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: getServiceName()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 int clientUid, 113 status_t *status); 114 115 virtual sp<IAudioRecord> openRecord( 116 audio_io_handle_t input, 117 uint32_t sampleRate, 118 audio_format_t format, 119 audio_channel_mask_t channelMask, 120 size_t frameCount, 121 IAudioFlinger::track_flags_t *flags, 122 pid_t tid, 123 int *sessionId, 124 status_t *status); 125 126 virtual uint32_t sampleRate(audio_io_handle_t output) const; 127 virtual int channelCount(audio_io_handle_t output) const; 128 virtual audio_format_t format(audio_io_handle_t output) const; 129 virtual size_t frameCount(audio_io_handle_t output) const; 130 virtual uint32_t latency(audio_io_handle_t output) const; 131 132 virtual status_t setMasterVolume(float value); 133 virtual status_t setMasterMute(bool muted); 134 135 virtual float masterVolume() const; 136 virtual bool masterMute() const; 137 138 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 139 audio_io_handle_t output); 140 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 141 142 virtual float streamVolume(audio_stream_type_t stream, 143 audio_io_handle_t output) const; 144 virtual bool streamMute(audio_stream_type_t stream) const; 145 146 virtual status_t setMode(audio_mode_t mode); 147 148 virtual status_t setMicMute(bool state); 149 virtual bool getMicMute() const; 150 151 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 152 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 153 154 virtual void registerClient(const sp<IAudioFlingerClient>& client); 155 156 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 157 audio_channel_mask_t channelMask) const; 158 159 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 160 audio_devices_t *pDevices, 161 uint32_t *pSamplingRate, 162 audio_format_t *pFormat, 163 audio_channel_mask_t *pChannelMask, 164 uint32_t *pLatencyMs, 165 audio_output_flags_t flags, 166 const audio_offload_info_t *offloadInfo); 167 168 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 169 audio_io_handle_t output2); 170 171 virtual status_t closeOutput(audio_io_handle_t output); 172 173 virtual status_t suspendOutput(audio_io_handle_t output); 174 175 virtual status_t restoreOutput(audio_io_handle_t output); 176 177 virtual audio_io_handle_t openInput(audio_module_handle_t module, 178 audio_devices_t *pDevices, 179 uint32_t *pSamplingRate, 180 audio_format_t *pFormat, 181 audio_channel_mask_t *pChannelMask); 182 183 virtual status_t closeInput(audio_io_handle_t input); 184 185 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 186 187 virtual status_t setVoiceVolume(float volume); 188 189 virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, 190 audio_io_handle_t output) const; 191 192 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 193 194 virtual int newAudioSessionId(); 195 196 virtual void acquireAudioSessionId(int audioSession); 197 198 virtual void releaseAudioSessionId(int audioSession); 199 200 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 201 202 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 203 204 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 205 effect_descriptor_t *descriptor) const; 206 207 virtual sp<IEffect> createEffect( 208 effect_descriptor_t *pDesc, 209 const sp<IEffectClient>& effectClient, 210 int32_t priority, 211 audio_io_handle_t io, 212 int sessionId, 213 status_t *status, 214 int *id, 215 int *enabled); 216 217 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 218 audio_io_handle_t dstOutput); 219 220 virtual audio_module_handle_t loadHwModule(const char *name); 221 222 virtual uint32_t getPrimaryOutputSamplingRate(); 223 virtual size_t getPrimaryOutputFrameCount(); 224 225 virtual status_t setLowRamDevice(bool isLowRamDevice); 226 227 virtual status_t onTransact( 228 uint32_t code, 229 const Parcel& data, 230 Parcel* reply, 231 uint32_t flags); 232 233 // end of IAudioFlinger interface 234 235 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 236 void unregisterWriter(const sp<NBLog::Writer>& writer); 237 private: 238 static const size_t kLogMemorySize = 10 * 1024; 239 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 240 public: 241 242 class SyncEvent; 243 244 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 245 246 class SyncEvent : public RefBase { 247 public: SyncEvent(AudioSystem::sync_event_t type,int triggerSession,int listenerSession,sync_event_callback_t callBack,void * cookie)248 SyncEvent(AudioSystem::sync_event_t type, 249 int triggerSession, 250 int listenerSession, 251 sync_event_callback_t