AudioFlinger.h revision c263ca0ad8b6bdf5b0693996bc5f2f5916e0cd49
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 54#include "FastCapture.h" 55#include "FastMixer.h" 56#include <media/nbaio/NBAIO.h> 57#include "AudioWatchdog.h" 58 59#include <powermanager/IPowerManager.h> 60 61#include <media/nbaio/NBLog.h> 62#include <private/media/AudioTrackShared.h> 63 64namespace android { 65 66struct audio_track_cblk_t; 67struct effect_param_cblk_t; 68class AudioMixer; 69class AudioBuffer; 70class AudioResampler; 71class FastMixer; 72class ServerProxy; 73 74// ---------------------------------------------------------------------------- 75 76// AudioFlinger has a hard-coded upper limit of 2 channels for capture and playback. 77// There is support for > 2 channel tracks down-mixed to 2 channel output via a down-mix effect. 78// Adding full support for > 2 channel capture or playback would require more than simply changing 79// this #define. There is an independent hard-coded upper limit in AudioMixer; 80// removing that AudioMixer limit would be necessary but insufficient to support > 2 channels. 81// The macro FCC_2 highlights some (but not all) places where there is are 2-channel assumptions. 82// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 83#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 84 85static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 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 /* List available audio ports and their attributes */ 229 virtual status_t listAudioPorts(unsigned int *num_ports, 230 struct audio_port *ports); 231 232 /* Get attributes for a given audio port */ 233 virtual status_t getAudioPort(struct audio_port *port); 234 235 /* Create an audio patch between several source and sink ports */ 236 virtual status_t createAudioPatch(const struct audio_patch *patch, 237 audio_patch_handle_t *handle); 238 239 /* Release an audio patch */ 240 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 241 242 /* List existing audio patches */ 243 virtual status_t listAudioPatches(unsigned int *num_patches, 244 struct audio_patch *patches); 245 246 /* Set audio port configuration */ 247 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 248 249 virtual status_t onTransact( 250 uint32_t code, 251 const Parcel& data, 252 Parcel* reply, 253 uint32_t flags); 254 255 // end of IAudioFlinger interface 256 257 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 258 void unregisterWriter(const sp<NBLog::Writer>& writer); 259private: 260 static const size_t kLogMemorySize = 40 * 1024; 261 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 262 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 263 // for as long as possible. The memory is only freed when it is needed for another log writer. 264 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 265 Mutex mUnregisteredWritersLock; 266public: 267 268 class SyncEvent; 269 270 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 271 272 class SyncEvent : public RefBase { 273 public: 274 SyncEvent(AudioSystem::sync_event_t type, 275 int triggerSession, 276 int listenerSession, 277 sync_event_callback_t callBack, 278 wp<RefBase> cookie) 279 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 280 mCallback(callBack), mCookie(cookie) 281 {} 282 283 virtual ~SyncEvent() {} 284 285 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 286 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 287 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 288 AudioSystem::sync_event_t type() const { return mType; } 289 int triggerSession() const { return mTriggerSession; } 290 int listenerSession() const { return mListenerSession; } 291 wp<RefBase> cookie() const { return mCookie; } 292 293 private: 294 const AudioSystem::sync_event_t mType; 295 const int mTriggerSession; 296 const int mListenerSession; 297 sync_event_callback_t mCallback; 298 const wp<RefBase> mCookie; 299 mutable Mutex mLock; 300 }; 301 302 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 303 int triggerSession, 304 int listenerSession, 305 sync_event_callback_t callBack, 306 wp<RefBase> cookie); 307 308private: 309 class AudioHwDevice; // fwd declaration for findSuitableHwDev_l 310 311 audio_mode_t getMode() const { return mMode; } 312 313 bool btNrecIsOff() const { return mBtNrecIsOff; } 314 315 AudioFlinger() ANDROID_API; 316 virtual ~AudioFlinger(); 317 318 