AudioFlinger.h revision 1035194cee4fbd57e35ea15c56e66cd09b63d56e
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 status_t *status /*non-NULL*/); 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 invalidateStream(audio_stream_type_t stream); 185 186 virtual status_t setVoiceVolume(float volume); 187 188 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 189 audio_io_handle_t output) const; 190 191 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 192 193 virtual int newAudioSessionId(); 194 195 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 196 197 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 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 /*non-NULL*/, 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); 236private: 237 static const size_t kLogMemorySize = 40 * 1024; 238 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 239 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 240 // for as long as possible. The memory is only freed when it is needed for another log writer. 241 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 242 Mutex mUnregisteredWritersLock; 243public: 244 245 class SyncEvent; 246 247 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 248 249 class SyncEvent : public RefBase { 250 public: 251 SyncEvent(AudioSystem::sync_event_t type, 252 int triggerSession, 253 int listenerSession, 254 sync_event_callback_t callBack, 255 wp<RefBase> cookie) 256 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 257 mCallback(callBack), mCookie(cookie) 258 {} 259 260 virtual ~SyncEvent() {} 261 262 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 263 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 264 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 265 AudioSystem::sync_event_t type() const { return mType; } 266 int triggerSession() const { return mTriggerSession; } 267 int listenerSession() const { return mListenerSession; } 268 wp<RefBase> cookie() const { return mCookie; } 269 270 private: 271 const AudioSystem::sync_event_t mType; 272 const int mTriggerSession; 273 const int mListenerSession; 274 sync_event_callback_t mCallback; 275 const wp<RefBase> mCookie; 276 mutable Mutex mLock; 277 }; 278 279 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 280 int triggerSession, 281 int listenerSession, 282 sync_event_callback_t callBack, 283 wp<RefBase> cookie); 284 285private: 286 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 287 288 audio_mode_t getMode() const { return mMode; } 289 290 bool btNrecIsOff() const { return mBtNrecIsOff; } 291 292 AudioFlinger() ANDROID_API; 293 virtual ~AudioFlinger(); 294 295 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 296 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 297 NO_INIT : NO_ERROR; } 298 299 // RefBase 300 virtual void onFirstRef(); 301 302 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 303 audio_devices_t devices); 304 void purgeStaleEffects_l(); 305 306 // standby delay for MIXER and DUPLICATING playback threads is read from property 307 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 308 static nsecs_t mStandbyTimeInNsecs; 309 310 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 311 // AudioFlinger::setParameters() updates, other threads read w/o lock 312 static uint32_t mScreenState; 313 314 // Internal dump utilities. 315 static const int kDumpLockRetries = 50; 316 static const int kDumpLockSleepUs = 20000; 317 static bool dumpTryLock(Mutex& mutex); 318 void dumpPermissionDenial(int fd, const Vector<String16>& args); 319 void dumpClients(int fd, const Vector<String16>& args); 320 void dumpInternals(int fd, const Vector<String16>& args); 321 322 // --- Client --- 323 class Client : public RefBase { 324 public: 325 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 326 virtual ~Client(); 327 sp<MemoryDealer> heap() const; 328 pid_t pid() const { return mPid; } 329 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 330 331 bool reserveTimedTrack(); 332 void releaseTimedTrack(); 333 334 private: 335 Client(const Client&); 336 Client& operator = (const Client&); 337 const sp<AudioFlinger> mAudioFlinger; 338 const sp<MemoryDealer> mMemoryDealer; 339 const pid_t mPid; 340 341 Mutex mTimedTrackLock; 342 int mTimedTrackCount; 343 }; 344 345 // --- Notification Client --- 346 class NotificationClient : public IBinder::DeathRecipient { 347 public: 348 NotificationClient(const sp<AudioFlinger>& audioFlinger, 349 const sp<IAudioFlingerClient>& client, 350 pid_t pid); 351 virtual ~NotificationClient(); 352 353 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 354 355 // IBinder::DeathRecipient 356 virtual void binderDied(const wp<IBinder>& who); 357 358 private: 359 NotificationClient(const NotificationClient&); 360 NotificationClient& operator = (const NotificationClient&); 361 362 const sp<AudioFlinger> mAudioFlinger; 363 const pid_t mPid; 364 const sp<IAudioFlingerClient> mAudioFlingerClient; 365 }; 366 367 class TrackHandle; 368 class RecordHandle; 369 class RecordThread; 370 class PlaybackThread; 371 class MixerThread; 372 class DirectOutputThread; 373 class OffloadThread; 374 class DuplicatingThread; 375 class AsyncCallbackThread; 376 class Track; 377 class RecordTrack; 378 class EffectModule; 379 class EffectHandle; 380 class EffectChain; 381 struct AudioStreamOut; 382 struct AudioStreamIn; 383 384 struct stream_type_t { 385 stream_type_t() 386 : volume(1.0f), 387 mute(false) 388 { 389 } 390 float volume; 391 bool mute; 392 }; 393 394 // --- PlaybackThread --- 395 396#include "Threads.h" 397 398#include "Effects.h" 399 400 // server side of the client's IAudioTrack 401 class TrackHandle : public android::BnAudioTrack { 402 public: 403 TrackHandle(const sp<PlaybackThread::Track>& track); 404 virtual ~TrackHandle(); 405 virtual sp<IMemory> getCblk() const; 406 virtual status_t start(); 407 virtual void stop(); 408 virtual void flush(); 409 virtual void pause(); 410 virtual status_t attachAuxEffect(int effectId); 411 virtual status_t allocateTimedBuffer(size_t size, 412 sp<IMemory>* buffer); 413 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 414 int64_t pts); 415 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 416 int target); 417 virtual status_t setParameters(const String8& keyValuePairs); 418 virtual status_t getTimestamp(AudioTimestamp& timestamp); 419 virtual void signal(); // signal playback thread for a change in control block 420 421 virtual status_t onTransact( 422 