AudioFlinger.h revision 1cda6afaf6207a41303e653a6ecd7909d73186eb
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 "Configuration.h" 22#include <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <common_time/cc_helper.h> 27 28#include <cutils/compiler.h> 29 30#include <media/IAudioFlinger.h> 31#include <media/IAudioFlingerClient.h> 32#include <media/IAudioTrack.h> 33#include <media/IAudioRecord.h> 34#include <media/AudioSystem.h> 35#include <media/AudioTrack.h> 36 37#include <utils/Atomic.h> 38#include <utils/Errors.h> 39#include <utils/threads.h> 40#include <utils/SortedVector.h> 41#include <utils/TypeHelpers.h> 42#include <utils/Vector.h> 43 44#include <binder/BinderService.h> 45#include <binder/MemoryDealer.h> 46 47#include <system/audio.h> 48#include <hardware/audio.h> 49#include <hardware/audio_policy.h> 50 51#include <media/AudioBufferProvider.h> 52#include <media/ExtendedAudioBufferProvider.h> 53#include "FastMixer.h" 54#include <media/nbaio/NBAIO.h> 55#include "AudioWatchdog.h" 56 57#include <powermanager/IPowerManager.h> 58 59#include <media/nbaio/NBLog.h> 60#include <private/media/AudioTrackShared.h> 61 62namespace android { 63 64struct audio_track_cblk_t; 65struct effect_param_cblk_t; 66class AudioMixer; 67class AudioBuffer; 68class AudioResampler; 69class FastMixer; 70class ServerProxy; 71 72// ---------------------------------------------------------------------------- 73 74// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 75// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 76// Adding full support for > 2 channel capture or playback would require more than simply changing 77// this #define. There is an independent hard-coded upper limit in AudioMixer; 78// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 79// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 80// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 81#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 82 83static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 84 85#define INCLUDING_FROM_AUDIOFLINGER_H 86 87class AudioFlinger : 88 public BinderService<AudioFlinger>, 89 public BnAudioFlinger 90{ 91 friend class BinderService<AudioFlinger>; // for AudioFlinger() 92public: 93 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 94 95 virtual status_t dump(int fd, const Vector<String16>& args); 96 97 // IAudioFlinger interface, in binder opcode order 98 virtual sp<IAudioTrack> createTrack( 99 audio_stream_type_t streamType, 100 uint32_t sampleRate, 101 audio_format_t format, 102 audio_channel_mask_t channelMask, 103 size_t *pFrameCount, 104 IAudioFlinger::track_flags_t *flags, 105 const sp<IMemory>& sharedBuffer, 106 audio_io_handle_t output, 107 pid_t tid, 108 int *sessionId, 109 int clientUid, 110 status_t *status /*non-NULL*/); 111 112 virtual sp<IAudioRecord> openRecord( 113 audio_io_handle_t input, 114 uint32_t sampleRate, 115 audio_format_t format, 116 audio_channel_mask_t channelMask, 117 size_t *pFrameCount, 118 IAudioFlinger::track_flags_t *flags, 119 pid_t tid, 120 int *sessionId, 121 sp<IMemory>& cblk, 122 sp<IMemory>& buffers, 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 /* List available audio ports and their attributes */ 227 virtual status_t listAudioPorts(unsigned int *num_ports, 228 struct audio_port *ports) 229 { 230 return INVALID_OPERATION; 231 } 232 233 /* Get attributes for a given audio port */ 234 virtual status_t getAudioPort(struct audio_port *port) 235 { 236 return INVALID_OPERATION; 237 } 238 239 /* Create an audio patch between several source and sink ports */ 240 virtual status_t createAudioPatch(const struct audio_patch *patch, 241 audio_patch_handle_t *handle) 242 { 243 return INVALID_OPERATION; 244 } 245 246 /* Release an audio patch */ 247 virtual status_t releaseAudioPatch(audio_patch_handle_t handle) 248 { 249 return INVALID_OPERATION; 250 } 251 252 /* List existing audio patches */ 253 virtual status_t listAudioPatches(unsigned int *num_patches, 254 struct audio_patch *patches) 255 { 256 return INVALID_OPERATION; 257 } 258 /* Set audio port configuration */ 259 virtual status_t setAudioPortConfig(const struct audio_port_config *config) 260 { 261 return INVALID_OPERATION; 262 } 263 264 virtual status_t onTransact( 265 uint32_t code, 266 const Parcel& data, 267 Parcel* reply, 268 uint32_t flags); 269 270 // end of IAudioFlinger interface 271 272 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 273 void unregisterWriter(const sp<NBLog::Writer>& writer); 274private: 275 static const size_t kLogMemorySize = 40 * 1024; 276 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 277 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 278 // for as long as possible. The memory is only freed when it is needed for another log writer. 