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 "AudioFlinger.h" 60 #include "FastMixer.h" 61 #include "FastCapture.h" 62 #include "ServiceUtilities.h" 63 #include "mediautils/SchedulingPolicyService.h" 64 65 #ifdef ADD_BATTERY_DATA 66 #include <media/IMediaPlayerService.h> 67 #include <media/IMediaDeathNotifier.h> 68 #endif 69 70 #ifdef DEBUG_CPU_USAGE 71 #include <cpustats/CentralTendencyStatistics.h> 72 #include <cpustats/ThreadCpuUsage.h> 73 #endif 74 75 #include "AutoPark.h" 76 77 #include <pthread.h> 78 #include "TypedLogger.h" 79 80 // ---------------------------------------------------------------------------- 81 82 // Note: the following macro is used for extremely verbose logging message. In 83 // order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 84 // 0; but one side effect of this is to turn all LOGV's as well. Some messages 85 // are so verbose that we want to suppress them even when we have ALOG_ASSERT 86 // turned on. Do not uncomment the #def below unless you really know what you 87 // are doing and want to see all of the extremely verbose messages. 88 //#define VERY_VERY_VERBOSE_LOGGING 89 #ifdef VERY_VERY_VERBOSE_LOGGING 90 #define ALOGVV ALOGV 91 #else 92 #define ALOGVV(a...) do { } while(0) 93 #endif 94 95 // TODO: Move these macro/inlines to a header file. 96 #define max(a, b) ((a) > (b) ? (a) : (b)) 97 template <typename T> 98 static inline T min(const T& a, const T& b) 99 { 100 return a < b ? a : b; 101 } 102 103 namespace android { 104 105 // retry counts for buffer fill timeout 106 // 50 * ~20msecs = 1 second 107 static const int8_t kMaxTrackRetries = 50; 108 static const int8_t kMaxTrackStartupRetries = 50; 109 // allow less retry attempts on direct output thread. 110 // direct outputs can be a scarce resource in audio hardware and should 111 // be released as quickly as possible. 112 static const int8_t kMaxTrackRetriesDirect = 2; 113 114 115 116 // don't warn about blocked writes or record buffer overflows more often than this 117 static const nsecs_t kWarningThrottleNs = seconds(5); 118 119 // RecordThread loop sleep time upon application overrun or audio HAL read error 120 static const int kRecordThreadSleepUs = 5000; 121 122 // maximum time to wait in sendConfigEvent_l() for a status to be received 123 static const nsecs_t kConfigEventTimeoutNs = seconds(2); 124 125 // minimum sleep time for the mixer thread loop when tracks are active but in underrun 126 static const uint32_t kMinThreadSleepTimeUs = 5000; 127 // maximum divider applied to the active sleep time in the mixer thread loop 128 static const uint32_t kMaxThreadSleepTimeShift = 2; 129 130 // minimum normal sink buffer size, expressed in milliseconds rather than frames 131 // FIXME This should be based on experimentally observed scheduling jitter 132 static const uint32_t kMinNormalSinkBufferSizeMs = 20; 133 // maximum normal sink buffer size 134 static const uint32_t kMaxNormalSinkBufferSizeMs = 24; 135 136 // minimum capture buffer size in milliseconds to _not_ need a fast capture thread 137 // FIXME This should be based on experimentally observed scheduling jitter 138 static const uint32_t kMinNormalCaptureBufferSizeMs = 12; 139 140 // Offloaded output thread standby delay: allows track transition without going to standby 141 static const nsecs_t kOffloadStandbyDelayNs = seconds(1); 142 143 // Direct output thread minimum sleep time in idle or active(underrun) state 144 static const nsecs_t kDirectMinSleepTimeUs = 10000; 145 146 // The universal constant for ubiquitous 20ms value. The value of 20ms seems to provide a good 147 // balance between power consumption and latency, and allows threads to be scheduled reliably 148 // by the CFS scheduler. 149 // FIXME Express other hardcoded references to 20ms with references to this constant and move 150 // it appropriately. 151 #define FMS_20 20 152 153 // Whether to use fast mixer 154 static const enum { 155 FastMixer_Never, // never initialize or use: for debugging only 156 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 157 // normal mixer multiplier is 1 158 FastMixer_Static, // initialize if needed, then use all the time if initialized, 159 // multiplier is calculated based on min & max normal mixer buffer size 160 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 161 // multiplier is calculated based on min & max normal mixer buffer size 162 // FIXME for FastMixer_Dynamic: 163 // Supporting this option will require fixing HALs that can't handle large writes. 164 // For example, one HAL implementation returns an error from a large write, 165 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 166 // We could either fix the HAL implementations, or provide a wrapper that breaks 167 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 168 } kUseFastMixer = FastMixer_Static; 169 170 // Whether to use fast capture 171 static const enum { 172 FastCapture_Never, // never initialize or use: for debugging only 173 FastCapture_Always, // always initialize and use, even if not needed: for debugging only 174 FastCapture_Static, // initialize if needed, then use all the time if initialized 175 } kUseFastCapture = FastCapture_Static; 176 177 // Priorities for requestPriority 178 static const int kPriorityAudioApp = 2; 179 static const int kPriorityFastMixer = 3; 180 static const int kPriorityFastCapture = 3; 181 182 // IAudioFlinger::createTrack() has an in/out parameter 'pFrameCount' for the total size of the 183 // track buffer in shared memory. Zero on input means to use a default value. For fast tracks, 184 // AudioFlinger derives the default from HAL buffer size and 'fast track multiplier'. 185 186 // This is the default value, if not specified by property. 187 static const int kFastTrackMultiplier = 2; 188 189 // The minimum and maximum allowed values 190 static const int kFastTrackMultiplierMin = 1; 191 static const int kFastTrackMultiplierMax = 2; 192 193 // The actual value to use, which can be specified per-device via property af.fast_track_multiplier. 194 static int sFastTrackMultiplier = kFastTrackMultiplier; 195 196 // See Thread::readOnlyHeap(). 197 // Initially this heap is used to allocate client buffers for "fast" AudioRecord. 198 // Eventually it will be the single buffer that FastCapture writes into via HAL read(), 199 // and that all "fast" AudioRecord clients read from. In either case, the size can be small. 200 static const size_t kRecordThreadReadOnlyHeapSize = 0x4000; 201 202 // ---------------------------------------------------------------------------- 203 204 static pthread_once_t sFastTrackMultiplierOnce = PTHREAD_ONCE_INIT; 205 206 static void sFastTrackMultiplierInit() 207 { 208 char value[PROPERTY_VALUE_MAX]; 209 if (property_get("af.fast_track_multiplier", value, NULL) > 0) { 210 char *endptr; 211 unsigned long ul = strtoul(value, &endptr, 0); 212 if (*endptr == '\0' && kFastTrackMultiplierMin <= ul && ul <= kFastTrackMultiplierMax) { 213 sFastTrackMultiplier = (int) ul; 214 } 215 } 216 } 217 218 // ---------------------------------------------------------------------------- 219 220 #ifdef ADD_BATTERY_DATA 221 // To collect the amplifier usage 222 static void addBatteryData(uint32_t params) { 223 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 224 if (service == NULL) { 225 // it already logged 226 return; 227 } 228 229 service->addBatteryData(params); 230 } 231 #endif 232 233 // Track the CLOCK_BOOTTIME versus CLOCK_MONOTONIC timebase offset 234 struct { 235 // call when you acquire a partial wakelock 236 void acquire(const sp<IBinder> &wakeLockToken) { 237 pthread_mutex_lock(&mLock); 238 if (wakeLockToken.get() == nullptr) { 239 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 240 } else { 241 if (mCount == 0) { 242 adjustTimebaseOffset(&mBoottimeOffset, ExtendedTimestamp::TIMEBASE_BOOTTIME); 243 } 244 ++mCount; 245 } 246 pthread_mutex_unlock(&mLock); 247 } 248 249 // call when you release a partial wakelock. 250 void release(const sp<IBinder> &wakeLockToken) { 251 if (wakeLockToken.get() == nullptr) { 252 return; 253 } 254 pthread_mutex_lock(&mLock); 255 if (--mCount < 0) { 256 ALOGE("negative wakelock count"); 257 mCount = 0; 258 } 259 pthread_mutex_unlock(&mLock); 260 } 261 262 // retrieves the boottime timebase offset from monotonic. 263 int64_t getBoottimeOffset() { 264 pthread_mutex_lock(&mLock); 265 int64_t boottimeOffset = mBoottimeOffset; 266 pthread_mutex_unlock(&mLock); 267 return boottimeOffset; 268 } 269 270 // Adjusts the timebase offset between TIMEBASE_MONOTONIC 271 // and the selected timebase. 272 // Currently only TIMEBASE_BOOTTIME is allowed. 273 // 274 // This only needs to be called upon acquiring the first partial wakelock 275 // after all other partial wakelocks are released. 276 // 277 // We do an empirical measurement of the offset rather than parsing 278 // /proc/timer_list since the latter is not a formal kernel ABI. 279 static void adjustTimebaseOffset(int64_t *offset, ExtendedTimestamp::Timebase timebase) { 280 int clockbase; 281 switch (timebase) { 282 case ExtendedTimestamp::TIMEBASE_BOOTTIME: 283 clockbase = SYSTEM_TIME_BOOTTIME; 284 break; 285 default: 286 LOG_ALWAYS_FATAL("invalid timebase %d", timebase); 287 break; 288 } 289 // try three times to get the clock offset, choose the one 290 // with the minimum gap in measurements. 291 const int tries = 3; 292 nsecs_t bestGap, measured; 293 for (int i = 0; i < tries; ++i) { 294 const nsecs_t tmono = systemTime(SYSTEM_TIME_MONOTONIC); 295 const nsecs_t tbase = systemTime(clockbase); 296 const nsecs_t tmono2 = systemTime(SYSTEM_TIME_MONOTONIC); 297 const nsecs_t gap = tmono2 - tmono; 298 if (i == 0 || gap < bestGap) { 299 bestGap = gap; 300 measured = tbase - ((tmono + tmono2) >> 1); 301 } 302 } 303 304 // to avoid micro-adjusting, we don't change the timebase 305 // unless it is significantly different. 306 // 307 // Assumption: It probably takes more than toleranceNs to 308 // suspend and resume the device. 309 static int64_t toleranceNs = 10000; // 10 us 310 if (llabs(*offset - measured) > toleranceNs) { 311 ALOGV("Adjusting timebase offset old: %lld new: %lld", 312 (long long)*offset, (long long)measured); 313 *offset = measured; 314 } 315 } 316 317 pthread_mutex_t mLock; 318 int32_t mCount; 319 int64_t mBoottimeOffset; 320 } gBoottime = { PTHREAD_MUTEX_INITIALIZER, 0, 0 }; // static, so use POD initialization 321 322 // ---------------------------------------------------------------------------- 323 // CPU Stats 324 // ---------------------------------------------------------------------------- 325 326 class CpuStats { 327 public: 328 CpuStats(); 329 void sample(const String8 &title); 330 #ifdef DEBUG_CPU_USAGE 331 private: 332 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 333 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 334 335 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 336 337 int mCpuNum; // thread's current CPU number 338 int mCpukHz; // frequency of thread's current CPU in kHz 339 #endif 340 }; 341 342 CpuStats::CpuStats() 343 #ifdef DEBUG_CPU_USAGE 344 : mCpuNum(-1), mCpukHz(-1) 345 #endif 346 { 347 } 348 349 void CpuStats::sample(const String8 &title 350 #ifndef DEBUG_CPU_USAGE 351 __unused 352 #endif 353 ) { 354 #ifdef DEBUG_CPU_USAGE 355 // get current thread's delta CPU time in wall clock ns 356 double wcNs; 357 bool valid = mCpuUsage.sampleAndEnable(wcNs); 358 359 // record sample for wall clock statistics 360 if (valid) { 361 mWcStats.sample(wcNs); 362 } 363 364 // get the current CPU number 365 int cpuNum = sched_getcpu(); 366 367 // get the current CPU frequency in kHz 368 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 369 370 // check if either CPU number or frequency changed 371 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 372 mCpuNum = cpuNum; 373 mCpukHz = cpukHz; 374 // ignore sample for purposes of cycles 375 valid = false; 376 } 377 378 // if no change in CPU number or frequency, then record sample for cycle statistics 379 if (valid && mCpukHz > 0) { 380 double cycles = wcNs * cpukHz * 0.000001; 381 mHzStats.sample(cycles); 382 } 383 384 unsigned n = mWcStats.n(); 385 // mCpuUsage.elapsed() is expensive, so don't call it every loop 386 if ((n & 127) == 1) { 387 long long elapsed = mCpuUsage.elapsed(); 388 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 389 double perLoop = elapsed / (double) n; 390 double perLoop100 = perLoop * 0.01; 391 double perLoop1k = perLoop * 0.001; 392 double mean = mWcStats.mean(); 393 double stddev = mWcStats.stddev(); 394 double minimum = mWcStats.minimum(); 395 double maximum = mWcStats.maximum(); 396 double meanCycles = mHzStats.mean(); 397 double stddevCycles = mHzStats.stddev(); 398 double minCycles = mHzStats.minimum(); 399 double maxCycles = mHzStats.maximum(); 400 mCpuUsage.resetElapsed(); 401 mWcStats.reset(); 402 mHzStats.reset(); 403 ALOGD("CPU usage for %s over past %.1f secs\n" 404 " (%u mixer loops at %.1f mean ms per loop):\n" 405 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 406 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 407 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 408 title.string(), 409 elapsed * .000000001, n, perLoop * .000001, 410 mean * .001, 411 stddev * .001, 412 minimum * .001, 413 maximum * .001, 414 mean / perLoop100, 415 stddev / perLoop100, 416 minimum / perLoop100, 417 maximum / perLoop100, 418 meanCycles / perLoop1k, 419 stddevCycles / perLoop1k, 420 minCycles / perLoop1k, 421 maxCycles / perLoop1k); 422 423 } 424 } 425 #endif 426 }; 427 428 // ---------------------------------------------------------------------------- 429 // ThreadBase 430 // ---------------------------------------------------------------------------- 431 432 // static 433 const char *AudioFlinger::ThreadBase::threadTypeToString(AudioFlinger::ThreadBase::type_t type) 434 { 435 switch (type) { 436 case MIXER: 437 return "MIXER"; 438 case DIRECT: 439 return "DIRECT"; 440 case DUPLICATING: 441 return "DUPLICATING"; 442 case RECORD: 443 return "RECORD"; 444 case OFFLOAD: 445 return "OFFLOAD"; 446 case MMAP: 447 return "MMAP"; 448 default: 449 return "unknown"; 450 } 451 } 452 453 std::string devicesToString(audio_devices_t devices) 454 { 455 std::string result; 456 if (devices & AUDIO_DEVICE_BIT_IN) { 457 InputDeviceConverter::maskToString(devices, result); 458 } else { 459 OutputDeviceConverter::maskToString(devices, result); 460 } 461 return result; 462 } 463 464 std::string inputFlagsToString(audio_input_flags_t flags) 465 { 466 std::string result; 467 InputFlagConverter::maskToString(flags, result); 468 return result; 469 } 470 471 std::string outputFlagsToString(audio_output_flags_t flags) 472 { 473 std::string result; 474 OutputFlagConverter::maskToString(flags, result); 475 return result; 476 } 477 478 const char *sourceToString(audio_source_t source) 479 { 480 switch (source) { 481 case AUDIO_SOURCE_DEFAULT: return "default"; 482 case AUDIO_SOURCE_MIC: return "mic"; 483 case AUDIO_SOURCE_VOICE_UPLINK: return "voice uplink"; 484 case AUDIO_SOURCE_VOICE_DOWNLINK: return "voice downlink"; 485 case AUDIO_SOURCE_VOICE_CALL: return "voice call"; 486 case AUDIO_SOURCE_CAMCORDER: return "camcorder"; 487 case AUDIO_SOURCE_VOICE_RECOGNITION: return "voice recognition"; 488 case AUDIO_SOURCE_VOICE_COMMUNICATION: return "voice communication"; 489 case AUDIO_SOURCE_REMOTE_SUBMIX: return "remote submix"; 490 case AUDIO_SOURCE_UNPROCESSED: return "unprocessed"; 491 case AUDIO_SOURCE_FM_TUNER: return "FM tuner"; 492 case AUDIO_SOURCE_HOTWORD: return "hotword"; 493 default: return "unknown"; 494 } 495 } 496 497 AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 498 audio_devices_t outDevice, audio_devices_t inDevice, type_t type, bool systemReady) 499 : Thread(false /*canCallJava*/), 500 mType(type), 501 mAudioFlinger(audioFlinger), 502 // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize 503 // are set by PlaybackThread::readOutputParameters_l() or 504 // RecordThread::readInputParameters_l() 505 //FIXME: mStandby should be true here. Is this some kind of hack? 506 mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), 507 mPrevOutDevice(AUDIO_DEVICE_NONE), mPrevInDevice(AUDIO_DEVICE_NONE), 508 mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), 509 // mName will be set by concrete (non-virtual) subclass 510 mDeathRecipient(new PMDeathRecipient(this)), 511 mSystemReady(systemReady), 512 mSignalPending(false) 513 { 514 memset(&mPatch, 0, sizeof(struct audio_patch)); 515 } 516 517 AudioFlinger::ThreadBase::~ThreadBase() 518 { 519 // mConfigEvents should be empty, but just in case it isn't, free the memory it owns 520 mConfigEvents.clear(); 521 522 // do not lock the mutex in destructor 523 releaseWakeLock_l(); 524 if (mPowerManager != 0) { 525 sp<IBinder> binder = IInterface::asBinder(mPowerManager); 526 binder->unlinkToDeath(mDeathRecipient); 527 } 528 } 529 530 status_t AudioFlinger::ThreadBase::readyToRun() 531 { 532 status_t status = initCheck(); 533 if (status == NO_ERROR) { 534 ALOGI("AudioFlinger's thread %p tid=%d ready to run", this, getTid()); 535 } else { 536 ALOGE("No working audio driver found."); 537 } 538 return status; 539 } 540 541 void AudioFlinger::ThreadBase::exit() 542 { 543 ALOGV("ThreadBase::exit"); 544 // do any cleanup required for exit to succeed 545 preExit(); 546 { 547 // This lock prevents the following race in thread (uniprocessor for illustration): 548 // if (!exitPending()) { 549 // // context switch from here to exit() 550 // // exit() calls requestExit(), what exitPending() observes 551 // // exit() calls signal(), which is dropped since no waiters 552 // // context switch back from exit() to here 553 // mWaitWorkCV.wait(...); 554 // // now thread is hung 555 // } 556 AutoMutex lock(mLock); 557 requestExit(); 558 mWaitWorkCV.broadcast(); 559 } 560 // When Thread::requestExitAndWait is made virtual and this method is renamed to 561 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 562 requestExitAndWait(); 563 } 564 565 status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 566 { 567 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 568 Mutex::Autolock _l(mLock); 569 570 return sendSetParameterConfigEvent_l(keyValuePairs); 571 } 572 573 // sendConfigEvent_l() must be called with ThreadBase::mLock held 574 // Can temporarily release the lock if waiting for a reply from processConfigEvents_l(). 575 status_t AudioFlinger::ThreadBase::sendConfigEvent_l(sp<ConfigEvent>& event) 576 { 577 status_t status = NO_ERROR; 578 579 if (event->mRequiresSystemReady && !mSystemReady) { 580 event->mWaitStatus = false; 581 mPendingConfigEvents.add(event); 582 return status; 583 } 584 mConfigEvents.add(event); 585 ALOGV("sendConfigEvent_l() num events %zu event %d", mConfigEvents.size(), event->mType); 586 mWaitWorkCV.signal(); 587 mLock.unlock(); 588 { 589 Mutex::Autolock _l(event->mLock); 590 while (event->mWaitStatus) { 591 if (event->mCond.waitRelative(event->mLock, kConfigEventTimeoutNs) != NO_ERROR) { 592 event->mStatus = TIMED_OUT; 593 event->mWaitStatus = false; 594 } 595 } 596 status = event->mStatus; 597 } 598 mLock.lock(); 599 return status; 600 } 601 602 void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid) 603 { 604 Mutex::Autolock _l(mLock); 605 sendIoConfigEvent_l(event, pid); 606 } 607 608 // sendIoConfigEvent_l() must be called with ThreadBase::mLock held 609 void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid) 610 { 611 sp<ConfigEvent> configEvent = (ConfigEvent *)new IoConfigEvent(event, pid); 612 sendConfigEvent_l(configEvent); 613 } 614 615 void AudioFlinger::ThreadBase::sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp) 616 { 617 Mutex::Autolock _l(mLock); 618 sendPrioConfigEvent_l(pid, tid, prio, forApp); 619 } 620 621 // sendPrioConfigEvent_l() must be called with ThreadBase::mLock held 622 void AudioFlinger::ThreadBase::sendPrioConfigEvent_l( 623 pid_t pid, pid_t tid, int32_t prio, bool forApp) 624 { 625 sp<ConfigEvent> configEvent = (ConfigEvent *)new PrioConfigEvent(pid, tid, prio, forApp); 626 sendConfigEvent_l(configEvent); 627 } 628 629 // sendSetParameterConfigEvent_l() must be called with ThreadBase::mLock held 630 status_t AudioFlinger::ThreadBase::sendSetParameterConfigEvent_l(const String8& keyValuePair) 631 { 632 sp<ConfigEvent> configEvent; 633 AudioParameter param(keyValuePair); 634 int value; 635 if (param.getInt(String8(AudioParameter::keyMonoOutput), value) == NO_ERROR) { 636 setMasterMono_l(value != 0); 637 if (param.size() == 1) { 638 return NO_ERROR; // should be a solo parameter - we don't pass down 639 } 640 param.remove(String8(AudioParameter::keyMonoOutput)); 641 configEvent = new SetParameterConfigEvent(param.toString()); 642 } else { 643 configEvent = new SetParameterConfigEvent(keyValuePair); 644 } 645 return sendConfigEvent_l(configEvent); 646 } 647 648 status_t AudioFlinger::ThreadBase::sendCreateAudioPatchConfigEvent( 649 const struct audio_patch *patch, 650 audio_patch_handle_t *handle) 651 { 652 Mutex::Autolock _l(mLock); 653 sp<ConfigEvent> configEvent = (ConfigEvent *)new CreateAudioPatchConfigEvent(*patch, *handle); 654 status_t status = sendConfigEvent_l(configEvent); 655 if (status == NO_ERROR) { 656 CreateAudioPatchConfigEventData *data = 657 (CreateAudioPatchConfigEventData *)configEvent->mData.get(); 658 *handle = data->mHandle; 659 } 660 return status; 661 } 662 663 status_t AudioFlinger::ThreadBase::sendReleaseAudioPatchConfigEvent( 664 const audio_patch_handle_t handle) 665 { 666 Mutex::Autolock _l(mLock); 667 sp<ConfigEvent> configEvent = (ConfigEvent *)new ReleaseAudioPatchConfigEvent(handle); 668 return sendConfigEvent_l(configEvent); 669 } 670 671 672 // post condition: mConfigEvents.isEmpty() 673 void AudioFlinger::ThreadBase::processConfigEvents_l() 674 { 675 bool configChanged = false; 676 677 while (!mConfigEvents.isEmpty()) { 678 ALOGV("processConfigEvents_l() remaining events %zu", mConfigEvents.size()); 679 sp<ConfigEvent> event = mConfigEvents[0]; 680 mConfigEvents.removeAt(0); 681 switch (event->mType) { 682 case CFG_EVENT_PRIO: { 683 PrioConfigEventData *data = (PrioConfigEventData *)event->mData.get(); 684 // FIXME Need to understand why this has to be done asynchronously 685 int err = requestPriority(data->mPid, data->mTid, data->mPrio, data->mForApp, 686 true /*asynchronous*/); 687 if (err != 0) { 688 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 689 data->mPrio, data->mPid, data->mTid, err); 690 } 691 } break; 692 case CFG_EVENT_IO: { 693 IoConfigEventData *data = (IoConfigEventData *)event->mData.get(); 694 ioConfigChanged(data->mEvent, data->mPid); 695 } break; 696 case CFG_EVENT_SET_PARAMETER: { 697 SetParameterConfigEventData *data = (SetParameterConfigEventData *)event->mData.get(); 698 if (checkForNewParameter_l(data->mKeyValuePairs, event->mStatus)) { 699 configChanged = true; 700 mLocalLog.log("CFG_EVENT_SET_PARAMETER: (%s) configuration changed", 701 data->mKeyValuePairs.string()); 702 } 703 } break; 704 case CFG_EVENT_CREATE_AUDIO_PATCH: { 705 const audio_devices_t oldDevice = getDevice(); 706 CreateAudioPatchConfigEventData *data = 707 (CreateAudioPatchConfigEventData *)event->mData.get(); 708 event->mStatus = createAudioPatch_l(&data->mPatch, &data->mHandle); 709 const audio_devices_t newDevice = getDevice(); 710 mLocalLog.log("CFG_EVENT_CREATE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)", 711 (unsigned)oldDevice, devicesToString(oldDevice).c_str(), 712 (unsigned)newDevice, devicesToString(newDevice).c_str()); 713 } break; 714 case CFG_EVENT_RELEASE_AUDIO_PATCH: { 715 const audio_devices_t oldDevice = getDevice(); 716 ReleaseAudioPatchConfigEventData *data = 717 (ReleaseAudioPatchConfigEventData *)event->mData.get(); 718 event->mStatus = releaseAudioPatch_l(data->mHandle); 719 const audio_devices_t newDevice = getDevice(); 720 mLocalLog.log("CFG_EVENT_RELEASE_AUDIO_PATCH: old device %#x (%s) new device %#x (%s)", 721 (unsigned)oldDevice, devicesToString(oldDevice).c_str(), 722 (unsigned)newDevice, devicesToString(newDevice).c_str()); 723 } break; 724 default: 725 ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType); 726 break; 727 } 728 { 729 Mutex::Autolock _l(event->mLock); 730 if (event->mWaitStatus) { 731 event->mWaitStatus = false; 732 event->mCond.signal(); 733 } 734 } 735 ALOGV_IF(mConfigEvents.isEmpty(), "processConfigEvents_l() DONE thread %p", this); 736 } 737 738 if (configChanged) { 739 cacheParameters_l(); 740 } 741 } 742 743 String8 channelMaskToString(audio_channel_mask_t mask, bool output) { 744 String8 s; 745 const audio_channel_representation_t representation = 746 audio_channel_mask_get_representation(mask); 747 748 switch (representation) { 749 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 750 if (output) { 751 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT) s.append("front-left, "); 752 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT) s.append("front-right, "); 753 if (mask & AUDIO_CHANNEL_OUT_FRONT_CENTER) s.append("front-center, "); 754 if (mask & AUDIO_CHANNEL_OUT_LOW_FREQUENCY) s.append("low freq, "); 755 if (mask & AUDIO_CHANNEL_OUT_BACK_LEFT) s.append("back-left, "); 756 if (mask & AUDIO_CHANNEL_OUT_BACK_RIGHT) s.append("back-right, "); 757 if (mask & AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER) s.append("front-left-of-center, "); 758 if (mask & AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER) s.append("front-right-of-center, "); 759 if (mask & AUDIO_CHANNEL_OUT_BACK_CENTER) s.append("back-center, "); 760 if (mask & AUDIO_CHANNEL_OUT_SIDE_LEFT) s.append("side-left, "); 761 if (mask & AUDIO_CHANNEL_OUT_SIDE_RIGHT) s.append("side-right, "); 762 if (mask & AUDIO_CHANNEL_OUT_TOP_CENTER) s.append("top-center ,"); 763 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT) s.append("top-front-left, "); 764 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER) s.append("top-front-center, "); 765 if (mask & AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT) s.append("top-front-right, "); 766 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, "); 767 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " ); 768 if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " ); 769 if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown, "); 770 } else { 771 if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, "); 772 if (mask & AUDIO_CHANNEL_IN_RIGHT) s.append("right, "); 773 if (mask & AUDIO_CHANNEL_IN_FRONT) s.append("front, "); 774 if (mask & AUDIO_CHANNEL_IN_BACK) s.append("back, "); 775 if (mask & AUDIO_CHANNEL_IN_LEFT_PROCESSED) s.append("left-processed, "); 776 if (mask & AUDIO_CHANNEL_IN_RIGHT_PROCESSED) s.append("right-processed, "); 777 if (mask & AUDIO_CHANNEL_IN_FRONT_PROCESSED) s.append("front-processed, "); 778 if (mask & AUDIO_CHANNEL_IN_BACK_PROCESSED) s.append("back-processed, "); 779 if (mask & AUDIO_CHANNEL_IN_PRESSURE) s.append("pressure, "); 780 if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, "); 781 if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, "); 782 if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, "); 783 if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, "); 784 if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, "); 785 if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown, "); 786 } 787 const int len = s.length(); 788 if (len > 2) { 789 (void) s.lockBuffer(len); // needed? 790 s.unlockBuffer(len - 2); // remove trailing ", " 791 } 792 return s; 793 } 794 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 795 s.appendFormat("index mask, bits:%#x", audio_channel_mask_get_bits(mask)); 796 return s; 797 default: 798 s.appendFormat("unknown mask, representation:%d bits:%#x", 799 representation, audio_channel_mask_get_bits(mask)); 800 return s; 801 } 802 } 803 804 void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args __unused) 805 { 806 const size_t SIZE = 256; 807 char buffer[SIZE]; 808 String8 result; 809 810 dprintf(fd, "\n%s thread %p, name %s, tid %d, type %d (%s):\n", isOutput() ? "Output" : "Input", 811 this, mThreadName, getTid(), type(), threadTypeToString(type())); 812 813 bool locked = AudioFlinger::dumpTryLock(mLock); 814 if (!locked) { 815 dprintf(fd, " Thread may be deadlocked\n"); 816 } 817 818 dprintf(fd, " I/O handle: %d\n", mId); 819 dprintf(fd, " Standby: %s\n", mStandby ? "yes" : "no"); 820 dprintf(fd, " Sample rate: %u Hz\n", mSampleRate); 821 dprintf(fd, " HAL frame count: %zu\n", mFrameCount); 822 dprintf(fd, " HAL format: 0x%x (%s)\n", mHALFormat, formatToString(mHALFormat).c_str()); 823 dprintf(fd, " HAL buffer size: %zu bytes\n", mBufferSize); 824 dprintf(fd, " Channel count: %u\n", mChannelCount); 825 dprintf(fd, " Channel mask: 0x%08x (%s)\n", mChannelMask, 826 channelMaskToString(mChannelMask, mType != RECORD).string()); 827 dprintf(fd, " Processing format: 0x%x (%s)\n", mFormat, formatToString(mFormat).c_str()); 828 dprintf(fd, " Processing frame size: %zu bytes\n", mFrameSize); 829 dprintf(fd, " Pending config events:"); 830 size_t numConfig = mConfigEvents.size(); 831 if (numConfig) { 832 for (size_t i = 0; i < numConfig; i++) { 833 mConfigEvents[i]->dump(buffer, SIZE); 834 dprintf(fd, "\n %s", buffer); 835 } 836 dprintf(fd, "\n"); 837 } else { 838 dprintf(fd, " none\n"); 839 } 840 // Note: output device may be used by capture threads for effects such as AEC. 841 dprintf(fd, " Output device: %#x (%s)\n", mOutDevice, devicesToString(mOutDevice).c_str()); 842 dprintf(fd, " Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).c_str()); 843 dprintf(fd, " Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource)); 844 845 if (locked) { 846 mLock.unlock(); 847 } 848 } 849 850 void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 851 { 852 const size_t SIZE = 256; 853 char buffer[SIZE]; 854 String8 result; 855 856 size_t numEffectChains = mEffectChains.size(); 857 snprintf(buffer, SIZE, " %zu Effect Chains\n", numEffectChains); 858 write(fd, buffer, strlen(buffer)); 859 860 for (size_t i = 0; i < numEffectChains; ++i) { 861 sp<EffectChain> chain = mEffectChains[i]; 862 if (chain != 0) { 863 chain->dump(fd, args); 864 } 865 } 866 } 867 868 void AudioFlinger::ThreadBase::acquireWakeLock() 869 { 870 Mutex::Autolock _l(mLock); 871 acquireWakeLock_l(); 872 } 873 874 String16 AudioFlinger::ThreadBase::getWakeLockTag() 875 { 876 switch (mType) { 877 case MIXER: 878 return String16("AudioMix"); 879 case DIRECT: 880 return String16("AudioDirectOut"); 881 case DUPLICATING: 882 return String16("AudioDup"); 883 case RECORD: 884 return String16("AudioIn"); 885 case OFFLOAD: 886 return String16("AudioOffload"); 887 case MMAP: 888 return String16("Mmap"); 889 default: 890 ALOG_ASSERT(false); 891 return String16("AudioUnknown"); 892 } 893 } 894 895 void AudioFlinger::ThreadBase::acquireWakeLock_l() 896 { 897 getPowerManager_l(); 898 if (mPowerManager != 0) { 899 sp<IBinder> binder = new BBinder(); 900 // Uses AID_AUDIOSERVER for wakelock. updateWakeLockUids_l() updates with client uids. 901 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 902 binder, 903 getWakeLockTag(), 904 String16("audioserver"), 905 true /* FIXME force oneway contrary to .aidl */); 906 if (status == NO_ERROR) { 907 mWakeLockToken = binder; 908 } 909 ALOGV("acquireWakeLock_l() %s status %d", mThreadName, status); 910 } 911 912 gBoottime.acquire(mWakeLockToken); 913 mTimestamp.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_BOOTTIME] = 914 gBoottime.getBoottimeOffset(); 915 } 916 917 void AudioFlinger::ThreadBase::releaseWakeLock() 918 { 919 Mutex::Autolock _l(mLock); 920 releaseWakeLock_l(); 921 } 922 923 void AudioFlinger::ThreadBase::releaseWakeLock_l() 924 { 925 gBoottime.release(mWakeLockToken); 926 if (mWakeLockToken != 0) { 927 ALOGV("releaseWakeLock_l() %s", mThreadName); 928 if (mPowerManager != 0) { 929 mPowerManager->releaseWakeLock(mWakeLockToken, 0, 930 true /* FIXME force oneway contrary to .aidl */); 931 } 932 mWakeLockToken.clear(); 933 } 934 } 935 936 void AudioFlinger::ThreadBase::getPowerManager_l() { 937 if (mSystemReady && mPowerManager == 0) { 938 // use checkService() to avoid blocking if power service is not up yet 939 sp<IBinder> binder = 940 defaultServiceManager()->checkService(String16("power")); 941 if (binder == 0) { 942 ALOGW("Thread %s cannot connect to the power manager service", mThreadName); 943 } else { 944 mPowerManager = interface_cast<IPowerManager>(binder); 945 binder->linkToDeath(mDeathRecipient); 946 } 947 } 948 } 949 950 void AudioFlinger::ThreadBase::updateWakeLockUids_l(const SortedVector<uid_t> &uids) { 951 getPowerManager_l(); 952 953 #if !LOG_NDEBUG 954 std::stringstream s; 955 for (uid_t uid : uids) { 956 s << uid << " "; 957 } 958 ALOGD("updateWakeLockUids_l %s uids:%s", mThreadName, s.str().c_str()); 959 #endif 960 961 if (mWakeLockToken == NULL) { // token may be NULL if AudioFlinger::systemReady() not called. 962 if (mSystemReady) { 963 ALOGE("no wake lock to update, but system ready!"); 964 } else { 965 ALOGW("no wake lock to update, system not ready yet"); 966 } 967 return; 968 } 969 if (mPowerManager != 0) { 970 std::vector<int> uidsAsInt(uids.begin(), uids.end()); // powermanager expects uids as ints 971 status_t status = mPowerManager->updateWakeLockUids( 972 mWakeLockToken, uidsAsInt.size(), uidsAsInt.data(), 973 true /* FIXME force oneway contrary to .aidl */); 974 ALOGV("updateWakeLockUids_l() %s status %d", mThreadName, status); 975 } 976 } 977 978 void AudioFlinger::ThreadBase::clearPowerManager() 979 { 980 Mutex::Autolock _l(mLock); 981 releaseWakeLock_l(); 982 mPowerManager.clear(); 983 } 984 985 void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who __unused) 986 { 987 sp<ThreadBase> thread = mThread.promote(); 988 if (thread != 0) { 989 thread->clearPowerManager(); 990 } 991 ALOGW("power manager service died !!!"); 992 } 993 994 void AudioFlinger::ThreadBase::setEffectSuspended_l( 995 const effect_uuid_t *type, bool suspend, audio_session_t sessionId) 996 { 997 sp<EffectChain> chain = getEffectChain_l(sessionId); 998 if (chain != 0) { 999 if (type != NULL) { 1000 chain->setEffectSuspended_l(type, suspend); 1001 } else { 1002 chain->setEffectSuspendedAll_l(suspend); 1003 } 1004 } 1005 1006 updateSuspendedSessions_l(type, suspend, sessionId); 1007 } 1008 1009 void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1010 { 1011 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1012 if (index < 0) { 1013 return; 1014 } 1015 1016 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1017 mSuspendedSessions.valueAt(index); 1018 1019 for (size_t i = 0; i < sessionEffects.size(); i++) { 1020 const sp<SuspendedSessionDesc>& desc = sessionEffects.valueAt(i); 1021 for (int j = 0; j < desc->mRefCount; j++) { 1022 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1023 chain->setEffectSuspendedAll_l(true); 1024 } else { 1025 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1026 desc->mType.timeLow); 1027 chain->setEffectSuspended_l(&desc->mType, true); 1028 } 1029 } 1030 } 1031 } 1032 1033 void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1034 bool suspend, 1035 audio_session_t sessionId) 1036 { 1037 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1038 1039 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1040 1041 if (suspend) { 1042 if (index >= 0) { 1043 sessionEffects = mSuspendedSessions.valueAt(index); 1044 } else { 1045 mSuspendedSessions.add(sessionId, sessionEffects); 1046 } 1047 } else { 1048 if (index < 0) { 1049 return; 1050 } 1051 sessionEffects = mSuspendedSessions.valueAt(index); 1052 } 1053 1054 1055 int key = EffectChain::kKeyForSuspendAll; 1056 if (type != NULL) { 1057 key = type->timeLow; 1058 } 1059 index = sessionEffects.indexOfKey(key); 1060 1061 sp<SuspendedSessionDesc> desc; 1062 if (suspend) { 1063 if (index >= 0) { 1064 desc = sessionEffects.valueAt(index); 1065 } else { 1066 desc = new SuspendedSessionDesc(); 1067 if (type != NULL) { 1068 desc->mType = *type; 1069 } 1070 sessionEffects.add(key, desc); 1071 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1072 } 1073 desc->mRefCount++; 1074 } else { 1075 if (index < 0) { 1076 return; 1077 } 1078 desc = sessionEffects.valueAt(index); 1079 if (--desc->mRefCount == 0) { 1080 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1081 sessionEffects.removeItemsAt(index); 1082 if (sessionEffects.isEmpty()) { 1083 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1084 sessionId); 1085 mSuspendedSessions.removeItem(sessionId); 1086 } 1087 } 1088 } 1089 if (!sessionEffects.isEmpty()) { 1090 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1091 } 1092 } 1093 1094 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1095 bool enabled, 1096 audio_session_t sessionId) 1097 { 1098 Mutex::Autolock _l(mLock); 1099 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1100 } 1101 1102 void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1103 bool enabled, 1104 audio_session_t sessionId) 1105 { 1106 if (mType != RECORD) { 1107 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1108 // another session. This gives the priority to well behaved effect control panels 1109 // and applications not using global effects. 