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 319 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 320 NO_INIT : NO_ERROR; } 321 322 // RefBase 323 virtual void onFirstRef(); 324 325 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 326 audio_devices_t devices); 327 void purgeStaleEffects_l(); 328 329 // standby delay for MIXER and DUPLICATING playback threads is read from property 330 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 331 static nsecs_t mStandbyTimeInNsecs; 332 333 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 334 // AudioFlinger::setParameters() updates, other threads read w/o lock 335 static uint32_t mScreenState; 336 337 // Internal dump utilities. 338 static const int kDumpLockRetries = 50; 339 static const int kDumpLockSleepUs = 20000; 340 static bool dumpTryLock(Mutex& mutex); 341 void dumpPermissionDenial(int fd, const Vector<String16>& args); 342 void dumpClients(int fd, const Vector<String16>& args); 343 void dumpInternals(int fd, const Vector<String16>& args); 344 345 // --- Client --- 346 class Client : public RefBase { 347 public: 348 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 349 virtual ~Client(); 350 sp<MemoryDealer> heap() const; 351 pid_t pid() const { return mPid; } 352 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 353 354 bool reserveTimedTrack(); 355 void releaseTimedTrack(); 356 357 private: 358 Client(const Client&); 359 Client& operator = (const Client&); 360 const sp<AudioFlinger> mAudioFlinger; 361 const sp<MemoryDealer> mMemoryDealer; 362 const pid_t mPid; 363 364 Mutex mTimedTrackLock; 365 int mTimedTrackCount; 366 }; 367 368 // --- Notification Client --- 369 class NotificationClient : public IBinder::DeathRecipient { 370 public: 371 NotificationClient(const sp<AudioFlinger>& audioFlinger, 372 const sp<IAudioFlingerClient>& client, 373 pid_t pid); 374 virtual ~NotificationClient(); 375 376 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 377 378 // IBinder::DeathRecipient 379 virtual void binderDied(const wp<IBinder>& who); 380 381 private: 382 NotificationClient(const NotificationClient&); 383 NotificationClient& operator = (const NotificationClient&); 384 385 const sp<AudioFlinger> mAudioFlinger; 386 const pid_t mPid; 387 const sp<IAudioFlingerClient> mAudioFlingerClient; 388 }; 389 390 class TrackHandle; 391 class RecordHandle; 392 class RecordThread; 393 class PlaybackThread; 394 class MixerThread; 395 class DirectOutputThread; 396 class OffloadThread; 397 class DuplicatingThread; 398 class AsyncCallbackThread; 399 class Track; 400 class RecordTrack; 401 class EffectModule; 402 class EffectHandle; 403 class EffectChain; 404 struct AudioStreamOut; 405 struct AudioStreamIn; 406 407 struct stream_type_t { 408 stream_type_t() 409 : volume(1.0f), 410 mute(false) 411 { 412 } 413 float volume; 414 bool mute; 415 }; 416 417 // --- PlaybackThread --- 418 419#include "Threads.h" 420 421#include "Effects.h" 422 423#include "PatchPanel.h" 424 425 // server side of the client's IAudioTrack 426 class TrackHandle : public android::BnAudioTrack { 427 public: 428 TrackHandle(const sp<PlaybackThread::Track>& track); 429 virtual ~TrackHandle(); 430 virtual sp<IMemory> getCblk() const; 431 virtual status_t start(); 432 virtual void stop(); 433 virtual void flush(); 434 virtual void pause(); 435 virtual status_t attachAuxEffect(int effectId); 436 virtual status_t allocateTimedBuffer(size_t size, 437 sp<IMemory>* buffer); 438 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 439 int64_t pts); 440 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 441 int target); 442 virtual status_t setParameters(const String8& keyValuePairs); 443 virtual status_t getTimestamp(AudioTimestamp& timestamp); 444 virtual void signal(); // signal playback thread for a change in control block 445 446 virtual status_t onTransact( 447 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 448 449 private: 450 const sp<PlaybackThread::Track> mTrack; 451 }; 452 453 // server side of the client's IAudioRecord 454 class RecordHandle : public android::BnAudioRecord { 455 public: 456 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 457 virtual ~RecordHandle(); 458 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 459 virtual void stop(); 460 virtual status_t onTransact( 461 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 462 private: 463 const sp<RecordThread::RecordTrack> mRecordTrack; 