279 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 280 Mutex mUnregisteredWritersLock; 281public: 282 283 class SyncEvent; 284 285 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 286 287 class SyncEvent : public RefBase { 288 public: 289 SyncEvent(AudioSystem::sync_event_t type, 290 int triggerSession, 291 int listenerSession, 292 sync_event_callback_t callBack, 293 wp<RefBase> cookie) 294 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 295 mCallback(callBack), mCookie(cookie) 296 {} 297 298 virtual ~SyncEvent() {} 299 300 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 301 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 302 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 303 AudioSystem::sync_event_t type() const { return mType; } 304 int triggerSession() const { return mTriggerSession; } 305 int listenerSession() const { return mListenerSession; } 306 wp<RefBase> cookie() const { return mCookie; } 307 308 private: 309 const AudioSystem::sync_event_t mType; 310 const int mTriggerSession; 311 const int mListenerSession; 312 sync_event_callback_t mCallback; 313 const wp<RefBase> mCookie; 314 mutable Mutex mLock; 315 }; 316 317 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 318 int triggerSession, 319 int listenerSession, 320 sync_event_callback_t callBack, 321 wp<RefBase> cookie); 322 323private: 324 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 325 326 audio_mode_t getMode() const { return mMode; } 327 328 bool btNrecIsOff() const { return mBtNrecIsOff; } 329 330 AudioFlinger() ANDROID_API; 331 virtual ~AudioFlinger(); 332 333 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 334 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 335 NO_INIT : NO_ERROR; } 336 337 // RefBase 338 virtual void onFirstRef(); 339 340 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 341 audio_devices_t devices); 342 void purgeStaleEffects_l(); 343 344 // standby delay for MIXER and DUPLICATING playback threads is read from property 345 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 346 static nsecs_t mStandbyTimeInNsecs; 347 348 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 349 // AudioFlinger::setParameters() updates, other threads read w/o lock 350 static uint32_t mScreenState; 351 352 // Internal dump utilities. 353 static const int kDumpLockRetries = 50; 354 static const int kDumpLockSleepUs = 20000; 355 static bool dumpTryLock(Mutex& mutex); 356 void dumpPermissionDenial(int fd, const Vector<String16>& args); 357 void dumpClients(int fd, const Vector<String16>& args); 358 void dumpInternals(int fd, const Vector<String16>& args); 359 360 // --- Client --- 361 class Client : public RefBase { 362 public: 363 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 364 virtual ~Client(); 365 sp<MemoryDealer> heap() const; 366 pid_t pid() const { return mPid; } 367 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 368 369 bool reserveTimedTrack(); 370 void releaseTimedTrack(); 371 372 private: 373 Client(const Client&); 374 Client& operator = (const Client&); 375 const sp<AudioFlinger> mAudioFlinger; 376 const sp<MemoryDealer> mMemoryDealer; 377 const pid_t mPid; 378 379 Mutex mTimedTrackLock; 380 int mTimedTrackCount; 381 }; 382 383 // --- Notification Client --- 384 class NotificationClient : public IBinder::DeathRecipient { 385 public: 386 NotificationClient(const sp<AudioFlinger>& audioFlinger, 387 const sp<IAudioFlingerClient>& client, 388 pid_t pid); 389 virtual ~NotificationClient(); 390 391 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 392 393 // IBinder::DeathRecipient 394 virtual void binderDied(const wp<IBinder>& who); 395 396 private: 397 NotificationClient(const NotificationClient&); 398 NotificationClient& operator = (const NotificationClient&); 399 400 const sp<AudioFlinger> mAudioFlinger; 401 const pid_t mPid; 402 const sp<IAudioFlingerClient> mAudioFlingerClient; 403 }; 404 405 class TrackHandle; 406 class RecordHandle; 407 class RecordThread; 408 class PlaybackThread; 409 class MixerThread; 410 class DirectOutputThread; 411 class OffloadThread; 412 class DuplicatingThread; 413 class AsyncCallbackThread; 414 class Track; 415 class RecordTrack; 416 class EffectModule; 417 class EffectHandle; 418 class EffectChain; 419 struct AudioStreamOut; 420 struct AudioStreamIn; 421 422 struct stream_type_t { 423 stream_type_t() 424 : volume(1.0f), 425 mute(false) 426 { 427 } 428 float volume; 429 bool mute; 430 }; 431 432 // --- PlaybackThread --- 433 434#include "Threads.h" 435 436#include "Effects.h" 437 438 // server side of the client's IAudioTrack 439 class TrackHandle : public android::BnAudioTrack { 440 public: 441 TrackHandle(const sp<PlaybackThread::Track>& track); 442 virtual ~TrackHandle(); 443 virtual sp<IMemory> getCblk() const; 444 virtual status_t start(); 445 virtual void stop(); 446 virtual void flush(); 447 