1110 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1111 // global effects 1112 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1113 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1114 } 1115 } 1116 1117 sp<EffectChain> chain = getEffectChain_l(sessionId); 1118 if (chain != 0) { 1119 chain->checkSuspendOnEffectEnabled(effect, enabled); 1120 } 1121 } 1122 1123 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1124 status_t AudioFlinger::RecordThread::checkEffectCompatibility_l( 1125 const effect_descriptor_t *desc, audio_session_t sessionId) 1126 { 1127 // No global effect sessions on record threads 1128 if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1129 ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s", 1130 desc->name, mThreadName); 1131 return BAD_VALUE; 1132 } 1133 // only pre processing effects on record thread 1134 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) { 1135 ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on record thread %s", 1136 desc->name, mThreadName); 1137 return BAD_VALUE; 1138 } 1139 1140 // always allow effects without processing load or latency 1141 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1142 return NO_ERROR; 1143 } 1144 1145 audio_input_flags_t flags = mInput->flags; 1146 if (hasFastCapture() || (flags & AUDIO_INPUT_FLAG_FAST)) { 1147 if (flags & AUDIO_INPUT_FLAG_RAW) { 1148 ALOGW("checkEffectCompatibility_l(): effect %s on record thread %s in raw mode", 1149 desc->name, mThreadName); 1150 return BAD_VALUE; 1151 } 1152 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1153 ALOGW("checkEffectCompatibility_l(): non HW effect %s on record thread %s in fast mode", 1154 desc->name, mThreadName); 1155 return BAD_VALUE; 1156 } 1157 } 1158 return NO_ERROR; 1159 } 1160 1161 // checkEffectCompatibility_l() must be called with ThreadBase::mLock held 1162 status_t AudioFlinger::PlaybackThread::checkEffectCompatibility_l( 1163 const effect_descriptor_t *desc, audio_session_t sessionId) 1164 { 1165 // no preprocessing on playback threads 1166 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC) { 1167 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback" 1168 " thread %s", desc->name, mThreadName); 1169 return BAD_VALUE; 1170 } 1171 1172 // always allow effects without processing load or latency 1173 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) == EFFECT_FLAG_NO_PROCESS) { 1174 return NO_ERROR; 1175 } 1176 1177 switch (mType) { 1178 case MIXER: { 1179 // Reject any effect on mixer multichannel sinks. 1180 // TODO: fix both format and multichannel issues with effects. 1181 if (mChannelCount != FCC_2) { 1182 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d) on MIXER" 1183 " thread %s", desc->name, mChannelCount, mThreadName); 1184 return BAD_VALUE; 1185 } 1186 audio_output_flags_t flags = mOutput->flags; 1187 if (hasFastMixer() || (flags & AUDIO_OUTPUT_FLAG_FAST)) { 1188 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1189 // global effects are applied only to non fast tracks if they are SW 1190 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1191 break; 1192 } 1193 } else if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 1194 // only post processing on output stage session 1195 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_POST_PROC) { 1196 ALOGW("checkEffectCompatibility_l(): non post processing effect %s not allowed" 1197 " on output stage session", desc->name); 1198 return BAD_VALUE; 1199 } 1200 } else { 1201 // no restriction on effects applied on non fast tracks 1202 if ((hasAudioSession_l(sessionId) & ThreadBase::FAST_SESSION) == 0) { 1203 break; 1204 } 1205 } 1206 1207 if (flags & AUDIO_OUTPUT_FLAG_RAW) { 1208 ALOGW("checkEffectCompatibility_l(): effect %s on playback thread in raw mode", 1209 desc->name); 1210 return BAD_VALUE; 1211 } 1212 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) == 0) { 1213 ALOGW("checkEffectCompatibility_l(): non HW effect %s on playback thread" 1214 " in fast mode", desc->name); 1215 return BAD_VALUE; 1216 } 1217 } 1218 } break; 1219 case OFFLOAD: 1220 // nothing actionable on offload threads, if the effect: 1221 // - is offloadable: the effect can be created 1222 // - is NOT offloadable: the effect should still be created, but EffectHandle::enable() 1223 // will take care of invalidating the tracks of the thread 1224 break; 1225 case DIRECT: 1226 // Reject any effect on Direct output threads for now, since the format of 1227 // mSinkBuffer is not guaranteed to be compatible with effect processing (PCM 16 stereo). 1228 ALOGW("checkEffectCompatibility_l(): effect %s on DIRECT output thread %s", 1229 desc->name, mThreadName); 1230 return BAD_VALUE; 1231 case DUPLICATING: 1232 // Reject any effect on mixer multichannel sinks. 1233 // TODO: fix both format and multichannel issues with effects. 1234 if (mChannelCount != FCC_2) { 1235 ALOGW("checkEffectCompatibility_l(): effect %s for multichannel(%d)" 1236 " on DUPLICATING thread %s", desc->name, mChannelCount, mThreadName); 1237 return BAD_VALUE; 1238 } 1239 if ((sessionId == AUDIO_SESSION_OUTPUT_STAGE) || (sessionId == AUDIO_SESSION_OUTPUT_MIX)) { 1240 ALOGW("checkEffectCompatibility_l(): global effect %s on DUPLICATING" 1241 " thread %s", desc->name, mThreadName); 1242 return BAD_VALUE; 1243 } 1244 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 1245 ALOGW("checkEffectCompatibility_l(): post processing effect %s on" 1246 " DUPLICATING thread %s", desc->name, mThreadName); 1247 return BAD_VALUE; 1248 } 1249 if ((desc->flags & EFFECT_FLAG_HW_ACC_TUNNEL) != 0) { 1250 ALOGW("checkEffectCompatibility_l(): HW tunneled effect %s on" 1251 " DUPLICATING thread %s", desc->name, mThreadName); 1252 return BAD_VALUE; 1253 } 1254 break; 1255 default: 1256 LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType); 1257 } 1258 1259 return NO_ERROR; 1260 } 1261 1262 // ThreadBase::createEffect_l() must be called with AudioFlinger::mLock held 1263 sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 1264 const sp<AudioFlinger::Client>& client, 1265 const sp<IEffectClient>& effectClient, 1266 int32_t priority, 1267 audio_session_t sessionId, 1268 effect_descriptor_t *desc, 1269 int *enabled, 1270 status_t *status, 1271 bool pinned) 1272 { 1273 sp<EffectModule> effect; 1274 sp<EffectHandle> handle; 1275 status_t lStatus; 1276 sp<EffectChain> chain; 1277 bool chainCreated = false; 1278 bool effectCreated = false; 1279 bool effectRegistered = false; 1280 audio_unique_id_t effectId = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; 1281 1282 lStatus = initCheck(); 1283 if (lStatus != NO_ERROR) { 1284 ALOGW("createEffect_l() Audio driver not initialized."); 1285 goto Exit; 1286 } 1287 1288 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 1289 1290 { // scope for mLock 1291 Mutex::Autolock _l(mLock); 1292 1293 lStatus = checkEffectCompatibility_l(desc, sessionId); 1294 if (lStatus != NO_ERROR) { 1295 goto Exit; 1296 } 1297 1298 // check for existing effect chain with the requested audio session 1299 chain = getEffectChain_l(sessionId); 1300 if (chain == 0) { 1301 // create a new chain for this session 1302 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 1303 chain = new EffectChain(this, sessionId); 1304 addEffectChain_l(chain); 1305 chain->setStrategy(getStrategyForSession_l(sessionId)); 1306 chainCreated = true; 1307 } else { 1308 effect = chain->getEffectFromDesc_l(desc); 1309 } 1310 1311 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 1312 1313 if (effect == 0) { 1314 effectId = mAudioFlinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); 1315 // Check CPU and memory usage 1316 lStatus = AudioSystem::registerEffect( 1317 desc, mId, chain->strategy(), sessionId, effectId); 1318 if (lStatus != NO_ERROR) { 1319 goto Exit; 1320 } 1321 effectRegistered = true; 1322 // create a new effect module if none present in the chain 1323 lStatus = chain->createEffect_l(effect, this, desc, effectId, sessionId, pinned); 1324 if (lStatus != NO_ERROR) { 1325 goto Exit; 1326 } 1327 effectCreated = true; 1328 1329 effect->setDevice(mOutDevice); 1330 effect->setDevice(mInDevice); 1331 effect->setMode(mAudioFlinger->getMode()); 1332 effect->setAudioSource(mAudioSource); 1333 } 1334 // create effect handle and connect it to effect module 1335 handle = new EffectHandle(effect, client, effectClient, priority); 1336 lStatus = handle->initCheck(); 1337 if (lStatus == OK) { 1338 lStatus = effect->addHandle(handle.get()); 1339 } 1340 if (enabled != NULL) { 1341 *enabled = (int)effect->isEnabled(); 1342 } 1343 } 1344 1345 Exit: 1346 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 1347 Mutex::Autolock _l(mLock); 1348 if (effectCreated) { 1349 chain->removeEffect_l(effect); 1350 } 1351 if (effectRegistered) { 1352 AudioSystem::unregisterEffect(effectId); 1353 } 1354 if (chainCreated) { 1355 removeEffectChain_l(chain); 1356 } 1357 handle.clear(); 1358 } 1359 1360 *status = lStatus; 1361 return handle; 1362 } 1363 1364 void AudioFlinger::ThreadBase::disconnectEffectHandle(EffectHandle *handle, 1365 bool unpinIfLast) 1366 { 1367 bool remove = false; 1368 sp<EffectModule> effect; 1369 { 1370 Mutex::Autolock _l(mLock); 1371 1372 effect = handle->effect().promote(); 1373 if (effect == 0) { 1374 return; 1375 } 1376 // restore suspended effects if the disconnected handle was enabled and the last one. 1377 remove = (effect->removeHandle(handle) == 0) && (!effect->isPinned() || unpinIfLast); 1378 if (remove) { 1379 removeEffect_l(effect, true); 1380 } 1381 } 1382 if (remove) { 1383 mAudioFlinger->updateOrphanEffectChains(effect); 1384 AudioSystem::unregisterEffect(effect->id()); 1385 if (handle->enabled()) { 1386 checkSuspendOnEffectEnabled(effect, false, effect->sessionId()); 1387 } 1388 } 1389 } 1390 1391 sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(audio_session_t sessionId, 1392 int effectId) 1393 { 1394 Mutex::Autolock _l(mLock); 1395 return getEffect_l(sessionId, effectId); 1396 } 1397 1398 sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(audio_session_t sessionId, 1399 int effectId) 1400 { 1401 sp<EffectChain> chain = getEffectChain_l(sessionId); 1402 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 1403 } 1404 1405 // PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 1406 // PlaybackThread::mLock held 1407 status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 1408 { 1409 // check for existing effect chain with the requested audio session 1410 audio_session_t sessionId = effect->sessionId(); 1411 sp<EffectChain> chain = getEffectChain_l(sessionId); 1412 bool chainCreated = false; 1413 1414 ALOGD_IF((mType == OFFLOAD) && !effect->isOffloadable(), 1415 "addEffect_l() on offloaded thread %p: effect %s does not support offload flags %x", 1416 this, effect->desc().name, effect->desc().flags); 1417 1418 if (chain == 0) { 1419 // create a new chain for this session 1420 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 1421 chain = new EffectChain(this, sessionId); 1422 addEffectChain_l(chain); 1423 chain->setStrategy(getStrategyForSession_l(sessionId)); 1424 chainCreated = true; 1425 } 1426 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 1427 1428 if (chain->getEffectFromId_l(effect->id()) != 0) { 1429 ALOGW("addEffect_l() %p effect %s already present in chain %p", 1430 this, effect->desc().name, chain.get()); 1431 return BAD_VALUE; 1432 } 1433 1434 effect->setOffloaded(mType == OFFLOAD, mId); 1435 1436 status_t status = chain->addEffect_l(effect); 1437 if (status != NO_ERROR) { 1438 if (chainCreated) { 1439 removeEffectChain_l(chain); 1440 } 1441 return status; 1442 } 1443 1444 effect->setDevice(mOutDevice); 1445 effect->setDevice(mInDevice); 1446 effect->setMode(mAudioFlinger->getMode()); 1447 effect->setAudioSource(mAudioSource); 1448 1449 return NO_ERROR; 1450 } 1451 1452 void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect, bool release) { 1453 1454 ALOGV("%s %p effect %p", __FUNCTION__, this, effect.get()); 1455 effect_descriptor_t desc = effect->desc(); 1456 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 1457 detachAuxEffect_l(effect->id()); 1458 } 1459 1460 sp<EffectChain> chain = effect->chain().promote(); 1461 if (chain != 0) { 1462 // remove effect chain if removing last effect 1463 if (chain->removeEffect_l(effect, release) == 0) { 1464 removeEffectChain_l(chain); 1465 } 1466 } else { 1467 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 1468 } 1469 } 1470 1471 void AudioFlinger::ThreadBase::lockEffectChains_l( 1472 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1473 { 1474 effectChains = mEffectChains; 1475 for (size_t i = 0; i < mEffectChains.size(); i++) { 1476 mEffectChains[i]->lock(); 1477 } 1478 } 1479 1480 void AudioFlinger::ThreadBase::unlockEffectChains( 1481 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 1482 { 1483 for (size_t i = 0; i < effectChains.size(); i++) { 1484 effectChains[i]->unlock(); 1485 } 1486 } 1487 1488 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(audio_session_t sessionId) 1489 { 1490 Mutex::Autolock _l(mLock); 1491 return getEffectChain_l(sessionId); 1492 } 1493 1494 sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(audio_session_t sessionId) 1495 const 1496 { 1497 size_t size = mEffectChains.size(); 1498 for (size_t i = 0; i < size; i++) { 1499 if (mEffectChains[i]->sessionId() == sessionId) { 1500 return mEffectChains[i]; 1501 } 1502 } 1503 return 0; 1504 } 1505 1506 void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 1507 { 1508 Mutex::Autolock _l(mLock); 1509 size_t size = mEffectChains.size(); 1510 for (size_t i = 0; i < size; i++) { 1511 mEffectChains[i]->setMode_l(mode); 1512 } 1513 } 1514 1515 void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config) 1516 { 1517 config->type = AUDIO_PORT_TYPE_MIX; 1518 config->ext.mix.handle = mId; 1519 config->sample_rate = mSampleRate; 1520 config->format = mFormat; 1521 config->channel_mask = mChannelMask; 1522 config->config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE|AUDIO_PORT_CONFIG_CHANNEL_MASK| 1523 AUDIO_PORT_CONFIG_FORMAT; 1524 } 1525 1526 void AudioFlinger::ThreadBase::systemReady() 1527 { 1528 Mutex::Autolock _l(mLock); 1529 if (mSystemReady) { 1530 return; 1531 } 1532 mSystemReady = true; 1533 1534 for (size_t i = 0; i < mPendingConfigEvents.size(); i++) { 1535 sendConfigEvent_l(mPendingConfigEvents.editItemAt(i)); 1536 } 1537 mPendingConfigEvents.clear(); 1538 } 1539 1540 template <typename T> 1541 ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::add(const sp<T> &track) { 1542 ssize_t index = mActiveTracks.indexOf(track); 1543 if (index >= 0) { 1544 ALOGW("ActiveTracks<T>::add track %p already there", track.get()); 1545 return index; 1546 } 1547 logTrack("add", track); 1548 mActiveTracksGeneration++; 1549 mLatestActiveTrack = track; 1550 ++mBatteryCounter[track->uid()].second; 1551 return mActiveTracks.add(track); 1552 } 1553 1554 template <typename T> 1555 ssize_t AudioFlinger::ThreadBase::ActiveTracks<T>::remove(const sp<T> &track) { 1556 ssize_t index = mActiveTracks.remove(track); 1557 if (index < 0) { 1558 ALOGW("ActiveTracks<T>::remove nonexistent track %p", track.get()); 1559 return index; 1560 } 1561 logTrack("remove", track); 1562 mActiveTracksGeneration++; 1563 --mBatteryCounter[track->uid()].second; 1564 // mLatestActiveTrack is not cleared even if is the same as track. 1565 return index; 1566 } 1567 1568 template <typename T> 1569 void AudioFlinger::ThreadBase::ActiveTracks<T>::clear() { 1570 for (const sp<T> &track : mActiveTracks) { 1571 BatteryNotifier::getInstance().noteStopAudio(track->uid()); 1572 logTrack("clear", track); 1573 } 1574 mLastActiveTracksGeneration = mActiveTracksGeneration; 1575 mActiveTracks.clear(); 1576 mLatestActiveTrack.clear(); 1577 mBatteryCounter.clear(); 1578 } 1579 1580 template <typename T> 1581 void AudioFlinger::ThreadBase::ActiveTracks<T>::updatePowerState( 1582 sp<ThreadBase> thread, bool force) { 1583 // Updates ActiveTracks client uids to the thread wakelock. 1584 if (mActiveTracksGeneration != mLastActiveTracksGeneration || force) { 1585 thread->updateWakeLockUids_l(getWakeLockUids()); 1586 mLastActiveTracksGeneration = mActiveTracksGeneration; 1587 } 1588 1589 // Updates BatteryNotifier uids 1590 for (auto it = mBatteryCounter.begin(); it != mBatteryCounter.end();) { 1591 const uid_t uid = it->first; 1592 ssize_t &previous = it->second.first; 1593 ssize_t ¤t = it->second.second; 1594 if (current > 0) { 1595 if (previous == 0) { 1596 BatteryNotifier::getInstance().noteStartAudio(uid); 1597 } 1598 previous = current; 1599 ++it; 1600 } else if (current == 0) { 1601 if (previous > 0) { 1602 BatteryNotifier::getInstance().noteStopAudio(uid); 1603 } 1604 it = mBatteryCounter.erase(it); // std::map<> is stable on iterator erase. 1605 } else /* (current < 0) */ { 1606 LOG_ALWAYS_FATAL("negative battery count %zd", current); 1607 } 1608 } 1609 } 1610 1611 template <typename T> 1612 void AudioFlinger::ThreadBase::ActiveTracks<T>::logTrack( 1613 const char *funcName, const sp<T> &track) const { 1614 if (mLocalLog != nullptr) { 1615 String8 result; 1616 track->appendDump(result, false /* active */); 1617 mLocalLog->log("AT::%-10s(%p) %s", funcName, track.get(), result.string()); 1618 } 1619 } 1620 1621 void AudioFlinger::ThreadBase::broadcast_l() 1622 { 1623 // Thread could be blocked waiting for async 1624 // so signal it to handle state changes immediately 1625 // If threadLoop is currently unlocked a signal of mWaitWorkCV will 1626 // be lost so we also flag to prevent it blocking on mWaitWorkCV 1627 mSignalPending = true; 1628 mWaitWorkCV.broadcast(); 1629 } 1630 1631 // ---------------------------------------------------------------------------- 1632 // Playback 1633 // ---------------------------------------------------------------------------- 1634 1635 AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1636 AudioStreamOut* output, 1637 audio_io_handle_t id, 1638 audio_devices_t device, 1639 type_t type, 1640 bool systemReady) 1641 : ThreadBase(audioFlinger, id, device, AUDIO_DEVICE_NONE, type, systemReady), 1642 mNormalFrameCount(0), mSinkBuffer(NULL), 1643 mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1644 mMixerBuffer(NULL), 1645 mMixerBufferSize(0), 1646 mMixerBufferFormat(AUDIO_FORMAT_INVALID), 1647 mMixerBufferValid(false), 1648 mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision), 1649 mEffectBuffer(NULL), 1650 mEffectBufferSize(0), 1651 mEffectBufferFormat(AUDIO_FORMAT_INVALID), 1652 mEffectBufferValid(false), 1653 mSuspended(0), mBytesWritten(0), 1654 mFramesWritten(0), 1655 mSuspendedFrames(0), 1656 mActiveTracks(&this->mLocalLog), 1657 // mStreamTypes[] initialized in constructor body 1658 mOutput(output), 1659 mLastWriteTime(-1), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1660 mMixerStatus(MIXER_IDLE), 1661 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1662 mStandbyDelayNs(AudioFlinger::mStandbyTimeInNsecs), 1663 mBytesRemaining(0), 1664 mCurrentWriteLength(0), 1665 mUseAsyncWrite(false), 1666 mWriteAckSequence(0), 1667 mDrainSequence(0), 1668 mScreenState(AudioFlinger::mScreenState), 1669 // index 0 is reserved for normal mixer's submix 1670 mFastTrackAvailMask(((1 << FastMixerState::sMaxFastTracks) - 1) & ~1), 1671 mHwSupportsPause(false), mHwPaused(false), mFlushPending(false), 1672 mLeftVolFloat(-1.0), mRightVolFloat(-1.0) 1673 { 1674 snprintf(mThreadName, kThreadNameLength, "AudioOut_%X", id); 1675 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 1676 1677 // Assumes constructor is called by AudioFlinger with it's mLock held, but 1678 // it would be safer to explicitly pass initial masterVolume/masterMute as 1679 // parameter. 1680 // 1681 // If the HAL we are using has support for master volume or master mute, 1682 // then do not attenuate or mute during mixing (just leave the volume at 1.0 1683 // and the mute set to false). 1684 mMasterVolume = audioFlinger->masterVolume_l(); 1685 mMasterMute = audioFlinger->masterMute_l(); 1686 if (mOutput && mOutput->audioHwDev) { 1687 if (mOutput->audioHwDev->canSetMasterVolume()) { 1688 mMasterVolume = 1.0; 1689 } 1690 1691 if (mOutput->audioHwDev->canSetMasterMute()) { 1692 mMasterMute = false; 1693 } 1694 } 1695 1696 readOutputParameters_l(); 1697 1698 // ++ operator does not compile 1699 for (audio_stream_type_t stream = AUDIO_STREAM_MIN; stream < AUDIO_STREAM_CNT; 1700 stream = (audio_stream_type_t) (stream + 1)) { 1701 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1702 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1703 } 1704 } 1705 1706 AudioFlinger::PlaybackThread::~PlaybackThread() 1707 { 1708 mAudioFlinger->unregisterWriter(mNBLogWriter); 1709 free(mSinkBuffer); 1710 free(mMixerBuffer); 1711 free(mEffectBuffer); 1712 } 1713 1714 void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1715 { 1716 dumpInternals(fd, args); 1717 dumpTracks(fd, args); 1718 dumpEffectChains(fd, args); 1719 dprintf(fd, " Local log:\n"); 1720 mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */); 1721 } 1722 1723 void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused) 1724 { 1725 String8 result; 1726 1727 result.appendFormat(" Stream volumes in dB: "); 1728 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1729 const stream_type_t *st = &mStreamTypes[i]; 1730 if (i > 0) { 1731 result.appendFormat(", "); 1732 } 1733 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1734 if (st->mute) { 1735 result.append("M"); 1736 } 1737 } 1738 result.append("\n"); 1739 write(fd, result.string(), result.length()); 1740 result.clear(); 1741 1742 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1743 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1744 dprintf(fd, " Normal mixer raw underrun counters: partial=%u empty=%u\n", 1745 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1746 1747 size_t numtracks = mTracks.size(); 1748 size_t numactive = mActiveTracks.size(); 1749 dprintf(fd, " %zu Tracks", numtracks); 1750 size_t numactiveseen = 0; 1751 const char *prefix = " "; 1752 if (numtracks) { 1753 dprintf(fd, " of which %zu are active\n", numactive); 1754 result.append(prefix); 1755 Track::appendDumpHeader(result); 1756 for (size_t i = 0; i < numtracks; ++i) { 1757 sp<Track> track = mTracks[i]; 1758 if (track != 0) { 1759 bool active = mActiveTracks.indexOf(track) >= 0; 1760 if (active) { 1761 numactiveseen++; 1762 } 1763 result.append(prefix); 1764 track->appendDump(result, active); 1765 } 1766 } 1767 } else { 1768 result.append("\n"); 1769 } 1770 if (numactiveseen != numactive) { 1771 // some tracks in the active list were not in the tracks list 1772 result.append(" The following tracks are in the active list but" 1773 " not in the track list\n"); 1774 result.append(prefix); 1775 Track::appendDumpHeader(result); 1776 for (size_t i = 0; i < numactive; ++i) { 1777 sp<Track> track = mActiveTracks[i]; 1778 if (mTracks.indexOf(track) < 0) { 1779 result.append(prefix); 1780 track->appendDump(result, true /* active */); 1781 } 1782 } 1783 } 1784 1785 write(fd, result.string(), result.size()); 1786 } 1787 1788 void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1789 { 1790 dumpBase(fd, args); 1791 1792 dprintf(fd, " Normal frame count: %zu\n", mNormalFrameCount); 1793 dprintf(fd, " Last write occurred (msecs): %llu\n", 1794 (unsigned long long) ns2ms(systemTime() - mLastWriteTime)); 1795 dprintf(fd, " Total writes: %d\n", mNumWrites); 1796 dprintf(fd, " Delayed writes: %d\n", mNumDelayedWrites); 1797 dprintf(fd, " Blocked in write: %s\n", mInWrite ? "yes" : "no"); 1798 dprintf(fd, " Suspend count: %d\n", mSuspended); 1799 dprintf(fd, " Sink buffer : %p\n", mSinkBuffer); 1800 dprintf(fd, " Mixer buffer: %p\n", mMixerBuffer); 1801 dprintf(fd, " Effect buffer: %p\n", mEffectBuffer); 1802 dprintf(fd, " Fast track availMask=%#x\n", mFastTrackAvailMask); 1803 dprintf(fd, " Standby delay ns=%lld\n", (long long)mStandbyDelayNs); 1804 AudioStreamOut *output = mOutput; 1805 audio_output_flags_t flags = output != NULL ? output->flags : AUDIO_OUTPUT_FLAG_NONE; 1806 dprintf(fd, " AudioStreamOut: %p flags %#x (%s)\n", 1807 output, flags, outputFlagsToString(flags).c_str()); 1808 dprintf(fd, " Frames written: %lld\n", (long long)mFramesWritten); 1809 dprintf(fd, " Suspended frames: %lld\n", (long long)mSuspendedFrames); 1810 if (mPipeSink.get() != nullptr) { 1811 dprintf(fd, " PipeSink frames written: %lld\n", (long long)mPipeSink->framesWritten()); 1812 } 1813 if (output != nullptr) { 1814 dprintf(fd, " Hal stream dump:\n"); 1815 (void)output->stream->dump(fd); 1816 } 1817 } 1818 1819 // Thread virtuals 1820 1821 void AudioFlinger::PlaybackThread::onFirstRef() 1822 { 1823 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 1824 } 1825 1826 // ThreadBase virtuals 1827 void AudioFlinger::PlaybackThread::preExit() 1828 { 1829 ALOGV(" preExit()"); 1830 // FIXME this is using hard-coded strings but in the future, this functionality will be 1831 // converted to use audio HAL extensions required to support tunneling 1832 status_t result = mOutput->stream->setParameters(String8("exiting=1")); 1833 ALOGE_IF(result != OK, "Error when setting parameters on exit: %d", result); 1834 } 1835 1836 // PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1837 sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1838 const sp<AudioFlinger::Client>& client, 1839 audio_stream_type_t streamType, 1840 uint32_t sampleRate, 1841 audio_format_t format, 1842 audio_channel_mask_t channelMask, 1843 size_t *pFrameCount, 1844 const sp<IMemory>& sharedBuffer, 1845 audio_session_t sessionId, 1846 audio_output_flags_t *flags, 1847 pid_t tid, 1848 uid_t uid, 1849 status_t *status, 1850 audio_port_handle_t portId) 1851 { 1852 size_t frameCount = *pFrameCount; 1853 sp<Track> track; 1854 status_t lStatus; 1855 audio_output_flags_t outputFlags = mOutput->flags; 1856 1857 // special case for FAST flag considered OK if fast mixer is present 1858 if (hasFastMixer()) { 1859 outputFlags = (audio_output_flags_t)(outputFlags | AUDIO_OUTPUT_FLAG_FAST); 1860 } 1861 1862 // Check if requested flags are compatible with output stream flags 1863 if ((*flags & outputFlags) != *flags) { 1864 ALOGW("createTrack_l(): mismatch between requested flags (%08x) and output flags (%08x)", 1865 *flags, outputFlags); 1866 *flags = (audio_output_flags_t)(*flags & outputFlags); 1867 } 1868 1869 // client expresses a preference for FAST, but we get the final say 1870 if (*flags & AUDIO_OUTPUT_FLAG_FAST) { 1871 if ( 1872 // PCM data 1873 audio_is_linear_pcm(format) && 1874 // TODO: extract as a data library function that checks that a computationally 1875 // expensive downmixer is not required: isFastOutputChannelConversion() 1876 (channelMask == mChannelMask || 1877 mChannelMask != AUDIO_CHANNEL_OUT_STEREO || 1878 (channelMask == AUDIO_CHANNEL_OUT_MONO 1879 /* && mChannelMask == AUDIO_CHANNEL_OUT_STEREO */)) && 1880 // hardware sample rate 1881 (sampleRate == mSampleRate) && 1882 // normal mixer has an associated fast mixer 1883 hasFastMixer() && 1884 // there are sufficient fast track slots available 1885 (mFastTrackAvailMask != 0) 1886 // FIXME test that MixerThread for this fast track has a capable output HAL 1887 // FIXME add a permission test also? 1888 ) { 1889 // static tracks can have any nonzero framecount, streaming tracks check against minimum. 1890 if (sharedBuffer == 0) { 1891 // read the fast track multiplier property the first time it is needed 1892 int ok = pthread_once(&sFastTrackMultiplierOnce, sFastTrackMultiplierInit); 1893 if (ok != 0) { 1894 ALOGE("%s pthread_once failed: %d", __func__, ok); 1895 } 1896 frameCount = max(frameCount, mFrameCount * sFastTrackMultiplier); // incl framecount 0 1897 } 1898 1899 // check compatibility with audio effects. 1900 { // scope for mLock 1901 Mutex::Autolock _l(mLock); 1902 for (audio_session_t session : { 1903 AUDIO_SESSION_OUTPUT_STAGE, 1904 AUDIO_SESSION_OUTPUT_MIX, 1905 sessionId, 1906 }) { 1907 sp<EffectChain> chain = getEffectChain_l(session); 1908 if (chain.get() != nullptr) { 1909 audio_output_flags_t old = *flags; 1910 chain->checkOutputFlagCompatibility(flags); 1911 if (old != *flags) { 1912 ALOGV("AUDIO_OUTPUT_FLAGS denied by effect, session=%d old=%#x new=%#x", 1913 (int)session, (int)old, (int)*flags); 1914 } 1915 } 1916 } 1917 } 1918 ALOGV_IF((*flags & AUDIO_OUTPUT_FLAG_FAST) != 0, 1919 "AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 1920 frameCount, mFrameCount); 1921 } else { 1922 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: sharedBuffer=%p frameCount=%zu " 1923 "mFrameCount=%zu format=%#x mFormat=%#x isLinear=%d channelMask=%#x " 1924 "sampleRate=%u mSampleRate=%u " 1925 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1926 sharedBuffer.get(), frameCount, mFrameCount, format, mFormat, 1927 audio_is_linear_pcm(format), 1928 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1929 *flags = (audio_output_flags_t)(*flags & ~AUDIO_OUTPUT_FLAG_FAST); 1930 } 1931 } 1932 // For normal PCM streaming tracks, update minimum frame count. 1933 // For compatibility with AudioTrack calculation, buffer depth is forced 1934 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1935 // This is probably too conservative, but legacy application code may depend on it. 1936 // If you change this calculation, also review the start threshold which is related. 1937 if (!(*flags & AUDIO_OUTPUT_FLAG_FAST) 1938 && audio_has_proportional_frames(format) && sharedBuffer == 0) { 1939 // this must match AudioTrack.cpp calculateMinFrameCount(). 1940 // TODO: Move to a common library 1941 uint32_t latencyMs = 0; 1942 lStatus = mOutput->stream->getLatency(&latencyMs); 1943 if (lStatus != OK) { 1944 ALOGE("Error when retrieving output stream latency: %d", lStatus); 1945 goto Exit; 1946 } 1947 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1948 if (minBufCount < 2) { 1949 minBufCount = 2; 1950 } 1951 // For normal mixing tracks, if speed is > 1.0f (normal), AudioTrack 1952 // or the client should compute and pass in a larger buffer request. 1953 size_t minFrameCount = 1954 minBufCount * sourceFramesNeededWithTimestretch( 1955 sampleRate, mNormalFrameCount, 1956 mSampleRate, AUDIO_TIMESTRETCH_SPEED_NORMAL /*speed*/); 1957 if (frameCount < minFrameCount) { // including frameCount == 0 1958 frameCount = minFrameCount; 1959 } 1960 } 1961 *pFrameCount = frameCount; 1962 1963 switch (mType) { 1964 1965 case DIRECT: 1966 if (audio_is_linear_pcm(format)) { // TODO maybe use audio_has_proportional_frames()? 1967 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1968 ALOGE("createTrack_l() Bad parameter: sampleRate %u format %#x, channelMask 0x%08x " 1969 "for output %p with format %#x", 1970 sampleRate, format, channelMask, mOutput, mFormat); 1971 lStatus = BAD_VALUE; 1972 goto Exit; 1973 } 1974 } 1975 break; 1976 1977 case OFFLOAD: 1978 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1979 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %#x, channelMask 0x%08x \"" 1980 "for output %p with format %#x", 1981 sampleRate, format, channelMask, mOutput, mFormat); 1982 lStatus = BAD_VALUE; 1983 goto Exit; 1984 } 1985 break; 1986 1987 default: 1988 if (!audio_is_linear_pcm(format)) { 1989 ALOGE("createTrack_l() Bad parameter: format %#x \"" 1990 "for output %p with format %#x", 1991 format, mOutput, mFormat); 1992 lStatus = BAD_VALUE; 1993 goto Exit; 1994 } 1995 if (sampleRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) { 1996 ALOGE("Sample rate out of range: %u mSampleRate %u", sampleRate, mSampleRate); 1997 lStatus = BAD_VALUE; 1998 goto Exit; 1999 } 2000 break; 2001 2002 } 2003 2004 lStatus = initCheck(); 2005 if (lStatus != NO_ERROR) { 2006 ALOGE("createTrack_l() audio driver not initialized"); 2007 goto Exit; 2008 } 2009 2010 { // scope for mLock 2011 Mutex::Autolock _l(mLock); 2012 2013 // all tracks in same audio session must share the same routing strategy otherwise 2014 // conflicts will happen when tracks are moved from one output to another by audio policy 2015 // manager 2016 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 2017 for (size_t i = 0; i < mTracks.size(); ++i) { 2018 sp<Track> t = mTracks[i]; 2019 if (t != 0 && t->isExternalTrack()) { 2020 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 2021 if (sessionId == t->sessionId() && strategy != actual) { 2022 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 2023 strategy, actual); 2024 lStatus = BAD_VALUE; 2025 goto Exit; 2026 } 2027 } 2028 } 2029 2030 track = new Track(this, client, streamType, sampleRate, format, 2031 channelMask, frameCount, 2032 nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer, 2033 sessionId, uid, *flags, TrackBase::TYPE_DEFAULT, portId); 2034 2035 lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY; 2036 if (lStatus != NO_ERROR) { 2037 ALOGE("createTrack_l() initCheck failed %d; no control block?", lStatus); 2038 // track must be cleared from the caller as the caller has the AF lock 2039 goto Exit; 2040 } 2041 mTracks.add(track); 2042 2043 sp<EffectChain> chain = getEffectChain_l(sessionId); 2044 if (chain != 0) { 2045 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 2046 track->setMainBuffer(chain->inBuffer()); 2047 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 2048 chain->incTrackCnt(); 2049 } 2050 2051 if ((*flags & AUDIO_OUTPUT_FLAG_FAST) && (tid != -1)) { 2052 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 2053 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 2054 // so ask activity manager to do this on our behalf 2055 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/); 2056 } 2057 } 2058 2059 lStatus = NO_ERROR; 2060 2061 Exit: 2062 *status = lStatus; 2063 return track; 2064 } 2065 2066 uint32_t AudioFlinger::PlaybackThread::correctLatency_l(uint32_t latency) const 2067 { 2068 return latency; 2069 } 2070 2071 uint32_t AudioFlinger::PlaybackThread::latency() const 2072 { 2073 Mutex::Autolock _l(mLock); 2074 return latency_l(); 2075 } 2076 uint32_t AudioFlinger::PlaybackThread::latency_l() const 2077 { 2078 uint32_t latency; 2079 if (initCheck() == NO_ERROR && mOutput->stream->getLatency(&latency) == OK) { 2080 return correctLatency_l(latency); 2081 } 2082 return 0; 2083 } 2084 2085 void AudioFlinger::PlaybackThread::setMasterVolume(float value) 2086 { 2087 Mutex::Autolock _l(mLock); 2088 // Don't apply master volume in SW if our HAL can do it for us. 2089 if (mOutput && mOutput->audioHwDev && 2090 mOutput->audioHwDev->canSetMasterVolume()) { 2091 mMasterVolume = 1.0; 2092 } else { 2093 mMasterVolume = value; 2094 } 2095 } 2096 2097 void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 2098 { 2099 if (isDuplicating()) { 2100 return; 2101 } 2102 Mutex::Autolock _l(mLock); 2103 // Don't apply master mute in SW if our HAL can do it for us. 2104 if (mOutput && mOutput->audioHwDev && 2105 mOutput->audioHwDev->canSetMasterMute()) { 2106 mMasterMute = false; 2107 } else { 2108 mMasterMute = muted; 2109 } 2110 } 2111 2112 void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 2113 { 2114 Mutex::Autolock _l(mLock); 2115 mStreamTypes[stream].volume = value; 2116 broadcast_l(); 2117 } 2118 2119 void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 2120 { 2121 Mutex::Autolock _l(mLock); 2122 mStreamTypes[stream].mute = muted; 2123 broadcast_l(); 2124 } 2125 2126 float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 2127 { 2128 Mutex::Autolock _l(mLock); 2129 return mStreamTypes[stream].volume; 2130 } 2131 2132 // addTrack_l() must be called with ThreadBase::mLock held 2133 status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 2134 { 2135 status_t status = ALREADY_EXISTS; 2136 2137 if (mActiveTracks.indexOf(track) < 0) { 2138 // the track is newly added, make sure it fills up all its 2139 // buffers before playing. This is to ensure the client will 2140 // effectively get the latency it requested. 