AudioFlinger.h revision 021cf9634ab09c0753a40b7c9ef4ba603be5c3da
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 61namespace android { 62 63struct audio_track_cblk_t; 64struct effect_param_cblk_t; 65class AudioMixer; 66class AudioBuffer; 67class AudioResampler; 68class FastMixer; 69class 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 82static 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 89class AudioFlinger : 90 public BinderService<AudioFlinger>, 91 public BnAudioFlinger 92{ 93 friend class BinderService<AudioFlinger>; // for AudioFlinger() 94public: 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 *pFrameCount, 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 int clientUid, 112 status_t *status /*non-NULL*/); 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 *pFrameCount, 120 IAudioFlinger::track_flags_t *flags, 121 pid_t tid, 122 int *sessionId, 123 sp<IMemory>& cblk, 124 sp<IMemory>& buffers, 125 status_t *status /*non-NULL*/); 126 127 virtual uint32_t sampleRate(audio_io_handle_t output) const; 128 virtual int channelCount(audio_io_handle_t output) const; 129 virtual audio_format_t format(audio_io_handle_t output) const; 130 virtual size_t frameCount(audio_io_handle_t output) const; 131 virtual uint32_t latency(audio_io_handle_t output) const; 132 133 virtual status_t setMasterVolume(float value); 134 virtual status_t setMasterMute(bool muted); 135 136 virtual float masterVolume() const; 137 virtual bool masterMute() const; 138 139 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 140 audio_io_handle_t output); 141 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 142 143 virtual float streamVolume(audio_stream_type_t stream, 144 audio_io_handle_t output) const; 145 virtual bool streamMute(audio_stream_type_t stream) const; 146 147 virtual status_t setMode(audio_mode_t mode); 148 149 virtual status_t setMicMute(bool state); 150 virtual bool getMicMute() const; 151 152 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 153 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 154 155 virtual void registerClient(const sp<IAudioFlingerClient>& client); 156 157 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 158 audio_channel_mask_t channelMask) const; 159 160 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 161 audio_devices_t *pDevices, 162 uint32_t *pSamplingRate, 163 audio_format_t *pFormat, 164 audio_channel_mask_t *pChannelMask, 165 uint32_t *pLatencyMs, 166 audio_output_flags_t flags, 167 const audio_offload_info_t *offloadInfo); 168 169 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 170 audio_io_handle_t output2); 171 172 virtual status_t closeOutput(audio_io_handle_t output); 173 174 virtual status_t suspendOutput(audio_io_handle_t output); 175 176 virtual status_t restoreOutput(audio_io_handle_t output); 177 178 virtual audio_io_handle_t openInput(audio_module_handle_t module, 179 audio_devices_t *pDevices, 180 uint32_t *pSamplingRate, 181 audio_format_t *pFormat, 182 audio_channel_mask_t *pChannelMask); 183 184 virtual status_t closeInput(audio_io_handle_t input); 185 186 virtual status_t invalidateStream(audio_stream_type_t stream); 187 188 virtual status_t setVoiceVolume(float volume); 189 190 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 191 audio_io_handle_t output) const; 192 193 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 194 195 virtual int newAudioSessionId(); 196 197 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 198 199 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 200 201 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 202 203 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 204 205 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 206 effect_descriptor_t *descriptor) const; 207 208 virtual sp<IEffect> createEffect( 209 effect_descriptor_t *pDesc, 210 const sp<IEffectClient>& effectClient, 211 int32_t priority, 212 audio_io_handle_t io, 213 int sessionId, 214 status_t *status /*non-NULL*/, 215 int *id, 216 int *enabled); 217 218 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 219 audio_io_handle_t dstOutput); 220 221 virtual audio_module_handle_t loadHwModule(const char *name); 222 223 virtual uint32_t getPrimaryOutputSamplingRate(); 224 virtual size_t getPrimaryOutputFrameCount(); 225 226 virtual status_t setLowRamDevice(bool isLowRamDevice); 227 228 virtual status_t onTransact( 229 uint32_t code, 230 const Parcel& data, 231 Parcel* reply, 232 uint32_t flags); 233 234 // end of IAudioFlinger interface 235 236 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 237 void unregisterWriter(const sp<NBLog::Writer>& writer); 238private: 239 static const size_t kLogMemorySize = 40 * 1024; 240 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 241 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 242 // for as long as possible. The memory is only freed when it is needed for another log writer. 