callBack, 252 void *cookie) 253 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 254 mCallback(callBack), mCookie(cookie) 255 {} 256 ~SyncEvent()257 virtual ~SyncEvent() {} 258 trigger()259 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } isCancelled()260 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } cancel()261 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } type()262 AudioSystem::sync_event_t type() const { return mType; } triggerSession()263 int triggerSession() const { return mTriggerSession; } listenerSession()264 int listenerSession() const { return mListenerSession; } cookie()265 void *cookie() const { return mCookie; } 266 267 private: 268 const AudioSystem::sync_event_t mType; 269 const int mTriggerSession; 270 const int mListenerSession; 271 sync_event_callback_t mCallback; 272 void * const mCookie; 273 mutable Mutex mLock; 274 }; 275 276 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 277 int triggerSession, 278 int listenerSession, 279 sync_event_callback_t callBack, 280 void *cookie); 281 282 private: 283 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 284 getMode()285 audio_mode_t getMode() const { return mMode; } 286 btNrecIsOff()287 bool btNrecIsOff() const { return mBtNrecIsOff; } 288 289 AudioFlinger() ANDROID_API; 290 virtual ~AudioFlinger(); 291 292 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev initCheck()293 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 294 NO_INIT : NO_ERROR; } 295 296 // RefBase 297 virtual void onFirstRef(); 298 299 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 300 audio_devices_t devices); 301 void purgeStaleEffects_l(); 302 303 // standby delay for MIXER and DUPLICATING playback threads is read from property 304 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 305 static nsecs_t mStandbyTimeInNsecs; 306 307 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 308 // AudioFlinger::setParameters() updates, other threads read w/o lock 309 static uint32_t mScreenState; 310 311 // Internal dump utilities. 312 static const int kDumpLockRetries = 50; 313 static const int kDumpLockSleepUs = 20000; 314 static bool dumpTryLock(Mutex& mutex); 315 void dumpPermissionDenial(int fd, const Vector<String16>& args); 316 void dumpClients(int fd, const Vector<String16>& args); 317 void dumpInternals(int fd, const Vector<String16>& args); 318 319 // --- Client --- 320 class Client : public RefBase { 321 public: 322 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 323 virtual ~Client(); 324 sp<MemoryDealer> heap() const; pid()325 pid_t pid() const { return mPid; } audioFlinger()326 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 327 328 bool reserveTimedTrack(); 329 void releaseTimedTrack(); 330 331 private: 332 Client(const Client&); 333 Client& operator = (const Client&); 334 const sp<AudioFlinger> mAudioFlinger; 335 const sp<MemoryDealer> mMemoryDealer; 336 const pid_t mPid; 337 338 Mutex mTimedTrackLock; 339 int mTimedTrackCount; 340 }; 341 342 // --- Notification Client --- 343 class NotificationClient : public IBinder::DeathRecipient { 344 public: 345 NotificationClient(const sp<AudioFlinger>& audioFlinger, 346 const sp<IAudioFlingerClient>& client, 347 pid_t pid); 348 virtual ~NotificationClient(); 349 audioFlingerClient()350 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 351 352 // IBinder::DeathRecipient 353 virtual void binderDied(const wp<IBinder>& who); 354 355 private: 356 NotificationClient(const NotificationClient&); 357 NotificationClient& operator = (const NotificationClient&); 358 359 const sp<AudioFlinger> mAudioFlinger; 360 const pid_t mPid; 361 const sp<IAudioFlingerClient> mAudioFlingerClient; 362 }; 363 364 class TrackHandle; 365 class RecordHandle; 366 class RecordThread; 367 class PlaybackThread; 368 class MixerThread; 369 class DirectOutputThread; 370 class OffloadThread; 371 class DuplicatingThread; 372 class AsyncCallbackThread; 373 class Track; 374 class RecordTrack; 375 class EffectModule; 376 class EffectHandle; 377 class EffectChain; 378 struct AudioStreamOut; 379 struct AudioStreamIn; 380 381 struct stream_type_t { stream_type_tstream_type_t382 stream_type_t() 383 : volume(1.0f), 384 mute(false) 385 { 386 } 387 float volume; 388 bool mute; 389 }; 390 391 // --- PlaybackThread --- 392 393 #include "Threads.h" 394 395 #include "Effects.h" 396 397 // server side of the client's IAudioTrack 398 class TrackHandle : public android::BnAudioTrack { 399 public: 400 TrackHandle(const sp<PlaybackThread::Track>& track); 401 virtual ~TrackHandle(); 402 virtual sp<IMemory> getCblk() const; 403 virtual