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 423 424 private: 425 const sp<PlaybackThread::Track> mTrack; 426 }; 427 428 // server side of the client's IAudioRecord 429 class RecordHandle : public android::BnAudioRecord { 430 public: 431 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 432 virtual ~RecordHandle(); 433 virtual sp<IMemory> getCblk() const; 434 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 435 virtual void stop(); 436 virtual status_t onTransact( 437 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 438 private: 439 const sp<RecordThread::RecordTrack> mRecordTrack; 440 441 // for use from destructor 442 void stop_nonvirtual(); 443 }; 444 445 446 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 447 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 448 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 449 // no range check, AudioFlinger::mLock held 450 bool streamMute_l(audio_stream_type_t stream) const 451 { return mStreamTypes[stream].mute; } 452 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 453 float streamVolume_l(audio_stream_type_t stream) const 454 { return mStreamTypes[stream].volume; } 455 void audioConfigChanged_l(const DefaultKeyedVector< pid_t,sp<NotificationClient> >& 456 notificationClients, 457 int event, 458 audio_io_handle_t ioHandle, 459 const void *param2); 460 461 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 462 // They all share the same ID space, but the namespaces are actually independent 463 // because there are separate KeyedVectors for each kind of ID. 464 // The return value is uint32_t, but is cast to signed for some IDs. 465 // FIXME This API does not handle rollover to zero (for unsigned IDs), 466 // or from positive to negative (for signed IDs). 467 // Thus it may fail by returning an ID of the wrong sign, 468 // or by returning a non-unique ID. 469 uint32_t nextUniqueId(); 470 471 status_t moveEffectChain_l(int sessionId, 472 PlaybackThread *srcThread, 473 PlaybackThread *dstThread, 474 bool reRegister); 475 // return thread associated with primary hardware device, or NULL 476 PlaybackThread *primaryPlaybackThread_l() const; 477 audio_devices_t primaryOutputDevice_l() const; 478 479 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 480 481 482 void removeClient_l(pid_t pid); 483 void removeNotificationClient(pid_t pid); 484 DefaultKeyedVector< pid_t,sp<NotificationClient> > notificationClients() { 485 Mutex::Autolock _l(mLock); return mNotificationClients; } 486 bool isNonOffloadableGlobalEffectEnabled_l(); 487 void onNonOffloadableGlobalEffectEnable(); 488 489 class AudioHwDevice { 490 public: 491 enum Flags { 492 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 493 AHWD_CAN_SET_MASTER_MUTE = 0x2, 494 }; 495 496 AudioHwDevice(const char *moduleName, 497 audio_hw_device_t *hwDevice, 498 Flags flags) 499 : mModuleName(strdup(moduleName)) 500 , mHwDevice(hwDevice) 501 , mFlags(flags) { } 502 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 503 504 bool canSetMasterVolume() const { 505 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 506 } 507 508 bool canSetMasterMute() const { 509 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 510 } 511 512 const char *moduleName() const { return mModuleName; } 513 audio_hw_device_t *hwDevice() const { return mHwDevice; } 514 private: 515 const char * const mModuleName; 516 audio_hw_device_t * const mHwDevice; 517 const Flags mFlags; 518 }; 519 520 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 521 // For emphasis, we could also make all pointers to them be "const *", 522 // but that would clutter the code unnecessarily. 523 524 struct AudioStreamOut { 525 AudioHwDevice* const audioHwDev; 526 audio_stream_out_t* const stream; 527 const audio_output_flags_t flags; 528 529 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 530 531 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 532 audioHwDev(dev), stream(out), flags(flags) {} 533 }; 534 535 struct AudioStreamIn { 536 AudioHwDevice* const audioHwDev; 537 audio_stream_in_t* const stream; 538 539 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 540 541 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 542 audioHwDev(dev), stream(in) {} 543 }; 544 545 // for mAudioSessionRefs only 546 struct AudioSessionRef { 547 AudioSessionRef(int sessionid, pid_t pid) : 548 mSessionid(sessionid), mPid(pid), mCnt(1) {} 549 const int mSessionid; 550 const pid_t mPid; 551 int mCnt; 552 }; 553 554 mutable Mutex mLock; 555 556 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 557 558 mutable Mutex mHardwareLock; 559 // NOTE: If both mLock and mHardwareLock mutexes must be held, 560 // always take mLock before mHardwareLock 561 562 // These two fields are immutable after onFirstRef(), so no lock needed to access 563 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 564 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 565 566 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 567 enum hardware_call_state { 568 AUDIO_HW_IDLE = 0, // no operation in progress 569 AUDIO_HW_INIT, // init_check 570 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 571 AUDIO_HW_OUTPUT_CLOSE, // unused 572 AUDIO_HW_INPUT_OPEN, // unused 573 AUDIO_HW_INPUT_CLOSE, // unused 574 AUDIO_HW_STANDBY, // unused 575 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 576 AUDIO_HW_GET_ROUTING, // unused 577 AUDIO_HW_SET_ROUTING, // unused 578 AUDIO_HW_GET_MODE, // unused 579 AUDIO_HW_SET_MODE, // set_mode 580 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 581 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 582 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 583 AUDIO_HW_SET_PARAMETER, // set_parameters 584 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 585 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 586 AUDIO_HW_GET_PARAMETER, // get_parameters 587 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 588 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 589 }; 590 591 mutable hardware_call_state mHardwareStatus; // for dump only 592 593 594 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 595 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 596 597 // member variables below are protected by mLock 598 float mMasterVolume; 599 bool mMasterMute; 600 // end of variables protected by mLock 601 602 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 603 604 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_l(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