&mAttr, &io, 8057 mSessionId, 8058 client.clientPid, 8059 client.clientUid, 8060 client.packageName, 8061 &config, 8062 AUDIO_INPUT_FLAG_MMAP_NOIRQ, 8063 &deviceId, 8064 &portId); 8065 } 8066 // APM should not chose a different input or output stream for the same set of attributes 8067 // and audo configuration 8068 if (ret != NO_ERROR || io != mId) { 8069 ALOGE("%s: error getting output or input from APM (error %d, io %d expected io %d)", 8070 __FUNCTION__, ret, io, mId); 8071 return BAD_VALUE; 8072 } 8073 8074 bool silenced = false; 8075 if (isOutput()) { 8076 ret = AudioSystem::startOutput(mId, streamType(), mSessionId); 8077 } else { 8078 ret = AudioSystem::startInput(portId, &silenced); 8079 } 8080 8081 Mutex::Autolock _l(mLock); 8082 // abort if start is rejected by audio policy manager 8083 if (ret != NO_ERROR) { 8084 ALOGE("%s: error start rejected by AudioPolicyManager = %d", __FUNCTION__, ret); 8085 if (mActiveTracks.size() != 0) { 8086 mLock.unlock(); 8087 if (isOutput()) { 8088 AudioSystem::releaseOutput(mId, streamType(), mSessionId); 8089 } else { 8090 AudioSystem::releaseInput(portId); 8091 } 8092 mLock.lock(); 8093 } else { 8094 mHalStream->stop(); 8095 } 8096 return PERMISSION_DENIED; 8097 } 8098 8099 if (isOutput()) { 8100 // force volume update when a new track is added 8101 mHalVolFloat = -1.0f; 8102 } else if (!silenced) { 8103 for (const sp<MmapTrack> &track : mActiveTracks) { 8104 if (track->isSilenced_l() && track->uid() != client.clientUid) 8105 track->invalidate(); 8106 } 8107 } 8108 8109 // Given that MmapThread::mAttr is mutable, should a MmapTrack have attributes ? 8110 sp<MmapTrack> track = new MmapTrack(this, mAttr, mSampleRate, mFormat, mChannelMask, mSessionId, 8111 client.clientUid, client.clientPid, portId); 8112 8113 track->setSilenced_l(silenced); 8114 mActiveTracks.add(track); 8115 sp<EffectChain> chain = getEffectChain_l(mSessionId); 8116 if (chain != 0) { 8117 chain->setStrategy(AudioSystem::getStrategyForStream(streamType())); 8118 chain->incTrackCnt(); 8119 chain->incActiveTrackCnt(); 8120 } 8121 8122 *handle = portId; 8123 broadcast_l(); 8124 8125 ALOGV("%s DONE handle %d stream %p", __FUNCTION__, *handle, mHalStream.get()); 8126 8127 return NO_ERROR; 8128 } 8129 8130 status_t AudioFlinger::MmapThread::stop(audio_port_handle_t handle) 8131 { 8132 ALOGV("%s handle %d", __FUNCTION__, handle); 8133 8134 if (mHalStream == 0) { 8135 return NO_INIT; 8136 } 8137 8138 if (handle == mPortId) { 8139 mHalStream->stop(); 8140 return NO_ERROR; 8141 } 8142 8143 Mutex::Autolock _l(mLock); 8144 8145 sp<MmapTrack> track; 8146 for (const sp<MmapTrack> &t : mActiveTracks) { 8147 if (handle == t->portId()) { 8148 track = t; 8149 break; 8150 } 8151 } 8152 if (track == 0) { 8153 return BAD_VALUE; 8154 } 8155 8156 mActiveTracks.remove(track); 8157 8158 mLock.unlock(); 8159 if (isOutput()) { 8160 AudioSystem::stopOutput(mId, streamType(), track->sessionId()); 8161 AudioSystem::releaseOutput(mId, streamType(), track->sessionId()); 8162 } else { 8163 AudioSystem::stopInput(track->portId()); 8164 AudioSystem::releaseInput(track->portId()); 8165 } 8166 mLock.lock(); 8167 8168 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 8169 if (chain != 0) { 8170 chain->decActiveTrackCnt(); 8171 chain->decTrackCnt(); 8172 } 8173 8174 broadcast_l(); 8175 8176 