1 /* 2 ** 3 ** Copyright 2007, 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 #ifndef ANDROID_AUDIO_FLINGER_H 19 #define ANDROID_AUDIO_FLINGER_H 20 21 #include <stdint.h> 22 #include <sys/types.h> 23 #include <limits.h> 24 25 #include <common_time/cc_helper.h> 26 27 #include <cutils/compiler.h> 28 29 #include <media/IAudioFlinger.h> 30 #include <media/IAudioFlingerClient.h> 31 #include <media/IAudioTrack.h> 32 #include <media/IAudioRecord.h> 33 #include <media/AudioSystem.h> 34 #include <media/AudioTrack.h> 35 36 #include <utils/Atomic.h> 37 #include <utils/Errors.h> 38 #include <utils/threads.h> 39 #include <utils/SortedVector.h> 40 #include <utils/TypeHelpers.h> 41 #include <utils/Vector.h> 42 43 #include <binder/BinderService.h> 44 #include <binder/MemoryDealer.h> 45 46 #include <system/audio.h> 47 #include <hardware/audio.h> 48 #include <hardware/audio_policy.h> 49 50 #include <media/AudioBufferProvider.h> 51 #include <media/ExtendedAudioBufferProvider.h> 52 #include "FastMixer.h" 53 #include <media/nbaio/NBAIO.h> 54 #include "AudioWatchdog.h" 55 56 #include <powermanager/IPowerManager.h> 57 58 #include <media/nbaio/NBLog.h> 59 #include <private/media/AudioTrackShared.h> 60 61 namespace android { 62 63 class audio_track_cblk_t; 64 class effect_param_cblk_t; 65 class AudioMixer; 66 class AudioBuffer; 67 class AudioResampler; 68 class FastMixer; 69 class ServerProxy; 70 71 // ---------------------------------------------------------------------------- 72 73 // AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 74 // There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 75 // Adding full support for > 2 channel capture or playback would require more than simply changing 76 // this #define. There is an independent hard-coded upper limit in AudioMixer; 77 // removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 78 // The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 79 // Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 80 #define FCC_2 2 // FCC_2 = Fixed Channel Count 2 81 82 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 83 84 #define MAX_GAIN 4096.0f 85 #define MAX_GAIN_INT 0x1000 86 87 #define INCLUDING_FROM_AUDIOFLINGER_H 88 89 class AudioFlinger : 90 public BinderService<AudioFlinger>, 91 public BnAudioFlinger 92 { 93 friend class BinderService<AudioFlinger>; // for AudioFlinger() 94 public: 95 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 96 97 virtual status_t dump(int fd, const Vector<String16>& args); 98 99 // IAudioFlinger interface, in binder opcode order 100 virtual sp<IAudioTrack> createTrack( 101 audio_stream_type_t streamType, 102 uint32_t sampleRate, 103 audio_format_t format, 104 audio_channel_mask_t channelMask, 105 size_t frameCount, 106 IAudioFlinger::track_flags_t *flags, 107 const sp<IMemory>& sharedBuffer, 108 audio_io_handle_t output, 109 pid_t tid, 110 int *sessionId, 111 String8& name, 112 status_t *status); 113 114 virtual sp<IAudioRecord> openRecord( 115 audio_io_handle_t input, 116 uint32_t sampleRate, 117 audio_format_t format, 118 audio_channel_mask_t channelMask, 119 size_t frameCount, 120 IAudioFlinger::track_flags_t *flags, 121 pid_t tid, 122 int *sessionId, 123 status_t *status); 124 125 virtual uint32_t sampleRate(audio_io_handle_t output) const; 126 virtual int channelCount(audio_io_handle_t output) const; 127 virtual audio_format_t format(audio_io_handle_t output) const; 128 virtual size_t frameCount(audio_io_handle_t output) const; 129 virtual uint32_t latency(audio_io_handle_t output) const; 130 131 virtual status_t setMasterVolume(float value); 132 virtual status_t setMasterMute(bool muted); 133 134 virtual float masterVolume() const; 135 virtual bool masterMute() const; 136 137 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 138 audio_io_handle_t output); 139 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 140 141 virtual float streamVolume(audio_stream_type_t stream, 142 audio_io_handle_t output) const; 143 virtual bool streamMute(audio_stream_type_t stream) const; 144 145 virtual status_t setMode(audio_mode_t mode); 146 147 virtual