AudioFlinger.h revision 4182c4e2a07e2441fcd5c22eaff0ddfe7f826f61
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 63class audio_track_cblk_t; 64class 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 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 status_t *status); 112 113 virtual sp<IAudioRecord> openRecord( 114 audio_io_handle_t input, 115 uint32_t sampleRate, 116 audio_format_t format, 117 audio_channel_mask_t channelMask, 118 size_t frameCount, 119 IAudioFlinger::track_flags_t flags, 120 pid_t tid, 121 int *sessionId, 122 status_t *status); 123 124 virtual uint32_t sampleRate(audio_io_handle_t output) const; 125 virtual int channelCount(audio_io_handle_t output) const; 126 virtual audio_format_t format(audio_io_handle_t output) const; 127 virtual size_t frameCount(audio_io_handle_t output) const; 128 virtual uint32_t latency(audio_io_handle_t output) const; 129 130 virtual status_t setMasterVolume(float value); 131 virtual status_t setMasterMute(bool muted); 132 133 virtual float masterVolume() const; 134 virtual bool masterMute() const; 135 136 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 137 audio_io_handle_t output); 138 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 139 140 virtual float streamVolume(audio_stream_type_t stream, 141 audio_io_handle_t output) const; 142 virtual bool streamMute(audio_stream_type_t stream) const; 143 144 virtual status_t setMode(audio_mode_t mode); 145 146 virtual status_t setMicMute(bool state); 147 virtual bool getMicMute() const; 148 149 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 150 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 151 152 virtual void registerClient(const sp<IAudioFlingerClient>& client); 153 154 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 155 audio_channel_mask_t channelMask) const; 156 157 virtual audio_io_handle_t openOutput(audio_module_handle_t module, 158 audio_devices_t *pDevices, 159 uint32_t *pSamplingRate, 160 audio_format_t *pFormat, 161 audio_channel_mask_t *pChannelMask, 162 uint32_t *pLatencyMs, 163 audio_output_flags_t flags, 164 const audio_offload_info_t *offloadInfo); 165 166 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 167 audio_io_handle_t output2); 168 169 virtual status_t closeOutput(audio_io_handle_t output); 170 171 virtual status_t suspendOutput(audio_io_handle_t output); 172 173 virtual status_t restoreOutput(audio_io_handle_t output); 174 175 virtual audio_io_handle_t openInput(audio_module_handle_t module, 176 audio_devices_t *pDevices, 177 uint32_t *pSamplingRate, 178 audio_format_t *pFormat, 179 audio_channel_mask_t *pChannelMask); 180 181 virtual status_t closeInput(audio_io_handle_t input); 182 183 virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output); 184 185 virtual status_t setVoiceVolume(float volume); 186 187 virtual status_t getRenderPosition(size_t *halFrames, size_t *dspFrames, 188 audio_io_handle_t output) const; 189 190 virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const; 191 192 virtual int newAudioSessionId(); 193 194 virtual void acquireAudioSessionId(int audioSession); 195 196 virtual void releaseAudioSessionId(int audioSession); 197 198 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 199 200 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 201 202 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 203 effect_descriptor_t *descriptor) const; 204 205 virtual sp<IEffect> createEffect( 206 effect_descriptor_t *pDesc, 207 const sp<IEffectClient>& effectClient, 208 int32_t priority, 209 audio_io_handle_t io, 210 int sessionId, 211 status_t *status, 212 int *id, 213 int *enabled); 214 215 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 216 audio_io_handle_t dstOutput); 217 218 virtual audio_module_handle_t loadHwModule(const char *name); 219 220 virtual uint32_t getPrimaryOutputSamplingRate(); 221 virtual size_t getPrimaryOutputFrameCount(); 222 223 virtual status_t setLowRamDevice(bool isLowRamDevice); 224 225 virtual status_t onTransact( 226 uint32_t code, 227 const Parcel& data, 228 Parcel* reply, 229 uint32_t flags); 230 231 // end of IAudioFlinger interface 232 233 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 234 void unregisterWriter(const sp<NBLog::Writer>& writer); 235private: 236 static const size_t kLogMemorySize = 10 * 1024; 237 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 