243 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 244 Mutex mUnregisteredWritersLock; 245public: 246 247 class SyncEvent; 248 249 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 250 251 class SyncEvent : public RefBase { 252 public: 253 SyncEvent(AudioSystem::sync_event_t type, 254 int triggerSession, 255 int listenerSession, 256 sync_event_callback_t callBack, 257 wp<RefBase> cookie) 258 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 259 mCallback(callBack), mCookie(cookie) 260 {} 261 262 virtual ~SyncEvent() {} 263 264 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 265 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 266 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 267 AudioSystem::sync_event_t type() const { return mType; } 268 int triggerSession() const { return mTriggerSession; } 269 int listenerSession() const { return mListenerSession; } 270 wp<RefBase> cookie() const { return mCookie; } 271 272 private: 273 const AudioSystem::sync_event_t mType; 274 const int mTriggerSession; 275 const int mListenerSession; 276 sync_event_callback_t mCallback; 277 const wp<RefBase> mCookie; 278 mutable Mutex mLock; 279 }; 280 281 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 282 int triggerSession, 283 int listenerSession, 284 sync_event_callback_t callBack, 285 wp<RefBase> cookie); 286 287private: 288 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 289 290 audio_mode_t getMode() const { return mMode; } 291 292 bool btNrecIsOff() const { return mBtNrecIsOff; } 293 294 AudioFlinger() ANDROID_API; 295 virtual ~AudioFlinger(); 296 297 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 298 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 299 NO_INIT : NO_ERROR; } 300 301 // RefBase 302 virtual void onFirstRef(); 303 304 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 305 audio_devices_t devices); 306 void purgeStaleEffects_l(); 307 308 // standby delay for MIXER and DUPLICATING playback threads is read from property 309 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 310 static nsecs_t mStandbyTimeInNsecs; 311 312 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 313 // AudioFlinger::setParameters() updates, other threads read w/o lock 314 static uint32_t mScreenState; 315 316 // Internal dump utilities. 317 static const int kDumpLockRetries = 50; 318 static const int kDumpLockSleepUs = 20000; 319 static bool dumpTryLock(Mutex& mutex); 320 void dumpPermissionDenial(int fd, const Vector<String16>& args); 321 void dumpClients(int fd, const Vector<String16>& args); 322 void dumpInternals(int fd, const Vector<String16>& args); 323 324 // --- Client --- 325 class Client : public RefBase { 326 public: 327 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 328 virtual ~Client(); 329 sp<MemoryDealer> heap() const; 330 pid_t pid() const { return mPid; } 331 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 332 333 bool reserveTimedTrack(); 334 void releaseTimedTrack(); 335 336 private: 337 Client(const Client&); 338 Client& operator = (const Client&); 339 const sp<AudioFlinger> mAudioFlinger; 340 const sp<MemoryDealer> mMemoryDealer; 341 const pid_t mPid; 342 343 Mutex mTimedTrackLock; 344 int mTimedTrackCount; 345 }; 346 347 // --- Notification Client --- 348 class NotificationClient : public IBinder::DeathRecipient { 349 public: 350 NotificationClient(const sp<AudioFlinger>& audioFlinger, 351 const sp<IAudioFlingerClient>& client, 352 pid_t pid); 353 virtual ~NotificationClient(); 354 355 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 356 357 // IBinder::DeathRecipient 358 virtual void binderDied(const wp<IBinder>& who); 359 360 private: 361 NotificationClient(const NotificationClient&); 362 NotificationClient& operator = (const NotificationClient&); 363 364 const sp<AudioFlinger> mAudioFlinger; 365 const pid_t mPid; 366 const sp<IAudioFlingerClient> mAudioFlingerClient; 367 }; 368 369 class TrackHandle; 370 class RecordHandle; 371 class RecordThread; 372 class PlaybackThread; 373 class MixerThread; 374 class DirectOutputThread; 375 class OffloadThread; 376 class DuplicatingThread; 377 class AsyncCallbackThread; 378 class Track; 379 class RecordTrack; 380 class EffectModule; 381 class EffectHandle; 382 class EffectChain; 383 struct AudioStreamOut; 384 struct AudioStreamIn; 385 386 struct stream_type_t { 387 stream_type_t() 388 : volume(1.0f), 389 mute(false) 390 { 391 } 392 float volume; 393 bool mute; 394 }; 395 396 // --- PlaybackThread --- 397 398#include "Threads.h" 399 400#include "Effects.h" 401 402 // server side of the client's IAudioTrack 403 class TrackHandle : public android::BnAudioTrack { 404 public: 405 TrackHandle(const sp<PlaybackThread::Track>& track); 406 virtual ~TrackHandle(); 407 virtual sp<IMemory> getCblk() const; 408 virtual