return NO_ERROR; 8177 } 8178 8179 status_t AudioFlinger::MmapThread::standby() 8180 { 8181 ALOGV("%s", __FUNCTION__); 8182 8183 if (mHalStream == 0) { 8184 return NO_INIT; 8185 } 8186 if (mActiveTracks.size() != 0) { 8187 return INVALID_OPERATION; 8188 } 8189 mHalStream->standby(); 8190 mStandby = true; 8191 releaseWakeLock(); 8192 return NO_ERROR; 8193 } 8194 8195 8196 void AudioFlinger::MmapThread::readHalParameters_l() 8197 { 8198 status_t result = mHalStream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat); 8199 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result); 8200 mFormat = mHALFormat; 8201 LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat); 8202 result = mHalStream->getFrameSize(&mFrameSize); 8203 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result); 8204 result = mHalStream->getBufferSize(&mBufferSize); 8205 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result); 8206 mFrameCount = mBufferSize / mFrameSize; 8207 } 8208 8209 bool AudioFlinger::MmapThread::threadLoop() 8210 { 8211 checkSilentMode_l(); 8212 8213 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 8214 8215 while (!exitPending()) 8216 { 8217 Mutex::Autolock _l(mLock); 8218 Vector< sp<EffectChain> > effectChains; 8219 8220 if (mSignalPending) { 8221 // A signal was raised while we were unlocked 8222 mSignalPending = false; 8223 } else { 8224 if (mConfigEvents.isEmpty()) { 8225 // we're about to wait, flush the binder command buffer 8226 IPCThreadState::self()->flushCommands(); 8227 8228 if (exitPending()) { 8229 break; 8230 } 8231 8232 // wait until we have something to do... 8233 ALOGV("%s going to sleep", myName.string()); 8234 mWaitWorkCV.wait(mLock); 8235 ALOGV("%s waking up", myName.string()); 8236 8237 checkSilentMode_l(); 8238 8239 continue; 8240 } 8241 } 8242 8243 processConfigEvents_l(); 8244 8245 processVolume_l(); 8246 8247 checkInvalidTracks_l(); 8248 8249 mActiveTracks.updatePowerState(this); 8250 8251 updateMetadata_l(); 8252 8253 lockEffectChains_l(effectChains); 8254 for (size_t i = 0; i < effectChains.size(); i ++) { 8255 effectChains[i]->process_l(); 8256 } 8257 // enable changes in effect chain 8258 unlockEffectChains(effectChains); 8259 // Effect chains will be actually deleted here if they were removed from 8260 // mEffectChains list during mixing or effects processing 8261 } 8262 8263 threadLoop_exit(); 8264 8265 if (!mStandby) { 8266 threadLoop_standby(); 8267 mStandby = true; 8268 } 8269 8270 ALOGV("Thread %p type %d exiting", this, mType); 8271 return false; 8272 } 8273 8274 // checkForNewParameter_l() must be called with ThreadBase::mLock held 8275 bool AudioFlinger::MmapThread::checkForNewParameter_l(const String8& keyValuePair, 8276 status_t& status) 8277 { 8278 AudioParameter param = AudioParameter(keyValuePair); 8279 int value; 8280 bool sendToHal = true; 8281 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 8282 audio_devices_t device = (audio_devices_t)value; 8283 // forward device change to effects that have requested to be 8284 // aware of attached audio device. 