238public: 239 240 class SyncEvent; 241 242 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 243 244 class SyncEvent : public RefBase { 245 public: 246 SyncEvent(AudioSystem::sync_event_t type, 247 int triggerSession, 248 int listenerSession, 249 sync_event_callback_t callBack, 250 void *cookie) 251 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 252 mCallback(callBack), mCookie(cookie) 253 {} 254 255 virtual ~SyncEvent() {} 256 257 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 258 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 259 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 260 AudioSystem::sync_event_t type() const { return mType; } 261 int triggerSession() const { return mTriggerSession; } 262 int listenerSession() const { return mListenerSession; } 263 void *cookie() const { return mCookie; } 264 265 private: 266 const AudioSystem::sync_event_t mType; 267 const int mTriggerSession; 268 const int mListenerSession; 269 sync_event_callback_t mCallback; 270 void * const mCookie; 271 mutable Mutex mLock; 272 }; 273 274 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 275 int triggerSession, 276 int listenerSession, 277 sync_event_callback_t callBack, 278 void *cookie); 279 280private: 281 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 282 283 audio_mode_t getMode() const { return mMode; } 284 285 bool btNrecIsOff() const { return mBtNrecIsOff; } 286 287 AudioFlinger() ANDROID_API; 288 virtual ~AudioFlinger(); 289 290 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 291 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 292 NO_INIT : NO_ERROR; } 293 294 // RefBase 295 virtual void onFirstRef(); 296 297 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 298 audio_devices_t devices); 299 void purgeStaleEffects_l(); 300 301 // standby delay for MIXER and DUPLICATING playback threads is read from property 302 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 303 static nsecs_t mStandbyTimeInNsecs; 304 305 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 306 // AudioFlinger::setParameters() updates, other threads read w/o lock 307 static uint32_t mScreenState; 308 309 // Internal dump utilities. 310 static const int kDumpLockRetries = 50; 311 static const int kDumpLockSleepUs = 20000; 312 static bool dumpTryLock(Mutex& mutex); 313 void dumpPermissionDenial(int fd, const Vector<String16>& args); 314 void dumpClients(int fd, const Vector<String16>& args); 315 void dumpInternals(int fd, const Vector<String16>& args); 316 317 // --- Client --- 318 class Client : public RefBase { 319 public: 320 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 321 virtual ~Client(); 322 sp<MemoryDealer> heap() const; 323 pid_t pid() const { return mPid; } 324 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 325 326 bool reserveTimedTrack(); 327 void releaseTimedTrack(); 328 329 private: 330 Client(const Client&); 331 Client& operator = (const Client&); 332 const sp<AudioFlinger> mAudioFlinger; 333 const sp<MemoryDealer> mMemoryDealer; 334 const pid_t mPid; 335 336 Mutex mTimedTrackLock; 337 int mTimedTrackCount; 338 }; 339 340 // --- Notification Client --- 341 class NotificationClient : public IBinder::DeathRecipient { 342 public: 343 NotificationClient(const sp<AudioFlinger>& audioFlinger, 344 const sp<IAudioFlingerClient>& client, 345 pid_t pid); 346 virtual ~NotificationClient(); 347 348 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 349 350 // IBinder::DeathRecipient 351 virtual void binderDied(const wp<IBinder>& who); 352 353 private: 354 NotificationClient(const NotificationClient&); 355 NotificationClient& operator = (const NotificationClient&); 356 357 const sp<AudioFlinger> mAudioFlinger; 358 const pid_t mPid; 359 const sp<IAudioFlingerClient> mAudioFlingerClient; 360 }; 361 362 class TrackHandle; 363 class RecordHandle; 364 class RecordThread; 365 class PlaybackThread; 366 class MixerThread; 367 class DirectOutputThread; 368 class DuplicatingThread; 369 class Track; 370 class RecordTrack; 371 class EffectModule; 372 class EffectHandle; 373 class EffectChain; 374 struct AudioStreamOut; 375 struct AudioStreamIn; 376 377 struct stream_type_t { 378 stream_type_t() 379 : volume(1.0f), 380 mute(false) 381 { 382 } 383 float volume; 384 bool mute; 385 }; 386 387 // --- PlaybackThread --- 388 389#include "Threads.h" 390 391#include "Effects.h" 392 393 // server side of the client's IAudioTrack 394 class TrackHandle : public android::BnAudioTrack { 395 public: 396 TrackHandle(const sp<PlaybackThread::Track>& track); 397 virtual ~TrackHandle(); 398 virtual sp<IMemory> getCblk() const; 399 virtual status_t start(); 400 virtual void stop(); 401 virtual void flush(); 402 virtual void pause(); 403 virtual status_t attachAuxEffect(int effectId); 404 virtual status_t allocateTimedBuffer(size_t size, 405 sp<IMemory>* buffer); 406 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 407 int64_t pts); 408 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 409 int target); 410 virtual status_t onTransact( 411 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 412 413 virtual status_t setParameters(const String8& keyValuePairs); 414 private: 415 const sp<PlaybackThread::Track> mTrack; 416 }; 417 418 // server side of the client's IAudioRecord 419 class RecordHandle : public android::BnAudioRecord { 420 public: 421 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 422 virtual ~RecordHandle(); 423 virtual sp<IMemory> getCblk() const; 424 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 425 virtual void stop(); 426 virtual status_t onTransact( 427 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 428 private: 429 const sp<RecordThread::RecordTrack> mRecordTrack; 430 431 // for use from destructor 432 void stop_nonvirtual(); 433 }; 434 435 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 436 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 437 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 438 // no range check, AudioFlinger::mLock held 439 bool streamMute_l(audio_stream_type_t stream) const 440 { return mStreamTypes[stream].mute; } 441 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 442 float streamVolume_l(audio_stream_type_t stream) const 443 { return mStreamTypes[stream].volume; } 444 void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2); 445 446 // allocate an audio_io_handle_t, session ID, or effect ID 447 uint32_t nextUniqueId(); 448 449 status_t moveEffectChain_l(int sessionId, 450 PlaybackThread *srcThread, 451 PlaybackThread *dstThread, 452 bool reRegister); 453 // return thread associated with primary hardware device, or NULL 454 PlaybackThread *primaryPlaybackThread_l() const; 455 audio_devices_t primaryOutputDevice_l() const; 456 457 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 458 459 460 void removeClient_l(pid_t pid); 461 void removeNotificationClient(pid_t pid); 462 463 class AudioHwDevice { 464 public: 465 enum Flags { 466 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 467 AHWD_CAN_SET_MASTER_MUTE = 0x2, 468 }; 469 470 AudioHwDevice(const char *moduleName, 471 audio_hw_device_t *hwDevice, 472 Flags flags) 473 : mModuleName(strdup(moduleName)) 474 , mHwDevice(hwDevice) 475 , mFlags(flags) { } 476 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 477 478 bool canSetMasterVolume() const { 479 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 480 } 481 482 bool canSetMasterMute() const { 483 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 484 } 485 486 const char *moduleName() const { return mModuleName; } 487 audio_hw_device_t *hwDevice() const { return mHwDevice; } 488 private: 489 const char * const mModuleName; 490 audio_hw_device_t * const mHwDevice; 491 Flags mFlags; 492 }; 493 494 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 495 // For emphasis, we could also make all pointers to them be "const *", 496 // but that would clutter the code unnecessarily. 497 498 struct AudioStreamOut { 499 AudioHwDevice* const audioHwDev; 500 audio_stream_out_t* const stream; 501 502 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 503 504 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out) : 505 audioHwDev(dev), stream(out) {} 506 }; 507 508 struct AudioStreamIn { 509 AudioHwDevice* const audioHwDev; 510 audio_stream_in_t* const stream; 511 512 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 513 514 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 515 audioHwDev(dev), stream(in) {} 516 }; 517 518 // for mAudioSessionRefs only 519 struct AudioSessionRef { 520 AudioSessionRef(int sessionid, pid_t pid) : 521 mSessionid(sessionid), mPid(pid), mCnt(1) {} 522 const int mSessionid; 523 const pid_t mPid; 524 int mCnt; 525 }; 526 527 mutable Mutex