2141 if (track->isExternalTrack()) { 2142 TrackBase::track_state state = track->mState; 2143 mLock.unlock(); 2144 status = AudioSystem::startOutput(mId, track->streamType(), 2145 track->sessionId()); 2146 mLock.lock(); 2147 // abort track was stopped/paused while we released the lock 2148 if (state != track->mState) { 2149 if (status == NO_ERROR) { 2150 mLock.unlock(); 2151 AudioSystem::stopOutput(mId, track->streamType(), 2152 track->sessionId()); 2153 mLock.lock(); 2154 } 2155 return INVALID_OPERATION; 2156 } 2157 // abort if start is rejected by audio policy manager 2158 if (status != NO_ERROR) { 2159 return PERMISSION_DENIED; 2160 } 2161 #ifdef ADD_BATTERY_DATA 2162 // to track the speaker usage 2163 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 2164 #endif 2165 } 2166 2167 // set retry count for buffer fill 2168 if (track->isOffloaded()) { 2169 if (track->isStopping_1()) { 2170 track->mRetryCount = kMaxTrackStopRetriesOffload; 2171 } else { 2172 track->mRetryCount = kMaxTrackStartupRetriesOffload; 2173 } 2174 track->mFillingUpStatus = mStandby ? Track::FS_FILLING : Track::FS_FILLED; 2175 } else { 2176 track->mRetryCount = kMaxTrackStartupRetries; 2177 track->mFillingUpStatus = 2178 track->sharedBuffer() != 0 ? Track::FS_FILLED : Track::FS_FILLING; 2179 } 2180 2181 track->mResetDone = false; 2182 track->mPresentationCompleteFrames = 0; 2183 mActiveTracks.add(track); 2184 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2185 if (chain != 0) { 2186 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), 2187 track->sessionId()); 2188 chain->incActiveTrackCnt(); 2189 } 2190 2191 status = NO_ERROR; 2192 } 2193 2194 onAddNewTrack_l(); 2195 return status; 2196 } 2197 2198 bool AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 2199 { 2200 track->terminate(); 2201 // active tracks are removed by threadLoop() 2202 bool trackActive = (mActiveTracks.indexOf(track) >= 0); 2203 track->mState = TrackBase::STOPPED; 2204 if (!trackActive) { 2205 removeTrack_l(track); 2206 } else if (track->isFastTrack() || track->isOffloaded() || track->isDirect()) { 2207 track->mState = TrackBase::STOPPING_1; 2208 } 2209 2210 return trackActive; 2211 } 2212 2213 void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 2214 { 2215 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 2216 2217 String8 result; 2218 track->appendDump(result, false /* active */); 2219 mLocalLog.log("removeTrack_l (%p) %s", track.get(), result.string()); 2220 2221 mTracks.remove(track); 2222 deleteTrackName_l(track->name()); 2223 // redundant as track is about to be destroyed, for dumpsys only 2224 track->mName = -1; 2225 if (track->isFastTrack()) { 2226 int index = track->mFastIndex; 2227 ALOG_ASSERT(0 < index && index < (int)FastMixerState::sMaxFastTracks); 2228 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 2229 mFastTrackAvailMask |= 1 << index; 2230 // redundant as track is about to be destroyed, for dumpsys only 2231 track->mFastIndex = -1; 2232 } 2233 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 2234 if (chain != 0) { 2235 chain->decTrackCnt(); 2236 } 2237 } 2238 2239 String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 2240 { 2241 Mutex::Autolock _l(mLock); 2242 String8 out_s8; 2243 if (initCheck() == NO_ERROR && mOutput->stream->getParameters(keys, &out_s8) == OK) { 2244 return out_s8; 2245 } 2246 return String8(); 2247 } 2248 2249 void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 2250 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 2251 ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event); 2252 2253 desc->mIoHandle = mId; 2254 2255 switch (event) { 2256 case AUDIO_OUTPUT_OPENED: 2257 case AUDIO_OUTPUT_REGISTERED: 2258 case AUDIO_OUTPUT_CONFIG_CHANGED: 2259 desc->mPatch = mPatch; 2260 desc->mChannelMask = mChannelMask; 2261 desc->mSamplingRate = mSampleRate; 2262 desc->mFormat = mFormat; 2263 desc->mFrameCount = mNormalFrameCount; // FIXME see 2264 // AudioFlinger::frameCount(audio_io_handle_t) 2265 desc->mFrameCountHAL = mFrameCount; 2266 desc->mLatency = latency_l(); 2267 break; 2268 2269 case AUDIO_OUTPUT_CLOSED: 2270 default: 2271 break; 2272 } 2273 mAudioFlinger->ioConfigChanged(event, desc, pid); 2274 } 2275 2276 void AudioFlinger::PlaybackThread::onWriteReady() 2277 { 2278 mCallbackThread->resetWriteBlocked(); 2279 } 2280 2281 void AudioFlinger::PlaybackThread::onDrainReady() 2282 { 2283 mCallbackThread->resetDraining(); 2284 } 2285 2286 void AudioFlinger::PlaybackThread::onError() 2287 { 2288 mCallbackThread->setAsyncError(); 2289 } 2290 2291 void AudioFlinger::PlaybackThread::resetWriteBlocked(uint32_t sequence) 2292 { 2293 Mutex::Autolock _l(mLock); 2294 // reject out of sequence requests 2295 if ((mWriteAckSequence & 1) && (sequence == mWriteAckSequence)) { 2296 mWriteAckSequence &= ~1; 2297 mWaitWorkCV.signal(); 2298 } 2299 } 2300 2301 void AudioFlinger::PlaybackThread::resetDraining(uint32_t sequence) 2302 { 2303 Mutex::Autolock _l(mLock); 2304 // reject out of sequence requests 2305 if ((mDrainSequence & 1) && (sequence == mDrainSequence)) { 2306 mDrainSequence &= ~1; 2307 mWaitWorkCV.signal(); 2308 } 2309 } 2310 2311 void AudioFlinger::PlaybackThread::readOutputParameters_l() 2312 { 2313 // unfortunately we have no way of recovering from errors here, hence the LOG_ALWAYS_FATAL 2314 mSampleRate = mOutput->getSampleRate(); 2315 mChannelMask = mOutput->getChannelMask(); 2316 if (!audio_is_output_channel(mChannelMask)) { 2317 LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask); 2318 } 2319 if ((mType == MIXER || mType == DUPLICATING) 2320 && !isValidPcmSinkChannelMask(mChannelMask)) { 2321 LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output", 2322 mChannelMask); 2323 } 2324 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 2325 2326 // Get actual HAL format. 2327 status_t result = mOutput->stream->getFormat(&mHALFormat); 2328 LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result); 2329 // Get format from the shim, which will be different than the HAL format 2330 // if playing compressed audio over HDMI passthrough. 2331 mFormat = mOutput->getFormat(); 2332 if (!audio_is_valid_format(mFormat)) { 2333 LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat); 2334 } 2335 if ((mType == MIXER || mType == DUPLICATING) 2336 && !isValidPcmSinkFormat(mFormat)) { 2337 LOG_FATAL("HAL format %#x not supported for mixed output", 2338 mFormat); 2339 } 2340 mFrameSize = mOutput->getFrameSize(); 2341 result = mOutput->stream->getBufferSize(&mBufferSize); 2342 LOG_ALWAYS_FATAL_IF(result != OK, 2343 "Error when retrieving output stream buffer size: %d", result); 2344 mFrameCount = mBufferSize / mFrameSize; 2345 if (mFrameCount & 15) { 2346 ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames", 2347 mFrameCount); 2348 } 2349 2350 if (mOutput->flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) { 2351 if (mOutput->stream->setCallback(this) == OK) { 2352 mUseAsyncWrite = true; 2353 mCallbackThread = new AudioFlinger::AsyncCallbackThread(this); 2354 } 2355 } 2356 2357 mHwSupportsPause = false; 2358 if (mOutput->flags & AUDIO_OUTPUT_FLAG_DIRECT) { 2359 bool supportsPause = false, supportsResume = false; 2360 if (mOutput->stream->supportsPauseAndResume(&supportsPause, &supportsResume) == OK) { 2361 if (supportsPause && supportsResume) { 2362 mHwSupportsPause = true; 2363 } else if (supportsPause) { 2364 ALOGW("direct output implements pause but not resume"); 2365 } else if (supportsResume) { 2366 ALOGW("direct output implements resume but not pause"); 2367 } 2368 } 2369 } 2370 if (!mHwSupportsPause && mOutput->flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) { 2371 LOG_ALWAYS_FATAL("HW_AV_SYNC requested but HAL does not implement pause and resume"); 2372 } 2373 2374 if (mType == DUPLICATING && mMixerBufferEnabled && mEffectBufferEnabled) { 2375 // For best precision, we use float instead of the associated output 2376 // device format (typically PCM 16 bit). 2377 2378 mFormat = AUDIO_FORMAT_PCM_FLOAT; 2379 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 2380 mBufferSize = mFrameSize * mFrameCount; 2381 2382 // TODO: We currently use the associated output device channel mask and sample rate. 2383 // (1) Perhaps use the ORed channel mask of all downstream MixerThreads 2384 // (if a valid mask) to avoid premature downmix. 2385 // (2) Perhaps use the maximum sample rate of all downstream MixerThreads 2386 // instead of the output device sample rate to avoid loss of high frequency information. 2387 // This may need to be updated as MixerThread/OutputTracks are added and not here. 2388 } 2389 2390 // Calculate size of normal sink buffer relative to the HAL output buffer size 2391 double multiplier = 1.0; 2392 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || 2393 kUseFastMixer == FastMixer_Dynamic)) { 2394 size_t minNormalFrameCount = (kMinNormalSinkBufferSizeMs * mSampleRate) / 1000; 2395 size_t maxNormalFrameCount = (kMaxNormalSinkBufferSizeMs * mSampleRate) / 1000; 2396 2397 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2398 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2399 maxNormalFrameCount = maxNormalFrameCount & ~15; 2400 if (maxNormalFrameCount < minNormalFrameCount) { 2401 maxNormalFrameCount = minNormalFrameCount; 2402 } 2403 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2404 if (multiplier <= 1.0) { 2405 multiplier = 1.0; 2406 } else if (multiplier <= 2.0) { 2407 if (2 * mFrameCount <= maxNormalFrameCount) { 2408 multiplier = 2.0; 2409 } else { 2410 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2411 } 2412 } else { 2413 multiplier = floor(multiplier); 2414 } 2415 } 2416 mNormalFrameCount = multiplier * mFrameCount; 2417 // round up to nearest 16 frames to satisfy AudioMixer 2418 if (mType == MIXER || mType == DUPLICATING) { 2419 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2420 } 2421 ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount, 2422 mNormalFrameCount); 2423 2424 // Check if we want to throttle the processing to no more than 2x normal rate 2425 mThreadThrottle = property_get_bool("af.thread.throttle", true /* default_value */); 2426 mThreadThrottleTimeMs = 0; 2427 mThreadThrottleEndMs = 0; 2428 mHalfBufferMs = mNormalFrameCount * 1000 / (2 * mSampleRate); 2429 2430 // mSinkBuffer is the sink buffer. Size is always multiple-of-16 frames. 2431 // Originally this was int16_t[] array, need to remove legacy implications. 2432 free(mSinkBuffer); 2433 mSinkBuffer = NULL; 2434 // For sink buffer size, we use the frame size from the downstream sink to avoid problems 2435 // with non PCM formats for compressed music, e.g. AAC, and Offload threads. 2436 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 2437 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 2438 2439 // We resize the mMixerBuffer according to the requirements of the sink buffer which 2440 // drives the output. 2441 free(mMixerBuffer); 2442 mMixerBuffer = NULL; 2443 if (mMixerBufferEnabled) { 2444 mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // also valid: AUDIO_FORMAT_PCM_16_BIT. 2445 mMixerBufferSize = mNormalFrameCount * mChannelCount 2446 * audio_bytes_per_sample(mMixerBufferFormat); 2447 (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize); 2448 } 2449 free(mEffectBuffer); 2450 mEffectBuffer = NULL; 2451 if (mEffectBufferEnabled) { 2452 mEffectBufferFormat = AUDIO_FORMAT_PCM_16_BIT; // Note: Effects support 16b only 2453 mEffectBufferSize = mNormalFrameCount * mChannelCount 2454 * audio_bytes_per_sample(mEffectBufferFormat); 2455 (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize); 2456 } 2457 2458 // force reconfiguration of effect chains and engines to take new buffer size and audio 2459 // parameters into account 2460 // Note that mLock is not held when readOutputParameters_l() is called from the constructor 2461 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2462 // matter. 2463 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2464 Vector< sp<EffectChain> > effectChains = mEffectChains; 2465 for (size_t i = 0; i < effectChains.size(); i ++) { 2466 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2467 } 2468 } 2469 2470 2471 status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2472 { 2473 if (halFrames == NULL || dspFrames == NULL) { 2474 return BAD_VALUE; 2475 } 2476 Mutex::Autolock _l(mLock); 2477 if (initCheck() != NO_ERROR) { 2478 return INVALID_OPERATION; 2479 } 2480 int64_t framesWritten = mBytesWritten / mFrameSize; 2481 *halFrames = framesWritten; 2482 2483 if (isSuspended()) { 2484 // return an estimation of rendered frames when the output is suspended 2485 size_t latencyFrames = (latency_l() * mSampleRate) / 1000; 2486 *dspFrames = (uint32_t) 2487 (framesWritten >= (int64_t)latencyFrames ? framesWritten - latencyFrames : 0); 2488 return NO_ERROR; 2489 } else { 2490 status_t status; 2491 uint32_t frames; 2492 status = mOutput->getRenderPosition(&frames); 2493 *dspFrames = (size_t)frames; 2494 return status; 2495 } 2496 } 2497 2498 // hasAudioSession_l() must be called with ThreadBase::mLock held 2499 uint32_t AudioFlinger::PlaybackThread::hasAudioSession_l(audio_session_t sessionId) const 2500 { 2501 uint32_t result = 0; 2502 if (getEffectChain_l(sessionId) != 0) { 2503 result = EFFECT_SESSION; 2504 } 2505 2506 for (size_t i = 0; i < mTracks.size(); ++i) { 2507 sp<Track> track = mTracks[i]; 2508 if (sessionId == track->sessionId() && !track->isInvalid()) { 2509 result |= TRACK_SESSION; 2510 if (track->isFastTrack()) { 2511 result |= FAST_SESSION; 2512 } 2513 break; 2514 } 2515 } 2516 2517 return result; 2518 } 2519 2520 uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(audio_session_t sessionId) 2521 { 2522 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2523 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2524 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2525 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2526 } 2527 for (size_t i = 0; i < mTracks.size(); i++) { 2528 sp<Track> track = mTracks[i]; 2529 if (sessionId == track->sessionId() && !track->isInvalid()) { 2530 return AudioSystem::getStrategyForStream(track->streamType()); 2531 } 2532 } 2533 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2534 } 2535 2536 2537 AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2538 { 2539 Mutex::Autolock _l(mLock); 2540 return mOutput; 2541 } 2542 2543 AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2544 { 2545 Mutex::Autolock _l(mLock); 2546 AudioStreamOut *output = mOutput; 2547 mOutput = NULL; 2548 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2549 // must push a NULL and wait for ack 2550 mOutputSink.clear(); 2551 mPipeSink.clear(); 2552 mNormalSink.clear(); 2553 return output; 2554 } 2555 2556 // this method must always be called either with ThreadBase mLock held or inside the thread loop 2557 sp<StreamHalInterface> AudioFlinger::PlaybackThread::stream() const 2558 { 2559 if (mOutput == NULL) { 2560 return NULL; 2561 } 2562 return mOutput->stream; 2563 } 2564 2565 uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2566 { 2567 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2568 } 2569 2570 status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2571 { 2572 if (!isValidSyncEvent(event)) { 2573 return BAD_VALUE; 2574 } 2575 2576 Mutex::Autolock _l(mLock); 2577 2578 for (size_t i = 0; i < mTracks.size(); ++i) { 2579 sp<Track> track = mTracks[i]; 2580 if (event->triggerSession() == track->sessionId()) { 2581 (void) track->setSyncEvent(event); 2582 return NO_ERROR; 2583 } 2584 } 2585 2586 return NAME_NOT_FOUND; 2587 } 2588 2589 bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) const 2590 { 2591 return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE; 2592 } 2593 2594 void AudioFlinger::PlaybackThread::threadLoop_removeTracks( 2595 const Vector< sp<Track> >& tracksToRemove) 2596 { 2597 size_t count = tracksToRemove.size(); 2598 if (count > 0) { 2599 for (size_t i = 0 ; i < count ; i++) { 2600 const sp<Track>& track = tracksToRemove.itemAt(i); 2601 if (track->isExternalTrack()) { 2602 AudioSystem::stopOutput(mId, track->streamType(), 2603 track->sessionId()); 2604 #ifdef ADD_BATTERY_DATA 2605 // to track the speaker usage 2606 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2607 #endif 2608 if (track->isTerminated()) { 2609 AudioSystem::releaseOutput(mId, track->streamType(), 2610 track->sessionId()); 2611 } 2612 } 2613 } 2614 } 2615 } 2616 2617 void AudioFlinger::PlaybackThread::checkSilentMode_l() 2618 { 2619 if (!mMasterMute) { 2620 char value[PROPERTY_VALUE_MAX]; 2621 if (mOutDevice == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) { 2622 ALOGD("ro.audio.silent will be ignored for threads on AUDIO_DEVICE_OUT_REMOTE_SUBMIX"); 2623 return; 2624 } 2625 if (property_get("ro.audio.silent", value, "0") > 0) { 2626 char *endptr; 2627 unsigned long ul = strtoul(value, &endptr, 0); 2628 if (*endptr == '\0' && ul != 0) { 2629 ALOGD("Silence is golden"); 2630 // The setprop command will not allow a property to be changed after 2631 // the first time it is set, so we don't have to worry about un-muting. 2632 setMasterMute_l(true); 2633 } 2634 } 2635 } 2636 } 2637 2638 // shared by MIXER and DIRECT, overridden by DUPLICATING 2639 ssize_t AudioFlinger::PlaybackThread::threadLoop_write() 2640 { 2641 mInWrite = true; 2642 ssize_t bytesWritten; 2643 const size_t offset = mCurrentWriteLength - mBytesRemaining; 2644 2645 // If an NBAIO sink is present, use it to write the normal mixer's submix 2646 if (mNormalSink != 0) { 2647 2648 const size_t count = mBytesRemaining / mFrameSize; 2649 2650 ATRACE_BEGIN("write"); 2651 // update the setpoint when AudioFlinger::mScreenState changes 2652 uint32_t screenState = AudioFlinger::mScreenState; 2653 if (screenState != mScreenState) { 2654 mScreenState = screenState; 2655 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2656 if (pipe != NULL) { 2657 pipe->setAvgFrames((mScreenState & 1) ? 2658 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2659 } 2660 } 2661 ssize_t framesWritten = mNormalSink->write((char *)mSinkBuffer + offset, count); 2662 ATRACE_END(); 2663 if (framesWritten > 0) { 2664 bytesWritten = framesWritten * mFrameSize; 2665 } else { 2666 bytesWritten = framesWritten; 2667 } 2668 // otherwise use the HAL / AudioStreamOut directly 2669 } else { 2670 // Direct output and offload threads 2671 2672 if (mUseAsyncWrite) { 2673 ALOGW_IF(mWriteAckSequence & 1, "threadLoop_write(): out of sequence write request"); 2674 mWriteAckSequence += 2; 2675 mWriteAckSequence |= 1; 2676 ALOG_ASSERT(mCallbackThread != 0); 2677 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2678 } 2679 // FIXME We should have an implementation of timestamps for direct output threads. 2680 // They are used e.g for multichannel PCM playback over HDMI. 2681 bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining); 2682 2683 if (mUseAsyncWrite && 2684 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) { 2685 // do not wait for async callback in case of error of full write 2686 mWriteAckSequence &= ~1; 2687 ALOG_ASSERT(mCallbackThread != 0); 2688 mCallbackThread->setWriteBlocked(mWriteAckSequence); 2689 } 2690 } 2691 2692 mNumWrites++; 2693 mInWrite = false; 2694 mStandby = false; 2695 return bytesWritten; 2696 } 2697 2698 void AudioFlinger::PlaybackThread::threadLoop_drain() 2699 { 2700 bool supportsDrain = false; 2701 if (mOutput->stream->supportsDrain(&supportsDrain) == OK && supportsDrain) { 2702 ALOGV("draining %s", (mMixerStatus == MIXER_DRAIN_TRACK) ? "early" : "full"); 2703 if (mUseAsyncWrite) { 2704 ALOGW_IF(mDrainSequence & 1, "threadLoop_drain(): out of sequence drain request"); 2705 mDrainSequence |= 1; 2706 ALOG_ASSERT(mCallbackThread != 0); 2707 mCallbackThread->setDraining(mDrainSequence); 2708 } 2709 status_t result = mOutput->stream->drain(mMixerStatus == MIXER_DRAIN_TRACK); 2710 ALOGE_IF(result != OK, "Error when draining stream: %d", result); 2711 } 2712 } 2713 2714 void AudioFlinger::PlaybackThread::threadLoop_exit() 2715 { 2716 { 2717 Mutex::Autolock _l(mLock); 2718 for (size_t i = 0; i < mTracks.size(); i++) { 2719 sp<Track> track = mTracks[i]; 2720 track->invalidate(); 2721 } 2722 // Clear ActiveTracks to update BatteryNotifier in case active tracks remain. 2723 // After we exit there are no more track changes sent to BatteryNotifier 2724 // because that requires an active threadLoop. 2725 // TODO: should we decActiveTrackCnt() of the cleared track effect chain? 2726 mActiveTracks.clear(); 2727 } 2728 } 2729 2730 /* 2731 The derived values that are cached: 2732 - mSinkBufferSize from frame count * frame size 2733 - mActiveSleepTimeUs from activeSleepTimeUs() 2734 - mIdleSleepTimeUs from idleSleepTimeUs() 2735 - mStandbyDelayNs from mActiveSleepTimeUs (DIRECT only) or forced to at least 2736 kDefaultStandbyTimeInNsecs when connected to an A2DP device. 2737 - maxPeriod from frame count and sample rate (MIXER only) 2738 2739 The parameters that affect these derived values are: 2740 - frame count 2741 - frame size 2742 - sample rate 2743 - device type: A2DP or not 2744 - device latency 2745 - format: PCM or not 2746 - active sleep time 2747 - idle sleep time 2748 */ 2749 2750 void AudioFlinger::PlaybackThread::cacheParameters_l() 2751 { 2752 mSinkBufferSize = mNormalFrameCount * mFrameSize; 2753 mActiveSleepTimeUs = activeSleepTimeUs(); 2754 mIdleSleepTimeUs = idleSleepTimeUs(); 2755 2756 // make sure standby delay is not too short when connected to an A2DP sink to avoid 2757 // truncating audio when going to standby. 2758 mStandbyDelayNs = AudioFlinger::mStandbyTimeInNsecs; 2759 if ((mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != 0) { 2760 if (mStandbyDelayNs < kDefaultStandbyTimeInNsecs) { 2761 mStandbyDelayNs = kDefaultStandbyTimeInNsecs; 2762 } 2763 } 2764 } 2765 2766 bool AudioFlinger::PlaybackThread::invalidateTracks_l(audio_stream_type_t streamType) 2767 { 2768 ALOGV("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %zu", 2769 this, streamType, mTracks.size()); 2770 bool trackMatch = false; 2771 size_t size = mTracks.size(); 2772 for (size_t i = 0; i < size; i++) { 2773 sp<Track> t = mTracks[i]; 2774 if (t->streamType() == streamType && t->isExternalTrack()) { 2775 t->invalidate(); 2776 trackMatch = true; 2777 } 2778 } 2779 return trackMatch; 2780 } 2781 2782 void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 2783 { 2784 Mutex::Autolock _l(mLock); 2785 invalidateTracks_l(streamType); 2786 } 2787 2788 status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 2789 { 2790 audio_session_t session = chain->sessionId(); 2791 sp<EffectBufferHalInterface> halInBuffer, halOutBuffer; 2792 status_t result = EffectBufferHalInterface::mirror( 2793 mEffectBufferEnabled ? mEffectBuffer : mSinkBuffer, 2794 mEffectBufferEnabled ? mEffectBufferSize : mSinkBufferSize, 2795 &halInBuffer); 2796 if (result != OK) return result; 2797 halOutBuffer = halInBuffer; 2798 int16_t *buffer = reinterpret_cast<int16_t*>(halInBuffer->externalData()); 2799 2800 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 2801 if (session > AUDIO_SESSION_OUTPUT_MIX) { 2802 // Only one effect chain can be present in direct output thread and it uses 2803 // the sink buffer as input 2804 if (mType != DIRECT) { 2805 size_t numSamples = mNormalFrameCount * mChannelCount; 2806 status_t result = EffectBufferHalInterface::allocate( 2807 numSamples * sizeof(int16_t), 2808 &halInBuffer); 2809 if (result != OK) return result; 2810 buffer = halInBuffer->audioBuffer()->s16; 2811 ALOGV("addEffectChain_l() creating new input buffer %p session %d", 2812 buffer, session); 2813 } 2814 2815 // Attach all tracks with same session ID to this chain. 2816 for (size_t i = 0; i < mTracks.size(); ++i) { 2817 sp<Track> track = mTracks[i]; 2818 if (session == track->sessionId()) { 2819 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), 2820 buffer); 2821 track->setMainBuffer(buffer); 2822 chain->incTrackCnt(); 2823 } 2824 } 2825 2826 // indicate all active tracks in the chain 2827 for (const sp<Track> &track : mActiveTracks) { 2828 if (session == track->sessionId()) { 2829 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 2830 chain->incActiveTrackCnt(); 2831 } 2832 } 2833 } 2834 chain->setThread(this); 2835 chain->setInBuffer(halInBuffer); 2836 chain->setOutBuffer(halOutBuffer); 2837 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 2838 // chains list in order to be processed last as it contains output stage effects. 2839 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 2840 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 2841 // after track specific effects and before output stage. 2842 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 2843 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX. 2844 // Effect chain for other sessions are inserted at beginning of effect 2845 // chains list to be processed before output mix effects. Relative order between other 2846 // sessions is not important. 2847 static_assert(AUDIO_SESSION_OUTPUT_MIX == 0 && 2848 AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX, 2849 "audio_session_t constants misdefined"); 2850 size_t size = mEffectChains.size(); 2851 size_t i = 0; 2852 for (i = 0; i < size; i++) { 2853 if (mEffectChains[i]->sessionId() < session) { 2854 break; 2855 } 2856 } 2857 mEffectChains.insertAt(chain, i); 2858 checkSuspendOnAddEffectChain_l(chain); 2859 2860 return NO_ERROR; 2861 } 2862 2863 size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 2864 { 2865 audio_session_t session = chain->sessionId(); 2866 2867 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 2868 2869 for (size_t i = 0; i < mEffectChains.size(); i++) { 2870 if (chain == mEffectChains[i]) { 2871 mEffectChains.removeAt(i); 2872 // detach all active tracks from the chain 2873 for (const sp<Track> &track : mActiveTracks) { 2874 if (session == track->sessionId()) { 2875 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 2876 chain.get(), session); 2877 chain->decActiveTrackCnt(); 2878 } 2879 } 2880 2881 // detach all tracks with same session ID from this chain 2882 for (size_t i = 0; i < mTracks.size(); ++i) { 2883 sp<Track> track = mTracks[i]; 2884 if (session == track->sessionId()) { 2885 track->setMainBuffer(reinterpret_cast<int16_t*>(mSinkBuffer)); 2886 chain->decTrackCnt(); 2887 } 2888 } 2889 break; 2890 } 2891 } 2892 return mEffectChains.size(); 2893 } 2894 2895 status_t AudioFlinger::PlaybackThread::attachAuxEffect( 2896 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId) 2897 { 2898 Mutex::Autolock _l(mLock); 2899 return attachAuxEffect_l(track, EffectId); 2900 } 2901 2902 status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 2903 const sp<AudioFlinger::PlaybackThread::Track>& track, int EffectId) 2904 { 2905 status_t status = NO_ERROR; 2906 2907 if (EffectId == 0) { 2908 track->setAuxBuffer(0, NULL); 2909 } else { 2910 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 2911 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 2912 if (effect != 0) { 2913 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2914 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 2915 } else { 2916 status = INVALID_OPERATION; 2917 } 2918 } else { 2919 status = BAD_VALUE; 2920 } 2921 } 2922 return status; 2923 } 2924 2925 void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 2926 { 2927 for (size_t i = 0; i < mTracks.size(); ++i) { 2928 sp<Track> track = mTracks[i]; 2929 if (track->auxEffectId() == effectId) { 2930 attachAuxEffect_l(track, 0); 2931 } 2932 } 2933 } 2934 2935 bool AudioFlinger::PlaybackThread::threadLoop() 2936 { 2937 tlNBLogWriter = mNBLogWriter.get(); 2938 2939 Vector< sp<Track> > tracksToRemove; 2940 2941 mStandbyTimeNs = systemTime(); 2942 nsecs_t lastWriteFinished = -1; // time last server write completed 2943 int64_t lastFramesWritten = -1; // track changes in timestamp server frames written 2944 2945 // MIXER 2946 nsecs_t lastWarning = 0; 2947 2948 // DUPLICATING 2949 // FIXME could this be made local to while loop? 2950 writeFrames = 0; 2951 2952 cacheParameters_l(); 2953 mSleepTimeUs = mIdleSleepTimeUs; 2954 2955 if (mType == MIXER) { 2956 sleepTimeShift = 0; 2957 } 2958 2959 CpuStats cpuStats; 2960 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2961 2962 acquireWakeLock(); 2963 2964 // mNBLogWriter logging APIs can only be called by a single thread, typically the 2965 // thread associated with this PlaybackThread. 2966 // If you want to share the mNBLogWriter with other threads (for example, binder threads) 2967 // then all such threads must agree to hold a common mutex before logging. 2968 // So if you need to log when mutex is unlocked, set logString to a non-NULL string, 2969 // and then that string will be logged at the next convenient opportunity. 2970 // See reference to logString below. 2971 const char *logString = NULL; 2972 2973 // Estimated time for next buffer to be written to hal. This is used only on 2974 // suspended mode (for now) to help schedule the wait time until next iteration. 2975 nsecs_t timeLoopNextNs = 0; 2976 2977 checkSilentMode_l(); 2978 2979 while (!exitPending()) 2980 { 2981 // Log merge requests are performed during AudioFlinger binder transactions, but 2982 // that does not cover audio playback. It's requested here for that reason. 2983 mAudioFlinger->requestLogMerge(); 2984 2985 cpuStats.sample(myName); 2986 2987 Vector< sp<EffectChain> > effectChains; 2988 2989 { // scope for mLock 2990 2991 Mutex::Autolock _l(mLock); 2992 2993 processConfigEvents_l(); 2994 2995 // See comment at declaration of logString for why this is done under mLock 2996 if (logString != NULL) { 2997 mNBLogWriter->logTimestamp(); 2998 mNBLogWriter->log(logString); 2999 logString = NULL; 3000 } 3001 3002 // Gather the framesReleased counters for all active tracks, 3003 // and associate with the sink frames written out. We need 3004 // this to convert the sink timestamp to the track timestamp. 3005 bool kernelLocationUpdate = false; 3006 if (mNormalSink != 0) { 3007 // Note: The DuplicatingThread may not have a mNormalSink. 3008 // We always fetch the timestamp here because often the downstream 3009 // sink will block while writing. 3010 ExtendedTimestamp timestamp; // use private copy to fetch 3011 (void) mNormalSink->getTimestamp(timestamp); 3012 3013 // We keep track of the last valid kernel position in case we are in underrun 3014 // and the normal mixer period is the same as the fast mixer period, or there 3015 // is some error from the HAL. 3016 if (mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3017 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3018 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL]; 3019 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL_LASTKERNELOK] = 3020 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3021 3022 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3023 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER]; 3024 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER_LASTKERNELOK] = 3025 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER]; 3026 } 3027 3028 if (timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] >= 0) { 3029 kernelLocationUpdate = true; 3030 } else { 3031 ALOGVV("getTimestamp error - no valid kernel position"); 3032 } 3033 3034 // copy over kernel info 3035 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 3036 timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] 3037 + mSuspendedFrames; // add frames discarded when suspended 3038 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = 3039 timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]; 3040 } 3041 // mFramesWritten for non-offloaded tracks are contiguous 3042 // even after standby() is called. This is useful for the track frame 3043 // to sink frame mapping. 3044 bool serverLocationUpdate = false; 3045 if (mFramesWritten != lastFramesWritten) { 3046 serverLocationUpdate = true; 3047 lastFramesWritten = mFramesWritten; 3048 } 3049 // Only update timestamps if there is a meaningful change. 3050 // Either the kernel timestamp must be valid or we have written something. 3051 if (kernelLocationUpdate || serverLocationUpdate) { 3052 if (serverLocationUpdate) { 3053 // use the time before we called the HAL write - it is a bit more accurate 3054 // to when the server last read data than the current time here. 3055 // 3056 // If we haven't written anything, mLastWriteTime will be -1 3057 // and we use systemTime(). 3058 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] = mFramesWritten; 3059 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = mLastWriteTime == -1 3060 ? systemTime() : mLastWriteTime; 3061 } 3062 3063 for (const sp<Track> &t : mActiveTracks) { 3064 if (!t->isFastTrack()) { 3065 t->updateTrackFrameInfo( 3066 t->mAudioTrackServerProxy->framesReleased(), 3067 mFramesWritten, 3068 mTimestamp); 3069 } 3070 } 3071 } 3072 #if 0 3073 // logFormat example 3074 if (z % 100 == 0) { 3075 timespec ts; 3076 clock_gettime(CLOCK_MONOTONIC, &ts); 3077 LOGT("This is an integer %d, this is a float %f, this is my " 3078 "pid %p %% %s %t", 42, 3.14, "and this is a timestamp", ts); 3079 LOGT("A deceptive null-terminated string %\0"); 3080 } 3081 ++z; 3082 #endif 3083 saveOutputTracks(); 3084 if (mSignalPending) { 3085 // A signal was raised while we were unlocked 3086 mSignalPending = false; 3087 } else if (waitingAsyncCallback_l()) { 3088 if (exitPending()) { 3089 break; 3090 } 3091 bool released = false; 3092 if (!keepWakeLock()) { 3093 releaseWakeLock_l(); 3094 released = true; 3095 } 3096 3097 const int64_t waitNs = computeWaitTimeNs_l(); 3098 ALOGV("wait async completion (wait time: %lld)", (long long)waitNs); 3099 status_t status = mWaitWorkCV.waitRelative(mLock, waitNs); 3100 if (status == TIMED_OUT) { 3101 mSignalPending = true; // if timeout recheck everything 3102 } 3103 ALOGV("async completion/wake"); 3104 if (released) { 3105 acquireWakeLock_l(); 3106 } 3107 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3108 mSleepTimeUs = 0; 3109 3110 continue; 3111 } 3112 if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) || 3113 isSuspended()) { 3114 // put audio hardware into standby after short delay 3115 if (shouldStandby_l()) { 3116 3117 threadLoop_standby(); 3118 3119 mStandby = true; 3120 } 3121 3122 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 3123 // we're about to wait, flush the binder command buffer 3124 IPCThreadState::self()->flushCommands(); 3125 3126 clearOutputTracks(); 3127 3128 if (exitPending()) { 3129 break; 3130 } 3131 3132 releaseWakeLock_l(); 3133 // wait until we have something to do... 3134 ALOGV("%s going to sleep", myName.string()); 3135 mWaitWorkCV.wait(mLock); 3136 ALOGV("%s waking up", myName.string()); 3137 acquireWakeLock_l(); 3138 3139 mMixerStatus = MIXER_IDLE; 3140 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 3141 mBytesWritten = 0; 3142 mBytesRemaining = 0; 3143 checkSilentMode_l(); 3144 3145 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3146 mSleepTimeUs = mIdleSleepTimeUs; 3147 if (mType == MIXER) { 3148 sleepTimeShift = 0; 3149 } 3150 3151 continue; 3152 } 3153 } 3154 // mMixerStatusIgnoringFastTracks is also updated internally 3155 mMixerStatus = prepareTracks_l(&tracksToRemove); 3156 3157 mActiveTracks.updatePowerState(this); 3158 3159 // prevent any changes in effect chain list and in each effect chain 3160 // during mixing and effect process as the audio buffers could be deleted 3161 // or modified if an effect is created or deleted 3162 lockEffectChains_l(effectChains); 3163 } // mLock scope ends 3164 3165 if (mBytesRemaining == 0) { 3166 mCurrentWriteLength = 0; 3167 if (mMixerStatus == MIXER_TRACKS_READY) { 3168 // threadLoop_mix() sets mCurrentWriteLength 3169 threadLoop_mix(); 3170 } else if ((mMixerStatus != MIXER_DRAIN_TRACK) 3171 && (mMixerStatus != MIXER_DRAIN_ALL)) { 3172 // threadLoop_sleepTime sets mSleepTimeUs to 0 if data 3173 // must be written to HAL 3174 threadLoop_sleepTime(); 3175 if (mSleepTimeUs == 0) { 3176 mCurrentWriteLength = mSinkBufferSize; 3177 } 3178 } 3179 // Either threadLoop_mix() or threadLoop_sleepTime() should have set 3180 // mMixerBuffer with data if mMixerBufferValid is true and mSleepTimeUs == 0. 