464 465 // for use from destructor 466 void stop_nonvirtual(); 467 }; 468 469 470 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 471 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 472 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 473 // no range check, AudioFlinger::mLock held 474 bool streamMute_l(audio_stream_type_t stream) const 475 { return mStreamTypes[stream].mute; } 476 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 477 float streamVolume_l(audio_stream_type_t stream) const 478 { return mStreamTypes[stream].volume; } 479 void audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 480 481 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 482 // They all share the same ID space, but the namespaces are actually independent 483 // because there are separate KeyedVectors for each kind of ID. 484 // The return value is uint32_t, but is cast to signed for some IDs. 485 // FIXME This API does not handle rollover to zero (for unsigned IDs), 486 // or from positive to negative (for signed IDs). 487 // Thus it may fail by returning an ID of the wrong sign, 488 // or by returning a non-unique ID. 489 uint32_t nextUniqueId(); 490 491 status_t moveEffectChain_l(int sessionId, 492 PlaybackThread *srcThread, 493 PlaybackThread *dstThread, 494 bool reRegister); 495 // return thread associated with primary hardware device, or NULL 496 PlaybackThread *primaryPlaybackThread_l() const; 497 audio_devices_t primaryOutputDevice_l() const; 498 499 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 500 501 502 void removeClient_l(pid_t pid); 503 void removeNotificationClient(pid_t pid); 504 bool isNonOffloadableGlobalEffectEnabled_l(); 505 void onNonOffloadableGlobalEffectEnable(); 506 507 class AudioHwDevice { 508 public: 509 enum Flags { 510 AHWD_CAN_SET_MASTER_VOLUME = 0x1, 511 AHWD_CAN_SET_MASTER_MUTE = 0x2, 512 }; 513 514 AudioHwDevice(const char *moduleName, 515 audio_hw_device_t *hwDevice, 516 Flags flags) 517 : mModuleName(strdup(moduleName)) 518 , mHwDevice(hwDevice) 519 , mFlags(flags) { } 520 /*virtual*/ ~AudioHwDevice() { free((void *)mModuleName); } 521 522 bool canSetMasterVolume() const { 523 return (0 != (mFlags & AHWD_CAN_SET_MASTER_VOLUME)); 524 } 525 526 bool canSetMasterMute() const { 527 return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE)); 528 } 529 530 const char *moduleName() const { return mModuleName; } 531 audio_hw_device_t *hwDevice() const { return mHwDevice; } 532 uint32_t version() const { return mHwDevice->common.version; } 533 534 private: 535 const char * const mModuleName; 536 audio_hw_device_t * const mHwDevice; 537 const Flags mFlags; 538 }; 539 540 // AudioStreamOut and AudioStreamIn are immutable, so their fields are const. 541 // For emphasis, we could also make all pointers to them be "const *", 542 // but that would clutter the code unnecessarily. 543 544 struct AudioStreamOut { 545 AudioHwDevice* const audioHwDev; 546 audio_stream_out_t* const stream; 547 const audio_output_flags_t flags; 548 549 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 550 551 AudioStreamOut(AudioHwDevice *dev, audio_stream_out_t *out, audio_output_flags_t flags) : 552 audioHwDev(dev), stream(out), flags(flags) {} 553 }; 554 555 struct AudioStreamIn { 556 AudioHwDevice* const audioHwDev; 557 audio_stream_in_t* const stream; 558 559 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 560 561 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 562 audioHwDev(dev), stream(in) {} 563 }; 564 565 // for mAudioSessionRefs only 566 struct AudioSessionRef { 567 AudioSessionRef(int sessionid, pid_t pid) : 568 mSessionid(sessionid), mPid(pid), mCnt(1) {} 569 const int mSessionid; 570 const pid_t mPid; 571 int mCnt; 572 }; 573 574 mutable Mutex mLock; 575 // protects mClients and mNotificationClients. 