status_t setMicMute(bool state); 148 virtual bool getMicMute() const; 149 150 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 151 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 152 153 virtual void registerClient(const sp<IAudioFlingerClient>& client); 154 155 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 156 audio_channel_mask_t channelMask) const; 157 158 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 159 audio_devices_t *pDevices, 160 uint32_t *pSamplingRate, 161 audio_format_t *pFormat, 162 audio_channel_mask_t *pChannelMask, 163 uint32_t *pLatencyMs, 164 audio_output_flags_t flags, 165 const audio_offload_info_t *offloadInfo); 166 167 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 168 audio_io_handle_t output2); 169 170 virtual status_t closeOutput(audio_io_handle_t output); 171 172 virtual status_t suspendOutput(audio_io_handle_t output); 173 174 virtual status_t restoreOutput(audio_io_handle_t output); 175 176 virtual audio_io_handle_t openInput(audio_module_handle_t module, 177 audio_devices_t *pDevices, 178 uint32_t *pSamplingRate, 179 audio_format_t *pFormat, 180 audio_channel_mask_t *pChannelMask); 181 182 virtual status_t closeInput(audio_io_handle_t input); 183 184 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 185 186 virtual status_t setVoiceVolume(float volume); 187 188 virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, 189 audio_io_handle_t output) const; 190 191 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 192 193 virtual int newAudioSessionId(); 194 195 virtual void acquireAudioSessionId(int audioSession); 196 197 virtual void releaseAudioSessionId(int audioSession); 198 199 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 200 201 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 202 203 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 204 effect_descriptor_t *descriptor) const; 205 206 virtual sp<IEffect> createEffect( 207 effect_descriptor_t *pDesc, 208 const sp<IEffectClient>& effectClient, 209 int32_t priority, 210 audio_io_handle_t io, 211 int sessionId, 212 status_t *status, 213 int *id, 214 int *enabled); 215 216 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 217 audio_io_handle_t dstOutput); 218 219 virtual audio_module_handle_t loadHwModule(const char *name); 220 221 virtual uint32_t getPrimaryOutputSamplingRate(); 222 virtual size_t getPrimaryOutputFrameCount(); 223 224 virtual status_t setLowRamDevice(bool isLowRamDevice); 225 226 virtual status_t onTransact( 227 uint32_t code, 228 const Parcel& data, 229 Parcel* reply, 230 uint32_t flags); 231 232 // end of IAudioFlinger interface 233 234 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 235 void unregisterWriter(const sp<NBLog::Writer>& writer); 236 private: 237 static const size_t kLogMemorySize = 10 * 1024; 238 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 239 public: 240 241 class SyncEvent; 242 243 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 244 245 class SyncEvent : public RefBase { 246 public: 247 SyncEvent(AudioSystem::sync_event_t type, 248 int triggerSession, 249 int listenerSession, 250 sync_event_callback_t callBack, 251 void *cookie) 252 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 253 mCallback(callBack), mCookie(cookie) 254 {} 255 256 virtual ~SyncEvent() {} 257 258 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 259 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 260 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 261 AudioSystem::sync_event_t type() const { return mType; } 262 int triggerSession() const { return mTriggerSession; } 263 int listenerSession() const { return mListenerSession; } 264 void *cookie() const { return mCookie; } 265 266 private: 267 const AudioSystem::sync_event_t mType; 268 const int mTriggerSession; 269 const int mListenerSession; 270 sync_event_callback_t mCallback; 271 void * const mCookie; 272 mutable Mutex mLock; 273 }; 274 275 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 276 int triggerSession, 277 int listenerSession, 278 sync_event_callback_t callBack, 279 void *cookie); 