virtual void pause(); 448 virtual status_t attachAuxEffect(int effectId); 449 virtual status_t allocateTimedBuffer(size_t size, 450 sp<IMemory>* buffer); 451 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 452 int64_t pts); 453 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 454 int target); 455 virtual status_t setParameters(const String8& keyValuePairs); 456 virtual status_t getTimestamp(AudioTimestamp& timestamp); 457 virtual void signal(); // signal playback thread for a change in control block 458 459 virtual status_t onTransact( 460 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 461 462 private: 463 const sp<PlaybackThread::Track> mTrack; 464 }; 465 466 // server side of the client's IAudioRecord 467 class RecordHandle : public android::BnAudioRecord { 468 public: 469 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 470 virtual ~RecordHandle(); 471 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 472 virtual void stop(); 473 virtual status_t onTransact( 474 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 475 private: 476 const sp<RecordThread::RecordTrack> mRecordTrack; 477 478 // for use from destructor 479 void stop_nonvirtual(); 480 }; 481 482 483 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 484 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 485 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 486 // no range check, AudioFlinger::mLock held 487 bool streamMute_l(audio_stream_type_t stream) const 488 { return mStreamTypes[stream].mute; } 489 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 490 float streamVolume_l(audio_stream_type_t stream) const 491 { return mStreamTypes[stream].volume; } 492 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 493 494 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 495 // They all share the same ID space, but the namespaces are actually independent 496 // because there are separate KeyedVectors for each kind of ID. 497 // The return value is uint32_t, but is cast to signed for some IDs. 498 // FIXME This API does not handle rollover to zero (for unsigned IDs), 499 // or from positive to negative (for signed IDs). 500 // Thus it may fail by returning an ID of the wrong sign, 501 // or by returning a non-unique ID. 502 uint32_t nextUniqueId(); 503 504 status_t moveEffectChain_l(int sessionId, 505 PlaybackThread *srcThread, 506 PlaybackThread *dstThread, 507 bool reRegister); 508 // return thread associated with primary hardware device, or NULL 509 PlaybackThread *primaryPlaybackThread_l() const; 510 audio_devices_t primaryOutputDevice_l() const; 511 512 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 513 514 515 void removeClient_l(pid_t pid); 516 void removeNotificationClient(pid_t pid); 517 bool isNonOffloadableGlobalEffectEnabled_l(); 518 void onNonOffloadableGlobalEffectEnable(); 519 520 class AudioHwDevice { 521 public: 522 enum Flags { 523 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 524 AHWD_CAN_SET_MASTER_MUTE = 0x2, 525 }; 526 527 AudioHwDevice(const char *moduleName, 528 audio_hw_device_t *hwDevice, 529 Flags flags) 530 : mModuleName(strdup(moduleName)) 531 , mHwDevice(hwDevice) 532 , mFlags(flags) { } 533 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 534 535 bool canSetMasterVolume() const { 536 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 537 } 538 539 bool canSetMasterMute() const { 540 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 541 } 542 543 const char *moduleName() const { return mModuleName; } 544 audio_hw_device_t *hwDevice() const { return mHwDevice; } 545 private: 546 const char * const mModuleName; 547 audio_hw_device_t * const mHwDevice; 548 const Flags mFlags; 549 }; 550 551 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 552 // For emphasis, we could also make all pointers to them be "const *", 553 // but that would clutter the code unnecessarily. 554 555 struct AudioStreamOut { 556 AudioHwDevice* const audioHwDev; 557 audio_stream_out_t* const stream; 558 const audio_output_flags_t flags; 559 560 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 561 562 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 563 audioHwDev(dev), stream(out), flags(flags) {} 564 }; 565 566 struct AudioStreamIn { 567 AudioHwDevice* const audioHwDev; 568 audio_stream_in_t* const stream; 569 570 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 571 572 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 573 audioHwDev(dev), stream(in) {} 574 }; 575 576 // for mAudioSessionRefs only 577 struct AudioSessionRef { 578 AudioSessionRef(int sessionid, pid_t pid) : 579 mSessionid(sessionid), mPid(pid), mCnt(1) {} 580 const int mSessionid; 581 const pid_t mPid; 582 int mCnt; 583 }; 584 585 mutable Mutex mLock; 586 // protects mClients and mNotificationClients. 