3181 // Merge mMixerBuffer data into mEffectBuffer (if any effects are valid) 3182 // or mSinkBuffer (if there are no effects). 3183 // 3184 // This is done pre-effects computation; if effects change to 3185 // support higher precision, this needs to move. 3186 // 3187 // mMixerBufferValid is only set true by MixerThread::prepareTracks_l(). 3188 // TODO use mSleepTimeUs == 0 as an additional condition. 3189 if (mMixerBufferValid) { 3190 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer; 3191 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat; 3192 3193 // mono blend occurs for mixer threads only (not direct or offloaded) 3194 // and is handled here if we're going directly to the sink. 3195 if (requireMonoBlend() && !mEffectBufferValid) { 3196 mono_blend(mMixerBuffer, mMixerBufferFormat, mChannelCount, mNormalFrameCount, 3197 true /*limit*/); 3198 } 3199 3200 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat, 3201 mNormalFrameCount * mChannelCount); 3202 } 3203 3204 mBytesRemaining = mCurrentWriteLength; 3205 if (isSuspended()) { 3206 // Simulate write to HAL when suspended (e.g. BT SCO phone call). 3207 mSleepTimeUs = suspendSleepTimeUs(); // assumes full buffer. 3208 const size_t framesRemaining = mBytesRemaining / mFrameSize; 3209 mBytesWritten += mBytesRemaining; 3210 mFramesWritten += framesRemaining; 3211 mSuspendedFrames += framesRemaining; // to adjust kernel HAL position 3212 mBytesRemaining = 0; 3213 } 3214 3215 // only process effects if we're going to write 3216 if (mSleepTimeUs == 0 && mType != OFFLOAD) { 3217 for (size_t i = 0; i < effectChains.size(); i ++) { 3218 effectChains[i]->process_l(); 3219 } 3220 } 3221 } 3222 // Process effect chains for offloaded thread even if no audio 3223 // was read from audio track: process only updates effect state 3224 // and thus does have to be synchronized with audio writes but may have 3225 // to be called while waiting for async write callback 3226 if (mType == OFFLOAD) { 3227 for (size_t i = 0; i < effectChains.size(); i ++) { 3228 effectChains[i]->process_l(); 3229 } 3230 } 3231 3232 // Only if the Effects buffer is enabled and there is data in the 3233 // Effects buffer (buffer valid), we need to 3234 // copy into the sink buffer. 3235 // TODO use mSleepTimeUs == 0 as an additional condition. 3236 if (mEffectBufferValid) { 3237 //ALOGV("writing effect buffer to sink buffer format %#x", mFormat); 3238 3239 if (requireMonoBlend()) { 3240 mono_blend(mEffectBuffer, mEffectBufferFormat, mChannelCount, mNormalFrameCount, 3241 true /*limit*/); 3242 } 3243 3244 memcpy_by_audio_format(mSinkBuffer, mFormat, mEffectBuffer, mEffectBufferFormat, 3245 mNormalFrameCount * mChannelCount); 3246 } 3247 3248 // enable changes in effect chain 3249 unlockEffectChains(effectChains); 3250 3251 if (!waitingAsyncCallback()) { 3252 // mSleepTimeUs == 0 means we must write to audio hardware 3253 if (mSleepTimeUs == 0) { 3254 ssize_t ret = 0; 3255 // We save lastWriteFinished here, as previousLastWriteFinished, 3256 // for throttling. On thread start, previousLastWriteFinished will be 3257 // set to -1, which properly results in no throttling after the first write. 3258 nsecs_t previousLastWriteFinished = lastWriteFinished; 3259 nsecs_t delta = 0; 3260 if (mBytesRemaining) { 3261 // FIXME rewrite to reduce number of system calls 3262 mLastWriteTime = systemTime(); // also used for dumpsys 3263 ret = threadLoop_write(); 3264 lastWriteFinished = systemTime(); 3265 delta = lastWriteFinished - mLastWriteTime; 3266 if (ret < 0) { 3267 mBytesRemaining = 0; 3268 } else { 3269 mBytesWritten += ret; 3270 mBytesRemaining -= ret; 3271 mFramesWritten += ret / mFrameSize; 3272 } 3273 } else if ((mMixerStatus == MIXER_DRAIN_TRACK) || 3274 (mMixerStatus == MIXER_DRAIN_ALL)) { 3275 threadLoop_drain(); 3276 } 3277 if (mType == MIXER && !mStandby) { 3278 // write blocked detection 3279 if (delta > maxPeriod) { 3280 mNumDelayedWrites++; 3281 if ((lastWriteFinished - lastWarning) > kWarningThrottleNs) { 3282 ATRACE_NAME("underrun"); 3283 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 3284 (unsigned long long) ns2ms(delta), mNumDelayedWrites, this); 3285 lastWarning = lastWriteFinished; 3286 } 3287 } 3288 3289 if (mThreadThrottle 3290 && mMixerStatus == MIXER_TRACKS_READY // we are mixing (active tracks) 3291 && ret > 0) { // we wrote something 3292 // Limit MixerThread data processing to no more than twice the 3293 // expected processing rate. 3294 // 3295 // This helps prevent underruns with NuPlayer and other applications 3296 // which may set up buffers that are close to the minimum size, or use 3297 // deep buffers, and rely on a double-buffering sleep strategy to fill. 3298 // 3299 // The throttle smooths out sudden large data drains from the device, 3300 // e.g. when it comes out of standby, which often causes problems with 3301 // (1) mixer threads without a fast mixer (which has its own warm-up) 3302 // (2) minimum buffer sized tracks (even if the track is full, 3303 // the app won't fill fast enough to handle the sudden draw). 3304 // 3305 // Total time spent in last processing cycle equals time spent in 3306 // 1. threadLoop_write, as well as time spent in 3307 // 2. threadLoop_mix (significant for heavy mixing, especially 3308 // on low tier processors) 3309 3310 // it's OK if deltaMs is an overestimate. 3311 const int32_t deltaMs = 3312 (lastWriteFinished - previousLastWriteFinished) / 1000000; 3313 const int32_t throttleMs = mHalfBufferMs - deltaMs; 3314 if ((signed)mHalfBufferMs >= throttleMs && throttleMs > 0) { 3315 usleep(throttleMs * 1000); 3316 // notify of throttle start on verbose log 3317 ALOGV_IF(mThreadThrottleEndMs == mThreadThrottleTimeMs, 3318 "mixer(%p) throttle begin:" 3319 " ret(%zd) deltaMs(%d) requires sleep %d ms", 3320 this, ret, deltaMs, throttleMs); 3321 mThreadThrottleTimeMs += throttleMs; 3322 // Throttle must be attributed to the previous mixer loop's write time 3323 // to allow back-to-back throttling. 3324 lastWriteFinished += throttleMs * 1000000; 3325 } else { 3326 uint32_t diff = mThreadThrottleTimeMs - mThreadThrottleEndMs; 3327 if (diff > 0) { 3328 // notify of throttle end on debug log 3329 // but prevent spamming for bluetooth 3330 ALOGD_IF(!audio_is_a2dp_out_device(outDevice()), 3331 "mixer(%p) throttle end: throttle time(%u)", this, diff); 3332 mThreadThrottleEndMs = mThreadThrottleTimeMs; 3333 } 3334 } 3335 } 3336 } 3337 3338 } else { 3339 ATRACE_BEGIN("sleep"); 3340 Mutex::Autolock _l(mLock); 3341 // suspended requires accurate metering of sleep time. 3342 if (isSuspended()) { 3343 // advance by expected sleepTime 3344 timeLoopNextNs += microseconds((nsecs_t)mSleepTimeUs); 3345 const nsecs_t nowNs = systemTime(); 3346 3347 // compute expected next time vs current time. 3348 // (negative deltas are treated as delays). 3349 nsecs_t deltaNs = timeLoopNextNs - nowNs; 3350 if (deltaNs < -kMaxNextBufferDelayNs) { 3351 // Delays longer than the max allowed trigger a reset. 3352 ALOGV("DelayNs: %lld, resetting timeLoopNextNs", (long long) deltaNs); 3353 deltaNs = microseconds((nsecs_t)mSleepTimeUs); 3354 timeLoopNextNs = nowNs + deltaNs; 3355 } else if (deltaNs < 0) { 3356 // Delays within the max delay allowed: zero the delta/sleepTime 3357 // to help the system catch up in the next iteration(s) 3358 ALOGV("DelayNs: %lld, catching-up", (long long) deltaNs); 3359 deltaNs = 0; 3360 } 3361 // update sleep time (which is >= 0) 3362 mSleepTimeUs = deltaNs / 1000; 3363 } 3364 if (!mSignalPending && mConfigEvents.isEmpty() && !exitPending()) { 3365 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)mSleepTimeUs)); 3366 } 3367 ATRACE_END(); 3368 } 3369 } 3370 3371 // Finally let go of removed track(s), without the lock held 3372 // since we can't guarantee the destructors won't acquire that 3373 // same lock. This will also mutate and push a new fast mixer state. 3374 threadLoop_removeTracks(tracksToRemove); 3375 tracksToRemove.clear(); 3376 3377 // FIXME I don't understand the need for this here; 3378 // it was in the original code but maybe the 3379 // assignment in saveOutputTracks() makes this unnecessary? 3380 clearOutputTracks(); 3381 3382 // Effect chains will be actually deleted here if they were removed from 3383 // mEffectChains list during mixing or effects processing 3384 effectChains.clear(); 3385 3386 // FIXME Note that the above .clear() is no longer necessary since effectChains 3387 // is now local to this block, but will keep it for now (at least until merge done). 3388 } 3389 3390 threadLoop_exit(); 3391 3392 if (!mStandby) { 3393 threadLoop_standby(); 3394 mStandby = true; 3395 } 3396 3397 releaseWakeLock(); 3398 3399 ALOGV("Thread %p type %d exiting", this, mType); 3400 return false; 3401 } 3402 3403 // removeTracks_l() must be called with ThreadBase::mLock held 3404 void AudioFlinger::PlaybackThread::removeTracks_l(const Vector< sp<Track> >& tracksToRemove) 3405 { 3406 size_t count = tracksToRemove.size(); 3407 if (count > 0) { 3408 for (size_t i=0 ; i<count ; i++) { 3409 const sp<Track>& track = tracksToRemove.itemAt(i); 3410 mActiveTracks.remove(track); 3411 ALOGV("removeTracks_l removing track on session %d", track->sessionId()); 3412 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 3413 if (chain != 0) { 3414 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), 3415 track->sessionId()); 3416 chain->decActiveTrackCnt(); 3417 } 3418 if (track->isTerminated()) { 3419 removeTrack_l(track); 3420 } 3421 } 3422 } 3423 3424 } 3425 3426 status_t AudioFlinger::PlaybackThread::getTimestamp_l(AudioTimestamp& timestamp) 3427 { 3428 if (mNormalSink != 0) { 3429 ExtendedTimestamp ets; 3430 status_t status = mNormalSink->getTimestamp(ets); 3431 if (status == NO_ERROR) { 3432 status = ets.getBestTimestamp(×tamp); 3433 } 3434 return status; 3435 } 3436 if ((mType == OFFLOAD || mType == DIRECT) && mOutput != NULL) { 3437 uint64_t position64; 3438 if (mOutput->getPresentationPosition(&position64, ×tamp.mTime) == OK) { 3439 timestamp.mPosition = (uint32_t)position64; 3440 return NO_ERROR; 3441 } 3442 } 3443 return INVALID_OPERATION; 3444 } 3445 3446 status_t AudioFlinger::MixerThread::createAudioPatch_l(const struct audio_patch *patch, 3447 audio_patch_handle_t *handle) 3448 { 3449 status_t status; 3450 if (property_get_bool("af.patch_park", false /* default_value */)) { 3451 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3452 // or if HAL does not properly lock against access. 3453 AutoPark<FastMixer> park(mFastMixer); 3454 status = PlaybackThread::createAudioPatch_l(patch, handle); 3455 } else { 3456 status = PlaybackThread::createAudioPatch_l(patch, handle); 3457 } 3458 return status; 3459 } 3460 3461 status_t AudioFlinger::PlaybackThread::createAudioPatch_l(const struct audio_patch *patch, 3462 audio_patch_handle_t *handle) 3463 { 3464 status_t status = NO_ERROR; 3465 3466 // store new device and send to effects 3467 audio_devices_t type = AUDIO_DEVICE_NONE; 3468 for (unsigned int i = 0; i < patch->num_sinks; i++) { 3469 type |= patch->sinks[i].ext.device.type; 3470 } 3471 3472 #ifdef ADD_BATTERY_DATA 3473 // when changing the audio output device, call addBatteryData to notify 3474 // the change 3475 if (mOutDevice != type) { 3476 uint32_t params = 0; 3477 // check whether speaker is on 3478 if (type & AUDIO_DEVICE_OUT_SPEAKER) { 3479 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3480 } 3481 3482 audio_devices_t deviceWithoutSpeaker 3483 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3484 // check if any other device (except speaker) is on 3485 if (type & deviceWithoutSpeaker) { 3486 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3487 } 3488 3489 if (params != 0) { 3490 addBatteryData(params); 3491 } 3492 } 3493 #endif 3494 3495 for (size_t i = 0; i < mEffectChains.size(); i++) { 3496 mEffectChains[i]->setDevice_l(type); 3497 } 3498 3499 // mPrevOutDevice is the latest device set by createAudioPatch_l(). It is not set when 3500 // the thread is created so that the first patch creation triggers an ioConfigChanged callback 3501 bool configChanged = mPrevOutDevice != type; 3502 mOutDevice = type; 3503 mPatch = *patch; 3504 3505 if (mOutput->audioHwDev->supportsAudioPatches()) { 3506 sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice(); 3507 status = hwDevice->createAudioPatch(patch->num_sources, 3508 patch->sources, 3509 patch->num_sinks, 3510 patch->sinks, 3511 handle); 3512 } else { 3513 char *address; 3514 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 3515 //FIXME: we only support address on first sink with HAL version < 3.0 3516 address = audio_device_address_to_parameter( 3517 patch->sinks[0].ext.device.type, 3518 patch->sinks[0].ext.device.address); 3519 } else { 3520 address = (char *)calloc(1, 1); 3521 } 3522 AudioParameter param = AudioParameter(String8(address)); 3523 free(address); 3524 param.addInt(String8(AudioParameter::keyRouting), (int)type); 3525 status = mOutput->stream->setParameters(param.toString()); 3526 *handle = AUDIO_PATCH_HANDLE_NONE; 3527 } 3528 if (configChanged) { 3529 mPrevOutDevice = type; 3530 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 3531 } 3532 return status; 3533 } 3534 3535 status_t AudioFlinger::MixerThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3536 { 3537 status_t status; 3538 if (property_get_bool("af.patch_park", false /* default_value */)) { 3539 // Park FastMixer to avoid potential DOS issues with writing to the HAL 3540 // or if HAL does not properly lock against access. 3541 AutoPark<FastMixer> park(mFastMixer); 3542 status = PlaybackThread::releaseAudioPatch_l(handle); 3543 } else { 3544 status = PlaybackThread::releaseAudioPatch_l(handle); 3545 } 3546 return status; 3547 } 3548 3549 status_t AudioFlinger::PlaybackThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 3550 { 3551 status_t status = NO_ERROR; 3552 3553 mOutDevice = AUDIO_DEVICE_NONE; 3554 3555 if (mOutput->audioHwDev->supportsAudioPatches()) { 3556 sp<DeviceHalInterface> hwDevice = mOutput->audioHwDev->hwDevice(); 3557 status = hwDevice->releaseAudioPatch(handle); 3558 } else { 3559 AudioParameter param; 3560 param.addInt(String8(AudioParameter::keyRouting), 0); 3561 status = mOutput->stream->setParameters(param.toString()); 3562 } 3563 return status; 3564 } 3565 3566 void AudioFlinger::PlaybackThread::addPatchTrack(const sp<PatchTrack>& track) 3567 { 3568 Mutex::Autolock _l(mLock); 3569 mTracks.add(track); 3570 } 3571 3572 void AudioFlinger::PlaybackThread::deletePatchTrack(const sp<PatchTrack>& track) 3573 { 3574 Mutex::Autolock _l(mLock); 3575 destroyTrack_l(track); 3576 } 3577 3578 void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config) 3579 { 3580 ThreadBase::getAudioPortConfig(config); 3581 config->role = AUDIO_PORT_ROLE_SOURCE; 3582 config->ext.mix.hw_module = mOutput->audioHwDev->handle(); 3583 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 3584 } 3585 3586 // ---------------------------------------------------------------------------- 3587 3588 AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 3589 audio_io_handle_t id, audio_devices_t device, bool systemReady, type_t type) 3590 : PlaybackThread(audioFlinger, output, id, device, type, systemReady), 3591 // mAudioMixer below 3592 // mFastMixer below 3593 mFastMixerFutex(0), 3594 mMasterMono(false) 3595 // mOutputSink below 3596 // mPipeSink below 3597 // mNormalSink below 3598 { 3599 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 3600 ALOGV("mSampleRate=%u, mChannelMask=%#x, mChannelCount=%u, mFormat=%d, mFrameSize=%zu, " 3601 "mFrameCount=%zu, mNormalFrameCount=%zu", 3602 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 3603 mNormalFrameCount); 3604 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3605 3606 if (type == DUPLICATING) { 3607 // The Duplicating thread uses the AudioMixer and delivers data to OutputTracks 3608 // (downstream MixerThreads) in DuplicatingThread::threadLoop_write(). 3609 // Do not create or use mFastMixer, mOutputSink, mPipeSink, or mNormalSink. 3610 return; 3611 } 3612 // create an NBAIO sink for the HAL output stream, and negotiate 3613 mOutputSink = new AudioStreamOutSink(output->stream); 3614 size_t numCounterOffers = 0; 3615 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 3616 #if !LOG_NDEBUG 3617 ssize_t index = 3618 #else 3619 (void) 3620 #endif 3621 mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 3622 ALOG_ASSERT(index == 0); 3623 3624 // initialize fast mixer depending on configuration 3625 bool initFastMixer; 3626 switch (kUseFastMixer) { 3627 case FastMixer_Never: 3628 initFastMixer = false; 3629 break; 3630 case FastMixer_Always: 3631 initFastMixer = true; 3632 break; 3633 case FastMixer_Static: 3634 case FastMixer_Dynamic: 3635 // FastMixer was designed to operate with a HAL that pulls at a regular rate, 3636 // where the period is less than an experimentally determined threshold that can be 3637 // scheduled reliably with CFS. However, the BT A2DP HAL is 3638 // bursty (does not pull at a regular rate) and so cannot operate with FastMixer. 3639 initFastMixer = mFrameCount < mNormalFrameCount 3640 && (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) == 0; 3641 break; 3642 } 3643 ALOGW_IF(initFastMixer == false && mFrameCount < mNormalFrameCount, 3644 "FastMixer is preferred for this sink as frameCount %zu is less than threshold %zu", 3645 mFrameCount, mNormalFrameCount); 3646 if (initFastMixer) { 3647 audio_format_t fastMixerFormat; 3648 if (mMixerBufferEnabled && mEffectBufferEnabled) { 3649 fastMixerFormat = AUDIO_FORMAT_PCM_FLOAT; 3650 } else { 3651 fastMixerFormat = AUDIO_FORMAT_PCM_16_BIT; 3652 } 3653 if (mFormat != fastMixerFormat) { 3654 // change our Sink format to accept our intermediate precision 3655 mFormat = fastMixerFormat; 3656 free(mSinkBuffer); 3657 mFrameSize = mChannelCount * audio_bytes_per_sample(mFormat); 3658 const size_t sinkBufferSize = mNormalFrameCount * mFrameSize; 3659 (void)posix_memalign(&mSinkBuffer, 32, sinkBufferSize); 3660 } 3661 3662 // create a MonoPipe to connect our submix to FastMixer 3663 NBAIO_Format format = mOutputSink->format(); 3664 #ifdef TEE_SINK 3665 NBAIO_Format origformat = format; 3666 #endif 3667 // adjust format to match that of the Fast Mixer 3668 ALOGV("format changed from %d to %d", format.mFormat, fastMixerFormat); 3669 format.mFormat = fastMixerFormat; 3670 format.mFrameSize = audio_bytes_per_sample(format.mFormat) * format.mChannelCount; 3671 3672 // This pipe depth compensates for scheduling latency of the normal mixer thread. 3673 // When it wakes up after a maximum latency, it runs a few cycles quickly before 3674 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 3675 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 3676 const NBAIO_Format offers[1] = {format}; 3677 size_t numCounterOffers = 0; 3678 #if !LOG_NDEBUG || defined(TEE_SINK) 3679 ssize_t index = 3680 #else 3681 (void) 3682 #endif 3683 monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 3684 ALOG_ASSERT(index == 0); 3685 monoPipe->setAvgFrames((mScreenState & 1) ? 3686 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 3687 mPipeSink = monoPipe; 3688 3689 #ifdef TEE_SINK 3690 if (mTeeSinkOutputEnabled) { 3691 // create a Pipe to archive a copy of FastMixer's output for dumpsys 3692 Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat); 3693 const NBAIO_Format offers2[1] = {origformat}; 3694 numCounterOffers = 0; 3695 index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers); 3696 ALOG_ASSERT(index == 0); 3697 mTeeSink = teeSink; 3698 PipeReader *teeSource = new PipeReader(*teeSink); 3699 numCounterOffers = 0; 3700 index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers); 3701 ALOG_ASSERT(index == 0); 3702 mTeeSource = teeSource; 3703 } 3704 #endif 3705 3706 // create fast mixer and configure it initially with just one fast track for our submix 3707 mFastMixer = new FastMixer(); 3708 FastMixerStateQueue *sq = mFastMixer->sq(); 3709 #ifdef STATE_QUEUE_DUMP 3710 sq->setObserverDump(&mStateQueueObserverDump); 3711 sq->setMutatorDump(&mStateQueueMutatorDump); 3712 #endif 3713 FastMixerState *state = sq->begin(); 3714 FastTrack *fastTrack = &state->mFastTracks[0]; 3715 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 3716 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 3717 fastTrack->mVolumeProvider = NULL; 3718 fastTrack->mChannelMask = mChannelMask; // mPipeSink channel mask for audio to FastMixer 3719 fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer 3720 fastTrack->mGeneration++; 3721 state->mFastTracksGen++; 3722 state->mTrackMask = 1; 3723 // fast mixer will use the HAL output sink 3724 state->mOutputSink = mOutputSink.get(); 3725 state->mOutputSinkGen++; 3726 state->mFrameCount = mFrameCount; 3727 state->mCommand = FastMixerState::COLD_IDLE; 3728 // already done in constructor initialization list 3729 //mFastMixerFutex = 0; 3730 state->mColdFutexAddr = &mFastMixerFutex; 3731 state->mColdGen++; 3732 state->mDumpState = &mFastMixerDumpState; 3733 #ifdef TEE_SINK 3734 state->mTeeSink = mTeeSink.get(); 3735 #endif 3736 mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer"); 3737 state->mNBLogWriter = mFastMixerNBLogWriter.get(); 3738 sq->end(); 3739 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3740 3741 // start the fast mixer 3742 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 3743 pid_t tid = mFastMixer->getTid(); 3744 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer, false /*forApp*/); 3745 stream()->setHalThreadPriority(kPriorityFastMixer); 3746 3747 #ifdef AUDIO_WATCHDOG 3748 // create and start the watchdog 3749 mAudioWatchdog = new AudioWatchdog(); 3750 mAudioWatchdog->setDump(&mAudioWatchdogDump); 3751 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 3752 tid = mAudioWatchdog->getTid(); 3753 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer, false /*forApp*/); 3754 #endif 3755 3756 } 3757 3758 switch (kUseFastMixer) { 3759 case FastMixer_Never: 3760 case FastMixer_Dynamic: 3761 mNormalSink = mOutputSink; 3762 break; 3763 case FastMixer_Always: 3764 mNormalSink = mPipeSink; 3765 break; 3766 case FastMixer_Static: 3767 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 3768 break; 3769 } 3770 } 3771 3772 AudioFlinger::MixerThread::~MixerThread() 3773 { 3774 if (mFastMixer != 0) { 3775 FastMixerStateQueue *sq = mFastMixer->sq(); 3776 FastMixerState *state = sq->begin(); 3777 if (state->mCommand == FastMixerState::COLD_IDLE) { 3778 int32_t old = android_atomic_inc(&mFastMixerFutex); 3779 if (old == -1) { 3780 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3781 } 3782 } 3783 state->mCommand = FastMixerState::EXIT; 3784 sq->end(); 3785 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3786 mFastMixer->join(); 3787 // Though the fast mixer thread has exited, it's state queue is still valid. 3788 // We'll use that extract the final state which contains one remaining fast track 3789 // corresponding to our sub-mix. 3790 state = sq->begin(); 3791 ALOG_ASSERT(state->mTrackMask == 1); 3792 FastTrack *fastTrack = &state->mFastTracks[0]; 3793 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 3794 delete fastTrack->mBufferProvider; 3795 sq->end(false /*didModify*/); 3796 mFastMixer.clear(); 3797 #ifdef AUDIO_WATCHDOG 3798 if (mAudioWatchdog != 0) { 3799 mAudioWatchdog->requestExit(); 3800 mAudioWatchdog->requestExitAndWait(); 3801 mAudioWatchdog.clear(); 3802 } 3803 #endif 3804 } 3805 mAudioFlinger->unregisterWriter(mFastMixerNBLogWriter); 3806 delete mAudioMixer; 3807 } 3808 3809 3810 uint32_t AudioFlinger::MixerThread::correctLatency_l(uint32_t latency) const 3811 { 3812 if (mFastMixer != 0) { 3813 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 3814 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 3815 } 3816 return latency; 3817 } 3818 3819 3820 void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 3821 { 3822 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 3823 } 3824 3825 ssize_t AudioFlinger::MixerThread::threadLoop_write() 3826 { 3827 // FIXME we should only do one push per cycle; confirm this is true 3828 // Start the fast mixer if it's not already running 3829 if (mFastMixer != 0) { 3830 FastMixerStateQueue *sq = mFastMixer->sq(); 3831 FastMixerState *state = sq->begin(); 3832 if (state->mCommand != FastMixerState::MIX_WRITE && 3833 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 3834 if (state->mCommand == FastMixerState::COLD_IDLE) { 3835 3836 // FIXME workaround for first HAL write being CPU bound on some devices 3837 ATRACE_BEGIN("write"); 3838 mOutput->write((char *)mSinkBuffer, 0); 3839 ATRACE_END(); 3840 3841 int32_t old = android_atomic_inc(&mFastMixerFutex); 3842 if (old == -1) { 3843 (void) syscall(__NR_futex, &mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 3844 } 3845 #ifdef AUDIO_WATCHDOG 3846 if (mAudioWatchdog != 0) { 3847 mAudioWatchdog->resume(); 3848 } 3849 #endif 3850 } 3851 state->mCommand = FastMixerState::MIX_WRITE; 3852 #ifdef FAST_THREAD_STATISTICS 3853 mFastMixerDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 3854 FastThreadDumpState::kSamplingNforLowRamDevice : FastThreadDumpState::kSamplingN); 3855 #endif 3856 sq->end(); 3857 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3858 if (kUseFastMixer == FastMixer_Dynamic) { 3859 mNormalSink = mPipeSink; 3860 } 3861 } else { 3862 sq->end(false /*didModify*/); 3863 } 3864 } 3865 return PlaybackThread::threadLoop_write(); 3866 } 3867 3868 void AudioFlinger::MixerThread::threadLoop_standby() 3869 { 3870 // Idle the fast mixer if it's currently running 3871 if (mFastMixer != 0) { 3872 FastMixerStateQueue *sq = mFastMixer->sq(); 3873 FastMixerState *state = sq->begin(); 3874 if (!(state->mCommand & FastMixerState::IDLE)) { 3875 // Report any frames trapped in the Monopipe 3876 MonoPipe *monoPipe = (MonoPipe *)mPipeSink.get(); 3877 const long long pipeFrames = monoPipe->maxFrames() - monoPipe->availableToWrite(); 3878 mLocalLog.log("threadLoop_standby: framesWritten:%lld suspendedFrames:%lld " 3879 "monoPipeWritten:%lld monoPipeLeft:%lld", 3880 (long long)mFramesWritten, (long long)mSuspendedFrames, 3881 (long long)mPipeSink->framesWritten(), pipeFrames); 3882 mLocalLog.log("threadLoop_standby: %s", mTimestamp.toString().c_str()); 3883 3884 state->mCommand = FastMixerState::COLD_IDLE; 3885 state->mColdFutexAddr = &mFastMixerFutex; 3886 state->mColdGen++; 3887 mFastMixerFutex = 0; 3888 sq->end(); 3889 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 3890 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3891 if (kUseFastMixer == FastMixer_Dynamic) { 3892 mNormalSink = mOutputSink; 3893 } 3894 #ifdef AUDIO_WATCHDOG 3895 if (mAudioWatchdog != 0) { 3896 mAudioWatchdog->pause(); 3897 } 3898 #endif 3899 } else { 3900 sq->end(false /*didModify*/); 3901 } 3902 } 3903 PlaybackThread::threadLoop_standby(); 3904 } 3905 3906 bool AudioFlinger::PlaybackThread::waitingAsyncCallback_l() 3907 { 3908 return false; 3909 } 3910 3911 bool AudioFlinger::PlaybackThread::shouldStandby_l() 3912 { 3913 return !mStandby; 3914 } 3915 3916 bool AudioFlinger::PlaybackThread::waitingAsyncCallback() 3917 { 3918 Mutex::Autolock _l(mLock); 3919 return waitingAsyncCallback_l(); 3920 } 3921 3922 // shared by MIXER and DIRECT, overridden by DUPLICATING 3923 void AudioFlinger::PlaybackThread::threadLoop_standby() 3924 { 3925 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 3926 mOutput->standby(); 3927 if (mUseAsyncWrite != 0) { 3928 // discard any pending drain or write ack by incrementing sequence 3929 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 3930 mDrainSequence = (mDrainSequence + 2) & ~1; 3931 ALOG_ASSERT(mCallbackThread != 0); 3932 mCallbackThread->setWriteBlocked(mWriteAckSequence); 3933 mCallbackThread->setDraining(mDrainSequence); 3934 } 3935 mHwPaused = false; 3936 } 3937 3938 void AudioFlinger::PlaybackThread::onAddNewTrack_l() 3939 { 3940 ALOGV("signal playback thread"); 3941 broadcast_l(); 3942 } 3943 3944 void AudioFlinger::PlaybackThread::onAsyncError() 3945 { 3946 for (int i = AUDIO_STREAM_SYSTEM; i < (int)AUDIO_STREAM_CNT; i++) { 3947 invalidateTracks((audio_stream_type_t)i); 3948 } 3949 } 3950 3951 void AudioFlinger::MixerThread::threadLoop_mix() 3952 { 3953 // mix buffers... 3954 mAudioMixer->process(); 3955 mCurrentWriteLength = mSinkBufferSize; 3956 // increase sleep time progressively when application underrun condition clears. 3957 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 3958 // that a steady state of alternating ready/not ready conditions keeps the sleep time 3959 // such that we would underrun the audio HAL. 3960 if ((mSleepTimeUs == 0) && (sleepTimeShift > 0)) { 3961 sleepTimeShift--; 3962 } 3963 mSleepTimeUs = 0; 3964 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 3965 //TODO: delay standby when effects have a tail 3966 3967 } 3968 3969 void AudioFlinger::MixerThread::threadLoop_sleepTime() 3970 { 3971 // If no tracks are ready, sleep once for the duration of an output 3972 // buffer size, then write 0s to the output 3973 if (mSleepTimeUs == 0) { 3974 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3975 mSleepTimeUs = mActiveSleepTimeUs >> sleepTimeShift; 3976 if (mSleepTimeUs < kMinThreadSleepTimeUs) { 3977 mSleepTimeUs = kMinThreadSleepTimeUs; 3978 } 3979 // reduce sleep time in case of consecutive application underruns to avoid 3980 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 3981 // duration we would end up writing less data than needed by the audio HAL if 3982 // the condition persists. 3983 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 3984 sleepTimeShift++; 3985 } 3986 } else { 3987 mSleepTimeUs = mIdleSleepTimeUs; 3988 } 3989 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 3990 // clear out mMixerBuffer or mSinkBuffer, to ensure buffers are cleared 3991 // before effects processing or output. 3992 if (mMixerBufferValid) { 3993 memset(mMixerBuffer, 0, mMixerBufferSize); 3994 } else { 3995 memset(mSinkBuffer, 0, mSinkBufferSize); 3996 } 3997 mSleepTimeUs = 0; 3998 ALOGV_IF(mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED), 3999 "anticipated start"); 4000 } 4001 // TODO add standby time extension fct of effect tail 4002 } 4003 4004 // prepareTracks_l() must be called with ThreadBase::mLock held 4005 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 4006 Vector< sp<Track> > *tracksToRemove) 4007 { 4008 4009 mixer_state mixerStatus = MIXER_IDLE; 4010 // find out which tracks need to be processed 4011 size_t count = mActiveTracks.size(); 4012 size_t mixedTracks = 0; 4013 size_t tracksWithEffect = 0; 4014 // counts only _active_ fast tracks 4015 size_t fastTracks = 0; 4016 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 4017 4018 float masterVolume = mMasterVolume; 4019 bool masterMute = mMasterMute; 4020 4021 if (masterMute) { 4022 masterVolume = 0; 4023 } 4024 // Delegate master volume control to effect in output mix effect chain if needed 4025 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4026 if (chain != 0) { 4027 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 4028 chain->setVolume_l(&v, &v); 4029 masterVolume = (float)((v + (1 << 23)) >> 24); 4030 chain.clear(); 4031 } 4032 4033 // prepare a new state to push 4034 FastMixerStateQueue *sq = NULL; 4035 FastMixerState *state = NULL; 4036 bool didModify = false; 4037 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 4038 bool coldIdle = false; 4039 if (mFastMixer != 0) { 4040 sq = mFastMixer->sq(); 4041 state = sq->begin(); 4042 coldIdle = state->mCommand == FastMixerState::COLD_IDLE; 4043 } 4044 4045 mMixerBufferValid = false; // mMixerBuffer has no valid data until appropriate tracks found. 4046 mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found. 4047 4048 for (size_t i=0 ; i<count ; i++) { 4049 const sp<Track> t = mActiveTracks[i]; 4050 4051 // this const just means the local variable doesn't change 4052 Track* const track = t.get(); 4053 4054 // process fast tracks 4055 if (track->isFastTrack()) { 4056 4057 // It's theoretically possible (though unlikely) for a fast track to be created 4058 // and then removed within the same normal mix cycle. This is not a problem, as 4059 // the track never becomes active so it's fast mixer slot is never touched. 4060 // The converse, of removing an (active) track and then creating a new track 4061 // at the identical fast mixer slot within the same normal mix cycle, 4062 // is impossible because the slot isn't marked available until the end of each cycle. 4063 int j = track->mFastIndex; 4064 ALOG_ASSERT(0 < j && j < (int)FastMixerState::sMaxFastTracks); 4065 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 4066 FastTrack *fastTrack = &state->mFastTracks[j]; 4067 4068 // Determine whether the track is currently in underrun condition, 4069 // and whether it had a recent underrun. 4070 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 4071 FastTrackUnderruns underruns = ftDump->mUnderruns; 4072 uint32_t recentFull = (underruns.mBitFields.mFull - 4073 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 4074 uint32_t recentPartial = (underruns.mBitFields.mPartial - 4075 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 4076 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 4077 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 4078 uint32_t recentUnderruns = recentPartial + recentEmpty; 4079 track->mObservedUnderruns = underruns; 4080 // don't count underruns that occur while stopping or pausing 4081 // or stopped which can occur when flush() is called while active 4082 if (!(track->isStopping() || track->isPausing() || track->isStopped()) && 4083 recentUnderruns > 0) { 4084 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun 4085 track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount); 4086 } else { 4087 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4088 } 4089 4090 // This is similar to the state machine for normal tracks, 4091 // with a few modifications for fast tracks. 4092 bool isActive = true; 4093 switch (track->mState) { 4094 case TrackBase::STOPPING_1: 4095 // track stays active in STOPPING_1 state until first underrun 4096 if (recentUnderruns > 0 || track->isTerminated()) { 4097 track->mState = TrackBase::STOPPING_2; 4098 } 4099 break; 4100 case TrackBase::PAUSING: 4101 // ramp down is not yet implemented 4102 track->setPaused(); 4103 break; 4104 case TrackBase::RESUMING: 4105 // ramp up is not yet implemented 4106 track->mState = TrackBase::ACTIVE; 4107 break; 4108 case TrackBase::ACTIVE: 4109 if (recentFull > 0 || recentPartial > 0) { 4110 // track has provided at least some frames recently: reset retry count 4111 track->mRetryCount = kMaxTrackRetries; 4112 } 4113 if (recentUnderruns == 0) { 4114 // no recent underruns: stay active 4115 break; 4116 } 4117 // there has recently been an underrun of some kind 4118 if (track->sharedBuffer() == 0) { 4119 // were any of the recent underruns "empty" (no frames available)? 