280 281 private: 282 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 283 284 audio_mode_t getMode() const { return mMode; } 285 286 bool btNrecIsOff() const { return mBtNrecIsOff; } 287 288 AudioFlinger() ANDROID_API; 289 virtual ~AudioFlinger(); 290 291 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 292 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 293 NO_INIT : NO_ERROR; } 294 295 // RefBase 296 virtual void onFirstRef(); 297 298 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 299 audio_devices_t devices); 300 void purgeStaleEffects_l(); 301 302 // standby delay for MIXER and DUPLICATING playback threads is read from property 303 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 304 static nsecs_t mStandbyTimeInNsecs; 305 306 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 307 // AudioFlinger::setParameters() updates, other threads read w/o lock 308 static uint32_t mScreenState; 309 310 // Internal dump utilities. 311 static const int kDumpLockRetries = 50; 312 static const int kDumpLockSleepUs = 20000; 313 static bool dumpTryLock(Mutex& mutex); 314 void dumpPermissionDenial(int fd, const Vector<String16>& args); 315 void dumpClients(int fd, const Vector<String16>& args); 316 void dumpInternals(int fd, const Vector<String16>& args); 317 318 // --- Client --- 319 class Client : public RefBase { 320 public: 321 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 322 virtual ~Client(); 323 sp<MemoryDealer> heap() const; 324 pid_t pid() const { return mPid; } 325 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 326 327 bool reserveTimedTrack(); 328 void releaseTimedTrack(); 329 330 private: 331 Client(const Client&); 332 Client& operator = (const Client&); 333 const sp<AudioFlinger> mAudioFlinger; 334 const sp<MemoryDealer> mMemoryDealer; 335 const pid_t mPid; 336 337 Mutex mTimedTrackLock; 338 int mTimedTrackCount; 339 }; 340 341 // --- Notification Client --- 342 class NotificationClient : public IBinder::DeathRecipient { 343 public: 344 NotificationClient(const sp<AudioFlinger>& audioFlinger, 345 const sp<IAudioFlingerClient>& client, 346 pid_t pid); 347 virtual ~NotificationClient(); 348 349 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 350 351 // IBinder::DeathRecipient 352 virtual void binderDied(const wp<IBinder>& who); 353 354 private: 355 NotificationClient(const NotificationClient&); 356 NotificationClient& operator = (const NotificationClient&); 357 358 const sp<AudioFlinger> mAudioFlinger; 359 const pid_t mPid; 360 const sp<IAudioFlingerClient> mAudioFlingerClient; 361 }; 362 363 class TrackHandle; 364 class RecordHandle; 365 class RecordThread; 366 class PlaybackThread; 367 class MixerThread; 368 class DirectOutputThread; 369 class OffloadThread; 370 class DuplicatingThread; 371 class AsyncCallbackThread; 372 class Track; 373 class RecordTrack; 374 class EffectModule; 375 class EffectHandle; 376 class EffectChain; 377 struct AudioStreamOut; 378 struct AudioStreamIn; 379 380 struct stream_type_t { 381 stream_type_t() 382 : volume(1.0f), 383 mute(false) 384 { 385 } 386 float volume; 387 bool mute; 388 }; 389 390 // --- PlaybackThread --- 391 392 #include "Threads.h" 393 394 #include "Effects.h" 395 396 // server side of the client's IAudioTrack 397 class TrackHandle : public android::BnAudioTrack { 398 public: 399 TrackHandle(const sp<PlaybackThread::Track>& track); 400 virtual ~TrackHandle(); 401 virtual sp<IMemory> getCblk() const; 402 virtual status_t start(); 403 virtual void stop(); 404 virtual void flush(); 405 virtual void pause(); 406 virtual status_t attachAuxEffect(int effectId); 407 virtual status_t allocateTimedBuffer(size_t size, 408 sp<IMemory>* buffer); 409 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 410 int64_t pts); 411 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 412 int target); 413 virtual status_t setParameters(const String8& keyValuePairs); 414 virtual status_t getTimestamp(AudioTimestamp& timestamp); 415 virtual void signal(); // signal playback thread for a change in control block 416 417 virtual status_t onTransact( 418 