8285 if (device != AUDIO_DEVICE_NONE) { 8286 for (size_t i = 0; i < mEffectChains.size(); i++) { 8287 mEffectChains[i]->setDevice_l(device); 8288 } 8289 } 8290 if (audio_is_output_devices(device)) { 8291 mOutDevice = device; 8292 if (!isOutput()) { 8293 sendToHal = false; 8294 } 8295 } else { 8296 mInDevice = device; 8297 if (device != AUDIO_DEVICE_NONE) { 8298 mPrevInDevice = value; 8299 } 8300 // TODO: implement and call checkBtNrec_l(); 8301 } 8302 } 8303 if (sendToHal) { 8304 status = mHalStream->setParameters(keyValuePair); 8305 } else { 8306 status = NO_ERROR; 8307 } 8308 8309 return false; 8310 } 8311 8312 String8 AudioFlinger::MmapThread::getParameters(const String8& keys) 8313 { 8314 Mutex::Autolock _l(mLock); 8315 String8 out_s8; 8316 if (initCheck() == NO_ERROR && mHalStream->getParameters(keys, &out_s8) == OK) { 8317 return out_s8; 8318 } 8319 return String8(); 8320 } 8321 8322 void AudioFlinger::MmapThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 8323 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 8324 8325 desc->mIoHandle = mId; 8326 8327 switch (event) { 8328 case AUDIO_INPUT_OPENED: 8329 case AUDIO_INPUT_REGISTERED: 8330 case AUDIO_INPUT_CONFIG_CHANGED: 8331 case AUDIO_OUTPUT_OPENED: 8332 case AUDIO_OUTPUT_REGISTERED: 8333 case AUDIO_OUTPUT_CONFIG_CHANGED: 8334 desc->mPatch = mPatch; 8335 desc->mChannelMask = mChannelMask; 8336 desc->mSamplingRate = mSampleRate; 8337 desc->mFormat = mFormat; 8338 desc->mFrameCount = mFrameCount; 8339 desc->mFrameCountHAL = mFrameCount; 8340 desc->mLatency = 0; 8341 break; 8342 8343 case AUDIO_INPUT_CLOSED: 8344 case AUDIO_OUTPUT_CLOSED: 8345 default: 8346 break; 8347 } 8348 mAudioFlinger->ioConfigChanged(event, desc, pid); 8349 } 8350 8351 status_t AudioFlinger::MmapThread::createAudioPatch_l(const struct audio_patch *patch, 8352 audio_patch_handle_t *handle) 8353 { 8354 status_t status = NO_ERROR; 8355 8356 // store new device and send to effects 8357 audio_devices_t type = AUDIO_DEVICE_NONE; 8358 audio_port_handle_t deviceId; 8359 if (isOutput()) { 8360 for (unsigned int i = 0; i < patch->num_sinks; i++) { 8361 type |= patch->sinks[i].ext.device.type; 8362 } 8363 deviceId = patch->sinks[0].id; 8364 } else { 8365 type = patch->sources[0].ext.device.type; 8366 deviceId = patch->sources[0].id; 8367 } 8368 8369 for (size_t i = 0; i < mEffectChains.size(); i++) { 8370 mEffectChains[i]->setDevice_l(type); 8371 } 8372 8373 if (isOutput()) { 8374 mOutDevice = type; 8375 } else { 8376 mInDevice = type; 8377 // store new source and send to effects 8378 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 8379 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 8380 for (size_t i = 0; i < mEffectChains.size(); i++) { 8381 mEffectChains[i]->setAudioSource_l(mAudioSource); 8382 } 8383 } 8384 } 8385 8386 if (mAudioHwDev->supportsAudioPatches()) { 8387 status = mHalDevice->createAudioPatch(patch->num_sources, 8388 patch->sources, 8389 patch->num_sinks, 8390 patch->sinks, 8391 handle); 8392 } else { 8393 char *address; 8394 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 8395 //FIXME: we only support address on first sink with HAL version < 3.0 8396 address = audio_device_address_to_parameter( 8397 patch->sinks[0].ext.device.type, 8398 patch->sinks[0].ext.device.address); 8399 } else { 8400 address = (char *)calloc(1, 1); 8401 } 8402 AudioParameter param = AudioParameter(String8(address)); 8403 free(address); 8404 param.addInt(String8(AudioParameter::keyRouting), (int)type); 8405 if (!isOutput()) { 8406 param.addInt(String8(AudioParameter::keyInputSource), 8407 (int)patch->sinks[0].ext.mix.usecase.source); 8408 } 8409 status = mHalStream->setParameters(param.toString()); 8410 *handle = AUDIO_PATCH_HANDLE_NONE; 8411 } 8412 8413 if (isOutput() && mPrevOutDevice != mOutDevice) { 8414 mPrevOutDevice = type; 8415 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 8416 sp<MmapStreamCallback> callback = mCallback.promote(); 8417 if (mDeviceId != deviceId && callback != 0) { 8418 mLock.unlock(); 8419 callback->onRoutingChanged(deviceId); 8420 mLock.lock(); 8421 } 8422 mDeviceId = deviceId; 8423 } 8424 if (!isOutput() && mPrevInDevice != mInDevice) { 8425 mPrevInDevice = type; 8426 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 8427 sp<MmapStreamCallback> callback = mCallback.promote(); 8428 if (mDeviceId != deviceId && callback != 0) { 8429 mLock.unlock(); 8430 callback->onRoutingChanged(deviceId); 8431 mLock.lock(); 8432 } 8433 mDeviceId = deviceId; 8434 } 8435 return status; 8436 } 8437 8438 status_t AudioFlinger::MmapThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 8439 { 8440 status_t status = NO_ERROR; 8441 8442 mInDevice = AUDIO_DEVICE_NONE; 8443 8444 bool supportsAudioPatches = mHalDevice->supportsAudioPatches(&supportsAudioPatches) == OK ? 8445 supportsAudioPatches : false; 8446 8447 if (supportsAudioPatches) { 8448 status = mHalDevice->releaseAudioPatch(handle); 8449 } else { 8450 AudioParameter param; 8451 param.addInt(String8(AudioParameter::keyRouting), 0); 8452 status = mHalStream->setParameters(param.toString()); 8453 } 8454 return status; 8455 } 8456 8457 void AudioFlinger::MmapThread::getAudioPortConfig(struct audio_port_config *config) 8458 { 8459 ThreadBase::getAudioPortConfig(config); 8460 if (isOutput()) { 8461 config->role = AUDIO_PORT_ROLE_SOURCE; 8462 config->ext.mix.hw_module = mAudioHwDev->handle(); 8463 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 8464 } else { 8465 config->role = AUDIO_PORT_ROLE_SINK; 8466 config->ext.mix.hw_module = mAudioHwDev->handle(); 8467 config->ext.mix.usecase.source = mAudioSource; 8468 } 8469 } 8470 8471 status_t AudioFlinger::MmapThread::addEffectChain_l(const sp<EffectChain>& chain) 8472 { 8473 audio_session_t session = chain->sessionId(); 8474 8475 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8476 // Attach all tracks with same session ID to this chain. 8477 // indicate all active tracks in the chain 8478 for (const sp<MmapTrack> &track : mActiveTracks) { 8479 if (session == track->sessionId()) { 8480 chain->incTrackCnt(); 8481 chain->incActiveTrackCnt(); 8482 } 8483 } 8484 8485 chain->setThread(this); 8486 chain->setInBuffer(nullptr); 8487 chain->setOutBuffer(nullptr); 8488 chain->syncHalEffectsState(); 8489 8490 mEffectChains.add(chain); 8491 checkSuspendOnAddEffectChain_l(chain); 8492 return NO_ERROR; 8493 } 8494 8495 size_t AudioFlinger::MmapThread::removeEffectChain_l(const sp<EffectChain>& chain) 8496 { 8497 audio_session_t session = chain->sessionId(); 8498 8499 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8500 8501 for (size_t i = 0; i < mEffectChains.size(); i++) { 8502 if (chain == mEffectChains[i]) { 8503 mEffectChains.removeAt(i); 8504 // detach all active tracks from the chain 8505 // detach all tracks with same session ID from this chain 8506 for (const sp<MmapTrack> &track : mActiveTracks) { 8507 if (session == track->sessionId()) { 