status_t start(); 404 virtual void stop(); 405 virtual void flush(); 406 virtual void pause(); 407 virtual status_t attachAuxEffect(int effectId); 408 virtual status_t allocateTimedBuffer(size_t size, 409 sp<IMemory>* buffer); 410 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 411 int64_t pts); 412 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 413 int target); 414 virtual status_t setParameters(const String8& keyValuePairs); 415 virtual status_t getTimestamp(AudioTimestamp& timestamp); 416 virtual void signal(); // signal playback thread for a change in control block 417 418 virtual status_t onTransact( 419 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 420 421 private: 422 const sp<PlaybackThread::Track> mTrack; 423 }; 424 425 // server side of the client's IAudioRecord 426 class RecordHandle : public android::BnAudioRecord { 427 public: 428 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 429 virtual ~RecordHandle(); 430 virtual sp<IMemory> getCblk() const; 431 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 432 virtual void stop(); 433 virtual status_t onTransact( 434 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 435 private: 436 const sp<RecordThread::RecordTrack> mRecordTrack; 437 438 // for use from destructor 439 void stop_nonvirtual(); 440 }; 441 442 443 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 444 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 445 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 446 // no range check, AudioFlinger::mLock held streamMute_l(audio_stream_type_t stream)447 bool streamMute_l(audio_stream_type_t stream) const 448 { return mStreamTypes[stream].mute; } 449 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held streamVolume_l(audio_stream_type_t stream)450 float streamVolume_l(audio_stream_type_t stream) const 451 { return mStreamTypes[stream].volume; } 452 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 453 454 // allocate an audio_io_handle_t, session ID, or effect ID 455 uint32_t nextUniqueId(); 456 457 status_t moveEffectChain_l(int sessionId, 458 PlaybackThread *srcThread, 459 PlaybackThread *dstThread, 460 bool reRegister); 461 // return thread associated with primary hardware device, or NULL 462 PlaybackThread *primaryPlaybackThread_l() const; 463 audio_devices_t primaryOutputDevice_l() const; 464 465 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 466 467 468 void removeClient_l(pid_t pid); 469 void removeNotificationClient(pid_t pid); 470 471 bool isNonOffloadableGlobalEffectEnabled_l(); 472 void onNonOffloadableGlobalEffectEnable(); 473 474 class AudioHwDevice { 475 public: 476 enum Flags { 477 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 478 AHWD_CAN_SET_MASTER_MUTE = 0x2, 479 }; 480 AudioHwDevice(const char * moduleName,audio_hw_device_t * hwDevice,Flags flags)481 AudioHwDevice(const char *moduleName, 482 audio_hw_device_t *hwDevice, 483 Flags flags) 484 : mModuleName(strdup(moduleName)) 485 , mHwDevice(hwDevice) 486 , mFlags(flags) { } ~AudioHwDevice()487 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 488 canSetMasterVolume()489 bool canSetMasterVolume() const { 490 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 491 } 492 canSetMasterMute()493 bool canSetMasterMute() const { 494 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 495 } 496 moduleName()497 const char *moduleName() const { return mModuleName; } hwDevice()498 audio_hw_device_t *hwDevice() const { return mHwDevice; } 499 private: 500 const char * const mModuleName; 501 audio_hw_device_t * const mHwDevice; 502 Flags mFlags; 503 }; 504 505 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 506 // For emphasis, we could also make all pointers to them be "const *", 507 // but that would clutter the code unnecessarily. 508 509 struct AudioStreamOut { 510 AudioHwDevice* const audioHwDev; 511 audio_stream_out_t* const stream; 512 audio_output_flags_t flags; 513 hwDevAudioStreamOut514 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 515 AudioStreamOutAudioStreamOut516 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 517 audioHwDev(dev), stream(out), flags(flags) {} 518 }; 519 520 struct AudioStreamIn { 521 AudioHwDevice* const audioHwDev; 522 audio_stream_in_t* const stream; 523 hwDevAudioStreamIn524 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 525 AudioStreamInAudioStreamIn526 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 527 audioHwDev(dev), stream(in) {} 528 }; 529 530 // for mAudioSessionRefs only 531 struct AudioSessionRef { AudioSessionRefAudioSessionRef532 AudioSessionRef(int sessionid, pid_t