4120 if (recentEmpty == 0) { 4121 // no, then ignore the partial underruns as they are allowed indefinitely 4122 break; 4123 } 4124 // there has recently been an "empty" underrun: decrement the retry counter 4125 if (--(track->mRetryCount) > 0) { 4126 break; 4127 } 4128 // indicate to client process that the track was disabled because of underrun; 4129 // it will then automatically call start() when data is available 4130 track->disable(); 4131 // remove from active list, but state remains ACTIVE [confusing but true] 4132 isActive = false; 4133 break; 4134 } 4135 // fall through 4136 case TrackBase::STOPPING_2: 4137 case TrackBase::PAUSED: 4138 case TrackBase::STOPPED: 4139 case TrackBase::FLUSHED: // flush() while active 4140 // Check for presentation complete if track is inactive 4141 // We have consumed all the buffers of this track. 4142 // This would be incomplete if we auto-paused on underrun 4143 { 4144 uint32_t latency = 0; 4145 status_t result = mOutput->stream->getLatency(&latency); 4146 ALOGE_IF(result != OK, 4147 "Error when retrieving output stream latency: %d", result); 4148 size_t audioHALFrames = (latency * mSampleRate) / 1000; 4149 int64_t framesWritten = mBytesWritten / mFrameSize; 4150 if (!(mStandby || track->presentationComplete(framesWritten, audioHALFrames))) { 4151 // track stays in active list until presentation is complete 4152 break; 4153 } 4154 } 4155 if (track->isStopping_2()) { 4156 track->mState = TrackBase::STOPPED; 4157 } 4158 if (track->isStopped()) { 4159 // Can't reset directly, as fast mixer is still polling this track 4160 // track->reset(); 4161 // So instead mark this track as needing to be reset after push with ack 4162 resetMask |= 1 << i; 4163 } 4164 isActive = false; 4165 break; 4166 case TrackBase::IDLE: 4167 default: 4168 LOG_ALWAYS_FATAL("unexpected track state %d", track->mState); 4169 } 4170 4171 if (isActive) { 4172 // was it previously inactive? 4173 if (!(state->mTrackMask & (1 << j))) { 4174 ExtendedAudioBufferProvider *eabp = track; 4175 VolumeProvider *vp = track; 4176 fastTrack->mBufferProvider = eabp; 4177 fastTrack->mVolumeProvider = vp; 4178 fastTrack->mChannelMask = track->mChannelMask; 4179 fastTrack->mFormat = track->mFormat; 4180 fastTrack->mGeneration++; 4181 state->mTrackMask |= 1 << j; 4182 didModify = true; 4183 // no acknowledgement required for newly active tracks 4184 } 4185 // cache the combined master volume and stream type volume for fast mixer; this 4186 // lacks any synchronization or barrier so VolumeProvider may read a stale value 4187 const float vh = track->getVolumeHandler()->getVolume( 4188 track->mAudioTrackServerProxy->framesReleased()).first; 4189 track->mCachedVolume = masterVolume 4190 * mStreamTypes[track->streamType()].volume 4191 * vh; 4192 ++fastTracks; 4193 } else { 4194 // was it previously active? 4195 if (state->mTrackMask & (1 << j)) { 4196 fastTrack->mBufferProvider = NULL; 4197 fastTrack->mGeneration++; 4198 state->mTrackMask &= ~(1 << j); 4199 didModify = true; 4200 // If any fast tracks were removed, we must wait for acknowledgement 4201 // because we're about to decrement the last sp<> on those tracks. 4202 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4203 } else { 4204 LOG_ALWAYS_FATAL("fast track %d should have been active; " 4205 "mState=%d, mTrackMask=%#x, recentUnderruns=%u, isShared=%d", 4206 j, track->mState, state->mTrackMask, recentUnderruns, 4207 track->sharedBuffer() != 0); 4208 } 4209 tracksToRemove->add(track); 4210 // Avoids a misleading display in dumpsys 4211 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 4212 } 4213 continue; 4214 } 4215 4216 { // local variable scope to avoid goto warning 4217 4218 audio_track_cblk_t* cblk = track->cblk(); 4219 4220 // The first time a track is added we wait 4221 // for all its buffers to be filled before processing it 4222 int name = track->name(); 4223 // make sure that we have enough frames to mix one full buffer. 4224 // enforce this condition only once to enable draining the buffer in case the client 4225 // app does not call stop() and relies on underrun to stop: 4226 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 4227 // during last round 4228 size_t desiredFrames; 4229 const uint32_t sampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4230 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4231 4232 desiredFrames = sourceFramesNeededWithTimestretch( 4233 sampleRate, mNormalFrameCount, mSampleRate, playbackRate.mSpeed); 4234 // TODO: ONLY USED FOR LEGACY RESAMPLERS, remove when they are removed. 4235 // add frames already consumed but not yet released by the resampler 4236 // because mAudioTrackServerProxy->framesReady() will include these frames 4237 desiredFrames += mAudioMixer->getUnreleasedFrames(track->name()); 4238 4239 uint32_t minFrames = 1; 4240 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 4241 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 4242 minFrames = desiredFrames; 4243 } 4244 4245 size_t framesReady = track->framesReady(); 4246 if (ATRACE_ENABLED()) { 4247 // I wish we had formatted trace names 4248 char traceName[16]; 4249 strcpy(traceName, "nRdy"); 4250 int name = track->name(); 4251 if (AudioMixer::TRACK0 <= name && 4252 name < (int) (AudioMixer::TRACK0 + AudioMixer::MAX_NUM_TRACKS)) { 4253 name -= AudioMixer::TRACK0; 4254 traceName[4] = (name / 10) + '0'; 4255 traceName[5] = (name % 10) + '0'; 4256 } else { 4257 traceName[4] = '?'; 4258 traceName[5] = '?'; 4259 } 4260 traceName[6] = '\0'; 4261 ATRACE_INT(traceName, framesReady); 4262 } 4263 if ((framesReady >= minFrames) && track->isReady() && 4264 !track->isPaused() && !track->isTerminated()) 4265 { 4266 ALOGVV("track %d s=%08x [OK] on thread %p", name, cblk->mServer, this); 4267 4268 mixedTracks++; 4269 4270 // track->mainBuffer() != mSinkBuffer or mMixerBuffer means 4271 // there is an effect chain connected to the track 4272 chain.clear(); 4273 if (track->mainBuffer() != mSinkBuffer && 4274 track->mainBuffer() != mMixerBuffer) { 4275 if (mEffectBufferEnabled) { 4276 mEffectBufferValid = true; // Later can set directly. 4277 } 4278 chain = getEffectChain_l(track->sessionId()); 4279 // Delegate volume control to effect in track effect chain if needed 4280 if (chain != 0) { 4281 tracksWithEffect++; 4282 } else { 4283 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on " 4284 "session %d", 4285 name, track->sessionId()); 4286 } 4287 } 4288 4289 4290 int param = AudioMixer::VOLUME; 4291 if (track->mFillingUpStatus == Track::FS_FILLED) { 4292 // no ramp for the first volume setting 4293 track->mFillingUpStatus = Track::FS_ACTIVE; 4294 if (track->mState == TrackBase::RESUMING) { 4295 track->mState = TrackBase::ACTIVE; 4296 param = AudioMixer::RAMP_VOLUME; 4297 } 4298 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 4299 mLeftVolFloat = -1.0; 4300 // FIXME should not make a decision based on mServer 4301 } else if (cblk->mServer != 0) { 4302 // If the track is stopped before the first frame was mixed, 4303 // do not apply ramp 4304 param = AudioMixer::RAMP_VOLUME; 4305 } 4306 4307 // compute volume for this track 4308 uint32_t vl, vr; // in U8.24 integer format 4309 float vlf, vrf, vaf; // in [0.0, 1.0] float format 4310 // read original volumes with volume control 4311 float typeVolume = mStreamTypes[track->streamType()].volume; 4312 float v = masterVolume * typeVolume; 4313 4314 if (track->isPausing() || mStreamTypes[track->streamType()].mute) { 4315 vl = vr = 0; 4316 vlf = vrf = vaf = 0.; 4317 if (track->isPausing()) { 4318 track->setPaused(); 4319 } 4320 } else { 4321 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy; 4322 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4323 vlf = float_from_gain(gain_minifloat_unpack_left(vlr)); 4324 vrf = float_from_gain(gain_minifloat_unpack_right(vlr)); 4325 // track volumes come from shared memory, so can't be trusted and must be clamped 4326 if (vlf > GAIN_FLOAT_UNITY) { 4327 ALOGV("Track left volume out of range: %.3g", vlf); 4328 vlf = GAIN_FLOAT_UNITY; 4329 } 4330 if (vrf > GAIN_FLOAT_UNITY) { 4331 ALOGV("Track right volume out of range: %.3g", vrf); 4332 vrf = GAIN_FLOAT_UNITY; 4333 } 4334 const float vh = track->getVolumeHandler()->getVolume( 4335 track->mAudioTrackServerProxy->framesReleased()).first; 4336 // now apply the master volume and stream type volume and shaper volume 4337 vlf *= v * vh; 4338 vrf *= v * vh; 4339 // assuming master volume and stream type volume each go up to 1.0, 4340 // then derive vl and vr as U8.24 versions for the effect chain 4341 const float scaleto8_24 = MAX_GAIN_INT * MAX_GAIN_INT; 4342 vl = (uint32_t) (scaleto8_24 * vlf); 4343 vr = (uint32_t) (scaleto8_24 * vrf); 4344 // vl and vr are now in U8.24 format 4345 uint16_t sendLevel = proxy->getSendLevel_U4_12(); 4346 // send level comes from shared memory and so may be corrupt 4347 if (sendLevel > MAX_GAIN_INT) { 4348 ALOGV("Track send level out of range: %04X", sendLevel); 4349 sendLevel = MAX_GAIN_INT; 4350 } 4351 // vaf is represented as [0.0, 1.0] float by rescaling sendLevel 4352 vaf = v * sendLevel * (1. / MAX_GAIN_INT); 4353 } 4354 4355 // Delegate volume control to effect in track effect chain if needed 4356 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 4357 // Do not ramp volume if volume is controlled by effect 4358 param = AudioMixer::VOLUME; 4359 // Update remaining floating point volume levels 4360 vlf = (float)vl / (1 << 24); 4361 vrf = (float)vr / (1 << 24); 4362 track->mHasVolumeController = true; 4363 } else { 4364 // force no volume ramp when volume controller was just disabled or removed 4365 // from effect chain to avoid volume spike 4366 if (track->mHasVolumeController) { 4367 param = AudioMixer::VOLUME; 4368 } 4369 track->mHasVolumeController = false; 4370 } 4371 4372 // For dedicated VoIP outputs, let the HAL apply the stream volume. Track volume is 4373 // still applied by the mixer. 4374 if ((mOutput->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) != 0) { 4375 v = mStreamTypes[track->streamType()].mute ? 0.0f : v; 4376 if (v != mLeftVolFloat) { 4377 status_t result = mOutput->stream->setVolume(v, v); 4378 ALOGE_IF(result != OK, "Error when setting output stream volume: %d", result); 4379 if (result == OK) { 4380 mLeftVolFloat = v; 4381 } 4382 } 4383 // if stream volume was successfully sent to the HAL, mLeftVolFloat == v here and we 4384 // remove stream volume contribution from software volume. 4385 if (v != 0.0f && mLeftVolFloat == v) { 4386 vlf = min(1.0f, vlf / v); 4387 vrf = min(1.0f, vrf / v); 4388 vaf = min(1.0f, vaf / v); 4389 } 4390 } 4391 // XXX: these things DON'T need to be done each time 4392 mAudioMixer->setBufferProvider(name, track); 4393 mAudioMixer->enable(name); 4394 4395 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, &vlf); 4396 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, &vrf); 4397 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, &vaf); 4398 mAudioMixer->setParameter( 4399 name, 4400 AudioMixer::TRACK, 4401 AudioMixer::FORMAT, (void *)track->format()); 4402 mAudioMixer->setParameter( 4403 name, 4404 AudioMixer::TRACK, 4405 AudioMixer::CHANNEL_MASK, (void *)(uintptr_t)track->channelMask()); 4406 mAudioMixer->setParameter( 4407 name, 4408 AudioMixer::TRACK, 4409 AudioMixer::MIXER_CHANNEL_MASK, (void *)(uintptr_t)mChannelMask); 4410 // limit track sample rate to 2 x output sample rate, which changes at re-configuration 4411 uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX; 4412 uint32_t reqSampleRate = track->mAudioTrackServerProxy->getSampleRate(); 4413 if (reqSampleRate == 0) { 4414 reqSampleRate = mSampleRate; 4415 } else if (reqSampleRate > maxSampleRate) { 4416 reqSampleRate = maxSampleRate; 4417 } 4418 mAudioMixer->setParameter( 4419 name, 4420 AudioMixer::RESAMPLE, 4421 AudioMixer::SAMPLE_RATE, 4422 (void *)(uintptr_t)reqSampleRate); 4423 4424 AudioPlaybackRate playbackRate = track->mAudioTrackServerProxy->getPlaybackRate(); 4425 mAudioMixer->setParameter( 4426 name, 4427 AudioMixer::TIMESTRETCH, 4428 AudioMixer::PLAYBACK_RATE, 4429 &playbackRate); 4430 4431 /* 4432 * Select the appropriate output buffer for the track. 4433 * 4434 * Tracks with effects go into their own effects chain buffer 4435 * and from there into either mEffectBuffer or mSinkBuffer. 4436 * 4437 * Other tracks can use mMixerBuffer for higher precision 4438 * channel accumulation. If this buffer is enabled 4439 * (mMixerBufferEnabled true), then selected tracks will accumulate 4440 * into it. 4441 * 4442 */ 4443 if (mMixerBufferEnabled 4444 && (track->mainBuffer() == mSinkBuffer 4445 || track->mainBuffer() == mMixerBuffer)) { 4446 mAudioMixer->setParameter( 4447 name, 4448 AudioMixer::TRACK, 4449 AudioMixer::MIXER_FORMAT, (void *)mMixerBufferFormat); 4450 mAudioMixer->setParameter( 4451 name, 4452 AudioMixer::TRACK, 4453 AudioMixer::MAIN_BUFFER, (void *)mMixerBuffer); 4454 // TODO: override track->mainBuffer()? 4455 mMixerBufferValid = true; 4456 } else { 4457 mAudioMixer->setParameter( 4458 name, 4459 AudioMixer::TRACK, 4460 AudioMixer::MIXER_FORMAT, (void *)AUDIO_FORMAT_PCM_16_BIT); 4461 mAudioMixer->setParameter( 4462 name, 4463 AudioMixer::TRACK, 4464 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 4465 } 4466 mAudioMixer->setParameter( 4467 name, 4468 AudioMixer::TRACK, 4469 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 4470 4471 // reset retry count 4472 track->mRetryCount = kMaxTrackRetries; 4473 4474 // If one track is ready, set the mixer ready if: 4475 // - the mixer was not ready during previous round OR 4476 // - no other track is not ready 4477 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 4478 mixerStatus != MIXER_TRACKS_ENABLED) { 4479 mixerStatus = MIXER_TRACKS_READY; 4480 } 4481 } else { 4482 if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) { 4483 ALOGV("track(%p) underrun, framesReady(%zu) < framesDesired(%zd)", 4484 track, framesReady, desiredFrames); 4485 track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames); 4486 } else { 4487 track->mAudioTrackServerProxy->tallyUnderrunFrames(0); 4488 } 4489 4490 // clear effect chain input buffer if an active track underruns to avoid sending 4491 // previous audio buffer again to effects 4492 chain = getEffectChain_l(track->sessionId()); 4493 if (chain != 0) { 4494 chain->clearInputBuffer(); 4495 } 4496 4497 ALOGVV("track %d s=%08x [NOT READY] on thread %p", name, cblk->mServer, this); 4498 if ((track->sharedBuffer() != 0) || track->isTerminated() || 4499 track->isStopped() || track->isPaused()) { 4500 // We have consumed all the buffers of this track. 4501 // Remove it from the list of active tracks. 4502 // TODO: use actual buffer filling status instead of latency when available from 4503 // audio HAL 4504 size_t audioHALFrames = (latency_l() * mSampleRate) / 1000; 4505 int64_t framesWritten = mBytesWritten / mFrameSize; 4506 if (mStandby || track->presentationComplete(framesWritten, audioHALFrames)) { 4507 if (track->isStopped()) { 4508 track->reset(); 4509 } 4510 tracksToRemove->add(track); 4511 } 4512 } else { 4513 // No buffers for this track. Give it a few chances to 4514 // fill a buffer, then remove it from active list. 4515 if (--(track->mRetryCount) <= 0) { 4516 ALOGI("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 4517 tracksToRemove->add(track); 4518 // indicate to client process that the track was disabled because of underrun; 4519 // it will then automatically call start() when data is available 4520 track->disable(); 4521 // If one track is not ready, mark the mixer also not ready if: 4522 // - the mixer was ready during previous round OR 4523 // - no other track is ready 4524 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 4525 mixerStatus != MIXER_TRACKS_READY) { 4526 mixerStatus = MIXER_TRACKS_ENABLED; 4527 } 4528 } 4529 mAudioMixer->disable(name); 4530 } 4531 4532 } // local variable scope to avoid goto warning 4533 4534 } 4535 4536 // Push the new FastMixer state if necessary 4537 bool pauseAudioWatchdog = false; 4538 if (didModify) { 4539 state->mFastTracksGen++; 4540 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 4541 if (kUseFastMixer == FastMixer_Dynamic && 4542 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 4543 state->mCommand = FastMixerState::COLD_IDLE; 4544 state->mColdFutexAddr = &mFastMixerFutex; 4545 state->mColdGen++; 4546 mFastMixerFutex = 0; 4547 if (kUseFastMixer == FastMixer_Dynamic) { 4548 mNormalSink = mOutputSink; 4549 } 4550 // If we go into cold idle, need to wait for acknowledgement 4551 // so that fast mixer stops doing I/O. 4552 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 4553 pauseAudioWatchdog = true; 4554 } 4555 } 4556 if (sq != NULL) { 4557 sq->end(didModify); 4558 // No need to block if the FastMixer is in COLD_IDLE as the FastThread 4559 // is not active. (We BLOCK_UNTIL_ACKED when entering COLD_IDLE 4560 // when bringing the output sink into standby.) 4561 // 4562 // We will get the latest FastMixer state when we come out of COLD_IDLE. 4563 // 4564 // This occurs with BT suspend when we idle the FastMixer with 4565 // active tracks, which may be added or removed. 4566 sq->push(coldIdle ? FastMixerStateQueue::BLOCK_NEVER : block); 4567 } 4568 #ifdef AUDIO_WATCHDOG 4569 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 4570 mAudioWatchdog->pause(); 4571 } 4572 #endif 4573 4574 // Now perform the deferred reset on fast tracks that have stopped 4575 while (resetMask != 0) { 4576 size_t i = __builtin_ctz(resetMask); 4577 ALOG_ASSERT(i < count); 4578 resetMask &= ~(1 << i); 4579 sp<Track> track = mActiveTracks[i]; 4580 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 4581 track->reset(); 4582 } 4583 4584 // remove all the tracks that need to be... 4585 removeTracks_l(*tracksToRemove); 4586 4587 if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) { 4588 mEffectBufferValid = true; 4589 } 4590 4591 if (mEffectBufferValid) { 4592 // as long as there are effects we should clear the effects buffer, to avoid 4593 // passing a non-clean buffer to the effect chain 4594 memset(mEffectBuffer, 0, mEffectBufferSize); 4595 } 4596 // sink or mix buffer must be cleared if all tracks are connected to an 4597 // effect chain as in this case the mixer will not write to the sink or mix buffer 4598 // and track effects will accumulate into it 4599 if ((mBytesRemaining == 0) && ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4600 (mixedTracks == 0 && fastTracks > 0))) { 4601 // FIXME as a performance optimization, should remember previous zero status 4602 if (mMixerBufferValid) { 4603 memset(mMixerBuffer, 0, mMixerBufferSize); 4604 // TODO: In testing, mSinkBuffer below need not be cleared because 4605 // the PlaybackThread::threadLoop() copies mMixerBuffer into mSinkBuffer 4606 // after mixing. 4607 // 4608 // To enforce this guarantee: 4609 // ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || 4610 // (mixedTracks == 0 && fastTracks > 0)) 4611 // must imply MIXER_TRACKS_READY. 4612 // Later, we may clear buffers regardless, and skip much of this logic. 4613 } 4614 // FIXME as a performance optimization, should remember previous zero status 4615 memset(mSinkBuffer, 0, mNormalFrameCount * mFrameSize); 4616 } 4617 4618 // if any fast tracks, then status is ready 4619 mMixerStatusIgnoringFastTracks = mixerStatus; 4620 if (fastTracks > 0) { 4621 mixerStatus = MIXER_TRACKS_READY; 4622 } 4623 return mixerStatus; 4624 } 4625 4626 // trackCountForUid_l() must be called with ThreadBase::mLock held 4627 uint32_t AudioFlinger::PlaybackThread::trackCountForUid_l(uid_t uid) 4628 { 4629 uint32_t trackCount = 0; 4630 for (size_t i = 0; i < mTracks.size() ; i++) { 4631 if (mTracks[i]->uid() == uid) { 4632 trackCount++; 4633 } 4634 } 4635 return trackCount; 4636 } 4637 4638 // getTrackName_l() must be called with ThreadBase::mLock held 4639 int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask, 4640 audio_format_t format, audio_session_t sessionId, uid_t uid) 4641 { 4642 if (trackCountForUid_l(uid) > (PlaybackThread::kMaxTracksPerUid - 1)) { 4643 return -1; 4644 } 4645 return mAudioMixer->getTrackName(channelMask, format, sessionId); 4646 } 4647 4648 // deleteTrackName_l() must be called with ThreadBase::mLock held 4649 void AudioFlinger::MixerThread::deleteTrackName_l(int name) 4650 { 4651 ALOGV("remove track (%d) and delete from mixer", name); 4652 mAudioMixer->deleteTrackName(name); 4653 } 4654 4655 // checkForNewParameter_l() must be called with ThreadBase::mLock held 4656 bool AudioFlinger::MixerThread::checkForNewParameter_l(const String8& keyValuePair, 4657 status_t& status) 4658 { 4659 bool reconfig = false; 4660 bool a2dpDeviceChanged = false; 4661 4662 status = NO_ERROR; 4663 4664 AutoPark<FastMixer> park(mFastMixer); 4665 4666 AudioParameter param = AudioParameter(keyValuePair); 4667 int value; 4668 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4669 reconfig = true; 4670 } 4671 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4672 if (!isValidPcmSinkFormat((audio_format_t) value)) { 4673 status = BAD_VALUE; 4674 } else { 4675 // no need to save value, since it's constant 4676 reconfig = true; 4677 } 4678 } 4679 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4680 if (!isValidPcmSinkChannelMask((audio_channel_mask_t) value)) { 4681 status = BAD_VALUE; 4682 } else { 4683 // no need to save value, since it's constant 4684 reconfig = true; 4685 } 4686 } 4687 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4688 // do not accept frame count changes if tracks are open as the track buffer 4689 // size depends on frame count and correct behavior would not be guaranteed 4690 // if frame count is changed after track creation 4691 if (!mTracks.isEmpty()) { 4692 status = INVALID_OPERATION; 4693 } else { 4694 reconfig = true; 4695 } 4696 } 4697 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4698 #ifdef ADD_BATTERY_DATA 4699 // when changing the audio output device, call addBatteryData to notify 4700 // the change 4701 if (mOutDevice != value) { 4702 uint32_t params = 0; 4703 // check whether speaker is on 4704 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 4705 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 4706 } 4707 4708 audio_devices_t deviceWithoutSpeaker 4709 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 4710 // check if any other device (except speaker) is on 4711 if (value & deviceWithoutSpeaker) { 4712 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 4713 } 4714 4715 if (params != 0) { 4716 addBatteryData(params); 4717 } 4718 } 4719 #endif 4720 4721 // forward device change to effects that have requested to be 4722 // aware of attached audio device. 4723 if (value != AUDIO_DEVICE_NONE) { 4724 a2dpDeviceChanged = 4725 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 4726 mOutDevice = value; 4727 for (size_t i = 0; i < mEffectChains.size(); i++) { 4728 mEffectChains[i]->setDevice_l(mOutDevice); 4729 } 4730 } 4731 } 4732 4733 if (status == NO_ERROR) { 4734 status = mOutput->stream->setParameters(keyValuePair); 4735 if (!mStandby && status == INVALID_OPERATION) { 4736 mOutput->standby(); 4737 mStandby = true; 4738 mBytesWritten = 0; 4739 status = mOutput->stream->setParameters(keyValuePair); 4740 } 4741 if (status == NO_ERROR && reconfig) { 4742 readOutputParameters_l(); 4743 delete mAudioMixer; 4744 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 4745 for (size_t i = 0; i < mTracks.size() ; i++) { 4746 int name = getTrackName_l(mTracks[i]->mChannelMask, 4747 mTracks[i]->mFormat, mTracks[i]->mSessionId, mTracks[i]->uid()); 4748 if (name < 0) { 4749 break; 4750 } 4751 mTracks[i]->mName = name; 4752 } 4753 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 4754 } 4755 } 4756 4757 return reconfig || a2dpDeviceChanged; 4758 } 4759 4760 4761 void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 4762 { 4763 PlaybackThread::dumpInternals(fd, args); 4764 dprintf(fd, " Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs); 4765 dprintf(fd, " AudioMixer tracks: 0x%08x\n", mAudioMixer->trackNames()); 4766 dprintf(fd, " Master mono: %s\n", mMasterMono ? "on" : "off"); 4767 4768 if (hasFastMixer()) { 4769 dprintf(fd, " FastMixer thread %p tid=%d", mFastMixer.get(), mFastMixer->getTid()); 4770 4771 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 4772 // while we are dumping it. It may be inconsistent, but it won't mutate! 4773 // This is a large object so we place it on the heap. 4774 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 4775 const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState); 4776 copy->dump(fd); 4777 delete copy; 4778 4779 #ifdef STATE_QUEUE_DUMP 4780 // Similar for state queue 4781 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 4782 observerCopy.dump(fd); 4783 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 4784 mutatorCopy.dump(fd); 4785 #endif 4786 4787 #ifdef AUDIO_WATCHDOG 4788 if (mAudioWatchdog != 0) { 4789 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 4790 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 4791 wdCopy.dump(fd); 4792 } 4793 #endif 4794 4795 } else { 4796 dprintf(fd, " No FastMixer\n"); 4797 } 4798 4799 #ifdef TEE_SINK 4800 // Write the tee output to a .wav file 4801 dumpTee(fd, mTeeSource, mId, 'M'); 4802 #endif 4803 4804 } 4805 4806 uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 4807 { 4808 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 4809 } 4810 4811 uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 4812 { 4813 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 4814 } 4815 4816 void AudioFlinger::MixerThread::cacheParameters_l() 4817 { 4818 PlaybackThread::cacheParameters_l(); 4819 4820 // FIXME: Relaxed timing because of a certain device that can't meet latency 4821 // Should be reduced to 2x after the vendor fixes the driver issue 4822 // increase threshold again due to low power audio mode. The way this warning 4823 // threshold is calculated and its usefulness should be reconsidered anyway. 4824 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 4825 } 4826 4827 // ---------------------------------------------------------------------------- 4828 4829 AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4830 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device, bool systemReady) 4831 : PlaybackThread(audioFlinger, output, id, device, DIRECT, systemReady) 4832 { 4833 } 4834 4835 AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 4836 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, 4837 ThreadBase::type_t type, bool systemReady) 4838 : PlaybackThread(audioFlinger, output, id, device, type, systemReady) 4839 , mVolumeShaperActive(false) 4840 { 4841 } 4842 4843 AudioFlinger::DirectOutputThread::~DirectOutputThread() 4844 { 4845 } 4846 4847 void AudioFlinger::DirectOutputThread::processVolume_l(Track *track, bool lastTrack) 4848 { 4849 float left, right; 4850 4851 if (mMasterMute || mStreamTypes[track->streamType()].mute) { 4852 left = right = 0; 4853 } else { 4854 float typeVolume = mStreamTypes[track->streamType()].volume; 4855 float v = mMasterVolume * typeVolume; 4856 sp<AudioTrackServerProxy> proxy = track->mAudioTrackServerProxy; 4857 4858 // Get volumeshaper scaling 4859 std::pair<float /* volume */, bool /* active */> 4860 vh = track->getVolumeHandler()->getVolume( 4861 track->mAudioTrackServerProxy->framesReleased()); 4862 v *= vh.first; 4863 mVolumeShaperActive = vh.second; 4864 4865 gain_minifloat_packed_t vlr = proxy->getVolumeLR(); 4866 left = float_from_gain(gain_minifloat_unpack_left(vlr)); 4867 if (left > GAIN_FLOAT_UNITY) { 4868 left = GAIN_FLOAT_UNITY; 4869 } 4870 left *= v; 4871 right = float_from_gain(gain_minifloat_unpack_right(vlr)); 4872 if (right > GAIN_FLOAT_UNITY) { 4873 right = GAIN_FLOAT_UNITY; 4874 } 4875 right *= v; 4876 } 4877 4878 if (lastTrack) { 4879 if (left != mLeftVolFloat || right != mRightVolFloat) { 4880 mLeftVolFloat = left; 4881 mRightVolFloat = right; 4882 4883 // Convert volumes from float to 8.24 4884 uint32_t vl = (uint32_t)(left * (1 << 24)); 4885 uint32_t vr = (uint32_t)(right * (1 << 24)); 4886 4887 // Delegate volume control to effect in track effect chain if needed 4888 // only one effect chain can be present on DirectOutputThread, so if 4889 // there is one, the track is connected to it 4890 if (!mEffectChains.isEmpty()) { 4891 mEffectChains[0]->setVolume_l(&vl, &vr); 4892 left = (float)vl / (1 << 24); 4893 right = (float)vr / (1 << 24); 4894 } 4895 status_t result = mOutput->stream->setVolume(left, right); 4896 ALOGE_IF(result != OK, "Error when setting output stream volume: %d", result); 4897 } 4898 } 4899 } 4900 4901 void AudioFlinger::DirectOutputThread::onAddNewTrack_l() 4902 { 4903 sp<Track> previousTrack = mPreviousTrack.promote(); 4904 sp<Track> latestTrack = mActiveTracks.getLatest(); 4905 4906 if (previousTrack != 0 && latestTrack != 0) { 4907 if (mType == DIRECT) { 4908 if (previousTrack.get() != latestTrack.get()) { 4909 mFlushPending = true; 4910 } 4911 } else /* mType == OFFLOAD */ { 4912 if (previousTrack->sessionId() != latestTrack->sessionId()) { 4913 mFlushPending = true; 4914 } 4915 } 4916 } 4917 PlaybackThread::onAddNewTrack_l(); 4918 } 4919 4920 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 4921 Vector< sp<Track> > *tracksToRemove 4922 ) 4923 { 4924 size_t count = mActiveTracks.size(); 4925 mixer_state mixerStatus = MIXER_IDLE; 4926 bool doHwPause = false; 4927 bool doHwResume = false; 4928 4929 // find out which tracks need to be processed 4930 for (const sp<Track> &t : mActiveTracks) { 4931 if (t->isInvalid()) { 4932 ALOGW("An invalidated track shouldn't be in active list"); 4933 tracksToRemove->add(t); 4934 continue; 4935 } 4936 4937 Track* const track = t.get(); 4938 #ifdef VERY_VERY_VERBOSE_LOGGING 4939 audio_track_cblk_t* cblk = track->cblk(); 4940 #endif 4941 // Only consider last track started for volume and mixer state control. 4942 // In theory an older track could underrun and restart after the new one starts 4943 // but as we only care about the transition phase between two tracks on a 4944 // direct output, it is not a problem to ignore the underrun case. 4945 sp<Track> l = mActiveTracks.getLatest(); 4946 bool last = l.get() == track; 4947 4948 if (track->isPausing()) { 4949 track->setPaused(); 4950 if (mHwSupportsPause && last && !mHwPaused) { 4951 doHwPause = true; 4952 mHwPaused = true; 4953 } 4954 tracksToRemove->add(track); 4955 } else if (track->isFlushPending()) { 4956 track->flushAck(); 4957 if (last) { 4958 mFlushPending = true; 4959 } 4960 } else if (track->isResumePending()) { 4961 track->resumeAck(); 4962 if (last) { 4963 mLeftVolFloat = mRightVolFloat = -1.0; 4964 if (mHwPaused) { 4965 doHwResume = true; 4966 mHwPaused = false; 4967 } 4968 } 4969 } 4970 4971 // The first time a track is added we wait 4972 // for all its buffers to be filled before processing it. 4973 // Allow draining the buffer in case the client 4974 // app does not call stop() and relies on underrun to stop: 4975 // hence the test on (track->mRetryCount > 1). 4976 // If retryCount<=1 then track is about to underrun and be removed. 4977 // Do not use a high threshold for compressed audio. 4978 uint32_t minFrames; 4979 if ((track->sharedBuffer() == 0) && !track->isStopping_1() && !track->isPausing() 4980 && (track->mRetryCount > 1) && audio_has_proportional_frames(mFormat)) { 4981 minFrames = mNormalFrameCount; 4982 } else { 4983 minFrames = 1; 4984 } 4985 4986 if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() && 4987 !track->isStopping_2() && !track->isStopped()) 4988 { 4989 ALOGVV("track %d s=%08x [OK]", track->name(), cblk->mServer); 4990 4991 if (track->mFillingUpStatus == Track::FS_FILLED) { 4992 track->mFillingUpStatus = Track::FS_ACTIVE; 4993 if (last) { 4994 // make sure processVolume_l() will apply new volume even if 0 4995 mLeftVolFloat = mRightVolFloat = -1.0; 4996 } 4997 if (!mHwSupportsPause) { 4998 track->resumeAck(); 4999 } 5000 } 5001 5002 // compute volume for this track 5003 processVolume_l(track, last); 5004 if (last) { 5005 sp<Track> previousTrack = mPreviousTrack.promote(); 5006 if (previousTrack != 0) { 5007 if (track != previousTrack.get()) { 5008 // Flush any data still being written from last track 5009 mBytesRemaining = 0; 5010 // Invalidate previous track to force a seek when resuming. 5011 previousTrack->invalidate(); 5012 } 5013 } 5014 mPreviousTrack = track; 5015 5016 // reset retry count 5017 track->mRetryCount = kMaxTrackRetriesDirect; 5018 mActiveTrack = t; 5019 mixerStatus = MIXER_TRACKS_READY; 5020 if (mHwPaused) { 5021 doHwResume = true; 5022 mHwPaused = false; 5023 } 5024 } 5025 } else { 5026 // clear effect chain input buffer if the last active track started underruns 5027 // to avoid sending previous audio buffer again to effects 5028 if (!mEffectChains.isEmpty() && last) { 5029 mEffectChains[0]->clearInputBuffer(); 5030 } 5031 if (track->isStopping_1()) { 5032 track->mState = TrackBase::STOPPING_2; 5033 if (last && mHwPaused) { 5034 doHwResume = true; 5035 mHwPaused = false; 5036 } 5037 } 5038 if ((track->sharedBuffer() != 0) || track->isStopped() || 5039 track->isStopping_2() || track->isPaused()) { 5040 // We have consumed all the buffers of this track. 5041 // Remove it from the list of active tracks. 5042 size_t audioHALFrames; 5043 if (audio_has_proportional_frames(mFormat)) { 5044 audioHALFrames = (latency_l() * mSampleRate) / 1000; 5045 } else { 5046 audioHALFrames = 0; 5047 } 5048 5049 int64_t framesWritten = mBytesWritten / mFrameSize; 5050 if (mStandby || !last || 5051 track->presentationComplete(framesWritten, audioHALFrames)) { 5052 if (track->isStopping_2()) { 5053 track->mState = TrackBase::STOPPED; 5054 } 5055 if (track->isStopped()) { 5056 track->reset(); 5057 } 5058 tracksToRemove->add(track); 5059 } 5060 } else { 5061 // No buffers for this track. Give it a few chances to 5062 // fill a buffer, then remove it from active list. 5063 // Only consider last track started for mixer state control 5064 if (--(track->mRetryCount) <= 0) { 5065 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 5066 tracksToRemove->add(track); 5067 // indicate to client process that the track was disabled because of underrun; 5068 // it will then automatically call start() when data is available 5069 track->disable(); 5070 } else if (last) { 5071 ALOGW("pause because of UNDERRUN, framesReady = %zu," 5072 "minFrames = %u, mFormat = %#x", 5073 track->framesReady(), minFrames, mFormat); 5074 mixerStatus = MIXER_TRACKS_ENABLED; 5075 if (mHwSupportsPause && !mHwPaused && !mStandby) { 5076 doHwPause = true; 5077 mHwPaused = true; 5078 } 5079 } 5080 } 5081 } 5082 } 5083 5084 // if an active track did not command a flush, check for pending flush on stopped tracks 5085 if (!mFlushPending) { 5086 for (size_t i = 0; i < mTracks.size(); i++) { 5087 if (mTracks[i]->isFlushPending()) { 5088 mTracks[i]->flushAck(); 5089 mFlushPending = true; 5090 } 5091 } 5092 } 5093 5094 // make sure the pause/flush/resume sequence is executed in the right order. 5095 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5096 // before flush and then resume HW. This can happen in case of pause/flush/resume 5097 // if resume is received before pause is executed. 