8508 chain->decActiveTrackCnt(); 8509 chain->decTrackCnt(); 8510 } 8511 } 8512 break; 8513 } 8514 } 8515 return mEffectChains.size(); 8516 } 8517 8518 // hasAudioSession_l() must be called with ThreadBase::mLock held 8519 uint32_t AudioFlinger::MmapThread::hasAudioSession_l(audio_session_t sessionId) const 8520 { 8521 uint32_t result = 0; 8522 if (getEffectChain_l(sessionId) != 0) { 8523 result = EFFECT_SESSION; 8524 } 8525 8526 for (size_t i = 0; i < mActiveTracks.size(); i++) { 8527 sp<MmapTrack> track = mActiveTracks[i]; 8528 if (sessionId == track->sessionId()) { 8529 result |= TRACK_SESSION; 8530 if (track->isFastTrack()) { 8531 result |= FAST_SESSION; 8532 } 8533 break; 8534 } 8535 } 8536 8537 return result; 8538 } 8539 8540 void AudioFlinger::MmapThread::threadLoop_standby() 8541 { 8542 mHalStream->standby(); 8543 } 8544 8545 void AudioFlinger::MmapThread::threadLoop_exit() 8546 { 8547 // Do not call callback->onTearDown() because it is redundant for thread exit 8548 // and because it can cause a recursive mutex lock on stop(). 8549 } 8550 8551 status_t AudioFlinger::MmapThread::setSyncEvent(const sp<SyncEvent>& event __unused) 8552 { 8553 return BAD_VALUE; 8554 } 8555 8556 bool AudioFlinger::MmapThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 8557 { 8558 return false; 8559 } 8560 8561 status_t AudioFlinger::MmapThread::checkEffectCompatibility_l( 8562 const effect_descriptor_t *desc, audio_session_t sessionId) 8563 { 8564 // No global effect sessions on mmap threads 8565 if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 8566 ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s", 8567 desc->name, mThreadName); 8568 return BAD_VALUE; 8569 } 8570 8571 if (!isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC)) { 8572 ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on capture mmap thread", 8573 desc->name); 8574 return BAD_VALUE; 8575 } 8576 if (isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 8577 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback mmap " 8578 "thread", desc->name); 8579 return BAD_VALUE; 8580 } 8581 8582 // Only allow effects without processing load or latency 8583 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) != EFFECT_FLAG_NO_PROCESS) { 8584 return BAD_VALUE; 8585 } 8586 8587 return NO_ERROR; 8588 8589 } 8590 8591 void AudioFlinger::MmapThread::checkInvalidTracks_l() 8592 { 8593 for (const sp<MmapTrack> &track : mActiveTracks) { 8594 if (track->isInvalid()) { 8595 sp<MmapStreamCallback> callback = mCallback.promote(); 8596 if (callback != 0) { 8597 mLock.unlock(); 8598 callback->onTearDown(track->portId()); 8599 mLock.lock(); 8600 } else if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) { 8601 ALOGW("Could not notify MMAP stream tear down: no onTearDown callback!"); 8602 mNoCallbackWarningCount++; 8603 } 8604 } 8605 } 8606 } 8607 8608 void AudioFlinger::MmapThread::dump(int fd, const Vector<String16>& args) 8609 { 8610 dumpInternals(fd, args); 8611 dumpTracks(fd, args); 8612 dumpEffectChains(fd, args); 8613 dprintf(fd, " Local log:\n"); 8614 mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */); 8615 } 8616 8617 void AudioFlinger::MmapThread::dumpInternals(int fd, const Vector<String16>& args) 8618 { 8619 dumpBase(fd, args); 8620 8621 dprintf(fd, " Attributes: content type %d usage %d