status_t start(); 409 virtual void stop(); 410 virtual void flush(); 411 virtual void pause(); 412 virtual status_t attachAuxEffect(int effectId); 413 virtual status_t allocateTimedBuffer(size_t size, 414 sp<IMemory>* buffer); 415 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 416 int64_t pts); 417 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 418 int target); 419 virtual status_t setParameters(const String8& keyValuePairs); 420 virtual status_t getTimestamp(AudioTimestamp& timestamp); 421 virtual void signal(); // signal playback thread for a change in control block 422 423 virtual status_t onTransact( 424 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 425 426 private: 427 const sp<PlaybackThread::Track> mTrack; 428 }; 429 430 // server side of the client's IAudioRecord 431 class RecordHandle : public android::BnAudioRecord { 432 public: 433 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 434 virtual ~RecordHandle(); 435 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 436 virtual void stop(); 437 virtual status_t onTransact( 438 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 439 private: 440 const sp<RecordThread::RecordTrack> mRecordTrack; 441 442 // for use from destructor 443 void stop_nonvirtual(); 444 }; 445 446 447 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 448 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 449 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 450 // no range check, AudioFlinger::mLock held 451 bool streamMute_l(audio_stream_type_t stream) const 452 { return mStreamTypes[stream].mute; } 453 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 454 float streamVolume_l(audio_stream_type_t stream) const 455 { return mStreamTypes[stream].volume; } 456 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 457 458 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 459 // They all share the same ID space, but the namespaces are actually independent 460 // because there are separate KeyedVectors for each kind of ID. 461 // The return value is uint32_t, but is cast to signed for some IDs. 462 // FIXME This API does not handle rollover to zero (for unsigned IDs), 463 // or from positive to negative (for signed IDs). 464 // Thus it may fail by returning an ID of the wrong sign, 465 // or by returning a non-unique ID. 466 uint32_t nextUniqueId(); 467 468 status_t moveEffectChain_l(int sessionId, 469 PlaybackThread *srcThread, 470 PlaybackThread *dstThread, 471 bool reRegister); 472 // return thread associated with primary hardware device, or NULL 473 PlaybackThread *primaryPlaybackThread_l() const; 474 audio_devices_t primaryOutputDevice_l() const; 475 476 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 477 478 479 void removeClient_l(pid_t pid); 480 void removeNotificationClient(pid_t pid); 481 bool isNonOffloadableGlobalEffectEnabled_l(); 482 void onNonOffloadableGlobalEffectEnable(); 483 484 class AudioHwDevice { 485 public: 486 enum Flags { 487 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 488 AHWD_CAN_SET_MASTER_MUTE = 0x2, 489 }; 490 491 AudioHwDevice(const char *moduleName, 492 audio_hw_device_t *hwDevice, 493 Flags flags) 494 : mModuleName(strdup(moduleName)) 495 , mHwDevice(hwDevice) 496 , mFlags(flags) { } 497 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 498 499 bool canSetMasterVolume() const { 500 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 501 } 502 503 bool canSetMasterMute() const { 504 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 505 } 506 507 const char *moduleName() const { return mModuleName; } 508 audio_hw_device_t *hwDevice() const { return mHwDevice; } 509 private: 510 const char * const mModuleName; 511 audio_hw_device_t * const mHwDevice; 512 const Flags mFlags; 513 }; 514 515 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 516 // For emphasis, we could also make all pointers to them be "const *", 517 // but that would clutter the code unnecessarily. 518 519 struct AudioStreamOut { 520 AudioHwDevice* const audioHwDev; 521 audio_stream_out_t* const stream; 522 const audio_output_flags_t flags; 523 524 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 525 526 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 527 audioHwDev(dev), stream(out), flags(flags) {} 528 }; 529 530 struct AudioStreamIn { 531 AudioHwDevice* const audioHwDev; 532 audio_stream_in_t* const stream; 533 534 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 535 536 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 537 audioHwDev(dev), stream(in) {} 538 }; 539 540 // for mAudioSessionRefs only 541 struct AudioSessionRef { 542 AudioSessionRef(int sessionid, pid_t pid) : 543 mSessionid(sessionid), mPid(pid), mCnt(1) {} 544 const int mSessionid; 545 const pid_t mPid; 546 int mCnt; 547 }; 548 549 mutable Mutex mLock; 550 // protects mClients and mNotificationClients. 