5098 if (mHwSupportsPause && !mStandby && 5099 (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5100 status_t result = mOutput->stream->pause(); 5101 ALOGE_IF(result != OK, "Error when pausing output stream: %d", result); 5102 } 5103 if (mFlushPending) { 5104 flushHw_l(); 5105 } 5106 if (mHwSupportsPause && !mStandby && doHwResume) { 5107 status_t result = mOutput->stream->resume(); 5108 ALOGE_IF(result != OK, "Error when resuming output stream: %d", result); 5109 } 5110 // remove all the tracks that need to be... 5111 removeTracks_l(*tracksToRemove); 5112 5113 return mixerStatus; 5114 } 5115 5116 void AudioFlinger::DirectOutputThread::threadLoop_mix() 5117 { 5118 size_t frameCount = mFrameCount; 5119 int8_t *curBuf = (int8_t *)mSinkBuffer; 5120 // output audio to hardware 5121 while (frameCount) { 5122 AudioBufferProvider::Buffer buffer; 5123 buffer.frameCount = frameCount; 5124 status_t status = mActiveTrack->getNextBuffer(&buffer); 5125 if (status != NO_ERROR || buffer.raw == NULL) { 5126 // no need to pad with 0 for compressed audio 5127 if (audio_has_proportional_frames(mFormat)) { 5128 memset(curBuf, 0, frameCount * mFrameSize); 5129 } 5130 break; 5131 } 5132 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 5133 frameCount -= buffer.frameCount; 5134 curBuf += buffer.frameCount * mFrameSize; 5135 mActiveTrack->releaseBuffer(&buffer); 5136 } 5137 mCurrentWriteLength = curBuf - (int8_t *)mSinkBuffer; 5138 mSleepTimeUs = 0; 5139 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5140 mActiveTrack.clear(); 5141 } 5142 5143 void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 5144 { 5145 // do not write to HAL when paused 5146 if (mHwPaused || (usesHwAvSync() && mStandby)) { 5147 mSleepTimeUs = mIdleSleepTimeUs; 5148 return; 5149 } 5150 if (mSleepTimeUs == 0) { 5151 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5152 mSleepTimeUs = mActiveSleepTimeUs; 5153 } else { 5154 mSleepTimeUs = mIdleSleepTimeUs; 5155 } 5156 } else if (mBytesWritten != 0 && audio_has_proportional_frames(mFormat)) { 5157 memset(mSinkBuffer, 0, mFrameCount * mFrameSize); 5158 mSleepTimeUs = 0; 5159 } 5160 } 5161 5162 void AudioFlinger::DirectOutputThread::threadLoop_exit() 5163 { 5164 { 5165 Mutex::Autolock _l(mLock); 5166 for (size_t i = 0; i < mTracks.size(); i++) { 5167 if (mTracks[i]->isFlushPending()) { 5168 mTracks[i]->flushAck(); 5169 mFlushPending = true; 5170 } 5171 } 5172 if (mFlushPending) { 5173 flushHw_l(); 5174 } 5175 } 5176 PlaybackThread::threadLoop_exit(); 5177 } 5178 5179 // must be called with thread mutex locked 5180 bool AudioFlinger::DirectOutputThread::shouldStandby_l() 5181 { 5182 bool trackPaused = false; 5183 bool trackStopped = false; 5184 5185 if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) { 5186 return !mStandby; 5187 } 5188 5189 // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack 5190 // after a timeout and we will enter standby then. 5191 if (mTracks.size() > 0) { 5192 trackPaused = mTracks[mTracks.size() - 1]->isPaused(); 5193 trackStopped = mTracks[mTracks.size() - 1]->isStopped() || 5194 mTracks[mTracks.size() - 1]->mState == TrackBase::IDLE; 5195 } 5196 5197 return !mStandby && !(trackPaused || (mHwPaused && !trackStopped)); 5198 } 5199 5200 // getTrackName_l() must be called with ThreadBase::mLock held 5201 int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask __unused, 5202 audio_format_t format __unused, audio_session_t sessionId __unused, uid_t uid) 5203 { 5204 if (trackCountForUid_l(uid) > (PlaybackThread::kMaxTracksPerUid - 1)) { 5205 return -1; 5206 } 5207 return 0; 5208 } 5209 5210 // deleteTrackName_l() must be called with ThreadBase::mLock held 5211 void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name __unused) 5212 { 5213 } 5214 5215 // checkForNewParameter_l() must be called with ThreadBase::mLock held 5216 bool AudioFlinger::DirectOutputThread::checkForNewParameter_l(const String8& keyValuePair, 5217 status_t& status) 5218 { 5219 bool reconfig = false; 5220 bool a2dpDeviceChanged = false; 5221 5222 status = NO_ERROR; 5223 5224 AudioParameter param = AudioParameter(keyValuePair); 5225 int value; 5226 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5227 // forward device change to effects that have requested to be 5228 // aware of attached audio device. 5229 if (value != AUDIO_DEVICE_NONE) { 5230 a2dpDeviceChanged = 5231 (mOutDevice & AUDIO_DEVICE_OUT_ALL_A2DP) != (value & AUDIO_DEVICE_OUT_ALL_A2DP); 5232 mOutDevice = value; 5233 for (size_t i = 0; i < mEffectChains.size(); i++) { 5234 mEffectChains[i]->setDevice_l(mOutDevice); 5235 } 5236 } 5237 } 5238 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5239 // do not accept frame count changes if tracks are open as the track buffer 5240 // size depends on frame count and correct behavior would not be garantied 5241 // if frame count is changed after track creation 5242 if (!mTracks.isEmpty()) { 5243 status = INVALID_OPERATION; 5244 } else { 5245 reconfig = true; 5246 } 5247 } 5248 if (status == NO_ERROR) { 5249 status = mOutput->stream->setParameters(keyValuePair); 5250 if (!mStandby && status == INVALID_OPERATION) { 5251 mOutput->standby(); 5252 mStandby = true; 5253 mBytesWritten = 0; 5254 status = mOutput->stream->setParameters(keyValuePair); 5255 } 5256 if (status == NO_ERROR && reconfig) { 5257 readOutputParameters_l(); 5258 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 5259 } 5260 } 5261 5262 return reconfig || a2dpDeviceChanged; 5263 } 5264 5265 uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 5266 { 5267 uint32_t time; 5268 if (audio_has_proportional_frames(mFormat)) { 5269 time = PlaybackThread::activeSleepTimeUs(); 5270 } else { 5271 time = kDirectMinSleepTimeUs; 5272 } 5273 return time; 5274 } 5275 5276 uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 5277 { 5278 uint32_t time; 5279 if (audio_has_proportional_frames(mFormat)) { 5280 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 5281 } else { 5282 time = kDirectMinSleepTimeUs; 5283 } 5284 return time; 5285 } 5286 5287 uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 5288 { 5289 uint32_t time; 5290 if (audio_has_proportional_frames(mFormat)) { 5291 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 5292 } else { 5293 time = kDirectMinSleepTimeUs; 5294 } 5295 return time; 5296 } 5297 5298 void AudioFlinger::DirectOutputThread::cacheParameters_l() 5299 { 5300 PlaybackThread::cacheParameters_l(); 5301 5302 // use shorter standby delay as on normal output to release 5303 // hardware resources as soon as possible 5304 // no delay on outputs with HW A/V sync 5305 if (usesHwAvSync()) { 5306 mStandbyDelayNs = 0; 5307 } else if ((mType == OFFLOAD) && !audio_has_proportional_frames(mFormat)) { 5308 mStandbyDelayNs = kOffloadStandbyDelayNs; 5309 } else { 5310 mStandbyDelayNs = microseconds(mActiveSleepTimeUs*2); 5311 } 5312 } 5313 5314 void AudioFlinger::DirectOutputThread::flushHw_l() 5315 { 5316 mOutput->flush(); 5317 mHwPaused = false; 5318 mFlushPending = false; 5319 } 5320 5321 int64_t AudioFlinger::DirectOutputThread::computeWaitTimeNs_l() const { 5322 // If a VolumeShaper is active, we must wake up periodically to update volume. 5323 const int64_t NS_PER_MS = 1000000; 5324 return mVolumeShaperActive ? 5325 kMinNormalSinkBufferSizeMs * NS_PER_MS : PlaybackThread::computeWaitTimeNs_l(); 5326 } 5327 5328 // ---------------------------------------------------------------------------- 5329 5330 AudioFlinger::AsyncCallbackThread::AsyncCallbackThread( 5331 const wp<AudioFlinger::PlaybackThread>& playbackThread) 5332 : Thread(false /*canCallJava*/), 5333 mPlaybackThread(playbackThread), 5334 mWriteAckSequence(0), 5335 mDrainSequence(0), 5336 mAsyncError(false) 5337 { 5338 } 5339 5340 AudioFlinger::AsyncCallbackThread::~AsyncCallbackThread() 5341 { 5342 } 5343 5344 void AudioFlinger::AsyncCallbackThread::onFirstRef() 5345 { 5346 run("Offload Cbk", ANDROID_PRIORITY_URGENT_AUDIO); 5347 } 5348 5349 bool AudioFlinger::AsyncCallbackThread::threadLoop() 5350 { 5351 while (!exitPending()) { 5352 uint32_t writeAckSequence; 5353 uint32_t drainSequence; 5354 bool asyncError; 5355 5356 { 5357 Mutex::Autolock _l(mLock); 5358 while (!((mWriteAckSequence & 1) || 5359 (mDrainSequence & 1) || 5360 mAsyncError || 5361 exitPending())) { 5362 mWaitWorkCV.wait(mLock); 5363 } 5364 5365 if (exitPending()) { 5366 break; 5367 } 5368 ALOGV("AsyncCallbackThread mWriteAckSequence %d mDrainSequence %d", 5369 mWriteAckSequence, mDrainSequence); 5370 writeAckSequence = mWriteAckSequence; 5371 mWriteAckSequence &= ~1; 5372 drainSequence = mDrainSequence; 5373 mDrainSequence &= ~1; 5374 asyncError = mAsyncError; 5375 mAsyncError = false; 5376 } 5377 { 5378 sp<AudioFlinger::PlaybackThread> playbackThread = mPlaybackThread.promote(); 5379 if (playbackThread != 0) { 5380 if (writeAckSequence & 1) { 5381 playbackThread->resetWriteBlocked(writeAckSequence >> 1); 5382 } 5383 if (drainSequence & 1) { 5384 playbackThread->resetDraining(drainSequence >> 1); 5385 } 5386 if (asyncError) { 5387 playbackThread->onAsyncError(); 5388 } 5389 } 5390 } 5391 } 5392 return false; 5393 } 5394 5395 void AudioFlinger::AsyncCallbackThread::exit() 5396 { 5397 ALOGV("AsyncCallbackThread::exit"); 5398 Mutex::Autolock _l(mLock); 5399 requestExit(); 5400 mWaitWorkCV.broadcast(); 5401 } 5402 5403 void AudioFlinger::AsyncCallbackThread::setWriteBlocked(uint32_t sequence) 5404 { 5405 Mutex::Autolock _l(mLock); 5406 // bit 0 is cleared 5407 mWriteAckSequence = sequence << 1; 5408 } 5409 5410 void AudioFlinger::AsyncCallbackThread::resetWriteBlocked() 5411 { 5412 Mutex::Autolock _l(mLock); 5413 // ignore unexpected callbacks 5414 if (mWriteAckSequence & 2) { 5415 mWriteAckSequence |= 1; 5416 mWaitWorkCV.signal(); 5417 } 5418 } 5419 5420 void AudioFlinger::AsyncCallbackThread::setDraining(uint32_t sequence) 5421 { 5422 Mutex::Autolock _l(mLock); 5423 // bit 0 is cleared 5424 mDrainSequence = sequence << 1; 5425 } 5426 5427 void AudioFlinger::AsyncCallbackThread::resetDraining() 5428 { 5429 Mutex::Autolock _l(mLock); 5430 // ignore unexpected callbacks 5431 if (mDrainSequence & 2) { 5432 mDrainSequence |= 1; 5433 mWaitWorkCV.signal(); 5434 } 5435 } 5436 5437 void AudioFlinger::AsyncCallbackThread::setAsyncError() 5438 { 5439 Mutex::Autolock _l(mLock); 5440 mAsyncError = true; 5441 mWaitWorkCV.signal(); 5442 } 5443 5444 5445 // ---------------------------------------------------------------------------- 5446 AudioFlinger::OffloadThread::OffloadThread(const sp<AudioFlinger>& audioFlinger, 5447 AudioStreamOut* output, audio_io_handle_t id, uint32_t device, bool systemReady) 5448 : DirectOutputThread(audioFlinger, output, id, device, OFFLOAD, systemReady), 5449 mPausedWriteLength(0), mPausedBytesRemaining(0), mKeepWakeLock(true), 5450 mOffloadUnderrunPosition(~0LL) 5451 { 5452 //FIXME: mStandby should be set to true by ThreadBase constructor 5453 mStandby = true; 5454 mKeepWakeLock = property_get_bool("ro.audio.offload_wakelock", true /* default_value */); 5455 } 5456 5457 void AudioFlinger::OffloadThread::threadLoop_exit() 5458 { 5459 if (mFlushPending || mHwPaused) { 5460 // If a flush is pending or track was paused, just discard buffered data 5461 flushHw_l(); 5462 } else { 5463 mMixerStatus = MIXER_DRAIN_ALL; 5464 threadLoop_drain(); 5465 } 5466 if (mUseAsyncWrite) { 5467 ALOG_ASSERT(mCallbackThread != 0); 5468 mCallbackThread->exit(); 5469 } 5470 PlaybackThread::threadLoop_exit(); 5471 } 5472 5473 AudioFlinger::PlaybackThread::mixer_state AudioFlinger::OffloadThread::prepareTracks_l( 5474 Vector< sp<Track> > *tracksToRemove 5475 ) 5476 { 5477 size_t count = mActiveTracks.size(); 5478 5479 mixer_state mixerStatus = MIXER_IDLE; 5480 bool doHwPause = false; 5481 bool doHwResume = false; 5482 5483 ALOGV("OffloadThread::prepareTracks_l active tracks %zu", count); 5484 5485 // find out which tracks need to be processed 5486 for (const sp<Track> &t : mActiveTracks) { 5487 Track* const track = t.get(); 5488 #ifdef VERY_VERY_VERBOSE_LOGGING 5489 audio_track_cblk_t* cblk = track->cblk(); 5490 #endif 5491 // Only consider last track started for volume and mixer state control. 5492 // In theory an older track could underrun and restart after the new one starts 5493 // but as we only care about the transition phase between two tracks on a 5494 // direct output, it is not a problem to ignore the underrun case. 5495 sp<Track> l = mActiveTracks.getLatest(); 5496 bool last = l.get() == track; 5497 5498 if (track->isInvalid()) { 5499 ALOGW("An invalidated track shouldn't be in active list"); 5500 tracksToRemove->add(track); 5501 continue; 5502 } 5503 5504 if (track->mState == TrackBase::IDLE) { 5505 ALOGW("An idle track shouldn't be in active list"); 5506 continue; 5507 } 5508 5509 if (track->isPausing()) { 5510 track->setPaused(); 5511 if (last) { 5512 if (mHwSupportsPause && !mHwPaused) { 5513 doHwPause = true; 5514 mHwPaused = true; 5515 } 5516 // If we were part way through writing the mixbuffer to 5517 // the HAL we must save this until we resume 5518 // BUG - this will be wrong if a different track is made active, 5519 // in that case we want to discard the pending data in the 5520 // mixbuffer and tell the client to present it again when the 5521 // track is resumed 5522 mPausedWriteLength = mCurrentWriteLength; 5523 mPausedBytesRemaining = mBytesRemaining; 5524 mBytesRemaining = 0; // stop writing 5525 } 5526 tracksToRemove->add(track); 5527 } else if (track->isFlushPending()) { 5528 if (track->isStopping_1()) { 5529 track->mRetryCount = kMaxTrackStopRetriesOffload; 5530 } else { 5531 track->mRetryCount = kMaxTrackRetriesOffload; 5532 } 5533 track->flushAck(); 5534 if (last) { 5535 mFlushPending = true; 5536 } 5537 } else if (track->isResumePending()){ 5538 track->resumeAck(); 5539 if (last) { 5540 if (mPausedBytesRemaining) { 5541 // Need to continue write that was interrupted 5542 mCurrentWriteLength = mPausedWriteLength; 5543 mBytesRemaining = mPausedBytesRemaining; 5544 mPausedBytesRemaining = 0; 5545 } 5546 if (mHwPaused) { 5547 doHwResume = true; 5548 mHwPaused = false; 5549 // threadLoop_mix() will handle the case that we need to 5550 // resume an interrupted write 5551 } 5552 // enable write to audio HAL 5553 mSleepTimeUs = 0; 5554 5555 mLeftVolFloat = mRightVolFloat = -1.0; 5556 5557 // Do not handle new data in this iteration even if track->framesReady() 5558 mixerStatus = MIXER_TRACKS_ENABLED; 5559 } 5560 } else if (track->framesReady() && track->isReady() && 5561 !track->isPaused() && !track->isTerminated() && !track->isStopping_2()) { 5562 ALOGVV("OffloadThread: track %d s=%08x [OK]", track->name(), cblk->mServer); 5563 if (track->mFillingUpStatus == Track::FS_FILLED) { 5564 track->mFillingUpStatus = Track::FS_ACTIVE; 5565 if (last) { 5566 // make sure processVolume_l() will apply new volume even if 0 5567 mLeftVolFloat = mRightVolFloat = -1.0; 5568 } 5569 } 5570 5571 if (last) { 5572 sp<Track> previousTrack = mPreviousTrack.promote(); 5573 if (previousTrack != 0) { 5574 if (track != previousTrack.get()) { 5575 // Flush any data still being written from last track 5576 mBytesRemaining = 0; 5577 if (mPausedBytesRemaining) { 5578 // Last track was paused so we also need to flush saved 5579 // mixbuffer state and invalidate track so that it will 5580 // re-submit that unwritten data when it is next resumed 5581 mPausedBytesRemaining = 0; 5582 // Invalidate is a bit drastic - would be more efficient 5583 // to have a flag to tell client that some of the 5584 // previously written data was lost 5585 previousTrack->invalidate(); 5586 } 5587 // flush data already sent to the DSP if changing audio session as audio 5588 // comes from a different source. Also invalidate previous track to force a 5589 // seek when resuming. 5590 if (previousTrack->sessionId() != track->sessionId()) { 5591 previousTrack->invalidate(); 5592 } 5593 } 5594 } 5595 mPreviousTrack = track; 5596 // reset retry count 5597 if (track->isStopping_1()) { 5598 track->mRetryCount = kMaxTrackStopRetriesOffload; 5599 } else { 5600 track->mRetryCount = kMaxTrackRetriesOffload; 5601 } 5602 mActiveTrack = t; 5603 mixerStatus = MIXER_TRACKS_READY; 5604 } 5605 } else { 5606 ALOGVV("OffloadThread: track %d s=%08x [NOT READY]", track->name(), cblk->mServer); 5607 if (track->isStopping_1()) { 5608 if (--(track->mRetryCount) <= 0) { 5609 // Hardware buffer can hold a large amount of audio so we must 5610 // wait for all current track's data to drain before we say 5611 // that the track is stopped. 5612 if (mBytesRemaining == 0) { 5613 // Only start draining when all data in mixbuffer 5614 // has been written 5615 ALOGV("OffloadThread: underrun and STOPPING_1 -> draining, STOPPING_2"); 5616 track->mState = TrackBase::STOPPING_2; // so presentation completes after 5617 // drain do not drain if no data was ever sent to HAL (mStandby == true) 5618 if (last && !mStandby) { 5619 // do not modify drain sequence if we are already draining. This happens 5620 // when resuming from pause after drain. 5621 if ((mDrainSequence & 1) == 0) { 5622 mSleepTimeUs = 0; 5623 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5624 mixerStatus = MIXER_DRAIN_TRACK; 5625 mDrainSequence += 2; 5626 } 5627 if (mHwPaused) { 5628 // It is possible to move from PAUSED to STOPPING_1 without 5629 // a resume so we must ensure hardware is running 5630 doHwResume = true; 5631 mHwPaused = false; 5632 } 5633 } 5634 } 5635 } else if (last) { 5636 ALOGV("stopping1 underrun retries left %d", track->mRetryCount); 5637 mixerStatus = MIXER_TRACKS_ENABLED; 5638 } 5639 } else if (track->isStopping_2()) { 5640 // Drain has completed or we are in standby, signal presentation complete 5641 if (!(mDrainSequence & 1) || !last || mStandby) { 5642 track->mState = TrackBase::STOPPED; 5643 uint32_t latency = 0; 5644 status_t result = mOutput->stream->getLatency(&latency); 5645 ALOGE_IF(result != OK, 5646 "Error when retrieving output stream latency: %d", result); 5647 size_t audioHALFrames = (latency * mSampleRate) / 1000; 5648 int64_t framesWritten = 5649 mBytesWritten / mOutput->getFrameSize(); 5650 track->presentationComplete(framesWritten, audioHALFrames); 5651 track->reset(); 5652 tracksToRemove->add(track); 5653 } 5654 } else { 5655 // No buffers for this track. Give it a few chances to 5656 // fill a buffer, then remove it from active list. 5657 if (--(track->mRetryCount) <= 0) { 5658 bool running = false; 5659 uint64_t position = 0; 5660 struct timespec unused; 5661 // The running check restarts the retry counter at least once. 5662 status_t ret = mOutput->stream->getPresentationPosition(&position, &unused); 5663 if (ret == NO_ERROR && position != mOffloadUnderrunPosition) { 5664 running = true; 5665 mOffloadUnderrunPosition = position; 5666 } 5667 if (ret == NO_ERROR) { 5668 ALOGVV("underrun counter, running(%d): %lld vs %lld", running, 5669 (long long)position, (long long)mOffloadUnderrunPosition); 5670 } 5671 if (running) { // still running, give us more time. 5672 track->mRetryCount = kMaxTrackRetriesOffload; 5673 } else { 5674 ALOGV("OffloadThread: BUFFER TIMEOUT: remove(%d) from active list", 5675 track->name()); 5676 tracksToRemove->add(track); 5677 // tell client process that the track was disabled because of underrun; 5678 // it will then automatically call start() when data is available 5679 track->disable(); 5680 } 5681 } else if (last){ 5682 mixerStatus = MIXER_TRACKS_ENABLED; 5683 } 5684 } 5685 } 5686 // compute volume for this track 5687 processVolume_l(track, last); 5688 } 5689 5690 // make sure the pause/flush/resume sequence is executed in the right order. 5691 // If a flush is pending and a track is active but the HW is not paused, force a HW pause 5692 // before flush and then resume HW. This can happen in case of pause/flush/resume 5693 // if resume is received before pause is executed. 5694 if (!mStandby && (doHwPause || (mFlushPending && !mHwPaused && (count != 0)))) { 5695 status_t result = mOutput->stream->pause(); 5696 ALOGE_IF(result != OK, "Error when pausing output stream: %d", result); 5697 } 5698 if (mFlushPending) { 5699 flushHw_l(); 5700 } 5701 if (!mStandby && doHwResume) { 5702 status_t result = mOutput->stream->resume(); 5703 ALOGE_IF(result != OK, "Error when resuming output stream: %d", result); 5704 } 5705 5706 // remove all the tracks that need to be... 5707 removeTracks_l(*tracksToRemove); 5708 5709 return mixerStatus; 5710 } 5711 5712 // must be called with thread mutex locked 5713 bool AudioFlinger::OffloadThread::waitingAsyncCallback_l() 5714 { 5715 ALOGVV("waitingAsyncCallback_l mWriteAckSequence %d mDrainSequence %d", 5716 mWriteAckSequence, mDrainSequence); 5717 if (mUseAsyncWrite && ((mWriteAckSequence & 1) || (mDrainSequence & 1))) { 5718 return true; 5719 } 5720 return false; 5721 } 5722 5723 bool AudioFlinger::OffloadThread::waitingAsyncCallback() 5724 { 5725 Mutex::Autolock _l(mLock); 5726 return waitingAsyncCallback_l(); 5727 } 5728 5729 void AudioFlinger::OffloadThread::flushHw_l() 5730 { 5731 DirectOutputThread::flushHw_l(); 5732 // Flush anything still waiting in the mixbuffer 5733 mCurrentWriteLength = 0; 5734 mBytesRemaining = 0; 5735 mPausedWriteLength = 0; 5736 mPausedBytesRemaining = 0; 5737 // reset bytes written count to reflect that DSP buffers are empty after flush. 5738 mBytesWritten = 0; 5739 mOffloadUnderrunPosition = ~0LL; 5740 5741 if (mUseAsyncWrite) { 5742 // discard any pending drain or write ack by incrementing sequence 5743 mWriteAckSequence = (mWriteAckSequence + 2) & ~1; 5744 mDrainSequence = (mDrainSequence + 2) & ~1; 5745 ALOG_ASSERT(mCallbackThread != 0); 5746 mCallbackThread->setWriteBlocked(mWriteAckSequence); 5747 mCallbackThread->setDraining(mDrainSequence); 5748 } 5749 } 5750 5751 void AudioFlinger::OffloadThread::invalidateTracks(audio_stream_type_t streamType) 5752 { 5753 Mutex::Autolock _l(mLock); 5754 if (PlaybackThread::invalidateTracks_l(streamType)) { 5755 mFlushPending = true; 5756 } 5757 } 5758 5759 // ---------------------------------------------------------------------------- 5760 5761 AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 5762 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id, bool systemReady) 5763 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), 5764 systemReady, DUPLICATING), 5765 mWaitTimeMs(UINT_MAX) 5766 { 5767 addOutputTrack(mainThread); 5768 } 5769 5770 AudioFlinger::DuplicatingThread::~DuplicatingThread() 5771 { 5772 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5773 mOutputTracks[i]->destroy(); 5774 } 5775 } 5776 5777 void AudioFlinger::DuplicatingThread::threadLoop_mix() 5778 { 5779 // mix buffers... 5780 if (outputsReady(outputTracks)) { 5781 mAudioMixer->process(); 5782 } else { 5783 if (mMixerBufferValid) { 5784 memset(mMixerBuffer, 0, mMixerBufferSize); 5785 } else { 5786 memset(mSinkBuffer, 0, mSinkBufferSize); 5787 } 5788 } 5789 mSleepTimeUs = 0; 5790 writeFrames = mNormalFrameCount; 5791 mCurrentWriteLength = mSinkBufferSize; 5792 mStandbyTimeNs = systemTime() + mStandbyDelayNs; 5793 } 5794 5795 void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 5796 { 5797 if (mSleepTimeUs == 0) { 5798 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5799 mSleepTimeUs = mActiveSleepTimeUs; 5800 } else { 5801 mSleepTimeUs = mIdleSleepTimeUs; 5802 } 5803 } else if (mBytesWritten != 0) { 5804 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 5805 writeFrames = mNormalFrameCount; 5806 memset(mSinkBuffer, 0, mSinkBufferSize); 5807 } else { 5808 // flush remaining overflow buffers in output tracks 5809 writeFrames = 0; 5810 } 5811 mSleepTimeUs = 0; 5812 } 5813 } 5814 5815 ssize_t AudioFlinger::DuplicatingThread::threadLoop_write() 5816 { 5817 for (size_t i = 0; i < outputTracks.size(); i++) { 5818 outputTracks[i]->write(mSinkBuffer, writeFrames); 5819 } 5820 mStandby = false; 5821 return (ssize_t)mSinkBufferSize; 5822 } 5823 5824 void AudioFlinger::DuplicatingThread::threadLoop_standby() 5825 { 5826 // DuplicatingThread implements standby by stopping all tracks 5827 for (size_t i = 0; i < outputTracks.size(); i++) { 5828 outputTracks[i]->stop(); 5829 } 5830 } 5831 5832 void AudioFlinger::DuplicatingThread::saveOutputTracks() 5833 { 5834 outputTracks = mOutputTracks; 5835 } 5836 5837 void AudioFlinger::DuplicatingThread::clearOutputTracks() 5838 { 5839 outputTracks.clear(); 5840 } 5841 5842 void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 5843 { 5844 Mutex::Autolock _l(mLock); 5845 // The downstream MixerThread consumes thread->frameCount() amount of frames per mix pass. 5846 // Adjust for thread->sampleRate() to determine minimum buffer frame count. 5847 // Then triple buffer because Threads do not run synchronously and may not be clock locked. 5848 const size_t frameCount = 5849 3 * sourceFramesNeeded(mSampleRate, thread->frameCount(), thread->sampleRate()); 5850 // TODO: Consider asynchronous sample rate conversion to handle clock disparity 5851 // from different OutputTracks and their associated MixerThreads (e.g. one may 5852 // nearly empty and the other may be dropping data). 5853 5854 sp<OutputTrack> outputTrack = new OutputTrack(thread, 5855 this, 5856 mSampleRate, 5857 mFormat, 5858 mChannelMask, 5859 frameCount, 5860 IPCThreadState::self()->getCallingUid()); 5861 status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY; 5862 if (status != NO_ERROR) { 5863 ALOGE("addOutputTrack() initCheck failed %d", status); 5864 return; 5865 } 5866 thread->setStreamVolume(AUDIO_STREAM_PATCH, 1.0f); 5867 mOutputTracks.add(outputTrack); 5868 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack.get(), thread); 5869 updateWaitTime_l(); 5870 } 5871 5872 void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 5873 { 5874 Mutex::Autolock _l(mLock); 5875 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5876 if (mOutputTracks[i]->thread() == thread) { 5877 mOutputTracks[i]->destroy(); 5878 mOutputTracks.removeAt(i); 5879 updateWaitTime_l(); 5880 if (thread->getOutput() == mOutput) { 5881 mOutput = NULL; 5882 } 5883 return; 5884 } 5885 } 5886 ALOGV("removeOutputTrack(): unknown thread: %p", thread); 5887 } 5888 5889 // caller must hold mLock 5890 void AudioFlinger::DuplicatingThread::updateWaitTime_l() 5891 { 5892 mWaitTimeMs = UINT_MAX; 5893 for (size_t i = 0; i < mOutputTracks.size(); i++) { 5894 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 5895 if (strong != 0) { 5896 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 5897 if (waitTimeMs < mWaitTimeMs) { 5898 mWaitTimeMs = waitTimeMs; 5899 } 5900 } 5901 } 5902 } 5903 5904 5905 bool AudioFlinger::DuplicatingThread::outputsReady( 5906 const SortedVector< sp<OutputTrack> > &outputTracks) 5907 { 5908 for (size_t i = 0; i < outputTracks.size(); i++) { 5909 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 5910 if (thread == 0) { 5911 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", 5912 outputTracks[i].get()); 5913 return false; 5914 } 5915 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5916 // see note at standby() declaration 5917 if (playbackThread->standby() && !playbackThread->isSuspended()) { 5918 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), 5919 thread.get()); 5920 return false; 5921 } 5922 } 5923 return true; 5924 } 5925 5926 uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 5927 { 5928 return (mWaitTimeMs * 1000) / 2; 5929 } 5930 5931 void AudioFlinger::DuplicatingThread::cacheParameters_l() 5932 { 5933 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 5934 updateWaitTime_l(); 5935 5936 MixerThread::cacheParameters_l(); 5937 } 5938 5939 5940 // ---------------------------------------------------------------------------- 5941 // Record 5942 // ---------------------------------------------------------------------------- 5943 5944 AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5945 AudioStreamIn *input, 5946 audio_io_handle_t id, 5947 audio_devices_t outDevice, 5948 audio_devices_t inDevice, 5949 bool systemReady 5950 #ifdef TEE_SINK 5951 , const sp<NBAIO_Sink>& teeSink 5952 #endif 5953 ) : 5954 ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady), 5955 mInput(input), 5956 mActiveTracks(&this->mLocalLog), 5957 mRsmpInBuffer(NULL), 5958 // mRsmpInFrames, mRsmpInFramesP2, and mRsmpInFramesOA are set by readInputParameters_l() 5959 mRsmpInRear(0) 5960 #ifdef TEE_SINK 5961 , mTeeSink(teeSink) 5962 #endif 5963 , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize, 5964 "RecordThreadRO", MemoryHeapBase::READ_ONLY)) 5965 // mFastCapture below 5966 , mFastCaptureFutex(0) 5967 // mInputSource 5968 // mPipeSink 5969 // mPipeSource 5970 , mPipeFramesP2(0) 5971 // mPipeMemory 5972 // mFastCaptureNBLogWriter 5973 , mFastTrackAvail(false) 5974 , mBtNrecSuspended(false) 5975 { 5976 snprintf(mThreadName, kThreadNameLength, "AudioIn_%X", id); 5977 mNBLogWriter = audioFlinger->newWriter_l(kLogSize, mThreadName); 5978 5979 readInputParameters_l(); 5980 5981 // create an NBAIO source for the HAL input stream, and negotiate 5982 mInputSource = new AudioStreamInSource(input->stream); 5983 size_t numCounterOffers = 0; 5984 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount, mFormat)}; 5985 #if !LOG_NDEBUG 5986 ssize_t index = 5987 #else 5988 (void) 5989 #endif 5990 mInputSource->negotiate(offers, 1, NULL, numCounterOffers); 5991 ALOG_ASSERT(index == 0); 5992 5993 // initialize fast capture depending on configuration 5994 bool initFastCapture; 5995 switch (kUseFastCapture) { 5996 case FastCapture_Never: 5997 initFastCapture = false; 5998 ALOGV("%p kUseFastCapture = Never, initFastCapture = false", this); 5999 break; 6000 case FastCapture_Always: 6001 initFastCapture = true; 6002 ALOGV("%p kUseFastCapture = Always, initFastCapture = true", this); 6003 break; 6004 case FastCapture_Static: 6005 initFastCapture = (mFrameCount * 1000) / mSampleRate < kMinNormalCaptureBufferSizeMs; 6006 ALOGV("%p kUseFastCapture = Static, (%lld * 1000) / %u vs %u, initFastCapture = %d", 6007 this, (long long)mFrameCount, mSampleRate, kMinNormalCaptureBufferSizeMs, 6008 initFastCapture); 6009 break; 6010 // case FastCapture_Dynamic: 6011 } 6012 6013 if (initFastCapture) { 6014 // create a Pipe for FastCapture to write to, and for us and fast tracks to read from 6015 NBAIO_Format format = mInputSource->format(); 6016 // quadruple-buffering of 20 ms each; this ensures we can sleep for 20ms in RecordThread 6017 size_t pipeFramesP2 = roundup(4 * FMS_20 * mSampleRate / 1000); 6018 size_t pipeSize = pipeFramesP2 * Format_frameSize(format); 6019 void *pipeBuffer = nullptr; 6020 const sp<MemoryDealer> roHeap(readOnlyHeap()); 6021 sp<IMemory> pipeMemory; 6022 if ((roHeap == 0) || 6023 (pipeMemory = roHeap->allocate(pipeSize)) == 0 || 6024 (pipeBuffer = pipeMemory->pointer()) == nullptr) { 6025 ALOGE("not enough memory for pipe buffer size=%zu; " 6026 "roHeap=%p, pipeMemory=%p, pipeBuffer=%p; roHeapSize: %lld", 6027 pipeSize, roHeap.get(), pipeMemory.get(), pipeBuffer, 6028 (long long)kRecordThreadReadOnlyHeapSize); 6029 goto failed; 6030 } 6031 // pipe will be shared directly with fast clients, so clear to avoid leaking old information 6032 memset(pipeBuffer, 0, pipeSize); 6033 Pipe *pipe = new Pipe(pipeFramesP2, format, pipeBuffer); 6034 const NBAIO_Format offers[1] = {format}; 6035 size_t numCounterOffers = 0; 6036 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 6037 ALOG_ASSERT(index == 0); 6038 mPipeSink = pipe; 6039 PipeReader *pipeReader = new PipeReader(*pipe); 6040 numCounterOffers = 0; 6041 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 6042 ALOG_ASSERT(index == 0); 6043 mPipeSource = pipeReader; 6044 mPipeFramesP2 = pipeFramesP2; 6045 mPipeMemory = pipeMemory; 6046 6047 // create fast capture 6048 mFastCapture = new FastCapture(); 6049 FastCaptureStateQueue *sq = mFastCapture->sq(); 6050 #ifdef STATE_QUEUE_DUMP 6051 // FIXME 6052 #endif 6053 FastCaptureState *state = sq->begin(); 6054 state->mCblk = NULL; 6055 state->mInputSource = mInputSource.get(); 6056 state->mInputSourceGen++; 6057 state->mPipeSink = pipe; 6058 state->mPipeSinkGen++; 6059 state->mFrameCount = mFrameCount; 6060 state->mCommand = FastCaptureState::COLD_IDLE; 6061 // already done in constructor initialization list 6062 //mFastCaptureFutex = 0; 6063 state->mColdFutexAddr = &mFastCaptureFutex; 6064 state->mColdGen++; 6065 state->mDumpState = &mFastCaptureDumpState; 6066 #ifdef TEE_SINK 6067 // FIXME 6068 #endif 6069 mFastCaptureNBLogWriter = audioFlinger->newWriter_l(kFastCaptureLogSize, "FastCapture"); 6070 state->mNBLogWriter = mFastCaptureNBLogWriter.get(); 6071 sq->end(); 6072 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6073 6074 // start the fast capture 6075 mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO); 6076 pid_t tid = mFastCapture->getTid(); 6077 sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture, false /*forApp*/); 6078 stream()->setHalThreadPriority(kPriorityFastCapture); 6079 #ifdef AUDIO_WATCHDOG 6080 // FIXME 6081 #endif 6082 6083 mFastTrackAvail = true; 6084 } 6085 failed: ; 6086 6087 // FIXME mNormalSource 6088 } 6089 6090 AudioFlinger::RecordThread::~RecordThread() 6091 { 6092 if (mFastCapture != 0) { 6093 FastCaptureStateQueue *sq = mFastCapture->sq(); 6094 FastCaptureState *state = sq->begin(); 6095 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6096 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6097 if (old == -1) { 6098 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6099 } 6100 } 6101 state->mCommand = FastCaptureState::EXIT; 6102 sq->end(); 6103 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_PUSHED); 6104 mFastCapture->join(); 6105 mFastCapture.clear(); 6106 } 6107 mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter); 6108 mAudioFlinger->unregisterWriter(mNBLogWriter); 6109 free(mRsmpInBuffer); 6110 } 6111 6112 void AudioFlinger::RecordThread::onFirstRef() 6113 { 6114 run(mThreadName, PRIORITY_URGENT_AUDIO); 6115 } 6116 6117 void AudioFlinger::RecordThread::preExit() 6118 { 6119 ALOGV(" preExit()"); 6120 Mutex::Autolock _l(mLock); 6121 for (size_t i = 0; i < mTracks.size(); i++) { 6122 sp<RecordTrack> track = mTracks[i]; 6123 track->invalidate(); 6124 } 6125 mActiveTracks.clear(); 6126 mStartStopCond.broadcast(); 6127 } 6128 6129 bool AudioFlinger::RecordThread::threadLoop() 6130 { 6131 nsecs_t lastWarning = 0; 6132 6133 inputStandBy(); 6134 6135 reacquire_wakelock: 6136 sp<RecordTrack> activeTrack; 6137 { 6138 Mutex::Autolock _l(mLock); 6139 acquireWakeLock_l(); 6140 } 6141 6142 // used to request a deferred sleep, to be executed later while mutex is unlocked 6143 uint32_t sleepUs = 0; 6144 6145 // loop while there is work to do 6146 for (;;) { 6147 Vector< sp<EffectChain> > effectChains; 6148 6149 // activeTracks accumulates a copy of a subset of mActiveTracks 6150 Vector< sp<RecordTrack> > activeTracks; 6151 6152 // reference to the (first and only) active fast track 6153 sp<RecordTrack> fastTrack; 6154 6155 // reference to a fast track which is about to be removed 6156 sp<RecordTrack> fastTrackToRemove; 6157 6158 { // scope for mLock 6159 Mutex::Autolock _l(mLock); 6160 6161 processConfigEvents_l(); 6162 6163 // check exitPending here because checkForNewParameters_l() and 6164 // checkForNewParameters_l() can temporarily release mLock 6165 if (exitPending()) { 6166 break; 6167 } 6168 6169 // sleep with mutex unlocked 6170 if (sleepUs > 0) { 6171 ATRACE_BEGIN("sleepC"); 6172 mWaitWorkCV.waitRelative(mLock, microseconds((nsecs_t)sleepUs)); 6173 ATRACE_END(); 6174 sleepUs = 0; 6175 continue; 6176 } 6177 6178 // if no active track(s), then standby and release wakelock 6179 size_t size = mActiveTracks.size(); 6180 if (size == 0) { 6181 standbyIfNotAlreadyInStandby(); 6182 // exitPending() can't become true here 6183 releaseWakeLock_l(); 6184 ALOGV("RecordThread: loop stopping"); 6185 // go to sleep 6186 mWaitWorkCV.wait(mLock); 6187 ALOGV("RecordThread: loop starting"); 6188 goto reacquire_wakelock; 6189 } 6190 6191 bool doBroadcast = false; 6192 bool allStopped = true; 6193 for (size_t i = 0; i < size; ) { 6194 6195 activeTrack = mActiveTracks[i]; 6196 if (activeTrack->isTerminated()) { 6197 if (activeTrack->isFastTrack()) { 6198 ALOG_ASSERT(fastTrackToRemove == 0); 6199 fastTrackToRemove = activeTrack; 6200 } 6201 removeTrack_l(activeTrack); 6202 mActiveTracks.remove(activeTrack); 6203 size--; 6204 continue; 6205 } 6206 6207 TrackBase::track_state activeTrackState = activeTrack->mState; 6208 switch (activeTrackState) { 6209 6210 case TrackBase::PAUSING: 6211 mActiveTracks.remove(activeTrack); 6212 doBroadcast = true; 6213 size--; 6214 continue; 6215 6216 case TrackBase::STARTING_1: 6217 sleepUs = 10000; 6218 i++; 6219 allStopped = false; 6220 continue; 6221 6222 case TrackBase::STARTING_2: 6223 doBroadcast = true; 6224 mStandby = false; 6225 activeTrack->mState = TrackBase::ACTIVE; 6226 allStopped = false; 6227 break; 6228 6229 case TrackBase::ACTIVE: 6230 allStopped = false; 6231 break; 6232 6233 case TrackBase::IDLE: 6234 i++; 6235 continue; 6236 6237 default: 6238 LOG_ALWAYS_FATAL("Unexpected activeTrackState %d", activeTrackState); 6239 } 6240 6241 activeTracks.add(activeTrack); 6242 i++; 6243 6244 if (activeTrack->isFastTrack()) { 6245 ALOG_ASSERT(!mFastTrackAvail); 6246 ALOG_ASSERT(fastTrack == 0); 6247 fastTrack = activeTrack; 6248 } 6249 } 6250 6251 mActiveTracks.updatePowerState(this); 6252 6253 if (allStopped) { 6254 standbyIfNotAlreadyInStandby(); 6255 } 6256 if (doBroadcast) { 6257 mStartStopCond.broadcast(); 6258 } 6259 6260 // sleep if there are no active tracks to process 6261 if (activeTracks.size() == 0) { 6262 if (sleepUs == 0) { 6263 sleepUs = kRecordThreadSleepUs; 6264 } 6265 continue; 6266 } 6267 sleepUs = 0; 6268 6269 lockEffectChains_l(effectChains); 6270 } 6271 6272 // thread mutex is now unlocked, mActiveTracks unknown, activeTracks.size() > 0 6273 6274 size_t size = effectChains.size(); 6275 for (size_t i = 0; i < size; i++) { 6276 // thread mutex is not locked, but effect chain is locked 6277 effectChains[i]->process_l(); 6278 } 6279 6280 // Push a new fast capture state if fast capture is not already running, or cblk change 6281 if (mFastCapture != 0) { 6282 FastCaptureStateQueue *sq = mFastCapture->sq(); 6283 FastCaptureState *state = sq->begin(); 6284 bool didModify = false; 6285 FastCaptureStateQueue::block_t block = FastCaptureStateQueue::BLOCK_UNTIL_PUSHED; 6286 if (state->mCommand != FastCaptureState::READ_WRITE /* FIXME && 6287 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)*/) { 6288 if (state->mCommand == FastCaptureState::COLD_IDLE) { 6289 int32_t old = android_atomic_inc(&mFastCaptureFutex); 6290 if (old == -1) { 6291 (void) syscall(__NR_futex, &mFastCaptureFutex, FUTEX_WAKE_PRIVATE, 1); 6292 } 6293 } 6294 state->mCommand = FastCaptureState::READ_WRITE; 6295 #if 0 // FIXME 6296 mFastCaptureDumpState.increaseSamplingN(mAudioFlinger->isLowRamDevice() ? 