pid) : 533 mSessionid(sessionid), mPid(pid), mCnt(1) {} 534 const int mSessionid; 535 const pid_t mPid; 536 int mCnt; 537 }; 538 539 mutable Mutex mLock; 540 541 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 542 543 mutable Mutex mHardwareLock; 544 // NOTE: If both mLock and mHardwareLock mutexes must be held, 545 // always take mLock before mHardwareLock 546 547 // These two fields are immutable after onFirstRef(), so no lock needed to access 548 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 549 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 550 551 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 552 enum hardware_call_state { 553 AUDIO_HW_IDLE = 0, // no operation in progress 554 AUDIO_HW_INIT, // init_check 555 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 556 AUDIO_HW_OUTPUT_CLOSE, // unused 557 AUDIO_HW_INPUT_OPEN, // unused 558 AUDIO_HW_INPUT_CLOSE, // unused 559 AUDIO_HW_STANDBY, // unused 560 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 561 AUDIO_HW_GET_ROUTING, // unused 562 AUDIO_HW_SET_ROUTING, // unused 563 AUDIO_HW_GET_MODE, // unused 564 AUDIO_HW_SET_MODE, // set_mode 565 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 566 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 567 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 568 AUDIO_HW_SET_PARAMETER, // set_parameters 569 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 570 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 571 AUDIO_HW_GET_PARAMETER, // get_parameters 572 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 573 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 574 }; 575 576 mutable hardware_call_state mHardwareStatus; // for dump only 577 578 579 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 580 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 581 582 // member variables below are protected by mLock 583 float mMasterVolume; 584 bool mMasterMute; 585 // end of variables protected by mLock 586 587 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 588 589 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 590 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 591 audio_mode_t mMode; 592 bool mBtNrecIsOff; 593 594 // protected by mLock 595 Vector<AudioSessionRef*> mAudioSessionRefs; 596 597 float masterVolume_l() const; 598 bool masterMute_l() const; 599 audio_module_handle_t loadHwModule_l(const char *name); 600 601 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 602 // to be created 603 604 private: 605 sp<Client> registerPid_l(pid_t pid); // always returns non-0 606 607 // for use from destructor 608 status_t closeOutput_nonvirtual(audio_io_handle_t output); 609 status_t closeInput_nonvirtual(audio_io_handle_t input); 610 611 #ifdef TEE_SINK 612 // all record threads serially share a common tee sink, which is re-created on format change 613 sp<NBAIO_Sink> mRecordTeeSink; 614 sp<NBAIO_Source> mRecordTeeSource; 615 #endif 616 617 public: 618 619 #ifdef TEE_SINK 620 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 621 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 622 623 // whether tee sink is enabled by property 624 static bool mTeeSinkInputEnabled; 625 static bool mTeeSinkOutputEnabled; 626 static bool mTeeSinkTrackEnabled; 627 628 // runtime configured size of each tee sink pipe, in frames 629 static size_t mTeeSinkInputFrames; 630 static size_t mTeeSinkOutputFrames; 631 static size_t mTeeSinkTrackFrames; 632 633 // compile-time default size of tee sink pipes, in frames 634 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 635 static const size_t kTeeSinkInputFramesDefault = 0x200000; 636 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 637 static const size_t kTeeSinkTrackFramesDefault = 0x1000; 638 #endif 639 640 // This method reads from a variable without mLock, but the variable is updated under mLock. So 641 // we might read a stale value, or a value that's inconsistent with respect to other variables. 642 // In this case, it's safe because the return value isn't used for making an important decision. 643 // The reason we don't want to take mLock is because it could block the caller for a long time. isLowRamDevice()644 bool isLowRamDevice() const { return mIsLowRamDevice; } 645 646 private: 647 bool mIsLowRamDevice; 648 bool mIsDeviceTypeKnown; 649 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 650 }; 651 652 #undef INCLUDING_FROM_AUDIOFLINGER_H 653 654 // ---------------------------------------------------------------------------- 655 656 }; // namespace android 657 658 #endif // ANDROID_AUDIO_FLINGER_H 659