587 // must be locked after mLock and ThreadBase::mLock if both must be locked 588 // avoids acquiring AudioFlinger::mLock from inside thread loop. 589 mutable Mutex mClientLock; 590 // protected by mClientLock 591 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 592 593 mutable Mutex mHardwareLock; 594 // NOTE: If both mLock and mHardwareLock mutexes must be held, 595 // always take mLock before mHardwareLock 596 597 // These two fields are immutable after onFirstRef(), so no lock needed to access 598 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 599 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 600 601 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 602 enum hardware_call_state { 603 AUDIO_HW_IDLE = 0, // no operation in progress 604 AUDIO_HW_INIT, // init_check 605 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 606 AUDIO_HW_OUTPUT_CLOSE, // unused 607 AUDIO_HW_INPUT_OPEN, // unused 608 AUDIO_HW_INPUT_CLOSE, // unused 609 AUDIO_HW_STANDBY, // unused 610 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 611 AUDIO_HW_GET_ROUTING, // unused 612 AUDIO_HW_SET_ROUTING, // unused 613 AUDIO_HW_GET_MODE, // unused 614 AUDIO_HW_SET_MODE, // set_mode 615 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 616 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 617 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 618 AUDIO_HW_SET_PARAMETER, // set_parameters 619 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 620 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 621 AUDIO_HW_GET_PARAMETER, // get_parameters 622 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 623 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 624 }; 625 626 mutable hardware_call_state mHardwareStatus; // for dump only 627 628 629 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 630 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 631 632 // member variables below are protected by mLock 633 float mMasterVolume; 634 bool mMasterMute; 635 // end of variables protected by mLock 636 637 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 638 639 // protected by mClientLock 640 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 641 642 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 643 // nextUniqueId() returns uint32_t, but this is declared int32_t 644 // because the atomic operations require an int32_t 645 646 audio_mode_t mMode; 647 bool mBtNrecIsOff; 648 649 // protected by mLock 650 Vector<AudioSessionRef*> mAudioSessionRefs; 651 652 float masterVolume_l() const; 653 bool masterMute_l() const; 654 audio_module_handle_t loadHwModule_l(const char *name); 655 656 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 657 // to be created 658 659private: 660 sp<Client> registerPid(pid_t pid); // always returns non-0 661 662 // for use from destructor 663 status_t closeOutput_nonvirtual(audio_io_handle_t output); 664 status_t closeInput_nonvirtual(audio_io_handle_t input); 665 666#ifdef TEE_SINK 667 // all record threads serially share a common tee sink, which is re-created on format change 668 sp<NBAIO_Sink> mRecordTeeSink; 669 sp<NBAIO_Source> mRecordTeeSource; 670#endif 671 672public: 673 674#ifdef TEE_SINK 675 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 676 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 677 678 // whether tee sink is enabled by property 679 static bool mTeeSinkInputEnabled; 680 static bool mTeeSinkOutputEnabled; 681 static bool mTeeSinkTrackEnabled; 682 683 // runtime configured size of each tee sink pipe, in frames 684 static size_t mTeeSinkInputFrames; 685 static size_t mTeeSinkOutputFrames; 686 static size_t mTeeSinkTrackFrames; 687 688 // compile-time default size of tee sink pipes, in frames 689 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 690 static const size_t kTeeSinkInputFramesDefault = 0x200000; 691 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 692 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 693#endif 694 695 // This method reads from a variable without mLock, but the variable is updated under mLock. So 696 // we might read a stale value, or a value that's inconsistent with respect to other variables. 697 // In this case, it's safe because the return value isn't used for making an important decision. 698 // The reason we don't want to take mLock is because it could block the caller for a long time. 699 bool isLowRamDevice() const { return mIsLowRamDevice; } 700 701private: 702 bool mIsLowRamDevice; 703 bool mIsDeviceTypeKnown; 704 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 705}; 706 707#undef INCLUDING_FROM_AUDIOFLINGER_H 708 709const char *formatToString(audio_format_t format); 710 711// ---------------------------------------------------------------------------- 712 713}; // namespace android 714 715#endif // ANDROID_AUDIO_FLINGER_H 716