551 // must be locked after mLock and ThreadBase::mLock if both must be locked 552 // avoids acquiring AudioFlinger::mLock from inside thread loop. 553 mutable Mutex mClientLock; 554 // protected by mClientLock 555 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 556 557 mutable Mutex mHardwareLock; 558 // NOTE: If both mLock and mHardwareLock mutexes must be held, 559 // always take mLock before mHardwareLock 560 561 // These two fields are immutable after onFirstRef(), so no lock needed to access 562 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 563 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 564 565 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 566 enum hardware_call_state { 567 AUDIO_HW_IDLE = 0, // no operation in progress 568 AUDIO_HW_INIT, // init_check 569 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 570 AUDIO_HW_OUTPUT_CLOSE, // unused 571 AUDIO_HW_INPUT_OPEN, // unused 572 AUDIO_HW_INPUT_CLOSE, // unused 573 AUDIO_HW_STANDBY, // unused 574 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 575 AUDIO_HW_GET_ROUTING, // unused 576 AUDIO_HW_SET_ROUTING, // unused 577 AUDIO_HW_GET_MODE, // unused 578 AUDIO_HW_SET_MODE, // set_mode 579 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 580 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 581 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 582 AUDIO_HW_SET_PARAMETER, // set_parameters 583 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 584 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 585 AUDIO_HW_GET_PARAMETER, // get_parameters 586 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 587 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 588 }; 589 590 mutable hardware_call_state mHardwareStatus; // for dump only 591 592 593 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 594 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 595 596 // member variables below are protected by mLock 597 float mMasterVolume; 598 bool mMasterMute; 599 // end of variables protected by mLock 600 601 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 602 603 // protected by mClientLock 604 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 605 606 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 607 // nextUniqueId() returns uint32_t, but this is declared int32_t 608 // because the atomic operations require an int32_t 609 610 audio_mode_t mMode; 611 bool mBtNrecIsOff; 612 613 // protected by mLock 614 Vector<AudioSessionRef*> mAudioSessionRefs; 615 616 float masterVolume_l() const; 617 bool masterMute_l() const; 618 audio_module_handle_t loadHwModule_l(const char *name); 619 620 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 621 // to be created 622 623private: 624 sp<Client> registerPid(pid_t pid); // always returns non-0 625 626 // for use from destructor 627 status_t closeOutput_nonvirtual(audio_io_handle_t output); 628 status_t closeInput_nonvirtual(audio_io_handle_t input); 629 630#ifdef TEE_SINK 631 // all record threads serially share a common tee sink, which is re-created on format change 632 sp<NBAIO_Sink> mRecordTeeSink; 633 sp<NBAIO_Source> mRecordTeeSource; 634#endif 635 636public: 637 638#ifdef TEE_SINK 639 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 640 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 641 642 // whether tee sink is enabled by property 643 static bool mTeeSinkInputEnabled; 644 static bool mTeeSinkOutputEnabled; 645 static bool mTeeSinkTrackEnabled; 646 647 // runtime configured size of each tee sink pipe, in frames 648 static size_t mTeeSinkInputFrames; 649 static size_t mTeeSinkOutputFrames; 650 static size_t mTeeSinkTrackFrames; 651 652 // compile-time default size of tee sink pipes, in frames 653 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 654 static const size_t kTeeSinkInputFramesDefault = 0x200000; 655 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 656 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 657#endif 658 659 // This method reads from a variable without mLock, but the variable is updated under mLock. So 660 // we might read a stale value, or a value that's inconsistent with respect to other variables. 661 // In this case, it's safe because the return value isn't used for making an important decision. 662 // The reason we don't want to take mLock is because it could block the caller for a long time. 663 bool isLowRamDevice() const { return mIsLowRamDevice; } 664 665private: 666 bool mIsLowRamDevice; 667 bool mIsDeviceTypeKnown; 668 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 669}; 670 671#undef INCLUDING_FROM_AUDIOFLINGER_H 672 673const char *formatToString(audio_format_t format); 674 675// ---------------------------------------------------------------------------- 676 677}; // namespace android 678 679#endif // ANDROID_AUDIO_FLINGER_H 680