source %d\n", 8622 mAttr.content_type, mAttr.usage, mAttr.source); 8623 dprintf(fd, " Session: %d port Id: %d\n", mSessionId, mPortId); 8624 if (mActiveTracks.size() == 0) { 8625 dprintf(fd, " No active clients\n"); 8626 } 8627 } 8628 8629 void AudioFlinger::MmapThread::dumpTracks(int fd, const Vector<String16>& args __unused) 8630 { 8631 String8 result; 8632 size_t numtracks = mActiveTracks.size(); 8633 dprintf(fd, " %zu Tracks\n", numtracks); 8634 const char *prefix = " "; 8635 if (numtracks) { 8636 result.append(prefix); 8637 MmapTrack::appendDumpHeader(result); 8638 for (size_t i = 0; i < numtracks ; ++i) { 8639 sp<MmapTrack> track = mActiveTracks[i]; 8640 result.append(prefix); 8641 track->appendDump(result, true /* active */); 8642 } 8643 } else { 8644 dprintf(fd, "\n"); 8645 } 8646 write(fd, result.string(), result.size()); 8647 } 8648 8649 AudioFlinger::MmapPlaybackThread::MmapPlaybackThread( 8650 const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 8651 AudioHwDevice *hwDev, AudioStreamOut *output, 8652 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady) 8653 : MmapThread(audioFlinger, id, hwDev, output->stream, outDevice, inDevice, systemReady), 8654 mStreamType(AUDIO_STREAM_MUSIC), 8655 mStreamVolume(1.0), 8656 mStreamMute(false), 8657 mOutput(output) 8658 { 8659 snprintf(mThreadName, kThreadNameLength, "AudioMmapOut_%X", id); 8660 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 8661 mMasterVolume = audioFlinger->masterVolume_l(); 8662 mMasterMute = audioFlinger->masterMute_l(); 8663 if (mAudioHwDev) { 8664 if (mAudioHwDev->canSetMasterVolume()) { 8665 mMasterVolume = 1.0; 8666 } 8667 8668 if (mAudioHwDev->canSetMasterMute()) { 8669 mMasterMute = false; 8670 } 8671 } 8672 } 8673 8674 void AudioFlinger::MmapPlaybackThread::configure(const audio_attributes_t *attr, 8675 audio_stream_type_t streamType, 8676 audio_session_t sessionId, 8677 const sp<MmapStreamCallback>& callback, 8678 audio_port_handle_t deviceId, 8679 audio_port_handle_t portId) 8680 { 8681 MmapThread::configure(attr, streamType, sessionId, callback, deviceId, portId); 8682 mStreamType = streamType; 8683 } 8684 8685 AudioStreamOut* AudioFlinger::MmapPlaybackThread::clearOutput() 8686 { 8687 Mutex::Autolock _l(mLock); 8688 AudioStreamOut *output = mOutput; 8689 mOutput = NULL; 8690 return output; 8691 } 8692 8693 void AudioFlinger::MmapPlaybackThread::setMasterVolume(float value) 8694 { 8695 Mutex::Autolock _l(mLock); 8696 // Don't apply master volume in SW if our HAL can do it for us. 8697 if (mAudioHwDev && 8698 mAudioHwDev->canSetMasterVolume()) { 8699 mMasterVolume = 1.0; 8700 } else { 8701 mMasterVolume = value; 8702 } 8703 } 8704 8705 void AudioFlinger::MmapPlaybackThread::setMasterMute(bool muted) 8706 { 8707 Mutex::Autolock _l(mLock); 8708 // Don't apply master mute in SW if our HAL can do it for us. 8709 if (mAudioHwDev && mAudioHwDev->canSetMasterMute()) { 8710 mMasterMute = false; 8711 } else { 8712 mMasterMute = muted; 8713 } 8714 } 8715 8716 void AudioFlinger::MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 8717 { 8718 Mutex::Autolock _l(mLock); 8719 if (stream == mStreamType) { 8720 mStreamVolume = value; 8721 broadcast_l(); 8722 } 8723 } 8724 8725 float AudioFlinger::MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const 8726 { 8727 Mutex::Autolock _l(mLock); 