mLock; 528 529 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 530 531 mutable Mutex mHardwareLock; 532 // NOTE: If both mLock and mHardwareLock mutexes must be held, 533 // always take mLock before mHardwareLock 534 535 // These two fields are immutable after onFirstRef(), so no lock needed to access 536 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 537 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 538 539 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 540 enum hardware_call_state { 541 AUDIO_HW_IDLE = 0, // no operation in progress 542 AUDIO_HW_INIT, // init_check 543 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 544 AUDIO_HW_OUTPUT_CLOSE, // unused 545 AUDIO_HW_INPUT_OPEN, // unused 546 AUDIO_HW_INPUT_CLOSE, // unused 547 AUDIO_HW_STANDBY, // unused 548 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 549 AUDIO_HW_GET_ROUTING, // unused 550 AUDIO_HW_SET_ROUTING, // unused 551 AUDIO_HW_GET_MODE, // unused 552 AUDIO_HW_SET_MODE, // set_mode 553 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 554 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 555 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 556 AUDIO_HW_SET_PARAMETER, // set_parameters 557 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 558 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 559 AUDIO_HW_GET_PARAMETER, // get_parameters 560 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 561 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 562 }; 563 564 mutable hardware_call_state mHardwareStatus; // for dump only 565 566 567 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 568 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 569 570 // member variables below are protected by mLock 571 float mMasterVolume; 572 bool mMasterMute; 573 // end of variables protected by mLock 574 575 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 576 577 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 578 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 579 audio_mode_t mMode; 580 bool mBtNrecIsOff; 581 582 // protected by mLock 583 Vector<AudioSessionRef*> mAudioSessionRefs; 584 585 float masterVolume_l() const; 586 bool masterMute_l() const; 587 audio_module_handle_t loadHwModule_l(const char *name); 588 589 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 590 // to be created 591 592private: 593 sp<Client> registerPid_l(pid_t pid); // always returns non-0 594 595 // for use from destructor 596 status_t closeOutput_nonvirtual(audio_io_handle_t output); 597 status_t closeInput_nonvirtual(audio_io_handle_t input); 598 599#ifdef TEE_SINK 600 // all record threads serially share a common tee sink, which is re-created on format change 601 sp<NBAIO_Sink> mRecordTeeSink; 602 sp<NBAIO_Source> mRecordTeeSource; 603#endif 604 605public: 606 607#ifdef TEE_SINK 608 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 609 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 610 611 // whether tee sink is enabled by property 612 static bool mTeeSinkInputEnabled; 613 static bool mTeeSinkOutputEnabled; 614 static bool mTeeSinkTrackEnabled; 615 616 // runtime configured size of each tee sink pipe, in frames 617 static size_t mTeeSinkInputFrames; 618 static size_t mTeeSinkOutputFrames; 619 static size_t mTeeSinkTrackFrames; 620 621 // compile-time default size of tee sink pipes, in frames 622 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 623 static const size_t kTeeSinkInputFramesDefault = 0x200000; 624 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 625 static const size_t kTeeSinkTrackFramesDefault = 0x1000; 626#endif 627 628 // This method reads from a variable without mLock, but the variable is updated under mLock. So 629 // we might read a stale value, or a value that's inconsistent with respect to other variables. 630 // In this case, it's safe because the return value isn't used for making an important decision. 631 // The reason we don't want to take mLock is because it could block the caller for a long time. 632 bool isLowRamDevice() const { return mIsLowRamDevice; } 633 634private: 635 bool mIsLowRamDevice; 636 bool mIsDeviceTypeKnown; 637}; 638 639#undef INCLUDING_FROM_AUDIOFLINGER_H 640 641// ---------------------------------------------------------------------------- 642 643}; // namespace android 644 645#endif // ANDROID_AUDIO_FLINGER_H 646