576 // must be locked after mLock and ThreadBase::mLock if both must be locked 577 // avoids acquiring AudioFlinger::mLock from inside thread loop. 578 mutable Mutex mClientLock; 579 // protected by mClientLock 580 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 581 582 mutable Mutex mHardwareLock; 583 // NOTE: If both mLock and mHardwareLock mutexes must be held, 584 // always take mLock before mHardwareLock 585 586 // These two fields are immutable after onFirstRef(), so no lock needed to access 587 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 588 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 589 590 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 591 enum hardware_call_state { 592 AUDIO_HW_IDLE = 0, // no operation in progress 593 AUDIO_HW_INIT, // init_check 594 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 595 AUDIO_HW_OUTPUT_CLOSE, // unused 596 AUDIO_HW_INPUT_OPEN, // unused 597 AUDIO_HW_INPUT_CLOSE, // unused 598 AUDIO_HW_STANDBY, // unused 599 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 600 AUDIO_HW_GET_ROUTING, // unused 601 AUDIO_HW_SET_ROUTING, // unused 602 AUDIO_HW_GET_MODE, // unused 603 AUDIO_HW_SET_MODE, // set_mode 604 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 605 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 606 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 607 AUDIO_HW_SET_PARAMETER, // set_parameters 608 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 609 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 610 AUDIO_HW_GET_PARAMETER, // get_parameters 611 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 612 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 613 }; 614 615 mutable hardware_call_state mHardwareStatus; // for dump only 616 617 618 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 619 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 620 621 // member variables below are protected by mLock 622 float mMasterVolume; 623 bool mMasterMute; 624 // end of variables protected by mLock 625 626 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 627 628 // protected by mClientLock 629 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 630 631 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 632 // nextUniqueId() returns uint32_t, but this is declared int32_t 633 // because the atomic operations require an int32_t 634 635 audio_mode_t mMode; 636 bool mBtNrecIsOff; 637 638 // protected by mLock 639 Vector<AudioSessionRef*> mAudioSessionRefs; 640 641 float masterVolume_l() const; 642 bool masterMute_l() const; 643 audio_module_handle_t loadHwModule_l(const char *name); 644 645 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 646 // to be created 647 648private: 649 sp<Client> registerPid(pid_t pid); // always returns non-0 650 651 // for use from destructor 652 status_t closeOutput_nonvirtual(audio_io_handle_t output); 653 status_t closeInput_nonvirtual(audio_io_handle_t input); 654 655#ifdef TEE_SINK 656 // all record threads serially share a common tee sink, which is re-created on format change 657 sp<NBAIO_Sink> mRecordTeeSink; 658 sp<NBAIO_Source> mRecordTeeSource; 659#endif 660 661public: 662 663#ifdef TEE_SINK 664 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 665 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 666 667 // whether tee sink is enabled by property 668 static bool mTeeSinkInputEnabled; 669 static bool mTeeSinkOutputEnabled; 670 static bool mTeeSinkTrackEnabled; 671 672 // runtime configured size of each tee sink pipe, in frames 673 static size_t mTeeSinkInputFrames; 674 static size_t mTeeSinkOutputFrames; 675 static size_t mTeeSinkTrackFrames; 676 677 // compile-time default size of tee sink pipes, in frames 678 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 679 static const size_t kTeeSinkInputFramesDefault = 0x200000; 680 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 681 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 682#endif 683 684 // This method reads from a variable without mLock, but the variable is updated under mLock. So 685 // we might read a stale value, or a value that's inconsistent with respect to other variables. 686 // In this case, it's safe because the return value isn't used for making an important decision. 687 // The reason we don't want to take mLock is because it could block the caller for a long time. 688 bool isLowRamDevice() const { return mIsLowRamDevice; } 689 690private: 691 bool mIsLowRamDevice; 692 bool mIsDeviceTypeKnown; 693 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 694 695 sp<PatchPanel> mPatchPanel; 696 697 uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none 698 // protected by mHardwareLock 699}; 700 701#undef INCLUDING_FROM_AUDIOFLINGER_H 702 703const char *formatToString(audio_format_t format); 704 705// ---------------------------------------------------------------------------- 706 707}; // namespace android 708 709#endif // ANDROID_AUDIO_FLINGER_H 710