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 419 420 private: 421 const sp<PlaybackThread::Track> mTrack; 422 }; 423 424 // server side of the client's IAudioRecord 425 class RecordHandle : public android::BnAudioRecord { 426 public: 427 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 428 virtual ~RecordHandle(); 429 virtual sp<IMemory> getCblk() const; 430 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 431 virtual void stop(); 432 virtual status_t onTransact( 433 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 434 private: 435 const sp<RecordThread::RecordTrack> mRecordTrack; 436 437 // for use from destructor 438 void stop_nonvirtual(); 439 }; 440 441 442 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 443 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 444 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 445 // no range check, AudioFlinger::mLock held 446 bool streamMute_l(audio_stream_type_t stream) const 447 { return mStreamTypes[stream].mute; } 448 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 449 float streamVolume_l(audio_stream_type_t stream) const 450 { return mStreamTypes[stream].volume; } 451 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 452 453 // allocate an audio_io_handle_t, session ID, or effect ID 454 uint32_t nextUniqueId(); 455 456 status_t moveEffectChain_l(int sessionId, 457 PlaybackThread *srcThread, 458 PlaybackThread *dstThread, 459 bool reRegister); 460 // return thread associated with primary hardware device, or NULL 461 PlaybackThread *primaryPlaybackThread_l() const; 462 audio_devices_t primaryOutputDevice_l() const; 463 464 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 465 466 467 void removeClient_l(pid_t pid); 468 void removeNotificationClient(pid_t pid); 469 470 bool isNonOffloadableGlobalEffectEnabled_l(); 471 void onNonOffloadableGlobalEffectEnable(); 472 473 class AudioHwDevice { 474 public: 475 enum Flags { 476 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 477 AHWD_CAN_SET_MASTER_MUTE = 0x2, 478 }; 479 480 AudioHwDevice(const char *moduleName, 481 audio_hw_device_t *hwDevice, 482 Flags flags) 483 : mModuleName(strdup(moduleName)) 484 , mHwDevice(hwDevice) 485 , mFlags(flags) { } 486 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 487 488 bool canSetMasterVolume() const { 489 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 490 } 491 492 bool canSetMasterMute() const { 493 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 494 } 495 496 const char *moduleName() const { return mModuleName; } 497 audio_hw_device_t *hwDevice() const { return mHwDevice; } 498 private: 499 const char * const mModuleName; 500 audio_hw_device_t * const mHwDevice; 501 Flags mFlags; 502 }; 503 504 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 505 // For emphasis, we could also make all pointers to them be "const *", 506 // but that would clutter the code unnecessarily. 507 508 struct AudioStreamOut { 509 AudioHwDevice* const audioHwDev; 510 audio_stream_out_t* const stream; 511 audio_output_flags_t flags; 512 513 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 514 515 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 516 audioHwDev(dev), stream(out), flags(flags) {} 517 }; 518 519 struct AudioStreamIn { 520 AudioHwDevice* const audioHwDev; 521 audio_stream_in_t* const stream; 522 523 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 524 525 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 526 audioHwDev(dev), stream(in) {} 527 }; 528 529 // for mAudioSessionRefs only 530 struct AudioSessionRef { 531 AudioSessionRef(int sessionid, pid_t pid) : 532 mSessionid(sessionid), mPid(pid), mCnt(1) {} 533 const int mSessionid; 534 const pid_t mPid; 535 int mCnt; 536 }; 537 538 mutable Mutex mLock; 539 540 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 541 542 mutable Mutex mHardwareLock; 543 // NOTE: If both mLock and mHardwareLock mutexes must be held, 544 // always take mLock before mHardwareLock 545 546 // These two fields are immutable after onFirstRef(), so no lock needed to access 547 