6297 FastThreadDumpState::kSamplingNforLowRamDevice : 6298 FastThreadDumpState::kSamplingN); 6299 #endif 6300 didModify = true; 6301 } 6302 audio_track_cblk_t *cblkOld = state->mCblk; 6303 audio_track_cblk_t *cblkNew = fastTrack != 0 ? fastTrack->cblk() : NULL; 6304 if (cblkNew != cblkOld) { 6305 state->mCblk = cblkNew; 6306 // block until acked if removing a fast track 6307 if (cblkOld != NULL) { 6308 block = FastCaptureStateQueue::BLOCK_UNTIL_ACKED; 6309 } 6310 didModify = true; 6311 } 6312 sq->end(didModify); 6313 if (didModify) { 6314 sq->push(block); 6315 #if 0 6316 if (kUseFastCapture == FastCapture_Dynamic) { 6317 mNormalSource = mPipeSource; 6318 } 6319 #endif 6320 } 6321 } 6322 6323 // now run the fast track destructor with thread mutex unlocked 6324 fastTrackToRemove.clear(); 6325 6326 // Read from HAL to keep up with fastest client if multiple active tracks, not slowest one. 6327 // Only the client(s) that are too slow will overrun. But if even the fastest client is too 6328 // slow, then this RecordThread will overrun by not calling HAL read often enough. 6329 // If destination is non-contiguous, first read past the nominal end of buffer, then 6330 // copy to the right place. Permitted because mRsmpInBuffer was over-allocated. 6331 6332 int32_t rear = mRsmpInRear & (mRsmpInFramesP2 - 1); 6333 ssize_t framesRead; 6334 6335 // If an NBAIO source is present, use it to read the normal capture's data 6336 if (mPipeSource != 0) { 6337 size_t framesToRead = mBufferSize / mFrameSize; 6338 framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2); 6339 framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize, 6340 framesToRead); 6341 // since pipe is non-blocking, simulate blocking input by waiting for 1/2 of 6342 // buffer size or at least for 20ms. 6343 size_t sleepFrames = max( 6344 min(mPipeFramesP2, mRsmpInFramesP2) / 2, FMS_20 * mSampleRate / 1000); 6345 if (framesRead <= (ssize_t) sleepFrames) { 6346 sleepUs = (sleepFrames * 1000000LL) / mSampleRate; 6347 } 6348 if (framesRead < 0) { 6349 status_t status = (status_t) framesRead; 6350 switch (status) { 6351 case OVERRUN: 6352 ALOGW("overrun on read from pipe"); 6353 framesRead = 0; 6354 break; 6355 case NEGOTIATE: 6356 ALOGE("re-negotiation is needed"); 6357 framesRead = -1; // Will cause an attempt to recover. 6358 break; 6359 default: 6360 ALOGE("unknown error %d on read from pipe", status); 6361 break; 6362 } 6363 } 6364 // otherwise use the HAL / AudioStreamIn directly 6365 } else { 6366 ATRACE_BEGIN("read"); 6367 size_t bytesRead; 6368 status_t result = mInput->stream->read( 6369 (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize, &bytesRead); 6370 ATRACE_END(); 6371 if (result < 0) { 6372 framesRead = result; 6373 } else { 6374 framesRead = bytesRead / mFrameSize; 6375 } 6376 } 6377 6378 // Update server timestamp with server stats 6379 // systemTime() is optional if the hardware supports timestamps. 6380 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += framesRead; 6381 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6382 6383 // Update server timestamp with kernel stats 6384 if (mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) { 6385 int64_t position, time; 6386 int ret = mInput->stream->getCapturePosition(&position, &time); 6387 if (ret == NO_ERROR) { 6388 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position; 6389 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time; 6390 // Note: In general record buffers should tend to be empty in 6391 // a properly running pipeline. 6392 // 6393 // Also, it is not advantageous to call get_presentation_position during the read 6394 // as the read obtains a lock, preventing the timestamp call from executing. 6395 } 6396 } 6397 // Use this to track timestamp information 6398 // ALOGD("%s", mTimestamp.toString().c_str()); 6399 6400 if (framesRead < 0 || (framesRead == 0 && mPipeSource == 0)) { 6401 ALOGE("read failed: framesRead=%zd", framesRead); 6402 // Force input into standby so that it tries to recover at next read attempt 6403 inputStandBy(); 6404 sleepUs = kRecordThreadSleepUs; 6405 } 6406 if (framesRead <= 0) { 6407 goto unlock; 6408 } 6409 ALOG_ASSERT(framesRead > 0); 6410 6411 if (mTeeSink != 0) { 6412 (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead); 6413 } 6414 // If destination is non-contiguous, we now correct for reading past end of buffer. 6415 { 6416 size_t part1 = mRsmpInFramesP2 - rear; 6417 if ((size_t) framesRead > part1) { 6418 memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize, 6419 (framesRead - part1) * mFrameSize); 6420 } 6421 } 6422 rear = mRsmpInRear += framesRead; 6423 6424 size = activeTracks.size(); 6425 // loop over each active track 6426 for (size_t i = 0; i < size; i++) { 6427 activeTrack = activeTracks[i]; 6428 6429 // skip fast tracks, as those are handled directly by FastCapture 6430 if (activeTrack->isFastTrack()) { 6431 continue; 6432 } 6433 6434 // TODO: This code probably should be moved to RecordTrack. 6435 // TODO: Update the activeTrack buffer converter in case of reconfigure. 6436 6437 enum { 6438 OVERRUN_UNKNOWN, 6439 OVERRUN_TRUE, 6440 OVERRUN_FALSE 6441 } overrun = OVERRUN_UNKNOWN; 6442 6443 // loop over getNextBuffer to handle circular sink 6444 for (;;) { 6445 6446 activeTrack->mSink.frameCount = ~0; 6447 status_t status = activeTrack->getNextBuffer(&activeTrack->mSink); 6448 size_t framesOut = activeTrack->mSink.frameCount; 6449 LOG_ALWAYS_FATAL_IF((status == OK) != (framesOut > 0)); 6450 6451 // check available frames and handle overrun conditions 6452 // if the record track isn't draining fast enough. 6453 bool hasOverrun; 6454 size_t framesIn; 6455 activeTrack->mResamplerBufferProvider->sync(&framesIn, &hasOverrun); 6456 if (hasOverrun) { 6457 overrun = OVERRUN_TRUE; 6458 } 6459 if (framesOut == 0 || framesIn == 0) { 6460 break; 6461 } 6462 6463 // Don't allow framesOut to be larger than what is possible with resampling 6464 // from framesIn. 6465 // This isn't strictly necessary but helps limit buffer resizing in 6466 // RecordBufferConverter. TODO: remove when no longer needed. 6467 framesOut = min(framesOut, 6468 destinationFramesPossible( 6469 framesIn, mSampleRate, activeTrack->mSampleRate)); 6470 // process frames from the RecordThread buffer provider to the RecordTrack buffer 6471 framesOut = activeTrack->mRecordBufferConverter->convert( 6472 activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut); 6473 6474 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) { 6475 overrun = OVERRUN_FALSE; 6476 } 6477 6478 if (activeTrack->mFramesToDrop == 0) { 6479 if (framesOut > 0) { 6480 activeTrack->mSink.frameCount = framesOut; 6481 activeTrack->releaseBuffer(&activeTrack->mSink); 6482 } 6483 } else { 6484 // FIXME could do a partial drop of framesOut 6485 if (activeTrack->mFramesToDrop > 0) { 6486 activeTrack->mFramesToDrop -= framesOut; 6487 if (activeTrack->mFramesToDrop <= 0) { 6488 activeTrack->clearSyncStartEvent(); 6489 } 6490 } else { 6491 activeTrack->mFramesToDrop += framesOut; 6492 if (activeTrack->mFramesToDrop >= 0 || activeTrack->mSyncStartEvent == 0 || 6493 activeTrack->mSyncStartEvent->isCancelled()) { 6494 ALOGW("Synced record %s, session %d, trigger session %d", 6495 (activeTrack->mFramesToDrop >= 0) ? "timed out" : "cancelled", 6496 activeTrack->sessionId(), 6497 (activeTrack->mSyncStartEvent != 0) ? 6498 activeTrack->mSyncStartEvent->triggerSession() : 6499 AUDIO_SESSION_NONE); 6500 activeTrack->clearSyncStartEvent(); 6501 } 6502 } 6503 } 6504 6505 if (framesOut == 0) { 6506 break; 6507 } 6508 } 6509 6510 switch (overrun) { 6511 case OVERRUN_TRUE: 6512 // client isn't retrieving buffers fast enough 6513 if (!activeTrack->setOverflow()) { 6514 nsecs_t now = systemTime(); 6515 // FIXME should lastWarning per track? 6516 if ((now - lastWarning) > kWarningThrottleNs) { 6517 ALOGW("RecordThread: buffer overflow"); 6518 lastWarning = now; 6519 } 6520 } 6521 break; 6522 case OVERRUN_FALSE: 6523 activeTrack->clearOverflow(); 6524 break; 6525 case OVERRUN_UNKNOWN: 6526 break; 6527 } 6528 6529 // update frame information and push timestamp out 6530 activeTrack->updateTrackFrameInfo( 6531 activeTrack->mServerProxy->framesReleased(), 6532 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER], 6533 mSampleRate, mTimestamp); 6534 } 6535 6536 unlock: 6537 // enable changes in effect chain 6538 unlockEffectChains(effectChains); 6539 // effectChains doesn't need to be cleared, since it is cleared by destructor at scope end 6540 } 6541 6542 standbyIfNotAlreadyInStandby(); 6543 6544 { 6545 Mutex::Autolock _l(mLock); 6546 for (size_t i = 0; i < mTracks.size(); i++) { 6547 sp<RecordTrack> track = mTracks[i]; 6548 track->invalidate(); 6549 } 6550 mActiveTracks.clear(); 6551 mStartStopCond.broadcast(); 6552 } 6553 6554 releaseWakeLock(); 6555 6556 ALOGV("RecordThread %p exiting", this); 6557 return false; 6558 } 6559 6560 void AudioFlinger::RecordThread::standbyIfNotAlreadyInStandby() 6561 { 6562 if (!mStandby) { 6563 inputStandBy(); 6564 mStandby = true; 6565 } 6566 } 6567 6568 void AudioFlinger::RecordThread::inputStandBy() 6569 { 6570 // Idle the fast capture if it's currently running 6571 if (mFastCapture != 0) { 6572 FastCaptureStateQueue *sq = mFastCapture->sq(); 6573 FastCaptureState *state = sq->begin(); 6574 if (!(state->mCommand & FastCaptureState::IDLE)) { 6575 state->mCommand = FastCaptureState::COLD_IDLE; 6576 state->mColdFutexAddr = &mFastCaptureFutex; 6577 state->mColdGen++; 6578 mFastCaptureFutex = 0; 6579 sq->end(); 6580 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 6581 sq->push(FastCaptureStateQueue::BLOCK_UNTIL_ACKED); 6582 #if 0 6583 if (kUseFastCapture == FastCapture_Dynamic) { 6584 // FIXME 6585 } 6586 #endif 6587 #ifdef AUDIO_WATCHDOG 6588 // FIXME 6589 #endif 6590 } else { 6591 sq->end(false /*didModify*/); 6592 } 6593 } 6594 status_t result = mInput->stream->standby(); 6595 ALOGE_IF(result != OK, "Error when putting input stream into standby: %d", result); 6596 6597 // If going into standby, flush the pipe source. 6598 if (mPipeSource.get() != nullptr) { 6599 const ssize_t flushed = mPipeSource->flush(); 6600 if (flushed > 0) { 6601 ALOGV("Input standby flushed PipeSource %zd frames", flushed); 6602 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_SERVER] += flushed; 6603 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] = systemTime(); 6604 } 6605 } 6606 } 6607 6608 // RecordThread::createRecordTrack_l() must be called with AudioFlinger::mLock held 6609 sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6610 const sp<AudioFlinger::Client>& client, 6611 uint32_t sampleRate, 6612 audio_format_t format, 6613 audio_channel_mask_t channelMask, 6614 size_t *pFrameCount, 6615 audio_session_t sessionId, 6616 size_t *notificationFrames, 6617 uid_t uid, 6618 audio_input_flags_t *flags, 6619 pid_t tid, 6620 status_t *status, 6621 audio_port_handle_t portId) 6622 { 6623 size_t frameCount = *pFrameCount; 6624 sp<RecordTrack> track; 6625 status_t lStatus; 6626 audio_input_flags_t inputFlags = mInput->flags; 6627 6628 // special case for FAST flag considered OK if fast capture is present 6629 if (hasFastCapture()) { 6630 inputFlags = (audio_input_flags_t)(inputFlags | AUDIO_INPUT_FLAG_FAST); 6631 } 6632 6633 // Check if requested flags are compatible with output stream flags 6634 if ((*flags & inputFlags) != *flags) { 6635 ALOGW("createRecordTrack_l(): mismatch between requested flags (%08x) and" 6636 " input flags (%08x)", 6637 *flags, inputFlags); 6638 *flags = (audio_input_flags_t)(*flags & inputFlags); 6639 } 6640 6641 // client expresses a preference for FAST, but we get the final say 6642 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6643 if ( 6644 // we formerly checked for a callback handler (non-0 tid), 6645 // but that is no longer required for TRANSFER_OBTAIN mode 6646 // 6647 // frame count is not specified, or is exactly the pipe depth 6648 ((frameCount == 0) || (frameCount == mPipeFramesP2)) && 6649 // PCM data 6650 audio_is_linear_pcm(format) && 6651 // hardware format 6652 (format == mFormat) && 6653 // hardware channel mask 6654 (channelMask == mChannelMask) && 6655 // hardware sample rate 6656 (sampleRate == mSampleRate) && 6657 // record thread has an associated fast capture 6658 hasFastCapture() && 6659 // there are sufficient fast track slots available 6660 mFastTrackAvail 6661 ) { 6662 // check compatibility with audio effects. 6663 Mutex::Autolock _l(mLock); 6664 // Do not accept FAST flag if the session has software effects 6665 sp<EffectChain> chain = getEffectChain_l(sessionId); 6666 if (chain != 0) { 6667 audio_input_flags_t old = *flags; 6668 chain->checkInputFlagCompatibility(flags); 6669 if (old != *flags) { 6670 ALOGV("%p AUDIO_INPUT_FLAGS denied by effect old=%#x new=%#x", 6671 this, (int)old, (int)*flags); 6672 } 6673 } 6674 ALOGV_IF((*flags & AUDIO_INPUT_FLAG_FAST) != 0, 6675 "%p AUDIO_INPUT_FLAG_FAST accepted: frameCount=%zu mFrameCount=%zu", 6676 this, frameCount, mFrameCount); 6677 } else { 6678 ALOGV("%p AUDIO_INPUT_FLAG_FAST denied: frameCount=%zu mFrameCount=%zu mPipeFramesP2=%zu " 6679 "format=%#x isLinear=%d mFormat=%#x channelMask=%#x sampleRate=%u mSampleRate=%u " 6680 "hasFastCapture=%d tid=%d mFastTrackAvail=%d", 6681 this, frameCount, mFrameCount, mPipeFramesP2, 6682 format, audio_is_linear_pcm(format), mFormat, channelMask, sampleRate, mSampleRate, 6683 hasFastCapture(), tid, mFastTrackAvail); 6684 *flags = (audio_input_flags_t)(*flags & ~AUDIO_INPUT_FLAG_FAST); 6685 } 6686 } 6687 6688 // compute track buffer size in frames, and suggest the notification frame count 6689 if (*flags & AUDIO_INPUT_FLAG_FAST) { 6690 // fast track: frame count is exactly the pipe depth 6691 frameCount = mPipeFramesP2; 6692 // ignore requested notificationFrames, and always notify exactly once every HAL buffer 6693 *notificationFrames = mFrameCount; 6694 } else { 6695 // not fast track: max notification period is resampled equivalent of one HAL buffer time 6696 // or 20 ms if there is a fast capture 6697 // TODO This could be a roundupRatio inline, and const 6698 size_t maxNotificationFrames = ((int64_t) (hasFastCapture() ? mSampleRate/50 : mFrameCount) 6699 * sampleRate + mSampleRate - 1) / mSampleRate; 6700 // minimum number of notification periods is at least kMinNotifications, 6701 // and at least kMinMs rounded up to a whole notification period (minNotificationsByMs) 6702 static const size_t kMinNotifications = 3; 6703 static const uint32_t kMinMs = 30; 6704 // TODO This could be a roundupRatio inline 6705 const size_t minFramesByMs = (sampleRate * kMinMs + 1000 - 1) / 1000; 6706 // TODO This could be a roundupRatio inline 6707 const size_t minNotificationsByMs = (minFramesByMs + maxNotificationFrames - 1) / 6708 maxNotificationFrames; 6709 const size_t minFrameCount = maxNotificationFrames * 6710 max(kMinNotifications, minNotificationsByMs); 6711 frameCount = max(frameCount, minFrameCount); 6712 if (*notificationFrames == 0 || *notificationFrames > maxNotificationFrames) { 6713 *notificationFrames = maxNotificationFrames; 6714 } 6715 } 6716 *pFrameCount = frameCount; 6717 6718 lStatus = initCheck(); 6719 if (lStatus != NO_ERROR) { 6720 ALOGE("createRecordTrack_l() audio driver not initialized"); 6721 goto Exit; 6722 } 6723 6724 { // scope for mLock 6725 Mutex::Autolock _l(mLock); 6726 6727 track = new RecordTrack(this, client, sampleRate, 6728 format, channelMask, frameCount, 6729 nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, uid, 6730 *flags, TrackBase::TYPE_DEFAULT, portId); 6731 6732 lStatus = track->initCheck(); 6733 if (lStatus != NO_ERROR) { 6734 ALOGE("createRecordTrack_l() initCheck failed %d; no control block?", lStatus); 6735 // track must be cleared from the caller as the caller has the AF lock 6736 goto Exit; 6737 } 6738 mTracks.add(track); 6739 6740 if ((*flags & AUDIO_INPUT_FLAG_FAST) && (tid != -1)) { 6741 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 6742 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 6743 // so ask activity manager to do this on our behalf 6744 sendPrioConfigEvent_l(callingPid, tid, kPriorityAudioApp, true /*forApp*/); 6745 } 6746 } 6747 6748 lStatus = NO_ERROR; 6749 6750 Exit: 6751 *status = lStatus; 6752 return track; 6753 } 6754 6755 status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6756 AudioSystem::sync_event_t event, 6757 audio_session_t triggerSession) 6758 { 6759 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6760 sp<ThreadBase> strongMe = this; 6761 status_t status = NO_ERROR; 6762 6763 if (event == AudioSystem::SYNC_EVENT_NONE) { 6764 recordTrack->clearSyncStartEvent(); 6765 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6766 recordTrack->mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6767 triggerSession, 6768 recordTrack->sessionId(), 6769 syncStartEventCallback, 6770 recordTrack); 6771 // Sync event can be cancelled by the trigger session if the track is not in a 6772 // compatible state in which case we start record immediately 6773 if (recordTrack->mSyncStartEvent->isCancelled()) { 6774 recordTrack->clearSyncStartEvent(); 6775 } else { 6776 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6777 recordTrack->mFramesToDrop = - 6778 ((AudioSystem::kSyncRecordStartTimeOutMs * recordTrack->mSampleRate) / 1000); 6779 } 6780 } 6781 6782 { 6783 // This section is a rendezvous between binder thread executing start() and RecordThread 6784 AutoMutex lock(mLock); 6785 if (mActiveTracks.indexOf(recordTrack) >= 0) { 6786 if (recordTrack->mState == TrackBase::PAUSING) { 6787 ALOGV("active record track PAUSING -> ACTIVE"); 6788 recordTrack->mState = TrackBase::ACTIVE; 6789 } else { 6790 ALOGV("active record track state %d", recordTrack->mState); 6791 } 6792 return status; 6793 } 6794 6795 // TODO consider other ways of handling this, such as changing the state to :STARTING and 6796 // adding the track to mActiveTracks after returning from AudioSystem::startInput(), 6797 // or using a separate command thread 6798 recordTrack->mState = TrackBase::STARTING_1; 6799 mActiveTracks.add(recordTrack); 6800 status_t status = NO_ERROR; 6801 if (recordTrack->isExternalTrack()) { 6802 mLock.unlock(); 6803 status = AudioSystem::startInput(mId, recordTrack->sessionId()); 6804 mLock.lock(); 6805 // FIXME should verify that recordTrack is still in mActiveTracks 6806 if (status != NO_ERROR) { 6807 mActiveTracks.remove(recordTrack); 6808 recordTrack->clearSyncStartEvent(); 6809 ALOGV("RecordThread::start error %d", status); 6810 return status; 6811 } 6812 } 6813 // Catch up with current buffer indices if thread is already running. 6814 // This is what makes a new client discard all buffered data. If the track's mRsmpInFront 6815 // was initialized to some value closer to the thread's mRsmpInFront, then the track could 6816 // see previously buffered data before it called start(), but with greater risk of overrun. 6817 6818 recordTrack->mResamplerBufferProvider->reset(); 6819 // clear any converter state as new data will be discontinuous 6820 recordTrack->mRecordBufferConverter->reset(); 6821 recordTrack->mState = TrackBase::STARTING_2; 6822 // signal thread to start 6823 mWaitWorkCV.broadcast(); 6824 if (mActiveTracks.indexOf(recordTrack) < 0) { 6825 ALOGV("Record failed to start"); 6826 status = BAD_VALUE; 6827 goto startError; 6828 } 6829 return status; 6830 } 6831 6832 startError: 6833 if (recordTrack->isExternalTrack()) { 6834 AudioSystem::stopInput(mId, recordTrack->sessionId()); 6835 } 6836 recordTrack->clearSyncStartEvent(); 6837 // FIXME I wonder why we do not reset the state here? 6838 return status; 6839 } 6840 6841 void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6842 { 6843 sp<SyncEvent> strongEvent = event.promote(); 6844 6845 if (strongEvent != 0) { 6846 sp<RefBase> ptr = strongEvent->cookie().promote(); 6847 if (ptr != 0) { 6848 RecordTrack *recordTrack = (RecordTrack *)ptr.get(); 6849 recordTrack->handleSyncStartEvent(strongEvent); 6850 } 6851 } 6852 } 6853 6854 bool AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6855 ALOGV("RecordThread::stop"); 6856 AutoMutex _l(mLock); 6857 if (mActiveTracks.indexOf(recordTrack) < 0 || recordTrack->mState == TrackBase::PAUSING) { 6858 return false; 6859 } 6860 // note that threadLoop may still be processing the track at this point [without lock] 6861 recordTrack->mState = TrackBase::PAUSING; 6862 // signal thread to stop 6863 mWaitWorkCV.broadcast(); 6864 // do not wait for mStartStopCond if exiting 6865 if (exitPending()) { 6866 return true; 6867 } 6868 // FIXME incorrect usage of wait: no explicit predicate or loop 6869 mStartStopCond.wait(mLock); 6870 // if we have been restarted, recordTrack is in mActiveTracks here 6871 if (exitPending() || mActiveTracks.indexOf(recordTrack) < 0) { 6872 ALOGV("Record stopped OK"); 6873 return true; 6874 } 6875 return false; 6876 } 6877 6878 bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 6879 { 6880 return false; 6881 } 6882 6883 status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event __unused) 6884 { 6885 #if 0 // This branch is currently dead code, but is preserved in case it will be needed in future 6886 if (!isValidSyncEvent(event)) { 6887 return BAD_VALUE; 6888 } 6889 6890 audio_session_t eventSession = event->triggerSession(); 6891 status_t ret = NAME_NOT_FOUND; 6892 6893 Mutex::Autolock _l(mLock); 6894 6895 for (size_t i = 0; i < mTracks.size(); i++) { 6896 sp<RecordTrack> track = mTracks[i]; 6897 if (eventSession == track->sessionId()) { 6898 (void) track->setSyncEvent(event); 6899 ret = NO_ERROR; 6900 } 6901 } 6902 return ret; 6903 #else 6904 return BAD_VALUE; 6905 #endif 6906 } 6907 6908 // destroyTrack_l() must be called with ThreadBase::mLock held 6909 void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6910 { 6911 track->terminate(); 6912 track->mState = TrackBase::STOPPED; 6913 // active tracks are removed by threadLoop() 6914 if (mActiveTracks.indexOf(track) < 0) { 6915 removeTrack_l(track); 6916 } 6917 } 6918 6919 void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6920 { 6921 String8 result; 6922 track->appendDump(result, false /* active */); 6923 mLocalLog.log("removeTrack_l (%p) %s", track.get(), result.string()); 6924 6925 mTracks.remove(track); 6926 // need anything related to effects here? 6927 if (track->isFastTrack()) { 6928 ALOG_ASSERT(!mFastTrackAvail); 6929 mFastTrackAvail = true; 6930 } 6931 } 6932 6933 void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6934 { 6935 dumpInternals(fd, args); 6936 dumpTracks(fd, args); 6937 dumpEffectChains(fd, args); 6938 dprintf(fd, " Local log:\n"); 6939 mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */); 6940 } 6941 6942 void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6943 { 6944 dumpBase(fd, args); 6945 6946 AudioStreamIn *input = mInput; 6947 audio_input_flags_t flags = input != NULL ? input->flags : AUDIO_INPUT_FLAG_NONE; 6948 dprintf(fd, " AudioStreamIn: %p flags %#x (%s)\n", 6949 input, flags, inputFlagsToString(flags).c_str()); 6950 if (mActiveTracks.size() == 0) { 6951 dprintf(fd, " No active record clients\n"); 6952 } 6953 6954 if (input != nullptr) { 6955 dprintf(fd, " Hal stream dump:\n"); 6956 (void)input->stream->dump(fd); 6957 } 6958 6959 dprintf(fd, " Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no"); 6960 dprintf(fd, " Fast track available: %s\n", mFastTrackAvail ? "yes" : "no"); 6961 6962 // Make a non-atomic copy of fast capture dump state so it won't change underneath us 6963 // while we are dumping it. It may be inconsistent, but it won't mutate! 6964 // This is a large object so we place it on the heap. 6965 // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages. 6966 const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState); 6967 copy->dump(fd); 6968 delete copy; 6969 } 6970 6971 void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused) 6972 { 6973 String8 result; 6974 size_t numtracks = mTracks.size(); 6975 size_t numactive = mActiveTracks.size(); 6976 size_t numactiveseen = 0; 6977 dprintf(fd, " %zu Tracks", numtracks); 6978 const char *prefix = " "; 6979 if (numtracks) { 6980 dprintf(fd, " of which %zu are active\n", numactive); 6981 result.append(prefix); 6982 RecordTrack::appendDumpHeader(result); 6983 for (size_t i = 0; i < numtracks ; ++i) { 6984 sp<RecordTrack> track = mTracks[i]; 6985 if (track != 0) { 6986 bool active = mActiveTracks.indexOf(track) >= 0; 6987 if (active) { 6988 numactiveseen++; 6989 } 6990 result.append(prefix); 6991 track->appendDump(result, active); 6992 } 6993 } 6994 } else { 6995 dprintf(fd, "\n"); 6996 } 6997 6998 if (numactiveseen != numactive) { 6999 result.append(" The following tracks are in the active list but" 7000 " not in the track list\n"); 7001 result.append(prefix); 7002 RecordTrack::appendDumpHeader(result); 7003 for (size_t i = 0; i < numactive; ++i) { 7004 sp<RecordTrack> track = mActiveTracks[i]; 7005 if (mTracks.indexOf(track) < 0) { 7006 result.append(prefix); 7007 track->appendDump(result, true /* active */); 7008 } 7009 } 7010 7011 } 7012 write(fd, result.string(), result.size()); 7013 } 7014 7015 7016 void AudioFlinger::RecordThread::ResamplerBufferProvider::reset() 7017 { 7018 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 7019 RecordThread *recordThread = (RecordThread *) threadBase.get(); 7020 mRsmpInFront = recordThread->mRsmpInRear; 7021 mRsmpInUnrel = 0; 7022 } 7023 7024 void AudioFlinger::RecordThread::ResamplerBufferProvider::sync( 7025 size_t *framesAvailable, bool *hasOverrun) 7026 { 7027 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 7028 RecordThread *recordThread = (RecordThread *) threadBase.get(); 7029 const int32_t rear = recordThread->mRsmpInRear; 7030 const int32_t front = mRsmpInFront; 7031 const ssize_t filled = rear - front; 7032 7033 size_t framesIn; 7034 bool overrun = false; 7035 if (filled < 0) { 7036 // should not happen, but treat like a massive overrun and re-sync 7037 framesIn = 0; 7038 mRsmpInFront = rear; 7039 overrun = true; 7040 } else if ((size_t) filled <= recordThread->mRsmpInFrames) { 7041 framesIn = (size_t) filled; 7042 } else { 7043 // client is not keeping up with server, but give it latest data 7044 framesIn = recordThread->mRsmpInFrames; 7045 mRsmpInFront = /* front = */ rear - framesIn; 7046 overrun = true; 7047 } 7048 if (framesAvailable != NULL) { 7049 *framesAvailable = framesIn; 7050 } 7051 if (hasOverrun != NULL) { 7052 *hasOverrun = overrun; 7053 } 7054 } 7055 7056 // AudioBufferProvider interface 7057 status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer( 7058 AudioBufferProvider::Buffer* buffer) 7059 { 7060 sp<ThreadBase> threadBase = mRecordTrack->mThread.promote(); 7061 if (threadBase == 0) { 7062 buffer->frameCount = 0; 7063 buffer->raw = NULL; 7064 return NOT_ENOUGH_DATA; 7065 } 7066 RecordThread *recordThread = (RecordThread *) threadBase.get(); 7067 int32_t rear = recordThread->mRsmpInRear; 7068 int32_t front = mRsmpInFront; 7069 ssize_t filled = rear - front; 7070 // FIXME should not be P2 (don't want to increase latency) 7071 // FIXME if client not keeping up, discard 7072 LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames)); 7073 // 'filled' may be non-contiguous, so return only the first contiguous chunk 7074 front &= recordThread->mRsmpInFramesP2 - 1; 7075 size_t part1 = recordThread->mRsmpInFramesP2 - front; 7076 if (part1 > (size_t) filled) { 7077 part1 = filled; 7078 } 7079 size_t ask = buffer->frameCount; 7080 ALOG_ASSERT(ask > 0); 7081 if (part1 > ask) { 7082 part1 = ask; 7083 } 7084 if (part1 == 0) { 7085 // out of data is fine since the resampler will return a short-count. 7086 buffer->raw = NULL; 7087 buffer->frameCount = 0; 7088 mRsmpInUnrel = 0; 7089 return NOT_ENOUGH_DATA; 7090 } 7091 7092 buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize; 7093 buffer->frameCount = part1; 7094 mRsmpInUnrel = part1; 7095 return NO_ERROR; 7096 } 7097 7098 // AudioBufferProvider interface 7099 void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer( 7100 AudioBufferProvider::Buffer* buffer) 7101 { 7102 size_t stepCount = buffer->frameCount; 7103 if (stepCount == 0) { 7104 return; 7105 } 7106 ALOG_ASSERT(stepCount <= mRsmpInUnrel); 7107 mRsmpInUnrel -= stepCount; 7108 mRsmpInFront += stepCount; 7109 buffer->raw = NULL; 7110 buffer->frameCount = 0; 7111 } 7112 7113 void AudioFlinger::RecordThread::checkBtNrec() 7114 { 7115 Mutex::Autolock _l(mLock); 7116 checkBtNrec_l(); 7117 } 7118 7119 void AudioFlinger::RecordThread::checkBtNrec_l() 7120 { 7121 // disable AEC and NS if the device is a BT SCO headset supporting those 7122 // pre processings 7123 bool suspend = audio_is_bluetooth_sco_device(mInDevice) && 7124 mAudioFlinger->btNrecIsOff(); 7125 if (mBtNrecSuspended.exchange(suspend) != suspend) { 7126 for (size_t i = 0; i < mEffectChains.size(); i++) { 7127 setEffectSuspended_l(FX_IID_AEC, suspend, mEffectChains[i]->sessionId()); 7128 setEffectSuspended_l(FX_IID_NS, suspend, mEffectChains[i]->sessionId()); 7129 } 7130 } 7131 } 7132 7133 7134 bool AudioFlinger::RecordThread::checkForNewParameter_l(const String8& keyValuePair, 7135 status_t& status) 7136 { 7137 bool reconfig = false; 7138 7139 status = NO_ERROR; 7140 7141 audio_format_t reqFormat = mFormat; 7142 uint32_t samplingRate = mSampleRate; 7143 // TODO this may change if we want to support capture from HDMI PCM multi channel (e.g on TVs). 7144 audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(mChannelCount); 7145 7146 AudioParameter param = AudioParameter(keyValuePair); 7147 int value; 7148 7149 // scope for AutoPark extends to end of method 7150 AutoPark<FastCapture> park(mFastCapture); 7151 7152 // TODO Investigate when this code runs. Check with audio policy when a sample rate and 7153 // channel count change can be requested. Do we mandate the first client defines the 7154 // HAL sampling rate and channel count or do we allow changes on the fly? 7155 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 7156 samplingRate = value; 7157 reconfig = true; 7158 } 7159 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 7160 if (!audio_is_linear_pcm((audio_format_t) value)) { 7161 status = BAD_VALUE; 7162 } else { 7163 reqFormat = (audio_format_t) value; 7164 reconfig = true; 7165 } 7166 } 7167 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 7168 audio_channel_mask_t mask = (audio_channel_mask_t) value; 7169 if (!audio_is_input_channel(mask) || 7170 audio_channel_count_from_in_mask(mask) > FCC_8) { 7171 status = BAD_VALUE; 7172 } else { 7173 channelMask = mask; 7174 reconfig = true; 7175 } 7176 } 7177 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 7178 // do not accept frame count changes if tracks are open as the track buffer 7179 // size depends on frame count and correct behavior would not be guaranteed 7180 // if frame count is changed after track creation 7181 if (mActiveTracks.size() > 0) { 7182 status = INVALID_OPERATION; 7183 } else { 7184 reconfig = true; 7185 } 7186 } 7187 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7188 // forward device change to effects that have requested to be 7189 // aware of attached audio device. 7190 for (size_t i = 0; i < mEffectChains.size(); i++) { 7191 mEffectChains[i]->setDevice_l(value); 7192 } 7193 7194 // store input device and output device but do not forward output device to audio HAL. 7195 // Note that status is ignored by the caller for output device 7196 // (see AudioFlinger::setParameters() 7197 if (audio_is_output_devices(value)) { 7198 mOutDevice = value; 7199 status = BAD_VALUE; 7200 } else { 7201 mInDevice = value; 7202 if (value != AUDIO_DEVICE_NONE) { 7203 mPrevInDevice = value; 7204 } 7205 checkBtNrec_l(); 7206 } 7207 } 7208 if (param.getInt(String8(AudioParameter::keyInputSource), value) == NO_ERROR && 7209 mAudioSource != (audio_source_t)value) { 7210 // forward device change to effects that have requested to be 7211 // aware of attached audio device. 7212 for (size_t i = 0; i < mEffectChains.size(); i++) { 7213 mEffectChains[i]->setAudioSource_l((audio_source_t)value); 7214 } 7215 mAudioSource = (audio_source_t)value; 7216 } 7217 7218 if (status == NO_ERROR) { 7219 status = mInput->stream->setParameters(keyValuePair); 7220 if (status == INVALID_OPERATION) { 7221 inputStandBy(); 7222 status = mInput->stream->setParameters(keyValuePair); 7223 } 7224 if (reconfig) { 7225 if (status == BAD_VALUE) { 7226 uint32_t sRate; 7227 audio_channel_mask_t channelMask; 7228 audio_format_t format; 7229 if (mInput->stream->getAudioProperties(&sRate, &channelMask, &format) == OK && 7230 audio_is_linear_pcm(format) && audio_is_linear_pcm(reqFormat) && 7231 sRate <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate) && 7232 audio_channel_count_from_in_mask(channelMask) <= FCC_8) { 7233 status = NO_ERROR; 7234 } 7235 } 7236 if (status == NO_ERROR) { 7237 readInputParameters_l(); 7238 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7239 } 7240 } 7241 } 7242 7243 return reconfig; 7244 } 7245 7246 String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 7247 { 7248 Mutex::Autolock _l(mLock); 7249 if (initCheck() == NO_ERROR) { 7250 String8 out_s8; 7251 if (mInput->stream->getParameters(keys, &out_s8) == OK) { 7252 return out_s8; 7253 } 7254 } 7255 return String8(); 7256 } 7257 7258 void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7259 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7260 7261 desc->mIoHandle = mId; 7262 7263 switch (event) { 7264 case AUDIO_INPUT_OPENED: 7265 case AUDIO_INPUT_REGISTERED: 7266 case AUDIO_INPUT_CONFIG_CHANGED: 7267 desc->mPatch = mPatch; 7268 desc->mChannelMask = mChannelMask; 7269 desc->mSamplingRate = mSampleRate; 7270 desc->mFormat = mFormat; 7271 desc->mFrameCount = mFrameCount; 7272 desc->mFrameCountHAL = mFrameCount; 7273 desc->mLatency = 0; 7274 break; 7275 7276 case AUDIO_INPUT_CLOSED: 7277 default: 7278 break; 7279 } 7280 mAudioFlinger->ioConfigChanged(event, desc, pid); 7281 } 7282 7283 void AudioFlinger::RecordThread::readInputParameters_l() 7284 { 7285 status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat); 7286 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result); 7287 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 7288 LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_8, "HAL channel count %d > %d", mChannelCount, FCC_8); 7289 mFormat = mHALFormat; 7290 LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat); 7291 result = mInput->stream->getFrameSize(&mFrameSize); 7292 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result); 7293 result = mInput->stream->getBufferSize(&mBufferSize); 7294 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result); 7295 mFrameCount = mBufferSize / mFrameSize; 7296 ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%lld, " 7297 "mBufferSize=%lld, mFrameCount=%lld", 7298 this, mChannelCount, mFormat, (long long)mFrameSize, (long long)mBufferSize, 7299 (long long)mFrameCount); 7300 // This is the formula for calculating the temporary buffer size. 7301 // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to 7302 // 1 full output buffer, regardless of the alignment of the available input. 