8728 if (stream == mStreamType) { 8729 return mStreamVolume; 8730 } 8731 return 0.0f; 8732 } 8733 8734 void AudioFlinger::MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 8735 { 8736 Mutex::Autolock _l(mLock); 8737 if (stream == mStreamType) { 8738 mStreamMute= muted; 8739 broadcast_l(); 8740 } 8741 } 8742 8743 void AudioFlinger::MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType) 8744 { 8745 Mutex::Autolock _l(mLock); 8746 if (streamType == mStreamType) { 8747 for (const sp<MmapTrack> &track : mActiveTracks) { 8748 track->invalidate(); 8749 } 8750 broadcast_l(); 8751 } 8752 } 8753 8754 void AudioFlinger::MmapPlaybackThread::processVolume_l() 8755 { 8756 float volume; 8757 8758 if (mMasterMute || mStreamMute) { 8759 volume = 0; 8760 } else { 8761 volume = mMasterVolume * mStreamVolume; 8762 } 8763 8764 if (volume != mHalVolFloat) { 8765 8766 // Convert volumes from float to 8.24 8767 uint32_t vol = (uint32_t)(volume * (1 << 24)); 8768 8769 // Delegate volume control to effect in track effect chain if needed 8770 // only one effect chain can be present on DirectOutputThread, so if 8771 // there is one, the track is connected to it 8772 if (!mEffectChains.isEmpty()) { 8773 mEffectChains[0]->setVolume_l(&vol, &vol); 8774 volume = (float)vol / (1 << 24); 8775 } 8776 // Try to use HW volume control and fall back to SW control if not implemented 8777 if (mOutput->stream->setVolume(volume, volume) == NO_ERROR) { 8778 mHalVolFloat = volume; // HW volume control worked, so update value. 8779 mNoCallbackWarningCount = 0; 8780 } else { 8781 sp<MmapStreamCallback> callback = mCallback.promote(); 8782 if (callback != 0) { 8783 int channelCount; 8784 if (isOutput()) { 8785 channelCount = audio_channel_count_from_out_mask(mChannelMask); 8786 } else { 8787 channelCount = audio_channel_count_from_in_mask(mChannelMask); 8788 } 8789 Vector<float> values; 8790 for (int i = 0; i < channelCount; i++) { 8791 values.add(volume); 8792 } 8793 mHalVolFloat = volume; // SW volume control worked, so update value. 8794 mNoCallbackWarningCount = 0; 8795 mLock.unlock(); 8796 callback->onVolumeChanged(mChannelMask, values); 8797 mLock.lock(); 8798 } else { 8799 if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) { 8800 ALOGW("Could not set MMAP stream volume: no volume callback!"); 8801 mNoCallbackWarningCount++; 8802 } 8803 } 8804 } 8805 } 8806 } 8807 8808 void AudioFlinger::MmapPlaybackThread::updateMetadata_l() 8809 { 8810 if (mOutput == nullptr || mOutput->stream == nullptr || 8811 !mActiveTracks.readAndClearHasChanged()) { 8812 return; 8813 } 8814 StreamOutHalInterface::SourceMetadata metadata; 8815 for (const sp<MmapTrack> &track : mActiveTracks) { 8816 // No track is invalid as this is called after prepareTrack_l in the same critical section 8817 metadata.tracks.push_back({ 8818 .usage = track->attributes().usage, 8819 .content_type = track->attributes().content_type, 8820 .gain = mHalVolFloat, // TODO: propagate from aaudio pre-mix volume 8821 }); 8822 } 8823 mOutput->stream->updateSourceMetadata(metadata); 8824 } 8825 8826 void AudioFlinger::MmapPlaybackThread::checkSilentMode_l() 8827 { 8828 if (!mMasterMute) { 8829 char value[PROPERTY_VALUE_MAX]; 8830 if (property_get("ro.audio.silent", value, "0") > 0) { 8831 char *endptr; 8832 unsigned long ul = strtoul(value, &endptr, 0); 8833 if (*endptr == '\0' && ul != 0) { 8834 ALOGD("Silence is golden"); 8835 // The setprop command will not allow a property to be changed after 8836 // the first time it is set, so we don't have to worry about un-muting. 