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 548 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 549 550 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 551 enum hardware_call_state { 552 AUDIO_HW_IDLE = 0, // no operation in progress 553 AUDIO_HW_INIT, // init_check 554 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 555 AUDIO_HW_OUTPUT_CLOSE, // unused 556 AUDIO_HW_INPUT_OPEN, // unused 557 AUDIO_HW_INPUT_CLOSE, // unused 558 AUDIO_HW_STANDBY, // unused 559 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 560 AUDIO_HW_GET_ROUTING, // unused 561 AUDIO_HW_SET_ROUTING, // unused 562 AUDIO_HW_GET_MODE, // unused 563 AUDIO_HW_SET_MODE, // set_mode 564 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 565 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 566 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 567 AUDIO_HW_SET_PARAMETER, // set_parameters 568 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 569 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 570 AUDIO_HW_GET_PARAMETER, // get_parameters 571 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 572 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 573 }; 574 575 mutable hardware_call_state mHardwareStatus; // for dump only 576 577 578 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 579 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 580 581 // member variables below are protected by mLock 582 float mMasterVolume; 583 bool mMasterMute; 584 // end of variables protected by mLock 585 586 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 587 588 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 589 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 590 audio_mode_t mMode; 591 bool mBtNrecIsOff; 592 593 // protected by mLock 594 Vector<AudioSessionRef*> mAudioSessionRefs; 595 596 float masterVolume_l() const; 597 bool masterMute_l() const; 598 audio_module_handle_t loadHwModule_l(const char *name); 599 600 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 601 // to be created 602 603 private: 604 sp<Client> registerPid_l(pid_t pid); // always returns non-0 605 606 // for use from destructor 607 status_t closeOutput_nonvirtual(audio_io_handle_t output); 608 status_t closeInput_nonvirtual(audio_io_handle_t input); 609 610 #ifdef TEE_SINK 611 // all record threads serially share a common tee sink, which is re-created on format change 612 sp<NBAIO_Sink> mRecordTeeSink; 613 sp<NBAIO_Source> mRecordTeeSource; 614 #endif 615 616 public: 617 618 #ifdef TEE_SINK 619 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 620 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 621 622 // whether tee sink is enabled by property 623 static bool mTeeSinkInputEnabled; 624 static bool mTeeSinkOutputEnabled; 625 static bool mTeeSinkTrackEnabled; 626 627 // runtime configured size of each tee sink pipe, in frames 628 static size_t mTeeSinkInputFrames; 629 static size_t mTeeSinkOutputFrames; 630 static size_t mTeeSinkTrackFrames; 631 632 // compile-time default size of tee sink pipes, in frames 633 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 634 static const size_t kTeeSinkInputFramesDefault = 0x200000; 635 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 636 static const size_t kTeeSinkTrackFramesDefault = 0x1000; 637 #endif 638 639 // This method reads from a variable without mLock, but the variable is updated under mLock. So 640 // we might read a stale value, or a value that's inconsistent with respect to other variables. 641 // In this case, it's safe because the return value isn't used for making an important decision. 642 // The reason we don't want to take mLock is because it could block the caller for a long time. 643 bool isLowRamDevice() const { return mIsLowRamDevice; } 644 645 private: 646 bool mIsLowRamDevice; 647 bool mIsDeviceTypeKnown; 648 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 649 }; 650 651 #undef INCLUDING_FROM_AUDIOFLINGER_H 652 653 // ---------------------------------------------------------------------------- 654 655 }; // namespace android 656 657 #endif // ANDROID_AUDIO_FLINGER_H 658