7303 // The value is somewhat arbitrary, and could probably be even larger. 7304 // A larger value should allow more old data to be read after a track calls start(), 7305 // without increasing latency. 7306 // 7307 // Note this is independent of the maximum downsampling ratio permitted for capture. 7308 mRsmpInFrames = mFrameCount * 7; 7309 mRsmpInFramesP2 = roundup(mRsmpInFrames); 7310 free(mRsmpInBuffer); 7311 mRsmpInBuffer = NULL; 7312 7313 // TODO optimize audio capture buffer sizes ... 7314 // Here we calculate the size of the sliding buffer used as a source 7315 // for resampling. mRsmpInFramesP2 is currently roundup(mFrameCount * 7). 7316 // For current HAL frame counts, this is usually 2048 = 40 ms. It would 7317 // be better to have it derived from the pipe depth in the long term. 7318 // The current value is higher than necessary. However it should not add to latency. 7319 7320 // Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer 7321 mRsmpInFramesOA = mRsmpInFramesP2 + mFrameCount - 1; 7322 (void)posix_memalign(&mRsmpInBuffer, 32, mRsmpInFramesOA * mFrameSize); 7323 // if posix_memalign fails, will segv here. 7324 memset(mRsmpInBuffer, 0, mRsmpInFramesOA * mFrameSize); 7325 7326 // AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints. 7327 // But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks? 7328 } 7329 7330 uint32_t AudioFlinger::RecordThread::getInputFramesLost() 7331 { 7332 Mutex::Autolock _l(mLock); 7333 uint32_t result; 7334 if (initCheck() == NO_ERROR && mInput->stream->getInputFramesLost(&result) == OK) { 7335 return result; 7336 } 7337 return 0; 7338 } 7339 7340 // hasAudioSession_l() must be called with ThreadBase::mLock held 7341 uint32_t AudioFlinger::RecordThread::hasAudioSession_l(audio_session_t sessionId) const 7342 { 7343 uint32_t result = 0; 7344 if (getEffectChain_l(sessionId) != 0) { 7345 result = EFFECT_SESSION; 7346 } 7347 7348 for (size_t i = 0; i < mTracks.size(); ++i) { 7349 if (sessionId == mTracks[i]->sessionId()) { 7350 result |= TRACK_SESSION; 7351 if (mTracks[i]->isFastTrack()) { 7352 result |= FAST_SESSION; 7353 } 7354 break; 7355 } 7356 } 7357 7358 return result; 7359 } 7360 7361 KeyedVector<audio_session_t, bool> AudioFlinger::RecordThread::sessionIds() const 7362 { 7363 KeyedVector<audio_session_t, bool> ids; 7364 Mutex::Autolock _l(mLock); 7365 for (size_t j = 0; j < mTracks.size(); ++j) { 7366 sp<RecordThread::RecordTrack> track = mTracks[j]; 7367 audio_session_t sessionId = track->sessionId(); 7368 if (ids.indexOfKey(sessionId) < 0) { 7369 ids.add(sessionId, true); 7370 } 7371 } 7372 return ids; 7373 } 7374 7375 AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 7376 { 7377 Mutex::Autolock _l(mLock); 7378 AudioStreamIn *input = mInput; 7379 mInput = NULL; 7380 return input; 7381 } 7382 7383 // this method must always be called either with ThreadBase mLock held or inside the thread loop 7384 sp<StreamHalInterface> AudioFlinger::RecordThread::stream() const 7385 { 7386 if (mInput == NULL) { 7387 return NULL; 7388 } 7389 return mInput->stream; 7390 } 7391 7392 status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7393 { 7394 // only one chain per input thread 7395 if (mEffectChains.size() != 0) { 7396 ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this); 7397 return INVALID_OPERATION; 7398 } 7399 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7400 chain->setThread(this); 7401 chain->setInBuffer(NULL); 7402 chain->setOutBuffer(NULL); 7403 7404 checkSuspendOnAddEffectChain_l(chain); 7405 7406 // make sure enabled pre processing effects state is communicated to the HAL as we 7407 // just moved them to a new input stream. 7408 chain->syncHalEffectsState(); 7409 7410 mEffectChains.add(chain); 7411 7412 return NO_ERROR; 7413 } 7414 7415 size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7416 { 7417 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7418 ALOGW_IF(mEffectChains.size() != 1, 7419 "removeEffectChain_l() %p invalid chain size %zu on thread %p", 7420 chain.get(), mEffectChains.size(), this); 7421 if (mEffectChains.size() == 1) { 7422 mEffectChains.removeAt(0); 7423 } 7424 return 0; 7425 } 7426 7427 status_t AudioFlinger::RecordThread::createAudioPatch_l(const struct audio_patch *patch, 7428 audio_patch_handle_t *handle) 7429 { 7430 status_t status = NO_ERROR; 7431 7432 // store new device and send to effects 7433 mInDevice = patch->sources[0].ext.device.type; 7434 mPatch = *patch; 7435 for (size_t i = 0; i < mEffectChains.size(); i++) { 7436 mEffectChains[i]->setDevice_l(mInDevice); 7437 } 7438 7439 checkBtNrec_l(); 7440 7441 // store new source and send to effects 7442 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7443 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7444 for (size_t i = 0; i < mEffectChains.size(); i++) { 7445 mEffectChains[i]->setAudioSource_l(mAudioSource); 7446 } 7447 } 7448 7449 if (mInput->audioHwDev->supportsAudioPatches()) { 7450 sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice(); 7451 status = hwDevice->createAudioPatch(patch->num_sources, 7452 patch->sources, 7453 patch->num_sinks, 7454 patch->sinks, 7455 handle); 7456 } else { 7457 char *address; 7458 if (strcmp(patch->sources[0].ext.device.address, "") != 0) { 7459 address = audio_device_address_to_parameter( 7460 patch->sources[0].ext.device.type, 7461 patch->sources[0].ext.device.address); 7462 } else { 7463 address = (char *)calloc(1, 1); 7464 } 7465 AudioParameter param = AudioParameter(String8(address)); 7466 free(address); 7467 param.addInt(String8(AudioParameter::keyRouting), 7468 (int)patch->sources[0].ext.device.type); 7469 param.addInt(String8(AudioParameter::keyInputSource), 7470 (int)patch->sinks[0].ext.mix.usecase.source); 7471 status = mInput->stream->setParameters(param.toString()); 7472 *handle = AUDIO_PATCH_HANDLE_NONE; 7473 } 7474 7475 if (mInDevice != mPrevInDevice) { 7476 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 7477 mPrevInDevice = mInDevice; 7478 } 7479 7480 return status; 7481 } 7482 7483 status_t AudioFlinger::RecordThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 7484 { 7485 status_t status = NO_ERROR; 7486 7487 mInDevice = AUDIO_DEVICE_NONE; 7488 7489 if (mInput->audioHwDev->supportsAudioPatches()) { 7490 sp<DeviceHalInterface> hwDevice = mInput->audioHwDev->hwDevice(); 7491 status = hwDevice->releaseAudioPatch(handle); 7492 } else { 7493 AudioParameter param; 7494 param.addInt(String8(AudioParameter::keyRouting), 0); 7495 status = mInput->stream->setParameters(param.toString()); 7496 } 7497 return status; 7498 } 7499 7500 void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record) 7501 { 7502 Mutex::Autolock _l(mLock); 7503 mTracks.add(record); 7504 } 7505 7506 void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record) 7507 { 7508 Mutex::Autolock _l(mLock); 7509 destroyTrack_l(record); 7510 } 7511 7512 void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config) 7513 { 7514 ThreadBase::getAudioPortConfig(config); 7515 config->role = AUDIO_PORT_ROLE_SINK; 7516 config->ext.mix.hw_module = mInput->audioHwDev->handle(); 7517 config->ext.mix.usecase.source = mAudioSource; 7518 } 7519 7520 // ---------------------------------------------------------------------------- 7521 // Mmap 7522 // ---------------------------------------------------------------------------- 7523 7524 AudioFlinger::MmapThreadHandle::MmapThreadHandle(const sp<MmapThread>& thread) 7525 : mThread(thread) 7526 { 7527 assert(thread != 0); // thread must start non-null and stay non-null 7528 } 7529 7530 AudioFlinger::MmapThreadHandle::~MmapThreadHandle() 7531 { 7532 mThread->disconnect(); 7533 } 7534 7535 status_t AudioFlinger::MmapThreadHandle::createMmapBuffer(int32_t minSizeFrames, 7536 struct audio_mmap_buffer_info *info) 7537 { 7538 return mThread->createMmapBuffer(minSizeFrames, info); 7539 } 7540 7541 status_t AudioFlinger::MmapThreadHandle::getMmapPosition(struct audio_mmap_position *position) 7542 { 7543 return mThread->getMmapPosition(position); 7544 } 7545 7546 status_t AudioFlinger::MmapThreadHandle::start(const AudioClient& client, 7547 audio_port_handle_t *handle) 7548 7549 { 7550 return mThread->start(client, handle); 7551 } 7552 7553 status_t AudioFlinger::MmapThreadHandle::stop(audio_port_handle_t handle) 7554 { 7555 return mThread->stop(handle); 7556 } 7557 7558 status_t AudioFlinger::MmapThreadHandle::standby() 7559 { 7560 return mThread->standby(); 7561 } 7562 7563 7564 AudioFlinger::MmapThread::MmapThread( 7565 const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 7566 AudioHwDevice *hwDev, sp<StreamHalInterface> stream, 7567 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady) 7568 : ThreadBase(audioFlinger, id, outDevice, inDevice, MMAP, systemReady), 7569 mSessionId(AUDIO_SESSION_NONE), 7570 mDeviceId(AUDIO_PORT_HANDLE_NONE), mPortId(AUDIO_PORT_HANDLE_NONE), 7571 mHalStream(stream), mHalDevice(hwDev->hwDevice()), mAudioHwDev(hwDev), 7572 mActiveTracks(&this->mLocalLog) 7573 { 7574 mStandby = true; 7575 readHalParameters_l(); 7576 } 7577 7578 AudioFlinger::MmapThread::~MmapThread() 7579 { 7580 releaseWakeLock_l(); 7581 } 7582 7583 void AudioFlinger::MmapThread::onFirstRef() 7584 { 7585 run(mThreadName, ANDROID_PRIORITY_URGENT_AUDIO); 7586 } 7587 7588 void AudioFlinger::MmapThread::disconnect() 7589 { 7590 for (const sp<MmapTrack> &t : mActiveTracks) { 7591 stop(t->portId()); 7592 } 7593 // This will decrement references and may cause the destruction of this thread. 7594 if (isOutput()) { 7595 AudioSystem::releaseOutput(mId, streamType(), mSessionId); 7596 } else { 7597 AudioSystem::releaseInput(mId, mSessionId); 7598 } 7599 } 7600 7601 7602 void AudioFlinger::MmapThread::configure(const audio_attributes_t *attr, 7603 audio_stream_type_t streamType __unused, 7604 audio_session_t sessionId, 7605 const sp<MmapStreamCallback>& callback, 7606 audio_port_handle_t deviceId, 7607 audio_port_handle_t portId) 7608 { 7609 mAttr = *attr; 7610 mSessionId = sessionId; 7611 mCallback = callback; 7612 mDeviceId = deviceId; 7613 mPortId = portId; 7614 } 7615 7616 status_t AudioFlinger::MmapThread::createMmapBuffer(int32_t minSizeFrames, 7617 struct audio_mmap_buffer_info *info) 7618 { 7619 if (mHalStream == 0) { 7620 return NO_INIT; 7621 } 7622 mStandby = true; 7623 acquireWakeLock(); 7624 return mHalStream->createMmapBuffer(minSizeFrames, info); 7625 } 7626 7627 status_t AudioFlinger::MmapThread::getMmapPosition(struct audio_mmap_position *position) 7628 { 7629 if (mHalStream == 0) { 7630 return NO_INIT; 7631 } 7632 return mHalStream->getMmapPosition(position); 7633 } 7634 7635 status_t AudioFlinger::MmapThread::start(const AudioClient& client, 7636 audio_port_handle_t *handle) 7637 { 7638 ALOGV("%s clientUid %d mStandby %d mPortId %d *handle %d", __FUNCTION__, 7639 client.clientUid, mStandby, mPortId, *handle); 7640 if (mHalStream == 0) { 7641 return NO_INIT; 7642 } 7643 7644 status_t ret; 7645 7646 if (*handle == mPortId) { 7647 // for the first track, reuse portId and session allocated when the stream was opened 7648 ret = mHalStream->start(); 7649 if (ret != NO_ERROR) { 7650 ALOGE("%s: error mHalStream->start() = %d for first track", __FUNCTION__, ret); 7651 return ret; 7652 } 7653 mStandby = false; 7654 return NO_ERROR; 7655 } 7656 7657 if (!isOutput() && !recordingAllowed(client.packageName, client.clientPid, client.clientUid)) { 7658 return PERMISSION_DENIED; 7659 } 7660 7661 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE; 7662 7663 audio_io_handle_t io = mId; 7664 if (isOutput()) { 7665 audio_config_t config = AUDIO_CONFIG_INITIALIZER; 7666 config.sample_rate = mSampleRate; 7667 config.channel_mask = mChannelMask; 7668 config.format = mFormat; 7669 audio_stream_type_t stream = streamType(); 7670 audio_output_flags_t flags = 7671 (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ | AUDIO_OUTPUT_FLAG_DIRECT); 7672 audio_port_handle_t deviceId = mDeviceId; 7673 ret = AudioSystem::getOutputForAttr(&mAttr, &io, 7674 mSessionId, 7675 &stream, 7676 client.clientUid, 7677 &config, 7678 flags, 7679 &deviceId, 7680 &portId); 7681 } else { 7682 audio_config_base_t config; 7683 config.sample_rate = mSampleRate; 7684 config.channel_mask = mChannelMask; 7685 config.format = mFormat; 7686 audio_port_handle_t deviceId = mDeviceId; 7687 ret = AudioSystem::getInputForAttr(&mAttr, &io, 7688 mSessionId, 7689 client.clientPid, 7690 client.clientUid, 7691 &config, 7692 AUDIO_INPUT_FLAG_MMAP_NOIRQ, 7693 &deviceId, 7694 &portId); 7695 } 7696 // APM should not chose a different input or output stream for the same set of attributes 7697 // and audo configuration 7698 if (ret != NO_ERROR || io != mId) { 7699 ALOGE("%s: error getting output or input from APM (error %d, io %d expected io %d)", 7700 __FUNCTION__, ret, io, mId); 7701 return BAD_VALUE; 7702 } 7703 7704 if (isOutput()) { 7705 ret = AudioSystem::startOutput(mId, streamType(), mSessionId); 7706 } else { 7707 ret = AudioSystem::startInput(mId, mSessionId); 7708 } 7709 7710 // abort if start is rejected by audio policy manager 7711 if (ret != NO_ERROR) { 7712 ALOGE("%s: error start rejected by AudioPolicyManager = %d", __FUNCTION__, ret); 7713 if (mActiveTracks.size() != 0) { 7714 if (isOutput()) { 7715 AudioSystem::releaseOutput(mId, streamType(), mSessionId); 7716 } else { 7717 AudioSystem::releaseInput(mId, mSessionId); 7718 } 7719 } else { 7720 mHalStream->stop(); 7721 } 7722 return PERMISSION_DENIED; 7723 } 7724 7725 sp<MmapTrack> track = new MmapTrack(this, mSampleRate, mFormat, mChannelMask, mSessionId, 7726 client.clientUid, client.clientPid, portId); 7727 7728 mActiveTracks.add(track); 7729 sp<EffectChain> chain = getEffectChain_l(mSessionId); 7730 if (chain != 0) { 7731 chain->setStrategy(AudioSystem::getStrategyForStream(streamType())); 7732 chain->incTrackCnt(); 7733 chain->incActiveTrackCnt(); 7734 } 7735 7736 *handle = portId; 7737 broadcast_l(); 7738 7739 ALOGV("%s DONE handle %d stream %p", __FUNCTION__, *handle, mHalStream.get()); 7740 7741 return NO_ERROR; 7742 } 7743 7744 status_t AudioFlinger::MmapThread::stop(audio_port_handle_t handle) 7745 { 7746 ALOGV("%s handle %d", __FUNCTION__, handle); 7747 7748 if (mHalStream == 0) { 7749 return NO_INIT; 7750 } 7751 7752 if (handle == mPortId) { 7753 mHalStream->stop(); 7754 return NO_ERROR; 7755 } 7756 7757 sp<MmapTrack> track; 7758 for (const sp<MmapTrack> &t : mActiveTracks) { 7759 if (handle == t->portId()) { 7760 track = t; 7761 break; 7762 } 7763 } 7764 if (track == 0) { 7765 return BAD_VALUE; 7766 } 7767 7768 mActiveTracks.remove(track); 7769 7770 if (isOutput()) { 7771 AudioSystem::stopOutput(mId, streamType(), track->sessionId()); 7772 AudioSystem::releaseOutput(mId, streamType(), track->sessionId()); 7773 } else { 7774 AudioSystem::stopInput(mId, track->sessionId()); 7775 AudioSystem::releaseInput(mId, track->sessionId()); 7776 } 7777 7778 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 7779 if (chain != 0) { 7780 chain->decActiveTrackCnt(); 7781 chain->decTrackCnt(); 7782 } 7783 7784 broadcast_l(); 7785 7786 return NO_ERROR; 7787 } 7788 7789 status_t AudioFlinger::MmapThread::standby() 7790 { 7791 ALOGV("%s", __FUNCTION__); 7792 7793 if (mHalStream == 0) { 7794 return NO_INIT; 7795 } 7796 if (mActiveTracks.size() != 0) { 7797 return INVALID_OPERATION; 7798 } 7799 mHalStream->standby(); 7800 mStandby = true; 7801 releaseWakeLock(); 7802 return NO_ERROR; 7803 } 7804 7805 7806 void AudioFlinger::MmapThread::readHalParameters_l() 7807 { 7808 status_t result = mHalStream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat); 7809 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result); 7810 mFormat = mHALFormat; 7811 LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat); 7812 result = mHalStream->getFrameSize(&mFrameSize); 7813 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result); 7814 result = mHalStream->getBufferSize(&mBufferSize); 7815 LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result); 7816 mFrameCount = mBufferSize / mFrameSize; 7817 } 7818 7819 bool AudioFlinger::MmapThread::threadLoop() 7820 { 7821 checkSilentMode_l(); 7822 7823 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 7824 7825 while (!exitPending()) 7826 { 7827 Mutex::Autolock _l(mLock); 7828 Vector< sp<EffectChain> > effectChains; 7829 7830 if (mSignalPending) { 7831 // A signal was raised while we were unlocked 7832 mSignalPending = false; 7833 } else { 7834 if (mConfigEvents.isEmpty()) { 7835 // we're about to wait, flush the binder command buffer 7836 IPCThreadState::self()->flushCommands(); 7837 7838 if (exitPending()) { 7839 break; 7840 } 7841 7842 // wait until we have something to do... 7843 ALOGV("%s going to sleep", myName.string()); 7844 mWaitWorkCV.wait(mLock); 7845 ALOGV("%s waking up", myName.string()); 7846 7847 checkSilentMode_l(); 7848 7849 continue; 7850 } 7851 } 7852 7853 processConfigEvents_l(); 7854 7855 processVolume_l(); 7856 7857 checkInvalidTracks_l(); 7858 7859 mActiveTracks.updatePowerState(this); 7860 7861 lockEffectChains_l(effectChains); 7862 for (size_t i = 0; i < effectChains.size(); i ++) { 7863 effectChains[i]->process_l(); 7864 } 7865 // enable changes in effect chain 7866 unlockEffectChains(effectChains); 7867 // Effect chains will be actually deleted here if they were removed from 7868 // mEffectChains list during mixing or effects processing 7869 } 7870 7871 threadLoop_exit(); 7872 7873 if (!mStandby) { 7874 threadLoop_standby(); 7875 mStandby = true; 7876 } 7877 7878 ALOGV("Thread %p type %d exiting", this, mType); 7879 return false; 7880 } 7881 7882 // checkForNewParameter_l() must be called with ThreadBase::mLock held 7883 bool AudioFlinger::MmapThread::checkForNewParameter_l(const String8& keyValuePair, 7884 status_t& status) 7885 { 7886 AudioParameter param = AudioParameter(keyValuePair); 7887 int value; 7888 bool sendToHal = true; 7889 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 7890 audio_devices_t device = (audio_devices_t)value; 7891 // forward device change to effects that have requested to be 7892 // aware of attached audio device. 7893 if (device != AUDIO_DEVICE_NONE) { 7894 for (size_t i = 0; i < mEffectChains.size(); i++) { 7895 mEffectChains[i]->setDevice_l(device); 7896 } 7897 } 7898 if (audio_is_output_devices(device)) { 7899 mOutDevice = device; 7900 if (!isOutput()) { 7901 sendToHal = false; 7902 } 7903 } else { 7904 mInDevice = device; 7905 if (device != AUDIO_DEVICE_NONE) { 7906 mPrevInDevice = value; 7907 } 7908 // TODO: implement and call checkBtNrec_l(); 7909 } 7910 } 7911 if (sendToHal) { 7912 status = mHalStream->setParameters(keyValuePair); 7913 } else { 7914 status = NO_ERROR; 7915 } 7916 7917 return false; 7918 } 7919 7920 String8 AudioFlinger::MmapThread::getParameters(const String8& keys) 7921 { 7922 Mutex::Autolock _l(mLock); 7923 String8 out_s8; 7924 if (initCheck() == NO_ERROR && mHalStream->getParameters(keys, &out_s8) == OK) { 7925 return out_s8; 7926 } 7927 return String8(); 7928 } 7929 7930 void AudioFlinger::MmapThread::ioConfigChanged(audio_io_config_event event, pid_t pid) { 7931 sp<AudioIoDescriptor> desc = new AudioIoDescriptor(); 7932 7933 desc->mIoHandle = mId; 7934 7935 switch (event) { 7936 case AUDIO_INPUT_OPENED: 7937 case AUDIO_INPUT_REGISTERED: 7938 case AUDIO_INPUT_CONFIG_CHANGED: 7939 case AUDIO_OUTPUT_OPENED: 7940 case AUDIO_OUTPUT_REGISTERED: 7941 case AUDIO_OUTPUT_CONFIG_CHANGED: 7942 desc->mPatch = mPatch; 7943 desc->mChannelMask = mChannelMask; 7944 desc->mSamplingRate = mSampleRate; 7945 desc->mFormat = mFormat; 7946 desc->mFrameCount = mFrameCount; 7947 desc->mFrameCountHAL = mFrameCount; 7948 desc->mLatency = 0; 7949 break; 7950 7951 case AUDIO_INPUT_CLOSED: 7952 case AUDIO_OUTPUT_CLOSED: 7953 default: 7954 break; 7955 } 7956 mAudioFlinger->ioConfigChanged(event, desc, pid); 7957 } 7958 7959 status_t AudioFlinger::MmapThread::createAudioPatch_l(const struct audio_patch *patch, 7960 audio_patch_handle_t *handle) 7961 { 7962 status_t status = NO_ERROR; 7963 7964 // store new device and send to effects 7965 audio_devices_t type = AUDIO_DEVICE_NONE; 7966 audio_port_handle_t deviceId; 7967 if (isOutput()) { 7968 for (unsigned int i = 0; i < patch->num_sinks; i++) { 7969 type |= patch->sinks[i].ext.device.type; 7970 } 7971 deviceId = patch->sinks[0].id; 7972 } else { 7973 type = patch->sources[0].ext.device.type; 7974 deviceId = patch->sources[0].id; 7975 } 7976 7977 for (size_t i = 0; i < mEffectChains.size(); i++) { 7978 mEffectChains[i]->setDevice_l(type); 7979 } 7980 7981 if (isOutput()) { 7982 mOutDevice = type; 7983 } else { 7984 mInDevice = type; 7985 // store new source and send to effects 7986 if (mAudioSource != patch->sinks[0].ext.mix.usecase.source) { 7987 mAudioSource = patch->sinks[0].ext.mix.usecase.source; 7988 for (size_t i = 0; i < mEffectChains.size(); i++) { 7989 mEffectChains[i]->setAudioSource_l(mAudioSource); 7990 } 7991 } 7992 } 7993 7994 if (mAudioHwDev->supportsAudioPatches()) { 7995 status = mHalDevice->createAudioPatch(patch->num_sources, 7996 patch->sources, 7997 patch->num_sinks, 7998 patch->sinks, 7999 handle); 8000 } else { 8001 char *address; 8002 if (strcmp(patch->sinks[0].ext.device.address, "") != 0) { 8003 //FIXME: we only support address on first sink with HAL version < 3.0 8004 address = audio_device_address_to_parameter( 8005 patch->sinks[0].ext.device.type, 8006 patch->sinks[0].ext.device.address); 8007 } else { 8008 address = (char *)calloc(1, 1); 8009 } 8010 AudioParameter param = AudioParameter(String8(address)); 8011 free(address); 8012 param.addInt(String8(AudioParameter::keyRouting), (int)type); 8013 if (!isOutput()) { 8014 param.addInt(String8(AudioParameter::keyInputSource), 8015 (int)patch->sinks[0].ext.mix.usecase.source); 8016 } 8017 status = mHalStream->setParameters(param.toString()); 8018 *handle = AUDIO_PATCH_HANDLE_NONE; 8019 } 8020 8021 if (isOutput() && mPrevOutDevice != mOutDevice) { 8022 mPrevOutDevice = type; 8023 sendIoConfigEvent_l(AUDIO_OUTPUT_CONFIG_CHANGED); 8024 sp<MmapStreamCallback> callback = mCallback.promote(); 8025 if (mDeviceId != deviceId && callback != 0) { 8026 callback->onRoutingChanged(deviceId); 8027 } 8028 mDeviceId = deviceId; 8029 } 8030 if (!isOutput() && mPrevInDevice != mInDevice) { 8031 mPrevInDevice = type; 8032 sendIoConfigEvent_l(AUDIO_INPUT_CONFIG_CHANGED); 8033 sp<MmapStreamCallback> callback = mCallback.promote(); 8034 if (mDeviceId != deviceId && callback != 0) { 8035 callback->onRoutingChanged(deviceId); 8036 } 8037 mDeviceId = deviceId; 8038 } 8039 return status; 8040 } 8041 8042 status_t AudioFlinger::MmapThread::releaseAudioPatch_l(const audio_patch_handle_t handle) 8043 { 8044 status_t status = NO_ERROR; 8045 8046 mInDevice = AUDIO_DEVICE_NONE; 8047 8048 bool supportsAudioPatches = mHalDevice->supportsAudioPatches(&supportsAudioPatches) == OK ? 8049 supportsAudioPatches : false; 8050 8051 if (supportsAudioPatches) { 8052 status = mHalDevice->releaseAudioPatch(handle); 8053 } else { 8054 AudioParameter param; 8055 param.addInt(String8(AudioParameter::keyRouting), 0); 8056 status = mHalStream->setParameters(param.toString()); 8057 } 8058 return status; 8059 } 8060 8061 void AudioFlinger::MmapThread::getAudioPortConfig(struct audio_port_config *config) 8062 { 8063 ThreadBase::getAudioPortConfig(config); 8064 if (isOutput()) { 8065 config->role = AUDIO_PORT_ROLE_SOURCE; 8066 config->ext.mix.hw_module = mAudioHwDev->handle(); 8067 config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT; 8068 } else { 8069 config->role = AUDIO_PORT_ROLE_SINK; 8070 config->ext.mix.hw_module = mAudioHwDev->handle(); 8071 config->ext.mix.usecase.source = mAudioSource; 8072 } 8073 } 8074 8075 status_t AudioFlinger::MmapThread::addEffectChain_l(const sp<EffectChain>& chain) 8076 { 8077 audio_session_t session = chain->sessionId(); 8078 8079 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 8080 // Attach all tracks with same session ID to this chain. 8081 // indicate all active tracks in the chain 8082 for (const sp<MmapTrack> &track : mActiveTracks) { 8083 if (session == track->sessionId()) { 8084 chain->incTrackCnt(); 8085 chain->incActiveTrackCnt(); 8086 } 8087 } 8088 8089 chain->setThread(this); 8090 chain->setInBuffer(nullptr); 8091 chain->setOutBuffer(nullptr); 8092 chain->syncHalEffectsState(); 8093 8094 mEffectChains.add(chain); 8095 checkSuspendOnAddEffectChain_l(chain); 8096 return NO_ERROR; 8097 } 8098 8099 size_t AudioFlinger::MmapThread::removeEffectChain_l(const sp<EffectChain>& chain) 8100 { 8101 audio_session_t session = chain->sessionId(); 8102 8103 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 8104 8105 for (size_t i = 0; i < mEffectChains.size(); i++) { 8106 if (chain == mEffectChains[i]) { 8107 mEffectChains.removeAt(i); 8108 // detach all active tracks from the chain 8109 // detach all tracks with same session ID from this chain 8110 for (const sp<MmapTrack> &track : mActiveTracks) { 8111 if (session == track->sessionId()) { 8112 chain->decActiveTrackCnt(); 8113 chain->decTrackCnt(); 8114 } 8115 } 8116 break; 8117 } 8118 } 8119 return mEffectChains.size(); 8120 } 8121 8122 // hasAudioSession_l() must be called with ThreadBase::mLock held 8123 uint32_t AudioFlinger::MmapThread::hasAudioSession_l(audio_session_t sessionId) const 8124 { 8125 uint32_t result = 0; 8126 if (getEffectChain_l(sessionId) != 0) { 8127 result = EFFECT_SESSION; 8128 } 8129 8130 for (size_t i = 0; i < mActiveTracks.size(); i++) { 8131 sp<MmapTrack> track = mActiveTracks[i]; 8132 if (sessionId == track->sessionId()) { 8133 result |= TRACK_SESSION; 8134 if (track->isFastTrack()) { 8135 result |= FAST_SESSION; 8136 } 8137 break; 8138 } 8139 } 8140 8141 return result; 8142 } 8143 8144 void AudioFlinger::MmapThread::threadLoop_standby() 8145 { 8146 mHalStream->standby(); 8147 } 8148 8149 void AudioFlinger::MmapThread::threadLoop_exit() 8150 { 8151 // Do not call callback->onTearDown() because it is redundant for thread exit 8152 // and because it can cause a recursive mutex lock on stop(). 8153 } 8154 8155 status_t AudioFlinger::MmapThread::setSyncEvent(const sp<SyncEvent>& event __unused) 8156 { 8157 return BAD_VALUE; 8158 } 8159 8160 bool AudioFlinger::MmapThread::isValidSyncEvent(const sp<SyncEvent>& event __unused) const 8161 { 8162 return false; 8163 } 8164 8165 status_t AudioFlinger::MmapThread::checkEffectCompatibility_l( 8166 const effect_descriptor_t *desc, audio_session_t sessionId) 8167 { 8168 // No global effect sessions on mmap threads 8169 if (sessionId == AUDIO_SESSION_OUTPUT_MIX || sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 8170 ALOGW("checkEffectCompatibility_l(): global effect %s on record thread %s", 8171 desc->name, mThreadName); 8172 return BAD_VALUE; 8173 } 8174 8175 if (!isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC)) { 8176 ALOGW("checkEffectCompatibility_l(): non pre processing effect %s on capture mmap thread", 8177 desc->name); 8178 return BAD_VALUE; 8179 } 8180 if (isOutput() && ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 8181 ALOGW("checkEffectCompatibility_l(): pre processing effect %s created on playback mmap " 8182 "thread", desc->name); 8183 return BAD_VALUE; 8184 } 8185 8186 // Only allow effects without processing load or latency 8187 if ((desc->flags & EFFECT_FLAG_NO_PROCESS_MASK) != EFFECT_FLAG_NO_PROCESS) { 8188 return BAD_VALUE; 8189 } 8190 8191 return NO_ERROR; 8192 8193 } 8194 8195 void AudioFlinger::MmapThread::checkInvalidTracks_l() 8196 { 8197 for (const sp<MmapTrack> &track : mActiveTracks) { 8198 if (track->isInvalid()) { 8199 sp<MmapStreamCallback> callback = mCallback.promote(); 8200 if (callback != 0) { 8201 callback->onTearDown(); 8202 } 8203 break; 8204 } 8205 } 8206 } 8207 8208 void AudioFlinger::MmapThread::dump(int fd, const Vector<String16>& args) 8209 { 8210 dumpInternals(fd, args); 8211 dumpTracks(fd, args); 8212 dumpEffectChains(fd, args); 8213 dprintf(fd, " Local log:\n"); 8214 mLocalLog.dump(fd, " " /* prefix */, 40 /* lines */); 8215 } 8216 8217 void AudioFlinger::MmapThread::dumpInternals(int fd, const Vector<String16>& args) 8218 { 8219 dumpBase(fd, args); 8220 8221 dprintf(fd, " Attributes: content type %d usage %d source %d\n", 8222 mAttr.content_type, mAttr.usage, mAttr.source); 8223 dprintf(fd, " Session: %d port Id: %d\n", mSessionId, mPortId); 8224 if (mActiveTracks.size() == 0) { 8225 dprintf(fd, " No active clients\n"); 8226 } 8227 } 8228 8229 void AudioFlinger::MmapThread::dumpTracks(int fd, const Vector<String16>& args __unused) 8230 { 8231 String8 result; 8232 size_t numtracks = mActiveTracks.size(); 8233 dprintf(fd, " %zu Tracks\n", numtracks); 8234 const char *prefix = " "; 8235 if (numtracks) { 8236 result.append(prefix); 8237 MmapTrack::appendDumpHeader(result); 8238 for (size_t i = 0; i < numtracks ; ++i) { 8239 sp<MmapTrack> track = mActiveTracks[i]; 8240 result.append(prefix); 8241 track->appendDump(result, true /* active */); 8242 } 8243 } else { 8244 dprintf(fd, "\n"); 8245 } 8246 write(fd, result.string(), result.size()); 8247 } 8248 8249 AudioFlinger::MmapPlaybackThread::MmapPlaybackThread( 8250 const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 8251 AudioHwDevice *hwDev, AudioStreamOut *output, 8252 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady) 8253 : MmapThread(audioFlinger, id, hwDev, output->stream, outDevice, inDevice, systemReady), 8254 mStreamType(AUDIO_STREAM_MUSIC), 8255 mStreamVolume(1.0), mStreamMute(false), mOutput(output) 8256 { 8257 snprintf(mThreadName, kThreadNameLength, "AudioMmapOut_%X", id); 8258 mChannelCount = audio_channel_count_from_out_mask(mChannelMask); 8259 mMasterVolume = audioFlinger->masterVolume_l(); 8260 mMasterMute = audioFlinger->masterMute_l(); 8261 if (mAudioHwDev) { 8262 if (mAudioHwDev->canSetMasterVolume()) { 8263 mMasterVolume = 1.0; 8264 } 8265 8266 if (mAudioHwDev->canSetMasterMute()) { 8267 mMasterMute = false; 8268 } 8269 } 8270 } 8271 8272 void AudioFlinger::MmapPlaybackThread::configure(const audio_attributes_t *attr, 8273 audio_stream_type_t streamType, 8274 audio_session_t sessionId, 8275 const sp<MmapStreamCallback>& callback, 8276 audio_port_handle_t deviceId, 8277 audio_port_handle_t portId) 8278 { 8279 MmapThread::configure(attr, streamType, sessionId, callback, deviceId, portId); 8280 mStreamType = streamType; 8281 } 8282 8283 AudioStreamOut* AudioFlinger::MmapPlaybackThread::clearOutput() 8284 { 8285 Mutex::Autolock _l(mLock); 8286 AudioStreamOut *output = mOutput; 8287 mOutput = NULL; 8288 return output; 8289 } 8290 8291 void AudioFlinger::MmapPlaybackThread::setMasterVolume(float value) 8292 { 8293 Mutex::Autolock _l(mLock); 8294 // Don't apply master volume in SW if our HAL can do it for us. 8295 if (mAudioHwDev && 8296 mAudioHwDev->canSetMasterVolume()) { 8297 mMasterVolume = 1.0; 8298 } else { 8299 mMasterVolume = value; 8300 } 8301 } 8302 8303 void AudioFlinger::MmapPlaybackThread::setMasterMute(bool muted) 8304 { 8305 Mutex::Autolock _l(mLock); 8306 // Don't apply master mute in SW if our HAL can do it for us. 8307 if (mAudioHwDev && mAudioHwDev->canSetMasterMute()) { 8308 mMasterMute = false; 8309 } else { 8310 mMasterMute = muted; 8311 } 8312 } 8313 8314 void AudioFlinger::MmapPlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 8315 { 8316 Mutex::Autolock _l(mLock); 8317 if (stream == mStreamType) { 8318 mStreamVolume = value; 8319 broadcast_l(); 8320 } 8321 } 8322 8323 float AudioFlinger::MmapPlaybackThread::streamVolume(audio_stream_type_t stream) const 8324 { 8325 Mutex::Autolock _l(mLock); 8326 if (stream == mStreamType) { 8327 return mStreamVolume; 8328 } 8329 return 0.0f; 8330 } 8331 8332 void AudioFlinger::MmapPlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 8333 { 8334 Mutex::Autolock _l(mLock); 8335 if (stream == mStreamType) { 8336 mStreamMute= muted; 8337 broadcast_l(); 8338 } 8339 } 8340 8341 void AudioFlinger::MmapPlaybackThread::invalidateTracks(audio_stream_type_t streamType) 8342 { 8343 Mutex::Autolock _l(mLock); 8344 if (streamType == mStreamType) { 8345 for (const sp<MmapTrack> &track : mActiveTracks) { 8346 track->invalidate(); 8347 } 8348 broadcast_l(); 8349 } 8350 } 8351 8352 void AudioFlinger::MmapPlaybackThread::processVolume_l() 8353 { 8354 float volume; 8355 8356 if (mMasterMute || mStreamMute) { 8357 volume = 0; 8358 } else { 8359 volume = mMasterVolume * mStreamVolume; 8360 } 8361 8362 if (volume != mHalVolFloat) { 8363 mHalVolFloat = volume; 8364 8365 // Convert volumes from float to 8.24 8366 uint32_t vol = (uint32_t)(volume * (1 << 24)); 8367 8368 // Delegate volume control to effect in track effect chain if needed 8369 // only one effect chain can be present on DirectOutputThread, so if 8370 // there is one, the track is connected to it 8371 if (!mEffectChains.isEmpty()) { 8372 mEffectChains[0]->setVolume_l(&vol, &vol); 8373 volume = (float)vol / (1 << 24); 8374 } 8375 // Try to use HW volume control and fall back to SW control if not implemented 8376 if (mOutput->stream->setVolume(volume, volume) != NO_ERROR) { 8377 sp<MmapStreamCallback> callback = mCallback.promote(); 8378 if (callback != 0) { 8379 int channelCount; 8380 if (isOutput()) { 8381 channelCount = audio_channel_count_from_out_mask(mChannelMask); 8382 } else { 8383 channelCount = audio_channel_count_from_in_mask(mChannelMask); 8384 } 8385 Vector<float> values; 8386 for (int i = 0; i < channelCount; i++) { 8387 values.add(volume); 8388 } 8389 callback->onVolumeChanged(mChannelMask, values); 8390 } else { 8391 ALOGW("Could not set MMAP stream volume: no volume callback!"); 8392 } 8393 } 8394 } 8395 } 8396 8397 void AudioFlinger::MmapPlaybackThread::checkSilentMode_l() 8398 { 8399 if (!mMasterMute) { 8400 char value[PROPERTY_VALUE_MAX]; 8401 if (property_get("ro.audio.silent", value, "0") > 0) { 8402 char *endptr; 8403 unsigned long ul = strtoul(value, &endptr, 0); 8404 if (*endptr == '\0' && ul != 0) { 8405 ALOGD("Silence is golden"); 8406 // The setprop command will not allow a property to be changed after 8407 // the first time it is set, so we don't have to worry about un-muting. 8408 setMasterMute_l(true); 8409 } 8410 } 8411 } 8412 } 8413 8414 void AudioFlinger::MmapPlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 8415 { 8416 MmapThread::dumpInternals(fd, args); 8417 8418 dprintf(fd, " Stream type: %d Stream volume: %f HAL volume: %f Stream mute %d\n", 8419 mStreamType, mStreamVolume, mHalVolFloat, mStreamMute); 8420 dprintf(fd, " Master volume: %f Master mute %d\n", mMasterVolume, mMasterMute); 8421 } 8422 8423 AudioFlinger::MmapCaptureThread::MmapCaptureThread( 8424 const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 8425 AudioHwDevice *hwDev, AudioStreamIn *input, 8426 audio_devices_t outDevice, audio_devices_t inDevice, bool systemReady) 8427 : MmapThread(audioFlinger, id, hwDev, input->stream, outDevice, inDevice, systemReady), 8428 mInput(input) 8429 { 8430 snprintf(mThreadName, kThreadNameLength, "AudioMmapIn_%X", id); 8431 mChannelCount = audio_channel_count_from_in_mask(mChannelMask); 8432 } 8433 8434 AudioFlinger::AudioStreamIn* AudioFlinger::MmapCaptureThread::clearInput() 8435 { 8436 Mutex::Autolock _l(mLock); 8437 AudioStreamIn *input = mInput; 8438 mInput = NULL; 8439 return input; 8440 } 8441 } // namespace android 8442