8837 setMasterMute_l(true); 8838 } 8839 } 8840 } 8841 } 8842 8843 void AudioFlinger::MmapPlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 8844 { 8845 MmapThread::dumpInternals(fd, args); 8846 8847 dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n", 8848 mStreamType, mStreamVolume, mHalVolFloat, mStreamMute); 8849 dprintf(fd, " Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute); 8850 } 8851 8852 AudioFlinger::MmapCaptureThread::MmapCaptureThread( 8853 const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 8854 AudioHwDevice *hwDev, AudioStreamIn *input, 8855 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady) 8856 : MmapThread(audioFlinger, id, hwDev, input->stream, outDevice, inDevice, systemReady), 8857 mInput(input) 8858 { 8859 snprintf(mThreadName, kThreadNameLength, "AudioMmapIn_%X", id); 8860 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 8861 } 8862 8863 status_t AudioFlinger::MmapCaptureThread::exitStandby() 8864 { 8865 mInput->stream->setGain(1.0f); 8866 return MmapThread::exitStandby(); 8867 } 8868 8869 AudioFlinger::AudioStreamIn* AudioFlinger::MmapCaptureThread::clearInput() 8870 { 8871 Mutex::Autolock _l(mLock); 8872 AudioStreamIn *input = mInput; 8873 mInput = NULL; 8874 return input; 8875 } 8876 8877 8878 void AudioFlinger::MmapCaptureThread::processVolume_l() 8879 { 8880 bool changed = false; 8881 bool silenced = false; 8882 8883 sp<MmapStreamCallback> callback = mCallback.promote(); 8884 if (callback == 0) { 8885 if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) { 8886 ALOGW("Could not set MMAP stream silenced: no onStreamSilenced callback!"); 8887 mNoCallbackWarningCount++; 8888 } 8889 } 8890 8891 // After a change occurred in track silenced state, mute capture in audio DSP if at least one 8892 // track is silenced and unmute otherwise 8893 for (size_t i = 0; i < mActiveTracks.size() && !silenced; i++) { 8894 if (!mActiveTracks[i]->getAndSetSilencedNotified_l()) { 8895 changed = true; 8896 silenced = mActiveTracks[i]->isSilenced_l(); 8897 } 8898 } 8899 8900 if (changed) { 8901 mInput->stream->setGain(silenced ? 0.0f: 1.0f); 8902 } 8903 } 8904 8905 void AudioFlinger::MmapCaptureThread::updateMetadata_l() 8906 { 8907 if (mInput == nullptr || mInput->stream == nullptr || 8908 !mActiveTracks.readAndClearHasChanged()) { 8909 return; 8910 } 8911 StreamInHalInterface::SinkMetadata metadata; 8912 for (const sp<MmapTrack> &track : mActiveTracks) { 8913 // No track is invalid as this is called after prepareTrack_l in the same critical section 8914 metadata.tracks.push_back({ 8915 .source = track->attributes().source, 8916 .gain = 1, // capture tracks do not have volumes 8917 }); 8918 } 8919 mInput->stream->updateSinkMetadata(metadata); 8920 } 8921 8922 void AudioFlinger::MmapCaptureThread::setRecordSilenced(uid_t uid, bool silenced) 8923 { 8924 Mutex::Autolock _l(mLock); 8925 for (size_t i = 0; i < mActiveTracks.size() ; i++) { 8926 if (mActiveTracks[i]->uid() == uid) { 8927 mActiveTracks[i]->setSilenced_l(silenced); 8928 broadcast_l(); 8929 } 8930 } 8931 } 8932 8933 } // namespace android 8934